From fed3d9297a9bf8b342c034e74a1fdba6940fe84a Mon Sep 17 00:00:00 2001 From: Hui Wang Date: Mon, 30 May 2022 12:01:51 +0800 Subject: ASoC: nau8822: Disable internal PLL if freq_out is zero After finishing the playback or recording, the machine driver might call snd_soc_dai_set_pll(codec, pll_id, 0, 0, 0) to stop the internal PLL, but with the codec driver nau8822, it will print error as below: nau8822 0-001a: Unsupported input clock 0 fsl-asoc-card sound-nau8822: failed to stop FLL: -22 Refer to the function wm8962_set_fll() in the codec driver wm8962, if the freq_out is zero, turn off the internal PLL and return 0. Cc: David Lin Cc: John Hsu Cc: Seven Li Signed-off-by: Hui Wang Link: https://lore.kernel.org/r/20220530040151.95221-3-hui.wang@canonical.com Signed-off-by: Mark Brown --- sound/soc/codecs/nau8822.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/nau8822.c b/sound/soc/codecs/nau8822.c index 08f6c56dc387..fd9c766e277f 100644 --- a/sound/soc/codecs/nau8822.c +++ b/sound/soc/codecs/nau8822.c @@ -726,6 +726,13 @@ static int nau8822_set_pll(struct snd_soc_dai *dai, int pll_id, int source, struct nau8822_pll *pll_param = &nau8822->pll; int ret, fs; + if (freq_out == 0) { + dev_dbg(component->dev, "PLL disabled\n"); + snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, NAU8822_PLL_EN_MASK, NAU8822_PLL_OFF); + return 0; + } + fs = freq_out / 256; ret = nau8822_calc_pll(freq_in, fs, pll_param); -- cgit v1.2.3 From 7358a803c778f28314721e78339f3fa5b787f55c Mon Sep 17 00:00:00 2001 From: Sameer Pujar Date: Fri, 3 Jun 2022 12:06:08 +0530 Subject: ASoC: tegra: Add Tegra210 based OPE driver The Output Processing Engine (OPE) is one of the AHUB client. It has PEQ (Parametric Equalizer) and MBDRC (Multi Band Dynamic Range Compressor) sub blocks for data processing. The PEQ block gets samples from the MBDRC block. This patch registers OPE driver with ASoC framework. The component driver exposes DAPM widgets, routes and kcontrols for the device. The DAI driver exposes OPE interfaces, which can be used to connect different components in the ASoC layer. Makefile and Kconfig support is added to allow build the driver. Signed-off-by: Sameer Pujar Link: https://lore.kernel.org/r/1654238172-16293-3-git-send-email-spujar@nvidia.com Signed-off-by: Mark Brown --- sound/soc/tegra/Kconfig | 9 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra210_mbdrc.c | 1012 ++++++++++++++++++++++++++++++++++++++ sound/soc/tegra/tegra210_mbdrc.h | 215 ++++++++ sound/soc/tegra/tegra210_ope.c | 419 ++++++++++++++++ sound/soc/tegra/tegra210_ope.h | 90 ++++ sound/soc/tegra/tegra210_peq.c | 434 ++++++++++++++++ sound/soc/tegra/tegra210_peq.h | 56 +++ 8 files changed, 2237 insertions(+) create mode 100644 sound/soc/tegra/tegra210_mbdrc.c create mode 100644 sound/soc/tegra/tegra210_mbdrc.h create mode 100644 sound/soc/tegra/tegra210_ope.c create mode 100644 sound/soc/tegra/tegra210_ope.h create mode 100644 sound/soc/tegra/tegra210_peq.c create mode 100644 sound/soc/tegra/tegra210_peq.h (limited to 'sound') diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index 2482d9867357..b6712a3d1fa1 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -85,6 +85,15 @@ config SND_SOC_TEGRA210_I2S compatible devices. Say Y or M if you want to add support for Tegra210 I2S module. +config SND_SOC_TEGRA210_OPE + tristate "Tegra210 OPE module" + help + Config to enable the Output Processing Engine (OPE) which includes + Parametric Equalizer (PEQ) and Multi Band Dynamic Range Compressor + (MBDRC) sub blocks for data processing. It can support up to 8 + channels. + Say Y or M if you want to add support for Tegra210 OPE module. + config SND_SOC_TEGRA186_ASRC tristate "Tegra186 ASRC module" help diff --git a/sound/soc/tegra/Makefile b/sound/soc/tegra/Makefile index 70a498ddb2fa..b723c78e665d 100644 --- a/sound/soc/tegra/Makefile +++ b/sound/soc/tegra/Makefile @@ -19,6 +19,7 @@ snd-soc-tegra210-sfc-objs := tegra210_sfc.o snd-soc-tegra210-amx-objs := tegra210_amx.o snd-soc-tegra210-adx-objs := tegra210_adx.o snd-soc-tegra210-mixer-objs := tegra210_mixer.o +snd-soc-tegra210-ope-objs := tegra210_ope.o tegra210_mbdrc.o tegra210_peq.o obj-$(CONFIG_SND_SOC_TEGRA) += snd-soc-tegra-pcm.o obj-$(CONFIG_SND_SOC_TEGRA20_AC97) += snd-soc-tegra20-ac97.o @@ -38,6 +39,7 @@ obj-$(CONFIG_SND_SOC_TEGRA210_SFC) += snd-soc-tegra210-sfc.o obj-$(CONFIG_SND_SOC_TEGRA210_AMX) += snd-soc-tegra210-amx.o obj-$(CONFIG_SND_SOC_TEGRA210_ADX) += snd-soc-tegra210-adx.o obj-$(CONFIG_SND_SOC_TEGRA210_MIXER) += snd-soc-tegra210-mixer.o +obj-$(CONFIG_SND_SOC_TEGRA210_OPE) += snd-soc-tegra210-ope.o # Tegra machine Support snd-soc-tegra-wm8903-objs := tegra_wm8903.o diff --git a/sound/soc/tegra/tegra210_mbdrc.c b/sound/soc/tegra/tegra210_mbdrc.c new file mode 100644 index 000000000000..7d9da33a9951 --- /dev/null +++ b/sound/soc/tegra/tegra210_mbdrc.c @@ -0,0 +1,1012 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// tegra210_mbdrc.c - Tegra210 MBDRC driver +// +// Copyright (c) 2022, NVIDIA CORPORATION. All rights reserved. + +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "tegra210_mbdrc.h" +#include "tegra210_ope.h" + +#define MBDRC_FILTER_REG(reg, id) \ + ((reg) + ((id) * TEGRA210_MBDRC_FILTER_PARAM_STRIDE)) + +#define MBDRC_FILTER_REG_DEFAULTS(id) \ + { MBDRC_FILTER_REG(TEGRA210_MBDRC_IIR_CFG, id), 0x00000005}, \ + { MBDRC_FILTER_REG(TEGRA210_MBDRC_IN_ATTACK, id), 0x3e48590c}, \ + { MBDRC_FILTER_REG(TEGRA210_MBDRC_IN_RELEASE, id), 0x08414e9f}, \ + { MBDRC_FILTER_REG(TEGRA210_MBDRC_FAST_ATTACK, id), 0x7fffffff}, \ + { MBDRC_FILTER_REG(TEGRA210_MBDRC_IN_THRESHOLD, id), 0x06145082}, \ + { MBDRC_FILTER_REG(TEGRA210_MBDRC_OUT_THRESHOLD, id), 0x060d379b}, \ + { MBDRC_FILTER_REG(TEGRA210_MBDRC_RATIO_1ST, id), 0x0000a000}, \ + { MBDRC_FILTER_REG(TEGRA210_MBDRC_RATIO_2ND, id), 0x00002000}, \ + { MBDRC_FILTER_REG(TEGRA210_MBDRC_RATIO_3RD, id), 0x00000b33}, \ + { MBDRC_FILTER_REG(TEGRA210_MBDRC_RATIO_4TH, id), 0x00000800}, \ + { MBDRC_FILTER_REG(TEGRA210_MBDRC_RATIO_5TH, id), 0x0000019a}, \ + { MBDRC_FILTER_REG(TEGRA210_MBDRC_MAKEUP_GAIN, id), 0x00000002}, \ + { MBDRC_FILTER_REG(TEGRA210_MBDRC_INIT_GAIN, id), 0x00066666}, \ + { MBDRC_FILTER_REG(TEGRA210_MBDRC_GAIN_ATTACK, id), 0x00d9ba0e}, \ + { MBDRC_FILTER_REG(TEGRA210_MBDRC_GAIN_RELEASE, id), 0x3e48590c}, \ + { MBDRC_FILTER_REG(TEGRA210_MBDRC_FAST_RELEASE, id), 0x7ffff26a}, \ + { MBDRC_FILTER_REG(TEGRA210_MBDRC_CFG_RAM_CTRL, id), 0x4000} + +static const struct reg_default tegra210_mbdrc_reg_defaults[] = { + { TEGRA210_MBDRC_CFG, 0x0030de51}, + { TEGRA210_MBDRC_CHANNEL_MASK, 0x00000003}, + { TEGRA210_MBDRC_FAST_FACTOR, 0x30000800}, + + MBDRC_FILTER_REG_DEFAULTS(0), + MBDRC_FILTER_REG_DEFAULTS(1), + MBDRC_FILTER_REG_DEFAULTS(2), +}; + +/* Default MBDRC parameters */ +static const struct tegra210_mbdrc_config mbdrc_init_config = { + .mode = 0, /* Bypass */ + .rms_off = 48, + .peak_rms_mode = 1, /* PEAK */ + .fliter_structure = 0, /* All-pass tree */ + .shift_ctrl = 30, + .frame_size = 32, + .channel_mask = 0x3, + .fa_factor = 2048, + .fr_factor = 14747, + + .band_params[MBDRC_LOW_BAND] = { + .band = MBDRC_LOW_BAND, + .iir_stages = 5, + .in_attack_tc = 1044928780, + .in_release_tc = 138497695, + .fast_attack_tc = 2147483647, + .in_threshold = {130, 80, 20, 6}, + .out_threshold = {155, 55, 13, 6}, + .ratio = {40960, 8192, 2867, 2048, 410}, + .makeup_gain = 4, + .gain_init = 419430, + .gain_attack_tc = 14268942, + .gain_release_tc = 1440547090, + .fast_release_tc = 2147480170, + + .biquad_params = { + /* + * Gains: + * + * b0, b1, a0, + * a1, a2, + */ + + /* Band-0 */ + 961046798, -2030431983, 1073741824, + 2030431983, -961046798, + /* Band-1 */ + 1030244425, -2099481453, 1073741824, + 2099481453, -1030244425, + /* Band-2 */ + 1067169294, -2136327263, 1073741824, + 2136327263, -1067169294, + /* Band-3 */ + 434951949, -1306567134, 1073741824, + 1306567134, -434951949, + /* Band-4 */ + 780656019, -1605955641, 1073741824, + 1605955641, -780656019, + /* Band-5 */ + 1024497031, -1817128152, 1073741824, + 1817128152, -1024497031, + /* Band-6 */ + 1073741824, 0, 0, + 0, 0, + /* Band-7 */ + 1073741824, 0, 0, + 0, 0, + } + }, + + .band_params[MBDRC_MID_BAND] = { + .band = MBDRC_MID_BAND, + .iir_stages = 5, + .in_attack_tc = 1581413104, + .in_release_tc = 35494783, + .fast_attack_tc = 2147483647, + .in_threshold = {130, 50, 30, 6}, + .out_threshold = {106, 50, 30, 13}, + .ratio = {40960, 2867, 4096, 2867, 410}, + .makeup_gain = 6, + .gain_init = 419430, + .gain_attack_tc = 4766887, + .gain_release_tc = 1044928780, + .fast_release_tc = 2147480170, + + .biquad_params = { + /* + * Gains: + * + * b0, b1, a0, + * a1, a2, + */ + + /* Band-0 */ + -1005668963, 1073741824, 0, + 1005668963, 0, + /* Band-1 */ + 998437058, -2067742187, 1073741824, + 2067742187, -998437058, + /* Band-2 */ + 1051963422, -2121153948, 1073741824, + 2121153948, -1051963422, + /* Band-3 */ + 434951949, -1306567134, 1073741824, + 1306567134, -434951949, + /* Band-4 */ + 780656019, -1605955641, 1073741824, + 1605955641, -780656019, + /* Band-5 */ + 1024497031, -1817128152, 1073741824, + 1817128152, -1024497031, + /* Band-6 */ + 1073741824, 0, 0, + 0, 0, + /* Band-7 */ + 1073741824, 0, 0, + 0, 0, + } + }, + + .band_params[MBDRC_HIGH_BAND] = { + .band = MBDRC_HIGH_BAND, + .iir_stages = 5, + .in_attack_tc = 2144750688, + .in_release_tc = 70402888, + .fast_attack_tc = 2147483647, + .in_threshold = {130, 50, 30, 6}, + .out_threshold = {106, 50, 30, 13}, + .ratio = {40960, 2867, 4096, 2867, 410}, + .makeup_gain = 6, + .gain_init = 419430, + .gain_attack_tc = 4766887, + .gain_release_tc = 1044928780, + .fast_release_tc = 2147480170, + + .biquad_params = { + /* + * Gains: + * + * b0, b1, a0, + * a1, a2, + */ + + /* Band-0 */ + 1073741824, 0, 0, + 0, 0, + /* Band-1 */ + 1073741824, 0, 0, + 0, 0, + /* Band-2 */ + 1073741824, 0, 0, + 0, 0, + /* Band-3 */ + -619925131, 1073741824, 0, + 619925131, 0, + /* Band-4 */ + 606839335, -1455425976, 1073741824, + 1455425976, -606839335, + /* Band-5 */ + 917759617, -1724690840, 1073741824, + 1724690840, -917759617, + /* Band-6 */ + 1073741824, 0, 0, + 0, 0, + /* Band-7 */ + 1073741824, 0, 0, + 0, 0, + } + } +}; + +static void tegra210_mbdrc_write_ram(struct regmap *regmap, unsigned int reg_ctrl, + unsigned int reg_data, unsigned int ram_offset, + unsigned int *data, size_t size) +{ + unsigned int val; + unsigned int i; + + val = ram_offset & TEGRA210_MBDRC_RAM_CTRL_RAM_ADDR_MASK; + val |= TEGRA210_MBDRC_RAM_CTRL_ADDR_INIT_EN; + val |= TEGRA210_MBDRC_RAM_CTRL_SEQ_ACCESS_EN; + val |= TEGRA210_MBDRC_RAM_CTRL_RW_WRITE; + + regmap_write(regmap, reg_ctrl, val); + + for (i = 0; i < size; i++) + regmap_write(regmap, reg_data, data[i]); +} + +static int tegra210_mbdrc_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct tegra210_ope *ope = snd_soc_component_get_drvdata(cmpnt); + unsigned int val; + + regmap_read(ope->mbdrc_regmap, mc->reg, &val); + + ucontrol->value.integer.value[0] = (val >> mc->shift) & mc->max; + + return 0; +} + +static int tegra210_mbdrc_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct tegra210_ope *ope = snd_soc_component_get_drvdata(cmpnt); + unsigned int val = ucontrol->value.integer.value[0]; + bool change = false; + + val = val << mc->shift; + + regmap_update_bits_check(ope->mbdrc_regmap, mc->reg, + (mc->max << mc->shift), val, &change); + + return change ? 1 : 0; +} + +static int tegra210_mbdrc_get_enum(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct tegra210_ope *ope = snd_soc_component_get_drvdata(cmpnt); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int val; + + regmap_read(ope->mbdrc_regmap, e->reg, &val); + + ucontrol->value.enumerated.item[0] = (val >> e->shift_l) & e->mask; + + return 0; +} + +static int tegra210_mbdrc_put_enum(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct tegra210_ope *ope = snd_soc_component_get_drvdata(cmpnt); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + bool change = false; + unsigned int val; + unsigned int mask; + + if (ucontrol->value.enumerated.item[0] > e->items - 1) + return -EINVAL; + + val = ucontrol->value.enumerated.item[0] << e->shift_l; + mask = e->mask << e->shift_l; + + regmap_update_bits_check(ope->mbdrc_regmap, e->reg, mask, val, + &change); + + return change ? 1 : 0; +} + +static int tegra210_mbdrc_band_params_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tegra_soc_bytes *params = (void *)kcontrol->private_value; + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct tegra210_ope *ope = snd_soc_component_get_drvdata(cmpnt); + u32 *data = (u32 *)ucontrol->value.bytes.data; + u32 regs = params->soc.base; + u32 mask = params->soc.mask; + u32 shift = params->shift; + unsigned int i; + + for (i = 0; i < params->soc.num_regs; i++, regs += cmpnt->val_bytes) { + regmap_read(ope->mbdrc_regmap, regs, &data[i]); + + data[i] = ((data[i] & mask) >> shift); + } + + return 0; +} + +static int tegra210_mbdrc_band_params_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tegra_soc_bytes *params = (void *)kcontrol->private_value; + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct tegra210_ope *ope = snd_soc_component_get_drvdata(cmpnt); + u32 *data = (u32 *)ucontrol->value.bytes.data; + u32 regs = params->soc.base; + u32 mask = params->soc.mask; + u32 shift = params->shift; + bool change = false; + unsigned int i; + + for (i = 0; i < params->soc.num_regs; i++, regs += cmpnt->val_bytes) { + bool update = false; + + regmap_update_bits_check(ope->mbdrc_regmap, regs, mask, + data[i] << shift, &update); + + change |= update; + } + + return change ? 1 : 0; +} + +static int tegra210_mbdrc_threshold_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tegra_soc_bytes *params = (void *)kcontrol->private_value; + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct tegra210_ope *ope = snd_soc_component_get_drvdata(cmpnt); + u32 *data = (u32 *)ucontrol->value.bytes.data; + u32 regs = params->soc.base; + u32 num_regs = params->soc.num_regs; + u32 val; + unsigned int i; + + for (i = 0; i < num_regs; i += 4, regs += cmpnt->val_bytes) { + regmap_read(ope->mbdrc_regmap, regs, &val); + + data[i] = (val & TEGRA210_MBDRC_THRESH_1ST_MASK) >> + TEGRA210_MBDRC_THRESH_1ST_SHIFT; + data[i + 1] = (val & TEGRA210_MBDRC_THRESH_2ND_MASK) >> + TEGRA210_MBDRC_THRESH_2ND_SHIFT; + data[i + 2] = (val & TEGRA210_MBDRC_THRESH_3RD_MASK) >> + TEGRA210_MBDRC_THRESH_3RD_SHIFT; + data[i + 3] = (val & TEGRA210_MBDRC_THRESH_4TH_MASK) >> + TEGRA210_MBDRC_THRESH_4TH_SHIFT; + } + + return 0; +} + +static int tegra210_mbdrc_threshold_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tegra_soc_bytes *params = (void *)kcontrol->private_value; + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct tegra210_ope *ope = snd_soc_component_get_drvdata(cmpnt); + u32 *data = (u32 *)ucontrol->value.bytes.data; + u32 regs = params->soc.base; + u32 num_regs = params->soc.num_regs; + bool change = false; + unsigned int i; + + for (i = 0; i < num_regs; i += 4, regs += cmpnt->val_bytes) { + bool update = false; + + data[i] = (((data[i] >> TEGRA210_MBDRC_THRESH_1ST_SHIFT) & + TEGRA210_MBDRC_THRESH_1ST_MASK) | + ((data[i + 1] >> TEGRA210_MBDRC_THRESH_2ND_SHIFT) & + TEGRA210_MBDRC_THRESH_2ND_MASK) | + ((data[i + 2] >> TEGRA210_MBDRC_THRESH_3RD_SHIFT) & + TEGRA210_MBDRC_THRESH_3RD_MASK) | + ((data[i + 3] >> TEGRA210_MBDRC_THRESH_4TH_SHIFT) & + TEGRA210_MBDRC_THRESH_4TH_MASK)); + + regmap_update_bits_check(ope->mbdrc_regmap, regs, 0xffffffff, + data[i], &update); + + change |= update; + } + + return change ? 1 : 0; +} + +static int tegra210_mbdrc_biquad_coeffs_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tegra_soc_bytes *params = (void *)kcontrol->private_value; + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + u32 *data = (u32 *)ucontrol->value.bytes.data; + + memset(data, 0, params->soc.num_regs * cmpnt->val_bytes); + + return 0; +} + +static int tegra210_mbdrc_biquad_coeffs_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tegra_soc_bytes *params = (void *)kcontrol->private_value; + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct tegra210_ope *ope = snd_soc_component_get_drvdata(cmpnt); + u32 reg_ctrl = params->soc.base; + u32 reg_data = reg_ctrl + cmpnt->val_bytes; + u32 *data = (u32 *)ucontrol->value.bytes.data; + + tegra210_mbdrc_write_ram(ope->mbdrc_regmap, reg_ctrl, reg_data, + params->shift, data, params->soc.num_regs); + + return 1; +} + +static int tegra210_mbdrc_param_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct soc_bytes *params = (void *)kcontrol->private_value; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; + uinfo->count = params->num_regs * sizeof(u32); + + return 0; +} + +static int tegra210_mbdrc_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct tegra210_ope *ope = snd_soc_component_get_drvdata(cmpnt); + int val; + + regmap_read(ope->mbdrc_regmap, mc->reg, &val); + + ucontrol->value.integer.value[0] = + ((val >> mc->shift) - TEGRA210_MBDRC_MASTER_VOL_MIN); + + return 0; +} + +static int tegra210_mbdrc_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct tegra210_ope *ope = snd_soc_component_get_drvdata(cmpnt); + int val = ucontrol->value.integer.value[0]; + bool change = false; + + val += TEGRA210_MBDRC_MASTER_VOL_MIN; + + regmap_update_bits_check(ope->mbdrc_regmap, mc->reg, + mc->max << mc->shift, val << mc->shift, + &change); + + regmap_read(ope->mbdrc_regmap, mc->reg, &val); + + return change ? 1 : 0; +} + +static const char * const tegra210_mbdrc_mode_text[] = { + "Bypass", "Fullband", "Dualband", "Multiband" +}; + +static const struct soc_enum tegra210_mbdrc_mode_enum = + SOC_ENUM_SINGLE(TEGRA210_MBDRC_CFG, TEGRA210_MBDRC_CFG_MBDRC_MODE_SHIFT, + 4, tegra210_mbdrc_mode_text); + +static const char * const tegra210_mbdrc_peak_rms_text[] = { + "Peak", "RMS" +}; + +static const struct soc_enum tegra210_mbdrc_peak_rms_enum = + SOC_ENUM_SINGLE(TEGRA210_MBDRC_CFG, TEGRA210_MBDRC_CFG_PEAK_RMS_SHIFT, + 2, tegra210_mbdrc_peak_rms_text); + +static const char * const tegra210_mbdrc_filter_structure_text[] = { + "All-pass-tree", "Flexible" +}; + +static const struct soc_enum tegra210_mbdrc_filter_structure_enum = + SOC_ENUM_SINGLE(TEGRA210_MBDRC_CFG, + TEGRA210_MBDRC_CFG_FILTER_STRUCTURE_SHIFT, 2, + tegra210_mbdrc_filter_structure_text); + +static const char * const tegra210_mbdrc_frame_size_text[] = { + "N1", "N2", "N4", "N8", "N16", "N32", "N64" +}; + +static const struct soc_enum tegra210_mbdrc_frame_size_enum = + SOC_ENUM_SINGLE(TEGRA210_MBDRC_CFG, TEGRA210_MBDRC_CFG_FRAME_SIZE_SHIFT, + 7, tegra210_mbdrc_frame_size_text); + +#define TEGRA_MBDRC_BYTES_EXT(xname, xbase, xregs, xshift, xmask, xinfo) \ + TEGRA_SOC_BYTES_EXT(xname, xbase, xregs, xshift, xmask, \ + tegra210_mbdrc_band_params_get, \ + tegra210_mbdrc_band_params_put, \ + tegra210_mbdrc_param_info) + +#define TEGRA_MBDRC_BAND_BYTES_EXT(xname, xbase, xshift, xmask, xinfo) \ + TEGRA_MBDRC_BYTES_EXT(xname, xbase, TEGRA210_MBDRC_FILTER_COUNT, \ + xshift, xmask, xinfo) + +static const DECLARE_TLV_DB_MINMAX(mdbrc_vol_tlv, -25600, 25500); + +static const struct snd_kcontrol_new tegra210_mbdrc_controls[] = { + SOC_ENUM_EXT("MBDRC Peak RMS Mode", tegra210_mbdrc_peak_rms_enum, + tegra210_mbdrc_get_enum, tegra210_mbdrc_put_enum), + + SOC_ENUM_EXT("MBDRC Filter Structure", + tegra210_mbdrc_filter_structure_enum, + tegra210_mbdrc_get_enum, tegra210_mbdrc_put_enum), + + SOC_ENUM_EXT("MBDRC Frame Size", tegra210_mbdrc_frame_size_enum, + tegra210_mbdrc_get_enum, tegra210_mbdrc_put_enum), + + SOC_ENUM_EXT("MBDRC Mode", tegra210_mbdrc_mode_enum, + tegra210_mbdrc_get_enum, tegra210_mbdrc_put_enum), + + SOC_SINGLE_EXT("MBDRC RMS Offset", TEGRA210_MBDRC_CFG, + TEGRA210_MBDRC_CFG_RMS_OFFSET_SHIFT, 0x1ff, 0, + tegra210_mbdrc_get, tegra210_mbdrc_put), + + SOC_SINGLE_EXT("MBDRC Shift Control", TEGRA210_MBDRC_CFG, + TEGRA210_MBDRC_CFG_SHIFT_CTRL_SHIFT, 0x1f, 0, + tegra210_mbdrc_get, tegra210_mbdrc_put), + + SOC_SINGLE_EXT("MBDRC Fast Attack Factor", TEGRA210_MBDRC_FAST_FACTOR, + TEGRA210_MBDRC_FAST_FACTOR_ATTACK_SHIFT, 0xffff, 0, + tegra210_mbdrc_get, tegra210_mbdrc_put), + + SOC_SINGLE_EXT("MBDRC Fast Release Factor", TEGRA210_MBDRC_FAST_FACTOR, + TEGRA210_MBDRC_FAST_FACTOR_RELEASE_SHIFT, 0xffff, 0, + tegra210_mbdrc_get, tegra210_mbdrc_put), + + SOC_SINGLE_RANGE_EXT_TLV("MBDRC Master Volume", + TEGRA210_MBDRC_MASTER_VOL, + TEGRA210_MBDRC_MASTER_VOL_SHIFT, + 0, 0x1ff, 0, + tegra210_mbdrc_vol_get, tegra210_mbdrc_vol_put, + mdbrc_vol_tlv), + + TEGRA_SOC_BYTES_EXT("MBDRC IIR Stages", TEGRA210_MBDRC_IIR_CFG, + TEGRA210_MBDRC_FILTER_COUNT, + TEGRA210_MBDRC_IIR_CFG_NUM_STAGES_SHIFT, + TEGRA210_MBDRC_IIR_CFG_NUM_STAGES_MASK, + tegra210_mbdrc_band_params_get, + tegra210_mbdrc_band_params_put, + tegra210_mbdrc_param_info), + + TEGRA_SOC_BYTES_EXT("MBDRC In Attack Time Const", TEGRA210_MBDRC_IN_ATTACK, + TEGRA210_MBDRC_FILTER_COUNT, + TEGRA210_MBDRC_IN_ATTACK_TC_SHIFT, + TEGRA210_MBDRC_IN_ATTACK_TC_MASK, + tegra210_mbdrc_band_params_get, + tegra210_mbdrc_band_params_put, + tegra210_mbdrc_param_info), + + TEGRA_SOC_BYTES_EXT("MBDRC In Release Time Const", TEGRA210_MBDRC_IN_RELEASE, + TEGRA210_MBDRC_FILTER_COUNT, + TEGRA210_MBDRC_IN_RELEASE_TC_SHIFT, + TEGRA210_MBDRC_IN_RELEASE_TC_MASK, + tegra210_mbdrc_band_params_get, + tegra210_mbdrc_band_params_put, + tegra210_mbdrc_param_info), + + TEGRA_SOC_BYTES_EXT("MBDRC Fast Attack Time Const", TEGRA210_MBDRC_FAST_ATTACK, + TEGRA210_MBDRC_FILTER_COUNT, + TEGRA210_MBDRC_FAST_ATTACK_TC_SHIFT, + TEGRA210_MBDRC_FAST_ATTACK_TC_MASK, + tegra210_mbdrc_band_params_get, + tegra210_mbdrc_band_params_put, + tegra210_mbdrc_param_info), + + TEGRA_SOC_BYTES_EXT("MBDRC In Threshold", TEGRA210_MBDRC_IN_THRESHOLD, + TEGRA210_MBDRC_FILTER_COUNT * 4, 0, 0xffffffff, + tegra210_mbdrc_threshold_get, + tegra210_mbdrc_threshold_put, + tegra210_mbdrc_param_info), + + TEGRA_SOC_BYTES_EXT("MBDRC Out Threshold", TEGRA210_MBDRC_OUT_THRESHOLD, + TEGRA210_MBDRC_FILTER_COUNT * 4, 0, 0xffffffff, + tegra210_mbdrc_threshold_get, + tegra210_mbdrc_threshold_put, + tegra210_mbdrc_param_info), + + TEGRA_SOC_BYTES_EXT("MBDRC Ratio", TEGRA210_MBDRC_RATIO_1ST, + TEGRA210_MBDRC_FILTER_COUNT * 5, + TEGRA210_MBDRC_RATIO_1ST_SHIFT, TEGRA210_MBDRC_RATIO_1ST_MASK, + tegra210_mbdrc_band_params_get, + tegra210_mbdrc_band_params_put, + tegra210_mbdrc_param_info), + + TEGRA_SOC_BYTES_EXT("MBDRC Makeup Gain", TEGRA210_MBDRC_MAKEUP_GAIN, + TEGRA210_MBDRC_FILTER_COUNT, + TEGRA210_MBDRC_MAKEUP_GAIN_SHIFT, + TEGRA210_MBDRC_MAKEUP_GAIN_MASK, + tegra210_mbdrc_band_params_get, + tegra210_mbdrc_band_params_put, + tegra210_mbdrc_param_info), + + TEGRA_SOC_BYTES_EXT("MBDRC Init Gain", TEGRA210_MBDRC_INIT_GAIN, + TEGRA210_MBDRC_FILTER_COUNT, + TEGRA210_MBDRC_INIT_GAIN_SHIFT, + TEGRA210_MBDRC_INIT_GAIN_MASK, + tegra210_mbdrc_band_params_get, + tegra210_mbdrc_band_params_put, + tegra210_mbdrc_param_info), + + TEGRA_SOC_BYTES_EXT("MBDRC Attack Gain", TEGRA210_MBDRC_GAIN_ATTACK, + TEGRA210_MBDRC_FILTER_COUNT, + TEGRA210_MBDRC_GAIN_ATTACK_SHIFT, + TEGRA210_MBDRC_GAIN_ATTACK_MASK, + tegra210_mbdrc_band_params_get, + tegra210_mbdrc_band_params_put, + tegra210_mbdrc_param_info), + + TEGRA_SOC_BYTES_EXT("MBDRC Release Gain", TEGRA210_MBDRC_GAIN_RELEASE, + TEGRA210_MBDRC_FILTER_COUNT, + TEGRA210_MBDRC_GAIN_RELEASE_SHIFT, + TEGRA210_MBDRC_GAIN_RELEASE_MASK, + tegra210_mbdrc_band_params_get, + tegra210_mbdrc_band_params_put, + tegra210_mbdrc_param_info), + + TEGRA_SOC_BYTES_EXT("MBDRC Fast Release Gain", + TEGRA210_MBDRC_FAST_RELEASE, + TEGRA210_MBDRC_FILTER_COUNT, + TEGRA210_MBDRC_FAST_RELEASE_SHIFT, + TEGRA210_MBDRC_FAST_RELEASE_MASK, + tegra210_mbdrc_band_params_get, + tegra210_mbdrc_band_params_put, + tegra210_mbdrc_param_info), + + TEGRA_SOC_BYTES_EXT("MBDRC Low Band Biquad Coeffs", + TEGRA210_MBDRC_CFG_RAM_CTRL, + TEGRA210_MBDRC_MAX_BIQUAD_STAGES * 5, 0, 0xffffffff, + tegra210_mbdrc_biquad_coeffs_get, + tegra210_mbdrc_biquad_coeffs_put, + tegra210_mbdrc_param_info), + + TEGRA_SOC_BYTES_EXT("MBDRC Mid Band Biquad Coeffs", + TEGRA210_MBDRC_CFG_RAM_CTRL + + TEGRA210_MBDRC_FILTER_PARAM_STRIDE, + TEGRA210_MBDRC_MAX_BIQUAD_STAGES * 5, 0, 0xffffffff, + tegra210_mbdrc_biquad_coeffs_get, + tegra210_mbdrc_biquad_coeffs_put, + tegra210_mbdrc_param_info), + + TEGRA_SOC_BYTES_EXT("MBDRC High Band Biquad Coeffs", + TEGRA210_MBDRC_CFG_RAM_CTRL + + (TEGRA210_MBDRC_FILTER_PARAM_STRIDE * 2), + TEGRA210_MBDRC_MAX_BIQUAD_STAGES * 5, 0, 0xffffffff, + tegra210_mbdrc_biquad_coeffs_get, + tegra210_mbdrc_biquad_coeffs_put, + tegra210_mbdrc_param_info), +}; + +static bool tegra210_mbdrc_wr_reg(struct device *dev, unsigned int reg) +{ + if (reg >= TEGRA210_MBDRC_IIR_CFG) + reg -= ((reg - TEGRA210_MBDRC_IIR_CFG) % + (TEGRA210_MBDRC_FILTER_PARAM_STRIDE * + TEGRA210_MBDRC_FILTER_COUNT)); + + switch (reg) { + case TEGRA210_MBDRC_SOFT_RESET: + case TEGRA210_MBDRC_CG: + case TEGRA210_MBDRC_CFG ... TEGRA210_MBDRC_CFG_RAM_DATA: + return true; + default: + return false; + } +} + +static bool tegra210_mbdrc_rd_reg(struct device *dev, unsigned int reg) +{ + if (tegra210_mbdrc_wr_reg(dev, reg)) + return true; + + if (reg >= TEGRA210_MBDRC_IIR_CFG) + reg -= ((reg - TEGRA210_MBDRC_IIR_CFG) % + (TEGRA210_MBDRC_FILTER_PARAM_STRIDE * + TEGRA210_MBDRC_FILTER_COUNT)); + + switch (reg) { + case TEGRA210_MBDRC_STATUS: + return true; + default: + return false; + } +} + +static bool tegra210_mbdrc_volatile_reg(struct device *dev, unsigned int reg) +{ + if (reg >= TEGRA210_MBDRC_IIR_CFG) + reg -= ((reg - TEGRA210_MBDRC_IIR_CFG) % + (TEGRA210_MBDRC_FILTER_PARAM_STRIDE * + TEGRA210_MBDRC_FILTER_COUNT)); + + switch (reg) { + case TEGRA210_MBDRC_SOFT_RESET: + case TEGRA210_MBDRC_STATUS: + case TEGRA210_MBDRC_CFG_RAM_CTRL: + case TEGRA210_MBDRC_CFG_RAM_DATA: + return true; + default: + return false; + } +} + +static bool tegra210_mbdrc_precious_reg(struct device *dev, unsigned int reg) +{ + if (reg >= TEGRA210_MBDRC_IIR_CFG) + reg -= ((reg - TEGRA210_MBDRC_IIR_CFG) % + (TEGRA210_MBDRC_FILTER_PARAM_STRIDE * + TEGRA210_MBDRC_FILTER_COUNT)); + + switch (reg) { + case TEGRA210_MBDRC_CFG_RAM_DATA: + return true; + default: + return false; + } +} + +static const struct regmap_config tegra210_mbdrc_regmap_cfg = { + .name = "mbdrc", + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = TEGRA210_MBDRC_MAX_REG, + .writeable_reg = tegra210_mbdrc_wr_reg, + .readable_reg = tegra210_mbdrc_rd_reg, + .volatile_reg = tegra210_mbdrc_volatile_reg, + .precious_reg = tegra210_mbdrc_precious_reg, + .reg_defaults = tegra210_mbdrc_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(tegra210_mbdrc_reg_defaults), + .cache_type = REGCACHE_FLAT, +}; + +int tegra210_mbdrc_hw_params(struct snd_soc_component *cmpnt) +{ + struct tegra210_ope *ope = snd_soc_component_get_drvdata(cmpnt); + const struct tegra210_mbdrc_config *conf = &mbdrc_init_config; + u32 val = 0; + unsigned int i; + + regmap_read(ope->mbdrc_regmap, TEGRA210_MBDRC_CFG, &val); + + if (val & TEGRA210_MBDRC_CFG_MBDRC_MODE_BYPASS) + return 0; + + for (i = 0; i < MBDRC_NUM_BAND; i++) { + const struct tegra210_mbdrc_band_params *params = + &conf->band_params[i]; + + u32 reg_off = i * TEGRA210_MBDRC_FILTER_PARAM_STRIDE; + + tegra210_mbdrc_write_ram(ope->mbdrc_regmap, + reg_off + TEGRA210_MBDRC_CFG_RAM_CTRL, + reg_off + TEGRA210_MBDRC_CFG_RAM_DATA, + 0, (u32 *)¶ms->biquad_params[0], + TEGRA210_MBDRC_MAX_BIQUAD_STAGES * 5); + } + return 0; +} + +int tegra210_mbdrc_component_init(struct snd_soc_component *cmpnt) +{ + struct tegra210_ope *ope = snd_soc_component_get_drvdata(cmpnt); + const struct tegra210_mbdrc_config *conf = &mbdrc_init_config; + unsigned int i; + u32 val; + + pm_runtime_get_sync(cmpnt->dev); + + /* Initialize MBDRC registers and AHUB RAM with default params */ + regmap_update_bits(ope->mbdrc_regmap, TEGRA210_MBDRC_CFG, + TEGRA210_MBDRC_CFG_MBDRC_MODE_MASK, + conf->mode << TEGRA210_MBDRC_CFG_MBDRC_MODE_SHIFT); + + regmap_update_bits(ope->mbdrc_regmap, TEGRA210_MBDRC_CFG, + TEGRA210_MBDRC_CFG_RMS_OFFSET_MASK, + conf->rms_off << TEGRA210_MBDRC_CFG_RMS_OFFSET_SHIFT); + + regmap_update_bits(ope->mbdrc_regmap, TEGRA210_MBDRC_CFG, + TEGRA210_MBDRC_CFG_PEAK_RMS_MASK, + conf->peak_rms_mode << TEGRA210_MBDRC_CFG_PEAK_RMS_SHIFT); + + regmap_update_bits(ope->mbdrc_regmap, TEGRA210_MBDRC_CFG, + TEGRA210_MBDRC_CFG_FILTER_STRUCTURE_MASK, + conf->fliter_structure << + TEGRA210_MBDRC_CFG_FILTER_STRUCTURE_SHIFT); + + regmap_update_bits(ope->mbdrc_regmap, TEGRA210_MBDRC_CFG, + TEGRA210_MBDRC_CFG_SHIFT_CTRL_MASK, + conf->shift_ctrl << TEGRA210_MBDRC_CFG_SHIFT_CTRL_SHIFT); + + regmap_update_bits(ope->mbdrc_regmap, TEGRA210_MBDRC_CFG, + TEGRA210_MBDRC_CFG_FRAME_SIZE_MASK, + __ffs(conf->frame_size) << + TEGRA210_MBDRC_CFG_FRAME_SIZE_SHIFT); + + regmap_update_bits(ope->mbdrc_regmap, TEGRA210_MBDRC_CHANNEL_MASK, + TEGRA210_MBDRC_CHANNEL_MASK_MASK, + conf->channel_mask << TEGRA210_MBDRC_CHANNEL_MASK_SHIFT); + + regmap_update_bits(ope->mbdrc_regmap, TEGRA210_MBDRC_FAST_FACTOR, + TEGRA210_MBDRC_FAST_FACTOR_ATTACK_MASK, + conf->fa_factor << TEGRA210_MBDRC_FAST_FACTOR_ATTACK_SHIFT); + + regmap_update_bits(ope->mbdrc_regmap, TEGRA210_MBDRC_FAST_FACTOR, + TEGRA210_MBDRC_FAST_FACTOR_ATTACK_MASK, + conf->fr_factor << TEGRA210_MBDRC_FAST_FACTOR_ATTACK_SHIFT); + + for (i = 0; i < MBDRC_NUM_BAND; i++) { + const struct tegra210_mbdrc_band_params *params = + &conf->band_params[i]; + u32 reg_off = i * TEGRA210_MBDRC_FILTER_PARAM_STRIDE; + + regmap_update_bits(ope->mbdrc_regmap, + reg_off + TEGRA210_MBDRC_IIR_CFG, + TEGRA210_MBDRC_IIR_CFG_NUM_STAGES_MASK, + params->iir_stages << + TEGRA210_MBDRC_IIR_CFG_NUM_STAGES_SHIFT); + + regmap_update_bits(ope->mbdrc_regmap, + reg_off + TEGRA210_MBDRC_IN_ATTACK, + TEGRA210_MBDRC_IN_ATTACK_TC_MASK, + params->in_attack_tc << + TEGRA210_MBDRC_IN_ATTACK_TC_SHIFT); + + regmap_update_bits(ope->mbdrc_regmap, + reg_off + TEGRA210_MBDRC_IN_RELEASE, + TEGRA210_MBDRC_IN_RELEASE_TC_MASK, + params->in_release_tc << + TEGRA210_MBDRC_IN_RELEASE_TC_SHIFT); + + regmap_update_bits(ope->mbdrc_regmap, + reg_off + TEGRA210_MBDRC_FAST_ATTACK, + TEGRA210_MBDRC_FAST_ATTACK_TC_MASK, + params->fast_attack_tc << + TEGRA210_MBDRC_FAST_ATTACK_TC_SHIFT); + + val = (((params->in_threshold[0] >> + TEGRA210_MBDRC_THRESH_1ST_SHIFT) & + TEGRA210_MBDRC_THRESH_1ST_MASK) | + ((params->in_threshold[1] >> + TEGRA210_MBDRC_THRESH_2ND_SHIFT) & + TEGRA210_MBDRC_THRESH_2ND_MASK) | + ((params->in_threshold[2] >> + TEGRA210_MBDRC_THRESH_3RD_SHIFT) & + TEGRA210_MBDRC_THRESH_3RD_MASK) | + ((params->in_threshold[3] >> + TEGRA210_MBDRC_THRESH_4TH_SHIFT) & + TEGRA210_MBDRC_THRESH_4TH_MASK)); + + regmap_update_bits(ope->mbdrc_regmap, + reg_off + TEGRA210_MBDRC_IN_THRESHOLD, + 0xffffffff, val); + + val = (((params->out_threshold[0] >> + TEGRA210_MBDRC_THRESH_1ST_SHIFT) & + TEGRA210_MBDRC_THRESH_1ST_MASK) | + ((params->out_threshold[1] >> + TEGRA210_MBDRC_THRESH_2ND_SHIFT) & + TEGRA210_MBDRC_THRESH_2ND_MASK) | + ((params->out_threshold[2] >> + TEGRA210_MBDRC_THRESH_3RD_SHIFT) & + TEGRA210_MBDRC_THRESH_3RD_MASK) | + ((params->out_threshold[3] >> + TEGRA210_MBDRC_THRESH_4TH_SHIFT) & + TEGRA210_MBDRC_THRESH_4TH_MASK)); + + regmap_update_bits(ope->mbdrc_regmap, + reg_off + TEGRA210_MBDRC_OUT_THRESHOLD, + 0xffffffff, val); + + regmap_update_bits(ope->mbdrc_regmap, + reg_off + TEGRA210_MBDRC_RATIO_1ST, + TEGRA210_MBDRC_RATIO_1ST_MASK, + params->ratio[0] << TEGRA210_MBDRC_RATIO_1ST_SHIFT); + + regmap_update_bits(ope->mbdrc_regmap, + reg_off + TEGRA210_MBDRC_RATIO_2ND, + TEGRA210_MBDRC_RATIO_2ND_MASK, + params->ratio[1] << TEGRA210_MBDRC_RATIO_2ND_SHIFT); + + regmap_update_bits(ope->mbdrc_regmap, + reg_off + TEGRA210_MBDRC_RATIO_3RD, + TEGRA210_MBDRC_RATIO_3RD_MASK, + params->ratio[2] << TEGRA210_MBDRC_RATIO_3RD_SHIFT); + + regmap_update_bits(ope->mbdrc_regmap, + reg_off + TEGRA210_MBDRC_RATIO_4TH, + TEGRA210_MBDRC_RATIO_4TH_MASK, + params->ratio[3] << TEGRA210_MBDRC_RATIO_4TH_SHIFT); + + regmap_update_bits(ope->mbdrc_regmap, + reg_off + TEGRA210_MBDRC_RATIO_5TH, + TEGRA210_MBDRC_RATIO_5TH_MASK, + params->ratio[4] << TEGRA210_MBDRC_RATIO_5TH_SHIFT); + + regmap_update_bits(ope->mbdrc_regmap, + reg_off + TEGRA210_MBDRC_MAKEUP_GAIN, + TEGRA210_MBDRC_MAKEUP_GAIN_MASK, + params->makeup_gain << + TEGRA210_MBDRC_MAKEUP_GAIN_SHIFT); + + regmap_update_bits(ope->mbdrc_regmap, + reg_off + TEGRA210_MBDRC_INIT_GAIN, + TEGRA210_MBDRC_INIT_GAIN_MASK, + params->gain_init << + TEGRA210_MBDRC_INIT_GAIN_SHIFT); + + regmap_update_bits(ope->mbdrc_regmap, + reg_off + TEGRA210_MBDRC_GAIN_ATTACK, + TEGRA210_MBDRC_GAIN_ATTACK_MASK, + params->gain_attack_tc << + TEGRA210_MBDRC_GAIN_ATTACK_SHIFT); + + regmap_update_bits(ope->mbdrc_regmap, + reg_off + TEGRA210_MBDRC_GAIN_RELEASE, + TEGRA210_MBDRC_GAIN_RELEASE_MASK, + params->gain_release_tc << + TEGRA210_MBDRC_GAIN_RELEASE_SHIFT); + + regmap_update_bits(ope->mbdrc_regmap, + reg_off + TEGRA210_MBDRC_FAST_RELEASE, + TEGRA210_MBDRC_FAST_RELEASE_MASK, + params->fast_release_tc << + TEGRA210_MBDRC_FAST_RELEASE_SHIFT); + + tegra210_mbdrc_write_ram(ope->mbdrc_regmap, + reg_off + TEGRA210_MBDRC_CFG_RAM_CTRL, + reg_off + TEGRA210_MBDRC_CFG_RAM_DATA, 0, + (u32 *)¶ms->biquad_params[0], + TEGRA210_MBDRC_MAX_BIQUAD_STAGES * 5); + } + + pm_runtime_put_sync(cmpnt->dev); + + snd_soc_add_component_controls(cmpnt, tegra210_mbdrc_controls, + ARRAY_SIZE(tegra210_mbdrc_controls)); + + return 0; +} + +int tegra210_mbdrc_regmap_init(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct tegra210_ope *ope = dev_get_drvdata(dev); + struct device_node *child; + struct resource mem; + void __iomem *regs; + int err; + + child = of_get_child_by_name(dev->of_node, "dynamic-range-compressor"); + if (!child) + return -ENODEV; + + err = of_address_to_resource(child, 0, &mem); + of_node_put(child); + if (err < 0) { + dev_err(dev, "fail to get MBDRC resource\n"); + return err; + } + + mem.flags = IORESOURCE_MEM; + regs = devm_ioremap_resource(dev, &mem); + if (IS_ERR(regs)) + return PTR_ERR(regs); + + ope->mbdrc_regmap = devm_regmap_init_mmio(dev, regs, + &tegra210_mbdrc_regmap_cfg); + if (IS_ERR(ope->mbdrc_regmap)) { + dev_err(dev, "regmap init failed\n"); + return PTR_ERR(ope->mbdrc_regmap); + } + + regcache_cache_only(ope->mbdrc_regmap, true); + + return 0; +} diff --git a/sound/soc/tegra/tegra210_mbdrc.h b/sound/soc/tegra/tegra210_mbdrc.h new file mode 100644 index 000000000000..4c48da0e1dea --- /dev/null +++ b/sound/soc/tegra/tegra210_mbdrc.h @@ -0,0 +1,215 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * tegra210_mbdrc.h - Definitions for Tegra210 MBDRC driver + * + * Copyright (c) 2022, NVIDIA CORPORATION. All rights reserved. + * + */ + +#ifndef __TEGRA210_MBDRC_H__ +#define __TEGRA210_MBDRC_H__ + +#include +#include + +/* Register offsets from TEGRA210_MBDRC*_BASE */ +#define TEGRA210_MBDRC_SOFT_RESET 0x4 +#define TEGRA210_MBDRC_CG 0x8 +#define TEGRA210_MBDRC_STATUS 0xc +#define TEGRA210_MBDRC_CFG 0x28 +#define TEGRA210_MBDRC_CHANNEL_MASK 0x2c +#define TEGRA210_MBDRC_MASTER_VOL 0x30 +#define TEGRA210_MBDRC_FAST_FACTOR 0x34 + +#define TEGRA210_MBDRC_FILTER_COUNT 3 +#define TEGRA210_MBDRC_FILTER_PARAM_STRIDE 0x4 + +#define TEGRA210_MBDRC_IIR_CFG 0x38 +#define TEGRA210_MBDRC_IN_ATTACK 0x44 +#define TEGRA210_MBDRC_IN_RELEASE 0x50 +#define TEGRA210_MBDRC_FAST_ATTACK 0x5c +#define TEGRA210_MBDRC_IN_THRESHOLD 0x68 +#define TEGRA210_MBDRC_OUT_THRESHOLD 0x74 +#define TEGRA210_MBDRC_RATIO_1ST 0x80 +#define TEGRA210_MBDRC_RATIO_2ND 0x8c +#define TEGRA210_MBDRC_RATIO_3RD 0x98 +#define TEGRA210_MBDRC_RATIO_4TH 0xa4 +#define TEGRA210_MBDRC_RATIO_5TH 0xb0 +#define TEGRA210_MBDRC_MAKEUP_GAIN 0xbc +#define TEGRA210_MBDRC_INIT_GAIN 0xc8 +#define TEGRA210_MBDRC_GAIN_ATTACK 0xd4 +#define TEGRA210_MBDRC_GAIN_RELEASE 0xe0 +#define TEGRA210_MBDRC_FAST_RELEASE 0xec +#define TEGRA210_MBDRC_CFG_RAM_CTRL 0xf8 +#define TEGRA210_MBDRC_CFG_RAM_DATA 0x104 + +#define TEGRA210_MBDRC_MAX_REG (TEGRA210_MBDRC_CFG_RAM_DATA + \ + (TEGRA210_MBDRC_FILTER_PARAM_STRIDE * \ + (TEGRA210_MBDRC_FILTER_COUNT - 1))) + +/* Fields for TEGRA210_MBDRC_CFG */ +#define TEGRA210_MBDRC_CFG_RMS_OFFSET_SHIFT 16 +#define TEGRA210_MBDRC_CFG_RMS_OFFSET_MASK (0x1ff << TEGRA210_MBDRC_CFG_RMS_OFFSET_SHIFT) + +#define TEGRA210_MBDRC_CFG_PEAK_RMS_SHIFT 14 +#define TEGRA210_MBDRC_CFG_PEAK_RMS_MASK (0x1 << TEGRA210_MBDRC_CFG_PEAK_RMS_SHIFT) +#define TEGRA210_MBDRC_CFG_PEAK (1 << TEGRA210_MBDRC_CFG_PEAK_RMS_SHIFT) + +#define TEGRA210_MBDRC_CFG_FILTER_STRUCTURE_SHIFT 13 +#define TEGRA210_MBDRC_CFG_FILTER_STRUCTURE_MASK (0x1 << TEGRA210_MBDRC_CFG_FILTER_STRUCTURE_SHIFT) +#define TEGRA210_MBDRC_CFG_FILTER_STRUCTURE_FLEX (1 << TEGRA210_MBDRC_CFG_FILTER_STRUCTURE_SHIFT) + +#define TEGRA210_MBDRC_CFG_SHIFT_CTRL_SHIFT 8 +#define TEGRA210_MBDRC_CFG_SHIFT_CTRL_MASK (0x1f << TEGRA210_MBDRC_CFG_SHIFT_CTRL_SHIFT) + +#define TEGRA210_MBDRC_CFG_FRAME_SIZE_SHIFT 4 +#define TEGRA210_MBDRC_CFG_FRAME_SIZE_MASK (0xf << TEGRA210_MBDRC_CFG_FRAME_SIZE_SHIFT) + +#define TEGRA210_MBDRC_CFG_MBDRC_MODE_SHIFT 0 +#define TEGRA210_MBDRC_CFG_MBDRC_MODE_MASK (0x3 << TEGRA210_MBDRC_CFG_MBDRC_MODE_SHIFT) +#define TEGRA210_MBDRC_CFG_MBDRC_MODE_BYPASS (0 << TEGRA210_MBDRC_CFG_MBDRC_MODE_SHIFT) + +/* Fields for TEGRA210_MBDRC_CHANNEL_MASK */ +#define TEGRA210_MBDRC_CHANNEL_MASK_SHIFT 0 +#define TEGRA210_MBDRC_CHANNEL_MASK_MASK (0xff << TEGRA210_MBDRC_CHANNEL_MASK_SHIFT) + +/* Fields for TEGRA210_MBDRC_MASTER_VOL */ +#define TEGRA210_MBDRC_MASTER_VOL_SHIFT 23 +#define TEGRA210_MBDRC_MASTER_VOL_MIN -256 +#define TEGRA210_MBDRC_MASTER_VOL_MAX 256 + +/* Fields for TEGRA210_MBDRC_FAST_FACTOR */ +#define TEGRA210_MBDRC_FAST_FACTOR_RELEASE_SHIFT 16 +#define TEGRA210_MBDRC_FAST_FACTOR_RELEASE_MASK (0xffff << TEGRA210_MBDRC_FAST_FACTOR_RELEASE_SHIFT) + +#define TEGRA210_MBDRC_FAST_FACTOR_ATTACK_SHIFT 0 +#define TEGRA210_MBDRC_FAST_FACTOR_ATTACK_MASK (0xffff << TEGRA210_MBDRC_FAST_FACTOR_ATTACK_SHIFT) + +/* Fields for TEGRA210_MBDRC_IIR_CFG */ +#define TEGRA210_MBDRC_IIR_CFG_NUM_STAGES_SHIFT 0 +#define TEGRA210_MBDRC_IIR_CFG_NUM_STAGES_MASK (0xf << TEGRA210_MBDRC_IIR_CFG_NUM_STAGES_SHIFT) + +/* Fields for TEGRA210_MBDRC_IN_ATTACK */ +#define TEGRA210_MBDRC_IN_ATTACK_TC_SHIFT 0 +#define TEGRA210_MBDRC_IN_ATTACK_TC_MASK (0xffffffff << TEGRA210_MBDRC_IN_ATTACK_TC_SHIFT) + +/* Fields for TEGRA210_MBDRC_IN_RELEASE */ +#define TEGRA210_MBDRC_IN_RELEASE_TC_SHIFT 0 +#define TEGRA210_MBDRC_IN_RELEASE_TC_MASK (0xffffffff << TEGRA210_MBDRC_IN_RELEASE_TC_SHIFT) + +/* Fields for TEGRA210_MBDRC_FAST_ATTACK */ +#define TEGRA210_MBDRC_FAST_ATTACK_TC_SHIFT 0 +#define TEGRA210_MBDRC_FAST_ATTACK_TC_MASK (0xffffffff << TEGRA210_MBDRC_FAST_ATTACK_TC_SHIFT) + +/* Fields for TEGRA210_MBDRC_IN_THRESHOLD / TEGRA210_MBDRC_OUT_THRESHOLD */ +#define TEGRA210_MBDRC_THRESH_4TH_SHIFT 24 +#define TEGRA210_MBDRC_THRESH_4TH_MASK (0xff << TEGRA210_MBDRC_THRESH_4TH_SHIFT) + +#define TEGRA210_MBDRC_THRESH_3RD_SHIFT 16 +#define TEGRA210_MBDRC_THRESH_3RD_MASK (0xff << TEGRA210_MBDRC_THRESH_3RD_SHIFT) + +#define TEGRA210_MBDRC_THRESH_2ND_SHIFT 8 +#define TEGRA210_MBDRC_THRESH_2ND_MASK (0xff << TEGRA210_MBDRC_THRESH_2ND_SHIFT) + +#define TEGRA210_MBDRC_THRESH_1ST_SHIFT 0 +#define TEGRA210_MBDRC_THRESH_1ST_MASK (0xff << TEGRA210_MBDRC_THRESH_1ST_SHIFT) + +/* Fields for TEGRA210_MBDRC_RATIO_1ST */ +#define TEGRA210_MBDRC_RATIO_1ST_SHIFT 0 +#define TEGRA210_MBDRC_RATIO_1ST_MASK (0xffff << TEGRA210_MBDRC_RATIO_1ST_SHIFT) + +/* Fields for TEGRA210_MBDRC_RATIO_2ND */ +#define TEGRA210_MBDRC_RATIO_2ND_SHIFT 0 +#define TEGRA210_MBDRC_RATIO_2ND_MASK (0xffff << TEGRA210_MBDRC_RATIO_2ND_SHIFT) + +/* Fields for TEGRA210_MBDRC_RATIO_3RD */ +#define TEGRA210_MBDRC_RATIO_3RD_SHIFT 0 +#define TEGRA210_MBDRC_RATIO_3RD_MASK (0xffff << TEGRA210_MBDRC_RATIO_3RD_SHIFT) + +/* Fields for TEGRA210_MBDRC_RATIO_4TH */ +#define TEGRA210_MBDRC_RATIO_4TH_SHIFT 0 +#define TEGRA210_MBDRC_RATIO_4TH_MASK (0xffff << TEGRA210_MBDRC_RATIO_4TH_SHIFT) + +/* Fields for TEGRA210_MBDRC_RATIO_5TH */ +#define TEGRA210_MBDRC_RATIO_5TH_SHIFT 0 +#define TEGRA210_MBDRC_RATIO_5TH_MASK (0xffff << TEGRA210_MBDRC_RATIO_5TH_SHIFT) + +/* Fields for TEGRA210_MBDRC_MAKEUP_GAIN */ +#define TEGRA210_MBDRC_MAKEUP_GAIN_SHIFT 0 +#define TEGRA210_MBDRC_MAKEUP_GAIN_MASK (0x3f << TEGRA210_MBDRC_MAKEUP_GAIN_SHIFT) + +/* Fields for TEGRA210_MBDRC_INIT_GAIN */ +#define TEGRA210_MBDRC_INIT_GAIN_SHIFT 0 +#define TEGRA210_MBDRC_INIT_GAIN_MASK (0xffffffff << TEGRA210_MBDRC_INIT_GAIN_SHIFT) + +/* Fields for TEGRA210_MBDRC_GAIN_ATTACK */ +#define TEGRA210_MBDRC_GAIN_ATTACK_SHIFT 0 +#define TEGRA210_MBDRC_GAIN_ATTACK_MASK (0xffffffff << TEGRA210_MBDRC_GAIN_ATTACK_SHIFT) + +/* Fields for TEGRA210_MBDRC_GAIN_RELEASE */ +#define TEGRA210_MBDRC_GAIN_RELEASE_SHIFT 0 +#define TEGRA210_MBDRC_GAIN_RELEASE_MASK (0xffffffff << TEGRA210_MBDRC_GAIN_RELEASE_SHIFT) + +/* Fields for TEGRA210_MBDRC_FAST_RELEASE */ +#define TEGRA210_MBDRC_FAST_RELEASE_SHIFT 0 +#define TEGRA210_MBDRC_FAST_RELEASE_MASK (0xffffffff << TEGRA210_MBDRC_FAST_RELEASE_SHIFT) + +#define TEGRA210_MBDRC_RAM_CTRL_RW_READ 0 +#define TEGRA210_MBDRC_RAM_CTRL_RW_WRITE (1 << 14) +#define TEGRA210_MBDRC_RAM_CTRL_ADDR_INIT_EN (1 << 13) +#define TEGRA210_MBDRC_RAM_CTRL_SEQ_ACCESS_EN (1 << 12) +#define TEGRA210_MBDRC_RAM_CTRL_RAM_ADDR_MASK 0x1ff + +/* + * Order and size of each structure element for following structures should not + * be altered size order of elements and their size are based on PEQ co-eff ram + * and shift ram layout. + */ +#define TEGRA210_MBDRC_THRESHOLD_NUM 4 +#define TEGRA210_MBDRC_RATIO_NUM (TEGRA210_MBDRC_THRESHOLD_NUM + 1) +#define TEGRA210_MBDRC_MAX_BIQUAD_STAGES 8 + +/* Order of these enums are same as the order of band specific hw registers */ +enum { + MBDRC_LOW_BAND, + MBDRC_MID_BAND, + MBDRC_HIGH_BAND, + MBDRC_NUM_BAND, +}; + +struct tegra210_mbdrc_band_params { + u32 band; + u32 iir_stages; + u32 in_attack_tc; + u32 in_release_tc; + u32 fast_attack_tc; + u32 in_threshold[TEGRA210_MBDRC_THRESHOLD_NUM]; + u32 out_threshold[TEGRA210_MBDRC_THRESHOLD_NUM]; + u32 ratio[TEGRA210_MBDRC_RATIO_NUM]; + u32 makeup_gain; + u32 gain_init; + u32 gain_attack_tc; + u32 gain_release_tc; + u32 fast_release_tc; + /* For biquad_params[][5] order of coeff is b0, b1, a0, a1, a2 */ + u32 biquad_params[TEGRA210_MBDRC_MAX_BIQUAD_STAGES * 5]; +}; + +struct tegra210_mbdrc_config { + unsigned int mode; + unsigned int rms_off; + unsigned int peak_rms_mode; + unsigned int fliter_structure; + unsigned int shift_ctrl; + unsigned int frame_size; + unsigned int channel_mask; + unsigned int fa_factor; /* Fast attack factor */ + unsigned int fr_factor; /* Fast release factor */ + struct tegra210_mbdrc_band_params band_params[MBDRC_NUM_BAND]; +}; + +int tegra210_mbdrc_regmap_init(struct platform_device *pdev); +int tegra210_mbdrc_component_init(struct snd_soc_component *cmpnt); +int tegra210_mbdrc_hw_params(struct snd_soc_component *cmpnt); + +#endif diff --git a/sound/soc/tegra/tegra210_ope.c b/sound/soc/tegra/tegra210_ope.c new file mode 100644 index 000000000000..3dd2bdec657b --- /dev/null +++ b/sound/soc/tegra/tegra210_ope.c @@ -0,0 +1,419 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// tegra210_ope.c - Tegra210 OPE driver +// +// Copyright (c) 2022, NVIDIA CORPORATION. All rights reserved. + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "tegra210_mbdrc.h" +#include "tegra210_ope.h" +#include "tegra210_peq.h" +#include "tegra_cif.h" + +static const struct reg_default tegra210_ope_reg_defaults[] = { + { TEGRA210_OPE_RX_INT_MASK, 0x00000001}, + { TEGRA210_OPE_RX_CIF_CTRL, 0x00007700}, + { TEGRA210_OPE_TX_INT_MASK, 0x00000001}, + { TEGRA210_OPE_TX_CIF_CTRL, 0x00007700}, + { TEGRA210_OPE_CG, 0x1}, +}; + +static int tegra210_ope_set_audio_cif(struct tegra210_ope *ope, + struct snd_pcm_hw_params *params, + unsigned int reg) +{ + int channels, audio_bits; + struct tegra_cif_conf cif_conf; + + memset(&cif_conf, 0, sizeof(struct tegra_cif_conf)); + + channels = params_channels(params); + if (channels < 2) + return -EINVAL; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + audio_bits = TEGRA_ACIF_BITS_16; + break; + case SNDRV_PCM_FORMAT_S32_LE: + audio_bits = TEGRA_ACIF_BITS_32; + break; + default: + return -EINVAL; + } + + cif_conf.audio_ch = channels; + cif_conf.client_ch = channels; + cif_conf.audio_bits = audio_bits; + cif_conf.client_bits = audio_bits; + + tegra_set_cif(ope->regmap, reg, &cif_conf); + + return 0; +} + +static int tegra210_ope_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct device *dev = dai->dev; + struct tegra210_ope *ope = snd_soc_dai_get_drvdata(dai); + int err; + + /* Set RX and TX CIF */ + err = tegra210_ope_set_audio_cif(ope, params, + TEGRA210_OPE_RX_CIF_CTRL); + if (err) { + dev_err(dev, "Can't set OPE RX CIF: %d\n", err); + return err; + } + + err = tegra210_ope_set_audio_cif(ope, params, + TEGRA210_OPE_TX_CIF_CTRL); + if (err) { + dev_err(dev, "Can't set OPE TX CIF: %d\n", err); + return err; + } + + tegra210_mbdrc_hw_params(dai->component); + + return err; +} + +static int tegra210_ope_component_probe(struct snd_soc_component *cmpnt) +{ + struct tegra210_ope *ope = dev_get_drvdata(cmpnt->dev); + + tegra210_peq_component_init(cmpnt); + tegra210_mbdrc_component_init(cmpnt); + + /* + * The OPE, PEQ and MBDRC functionalities are combined under one + * device registered by OPE driver. In fact OPE HW block includes + * sub blocks PEQ and MBDRC. However driver registers separate + * regmap interfaces for each of these. ASoC core depends on + * dev_get_regmap() to populate the regmap field for a given ASoC + * component. A component can have one regmap reference and since + * the DAPM routes depend on OPE regmap only, below explicit + * assignment is done to highlight this. This is needed for ASoC + * core to access correct regmap during DAPM path setup. + */ + snd_soc_component_init_regmap(cmpnt, ope->regmap); + + return 0; +} + +static const struct snd_soc_dai_ops tegra210_ope_dai_ops = { + .hw_params = tegra210_ope_hw_params, +}; + +static struct snd_soc_dai_driver tegra210_ope_dais[] = { + { + .name = "OPE-RX-CIF", + .playback = { + .stream_name = "RX-CIF-Playback", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .capture = { + .stream_name = "RX-CIF-Capture", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + }, + { + .name = "OPE-TX-CIF", + .playback = { + .stream_name = "TX-CIF-Playback", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .capture = { + .stream_name = "TX-CIF-Capture", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .ops = &tegra210_ope_dai_ops, + } +}; + +static const struct snd_soc_dapm_widget tegra210_ope_widgets[] = { + SND_SOC_DAPM_AIF_IN("RX", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("TX", NULL, 0, TEGRA210_OPE_ENABLE, + TEGRA210_OPE_EN_SHIFT, 0), +}; + +#define OPE_ROUTES(sname) \ + { "RX XBAR-" sname, NULL, "XBAR-TX" }, \ + { "RX-CIF-" sname, NULL, "RX XBAR-" sname }, \ + { "RX", NULL, "RX-CIF-" sname }, \ + { "TX-CIF-" sname, NULL, "TX" }, \ + { "TX XBAR-" sname, NULL, "TX-CIF-" sname }, \ + { "XBAR-RX", NULL, "TX XBAR-" sname } + +static const struct snd_soc_dapm_route tegra210_ope_routes[] = { + { "TX", NULL, "RX" }, + OPE_ROUTES("Playback"), + OPE_ROUTES("Capture"), +}; + +static const char * const tegra210_ope_data_dir_text[] = { + "MBDRC to PEQ", + "PEQ to MBDRC" +}; + +static const struct soc_enum tegra210_ope_data_dir_enum = + SOC_ENUM_SINGLE(TEGRA210_OPE_DIR, TEGRA210_OPE_DIR_SHIFT, + 2, tegra210_ope_data_dir_text); + +static int tegra210_ope_get_data_dir(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct tegra210_ope *ope = snd_soc_component_get_drvdata(cmpnt); + + ucontrol->value.enumerated.item[0] = ope->data_dir; + + return 0; +} + +static int tegra210_ope_put_data_dir(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct tegra210_ope *ope = snd_soc_component_get_drvdata(cmpnt); + unsigned int value = ucontrol->value.enumerated.item[0]; + + if (value == ope->data_dir) + return 0; + + ope->data_dir = value; + + return 1; +} + +static const struct snd_kcontrol_new tegra210_ope_controls[] = { + SOC_ENUM_EXT("Data Flow Direction", tegra210_ope_data_dir_enum, + tegra210_ope_get_data_dir, tegra210_ope_put_data_dir), +}; + +static const struct snd_soc_component_driver tegra210_ope_cmpnt = { + .probe = tegra210_ope_component_probe, + .dapm_widgets = tegra210_ope_widgets, + .num_dapm_widgets = ARRAY_SIZE(tegra210_ope_widgets), + .dapm_routes = tegra210_ope_routes, + .num_dapm_routes = ARRAY_SIZE(tegra210_ope_routes), + .controls = tegra210_ope_controls, + .num_controls = ARRAY_SIZE(tegra210_ope_controls), +}; + +static bool tegra210_ope_wr_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TEGRA210_OPE_RX_INT_MASK ... TEGRA210_OPE_RX_CIF_CTRL: + case TEGRA210_OPE_TX_INT_MASK ... TEGRA210_OPE_TX_CIF_CTRL: + case TEGRA210_OPE_ENABLE ... TEGRA210_OPE_CG: + case TEGRA210_OPE_DIR: + return true; + default: + return false; + } +} + +static bool tegra210_ope_rd_reg(struct device *dev, unsigned int reg) +{ + if (tegra210_ope_wr_reg(dev, reg)) + return true; + + switch (reg) { + case TEGRA210_OPE_RX_STATUS: + case TEGRA210_OPE_RX_INT_STATUS: + case TEGRA210_OPE_TX_STATUS: + case TEGRA210_OPE_TX_INT_STATUS: + case TEGRA210_OPE_STATUS: + case TEGRA210_OPE_INT_STATUS: + return true; + default: + return false; + } +} + +static bool tegra210_ope_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TEGRA210_OPE_RX_STATUS: + case TEGRA210_OPE_RX_INT_STATUS: + case TEGRA210_OPE_TX_STATUS: + case TEGRA210_OPE_TX_INT_STATUS: + case TEGRA210_OPE_SOFT_RESET: + case TEGRA210_OPE_STATUS: + case TEGRA210_OPE_INT_STATUS: + return true; + default: + return false; + } +} + +static const struct regmap_config tegra210_ope_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = TEGRA210_OPE_DIR, + .writeable_reg = tegra210_ope_wr_reg, + .readable_reg = tegra210_ope_rd_reg, + .volatile_reg = tegra210_ope_volatile_reg, + .reg_defaults = tegra210_ope_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(tegra210_ope_reg_defaults), + .cache_type = REGCACHE_FLAT, +}; + +static int tegra210_ope_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct tegra210_ope *ope; + void __iomem *regs; + int err; + + ope = devm_kzalloc(dev, sizeof(*ope), GFP_KERNEL); + if (!ope) + return -ENOMEM; + + regs = devm_platform_ioremap_resource(pdev, 0); + if (IS_ERR(regs)) + return PTR_ERR(regs); + + ope->regmap = devm_regmap_init_mmio(dev, regs, + &tegra210_ope_regmap_config); + if (IS_ERR(ope->regmap)) { + dev_err(dev, "regmap init failed\n"); + return PTR_ERR(ope->regmap); + } + + regcache_cache_only(ope->regmap, true); + + dev_set_drvdata(dev, ope); + + err = tegra210_peq_regmap_init(pdev); + if (err < 0) { + dev_err(dev, "PEQ init failed\n"); + return err; + } + + err = tegra210_mbdrc_regmap_init(pdev); + if (err < 0) { + dev_err(dev, "MBDRC init failed\n"); + return err; + } + + err = devm_snd_soc_register_component(dev, &tegra210_ope_cmpnt, + tegra210_ope_dais, + ARRAY_SIZE(tegra210_ope_dais)); + if (err) { + dev_err(dev, "can't register OPE component, err: %d\n", err); + return err; + } + + pm_runtime_enable(dev); + + return 0; +} + +static int tegra210_ope_remove(struct platform_device *pdev) +{ + pm_runtime_disable(&pdev->dev); + + return 0; +} + +static int __maybe_unused tegra210_ope_runtime_suspend(struct device *dev) +{ + struct tegra210_ope *ope = dev_get_drvdata(dev); + + tegra210_peq_save(ope->peq_regmap, ope->peq_biquad_gains, + ope->peq_biquad_shifts); + + regcache_cache_only(ope->mbdrc_regmap, true); + regcache_cache_only(ope->peq_regmap, true); + regcache_cache_only(ope->regmap, true); + + regcache_mark_dirty(ope->regmap); + regcache_mark_dirty(ope->peq_regmap); + regcache_mark_dirty(ope->mbdrc_regmap); + + return 0; +} + +static int __maybe_unused tegra210_ope_runtime_resume(struct device *dev) +{ + struct tegra210_ope *ope = dev_get_drvdata(dev); + + regcache_cache_only(ope->regmap, false); + regcache_cache_only(ope->peq_regmap, false); + regcache_cache_only(ope->mbdrc_regmap, false); + + regcache_sync(ope->regmap); + regcache_sync(ope->peq_regmap); + regcache_sync(ope->mbdrc_regmap); + + tegra210_peq_restore(ope->peq_regmap, ope->peq_biquad_gains, + ope->peq_biquad_shifts); + + return 0; +} + +static const struct dev_pm_ops tegra210_ope_pm_ops = { + SET_RUNTIME_PM_OPS(tegra210_ope_runtime_suspend, + tegra210_ope_runtime_resume, NULL) + SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, + pm_runtime_force_resume) +}; + +static const struct of_device_id tegra210_ope_of_match[] = { + { .compatible = "nvidia,tegra210-ope" }, + {}, +}; +MODULE_DEVICE_TABLE(of, tegra210_ope_of_match); + +static struct platform_driver tegra210_ope_driver = { + .driver = { + .name = "tegra210-ope", + .of_match_table = tegra210_ope_of_match, + .pm = &tegra210_ope_pm_ops, + }, + .probe = tegra210_ope_probe, + .remove = tegra210_ope_remove, +}; +module_platform_driver(tegra210_ope_driver) + +MODULE_AUTHOR("Sumit Bhattacharya "); +MODULE_DESCRIPTION("Tegra210 OPE ASoC driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/tegra/tegra210_ope.h b/sound/soc/tegra/tegra210_ope.h new file mode 100644 index 000000000000..2835af6ce631 --- /dev/null +++ b/sound/soc/tegra/tegra210_ope.h @@ -0,0 +1,90 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * tegra210_ope.h - Definitions for Tegra210 OPE driver + * + * Copyright (c) 2022, NVIDIA CORPORATION. All rights reserved. + * + */ + +#ifndef __TEGRA210_OPE_H__ +#define __TEGRA210_OPE_H__ + +#include +#include + +#include "tegra210_peq.h" + +/* + * OPE_RX registers are with respect to XBAR. + * The data comes from XBAR to OPE + */ +#define TEGRA210_OPE_RX_STATUS 0xc +#define TEGRA210_OPE_RX_INT_STATUS 0x10 +#define TEGRA210_OPE_RX_INT_MASK 0x14 +#define TEGRA210_OPE_RX_INT_SET 0x18 +#define TEGRA210_OPE_RX_INT_CLEAR 0x1c +#define TEGRA210_OPE_RX_CIF_CTRL 0x20 + +/* + * OPE_TX registers are with respect to XBAR. + * The data goes out from OPE to XBAR + */ +#define TEGRA210_OPE_TX_STATUS 0x4c +#define TEGRA210_OPE_TX_INT_STATUS 0x50 +#define TEGRA210_OPE_TX_INT_MASK 0x54 +#define TEGRA210_OPE_TX_INT_SET 0x58 +#define TEGRA210_OPE_TX_INT_CLEAR 0x5c +#define TEGRA210_OPE_TX_CIF_CTRL 0x60 + +/* OPE Gloabal registers */ +#define TEGRA210_OPE_ENABLE 0x80 +#define TEGRA210_OPE_SOFT_RESET 0x84 +#define TEGRA210_OPE_CG 0x88 +#define TEGRA210_OPE_STATUS 0x8c +#define TEGRA210_OPE_INT_STATUS 0x90 +#define TEGRA210_OPE_DIR 0x94 + +/* Fields for TEGRA210_OPE_ENABLE */ +#define TEGRA210_OPE_EN_SHIFT 0 +#define TEGRA210_OPE_EN (1 << TEGRA210_OPE_EN_SHIFT) + +/* Fields for TEGRA210_OPE_SOFT_RESET */ +#define TEGRA210_OPE_SOFT_RESET_SHIFT 0 +#define TEGRA210_OPE_SOFT_RESET_EN (1 << TEGRA210_OPE_SOFT_RESET_SHIFT) + +#define TEGRA210_OPE_DIR_SHIFT 0 + +struct tegra210_ope { + struct regmap *regmap; + struct regmap *peq_regmap; + struct regmap *mbdrc_regmap; + u32 peq_biquad_gains[TEGRA210_PEQ_GAIN_PARAM_SIZE_PER_CH]; + u32 peq_biquad_shifts[TEGRA210_PEQ_SHIFT_PARAM_SIZE_PER_CH]; + unsigned int data_dir; +}; + +/* Extension of soc_bytes structure defined in sound/soc.h */ +struct tegra_soc_bytes { + struct soc_bytes soc; + u32 shift; /* Used as offset for AHUB RAM related programing */ +}; + +/* Utility structures for using mixer control of type snd_soc_bytes */ +#define TEGRA_SOC_BYTES_EXT(xname, xbase, xregs, xshift, xmask, \ + xhandler_get, xhandler_put, xinfo) \ +{ \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .info = xinfo, \ + .get = xhandler_get, \ + .put = xhandler_put, \ + .private_value = ((unsigned long)&(struct tegra_soc_bytes) \ + { \ + .soc.base = xbase, \ + .soc.num_regs = xregs, \ + .soc.mask = xmask, \ + .shift = xshift \ + }) \ +} + +#endif diff --git a/sound/soc/tegra/tegra210_peq.c b/sound/soc/tegra/tegra210_peq.c new file mode 100644 index 000000000000..205d956abb42 --- /dev/null +++ b/sound/soc/tegra/tegra210_peq.c @@ -0,0 +1,434 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// tegra210_peq.c - Tegra210 PEQ driver +// +// Copyright (c) 2022, NVIDIA CORPORATION. All rights reserved. + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "tegra210_ope.h" +#include "tegra210_peq.h" + +static const struct reg_default tegra210_peq_reg_defaults[] = { + { TEGRA210_PEQ_CFG, 0x00000013}, + { TEGRA210_PEQ_CFG_RAM_CTRL, 0x00004000}, + { TEGRA210_PEQ_CFG_RAM_SHIFT_CTRL, 0x00004000}, +}; + +static const u32 biquad_init_gains[TEGRA210_PEQ_GAIN_PARAM_SIZE_PER_CH] = { + 1495012349, /* Pre-gain */ + + /* Gains : b0, b1, a0, a1, a2 */ + 536870912, -1073741824, 536870912, 2143508246, -1069773768, /* Band-0 */ + 134217728, -265414508, 131766272, 2140402222, -1071252997, /* Band-1 */ + 268435456, -233515765, -33935948, 1839817267, -773826124, /* Band-2 */ + 536870912, -672537913, 139851540, 1886437554, -824433167, /* Band-3 */ + 268435456, -114439279, 173723964, 205743566, 278809729, /* Band-4 */ + 1, 0, 0, 0, 0, /* Band-5 */ + 1, 0, 0, 0, 0, /* Band-6 */ + 1, 0, 0, 0, 0, /* Band-7 */ + 1, 0, 0, 0, 0, /* Band-8 */ + 1, 0, 0, 0, 0, /* Band-9 */ + 1, 0, 0, 0, 0, /* Band-10 */ + 1, 0, 0, 0, 0, /* Band-11 */ + + 963423114, /* Post-gain */ +}; + +static const u32 biquad_init_shifts[TEGRA210_PEQ_SHIFT_PARAM_SIZE_PER_CH] = { + 23, /* Pre-shift */ + 30, 30, 30, 30, 30, 0, 0, 0, 0, 0, 0, 0, /* Shift for bands */ + 28, /* Post-shift */ +}; + +static s32 biquad_coeff_buffer[TEGRA210_PEQ_GAIN_PARAM_SIZE_PER_CH]; + +static void tegra210_peq_read_ram(struct regmap *regmap, unsigned int reg_ctrl, + unsigned int reg_data, unsigned int ram_offset, + unsigned int *data, size_t size) +{ + unsigned int val; + unsigned int i; + + val = ram_offset & TEGRA210_PEQ_RAM_CTRL_RAM_ADDR_MASK; + val |= TEGRA210_PEQ_RAM_CTRL_ADDR_INIT_EN; + val |= TEGRA210_PEQ_RAM_CTRL_SEQ_ACCESS_EN; + val |= TEGRA210_PEQ_RAM_CTRL_RW_READ; + + regmap_write(regmap, reg_ctrl, val); + + /* + * Since all ahub non-io modules work under same ahub clock it is not + * necessary to check ahub read busy bit after every read. + */ + for (i = 0; i < size; i++) + regmap_read(regmap, reg_data, &data[i]); +} + +static void tegra210_peq_write_ram(struct regmap *regmap, unsigned int reg_ctrl, + unsigned int reg_data, unsigned int ram_offset, + unsigned int *data, size_t size) +{ + unsigned int val; + unsigned int i; + + val = ram_offset & TEGRA210_PEQ_RAM_CTRL_RAM_ADDR_MASK; + val |= TEGRA210_PEQ_RAM_CTRL_ADDR_INIT_EN; + val |= TEGRA210_PEQ_RAM_CTRL_SEQ_ACCESS_EN; + val |= TEGRA210_PEQ_RAM_CTRL_RW_WRITE; + + regmap_write(regmap, reg_ctrl, val); + + for (i = 0; i < size; i++) + regmap_write(regmap, reg_data, data[i]); +} + +static int tegra210_peq_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct tegra210_ope *ope = snd_soc_component_get_drvdata(cmpnt); + unsigned int mask = (1 << fls(mc->max)) - 1; + unsigned int val; + + regmap_read(ope->peq_regmap, mc->reg, &val); + + ucontrol->value.integer.value[0] = (val >> mc->shift) & mask; + + if (!mc->invert) + return 0; + + ucontrol->value.integer.value[0] = + mc->max - ucontrol->value.integer.value[0]; + + return 0; +} + +static int tegra210_peq_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct tegra210_ope *ope = snd_soc_component_get_drvdata(cmpnt); + unsigned int mask = (1 << fls(mc->max)) - 1; + bool change = false; + unsigned int val; + + val = (ucontrol->value.integer.value[0] & mask); + + if (mc->invert) + val = mc->max - val; + + val = val << mc->shift; + + regmap_update_bits_check(ope->peq_regmap, mc->reg, (mask << mc->shift), + val, &change); + + return change ? 1 : 0; +} + +static int tegra210_peq_ram_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tegra_soc_bytes *params = (void *)kcontrol->private_value; + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct tegra210_ope *ope = snd_soc_component_get_drvdata(cmpnt); + u32 i, reg_ctrl = params->soc.base; + u32 reg_data = reg_ctrl + cmpnt->val_bytes; + s32 *data = (s32 *)biquad_coeff_buffer; + + pm_runtime_get_sync(cmpnt->dev); + + tegra210_peq_read_ram(ope->peq_regmap, reg_ctrl, reg_data, + params->shift, data, params->soc.num_regs); + + pm_runtime_put_sync(cmpnt->dev); + + for (i = 0; i < params->soc.num_regs; i++) + ucontrol->value.integer.value[i] = (long)data[i]; + + return 0; +} + +static int tegra210_peq_ram_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tegra_soc_bytes *params = (void *)kcontrol->private_value; + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct tegra210_ope *ope = snd_soc_component_get_drvdata(cmpnt); + u32 i, reg_ctrl = params->soc.base; + u32 reg_data = reg_ctrl + cmpnt->val_bytes; + s32 *data = (s32 *)biquad_coeff_buffer; + + for (i = 0; i < params->soc.num_regs; i++) + data[i] = (s32)ucontrol->value.integer.value[i]; + + pm_runtime_get_sync(cmpnt->dev); + + tegra210_peq_write_ram(ope->peq_regmap, reg_ctrl, reg_data, + params->shift, data, params->soc.num_regs); + + pm_runtime_put_sync(cmpnt->dev); + + return 1; +} + +static int tegra210_peq_param_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct soc_bytes *params = (void *)kcontrol->private_value; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->value.integer.min = INT_MIN; + uinfo->value.integer.max = INT_MAX; + uinfo->count = params->num_regs; + + return 0; +} + +#define TEGRA210_PEQ_GAIN_PARAMS_CTRL(chan) \ + TEGRA_SOC_BYTES_EXT("PEQ Channel-" #chan " Biquad Gain Params", \ + TEGRA210_PEQ_CFG_RAM_CTRL, \ + TEGRA210_PEQ_GAIN_PARAM_SIZE_PER_CH, \ + (TEGRA210_PEQ_GAIN_PARAM_SIZE_PER_CH * chan), 0xffffffff, \ + tegra210_peq_ram_get, tegra210_peq_ram_put, \ + tegra210_peq_param_info) + +#define TEGRA210_PEQ_SHIFT_PARAMS_CTRL(chan) \ + TEGRA_SOC_BYTES_EXT("PEQ Channel-" #chan " Biquad Shift Params", \ + TEGRA210_PEQ_CFG_RAM_SHIFT_CTRL, \ + TEGRA210_PEQ_SHIFT_PARAM_SIZE_PER_CH, \ + (TEGRA210_PEQ_SHIFT_PARAM_SIZE_PER_CH * chan), 0x1f, \ + tegra210_peq_ram_get, tegra210_peq_ram_put, \ + tegra210_peq_param_info) + +static const struct snd_kcontrol_new tegra210_peq_controls[] = { + SOC_SINGLE_EXT("PEQ Active", TEGRA210_PEQ_CFG, + TEGRA210_PEQ_CFG_MODE_SHIFT, 1, 0, + tegra210_peq_get, tegra210_peq_put), + + SOC_SINGLE_EXT("PEQ Biquad Stages", TEGRA210_PEQ_CFG, + TEGRA210_PEQ_CFG_BIQUAD_STAGES_SHIFT, + TEGRA210_PEQ_MAX_BIQUAD_STAGES - 1, 0, + tegra210_peq_get, tegra210_peq_put), + + TEGRA210_PEQ_GAIN_PARAMS_CTRL(0), + TEGRA210_PEQ_GAIN_PARAMS_CTRL(1), + TEGRA210_PEQ_GAIN_PARAMS_CTRL(2), + TEGRA210_PEQ_GAIN_PARAMS_CTRL(3), + TEGRA210_PEQ_GAIN_PARAMS_CTRL(4), + TEGRA210_PEQ_GAIN_PARAMS_CTRL(5), + TEGRA210_PEQ_GAIN_PARAMS_CTRL(6), + TEGRA210_PEQ_GAIN_PARAMS_CTRL(7), + + TEGRA210_PEQ_SHIFT_PARAMS_CTRL(0), + TEGRA210_PEQ_SHIFT_PARAMS_CTRL(1), + TEGRA210_PEQ_SHIFT_PARAMS_CTRL(2), + TEGRA210_PEQ_SHIFT_PARAMS_CTRL(3), + TEGRA210_PEQ_SHIFT_PARAMS_CTRL(4), + TEGRA210_PEQ_SHIFT_PARAMS_CTRL(5), + TEGRA210_PEQ_SHIFT_PARAMS_CTRL(6), + TEGRA210_PEQ_SHIFT_PARAMS_CTRL(7), +}; + +static bool tegra210_peq_wr_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TEGRA210_PEQ_SOFT_RESET: + case TEGRA210_PEQ_CG: + case TEGRA210_PEQ_CFG ... TEGRA210_PEQ_CFG_RAM_SHIFT_DATA: + return true; + default: + return false; + } +} + +static bool tegra210_peq_rd_reg(struct device *dev, unsigned int reg) +{ + if (tegra210_peq_wr_reg(dev, reg)) + return true; + + switch (reg) { + case TEGRA210_PEQ_STATUS: + return true; + default: + return false; + } +} + +static bool tegra210_peq_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TEGRA210_PEQ_SOFT_RESET: + case TEGRA210_PEQ_STATUS: + case TEGRA210_PEQ_CFG_RAM_CTRL ... TEGRA210_PEQ_CFG_RAM_SHIFT_DATA: + return true; + default: + return false; + } +} + +static bool tegra210_peq_precious_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TEGRA210_PEQ_CFG_RAM_DATA: + case TEGRA210_PEQ_CFG_RAM_SHIFT_DATA: + return true; + default: + return false; + } +} + +static const struct regmap_config tegra210_peq_regmap_config = { + .name = "peq", + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = TEGRA210_PEQ_CFG_RAM_SHIFT_DATA, + .writeable_reg = tegra210_peq_wr_reg, + .readable_reg = tegra210_peq_rd_reg, + .volatile_reg = tegra210_peq_volatile_reg, + .precious_reg = tegra210_peq_precious_reg, + .reg_defaults = tegra210_peq_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(tegra210_peq_reg_defaults), + .cache_type = REGCACHE_FLAT, +}; + +void tegra210_peq_restore(struct regmap *regmap, u32 *biquad_gains, + u32 *biquad_shifts) +{ + unsigned int i; + + for (i = 0; i < TEGRA210_PEQ_MAX_CHANNELS; i++) { + tegra210_peq_write_ram(regmap, TEGRA210_PEQ_CFG_RAM_CTRL, + TEGRA210_PEQ_CFG_RAM_DATA, + (i * TEGRA210_PEQ_GAIN_PARAM_SIZE_PER_CH), + biquad_gains, + TEGRA210_PEQ_GAIN_PARAM_SIZE_PER_CH); + + tegra210_peq_write_ram(regmap, + TEGRA210_PEQ_CFG_RAM_SHIFT_CTRL, + TEGRA210_PEQ_CFG_RAM_SHIFT_DATA, + (i * TEGRA210_PEQ_SHIFT_PARAM_SIZE_PER_CH), + biquad_shifts, + TEGRA210_PEQ_SHIFT_PARAM_SIZE_PER_CH); + + } +} + +void tegra210_peq_save(struct regmap *regmap, u32 *biquad_gains, + u32 *biquad_shifts) +{ + unsigned int i; + + for (i = 0; i < TEGRA210_PEQ_MAX_CHANNELS; i++) { + tegra210_peq_read_ram(regmap, + TEGRA210_PEQ_CFG_RAM_CTRL, + TEGRA210_PEQ_CFG_RAM_DATA, + (i * TEGRA210_PEQ_GAIN_PARAM_SIZE_PER_CH), + biquad_gains, + TEGRA210_PEQ_GAIN_PARAM_SIZE_PER_CH); + + tegra210_peq_read_ram(regmap, + TEGRA210_PEQ_CFG_RAM_SHIFT_CTRL, + TEGRA210_PEQ_CFG_RAM_SHIFT_DATA, + (i * TEGRA210_PEQ_SHIFT_PARAM_SIZE_PER_CH), + biquad_shifts, + TEGRA210_PEQ_SHIFT_PARAM_SIZE_PER_CH); + } +} + +int tegra210_peq_component_init(struct snd_soc_component *cmpnt) +{ + struct tegra210_ope *ope = snd_soc_component_get_drvdata(cmpnt); + unsigned int i; + + pm_runtime_get_sync(cmpnt->dev); + regmap_update_bits(ope->peq_regmap, TEGRA210_PEQ_CFG, + TEGRA210_PEQ_CFG_MODE_MASK, + 0 << TEGRA210_PEQ_CFG_MODE_SHIFT); + regmap_update_bits(ope->peq_regmap, TEGRA210_PEQ_CFG, + TEGRA210_PEQ_CFG_BIQUAD_STAGES_MASK, + (TEGRA210_PEQ_BIQUAD_INIT_STAGE - 1) << + TEGRA210_PEQ_CFG_BIQUAD_STAGES_SHIFT); + + /* Initialize PEQ AHUB RAM with default params */ + for (i = 0; i < TEGRA210_PEQ_MAX_CHANNELS; i++) { + + /* Set default gain params */ + tegra210_peq_write_ram(ope->peq_regmap, + TEGRA210_PEQ_CFG_RAM_CTRL, + TEGRA210_PEQ_CFG_RAM_DATA, + (i * TEGRA210_PEQ_GAIN_PARAM_SIZE_PER_CH), + (u32 *)&biquad_init_gains, + TEGRA210_PEQ_GAIN_PARAM_SIZE_PER_CH); + + /* Set default shift params */ + tegra210_peq_write_ram(ope->peq_regmap, + TEGRA210_PEQ_CFG_RAM_SHIFT_CTRL, + TEGRA210_PEQ_CFG_RAM_SHIFT_DATA, + (i * TEGRA210_PEQ_SHIFT_PARAM_SIZE_PER_CH), + (u32 *)&biquad_init_shifts, + TEGRA210_PEQ_SHIFT_PARAM_SIZE_PER_CH); + + } + + pm_runtime_put_sync(cmpnt->dev); + + snd_soc_add_component_controls(cmpnt, tegra210_peq_controls, + ARRAY_SIZE(tegra210_peq_controls)); + + return 0; +} + +int tegra210_peq_regmap_init(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct tegra210_ope *ope = dev_get_drvdata(dev); + struct device_node *child; + struct resource mem; + void __iomem *regs; + int err; + + child = of_get_child_by_name(dev->of_node, "equalizer"); + if (!child) + return -ENODEV; + + err = of_address_to_resource(child, 0, &mem); + of_node_put(child); + if (err < 0) { + dev_err(dev, "fail to get PEQ resource\n"); + return err; + } + + mem.flags = IORESOURCE_MEM; + regs = devm_ioremap_resource(dev, &mem); + if (IS_ERR(regs)) + return PTR_ERR(regs); + ope->peq_regmap = devm_regmap_init_mmio(dev, regs, + &tegra210_peq_regmap_config); + if (IS_ERR(ope->peq_regmap)) { + dev_err(dev, "regmap init failed\n"); + return PTR_ERR(ope->peq_regmap); + } + + regcache_cache_only(ope->peq_regmap, true); + + return 0; +} diff --git a/sound/soc/tegra/tegra210_peq.h b/sound/soc/tegra/tegra210_peq.h new file mode 100644 index 000000000000..6d3de4ff05cc --- /dev/null +++ b/sound/soc/tegra/tegra210_peq.h @@ -0,0 +1,56 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * tegra210_peq.h - Definitions for Tegra210 PEQ driver + * + * Copyright (c) 2022, NVIDIA CORPORATION. All rights reserved. + * + */ + +#ifndef __TEGRA210_PEQ_H__ +#define __TEGRA210_PEQ_H__ + +#include +#include +#include + +/* Register offsets from PEQ base */ +#define TEGRA210_PEQ_SOFT_RESET 0x0 +#define TEGRA210_PEQ_CG 0x4 +#define TEGRA210_PEQ_STATUS 0x8 +#define TEGRA210_PEQ_CFG 0xc +#define TEGRA210_PEQ_CFG_RAM_CTRL 0x10 +#define TEGRA210_PEQ_CFG_RAM_DATA 0x14 +#define TEGRA210_PEQ_CFG_RAM_SHIFT_CTRL 0x18 +#define TEGRA210_PEQ_CFG_RAM_SHIFT_DATA 0x1c + +/* Fields in TEGRA210_PEQ_CFG */ +#define TEGRA210_PEQ_CFG_BIQUAD_STAGES_SHIFT 2 +#define TEGRA210_PEQ_CFG_BIQUAD_STAGES_MASK (0xf << TEGRA210_PEQ_CFG_BIQUAD_STAGES_SHIFT) + +#define TEGRA210_PEQ_CFG_MODE_SHIFT 0 +#define TEGRA210_PEQ_CFG_MODE_MASK (0x1 << TEGRA210_PEQ_CFG_MODE_SHIFT) + +#define TEGRA210_PEQ_RAM_CTRL_RW_READ 0 +#define TEGRA210_PEQ_RAM_CTRL_RW_WRITE (1 << 14) +#define TEGRA210_PEQ_RAM_CTRL_ADDR_INIT_EN (1 << 13) +#define TEGRA210_PEQ_RAM_CTRL_SEQ_ACCESS_EN (1 << 12) +#define TEGRA210_PEQ_RAM_CTRL_RAM_ADDR_MASK 0x1ff + +/* PEQ register definition ends here */ +#define TEGRA210_PEQ_MAX_BIQUAD_STAGES 12 + +#define TEGRA210_PEQ_MAX_CHANNELS 8 + +#define TEGRA210_PEQ_BIQUAD_INIT_STAGE 5 + +#define TEGRA210_PEQ_GAIN_PARAM_SIZE_PER_CH (2 + TEGRA210_PEQ_MAX_BIQUAD_STAGES * 5) +#define TEGRA210_PEQ_SHIFT_PARAM_SIZE_PER_CH (2 + TEGRA210_PEQ_MAX_BIQUAD_STAGES) + +int tegra210_peq_regmap_init(struct platform_device *pdev); +int tegra210_peq_component_init(struct snd_soc_component *cmpnt); +void tegra210_peq_restore(struct regmap *regmap, u32 *biquad_gains, + u32 *biquad_shifts); +void tegra210_peq_save(struct regmap *regmap, u32 *biquad_gains, + u32 *biquad_shifts); + +#endif -- cgit v1.2.3 From 7ee0910d03168535ffeea2f4ce924eebb3b24863 Mon Sep 17 00:00:00 2001 From: Sameer Pujar Date: Fri, 3 Jun 2022 12:06:09 +0530 Subject: ASoC: tegra: AHUB routes for OPE module Add AHUB routes for OPE module. The OPE module can be plugged into audio path as per the need. The routing controls can be used to setup the audio path with OPE similar to the already existing routes. The support is added on Tegra210 and later Tegra SoCs where OPE module is present. Signed-off-by: Sameer Pujar Link: https://lore.kernel.org/r/1654238172-16293-4-git-send-email-spujar@nvidia.com Signed-off-by: Mark Brown --- sound/soc/tegra/tegra210_ahub.c | 39 +++++++++++++++++++++++++++++++++++---- 1 file changed, 35 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra210_ahub.c b/sound/soc/tegra/tegra210_ahub.c index e1f90daea7a1..b38d205b69cc 100644 --- a/sound/soc/tegra/tegra210_ahub.c +++ b/sound/soc/tegra/tegra210_ahub.c @@ -170,6 +170,11 @@ static struct snd_soc_dai_driver tegra210_ahub_dais[] = { DAI(MIXER1 TX3), DAI(MIXER1 TX4), DAI(MIXER1 TX5), + /* XBAR -> OPE -> XBAR */ + DAI(OPE1 RX), + DAI(OPE1 TX), + DAI(OPE2 RX), + DAI(OPE2 TX), }; static struct snd_soc_dai_driver tegra186_ahub_dais[] = { @@ -294,6 +299,9 @@ static struct snd_soc_dai_driver tegra186_ahub_dais[] = { DAI(ASRC1 RX6), DAI(ASRC1 TX6), DAI(ASRC1 RX7), + /* XBAR -> OPE -> XBAR */ + DAI(OPE1 RX), + DAI(OPE1 TX), }; static const char * const tegra210_ahub_mux_texts[] = { @@ -337,6 +345,8 @@ static const char * const tegra210_ahub_mux_texts[] = { "MIXER1 TX3", "MIXER1 TX4", "MIXER1 TX5", + "OPE1", + "OPE2", }; static const char * const tegra186_ahub_mux_texts[] = { @@ -408,6 +418,7 @@ static const char * const tegra186_ahub_mux_texts[] = { "ASRC1 TX4", "ASRC1 TX5", "ASRC1 TX6", + "OPE1", }; static const unsigned int tegra210_ahub_mux_values[] = { @@ -459,6 +470,9 @@ static const unsigned int tegra210_ahub_mux_values[] = { MUX_VALUE(1, 2), MUX_VALUE(1, 3), MUX_VALUE(1, 4), + /* OPE */ + MUX_VALUE(2, 0), + MUX_VALUE(2, 1), }; static const unsigned int tegra186_ahub_mux_values[] = { @@ -540,6 +554,8 @@ static const unsigned int tegra186_ahub_mux_values[] = { MUX_VALUE(3, 27), MUX_VALUE(3, 28), MUX_VALUE(3, 29), + /* OPE */ + MUX_VALUE(2, 0), }; /* Controls for t210 */ @@ -584,6 +600,8 @@ MUX_ENUM_CTRL_DECL(t210_mixer17_tx, 0x26); MUX_ENUM_CTRL_DECL(t210_mixer18_tx, 0x27); MUX_ENUM_CTRL_DECL(t210_mixer19_tx, 0x28); MUX_ENUM_CTRL_DECL(t210_mixer110_tx, 0x29); +MUX_ENUM_CTRL_DECL(t210_ope1_tx, 0x40); +MUX_ENUM_CTRL_DECL(t210_ope2_tx, 0x41); /* Controls for t186 */ MUX_ENUM_CTRL_DECL_186(t186_admaif1_tx, 0x00); @@ -657,6 +675,7 @@ MUX_ENUM_CTRL_DECL_186(t186_asrc14_tx, 0x6f); MUX_ENUM_CTRL_DECL_186(t186_asrc15_tx, 0x70); MUX_ENUM_CTRL_DECL_186(t186_asrc16_tx, 0x71); MUX_ENUM_CTRL_DECL_186(t186_asrc17_tx, 0x72); +MUX_ENUM_CTRL_DECL_186(t186_ope1_tx, 0x40); /* Controls for t234 */ MUX_ENUM_CTRL_DECL_234(t234_mvc1_tx, 0x44); @@ -758,6 +777,8 @@ static const struct snd_soc_dapm_widget tegra210_ahub_widgets[] = { TX_WIDGETS("MIXER1 TX3"), TX_WIDGETS("MIXER1 TX4"), TX_WIDGETS("MIXER1 TX5"), + WIDGETS("OPE1", t210_ope1_tx), + WIDGETS("OPE2", t210_ope2_tx), }; static const struct snd_soc_dapm_widget tegra186_ahub_widgets[] = { @@ -867,6 +888,7 @@ static const struct snd_soc_dapm_widget tegra186_ahub_widgets[] = { TX_WIDGETS("ASRC1 TX4"), TX_WIDGETS("ASRC1 TX5"), TX_WIDGETS("ASRC1 TX6"), + WIDGETS("OPE1", t186_ope1_tx), }; static const struct snd_soc_dapm_widget tegra234_ahub_widgets[] = { @@ -976,6 +998,7 @@ static const struct snd_soc_dapm_widget tegra234_ahub_widgets[] = { TX_WIDGETS("ASRC1 TX4"), TX_WIDGETS("ASRC1 TX5"), TX_WIDGETS("ASRC1 TX6"), + WIDGETS("OPE1", t186_ope1_tx), }; #define TEGRA_COMMON_MUX_ROUTES(name) \ @@ -1018,7 +1041,11 @@ static const struct snd_soc_dapm_widget tegra234_ahub_widgets[] = { { name " Mux", "MIXER1 TX2", "MIXER1 TX2 XBAR-RX" }, \ { name " Mux", "MIXER1 TX3", "MIXER1 TX3 XBAR-RX" }, \ { name " Mux", "MIXER1 TX4", "MIXER1 TX4 XBAR-RX" }, \ - { name " Mux", "MIXER1 TX5", "MIXER1 TX5 XBAR-RX" }, + { name " Mux", "MIXER1 TX5", "MIXER1 TX5 XBAR-RX" }, \ + { name " Mux", "OPE1", "OPE1 XBAR-RX" }, + +#define TEGRA210_ONLY_MUX_ROUTES(name) \ + { name " Mux", "OPE2", "OPE2 XBAR-RX" }, #define TEGRA186_ONLY_MUX_ROUTES(name) \ { name " Mux", "ADMAIF11", "ADMAIF11 XBAR-RX" }, \ @@ -1050,10 +1077,11 @@ static const struct snd_soc_dapm_widget tegra234_ahub_widgets[] = { { name " Mux", "ASRC1 TX5", "ASRC1 TX5 XBAR-RX" }, \ { name " Mux", "ASRC1 TX6", "ASRC1 TX6 XBAR-RX" }, -#define TEGRA210_MUX_ROUTES(name) \ - TEGRA_COMMON_MUX_ROUTES(name) +#define TEGRA210_MUX_ROUTES(name) \ + TEGRA_COMMON_MUX_ROUTES(name) \ + TEGRA210_ONLY_MUX_ROUTES(name) -#define TEGRA186_MUX_ROUTES(name) \ +#define TEGRA186_MUX_ROUTES(name) \ TEGRA_COMMON_MUX_ROUTES(name) \ TEGRA186_ONLY_MUX_ROUTES(name) @@ -1121,6 +1149,8 @@ static const struct snd_soc_dapm_route tegra210_ahub_routes[] = { TEGRA210_MUX_ROUTES("MIXER1 RX8") TEGRA210_MUX_ROUTES("MIXER1 RX9") TEGRA210_MUX_ROUTES("MIXER1 RX10") + TEGRA210_MUX_ROUTES("OPE1") + TEGRA210_MUX_ROUTES("OPE2") }; static const struct snd_soc_dapm_route tegra186_ahub_routes[] = { @@ -1215,6 +1245,7 @@ static const struct snd_soc_dapm_route tegra186_ahub_routes[] = { TEGRA186_MUX_ROUTES("ASRC1 RX5") TEGRA186_MUX_ROUTES("ASRC1 RX6") TEGRA186_MUX_ROUTES("ASRC1 RX7") + TEGRA186_MUX_ROUTES("OPE1") }; static const struct snd_soc_component_driver tegra210_ahub_component = { -- cgit v1.2.3 From b5df2a7dca1cc6c66eee0005c92094855dc2028c Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Wed, 11 May 2022 18:23:51 +0200 Subject: ASoC: codecs: Add HD-Audio codec driver Add generic ASoC equivalent of ALSA HD-Audio codec. This codec is designed to follow HDA_DEV_LEGACY convention. Driver wrapps existing hda_codec.c handlers to prevent code duplication within the newly added code. Number of DAIs created is dependent on capabilities exposed by the codec itself. Because of this, single solution can be applied to support every single HD-Audio codec type. At the same time, through the ASoC topology, platform drivers may limit the number of endpoints available to the userspace as codec driver exposes BE DAIs only. Both hda_codec_probe() and hda_codec_remove() declare their expectations on device's usage_count and suspended-status. This is to catch any unexpected behavior as PM-related code for HD-Audio has been changing quite a bit throughout the years. In order for codec DAI list to reflect its actual PCM capabilities, PCMs need to be built and that can only happen once codec device is constructed. To do that, a valid component->card->snd_card pointer is needed. Said pointer will be provided by the framework once all card components are accounted for and their probing can begin. Usage of "binder" BE DAI solves the problem - codec can be listed as one of HD-Audio card components without declaring any actual BE DAIs statically. Relation with hdac_hda: Addition of parallel solution is motivated by behavioral differences between hdac_hda.c and its legacy equivalent found in sound/pci/hda e.g.: lack of dynamic, based on codec capabilities, resource allocation and high cost of removing such differences on actively used targets. Major goal of codec driver presented here is to follow HD-Audio legacy behavior in 1:1 fashion by becoming a wrapper. Doing so increases code coverage of the legacy code and reduces the maintenance cost for both solutions. Signed-off-by: Cezary Rojewski Link: https://lore.kernel.org/r/20220511162403.3987658-3-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 10 ++ sound/soc/codecs/Makefile | 2 + sound/soc/codecs/hda-dai.c | 102 ++++++++++++ sound/soc/codecs/hda.c | 395 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/hda.h | 19 +++ 5 files changed, 528 insertions(+) create mode 100644 sound/soc/codecs/hda-dai.c create mode 100644 sound/soc/codecs/hda.c create mode 100644 sound/soc/codecs/hda.h (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 6165db92a629..5a60633a196c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -937,6 +937,16 @@ config SND_SOC_HDAC_HDA tristate select SND_HDA +config SND_SOC_HDA + tristate "HD-Audio codec driver" + select SND_HDA_EXT_CORE + select SND_HDA + help + This enables HD-Audio codec support in ASoC subsystem. Compared + to SND_SOC_HDAC_HDA, driver's behavior is identical to HD-Audio + legacy solution - including the dynamic resource allocation + based on actual codec capabilities. + config SND_SOC_ICS43432 tristate "ICS43423 and compatible i2s microphones" diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 28dc4edfd01f..d32026ae326f 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -106,6 +106,7 @@ snd-soc-es8328-spi-objs := es8328-spi.o snd-soc-gtm601-objs := gtm601.o snd-soc-hdac-hdmi-objs := hdac_hdmi.o snd-soc-hdac-hda-objs := hdac_hda.o +snd-soc-hda-codec-objs := hda.o hda-dai.o snd-soc-ics43432-objs := ics43432.o snd-soc-inno-rk3036-objs := inno_rk3036.o snd-soc-isabelle-objs := isabelle.o @@ -458,6 +459,7 @@ obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o obj-$(CONFIG_SND_SOC_GTM601) += snd-soc-gtm601.o obj-$(CONFIG_SND_SOC_HDAC_HDMI) += snd-soc-hdac-hdmi.o obj-$(CONFIG_SND_SOC_HDAC_HDA) += snd-soc-hdac-hda.o +obj-$(CONFIG_SND_SOC_HDA) += snd-soc-hda-codec.o obj-$(CONFIG_SND_SOC_ICS43432) += snd-soc-ics43432.o obj-$(CONFIG_SND_SOC_INNO_RK3036) += snd-soc-inno-rk3036.o obj-$(CONFIG_SND_SOC_ISABELLE) += snd-soc-isabelle.o diff --git a/sound/soc/codecs/hda-dai.c b/sound/soc/codecs/hda-dai.c new file mode 100644 index 000000000000..5371ff086261 --- /dev/null +++ b/sound/soc/codecs/hda-dai.c @@ -0,0 +1,102 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// +// Author: Cezary Rojewski +// + +#include +#include +#include "hda.h" + +static int hda_codec_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) +{ + struct hda_pcm_stream *stream_info; + struct hda_codec *codec; + struct hda_pcm *pcm; + int ret; + + codec = dev_to_hda_codec(dai->dev); + stream_info = snd_soc_dai_get_dma_data(dai, substream); + pcm = container_of(stream_info, struct hda_pcm, stream[substream->stream]); + + dev_dbg(dai->dev, "open stream codec: %08x, info: %p, pcm: %p %s substream: %p\n", + codec->core.vendor_id, stream_info, pcm, pcm->name, substream); + + snd_hda_codec_pcm_get(pcm); + + ret = stream_info->ops.open(stream_info, codec, substream); + if (ret < 0) { + dev_err(dai->dev, "codec open failed: %d\n", ret); + snd_hda_codec_pcm_put(pcm); + return ret; + } + + return 0; +} + +static void hda_codec_dai_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) +{ + struct hda_pcm_stream *stream_info; + struct hda_codec *codec; + struct hda_pcm *pcm; + int ret; + + codec = dev_to_hda_codec(dai->dev); + stream_info = snd_soc_dai_get_dma_data(dai, substream); + pcm = container_of(stream_info, struct hda_pcm, stream[substream->stream]); + + dev_dbg(dai->dev, "close stream codec: %08x, info: %p, pcm: %p %s substream: %p\n", + codec->core.vendor_id, stream_info, pcm, pcm->name, substream); + + ret = stream_info->ops.close(stream_info, codec, substream); + if (ret < 0) + dev_err(dai->dev, "codec close failed: %d\n", ret); + + snd_hda_codec_pcm_put(pcm); +} + +static int hda_codec_dai_hw_free(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) +{ + struct hda_pcm_stream *stream_info; + struct hda_codec *codec; + + codec = dev_to_hda_codec(dai->dev); + stream_info = snd_soc_dai_get_dma_data(dai, substream); + + snd_hda_codec_cleanup(codec, stream_info, substream); + + return 0; +} + +static int hda_codec_dai_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct hda_pcm_stream *stream_info; + struct hdac_stream *stream; + struct hda_codec *codec; + unsigned int format; + int ret; + + codec = dev_to_hda_codec(dai->dev); + stream = substream->runtime->private_data; + stream_info = snd_soc_dai_get_dma_data(dai, substream); + format = snd_hdac_calc_stream_format(runtime->rate, runtime->channels, runtime->format, + runtime->sample_bits, 0); + + ret = snd_hda_codec_prepare(codec, stream_info, stream->stream_tag, format, substream); + if (ret < 0) { + dev_err(dai->dev, "codec prepare failed: %d\n", ret); + return ret; + } + + return 0; +} + +const struct snd_soc_dai_ops snd_soc_hda_codec_dai_ops = { + .startup = hda_codec_dai_startup, + .shutdown = hda_codec_dai_shutdown, + .hw_free = hda_codec_dai_hw_free, + .prepare = hda_codec_dai_prepare, +}; +EXPORT_SYMBOL_GPL(snd_soc_hda_codec_dai_ops); diff --git a/sound/soc/codecs/hda.c b/sound/soc/codecs/hda.c new file mode 100644 index 000000000000..edcb8bc6806b --- /dev/null +++ b/sound/soc/codecs/hda.c @@ -0,0 +1,395 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// +// Author: Cezary Rojewski +// + +#include +#include +#include +#include +#include +#include +#include "hda.h" + +static int hda_codec_create_dais(struct hda_codec *codec, int pcm_count, + struct snd_soc_dai_driver **drivers) +{ + struct device *dev = &codec->core.dev; + struct snd_soc_dai_driver *drvs; + struct hda_pcm *pcm; + int i; + + drvs = devm_kcalloc(dev, pcm_count, sizeof(*drvs), GFP_KERNEL); + if (!drvs) + return -ENOMEM; + + pcm = list_first_entry(&codec->pcm_list_head, struct hda_pcm, list); + + for (i = 0; i < pcm_count; i++, pcm = list_next_entry(pcm, list)) { + struct snd_soc_pcm_stream *stream; + int dir; + + dev_info(dev, "creating for %s %d\n", pcm->name, i); + drvs[i].id = i; + drvs[i].name = pcm->name; + drvs[i].ops = &snd_soc_hda_codec_dai_ops; + + dir = SNDRV_PCM_STREAM_PLAYBACK; + stream = &drvs[i].playback; + if (!pcm->stream[dir].substreams) { + dev_info(dev, "skipping playback dai for %s\n", pcm->name); + goto capture_dais; + } + + stream->stream_name = + devm_kasprintf(dev, GFP_KERNEL, "%s %s", pcm->name, + snd_pcm_direction_name(dir)); + if (!stream->stream_name) + return -ENOMEM; + stream->channels_min = pcm->stream[dir].channels_min; + stream->channels_max = pcm->stream[dir].channels_max; + stream->rates = pcm->stream[dir].rates; + stream->formats = pcm->stream[dir].formats; + stream->sig_bits = pcm->stream[dir].maxbps; + +capture_dais: + dir = SNDRV_PCM_STREAM_CAPTURE; + stream = &drvs[i].capture; + if (!pcm->stream[dir].substreams) { + dev_info(dev, "skipping capture dai for %s\n", pcm->name); + continue; + } + + stream->stream_name = + devm_kasprintf(dev, GFP_KERNEL, "%s %s", pcm->name, + snd_pcm_direction_name(dir)); + if (!stream->stream_name) + return -ENOMEM; + stream->channels_min = pcm->stream[dir].channels_min; + stream->channels_max = pcm->stream[dir].channels_max; + stream->rates = pcm->stream[dir].rates; + stream->formats = pcm->stream[dir].formats; + stream->sig_bits = pcm->stream[dir].maxbps; + } + + *drivers = drvs; + return 0; +} + +static int hda_codec_register_dais(struct hda_codec *codec, struct snd_soc_component *component) +{ + struct snd_soc_dai_driver *drvs = NULL; + struct snd_soc_dapm_context *dapm; + struct hda_pcm *pcm; + int ret, pcm_count = 0; + + if (list_empty(&codec->pcm_list_head)) + return -EINVAL; + list_for_each_entry(pcm, &codec->pcm_list_head, list) + pcm_count++; + + ret = hda_codec_create_dais(codec, pcm_count, &drvs); + if (ret < 0) + return ret; + + dapm = snd_soc_component_get_dapm(component); + + list_for_each_entry(pcm, &codec->pcm_list_head, list) { + struct snd_soc_dai *dai; + + dai = snd_soc_register_dai(component, drvs, false); + if (!dai) { + dev_err(component->dev, "register dai for %s failed\n", pcm->name); + return -EINVAL; + } + + ret = snd_soc_dapm_new_dai_widgets(dapm, dai); + if (ret < 0) { + dev_err(component->dev, "create widgets failed: %d\n", ret); + snd_soc_unregister_dai(dai); + return ret; + } + + snd_soc_dai_init_dma_data(dai, &pcm->stream[0], &pcm->stream[1]); + drvs++; + } + + return 0; +} + +static void hda_codec_unregister_dais(struct hda_codec *codec, + struct snd_soc_component *component) +{ + struct snd_soc_dai *dai, *save; + struct hda_pcm *pcm; + + for_each_component_dais_safe(component, dai, save) { + list_for_each_entry(pcm, &codec->pcm_list_head, list) { + if (strcmp(dai->driver->name, pcm->name)) + continue; + + if (dai->playback_widget) + snd_soc_dapm_free_widget(dai->playback_widget); + if (dai->capture_widget) + snd_soc_dapm_free_widget(dai->capture_widget); + snd_soc_unregister_dai(dai); + break; + } + } +} + +int hda_codec_probe_complete(struct hda_codec *codec) +{ + struct hdac_device *hdev = &codec->core; + struct hdac_bus *bus = hdev->bus; + int ret; + + ret = snd_hda_codec_build_controls(codec); + if (ret < 0) { + dev_err(&hdev->dev, "unable to create controls %d\n", ret); + goto out; + } + + /* Bus suspended codecs as it does not manage their pm */ + pm_runtime_set_active(&hdev->dev); + /* rpm was forbidden in snd_hda_codec_device_new() */ + snd_hda_codec_set_power_save(codec, 2000); + snd_hda_codec_register(codec); +out: + /* Complement pm_runtime_get_sync(bus) in probe */ + pm_runtime_mark_last_busy(bus->dev); + pm_runtime_put_autosuspend(bus->dev); + + return ret; +} +EXPORT_SYMBOL_GPL(hda_codec_probe_complete); + +/* Expects codec with usage_count=1 and status=suspended */ +static int hda_codec_probe(struct snd_soc_component *component) +{ + struct hda_codec *codec = dev_to_hda_codec(component->dev); + struct hdac_device *hdev = &codec->core; + struct hdac_bus *bus = hdev->bus; + struct hdac_ext_link *hlink; + hda_codec_patch_t patch; + int ret; + +#ifdef CONFIG_PM + WARN_ON(atomic_read(&hdev->dev.power.usage_count) != 1 || + !pm_runtime_status_suspended(&hdev->dev)); +#endif + + hlink = snd_hdac_ext_bus_link_at(bus, hdev->addr); + if (!hlink) { + dev_err(&hdev->dev, "hdac link not found\n"); + return -EIO; + } + + pm_runtime_get_sync(bus->dev); + if (hda_codec_is_display(codec)) + snd_hdac_display_power(bus, hdev->addr, true); + snd_hdac_ext_bus_link_get(bus, hlink); + + ret = snd_hda_codec_device_new(codec->bus, component->card->snd_card, hdev->addr, codec, + false); + if (ret < 0) { + dev_err(&hdev->dev, "create hda codec failed: %d\n", ret); + goto device_new_err; + } + + ret = snd_hda_codec_set_name(codec, codec->preset->name); + if (ret < 0) { + dev_err(&hdev->dev, "name failed %s\n", codec->preset->name); + goto err; + } + + ret = snd_hdac_regmap_init(&codec->core); + if (ret < 0) { + dev_err(&hdev->dev, "regmap init failed\n"); + goto err; + } + + patch = (hda_codec_patch_t)codec->preset->driver_data; + if (!patch) { + dev_err(&hdev->dev, "no patch specified?\n"); + ret = -EINVAL; + goto err; + } + + ret = patch(codec); + if (ret < 0) { + dev_err(&hdev->dev, "patch failed %d\n", ret); + goto err; + } + + /* configure codec for 1:1 PCM:DAI mapping */ + codec->mst_no_extra_pcms = 1; + + ret = snd_hda_codec_parse_pcms(codec); + if (ret < 0) { + dev_err(&hdev->dev, "unable to map pcms to dai %d\n", ret); + goto parse_pcms_err; + } + + ret = hda_codec_register_dais(codec, component); + if (ret < 0) { + dev_err(&hdev->dev, "update dais failed: %d\n", ret); + goto parse_pcms_err; + } + + if (!hda_codec_is_display(codec)) { + ret = hda_codec_probe_complete(codec); + if (ret < 0) + goto complete_err; + } + + codec->core.lazy_cache = true; + + return 0; + +complete_err: + hda_codec_unregister_dais(codec, component); +parse_pcms_err: + if (codec->patch_ops.free) + codec->patch_ops.free(codec); +err: + snd_hda_codec_cleanup_for_unbind(codec); +device_new_err: + if (hda_codec_is_display(codec)) + snd_hdac_display_power(bus, hdev->addr, false); + + snd_hdac_ext_bus_link_put(bus, hlink); + + pm_runtime_mark_last_busy(bus->dev); + pm_runtime_put_autosuspend(bus->dev); + return ret; +} + +/* Leaves codec with usage_count=1 and status=suspended */ +static void hda_codec_remove(struct snd_soc_component *component) +{ + struct hda_codec *codec = dev_to_hda_codec(component->dev); + struct hdac_device *hdev = &codec->core; + struct hdac_bus *bus = hdev->bus; + struct hdac_ext_link *hlink; + bool was_registered = codec->registered; + + /* Don't allow any more runtime suspends */ + pm_runtime_forbid(&hdev->dev); + + hda_codec_unregister_dais(codec, component); + + if (codec->patch_ops.free) + codec->patch_ops.free(codec); + + snd_hda_codec_cleanup_for_unbind(codec); + pm_runtime_put_noidle(&hdev->dev); + /* snd_hdac_device_exit() is only called on bus remove */ + pm_runtime_set_suspended(&hdev->dev); + + if (hda_codec_is_display(codec)) + snd_hdac_display_power(bus, hdev->addr, false); + + hlink = snd_hdac_ext_bus_link_at(bus, hdev->addr); + if (hlink) + snd_hdac_ext_bus_link_put(bus, hlink); + /* + * HDMI card's hda_codec_probe_complete() (see late_probe()) may + * not be called due to early error, leaving bus uc unbalanced + */ + if (!was_registered) { + pm_runtime_mark_last_busy(bus->dev); + pm_runtime_put_autosuspend(bus->dev); + } + +#ifdef CONFIG_PM + WARN_ON(atomic_read(&hdev->dev.power.usage_count) != 1 || + !pm_runtime_status_suspended(&hdev->dev)); +#endif +} + +static const struct snd_soc_dapm_route hda_dapm_routes[] = { + {"AIF1TX", NULL, "Codec Input Pin1"}, + {"AIF2TX", NULL, "Codec Input Pin2"}, + {"AIF3TX", NULL, "Codec Input Pin3"}, + + {"Codec Output Pin1", NULL, "AIF1RX"}, + {"Codec Output Pin2", NULL, "AIF2RX"}, + {"Codec Output Pin3", NULL, "AIF3RX"}, +}; + +static const struct snd_soc_dapm_widget hda_dapm_widgets[] = { + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIF1RX", "Analog Codec Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF2RX", "Digital Codec Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF3RX", "Alt Analog Codec Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX", "Analog Codec Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF2TX", "Digital Codec Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF3TX", "Alt Analog Codec Capture", 0, SND_SOC_NOPM, 0, 0), + + /* Input Pins */ + SND_SOC_DAPM_INPUT("Codec Input Pin1"), + SND_SOC_DAPM_INPUT("Codec Input Pin2"), + SND_SOC_DAPM_INPUT("Codec Input Pin3"), + + /* Output Pins */ + SND_SOC_DAPM_OUTPUT("Codec Output Pin1"), + SND_SOC_DAPM_OUTPUT("Codec Output Pin2"), + SND_SOC_DAPM_OUTPUT("Codec Output Pin3"), +}; + +static struct snd_soc_dai_driver card_binder_dai = { + .id = -1, + .name = "codec-probing-DAI", +}; + +static int hda_hdev_attach(struct hdac_device *hdev) +{ + struct hda_codec *codec = dev_to_hda_codec(&hdev->dev); + struct snd_soc_component_driver *comp_drv; + + comp_drv = devm_kzalloc(&hdev->dev, sizeof(*comp_drv), GFP_KERNEL); + if (!comp_drv) + return -ENOMEM; + + /* + * It's save to rely on dev_name() rather than a copy as component + * driver's lifetime is directly tied to hda codec one + */ + comp_drv->name = dev_name(&hdev->dev); + comp_drv->probe = hda_codec_probe; + comp_drv->remove = hda_codec_remove; + comp_drv->idle_bias_on = false; + if (!hda_codec_is_display(codec)) { + comp_drv->dapm_widgets = hda_dapm_widgets; + comp_drv->num_dapm_widgets = ARRAY_SIZE(hda_dapm_widgets); + comp_drv->dapm_routes = hda_dapm_routes; + comp_drv->num_dapm_routes = ARRAY_SIZE(hda_dapm_routes); + } + + return snd_soc_register_component(&hdev->dev, comp_drv, &card_binder_dai, 1); +} + +static int hda_hdev_detach(struct hdac_device *hdev) +{ + struct hda_codec *codec = dev_to_hda_codec(&hdev->dev); + + if (codec->registered) + cancel_delayed_work_sync(&codec->jackpoll_work); + + snd_soc_unregister_component(&hdev->dev); + + return 0; +} + +const struct hdac_ext_bus_ops soc_hda_ext_bus_ops = { + .hdev_attach = hda_hdev_attach, + .hdev_detach = hda_hdev_detach, +}; +EXPORT_SYMBOL_GPL(soc_hda_ext_bus_ops); + +MODULE_DESCRIPTION("HD-Audio codec driver"); +MODULE_AUTHOR("Cezary Rojewski "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/hda.h b/sound/soc/codecs/hda.h new file mode 100644 index 000000000000..78a2be4945b1 --- /dev/null +++ b/sound/soc/codecs/hda.h @@ -0,0 +1,19 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Copyright(c) 2021-2022 Intel Corporation. All rights reserved. + * + * Author: Cezary Rojewski + */ + +#ifndef SND_SOC_CODECS_HDA_H +#define SND_SOC_CODECS_HDA_H + +#define hda_codec_is_display(codec) \ + ((((codec)->core.vendor_id >> 16) & 0xFFFF) == 0x8086) + +extern const struct snd_soc_dai_ops snd_soc_hda_codec_dai_ops; + +extern const struct hdac_ext_bus_ops soc_hda_ext_bus_ops; +int hda_codec_probe_complete(struct hda_codec *codec); + +#endif -- cgit v1.2.3 From 97030a43371ea29d65f332d288eb73e8f7bdb3a9 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Wed, 11 May 2022 18:23:52 +0200 Subject: ASoC: Intel: avs: Add HDAudio machine board MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Connect AVS driver with ASoC HDAudio codec with help of this machine board. Similarly to its platform and codec components, DAI links and routes are being created dynamically so single board can be used across all HDAudio codec types. Card makes use of "binder" BE DAI Link so HDAudio codec driver can be listed as one of its components. This allows for BE DAIs to be created dynamically, based on HDAudio codec capabilities. Signed-off-by: Amadeusz Sławiński Signed-off-by: Cezary Rojewski Link: https://lore.kernel.org/r/20220511162403.3987658-4-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/Kconfig | 3 + sound/soc/intel/avs/Makefile | 3 + sound/soc/intel/avs/boards/Kconfig | 15 ++ sound/soc/intel/avs/boards/Makefile | 5 + sound/soc/intel/avs/boards/hdaudio.c | 294 +++++++++++++++++++++++++++++++++++ 5 files changed, 320 insertions(+) create mode 100644 sound/soc/intel/avs/boards/Kconfig create mode 100644 sound/soc/intel/avs/boards/Makefile create mode 100644 sound/soc/intel/avs/boards/hdaudio.c (limited to 'sound') diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 7c85d1bb9c12..e5107a3ce16a 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -226,5 +226,8 @@ config SND_SOC_INTEL_AVS capabilities. This includes Skylake, Kabylake, Amberlake and Apollolake. +# Machine board drivers +source "sound/soc/intel/avs/boards/Kconfig" + # ASoC codec drivers source "sound/soc/intel/boards/Kconfig" diff --git a/sound/soc/intel/avs/Makefile b/sound/soc/intel/avs/Makefile index b6b93ae80304..919212825f21 100644 --- a/sound/soc/intel/avs/Makefile +++ b/sound/soc/intel/avs/Makefile @@ -10,3 +10,6 @@ snd-soc-avs-objs += trace.o CFLAGS_trace.o := -I$(src) obj-$(CONFIG_SND_SOC_INTEL_AVS) += snd-soc-avs.o + +# Machine support +obj-$(CONFIG_SND_SOC) += boards/ diff --git a/sound/soc/intel/avs/boards/Kconfig b/sound/soc/intel/avs/boards/Kconfig new file mode 100644 index 000000000000..de62c0437f6e --- /dev/null +++ b/sound/soc/intel/avs/boards/Kconfig @@ -0,0 +1,15 @@ +# SPDX-License-Identifier: GPL-2.0-only +menu "Intel AVS Machine drivers" + depends on SND_SOC_INTEL_AVS + +comment "Available DSP configurations" + +config SND_SOC_INTEL_AVS_MACH_HDAUDIO + tristate "HD-Audio generic board" + select SND_SOC_HDA + help + This adds support for AVS with HDAudio codec configuration. + Say Y or m if you have such a device. This is a recommended option. + If unsure select "N". + +endmenu diff --git a/sound/soc/intel/avs/boards/Makefile b/sound/soc/intel/avs/boards/Makefile new file mode 100644 index 000000000000..e5281148e5d4 --- /dev/null +++ b/sound/soc/intel/avs/boards/Makefile @@ -0,0 +1,5 @@ +# SPDX-License-Identifier: GPL-2.0-only + +snd-soc-avs-hdaudio-objs := hdaudio.o + +obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_HDAUDIO) += snd-soc-avs-hdaudio.o diff --git a/sound/soc/intel/avs/boards/hdaudio.c b/sound/soc/intel/avs/boards/hdaudio.c new file mode 100644 index 000000000000..d2fc41d39448 --- /dev/null +++ b/sound/soc/intel/avs/boards/hdaudio.c @@ -0,0 +1,294 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// +// Authors: Cezary Rojewski +// Amadeusz Slawinski +// + +#include +#include +#include +#include +#include +#include "../../../codecs/hda.h" + +static int avs_create_dai_links(struct device *dev, struct hda_codec *codec, int pcm_count, + const char *platform_name, struct snd_soc_dai_link **links) +{ + struct snd_soc_dai_link_component *platform; + struct snd_soc_dai_link *dl; + struct hda_pcm *pcm; + const char *cname = dev_name(&codec->core.dev); + int i; + + dl = devm_kcalloc(dev, pcm_count, sizeof(*dl), GFP_KERNEL); + platform = devm_kzalloc(dev, sizeof(*platform), GFP_KERNEL); + if (!dl || !platform) + return -ENOMEM; + + platform->name = platform_name; + pcm = list_first_entry(&codec->pcm_list_head, struct hda_pcm, list); + + for (i = 0; i < pcm_count; i++, pcm = list_next_entry(pcm, list)) { + dl[i].name = devm_kasprintf(dev, GFP_KERNEL, "%s link%d", cname, i); + if (!dl[i].name) + return -ENOMEM; + + dl[i].id = i; + dl[i].nonatomic = 1; + dl[i].no_pcm = 1; + dl[i].dpcm_playback = 1; + dl[i].dpcm_capture = 1; + dl[i].platforms = platform; + dl[i].num_platforms = 1; + + dl[i].codecs = devm_kzalloc(dev, sizeof(*dl->codecs), GFP_KERNEL); + dl[i].cpus = devm_kzalloc(dev, sizeof(*dl->cpus), GFP_KERNEL); + if (!dl[i].codecs || !dl[i].cpus) + return -ENOMEM; + + dl[i].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, "%s-cpu%d", cname, i); + if (!dl[i].cpus->dai_name) + return -ENOMEM; + + dl[i].codecs->name = devm_kstrdup(dev, cname, GFP_KERNEL); + dl[i].codecs->dai_name = pcm->name; + dl[i].num_codecs = 1; + dl[i].num_cpus = 1; + } + + *links = dl; + return 0; +} + +static int avs_create_dapm_routes(struct device *dev, struct hda_codec *codec, int pcm_count, + struct snd_soc_dapm_route **routes, int *num_routes) +{ + struct snd_soc_dapm_route *dr; + struct hda_pcm *pcm; + const char *cname = dev_name(&codec->core.dev); + int i, n = 0; + + /* at max twice the number of pcms */ + dr = devm_kcalloc(dev, pcm_count * 2, sizeof(*dr), GFP_KERNEL); + if (!dr) + return -ENOMEM; + + pcm = list_first_entry(&codec->pcm_list_head, struct hda_pcm, list); + + for (i = 0; i < pcm_count; i++, pcm = list_next_entry(pcm, list)) { + struct hda_pcm_stream *stream; + int dir; + + dir = SNDRV_PCM_STREAM_PLAYBACK; + stream = &pcm->stream[dir]; + if (!stream->substreams) + goto capture_routes; + + dr[n].sink = devm_kasprintf(dev, GFP_KERNEL, "%s %s", pcm->name, + snd_pcm_direction_name(dir)); + dr[n].source = devm_kasprintf(dev, GFP_KERNEL, "%s-cpu%d Tx", cname, i); + if (!dr[n].sink || !dr[n].source) + return -ENOMEM; + n++; + +capture_routes: + dir = SNDRV_PCM_STREAM_CAPTURE; + stream = &pcm->stream[dir]; + if (!stream->substreams) + continue; + + dr[n].sink = devm_kasprintf(dev, GFP_KERNEL, "%s-cpu%d Rx", cname, i); + dr[n].source = devm_kasprintf(dev, GFP_KERNEL, "%s %s", pcm->name, + snd_pcm_direction_name(dir)); + if (!dr[n].sink || !dr[n].source) + return -ENOMEM; + n++; + } + + *routes = dr; + *num_routes = n; + return 0; +} + +/* Should be aligned with SectionPCM's name from topology */ +#define FEDAI_NAME_PREFIX "HDMI" + +static struct snd_pcm * +avs_card_hdmi_pcm_at(struct snd_soc_card *card, int hdmi_idx) +{ + struct snd_soc_pcm_runtime *rtd; + int dir = SNDRV_PCM_STREAM_PLAYBACK; + + for_each_card_rtds(card, rtd) { + struct snd_pcm *spcm; + int ret, n; + + spcm = rtd->pcm ? rtd->pcm->streams[dir].pcm : NULL; + if (!spcm || !strstr(spcm->id, FEDAI_NAME_PREFIX)) + continue; + + ret = sscanf(spcm->id, FEDAI_NAME_PREFIX "%d", &n); + if (ret != 1) + continue; + if (n == hdmi_idx) + return rtd->pcm; + } + + return NULL; +} + +static int avs_card_late_probe(struct snd_soc_card *card) +{ + struct snd_soc_acpi_mach *mach = dev_get_platdata(card->dev); + struct hda_codec *codec = mach->pdata; + struct hda_pcm *hpcm; + /* Topology pcm indexing is 1-based */ + int i = 1; + + list_for_each_entry(hpcm, &codec->pcm_list_head, list) { + struct snd_pcm *spcm; + + spcm = avs_card_hdmi_pcm_at(card, i); + if (spcm) { + hpcm->pcm = spcm; + hpcm->device = spcm->device; + dev_info(card->dev, "%s: mapping HDMI converter %d to PCM %d (%p)\n", + __func__, i, hpcm->device, spcm); + } else { + hpcm->pcm = NULL; + hpcm->device = SNDRV_PCM_INVALID_DEVICE; + dev_warn(card->dev, "%s: no PCM in topology for HDMI converter %d\n", + __func__, i); + } + i++; + } + + return hda_codec_probe_complete(codec); +} + +static int avs_probing_link_init(struct snd_soc_pcm_runtime *rtm) +{ + struct snd_soc_dapm_route *routes; + struct snd_soc_acpi_mach *mach; + struct snd_soc_dai_link *links = NULL; + struct snd_soc_card *card = rtm->card; + struct hda_codec *codec; + struct hda_pcm *pcm; + int ret, n, pcm_count = 0; + + mach = dev_get_platdata(card->dev); + codec = mach->pdata; + + if (list_empty(&codec->pcm_list_head)) + return -EINVAL; + list_for_each_entry(pcm, &codec->pcm_list_head, list) + pcm_count++; + + ret = avs_create_dai_links(card->dev, codec, pcm_count, mach->mach_params.platform, &links); + if (ret < 0) { + dev_err(card->dev, "create links failed: %d\n", ret); + return ret; + } + + for (n = 0; n < pcm_count; n++) { + ret = snd_soc_add_pcm_runtime(card, &links[n]); + if (ret < 0) { + dev_err(card->dev, "add links failed: %d\n", ret); + return ret; + } + } + + ret = avs_create_dapm_routes(card->dev, codec, pcm_count, &routes, &n); + if (ret < 0) { + dev_err(card->dev, "create routes failed: %d\n", ret); + return ret; + } + + ret = snd_soc_dapm_add_routes(&card->dapm, routes, n); + if (ret < 0) { + dev_err(card->dev, "add routes failed: %d\n", ret); + return ret; + } + + return 0; +} + +SND_SOC_DAILINK_DEF(dummy, DAILINK_COMP_ARRAY(COMP_DUMMY())); + +static struct snd_soc_dai_link probing_link = { + .name = "probing-LINK", + .id = -1, + .nonatomic = 1, + .no_pcm = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + .cpus = dummy, + .num_cpus = ARRAY_SIZE(dummy), + .init = avs_probing_link_init, +}; + +static int avs_hdaudio_probe(struct platform_device *pdev) +{ + struct snd_soc_dai_link *binder; + struct snd_soc_acpi_mach *mach; + struct snd_soc_card *card; + struct device *dev = &pdev->dev; + struct hda_codec *codec; + + mach = dev_get_platdata(dev); + codec = mach->pdata; + + /* codec may be unloaded before card's probe() fires */ + if (!device_is_registered(&codec->core.dev)) + return -ENODEV; + + binder = devm_kmemdup(dev, &probing_link, sizeof(probing_link), GFP_KERNEL); + if (!binder) + return -ENOMEM; + + binder->platforms = devm_kzalloc(dev, sizeof(*binder->platforms), GFP_KERNEL); + binder->codecs = devm_kzalloc(dev, sizeof(*binder->codecs), GFP_KERNEL); + if (!binder->platforms || !binder->codecs) + return -ENOMEM; + + binder->codecs->name = devm_kstrdup(dev, dev_name(&codec->core.dev), GFP_KERNEL); + if (!binder->codecs->name) + return -ENOMEM; + + binder->platforms->name = mach->mach_params.platform; + binder->num_platforms = 1; + binder->codecs->dai_name = "codec-probing-DAI"; + binder->num_codecs = 1; + + card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); + if (!card) + return -ENOMEM; + + card->name = binder->codecs->name; + card->dev = dev; + card->owner = THIS_MODULE; + card->dai_link = binder; + card->num_links = 1; + card->fully_routed = true; + if (hda_codec_is_display(codec)) + card->late_probe = avs_card_late_probe; + + return devm_snd_soc_register_card(dev, card); +} + +static struct platform_driver avs_hdaudio_driver = { + .probe = avs_hdaudio_probe, + .driver = { + .name = "avs_hdaudio", + .pm = &snd_soc_pm_ops, + }, +}; + +module_platform_driver(avs_hdaudio_driver) + +MODULE_DESCRIPTION("Intel HD-Audio machine driver"); +MODULE_AUTHOR("Cezary Rojewski "); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:avs_hdaudio"); -- cgit v1.2.3 From 6575e5cae7525b07d0b5fbd7d42323363919a867 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Wed, 11 May 2022 18:23:53 +0200 Subject: ASoC: Intel: avs: Add DMIC machine board MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Support AVS-DMIC configuration by implementing board connecting AVS platform component with DMIC codec one. Signed-off-by: Amadeusz Sławiński Signed-off-by: Cezary Rojewski Link: https://lore.kernel.org/r/20220511162403.3987658-5-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/boards/Kconfig | 8 ++++ sound/soc/intel/avs/boards/Makefile | 2 + sound/soc/intel/avs/boards/dmic.c | 93 +++++++++++++++++++++++++++++++++++++ 3 files changed, 103 insertions(+) create mode 100644 sound/soc/intel/avs/boards/dmic.c (limited to 'sound') diff --git a/sound/soc/intel/avs/boards/Kconfig b/sound/soc/intel/avs/boards/Kconfig index de62c0437f6e..1d4597fa9814 100644 --- a/sound/soc/intel/avs/boards/Kconfig +++ b/sound/soc/intel/avs/boards/Kconfig @@ -4,6 +4,14 @@ menu "Intel AVS Machine drivers" comment "Available DSP configurations" +config SND_SOC_INTEL_AVS_MACH_DMIC + tristate "DMIC generic board" + select SND_SOC_DMIC + help + This adds support for AVS with Digital Mic array configuration. + Say Y or m if you have such a device. This is a recommended option. + If unsure select "N". + config SND_SOC_INTEL_AVS_MACH_HDAUDIO tristate "HD-Audio generic board" select SND_SOC_HDA diff --git a/sound/soc/intel/avs/boards/Makefile b/sound/soc/intel/avs/boards/Makefile index e5281148e5d4..2ff35d4d97d8 100644 --- a/sound/soc/intel/avs/boards/Makefile +++ b/sound/soc/intel/avs/boards/Makefile @@ -1,5 +1,7 @@ # SPDX-License-Identifier: GPL-2.0-only +snd-soc-avs-dmic-objs := dmic.o snd-soc-avs-hdaudio-objs := hdaudio.o +obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_DMIC) += snd-soc-avs-dmic.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_HDAUDIO) += snd-soc-avs-hdaudio.o diff --git a/sound/soc/intel/avs/boards/dmic.c b/sound/soc/intel/avs/boards/dmic.c new file mode 100644 index 000000000000..90a921638572 --- /dev/null +++ b/sound/soc/intel/avs/boards/dmic.c @@ -0,0 +1,93 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// +// Authors: Cezary Rojewski +// Amadeusz Slawinski +// + +#include +#include +#include +#include + +SND_SOC_DAILINK_DEF(dmic_pin, DAILINK_COMP_ARRAY(COMP_CPU("DMIC Pin"))); +SND_SOC_DAILINK_DEF(dmic_wov_pin, DAILINK_COMP_ARRAY(COMP_CPU("DMIC WoV Pin"))); +SND_SOC_DAILINK_DEF(dmic_codec, DAILINK_COMP_ARRAY(COMP_CODEC("dmic-codec", "dmic-hifi"))); +/* Name overridden on probe */ +SND_SOC_DAILINK_DEF(platform, DAILINK_COMP_ARRAY(COMP_PLATFORM(""))); + +static struct snd_soc_dai_link card_dai_links[] = { + /* Back ends */ + { + .name = "DMIC", + .id = 0, + .dpcm_capture = 1, + .nonatomic = 1, + .no_pcm = 1, + SND_SOC_DAILINK_REG(dmic_pin, dmic_codec, platform), + }, + { + .name = "DMIC WoV", + .id = 1, + .dpcm_capture = 1, + .nonatomic = 1, + .no_pcm = 1, + .ignore_suspend = 1, + SND_SOC_DAILINK_REG(dmic_wov_pin, dmic_codec, platform), + }, +}; + +static const struct snd_soc_dapm_widget card_widgets[] = { + SND_SOC_DAPM_MIC("SoC DMIC", NULL), +}; + +static const struct snd_soc_dapm_route card_routes[] = { + {"DMic", NULL, "SoC DMIC"}, + {"DMIC Rx", NULL, "Capture"}, + {"DMIC WoV Rx", NULL, "Capture"}, +}; + +static int avs_dmic_probe(struct platform_device *pdev) +{ + struct snd_soc_acpi_mach *mach; + struct snd_soc_card *card; + struct device *dev = &pdev->dev; + int ret; + + mach = dev_get_platdata(dev); + + card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); + if (!card) + return -ENOMEM; + + card->name = "avs_dmic"; + card->dev = dev; + card->owner = THIS_MODULE; + card->dai_link = card_dai_links; + card->num_links = ARRAY_SIZE(card_dai_links); + card->dapm_widgets = card_widgets; + card->num_dapm_widgets = ARRAY_SIZE(card_widgets); + card->dapm_routes = card_routes; + card->num_dapm_routes = ARRAY_SIZE(card_routes); + card->fully_routed = true; + + ret = snd_soc_fixup_dai_links_platform_name(card, mach->mach_params.platform); + if (ret) + return ret; + + return devm_snd_soc_register_card(dev, card); +} + +static struct platform_driver avs_dmic_driver = { + .probe = avs_dmic_probe, + .driver = { + .name = "avs_dmic", + .pm = &snd_soc_pm_ops, + }, +}; + +module_platform_driver(avs_dmic_driver); + +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:avs_dmic"); -- cgit v1.2.3 From e39acc4cfd9250e7b8ec01897570f3009659c3d6 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Wed, 11 May 2022 18:23:54 +0200 Subject: ASoC: Intel: avs: Add I2S-test machine board MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Allow for testing audio streaming over I2S interface through SSP loopback. No actual codec is needed here as board is intended for SSP loopback scenarios only. One playback and one capture endpoint is exposed per SSP port. Signed-off-by: Amadeusz Sławiński Signed-off-by: Cezary Rojewski Link: https://lore.kernel.org/r/20220511162403.3987658-6-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/boards/Kconfig | 6 ++ sound/soc/intel/avs/boards/Makefile | 2 + sound/soc/intel/avs/boards/i2s_test.c | 180 ++++++++++++++++++++++++++++++++++ 3 files changed, 188 insertions(+) create mode 100644 sound/soc/intel/avs/boards/i2s_test.c (limited to 'sound') diff --git a/sound/soc/intel/avs/boards/Kconfig b/sound/soc/intel/avs/boards/Kconfig index 1d4597fa9814..5b89fcb5f07f 100644 --- a/sound/soc/intel/avs/boards/Kconfig +++ b/sound/soc/intel/avs/boards/Kconfig @@ -20,4 +20,10 @@ config SND_SOC_INTEL_AVS_MACH_HDAUDIO Say Y or m if you have such a device. This is a recommended option. If unsure select "N". +config SND_SOC_INTEL_AVS_MACH_I2S_TEST + tristate "I2S test board" + help + This adds support for I2S test-board which can be used to verify + transfer over I2S interface with SSP loopback scenarios. + endmenu diff --git a/sound/soc/intel/avs/boards/Makefile b/sound/soc/intel/avs/boards/Makefile index 2ff35d4d97d8..fa1a279106be 100644 --- a/sound/soc/intel/avs/boards/Makefile +++ b/sound/soc/intel/avs/boards/Makefile @@ -2,6 +2,8 @@ snd-soc-avs-dmic-objs := dmic.o snd-soc-avs-hdaudio-objs := hdaudio.o +snd-soc-avs-i2s-test-objs := i2s_test.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_DMIC) += snd-soc-avs-dmic.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_HDAUDIO) += snd-soc-avs-hdaudio.o +obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_i2s_TEST) += snd-soc-avs-i2s-test.o diff --git a/sound/soc/intel/avs/boards/i2s_test.c b/sound/soc/intel/avs/boards/i2s_test.c new file mode 100644 index 000000000000..461b651cd331 --- /dev/null +++ b/sound/soc/intel/avs/boards/i2s_test.c @@ -0,0 +1,180 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// +// Authors: Cezary Rojewski +// Amadeusz Slawinski +// + +#include +#include +#include +#include +#include +#include + +static int avs_create_dai_link(struct device *dev, const char *platform_name, int ssp_port, + struct snd_soc_dai_link **dai_link) +{ + struct snd_soc_dai_link_component *platform; + struct snd_soc_dai_link *dl; + + dl = devm_kzalloc(dev, sizeof(*dl), GFP_KERNEL); + platform = devm_kzalloc(dev, sizeof(*platform), GFP_KERNEL); + if (!dl || !platform) + return -ENOMEM; + + platform->name = platform_name; + + dl->name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-Codec", ssp_port); + dl->cpus = devm_kzalloc(dev, sizeof(*dl->cpus), GFP_KERNEL); + dl->codecs = devm_kzalloc(dev, sizeof(*dl->codecs), GFP_KERNEL); + if (!dl->name || !dl->cpus || !dl->codecs) + return -ENOMEM; + + dl->cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", ssp_port); + dl->codecs->name = devm_kasprintf(dev, GFP_KERNEL, "snd-soc-dummy"); + dl->codecs->dai_name = devm_kasprintf(dev, GFP_KERNEL, "snd-soc-dummy-dai"); + if (!dl->cpus->dai_name || !dl->codecs->name || !dl->codecs->dai_name) + return -ENOMEM; + + dl->num_cpus = 1; + dl->num_codecs = 1; + dl->platforms = platform; + dl->num_platforms = 1; + dl->id = 0; + dl->nonatomic = 1; + dl->no_pcm = 1; + dl->dpcm_capture = 1; + dl->dpcm_playback = 1; + + *dai_link = dl; + + return 0; +} + +static int avs_create_dapm_routes(struct device *dev, int ssp_port, + struct snd_soc_dapm_route **routes, int *num_routes) +{ + struct snd_soc_dapm_route *dr; + const int num_dr = 2; + + dr = devm_kcalloc(dev, num_dr, sizeof(*dr), GFP_KERNEL); + if (!dr) + return -ENOMEM; + + dr[0].sink = devm_kasprintf(dev, GFP_KERNEL, "ssp%dpb", ssp_port); + dr[0].source = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Tx", ssp_port); + if (!dr[0].sink || !dr[0].source) + return -ENOMEM; + + dr[1].sink = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Rx", ssp_port); + dr[1].source = devm_kasprintf(dev, GFP_KERNEL, "ssp%dcp", ssp_port); + if (!dr[1].sink || !dr[1].source) + return -ENOMEM; + + *routes = dr; + *num_routes = num_dr; + + return 0; +} + +static int avs_create_dapm_widgets(struct device *dev, int ssp_port, + struct snd_soc_dapm_widget **widgets, int *num_widgets) +{ + struct snd_soc_dapm_widget *dw; + const int num_dw = 2; + + dw = devm_kcalloc(dev, num_dw, sizeof(*dw), GFP_KERNEL); + if (!dw) + return -ENOMEM; + + dw[0].id = snd_soc_dapm_hp; + dw[0].reg = SND_SOC_NOPM; + dw[0].name = devm_kasprintf(dev, GFP_KERNEL, "ssp%dpb", ssp_port); + if (!dw[0].name) + return -ENOMEM; + + dw[1].id = snd_soc_dapm_mic; + dw[1].reg = SND_SOC_NOPM; + dw[1].name = devm_kasprintf(dev, GFP_KERNEL, "ssp%dcp", ssp_port); + if (!dw[1].name) + return -ENOMEM; + + *widgets = dw; + *num_widgets = num_dw; + + return 0; +} + +static int avs_i2s_test_probe(struct platform_device *pdev) +{ + struct snd_soc_dapm_widget *widgets; + struct snd_soc_dapm_route *routes; + struct snd_soc_dai_link *dai_link; + struct snd_soc_acpi_mach *mach; + struct snd_soc_card *card; + struct device *dev = &pdev->dev; + const char *pname; + int num_routes, num_widgets; + int ssp_port, ret; + + mach = dev_get_platdata(dev); + pname = mach->mach_params.platform; + ssp_port = __ffs(mach->mach_params.i2s_link_mask); + + card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); + if (!card) + return -ENOMEM; + + card->name = devm_kasprintf(dev, GFP_KERNEL, "ssp%ld-loopback", ssp_port); + if (!card->name) + return -ENOMEM; + + ret = avs_create_dai_link(dev, pname, ssp_port, &dai_link); + if (ret) { + dev_err(dev, "Failed to create dai link: %d\n", ret); + return ret; + } + + ret = avs_create_dapm_routes(dev, ssp_port, &routes, &num_routes); + if (ret) { + dev_err(dev, "Failed to create dapm routes: %d\n", ret); + return ret; + } + + ret = avs_create_dapm_widgets(dev, ssp_port, &widgets, &num_widgets); + if (ret) { + dev_err(dev, "Failed to create dapm widgets: %d\n", ret); + return ret; + } + + card->dev = dev; + card->owner = THIS_MODULE; + card->dai_link = dai_link; + card->num_links = 1; + card->dapm_routes = routes; + card->num_dapm_routes = num_routes; + card->dapm_widgets = widgets; + card->num_dapm_widgets = num_widgets; + card->fully_routed = true; + + ret = snd_soc_fixup_dai_links_platform_name(card, pname); + if (ret) + return ret; + + return devm_snd_soc_register_card(dev, card); +} + +static struct platform_driver avs_i2s_test_driver = { + .probe = avs_i2s_test_probe, + .driver = { + .name = "avs_i2s_test", + .pm = &snd_soc_pm_ops, + }, +}; + +module_platform_driver(avs_i2s_test_driver); + +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:avs_i2s_test"); -- cgit v1.2.3 From e2a4cbf277c4561d01f1aafa3cfafe46bf3feec7 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Wed, 11 May 2022 18:23:55 +0200 Subject: ASoC: Intel: avs: Add rt274 machine board MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit To support AVS-rt274 configuration add machine board connecting AVS platform component driver with rt274 codec one. Signed-off-by: Amadeusz Sławiński Signed-off-by: Cezary Rojewski Link: https://lore.kernel.org/r/20220511162403.3987658-7-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/boards/Kconfig | 10 ++ sound/soc/intel/avs/boards/Makefile | 2 + sound/soc/intel/avs/boards/rt274.c | 310 ++++++++++++++++++++++++++++++++++++ 3 files changed, 322 insertions(+) create mode 100644 sound/soc/intel/avs/boards/rt274.c (limited to 'sound') diff --git a/sound/soc/intel/avs/boards/Kconfig b/sound/soc/intel/avs/boards/Kconfig index 5b89fcb5f07f..9058919c99a7 100644 --- a/sound/soc/intel/avs/boards/Kconfig +++ b/sound/soc/intel/avs/boards/Kconfig @@ -26,4 +26,14 @@ config SND_SOC_INTEL_AVS_MACH_I2S_TEST This adds support for I2S test-board which can be used to verify transfer over I2S interface with SSP loopback scenarios. +config SND_SOC_INTEL_AVS_MACH_RT274 + tristate "rt274 in I2S mode" + depends on I2C + depends on MFD_INTEL_LPSS || COMPILE_TEST + select SND_SOC_RT274 + help + This adds support for ASoC machine driver with RT274 I2S audio codec. + Say Y or m if you have such a device. This is a recommended option. + If unsure select "N". + endmenu diff --git a/sound/soc/intel/avs/boards/Makefile b/sound/soc/intel/avs/boards/Makefile index fa1a279106be..e94f04d00ffc 100644 --- a/sound/soc/intel/avs/boards/Makefile +++ b/sound/soc/intel/avs/boards/Makefile @@ -3,7 +3,9 @@ snd-soc-avs-dmic-objs := dmic.o snd-soc-avs-hdaudio-objs := hdaudio.o snd-soc-avs-i2s-test-objs := i2s_test.o +snd-soc-avs-rt274-objs := rt274.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_DMIC) += snd-soc-avs-dmic.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_HDAUDIO) += snd-soc-avs-hdaudio.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_i2s_TEST) += snd-soc-avs-i2s-test.o +obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_RT274) += snd-soc-avs-rt274.o diff --git a/sound/soc/intel/avs/boards/rt274.c b/sound/soc/intel/avs/boards/rt274.c new file mode 100644 index 000000000000..afef5a3ca60b --- /dev/null +++ b/sound/soc/intel/avs/boards/rt274.c @@ -0,0 +1,310 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// +// Authors: Cezary Rojewski +// Amadeusz Slawinski +// + +#include +#include +#include +#include +#include +#include +#include "../../../codecs/rt274.h" + +#define AVS_RT274_FREQ_OUT 24000000 +#define AVS_RT274_BE_FIXUP_RATE 48000 +#define RT274_CODEC_DAI "rt274-aif1" + +static const struct snd_kcontrol_new card_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Mic Jack"), +}; + +static int +avs_rt274_clock_control(struct snd_soc_dapm_widget *w, struct snd_kcontrol *control, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct snd_soc_dai *codec_dai; + int ret; + + codec_dai = snd_soc_card_get_codec_dai(card, RT274_CODEC_DAI); + if (!codec_dai) + return -EINVAL; + + /* Codec needs clock for Jack detection and button press */ + ret = snd_soc_dai_set_sysclk(codec_dai, RT274_SCLK_S_PLL2, AVS_RT274_FREQ_OUT, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "set codec sysclk failed: %d\n", ret); + return ret; + } + + if (SND_SOC_DAPM_EVENT_ON(event)) { + int ratio = 100; + + snd_soc_dai_set_bclk_ratio(codec_dai, ratio); + + ret = snd_soc_dai_set_pll(codec_dai, 0, RT274_PLL2_S_BCLK, + AVS_RT274_BE_FIXUP_RATE * ratio, AVS_RT274_FREQ_OUT); + if (ret) { + dev_err(codec_dai->dev, "failed to enable PLL2: %d\n", ret); + return ret; + } + } + + return 0; +} + +static const struct snd_soc_dapm_widget card_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, avs_rt274_clock_control, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route card_base_routes[] = { + {"Headphone Jack", NULL, "HPO Pin"}, + {"MIC", NULL, "Mic Jack"}, + + {"Headphone Jack", NULL, "Platform Clock"}, + {"MIC", NULL, "Platform Clock"}, +}; + +static struct snd_soc_jack_pin card_headset_pins[] = { + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Mic Jack", + .mask = SND_JACK_MICROPHONE, + }, +}; + +static int avs_rt274_codec_init(struct snd_soc_pcm_runtime *runtime) +{ + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(runtime, 0); + struct snd_soc_component *component = codec_dai->component; + struct snd_soc_jack_pin *pins; + struct snd_soc_jack *jack; + struct snd_soc_card *card = runtime->card; + int num_pins, ret; + + jack = snd_soc_card_get_drvdata(card); + num_pins = ARRAY_SIZE(card_headset_pins); + + pins = devm_kmemdup(card->dev, card_headset_pins, sizeof(*pins) * num_pins, GFP_KERNEL); + if (!pins) + return -ENOMEM; + + ret = snd_soc_card_jack_new_pins(card, "Headset", SND_JACK_HEADSET, jack, pins, num_pins); + if (ret) + return ret; + + snd_soc_component_set_jack(component, jack, NULL); + + /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */ + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0xF, 4, 24); + if (ret < 0) { + dev_err(card->dev, "can't set codec pcm format %d\n", ret); + return ret; + } + + card->dapm.idle_bias_off = true; + + return 0; +} + +static int avs_rt274_be_fixup(struct snd_soc_pcm_runtime *runtime, struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate, *channels; + struct snd_mask *fmt; + + rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + + /* The ADSP will convert the FE rate to 48k, stereo */ + rate->min = rate->max = AVS_RT274_BE_FIXUP_RATE; + channels->min = channels->max = 2; + + /* set SSPN to 24 bit */ + snd_mask_none(fmt); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); + + return 0; +} + +static int avs_create_dai_link(struct device *dev, const char *platform_name, int ssp_port, + struct snd_soc_dai_link **dai_link) +{ + struct snd_soc_dai_link_component *platform; + struct snd_soc_dai_link *dl; + + dl = devm_kzalloc(dev, sizeof(*dl), GFP_KERNEL); + platform = devm_kzalloc(dev, sizeof(*platform), GFP_KERNEL); + if (!dl || !platform) + return -ENOMEM; + + platform->name = platform_name; + + dl->name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-Codec", ssp_port); + dl->cpus = devm_kzalloc(dev, sizeof(*dl->cpus), GFP_KERNEL); + dl->codecs = devm_kzalloc(dev, sizeof(*dl->codecs), GFP_KERNEL); + if (!dl->name || !dl->cpus || !dl->codecs) + return -ENOMEM; + + dl->cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", ssp_port); + dl->codecs->name = devm_kasprintf(dev, GFP_KERNEL, "i2c-INT34C2:00"); + dl->codecs->dai_name = devm_kasprintf(dev, GFP_KERNEL, "rt274-aif1"); + if (!dl->cpus->dai_name || !dl->codecs->name || !dl->codecs->dai_name) + return -ENOMEM; + + dl->num_cpus = 1; + dl->num_codecs = 1; + dl->platforms = platform; + dl->num_platforms = 1; + dl->id = 0; + dl->dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS; + dl->init = avs_rt274_codec_init; + dl->be_hw_params_fixup = avs_rt274_be_fixup; + dl->nonatomic = 1; + dl->no_pcm = 1; + dl->dpcm_capture = 1; + dl->dpcm_playback = 1; + + *dai_link = dl; + + return 0; +} + +static int avs_create_dapm_routes(struct device *dev, int ssp_port, + struct snd_soc_dapm_route **routes, int *num_routes) +{ + struct snd_soc_dapm_route *dr; + const int num_base = ARRAY_SIZE(card_base_routes); + const int num_dr = num_base + 2; + int idx; + + dr = devm_kcalloc(dev, num_dr, sizeof(*dr), GFP_KERNEL); + if (!dr) + return -ENOMEM; + + memcpy(dr, card_base_routes, num_base * sizeof(*dr)); + + idx = num_base; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "AIF1 Playback"); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Tx", ssp_port); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + idx++; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Rx", ssp_port); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "AIF1 Capture"); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + *routes = dr; + *num_routes = num_dr; + + return 0; +} + +static int avs_card_set_jack(struct snd_soc_card *card, struct snd_soc_jack *jack) +{ + struct snd_soc_component *component; + + for_each_card_components(card, component) + snd_soc_component_set_jack(component, jack, NULL); + return 0; +} + +static int avs_card_remove(struct snd_soc_card *card) +{ + return avs_card_set_jack(card, NULL); +} + +static int avs_card_suspend_pre(struct snd_soc_card *card) +{ + return avs_card_set_jack(card, NULL); +} + +static int avs_card_resume_post(struct snd_soc_card *card) +{ + struct snd_soc_jack *jack = snd_soc_card_get_drvdata(card); + + return avs_card_set_jack(card, jack); +} + +static int avs_rt274_probe(struct platform_device *pdev) +{ + struct snd_soc_dapm_route *routes; + struct snd_soc_dai_link *dai_link; + struct snd_soc_acpi_mach *mach; + struct snd_soc_card *card; + struct snd_soc_jack *jack; + struct device *dev = &pdev->dev; + const char *pname; + int num_routes, ssp_port, ret; + + mach = dev_get_platdata(dev); + pname = mach->mach_params.platform; + ssp_port = __ffs(mach->mach_params.i2s_link_mask); + + ret = avs_create_dai_link(dev, pname, ssp_port, &dai_link); + if (ret) { + dev_err(dev, "Failed to create dai link: %d", ret); + return ret; + } + + ret = avs_create_dapm_routes(dev, ssp_port, &routes, &num_routes); + if (ret) { + dev_err(dev, "Failed to create dapm routes: %d", ret); + return ret; + } + + jack = devm_kzalloc(dev, sizeof(*jack), GFP_KERNEL); + card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); + if (!jack || !card) + return -ENOMEM; + + card->name = "avs_rt274"; + card->dev = dev; + card->owner = THIS_MODULE; + card->remove = avs_card_remove; + card->suspend_pre = avs_card_suspend_pre; + card->resume_post = avs_card_resume_post; + card->dai_link = dai_link; + card->num_links = 1; + card->controls = card_controls; + card->num_controls = ARRAY_SIZE(card_controls); + card->dapm_widgets = card_widgets; + card->num_dapm_widgets = ARRAY_SIZE(card_widgets); + card->dapm_routes = routes; + card->num_dapm_routes = num_routes; + card->fully_routed = true; + snd_soc_card_set_drvdata(card, jack); + + ret = snd_soc_fixup_dai_links_platform_name(card, pname); + if (ret) + return ret; + + return devm_snd_soc_register_card(dev, card); +} + +static struct platform_driver avs_rt274_driver = { + .probe = avs_rt274_probe, + .driver = { + .name = "avs_rt274", + .pm = &snd_soc_pm_ops, + }, +}; + +module_platform_driver(avs_rt274_driver); + +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:avs_rt274"); -- cgit v1.2.3 From 1d395ee2e19b33a1008acfc7af186f2851b63d01 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Wed, 11 May 2022 18:23:56 +0200 Subject: ASoC: Intel: avs: Add rt286 machine board MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit To support AVS-rt286 configuration add machine board connecting AVS platform component driver with rt286 codec one. Signed-off-by: Amadeusz Sławiński Signed-off-by: Cezary Rojewski Link: https://lore.kernel.org/r/20220511162403.3987658-8-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/boards/Kconfig | 10 ++ sound/soc/intel/avs/boards/Makefile | 2 + sound/soc/intel/avs/boards/rt286.c | 281 ++++++++++++++++++++++++++++++++++++ 3 files changed, 293 insertions(+) create mode 100644 sound/soc/intel/avs/boards/rt286.c (limited to 'sound') diff --git a/sound/soc/intel/avs/boards/Kconfig b/sound/soc/intel/avs/boards/Kconfig index 9058919c99a7..707e9e96746d 100644 --- a/sound/soc/intel/avs/boards/Kconfig +++ b/sound/soc/intel/avs/boards/Kconfig @@ -36,4 +36,14 @@ config SND_SOC_INTEL_AVS_MACH_RT274 Say Y or m if you have such a device. This is a recommended option. If unsure select "N". +config SND_SOC_INTEL_AVS_MACH_RT286 + tristate "rt286 in I2S mode" + depends on I2C + depends on MFD_INTEL_LPSS || COMPILE_TEST + select SND_SOC_RT286 + help + This adds support for ASoC machine driver with RT286 I2S audio codec. + Say Y or m if you have such a device. This is a recommended option. + If unsure select "N". + endmenu diff --git a/sound/soc/intel/avs/boards/Makefile b/sound/soc/intel/avs/boards/Makefile index e94f04d00ffc..7ea4ad38c7df 100644 --- a/sound/soc/intel/avs/boards/Makefile +++ b/sound/soc/intel/avs/boards/Makefile @@ -4,8 +4,10 @@ snd-soc-avs-dmic-objs := dmic.o snd-soc-avs-hdaudio-objs := hdaudio.o snd-soc-avs-i2s-test-objs := i2s_test.o snd-soc-avs-rt274-objs := rt274.o +snd-soc-avs-rt286-objs := rt286.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_DMIC) += snd-soc-avs-dmic.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_HDAUDIO) += snd-soc-avs-hdaudio.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_i2s_TEST) += snd-soc-avs-i2s-test.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_RT274) += snd-soc-avs-rt274.o +obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_RT286) += snd-soc-avs-rt286.o diff --git a/sound/soc/intel/avs/boards/rt286.c b/sound/soc/intel/avs/boards/rt286.c new file mode 100644 index 000000000000..e51d4e181274 --- /dev/null +++ b/sound/soc/intel/avs/boards/rt286.c @@ -0,0 +1,281 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// +// Authors: Cezary Rojewski +// Amadeusz Slawinski +// + +#include +#include +#include +#include +#include +#include +#include "../../../codecs/rt286.h" + +static const struct snd_kcontrol_new card_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Mic Jack"), + SOC_DAPM_PIN_SWITCH("Speaker"), +}; + +static const struct snd_soc_dapm_widget card_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), +}; + +static const struct snd_soc_dapm_route card_base_routes[] = { + /* HP jack connectors - unknown if we have jack detect */ + {"Headphone Jack", NULL, "HPO Pin"}, + {"MIC1", NULL, "Mic Jack"}, + + {"Speaker", NULL, "SPOR"}, + {"Speaker", NULL, "SPOL"}, +}; + +static struct snd_soc_jack_pin card_headset_pins[] = { + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Mic Jack", + .mask = SND_JACK_MICROPHONE, + }, +}; + +static int avs_rt286_codec_init(struct snd_soc_pcm_runtime *runtime) +{ + struct snd_soc_component *component = asoc_rtd_to_codec(runtime, 0)->component; + struct snd_soc_jack_pin *pins; + struct snd_soc_jack *jack; + struct snd_soc_card *card = runtime->card; + int num_pins, ret; + + jack = snd_soc_card_get_drvdata(card); + num_pins = ARRAY_SIZE(card_headset_pins); + + pins = devm_kmemdup(card->dev, card_headset_pins, sizeof(*pins) * num_pins, GFP_KERNEL); + if (!pins) + return -ENOMEM; + + ret = snd_soc_card_jack_new_pins(card, "Headset", SND_JACK_HEADSET | SND_JACK_BTN_0, jack, + pins, num_pins); + if (ret) + return ret; + + snd_soc_component_set_jack(component, jack, NULL); + + return 0; +} + +static int avs_rt286_be_fixup(struct snd_soc_pcm_runtime *runtime, struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate, *channels; + struct snd_mask *fmt; + + rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + + /* The ADSP will convert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP0 to 24 bit */ + snd_mask_none(fmt); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); + + return 0; +} + +static int +avs_rt286_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *runtime = substream->private_data; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(runtime, 0); + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, RT286_SCLK_S_PLL, 24000000, SND_SOC_CLOCK_IN); + if (ret < 0) + dev_err(runtime->dev, "Set codec sysclk failed: %d\n", ret); + + return ret; +} + +static const struct snd_soc_ops avs_rt286_ops = { + .hw_params = avs_rt286_hw_params, +}; + +static int avs_create_dai_link(struct device *dev, const char *platform_name, int ssp_port, + struct snd_soc_dai_link **dai_link) +{ + struct snd_soc_dai_link_component *platform; + struct snd_soc_dai_link *dl; + + dl = devm_kzalloc(dev, sizeof(*dl), GFP_KERNEL); + platform = devm_kzalloc(dev, sizeof(*platform), GFP_KERNEL); + if (!dl || !platform) + return -ENOMEM; + + platform->name = platform_name; + + dl->name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-Codec", ssp_port); + dl->cpus = devm_kzalloc(dev, sizeof(*dl->cpus), GFP_KERNEL); + dl->codecs = devm_kzalloc(dev, sizeof(*dl->codecs), GFP_KERNEL); + if (!dl->name || !dl->cpus || !dl->codecs) + return -ENOMEM; + + dl->cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", ssp_port); + dl->codecs->name = devm_kasprintf(dev, GFP_KERNEL, "i2c-INT343A:00"); + dl->codecs->dai_name = devm_kasprintf(dev, GFP_KERNEL, "rt286-aif1"); + if (!dl->cpus->dai_name || !dl->codecs->name || !dl->codecs->dai_name) + return -ENOMEM; + + dl->num_cpus = 1; + dl->num_codecs = 1; + dl->platforms = platform; + dl->num_platforms = 1; + dl->id = 0; + dl->dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS; + dl->init = avs_rt286_codec_init; + dl->be_hw_params_fixup = avs_rt286_be_fixup; + dl->ops = &avs_rt286_ops; + dl->nonatomic = 1; + dl->no_pcm = 1; + dl->dpcm_capture = 1; + dl->dpcm_playback = 1; + + *dai_link = dl; + + return 0; +} + +static int avs_create_dapm_routes(struct device *dev, int ssp_port, + struct snd_soc_dapm_route **routes, int *num_routes) +{ + struct snd_soc_dapm_route *dr; + const int num_base = ARRAY_SIZE(card_base_routes); + const int num_dr = num_base + 2; + int idx; + + dr = devm_kcalloc(dev, num_dr, sizeof(*dr), GFP_KERNEL); + if (!dr) + return -ENOMEM; + + memcpy(dr, card_base_routes, num_base * sizeof(*dr)); + + idx = num_base; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "AIF1 Playback"); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Tx", ssp_port); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + idx++; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Rx", ssp_port); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "AIF1 Capture"); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + *routes = dr; + *num_routes = num_dr; + + return 0; +} + +static int avs_card_set_jack(struct snd_soc_card *card, struct snd_soc_jack *jack) +{ + struct snd_soc_component *component; + + for_each_card_components(card, component) + snd_soc_component_set_jack(component, jack, NULL); + return 0; +} + +static int avs_card_remove(struct snd_soc_card *card) +{ + return avs_card_set_jack(card, NULL); +} + +static int avs_card_suspend_pre(struct snd_soc_card *card) +{ + return avs_card_set_jack(card, NULL); +} + +static int avs_card_resume_post(struct snd_soc_card *card) +{ + struct snd_soc_jack *jack = snd_soc_card_get_drvdata(card); + + return avs_card_set_jack(card, jack); +} + +static int avs_rt286_probe(struct platform_device *pdev) +{ + struct snd_soc_dapm_route *routes; + struct snd_soc_dai_link *dai_link; + struct snd_soc_acpi_mach *mach; + struct snd_soc_card *card; + struct snd_soc_jack *jack; + struct device *dev = &pdev->dev; + const char *pname; + int num_routes, ssp_port, ret; + + mach = dev_get_platdata(dev); + pname = mach->mach_params.platform; + ssp_port = __ffs(mach->mach_params.i2s_link_mask); + + ret = avs_create_dai_link(dev, pname, ssp_port, &dai_link); + if (ret) { + dev_err(dev, "Failed to create dai link: %d", ret); + return ret; + } + + ret = avs_create_dapm_routes(dev, ssp_port, &routes, &num_routes); + if (ret) { + dev_err(dev, "Failed to create dapm routes: %d", ret); + return ret; + } + + jack = devm_kzalloc(dev, sizeof(*jack), GFP_KERNEL); + card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); + if (!jack || !card) + return -ENOMEM; + + card->name = "avs_rt286"; + card->dev = dev; + card->owner = THIS_MODULE; + card->remove = avs_card_remove; + card->suspend_pre = avs_card_suspend_pre; + card->resume_post = avs_card_resume_post; + card->dai_link = dai_link; + card->num_links = 1; + card->controls = card_controls; + card->num_controls = ARRAY_SIZE(card_controls); + card->dapm_widgets = card_widgets; + card->num_dapm_widgets = ARRAY_SIZE(card_widgets); + card->dapm_routes = routes; + card->num_dapm_routes = num_routes; + card->fully_routed = true; + snd_soc_card_set_drvdata(card, jack); + + ret = snd_soc_fixup_dai_links_platform_name(card, pname); + if (ret) + return ret; + + return devm_snd_soc_register_card(dev, card); +} + +static struct platform_driver avs_rt286_driver = { + .probe = avs_rt286_probe, + .driver = { + .name = "avs_rt286", + .pm = &snd_soc_pm_ops, + }, +}; + +module_platform_driver(avs_rt286_driver); + +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:avs_rt286"); -- cgit v1.2.3 From 88429ab16df4cd4a1a77d45b90ec95cf62cc22d1 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Wed, 11 May 2022 18:23:57 +0200 Subject: ASoC: Intel: avs: Add rt298 machine board MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit To support AVS-rt298 configuration add machine board connecting AVS platform component driver with rt298 codec one. Signed-off-by: Amadeusz Sławiński Signed-off-by: Cezary Rojewski Link: https://lore.kernel.org/r/20220511162403.3987658-9-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/boards/Kconfig | 10 ++ sound/soc/intel/avs/boards/Makefile | 2 + sound/soc/intel/avs/boards/rt298.c | 281 ++++++++++++++++++++++++++++++++++++ 3 files changed, 293 insertions(+) create mode 100644 sound/soc/intel/avs/boards/rt298.c (limited to 'sound') diff --git a/sound/soc/intel/avs/boards/Kconfig b/sound/soc/intel/avs/boards/Kconfig index 707e9e96746d..b4dc2b02097d 100644 --- a/sound/soc/intel/avs/boards/Kconfig +++ b/sound/soc/intel/avs/boards/Kconfig @@ -46,4 +46,14 @@ config SND_SOC_INTEL_AVS_MACH_RT286 Say Y or m if you have such a device. This is a recommended option. If unsure select "N". +config SND_SOC_INTEL_AVS_MACH_RT298 + tristate "rt298 in I2S mode" + depends on I2C + depends on MFD_INTEL_LPSS || COMPILE_TEST + select SND_SOC_RT298 + help + This adds support for ASoC machine driver with RT298 I2S audio codec. + Say Y or m if you have such a device. This is a recommended option. + If unsure select "N". + endmenu diff --git a/sound/soc/intel/avs/boards/Makefile b/sound/soc/intel/avs/boards/Makefile index 7ea4ad38c7df..0fd664694c8c 100644 --- a/sound/soc/intel/avs/boards/Makefile +++ b/sound/soc/intel/avs/boards/Makefile @@ -5,9 +5,11 @@ snd-soc-avs-hdaudio-objs := hdaudio.o snd-soc-avs-i2s-test-objs := i2s_test.o snd-soc-avs-rt274-objs := rt274.o snd-soc-avs-rt286-objs := rt286.o +snd-soc-avs-rt298-objs := rt298.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_DMIC) += snd-soc-avs-dmic.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_HDAUDIO) += snd-soc-avs-hdaudio.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_i2s_TEST) += snd-soc-avs-i2s-test.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_RT274) += snd-soc-avs-rt274.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_RT286) += snd-soc-avs-rt286.o +obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_RT298) += snd-soc-avs-rt298.o diff --git a/sound/soc/intel/avs/boards/rt298.c b/sound/soc/intel/avs/boards/rt298.c new file mode 100644 index 000000000000..b28d36872dcb --- /dev/null +++ b/sound/soc/intel/avs/boards/rt298.c @@ -0,0 +1,281 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// +// Authors: Cezary Rojewski +// Amadeusz Slawinski +// + +#include +#include +#include +#include +#include +#include +#include "../../../codecs/rt298.h" + +static const struct snd_kcontrol_new card_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Mic Jack"), + SOC_DAPM_PIN_SWITCH("Speaker"), +}; + +static const struct snd_soc_dapm_widget card_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), +}; + +static const struct snd_soc_dapm_route card_base_routes[] = { + /* HP jack connectors - unknown if we have jack detect */ + {"Headphone Jack", NULL, "HPO Pin"}, + {"MIC1", NULL, "Mic Jack"}, + + {"Speaker", NULL, "SPOR"}, + {"Speaker", NULL, "SPOL"}, +}; + +static struct snd_soc_jack_pin card_headset_pins[] = { + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Mic Jack", + .mask = SND_JACK_MICROPHONE, + }, +}; + +static int avs_rt298_codec_init(struct snd_soc_pcm_runtime *runtime) +{ + struct snd_soc_component *component = asoc_rtd_to_codec(runtime, 0)->component; + struct snd_soc_jack_pin *pins; + struct snd_soc_jack *jack; + struct snd_soc_card *card = runtime->card; + int num_pins, ret; + + jack = snd_soc_card_get_drvdata(card); + num_pins = ARRAY_SIZE(card_headset_pins); + + pins = devm_kmemdup(card->dev, card_headset_pins, sizeof(*pins) * num_pins, GFP_KERNEL); + if (!pins) + return -ENOMEM; + + ret = snd_soc_card_jack_new_pins(card, "Headset", SND_JACK_HEADSET | SND_JACK_BTN_0, jack, + pins, num_pins); + if (ret) + return ret; + + snd_soc_component_set_jack(component, jack, NULL); + + return 0; +} + +static int avs_rt298_be_fixup(struct snd_soc_pcm_runtime *runtime, struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate, *channels; + struct snd_mask *fmt; + + rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + + /* The ADSP will convert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP0 to 24 bit */ + snd_mask_none(fmt); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); + + return 0; +} + +static int +avs_rt298_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, RT298_SCLK_S_PLL, 19200000, SND_SOC_CLOCK_IN); + if (ret < 0) + dev_err(rtd->dev, "Set codec sysclk failed: %d\n", ret); + + return ret; +} + +static const struct snd_soc_ops avs_rt298_ops = { + .hw_params = avs_rt298_hw_params, +}; + +static int avs_create_dai_link(struct device *dev, const char *platform_name, int ssp_port, + struct snd_soc_dai_link **dai_link) +{ + struct snd_soc_dai_link_component *platform; + struct snd_soc_dai_link *dl; + + dl = devm_kzalloc(dev, sizeof(*dl), GFP_KERNEL); + platform = devm_kzalloc(dev, sizeof(*platform), GFP_KERNEL); + if (!dl || !platform) + return -ENOMEM; + + platform->name = platform_name; + + dl->name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-Codec", ssp_port); + dl->cpus = devm_kzalloc(dev, sizeof(*dl->cpus), GFP_KERNEL); + dl->codecs = devm_kzalloc(dev, sizeof(*dl->codecs), GFP_KERNEL); + if (!dl->name || !dl->cpus || !dl->codecs) + return -ENOMEM; + + dl->cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", ssp_port); + dl->codecs->name = devm_kasprintf(dev, GFP_KERNEL, "i2c-INT343A:00"); + dl->codecs->dai_name = devm_kasprintf(dev, GFP_KERNEL, "rt298-aif1"); + if (!dl->cpus->dai_name || !dl->codecs->name || !dl->codecs->dai_name) + return -ENOMEM; + + dl->num_cpus = 1; + dl->num_codecs = 1; + dl->platforms = platform; + dl->num_platforms = 1; + dl->id = 0; + dl->dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS; + dl->init = avs_rt298_codec_init; + dl->be_hw_params_fixup = avs_rt298_be_fixup; + dl->ops = &avs_rt298_ops; + dl->nonatomic = 1; + dl->no_pcm = 1; + dl->dpcm_capture = 1; + dl->dpcm_playback = 1; + + *dai_link = dl; + + return 0; +} + +static int avs_create_dapm_routes(struct device *dev, int ssp_port, + struct snd_soc_dapm_route **routes, int *num_routes) +{ + struct snd_soc_dapm_route *dr; + const int num_base = ARRAY_SIZE(card_base_routes); + const int num_dr = num_base + 2; + int idx; + + dr = devm_kcalloc(dev, num_dr, sizeof(*dr), GFP_KERNEL); + if (!dr) + return -ENOMEM; + + memcpy(dr, card_base_routes, num_base * sizeof(*dr)); + + idx = num_base; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "AIF1 Playback"); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Tx", ssp_port); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + idx++; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Rx", ssp_port); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "AIF1 Capture"); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + *routes = dr; + *num_routes = num_dr; + + return 0; +} + +static int avs_card_set_jack(struct snd_soc_card *card, struct snd_soc_jack *jack) +{ + struct snd_soc_component *component; + + for_each_card_components(card, component) + snd_soc_component_set_jack(component, jack, NULL); + return 0; +} + +static int avs_card_remove(struct snd_soc_card *card) +{ + return avs_card_set_jack(card, NULL); +} + +static int avs_card_suspend_pre(struct snd_soc_card *card) +{ + return avs_card_set_jack(card, NULL); +} + +static int avs_card_resume_post(struct snd_soc_card *card) +{ + struct snd_soc_jack *jack = snd_soc_card_get_drvdata(card); + + return avs_card_set_jack(card, jack); +} + +static int avs_rt298_probe(struct platform_device *pdev) +{ + struct snd_soc_dapm_route *routes; + struct snd_soc_dai_link *dai_link; + struct snd_soc_acpi_mach *mach; + struct snd_soc_card *card; + struct snd_soc_jack *jack; + struct device *dev = &pdev->dev; + const char *pname; + int num_routes, ssp_port, ret; + + mach = dev_get_platdata(dev); + pname = mach->mach_params.platform; + ssp_port = __ffs(mach->mach_params.i2s_link_mask); + + ret = avs_create_dai_link(dev, pname, ssp_port, &dai_link); + if (ret) { + dev_err(dev, "Failed to create dai link: %d", ret); + return ret; + } + + ret = avs_create_dapm_routes(dev, ssp_port, &routes, &num_routes); + if (ret) { + dev_err(dev, "Failed to create dapm routes: %d", ret); + return ret; + } + + jack = devm_kzalloc(dev, sizeof(*jack), GFP_KERNEL); + card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); + if (!jack || !card) + return -ENOMEM; + + card->name = "avs_rt298"; + card->dev = dev; + card->owner = THIS_MODULE; + card->remove = avs_card_remove; + card->suspend_pre = avs_card_suspend_pre; + card->resume_post = avs_card_resume_post; + card->dai_link = dai_link; + card->num_links = 1; + card->controls = card_controls; + card->num_controls = ARRAY_SIZE(card_controls); + card->dapm_widgets = card_widgets; + card->num_dapm_widgets = ARRAY_SIZE(card_widgets); + card->dapm_routes = routes; + card->num_dapm_routes = num_routes; + card->fully_routed = true; + snd_soc_card_set_drvdata(card, jack); + + ret = snd_soc_fixup_dai_links_platform_name(card, pname); + if (ret) + return ret; + + return devm_snd_soc_register_card(dev, card); +} + +static struct platform_driver avs_rt298_driver = { + .probe = avs_rt298_probe, + .driver = { + .name = "avs_rt298", + .pm = &snd_soc_pm_ops, + }, +}; + +module_platform_driver(avs_rt298_driver); + +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:avs_rt298"); -- cgit v1.2.3 From 748102786b3ce0bf402c2dc42386cbfaab71ac39 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Wed, 11 May 2022 18:23:58 +0200 Subject: ASoC: Intel: avs: Add rt5682 machine board MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit To support AVS-rt5682 configuration add machine board connecting AVS platform component driver with rt5682 codec one. Signed-off-by: Amadeusz Sławiński Signed-off-by: Cezary Rojewski Link: https://lore.kernel.org/r/20220511162403.3987658-10-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/boards/Kconfig | 10 ++ sound/soc/intel/avs/boards/Makefile | 2 + sound/soc/intel/avs/boards/rt5682.c | 340 ++++++++++++++++++++++++++++++++++++ 3 files changed, 352 insertions(+) create mode 100644 sound/soc/intel/avs/boards/rt5682.c (limited to 'sound') diff --git a/sound/soc/intel/avs/boards/Kconfig b/sound/soc/intel/avs/boards/Kconfig index b4dc2b02097d..767eae57be57 100644 --- a/sound/soc/intel/avs/boards/Kconfig +++ b/sound/soc/intel/avs/boards/Kconfig @@ -56,4 +56,14 @@ config SND_SOC_INTEL_AVS_MACH_RT298 Say Y or m if you have such a device. This is a recommended option. If unsure select "N". +config SND_SOC_INTEL_AVS_MACH_RT5682 + tristate "rt5682 in I2S mode" + depends on I2C + depends on MFD_INTEL_LPSS || COMPILE_TEST + select SND_SOC_RT5682_I2C + help + This adds support for ASoC machine driver with RT5682 I2S audio codec. + Say Y or m if you have such a device. This is a recommended option. + If unsure select "N". + endmenu diff --git a/sound/soc/intel/avs/boards/Makefile b/sound/soc/intel/avs/boards/Makefile index 0fd664694c8c..fd49bd2a5876 100644 --- a/sound/soc/intel/avs/boards/Makefile +++ b/sound/soc/intel/avs/boards/Makefile @@ -6,6 +6,7 @@ snd-soc-avs-i2s-test-objs := i2s_test.o snd-soc-avs-rt274-objs := rt274.o snd-soc-avs-rt286-objs := rt286.o snd-soc-avs-rt298-objs := rt298.o +snd-soc-avs-rt5682-objs := rt5682.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_DMIC) += snd-soc-avs-dmic.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_HDAUDIO) += snd-soc-avs-hdaudio.o @@ -13,3 +14,4 @@ obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_i2s_TEST) += snd-soc-avs-i2s-test.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_RT274) += snd-soc-avs-rt274.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_RT286) += snd-soc-avs-rt286.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_RT298) += snd-soc-avs-rt298.o +obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_RT5682) += snd-soc-avs-rt5682.o diff --git a/sound/soc/intel/avs/boards/rt5682.c b/sound/soc/intel/avs/boards/rt5682.c new file mode 100644 index 000000000000..01f9b9f0c12b --- /dev/null +++ b/sound/soc/intel/avs/boards/rt5682.c @@ -0,0 +1,340 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// +// Authors: Cezary Rojewski +// Amadeusz Slawinski +// + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "../../common/soc-intel-quirks.h" +#include "../../../codecs/rt5682.h" + +#define AVS_RT5682_SSP_CODEC(quirk) ((quirk) & GENMASK(2, 0)) +#define AVS_RT5682_SSP_CODEC_MASK (GENMASK(2, 0)) +#define AVS_RT5682_MCLK_EN BIT(3) +#define AVS_RT5682_MCLK_24MHZ BIT(4) + +/* Default: MCLK on, MCLK 19.2M, SSP0 */ +static unsigned long avs_rt5682_quirk = AVS_RT5682_MCLK_EN | AVS_RT5682_SSP_CODEC(0); + +static int avs_rt5682_quirk_cb(const struct dmi_system_id *id) +{ + avs_rt5682_quirk = (unsigned long)id->driver_data; + return 1; +} + +static const struct dmi_system_id avs_rt5682_quirk_table[] = { + { + .callback = avs_rt5682_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_PRODUCT_NAME, "WhiskeyLake Client"), + }, + .driver_data = (void *)(AVS_RT5682_MCLK_EN | + AVS_RT5682_MCLK_24MHZ | + AVS_RT5682_SSP_CODEC(1)), + }, + { + .callback = avs_rt5682_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_PRODUCT_NAME, "Ice Lake Client"), + }, + .driver_data = (void *)(AVS_RT5682_MCLK_EN | + AVS_RT5682_SSP_CODEC(0)), + }, + {} +}; + +static const struct snd_kcontrol_new card_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), +}; + +static const struct snd_soc_dapm_widget card_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), +}; + +static const struct snd_soc_dapm_route card_base_routes[] = { + /* HP jack connectors - unknown if we have jack detect */ + { "Headphone Jack", NULL, "HPOL" }, + { "Headphone Jack", NULL, "HPOR" }, + + /* other jacks */ + { "IN1P", NULL, "Headset Mic" }, +}; + +static int avs_rt5682_codec_init(struct snd_soc_pcm_runtime *runtime) +{ + struct snd_soc_component *component = asoc_rtd_to_codec(runtime, 0)->component; + struct snd_soc_jack *jack; + struct snd_soc_card *card = runtime->card; + int ret; + + jack = snd_soc_card_get_drvdata(card); + + /* Need to enable ASRC function for 24MHz mclk rate */ + if ((avs_rt5682_quirk & AVS_RT5682_MCLK_EN) && + (avs_rt5682_quirk & AVS_RT5682_MCLK_24MHZ)) { + rt5682_sel_asrc_clk_src(component, RT5682_DA_STEREO1_FILTER | + RT5682_AD_STEREO1_FILTER, RT5682_CLK_SEL_I2S1_ASRC); + } + + /* + * Headset buttons map to the google Reference headset. + * These can be configured by userspace. + */ + ret = snd_soc_card_jack_new(card, "Headset", SND_JACK_HEADSET | SND_JACK_BTN_0 | + SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3, jack); + if (ret) { + dev_err(card->dev, "Headset Jack creation failed: %d\n", ret); + return ret; + } + + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); + + ret = snd_soc_component_set_jack(component, jack, NULL); + if (ret) { + dev_err(card->dev, "Headset Jack call-back failed: %d\n", ret); + return ret; + } + + return 0; +}; + +static int +avs_rt5682_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *runtime = asoc_substream_to_rtd(substream); + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(runtime, 0); + int clk_id, clk_freq; + int pll_out, ret; + + if (avs_rt5682_quirk & AVS_RT5682_MCLK_EN) { + clk_id = RT5682_PLL1_S_MCLK; + if (avs_rt5682_quirk & AVS_RT5682_MCLK_24MHZ) + clk_freq = 24000000; + else + clk_freq = 19200000; + } else { + clk_id = RT5682_PLL1_S_BCLK1; + clk_freq = params_rate(params) * 50; + } + + pll_out = params_rate(params) * 512; + + ret = snd_soc_dai_set_pll(codec_dai, 0, clk_id, clk_freq, pll_out); + if (ret < 0) + dev_err(runtime->dev, "snd_soc_dai_set_pll err = %d\n", ret); + + /* Configure sysclk for codec */ + ret = snd_soc_dai_set_sysclk(codec_dai, RT5682_SCLK_S_PLL1, pll_out, SND_SOC_CLOCK_IN); + if (ret < 0) + dev_err(runtime->dev, "snd_soc_dai_set_sysclk err = %d\n", ret); + + /* slot_width should equal or large than data length, set them be the same */ + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x0, 0x0, 2, params_width(params)); + if (ret < 0) { + dev_err(runtime->dev, "set TDM slot err:%d\n", ret); + return ret; + } + + return 0; +} + +static const struct snd_soc_ops avs_rt5682_ops = { + .hw_params = avs_rt5682_hw_params, +}; + +static int avs_create_dai_link(struct device *dev, const char *platform_name, int ssp_port, + struct snd_soc_dai_link **dai_link) +{ + struct snd_soc_dai_link_component *platform; + struct snd_soc_dai_link *dl; + + dl = devm_kzalloc(dev, sizeof(*dl), GFP_KERNEL); + platform = devm_kzalloc(dev, sizeof(*platform), GFP_KERNEL); + if (!dl || !platform) + return -ENOMEM; + + platform->name = platform_name; + + dl->name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-Codec", ssp_port); + dl->cpus = devm_kzalloc(dev, sizeof(*dl->cpus), GFP_KERNEL); + dl->codecs = devm_kzalloc(dev, sizeof(*dl->codecs), GFP_KERNEL); + if (!dl->name || !dl->cpus || !dl->codecs) + return -ENOMEM; + + dl->cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", ssp_port); + dl->codecs->name = devm_kasprintf(dev, GFP_KERNEL, "i2c-10EC5682:00"); + dl->codecs->dai_name = devm_kasprintf(dev, GFP_KERNEL, "rt5682-aif1"); + if (!dl->cpus->dai_name || !dl->codecs->name || !dl->codecs->dai_name) + return -ENOMEM; + + dl->num_cpus = 1; + dl->num_codecs = 1; + dl->platforms = platform; + dl->num_platforms = 1; + dl->id = 0; + dl->init = avs_rt5682_codec_init; + dl->ops = &avs_rt5682_ops; + dl->nonatomic = 1; + dl->no_pcm = 1; + dl->dpcm_capture = 1; + dl->dpcm_playback = 1; + + *dai_link = dl; + + return 0; +} + +static int avs_create_dapm_routes(struct device *dev, int ssp_port, + struct snd_soc_dapm_route **routes, int *num_routes) +{ + struct snd_soc_dapm_route *dr; + const int num_base = ARRAY_SIZE(card_base_routes); + const int num_dr = num_base + 2; + int idx; + + dr = devm_kcalloc(dev, num_dr, sizeof(*dr), GFP_KERNEL); + if (!dr) + return -ENOMEM; + + memcpy(dr, card_base_routes, num_base * sizeof(*dr)); + + idx = num_base; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "AIF1 Playback"); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Tx", ssp_port); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + idx++; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Rx", ssp_port); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "AIF1 Capture"); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + *routes = dr; + *num_routes = num_dr; + + return 0; +} + +static int avs_card_set_jack(struct snd_soc_card *card, struct snd_soc_jack *jack) +{ + struct snd_soc_component *component; + + for_each_card_components(card, component) + snd_soc_component_set_jack(component, jack, NULL); + return 0; +} + +static int avs_card_remove(struct snd_soc_card *card) +{ + return avs_card_set_jack(card, NULL); +} + +static int avs_card_suspend_pre(struct snd_soc_card *card) +{ + return avs_card_set_jack(card, NULL); +} + +static int avs_card_resume_post(struct snd_soc_card *card) +{ + struct snd_soc_jack *jack = snd_soc_card_get_drvdata(card); + + return avs_card_set_jack(card, jack); +} + +static int avs_rt5682_probe(struct platform_device *pdev) +{ + struct snd_soc_dapm_route *routes; + struct snd_soc_dai_link *dai_link; + struct snd_soc_acpi_mach *mach; + struct snd_soc_card *card; + struct snd_soc_jack *jack; + struct device *dev = &pdev->dev; + const char *pname; + int num_routes, ssp_port, ret; + + if (pdev->id_entry && pdev->id_entry->driver_data) + avs_rt5682_quirk = (unsigned long)pdev->id_entry->driver_data; + + dmi_check_system(avs_rt5682_quirk_table); + dev_dbg(dev, "avs_rt5682_quirk = %lx\n", avs_rt5682_quirk); + + mach = dev_get_platdata(dev); + pname = mach->mach_params.platform; + ssp_port = __ffs(mach->mach_params.i2s_link_mask); + + ret = avs_create_dai_link(dev, pname, ssp_port, &dai_link); + if (ret) { + dev_err(dev, "Failed to create dai link: %d", ret); + return ret; + } + + ret = avs_create_dapm_routes(dev, ssp_port, &routes, &num_routes); + if (ret) { + dev_err(dev, "Failed to create dapm routes: %d", ret); + return ret; + } + + jack = devm_kzalloc(dev, sizeof(*jack), GFP_KERNEL); + card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); + if (!jack || !card) + return -ENOMEM; + + card->name = "avs_rt5682"; + card->dev = dev; + card->owner = THIS_MODULE; + card->remove = avs_card_remove; + card->suspend_pre = avs_card_suspend_pre; + card->resume_post = avs_card_resume_post; + card->dai_link = dai_link; + card->num_links = 1; + card->controls = card_controls; + card->num_controls = ARRAY_SIZE(card_controls); + card->dapm_widgets = card_widgets; + card->num_dapm_widgets = ARRAY_SIZE(card_widgets); + card->dapm_routes = routes; + card->num_dapm_routes = num_routes; + card->fully_routed = true; + snd_soc_card_set_drvdata(card, jack); + + ret = snd_soc_fixup_dai_links_platform_name(card, pname); + if (ret) + return ret; + + return devm_snd_soc_register_card(dev, card); +} + +static struct platform_driver avs_rt5682_driver = { + .probe = avs_rt5682_probe, + .driver = { + .name = "avs_rt5682", + .pm = &snd_soc_pm_ops, + }, +}; + +module_platform_driver(avs_rt5682_driver) + +MODULE_AUTHOR("Cezary Rojewski "); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:avs_rt5682"); -- cgit v1.2.3 From 32ee40b5590081a6b38a55e4ab16b47085f93afe Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Wed, 11 May 2022 18:23:59 +0200 Subject: ASoC: Intel: avs: Add nau8825 machine board MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit To support AVS-nau8825 configuration add machine board connecting AVS platform component driver with nau8825 codec one. Signed-off-by: Amadeusz Sławiński Signed-off-by: Cezary Rojewski Link: https://lore.kernel.org/r/20220511162403.3987658-11-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/boards/Kconfig | 11 ++ sound/soc/intel/avs/boards/Makefile | 2 + sound/soc/intel/avs/boards/nau8825.c | 353 +++++++++++++++++++++++++++++++++++ 3 files changed, 366 insertions(+) create mode 100644 sound/soc/intel/avs/boards/nau8825.c (limited to 'sound') diff --git a/sound/soc/intel/avs/boards/Kconfig b/sound/soc/intel/avs/boards/Kconfig index 767eae57be57..6bf8fa1924a2 100644 --- a/sound/soc/intel/avs/boards/Kconfig +++ b/sound/soc/intel/avs/boards/Kconfig @@ -26,6 +26,17 @@ config SND_SOC_INTEL_AVS_MACH_I2S_TEST This adds support for I2S test-board which can be used to verify transfer over I2S interface with SSP loopback scenarios. +config SND_SOC_INTEL_AVS_MACH_NAU8825 + tristate "nau8825 I2S board" + depends on I2C + depends on MFD_INTEL_LPSS || COMPILE_TEST + select SND_SOC_NAU8825 + help + This adds support for ASoC machine driver with NAU8825 I2S audio codec. + It is meant to be used with AVS driver. + Say Y or m if you have such a device. This is a recommended option. + If unsure select "N". + config SND_SOC_INTEL_AVS_MACH_RT274 tristate "rt274 in I2S mode" depends on I2C diff --git a/sound/soc/intel/avs/boards/Makefile b/sound/soc/intel/avs/boards/Makefile index fd49bd2a5876..9ac14b269f56 100644 --- a/sound/soc/intel/avs/boards/Makefile +++ b/sound/soc/intel/avs/boards/Makefile @@ -3,6 +3,7 @@ snd-soc-avs-dmic-objs := dmic.o snd-soc-avs-hdaudio-objs := hdaudio.o snd-soc-avs-i2s-test-objs := i2s_test.o +snd-soc-avs-nau8825-objs := nau8825.o snd-soc-avs-rt274-objs := rt274.o snd-soc-avs-rt286-objs := rt286.o snd-soc-avs-rt298-objs := rt298.o @@ -11,6 +12,7 @@ snd-soc-avs-rt5682-objs := rt5682.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_DMIC) += snd-soc-avs-dmic.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_HDAUDIO) += snd-soc-avs-hdaudio.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_i2s_TEST) += snd-soc-avs-i2s-test.o +obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_NAU8825) += snd-soc-avs-nau8825.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_RT274) += snd-soc-avs-rt274.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_RT286) += snd-soc-avs-rt286.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_RT298) += snd-soc-avs-rt298.o diff --git a/sound/soc/intel/avs/boards/nau8825.c b/sound/soc/intel/avs/boards/nau8825.c new file mode 100644 index 000000000000..f76909e9f990 --- /dev/null +++ b/sound/soc/intel/avs/boards/nau8825.c @@ -0,0 +1,353 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// +// Authors: Cezary Rojewski +// Amadeusz Slawinski +// + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "../../../codecs/nau8825.h" + +#define SKL_NUVOTON_CODEC_DAI "nau8825-hifi" + +static int +avs_nau8825_clock_control(struct snd_soc_dapm_widget *w, struct snd_kcontrol *control, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct snd_soc_dai *codec_dai; + int ret; + + codec_dai = snd_soc_card_get_codec_dai(card, SKL_NUVOTON_CODEC_DAI); + if (!codec_dai) { + dev_err(card->dev, "Codec dai not found\n"); + return -EINVAL; + } + + if (!SND_SOC_DAPM_EVENT_ON(event)) { + ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_INTERNAL, 0, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(card->dev, "set sysclk err = %d\n", ret); + return ret; + } + } + + return 0; +} + +static const struct snd_kcontrol_new card_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), +}; + +static const struct snd_soc_dapm_widget card_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, avs_nau8825_clock_control, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route card_base_routes[] = { + { "Headphone Jack", NULL, "HPOL" }, + { "Headphone Jack", NULL, "HPOR" }, + + { "MIC", NULL, "Headset Mic" }, + + { "Headphone Jack", NULL, "Platform Clock" }, + { "Headset Mic", NULL, "Platform Clock" }, +}; + +static struct snd_soc_jack_pin card_headset_pins[] = { + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, +}; + +static int avs_nau8825_codec_init(struct snd_soc_pcm_runtime *runtime) +{ + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(runtime, 0); + struct snd_soc_component *component = codec_dai->component; + struct snd_soc_jack_pin *pins; + struct snd_soc_jack *jack; + struct snd_soc_card *card = runtime->card; + int num_pins, ret; + + jack = snd_soc_card_get_drvdata(card); + num_pins = ARRAY_SIZE(card_headset_pins); + + pins = devm_kmemdup(card->dev, card_headset_pins, sizeof(*pins) * num_pins, GFP_KERNEL); + if (!pins) + return -ENOMEM; + + /* + * 4 buttons here map to the google Reference headset. + * The use of these buttons can be decided by the user space. + */ + ret = snd_soc_card_jack_new_pins(card, "Headset", SND_JACK_HEADSET | SND_JACK_BTN_0 | + SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3, + jack, pins, num_pins); + if (ret) + return ret; + + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); + + //snd_soc_component_set_jack(component, jack, NULL); + // TODO: Fix nau8825 codec to use .set_jack, like everyone else + nau8825_enable_jack_detect(component, jack); + + return 0; +} + +static int +avs_nau8825_be_fixup(struct snd_soc_pcm_runtime *runtime, struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate, *channels; + struct snd_mask *fmt; + + rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + + /* The ADSP will convert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP to 24 bit */ + snd_mask_none(fmt); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); + + return 0; +} + +static int avs_nau8825_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtm = asoc_substream_to_rtd(substream); + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtm, 0); + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_FLL_FS, 0, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "can't set FS clock %d\n", ret); + break; + } + + ret = snd_soc_dai_set_pll(codec_dai, 0, 0, runtime->rate, runtime->rate * 256); + if (ret < 0) + dev_err(codec_dai->dev, "can't set FLL: %d\n", ret); + break; + + case SNDRV_PCM_TRIGGER_RESUME: + ret = snd_soc_dai_set_pll(codec_dai, 0, 0, runtime->rate, runtime->rate * 256); + if (ret < 0) + dev_err(codec_dai->dev, "can't set FLL: %d\n", ret); + break; + } + + return ret; +} + + +static const struct snd_soc_ops avs_nau8825_ops = { + .trigger = avs_nau8825_trigger, +}; + +static int avs_create_dai_link(struct device *dev, const char *platform_name, int ssp_port, + struct snd_soc_dai_link **dai_link) +{ + struct snd_soc_dai_link_component *platform; + struct snd_soc_dai_link *dl; + + dl = devm_kzalloc(dev, sizeof(*dl), GFP_KERNEL); + platform = devm_kzalloc(dev, sizeof(*platform), GFP_KERNEL); + if (!dl || !platform) + return -ENOMEM; + + platform->name = platform_name; + + dl->name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-Codec", ssp_port); + dl->cpus = devm_kzalloc(dev, sizeof(*dl->cpus), GFP_KERNEL); + dl->codecs = devm_kzalloc(dev, sizeof(*dl->codecs), GFP_KERNEL); + if (!dl->name || !dl->cpus || !dl->codecs) + return -ENOMEM; + + dl->cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", ssp_port); + dl->codecs->name = devm_kasprintf(dev, GFP_KERNEL, "i2c-10508825:00"); + dl->codecs->dai_name = devm_kasprintf(dev, GFP_KERNEL, SKL_NUVOTON_CODEC_DAI); + if (!dl->cpus->dai_name || !dl->codecs->name || !dl->codecs->dai_name) + return -ENOMEM; + + dl->num_cpus = 1; + dl->num_codecs = 1; + dl->platforms = platform; + dl->num_platforms = 1; + dl->id = 0; + dl->dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS; + dl->init = avs_nau8825_codec_init; + dl->be_hw_params_fixup = avs_nau8825_be_fixup; + dl->ops = &avs_nau8825_ops; + dl->nonatomic = 1; + dl->no_pcm = 1; + dl->dpcm_capture = 1; + dl->dpcm_playback = 1; + + *dai_link = dl; + + return 0; +} + +static int avs_create_dapm_routes(struct device *dev, int ssp_port, + struct snd_soc_dapm_route **routes, int *num_routes) +{ + struct snd_soc_dapm_route *dr; + const int num_base = ARRAY_SIZE(card_base_routes); + const int num_dr = num_base + 2; + int idx; + + dr = devm_kcalloc(dev, num_dr, sizeof(*dr), GFP_KERNEL); + if (!dr) + return -ENOMEM; + + memcpy(dr, card_base_routes, num_base * sizeof(*dr)); + + idx = num_base; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "Playback"); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Tx", ssp_port); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + idx++; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Rx", ssp_port); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "Capture"); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + *routes = dr; + *num_routes = num_dr; + + return 0; +} + +static int avs_card_set_jack(struct snd_soc_card *card, struct snd_soc_jack *jack) +{ + struct snd_soc_component *component; + + for_each_card_components(card, component) + snd_soc_component_set_jack(component, jack, NULL); + return 0; +} + +static int avs_card_remove(struct snd_soc_card *card) +{ + return avs_card_set_jack(card, NULL); +} + +static int avs_card_suspend_pre(struct snd_soc_card *card) +{ + return avs_card_set_jack(card, NULL); +} + +static int avs_card_resume_post(struct snd_soc_card *card) +{ + struct snd_soc_dai *codec_dai = snd_soc_card_get_codec_dai(card, SKL_NUVOTON_CODEC_DAI); + struct snd_soc_jack *jack = snd_soc_card_get_drvdata(card); + + if (!codec_dai) { + dev_err(card->dev, "Codec dai not found\n"); + return -EINVAL; + } + + if (codec_dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK] && + codec_dai->playback_widget->active) + snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_FLL_FS, 0, SND_SOC_CLOCK_IN); + + return avs_card_set_jack(card, jack); +} + +static int avs_nau8825_probe(struct platform_device *pdev) +{ + struct snd_soc_dapm_route *routes; + struct snd_soc_dai_link *dai_link; + struct snd_soc_acpi_mach *mach; + struct snd_soc_card *card; + struct snd_soc_jack *jack; + struct device *dev = &pdev->dev; + const char *pname; + int num_routes, ssp_port, ret; + + mach = dev_get_platdata(dev); + pname = mach->mach_params.platform; + ssp_port = __ffs(mach->mach_params.i2s_link_mask); + + ret = avs_create_dai_link(dev, pname, ssp_port, &dai_link); + if (ret) { + dev_err(dev, "Failed to create dai link: %d", ret); + return ret; + } + + ret = avs_create_dapm_routes(dev, ssp_port, &routes, &num_routes); + if (ret) { + dev_err(dev, "Failed to create dapm routes: %d", ret); + return ret; + } + + jack = devm_kzalloc(dev, sizeof(*jack), GFP_KERNEL); + card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); + if (!jack || !card) + return -ENOMEM; + + card->name = "avs_nau8825"; + card->dev = dev; + card->owner = THIS_MODULE; + card->remove = avs_card_remove; + card->suspend_pre = avs_card_suspend_pre; + card->resume_post = avs_card_resume_post; + card->dai_link = dai_link; + card->num_links = 1; + card->controls = card_controls; + card->num_controls = ARRAY_SIZE(card_controls); + card->dapm_widgets = card_widgets; + card->num_dapm_widgets = ARRAY_SIZE(card_widgets); + card->dapm_routes = routes; + card->num_dapm_routes = num_routes; + card->fully_routed = true; + snd_soc_card_set_drvdata(card, jack); + + ret = snd_soc_fixup_dai_links_platform_name(card, pname); + if (ret) + return ret; + + return devm_snd_soc_register_card(dev, card); +} + +static struct platform_driver avs_nau8825_driver = { + .probe = avs_nau8825_probe, + .driver = { + .name = "avs_nau8825", + .pm = &snd_soc_pm_ops, + }, +}; + +module_platform_driver(avs_nau8825_driver) + +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:avs_nau8825"); -- cgit v1.2.3 From 69ea14efe99b533652255b07a9736a9856f50ea5 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Wed, 11 May 2022 18:24:00 +0200 Subject: ASoC: Intel: avs: Add ssm4567 machine board MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit To support AVS-ssm4567 configuration add machine board connecting AVS platform component driver with ssm4567 codec one. Signed-off-by: Amadeusz Sławiński Signed-off-by: Cezary Rojewski Link: https://lore.kernel.org/r/20220511162403.3987658-12-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/boards/Kconfig | 11 ++ sound/soc/intel/avs/boards/Makefile | 2 + sound/soc/intel/avs/boards/ssm4567.c | 271 +++++++++++++++++++++++++++++++++++ 3 files changed, 284 insertions(+) create mode 100644 sound/soc/intel/avs/boards/ssm4567.c (limited to 'sound') diff --git a/sound/soc/intel/avs/boards/Kconfig b/sound/soc/intel/avs/boards/Kconfig index 6bf8fa1924a2..7020e7bf196e 100644 --- a/sound/soc/intel/avs/boards/Kconfig +++ b/sound/soc/intel/avs/boards/Kconfig @@ -77,4 +77,15 @@ config SND_SOC_INTEL_AVS_MACH_RT5682 Say Y or m if you have such a device. This is a recommended option. If unsure select "N". +config SND_SOC_INTEL_AVS_MACH_SSM4567 + tristate "ssm4567 I2S board" + depends on I2C + depends on MFD_INTEL_LPSS || COMPILE_TEST + select SND_SOC_SSM4567 + help + This adds support for ASoC machine driver with SSM4567 I2S audio codec. + It is meant to be used with AVS driver. + Say Y or m if you have such a device. This is a recommended option. + If unsure select "N". + endmenu diff --git a/sound/soc/intel/avs/boards/Makefile b/sound/soc/intel/avs/boards/Makefile index 9ac14b269f56..ea67fc711d9d 100644 --- a/sound/soc/intel/avs/boards/Makefile +++ b/sound/soc/intel/avs/boards/Makefile @@ -8,6 +8,7 @@ snd-soc-avs-rt274-objs := rt274.o snd-soc-avs-rt286-objs := rt286.o snd-soc-avs-rt298-objs := rt298.o snd-soc-avs-rt5682-objs := rt5682.o +snd-soc-avs-ssm4567-objs := ssm4567.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_DMIC) += snd-soc-avs-dmic.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_HDAUDIO) += snd-soc-avs-hdaudio.o @@ -17,3 +18,4 @@ obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_RT274) += snd-soc-avs-rt274.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_RT286) += snd-soc-avs-rt286.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_RT298) += snd-soc-avs-rt298.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_RT5682) += snd-soc-avs-rt5682.o +obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_SSM4567) += snd-soc-avs-ssm4567.o diff --git a/sound/soc/intel/avs/boards/ssm4567.c b/sound/soc/intel/avs/boards/ssm4567.c new file mode 100644 index 000000000000..9f84c8ab3447 --- /dev/null +++ b/sound/soc/intel/avs/boards/ssm4567.c @@ -0,0 +1,271 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// +// Authors: Cezary Rojewski +// Amadeusz Slawinski +// + +#include +#include +#include +#include +#include +#include +#include +#include "../../../codecs/nau8825.h" + +#define SKL_NUVOTON_CODEC_DAI "nau8825-hifi" +#define SKL_SSM_CODEC_DAI "ssm4567-hifi" + +static struct snd_soc_codec_conf card_codec_conf[] = { + { + .dlc = COMP_CODEC_CONF("i2c-INT343B:00"), + .name_prefix = "Left", + }, + { + .dlc = COMP_CODEC_CONF("i2c-INT343B:01"), + .name_prefix = "Right", + }, +}; + +static const struct snd_kcontrol_new card_controls[] = { + SOC_DAPM_PIN_SWITCH("Left Speaker"), + SOC_DAPM_PIN_SWITCH("Right Speaker"), +}; + +static int +platform_clock_control(struct snd_soc_dapm_widget *w, struct snd_kcontrol *control, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct snd_soc_dai *codec_dai; + int ret; + + codec_dai = snd_soc_card_get_codec_dai(card, SKL_NUVOTON_CODEC_DAI); + if (!codec_dai) { + dev_err(card->dev, "Codec dai not found\n"); + return -EINVAL; + } + + if (SND_SOC_DAPM_EVENT_ON(event)) { + ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_MCLK, 24000000, + SND_SOC_CLOCK_IN); + if (ret < 0) + dev_err(card->dev, "set sysclk err = %d\n", ret); + } else { + ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_INTERNAL, 0, SND_SOC_CLOCK_IN); + if (ret < 0) + dev_err(card->dev, "set sysclk err = %d\n", ret); + } + + return ret; +} + +static const struct snd_soc_dapm_widget card_widgets[] = { + SND_SOC_DAPM_SPK("Left Speaker", NULL), + SND_SOC_DAPM_SPK("Right Speaker", NULL), + SND_SOC_DAPM_SPK("DP1", NULL), + SND_SOC_DAPM_SPK("DP2", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, platform_clock_control, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route card_base_routes[] = { + {"Left Speaker", NULL, "Left OUT"}, + {"Right Speaker", NULL, "Right OUT"}, +}; + +static int avs_ssm4567_codec_init(struct snd_soc_pcm_runtime *runtime) +{ + int ret; + + /* Slot 1 for left */ + ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_codec(runtime, 0), 0x01, 0x01, 2, 48); + if (ret < 0) + return ret; + + /* Slot 2 for right */ + ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_codec(runtime, 1), 0x02, 0x02, 2, 48); + if (ret < 0) + return ret; + + return 0; +} + +static int +avs_ssm4567_be_fixup(struct snd_soc_pcm_runtime *runrime, struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate, *channels; + struct snd_mask *fmt; + + rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + + /* The ADSP will covert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP0 to 24 bit */ + snd_mask_none(fmt); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); + return 0; +} + +static int avs_create_dai_link(struct device *dev, const char *platform_name, int ssp_port, + struct snd_soc_dai_link **dai_link) +{ + struct snd_soc_dai_link_component *platform; + struct snd_soc_dai_link *dl; + + dl = devm_kzalloc(dev, sizeof(*dl), GFP_KERNEL); + platform = devm_kzalloc(dev, sizeof(*platform), GFP_KERNEL); + if (!dl || !platform) + return -ENOMEM; + + platform->name = platform_name; + + dl->name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-Codec", ssp_port); + dl->cpus = devm_kzalloc(dev, sizeof(*dl->cpus), GFP_KERNEL); + dl->codecs = devm_kzalloc(dev, sizeof(*dl->codecs) * 2, GFP_KERNEL); + if (!dl->name || !dl->cpus || !dl->codecs) + return -ENOMEM; + + dl->cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", ssp_port); + dl->codecs[0].name = devm_kasprintf(dev, GFP_KERNEL, "i2c-INT343B:00"); + dl->codecs[0].dai_name = devm_kasprintf(dev, GFP_KERNEL, "ssm4567-hifi"); + dl->codecs[1].name = devm_kasprintf(dev, GFP_KERNEL, "i2c-INT343B:01"); + dl->codecs[1].dai_name = devm_kasprintf(dev, GFP_KERNEL, "ssm4567-hifi"); + if (!dl->cpus->dai_name || !dl->codecs[0].name || !dl->codecs[0].dai_name || + !dl->codecs[1].name || !dl->codecs[1].dai_name) + return -ENOMEM; + + dl->num_cpus = 1; + dl->num_codecs = 2; + dl->platforms = platform; + dl->num_platforms = 1; + dl->id = 0; + dl->dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_CBS_CFS; + dl->init = avs_ssm4567_codec_init; + dl->be_hw_params_fixup = avs_ssm4567_be_fixup; + dl->nonatomic = 1; + dl->no_pcm = 1; + dl->dpcm_capture = 1; + dl->dpcm_playback = 1; + dl->ignore_pmdown_time = 1; + + *dai_link = dl; + + return 0; +} + +static int avs_create_dapm_routes(struct device *dev, int ssp_port, + struct snd_soc_dapm_route **routes, int *num_routes) +{ + struct snd_soc_dapm_route *dr; + const int num_base = ARRAY_SIZE(card_base_routes); + const int num_dr = num_base + 4; + int idx; + + dr = devm_kcalloc(dev, num_dr, sizeof(*dr), GFP_KERNEL); + if (!dr) + return -ENOMEM; + + memcpy(dr, card_base_routes, num_base * sizeof(*dr)); + + idx = num_base; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "Left Playback"); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Tx", ssp_port); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + idx++; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "Right Playback"); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Tx", ssp_port); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + idx++; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Rx", ssp_port); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "Left Capture Sense"); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + idx++; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Rx", ssp_port); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "Right Capture Sense"); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + *routes = dr; + *num_routes = num_dr; + + return 0; +} + +static int avs_ssm4567_probe(struct platform_device *pdev) +{ + struct snd_soc_dapm_route *routes; + struct snd_soc_dai_link *dai_link; + struct snd_soc_acpi_mach *mach; + struct snd_soc_card *card; + struct device *dev = &pdev->dev; + const char *pname; + int num_routes, ssp_port, ret; + + mach = dev_get_platdata(dev); + pname = mach->mach_params.platform; + ssp_port = __ffs(mach->mach_params.i2s_link_mask); + + ret = avs_create_dai_link(dev, pname, ssp_port, &dai_link); + if (ret) { + dev_err(dev, "Failed to create dai link: %d", ret); + return ret; + } + + ret = avs_create_dapm_routes(dev, ssp_port, &routes, &num_routes); + if (ret) { + dev_err(dev, "Failed to create dapm routes: %d", ret); + return ret; + } + + card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); + if (!card) + return -ENOMEM; + + card->name = "avs_ssm4567-adi"; + card->dev = dev; + card->owner = THIS_MODULE; + card->dai_link = dai_link; + card->num_links = 1; + card->codec_conf = card_codec_conf; + card->num_configs = ARRAY_SIZE(card_codec_conf); + card->controls = card_controls; + card->num_controls = ARRAY_SIZE(card_controls); + card->dapm_widgets = card_widgets; + card->num_dapm_widgets = ARRAY_SIZE(card_widgets); + card->dapm_routes = routes; + card->num_dapm_routes = num_routes; + card->fully_routed = true; + card->disable_route_checks = true; + + ret = snd_soc_fixup_dai_links_platform_name(card, pname); + if (ret) + return ret; + + return devm_snd_soc_register_card(dev, card); +} + +static struct platform_driver avs_ssm4567_driver = { + .probe = avs_ssm4567_probe, + .driver = { + .name = "avs_ssm4567", + .pm = &snd_soc_pm_ops, + }, +}; + +module_platform_driver(avs_ssm4567_driver) + +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:avs_ssm4567"); -- cgit v1.2.3 From 282c8f8de72f95325225d94caef61f3cc96401da Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Wed, 11 May 2022 18:24:01 +0200 Subject: ASoC: Intel: avs: Add max98357a machine board MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit To support AVS-max98357a configuration add machine board connecting AVS platform component driver with max98357a codec one. Signed-off-by: Amadeusz Sławiński Signed-off-by: Cezary Rojewski Link: https://lore.kernel.org/r/20220511162403.3987658-13-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/boards/Kconfig | 10 +++ sound/soc/intel/avs/boards/Makefile | 2 + sound/soc/intel/avs/boards/max98357a.c | 154 +++++++++++++++++++++++++++++++++ 3 files changed, 166 insertions(+) create mode 100644 sound/soc/intel/avs/boards/max98357a.c (limited to 'sound') diff --git a/sound/soc/intel/avs/boards/Kconfig b/sound/soc/intel/avs/boards/Kconfig index 7020e7bf196e..28e6691270d9 100644 --- a/sound/soc/intel/avs/boards/Kconfig +++ b/sound/soc/intel/avs/boards/Kconfig @@ -26,6 +26,16 @@ config SND_SOC_INTEL_AVS_MACH_I2S_TEST This adds support for I2S test-board which can be used to verify transfer over I2S interface with SSP loopback scenarios. +config SND_SOC_INTEL_AVS_MACH_MAX98357A + tristate "max98357A I2S board" + depends on I2C + depends on MFD_INTEL_LPSS || COMPILE_TEST + select SND_SOC_MAX98357A + help + This adds support for AVS with MAX98357A I2S codec configuration. + Say Y or m if you have such a device. This is a recommended option. + If unsure select "N". + config SND_SOC_INTEL_AVS_MACH_NAU8825 tristate "nau8825 I2S board" depends on I2C diff --git a/sound/soc/intel/avs/boards/Makefile b/sound/soc/intel/avs/boards/Makefile index ea67fc711d9d..f7ac1151a8f7 100644 --- a/sound/soc/intel/avs/boards/Makefile +++ b/sound/soc/intel/avs/boards/Makefile @@ -3,6 +3,7 @@ snd-soc-avs-dmic-objs := dmic.o snd-soc-avs-hdaudio-objs := hdaudio.o snd-soc-avs-i2s-test-objs := i2s_test.o +snd-soc-avs-max98357a-objs := max98357a.o snd-soc-avs-nau8825-objs := nau8825.o snd-soc-avs-rt274-objs := rt274.o snd-soc-avs-rt286-objs := rt286.o @@ -13,6 +14,7 @@ snd-soc-avs-ssm4567-objs := ssm4567.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_DMIC) += snd-soc-avs-dmic.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_HDAUDIO) += snd-soc-avs-hdaudio.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_i2s_TEST) += snd-soc-avs-i2s-test.o +obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_MAX98357A) += snd-soc-avs-max98357a.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_NAU8825) += snd-soc-avs-nau8825.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_RT274) += snd-soc-avs-rt274.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_RT286) += snd-soc-avs-rt286.o diff --git a/sound/soc/intel/avs/boards/max98357a.c b/sound/soc/intel/avs/boards/max98357a.c new file mode 100644 index 000000000000..921f42caf7e0 --- /dev/null +++ b/sound/soc/intel/avs/boards/max98357a.c @@ -0,0 +1,154 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// +// Authors: Cezary Rojewski +// Amadeusz Slawinski +// + +#include +#include +#include +#include +#include + +static const struct snd_kcontrol_new card_controls[] = { + SOC_DAPM_PIN_SWITCH("Spk"), +}; + +static const struct snd_soc_dapm_widget card_widgets[] = { + SND_SOC_DAPM_SPK("Spk", NULL), +}; + +static const struct snd_soc_dapm_route card_base_routes[] = { + { "Spk", NULL, "Speaker" }, +}; + +static int avs_create_dai_link(struct device *dev, const char *platform_name, int ssp_port, + struct snd_soc_dai_link **dai_link) +{ + struct snd_soc_dai_link_component *platform; + struct snd_soc_dai_link *dl; + + dl = devm_kzalloc(dev, sizeof(*dl), GFP_KERNEL); + platform = devm_kzalloc(dev, sizeof(*platform), GFP_KERNEL); + if (!dl || !platform) + return -ENOMEM; + + platform->name = platform_name; + + dl->name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-Codec", ssp_port); + dl->cpus = devm_kzalloc(dev, sizeof(*dl->cpus), GFP_KERNEL); + dl->codecs = devm_kzalloc(dev, sizeof(*dl->codecs), GFP_KERNEL); + if (!dl->name || !dl->cpus || !dl->codecs) + return -ENOMEM; + + dl->cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", ssp_port); + dl->codecs->name = devm_kasprintf(dev, GFP_KERNEL, "MX98357A:00"); + dl->codecs->dai_name = devm_kasprintf(dev, GFP_KERNEL, "HiFi"); + if (!dl->cpus->dai_name || !dl->codecs->name || !dl->codecs->dai_name) + return -ENOMEM; + + dl->num_cpus = 1; + dl->num_codecs = 1; + dl->platforms = platform; + dl->num_platforms = 1; + dl->id = 0; + dl->dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS; + dl->nonatomic = 1; + dl->no_pcm = 1; + dl->dpcm_playback = 1; + + *dai_link = dl; + + return 0; +} + +static int avs_create_dapm_routes(struct device *dev, int ssp_port, + struct snd_soc_dapm_route **routes, int *num_routes) +{ + struct snd_soc_dapm_route *dr; + const int num_base = ARRAY_SIZE(card_base_routes); + const int num_dr = num_base + 1; + int idx; + + dr = devm_kcalloc(dev, num_dr, sizeof(*dr), GFP_KERNEL); + if (!dr) + return -ENOMEM; + + memcpy(dr, card_base_routes, num_base * sizeof(*dr)); + + idx = num_base; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "HiFi Playback"); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Tx", ssp_port); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + *routes = dr; + *num_routes = num_dr; + + return 0; +} + +static int avs_max98357a_probe(struct platform_device *pdev) +{ + struct snd_soc_dapm_route *routes; + struct snd_soc_dai_link *dai_link; + struct snd_soc_acpi_mach *mach; + struct snd_soc_card *card; + struct device *dev = &pdev->dev; + const char *pname; + int num_routes, ssp_port, ret; + + mach = dev_get_platdata(dev); + pname = mach->mach_params.platform; + ssp_port = __ffs(mach->mach_params.i2s_link_mask); + + ret = avs_create_dai_link(dev, pname, ssp_port, &dai_link); + if (ret) { + dev_err(dev, "Failed to create dai link: %d", ret); + return ret; + } + + ret = avs_create_dapm_routes(dev, ssp_port, &routes, &num_routes); + if (ret) { + dev_err(dev, "Failed to create dapm routes: %d", ret); + return ret; + } + + card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); + if (!card) + return -ENOMEM; + + card->name = "avs_max98357a"; + card->dev = dev; + card->owner = THIS_MODULE; + card->dai_link = dai_link; + card->num_links = 1; + card->controls = card_controls; + card->num_controls = ARRAY_SIZE(card_controls); + card->dapm_widgets = card_widgets; + card->num_dapm_widgets = ARRAY_SIZE(card_widgets); + card->dapm_routes = routes; + card->num_dapm_routes = num_routes; + card->fully_routed = true; + + ret = snd_soc_fixup_dai_links_platform_name(card, pname); + if (ret) + return ret; + + return devm_snd_soc_register_card(dev, card); +} + +static struct platform_driver avs_max98357a_driver = { + .probe = avs_max98357a_probe, + .driver = { + .name = "avs_max98357a", + .pm = &snd_soc_pm_ops, + }, +}; + +module_platform_driver(avs_max98357a_driver) + +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:avs_max98357a"); -- cgit v1.2.3 From 223a0a945821b96f4ccd9940ee975499706e1794 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Wed, 11 May 2022 18:24:02 +0200 Subject: ASoC: Intel: avs: Add max98373 machine board MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit To support AVS-max98373 configuration add machine board connecting AVS platform component driver with max98373 codec one. Signed-off-by: Amadeusz Sławiński Signed-off-by: Cezary Rojewski Link: https://lore.kernel.org/r/20220511162403.3987658-14-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/boards/Kconfig | 10 ++ sound/soc/intel/avs/boards/Makefile | 2 + sound/soc/intel/avs/boards/max98373.c | 239 ++++++++++++++++++++++++++++++++++ 3 files changed, 251 insertions(+) create mode 100644 sound/soc/intel/avs/boards/max98373.c (limited to 'sound') diff --git a/sound/soc/intel/avs/boards/Kconfig b/sound/soc/intel/avs/boards/Kconfig index 28e6691270d9..d3be6dc1fc10 100644 --- a/sound/soc/intel/avs/boards/Kconfig +++ b/sound/soc/intel/avs/boards/Kconfig @@ -36,6 +36,16 @@ config SND_SOC_INTEL_AVS_MACH_MAX98357A Say Y or m if you have such a device. This is a recommended option. If unsure select "N". +config SND_SOC_INTEL_AVS_MACH_MAX98373 + tristate "max98373 I2S board" + depends on I2C + depends on MFD_INTEL_LPSS || COMPILE_TEST + select SND_SOC_MAX98373 + help + This adds support for AVS with MAX98373 I2S codec configuration. + Say Y or m if you have such a device. This is a recommended option. + If unsure select "N". + config SND_SOC_INTEL_AVS_MACH_NAU8825 tristate "nau8825 I2S board" depends on I2C diff --git a/sound/soc/intel/avs/boards/Makefile b/sound/soc/intel/avs/boards/Makefile index f7ac1151a8f7..0bce31e192ce 100644 --- a/sound/soc/intel/avs/boards/Makefile +++ b/sound/soc/intel/avs/boards/Makefile @@ -4,6 +4,7 @@ snd-soc-avs-dmic-objs := dmic.o snd-soc-avs-hdaudio-objs := hdaudio.o snd-soc-avs-i2s-test-objs := i2s_test.o snd-soc-avs-max98357a-objs := max98357a.o +snd-soc-avs-max98373-objs := max98373.o snd-soc-avs-nau8825-objs := nau8825.o snd-soc-avs-rt274-objs := rt274.o snd-soc-avs-rt286-objs := rt286.o @@ -15,6 +16,7 @@ obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_DMIC) += snd-soc-avs-dmic.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_HDAUDIO) += snd-soc-avs-hdaudio.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_i2s_TEST) += snd-soc-avs-i2s-test.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_MAX98357A) += snd-soc-avs-max98357a.o +obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_MAX98373) += snd-soc-avs-max98373.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_NAU8825) += snd-soc-avs-nau8825.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_RT274) += snd-soc-avs-rt274.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_RT286) += snd-soc-avs-rt286.o diff --git a/sound/soc/intel/avs/boards/max98373.c b/sound/soc/intel/avs/boards/max98373.c new file mode 100644 index 000000000000..0fa8f5606385 --- /dev/null +++ b/sound/soc/intel/avs/boards/max98373.c @@ -0,0 +1,239 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// Copyright(c) 2022 Intel Corporation. All rights reserved. +// +// Authors: Cezary Rojewski +// Amadeusz Slawinski +// + +#include +#include +#include +#include +#include +#include + +#define MAX98373_DEV0_NAME "i2c-MX98373:00" +#define MAX98373_DEV1_NAME "i2c-MX98373:01" +#define MAX98373_CODEC_NAME "max98373-aif1" + +static struct snd_soc_codec_conf card_codec_conf[] = { + { + .dlc = COMP_CODEC_CONF(MAX98373_DEV0_NAME), + .name_prefix = "Right", + }, + { + .dlc = COMP_CODEC_CONF(MAX98373_DEV1_NAME), + .name_prefix = "Left", + }, +}; + +static const struct snd_kcontrol_new card_controls[] = { + SOC_DAPM_PIN_SWITCH("Left Spk"), + SOC_DAPM_PIN_SWITCH("Right Spk"), +}; + +static const struct snd_soc_dapm_widget card_widgets[] = { + SND_SOC_DAPM_SPK("Left Spk", NULL), + SND_SOC_DAPM_SPK("Right Spk", NULL), +}; + +static const struct snd_soc_dapm_route card_base_routes[] = { + { "Left Spk", NULL, "Left BE_OUT" }, + { "Right Spk", NULL, "Right BE_OUT" }, +}; + +static int +avs_max98373_be_fixup(struct snd_soc_pcm_runtime *runrime, struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate, *channels; + struct snd_mask *fmt; + + rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + + /* The ADSP will covert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP0 to 16 bit */ + snd_mask_none(fmt); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE); + return 0; +} + +static int avs_max98373_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *runtime = asoc_substream_to_rtd(substream); + struct snd_soc_dai *codec_dai; + int ret, i; + + for_each_rtd_codec_dais(runtime, i, codec_dai) { + if (!strcmp(codec_dai->component->name, MAX98373_DEV0_NAME)) { + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x30, 3, 8, 16); + if (ret < 0) { + dev_err(runtime->dev, "DEV0 TDM slot err:%d\n", ret); + return ret; + } + } + if (!strcmp(codec_dai->component->name, MAX98373_DEV1_NAME)) { + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xC0, 3, 8, 16); + if (ret < 0) { + dev_err(runtime->dev, "DEV1 TDM slot err:%d\n", ret); + return ret; + } + } + } + + return 0; +} + +static const struct snd_soc_ops avs_max98373_ops = { + .hw_params = avs_max98373_hw_params, +}; + +static int avs_create_dai_link(struct device *dev, const char *platform_name, int ssp_port, + struct snd_soc_dai_link **dai_link) +{ + struct snd_soc_dai_link_component *platform; + struct snd_soc_dai_link *dl; + + dl = devm_kzalloc(dev, sizeof(*dl), GFP_KERNEL); + platform = devm_kzalloc(dev, sizeof(*platform), GFP_KERNEL); + if (!dl || !platform) + return -ENOMEM; + + platform->name = platform_name; + + dl->name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-Codec", ssp_port); + dl->cpus = devm_kzalloc(dev, sizeof(*dl->cpus), GFP_KERNEL); + dl->codecs = devm_kzalloc(dev, sizeof(*dl->codecs) * 2, GFP_KERNEL); + if (!dl->name || !dl->cpus || !dl->codecs) + return -ENOMEM; + + dl->cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", ssp_port); + dl->codecs[0].name = devm_kasprintf(dev, GFP_KERNEL, MAX98373_DEV0_NAME); + dl->codecs[0].dai_name = devm_kasprintf(dev, GFP_KERNEL, MAX98373_CODEC_NAME); + dl->codecs[1].name = devm_kasprintf(dev, GFP_KERNEL, MAX98373_DEV1_NAME); + dl->codecs[1].dai_name = devm_kasprintf(dev, GFP_KERNEL, MAX98373_CODEC_NAME); + if (!dl->cpus->dai_name || !dl->codecs[0].name || !dl->codecs[0].dai_name || + !dl->codecs[1].name || !dl->codecs[1].dai_name) + return -ENOMEM; + + dl->num_cpus = 1; + dl->num_codecs = 2; + dl->platforms = platform; + dl->num_platforms = 1; + dl->id = 0; + dl->dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBC_CFC; + dl->be_hw_params_fixup = avs_max98373_be_fixup; + dl->nonatomic = 1; + dl->no_pcm = 1; + dl->dpcm_capture = 1; + dl->dpcm_playback = 1; + dl->ignore_pmdown_time = 1; + dl->ops = &avs_max98373_ops; + + *dai_link = dl; + + return 0; +} + +static int avs_create_dapm_routes(struct device *dev, int ssp_port, + struct snd_soc_dapm_route **routes, int *num_routes) +{ + struct snd_soc_dapm_route *dr; + const int num_base = ARRAY_SIZE(card_base_routes); + const int num_dr = num_base + 2; + int idx; + + dr = devm_kcalloc(dev, num_dr, sizeof(*dr), GFP_KERNEL); + if (!dr) + return -ENOMEM; + + memcpy(dr, card_base_routes, num_base * sizeof(*dr)); + + idx = num_base; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "Left HiFi Playback"); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Tx", ssp_port); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + idx++; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "Right HiFi Playback"); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Tx", ssp_port); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + *routes = dr; + *num_routes = num_dr; + + return 0; +} + +static int avs_max98373_probe(struct platform_device *pdev) +{ + struct snd_soc_dapm_route *routes; + struct snd_soc_dai_link *dai_link; + struct snd_soc_acpi_mach *mach; + struct snd_soc_card *card; + struct device *dev = &pdev->dev; + const char *pname; + int num_routes, ssp_port, ret; + + mach = dev_get_platdata(dev); + pname = mach->mach_params.platform; + ssp_port = __ffs(mach->mach_params.i2s_link_mask); + + ret = avs_create_dai_link(dev, pname, ssp_port, &dai_link); + if (ret) { + dev_err(dev, "Failed to create dai link: %d", ret); + return ret; + } + + ret = avs_create_dapm_routes(dev, ssp_port, &routes, &num_routes); + if (ret) { + dev_err(dev, "Failed to create dapm routes: %d", ret); + return ret; + } + + card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); + if (!card) + return -ENOMEM; + + card->name = "avs_max98373"; + card->dev = dev; + card->owner = THIS_MODULE; + card->dai_link = dai_link; + card->num_links = 1; + card->codec_conf = card_codec_conf; + card->num_configs = ARRAY_SIZE(card_codec_conf); + card->controls = card_controls; + card->num_controls = ARRAY_SIZE(card_controls); + card->dapm_widgets = card_widgets; + card->num_dapm_widgets = ARRAY_SIZE(card_widgets); + card->dapm_routes = routes; + card->num_dapm_routes = num_routes; + card->fully_routed = true; + + ret = snd_soc_fixup_dai_links_platform_name(card, pname); + if (ret) + return ret; + + return devm_snd_soc_register_card(dev, card); +} + +static struct platform_driver avs_max98373_driver = { + .probe = avs_max98373_probe, + .driver = { + .name = "avs_max98373", + .pm = &snd_soc_pm_ops, + }, +}; + +module_platform_driver(avs_max98373_driver) + +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:avs_max98373"); -- cgit v1.2.3 From 6b5b0d6f36dd45e22f1710e8bcd97f28b4ba41f5 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Wed, 11 May 2022 18:24:03 +0200 Subject: ASoC: Intel: avs: Add da7219 machine board MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit To support AVS-da7219 configuration add machine board connecting AVS platform component driver with da7219 codec one. Signed-off-by: Amadeusz Sławiński Signed-off-by: Cezary Rojewski Link: https://lore.kernel.org/r/20220511162403.3987658-15-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/boards/Kconfig | 10 ++ sound/soc/intel/avs/boards/Makefile | 2 + sound/soc/intel/avs/boards/da7219.c | 282 ++++++++++++++++++++++++++++++++++++ 3 files changed, 294 insertions(+) create mode 100644 sound/soc/intel/avs/boards/da7219.c (limited to 'sound') diff --git a/sound/soc/intel/avs/boards/Kconfig b/sound/soc/intel/avs/boards/Kconfig index d3be6dc1fc10..4d68e3ef992b 100644 --- a/sound/soc/intel/avs/boards/Kconfig +++ b/sound/soc/intel/avs/boards/Kconfig @@ -4,6 +4,16 @@ menu "Intel AVS Machine drivers" comment "Available DSP configurations" +config SND_SOC_INTEL_AVS_MACH_DA7219 + tristate "da7219 I2S board" + depends on I2C + depends on MFD_INTEL_LPSS || COMPILE_TEST + select SND_SOC_DA7219 + help + This adds support for AVS with DA7219 I2S codec configuration. + Say Y or m if you have such a device. This is a recommended option. + If unsure select "N". + config SND_SOC_INTEL_AVS_MACH_DMIC tristate "DMIC generic board" select SND_SOC_DMIC diff --git a/sound/soc/intel/avs/boards/Makefile b/sound/soc/intel/avs/boards/Makefile index 0bce31e192ce..25e8c4bb07db 100644 --- a/sound/soc/intel/avs/boards/Makefile +++ b/sound/soc/intel/avs/boards/Makefile @@ -1,5 +1,6 @@ # SPDX-License-Identifier: GPL-2.0-only +snd-soc-avs-da7219-objs := da7219.o snd-soc-avs-dmic-objs := dmic.o snd-soc-avs-hdaudio-objs := hdaudio.o snd-soc-avs-i2s-test-objs := i2s_test.o @@ -12,6 +13,7 @@ snd-soc-avs-rt298-objs := rt298.o snd-soc-avs-rt5682-objs := rt5682.o snd-soc-avs-ssm4567-objs := ssm4567.o +obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_DA7219) += snd-soc-avs-da7219.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_DMIC) += snd-soc-avs-dmic.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_HDAUDIO) += snd-soc-avs-hdaudio.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_i2s_TEST) += snd-soc-avs-i2s-test.o diff --git a/sound/soc/intel/avs/boards/da7219.c b/sound/soc/intel/avs/boards/da7219.c new file mode 100644 index 000000000000..02ae542ad779 --- /dev/null +++ b/sound/soc/intel/avs/boards/da7219.c @@ -0,0 +1,282 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// +// Author: Cezary Rojewski +// + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "../../../codecs/da7219.h" +#include "../../../codecs/da7219-aad.h" + +#define DA7219_DAI_NAME "da7219-hifi" + +static const struct snd_kcontrol_new card_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), +}; + +static int platform_clock_control(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct snd_soc_dai *codec_dai; + int ret = 0; + + codec_dai = snd_soc_card_get_codec_dai(card, DA7219_DAI_NAME); + if (!codec_dai) { + dev_err(card->dev, "Codec dai not found. Unable to set/unset codec pll\n"); + return -EIO; + } + + if (SND_SOC_DAPM_EVENT_OFF(event)) { + ret = snd_soc_dai_set_pll(codec_dai, 0, DA7219_SYSCLK_MCLK, 0, 0); + if (ret) + dev_err(card->dev, "failed to stop PLL: %d\n", ret); + } else if (SND_SOC_DAPM_EVENT_ON(event)) { + ret = snd_soc_dai_set_pll(codec_dai, 0, DA7219_SYSCLK_PLL_SRM, + 0, DA7219_PLL_FREQ_OUT_98304); + if (ret) + dev_err(card->dev, "failed to start PLL: %d\n", ret); + } + + return ret; +} + +static const struct snd_soc_dapm_widget card_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, platform_clock_control, + SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_PRE_PMU), +}; + +static const struct snd_soc_dapm_route card_base_routes[] = { + /* HP jack connectors - unknown if we have jack detection */ + {"Headphone Jack", NULL, "HPL"}, + {"Headphone Jack", NULL, "HPR"}, + + {"MIC", NULL, "Headset Mic"}, + + { "Headphone Jack", NULL, "Platform Clock" }, + { "Headset Mic", NULL, "Platform Clock" }, +}; + +static int avs_da7219_codec_init(struct snd_soc_pcm_runtime *runtime) +{ + struct snd_soc_component *component = asoc_rtd_to_codec(runtime, 0)->component; + struct snd_soc_card *card = runtime->card; + struct snd_soc_jack *jack; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(runtime, 0); + int clk_freq; + int ret; + + jack = snd_soc_card_get_drvdata(card); + clk_freq = 19200000; + + ret = snd_soc_dai_set_sysclk(codec_dai, DA7219_CLKSRC_MCLK, clk_freq, SND_SOC_CLOCK_IN); + if (ret) { + dev_err(card->dev, "can't set codec sysclk configuration\n"); + return ret; + } + + /* + * Headset buttons map to the google Reference headset. + * These can be configured by userspace. + */ + ret = snd_soc_card_jack_new(card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | + SND_JACK_BTN_1 | SND_JACK_BTN_2 | + SND_JACK_BTN_3 | SND_JACK_LINEOUT, jack); + if (ret) { + dev_err(card->dev, "Headset Jack creation failed: %d\n", ret); + return ret; + } + + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); + + da7219_aad_jack_det(component, jack); + + return 0; +} + +static int avs_create_dai_link(struct device *dev, const char *platform_name, int ssp_port, + struct snd_soc_dai_link **dai_link) +{ + struct snd_soc_dai_link_component *platform; + struct snd_soc_dai_link *dl; + + dl = devm_kzalloc(dev, sizeof(*dl), GFP_KERNEL); + platform = devm_kzalloc(dev, sizeof(*platform), GFP_KERNEL); + if (!dl || !platform) + return -ENOMEM; + + platform->name = platform_name; + + dl->name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-Codec", ssp_port); + dl->cpus = devm_kzalloc(dev, sizeof(*dl->cpus), GFP_KERNEL); + dl->codecs = devm_kzalloc(dev, sizeof(*dl->codecs), GFP_KERNEL); + if (!dl->name || !dl->cpus || !dl->codecs) + return -ENOMEM; + + dl->cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", ssp_port); + dl->codecs->name = devm_kasprintf(dev, GFP_KERNEL, "i2c-DLGS7219:00"); + dl->codecs->dai_name = devm_kasprintf(dev, GFP_KERNEL, DA7219_DAI_NAME); + if (!dl->cpus->dai_name || !dl->codecs->name || !dl->codecs->dai_name) + return -ENOMEM; + + dl->num_cpus = 1; + dl->num_codecs = 1; + dl->platforms = platform; + dl->num_platforms = 1; + dl->id = 0; + dl->dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS; + dl->init = avs_da7219_codec_init; + dl->nonatomic = 1; + dl->no_pcm = 1; + dl->dpcm_capture = 1; + dl->dpcm_playback = 1; + + *dai_link = dl; + + return 0; +} + +static int avs_create_dapm_routes(struct device *dev, int ssp_port, + struct snd_soc_dapm_route **routes, int *num_routes) +{ + struct snd_soc_dapm_route *dr; + const int num_base = ARRAY_SIZE(card_base_routes); + const int num_dr = num_base + 2; + int idx; + + dr = devm_kcalloc(dev, num_dr, sizeof(*dr), GFP_KERNEL); + if (!dr) + return -ENOMEM; + + memcpy(dr, card_base_routes, num_base * sizeof(*dr)); + + idx = num_base; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "Playback"); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Tx", ssp_port); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + idx++; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Rx", ssp_port); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "Capture"); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + *routes = dr; + *num_routes = num_dr; + + return 0; +} + +static int avs_card_set_jack(struct snd_soc_card *card, struct snd_soc_jack *jack) +{ + struct snd_soc_component *component; + + for_each_card_components(card, component) + snd_soc_component_set_jack(component, jack, NULL); + return 0; +} + +static int avs_card_remove(struct snd_soc_card *card) +{ + return avs_card_set_jack(card, NULL); +} + +static int avs_card_suspend_pre(struct snd_soc_card *card) +{ + return avs_card_set_jack(card, NULL); +} + +static int avs_card_resume_post(struct snd_soc_card *card) +{ + struct snd_soc_jack *jack = snd_soc_card_get_drvdata(card); + + return avs_card_set_jack(card, jack); +} + +static int avs_da7219_probe(struct platform_device *pdev) +{ + struct snd_soc_dapm_route *routes; + struct snd_soc_dai_link *dai_link; + struct snd_soc_acpi_mach *mach; + struct snd_soc_card *card; + struct snd_soc_jack *jack; + struct device *dev = &pdev->dev; + const char *pname; + int num_routes, ssp_port, ret; + + mach = dev_get_platdata(dev); + pname = mach->mach_params.platform; + ssp_port = __ffs(mach->mach_params.i2s_link_mask); + + ret = avs_create_dai_link(dev, pname, ssp_port, &dai_link); + if (ret) { + dev_err(dev, "Failed to create dai link: %d", ret); + return ret; + } + + ret = avs_create_dapm_routes(dev, ssp_port, &routes, &num_routes); + if (ret) { + dev_err(dev, "Failed to create dapm routes: %d", ret); + return ret; + } + + jack = devm_kzalloc(dev, sizeof(*jack), GFP_KERNEL); + card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); + if (!jack || !card) + return -ENOMEM; + + card->name = "avs_da7219"; + card->dev = dev; + card->owner = THIS_MODULE; + card->remove = avs_card_remove; + card->suspend_pre = avs_card_suspend_pre; + card->resume_post = avs_card_resume_post; + card->dai_link = dai_link; + card->num_links = 1; + card->controls = card_controls; + card->num_controls = ARRAY_SIZE(card_controls); + card->dapm_widgets = card_widgets; + card->num_dapm_widgets = ARRAY_SIZE(card_widgets); + card->dapm_routes = routes; + card->num_dapm_routes = num_routes; + card->fully_routed = true; + snd_soc_card_set_drvdata(card, jack); + + ret = snd_soc_fixup_dai_links_platform_name(card, pname); + if (ret) + return ret; + + return devm_snd_soc_register_card(dev, card); +} + +static struct platform_driver avs_da7219_driver = { + .probe = avs_da7219_probe, + .driver = { + .name = "avs_da7219", + .pm = &snd_soc_pm_ops, + }, +}; + +module_platform_driver(avs_da7219_driver); + +MODULE_AUTHOR("Cezary Rojewski "); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:avs_da7219"); -- cgit v1.2.3 From 905f3a04e184854555fc248ca4e692fdbf2f2547 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 May 2022 16:42:23 +0100 Subject: ASoC: core: Add set_fmt_new callback that directly specifies provider The original set_fmt callback always passes clock provider/consumer with respect to the CODEC. This made sense when the framework was directly broken down into platforms and CODECs. Now everything is componentised it simplifies things if each side of the link is just told if it is provider or consumer of the clocks. To start this migration add a new callback that can be used to receive a direct specification of clocking. As there are more CODEC drivers than platform drivers, we make the new flags identical to the old CODEC flags meaning CODEC drivers will not require an update. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220519154318.2153729-2-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 3 ++- sound/soc/soc-dai.c | 5 ++++- 2 files changed, 6 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 9574f86dd4de..90f4265bea50 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1235,7 +1235,8 @@ int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd, for_each_rtd_cpu_dais(rtd, i, cpu_dai) { unsigned int fmt = dai_fmt; - if (snd_soc_component_is_codec(cpu_dai->component)) + if (cpu_dai->driver->ops->set_fmt_new || + snd_soc_component_is_codec(cpu_dai->component)) fmt = inv_dai_fmt; ret = snd_soc_dai_set_fmt(cpu_dai, fmt); diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 6078afe335f8..996712f4d9bf 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -209,7 +209,10 @@ int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) int ret = -ENOTSUPP; if (dai->driver->ops && - dai->driver->ops->set_fmt) + dai->driver->ops->set_fmt_new) + ret = dai->driver->ops->set_fmt_new(dai, fmt); + else if (dai->driver->ops && + dai->driver->ops->set_fmt) ret = dai->driver->ops->set_fmt(dai, fmt); return soc_dai_ret(dai, ret); -- cgit v1.2.3 From ab890e0f83a65624d20b0ca4a7cb6306b8511558 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 May 2022 16:42:24 +0100 Subject: ASoC: amd: vangogh: Update to use set_fmt_new callback As part of updating the core to directly tell drivers if they are clock provider or consumer update this CPU side driver to use the new direct callback. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220519154318.2153729-3-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/amd/vangogh/acp5x-i2s.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/amd/vangogh/acp5x-i2s.c b/sound/soc/amd/vangogh/acp5x-i2s.c index 59a98f89a669..40fbd0bc77fd 100644 --- a/sound/soc/amd/vangogh/acp5x-i2s.c +++ b/sound/soc/amd/vangogh/acp5x-i2s.c @@ -37,10 +37,10 @@ static int acp5x_i2s_set_fmt(struct snd_soc_dai *cpu_dai, } mode = fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK; switch (mode) { - case SND_SOC_DAIFMT_CBC_CFC: + case SND_SOC_DAIFMT_BP_FP: adata->master_mode = I2S_MASTER_MODE_ENABLE; break; - case SND_SOC_DAIFMT_CBP_CFP: + case SND_SOC_DAIFMT_BC_FC: adata->master_mode = I2S_MASTER_MODE_DISABLE; break; } @@ -339,7 +339,7 @@ static int acp5x_i2s_trigger(struct snd_pcm_substream *substream, static const struct snd_soc_dai_ops acp5x_i2s_dai_ops = { .hw_params = acp5x_i2s_hwparams, .trigger = acp5x_i2s_trigger, - .set_fmt = acp5x_i2s_set_fmt, + .set_fmt_new = acp5x_i2s_set_fmt, .set_tdm_slot = acp5x_i2s_set_tdm_slot, }; -- cgit v1.2.3 From 0fd054a577180cd807992e32c7cd394e54c85903 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 May 2022 16:42:25 +0100 Subject: ASoC: atmel: Update to use set_fmt_new callback As part of updating the core to directly tell drivers if they are clock provider or consumer update these CPU side drivers to use the new direct callback. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220519154318.2153729-4-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/atmel/atmel-i2s.c | 6 +++--- sound/soc/atmel/atmel_ssc_dai.c | 20 ++++++++++---------- sound/soc/atmel/mchp-i2s-mcc.c | 10 +++++----- sound/soc/atmel/mchp-pdmc.c | 6 +++--- 4 files changed, 21 insertions(+), 21 deletions(-) (limited to 'sound') diff --git a/sound/soc/atmel/atmel-i2s.c b/sound/soc/atmel/atmel-i2s.c index 1934767690b5..c5ce695da586 100644 --- a/sound/soc/atmel/atmel-i2s.c +++ b/sound/soc/atmel/atmel-i2s.c @@ -343,7 +343,7 @@ static int atmel_i2s_hw_params(struct snd_pcm_substream *substream, } switch (dev->fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { - case SND_SOC_DAIFMT_CBC_CFC: + case SND_SOC_DAIFMT_BP_FP: /* codec is slave, so cpu is master */ mr |= ATMEL_I2SC_MR_MODE_MASTER; ret = atmel_i2s_get_gck_param(dev, params_rate(params)); @@ -351,7 +351,7 @@ static int atmel_i2s_hw_params(struct snd_pcm_substream *substream, return ret; break; - case SND_SOC_DAIFMT_CBP_CFP: + case SND_SOC_DAIFMT_BC_FC: /* codec is master, so cpu is slave */ mr |= ATMEL_I2SC_MR_MODE_SLAVE; dev->gck_param = NULL; @@ -533,7 +533,7 @@ static const struct snd_soc_dai_ops atmel_i2s_dai_ops = { .prepare = atmel_i2s_prepare, .trigger = atmel_i2s_trigger, .hw_params = atmel_i2s_hw_params, - .set_fmt = atmel_i2s_set_dai_fmt, + .set_fmt_new = atmel_i2s_set_dai_fmt, }; static int atmel_i2s_dai_probe(struct snd_soc_dai *dai) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index c1dea8d62416..da094762dc99 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -210,7 +210,7 @@ static int atmel_ssc_hw_rule_rate(struct snd_pcm_hw_params *params, return frame_size; switch (ssc_p->daifmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { - case SND_SOC_DAIFMT_CBP_CFC: + case SND_SOC_DAIFMT_BC_FP: if ((ssc_p->dir_mask & SSC_DIR_MASK_CAPTURE) && ssc->clk_from_rk_pin) /* Receiver Frame Synchro (i.e. capture) @@ -220,7 +220,7 @@ static int atmel_ssc_hw_rule_rate(struct snd_pcm_hw_params *params, mck_div = 3; break; - case SND_SOC_DAIFMT_CBP_CFP: + case SND_SOC_DAIFMT_BC_FC: if ((ssc_p->dir_mask & SSC_DIR_MASK_PLAYBACK) && !ssc->clk_from_rk_pin) /* Transmit Frame Synchro (i.e. playback) @@ -233,7 +233,7 @@ static int atmel_ssc_hw_rule_rate(struct snd_pcm_hw_params *params, } switch (ssc_p->daifmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { - case SND_SOC_DAIFMT_CBC_CFC: + case SND_SOC_DAIFMT_BP_FP: r.num = ssc_p->mck_rate / mck_div / frame_size; ret = snd_interval_ratnum(i, 1, &r, &num, &den); @@ -243,8 +243,8 @@ static int atmel_ssc_hw_rule_rate(struct snd_pcm_hw_params *params, } break; - case SND_SOC_DAIFMT_CBP_CFC: - case SND_SOC_DAIFMT_CBP_CFP: + case SND_SOC_DAIFMT_BC_FP: + case SND_SOC_DAIFMT_BC_FC: t.min = 8000; t.max = ssc_p->mck_rate / mck_div / frame_size; t.openmin = t.openmax = 0; @@ -433,8 +433,8 @@ static int atmel_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, static int atmel_ssc_cfs(struct atmel_ssc_info *ssc_p) { switch (ssc_p->daifmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { - case SND_SOC_DAIFMT_CBP_CFC: - case SND_SOC_DAIFMT_CBC_CFC: + case SND_SOC_DAIFMT_BC_FP: + case SND_SOC_DAIFMT_BP_FP: return 1; } return 0; @@ -444,8 +444,8 @@ static int atmel_ssc_cfs(struct atmel_ssc_info *ssc_p) static int atmel_ssc_cbs(struct atmel_ssc_info *ssc_p) { switch (ssc_p->daifmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { - case SND_SOC_DAIFMT_CBC_CFP: - case SND_SOC_DAIFMT_CBC_CFC: + case SND_SOC_DAIFMT_BP_FC: + case SND_SOC_DAIFMT_BP_FP: return 1; } return 0; @@ -835,7 +835,7 @@ static const struct snd_soc_dai_ops atmel_ssc_dai_ops = { .prepare = atmel_ssc_prepare, .trigger = atmel_ssc_trigger, .hw_params = atmel_ssc_hw_params, - .set_fmt = atmel_ssc_set_dai_fmt, + .set_fmt_new = atmel_ssc_set_dai_fmt, .set_clkdiv = atmel_ssc_set_dai_clkdiv, }; diff --git a/sound/soc/atmel/mchp-i2s-mcc.c b/sound/soc/atmel/mchp-i2s-mcc.c index 6d1227a1d67b..48d434e0c331 100644 --- a/sound/soc/atmel/mchp-i2s-mcc.c +++ b/sound/soc/atmel/mchp-i2s-mcc.c @@ -350,7 +350,7 @@ static int mchp_i2s_mcc_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; /* We can't generate only FSYNC */ - if ((fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) == SND_SOC_DAIFMT_CBP_CFC) + if ((fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) == SND_SOC_DAIFMT_BC_FP) return -EINVAL; /* We can only reconfigure the IP when it's stopped */ @@ -547,19 +547,19 @@ static int mchp_i2s_mcc_hw_params(struct snd_pcm_substream *substream, } switch (dev->fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { - case SND_SOC_DAIFMT_CBC_CFC: + case SND_SOC_DAIFMT_BP_FP: /* cpu is BCLK and LRC master */ mra |= MCHP_I2SMCC_MRA_MODE_MASTER; if (dev->sysclk) mra |= MCHP_I2SMCC_MRA_IMCKMODE_GEN; set_divs = 1; break; - case SND_SOC_DAIFMT_CBC_CFP: + case SND_SOC_DAIFMT_BP_FC: /* cpu is BCLK master */ mrb |= MCHP_I2SMCC_MRB_CLKSEL_INT; set_divs = 1; fallthrough; - case SND_SOC_DAIFMT_CBP_CFP: + case SND_SOC_DAIFMT_BC_FC: /* cpu is slave */ mra |= MCHP_I2SMCC_MRA_MODE_SLAVE; if (dev->sysclk) @@ -877,7 +877,7 @@ static const struct snd_soc_dai_ops mchp_i2s_mcc_dai_ops = { .trigger = mchp_i2s_mcc_trigger, .hw_params = mchp_i2s_mcc_hw_params, .hw_free = mchp_i2s_mcc_hw_free, - .set_fmt = mchp_i2s_mcc_set_dai_fmt, + .set_fmt_new = mchp_i2s_mcc_set_dai_fmt, .set_tdm_slot = mchp_i2s_mcc_set_dai_tdm_slot, }; diff --git a/sound/soc/atmel/mchp-pdmc.c b/sound/soc/atmel/mchp-pdmc.c index a3856c73e221..b3f04fa2f608 100644 --- a/sound/soc/atmel/mchp-pdmc.c +++ b/sound/soc/atmel/mchp-pdmc.c @@ -492,8 +492,8 @@ static int mchp_pdmc_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) unsigned int fmt_format = fmt & SND_SOC_DAIFMT_FORMAT_MASK; /* IP needs to be bitclock master */ - if (fmt_master != SND_SOC_DAIFMT_CBS_CFS && - fmt_master != SND_SOC_DAIFMT_CBS_CFM) + if (fmt_master != SND_SOC_DAIFMT_BP_FP && + fmt_master != SND_SOC_DAIFMT_BP_FC) return -EINVAL; /* IP supports only PDM interface */ @@ -708,7 +708,7 @@ static int mchp_pdmc_trigger(struct snd_pcm_substream *substream, } static const struct snd_soc_dai_ops mchp_pdmc_dai_ops = { - .set_fmt = mchp_pdmc_set_fmt, + .set_fmt_new = mchp_pdmc_set_fmt, .startup = mchp_pdmc_startup, .shutdown = mchp_pdmc_shutdown, .hw_params = mchp_pdmc_hw_params, -- cgit v1.2.3 From fee11f70849b21a244e6e27d281f3858b671bfea Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 May 2022 16:42:26 +0100 Subject: ASoC: au1x: Update to use set_fmt_new callback As part of updating the core to directly tell drivers if they are clock provider or consumer update these CPU side drivers to use the new direct callback. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220519154318.2153729-5-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/au1x/i2sc.c | 4 ++-- sound/soc/au1x/psc-i2s.c | 6 +++--- 2 files changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/au1x/i2sc.c b/sound/soc/au1x/i2sc.c index 740d4e052e4d..72f16b7fda3e 100644 --- a/sound/soc/au1x/i2sc.c +++ b/sound/soc/au1x/i2sc.c @@ -121,7 +121,7 @@ static int au1xi2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) /* I2S controller only supports provider */ switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { - case SND_SOC_DAIFMT_CBC_CFC: /* CODEC consumer */ + case SND_SOC_DAIFMT_BP_FP: /* CODEC consumer */ break; default: goto out; @@ -206,7 +206,7 @@ static const struct snd_soc_dai_ops au1xi2s_dai_ops = { .startup = au1xi2s_startup, .trigger = au1xi2s_trigger, .hw_params = au1xi2s_hw_params, - .set_fmt = au1xi2s_set_fmt, + .set_fmt_new = au1xi2s_set_fmt, }; static struct snd_soc_dai_driver au1xi2s_dai_driver = { diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index b2b8896bb593..d82c1353f2f0 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -91,10 +91,10 @@ static int au1xpsc_i2s_set_fmt(struct snd_soc_dai *cpu_dai, } switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { - case SND_SOC_DAIFMT_CBP_CFP: /* CODEC provider */ + case SND_SOC_DAIFMT_BC_FC: /* CODEC provider */ ct |= PSC_I2SCFG_MS; /* PSC I2S consumer mode */ break; - case SND_SOC_DAIFMT_CBC_CFC: /* CODEC consumer */ + case SND_SOC_DAIFMT_BP_FP: /* CODEC consumer */ ct &= ~PSC_I2SCFG_MS; /* PSC I2S provider mode */ break; default: @@ -266,7 +266,7 @@ static const struct snd_soc_dai_ops au1xpsc_i2s_dai_ops = { .startup = au1xpsc_i2s_startup, .trigger = au1xpsc_i2s_trigger, .hw_params = au1xpsc_i2s_hw_params, - .set_fmt = au1xpsc_i2s_set_fmt, + .set_fmt_new = au1xpsc_i2s_set_fmt, }; static const struct snd_soc_dai_driver au1xpsc_i2s_dai_template = { -- cgit v1.2.3 From 04ea2404468b7885c560c3673f6f2fd368f305a2 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 May 2022 16:42:27 +0100 Subject: ASoC: bcm: Update to use set_fmt_new callback As part of updating the core to directly tell drivers if they are clock provider or consumer update these CPU side drivers to use the new direct callback. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220519154318.2153729-6-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/bcm/bcm2835-i2s.c | 22 +++++++++++----------- sound/soc/bcm/cygnus-ssp.c | 6 +++--- 2 files changed, 14 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/bcm/bcm2835-i2s.c b/sound/soc/bcm/bcm2835-i2s.c index e3fc4bee8cfd..aa7d8e081f89 100644 --- a/sound/soc/bcm/bcm2835-i2s.c +++ b/sound/soc/bcm/bcm2835-i2s.c @@ -133,8 +133,8 @@ static void bcm2835_i2s_start_clock(struct bcm2835_i2s_dev *dev) return; switch (provider) { - case SND_SOC_DAIFMT_CBC_CFC: - case SND_SOC_DAIFMT_CBC_CFP: + case SND_SOC_DAIFMT_BP_FP: + case SND_SOC_DAIFMT_BP_FC: clk_prepare_enable(dev->clk); dev->clk_prepared = true; break; @@ -385,12 +385,12 @@ static int bcm2835_i2s_hw_params(struct snd_pcm_substream *substream, /* Check if CPU is bit clock provider */ switch (dev->fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { - case SND_SOC_DAIFMT_CBC_CFC: - case SND_SOC_DAIFMT_CBC_CFP: + case SND_SOC_DAIFMT_BP_FP: + case SND_SOC_DAIFMT_BP_FC: bit_clock_provider = true; break; - case SND_SOC_DAIFMT_CBP_CFC: - case SND_SOC_DAIFMT_CBP_CFP: + case SND_SOC_DAIFMT_BC_FP: + case SND_SOC_DAIFMT_BC_FC: bit_clock_provider = false; break; default: @@ -399,12 +399,12 @@ static int bcm2835_i2s_hw_params(struct snd_pcm_substream *substream, /* Check if CPU is frame sync provider */ switch (dev->fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { - case SND_SOC_DAIFMT_CBC_CFC: - case SND_SOC_DAIFMT_CBP_CFC: + case SND_SOC_DAIFMT_BP_FP: + case SND_SOC_DAIFMT_BC_FP: frame_sync_provider = true; break; - case SND_SOC_DAIFMT_CBC_CFP: - case SND_SOC_DAIFMT_CBP_CFP: + case SND_SOC_DAIFMT_BP_FC: + case SND_SOC_DAIFMT_BC_FC: frame_sync_provider = false; break; default: @@ -743,7 +743,7 @@ static const struct snd_soc_dai_ops bcm2835_i2s_dai_ops = { .prepare = bcm2835_i2s_prepare, .trigger = bcm2835_i2s_trigger, .hw_params = bcm2835_i2s_hw_params, - .set_fmt = bcm2835_i2s_set_dai_fmt, + .set_fmt_new = bcm2835_i2s_set_dai_fmt, .set_bclk_ratio = bcm2835_i2s_set_dai_bclk_ratio, .set_tdm_slot = bcm2835_i2s_set_dai_tdm_slot, }; diff --git a/sound/soc/bcm/cygnus-ssp.c b/sound/soc/bcm/cygnus-ssp.c index 9698f4531c90..257f3657bcd6 100644 --- a/sound/soc/bcm/cygnus-ssp.c +++ b/sound/soc/bcm/cygnus-ssp.c @@ -849,11 +849,11 @@ static int cygnus_ssp_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) ssp_newcfg = 0; switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { - case SND_SOC_DAIFMT_CBP_CFP: + case SND_SOC_DAIFMT_BC_FC: ssp_newcfg |= BIT(I2S_OUT_CFGX_SLAVE_MODE); aio->is_slave = 1; break; - case SND_SOC_DAIFMT_CBC_CFC: + case SND_SOC_DAIFMT_BP_FP: ssp_newcfg &= ~BIT(I2S_OUT_CFGX_SLAVE_MODE); aio->is_slave = 0; break; @@ -1148,7 +1148,7 @@ static const struct snd_soc_dai_ops cygnus_ssp_dai_ops = { .shutdown = cygnus_ssp_shutdown, .trigger = cygnus_ssp_trigger, .hw_params = cygnus_ssp_hw_params, - .set_fmt = cygnus_ssp_set_fmt, + .set_fmt_new = cygnus_ssp_set_fmt, .set_sysclk = cygnus_ssp_set_sysclk, .set_tdm_slot = cygnus_set_dai_tdm_slot, }; -- cgit v1.2.3 From 5d6124e58d56818249a6266f56d9c3739e72e1bd Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 May 2022 16:42:28 +0100 Subject: ASoC: ep93xx: Update to use set_fmt_new callback As part of updating the core to directly tell drivers if they are clock provider or consumer update this CPU side driver to use the new direct callback. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220519154318.2153729-7-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/cirrus/ep93xx-i2s.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/cirrus/ep93xx-i2s.c b/sound/soc/cirrus/ep93xx-i2s.c index 2c8cd843d049..2c8b1c76b834 100644 --- a/sound/soc/cirrus/ep93xx-i2s.c +++ b/sound/soc/cirrus/ep93xx-i2s.c @@ -246,12 +246,12 @@ static int ep93xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, } switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { - case SND_SOC_DAIFMT_CBC_CFC: + case SND_SOC_DAIFMT_BP_FP: /* CPU is provider */ clk_cfg |= EP93XX_I2S_CLKCFG_MASTER; break; - case SND_SOC_DAIFMT_CBP_CFP: + case SND_SOC_DAIFMT_BC_FC: /* Codec is provider */ clk_cfg &= ~EP93XX_I2S_CLKCFG_MASTER; break; @@ -398,7 +398,7 @@ static const struct snd_soc_dai_ops ep93xx_i2s_dai_ops = { .shutdown = ep93xx_i2s_shutdown, .hw_params = ep93xx_i2s_hw_params, .set_sysclk = ep93xx_i2s_set_sysclk, - .set_fmt = ep93xx_i2s_set_dai_fmt, + .set_fmt_new = ep93xx_i2s_set_dai_fmt, }; #define EP93XX_I2S_FORMATS (SNDRV_PCM_FMTBIT_S32_LE) -- cgit v1.2.3 From ca0444f1f7b228ae3b8d1a5c0f0d1b4463171f98 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 May 2022 16:42:29 +0100 Subject: ASoC: dwc: Update to use set_fmt_new callback As part of updating the core to directly tell drivers if they are clock provider or consumer update this CPU side driver to use the new direct callback. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220519154318.2153729-8-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/dwc/dwc-i2s.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index 1edac3e10f34..d3778d2d739d 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -357,20 +357,20 @@ static int dw_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) int ret = 0; switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { - case SND_SOC_DAIFMT_CBP_CFP: + case SND_SOC_DAIFMT_BC_FC: if (dev->capability & DW_I2S_SLAVE) ret = 0; else ret = -EINVAL; break; - case SND_SOC_DAIFMT_CBC_CFC: + case SND_SOC_DAIFMT_BP_FP: if (dev->capability & DW_I2S_MASTER) ret = 0; else ret = -EINVAL; break; - case SND_SOC_DAIFMT_CBP_CFC: - case SND_SOC_DAIFMT_CBC_CFP: + case SND_SOC_DAIFMT_BC_FP: + case SND_SOC_DAIFMT_BP_FC: ret = -EINVAL; break; default: @@ -387,7 +387,7 @@ static const struct snd_soc_dai_ops dw_i2s_dai_ops = { .hw_params = dw_i2s_hw_params, .prepare = dw_i2s_prepare, .trigger = dw_i2s_trigger, - .set_fmt = dw_i2s_set_fmt, + .set_fmt_new = dw_i2s_set_fmt, }; #ifdef CONFIG_PM -- cgit v1.2.3 From 3b14c15a333b8225ea38479e13c0366539d3374a Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 May 2022 16:42:30 +0100 Subject: ASoC: fsl: Update to use set_fmt_new callback As part of updating the core to directly tell drivers if they are clock provider or consumer update these CPU side drivers to use the new direct callback. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220519154318.2153729-9-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_audmix.c | 6 +++--- sound/soc/fsl/fsl_esai.c | 10 +++++----- sound/soc/fsl/fsl_mqs.c | 4 ++-- sound/soc/fsl/fsl_sai.c | 10 +++++----- sound/soc/fsl/fsl_ssi.c | 24 ++++++++++++------------ sound/soc/fsl/imx-audmix.c | 4 ++-- sound/soc/fsl/imx-card.c | 2 +- 7 files changed, 30 insertions(+), 30 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_audmix.c b/sound/soc/fsl/fsl_audmix.c index 6dbb8c99f626..c580dcb9a4cf 100644 --- a/sound/soc/fsl/fsl_audmix.c +++ b/sound/soc/fsl/fsl_audmix.c @@ -259,8 +259,8 @@ static int fsl_audmix_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) /* For playback the AUDMIX is consumer, and for record is provider */ switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { - case SND_SOC_DAIFMT_CBP_CFP: - case SND_SOC_DAIFMT_CBC_CFC: + case SND_SOC_DAIFMT_BC_FC: + case SND_SOC_DAIFMT_BP_FP: break; default: return -EINVAL; @@ -317,7 +317,7 @@ static int fsl_audmix_dai_trigger(struct snd_pcm_substream *substream, int cmd, } static const struct snd_soc_dai_ops fsl_audmix_dai_ops = { - .set_fmt = fsl_audmix_dai_set_fmt, + .set_fmt_new = fsl_audmix_dai_set_fmt, .trigger = fsl_audmix_dai_trigger, }; diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 1a2bdf8e76f0..572bdaee73eb 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -480,16 +480,16 @@ static int fsl_esai_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) /* DAI clock provider masks */ switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { - case SND_SOC_DAIFMT_CBP_CFP: + case SND_SOC_DAIFMT_BC_FC: esai_priv->consumer_mode = true; break; - case SND_SOC_DAIFMT_CBC_CFP: + case SND_SOC_DAIFMT_BP_FC: xccr |= ESAI_xCCR_xCKD; break; - case SND_SOC_DAIFMT_CBP_CFC: + case SND_SOC_DAIFMT_BC_FP: xccr |= ESAI_xCCR_xFSD; break; - case SND_SOC_DAIFMT_CBC_CFC: + case SND_SOC_DAIFMT_BP_FP: xccr |= ESAI_xCCR_xFSD | ESAI_xCCR_xCKD; break; default: @@ -790,7 +790,7 @@ static const struct snd_soc_dai_ops fsl_esai_dai_ops = { .trigger = fsl_esai_trigger, .hw_params = fsl_esai_hw_params, .set_sysclk = fsl_esai_set_dai_sysclk, - .set_fmt = fsl_esai_set_dai_fmt, + .set_fmt_new = fsl_esai_set_dai_fmt, .set_tdm_slot = fsl_esai_set_dai_tdm_slot, }; diff --git a/sound/soc/fsl/fsl_mqs.c b/sound/soc/fsl/fsl_mqs.c index ceaecbe3a25e..371d441b1dbe 100644 --- a/sound/soc/fsl/fsl_mqs.c +++ b/sound/soc/fsl/fsl_mqs.c @@ -103,7 +103,7 @@ static int fsl_mqs_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) } switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { - case SND_SOC_DAIFMT_CBC_CFC: + case SND_SOC_DAIFMT_BP_FP: break; default: return -EINVAL; @@ -152,7 +152,7 @@ static const struct snd_soc_dai_ops fsl_mqs_dai_ops = { .startup = fsl_mqs_startup, .shutdown = fsl_mqs_shutdown, .hw_params = fsl_mqs_hw_params, - .set_fmt = fsl_mqs_set_dai_fmt, + .set_fmt_new = fsl_mqs_set_dai_fmt, }; static struct snd_soc_dai_driver fsl_mqs_dai = { diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index fa950dde5310..3edd302eb5c2 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -292,19 +292,19 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, /* DAI clock provider masks */ switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { - case SND_SOC_DAIFMT_CBC_CFC: + case SND_SOC_DAIFMT_BP_FP: val_cr2 |= FSL_SAI_CR2_BCD_MSTR; val_cr4 |= FSL_SAI_CR4_FSD_MSTR; sai->is_consumer_mode = false; break; - case SND_SOC_DAIFMT_CBP_CFP: + case SND_SOC_DAIFMT_BC_FC: sai->is_consumer_mode = true; break; - case SND_SOC_DAIFMT_CBC_CFP: + case SND_SOC_DAIFMT_BP_FC: val_cr2 |= FSL_SAI_CR2_BCD_MSTR; sai->is_consumer_mode = false; break; - case SND_SOC_DAIFMT_CBP_CFC: + case SND_SOC_DAIFMT_BC_FP: val_cr4 |= FSL_SAI_CR4_FSD_MSTR; sai->is_consumer_mode = true; break; @@ -704,7 +704,7 @@ static int fsl_sai_startup(struct snd_pcm_substream *substream, static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = { .set_bclk_ratio = fsl_sai_set_dai_bclk_ratio, .set_sysclk = fsl_sai_set_dai_sysclk, - .set_fmt = fsl_sai_set_dai_fmt, + .set_fmt_new = fsl_sai_set_dai_fmt, .set_tdm_slot = fsl_sai_set_dai_tdm_slot, .hw_params = fsl_sai_hw_params, .hw_free = fsl_sai_hw_free, diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 84cb36d9dfea..32e4cf37c202 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -93,7 +93,7 @@ */ #define FSLSSI_AC97_DAIFMT \ (SND_SOC_DAIFMT_AC97 | \ - SND_SOC_DAIFMT_CBM_CFS | \ + SND_SOC_DAIFMT_BC_FP | \ SND_SOC_DAIFMT_NB_NF) #define FSLSSI_SIER_DBG_RX_FLAGS \ @@ -358,13 +358,13 @@ static bool fsl_ssi_is_ac97(struct fsl_ssi *ssi) static bool fsl_ssi_is_i2s_clock_provider(struct fsl_ssi *ssi) { return (ssi->dai_fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) == - SND_SOC_DAIFMT_CBC_CFC; + SND_SOC_DAIFMT_BP_FP; } -static bool fsl_ssi_is_i2s_cbp_cfc(struct fsl_ssi *ssi) +static bool fsl_ssi_is_i2s_bc_fp(struct fsl_ssi *ssi) { return (ssi->dai_fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) == - SND_SOC_DAIFMT_CBP_CFC; + SND_SOC_DAIFMT_BC_FP; } /** @@ -847,7 +847,7 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, u8 i2s_net = ssi->i2s_net; /* Normal + Network mode to send 16-bit data in 32-bit frames */ - if (fsl_ssi_is_i2s_cbp_cfc(ssi) && sample_size == 16) + if (fsl_ssi_is_i2s_bc_fp(ssi) && sample_size == 16) i2s_net = SSI_SCR_I2S_MODE_NORMAL | SSI_SCR_NET; /* Use Normal mode to send mono data at 1st slot of 2 slots */ @@ -920,17 +920,17 @@ static int _fsl_ssi_set_dai_fmt(struct fsl_ssi *ssi, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { - case SND_SOC_DAIFMT_CBC_CFC: + case SND_SOC_DAIFMT_BP_FP: if (IS_ERR(ssi->baudclk)) { dev_err(ssi->dev, "missing baudclk for master mode\n"); return -EINVAL; } fallthrough; - case SND_SOC_DAIFMT_CBP_CFC: + case SND_SOC_DAIFMT_BC_FP: ssi->i2s_net |= SSI_SCR_I2S_MODE_MASTER; break; - case SND_SOC_DAIFMT_CBP_CFP: + case SND_SOC_DAIFMT_BC_FC: ssi->i2s_net |= SSI_SCR_I2S_MODE_SLAVE; break; default: @@ -992,15 +992,15 @@ static int _fsl_ssi_set_dai_fmt(struct fsl_ssi *ssi, unsigned int fmt) /* DAI clock provider masks */ switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { - case SND_SOC_DAIFMT_CBC_CFC: + case SND_SOC_DAIFMT_BP_FP: /* Output bit and frame sync clocks */ strcr |= SSI_STCR_TFDIR | SSI_STCR_TXDIR; scr |= SSI_SCR_SYS_CLK_EN; break; - case SND_SOC_DAIFMT_CBP_CFP: + case SND_SOC_DAIFMT_BC_FC: /* Input bit or frame sync clocks */ break; - case SND_SOC_DAIFMT_CBP_CFC: + case SND_SOC_DAIFMT_BC_FP: /* Input bit clock but output frame sync clock */ strcr |= SSI_STCR_TFDIR; break; @@ -1156,7 +1156,7 @@ static const struct snd_soc_dai_ops fsl_ssi_dai_ops = { .shutdown = fsl_ssi_shutdown, .hw_params = fsl_ssi_hw_params, .hw_free = fsl_ssi_hw_free, - .set_fmt = fsl_ssi_set_dai_fmt, + .set_fmt_new = fsl_ssi_set_dai_fmt, .set_tdm_slot = fsl_ssi_set_dai_tdm_slot, .trigger = fsl_ssi_trigger, }; diff --git a/sound/soc/fsl/imx-audmix.c b/sound/soc/fsl/imx-audmix.c index 502fe1b522ab..1292a845c424 100644 --- a/sound/soc/fsl/imx-audmix.c +++ b/sound/soc/fsl/imx-audmix.c @@ -81,7 +81,7 @@ static int imx_audmix_fe_hw_params(struct snd_pcm_substream *substream, int ret, dir; /* For playback the AUDMIX is consumer, and for record is provider */ - fmt |= tx ? SND_SOC_DAIFMT_CBC_CFC : SND_SOC_DAIFMT_CBP_CFP; + fmt |= tx ? SND_SOC_DAIFMT_BP_FP : SND_SOC_DAIFMT_BC_FC; dir = tx ? SND_SOC_CLOCK_OUT : SND_SOC_CLOCK_IN; /* set DAI configuration */ @@ -122,7 +122,7 @@ static int imx_audmix_be_hw_params(struct snd_pcm_substream *substream, return 0; /* For playback the AUDMIX is consumer */ - fmt |= SND_SOC_DAIFMT_CBP_CFP; + fmt |= SND_SOC_DAIFMT_BC_FC; /* set AUDMIX DAI configuration */ ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), fmt); diff --git a/sound/soc/fsl/imx-card.c b/sound/soc/fsl/imx-card.c index 6f8efd838fcc..1797d777b1b8 100644 --- a/sound/soc/fsl/imx-card.c +++ b/sound/soc/fsl/imx-card.c @@ -317,7 +317,7 @@ static int imx_aif_hw_params(struct snd_pcm_substream *substream, } } - ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + ret = snd_soc_dai_set_fmt(cpu_dai, snd_soc_daifmt_clock_provider_flipped(fmt)); if (ret && ret != -ENOTSUPP) { dev_err(dev, "failed to set cpu dai fmt: %d\n", ret); return ret; -- cgit v1.2.3 From 0f362524dd3face4865077a4f7e7e640a95702aa Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 May 2022 16:42:31 +0100 Subject: ASoC: hisilicon: Update to use set_fmt_new callback As part of updating the core to directly tell drivers if they are clock provider or consumer update this CPU side driver to use the new direct callback. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220519154318.2153729-10-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/hisilicon/hi6210-i2s.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/hisilicon/hi6210-i2s.c b/sound/soc/hisilicon/hi6210-i2s.c index a297d4af5099..51f98ae651a6 100644 --- a/sound/soc/hisilicon/hi6210-i2s.c +++ b/sound/soc/hisilicon/hi6210-i2s.c @@ -227,9 +227,9 @@ static int hi6210_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) * We don't actually set the hardware until the hw_params * call, but we need to validate the user input here. */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - case SND_SOC_DAIFMT_CBS_CFS: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_BC_FC: + case SND_SOC_DAIFMT_BP_FP: break; default: return -EINVAL; @@ -245,8 +245,8 @@ static int hi6210_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) } i2s->format = fmt; - i2s->master = (i2s->format & SND_SOC_DAIFMT_MASTER_MASK) == - SND_SOC_DAIFMT_CBS_CFS; + i2s->master = (i2s->format & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) == + SND_SOC_DAIFMT_BP_FP; return 0; } @@ -375,21 +375,21 @@ static int hi6210_i2s_hw_params(struct snd_pcm_substream *substream, hi6210_write_reg(i2s, HII2S_MUX_TOP_MODULE_CFG, val); - switch (i2s->format & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + switch (i2s->format & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_BC_FC: i2s->master = false; val = hi6210_read_reg(i2s, HII2S_I2S_CFG); val |= HII2S_I2S_CFG__S2_MST_SLV; hi6210_write_reg(i2s, HII2S_I2S_CFG, val); break; - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_BP_FP: i2s->master = true; val = hi6210_read_reg(i2s, HII2S_I2S_CFG); val &= ~HII2S_I2S_CFG__S2_MST_SLV; hi6210_write_reg(i2s, HII2S_I2S_CFG, val); break; default: - WARN_ONCE(1, "Invalid i2s->fmt MASTER_MASK. This shouldn't happen\n"); + WARN_ONCE(1, "Invalid i2s->fmt CLOCK_PROVIDER_MASK. This shouldn't happen\n"); return -EINVAL; } @@ -513,7 +513,7 @@ static int hi6210_i2s_dai_probe(struct snd_soc_dai *dai) static const struct snd_soc_dai_ops hi6210_i2s_dai_ops = { .trigger = hi6210_i2s_trigger, .hw_params = hi6210_i2s_hw_params, - .set_fmt = hi6210_i2s_set_fmt, + .set_fmt_new = hi6210_i2s_set_fmt, .startup = hi6210_i2s_startup, .shutdown = hi6210_i2s_shutdown, }; -- cgit v1.2.3 From ed2b384082a678a0c4c8c56deff9e5f46d5e3fca Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 May 2022 16:42:32 +0100 Subject: ASoC: img: Update to use set_fmt_new callback As part of updating the core to directly tell drivers if they are clock provider or consumer update these CPU side drivers to use the new direct callback. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220519154318.2153729-11-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/img/img-i2s-in.c | 6 +++--- sound/soc/img/img-i2s-out.c | 8 ++++---- 2 files changed, 7 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/img/img-i2s-in.c b/sound/soc/img/img-i2s-in.c index 09d23b11621c..79e733bc0ae6 100644 --- a/sound/soc/img/img-i2s-in.c +++ b/sound/soc/img/img-i2s-in.c @@ -333,8 +333,8 @@ static int img_i2s_in_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; } - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_BC_FC: break; default: return -EINVAL; @@ -373,7 +373,7 @@ static int img_i2s_in_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) static const struct snd_soc_dai_ops img_i2s_in_dai_ops = { .trigger = img_i2s_in_trigger, .hw_params = img_i2s_in_hw_params, - .set_fmt = img_i2s_in_set_fmt + .set_fmt_new = img_i2s_in_set_fmt }; static int img_i2s_in_dai_probe(struct snd_soc_dai *dai) diff --git a/sound/soc/img/img-i2s-out.c b/sound/soc/img/img-i2s-out.c index 28f48ca1508a..d92539603d6c 100644 --- a/sound/soc/img/img-i2s-out.c +++ b/sound/soc/img/img-i2s-out.c @@ -302,10 +302,10 @@ static int img_i2s_out_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) if (force_clk_active) control_set |= IMG_I2S_OUT_CTL_CLK_EN_MASK; - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_BC_FC: break; - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_BP_FP: control_set |= IMG_I2S_OUT_CTL_MASTER_MASK; break; default: @@ -381,7 +381,7 @@ static int img_i2s_out_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) static const struct snd_soc_dai_ops img_i2s_out_dai_ops = { .trigger = img_i2s_out_trigger, .hw_params = img_i2s_out_hw_params, - .set_fmt = img_i2s_out_set_fmt + .set_fmt_new = img_i2s_out_set_fmt }; static int img_i2s_out_dai_probe(struct snd_soc_dai *dai) -- cgit v1.2.3 From add9ee8c64c617f561a309cdda50104e9e2c12f6 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 May 2022 16:42:33 +0100 Subject: ASoC: Intel: Update to use set_fmt_new callback As part of updating the core to directly tell drivers if they are clock provider or consumer update these CPU side drivers to use the new direct callback. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220519154318.2153729-12-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/intel/atom/sst-atom-controls.c | 4 ++-- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 2 +- sound/soc/intel/boards/bytcht_cx2072x.c | 2 +- sound/soc/intel/boards/bytcht_da7213.c | 2 +- sound/soc/intel/boards/bytcht_es8316.c | 2 +- sound/soc/intel/boards/bytcht_nocodec.c | 2 +- sound/soc/intel/boards/bytcr_rt5640.c | 2 +- sound/soc/intel/boards/bytcr_rt5651.c | 2 +- sound/soc/intel/boards/bytcr_wm5102.c | 2 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 3 +-- sound/soc/intel/boards/cht_bsw_rt5645.c | 6 +++--- sound/soc/intel/boards/cht_bsw_rt5672.c | 2 +- sound/soc/intel/keembay/kmb_platform.c | 6 +++--- 13 files changed, 18 insertions(+), 19 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c index 335c32732994..406455ddcb96 100644 --- a/sound/soc/intel/atom/sst-atom-controls.c +++ b/sound/soc/intel/atom/sst-atom-controls.c @@ -831,9 +831,9 @@ static int sst_get_ssp_mode(struct snd_soc_dai *dai, unsigned int fmt) dev_dbg(dai->dev, "Enter:%s, format=%x\n", __func__, format); switch (format) { - case SND_SOC_DAIFMT_CBC_CFC: + case SND_SOC_DAIFMT_BP_FP: return SSP_MODE_PROVIDER; - case SND_SOC_DAIFMT_CBP_CFP: + case SND_SOC_DAIFMT_BC_FC: return SSP_MODE_CONSUMER; default: dev_err(dai->dev, "Invalid ssp protocol: %d\n", format); diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index a56dd48c045f..339d9440c150 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -473,7 +473,7 @@ static const struct snd_soc_dai_ops sst_compr_dai_ops = { static const struct snd_soc_dai_ops sst_be_dai_ops = { .startup = sst_enable_ssp, .hw_params = sst_be_hw_params, - .set_fmt = sst_set_format, + .set_fmt_new = sst_set_format, .set_tdm_slot = sst_platform_set_ssp_slot, .shutdown = sst_disable_ssp, }; diff --git a/sound/soc/intel/boards/bytcht_cx2072x.c b/sound/soc/intel/boards/bytcht_cx2072x.c index 0eed68a11f7e..ae899866863e 100644 --- a/sound/soc/intel/boards/bytcht_cx2072x.c +++ b/sound/soc/intel/boards/bytcht_cx2072x.c @@ -126,7 +126,7 @@ static int byt_cht_cx2072x_fixup(struct snd_soc_pcm_runtime *rtd, ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBC_CFC); + SND_SOC_DAIFMT_BP_FP); if (ret < 0) { dev_err(rtd->dev, "can't set format to I2S, err %d\n", ret); return ret; diff --git a/sound/soc/intel/boards/bytcht_da7213.c b/sound/soc/intel/boards/bytcht_da7213.c index eb19bf16afad..a0c8f1d3f8ce 100644 --- a/sound/soc/intel/boards/bytcht_da7213.c +++ b/sound/soc/intel/boards/bytcht_da7213.c @@ -81,7 +81,7 @@ static int codec_fixup(struct snd_soc_pcm_runtime *rtd, ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBC_CFC); + SND_SOC_DAIFMT_BP_FP); if (ret < 0) { dev_err(rtd->dev, "can't set format to I2S, err %d\n", ret); return ret; diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index a08507783e44..6432b83f616f 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -265,7 +265,7 @@ static int byt_cht_es8316_codec_fixup(struct snd_soc_pcm_runtime *rtd, ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBC_CFC + SND_SOC_DAIFMT_BP_FP ); if (ret < 0) { dev_err(rtd->dev, "can't set format to I2S, err %d\n", ret); diff --git a/sound/soc/intel/boards/bytcht_nocodec.c b/sound/soc/intel/boards/bytcht_nocodec.c index 115c2bcaabd4..7fc03f2efd35 100644 --- a/sound/soc/intel/boards/bytcht_nocodec.c +++ b/sound/soc/intel/boards/bytcht_nocodec.c @@ -61,7 +61,7 @@ static int codec_fixup(struct snd_soc_pcm_runtime *rtd, ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBC_CFC); + SND_SOC_DAIFMT_BP_FP); if (ret < 0) { dev_err(rtd->dev, "can't set format to I2S, err %d\n", ret); diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index ed9fa1728722..ce1f3eb5f83b 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -1413,7 +1413,7 @@ static int byt_rt5640_codec_fixup(struct snd_soc_pcm_runtime *rtd, ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBC_CFC); + SND_SOC_DAIFMT_BP_FP); if (ret < 0) { dev_err(rtd->dev, "can't set format to I2S, err %d\n", ret); return ret; diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index d467fcaa48ea..f72a597114bf 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -706,7 +706,7 @@ static int byt_rt5651_codec_fixup(struct snd_soc_pcm_runtime *rtd, ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBC_CFC + SND_SOC_DAIFMT_BP_FP ); if (ret < 0) { diff --git a/sound/soc/intel/boards/bytcr_wm5102.c b/sound/soc/intel/boards/bytcr_wm5102.c index 00384c6fbcaa..fe79f6e5f2bb 100644 --- a/sound/soc/intel/boards/bytcr_wm5102.c +++ b/sound/soc/intel/boards/bytcr_wm5102.c @@ -265,7 +265,7 @@ static int byt_wm5102_codec_fixup(struct snd_soc_pcm_runtime *rtd, ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBC_CFC); + SND_SOC_DAIFMT_BP_FP); if (ret) { dev_err(rtd->dev, "Error setting format to I2S: %d\n", ret); return ret; diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index a5160f27adea..64eb73525ee3 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -264,8 +264,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, return ret; } - fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBC_CFC; + fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_BP_FP; ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), fmt); if (ret < 0) { diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index 45c301ea5e00..56ee53e7ed3f 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -362,7 +362,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBC_CFC + SND_SOC_DAIFMT_BP_FP ); if (ret < 0) { dev_err(rtd->dev, "can't set format to I2S, err %d\n", ret); @@ -372,7 +372,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, ret = snd_soc_dai_set_fmt(asoc_rtd_to_codec(rtd, 0), SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBC_CFC + SND_SOC_DAIFMT_BC_FC ); if (ret < 0) { dev_err(rtd->dev, "can't set format to I2S, err %d\n", ret); @@ -396,7 +396,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, ret = snd_soc_dai_set_fmt(asoc_rtd_to_codec(rtd, 0), SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF | - SND_SOC_DAIFMT_CBC_CFC); + SND_SOC_DAIFMT_BC_FC); if (ret < 0) { dev_err(rtd->dev, "can't set format to TDM %d\n", ret); return ret; diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c index c80324f34b1b..ca47f6476b07 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -300,7 +300,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBC_CFC); + SND_SOC_DAIFMT_BP_FP); if (ret < 0) { dev_err(rtd->dev, "can't set format to I2S, err %d\n", ret); return ret; diff --git a/sound/soc/intel/keembay/kmb_platform.c b/sound/soc/intel/keembay/kmb_platform.c index a6fb74ba1c42..a65f03884d9a 100644 --- a/sound/soc/intel/keembay/kmb_platform.c +++ b/sound/soc/intel/keembay/kmb_platform.c @@ -497,11 +497,11 @@ static int kmb_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) int ret; switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { - case SND_SOC_DAIFMT_CBP_CFP: + case SND_SOC_DAIFMT_BC_FC: kmb_i2s->clock_provider = false; ret = 0; break; - case SND_SOC_DAIFMT_CBC_CFC: + case SND_SOC_DAIFMT_BP_FP: writel(CLOCK_PROVIDER_MODE, kmb_i2s->pss_base + I2S_GEN_CFG_0); ret = clk_prepare_enable(kmb_i2s->clk_i2s); @@ -736,7 +736,7 @@ static const struct snd_soc_dai_ops kmb_dai_ops = { .hw_params = kmb_dai_hw_params, .hw_free = kmb_dai_hw_free, .prepare = kmb_dai_prepare, - .set_fmt = kmb_set_dai_fmt, + .set_fmt_new = kmb_set_dai_fmt, }; static struct snd_soc_dai_driver intel_kmb_hdmi_dai[] = { -- cgit v1.2.3 From cbb3a19f090d5a41b822caf9ff2058e1c6bc7ea3 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 May 2022 16:42:34 +0100 Subject: ASoC: js4740-i2s: Update to use set_fmt_new callback As part of updating the core to directly tell drivers if they are clock provider or consumer update this CPU side driver to use the new direct callback. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220519154318.2153729-13-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/jz4740/jz4740-i2s.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index 7ad5d9a924d8..2c9dee241778 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -206,18 +206,18 @@ static int jz4740_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) conf &= ~(JZ_AIC_CONF_BIT_CLK_MASTER | JZ_AIC_CONF_SYNC_CLK_MASTER); - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_BP_FP: conf |= JZ_AIC_CONF_BIT_CLK_MASTER | JZ_AIC_CONF_SYNC_CLK_MASTER; format |= JZ_AIC_I2S_FMT_ENABLE_SYS_CLK; break; - case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_BC_FP: conf |= JZ_AIC_CONF_SYNC_CLK_MASTER; break; - case SND_SOC_DAIFMT_CBS_CFM: + case SND_SOC_DAIFMT_BP_FC: conf |= JZ_AIC_CONF_BIT_CLK_MASTER; break; - case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_BC_FC: break; default: return -EINVAL; @@ -433,7 +433,7 @@ static const struct snd_soc_dai_ops jz4740_i2s_dai_ops = { .shutdown = jz4740_i2s_shutdown, .trigger = jz4740_i2s_trigger, .hw_params = jz4740_i2s_hw_params, - .set_fmt = jz4740_i2s_set_fmt, + .set_fmt_new = jz4740_i2s_set_fmt, .set_sysclk = jz4740_i2s_set_sysclk, }; -- cgit v1.2.3 From 3af99430f8d948a41556156155b0295dec274d41 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 May 2022 16:42:35 +0100 Subject: ASoC: mediatek: Update to use set_fmt_new callback As part of updating the core to directly tell drivers if they are clock provider or consumer update these CPU side drivers to use the new direct callback. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220519154318.2153729-14-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8195/mt8195-dai-etdm.c | 10 +++++----- sound/soc/mediatek/mt8195/mt8195-dai-pcm.c | 8 ++++---- 2 files changed, 9 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/mediatek/mt8195/mt8195-dai-etdm.c b/sound/soc/mediatek/mt8195/mt8195-dai-etdm.c index c02c10da3600..5f7c9516dfa1 100644 --- a/sound/soc/mediatek/mt8195/mt8195-dai-etdm.c +++ b/sound/soc/mediatek/mt8195/mt8195-dai-etdm.c @@ -2172,11 +2172,11 @@ static int mtk_dai_etdm_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; } - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_BC_FC: etdm_data->slave_mode = true; break; - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_BP_FP: etdm_data->slave_mode = false; break; default: @@ -2346,7 +2346,7 @@ static const struct snd_soc_dai_ops mtk_dai_etdm_ops = { .hw_params = mtk_dai_etdm_hw_params, .trigger = mtk_dai_etdm_trigger, .set_sysclk = mtk_dai_etdm_set_sysclk, - .set_fmt = mtk_dai_etdm_set_fmt, + .set_fmt_new = mtk_dai_etdm_set_fmt, .set_tdm_slot = mtk_dai_etdm_set_tdm_slot, }; @@ -2356,7 +2356,7 @@ static const struct snd_soc_dai_ops mtk_dai_hdmitx_dptx_ops = { .hw_params = mtk_dai_hdmitx_dptx_hw_params, .trigger = mtk_dai_hdmitx_dptx_trigger, .set_sysclk = mtk_dai_hdmitx_dptx_set_sysclk, - .set_fmt = mtk_dai_etdm_set_fmt, + .set_fmt_new = mtk_dai_etdm_set_fmt, }; /* dai driver */ diff --git a/sound/soc/mediatek/mt8195/mt8195-dai-pcm.c b/sound/soc/mediatek/mt8195/mt8195-dai-pcm.c index 12644ded83d5..37a8968ac21d 100644 --- a/sound/soc/mediatek/mt8195/mt8195-dai-pcm.c +++ b/sound/soc/mediatek/mt8195/mt8195-dai-pcm.c @@ -266,11 +266,11 @@ static int mtk_dai_pcm_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; } - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_BC_FC: pcmif_priv->slave_mode = 1; break; - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_BP_FP: pcmif_priv->slave_mode = 0; break; default: @@ -282,7 +282,7 @@ static int mtk_dai_pcm_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) static const struct snd_soc_dai_ops mtk_dai_pcm_ops = { .prepare = mtk_dai_pcm_prepare, - .set_fmt = mtk_dai_pcm_set_fmt, + .set_fmt_new = mtk_dai_pcm_set_fmt, }; /* dai driver */ -- cgit v1.2.3 From f60442bf6eab47aa4ab127aab88afdcc29a09a73 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 May 2022 16:42:36 +0100 Subject: ASoC: meson: Update to use set_fmt_new callback As part of updating the core to directly tell drivers if they are clock provider or consumer update these CPU side drivers to use the new direct callback. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220519154318.2153729-15-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/meson/aiu-encoder-i2s.c | 4 ++-- sound/soc/meson/axg-tdm-interface.c | 16 ++++++++-------- 2 files changed, 10 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/meson/aiu-encoder-i2s.c b/sound/soc/meson/aiu-encoder-i2s.c index 67729de41a73..0ab991230dee 100644 --- a/sound/soc/meson/aiu-encoder-i2s.c +++ b/sound/soc/meson/aiu-encoder-i2s.c @@ -229,7 +229,7 @@ static int aiu_encoder_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) unsigned int skew; /* Only CPU Master / Codec Slave supported ATM */ - if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) + if ((fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) != SND_SOC_DAIFMT_BP_FP) return -EINVAL; if (inv == SND_SOC_DAIFMT_NB_IF || @@ -323,7 +323,7 @@ static void aiu_encoder_i2s_shutdown(struct snd_pcm_substream *substream, const struct snd_soc_dai_ops aiu_encoder_i2s_dai_ops = { .hw_params = aiu_encoder_i2s_hw_params, .hw_free = aiu_encoder_i2s_hw_free, - .set_fmt = aiu_encoder_i2s_set_fmt, + .set_fmt_new = aiu_encoder_i2s_set_fmt, .set_sysclk = aiu_encoder_i2s_set_sysclk, .startup = aiu_encoder_i2s_startup, .shutdown = aiu_encoder_i2s_shutdown, diff --git a/sound/soc/meson/axg-tdm-interface.c b/sound/soc/meson/axg-tdm-interface.c index e076ced30025..ffdb12d0e01e 100644 --- a/sound/soc/meson/axg-tdm-interface.c +++ b/sound/soc/meson/axg-tdm-interface.c @@ -119,19 +119,19 @@ static int axg_tdm_iface_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai); - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_BP_FP: if (!iface->mclk) { dev_err(dai->dev, "cpu clock master: mclk missing\n"); return -ENODEV; } break; - case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_BC_FC: break; - case SND_SOC_DAIFMT_CBS_CFM: - case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_BP_FC: + case SND_SOC_DAIFMT_BC_FP: dev_err(dai->dev, "only CBS_CFS and CBM_CFM are supported\n"); fallthrough; default: @@ -326,8 +326,8 @@ static int axg_tdm_iface_hw_params(struct snd_pcm_substream *substream, if (ret) return ret; - if ((iface->fmt & SND_SOC_DAIFMT_MASTER_MASK) == - SND_SOC_DAIFMT_CBS_CFS) { + if ((iface->fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) == + SND_SOC_DAIFMT_BP_FP) { ret = axg_tdm_iface_set_sclk(dai, params); if (ret) return ret; @@ -394,7 +394,7 @@ static int axg_tdm_iface_probe_dai(struct snd_soc_dai *dai) static const struct snd_soc_dai_ops axg_tdm_iface_ops = { .set_sysclk = axg_tdm_iface_set_sysclk, - .set_fmt = axg_tdm_iface_set_fmt, + .set_fmt_new = axg_tdm_iface_set_fmt, .startup = axg_tdm_iface_startup, .hw_params = axg_tdm_iface_hw_params, .prepare = axg_tdm_iface_prepare, -- cgit v1.2.3 From f3c0064f1f8e358799c70c7905a09d15c5ec5e5a Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 May 2022 16:42:37 +0100 Subject: ASoC: mxs-saif: Update to use set_fmt_new callback As part of updating the core to directly tell drivers if they are clock provider or consumer update this CPU side driver to use the new direct callback. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220519154318.2153729-16-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/mxs/mxs-saif.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 7afe1a1acc56..38de46ba1583 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -358,8 +358,8 @@ static int mxs_saif_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) * Saif internally could be slave when working on EXTMASTER mode. * We just hide this to machine driver. */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_BP_FP: if (saif->id == saif->master_id) scr &= ~BM_SAIF_CTRL_SLAVE_MODE; else @@ -642,7 +642,7 @@ static const struct snd_soc_dai_ops mxs_saif_dai_ops = { .prepare = mxs_saif_prepare, .hw_params = mxs_saif_hw_params, .set_sysclk = mxs_saif_set_dai_sysclk, - .set_fmt = mxs_saif_set_dai_fmt, + .set_fmt_new = mxs_saif_set_dai_fmt, }; static struct snd_soc_dai_driver mxs_saif_dai = { -- cgit v1.2.3 From 84c5b47c8ce4d5059d5e7539d3b44922cc0390e9 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 May 2022 16:42:38 +0100 Subject: ASoC: pxa: Update to use set_fmt_new callback As part of updating the core to directly tell drivers if they are clock provider or consumer update these CPU side drivers to use the new direct callback. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220519154318.2153729-17-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/pxa/magician.c | 8 ++++---- sound/soc/pxa/mmp-sspa.c | 8 ++++---- sound/soc/pxa/pxa-ssp.c | 24 ++++++++++++------------ sound/soc/pxa/pxa2xx-i2s.c | 8 ++++---- 4 files changed, 24 insertions(+), 24 deletions(-) (limited to 'sound') diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index 9433cc927755..b791a2ba5ce5 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -91,13 +91,13 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, /* set codec DAI configuration */ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_MSB | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_BC_FC); if (ret < 0) return ret; /* set cpu DAI configuration */ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A | - SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_CBS_CFS); + SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_BP_FP); if (ret < 0) return ret; @@ -129,14 +129,14 @@ static int magician_capture_hw_params(struct snd_pcm_substream *substream, /* set codec DAI configuration */ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS); + SND_SOC_DAIFMT_BC_FC); if (ret < 0) return ret; /* set cpu DAI configuration */ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS); + SND_SOC_DAIFMT_BP_FP); if (ret < 0) return ret; diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c index 7e39210a0b38..b746e52aaf85 100644 --- a/sound/soc/pxa/mmp-sspa.c +++ b/sound/soc/pxa/mmp-sspa.c @@ -171,11 +171,11 @@ static int mmp_sspa_set_dai_fmt(struct snd_soc_dai *cpu_dai, sspa->sp = SSPA_SP_WEN | SSPA_SP_S_RST | SSPA_SP_FFLUSH; sspa->ctrl = 0; - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_BP_FP: sspa->sp |= SSPA_SP_MSL; break; - case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_BC_FC: break; default: return -EINVAL; @@ -346,7 +346,7 @@ static const struct snd_soc_dai_ops mmp_sspa_dai_ops = { .hw_params = mmp_sspa_hw_params, .set_sysclk = mmp_sspa_set_dai_sysclk, .set_pll = mmp_sspa_set_dai_pll, - .set_fmt = mmp_sspa_set_dai_fmt, + .set_fmt_new = mmp_sspa_set_dai_fmt, }; static struct snd_soc_dai_driver mmp_sspa_dai = { diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 7f13a35e9cc1..52124be1778e 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -372,10 +372,10 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, { struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai); - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - case SND_SOC_DAIFMT_CBM_CFS: - case SND_SOC_DAIFMT_CBS_CFS: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_BC_FC: + case SND_SOC_DAIFMT_BC_FP: + case SND_SOC_DAIFMT_BP_FP: break; default: return -EINVAL; @@ -432,14 +432,14 @@ static int pxa_ssp_configure_dai_fmt(struct ssp_priv *priv) sscr1 |= SSCR1_RxTresh(8) | SSCR1_TxTresh(7); - switch (priv->dai_fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + switch (priv->dai_fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_BC_FC: sscr1 |= SSCR1_SCLKDIR | SSCR1_SFRMDIR | SSCR1_SCFR; break; - case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_BC_FP: sscr1 |= SSCR1_SCLKDIR | SSCR1_SCFR; break; - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_BP_FP: break; default: return -EINVAL; @@ -484,9 +484,9 @@ static int pxa_ssp_configure_dai_fmt(struct ssp_priv *priv) pxa_ssp_write_reg(ssp, SSCR1, sscr1); pxa_ssp_write_reg(ssp, SSPSP, sspsp); - switch (priv->dai_fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - case SND_SOC_DAIFMT_CBM_CFS: + switch (priv->dai_fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_BC_FC: + case SND_SOC_DAIFMT_BC_FP: scfr = pxa_ssp_read_reg(ssp, SSCR1) | SSCR1_SCFR; pxa_ssp_write_reg(ssp, SSCR1, scfr); @@ -824,7 +824,7 @@ static const struct snd_soc_dai_ops pxa_ssp_dai_ops = { .trigger = pxa_ssp_trigger, .hw_params = pxa_ssp_hw_params, .set_sysclk = pxa_ssp_set_dai_sysclk, - .set_fmt = pxa_ssp_set_dai_fmt, + .set_fmt_new = pxa_ssp_set_dai_fmt, .set_tdm_slot = pxa_ssp_set_dai_tdm_slot, .set_tristate = pxa_ssp_set_dai_tristate, }; diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 746e6ec9198b..9f12fc3615b6 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -129,11 +129,11 @@ static int pxa2xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, break; } - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_BP_FP: pxa_i2s.master = 1; break; - case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_BC_FP: pxa_i2s.master = 0; break; default: @@ -333,7 +333,7 @@ static const struct snd_soc_dai_ops pxa_i2s_dai_ops = { .shutdown = pxa2xx_i2s_shutdown, .trigger = pxa2xx_i2s_trigger, .hw_params = pxa2xx_i2s_hw_params, - .set_fmt = pxa2xx_i2s_set_dai_fmt, + .set_fmt_new = pxa2xx_i2s_set_dai_fmt, .set_sysclk = pxa2xx_i2s_set_dai_sysclk, }; -- cgit v1.2.3 From 1148e16b335f341f36475b646c692b4a71a1855e Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 May 2022 16:42:39 +0100 Subject: ASoC: qcom: Update to use set_fmt_new callback As part of updating the core to directly tell drivers if they are clock provider or consumer update these CPU side drivers to use the new direct callback. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220519154318.2153729-18-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/qcom/apq8016_sbc.c | 2 +- sound/soc/qcom/qdsp6/audioreach.c | 4 ++-- sound/soc/qcom/qdsp6/q6afe-dai.c | 2 +- sound/soc/qcom/qdsp6/q6afe.c | 6 +++--- sound/soc/qcom/qdsp6/q6apm-lpass-dais.c | 2 +- sound/soc/qcom/sc7180.c | 2 +- sound/soc/qcom/sdm845.c | 6 +++--- sound/soc/qcom/sm8250.c | 4 ++-- 8 files changed, 14 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/qcom/apq8016_sbc.c b/sound/soc/qcom/apq8016_sbc.c index b0a4f7ca2751..e54b8961112f 100644 --- a/sound/soc/qcom/apq8016_sbc.c +++ b/sound/soc/qcom/apq8016_sbc.c @@ -172,7 +172,7 @@ static int msm8916_qdsp6_dai_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); - snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_CBS_CFS); + snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_BP_FP); return apq8016_dai_init(rtd, qdsp6_dai_get_lpass_id(cpu_dai)); } diff --git a/sound/soc/qcom/qdsp6/audioreach.c b/sound/soc/qcom/qdsp6/audioreach.c index 98c0efa1d0fe..01dac32c50fd 100644 --- a/sound/soc/qcom/qdsp6/audioreach.c +++ b/sound/soc/qcom/qdsp6/audioreach.c @@ -732,10 +732,10 @@ static int audioreach_i2s_set_media_format(struct q6apm_graph *graph, intf_cfg->cfg.sd_line_idx = module->sd_line_idx; switch (cfg->fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { - case SND_SOC_DAIFMT_CBC_CFC: + case SND_SOC_DAIFMT_BP_FP: intf_cfg->cfg.ws_src = CONFIG_I2S_WS_SRC_INTERNAL; break; - case SND_SOC_DAIFMT_CBP_CFP: + case SND_SOC_DAIFMT_BC_FC: /* CPU is slave */ intf_cfg->cfg.ws_src = CONFIG_I2S_WS_SRC_EXTERNAL; break; diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c index 8bb7452b8f18..8f8794cffc1c 100644 --- a/sound/soc/qcom/qdsp6/q6afe-dai.c +++ b/sound/soc/qcom/qdsp6/q6afe-dai.c @@ -648,7 +648,7 @@ static const struct snd_soc_dai_ops q6hdmi_ops = { static const struct snd_soc_dai_ops q6i2s_ops = { .prepare = q6afe_dai_prepare, .hw_params = q6i2s_hw_params, - .set_fmt = q6i2s_set_fmt, + .set_fmt_new = q6i2s_set_fmt, .shutdown = q6afe_dai_shutdown, .set_sysclk = q6afe_mi2s_set_sysclk, }; diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c index 625724852a7f..919e326b9462 100644 --- a/sound/soc/qcom/qdsp6/q6afe.c +++ b/sound/soc/qcom/qdsp6/q6afe.c @@ -1328,11 +1328,11 @@ int q6afe_i2s_port_prepare(struct q6afe_port *port, struct q6afe_i2s_cfg *cfg) pcfg->i2s_cfg.bit_width = cfg->bit_width; pcfg->i2s_cfg.data_format = AFE_LINEAR_PCM_DATA; - switch (cfg->fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + switch (cfg->fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_BP_FP: pcfg->i2s_cfg.ws_src = AFE_PORT_CONFIG_I2S_WS_SRC_INTERNAL; break; - case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_BC_FC: /* CPU is slave */ pcfg->i2s_cfg.ws_src = AFE_PORT_CONFIG_I2S_WS_SRC_EXTERNAL; break; diff --git a/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c b/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c index ce9e5646d8f3..82ee52051f83 100644 --- a/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c +++ b/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c @@ -207,7 +207,7 @@ static const struct snd_soc_dai_ops q6i2s_ops = { .shutdown = q6apm_lpass_dai_shutdown, .set_channel_map = q6dma_set_channel_map, .hw_params = q6dma_hw_params, - .set_fmt = q6i2s_set_fmt, + .set_fmt_new = q6i2s_set_fmt, }; static const struct snd_soc_component_driver q6apm_lpass_dai_component = { diff --git a/sound/soc/qcom/sc7180.c b/sound/soc/qcom/sc7180.c index efccb5c0b3e0..f5f7c64b23a2 100644 --- a/sound/soc/qcom/sc7180.c +++ b/sound/soc/qcom/sc7180.c @@ -155,7 +155,7 @@ static int sc7180_snd_startup(struct snd_pcm_substream *substream) } snd_soc_dai_set_fmt(codec_dai, - SND_SOC_DAIFMT_CBS_CFS | + SND_SOC_DAIFMT_BC_FC | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_I2S); diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index 61fda790f375..d8d35563af00 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -316,8 +316,8 @@ static int sdm845_dai_init(struct snd_soc_pcm_runtime *rtd) static int sdm845_snd_startup(struct snd_pcm_substream *substream) { - unsigned int fmt = SND_SOC_DAIFMT_CBS_CFS; - unsigned int codec_dai_fmt = SND_SOC_DAIFMT_CBS_CFS; + unsigned int fmt = SND_SOC_DAIFMT_BP_FP; + unsigned int codec_dai_fmt = SND_SOC_DAIFMT_BC_FC; struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); struct snd_soc_card *card = rtd->card; struct sdm845_snd_data *data = snd_soc_card_get_drvdata(card); @@ -356,7 +356,7 @@ static int sdm845_snd_startup(struct snd_pcm_substream *substream) snd_soc_dai_set_sysclk(cpu_dai, Q6AFE_LPASS_CLK_ID_QUAD_MI2S_IBIT, MI2S_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK); - snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_CBS_CFS); + snd_soc_dai_set_fmt(cpu_dai, fmt); break; diff --git a/sound/soc/qcom/sm8250.c b/sound/soc/qcom/sm8250.c index 6e1184c8b672..ce4a5713386a 100644 --- a/sound/soc/qcom/sm8250.c +++ b/sound/soc/qcom/sm8250.c @@ -96,8 +96,8 @@ static int sm8250_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, static int sm8250_snd_startup(struct snd_pcm_substream *substream) { - unsigned int fmt = SND_SOC_DAIFMT_CBS_CFS; - unsigned int codec_dai_fmt = SND_SOC_DAIFMT_CBS_CFS; + unsigned int fmt = SND_SOC_DAIFMT_BP_FP; + unsigned int codec_dai_fmt = SND_SOC_DAIFMT_BC_FC; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); -- cgit v1.2.3 From 27646d265da1745b2d1d10fec18465631cb1135f Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 May 2022 16:42:40 +0100 Subject: ASoC: rockchip: Update to use set_fmt_new callback As part of updating the core to directly tell drivers if they are clock provider or consumer update these CPU side drivers to use the new direct callback. Signed-off-by: Charles Keepax Tested-by: Nicolas Frattaroli Link: https://lore.kernel.org/r/20220519154318.2153729-19-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/rockchip/rockchip_i2s.c | 8 ++++---- sound/soc/rockchip/rockchip_i2s_tdm.c | 8 ++++---- 2 files changed, 8 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 4ce5d2579387..0a66c7df323d 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -199,13 +199,13 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, pm_runtime_get_sync(cpu_dai->dev); mask = I2S_CKR_MSS_MASK; - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_BP_FP: /* Set source clock in Master mode */ val = I2S_CKR_MSS_MASTER; i2s->is_master_mode = true; break; - case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_BC_FC: val = I2S_CKR_MSS_SLAVE; i2s->is_master_mode = false; break; @@ -486,7 +486,7 @@ static const struct snd_soc_dai_ops rockchip_i2s_dai_ops = { .hw_params = rockchip_i2s_hw_params, .set_bclk_ratio = rockchip_i2s_set_bclk_ratio, .set_sysclk = rockchip_i2s_set_sysclk, - .set_fmt = rockchip_i2s_set_fmt, + .set_fmt_new = rockchip_i2s_set_fmt, .trigger = rockchip_i2s_trigger, }; diff --git a/sound/soc/rockchip/rockchip_i2s_tdm.c b/sound/soc/rockchip/rockchip_i2s_tdm.c index 98700e75b82a..c90afccdae36 100644 --- a/sound/soc/rockchip/rockchip_i2s_tdm.c +++ b/sound/soc/rockchip/rockchip_i2s_tdm.c @@ -411,12 +411,12 @@ static int rockchip_i2s_tdm_set_fmt(struct snd_soc_dai *cpu_dai, } mask = I2S_CKR_MSS_MASK; - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBC_CFC: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_BP_FP: val = I2S_CKR_MSS_MASTER; i2s_tdm->is_master_mode = true; break; - case SND_SOC_DAIFMT_CBP_CFP: + case SND_SOC_DAIFMT_BC_FC: val = I2S_CKR_MSS_SLAVE; i2s_tdm->is_master_mode = false; break; @@ -1113,7 +1113,7 @@ static const struct snd_soc_dai_ops rockchip_i2s_tdm_dai_ops = { .hw_params = rockchip_i2s_tdm_hw_params, .set_bclk_ratio = rockchip_i2s_tdm_set_bclk_ratio, .set_sysclk = rockchip_i2s_tdm_set_sysclk, - .set_fmt = rockchip_i2s_tdm_set_fmt, + .set_fmt_new = rockchip_i2s_tdm_set_fmt, .set_tdm_slot = rockchip_dai_tdm_slot, .trigger = rockchip_i2s_tdm_trigger, }; -- cgit v1.2.3 From 0b491c7c1b2555ef08285fd49a8567f2f9f34ff8 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 May 2022 16:42:41 +0100 Subject: ASoC: samsung: Update to use set_fmt_new callback As part of updating the core to directly tell drivers if they are clock provider or consumer update these CPU side drivers to use the new direct callback. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220519154318.2153729-20-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/samsung/i2s.c | 8 ++++---- sound/soc/samsung/pcm.c | 6 +++--- sound/soc/samsung/s3c-i2s-v2.c | 8 ++++---- sound/soc/samsung/s3c24xx-i2s.c | 8 ++++---- 4 files changed, 15 insertions(+), 15 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 70c827162be4..9ed275ebd744 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -671,11 +671,11 @@ static int i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; } - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_BC_FC: tmp |= mod_slave; break; - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_BP_FP: /* * Set default source clock in Master mode, only when the * CLK_I2S_RCLK_SRC clock is not exposed so we ensure any @@ -1107,7 +1107,7 @@ static int samsung_i2s_dai_remove(struct snd_soc_dai *dai) static const struct snd_soc_dai_ops samsung_i2s_dai_ops = { .trigger = i2s_trigger, .hw_params = i2s_hw_params, - .set_fmt = i2s_set_fmt, + .set_fmt_new = i2s_set_fmt, .set_clkdiv = i2s_set_clkdiv, .set_sysclk = i2s_set_sysclk, .startup = i2s_startup, diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index 4c4dfde0568f..818172d8832d 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -340,8 +340,8 @@ static int s3c_pcm_set_fmt(struct snd_soc_dai *cpu_dai, goto exit; } - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_BP_FP: /* Nothing to do, Master by default */ break; default: @@ -437,7 +437,7 @@ static const struct snd_soc_dai_ops s3c_pcm_dai_ops = { .set_clkdiv = s3c_pcm_set_clkdiv, .trigger = s3c_pcm_trigger, .hw_params = s3c_pcm_hw_params, - .set_fmt = s3c_pcm_set_fmt, + .set_fmt_new = s3c_pcm_set_fmt, }; static int s3c_pcm_dai_probe(struct snd_soc_dai *dai) diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c index de66cc422e6e..9c8a0697849d 100644 --- a/sound/soc/samsung/s3c-i2s-v2.c +++ b/sound/soc/samsung/s3c-i2s-v2.c @@ -252,12 +252,12 @@ static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai, iismod = readl(i2s->regs + S3C2412_IISMOD); pr_debug("hw_params r: IISMOD: %x \n", iismod); - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_BC_FC: i2s->master = 0; iismod |= S3C2412_IISMOD_SLAVE; break; - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_BP_FP: i2s->master = 1; iismod &= ~S3C2412_IISMOD_SLAVE; break; @@ -666,7 +666,7 @@ int s3c_i2sv2_register_component(struct device *dev, int id, ops->trigger = s3c2412_i2s_trigger; if (!ops->hw_params) ops->hw_params = s3c_i2sv2_hw_params; - ops->set_fmt = s3c2412_i2s_set_fmt; + ops->set_fmt_new = s3c2412_i2s_set_fmt; ops->set_clkdiv = s3c2412_i2s_set_clkdiv; ops->set_sysclk = s3c_i2sv2_set_sysclk; diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index 0f46304eaa4f..6226b3b585e5 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -169,11 +169,11 @@ static int s3c24xx_i2s_set_fmt(struct snd_soc_dai *cpu_dai, iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); pr_debug("hw_params r: IISMOD: %x \n", iismod); - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_BC_CFC: iismod |= S3C2410_IISMOD_SLAVE; break; - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_BP_FP: iismod &= ~S3C2410_IISMOD_SLAVE; break; default: @@ -394,7 +394,7 @@ static int s3c24xx_i2s_resume(struct snd_soc_component *component) static const struct snd_soc_dai_ops s3c24xx_i2s_dai_ops = { .trigger = s3c24xx_i2s_trigger, .hw_params = s3c24xx_i2s_hw_params, - .set_fmt = s3c24xx_i2s_set_fmt, + .set_fmt_new = s3c24xx_i2s_set_fmt, .set_clkdiv = s3c24xx_i2s_set_clkdiv, .set_sysclk = s3c24xx_i2s_set_sysclk, }; -- cgit v1.2.3 From 2d4dd776e902546389f2d7808ece7fd815aa829c Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 May 2022 16:42:42 +0100 Subject: ASoC: sh: Update to use set_fmt_new callback As part of updating the core to directly tell drivers if they are clock provider or consumer update these CPU side drivers to use the new direct callback. Signed-off-by: Charles Keepax Acked-by: Kuninori Morimoto Link: https://lore.kernel.org/r/20220519154318.2153729-21-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 8 ++++---- sound/soc/sh/rcar/core.c | 6 +++--- sound/soc/sh/rz-ssi.c | 4 ++-- sound/soc/sh/ssi.c | 12 ++++++------ 4 files changed, 15 insertions(+), 15 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index e9a1eb6bdf66..4058d60b7e93 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1646,10 +1646,10 @@ static int fsi_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) int ret; /* set clock master audio interface */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_BC_FC: break; - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_BP_FP: fsi->clk_master = 1; /* cpu is master */ break; default: @@ -1724,7 +1724,7 @@ static const struct snd_soc_dai_ops fsi_dai_ops = { .startup = fsi_dai_startup, .shutdown = fsi_dai_shutdown, .trigger = fsi_dai_trigger, - .set_fmt = fsi_dai_set_fmt, + .set_fmt_new = fsi_dai_set_fmt, .hw_params = fsi_dai_hw_params, .auto_selectable_formats = &fsi_dai_formats, .num_auto_selectable_formats = 1, diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index eb762ab94d3e..0ac15b74c58a 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -756,10 +756,10 @@ static int rsnd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) /* set clock master for audio interface */ switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { - case SND_SOC_DAIFMT_CBP_CFP: + case SND_SOC_DAIFMT_BC_FC: rdai->clk_master = 0; break; - case SND_SOC_DAIFMT_CBC_CFC: + case SND_SOC_DAIFMT_BP_FP: rdai->clk_master = 1; /* cpu is master */ break; default: @@ -1068,7 +1068,7 @@ static const struct snd_soc_dai_ops rsnd_soc_dai_ops = { .startup = rsnd_soc_dai_startup, .shutdown = rsnd_soc_dai_shutdown, .trigger = rsnd_soc_dai_trigger, - .set_fmt = rsnd_soc_dai_set_fmt, + .set_fmt_new = rsnd_soc_dai_set_fmt, .set_tdm_slot = rsnd_soc_set_dai_tdm_slot, .prepare = rsnd_soc_dai_prepare, .auto_selectable_formats = rsnd_soc_dai_formats, diff --git a/sound/soc/sh/rz-ssi.c b/sound/soc/sh/rz-ssi.c index e392de7a262e..0557d22a089f 100644 --- a/sound/soc/sh/rz-ssi.c +++ b/sound/soc/sh/rz-ssi.c @@ -767,7 +767,7 @@ static int rz_ssi_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) struct rz_ssi_priv *ssi = snd_soc_dai_get_drvdata(dai); switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { - case SND_SOC_DAIFMT_CBC_CFC: + case SND_SOC_DAIFMT_BP_FP: break; default: dev_err(ssi->dev, "Codec should be clk and frame consumer\n"); @@ -840,7 +840,7 @@ static int rz_ssi_dai_hw_params(struct snd_pcm_substream *substream, static const struct snd_soc_dai_ops rz_ssi_dai_ops = { .trigger = rz_ssi_dai_trigger, - .set_fmt = rz_ssi_dai_set_fmt, + .set_fmt_new = rz_ssi_dai_set_fmt, .hw_params = rz_ssi_dai_hw_params, }; diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c index 15b01bcefca5..95571cbeae29 100644 --- a/sound/soc/sh/ssi.c +++ b/sound/soc/sh/ssi.c @@ -291,16 +291,16 @@ static int ssi_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; } - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_BC_FC: break; - case SND_SOC_DAIFMT_CBS_CFM: + case SND_SOC_DAIFMT_BP_FC: ssicr |= CR_SCK_MASTER; break; - case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_BC_FP: ssicr |= CR_SWS_MASTER; break; - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_BP_FP: ssicr |= CR_SWS_MASTER | CR_SCK_MASTER; break; default: @@ -336,7 +336,7 @@ static const struct snd_soc_dai_ops ssi_dai_ops = { .hw_params = ssi_hw_params, .set_sysclk = ssi_set_sysclk, .set_clkdiv = ssi_set_clkdiv, - .set_fmt = ssi_set_fmt, + .set_fmt_new = ssi_set_fmt, }; static struct snd_soc_dai_driver sh4_ssi_dai[] = { -- cgit v1.2.3 From 0092dac91ec1c404787841bdd9ecbf3404d1a41c Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 May 2022 16:42:43 +0100 Subject: ASoC: stm: Update to use set_fmt_new callback As part of updating the core to directly tell drivers if they are clock provider or consumer update these CPU side drivers to use the new direct callback. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220519154318.2153729-22-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/stm/stm32_i2s.c | 10 +++++----- sound/soc/stm/stm32_sai_sub.c | 10 +++++----- 2 files changed, 10 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/stm/stm32_i2s.c b/sound/soc/stm/stm32_i2s.c index ac5dff4d1677..30c04f96ef1d 100644 --- a/sound/soc/stm/stm32_i2s.c +++ b/sound/soc/stm/stm32_i2s.c @@ -593,16 +593,16 @@ static int stm32_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) } /* DAI clock master masks */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_BC_FC: i2s->ms_flg = I2S_MS_SLAVE; break; - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_BP_FP: i2s->ms_flg = I2S_MS_MASTER; break; default: dev_err(cpu_dai->dev, "Unsupported mode %#x\n", - fmt & SND_SOC_DAIFMT_MASTER_MASK); + fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK); return -EINVAL; } @@ -954,7 +954,7 @@ static const struct regmap_config stm32_h7_i2s_regmap_conf = { static const struct snd_soc_dai_ops stm32_i2s_pcm_dai_ops = { .set_sysclk = stm32_i2s_set_sysclk, - .set_fmt = stm32_i2s_set_dai_fmt, + .set_fmt_new = stm32_i2s_set_dai_fmt, .startup = stm32_i2s_startup, .hw_params = stm32_i2s_hw_params, .trigger = stm32_i2s_trigger, diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index dd636af81c9b..9f169b93fa74 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -719,18 +719,18 @@ static int stm32_sai_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) stm32_sai_sub_reg_up(sai, STM_SAI_FRCR_REGX, frcr_mask, frcr); /* DAI clock master masks */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_BC_FC: /* codec is master */ cr1 |= SAI_XCR1_SLAVE; sai->master = false; break; - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_BP_FP: sai->master = true; break; default: dev_err(cpu_dai->dev, "Unsupported mode %#x\n", - fmt & SND_SOC_DAIFMT_MASTER_MASK); + fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK); return -EINVAL; } @@ -1225,7 +1225,7 @@ static int stm32_sai_dai_probe(struct snd_soc_dai *cpu_dai) static const struct snd_soc_dai_ops stm32_sai_pcm_dai_ops = { .set_sysclk = stm32_sai_set_sysclk, - .set_fmt = stm32_sai_set_dai_fmt, + .set_fmt_new = stm32_sai_set_dai_fmt, .set_tdm_slot = stm32_sai_set_dai_tdm_slot, .startup = stm32_sai_startup, .hw_params = stm32_sai_hw_params, -- cgit v1.2.3 From 7cc3965fde74c9c725ed01de4ac35bc7d562d16a Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 May 2022 16:42:44 +0100 Subject: ASoC: sunxi: Update to use set_fmt_new callback As part of updating the core to directly tell drivers if they are clock provider or consumer update these CPU side drivers to use the new direct callback. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220519154318.2153729-23-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/sunxi/sun4i-i2s.c | 20 ++++++++++---------- sound/soc/sunxi/sun8i-codec.c | 8 ++++---- 2 files changed, 14 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index 7047f71629ab..872838d3e0a9 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -702,13 +702,13 @@ static int sun4i_i2s_set_soc_fmt(const struct sun4i_i2s *i2s, SUN4I_I2S_FMT0_FMT_MASK, val); /* DAI clock master masks */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_BP_FP: /* BCLK and LRCLK master */ val = SUN4I_I2S_CTRL_MODE_MASTER; break; - case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_BC_FC: /* BCLK and LRCLK slave */ val = SUN4I_I2S_CTRL_MODE_SLAVE; break; @@ -802,13 +802,13 @@ static int sun8i_i2s_set_soc_fmt(const struct sun4i_i2s *i2s, SUN8I_I2S_TX_CHAN_OFFSET(offset)); /* DAI clock master masks */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_BP_FP: /* BCLK and LRCLK master */ val = SUN8I_I2S_CTRL_BCLK_OUT | SUN8I_I2S_CTRL_LRCK_OUT; break; - case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_BC_FC: /* BCLK and LRCLK slave */ val = 0; break; @@ -909,13 +909,13 @@ static int sun50i_h6_i2s_set_soc_fmt(const struct sun4i_i2s *i2s, SUN50I_H6_I2S_TX_CHAN_SEL_OFFSET(offset)); /* DAI clock master masks */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_BP_FP: /* BCLK and LRCLK master */ val = SUN8I_I2S_CTRL_BCLK_OUT | SUN8I_I2S_CTRL_LRCK_OUT; break; - case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_BC_FC: /* BCLK and LRCLK slave */ val = 0; break; @@ -1081,7 +1081,7 @@ static int sun4i_i2s_set_tdm_slot(struct snd_soc_dai *dai, static const struct snd_soc_dai_ops sun4i_i2s_dai_ops = { .hw_params = sun4i_i2s_hw_params, - .set_fmt = sun4i_i2s_set_fmt, + .set_fmt_new = sun4i_i2s_set_fmt, .set_sysclk = sun4i_i2s_set_sysclk, .set_tdm_slot = sun4i_i2s_set_tdm_slot, .trigger = sun4i_i2s_trigger, diff --git a/sound/soc/sunxi/sun8i-codec.c b/sound/soc/sunxi/sun8i-codec.c index 0bea2162f68d..6e9ef948d662 100644 --- a/sound/soc/sunxi/sun8i-codec.c +++ b/sound/soc/sunxi/sun8i-codec.c @@ -286,11 +286,11 @@ static int sun8i_codec_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) u32 dsp_format, format, invert, value; /* clock masters */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: /* Codec slave, DAI master */ + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_BP_FP: /* Codec slave, DAI master */ value = 0x1; break; - case SND_SOC_DAIFMT_CBM_CFM: /* Codec Master, DAI slave */ + case SND_SOC_DAIFMT_BC_FC: /* Codec Master, DAI slave */ value = 0x0; break; default: @@ -630,7 +630,7 @@ done: } static const struct snd_soc_dai_ops sun8i_codec_dai_ops = { - .set_fmt = sun8i_codec_set_fmt, + .set_fmt_new = sun8i_codec_set_fmt, .set_tdm_slot = sun8i_codec_set_tdm_slot, .startup = sun8i_codec_startup, .hw_params = sun8i_codec_hw_params, -- cgit v1.2.3 From d92ad6633fa77f9496840b77c8effeaa13ac78dc Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 May 2022 16:42:45 +0100 Subject: ASoC: tegra: Update to use set_fmt_new callback As part of updating the core to directly tell drivers if they are clock provider or consumer update these CPU side drivers to use the new direct callback. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220519154318.2153729-24-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/tegra/tegra20_i2s.c | 8 ++++---- sound/soc/tegra/tegra210_i2s.c | 8 ++++---- sound/soc/tegra/tegra30_i2s.c | 8 ++++---- 3 files changed, 12 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra20_i2s.c b/sound/soc/tegra/tegra20_i2s.c index 27365a877e47..9abb0e3536d8 100644 --- a/sound/soc/tegra/tegra20_i2s.c +++ b/sound/soc/tegra/tegra20_i2s.c @@ -95,11 +95,11 @@ static int tegra20_i2s_set_fmt(struct snd_soc_dai *dai, } mask |= TEGRA20_I2S_CTRL_MASTER_ENABLE; - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_BP_FP: val |= TEGRA20_I2S_CTRL_MASTER_ENABLE; break; - case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_BC_FC: break; default: return -EINVAL; @@ -311,7 +311,7 @@ static int tegra20_i2s_startup(struct snd_pcm_substream *substream, } static const struct snd_soc_dai_ops tegra20_i2s_dai_ops = { - .set_fmt = tegra20_i2s_set_fmt, + .set_fmt_new = tegra20_i2s_set_fmt, .hw_params = tegra20_i2s_hw_params, .trigger = tegra20_i2s_trigger, .startup = tegra20_i2s_startup, diff --git a/sound/soc/tegra/tegra210_i2s.c b/sound/soc/tegra/tegra210_i2s.c index 9552bbb939dd..a304948ee393 100644 --- a/sound/soc/tegra/tegra210_i2s.c +++ b/sound/soc/tegra/tegra210_i2s.c @@ -214,11 +214,11 @@ static int tegra210_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int mask, val; mask = I2S_CTRL_MASTER_EN_MASK; - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_BP_FP: val = 0; break; - case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_BC_FC: val = I2S_CTRL_MASTER_EN; break; default: @@ -678,7 +678,7 @@ static int tegra210_i2s_hw_params(struct snd_pcm_substream *substream, } static const struct snd_soc_dai_ops tegra210_i2s_dai_ops = { - .set_fmt = tegra210_i2s_set_fmt, + .set_fmt_new = tegra210_i2s_set_fmt, .hw_params = tegra210_i2s_hw_params, .set_bclk_ratio = tegra210_i2s_set_dai_bclk_ratio, .set_tdm_slot = tegra210_i2s_set_tdm_slot, diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c index 084a533bf4f2..a4ea5221de6b 100644 --- a/sound/soc/tegra/tegra30_i2s.c +++ b/sound/soc/tegra/tegra30_i2s.c @@ -87,11 +87,11 @@ static int tegra30_i2s_set_fmt(struct snd_soc_dai *dai, } mask |= TEGRA30_I2S_CTRL_MASTER_ENABLE; - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_BP_FP: val |= TEGRA30_I2S_CTRL_MASTER_ENABLE; break; - case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_BC_FC: break; default: return -EINVAL; @@ -304,7 +304,7 @@ static int tegra30_i2s_probe(struct snd_soc_dai *dai) } static const struct snd_soc_dai_ops tegra30_i2s_dai_ops = { - .set_fmt = tegra30_i2s_set_fmt, + .set_fmt_new = tegra30_i2s_set_fmt, .hw_params = tegra30_i2s_hw_params, .trigger = tegra30_i2s_trigger, .set_tdm_slot = tegra30_i2s_set_tdm, -- cgit v1.2.3 From d444c8d246a62392c0d249b1030c3ca271d47649 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 May 2022 16:42:46 +0100 Subject: ASoC: test-component: Update to use set_fmt_new callback As part of updating the core to directly tell drivers if they are clock provider or consumer update this CPU side driver to use the new direct callback. Signed-off-by: Charles Keepax Acked-by: Kuninori Morimoto Link: https://lore.kernel.org/r/20220519154318.2153729-25-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/generic/test-component.c | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/generic/test-component.c b/sound/soc/generic/test-component.c index 5da4725d9e16..3a992a6eba9b 100644 --- a/sound/soc/generic/test-component.c +++ b/sound/soc/generic/test-component.c @@ -66,7 +66,7 @@ static int test_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) unsigned int format = fmt & SND_SOC_DAIFMT_FORMAT_MASK; unsigned int clock = fmt & SND_SOC_DAIFMT_CLOCK_MASK; unsigned int inv = fmt & SND_SOC_DAIFMT_INV_MASK; - unsigned int master = fmt & SND_SOC_DAIFMT_MASTER_MASK; + unsigned int master = fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK; char *str; dev_info(dai->dev, "name : %s", dai->name); @@ -105,16 +105,16 @@ static int test_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) str = "unknown"; switch (master) { - case SND_SOC_DAIFMT_CBP_CFP: + case SND_SOC_DAIFMT_BP_FP: str = "clk provider, frame provider"; break; - case SND_SOC_DAIFMT_CBC_CFP: + case SND_SOC_DAIFMT_BC_FP: str = "clk consumer, frame provider"; break; - case SND_SOC_DAIFMT_CBP_CFC: + case SND_SOC_DAIFMT_BP_FC: str = "clk provider, frame consumer"; break; - case SND_SOC_DAIFMT_CBC_CFC: + case SND_SOC_DAIFMT_BC_FC: str = "clk consumer, frame consumer"; break; } @@ -192,10 +192,10 @@ static int test_dai_bespoke_trigger(struct snd_pcm_substream *substream, static u64 test_dai_formats = /* * Select below from Sound Card, not auto - * SND_SOC_POSSIBLE_DAIFMT_CBP_CFP - * SND_SOC_POSSIBLE_DAIFMT_CBC_CFP - * SND_SOC_POSSIBLE_DAIFMT_CBP_CFC - * SND_SOC_POSSIBLE_DAIFMT_CBC_CFC + * SND_SOC_POSSIBLE_DAIFMT_BP_FP + * SND_SOC_POSSIBLE_DAIFMT_BC_FP + * SND_SOC_POSSIBLE_DAIFMT_BP_FC + * SND_SOC_POSSIBLE_DAIFMT_BC_FC */ SND_SOC_POSSIBLE_DAIFMT_I2S | SND_SOC_POSSIBLE_DAIFMT_RIGHT_J | @@ -210,7 +210,7 @@ static u64 test_dai_formats = SND_SOC_POSSIBLE_DAIFMT_IB_IF; static const struct snd_soc_dai_ops test_ops = { - .set_fmt = test_dai_set_fmt, + .set_fmt_new = test_dai_set_fmt, .startup = test_dai_startup, .shutdown = test_dai_shutdown, .auto_selectable_formats = &test_dai_formats, @@ -221,7 +221,7 @@ static const struct snd_soc_dai_ops test_verbose_ops = { .set_sysclk = test_dai_set_sysclk, .set_pll = test_dai_set_pll, .set_clkdiv = test_dai_set_clkdiv, - .set_fmt = test_dai_set_fmt, + .set_fmt_new = test_dai_set_fmt, .mute_stream = test_dai_mute_stream, .startup = test_dai_startup, .shutdown = test_dai_shutdown, -- cgit v1.2.3 From 563ff63dc9fbb8ef4b8f145a53c84a5489bbd789 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 May 2022 16:42:47 +0100 Subject: ASoC: ti: Update to use set_fmt_new callback As part of updating the core to directly tell drivers if they are clock provider or consumer update these CPU side drivers to use the new direct callback. Signed-off-by: Charles Keepax Acked-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20220519154318.2153729-26-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/ti/davinci-i2s.c | 34 +++++++++++++++++----------------- sound/soc/ti/davinci-mcasp.c | 12 ++++++------ sound/soc/ti/omap-mcbsp.c | 14 +++++++------- 3 files changed, 30 insertions(+), 30 deletions(-) (limited to 'sound') diff --git a/sound/soc/ti/davinci-i2s.c b/sound/soc/ti/davinci-i2s.c index 0363a088d2e0..c7368d529668 100644 --- a/sound/soc/ti/davinci-i2s.c +++ b/sound/soc/ti/davinci-i2s.c @@ -230,15 +230,15 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, dev->fmt = fmt; /* set master/slave audio interface */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_BP_FP: /* cpu is master */ pcr = DAVINCI_MCBSP_PCR_FSXM | DAVINCI_MCBSP_PCR_FSRM | DAVINCI_MCBSP_PCR_CLKXM | DAVINCI_MCBSP_PCR_CLKRM; break; - case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_BC_FP: pcr = DAVINCI_MCBSP_PCR_FSRM | DAVINCI_MCBSP_PCR_FSXM; /* * Selection of the clock input pin that is the @@ -260,7 +260,7 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, } break; - case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_BC_FC: /* codec is master */ pcr = 0; break; @@ -395,12 +395,12 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr); } - master = dev->fmt & SND_SOC_DAIFMT_MASTER_MASK; + master = dev->fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK; fmt = params_format(params); mcbsp_word_length = asp_word_length[fmt]; switch (master) { - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_BP_FP: freq = clk_get_rate(dev->clk); srgr = DAVINCI_MCBSP_SRGR_FSGM | DAVINCI_MCBSP_SRGR_CLKSM; @@ -426,7 +426,7 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, clk_div &= 0xFF; srgr |= clk_div; break; - case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_BC_FP: srgr = DAVINCI_MCBSP_SRGR_FSGM; clk_div = dev->clk_div - 1; srgr |= DAVINCI_MCBSP_SRGR_FWID(mcbsp_word_length * 8 - 1); @@ -434,7 +434,7 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, clk_div &= 0xFF; srgr |= clk_div; break; - case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_BC_FC: /* Clock and frame sync given from external sources */ i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS); srgr = DAVINCI_MCBSP_SRGR_FSGM; @@ -473,15 +473,15 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, fmt = double_fmt[fmt]; } switch (master) { - case SND_SOC_DAIFMT_CBS_CFS: - case SND_SOC_DAIFMT_CBS_CFM: + case SND_SOC_DAIFMT_BP_FP: + case SND_SOC_DAIFMT_BP_FC: rcr |= DAVINCI_MCBSP_RCR_RFRLEN2(0); xcr |= DAVINCI_MCBSP_XCR_XFRLEN2(0); rcr |= DAVINCI_MCBSP_RCR_RPHASE; xcr |= DAVINCI_MCBSP_XCR_XPHASE; break; - case SND_SOC_DAIFMT_CBM_CFM: - case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_BC_FC: + case SND_SOC_DAIFMT_BC_FP: rcr |= DAVINCI_MCBSP_RCR_RFRLEN2(element_cnt - 1); xcr |= DAVINCI_MCBSP_XCR_XFRLEN2(element_cnt - 1); break; @@ -492,13 +492,13 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, mcbsp_word_length = asp_word_length[fmt]; switch (master) { - case SND_SOC_DAIFMT_CBS_CFS: - case SND_SOC_DAIFMT_CBS_CFM: + case SND_SOC_DAIFMT_BP_FP: + case SND_SOC_DAIFMT_BP_FC: rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(0); xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(0); break; - case SND_SOC_DAIFMT_CBM_CFM: - case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_BC_FC: + case SND_SOC_DAIFMT_BC_FP: rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(element_cnt - 1); xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(element_cnt - 1); break; @@ -606,7 +606,7 @@ static const struct snd_soc_dai_ops davinci_i2s_dai_ops = { .prepare = davinci_i2s_prepare, .trigger = davinci_i2s_trigger, .hw_params = davinci_i2s_hw_params, - .set_fmt = davinci_i2s_set_dai_fmt, + .set_fmt_new = davinci_i2s_set_dai_fmt, .set_clkdiv = davinci_i2s_dai_set_clkdiv, }; diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index 377be2e2b6ee..961bac696365 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -492,8 +492,8 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, FSRDLY(data_delay), FSRDLY(3)); - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_BP_FP: /* codec is clock and frame slave */ mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, AFSXE); @@ -510,7 +510,7 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, mcasp->bclk_master = 1; break; - case SND_SOC_DAIFMT_CBS_CFM: + case SND_SOC_DAIFMT_BP_FC: /* codec is clock slave and frame master */ mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); mcasp_clr_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, AFSXE); @@ -527,7 +527,7 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, mcasp->bclk_master = 1; break; - case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_BC_FP: /* codec is clock master and frame slave */ mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, AFSXE); @@ -544,7 +544,7 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, mcasp->bclk_master = 0; break; - case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_BC_FC: /* codec is clock and frame master */ mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); mcasp_clr_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, AFSXE); @@ -1620,7 +1620,7 @@ static const struct snd_soc_dai_ops davinci_mcasp_dai_ops = { .trigger = davinci_mcasp_trigger, .delay = davinci_mcasp_delay, .hw_params = davinci_mcasp_hw_params, - .set_fmt = davinci_mcasp_set_dai_fmt, + .set_fmt_new = davinci_mcasp_set_dai_fmt, .set_clkdiv = davinci_mcasp_set_clkdiv, .set_sysclk = davinci_mcasp_set_sysclk, .set_tdm_slot = davinci_mcasp_set_tdm_slot, diff --git a/sound/soc/ti/omap-mcbsp.c b/sound/soc/ti/omap-mcbsp.c index 4479d74f0a45..5bfb56d4ff84 100644 --- a/sound/soc/ti/omap-mcbsp.c +++ b/sound/soc/ti/omap-mcbsp.c @@ -1036,8 +1036,8 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, /* In McBSP master modes, FRAME (i.e. sample rate) is generated * by _counting_ BCLKs. Calculate frame size in BCLKs */ - master = mcbsp->fmt & SND_SOC_DAIFMT_MASTER_MASK; - if (master == SND_SOC_DAIFMT_CBS_CFS) { + master = mcbsp->fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK; + if (master == SND_SOC_DAIFMT_BP_FP) { div = mcbsp->clk_div ? mcbsp->clk_div : 1; framesize = (mcbsp->in_freq / div) / params_rate(params); @@ -1136,20 +1136,20 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, return -EINVAL; } - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_BP_FP: /* McBSP master. Set FS and bit clocks as outputs */ regs->pcr0 |= FSXM | FSRM | CLKXM | CLKRM; /* Sample rate generator drives the FS */ regs->srgr2 |= FSGM; break; - case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_BC_FP: /* McBSP slave. FS clock as output */ regs->srgr2 |= FSGM; regs->pcr0 |= FSXM | FSRM; break; - case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_BC_FC: /* McBSP slave */ break; default: @@ -1271,7 +1271,7 @@ static const struct snd_soc_dai_ops mcbsp_dai_ops = { .trigger = omap_mcbsp_dai_trigger, .delay = omap_mcbsp_dai_delay, .hw_params = omap_mcbsp_dai_hw_params, - .set_fmt = omap_mcbsp_dai_set_dai_fmt, + .set_fmt_new = omap_mcbsp_dai_set_dai_fmt, .set_clkdiv = omap_mcbsp_dai_set_clkdiv, .set_sysclk = omap_mcbsp_dai_set_dai_sysclk, }; -- cgit v1.2.3 From ce3467c78478e33927aea9043bf20f46fa4d5688 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 May 2022 16:42:48 +0100 Subject: ASoC: ux500: Update to use set_fmt_new callback As part of updating the core to directly tell drivers if they are clock provider or consumer update this CPU side driver to use the new direct callback. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220519154318.2153729-27-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/ux500/ux500_msp_dai.c | 38 +++++++++++++++++++------------------- 1 file changed, 19 insertions(+), 19 deletions(-) (limited to 'sound') diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c index 21052378a32e..cd6c4bdf5041 100644 --- a/sound/soc/ux500/ux500_msp_dai.c +++ b/sound/soc/ux500/ux500_msp_dai.c @@ -191,8 +191,8 @@ static int setup_clocking(struct snd_soc_dai *dai, return -EINVAL; } - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_BC_FC: dev_dbg(dai->dev, "%s: Codec is master.\n", __func__); msp_config->iodelay = 0x20; @@ -204,7 +204,7 @@ static int setup_clocking(struct snd_soc_dai *dai, break; - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_BP_FP: dev_dbg(dai->dev, "%s: Codec is slave.\n", __func__); msp_config->tx_clk_sel = TX_CLK_SEL_SRG; @@ -328,15 +328,15 @@ static int setup_msp_config(struct snd_pcm_substream *substream, dev_dbg(dai->dev, "%s: rate: %u, channels: %d.\n", __func__, runtime->rate, runtime->channels); switch (fmt & - (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_MASTER_MASK)) { - case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS: + (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK)) { + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_BP_FP: dev_dbg(dai->dev, "%s: SND_SOC_DAIFMT_I2S.\n", __func__); msp_config->default_protdesc = 1; msp_config->protocol = MSP_I2S_PROTOCOL; break; - case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_BC_FC: dev_dbg(dai->dev, "%s: SND_SOC_DAIFMT_I2S.\n", __func__); msp_config->data_size = MSP_DATA_BITS_16; @@ -348,10 +348,10 @@ static int setup_msp_config(struct snd_pcm_substream *substream, break; - case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS: - case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM: - case SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBS_CFS: - case SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_BP_FP: + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_BC_FC: + case SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_BP_FP: + case SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_BC_FC: dev_dbg(dai->dev, "%s: PCM format.\n", __func__); msp_config->data_size = MSP_DATA_BITS_16; @@ -477,7 +477,7 @@ static int ux500_msp_dai_prepare(struct snd_pcm_substream *substream, } /* Set OPP-level */ - if ((drvdata->fmt & SND_SOC_DAIFMT_MASTER_MASK) && + if ((drvdata->fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) && (drvdata->msp->f_bitclk > 19200000)) { /* If the bit-clock is higher than 19.2MHz, Vape should be * run in 100% OPP. Only when bit-clock is used (MSP master) @@ -544,13 +544,13 @@ static int ux500_msp_dai_set_dai_fmt(struct snd_soc_dai *dai, dev_dbg(dai->dev, "%s: MSP %d: Enter.\n", __func__, dai->id); switch (fmt & (SND_SOC_DAIFMT_FORMAT_MASK | - SND_SOC_DAIFMT_MASTER_MASK)) { - case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS: - case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM: - case SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBS_CFS: - case SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM: - case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS: - case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM: + SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK)) { + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_BP_FP: + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_BC_FC: + case SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_BP_FP: + case SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_BC_FC: + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_BP_FP: + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_BC_FC: break; default: @@ -707,7 +707,7 @@ static int ux500_msp_dai_probe(struct snd_soc_dai *dai) static const struct snd_soc_dai_ops ux500_msp_dai_ops[] = { { .set_sysclk = ux500_msp_dai_set_dai_sysclk, - .set_fmt = ux500_msp_dai_set_dai_fmt, + .set_fmt_new = ux500_msp_dai_set_dai_fmt, .set_tdm_slot = ux500_msp_dai_set_tdm_slot, .startup = ux500_msp_dai_startup, .shutdown = ux500_msp_dai_shutdown, -- cgit v1.2.3 From e945206a0a448ac81dde0609578508368946f7a6 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 May 2022 16:42:49 +0100 Subject: ASoC: xtensa: Update to use set_fmt_new callback As part of updating the core to directly tell drivers if they are clock provider or consumer update this CPU side driver to use the new direct callback. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220519154318.2153729-28-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/xtensa/xtfpga-i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/xtensa/xtfpga-i2s.c b/sound/soc/xtensa/xtfpga-i2s.c index aeb4b2c4d1d3..72935f491901 100644 --- a/sound/soc/xtensa/xtfpga-i2s.c +++ b/sound/soc/xtensa/xtfpga-i2s.c @@ -339,7 +339,7 @@ static int xtfpga_i2s_set_fmt(struct snd_soc_dai *cpu_dai, { if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF) return -EINVAL; - if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) + if ((fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) != SND_SOC_DAIFMT_BP_FP) return -EINVAL; if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) != SND_SOC_DAIFMT_I2S) return -EINVAL; @@ -487,7 +487,7 @@ static const struct snd_soc_component_driver xtfpga_i2s_component = { static const struct snd_soc_dai_ops xtfpga_i2s_dai_ops = { .startup = xtfpga_i2s_startup, .hw_params = xtfpga_i2s_hw_params, - .set_fmt = xtfpga_i2s_set_fmt, + .set_fmt_new = xtfpga_i2s_set_fmt, }; static struct snd_soc_dai_driver xtfpga_i2s_dai[] = { -- cgit v1.2.3 From 6c076273a326cc5b5162451aacf7b7744bb03c66 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 May 2022 16:42:50 +0100 Subject: ASoC: core: Always send the CPU DAI a direct clock specifier All CPU drivers are now updated to accept a direct indication of whether they are clock provider or consumer, rather than being told if the CODEC is clock provider. As such update the core to always pass this direct indication rather than only if the new set_fmt_new callback is defined. Note this makes no difference to the CODEC side of the DAI link as it was already being directly told if it was clock provider under the old system. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220519154318.2153729-29-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 15 +++------------ 1 file changed, 3 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 90f4265bea50..227540851ded 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1214,7 +1214,6 @@ int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd, { struct snd_soc_dai *cpu_dai; struct snd_soc_dai *codec_dai; - unsigned int inv_dai_fmt; unsigned int i; int ret; @@ -1227,19 +1226,11 @@ int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd, return ret; } - /* - * Flip the polarity for the "CPU" end of a CODEC<->CODEC link - */ - inv_dai_fmt = snd_soc_daifmt_clock_provider_flipped(dai_fmt); + /* Flip the polarity for the "CPU" end of link */ + dai_fmt = snd_soc_daifmt_clock_provider_flipped(dai_fmt); for_each_rtd_cpu_dais(rtd, i, cpu_dai) { - unsigned int fmt = dai_fmt; - - if (cpu_dai->driver->ops->set_fmt_new || - snd_soc_component_is_codec(cpu_dai->component)) - fmt = inv_dai_fmt; - - ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + ret = snd_soc_dai_set_fmt(cpu_dai, dai_fmt); if (ret != 0 && ret != -ENOTSUPP) return ret; } -- cgit v1.2.3 From 346f47e784cd48b456f267a66e0daf1ef10d21b3 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 May 2022 16:42:51 +0100 Subject: ASoC: amd: vangogh: Rename set_fmt_new back to set_fmt Now the core has been migrated across to the new direct clock specification we can move the drivers back to the normal set_fmt callback. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220519154318.2153729-30-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/amd/vangogh/acp5x-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/amd/vangogh/acp5x-i2s.c b/sound/soc/amd/vangogh/acp5x-i2s.c index 40fbd0bc77fd..72c8c68e5933 100644 --- a/sound/soc/amd/vangogh/acp5x-i2s.c +++ b/sound/soc/amd/vangogh/acp5x-i2s.c @@ -339,7 +339,7 @@ static int acp5x_i2s_trigger(struct snd_pcm_substream *substream, static const struct snd_soc_dai_ops acp5x_i2s_dai_ops = { .hw_params = acp5x_i2s_hwparams, .trigger = acp5x_i2s_trigger, - .set_fmt_new = acp5x_i2s_set_fmt, + .set_fmt = acp5x_i2s_set_fmt, .set_tdm_slot = acp5x_i2s_set_tdm_slot, }; -- cgit v1.2.3 From a839a53b9dc70f94032a671ee019599884612d4a Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 May 2022 16:42:52 +0100 Subject: ASoC: atmel: Rename set_fmt_new back to set_fmt Now the core has been migrated across to the new direct clock specification we can move the drivers back to the normal set_fmt callback. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220519154318.2153729-31-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/atmel/atmel-i2s.c | 2 +- sound/soc/atmel/atmel_ssc_dai.c | 2 +- sound/soc/atmel/mchp-i2s-mcc.c | 2 +- sound/soc/atmel/mchp-pdmc.c | 2 +- 4 files changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/atmel/atmel-i2s.c b/sound/soc/atmel/atmel-i2s.c index c5ce695da586..ba56d6ac7e57 100644 --- a/sound/soc/atmel/atmel-i2s.c +++ b/sound/soc/atmel/atmel-i2s.c @@ -533,7 +533,7 @@ static const struct snd_soc_dai_ops atmel_i2s_dai_ops = { .prepare = atmel_i2s_prepare, .trigger = atmel_i2s_trigger, .hw_params = atmel_i2s_hw_params, - .set_fmt_new = atmel_i2s_set_dai_fmt, + .set_fmt = atmel_i2s_set_dai_fmt, }; static int atmel_i2s_dai_probe(struct snd_soc_dai *dai) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index da094762dc99..c92905e343e7 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -835,7 +835,7 @@ static const struct snd_soc_dai_ops atmel_ssc_dai_ops = { .prepare = atmel_ssc_prepare, .trigger = atmel_ssc_trigger, .hw_params = atmel_ssc_hw_params, - .set_fmt_new = atmel_ssc_set_dai_fmt, + .set_fmt = atmel_ssc_set_dai_fmt, .set_clkdiv = atmel_ssc_set_dai_clkdiv, }; diff --git a/sound/soc/atmel/mchp-i2s-mcc.c b/sound/soc/atmel/mchp-i2s-mcc.c index 48d434e0c331..269eab56b6dd 100644 --- a/sound/soc/atmel/mchp-i2s-mcc.c +++ b/sound/soc/atmel/mchp-i2s-mcc.c @@ -877,7 +877,7 @@ static const struct snd_soc_dai_ops mchp_i2s_mcc_dai_ops = { .trigger = mchp_i2s_mcc_trigger, .hw_params = mchp_i2s_mcc_hw_params, .hw_free = mchp_i2s_mcc_hw_free, - .set_fmt_new = mchp_i2s_mcc_set_dai_fmt, + .set_fmt = mchp_i2s_mcc_set_dai_fmt, .set_tdm_slot = mchp_i2s_mcc_set_dai_tdm_slot, }; diff --git a/sound/soc/atmel/mchp-pdmc.c b/sound/soc/atmel/mchp-pdmc.c index b3f04fa2f608..b9f637059448 100644 --- a/sound/soc/atmel/mchp-pdmc.c +++ b/sound/soc/atmel/mchp-pdmc.c @@ -708,7 +708,7 @@ static int mchp_pdmc_trigger(struct snd_pcm_substream *substream, } static const struct snd_soc_dai_ops mchp_pdmc_dai_ops = { - .set_fmt_new = mchp_pdmc_set_fmt, + .set_fmt = mchp_pdmc_set_fmt, .startup = mchp_pdmc_startup, .shutdown = mchp_pdmc_shutdown, .hw_params = mchp_pdmc_hw_params, -- cgit v1.2.3 From 2c73f5fd20a845fcb48173578b7c83dbcbacdeda Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 May 2022 16:42:53 +0100 Subject: ASoC: au1x: Rename set_fmt_new back to set_fmt Now the core has been migrated across to the new direct clock specification we can move the drivers back to the normal set_fmt callback. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220519154318.2153729-32-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/au1x/i2sc.c | 2 +- sound/soc/au1x/psc-i2s.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/au1x/i2sc.c b/sound/soc/au1x/i2sc.c index 72f16b7fda3e..45bb7851e75d 100644 --- a/sound/soc/au1x/i2sc.c +++ b/sound/soc/au1x/i2sc.c @@ -206,7 +206,7 @@ static const struct snd_soc_dai_ops au1xi2s_dai_ops = { .startup = au1xi2s_startup, .trigger = au1xi2s_trigger, .hw_params = au1xi2s_hw_params, - .set_fmt_new = au1xi2s_set_fmt, + .set_fmt = au1xi2s_set_fmt, }; static struct snd_soc_dai_driver au1xi2s_dai_driver = { diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index d82c1353f2f0..530a072d7427 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -266,7 +266,7 @@ static const struct snd_soc_dai_ops au1xpsc_i2s_dai_ops = { .startup = au1xpsc_i2s_startup, .trigger = au1xpsc_i2s_trigger, .hw_params = au1xpsc_i2s_hw_params, - .set_fmt_new = au1xpsc_i2s_set_fmt, + .set_fmt = au1xpsc_i2s_set_fmt, }; static const struct snd_soc_dai_driver au1xpsc_i2s_dai_template = { -- cgit v1.2.3 From 1a267dd98c246237be00587b6e71f969bf75f10d Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 May 2022 16:42:54 +0100 Subject: ASoC: bcm: Rename set_fmt_new back to set_fmt Now the core has been migrated across to the new direct clock specification we can move the drivers back to the normal set_fmt callback. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220519154318.2153729-33-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/bcm/bcm2835-i2s.c | 2 +- sound/soc/bcm/cygnus-ssp.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/bcm/bcm2835-i2s.c b/sound/soc/bcm/bcm2835-i2s.c index aa7d8e081f89..e39c8d9f4099 100644 --- a/sound/soc/bcm/bcm2835-i2s.c +++ b/sound/soc/bcm/bcm2835-i2s.c @@ -743,7 +743,7 @@ static const struct snd_soc_dai_ops bcm2835_i2s_dai_ops = { .prepare = bcm2835_i2s_prepare, .trigger = bcm2835_i2s_trigger, .hw_params = bcm2835_i2s_hw_params, - .set_fmt_new = bcm2835_i2s_set_dai_fmt, + .set_fmt = bcm2835_i2s_set_dai_fmt, .set_bclk_ratio = bcm2835_i2s_set_dai_bclk_ratio, .set_tdm_slot = bcm2835_i2s_set_dai_tdm_slot, }; diff --git a/sound/soc/bcm/cygnus-ssp.c b/sound/soc/bcm/cygnus-ssp.c index 257f3657bcd6..4bfa2d715ff4 100644 --- a/sound/soc/bcm/cygnus-ssp.c +++ b/sound/soc/bcm/cygnus-ssp.c @@ -1148,7 +1148,7 @@ static const struct snd_soc_dai_ops cygnus_ssp_dai_ops = { .shutdown = cygnus_ssp_shutdown, .trigger = cygnus_ssp_trigger, .hw_params = cygnus_ssp_hw_params, - .set_fmt_new = cygnus_ssp_set_fmt, + .set_fmt = cygnus_ssp_set_fmt, .set_sysclk = cygnus_ssp_set_sysclk, .set_tdm_slot = cygnus_set_dai_tdm_slot, }; -- cgit v1.2.3 From 324a4db8de05290237793dc3d7da887846ae90c1 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 May 2022 16:42:55 +0100 Subject: ASoC: ep93xx: Rename set_fmt_new back to set_fmt Now the core has been migrated across to the new direct clock specification we can move the drivers back to the normal set_fmt callback. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220519154318.2153729-34-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/cirrus/ep93xx-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/cirrus/ep93xx-i2s.c b/sound/soc/cirrus/ep93xx-i2s.c index 2c8b1c76b834..47959794353a 100644 --- a/sound/soc/cirrus/ep93xx-i2s.c +++ b/sound/soc/cirrus/ep93xx-i2s.c @@ -398,7 +398,7 @@ static const struct snd_soc_dai_ops ep93xx_i2s_dai_ops = { .shutdown = ep93xx_i2s_shutdown, .hw_params = ep93xx_i2s_hw_params, .set_sysclk = ep93xx_i2s_set_sysclk, - .set_fmt_new = ep93xx_i2s_set_dai_fmt, + .set_fmt = ep93xx_i2s_set_dai_fmt, }; #define EP93XX_I2S_FORMATS (SNDRV_PCM_FMTBIT_S32_LE) -- cgit v1.2.3 From 765fb623a2cd925c550370f73efe2137c52a1b25 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 May 2022 16:42:56 +0100 Subject: ASoC: dwc: Rename set_fmt_new back to set_fmt Now the core has been migrated across to the new direct clock specification we can move the drivers back to the normal set_fmt callback. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220519154318.2153729-35-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/dwc/dwc-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index d3778d2d739d..e794e020052e 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -387,7 +387,7 @@ static const struct snd_soc_dai_ops dw_i2s_dai_ops = { .hw_params = dw_i2s_hw_params, .prepare = dw_i2s_prepare, .trigger = dw_i2s_trigger, - .set_fmt_new = dw_i2s_set_fmt, + .set_fmt = dw_i2s_set_fmt, }; #ifdef CONFIG_PM -- cgit v1.2.3 From 00778276cf4c611882219ab7aba9664c48981f1a Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 May 2022 16:42:57 +0100 Subject: ASoC: fsl: Rename set_fmt_new back to set_fmt Now the core has been migrated across to the new direct clock specification we can move the drivers back to the normal set_fmt callback. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220519154318.2153729-36-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_audmix.c | 2 +- sound/soc/fsl/fsl_esai.c | 2 +- sound/soc/fsl/fsl_mqs.c | 2 +- sound/soc/fsl/fsl_sai.c | 2 +- sound/soc/fsl/fsl_ssi.c | 2 +- 5 files changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_audmix.c b/sound/soc/fsl/fsl_audmix.c index c580dcb9a4cf..43857b7a81c9 100644 --- a/sound/soc/fsl/fsl_audmix.c +++ b/sound/soc/fsl/fsl_audmix.c @@ -317,7 +317,7 @@ static int fsl_audmix_dai_trigger(struct snd_pcm_substream *substream, int cmd, } static const struct snd_soc_dai_ops fsl_audmix_dai_ops = { - .set_fmt_new = fsl_audmix_dai_set_fmt, + .set_fmt = fsl_audmix_dai_set_fmt, .trigger = fsl_audmix_dai_trigger, }; diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 572bdaee73eb..75f7807df29a 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -790,7 +790,7 @@ static const struct snd_soc_dai_ops fsl_esai_dai_ops = { .trigger = fsl_esai_trigger, .hw_params = fsl_esai_hw_params, .set_sysclk = fsl_esai_set_dai_sysclk, - .set_fmt_new = fsl_esai_set_dai_fmt, + .set_fmt = fsl_esai_set_dai_fmt, .set_tdm_slot = fsl_esai_set_dai_tdm_slot, }; diff --git a/sound/soc/fsl/fsl_mqs.c b/sound/soc/fsl/fsl_mqs.c index 371d441b1dbe..fc539a139250 100644 --- a/sound/soc/fsl/fsl_mqs.c +++ b/sound/soc/fsl/fsl_mqs.c @@ -152,7 +152,7 @@ static const struct snd_soc_dai_ops fsl_mqs_dai_ops = { .startup = fsl_mqs_startup, .shutdown = fsl_mqs_shutdown, .hw_params = fsl_mqs_hw_params, - .set_fmt_new = fsl_mqs_set_dai_fmt, + .set_fmt = fsl_mqs_set_dai_fmt, }; static struct snd_soc_dai_driver fsl_mqs_dai = { diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 3edd302eb5c2..f67d8527876e 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -704,7 +704,7 @@ static int fsl_sai_startup(struct snd_pcm_substream *substream, static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = { .set_bclk_ratio = fsl_sai_set_dai_bclk_ratio, .set_sysclk = fsl_sai_set_dai_sysclk, - .set_fmt_new = fsl_sai_set_dai_fmt, + .set_fmt = fsl_sai_set_dai_fmt, .set_tdm_slot = fsl_sai_set_dai_tdm_slot, .hw_params = fsl_sai_hw_params, .hw_free = fsl_sai_hw_free, diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 32e4cf37c202..7dd0c48cd9ae 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1156,7 +1156,7 @@ static const struct snd_soc_dai_ops fsl_ssi_dai_ops = { .shutdown = fsl_ssi_shutdown, .hw_params = fsl_ssi_hw_params, .hw_free = fsl_ssi_hw_free, - .set_fmt_new = fsl_ssi_set_dai_fmt, + .set_fmt = fsl_ssi_set_dai_fmt, .set_tdm_slot = fsl_ssi_set_dai_tdm_slot, .trigger = fsl_ssi_trigger, }; -- cgit v1.2.3 From b9a7972818b84a15d46505df7808fd86c3fba5bb Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 May 2022 16:42:58 +0100 Subject: ASoC: hisilicon: Rename set_fmt_new back to set_fmt Now the core has been migrated across to the new direct clock specification we can move the drivers back to the normal set_fmt callback. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220519154318.2153729-37-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/hisilicon/hi6210-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/hisilicon/hi6210-i2s.c b/sound/soc/hisilicon/hi6210-i2s.c index 51f98ae651a6..689ae13f34f5 100644 --- a/sound/soc/hisilicon/hi6210-i2s.c +++ b/sound/soc/hisilicon/hi6210-i2s.c @@ -513,7 +513,7 @@ static int hi6210_i2s_dai_probe(struct snd_soc_dai *dai) static const struct snd_soc_dai_ops hi6210_i2s_dai_ops = { .trigger = hi6210_i2s_trigger, .hw_params = hi6210_i2s_hw_params, - .set_fmt_new = hi6210_i2s_set_fmt, + .set_fmt = hi6210_i2s_set_fmt, .startup = hi6210_i2s_startup, .shutdown = hi6210_i2s_shutdown, }; -- cgit v1.2.3 From 1830a30ec4cf1642a429e80dbbeb86aa7825c71a Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 May 2022 16:42:59 +0100 Subject: ASoC: img: Rename set_fmt_new back to set_fmt Now the core has been migrated across to the new direct clock specification we can move the drivers back to the normal set_fmt callback. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220519154318.2153729-38-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/img/img-i2s-in.c | 2 +- sound/soc/img/img-i2s-out.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/img/img-i2s-in.c b/sound/soc/img/img-i2s-in.c index 79e733bc0ae6..97cab6d95b16 100644 --- a/sound/soc/img/img-i2s-in.c +++ b/sound/soc/img/img-i2s-in.c @@ -373,7 +373,7 @@ static int img_i2s_in_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) static const struct snd_soc_dai_ops img_i2s_in_dai_ops = { .trigger = img_i2s_in_trigger, .hw_params = img_i2s_in_hw_params, - .set_fmt_new = img_i2s_in_set_fmt + .set_fmt = img_i2s_in_set_fmt }; static int img_i2s_in_dai_probe(struct snd_soc_dai *dai) diff --git a/sound/soc/img/img-i2s-out.c b/sound/soc/img/img-i2s-out.c index d92539603d6c..9ec6fc528e2b 100644 --- a/sound/soc/img/img-i2s-out.c +++ b/sound/soc/img/img-i2s-out.c @@ -381,7 +381,7 @@ static int img_i2s_out_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) static const struct snd_soc_dai_ops img_i2s_out_dai_ops = { .trigger = img_i2s_out_trigger, .hw_params = img_i2s_out_hw_params, - .set_fmt_new = img_i2s_out_set_fmt + .set_fmt = img_i2s_out_set_fmt }; static int img_i2s_out_dai_probe(struct snd_soc_dai *dai) -- cgit v1.2.3 From c14a6ce9848571cf67faff206b02e212bec82761 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 May 2022 16:43:00 +0100 Subject: ASoC: Intel: Rename set_fmt_new back to set_fmt Now the core has been migrated across to the new direct clock specification we can move the drivers back to the normal set_fmt callback. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220519154318.2153729-39-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 2 +- sound/soc/intel/keembay/kmb_platform.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 339d9440c150..a56dd48c045f 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -473,7 +473,7 @@ static const struct snd_soc_dai_ops sst_compr_dai_ops = { static const struct snd_soc_dai_ops sst_be_dai_ops = { .startup = sst_enable_ssp, .hw_params = sst_be_hw_params, - .set_fmt_new = sst_set_format, + .set_fmt = sst_set_format, .set_tdm_slot = sst_platform_set_ssp_slot, .shutdown = sst_disable_ssp, }; diff --git a/sound/soc/intel/keembay/kmb_platform.c b/sound/soc/intel/keembay/kmb_platform.c index a65f03884d9a..d10881fedc8b 100644 --- a/sound/soc/intel/keembay/kmb_platform.c +++ b/sound/soc/intel/keembay/kmb_platform.c @@ -736,7 +736,7 @@ static const struct snd_soc_dai_ops kmb_dai_ops = { .hw_params = kmb_dai_hw_params, .hw_free = kmb_dai_hw_free, .prepare = kmb_dai_prepare, - .set_fmt_new = kmb_set_dai_fmt, + .set_fmt = kmb_set_dai_fmt, }; static struct snd_soc_dai_driver intel_kmb_hdmi_dai[] = { -- cgit v1.2.3 From 1724cc38e7685ad8b01413acd70a4a731fc105ae Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 May 2022 16:43:01 +0100 Subject: ASoC: jz4740-i2s: Rename set_fmt_new back to set_fmt Now the core has been migrated across to the new direct clock specification we can move the drivers back to the normal set_fmt callback. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220519154318.2153729-40-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/jz4740/jz4740-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index 2c9dee241778..446c5e061564 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -433,7 +433,7 @@ static const struct snd_soc_dai_ops jz4740_i2s_dai_ops = { .shutdown = jz4740_i2s_shutdown, .trigger = jz4740_i2s_trigger, .hw_params = jz4740_i2s_hw_params, - .set_fmt_new = jz4740_i2s_set_fmt, + .set_fmt = jz4740_i2s_set_fmt, .set_sysclk = jz4740_i2s_set_sysclk, }; -- cgit v1.2.3 From 00ca2d152ef0fa9f4beb2a590e176499440de8fe Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 May 2022 16:43:02 +0100 Subject: ASoC: mediatek: Rename set_fmt_new back to set_fmt Now the core has been migrated across to the new direct clock specification we can move the drivers back to the normal set_fmt callback. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220519154318.2153729-41-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8195/mt8195-dai-etdm.c | 4 ++-- sound/soc/mediatek/mt8195/mt8195-dai-pcm.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/mediatek/mt8195/mt8195-dai-etdm.c b/sound/soc/mediatek/mt8195/mt8195-dai-etdm.c index 5f7c9516dfa1..c2e268054773 100644 --- a/sound/soc/mediatek/mt8195/mt8195-dai-etdm.c +++ b/sound/soc/mediatek/mt8195/mt8195-dai-etdm.c @@ -2346,7 +2346,7 @@ static const struct snd_soc_dai_ops mtk_dai_etdm_ops = { .hw_params = mtk_dai_etdm_hw_params, .trigger = mtk_dai_etdm_trigger, .set_sysclk = mtk_dai_etdm_set_sysclk, - .set_fmt_new = mtk_dai_etdm_set_fmt, + .set_fmt = mtk_dai_etdm_set_fmt, .set_tdm_slot = mtk_dai_etdm_set_tdm_slot, }; @@ -2356,7 +2356,7 @@ static const struct snd_soc_dai_ops mtk_dai_hdmitx_dptx_ops = { .hw_params = mtk_dai_hdmitx_dptx_hw_params, .trigger = mtk_dai_hdmitx_dptx_trigger, .set_sysclk = mtk_dai_hdmitx_dptx_set_sysclk, - .set_fmt_new = mtk_dai_etdm_set_fmt, + .set_fmt = mtk_dai_etdm_set_fmt, }; /* dai driver */ diff --git a/sound/soc/mediatek/mt8195/mt8195-dai-pcm.c b/sound/soc/mediatek/mt8195/mt8195-dai-pcm.c index 37a8968ac21d..caceb0deb467 100644 --- a/sound/soc/mediatek/mt8195/mt8195-dai-pcm.c +++ b/sound/soc/mediatek/mt8195/mt8195-dai-pcm.c @@ -282,7 +282,7 @@ static int mtk_dai_pcm_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) static const struct snd_soc_dai_ops mtk_dai_pcm_ops = { .prepare = mtk_dai_pcm_prepare, - .set_fmt_new = mtk_dai_pcm_set_fmt, + .set_fmt = mtk_dai_pcm_set_fmt, }; /* dai driver */ -- cgit v1.2.3 From eee6b5b9f3af0e906085022713ef41e56d03eca8 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 May 2022 16:43:03 +0100 Subject: ASoC: meson: Rename set_fmt_new back to set_fmt Now the core has been migrated across to the new direct clock specification we can move the drivers back to the normal set_fmt callback. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220519154318.2153729-42-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/meson/aiu-encoder-i2s.c | 2 +- sound/soc/meson/axg-tdm-interface.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/meson/aiu-encoder-i2s.c b/sound/soc/meson/aiu-encoder-i2s.c index 0ab991230dee..a0dd914c8ed1 100644 --- a/sound/soc/meson/aiu-encoder-i2s.c +++ b/sound/soc/meson/aiu-encoder-i2s.c @@ -323,7 +323,7 @@ static void aiu_encoder_i2s_shutdown(struct snd_pcm_substream *substream, const struct snd_soc_dai_ops aiu_encoder_i2s_dai_ops = { .hw_params = aiu_encoder_i2s_hw_params, .hw_free = aiu_encoder_i2s_hw_free, - .set_fmt_new = aiu_encoder_i2s_set_fmt, + .set_fmt = aiu_encoder_i2s_set_fmt, .set_sysclk = aiu_encoder_i2s_set_sysclk, .startup = aiu_encoder_i2s_startup, .shutdown = aiu_encoder_i2s_shutdown, diff --git a/sound/soc/meson/axg-tdm-interface.c b/sound/soc/meson/axg-tdm-interface.c index ffdb12d0e01e..c040c83637e0 100644 --- a/sound/soc/meson/axg-tdm-interface.c +++ b/sound/soc/meson/axg-tdm-interface.c @@ -394,7 +394,7 @@ static int axg_tdm_iface_probe_dai(struct snd_soc_dai *dai) static const struct snd_soc_dai_ops axg_tdm_iface_ops = { .set_sysclk = axg_tdm_iface_set_sysclk, - .set_fmt_new = axg_tdm_iface_set_fmt, + .set_fmt = axg_tdm_iface_set_fmt, .startup = axg_tdm_iface_startup, .hw_params = axg_tdm_iface_hw_params, .prepare = axg_tdm_iface_prepare, -- cgit v1.2.3 From 1a805faeb4915496671cd24bd2a75cc97a85dfc8 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 May 2022 16:43:04 +0100 Subject: ASoC: mxs-saif: Rename set_fmt_new back to set_fmt Now the core has been migrated across to the new direct clock specification we can move the drivers back to the normal set_fmt callback. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220519154318.2153729-43-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/mxs/mxs-saif.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 38de46ba1583..467b0f2ce0bb 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -642,7 +642,7 @@ static const struct snd_soc_dai_ops mxs_saif_dai_ops = { .prepare = mxs_saif_prepare, .hw_params = mxs_saif_hw_params, .set_sysclk = mxs_saif_set_dai_sysclk, - .set_fmt_new = mxs_saif_set_dai_fmt, + .set_fmt = mxs_saif_set_dai_fmt, }; static struct snd_soc_dai_driver mxs_saif_dai = { -- cgit v1.2.3 From 8e2cc2b241bc0bb905231f301e6dfc80dc79f8a8 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 May 2022 16:43:05 +0100 Subject: ASoC: pxa: Rename set_fmt_new back to set_fmt Now the core has been migrated across to the new direct clock specification we can move the drivers back to the normal set_fmt callback. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220519154318.2153729-44-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/pxa/mmp-sspa.c | 2 +- sound/soc/pxa/pxa-ssp.c | 2 +- sound/soc/pxa/pxa2xx-i2s.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c index b746e52aaf85..382e9d8608a3 100644 --- a/sound/soc/pxa/mmp-sspa.c +++ b/sound/soc/pxa/mmp-sspa.c @@ -346,7 +346,7 @@ static const struct snd_soc_dai_ops mmp_sspa_dai_ops = { .hw_params = mmp_sspa_hw_params, .set_sysclk = mmp_sspa_set_dai_sysclk, .set_pll = mmp_sspa_set_dai_pll, - .set_fmt_new = mmp_sspa_set_dai_fmt, + .set_fmt = mmp_sspa_set_dai_fmt, }; static struct snd_soc_dai_driver mmp_sspa_dai = { diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 52124be1778e..0f504a9f4983 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -824,7 +824,7 @@ static const struct snd_soc_dai_ops pxa_ssp_dai_ops = { .trigger = pxa_ssp_trigger, .hw_params = pxa_ssp_hw_params, .set_sysclk = pxa_ssp_set_dai_sysclk, - .set_fmt_new = pxa_ssp_set_dai_fmt, + .set_fmt = pxa_ssp_set_dai_fmt, .set_tdm_slot = pxa_ssp_set_dai_tdm_slot, .set_tristate = pxa_ssp_set_dai_tristate, }; diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 9f12fc3615b6..ffcf44e4dc8c 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -333,7 +333,7 @@ static const struct snd_soc_dai_ops pxa_i2s_dai_ops = { .shutdown = pxa2xx_i2s_shutdown, .trigger = pxa2xx_i2s_trigger, .hw_params = pxa2xx_i2s_hw_params, - .set_fmt_new = pxa2xx_i2s_set_dai_fmt, + .set_fmt = pxa2xx_i2s_set_dai_fmt, .set_sysclk = pxa2xx_i2s_set_dai_sysclk, }; -- cgit v1.2.3 From f1bd2fae856384f9377ca3faed0583d929002640 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 May 2022 16:43:06 +0100 Subject: ASoC: qcom: Rename set_fmt_new back to set_fmt Now the core has been migrated across to the new direct clock specification we can move the drivers back to the normal set_fmt callback. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220519154318.2153729-45-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6afe-dai.c | 2 +- sound/soc/qcom/qdsp6/q6apm-lpass-dais.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c index 8f8794cffc1c..8bb7452b8f18 100644 --- a/sound/soc/qcom/qdsp6/q6afe-dai.c +++ b/sound/soc/qcom/qdsp6/q6afe-dai.c @@ -648,7 +648,7 @@ static const struct snd_soc_dai_ops q6hdmi_ops = { static const struct snd_soc_dai_ops q6i2s_ops = { .prepare = q6afe_dai_prepare, .hw_params = q6i2s_hw_params, - .set_fmt_new = q6i2s_set_fmt, + .set_fmt = q6i2s_set_fmt, .shutdown = q6afe_dai_shutdown, .set_sysclk = q6afe_mi2s_set_sysclk, }; diff --git a/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c b/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c index 82ee52051f83..ce9e5646d8f3 100644 --- a/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c +++ b/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c @@ -207,7 +207,7 @@ static const struct snd_soc_dai_ops q6i2s_ops = { .shutdown = q6apm_lpass_dai_shutdown, .set_channel_map = q6dma_set_channel_map, .hw_params = q6dma_hw_params, - .set_fmt_new = q6i2s_set_fmt, + .set_fmt = q6i2s_set_fmt, }; static const struct snd_soc_component_driver q6apm_lpass_dai_component = { -- cgit v1.2.3 From 059f16bc0e02164617312435c31dffdc419f113f Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 May 2022 16:43:07 +0100 Subject: ASoC: rockchip: Rename set_fmt_new back to set_fmt Now the core has been migrated across to the new direct clock specification we can move the drivers back to the normal set_fmt callback. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220519154318.2153729-46-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/rockchip/rockchip_i2s.c | 2 +- sound/soc/rockchip/rockchip_i2s_tdm.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 0a66c7df323d..47a3971a9ce1 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -486,7 +486,7 @@ static const struct snd_soc_dai_ops rockchip_i2s_dai_ops = { .hw_params = rockchip_i2s_hw_params, .set_bclk_ratio = rockchip_i2s_set_bclk_ratio, .set_sysclk = rockchip_i2s_set_sysclk, - .set_fmt_new = rockchip_i2s_set_fmt, + .set_fmt = rockchip_i2s_set_fmt, .trigger = rockchip_i2s_trigger, }; diff --git a/sound/soc/rockchip/rockchip_i2s_tdm.c b/sound/soc/rockchip/rockchip_i2s_tdm.c index c90afccdae36..48b3ecfa58b4 100644 --- a/sound/soc/rockchip/rockchip_i2s_tdm.c +++ b/sound/soc/rockchip/rockchip_i2s_tdm.c @@ -1113,7 +1113,7 @@ static const struct snd_soc_dai_ops rockchip_i2s_tdm_dai_ops = { .hw_params = rockchip_i2s_tdm_hw_params, .set_bclk_ratio = rockchip_i2s_tdm_set_bclk_ratio, .set_sysclk = rockchip_i2s_tdm_set_sysclk, - .set_fmt_new = rockchip_i2s_tdm_set_fmt, + .set_fmt = rockchip_i2s_tdm_set_fmt, .set_tdm_slot = rockchip_dai_tdm_slot, .trigger = rockchip_i2s_tdm_trigger, }; -- cgit v1.2.3 From b99d00c724bcf395558cb3028e823bd8f554fee6 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 May 2022 16:43:08 +0100 Subject: ASoC: samsung: Rename set_fmt_new back to set_fmt Now the core has been migrated across to the new direct clock specification we can move the drivers back to the normal set_fmt callback. Signed-off-by: Charles Keepax Acked-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20220519154318.2153729-47-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/samsung/i2s.c | 2 +- sound/soc/samsung/pcm.c | 2 +- sound/soc/samsung/s3c-i2s-v2.c | 2 +- sound/soc/samsung/s3c24xx-i2s.c | 2 +- 4 files changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 9ed275ebd744..fdd9561c6a9f 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1107,7 +1107,7 @@ static int samsung_i2s_dai_remove(struct snd_soc_dai *dai) static const struct snd_soc_dai_ops samsung_i2s_dai_ops = { .trigger = i2s_trigger, .hw_params = i2s_hw_params, - .set_fmt_new = i2s_set_fmt, + .set_fmt = i2s_set_fmt, .set_clkdiv = i2s_set_clkdiv, .set_sysclk = i2s_set_sysclk, .startup = i2s_startup, diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index 818172d8832d..c2eb3534bfcc 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -437,7 +437,7 @@ static const struct snd_soc_dai_ops s3c_pcm_dai_ops = { .set_clkdiv = s3c_pcm_set_clkdiv, .trigger = s3c_pcm_trigger, .hw_params = s3c_pcm_hw_params, - .set_fmt_new = s3c_pcm_set_fmt, + .set_fmt = s3c_pcm_set_fmt, }; static int s3c_pcm_dai_probe(struct snd_soc_dai *dai) diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c index 9c8a0697849d..1bec72246ed0 100644 --- a/sound/soc/samsung/s3c-i2s-v2.c +++ b/sound/soc/samsung/s3c-i2s-v2.c @@ -666,7 +666,7 @@ int s3c_i2sv2_register_component(struct device *dev, int id, ops->trigger = s3c2412_i2s_trigger; if (!ops->hw_params) ops->hw_params = s3c_i2sv2_hw_params; - ops->set_fmt_new = s3c2412_i2s_set_fmt; + ops->set_fmt = s3c2412_i2s_set_fmt; ops->set_clkdiv = s3c2412_i2s_set_clkdiv; ops->set_sysclk = s3c_i2sv2_set_sysclk; diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index 6226b3b585e5..4082ad7cbcc1 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -394,7 +394,7 @@ static int s3c24xx_i2s_resume(struct snd_soc_component *component) static const struct snd_soc_dai_ops s3c24xx_i2s_dai_ops = { .trigger = s3c24xx_i2s_trigger, .hw_params = s3c24xx_i2s_hw_params, - .set_fmt_new = s3c24xx_i2s_set_fmt, + .set_fmt = s3c24xx_i2s_set_fmt, .set_clkdiv = s3c24xx_i2s_set_clkdiv, .set_sysclk = s3c24xx_i2s_set_sysclk, }; -- cgit v1.2.3 From adced68031f96642272fae4e8c36d45d13797306 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 May 2022 16:43:09 +0100 Subject: ASoC: sh: Rename set_fmt_new back to set_fmt Now the core has been migrated across to the new direct clock specification we can move the drivers back to the normal set_fmt callback. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220519154318.2153729-48-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 2 +- sound/soc/sh/rcar/core.c | 2 +- sound/soc/sh/rz-ssi.c | 2 +- sound/soc/sh/ssi.c | 2 +- 4 files changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 4058d60b7e93..f3edc2e3d9d7 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1724,7 +1724,7 @@ static const struct snd_soc_dai_ops fsi_dai_ops = { .startup = fsi_dai_startup, .shutdown = fsi_dai_shutdown, .trigger = fsi_dai_trigger, - .set_fmt_new = fsi_dai_set_fmt, + .set_fmt = fsi_dai_set_fmt, .hw_params = fsi_dai_hw_params, .auto_selectable_formats = &fsi_dai_formats, .num_auto_selectable_formats = 1, diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 0ac15b74c58a..a4180dc5a59b 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1068,7 +1068,7 @@ static const struct snd_soc_dai_ops rsnd_soc_dai_ops = { .startup = rsnd_soc_dai_startup, .shutdown = rsnd_soc_dai_shutdown, .trigger = rsnd_soc_dai_trigger, - .set_fmt_new = rsnd_soc_dai_set_fmt, + .set_fmt = rsnd_soc_dai_set_fmt, .set_tdm_slot = rsnd_soc_set_dai_tdm_slot, .prepare = rsnd_soc_dai_prepare, .auto_selectable_formats = rsnd_soc_dai_formats, diff --git a/sound/soc/sh/rz-ssi.c b/sound/soc/sh/rz-ssi.c index 0557d22a089f..beaf1a8d6da1 100644 --- a/sound/soc/sh/rz-ssi.c +++ b/sound/soc/sh/rz-ssi.c @@ -840,7 +840,7 @@ static int rz_ssi_dai_hw_params(struct snd_pcm_substream *substream, static const struct snd_soc_dai_ops rz_ssi_dai_ops = { .trigger = rz_ssi_dai_trigger, - .set_fmt_new = rz_ssi_dai_set_fmt, + .set_fmt = rz_ssi_dai_set_fmt, .hw_params = rz_ssi_dai_hw_params, }; diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c index 95571cbeae29..bf7a3c69920a 100644 --- a/sound/soc/sh/ssi.c +++ b/sound/soc/sh/ssi.c @@ -336,7 +336,7 @@ static const struct snd_soc_dai_ops ssi_dai_ops = { .hw_params = ssi_hw_params, .set_sysclk = ssi_set_sysclk, .set_clkdiv = ssi_set_clkdiv, - .set_fmt_new = ssi_set_fmt, + .set_fmt = ssi_set_fmt, }; static struct snd_soc_dai_driver sh4_ssi_dai[] = { -- cgit v1.2.3 From 02ba0d9680feee645a321d65012f38d0a368b559 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 May 2022 16:43:10 +0100 Subject: ASoC: stm: Rename set_fmt_new back to set_fmt Now the core has been migrated across to the new direct clock specification we can move the drivers back to the normal set_fmt callback. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220519154318.2153729-49-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/stm/stm32_i2s.c | 2 +- sound/soc/stm/stm32_sai_sub.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/stm/stm32_i2s.c b/sound/soc/stm/stm32_i2s.c index 30c04f96ef1d..32d885f84a92 100644 --- a/sound/soc/stm/stm32_i2s.c +++ b/sound/soc/stm/stm32_i2s.c @@ -954,7 +954,7 @@ static const struct regmap_config stm32_h7_i2s_regmap_conf = { static const struct snd_soc_dai_ops stm32_i2s_pcm_dai_ops = { .set_sysclk = stm32_i2s_set_sysclk, - .set_fmt_new = stm32_i2s_set_dai_fmt, + .set_fmt = stm32_i2s_set_dai_fmt, .startup = stm32_i2s_startup, .hw_params = stm32_i2s_hw_params, .trigger = stm32_i2s_trigger, diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 9f169b93fa74..e09221b5f1f3 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -1225,7 +1225,7 @@ static int stm32_sai_dai_probe(struct snd_soc_dai *cpu_dai) static const struct snd_soc_dai_ops stm32_sai_pcm_dai_ops = { .set_sysclk = stm32_sai_set_sysclk, - .set_fmt_new = stm32_sai_set_dai_fmt, + .set_fmt = stm32_sai_set_dai_fmt, .set_tdm_slot = stm32_sai_set_dai_tdm_slot, .startup = stm32_sai_startup, .hw_params = stm32_sai_hw_params, -- cgit v1.2.3 From 15011b2388d020a6cdb323539fc69c31b04d9f21 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 May 2022 16:43:11 +0100 Subject: ASoC: sunxi: Rename set_fmt_new back to set_fmt Now the core has been migrated across to the new direct clock specification we can move the drivers back to the normal set_fmt callback. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220519154318.2153729-50-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/sunxi/sun4i-i2s.c | 2 +- sound/soc/sunxi/sun8i-codec.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index 872838d3e0a9..f58aa6406a87 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -1081,7 +1081,7 @@ static int sun4i_i2s_set_tdm_slot(struct snd_soc_dai *dai, static const struct snd_soc_dai_ops sun4i_i2s_dai_ops = { .hw_params = sun4i_i2s_hw_params, - .set_fmt_new = sun4i_i2s_set_fmt, + .set_fmt = sun4i_i2s_set_fmt, .set_sysclk = sun4i_i2s_set_sysclk, .set_tdm_slot = sun4i_i2s_set_tdm_slot, .trigger = sun4i_i2s_trigger, diff --git a/sound/soc/sunxi/sun8i-codec.c b/sound/soc/sunxi/sun8i-codec.c index 6e9ef948d662..90d74a2d53f3 100644 --- a/sound/soc/sunxi/sun8i-codec.c +++ b/sound/soc/sunxi/sun8i-codec.c @@ -630,7 +630,7 @@ done: } static const struct snd_soc_dai_ops sun8i_codec_dai_ops = { - .set_fmt_new = sun8i_codec_set_fmt, + .set_fmt = sun8i_codec_set_fmt, .set_tdm_slot = sun8i_codec_set_tdm_slot, .startup = sun8i_codec_startup, .hw_params = sun8i_codec_hw_params, -- cgit v1.2.3 From 475f2af6a2ff33e828900601a162e324b9986f9a Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 May 2022 16:43:12 +0100 Subject: ASoC: tegra: Rename set_fmt_new back to set_fmt Now the core has been migrated across to the new direct clock specification we can move the drivers back to the normal set_fmt callback. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220519154318.2153729-51-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/tegra/tegra20_i2s.c | 2 +- sound/soc/tegra/tegra210_i2s.c | 2 +- sound/soc/tegra/tegra30_i2s.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra20_i2s.c b/sound/soc/tegra/tegra20_i2s.c index 9abb0e3536d8..2e1a726602f0 100644 --- a/sound/soc/tegra/tegra20_i2s.c +++ b/sound/soc/tegra/tegra20_i2s.c @@ -311,7 +311,7 @@ static int tegra20_i2s_startup(struct snd_pcm_substream *substream, } static const struct snd_soc_dai_ops tegra20_i2s_dai_ops = { - .set_fmt_new = tegra20_i2s_set_fmt, + .set_fmt = tegra20_i2s_set_fmt, .hw_params = tegra20_i2s_hw_params, .trigger = tegra20_i2s_trigger, .startup = tegra20_i2s_startup, diff --git a/sound/soc/tegra/tegra210_i2s.c b/sound/soc/tegra/tegra210_i2s.c index a304948ee393..a28945895466 100644 --- a/sound/soc/tegra/tegra210_i2s.c +++ b/sound/soc/tegra/tegra210_i2s.c @@ -678,7 +678,7 @@ static int tegra210_i2s_hw_params(struct snd_pcm_substream *substream, } static const struct snd_soc_dai_ops tegra210_i2s_dai_ops = { - .set_fmt_new = tegra210_i2s_set_fmt, + .set_fmt = tegra210_i2s_set_fmt, .hw_params = tegra210_i2s_hw_params, .set_bclk_ratio = tegra210_i2s_set_dai_bclk_ratio, .set_tdm_slot = tegra210_i2s_set_tdm_slot, diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c index a4ea5221de6b..3aa157c82ae2 100644 --- a/sound/soc/tegra/tegra30_i2s.c +++ b/sound/soc/tegra/tegra30_i2s.c @@ -304,7 +304,7 @@ static int tegra30_i2s_probe(struct snd_soc_dai *dai) } static const struct snd_soc_dai_ops tegra30_i2s_dai_ops = { - .set_fmt_new = tegra30_i2s_set_fmt, + .set_fmt = tegra30_i2s_set_fmt, .hw_params = tegra30_i2s_hw_params, .trigger = tegra30_i2s_trigger, .set_tdm_slot = tegra30_i2s_set_tdm, -- cgit v1.2.3 From 408c122ef9de99220f7919594ab8af98194a19e8 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 May 2022 16:43:13 +0100 Subject: ASoC: test-component: Rename set_fmt_new back to set_fmt Now the core has been migrated across to the new direct clock specification we can move the drivers back to the normal set_fmt callback. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220519154318.2153729-52-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/generic/test-component.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/generic/test-component.c b/sound/soc/generic/test-component.c index 3a992a6eba9b..d28712fabe3f 100644 --- a/sound/soc/generic/test-component.c +++ b/sound/soc/generic/test-component.c @@ -210,7 +210,7 @@ static u64 test_dai_formats = SND_SOC_POSSIBLE_DAIFMT_IB_IF; static const struct snd_soc_dai_ops test_ops = { - .set_fmt_new = test_dai_set_fmt, + .set_fmt = test_dai_set_fmt, .startup = test_dai_startup, .shutdown = test_dai_shutdown, .auto_selectable_formats = &test_dai_formats, @@ -221,7 +221,7 @@ static const struct snd_soc_dai_ops test_verbose_ops = { .set_sysclk = test_dai_set_sysclk, .set_pll = test_dai_set_pll, .set_clkdiv = test_dai_set_clkdiv, - .set_fmt_new = test_dai_set_fmt, + .set_fmt = test_dai_set_fmt, .mute_stream = test_dai_mute_stream, .startup = test_dai_startup, .shutdown = test_dai_shutdown, -- cgit v1.2.3 From 9ff1836023ae19013c01f230e6a091fad6835213 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 May 2022 16:43:14 +0100 Subject: ASoC: ti: Rename set_fmt_new back to set_fmt Now the core has been migrated across to the new direct clock specification we can move the drivers back to the normal set_fmt callback. Signed-off-by: Charles Keepax Acked-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20220519154318.2153729-53-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/ti/davinci-i2s.c | 2 +- sound/soc/ti/davinci-mcasp.c | 2 +- sound/soc/ti/omap-mcbsp.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/ti/davinci-i2s.c b/sound/soc/ti/davinci-i2s.c index c7368d529668..fe572b720b09 100644 --- a/sound/soc/ti/davinci-i2s.c +++ b/sound/soc/ti/davinci-i2s.c @@ -606,7 +606,7 @@ static const struct snd_soc_dai_ops davinci_i2s_dai_ops = { .prepare = davinci_i2s_prepare, .trigger = davinci_i2s_trigger, .hw_params = davinci_i2s_hw_params, - .set_fmt_new = davinci_i2s_set_dai_fmt, + .set_fmt = davinci_i2s_set_dai_fmt, .set_clkdiv = davinci_i2s_dai_set_clkdiv, }; diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index 961bac696365..e2aab4729f3a 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -1620,7 +1620,7 @@ static const struct snd_soc_dai_ops davinci_mcasp_dai_ops = { .trigger = davinci_mcasp_trigger, .delay = davinci_mcasp_delay, .hw_params = davinci_mcasp_hw_params, - .set_fmt_new = davinci_mcasp_set_dai_fmt, + .set_fmt = davinci_mcasp_set_dai_fmt, .set_clkdiv = davinci_mcasp_set_clkdiv, .set_sysclk = davinci_mcasp_set_sysclk, .set_tdm_slot = davinci_mcasp_set_tdm_slot, diff --git a/sound/soc/ti/omap-mcbsp.c b/sound/soc/ti/omap-mcbsp.c index 5bfb56d4ff84..58d8e200a7b9 100644 --- a/sound/soc/ti/omap-mcbsp.c +++ b/sound/soc/ti/omap-mcbsp.c @@ -1271,7 +1271,7 @@ static const struct snd_soc_dai_ops mcbsp_dai_ops = { .trigger = omap_mcbsp_dai_trigger, .delay = omap_mcbsp_dai_delay, .hw_params = omap_mcbsp_dai_hw_params, - .set_fmt_new = omap_mcbsp_dai_set_dai_fmt, + .set_fmt = omap_mcbsp_dai_set_dai_fmt, .set_clkdiv = omap_mcbsp_dai_set_clkdiv, .set_sysclk = omap_mcbsp_dai_set_dai_sysclk, }; -- cgit v1.2.3 From e24ba1a21e244e7174e75ca0c4020beaff0ad369 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 May 2022 16:43:15 +0100 Subject: ASoC: ux500: Rename set_fmt_new back to set_fmt Now the core has been migrated across to the new direct clock specification we can move the drivers back to the normal set_fmt callback. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220519154318.2153729-54-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/ux500/ux500_msp_dai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c index cd6c4bdf5041..851c3b8473fd 100644 --- a/sound/soc/ux500/ux500_msp_dai.c +++ b/sound/soc/ux500/ux500_msp_dai.c @@ -707,7 +707,7 @@ static int ux500_msp_dai_probe(struct snd_soc_dai *dai) static const struct snd_soc_dai_ops ux500_msp_dai_ops[] = { { .set_sysclk = ux500_msp_dai_set_dai_sysclk, - .set_fmt_new = ux500_msp_dai_set_dai_fmt, + .set_fmt = ux500_msp_dai_set_dai_fmt, .set_tdm_slot = ux500_msp_dai_set_tdm_slot, .startup = ux500_msp_dai_startup, .shutdown = ux500_msp_dai_shutdown, -- cgit v1.2.3 From 58e23e21d18532aaa404e1db87ec92762e1fecd5 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 May 2022 16:43:16 +0100 Subject: ASoC: xtensa: Rename set_fmt_new back to set_fmt Now the core has been migrated across to the new direct clock specification we can move the drivers back to the normal set_fmt callback. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220519154318.2153729-55-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/xtensa/xtfpga-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/xtensa/xtfpga-i2s.c b/sound/soc/xtensa/xtfpga-i2s.c index 72935f491901..8bd121546032 100644 --- a/sound/soc/xtensa/xtfpga-i2s.c +++ b/sound/soc/xtensa/xtfpga-i2s.c @@ -487,7 +487,7 @@ static const struct snd_soc_component_driver xtfpga_i2s_component = { static const struct snd_soc_dai_ops xtfpga_i2s_dai_ops = { .startup = xtfpga_i2s_startup, .hw_params = xtfpga_i2s_hw_params, - .set_fmt_new = xtfpga_i2s_set_fmt, + .set_fmt = xtfpga_i2s_set_fmt, }; static struct snd_soc_dai_driver xtfpga_i2s_dai[] = { -- cgit v1.2.3 From 19423951a4b5c4f0ca107d6a4bed23f3f63718ca Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 May 2022 16:43:17 +0100 Subject: ASoC: soc-dai: Remove set_fmt_new callback Now the behaviour of the core and all drivers is updated to the new direct clock specification the temporary set_fmt_new callback can be completely removed. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220519154318.2153729-56-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/soc-dai.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 996712f4d9bf..d530e8c2b77b 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -208,11 +208,7 @@ int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { int ret = -ENOTSUPP; - if (dai->driver->ops && - dai->driver->ops->set_fmt_new) - ret = dai->driver->ops->set_fmt_new(dai, fmt); - else if (dai->driver->ops && - dai->driver->ops->set_fmt) + if (dai->driver->ops && dai->driver->ops->set_fmt) ret = dai->driver->ops->set_fmt(dai, fmt); return soc_dai_ret(dai, ret); -- cgit v1.2.3 From 28086d05ada6d03daa886aad0e469854b811311c Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 May 2022 16:43:18 +0100 Subject: ASoC: simple-card-utils: Move snd_soc_component_is_codec to be local The helper function snd_soc_component_is_codec is based off the presence of the non_legacy_dai_naming flag. This isn't super robust as CPU side components may also specify this flag, and indeed the kernel already contains a couple that do. After componentisation there isn't really a totally robust solution to identifying what is a CODEC driver, without introducing a flag specifically for that purpose, and really the desirable direction to move in is that the distinction doesn't matter. This patch does two things to try to mitigate these problems. Firstly, now that all the other users of the helper function have been removed, it makes the helper function local to the driver rather, than being part of the core. This should help to discourage any new code from being created that depends on the CODEC driver distinction. Secondly, it updates the helper function itself to use the endianness flag rather than the non_legacy_dai_naming flag. The endianness flag is definitely invalid on a CPU side component, so it a more reliable indicator that the device is definitely a CODEC. The vast majority of buses require the CODEC to set the endianness flag, so the number of corner cases should be fairly minimal. It is worth noting that CODECs sending audio over SPI, or built into the CPU CODECs are potential corner cases, however the hope is that in most cases those types of devices do not consitute a simple audio card. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220519154318.2153729-57-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/generic/simple-card-utils.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 539d7f081bd7..50a982708933 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -513,6 +513,11 @@ static int asoc_simple_init_dai(struct snd_soc_dai *dai, return 0; } +static inline int asoc_simple_component_is_codec(struct snd_soc_component *component) +{ + return component->driver->endianness; +} + static int asoc_simple_init_dai_link_params(struct snd_soc_pcm_runtime *rtd, struct simple_dai_props *dai_props) { @@ -524,7 +529,7 @@ static int asoc_simple_init_dai_link_params(struct snd_soc_pcm_runtime *rtd, /* Only Codecs */ for_each_rtd_components(rtd, i, component) { - if (!snd_soc_component_is_codec(component)) + if (!asoc_simple_component_is_codec(component)) return 0; } -- cgit v1.2.3 From 60391d788a221f1866492a71929483790b772676 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Uwe=20Kleine-K=C3=B6nig?= Date: Fri, 3 Jun 2022 16:05:10 +0200 Subject: ASoC: ak4642: Drop no-op remove function MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit A remove callback that just returns 0 is equivalent to no callback at all as can be seen in i2c_device_remove(). So simplify accordingly. Signed-off-by: Uwe Kleine-König Link: https://lore.kernel.org/r/20220603140513.131142-2-u.kleine-koenig@pengutronix.de Signed-off-by: Mark Brown --- sound/soc/codecs/ak4613.c | 6 ------ 1 file changed, 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c index 55e773f92122..648208d40d06 100644 --- a/sound/soc/codecs/ak4613.c +++ b/sound/soc/codecs/ak4613.c @@ -919,18 +919,12 @@ static int ak4613_i2c_probe(struct i2c_client *i2c) &ak4613_dai, 1); } -static int ak4613_i2c_remove(struct i2c_client *client) -{ - return 0; -} - static struct i2c_driver ak4613_i2c_driver = { .driver = { .name = "ak4613-codec", .of_match_table = ak4613_of_match, }, .probe_new = ak4613_i2c_probe, - .remove = ak4613_i2c_remove, .id_table = ak4613_i2c_id, }; -- cgit v1.2.3 From 8a291eebeb633316edad2e80537a3c7df83ee8dc Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Uwe=20Kleine-K=C3=B6nig?= Date: Fri, 3 Jun 2022 16:05:11 +0200 Subject: ASoC: da7219: Drop no-op remove function MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit A remove callback that just returns 0 is equivalent to no callback at all as can be seen in i2c_device_remove(). So simplify accordingly. Signed-off-by: Uwe Kleine-König Link: https://lore.kernel.org/r/20220603140513.131142-3-u.kleine-koenig@pengutronix.de Signed-off-by: Mark Brown --- sound/soc/codecs/da7219.c | 6 ------ 1 file changed, 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 7fdef38ed8cd..c18f76f370fc 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -2693,11 +2693,6 @@ static int da7219_i2c_probe(struct i2c_client *i2c) return ret; } -static int da7219_i2c_remove(struct i2c_client *client) -{ - return 0; -} - static const struct i2c_device_id da7219_i2c_id[] = { { "da7219", }, { } @@ -2711,7 +2706,6 @@ static struct i2c_driver da7219_i2c_driver = { .acpi_match_table = ACPI_PTR(da7219_acpi_match), }, .probe_new = da7219_i2c_probe, - .remove = da7219_i2c_remove, .id_table = da7219_i2c_id, }; -- cgit v1.2.3 From 3cce931a5e4487f7339be559e2ea032478be021a Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Uwe=20Kleine-K=C3=B6nig?= Date: Fri, 3 Jun 2022 16:05:12 +0200 Subject: ASoC: lm49453: Drop no-op remove function MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit A remove callback that just returns 0 is equivalent to no callback at all as can be seen in i2c_device_remove(). So simplify accordingly. Signed-off-by: Uwe Kleine-König Link: https://lore.kernel.org/r/20220603140513.131142-4-u.kleine-koenig@pengutronix.de Signed-off-by: Mark Brown --- sound/soc/codecs/lm49453.c | 6 ------ 1 file changed, 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index bd0078e4499b..c4900ada8618 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -1442,11 +1442,6 @@ static int lm49453_i2c_probe(struct i2c_client *i2c) return ret; } -static int lm49453_i2c_remove(struct i2c_client *client) -{ - return 0; -} - static const struct i2c_device_id lm49453_i2c_id[] = { { "lm49453", 0 }, { } @@ -1458,7 +1453,6 @@ static struct i2c_driver lm49453_i2c_driver = { .name = "lm49453", }, .probe_new = lm49453_i2c_probe, - .remove = lm49453_i2c_remove, .id_table = lm49453_i2c_id, }; -- cgit v1.2.3 From fb68cb963bb78380166a98beea593d20b956e4c3 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Uwe=20Kleine-K=C3=B6nig?= Date: Fri, 3 Jun 2022 16:05:13 +0200 Subject: ASoC: da732x: Drop no-op remove function MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit A remove callback that just returns 0 is equivalent to no callback at all as can be seen in i2c_device_remove(). So simplify accordingly. Signed-off-by: Uwe Kleine-König Link: https://lore.kernel.org/r/20220603140513.131142-5-u.kleine-koenig@pengutronix.de Signed-off-by: Mark Brown --- sound/soc/codecs/da732x.c | 6 ------ 1 file changed, 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index f14cddf23f42..3f1cfee10df3 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -1546,11 +1546,6 @@ err: return ret; } -static int da732x_i2c_remove(struct i2c_client *client) -{ - return 0; -} - static const struct i2c_device_id da732x_i2c_id[] = { { "da7320", 0}, { } @@ -1562,7 +1557,6 @@ static struct i2c_driver da732x_i2c_driver = { .name = "da7320", }, .probe_new = da732x_i2c_probe, - .remove = da732x_i2c_remove, .id_table = da732x_i2c_id, }; -- cgit v1.2.3 From 94e0bc317ad241c022a6bb311b3a28b4d51ea8b6 Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Wed, 25 May 2022 14:16:30 +0100 Subject: ASoC: cs35l41: Move cs35l41 exit hibernate function into shared code CS35L41 HDA Driver will support hibernation using DSP firmware, move the exit hibernate function into shared code so this can be reused. Acked-by: Charles Keepax Signed-off-by: Stefan Binding Signed-off-by: Vitaly Rodionov Link: https://lore.kernel.org/r/20220525131638.5512-10-vitalyr@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs35l41-lib.c | 60 +++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs35l41.c | 61 +----------------------------------------- 2 files changed, 61 insertions(+), 60 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs35l41-lib.c b/sound/soc/codecs/cs35l41-lib.c index 6d3070ea9e06..cc5366c8bdd6 100644 --- a/sound/soc/codecs/cs35l41-lib.c +++ b/sound/soc/codecs/cs35l41-lib.c @@ -1321,6 +1321,66 @@ int cs35l41_write_fs_errata(struct device *dev, struct regmap *regmap) } EXPORT_SYMBOL_GPL(cs35l41_write_fs_errata); +static void cs35l41_wait_for_pwrmgt_sts(struct device *dev, struct regmap *regmap) +{ + const int pwrmgt_retries = 10; + unsigned int sts; + int i, ret; + + for (i = 0; i < pwrmgt_retries; i++) { + ret = regmap_read(regmap, CS35L41_PWRMGT_STS, &sts); + if (ret) + dev_err(dev, "Failed to read PWRMGT_STS: %d\n", ret); + else if (!(sts & CS35L41_WR_PEND_STS_MASK)) + return; + + udelay(20); + } + + dev_err(dev, "Timed out reading PWRMGT_STS\n"); +} + +int cs35l41_exit_hibernate(struct device *dev, struct regmap *regmap) +{ + const int wake_retries = 20; + const int sleep_retries = 5; + int ret, i, j; + + for (i = 0; i < sleep_retries; i++) { + dev_dbg(dev, "Exit hibernate\n"); + + for (j = 0; j < wake_retries; j++) { + ret = cs35l41_set_cspl_mbox_cmd(dev, regmap, + CSPL_MBOX_CMD_OUT_OF_HIBERNATE); + if (!ret) + break; + + usleep_range(100, 200); + } + + if (j < wake_retries) { + dev_dbg(dev, "Wake success at cycle: %d\n", j); + return 0; + } + + dev_err(dev, "Wake failed, re-enter hibernate: %d\n", ret); + + cs35l41_wait_for_pwrmgt_sts(dev, regmap); + regmap_write(regmap, CS35L41_WAKESRC_CTL, 0x0088); + + cs35l41_wait_for_pwrmgt_sts(dev, regmap); + regmap_write(regmap, CS35L41_WAKESRC_CTL, 0x0188); + + cs35l41_wait_for_pwrmgt_sts(dev, regmap); + regmap_write(regmap, CS35L41_PWRMGT_CTL, 0x3); + } + + dev_err(dev, "Timed out waking device\n"); + + return -ETIMEDOUT; +} +EXPORT_SYMBOL_GPL(cs35l41_exit_hibernate); + MODULE_DESCRIPTION("CS35L41 library"); MODULE_AUTHOR("David Rhodes, Cirrus Logic Inc, "); MODULE_AUTHOR("Lucas Tanure, Cirrus Logic Inc, "); diff --git a/sound/soc/codecs/cs35l41.c b/sound/soc/codecs/cs35l41.c index 3e68a07a3c8e..be7d02517739 100644 --- a/sound/soc/codecs/cs35l41.c +++ b/sound/soc/codecs/cs35l41.c @@ -1351,65 +1351,6 @@ static int __maybe_unused cs35l41_runtime_suspend(struct device *dev) return 0; } -static void cs35l41_wait_for_pwrmgt_sts(struct cs35l41_private *cs35l41) -{ - const int pwrmgt_retries = 10; - unsigned int sts; - int i, ret; - - for (i = 0; i < pwrmgt_retries; i++) { - ret = regmap_read(cs35l41->regmap, CS35L41_PWRMGT_STS, &sts); - if (ret) - dev_err(cs35l41->dev, "Failed to read PWRMGT_STS: %d\n", ret); - else if (!(sts & CS35L41_WR_PEND_STS_MASK)) - return; - - udelay(20); - } - - dev_err(cs35l41->dev, "Timed out reading PWRMGT_STS\n"); -} - -static int cs35l41_exit_hibernate(struct cs35l41_private *cs35l41) -{ - const int wake_retries = 20; - const int sleep_retries = 5; - int ret, i, j; - - for (i = 0; i < sleep_retries; i++) { - dev_dbg(cs35l41->dev, "Exit hibernate\n"); - - for (j = 0; j < wake_retries; j++) { - ret = cs35l41_set_cspl_mbox_cmd(cs35l41->dev, cs35l41->regmap, - CSPL_MBOX_CMD_OUT_OF_HIBERNATE); - if (!ret) - break; - - usleep_range(100, 200); - } - - if (j < wake_retries) { - dev_dbg(cs35l41->dev, "Wake success at cycle: %d\n", j); - return 0; - } - - dev_err(cs35l41->dev, "Wake failed, re-enter hibernate: %d\n", ret); - - cs35l41_wait_for_pwrmgt_sts(cs35l41); - regmap_write(cs35l41->regmap, CS35L41_WAKESRC_CTL, 0x0088); - - cs35l41_wait_for_pwrmgt_sts(cs35l41); - regmap_write(cs35l41->regmap, CS35L41_WAKESRC_CTL, 0x0188); - - cs35l41_wait_for_pwrmgt_sts(cs35l41); - regmap_write(cs35l41->regmap, CS35L41_PWRMGT_CTL, 0x3); - } - - dev_err(cs35l41->dev, "Timed out waking device\n"); - - return -ETIMEDOUT; -} - static int __maybe_unused cs35l41_runtime_resume(struct device *dev) { struct cs35l41_private *cs35l41 = dev_get_drvdata(dev); @@ -1422,7 +1363,7 @@ static int __maybe_unused cs35l41_runtime_resume(struct device *dev) regcache_cache_only(cs35l41->regmap, false); - ret = cs35l41_exit_hibernate(cs35l41); + ret = cs35l41_exit_hibernate(cs35l41->dev, cs35l41->regmap); if (ret) return ret; -- cgit v1.2.3 From e341efc308e5374ded6b471f9e1ec01450bcc93e Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Wed, 25 May 2022 14:16:32 +0100 Subject: ASoC: cs35l41: Add common cs35l41 enter hibernate function Since the CS35L41 HDA driver also support hibernation, it makes sense to move code from the ASoC driver to enter hibernation into common code. Since HDA must support laptops which do not support hibernation due to lack of external boost GPIO it is necessary to ensure the function returns an error when an unsupported boost type is in use. Acked-by: Charles Keepax Signed-off-by: Stefan Binding Signed-off-by: Vitaly Rodionov Link: https://lore.kernel.org/r/20220525131638.5512-12-vitalyr@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs35l41-lib.c | 19 +++++++++++++++++++ sound/soc/codecs/cs35l41.c | 10 +--------- 2 files changed, 20 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs35l41-lib.c b/sound/soc/codecs/cs35l41-lib.c index cc5366c8bdd6..10b754481ca2 100644 --- a/sound/soc/codecs/cs35l41-lib.c +++ b/sound/soc/codecs/cs35l41-lib.c @@ -1321,6 +1321,25 @@ int cs35l41_write_fs_errata(struct device *dev, struct regmap *regmap) } EXPORT_SYMBOL_GPL(cs35l41_write_fs_errata); +int cs35l41_enter_hibernate(struct device *dev, struct regmap *regmap, + enum cs35l41_boost_type b_type) +{ + if (!cs35l41_safe_reset(regmap, b_type)) { + dev_dbg(dev, "System does not support Suspend\n"); + return -EINVAL; + } + + dev_dbg(dev, "Enter hibernate\n"); + regmap_write(regmap, CS35L41_WAKESRC_CTL, 0x0088); + regmap_write(regmap, CS35L41_WAKESRC_CTL, 0x0188); + + // Don't wait for ACK since bus activity would wake the device + regmap_write(regmap, CS35L41_DSP_VIRT1_MBOX_1, CSPL_MBOX_CMD_HIBERNATE); + + return 0; +} +EXPORT_SYMBOL_GPL(cs35l41_enter_hibernate); + static void cs35l41_wait_for_pwrmgt_sts(struct device *dev, struct regmap *regmap) { const int pwrmgt_retries = 10; diff --git a/sound/soc/codecs/cs35l41.c b/sound/soc/codecs/cs35l41.c index be7d02517739..a115ea35b92d 100644 --- a/sound/soc/codecs/cs35l41.c +++ b/sound/soc/codecs/cs35l41.c @@ -1335,15 +1335,7 @@ static int __maybe_unused cs35l41_runtime_suspend(struct device *dev) if (!cs35l41->dsp.preloaded || !cs35l41->dsp.cs_dsp.running) return 0; - dev_dbg(cs35l41->dev, "Enter hibernate\n"); - - cs35l41_safe_reset(cs35l41->regmap, cs35l41->hw_cfg.bst_type); - regmap_write(cs35l41->regmap, CS35L41_WAKESRC_CTL, 0x0088); - regmap_write(cs35l41->regmap, CS35L41_WAKESRC_CTL, 0x0188); - - // Don't wait for ACK since bus activity would wake the device - regmap_write(cs35l41->regmap, CS35L41_DSP_VIRT1_MBOX_1, - CSPL_MBOX_CMD_HIBERNATE); + cs35l41_enter_hibernate(dev, cs35l41->regmap, cs35l41->hw_cfg.bst_type); regcache_cache_only(cs35l41->regmap, true); regcache_mark_dirty(cs35l41->regmap); -- cgit v1.2.3 From 97076475e2fdf471348b9ce73215cdbceeb4390f Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Wed, 25 May 2022 14:16:31 +0100 Subject: ASoC: cs35l41: Do not print error when waking from hibernation When waking from hibernation, it is possible for the function which sends the wake command to fail initially, but after a retry it will succeed. There is no need to print an error if the initial attempts fail. Signed-off-by: Stefan Binding Signed-off-by: Vitaly Rodionov Acked-by: Charles Keepax Link: https://lore.kernel.org/r/20220525131638.5512-11-vitalyr@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs35l41-lib.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs35l41-lib.c b/sound/soc/codecs/cs35l41-lib.c index 10b754481ca2..0c7d1c791279 100644 --- a/sound/soc/codecs/cs35l41-lib.c +++ b/sound/soc/codecs/cs35l41-lib.c @@ -1302,7 +1302,8 @@ int cs35l41_set_cspl_mbox_cmd(struct device *dev, struct regmap *regmap, return 0; } - dev_err(dev, "Failed to set mailbox cmd %u (status %u)\n", cmd, sts); + if (cmd != CSPL_MBOX_CMD_OUT_OF_HIBERNATE) + dev_err(dev, "Failed to set mailbox cmd %u (status %u)\n", cmd, sts); return -ENOMSG; } -- cgit v1.2.3 From 0439eb4d94e0fc17c6dd3829fabd88c11773381d Mon Sep 17 00:00:00 2001 From: V sujith kumar Reddy Date: Tue, 31 May 2022 17:38:11 +0530 Subject: ASoC: amd: acp: Add support for nau8825 and max98360 card We have new platform with nau8825 as a primary codec and max98360 as an amp codec. Add machine struct to register sof audio based sound card on such Chrome machine. Signed-off-by: V sujith kumar Reddy Reviewed-by: Akihiko Odaki Link: https://lore.kernel.org/r/20220531120813.47116-2-Vsujithkumar.Reddy@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp-config.c | 21 +++++ sound/soc/amd/acp/Kconfig | 1 + sound/soc/amd/acp/acp-mach-common.c | 166 ++++++++++++++++++++++++++++++++++-- sound/soc/amd/acp/acp-mach.h | 3 + sound/soc/amd/acp/acp-sof-mach.c | 15 ++++ sound/soc/amd/mach-config.h | 1 + 6 files changed, 201 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/amd/acp-config.c b/sound/soc/amd/acp-config.c index 5cbc82eca4c9..3b9f851bf50d 100644 --- a/sound/soc/amd/acp-config.c +++ b/sound/soc/amd/acp-config.c @@ -130,4 +130,25 @@ struct snd_soc_acpi_mach snd_soc_acpi_amd_sof_machines[] = { }; EXPORT_SYMBOL(snd_soc_acpi_amd_sof_machines); +struct snd_soc_acpi_mach snd_soc_acpi_amd_rmb_sof_machines[] = { + { + .id = "AMDI1019", + .drv_name = "rmb-dsp", + .pdata = &acp_quirk_data, + .fw_filename = "sof-rmb.ri", + .sof_tplg_filename = "sof-acp-rmb.tplg", + }, + { + .id = "10508825", + .drv_name = "nau8825-max", + .pdata = &acp_quirk_data, + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &_max, + .fw_filename = "sof-rmb.ri", + .sof_tplg_filename = "sof-rmb-nau8825-max98360.tplg", + }, + {}, +}; +EXPORT_SYMBOL(snd_soc_acpi_amd_rmb_sof_machines); + MODULE_LICENSE("Dual BSD/GPL"); diff --git a/sound/soc/amd/acp/Kconfig b/sound/soc/amd/acp/Kconfig index 9dae2719084c..7e56d2644105 100644 --- a/sound/soc/amd/acp/Kconfig +++ b/sound/soc/amd/acp/Kconfig @@ -49,6 +49,7 @@ config SND_SOC_AMD_MACH_COMMON select SND_SOC_RT1019 select SND_SOC_MAX98357A select SND_SOC_RT5682S + select SND_SOC_NAU8825 help This option enables common Machine driver module for ACP. diff --git a/sound/soc/amd/acp/acp-mach-common.c b/sound/soc/amd/acp/acp-mach-common.c index 6ae454bf60af..a03b396d96bb 100644 --- a/sound/soc/amd/acp/acp-mach-common.c +++ b/sound/soc/amd/acp/acp-mach-common.c @@ -24,6 +24,7 @@ #include "../../codecs/rt5682.h" #include "../../codecs/rt1019.h" #include "../../codecs/rt5682s.h" +#include "../../codecs/nau8825.h" #include "acp-mach.h" #define PCO_PLAT_CLK 48000000 @@ -175,7 +176,8 @@ static void acp_card_shutdown(struct snd_pcm_substream *substream) struct snd_soc_card *card = rtd->card; struct acp_card_drvdata *drvdata = card->drvdata; - clk_disable_unprepare(drvdata->wclk); + if (!drvdata->soc_mclk) + clk_disable_unprepare(drvdata->wclk); } static const struct snd_soc_ops acp_card_rt5682_ops = { @@ -363,7 +365,7 @@ static int acp_card_amp_startup(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); struct snd_soc_card *card = rtd->card; struct acp_card_drvdata *drvdata = card->drvdata; - int ret; + int ret = 0; runtime->hw.channels_max = DUAL_CHANNEL; snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, @@ -371,10 +373,13 @@ static int acp_card_amp_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); - ret = acp_clk_enable(drvdata); - if (ret < 0) - dev_err(rtd->card->dev, "Failed to enable AMP clk: %d\n", ret); - + if (!drvdata->soc_mclk) { + ret = acp_clk_enable(drvdata); + if (ret < 0) { + dev_err(rtd->card->dev, "Failed to enable AMP clk: %d\n", ret); + return ret; + } + } return ret; } @@ -409,6 +414,104 @@ static const struct snd_soc_ops acp_card_maxim_ops = { .shutdown = acp_card_shutdown, }; +/* Declare nau8825 codec components */ +SND_SOC_DAILINK_DEF(nau8825, + DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10508825:00", "nau8825-hifi"))); + +static const struct snd_soc_dapm_route nau8825_map[] = { + { "Headphone Jack", NULL, "HPOL" }, + { "Headphone Jack", NULL, "HPOR" }, +}; + +static int acp_card_nau8825_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + struct acp_card_drvdata *drvdata = card->drvdata; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_component *component = codec_dai->component; + unsigned int fmt; + int ret; + + dev_info(rtd->dev, "codec dai name = %s\n", codec_dai->name); + + if (drvdata->hs_codec_id != NAU8825) + return -EINVAL; + + if (drvdata->soc_mclk) + fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBC_CFC; + else + fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBP_CFP; + + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) { + dev_err(rtd->card->dev, "Failed to set dai fmt: %d\n", ret); + return ret; + } + ret = snd_soc_card_jack_new(card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_LINEOUT | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3, + &pco_jack); + if (ret) { + dev_err(card->dev, "HP jack creation failed %d\n", ret); + return ret; + } + + snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); + snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_2, KEY_VOLUMEUP); + snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); + + ret = snd_soc_component_set_jack(component, &pco_jack, NULL); + if (ret) { + dev_err(rtd->dev, "Headset Jack call-back failed: %d\n", ret); + return ret; + } + + return snd_soc_dapm_add_routes(&rtd->card->dapm, nau8825_map, ARRAY_SIZE(nau8825_map)); +} + +static int acp_nau8825_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_FLL_FS, + (48000 * 256), SND_SOC_CLOCK_IN); + if (ret < 0) + dev_err(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", ret); + + ret = snd_soc_dai_set_pll(codec_dai, 0, 0, params_rate(params), + params_rate(params) * 256); + if (ret < 0) { + dev_err(rtd->dev, "can't set FLL: %d\n", ret); + return ret; + } + + return ret; +} + +static int acp_nau8825_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw.channels_max = 2; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + + runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; + snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); + return 0; +} + +static const struct snd_soc_ops acp_card_nau8825_ops = { + .startup = acp_nau8825_startup, + .hw_params = acp_nau8825_hw_params, +}; + /* Declare DMIC codec components */ SND_SOC_DAILINK_DEF(dmic_codec, DAILINK_COMP_ARRAY(COMP_CODEC("dmic-codec", "dmic-hifi"))); @@ -437,6 +540,8 @@ SND_SOC_DAILINK_DEF(i2s_sp, DAILINK_COMP_ARRAY(COMP_CPU("acp-i2s-sp"))); SND_SOC_DAILINK_DEF(sof_sp, DAILINK_COMP_ARRAY(COMP_CPU("acp-sof-sp"))); +SND_SOC_DAILINK_DEF(sof_hs, + DAILINK_COMP_ARRAY(COMP_CPU("acp-sof-hs"))); SND_SOC_DAILINK_DEF(sof_dmic, DAILINK_COMP_ARRAY(COMP_CPU("acp-sof-dmic"))); SND_SOC_DAILINK_DEF(pdm_dmic, @@ -491,6 +596,31 @@ int acp_sofdsp_dai_links_create(struct snd_soc_card *card) i++; } + if (drv_data->hs_cpu_id == I2S_HS) { + links[i].name = "acp-headset-codec"; + links[i].id = HEADSET_BE_ID; + links[i].cpus = sof_hs; + links[i].num_cpus = ARRAY_SIZE(sof_hs); + links[i].platforms = sof_component; + links[i].num_platforms = ARRAY_SIZE(sof_component); + links[i].dpcm_playback = 1; + links[i].dpcm_capture = 1; + links[i].nonatomic = true; + links[i].no_pcm = 1; + if (!drv_data->hs_codec_id) { + /* Use dummy codec if codec id not specified */ + links[i].codecs = dummy_codec; + links[i].num_codecs = ARRAY_SIZE(dummy_codec); + } + if (drv_data->hs_codec_id == NAU8825) { + links[i].codecs = nau8825; + links[i].num_codecs = ARRAY_SIZE(nau8825); + links[i].init = acp_card_nau8825_init; + links[i].ops = &acp_card_nau8825_ops; + } + i++; + } + if (drv_data->amp_cpu_id == I2S_SP) { links[i].name = "acp-amp-codec"; links[i].id = AMP_BE_ID; @@ -523,6 +653,30 @@ int acp_sofdsp_dai_links_create(struct snd_soc_card *card) i++; } + if (drv_data->amp_cpu_id == I2S_HS) { + links[i].name = "acp-amp-codec"; + links[i].id = AMP_BE_ID; + links[i].cpus = sof_hs; + links[i].num_cpus = ARRAY_SIZE(sof_hs); + links[i].platforms = sof_component; + links[i].num_platforms = ARRAY_SIZE(sof_component); + links[i].dpcm_playback = 1; + links[i].nonatomic = true; + links[i].no_pcm = 1; + if (!drv_data->amp_codec_id) { + /* Use dummy codec if codec id not specified */ + links[i].codecs = dummy_codec; + links[i].num_codecs = ARRAY_SIZE(dummy_codec); + } + if (drv_data->amp_codec_id == MAX98360A) { + links[i].codecs = max98360a; + links[i].num_codecs = ARRAY_SIZE(max98360a); + links[i].ops = &acp_card_maxim_ops; + links[i].init = acp_card_maxim_init; + } + i++; + } + if (drv_data->dmic_cpu_id == DMIC) { links[i].name = "acp-dmic-codec"; links[i].id = DMIC_BE_ID; diff --git a/sound/soc/amd/acp/acp-mach.h b/sound/soc/amd/acp/acp-mach.h index 5dc47cfbff10..c95ee1c52eb1 100644 --- a/sound/soc/amd/acp/acp-mach.h +++ b/sound/soc/amd/acp/acp-mach.h @@ -26,6 +26,7 @@ enum be_id { enum cpu_endpoints { NONE = 0, + I2S_HS, I2S_SP, I2S_BT, DMIC, @@ -37,6 +38,7 @@ enum codec_endpoints { RT1019, MAX98360A, RT5682S, + NAU8825, }; struct acp_card_drvdata { @@ -49,6 +51,7 @@ struct acp_card_drvdata { unsigned int dai_fmt; struct clk *wclk; struct clk *bclk; + bool soc_mclk; }; int acp_sofdsp_dai_links_create(struct snd_soc_card *card); diff --git a/sound/soc/amd/acp/acp-sof-mach.c b/sound/soc/amd/acp/acp-sof-mach.c index d1531cdab110..adbae809f2aa 100644 --- a/sound/soc/amd/acp/acp-sof-mach.c +++ b/sound/soc/amd/acp/acp-sof-mach.c @@ -56,6 +56,16 @@ static struct acp_card_drvdata sof_rt5682s_max_data = { .dmic_codec_id = DMIC, }; +static struct acp_card_drvdata sof_nau8825_data = { + .hs_cpu_id = I2S_HS, + .amp_cpu_id = I2S_HS, + .dmic_cpu_id = DMIC, + .hs_codec_id = NAU8825, + .amp_codec_id = MAX98360A, + .dmic_codec_id = DMIC, + .soc_mclk = true, +}; + static const struct snd_kcontrol_new acp_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone Jack"), SOC_DAPM_PIN_SWITCH("Headset Mic"), @@ -124,6 +134,10 @@ static const struct platform_device_id board_ids[] = { .name = "rt5682s-rt1019", .driver_data = (kernel_ulong_t)&sof_rt5682s_rt1019_data }, + { + .name = "nau8825-max", + .driver_data = (kernel_ulong_t)&sof_nau8825_data + }, { } }; static struct platform_driver acp_asoc_audio = { @@ -143,4 +157,5 @@ MODULE_ALIAS("platform:rt5682-rt1019"); MODULE_ALIAS("platform:rt5682-max"); MODULE_ALIAS("platform:rt5682s-max"); MODULE_ALIAS("platform:rt5682s-rt1019"); +MODULE_ALIAS("platform:nau8825-max"); MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/amd/mach-config.h b/sound/soc/amd/mach-config.h index 0a54567a2841..7b4c625da40d 100644 --- a/sound/soc/amd/mach-config.h +++ b/sound/soc/amd/mach-config.h @@ -19,6 +19,7 @@ #define ACP_PCI_DEV_ID 0x15E2 extern struct snd_soc_acpi_mach snd_soc_acpi_amd_sof_machines[]; +extern struct snd_soc_acpi_mach snd_soc_acpi_amd_rmb_sof_machines[]; struct config_entry { u32 flags; -- cgit v1.2.3 From 4dc6737cfe882765d914fcb88b5eaa14551ffddd Mon Sep 17 00:00:00 2001 From: V sujith kumar Reddy Date: Tue, 31 May 2022 17:38:12 +0530 Subject: ASoC: amd: acp: Add support for rt5682s and rt1019 card with hs instance We have new platform with rt5682s as a primary codec and rt1019 as an amp codec. Add machine struct to register sof audio based sound card on such Chrome machine. Here we are configuring as a soc mclk master and codec slave. Signed-off-by: V sujith kumar Reddy Link: https://lore.kernel.org/r/20220531120813.47116-3-Vsujithkumar.Reddy@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp-config.c | 9 +++++ sound/soc/amd/acp/acp-mach-common.c | 67 +++++++++++++++++++++++++++++++------ sound/soc/amd/acp/acp-sof-mach.c | 15 +++++++++ 3 files changed, 81 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/amd/acp-config.c b/sound/soc/amd/acp-config.c index 3b9f851bf50d..0932473b6394 100644 --- a/sound/soc/amd/acp-config.c +++ b/sound/soc/amd/acp-config.c @@ -147,6 +147,15 @@ struct snd_soc_acpi_mach snd_soc_acpi_amd_rmb_sof_machines[] = { .fw_filename = "sof-rmb.ri", .sof_tplg_filename = "sof-rmb-nau8825-max98360.tplg", }, + { + .id = "RTL5682", + .drv_name = "rt5682s-hs-rt1019", + .pdata = &acp_quirk_data, + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &_rt1019, + .fw_filename = "sof-rmb.ri", + .sof_tplg_filename = "sof-rmb-rt5682s-rt1019.tplg", + }, {}, }; EXPORT_SYMBOL(snd_soc_acpi_amd_rmb_sof_machines); diff --git a/sound/soc/amd/acp/acp-mach-common.c b/sound/soc/amd/acp/acp-mach-common.c index a03b396d96bb..7530cab24bc8 100644 --- a/sound/soc/amd/acp/acp-mach-common.c +++ b/sound/soc/amd/acp/acp-mach-common.c @@ -149,9 +149,14 @@ static int acp_card_hs_startup(struct snd_pcm_substream *substream) struct acp_card_drvdata *drvdata = card->drvdata; struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret; + unsigned int fmt; - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBP_CFP); + if (drvdata->soc_mclk) + fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBC_CFC; + else + fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBP_CFP; + + ret = snd_soc_dai_set_fmt(codec_dai, fmt); if (ret < 0) { dev_err(rtd->card->dev, "Failed to set dai fmt: %d\n", ret); return ret; @@ -162,10 +167,13 @@ static int acp_card_hs_startup(struct snd_pcm_substream *substream) &constraints_channels); snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); - - ret = acp_clk_enable(drvdata); - if (ret < 0) - dev_err(rtd->card->dev, "Failed to enable HS clk: %d\n", ret); + if (!drvdata->soc_mclk) { + ret = acp_clk_enable(drvdata); + if (ret < 0) { + dev_err(rtd->card->dev, "Failed to enable HS clk: %d\n", ret); + return ret; + } + } return ret; } @@ -201,6 +209,7 @@ static int acp_card_rt5682s_init(struct snd_soc_pcm_runtime *rtd) struct acp_card_drvdata *drvdata = card->drvdata; struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); struct snd_soc_component *component = codec_dai->component; + unsigned int fmt; int ret; dev_info(rtd->dev, "codec dai name = %s\n", codec_dai->name); @@ -208,8 +217,12 @@ static int acp_card_rt5682s_init(struct snd_soc_pcm_runtime *rtd) if (drvdata->hs_codec_id != RT5682S) return -EINVAL; - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBP_CFP); + if (drvdata->soc_mclk) + fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBC_CFC; + else + fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBP_CFP; + + ret = snd_soc_dai_set_fmt(codec_dai, fmt); if (ret < 0) { dev_err(rtd->card->dev, "Failed to set dai fmt: %d\n", ret); return ret; @@ -236,8 +249,10 @@ static int acp_card_rt5682s_init(struct snd_soc_pcm_runtime *rtd) return ret; } - drvdata->wclk = clk_get(component->dev, "rt5682-dai-wclk"); - drvdata->bclk = clk_get(component->dev, "rt5682-dai-bclk"); + if (!drvdata->soc_mclk) { + drvdata->wclk = clk_get(component->dev, "rt5682-dai-wclk"); + drvdata->bclk = clk_get(component->dev, "rt5682-dai-bclk"); + } ret = snd_soc_card_jack_new(card, "Headset Jack", SND_JACK_HEADSET | SND_JACK_LINEOUT | @@ -298,6 +313,9 @@ static const struct snd_soc_ops acp_card_dmic_ops = { SND_SOC_DAILINK_DEF(rt1019, DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10EC1019:00", "rt1019-aif"), COMP_CODEC("i2c-10EC1019:01", "rt1019-aif"))); +SND_SOC_DAILINK_DEF(rt1019_1, + DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10EC1019:02", "rt1019-aif"), + COMP_CODEC("i2c-10EC1019:01", "rt1019-aif"))); static const struct snd_soc_dapm_route rt1019_map_lr[] = { { "Left Spk", NULL, "Left SPO" }, @@ -315,6 +333,17 @@ static struct snd_soc_codec_conf rt1019_conf[] = { }, }; +static struct snd_soc_codec_conf rt1019_1_conf[] = { + { + .dlc = COMP_CODEC_CONF("i2c-10EC1019:02"), + .name_prefix = "Left", + }, + { + .dlc = COMP_CODEC_CONF("i2c-10EC1019:01"), + .name_prefix = "Right", + }, +}; + static int acp_card_rt1019_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; @@ -618,6 +647,12 @@ int acp_sofdsp_dai_links_create(struct snd_soc_card *card) links[i].init = acp_card_nau8825_init; links[i].ops = &acp_card_nau8825_ops; } + if (drv_data->hs_codec_id == RT5682S) { + links[i].codecs = rt5682s; + links[i].num_codecs = ARRAY_SIZE(rt5682s); + links[i].init = acp_card_rt5682s_init; + links[i].ops = &acp_card_rt5682s_ops; + } i++; } @@ -674,6 +709,18 @@ int acp_sofdsp_dai_links_create(struct snd_soc_card *card) links[i].ops = &acp_card_maxim_ops; links[i].init = acp_card_maxim_init; } + if (drv_data->amp_codec_id == RT1019) { + links[i].codecs = rt1019; + links[i].num_codecs = ARRAY_SIZE(rt1019); + links[i].ops = &acp_card_rt1019_ops; + links[i].init = acp_card_rt1019_init; + card->codec_conf = rt1019_conf; + card->num_configs = ARRAY_SIZE(rt1019_conf); + links[i].codecs = rt1019_1; + links[i].num_codecs = ARRAY_SIZE(rt1019_1); + card->codec_conf = rt1019_1_conf; + card->num_configs = ARRAY_SIZE(rt1019_1_conf); + } i++; } diff --git a/sound/soc/amd/acp/acp-sof-mach.c b/sound/soc/amd/acp/acp-sof-mach.c index adbae809f2aa..f19f064a7527 100644 --- a/sound/soc/amd/acp/acp-sof-mach.c +++ b/sound/soc/amd/acp/acp-sof-mach.c @@ -66,6 +66,16 @@ static struct acp_card_drvdata sof_nau8825_data = { .soc_mclk = true, }; +static struct acp_card_drvdata sof_rt5682s_hs_rt1019_data = { + .hs_cpu_id = I2S_HS, + .amp_cpu_id = I2S_HS, + .dmic_cpu_id = DMIC, + .hs_codec_id = RT5682S, + .amp_codec_id = RT1019, + .dmic_codec_id = DMIC, + .soc_mclk = true, +}; + static const struct snd_kcontrol_new acp_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone Jack"), SOC_DAPM_PIN_SWITCH("Headset Mic"), @@ -138,6 +148,10 @@ static const struct platform_device_id board_ids[] = { .name = "nau8825-max", .driver_data = (kernel_ulong_t)&sof_nau8825_data }, + { + .name = "rt5682s-hs-rt1019", + .driver_data = (kernel_ulong_t)&sof_rt5682s_hs_rt1019_data + }, { } }; static struct platform_driver acp_asoc_audio = { @@ -158,4 +172,5 @@ MODULE_ALIAS("platform:rt5682-max"); MODULE_ALIAS("platform:rt5682s-max"); MODULE_ALIAS("platform:rt5682s-rt1019"); MODULE_ALIAS("platform:nau8825-max"); +MODULE_ALIAS("platform:rt5682s-hs-rt1019"); MODULE_LICENSE("GPL v2"); -- cgit v1.2.3 From 8dc51d009fad7aba0575e0eb4b684d25c0f01f37 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 2 Jun 2022 15:05:29 +0200 Subject: ASoC: ssm2518: Use modern ASoC DAI format terminology As part of moving to remove the old style defines for the bus clocks update the ssm2518 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown Link: https://lore.kernel.org/r/20220602130531.3552275-2-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/ssm2518.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c index 83acbdbb8e0d..012f209e76e9 100644 --- a/sound/soc/codecs/ssm2518.c +++ b/sound/soc/codecs/ssm2518.c @@ -409,8 +409,8 @@ static int ssm2518_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) bool invert_fclk; int ret; - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBC_CFC: break; default: return -EINVAL; -- cgit v1.2.3 From 0160e8835fab4d4a15abefe7509d0397890c0ffd Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 2 Jun 2022 15:05:30 +0200 Subject: ASoC: ssm2602: Use modern ASoC DAI format terminology As part of moving to remove the old style defines for the bus clocks update the ssm2602 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown Link: https://lore.kernel.org/r/20220602130531.3552275-3-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/ssm2602.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 7964e922b07f..1821854ca0f3 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -411,11 +411,11 @@ static int ssm2602_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int iface = 0; /* set master/slave audio interface */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: iface |= 0x0040; break; - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBC_CFC: break; default: return -EINVAL; -- cgit v1.2.3 From 627a18149250e13409079ffb6936e472c3766f44 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 2 Jun 2022 15:05:31 +0200 Subject: ASoC: ssm4567: Use modern ASoC DAI format terminology As part of moving to remove the old style defines for the bus clocks update the ssm4567 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown Link: https://lore.kernel.org/r/20220602130531.3552275-4-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/ssm4567.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ssm4567.c b/sound/soc/codecs/ssm4567.c index 08ced09ef001..b47321c597d0 100644 --- a/sound/soc/codecs/ssm4567.c +++ b/sound/soc/codecs/ssm4567.c @@ -278,8 +278,8 @@ static int ssm4567_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) unsigned int ctrl1 = 0; bool invert_fclk; - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBC_CFC: break; default: return -EINVAL; -- cgit v1.2.3 From 0511e2ac4e848ceac14b3ac4b476f0e26b48ddb2 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 20 May 2022 16:17:04 -0500 Subject: ASoC: cs35l45: typo in argument definition MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit cppcheck warning: sound/soc/codecs/cs35l45-tables.c:36:49: style:inconclusive: Function 'cs35l45_apply_patch' argument 1 names different: declaration 'cs43l45' definition 'cs35l45'. [funcArgNamesDifferent] Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Link: https://lore.kernel.org/r/20220520211719.607543-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs35l45.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs35l45.h b/sound/soc/codecs/cs35l45.h index 4e266d19cd1c..8f98cbbf6209 100644 --- a/sound/soc/codecs/cs35l45.h +++ b/sound/soc/codecs/cs35l45.h @@ -209,7 +209,7 @@ struct cs35l45_private { extern const struct dev_pm_ops cs35l45_pm_ops; extern const struct regmap_config cs35l45_i2c_regmap; extern const struct regmap_config cs35l45_spi_regmap; -int cs35l45_apply_patch(struct cs35l45_private *cs43l45); +int cs35l45_apply_patch(struct cs35l45_private *cs35l45); unsigned int cs35l45_get_clk_freq_id(unsigned int freq); int cs35l45_probe(struct cs35l45_private *cs35l45); int cs35l45_remove(struct cs35l45_private *cs35l45); -- cgit v1.2.3 From 94f8f2068ed0e3a5e367029f64ed76e6e65d5eb3 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 20 May 2022 16:17:05 -0500 Subject: ASoC: cs42l42: remove redundant test MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit cppcheck warning: sound/soc/codecs/cs42l42.c:1704:28: style: The statement 'if (cs42l42->plug_state!=CS42L42_TS_TRANS) cs42l42->plug_state=CS42L42_TS_TRANS' is logically equivalent to 'cs42l42->plug_state=CS42L42_TS_TRANS'. [duplicateConditionalAssign] if (cs42l42->plug_state != CS42L42_TS_TRANS) ^ sound/soc/codecs/cs42l42.c:1705:25: note: Assignment 'cs42l42->plug_state=CS42L42_TS_TRANS' cs42l42->plug_state = CS42L42_TS_TRANS; ^ Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Link: https://lore.kernel.org/r/20220520211719.607543-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l42.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index 4fade2388797..6ca74c0d46ea 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -1701,8 +1701,7 @@ static irqreturn_t cs42l42_irq_thread(int irq, void *data) break; default: - if (cs42l42->plug_state != CS42L42_TS_TRANS) - cs42l42->plug_state = CS42L42_TS_TRANS; + cs42l42->plug_state = CS42L42_TS_TRANS; } } -- cgit v1.2.3 From cac24a360a6b948ffb75c3d7ccc819064300454c Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 20 May 2022 16:17:06 -0500 Subject: ASoC: wcd-mbhc-v2: remove useless initialization MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit cppcheck warning: sound/soc/codecs/wcd-mbhc-v2.c:1309:17: style: Variable 'clamp_state' is assigned a value that is never used. [unreadVariable] u8 clamp_state = 0; ^ Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Link: https://lore.kernel.org/r/20220520211719.607543-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/wcd-mbhc-v2.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wcd-mbhc-v2.c b/sound/soc/codecs/wcd-mbhc-v2.c index c53c2ef33e1a..31009283e7d4 100644 --- a/sound/soc/codecs/wcd-mbhc-v2.c +++ b/sound/soc/codecs/wcd-mbhc-v2.c @@ -1306,7 +1306,7 @@ exit: static irqreturn_t wcd_mbhc_adc_hs_ins_irq(int irq, void *data) { struct wcd_mbhc *mbhc = data; - u8 clamp_state = 0; + u8 clamp_state; u8 clamp_retry = WCD_MBHC_FAKE_INS_RETRY; /* -- cgit v1.2.3 From 0016361dfcc93a70850c6909fb76f15305dda5ae Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 20 May 2022 16:17:07 -0500 Subject: ASoC: wcd9335: remove redundant tests MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit cppcheck warning: sound/soc/codecs/wcd9335.c:1810:23: style: Condition 'tx_port>=4' is always true [knownConditionTrueFalse] } else if ((tx_port >= 4) && (tx_port < 8)) { ^ sound/soc/codecs/wcd9335.c:1806:15: note: Assuming that condition 'tx_port<4' is not redundant if (tx_port < 4) { ^ sound/soc/codecs/wcd9335.c:1810:23: note: Condition 'tx_port>=4' is always true } else if ((tx_port >= 4) && (tx_port < 8)) { ^ Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Link: https://lore.kernel.org/r/20220520211719.607543-5-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/wcd9335.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index 617a36a89dfe..e1b693048084 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -1807,11 +1807,11 @@ static int wcd9335_set_decimator_rate(struct snd_soc_dai *dai, tx_port_reg = WCD9335_CDC_IF_ROUTER_TX_MUX_CFG0; shift = (tx_port << 1); shift_val = 0x03; - } else if ((tx_port >= 4) && (tx_port < 8)) { + } else if (tx_port < 8) { tx_port_reg = WCD9335_CDC_IF_ROUTER_TX_MUX_CFG1; shift = ((tx_port - 4) << 1); shift_val = 0x03; - } else if ((tx_port >= 8) && (tx_port < 11)) { + } else if (tx_port < 11) { tx_port_reg = WCD9335_CDC_IF_ROUTER_TX_MUX_CFG2; shift = ((tx_port - 8) << 1); shift_val = 0x03; -- cgit v1.2.3 From fb6ed937aaa0703bcdacfe013897d583a6eba365 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 20 May 2022 16:17:08 -0500 Subject: ASoC: Intel: atom: sst: remove useless initialization MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit cppcheck reports an invalid null pointer dereference but there's indeed no need to initialize a loop variable. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Link: https://lore.kernel.org/r/20220520211719.607543-6-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/atom/sst/sst.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/atom/sst/sst.c b/sound/soc/intel/atom/sst/sst.c index 3a42d68c0247..160b50f479fb 100644 --- a/sound/soc/intel/atom/sst/sst.c +++ b/sound/soc/intel/atom/sst/sst.c @@ -114,7 +114,7 @@ static irqreturn_t intel_sst_interrupt_mrfld(int irq, void *context) static irqreturn_t intel_sst_irq_thread_mrfld(int irq, void *context) { struct intel_sst_drv *drv = (struct intel_sst_drv *) context; - struct ipc_post *__msg, *msg = NULL; + struct ipc_post *__msg, *msg; unsigned long irq_flags; spin_lock_irqsave(&drv->rx_msg_lock, irq_flags); -- cgit v1.2.3 From d8af541139fa135a250c5ae743bfec3b49e97c3a Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 20 May 2022 16:17:09 -0500 Subject: ASoC: Intel: atom: sst_ipc: remove redundant test MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit cppcheck warning: sound/soc/intel/atom/sst/sst_ipc.c:344:30: style: Condition 'drv_id' is always true [knownConditionTrueFalse] if (msg_high.part.result && drv_id && !msg_high.part.large) { ^ sound/soc/intel/atom/sst/sst_ipc.c:337:13: note: Assuming that condition 'drv_id==0' is not redundant if (drv_id == SST_ASYNC_DRV_ID) { ^ sound/soc/intel/atom/sst/sst_ipc.c:344:30: note: Condition 'drv_id' is always true if (msg_high.part.result && drv_id && !msg_high.part.large) { ^ Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Link: https://lore.kernel.org/r/20220520211719.607543-7-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/atom/sst/sst_ipc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/atom/sst/sst_ipc.c b/sound/soc/intel/atom/sst/sst_ipc.c index 4e8382097e61..78ea67c7a128 100644 --- a/sound/soc/intel/atom/sst/sst_ipc.c +++ b/sound/soc/intel/atom/sst/sst_ipc.c @@ -341,7 +341,7 @@ void sst_process_reply_mrfld(struct intel_sst_drv *sst_drv_ctx, } /* FW sent short error response for an IPC */ - if (msg_high.part.result && drv_id && !msg_high.part.large) { + if (msg_high.part.result && !msg_high.part.large) { /* 32-bit FW error code in msg_low */ dev_err(sst_drv_ctx->dev, "FW sent error response 0x%x", msg_low); sst_wake_up_block(sst_drv_ctx, msg_high.part.result, -- cgit v1.2.3 From a140785b701d286374ea1b26762f333e4f5e9ee3 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 20 May 2022 16:17:10 -0500 Subject: ASoC: Intel: atom: sst_ipc: remove useless initializations MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit cppcheck throws invalid NULL dereference warnings but there's indeed no need to initialize a loop variable or initialize to NULL before allocating memory. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Link: https://lore.kernel.org/r/20220520211719.607543-8-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/atom/sst/sst_ipc.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/atom/sst/sst_ipc.c b/sound/soc/intel/atom/sst/sst_ipc.c index 78ea67c7a128..4e039c7173d8 100644 --- a/sound/soc/intel/atom/sst/sst_ipc.c +++ b/sound/soc/intel/atom/sst/sst_ipc.c @@ -28,7 +28,7 @@ struct sst_block *sst_create_block(struct intel_sst_drv *ctx, u32 msg_id, u32 drv_id) { - struct sst_block *msg = NULL; + struct sst_block *msg; dev_dbg(ctx->dev, "Enter\n"); msg = kzalloc(sizeof(*msg), GFP_KERNEL); @@ -63,7 +63,7 @@ struct sst_block *sst_create_block(struct intel_sst_drv *ctx, int sst_wake_up_block(struct intel_sst_drv *ctx, int result, u32 drv_id, u32 ipc, void *data, u32 size) { - struct sst_block *block = NULL; + struct sst_block *block; dev_dbg(ctx->dev, "Enter\n"); @@ -91,7 +91,7 @@ int sst_wake_up_block(struct intel_sst_drv *ctx, int result, int sst_free_block(struct intel_sst_drv *ctx, struct sst_block *freed) { - struct sst_block *block = NULL, *__block; + struct sst_block *block, *__block; dev_dbg(ctx->dev, "Enter\n"); spin_lock_bh(&ctx->block_lock); -- cgit v1.2.3 From f6cd55a19f3f46e3d36b1121f844956128c60b6a Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 20 May 2022 16:17:11 -0500 Subject: ASoC: Intel: atom: controls: remove useless initializations MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit cppcheck complains about invalid NULL dereferences but there's indeed no need to initialize loop variables or before allocating memory. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Link: https://lore.kernel.org/r/20220520211719.607543-9-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/atom/sst-atom-controls.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c index 335c32732994..34d63252debf 100644 --- a/sound/soc/intel/atom/sst-atom-controls.c +++ b/sound/soc/intel/atom/sst-atom-controls.c @@ -1328,7 +1328,7 @@ int sst_send_pipe_gains(struct snd_soc_dai *dai, int stream, int mute) { struct sst_data *drv = snd_soc_dai_get_drvdata(dai); struct snd_soc_dapm_widget *w; - struct snd_soc_dapm_path *p = NULL; + struct snd_soc_dapm_path *p; dev_dbg(dai->dev, "enter, dai-name=%s dir=%d\n", dai->name, stream); @@ -1392,7 +1392,7 @@ int sst_send_pipe_gains(struct snd_soc_dai *dai, int stream, int mute) static int sst_fill_module_list(struct snd_kcontrol *kctl, struct snd_soc_dapm_widget *w, int type) { - struct sst_module *module = NULL; + struct sst_module *module; struct snd_soc_component *c = snd_soc_dapm_to_component(w->dapm); struct sst_ids *ids = w->priv; int ret = 0; -- cgit v1.2.3 From 9972773c26125242b467f0062c1fee874c87ae68 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 20 May 2022 16:17:12 -0500 Subject: ASoC: Intel: boards: reset acpi_chan_package MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit cppcheck complains about possible tests of uninitialized 'aif_value' members. This isn't really possible but static analysis cannot know what ACPICA does, so make sure the acpi_chan_package structure is reset prior to use to make the warning go away. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Link: https://lore.kernel.org/r/20220520211719.607543-10-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5640.c | 2 +- sound/soc/intel/boards/bytcr_rt5651.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index ed9fa1728722..2371927fe836 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -1636,7 +1636,7 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) * with the codec driver/pdata are non-existent */ - struct acpi_chan_package chan_package; + struct acpi_chan_package chan_package = { 0 }; /* format specified: 2 64-bit integers */ struct acpi_buffer format = {sizeof("NN"), "NN"}; diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index d467fcaa48ea..03a52b1069a9 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -952,7 +952,7 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) * with the codec driver/pdata are non-existent */ - struct acpi_chan_package chan_package; + struct acpi_chan_package chan_package = { 0 }; /* format specified: 2 64-bit integers */ struct acpi_buffer format = {sizeof("NN"), "NN"}; diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index 45c301ea5e00..453281326c83 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -603,7 +603,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) * with the codec driver/pdata are non-existent */ - struct acpi_chan_package chan_package; + struct acpi_chan_package chan_package = { 0 }; /* format specified: 2 64-bit integers */ struct acpi_buffer format = {sizeof("NN"), "NN"}; -- cgit v1.2.3 From f057852fd351741d1efaadc48aa59ea49c79a087 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 20 May 2022 16:17:13 -0500 Subject: ASoC: Intel: sof_pcm512x: remove unnecessary init MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit cppcheck complains about an invalid NULL dereference but indeed there is no need to initialize a loop variable. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Link: https://lore.kernel.org/r/20220520211719.607543-11-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_pcm512x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/sof_pcm512x.c b/sound/soc/intel/boards/sof_pcm512x.c index 6815204e58d5..d4c67d5340a9 100644 --- a/sound/soc/intel/boards/sof_pcm512x.c +++ b/sound/soc/intel/boards/sof_pcm512x.c @@ -419,7 +419,7 @@ static int sof_audio_probe(struct platform_device *pdev) static int sof_pcm512x_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); - struct snd_soc_component *component = NULL; + struct snd_soc_component *component; for_each_card_components(card, component) { if (!strcmp(component->name, pcm512x_component[0].name)) { -- cgit v1.2.3 From 9e9fb5d3f387788d50f5eae4c01ff60429691e71 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 20 May 2022 16:17:14 -0500 Subject: ASoC: mediatek: mt8195: simplify error handling MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit cppcheck warnings: sound/soc/mediatek/mt8195/mt8195-afe-clk.c:311:9: warning: Identical condition and return expression 'ret', return value is always 0 [identicalConditionAfterEarlyExit] return ret; ^ sound/soc/mediatek/mt8195/mt8195-afe-clk.c:297:6: note: If condition 'ret' is true, the function will return/exit if (ret) ^ sound/soc/mediatek/mt8195/mt8195-afe-clk.c:311:9: note: Returning identical expression 'ret' return ret; ^ sound/soc/mediatek/mt8195/mt8195-afe-clk.c:341:9: warning: Identical condition and return expression 'ret', return value is always 0 [identicalConditionAfterEarlyExit] return ret; ^ sound/soc/mediatek/mt8195/mt8195-afe-clk.c:338:6: note: If condition 'ret' is true, the function will return/exit if (ret) ^ sound/soc/mediatek/mt8195/mt8195-afe-clk.c:341:9: note: Returning identical expression 'ret' return ret; ^ Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Link: https://lore.kernel.org/r/20220520211719.607543-12-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8195/mt8195-afe-clk.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/mediatek/mt8195/mt8195-afe-clk.c b/sound/soc/mediatek/mt8195/mt8195-afe-clk.c index efd5cc364a35..2ee3872c83c3 100644 --- a/sound/soc/mediatek/mt8195/mt8195-afe-clk.c +++ b/sound/soc/mediatek/mt8195/mt8195-afe-clk.c @@ -284,7 +284,7 @@ static int mt8195_afe_enable_apll_tuner(struct mtk_base_afe *afe, { struct mt8195_afe_tuner_cfg *cfg = mt8195_afe_found_apll_tuner(id); unsigned long flags; - int ret = 0; + int ret; if (!cfg) return -EINVAL; @@ -308,7 +308,7 @@ static int mt8195_afe_enable_apll_tuner(struct mtk_base_afe *afe, spin_unlock_irqrestore(&cfg->ctrl_lock, flags); - return ret; + return 0; } static int mt8195_afe_disable_apll_tuner(struct mtk_base_afe *afe, @@ -316,7 +316,7 @@ static int mt8195_afe_disable_apll_tuner(struct mtk_base_afe *afe, { struct mt8195_afe_tuner_cfg *cfg = mt8195_afe_found_apll_tuner(id); unsigned long flags; - int ret = 0; + int ret; if (!cfg) return -EINVAL; @@ -338,7 +338,7 @@ static int mt8195_afe_disable_apll_tuner(struct mtk_base_afe *afe, if (ret) return ret; - return ret; + return 0; } int mt8195_afe_get_mclk_source_clk_id(int sel) -- cgit v1.2.3 From 015d9ab7805fb1b3766d1dc487ed34dbc03bd4da Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 20 May 2022 16:17:15 -0500 Subject: ASoC: qcom: q6dsp: q6adm: remove useless initializations MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit cppcheck complains about invalid NULL dereferences but there's indeed no need to initialize loop variables. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Link: https://lore.kernel.org/r/20220520211719.607543-13-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6adm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/qcom/qdsp6/q6adm.c b/sound/soc/qcom/qdsp6/q6adm.c index 72c5719f1d25..22b408c3794e 100644 --- a/sound/soc/qcom/qdsp6/q6adm.c +++ b/sound/soc/qcom/qdsp6/q6adm.c @@ -90,7 +90,7 @@ struct q6adm_session_map_node_v5 { static struct q6copp *q6adm_find_copp(struct q6adm *adm, int port_idx, int copp_idx) { - struct q6copp *c = NULL; + struct q6copp *c; struct q6copp *ret = NULL; unsigned long flags; @@ -299,7 +299,7 @@ static struct q6copp *q6adm_find_matching_copp(struct q6adm *adm, int channel_mode, int bit_width, int app_type) { - struct q6copp *c = NULL; + struct q6copp *c; struct q6copp *ret = NULL; unsigned long flags; -- cgit v1.2.3 From 7518be0cc120d7617a8985787196cd5776b93688 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 20 May 2022 16:17:16 -0500 Subject: ASoC: qcom: q6dsp: remove spurious space MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit cppcheck warning: sound/soc/qcom/qdsp6/q6adm.c:183:14: warning:inconclusive: Found suspicious oper ator '*' [constStatement] } __packed * open = data->payload; ^ Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Link: https://lore.kernel.org/r/20220520211719.607543-14-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6adm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/qcom/qdsp6/q6adm.c b/sound/soc/qcom/qdsp6/q6adm.c index 22b408c3794e..01f383888b62 100644 --- a/sound/soc/qcom/qdsp6/q6adm.c +++ b/sound/soc/qcom/qdsp6/q6adm.c @@ -180,7 +180,7 @@ static int q6adm_callback(struct apr_device *adev, struct apr_resp_pkt *data) u32 status; u16 copp_id; u16 reserved; - } __packed * open = data->payload; + } __packed *open = data->payload; copp = q6adm_find_copp(adm, port_idx, copp_idx); if (!copp) -- cgit v1.2.3 From 59a6cc5c5d64ca20461fec46e450e0639b1e6410 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 20 May 2022 16:17:17 -0500 Subject: ASoC: rockchip: simplify error handling MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit cppcheck warning: sound/soc/rockchip/rk3288_hdmi_analog.c:256:9: warning: Identical condition and return expression 'ret', return value is always 0 [identicalConditionAfterEarlyExit] return ret; ^ sound/soc/rockchip/rk3288_hdmi_analog.c:252:6: note: If condition 'ret' is true, the function will return/exit if (ret) ^ sound/soc/rockchip/rk3288_hdmi_analog.c:256:9: note: Returning identical expression 'ret' return ret; ^ Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Link: https://lore.kernel.org/r/20220520211719.607543-15-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/rockchip/rk3288_hdmi_analog.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/rockchip/rk3288_hdmi_analog.c b/sound/soc/rockchip/rk3288_hdmi_analog.c index bcdeddeba80c..0c6bd9a019db 100644 --- a/sound/soc/rockchip/rk3288_hdmi_analog.c +++ b/sound/soc/rockchip/rk3288_hdmi_analog.c @@ -169,7 +169,7 @@ static struct snd_soc_card snd_soc_card_rk = { static int snd_rk_mc_probe(struct platform_device *pdev) { - int ret = 0; + int ret; struct snd_soc_card *card = &snd_soc_card_rk; struct device_node *np = pdev->dev.of_node; struct rk_drvdata *machine; @@ -253,7 +253,7 @@ static int snd_rk_mc_probe(struct platform_device *pdev) return dev_err_probe(&pdev->dev, ret, "Soc register card failed\n"); - return ret; + return 0; } static const struct of_device_id rockchip_sound_of_match[] = { -- cgit v1.2.3 From 0c57064e3fdba9bb76086b9a6e318eb0cef24b69 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 20 May 2022 16:17:18 -0500 Subject: ASoC: samsung: snow: simplify error handling MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit cppcheck warning: sound/soc/samsung/snow.c:219:9: warning: Identical condition and return expression 'ret', return value is always 0 [identicalConditionAfterEarlyExit] return ret; ^ sound/soc/samsung/snow.c:215:6: note: If condition 'ret' is true, the function will return/exit if (ret) ^ sound/soc/samsung/snow.c:219:9: note: Returning identical expression 'ret' return ret; ^ Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Link: https://lore.kernel.org/r/20220520211719.607543-16-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/samsung/snow.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/samsung/snow.c b/sound/soc/samsung/snow.c index 02372109c251..da342da03880 100644 --- a/sound/soc/samsung/snow.c +++ b/sound/soc/samsung/snow.c @@ -216,7 +216,7 @@ static int snow_probe(struct platform_device *pdev) return dev_err_probe(&pdev->dev, ret, "snd_soc_register_card failed\n"); - return ret; + return 0; } static int snow_remove(struct platform_device *pdev) -- cgit v1.2.3 From 7188b28f6686af0bc4aa1f96d720de782769a0a9 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 20 May 2022 16:17:19 -0500 Subject: ASoC: meson: remove useless initialization MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit cppcheck complains about invalid NULL dereferences but there's indeed no need to initialize a loop variable. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Link: https://lore.kernel.org/r/20220520211719.607543-17-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/meson/meson-codec-glue.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/meson/meson-codec-glue.c b/sound/soc/meson/meson-codec-glue.c index 2870cfad813a..80c5ed196961 100644 --- a/sound/soc/meson/meson-codec-glue.c +++ b/sound/soc/meson/meson-codec-glue.c @@ -13,7 +13,7 @@ static struct snd_soc_dapm_widget * meson_codec_glue_get_input(struct snd_soc_dapm_widget *w) { - struct snd_soc_dapm_path *p = NULL; + struct snd_soc_dapm_path *p; struct snd_soc_dapm_widget *in; snd_soc_dapm_widget_for_each_source_path(w, p) { -- cgit v1.2.3 From 0a034d93ee929a9ea89f3fa5f1d8492435b9ee6e Mon Sep 17 00:00:00 2001 From: Miaoqian Lin Date: Fri, 3 Jun 2022 17:10:43 +0400 Subject: ASoC: cros_ec_codec: Fix refcount leak in cros_ec_codec_platform_probe of_parse_phandle() returns a node pointer with refcount incremented, we should use of_node_put() on it when not need anymore. Add missing of_node_put() to avoid refcount leak. Fixes: b6bc07d4360d ("ASoC: cros_ec_codec: support WoV") Signed-off-by: Miaoqian Lin Reviewed-by: Tzung-Bi Shih Reviewed-by: Guenter Roeck Link: https://lore.kernel.org/r/20220603131043.38907-1-linmq006@gmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/cros_ec_codec.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/cros_ec_codec.c b/sound/soc/codecs/cros_ec_codec.c index 8b0a9c788a26..11e7b3f6d410 100644 --- a/sound/soc/codecs/cros_ec_codec.c +++ b/sound/soc/codecs/cros_ec_codec.c @@ -995,6 +995,7 @@ static int cros_ec_codec_platform_probe(struct platform_device *pdev) dev_dbg(dev, "ap_shm_phys_addr=%#llx len=%#x\n", priv->ap_shm_phys_addr, priv->ap_shm_len); } + of_node_put(node); } #endif -- cgit v1.2.3 From 3e2649c5e8643bea0867bb1dd970fedadb0eb7f3 Mon Sep 17 00:00:00 2001 From: Miaoqian Lin Date: Fri, 3 Jun 2022 17:06:39 +0400 Subject: ASoC: samsung: Fix error handling in aries_audio_probe of_get_child_by_name() returns a node pointer with refcount incremented, we should use of_node_put() on it when not need anymore. This function is missing of_node_put(cpu) in the error path. Fix this by goto out label. of_node_put() will check NULL pointer. Fixes: 7a3a7671fa6c ("ASoC: samsung: Add driver for Aries boards") Signed-off-by: Miaoqian Lin Reviewed-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20220603130640.37624-1-linmq006@gmail.com Signed-off-by: Mark Brown --- sound/soc/samsung/aries_wm8994.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/aries_wm8994.c b/sound/soc/samsung/aries_wm8994.c index bb0cf4244e00..edee02d7f100 100644 --- a/sound/soc/samsung/aries_wm8994.c +++ b/sound/soc/samsung/aries_wm8994.c @@ -628,8 +628,10 @@ static int aries_audio_probe(struct platform_device *pdev) return -EINVAL; codec = of_get_child_by_name(dev->of_node, "codec"); - if (!codec) - return -EINVAL; + if (!codec) { + ret = -EINVAL; + goto out; + } for_each_card_prelinks(card, i, dai_link) { dai_link->codecs->of_node = of_parse_phandle(codec, -- cgit v1.2.3 From 8466579b63cc9aa957b7b4f273087512f989d2a1 Mon Sep 17 00:00:00 2001 From: Christophe JAILLET Date: Sat, 28 May 2022 09:59:22 +0200 Subject: ASoC: ux500: Remove some leftover from the "Replace GPLv2 boilerplate/reference with SPDX" rules The "Replace GPLv2 boilerplate/reference with SPDX" has left some empty "License terms" paragraphs. Remove them as well. Signed-off-by: Christophe JAILLET Link: https://lore.kernel.org/r/84d94977c57deee9e85249f18394ebf8d72497bc.1653724723.git.christophe.jaillet@wanadoo.fr Signed-off-by: Mark Brown --- sound/soc/ux500/mop500.c | 2 -- sound/soc/ux500/mop500_ab8500.c | 2 -- sound/soc/ux500/mop500_ab8500.h | 2 -- sound/soc/ux500/ux500_msp_dai.c | 2 -- sound/soc/ux500/ux500_msp_dai.h | 2 -- sound/soc/ux500/ux500_msp_i2s.c | 2 -- sound/soc/ux500/ux500_msp_i2s.h | 2 -- sound/soc/ux500/ux500_pcm.c | 2 -- sound/soc/ux500/ux500_pcm.h | 2 -- 9 files changed, 18 deletions(-) (limited to 'sound') diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c index 4f41bb0ab2b0..fdd55d772b8e 100644 --- a/sound/soc/ux500/mop500.c +++ b/sound/soc/ux500/mop500.c @@ -4,8 +4,6 @@ * * Author: Ola Lilja (ola.o.lilja@stericsson.com) * for ST-Ericsson. - * - * License terms: */ #include diff --git a/sound/soc/ux500/mop500_ab8500.c b/sound/soc/ux500/mop500_ab8500.c index 1ea1729984a9..e5e73a2bd9fe 100644 --- a/sound/soc/ux500/mop500_ab8500.c +++ b/sound/soc/ux500/mop500_ab8500.c @@ -5,8 +5,6 @@ * Author: Ola Lilja , * Kristoffer Karlsson * for ST-Ericsson. - * - * License terms: */ #include diff --git a/sound/soc/ux500/mop500_ab8500.h b/sound/soc/ux500/mop500_ab8500.h index 087ef246d87d..98de80a9cc4f 100644 --- a/sound/soc/ux500/mop500_ab8500.h +++ b/sound/soc/ux500/mop500_ab8500.h @@ -4,8 +4,6 @@ * * Author: Ola Lilja * for ST-Ericsson. - * - * License terms: */ #ifndef MOP500_AB8500_H diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c index 21052378a32e..56532b62faf3 100644 --- a/sound/soc/ux500/ux500_msp_dai.c +++ b/sound/soc/ux500/ux500_msp_dai.c @@ -5,8 +5,6 @@ * Author: Ola Lilja , * Roger Nilsson * for ST-Ericsson. - * - * License terms: */ #include diff --git a/sound/soc/ux500/ux500_msp_dai.h b/sound/soc/ux500/ux500_msp_dai.h index fcd4b26f5d2d..30bf70838196 100644 --- a/sound/soc/ux500/ux500_msp_dai.h +++ b/sound/soc/ux500/ux500_msp_dai.h @@ -5,8 +5,6 @@ * Author: Ola Lilja , * Roger Nilsson * for ST-Ericsson. - * - * License terms: */ #ifndef UX500_msp_dai_H diff --git a/sound/soc/ux500/ux500_msp_i2s.c b/sound/soc/ux500/ux500_msp_i2s.c index fd0b88bb7921..d113411a19f8 100644 --- a/sound/soc/ux500/ux500_msp_i2s.c +++ b/sound/soc/ux500/ux500_msp_i2s.c @@ -6,8 +6,6 @@ * Roger Nilsson , * Sandeep Kaushik * for ST-Ericsson. - * - * License terms: */ #include diff --git a/sound/soc/ux500/ux500_msp_i2s.h b/sound/soc/ux500/ux500_msp_i2s.h index 756b3973af9a..d45b5e2831cc 100644 --- a/sound/soc/ux500/ux500_msp_i2s.h +++ b/sound/soc/ux500/ux500_msp_i2s.h @@ -4,8 +4,6 @@ * * Author: Ola Lilja , * for ST-Ericsson. - * - * License terms: */ diff --git a/sound/soc/ux500/ux500_pcm.c b/sound/soc/ux500/ux500_pcm.c index 18191084b8b8..d3802e5ef196 100644 --- a/sound/soc/ux500/ux500_pcm.c +++ b/sound/soc/ux500/ux500_pcm.c @@ -5,8 +5,6 @@ * Author: Ola Lilja , * Roger Nilsson * for ST-Ericsson. - * - * License terms: */ #include diff --git a/sound/soc/ux500/ux500_pcm.h b/sound/soc/ux500/ux500_pcm.h index ff3ef7223db6..bd4348ebf9a1 100644 --- a/sound/soc/ux500/ux500_pcm.h +++ b/sound/soc/ux500/ux500_pcm.h @@ -5,8 +5,6 @@ * Author: Ola Lilja , * Roger Nilsson * for ST-Ericsson. - * - * License terms: */ #ifndef UX500_PCM_H #define UX500_PCM_H -- cgit v1.2.3 From 2f4a8171da06609bb6a063630ed546ee3d93dad7 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Wed, 25 May 2022 22:05:43 -0300 Subject: ASoC: imx-audmux: Silence a clang warning Change the of_device_get_match_data() cast to (uintptr_t) to silence the following clang warning: sound/soc/fsl/imx-audmux.c:301:16: warning: cast to smaller integer type 'enum imx_audmux_type' from 'const void *' [-Wvoid-pointer-to-enum-cast] Reported-by: kernel test robot Fixes: 6a8b8b582db1 ("ASoC: imx-audmux: Remove unused .id_table") Signed-off-by: Fabio Estevam Link: https://lore.kernel.org/r/20220526010543.1164793-1-festevam@gmail.com Signed-off-by: Mark Brown --- sound/soc/fsl/imx-audmux.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index dfa05d40b276..a8e5e0f57faf 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -298,7 +298,7 @@ static int imx_audmux_probe(struct platform_device *pdev) audmux_clk = NULL; } - audmux_type = (enum imx_audmux_type)of_device_get_match_data(&pdev->dev); + audmux_type = (uintptr_t)of_device_get_match_data(&pdev->dev); switch (audmux_type) { case IMX31_AUDMUX: -- cgit v1.2.3 From b521e85eefa384a5c31984b1a7e0d71b762c9663 Mon Sep 17 00:00:00 2001 From: Christophe JAILLET Date: Sat, 28 May 2022 10:00:53 +0200 Subject: ASoC: ab8500: Remove some leftover from the "Replace GPLv2 boilerplate/reference with SPDX" rules The "Replace GPLv2 boilerplate/reference with SPDX" has left some empty "License terms" paragraphs. Remove them as well. Signed-off-by: Christophe JAILLET Link: https://lore.kernel.org/r/28c0833d4a11f8f75f385e5aad93c23721b06c7e.1653724847.git.christophe.jaillet@wanadoo.fr Signed-off-by: Mark Brown --- sound/soc/codecs/ab8500-codec.c | 2 -- sound/soc/codecs/ab8500-codec.h | 2 -- 2 files changed, 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index aefafb0b7b97..cbd4a92cb06c 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -12,8 +12,6 @@ * Mikko Sarmanne , * Jarmo K. Kuronen , * for ST-Ericsson. - * - * License terms: */ #include diff --git a/sound/soc/codecs/ab8500-codec.h b/sound/soc/codecs/ab8500-codec.h index 0ac87d0446c2..2a6f6409f1f8 100644 --- a/sound/soc/codecs/ab8500-codec.h +++ b/sound/soc/codecs/ab8500-codec.h @@ -11,8 +11,6 @@ * Mikko J. Lehto , * Mikko Sarmanne , * for ST-Ericsson. - * - * License terms: */ #ifndef AB8500_CODEC_REGISTERS_H -- cgit v1.2.3 From b661a848a50c0cc3e0b79795c74469d7b50ff4ac Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Sat, 21 May 2022 13:11:29 +0200 Subject: ASoC: amd: acp: fix typo in comment Spelling mistake (triple letters) in comment. Detected with the help of Coccinelle. Signed-off-by: Julia Lawall Link: https://lore.kernel.org/r/20220521111145.81697-79-Julia.Lawall@inria.fr Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp-pdm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/amd/acp/acp-pdm.c b/sound/soc/amd/acp/acp-pdm.c index 424c6e0bb9d6..7a0b26a30051 100644 --- a/sound/soc/amd/acp/acp-pdm.c +++ b/sound/soc/amd/acp/acp-pdm.c @@ -174,7 +174,7 @@ static void acp_dmic_dai_shutdown(struct snd_pcm_substream *substream, struct acp_dev_data *adata = dev_get_drvdata(dev); u32 ext_int_ctrl; - /* Disable DMIC interrrupts */ + /* Disable DMIC interrupts */ ext_int_ctrl = readl(adata->acp_base + ACP_EXTERNAL_INTR_CNTL); ext_int_ctrl |= ~PDM_DMA_INTR_MASK; writel(ext_int_ctrl, adata->acp_base + ACP_EXTERNAL_INTR_CNTL); -- cgit v1.2.3 From 99b5c107506c728b8a7d25742cf13f6c9c89d6ea Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 2 Jun 2022 11:29:20 +0200 Subject: ASoC: ops: Clarify snd_soc_info_volsw_sx() Currently snd_soc_info_volsw_sx() is implemented indirectly, wrapping snd_soc_info_volsw() and modifying the values it sets up rather than directly setting up the values reported to userspace. This makes it much harder to follow what the intended behaviour of these controls is. Let's rewrite the function to be self contained with a clarifying comment at the top in an effort to help maintainability. Signed-off-by: Mark Brown Reviewed-by: Charles Keepax Tested-by: Charles Keepax Link: https://lore.kernel.org/r/20220602092921.3302713-1-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/soc-ops.c | 22 ++++++++++++++++------ 1 file changed, 16 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index e693070f51fe..8c0e669fe92d 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -203,7 +203,8 @@ EXPORT_SYMBOL_GPL(snd_soc_info_volsw); * Callback to provide information about a single mixer control, or a double * mixer control that spans 2 registers of the SX TLV type. SX TLV controls * have a range that represents both positive and negative values either side - * of zero but without a sign bit. + * of zero but without a sign bit. min is the minimum register value, max is + * the number of steps. * * Returns 0 for success. */ @@ -212,12 +213,21 @@ int snd_soc_info_volsw_sx(struct snd_kcontrol *kcontrol, { struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; + int max; - snd_soc_info_volsw(kcontrol, uinfo); - /* Max represents the number of levels in an SX control not the - * maximum value, so add the minimum value back on - */ - uinfo->value.integer.max += mc->min; + if (mc->platform_max) + max = mc->platform_max; + else + max = mc->max; + + if (max == 1 && !strstr(kcontrol->id.name, " Volume")) + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + else + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + + uinfo->count = snd_soc_volsw_is_stereo(mc) ? 2 : 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = max; return 0; } -- cgit v1.2.3 From f53f50ee21d46094a8c48970e95e38a4deaa128e Mon Sep 17 00:00:00 2001 From: Marco Felsch Date: Wed, 1 Jun 2022 11:23:40 +0200 Subject: ASoC: fsl_sai: use local device pointer Use a local variable to dereference the device pointer once and use the local variable in further calls. No functional changes. Signed-off-by: Marco Felsch Acked-by: Shengjiu Wang Link: https://lore.kernel.org/r/20220601092342.3328644-1-m.felsch@pengutronix.de Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 53 +++++++++++++++++++++++++------------------------ 1 file changed, 27 insertions(+), 26 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index fa950dde5310..a7637d602f3c 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -1004,6 +1004,7 @@ static int fsl_sai_runtime_resume(struct device *dev); static int fsl_sai_probe(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; + struct device *dev = &pdev->dev; struct fsl_sai *sai; struct regmap *gpr; struct resource *res; @@ -1012,12 +1013,12 @@ static int fsl_sai_probe(struct platform_device *pdev) int irq, ret, i; int index; - sai = devm_kzalloc(&pdev->dev, sizeof(*sai), GFP_KERNEL); + sai = devm_kzalloc(dev, sizeof(*sai), GFP_KERNEL); if (!sai) return -ENOMEM; sai->pdev = pdev; - sai->soc_data = of_device_get_match_data(&pdev->dev); + sai->soc_data = of_device_get_match_data(dev); sai->is_lsb_first = of_property_read_bool(np, "lsb-first"); @@ -1032,18 +1033,18 @@ static int fsl_sai_probe(struct platform_device *pdev) ARRAY_SIZE(fsl_sai_reg_defaults_ofs8); } - sai->regmap = devm_regmap_init_mmio(&pdev->dev, base, &fsl_sai_regmap_config); + sai->regmap = devm_regmap_init_mmio(dev, base, &fsl_sai_regmap_config); if (IS_ERR(sai->regmap)) { - dev_err(&pdev->dev, "regmap init failed\n"); + dev_err(dev, "regmap init failed\n"); return PTR_ERR(sai->regmap); } - sai->bus_clk = devm_clk_get(&pdev->dev, "bus"); + sai->bus_clk = devm_clk_get(dev, "bus"); /* Compatible with old DTB cases */ if (IS_ERR(sai->bus_clk) && PTR_ERR(sai->bus_clk) != -EPROBE_DEFER) - sai->bus_clk = devm_clk_get(&pdev->dev, "sai"); + sai->bus_clk = devm_clk_get(dev, "sai"); if (IS_ERR(sai->bus_clk)) { - dev_err(&pdev->dev, "failed to get bus clock: %ld\n", + dev_err(dev, "failed to get bus clock: %ld\n", PTR_ERR(sai->bus_clk)); /* -EPROBE_DEFER */ return PTR_ERR(sai->bus_clk); @@ -1051,9 +1052,9 @@ static int fsl_sai_probe(struct platform_device *pdev) for (i = 1; i < FSL_SAI_MCLK_MAX; i++) { sprintf(tmp, "mclk%d", i); - sai->mclk_clk[i] = devm_clk_get(&pdev->dev, tmp); + sai->mclk_clk[i] = devm_clk_get(dev, tmp); if (IS_ERR(sai->mclk_clk[i])) { - dev_err(&pdev->dev, "failed to get mclk%d clock: %ld\n", + dev_err(dev, "failed to get mclk%d clock: %ld\n", i + 1, PTR_ERR(sai->mclk_clk[i])); sai->mclk_clk[i] = NULL; } @@ -1068,10 +1069,10 @@ static int fsl_sai_probe(struct platform_device *pdev) if (irq < 0) return irq; - ret = devm_request_irq(&pdev->dev, irq, fsl_sai_isr, IRQF_SHARED, + ret = devm_request_irq(dev, irq, fsl_sai_isr, IRQF_SHARED, np->name, sai); if (ret) { - dev_err(&pdev->dev, "failed to claim irq %u\n", irq); + dev_err(dev, "failed to claim irq %u\n", irq); return ret; } @@ -1088,7 +1089,7 @@ static int fsl_sai_probe(struct platform_device *pdev) if (of_find_property(np, "fsl,sai-synchronous-rx", NULL) && of_find_property(np, "fsl,sai-asynchronous", NULL)) { /* error out if both synchronous and asynchronous are present */ - dev_err(&pdev->dev, "invalid binding for synchronous mode\n"); + dev_err(dev, "invalid binding for synchronous mode\n"); return -EINVAL; } @@ -1109,7 +1110,7 @@ static int fsl_sai_probe(struct platform_device *pdev) of_device_is_compatible(np, "fsl,imx6ul-sai")) { gpr = syscon_regmap_lookup_by_compatible("fsl,imx6ul-iomuxc-gpr"); if (IS_ERR(gpr)) { - dev_err(&pdev->dev, "cannot find iomuxc registers\n"); + dev_err(dev, "cannot find iomuxc registers\n"); return PTR_ERR(gpr); } @@ -1127,23 +1128,23 @@ static int fsl_sai_probe(struct platform_device *pdev) sai->dma_params_tx.maxburst = FSL_SAI_MAXBURST_TX; platform_set_drvdata(pdev, sai); - pm_runtime_enable(&pdev->dev); - if (!pm_runtime_enabled(&pdev->dev)) { - ret = fsl_sai_runtime_resume(&pdev->dev); + pm_runtime_enable(dev); + if (!pm_runtime_enabled(dev)) { + ret = fsl_sai_runtime_resume(dev); if (ret) goto err_pm_disable; } - ret = pm_runtime_get_sync(&pdev->dev); + ret = pm_runtime_get_sync(dev); if (ret < 0) { - pm_runtime_put_noidle(&pdev->dev); + pm_runtime_put_noidle(dev); goto err_pm_get_sync; } /* Get sai version */ - ret = fsl_sai_check_version(&pdev->dev); + ret = fsl_sai_check_version(dev); if (ret < 0) - dev_warn(&pdev->dev, "Error reading SAI version: %d\n", ret); + dev_warn(dev, "Error reading SAI version: %d\n", ret); /* Select MCLK direction */ if (of_find_property(np, "fsl,sai-mclk-direction-output", NULL) && @@ -1152,7 +1153,7 @@ static int fsl_sai_probe(struct platform_device *pdev) FSL_SAI_MCTL_MCLK_EN, FSL_SAI_MCTL_MCLK_EN); } - ret = pm_runtime_put_sync(&pdev->dev); + ret = pm_runtime_put_sync(dev); if (ret < 0) goto err_pm_get_sync; @@ -1165,12 +1166,12 @@ static int fsl_sai_probe(struct platform_device *pdev) if (ret) goto err_pm_get_sync; } else { - ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + ret = devm_snd_dmaengine_pcm_register(dev, NULL, 0); if (ret) goto err_pm_get_sync; } - ret = devm_snd_soc_register_component(&pdev->dev, &fsl_component, + ret = devm_snd_soc_register_component(dev, &fsl_component, &sai->cpu_dai_drv, 1); if (ret) goto err_pm_get_sync; @@ -1178,10 +1179,10 @@ static int fsl_sai_probe(struct platform_device *pdev) return ret; err_pm_get_sync: - if (!pm_runtime_status_suspended(&pdev->dev)) - fsl_sai_runtime_suspend(&pdev->dev); + if (!pm_runtime_status_suspended(dev)) + fsl_sai_runtime_suspend(dev); err_pm_disable: - pm_runtime_disable(&pdev->dev); + pm_runtime_disable(dev); return ret; } -- cgit v1.2.3 From 22205521770ee740f64a3ec90301f50e34738cfd Mon Sep 17 00:00:00 2001 From: Marco Felsch Date: Wed, 1 Jun 2022 11:23:42 +0200 Subject: ASoC: fsl_sai: add error message in case of missing imx-pcm-dma support If the imx-pcm-dma is required we need to have the module enabled. For all NXP/FSL sound cards using the ASoC architecture this is the case but in case of using the simple-audio-card sound card this isn't the case. In such case the driver probe fails silently and the card isn't available. It took a while to find the missing Kconfig. Make this easier for others by printing a error if this the module isn't available but required by the HW. Signed-off-by: Marco Felsch Acked-by: Shengjiu Wang Link: https://lore.kernel.org/r/20220601092342.3328644-3-m.felsch@pengutronix.de Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index a7637d602f3c..b65c9c7cf54a 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -1163,8 +1163,11 @@ static int fsl_sai_probe(struct platform_device *pdev) */ if (sai->soc_data->use_imx_pcm) { ret = imx_pcm_dma_init(pdev); - if (ret) + if (ret) { + if (!IS_ENABLED(CONFIG_SND_SOC_IMX_PCM_DMA)) + dev_err(dev, "Error: You must enable the imx-pcm-dma support!\n"); goto err_pm_get_sync; + } } else { ret = devm_snd_dmaengine_pcm_register(dev, NULL, 0); if (ret) -- cgit v1.2.3 From ae4f11c1ed2d67192fdf3d89db719ee439827c11 Mon Sep 17 00:00:00 2001 From: Miaoqian Lin Date: Thu, 2 Jun 2022 07:41:42 +0400 Subject: ASoC: mediatek: mt8173: Fix refcount leak in mt8173_rt5650_rt5676_dev_probe of_parse_phandle() returns a node pointer with refcount incremented, we should use of_node_put() on it when not need anymore. Fix missing of_node_put() in error paths. Fixes: 94319ba10eca ("ASoC: mediatek: Use platform_of_node for machine drivers") Signed-off-by: Miaoqian Lin Link: https://lore.kernel.org/r/20220602034144.60159-1-linmq006@gmail.com Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c index 70bf312e855f..8794720cea3a 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c @@ -256,14 +256,16 @@ static int mt8173_rt5650_rt5676_dev_probe(struct platform_device *pdev) if (!mt8173_rt5650_rt5676_dais[DAI_LINK_CODEC_I2S].codecs[0].of_node) { dev_err(&pdev->dev, "Property 'audio-codec' missing or invalid\n"); - return -EINVAL; + ret = -EINVAL; + goto put_node; } mt8173_rt5650_rt5676_dais[DAI_LINK_CODEC_I2S].codecs[1].of_node = of_parse_phandle(pdev->dev.of_node, "mediatek,audio-codec", 1); if (!mt8173_rt5650_rt5676_dais[DAI_LINK_CODEC_I2S].codecs[1].of_node) { dev_err(&pdev->dev, "Property 'audio-codec' missing or invalid\n"); - return -EINVAL; + ret = -EINVAL; + goto put_node; } mt8173_rt5650_rt5676_codec_conf[0].dlc.of_node = mt8173_rt5650_rt5676_dais[DAI_LINK_CODEC_I2S].codecs[1].of_node; @@ -276,13 +278,15 @@ static int mt8173_rt5650_rt5676_dev_probe(struct platform_device *pdev) if (!mt8173_rt5650_rt5676_dais[DAI_LINK_HDMI_I2S].codecs->of_node) { dev_err(&pdev->dev, "Property 'audio-codec' missing or invalid\n"); - return -EINVAL; + ret = -EINVAL; + goto put_node; } card->dev = &pdev->dev; ret = devm_snd_soc_register_card(&pdev->dev, card); +put_node: of_node_put(platform_node); return ret; } -- cgit v1.2.3 From aa7407f807b250eca7697e5fe9a699bc6c2fab71 Mon Sep 17 00:00:00 2001 From: Randy Dunlap Date: Sun, 5 Jun 2022 09:31:23 -0700 Subject: ASoC: max98390: use linux/gpio/consumer.h to fix build Change the header file to fix build errors in max98390.c: ../sound/soc/codecs/max98390.c: In function 'max98390_i2c_probe': ../sound/soc/codecs/max98390.c:1076:22: error: implicit declaration of function 'devm_gpiod_get_optional'; did you mean 'devm_regulator_get_optional'? [-Werror=implicit-function-declaration] 1076 | reset_gpio = devm_gpiod_get_optional(&i2c->dev, ../sound/soc/codecs/max98390.c:1077:55: error: 'GPIOD_OUT_HIGH' undeclared (first use in this function); did you mean 'GPIOF_INIT_HIGH'? 1077 | "reset", GPIOD_OUT_HIGH); ../sound/soc/codecs/max98390.c:1077:55: note: each undeclared identifier is reported only once for each function it appears in ../sound/soc/codecs/max98390.c:1083:17: error: implicit declaration of function 'gpiod_set_value_cansleep'; did you mean 'gpio_set_value_cansleep'? [-Werror=implicit-function-declaration] 1083 | gpiod_set_value_cansleep(reset_gpio, 0); Fixes: 397ff0249606 ("ASoC: max98390: Add reset gpio control") Signed-off-by: Randy Dunlap Reported-by: kernel test robot Cc: Steve Lee Cc: Mark Brown Cc: Liam Girdwood Cc: Jaroslav Kysela Cc: Takashi Iwai Cc: alsa-devel@alsa-project.org Link: https://lore.kernel.org/r/20220605163123.23537-1-rdunlap@infradead.org Signed-off-by: Mark Brown --- sound/soc/codecs/max98390.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/max98390.c b/sound/soc/codecs/max98390.c index 2a6b1648c884..d83f81d9ff4e 100644 --- a/sound/soc/codecs/max98390.c +++ b/sound/soc/codecs/max98390.c @@ -10,7 +10,7 @@ #include #include #include -#include +#include #include #include #include -- cgit v1.2.3 From ef6c320942a2f057204702d769d507186fd7f0b7 Mon Sep 17 00:00:00 2001 From: Alexander Martinz Date: Thu, 2 Jun 2022 18:45:03 +0200 Subject: ASoC: codecs: tfa989x: Add support for tfa9890 The initialization sequence is taken from the version provided by the supplier [1]. This allows speakers using the TFA9890 amplifier to work, which are used by various mobile phones such as the SHIFT6mq. [1]: https://source.codeaurora.org/external/mas/tfa98xx/tree/src/tfa_init.c?id=d2cd12931fbc48df988b62931fb9960d4e9dc05d#n1827 Signed-off-by: Alexander Martinz Reviewed-by: Stephan Gerhold Link: https://lore.kernel.org/r/20220602164504.261361-1-amartinz@shiftphones.com Signed-off-by: Mark Brown --- sound/soc/codecs/tfa989x.c | 30 ++++++++++++++++++++++++++++++ 1 file changed, 30 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/tfa989x.c b/sound/soc/codecs/tfa989x.c index dc86852752c5..8ab2656de750 100644 --- a/sound/soc/codecs/tfa989x.c +++ b/sound/soc/codecs/tfa989x.c @@ -40,12 +40,14 @@ #define TFA989X_I2S_SEL_REG 0x0a #define TFA989X_I2S_SEL_REG_SPKR_MSK GENMASK(10, 9) /* speaker impedance */ #define TFA989X_I2S_SEL_REG_DCFG_MSK GENMASK(14, 11) /* DCDC compensation */ +#define TFA989X_HIDE_UNHIDE_KEY 0x40 #define TFA989X_PWM_CONTROL 0x41 #define TFA989X_CURRENTSENSE1 0x46 #define TFA989X_CURRENTSENSE2 0x47 #define TFA989X_CURRENTSENSE3 0x48 #define TFA989X_CURRENTSENSE4 0x49 +#define TFA9890_REVISION 0x80 #define TFA9895_REVISION 0x12 #define TFA9897_REVISION 0x97 @@ -188,6 +190,33 @@ static struct snd_soc_dai_driver tfa989x_dai = { .ops = &tfa989x_dai_ops, }; +static int tfa9890_init(struct regmap *regmap) +{ + int ret; + + /* unhide keys to allow updating them */ + ret = regmap_write(regmap, TFA989X_HIDE_UNHIDE_KEY, 0x5a6b); + if (ret) + return ret; + + /* update PLL registers */ + ret = regmap_set_bits(regmap, 0x59, 0x3); + if (ret) + return ret; + + /* hide keys again */ + ret = regmap_write(regmap, TFA989X_HIDE_UNHIDE_KEY, 0x0000); + if (ret) + return ret; + + return regmap_write(regmap, TFA989X_CURRENTSENSE2, 0x7BE1); +} + +static const struct tfa989x_rev tfa9890_rev = { + .rev = TFA9890_REVISION, + .init = tfa9890_init, +}; + static const struct reg_sequence tfa9895_reg_init[] = { /* some other registers must be set for optimal amplifier behaviour */ { TFA989X_BAT_PROT, 0x13ab }, @@ -376,6 +405,7 @@ static int tfa989x_i2c_probe(struct i2c_client *i2c) } static const struct of_device_id tfa989x_of_match[] = { + { .compatible = "nxp,tfa9890", .data = &tfa9890_rev }, { .compatible = "nxp,tfa9895", .data = &tfa9895_rev }, { .compatible = "nxp,tfa9897", .data = &tfa9897_rev }, { } -- cgit v1.2.3 From 6398b004cfcce38626f3ba6fa5853177a3501aae Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Fri, 20 May 2022 11:06:00 +0800 Subject: ASoC: fsl_asrc_dma: enable dual fifo for ASRC P2P The SSI and SPDIF has dual fifos, enhance P2P for these case with using the sdma_peripheral_config struct Signed-off-by: Shengjiu Wang Link: https://lore.kernel.org/r/1653015960-15474-1-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_asrc_dma.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c index 5038faf035cb..aaf7993935b7 100644 --- a/sound/soc/fsl/fsl_asrc_dma.c +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -129,6 +129,7 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component, struct snd_pcm_hw_params *params) { enum dma_slave_buswidth buswidth = DMA_SLAVE_BUSWIDTH_2_BYTES; + enum sdma_peripheral_type be_peripheral_type = IMX_DMATYPE_SSI; struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; struct snd_dmaengine_dai_dma_data *dma_params_fe = NULL; @@ -139,6 +140,7 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component, struct snd_soc_component *component_be = NULL; struct fsl_asrc *asrc = pair->asrc; struct dma_slave_config config_fe, config_be; + struct sdma_peripheral_config audio_config; enum asrc_pair_index index = pair->index; struct device *dev = component->dev; struct device_node *of_dma_node; @@ -221,6 +223,7 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component, /* Get DMA request of Back-End */ tmp_data = tmp_chan->private; pair->dma_data.dma_request = tmp_data->dma_request; + be_peripheral_type = tmp_data->peripheral_type; if (!be_chan) dma_release_channel(tmp_chan); @@ -268,6 +271,17 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component, config_be.dst_addr_width = buswidth; config_be.dst_maxburst = dma_params_be->maxburst; + memset(&audio_config, 0, sizeof(audio_config)); + config_be.peripheral_config = &audio_config; + config_be.peripheral_size = sizeof(audio_config); + + if (tx && (be_peripheral_type == IMX_DMATYPE_SSI_DUAL || + be_peripheral_type == IMX_DMATYPE_SPDIF)) + audio_config.n_fifos_dst = 2; + if (!tx && (be_peripheral_type == IMX_DMATYPE_SSI_DUAL || + be_peripheral_type == IMX_DMATYPE_SPDIF)) + audio_config.n_fifos_src = 2; + if (tx) { config_be.src_addr = asrc->paddr + asrc->get_fifo_addr(OUT, index); config_be.dst_addr = dma_params_be->addr; -- cgit v1.2.3 From ff31753fcb061b90bd8c356d5b27a6eb5f8ade15 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 30 May 2022 04:28:44 +0000 Subject: ASoC: simple-card-utils: rename asoc_simple_init_dai_link_params() to asoc_simple_init_for_codec2codec() commit 95cfc0a0aaf5752071 ("ASoC: simple-card: Add support for codec2codec DAI links") added the function asoc_simple_init_dai_link_params() to initialize dai_link "params". It is very straight naming, but difficult to noticed that it is for Codec2Codec support. Handling Codec2Codec is one of very tricky part on ALSA SoC, thus it is very important to clarify it. This patch renames the function name. Signed-off-by: Kuninori Morimoto Link: https://lore.kernel.org/r/87o7zflk3n.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/generic/simple-card-utils.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 539d7f081bd7..fa080f166345 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -513,7 +513,7 @@ static int asoc_simple_init_dai(struct snd_soc_dai *dai, return 0; } -static int asoc_simple_init_dai_link_params(struct snd_soc_pcm_runtime *rtd, +static int asoc_simple_init_for_codec2codec(struct snd_soc_pcm_runtime *rtd, struct simple_dai_props *dai_props) { struct snd_soc_dai_link *dai_link = rtd->dai_link; @@ -575,7 +575,7 @@ int asoc_simple_dai_init(struct snd_soc_pcm_runtime *rtd) return ret; } - ret = asoc_simple_init_dai_link_params(rtd, props); + ret = asoc_simple_init_for_codec2codec(rtd, props); if (ret < 0) return ret; -- cgit v1.2.3 From 3ae190edc5f6f64f296f8dd15f4b511f529ab402 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 3 Jun 2022 12:35:30 +0200 Subject: ASoC: nau8822: Don't reconfigure PLL to the same values When we configure the PLL record the input and output frequency, then if we get asked to configure the same values again just skip reprogramming the hardware. This makes things a bit easier to use for machine drivers since it means they don't need to keep track of if they've programmed the PLL so much. Signed-off-by: Mark Brown Link: https://lore.kernel.org/r/20220603103530.3844527-1-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/nau8822.c | 7 +++++++ sound/soc/codecs/nau8822.h | 2 ++ 2 files changed, 9 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/nau8822.c b/sound/soc/codecs/nau8822.c index 08f6c56dc387..f4f68b549e1a 100644 --- a/sound/soc/codecs/nau8822.c +++ b/sound/soc/codecs/nau8822.c @@ -726,6 +726,10 @@ static int nau8822_set_pll(struct snd_soc_dai *dai, int pll_id, int source, struct nau8822_pll *pll_param = &nau8822->pll; int ret, fs; + if (freq_in == pll_param->freq_in && + freq_out == pll_param->freq_out) + return 0; + fs = freq_out / 256; ret = nau8822_calc_pll(freq_in, fs, pll_param); @@ -762,6 +766,9 @@ static int nau8822_set_pll(struct snd_soc_dai *dai, int pll_id, int source, snd_soc_component_update_bits(component, NAU8822_REG_POWER_MANAGEMENT_1, NAU8822_PLL_EN_MASK, NAU8822_PLL_ON); + pll_param->freq_in = freq_in; + pll_param->freq_out = freq_out; + return 0; } diff --git a/sound/soc/codecs/nau8822.h b/sound/soc/codecs/nau8822.h index b45d42c15de6..547ec057f853 100644 --- a/sound/soc/codecs/nau8822.h +++ b/sound/soc/codecs/nau8822.h @@ -198,6 +198,8 @@ struct nau8822_pll { int mclk_scaler; int pll_frac; int pll_int; + int freq_in; + int freq_out; }; /* Codec Private Data */ -- cgit v1.2.3 From 84965cc60e643db7049eb75bb9a6cc5cd66ee3d8 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Uwe=20Kleine-K=C3=B6nig?= Date: Fri, 20 May 2022 19:33:49 +0200 Subject: ASoC: cs35l45: Make cs35l45_remove() return void MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit cs35l45_remove() always returns zero. Make it return no value which makes it easier to see in the callers that there is no error to handle. Also the return value of i2c driver remove callbacks is ignored anyway. This prepares making i2c remove callbacks return void, too. Signed-off-by: Uwe Kleine-König Acked-by: Charles Keepax Reviewed-by: Richard Fitzgerald Link: https://lore.kernel.org/r/20220520173349.774366-1-u.kleine-koenig@pengutronix.de Signed-off-by: Mark Brown --- sound/soc/codecs/cs35l45-i2c.c | 4 +++- sound/soc/codecs/cs35l45.c | 4 +--- sound/soc/codecs/cs35l45.h | 2 +- 3 files changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs35l45-i2c.c b/sound/soc/codecs/cs35l45-i2c.c index 38a4dbc9e9fe..06c2ddffb9c5 100644 --- a/sound/soc/codecs/cs35l45-i2c.c +++ b/sound/soc/codecs/cs35l45-i2c.c @@ -40,7 +40,9 @@ static int cs35l45_i2c_remove(struct i2c_client *client) { struct cs35l45_private *cs35l45 = i2c_get_clientdata(client); - return cs35l45_remove(cs35l45); + cs35l45_remove(cs35l45); + + return 0; } static const struct of_device_id cs35l45_of_match[] = { diff --git a/sound/soc/codecs/cs35l45.c b/sound/soc/codecs/cs35l45.c index 2367c1a4c10e..c94edfce4b72 100644 --- a/sound/soc/codecs/cs35l45.c +++ b/sound/soc/codecs/cs35l45.c @@ -665,7 +665,7 @@ err: } EXPORT_SYMBOL_NS_GPL(cs35l45_probe, SND_SOC_CS35L45); -int cs35l45_remove(struct cs35l45_private *cs35l45) +void cs35l45_remove(struct cs35l45_private *cs35l45) { pm_runtime_disable(cs35l45->dev); @@ -673,8 +673,6 @@ int cs35l45_remove(struct cs35l45_private *cs35l45) regulator_disable(cs35l45->vdd_a); /* VDD_BATT must be the last to power-off */ regulator_disable(cs35l45->vdd_batt); - - return 0; } EXPORT_SYMBOL_NS_GPL(cs35l45_remove, SND_SOC_CS35L45); diff --git a/sound/soc/codecs/cs35l45.h b/sound/soc/codecs/cs35l45.h index 4e266d19cd1c..680891bcfce9 100644 --- a/sound/soc/codecs/cs35l45.h +++ b/sound/soc/codecs/cs35l45.h @@ -212,6 +212,6 @@ extern const struct regmap_config cs35l45_spi_regmap; int cs35l45_apply_patch(struct cs35l45_private *cs43l45); unsigned int cs35l45_get_clk_freq_id(unsigned int freq); int cs35l45_probe(struct cs35l45_private *cs35l45); -int cs35l45_remove(struct cs35l45_private *cs35l45); +void cs35l45_remove(struct cs35l45_private *cs35l45); #endif /* CS35L45_H */ -- cgit v1.2.3 From 9c3148dec7d2d40ef727b8789d3e9410ad6d4a1f Mon Sep 17 00:00:00 2001 From: zhangqilong Date: Thu, 2 Jun 2022 15:20:24 +0800 Subject: ASoC: fsl_xcvr:Fix unbalanced pm_runtime_enable in fsl_xcvr_probe a) Add missing pm_runtime_disable() when probe error out. It could avoid pm_runtime implementation complains when removing and probing again the driver. b) Add remove for missing pm_runtime_disable(). Fix:c590fa80b3928 ("ASoC: fsl_xcvr: register platform component before registering cpu dai") Signed-off-by: Zhang Qilong Acked-by: Shengjiu Wang Link: https://lore.kernel.org/r/20220602072024.33236-1-zhangqilong3@huawei.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_xcvr.c | 9 +++++++++ 1 file changed, 9 insertions(+) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c index d0556c79fdb1..55e640cba87d 100644 --- a/sound/soc/fsl/fsl_xcvr.c +++ b/sound/soc/fsl/fsl_xcvr.c @@ -1228,6 +1228,7 @@ static int fsl_xcvr_probe(struct platform_device *pdev) */ ret = devm_snd_dmaengine_pcm_register(dev, NULL, 0); if (ret) { + pm_runtime_disable(dev); dev_err(dev, "failed to pcm register\n"); return ret; } @@ -1235,6 +1236,7 @@ static int fsl_xcvr_probe(struct platform_device *pdev) ret = devm_snd_soc_register_component(dev, &fsl_xcvr_comp, &fsl_xcvr_dai, 1); if (ret) { + pm_runtime_disable(dev); dev_err(dev, "failed to register component %s\n", fsl_xcvr_comp.name); } @@ -1242,6 +1244,12 @@ static int fsl_xcvr_probe(struct platform_device *pdev) return ret; } +static int fsl_xcvr_remove(struct platform_device *pdev) +{ + pm_runtime_disable(&pdev->dev); + return 0; +} + static __maybe_unused int fsl_xcvr_runtime_suspend(struct device *dev) { struct fsl_xcvr *xcvr = dev_get_drvdata(dev); @@ -1370,6 +1378,7 @@ static struct platform_driver fsl_xcvr_driver = { .pm = &fsl_xcvr_pm_ops, .of_match_table = fsl_xcvr_dt_ids, }, + .remove = fsl_xcvr_remove, }; module_platform_driver(fsl_xcvr_driver); -- cgit v1.2.3 From bf1ebcddcb19a1b6d6d8b75b75626197a5a76d4f Mon Sep 17 00:00:00 2001 From: Tang Bin Date: Wed, 25 May 2022 21:50:23 +0800 Subject: ASoC: stm32: sai: Remove useless define STM_SAI_IS_SUB_B(x) and STM_SAI_BLOCK_NAME(x) are not being used, so remove them. Signed-off-by: Tang Bin Acked-by: Olivier Moysan Link: https://lore.kernel.org/r/20220525135023.6792-1-tangbin@cmss.chinamobile.com Signed-off-by: Mark Brown --- sound/soc/stm/stm32_sai_sub.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index dd636af81c9b..4296fcb245c9 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -45,8 +45,6 @@ #define STM_SAI_B_ID 0x1 #define STM_SAI_IS_SUB_A(x) ((x)->id == STM_SAI_A_ID) -#define STM_SAI_IS_SUB_B(x) ((x)->id == STM_SAI_B_ID) -#define STM_SAI_BLOCK_NAME(x) (((x)->id == STM_SAI_A_ID) ? "A" : "B") #define SAI_SYNC_NONE 0x0 #define SAI_SYNC_INTERNAL 0x1 -- cgit v1.2.3 From fef94875a72bc63ba60d2e12421d7f49d31523f0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 2 Jun 2022 12:18:33 +0200 Subject: ASoC: ops: Remove unneeded delay.h inclusion The ops code does not do any sleeps or delays so does not need delay.h. Signed-off-by: Mark Brown Link: https://lore.kernel.org/r/20220602101833.3481641-1-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/soc-ops.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 8c0e669fe92d..2d5910b6ca54 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -14,7 +14,6 @@ #include #include #include -#include #include #include #include -- cgit v1.2.3 From 4f8ed19593872b710f27bbc3b7a9ce03310efc57 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 2 Jun 2022 15:10:58 +0200 Subject: ASoC: tfa9879: Use modern ASoC DAI format terminology As part of moving to remove the old style defines for the bus clocks update the tfa9879 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown Acked-by: Peter Rosin Link: https://lore.kernel.org/r/20220602131058.3552621-1-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/tfa9879.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tfa9879.c b/sound/soc/codecs/tfa9879.c index 3d8e8c2276f0..41a9b1b76e62 100644 --- a/sound/soc/codecs/tfa9879.c +++ b/sound/soc/codecs/tfa9879.c @@ -111,8 +111,8 @@ static int tfa9879_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) int i2s_set; int sck_pol; - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBC_CFC: break; default: return -EINVAL; -- cgit v1.2.3 From 7472eb8d7dd12b6b9b1a4f4527719cc9c7f5965f Mon Sep 17 00:00:00 2001 From: Miaoqian Lin Date: Fri, 3 Jun 2022 12:34:15 +0400 Subject: ASoC: mt6797-mt6351: Fix refcount leak in mt6797_mt6351_dev_probe of_parse_phandle() returns a node pointer with refcount incremented, we should use of_node_put() on it when not need anymore. Add missing of_node_put() to avoid refcount leak. Fixes: f0ab0bf250da ("ASoC: add mt6797-mt6351 driver and config option") Signed-off-by: Miaoqian Lin Link: https://lore.kernel.org/r/20220603083417.9011-1-linmq006@gmail.com Signed-off-by: Mark Brown --- sound/soc/mediatek/mt6797/mt6797-mt6351.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/mediatek/mt6797/mt6797-mt6351.c b/sound/soc/mediatek/mt6797/mt6797-mt6351.c index 496f32bcfb5e..d2f6213a6bfc 100644 --- a/sound/soc/mediatek/mt6797/mt6797-mt6351.c +++ b/sound/soc/mediatek/mt6797/mt6797-mt6351.c @@ -217,7 +217,8 @@ static int mt6797_mt6351_dev_probe(struct platform_device *pdev) if (!codec_node) { dev_err(&pdev->dev, "Property 'audio-codec' missing or invalid\n"); - return -EINVAL; + ret = -EINVAL; + goto put_platform_node; } for_each_card_prelinks(card, i, dai_link) { if (dai_link->codecs->name) @@ -230,6 +231,9 @@ static int mt6797_mt6351_dev_probe(struct platform_device *pdev) dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", __func__, ret); + of_node_put(codec_node); +put_platform_node: + of_node_put(platform_node); return ret; } -- cgit v1.2.3 From 82fa8f581a954ddeec1602bed9f8b4a09d100e6e Mon Sep 17 00:00:00 2001 From: Jiasheng Jiang Date: Tue, 31 May 2022 17:47:12 +0800 Subject: ASoC: codecs: da7210: add check for i2c_add_driver As i2c_add_driver could return error if fails, it should be better to check the return value. However, if the CONFIG_I2C and CONFIG_SPI_MASTER are both true, the return value of i2c_add_driver will be covered by spi_register_driver. Therefore, it is necessary to add check and return error if fails. Fixes: aa0e25caafb7 ("ASoC: da7210: Add support for spi regmap") Signed-off-by: Jiasheng Jiang Link: https://lore.kernel.org/r/20220531094712.2376759-1-jiasheng@iscas.ac.cn Signed-off-by: Mark Brown --- sound/soc/codecs/da7210.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 3fa3042e4424..76a21976ccdd 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -1335,6 +1335,8 @@ static int __init da7210_modinit(void) int ret = 0; #if IS_ENABLED(CONFIG_I2C) ret = i2c_add_driver(&da7210_i2c_driver); + if (ret) + return ret; #endif #if defined(CONFIG_SPI_MASTER) ret = spi_register_driver(&da7210_spi_driver); -- cgit v1.2.3 From 12ba5ceb4a08d5ea776d3eaf83c0cee63fafe952 Mon Sep 17 00:00:00 2001 From: Minghao Chi Date: Thu, 2 Jun 2022 07:18:09 +0000 Subject: ASoC: mediatek: remove unnecessary check of clk_disable_unprepare Because clk_disable_unprepare already checked NULL clock parameter, so the additional checks are unnecessary, just remove them. Reported-by: Zeal Robot Signed-off-by: Minghao Chi Link: https://lore.kernel.org/r/20220602071809.278134-1-chi.minghao@zte.com.cn Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c b/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c index 31494930433f..dcaeeeb8aac7 100644 --- a/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c +++ b/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c @@ -286,10 +286,8 @@ static int mt8173_afe_dais_set_clks(struct mtk_base_afe *afe, static void mt8173_afe_dais_disable_clks(struct mtk_base_afe *afe, struct clk *m_ck, struct clk *b_ck) { - if (m_ck) - clk_disable_unprepare(m_ck); - if (b_ck) - clk_disable_unprepare(b_ck); + clk_disable_unprepare(m_ck); + clk_disable_unprepare(b_ck); } static int mt8173_afe_i2s_startup(struct snd_pcm_substream *substream, -- cgit v1.2.3 From 8366d8ca0f7805be6cffe1e242822565aed509ae Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 2 Jun 2022 14:58:12 +0200 Subject: ASoC: max9860: Use modern ASoC DAI format terminology As part of moving to remove the old style defines for the bus clocks update the max9860 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown Acked-by: Peter Rosin Link: https://lore.kernel.org/r/20220602125812.3551947-1-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/max9860.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/max9860.c b/sound/soc/codecs/max9860.c index 82f20a8e27ad..a1d0179e12c7 100644 --- a/sound/soc/codecs/max9860.c +++ b/sound/soc/codecs/max9860.c @@ -448,9 +448,9 @@ static int max9860_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) struct snd_soc_component *component = dai->component; struct max9860_priv *max9860 = snd_soc_component_get_drvdata(component); - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - case SND_SOC_DAIFMT_CBS_CFS: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: + case SND_SOC_DAIFMT_CBC_CFC: max9860->fmt = fmt; return 0; -- cgit v1.2.3 From 063c915502b914a5a621458c763dfc28286f7606 Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Wed, 25 May 2022 13:23:41 +0800 Subject: ASoC: fsl_mqs: simplify the code with adding fsl_mqs_soc_data Add soc specific data struct fsl_mqs_soc_data, move the definition of control register, each function bits to it, then the code can be simplified. Signed-off-by: Shengjiu Wang Link: https://lore.kernel.org/r/1653456221-21613-1-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_mqs.c | 119 +++++++++++++++++++++++++++++------------------- 1 file changed, 71 insertions(+), 48 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_mqs.c b/sound/soc/fsl/fsl_mqs.c index ceaecbe3a25e..8a8d727319d6 100644 --- a/sound/soc/fsl/fsl_mqs.c +++ b/sound/soc/fsl/fsl_mqs.c @@ -10,6 +10,7 @@ #include #include #include +#include #include #include #include @@ -29,15 +30,41 @@ #define MQS_CLK_DIV_MASK (0xFF << 0) #define MQS_CLK_DIV_SHIFT (0) +/** + * struct fsl_mqs_soc_data - soc specific data + * + * @use_gpr: control register is in General Purpose Register group + * @ctrl_off: control register offset + * @en_mask: enable bit mask + * @en_shift: enable bit shift + * @rst_mask: reset bit mask + * @rst_shift: reset bit shift + * @osr_mask: oversample bit mask + * @osr_shift: oversample bit shift + * @div_mask: clock divider mask + * @div_shift: clock divider bit shift + */ +struct fsl_mqs_soc_data { + bool use_gpr; + int ctrl_off; + int en_mask; + int en_shift; + int rst_mask; + int rst_shift; + int osr_mask; + int osr_shift; + int div_mask; + int div_shift; +}; + /* codec private data */ struct fsl_mqs { struct regmap *regmap; struct clk *mclk; struct clk *ipg; + const struct fsl_mqs_soc_data *soc; - unsigned int reg_iomuxc_gpr2; unsigned int reg_mqs_ctrl; - bool use_gpr; }; #define FSL_MQS_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) @@ -65,19 +92,11 @@ static int fsl_mqs_hw_params(struct snd_pcm_substream *substream, res = mclk_rate % (32 * lrclk * 2 * 8); if (res == 0 && div > 0 && div <= 256) { - if (mqs_priv->use_gpr) { - regmap_update_bits(mqs_priv->regmap, IOMUXC_GPR2, - IMX6SX_GPR2_MQS_CLK_DIV_MASK, - (div - 1) << IMX6SX_GPR2_MQS_CLK_DIV_SHIFT); - regmap_update_bits(mqs_priv->regmap, IOMUXC_GPR2, - IMX6SX_GPR2_MQS_OVERSAMPLE_MASK, 0); - } else { - regmap_update_bits(mqs_priv->regmap, REG_MQS_CTRL, - MQS_CLK_DIV_MASK, - (div - 1) << MQS_CLK_DIV_SHIFT); - regmap_update_bits(mqs_priv->regmap, REG_MQS_CTRL, - MQS_OVERSAMPLE_MASK, 0); - } + regmap_update_bits(mqs_priv->regmap, mqs_priv->soc->ctrl_off, + mqs_priv->soc->div_mask, + (div - 1) << mqs_priv->soc->div_shift); + regmap_update_bits(mqs_priv->regmap, mqs_priv->soc->ctrl_off, + mqs_priv->soc->osr_mask, 0); } else { dev_err(component->dev, "can't get proper divider\n"); } @@ -118,14 +137,9 @@ static int fsl_mqs_startup(struct snd_pcm_substream *substream, struct snd_soc_component *component = dai->component; struct fsl_mqs *mqs_priv = snd_soc_component_get_drvdata(component); - if (mqs_priv->use_gpr) - regmap_update_bits(mqs_priv->regmap, IOMUXC_GPR2, - IMX6SX_GPR2_MQS_EN_MASK, - 1 << IMX6SX_GPR2_MQS_EN_SHIFT); - else - regmap_update_bits(mqs_priv->regmap, REG_MQS_CTRL, - MQS_EN_MASK, - 1 << MQS_EN_SHIFT); + regmap_update_bits(mqs_priv->regmap, mqs_priv->soc->ctrl_off, + mqs_priv->soc->en_mask, + 1 << mqs_priv->soc->en_shift); return 0; } @@ -135,12 +149,8 @@ static void fsl_mqs_shutdown(struct snd_pcm_substream *substream, struct snd_soc_component *component = dai->component; struct fsl_mqs *mqs_priv = snd_soc_component_get_drvdata(component); - if (mqs_priv->use_gpr) - regmap_update_bits(mqs_priv->regmap, IOMUXC_GPR2, - IMX6SX_GPR2_MQS_EN_MASK, 0); - else - regmap_update_bits(mqs_priv->regmap, REG_MQS_CTRL, - MQS_EN_MASK, 0); + regmap_update_bits(mqs_priv->regmap, mqs_priv->soc->ctrl_off, + mqs_priv->soc->en_mask, 0); } static const struct snd_soc_component_driver soc_codec_fsl_mqs = { @@ -191,12 +201,9 @@ static int fsl_mqs_probe(struct platform_device *pdev) * But in i.MX8QM/i.MX8QXP the control register is moved * to its own domain. */ - if (of_device_is_compatible(np, "fsl,imx8qm-mqs")) - mqs_priv->use_gpr = false; - else - mqs_priv->use_gpr = true; + mqs_priv->soc = of_device_get_match_data(&pdev->dev); - if (mqs_priv->use_gpr) { + if (mqs_priv->soc->use_gpr) { gpr_np = of_parse_phandle(np, "gpr", 0); if (!gpr_np) { dev_err(&pdev->dev, "failed to get gpr node by phandle\n"); @@ -280,12 +287,7 @@ static int fsl_mqs_runtime_resume(struct device *dev) return ret; } - if (mqs_priv->use_gpr) - regmap_write(mqs_priv->regmap, IOMUXC_GPR2, - mqs_priv->reg_iomuxc_gpr2); - else - regmap_write(mqs_priv->regmap, REG_MQS_CTRL, - mqs_priv->reg_mqs_ctrl); + regmap_write(mqs_priv->regmap, mqs_priv->soc->ctrl_off, mqs_priv->reg_mqs_ctrl); return 0; } @@ -293,12 +295,7 @@ static int fsl_mqs_runtime_suspend(struct device *dev) { struct fsl_mqs *mqs_priv = dev_get_drvdata(dev); - if (mqs_priv->use_gpr) - regmap_read(mqs_priv->regmap, IOMUXC_GPR2, - &mqs_priv->reg_iomuxc_gpr2); - else - regmap_read(mqs_priv->regmap, REG_MQS_CTRL, - &mqs_priv->reg_mqs_ctrl); + regmap_read(mqs_priv->regmap, mqs_priv->soc->ctrl_off, &mqs_priv->reg_mqs_ctrl); clk_disable_unprepare(mqs_priv->mclk); clk_disable_unprepare(mqs_priv->ipg); @@ -315,9 +312,35 @@ static const struct dev_pm_ops fsl_mqs_pm_ops = { pm_runtime_force_resume) }; +static const struct fsl_mqs_soc_data fsl_mqs_imx8qm_data = { + .use_gpr = false, + .ctrl_off = REG_MQS_CTRL, + .en_mask = MQS_EN_MASK, + .en_shift = MQS_EN_SHIFT, + .rst_mask = MQS_SW_RST_MASK, + .rst_shift = MQS_SW_RST_SHIFT, + .osr_mask = MQS_OVERSAMPLE_MASK, + .osr_shift = MQS_OVERSAMPLE_SHIFT, + .div_mask = MQS_CLK_DIV_MASK, + .div_shift = MQS_CLK_DIV_SHIFT, +}; + +static const struct fsl_mqs_soc_data fsl_mqs_imx6sx_data = { + .use_gpr = true, + .ctrl_off = IOMUXC_GPR2, + .en_mask = IMX6SX_GPR2_MQS_EN_MASK, + .en_shift = IMX6SX_GPR2_MQS_EN_SHIFT, + .rst_mask = IMX6SX_GPR2_MQS_SW_RST_MASK, + .rst_shift = IMX6SX_GPR2_MQS_SW_RST_SHIFT, + .osr_mask = IMX6SX_GPR2_MQS_OVERSAMPLE_MASK, + .osr_shift = IMX6SX_GPR2_MQS_OVERSAMPLE_SHIFT, + .div_mask = IMX6SX_GPR2_MQS_CLK_DIV_MASK, + .div_shift = IMX6SX_GPR2_MQS_CLK_DIV_SHIFT, +}; + static const struct of_device_id fsl_mqs_dt_ids[] = { - { .compatible = "fsl,imx8qm-mqs", }, - { .compatible = "fsl,imx6sx-mqs", }, + { .compatible = "fsl,imx8qm-mqs", .data = &fsl_mqs_imx8qm_data }, + { .compatible = "fsl,imx6sx-mqs", .data = &fsl_mqs_imx6sx_data }, {} }; MODULE_DEVICE_TABLE(of, fsl_mqs_dt_ids); -- cgit v1.2.3 From 2685d5046962f018b1a155b3eef316562414638b Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Sat, 21 May 2022 13:11:26 +0200 Subject: ASoC: stm32: dfsdm: fix typo in comment Spelling mistake (triple letters) in comment. Detected with the help of Coccinelle. Signed-off-by: Julia Lawall Link: https://lore.kernel.org/r/20220521111145.81697-76-Julia.Lawall@inria.fr Signed-off-by: Mark Brown --- sound/soc/stm/stm32_adfsdm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/stm/stm32_adfsdm.c b/sound/soc/stm/stm32_adfsdm.c index 6ee714542b84..122805160e70 100644 --- a/sound/soc/stm/stm32_adfsdm.c +++ b/sound/soc/stm/stm32_adfsdm.c @@ -296,7 +296,7 @@ static int stm32_adfsdm_pcm_new(struct snd_soc_component *component, static int stm32_adfsdm_dummy_cb(const void *data, void *private) { /* - * This dummmy callback is requested by iio_channel_get_all_cb() API, + * This dummy callback is requested by iio_channel_get_all_cb() API, * but the stm32_dfsdm_get_buff_cb() API is used instead, to optimize * DMA transfers. */ -- cgit v1.2.3 From ac8a2ea48001a4c336fbaaa977642d5ad79cdbd8 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 3 Jun 2022 13:50:03 +0200 Subject: ASoC: wm_adsp: Fix event generation for wm_adsp_fw_put() Currently wm_adsp_fw_put() returns 0 rather than 1 when updating the value of the control, meaning that no event is generated to userspace. Fix this by setting the default return value to 1, the code already exits early with a return value of 0 if the value is unchanged. Signed-off-by: Mark Brown Reviewed-by: Richard Fitzgerald Link: https://lore.kernel.org/r/20220603115003.3865834-1-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 7973a75cac05..6d7fd88243aa 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -333,7 +333,7 @@ int wm_adsp_fw_put(struct snd_kcontrol *kcontrol, struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; struct wm_adsp *dsp = snd_soc_component_get_drvdata(component); - int ret = 0; + int ret = 1; if (ucontrol->value.enumerated.item[0] == dsp[e->shift_l].fw) return 0; -- cgit v1.2.3 From 3929ead38d61abe6c5302adce1d490f5c041d4b3 Mon Sep 17 00:00:00 2001 From: Hui Wang Date: Mon, 30 May 2022 12:01:50 +0800 Subject: ASoC: nau8822: Add operation for internal PLL off and on We tried to enable the audio on an imx6sx EVB with the codec nau8822, after setting the internal PLL fractional parameters, the audio still couldn't work and the there was no sdma irq at all. After checking with the section "8.1.1 Phase Locked Loop (PLL) Design Example" of "NAU88C22 Datasheet Rev 0.6", we found we need to turn off the PLL before programming fractional parameters and turn on the PLL after programming. After this change, the audio driver could record and play sound and the sdma's irq is triggered when playing or recording. Cc: David Lin Cc: John Hsu Cc: Seven Li Signed-off-by: Hui Wang Link: https://lore.kernel.org/r/20220530040151.95221-2-hui.wang@canonical.com Signed-off-by: Mark Brown --- sound/soc/codecs/nau8822.c | 4 ++++ sound/soc/codecs/nau8822.h | 3 +++ 2 files changed, 7 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/nau8822.c b/sound/soc/codecs/nau8822.c index 66bbd8f4f1ad..08f6c56dc387 100644 --- a/sound/soc/codecs/nau8822.c +++ b/sound/soc/codecs/nau8822.c @@ -740,6 +740,8 @@ static int nau8822_set_pll(struct snd_soc_dai *dai, int pll_id, int source, pll_param->pll_int, pll_param->pll_frac, pll_param->mclk_scaler, pll_param->pre_factor); + snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, NAU8822_PLL_EN_MASK, NAU8822_PLL_OFF); snd_soc_component_update_bits(component, NAU8822_REG_PLL_N, NAU8822_PLLMCLK_DIV2 | NAU8822_PLLN_MASK, (pll_param->pre_factor ? NAU8822_PLLMCLK_DIV2 : 0) | @@ -757,6 +759,8 @@ static int nau8822_set_pll(struct snd_soc_dai *dai, int pll_id, int source, pll_param->mclk_scaler << NAU8822_MCLKSEL_SFT); snd_soc_component_update_bits(component, NAU8822_REG_CLOCKING, NAU8822_CLKM_MASK, NAU8822_CLKM_PLL); + snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, NAU8822_PLL_EN_MASK, NAU8822_PLL_ON); return 0; } diff --git a/sound/soc/codecs/nau8822.h b/sound/soc/codecs/nau8822.h index 489191ff187e..b45d42c15de6 100644 --- a/sound/soc/codecs/nau8822.h +++ b/sound/soc/codecs/nau8822.h @@ -90,6 +90,9 @@ #define NAU8822_REFIMP_3K 0x3 #define NAU8822_IOBUF_EN (0x1 << 2) #define NAU8822_ABIAS_EN (0x1 << 3) +#define NAU8822_PLL_EN_MASK (0x1 << 5) +#define NAU8822_PLL_ON (0x1 << 5) +#define NAU8822_PLL_OFF (0x0 << 5) /* NAU8822_REG_AUDIO_INTERFACE (0x4) */ #define NAU8822_AIFMT_MASK (0x3 << 3) -- cgit v1.2.3 From dd58365d43efccd87dbfc8f93eb3e61b9b4d64f8 Mon Sep 17 00:00:00 2001 From: Srinivasa Rao Mandadapu Date: Fri, 27 May 2022 19:40:08 +0530 Subject: ASoC: qcom: lpass-platform: Update VMA access permissions in mmap callback Replace page protection permissions from noncashed to writecombine, in lpass codec DMA path mmp callabck, to support 64 bit chromeOS. Avoid SIGBUS error in userspace caused by noncached permissions in 64 bit chromeOS. Signed-off-by: Srinivasa Rao Mandadapu Link: https://lore.kernel.org/r/1653660608-27245-1-git-send-email-quic_srivasam@quicinc.com Signed-off-by: Mark Brown --- sound/soc/qcom/lpass-platform.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c index f03a7ae49d50..b41ab7a321ae 100644 --- a/sound/soc/qcom/lpass-platform.c +++ b/sound/soc/qcom/lpass-platform.c @@ -898,7 +898,7 @@ static int lpass_platform_cdc_dma_mmap(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime = substream->runtime; unsigned long size, offset; - vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot); + vma->vm_page_prot = pgprot_writecombine(vma->vm_page_prot); size = vma->vm_end - vma->vm_start; offset = vma->vm_pgoff << PAGE_SHIFT; return io_remap_pfn_range(vma, vma->vm_start, -- cgit v1.2.3 From 33dbf3fc6942b53920296395bb4c81fb3cc5ebfd Mon Sep 17 00:00:00 2001 From: xliu Date: Thu, 2 Jun 2022 13:19:22 +0800 Subject: ASoC: Intel: cirrus-common: fix incorrect channel mapping The default mapping of ASPRX1 (DAC source) is slot 0. Change the slot mapping of right amplifiers (WR and TR) to slot 1 to receive right channel data. Also update the ACPI instance ID mapping according to HW configuration. Signed-off-by: xliu Signed-off-by: Brent Lu Acked-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220602051922.1232457-1-brent.lu@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_cirrus_common.c | 40 +++++++++++++++++++++++++++--- 1 file changed, 36 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/sof_cirrus_common.c b/sound/soc/intel/boards/sof_cirrus_common.c index e71d74ec1b0b..f4192df962d6 100644 --- a/sound/soc/intel/boards/sof_cirrus_common.c +++ b/sound/soc/intel/boards/sof_cirrus_common.c @@ -54,22 +54,29 @@ static struct snd_soc_dai_link_component cs35l41_components[] = { }, }; +/* + * Mapping between ACPI instance id and speaker position. + * + * Four speakers: + * 0: Tweeter left, 1: Woofer left + * 2: Tweeter right, 3: Woofer right + */ static struct snd_soc_codec_conf cs35l41_codec_conf[] = { { .dlc = COMP_CODEC_CONF(CS35L41_DEV0_NAME), - .name_prefix = "WL", + .name_prefix = "TL", }, { .dlc = COMP_CODEC_CONF(CS35L41_DEV1_NAME), - .name_prefix = "WR", + .name_prefix = "WL", }, { .dlc = COMP_CODEC_CONF(CS35L41_DEV2_NAME), - .name_prefix = "TL", + .name_prefix = "TR", }, { .dlc = COMP_CODEC_CONF(CS35L41_DEV3_NAME), - .name_prefix = "TR", + .name_prefix = "WR", }, }; @@ -101,6 +108,21 @@ static int cs35l41_init(struct snd_soc_pcm_runtime *rtd) return ret; } +/* + * Channel map: + * + * TL/WL: ASPRX1 on slot 0, ASPRX2 on slot 1 (default) + * TR/WR: ASPRX1 on slot 1, ASPRX2 on slot 0 + */ +static const struct { + unsigned int rx[2]; +} cs35l41_channel_map[] = { + {.rx = {0, 1}}, /* TL */ + {.rx = {0, 1}}, /* WL */ + {.rx = {1, 0}}, /* TR */ + {.rx = {1, 0}}, /* WR */ +}; + static int cs35l41_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -134,6 +156,16 @@ static int cs35l41_hw_params(struct snd_pcm_substream *substream, ret); return ret; } + + /* setup channel map */ + ret = snd_soc_dai_set_channel_map(codec_dai, 0, NULL, + ARRAY_SIZE(cs35l41_channel_map[i].rx), + (unsigned int *)cs35l41_channel_map[i].rx); + if (ret < 0) { + dev_err(codec_dai->dev, "fail to set channel map, ret %d\n", + ret); + return ret; + } } return 0; -- cgit v1.2.3 From 07c2307ce8b420e351e0635c690397ad7a9fab77 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 2 Jun 2022 17:21:14 +0100 Subject: ASoC: cs42l52: Fix TLV scales for mixer controls The datasheet specifies the range of the mixer volumes as between -51.5dB and 12dB with a 0.5dB step. Update the TLVs for this. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220602162119.3393857-2-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l52.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 9b182b585be4..02c25399cf8a 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -137,7 +137,7 @@ static DECLARE_TLV_DB_SCALE(mic_tlv, 1600, 100, 0); static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); -static DECLARE_TLV_DB_SCALE(mix_tlv, -50, 50, 0); +static DECLARE_TLV_DB_SCALE(mix_tlv, -5150, 50, 0); static DECLARE_TLV_DB_SCALE(beep_tlv, -56, 200, 0); @@ -364,7 +364,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { CS42L52_ADCB_VOL, 0, 0xA0, 0x78, ipd_tlv), SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", CS42L52_ADCA_MIXER_VOL, CS42L52_ADCB_MIXER_VOL, - 0, 0x19, 0x7F, ipd_tlv), + 0, 0x19, 0x7F, mix_tlv), SOC_DOUBLE("ADC Switch", CS42L52_ADC_MISC_CTL, 0, 1, 1, 0), -- cgit v1.2.3 From e9dad4de223ee5a4bd5e8b11931a2af8558da0bc Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 2 Jun 2022 17:21:15 +0100 Subject: ASoC: cs35l36: Update digital volume TLV The digital volume TLV specifies the step as 0.25dB but the actual step of the control is 0.125dB. Update the TLV to correct this. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220602162119.3393857-3-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs35l36.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs35l36.c b/sound/soc/codecs/cs35l36.c index 920190daa4d1..dfe85dc2cd20 100644 --- a/sound/soc/codecs/cs35l36.c +++ b/sound/soc/codecs/cs35l36.c @@ -444,7 +444,8 @@ static bool cs35l36_volatile_reg(struct device *dev, unsigned int reg) } } -static DECLARE_TLV_DB_SCALE(dig_vol_tlv, -10200, 25, 0); +static const DECLARE_TLV_DB_RANGE(dig_vol_tlv, 0, 912, + TLV_DB_MINMAX_ITEM(-10200, 1200)); static DECLARE_TLV_DB_SCALE(amp_gain_tlv, 0, 1, 1); static const char * const cs35l36_pcm_sftramp_text[] = { -- cgit v1.2.3 From 5a7f6cdd402e3da891d2768f1da1f3ea1664a2a2 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 2 Jun 2022 17:21:16 +0100 Subject: ASoC: cs53l30: Correct number of volume levels on SX controls This driver specified the maximum value rather than the number of volume levels on the SX controls, this is incorrect, so correct them. Reported-by: David Rhodes Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220602162119.3393857-4-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs53l30.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c index 703545273900..360ca2ffd506 100644 --- a/sound/soc/codecs/cs53l30.c +++ b/sound/soc/codecs/cs53l30.c @@ -348,22 +348,22 @@ static const struct snd_kcontrol_new cs53l30_snd_controls[] = { SOC_ENUM("ADC2 NG Delay", adc2_ng_delay_enum), SOC_SINGLE_SX_TLV("ADC1A PGA Volume", - CS53L30_ADC1A_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC1A_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC1B PGA Volume", - CS53L30_ADC1B_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC1B_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC2A PGA Volume", - CS53L30_ADC2A_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC2A_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC2B PGA Volume", - CS53L30_ADC2B_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC2B_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC1A Digital Volume", - CS53L30_ADC1A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC1A_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), SOC_SINGLE_SX_TLV("ADC1B Digital Volume", - CS53L30_ADC1B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC1B_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), SOC_SINGLE_SX_TLV("ADC2A Digital Volume", - CS53L30_ADC2A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC2A_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), SOC_SINGLE_SX_TLV("ADC2B Digital Volume", - CS53L30_ADC2B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC2B_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), }; static const struct snd_soc_dapm_widget cs53l30_dapm_widgets[] = { -- cgit v1.2.3 From cd6c0895b9d30b47d22293b9cddab3a8366e4a76 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 2 Jun 2022 17:21:17 +0100 Subject: ASoC: cs42l52: Correct TLV for Bypass Volume The Bypass Volume is accidentally using a -6dB minimum TLV rather than the correct -60dB minimum. Add a new TLV to correct this. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220602162119.3393857-5-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l52.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 02c25399cf8a..10e696406a71 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -137,6 +137,8 @@ static DECLARE_TLV_DB_SCALE(mic_tlv, 1600, 100, 0); static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); +static DECLARE_TLV_DB_SCALE(pass_tlv, -6000, 50, 0); + static DECLARE_TLV_DB_SCALE(mix_tlv, -5150, 50, 0); static DECLARE_TLV_DB_SCALE(beep_tlv, -56, 200, 0); @@ -351,7 +353,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { CS42L52_SPKB_VOL, 0, 0x40, 0xC0, hl_tlv), SOC_DOUBLE_R_SX_TLV("Bypass Volume", CS42L52_PASSTHRUA_VOL, - CS42L52_PASSTHRUB_VOL, 0, 0x88, 0x90, pga_tlv), + CS42L52_PASSTHRUB_VOL, 0, 0x88, 0x90, pass_tlv), SOC_DOUBLE("Bypass Mute", CS42L52_MISC_CTL, 4, 5, 1, 0), -- cgit v1.2.3 From 0c9495ee315e13cce3e3eb588efdcb107b566aab Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 2 Jun 2022 17:21:18 +0100 Subject: ASoC: cs42l56: Correct typo in minimum level for SX volume controls A couple of the SX volume controls specify 0x84 as the lowest volume value, however the correct value from the datasheet is 0x44. The datasheet don't include spaces in the value it displays as binary so this was almost certainly just a typo reading 1000100. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220602162119.3393857-6-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l56.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index dc23007336c5..510c94265b1f 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -391,9 +391,9 @@ static const struct snd_kcontrol_new cs42l56_snd_controls[] = { SOC_DOUBLE("ADC Boost Switch", CS42L56_GAIN_BIAS_CTL, 3, 2, 1, 1), SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L56_HPA_VOLUME, - CS42L56_HPB_VOLUME, 0, 0x84, 0x48, hl_tlv), + CS42L56_HPB_VOLUME, 0, 0x44, 0x48, hl_tlv), SOC_DOUBLE_R_SX_TLV("LineOut Volume", CS42L56_LOA_VOLUME, - CS42L56_LOB_VOLUME, 0, 0x84, 0x48, hl_tlv), + CS42L56_LOB_VOLUME, 0, 0x44, 0x48, hl_tlv), SOC_SINGLE_TLV("Bass Shelving Volume", CS42L56_TONE_CTL, 0, 0x00, 1, tone_tlv), -- cgit v1.2.3 From 513abe2460de2feaa56a66270efda5fa7a788459 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 2 Jun 2022 17:21:19 +0100 Subject: ASoC: cs42l51: Correct minimum value for SX volume control The minimum value for the PGA Volume is given as 0x1A, however the values from there to 0x19 are all the same volume and this is not represented in the TLV structure. The number of volumes given is correct so this leads to all the volumes being shifted. Move the minimum value up to 0x19 to fix this. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220602162119.3393857-7-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l51.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index aff618513c75..0e933181b5db 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -143,7 +143,7 @@ static const struct snd_kcontrol_new cs42l51_snd_controls[] = { 0, 0xA0, 96, adc_att_tlv), SOC_DOUBLE_R_SX_TLV("PGA Volume", CS42L51_ALC_PGA_CTL, CS42L51_ALC_PGB_CTL, - 0, 0x1A, 30, pga_tlv), + 0, 0x19, 30, pga_tlv), SOC_SINGLE("Playback Deemphasis Switch", CS42L51_DAC_CTL, 3, 1, 0), SOC_SINGLE("Auto-Mute Switch", CS42L51_DAC_CTL, 2, 1, 0), SOC_SINGLE("Soft Ramp Switch", CS42L51_DAC_CTL, 1, 1, 0), -- cgit v1.2.3 From eff8f2aeaf0c1b529d918c9f9569577dff600dc5 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 2 Jun 2022 15:52:57 +0200 Subject: ASoC: cx2072x: Use modern ASoC DAI format terminology As part of moving to remove the old style defines for the bus clocks update the cx2072x driver to use more modern terminology for clocking. Signed-off-by: Mark Brown Link: https://lore.kernel.org/r/20220602135316.3554400-2-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/cx2072x.c | 17 +++++++---------- 1 file changed, 7 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cx2072x.c b/sound/soc/codecs/cx2072x.c index b35debb5818d..b6667e8a6099 100644 --- a/sound/soc/codecs/cx2072x.c +++ b/sound/soc/codecs/cx2072x.c @@ -710,22 +710,19 @@ static int cx2072x_config_i2spcm(struct cx2072x_priv *cx2072x) regdbt2.ulval = 0xac; - /* set master/slave */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: reg2.r.tx_master = 1; reg3.r.rx_master = 1; - dev_dbg(dev, "Sets Master mode\n"); break; - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBC_CFC: reg2.r.tx_master = 0; reg3.r.rx_master = 0; - dev_dbg(dev, "Sets Slave mode\n"); break; default: - dev_err(dev, "Unsupported DAI master mode\n"); + dev_err(dev, "Unsupported DAI clocking mode\n"); return -EINVAL; } @@ -1009,9 +1006,9 @@ static int cx2072x_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) dev_dbg(dev, "set_dai_fmt- %08x\n", fmt); /* set master/slave */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - case SND_SOC_DAIFMT_CBS_CFS: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: + case SND_SOC_DAIFMT_CBC_CFC: break; default: -- cgit v1.2.3 From 573a9a37b6fcef6dc3977ca11a671f82b1c1b606 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 2 Jun 2022 15:52:58 +0200 Subject: ASoC: max98090: Use modern ASoC DAI format terminology As part of moving to remove the old style defines for the bus clocks update the max98090 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown Link: https://lore.kernel.org/r/20220602135316.3554400-3-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/max98090.c | 12 +++++------- 1 file changed, 5 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 576277a82d41..72471cdb2229 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -1591,9 +1591,9 @@ static int max98090_dai_set_fmt(struct snd_soc_dai *codec_dai, cdata->fmt = fmt; regval = 0; - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: - /* Set to slave mode PLL - MAS mode off */ + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBC_CFC: + /* Set to consumer mode PLL - MAS mode off */ snd_soc_component_write(component, M98090_REG_CLOCK_RATIO_NI_MSB, 0x00); snd_soc_component_write(component, @@ -1602,8 +1602,8 @@ static int max98090_dai_set_fmt(struct snd_soc_dai *codec_dai, M98090_USE_M1_MASK, 0); max98090->master = false; break; - case SND_SOC_DAIFMT_CBM_CFM: - /* Set to master mode */ + case SND_SOC_DAIFMT_CBP_CFP: + /* Set to provider mode */ if (max98090->tdm_slots == 4) { /* TDM */ regval |= M98090_MAS_MASK | @@ -1619,8 +1619,6 @@ static int max98090_dai_set_fmt(struct snd_soc_dai *codec_dai, } max98090->master = true; break; - case SND_SOC_DAIFMT_CBS_CFM: - case SND_SOC_DAIFMT_CBM_CFS: default: dev_err(component->dev, "DAI clock mode unsupported"); return -EINVAL; -- cgit v1.2.3 From cd0df1706d181bf103d0f02e6c008c2386772eb1 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 2 Jun 2022 15:52:59 +0200 Subject: ASoC: rk3328: Use modern ASoC DAI format terminology As part of moving to remove the old style defines for the bus clocks update the rk3328 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown Link: https://lore.kernel.org/r/20220602135316.3554400-4-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/rk3328_codec.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rk3328_codec.c b/sound/soc/codecs/rk3328_codec.c index 86b679cf7aef..1d523bfd9d84 100644 --- a/sound/soc/codecs/rk3328_codec.c +++ b/sound/soc/codecs/rk3328_codec.c @@ -69,11 +69,11 @@ static int rk3328_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) snd_soc_component_get_drvdata(dai->component); unsigned int val; - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBC_CFC: val = PIN_DIRECTION_IN | DAC_I2S_MODE_SLAVE; break; - case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBP_CFP: val = PIN_DIRECTION_OUT | DAC_I2S_MODE_MASTER; break; default: -- cgit v1.2.3 From ef08b481ae78eb89672bdf67ed306a43065253b3 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 2 Jun 2022 15:53:00 +0200 Subject: ASoC: sta32x: Use modern ASoC DAI format terminology As part of moving to remove the old style defines for the bus clocks update the sta32x driver to use more modern terminology for clocking. Signed-off-by: Mark Brown Link: https://lore.kernel.org/r/20220602135316.3554400-5-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/sta32x.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 8585cbef4c9b..17e5077f26b0 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -601,8 +601,8 @@ static int sta32x_set_dai_fmt(struct snd_soc_dai *codec_dai, struct sta32x_priv *sta32x = snd_soc_component_get_drvdata(component); u8 confb = 0; - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBC_CFC: break; default: return -EINVAL; -- cgit v1.2.3 From def5b3774a48ed06e69b56af8317cb563bbd9ceb Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 2 Jun 2022 15:53:01 +0200 Subject: ASoC: sta350: Use modern ASoC DAI format terminology As part of moving to remove the old style defines for the bus clocks update the sta350 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown Link: https://lore.kernel.org/r/20220602135316.3554400-6-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/sta350.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sta350.c b/sound/soc/codecs/sta350.c index 9189fb3648f7..b2d15d20fe63 100644 --- a/sound/soc/codecs/sta350.c +++ b/sound/soc/codecs/sta350.c @@ -630,8 +630,8 @@ static int sta350_set_dai_fmt(struct snd_soc_dai *codec_dai, struct sta350_priv *sta350 = snd_soc_component_get_drvdata(component); unsigned int confb = 0; - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBC_CFC: break; default: return -EINVAL; -- cgit v1.2.3 From d7e98b570e801375130ed4796bcbb35a39669d44 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 2 Jun 2022 15:53:02 +0200 Subject: ASoC: sti-sas: Use modern ASoC DAI format terminology As part of moving to remove the old style defines for the bus clocks update the sti-sas driver to use more modern terminology for clocking. Signed-off-by: Mark Brown Link: https://lore.kernel.org/r/20220602135316.3554400-7-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/sti-sas.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sti-sas.c b/sound/soc/codecs/sti-sas.c index 3be4940e3c77..10a6a112f4b4 100644 --- a/sound/soc/codecs/sti-sas.c +++ b/sound/soc/codecs/sti-sas.c @@ -199,10 +199,10 @@ static int stih407_sas_dac_mute(struct snd_soc_dai *dai, int mute, int stream) static int sti_sas_spdif_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { - if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) { + if ((fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) != SND_SOC_DAIFMT_CBC_CFC) { dev_err(dai->component->dev, - "%s: ERROR: Unsupporter master mask 0x%x\n", - __func__, fmt & SND_SOC_DAIFMT_MASTER_MASK); + "%s: ERROR: Unsupported clocking mask 0x%x\n", + __func__, fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK); return -EINVAL; } -- cgit v1.2.3 From 6b486af2ab946cbcad5c95f8daa1f4a8a53f25c5 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 2 Jun 2022 15:53:03 +0200 Subject: ASoC: tas2552: Use modern ASoC DAI format terminology As part of moving to remove the old style defines for the bus clocks update the tas2552 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown Link: https://lore.kernel.org/r/20220602135316.3554400-8-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/tas2552.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index b5c9c61ff5a8..c98a9332dcc0 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -347,17 +347,17 @@ static int tas2552_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) struct tas2552_data *tas2552 = dev_get_drvdata(component->dev); u8 serial_format; - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBC_CFC: serial_format = 0x00; break; - case SND_SOC_DAIFMT_CBS_CFM: + case SND_SOC_DAIFMT_CBC_CFP: serial_format = TAS2552_WCLKDIR; break; - case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBP_CFC: serial_format = TAS2552_BCLKDIR; break; - case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBP_CFP: serial_format = (TAS2552_BCLKDIR | TAS2552_WCLKDIR); break; default: -- cgit v1.2.3 From f8a4018c826fde6137425bbdbe524d5973feb173 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 2 Jun 2022 15:53:04 +0200 Subject: ASoC: tas2770: Use modern ASoC DAI format terminology As part of moving to remove the old style defines for the bus clocks update the tas2770 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown Link: https://lore.kernel.org/r/20220602135316.3554400-9-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/tas2770.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tas2770.c b/sound/soc/codecs/tas2770.c index c1dbd978d550..f6037a148cb6 100644 --- a/sound/soc/codecs/tas2770.c +++ b/sound/soc/codecs/tas2770.c @@ -340,11 +340,11 @@ static int tas2770_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) u8 tdm_rx_start_slot = 0, asi_cfg_1 = 0; int ret; - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBC_CFC: break; default: - dev_err(tas2770->dev, "ASI format master is not found\n"); + dev_err(tas2770->dev, "ASI invalid DAI clocking\n"); return -EINVAL; } -- cgit v1.2.3 From 7c5c399fb97e3f7a88d1b154f610cab4d9253955 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 2 Jun 2022 15:53:05 +0200 Subject: ASoC: tas5086: Use modern ASoC DAI format terminology As part of moving to remove the old style defines for the bus clocks update the tas5086 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown Link: https://lore.kernel.org/r/20220602135316.3554400-10-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/tas5086.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c index 5c0df3cd4832..05b57bb1aea0 100644 --- a/sound/soc/codecs/tas5086.c +++ b/sound/soc/codecs/tas5086.c @@ -318,7 +318,7 @@ static int tas5086_set_dai_fmt(struct snd_soc_dai *codec_dai, struct tas5086_private *priv = snd_soc_component_get_drvdata(component); /* The TAS5086 can only be slave to all clocks */ - if ((format & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) { + if ((format & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) != SND_SOC_DAIFMT_CBC_CFC) { dev_err(component->dev, "Invalid clocking mode\n"); return -EINVAL; } -- cgit v1.2.3 From 9f6654c3162a4e64265c62bea433550fce4beffd Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 2 Jun 2022 15:53:06 +0200 Subject: ASoC: tas5720: Use modern ASoC DAI format terminology As part of moving to remove the old style defines for the bus clocks update the tas5720 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown Link: https://lore.kernel.org/r/20220602135316.3554400-11-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/tas5720.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tas5720.c b/sound/soc/codecs/tas5720.c index 17034abef568..2ee06a95f3e4 100644 --- a/sound/soc/codecs/tas5720.c +++ b/sound/soc/codecs/tas5720.c @@ -89,8 +89,8 @@ static int tas5720_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) u8 serial_format; int ret; - if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) { - dev_vdbg(component->dev, "DAI Format master is not found\n"); + if ((fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) != SND_SOC_DAIFMT_CBC_CFC) { + dev_vdbg(component->dev, "DAI clocking invalid\n"); return -EINVAL; } -- cgit v1.2.3 From f025fcc466cc03fa4f5ae245b6848629b846edff Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 2 Jun 2022 15:53:07 +0200 Subject: ASoC: tas6424: Use modern ASoC DAI format terminology As part of moving to remove the old style defines for the bus clocks update the tas6424 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown Link: https://lore.kernel.org/r/20220602135316.3554400-12-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/tas6424.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tas6424.c b/sound/soc/codecs/tas6424.c index 22b53856e691..9c9a6ec4d977 100644 --- a/sound/soc/codecs/tas6424.c +++ b/sound/soc/codecs/tas6424.c @@ -160,11 +160,11 @@ static int tas6424_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) dev_dbg(component->dev, "%s() fmt=0x%0x\n", __func__, fmt); /* clock masters */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBC_CFC: break; default: - dev_err(component->dev, "Invalid DAI master/slave interface\n"); + dev_err(component->dev, "Invalid DAI clocking\n"); return -EINVAL; } -- cgit v1.2.3 From 5fc4ed4bda465fb826bea7c6a7b15657154787ce Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 2 Jun 2022 15:53:08 +0200 Subject: ASoC: uda1334: Use modern ASoC DAI format terminology As part of moving to remove the old style defines for the bus clocks update the uda1334 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown Link: https://lore.kernel.org/r/20220602135316.3554400-13-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/uda1334.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/uda1334.c b/sound/soc/codecs/uda1334.c index 8670a2a05a56..9d5ed34e5420 100644 --- a/sound/soc/codecs/uda1334.c +++ b/sound/soc/codecs/uda1334.c @@ -169,7 +169,7 @@ static int uda1334_set_dai_sysclk(struct snd_soc_dai *codec_dai, static int uda1334_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { fmt &= (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_INV_MASK | - SND_SOC_DAIFMT_MASTER_MASK); + SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK); if (fmt != (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBC_CFC)) { -- cgit v1.2.3 From ad60ff09801fa1841dcdcf1f6ad1fa0e09ad0693 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 2 Jun 2022 15:53:09 +0200 Subject: ASoC: tlv320adc3xxx: Use modern ASoC DAI format terminology As part of moving to remove the old style defines for the bus clocks update the tlv320aic3xxx driver to use more modern terminology for clocking. Signed-off-by: Mark Brown Link: https://lore.kernel.org/r/20220602135316.3554400-14-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320adc3xxx.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320adc3xxx.c b/sound/soc/codecs/tlv320adc3xxx.c index 82532ad00c3c..748998e48af9 100644 --- a/sound/soc/codecs/tlv320adc3xxx.c +++ b/sound/soc/codecs/tlv320adc3xxx.c @@ -1252,8 +1252,7 @@ static int adc3xxx_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) int master = 0; int ret; - /* set master/slave audio interface */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { case SND_SOC_DAIFMT_CBP_CFP: master = 1; clkdir = ADC3XXX_BCLK_MASTER | ADC3XXX_WCLK_MASTER; -- cgit v1.2.3 From 10649fa392c9abb6e9b258f7af9577596339fbe2 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 2 Jun 2022 15:53:10 +0200 Subject: ASoC: tlv320adcx140: Use modern ASoC DAI format terminology As part of moving to remove the old style defines for the bus clocks update the tlv320adcx140 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown Link: https://lore.kernel.org/r/20220602135316.3554400-15-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320adcx140.c | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c index b55f0b836932..de5b184a701e 100644 --- a/sound/soc/codecs/tlv320adcx140.c +++ b/sound/soc/codecs/tlv320adcx140.c @@ -713,16 +713,14 @@ static int adcx140_set_dai_fmt(struct snd_soc_dai *codec_dai, bool inverted_bclk = false; /* set master/slave audio interface */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: iface_reg2 |= ADCX140_BCLK_FSYNC_MASTER; break; - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBC_CFC: break; - case SND_SOC_DAIFMT_CBS_CFM: - case SND_SOC_DAIFMT_CBM_CFS: default: - dev_err(component->dev, "Invalid DAI master/slave interface\n"); + dev_err(component->dev, "Invalid DAI clock provider\n"); return -EINVAL; } -- cgit v1.2.3 From b9ff35c7afc6ae1bddca3f84fb23a3d903a62a23 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 2 Jun 2022 15:53:11 +0200 Subject: ASoC: tlv320aic23: Use modern ASoC DAI format terminology As part of moving to remove the old style defines for the bus clocks update the tlv320aic23 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown Reported-by: kernel test robot Link: https://lore.kernel.org/r/20220602135316.3554400-16-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic23.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 2400093e2c99..c86ca793a2b6 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -429,12 +429,11 @@ static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai, iface_reg = snd_soc_component_read(component, TLV320AIC23_DIGT_FMT) & (~0x03); - /* set master/slave audio interface */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: iface_reg |= TLV320AIC23_MS_MASTER; break; - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBC_CFC: iface_reg &= ~TLV320AIC23_MS_MASTER; break; default: -- cgit v1.2.3 From 8d322f170b09989f47614c1a663371647f03176f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 2 Jun 2022 15:53:12 +0200 Subject: ASoC: tlv320aic26: Use modern ASoC DAI format terminology As part of moving to remove the old style defines for the bus clocks update the tlv320aic26 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown Reported-by: kernel test robot Link: https://lore.kernel.org/r/20220602135316.3554400-17-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic26.c | 15 +++++++-------- 1 file changed, 7 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index 077415a57225..f85f8061639f 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -32,7 +32,7 @@ struct aic26 { struct spi_device *spi; struct regmap *regmap; struct snd_soc_component *component; - int master; + int clock_provider; int datfm; int mclk; @@ -117,8 +117,8 @@ static int aic26_hw_params(struct snd_pcm_substream *substream, reg = dval << 2; snd_soc_component_write(component, AIC26_REG_PLL_PROG2, reg); - /* Audio Control 3 (master mode, fsref rate) */ - if (aic26->master) + /* Audio Control 3 (clock provider mode, fsref rate) */ + if (aic26->clock_provider) reg = 0x0800; if (fsref == 48000) reg = 0x2000; @@ -178,10 +178,9 @@ static int aic26_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) dev_dbg(&aic26->spi->dev, "aic26_set_fmt(dai=%p, fmt==%i)\n", codec_dai, fmt); - /* set master/slave audio interface */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: aic26->master = 1; break; - case SND_SOC_DAIFMT_CBS_CFS: aic26->master = 0; break; + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: aic26->clock_provider = 1; break; + case SND_SOC_DAIFMT_CBC_CFC: aic26->clock_provider = 0; break; default: dev_dbg(&aic26->spi->dev, "bad master\n"); return -EINVAL; } @@ -363,7 +362,7 @@ static int aic26_spi_probe(struct spi_device *spi) /* Initialize the driver data */ aic26->spi = spi; dev_set_drvdata(&spi->dev, aic26); - aic26->master = 1; + aic26->clock_provider = 1; ret = devm_snd_soc_register_component(&spi->dev, &aic26_soc_component_dev, &aic26_dai, 1); -- cgit v1.2.3 From 2fd8298aed2228b8c6b94edf820121da25b3f5e2 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 2 Jun 2022 15:53:13 +0200 Subject: ASoC: tlv320aic31xx: Use modern ASoC DAI format terminology As part of moving to remove the old style defines for the bus clocks update the tlv320aic31xx driver to use more modern terminology for clocking. Signed-off-by: Mark Brown Link: https://lore.kernel.org/r/20220602135316.3554400-18-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic31xx.c | 19 +++++++++---------- 1 file changed, 9 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index b2e59581c17a..aacee2367992 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1033,8 +1033,8 @@ static int aic31xx_clock_master_routes(struct snd_soc_component *component, struct aic31xx_priv *aic31xx = snd_soc_component_get_drvdata(component); int ret; - fmt &= SND_SOC_DAIFMT_MASTER_MASK; - if (fmt == SND_SOC_DAIFMT_CBS_CFS && + fmt &= SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK; + if (fmt == SND_SOC_DAIFMT_CBC_CFC && aic31xx->master_dapm_route_applied) { /* * Remove the DAPM route(s) for codec clock master modes, @@ -1051,7 +1051,7 @@ static int aic31xx_clock_master_routes(struct snd_soc_component *component, return ret; aic31xx->master_dapm_route_applied = false; - } else if (fmt != SND_SOC_DAIFMT_CBS_CFS && + } else if (fmt != SND_SOC_DAIFMT_CBC_CFC && !aic31xx->master_dapm_route_applied) { /* * Add the needed DAPM route(s) for codec clock master modes, @@ -1083,21 +1083,20 @@ static int aic31xx_set_dai_fmt(struct snd_soc_dai *codec_dai, dev_dbg(component->dev, "## %s: fmt = 0x%x\n", __func__, fmt); - /* set master/slave audio interface */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: iface_reg1 |= AIC31XX_BCLK_MASTER | AIC31XX_WCLK_MASTER; break; - case SND_SOC_DAIFMT_CBS_CFM: + case SND_SOC_DAIFMT_CBC_CFP: iface_reg1 |= AIC31XX_WCLK_MASTER; break; - case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBP_CFC: iface_reg1 |= AIC31XX_BCLK_MASTER; break; - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBC_CFC: break; default: - dev_err(component->dev, "Invalid DAI master/slave interface\n"); + dev_err(component->dev, "Invalid DAI clock provider\n"); return -EINVAL; } -- cgit v1.2.3 From 0cc5a137f7a3ba6fec069d8d222020f0927a18ef Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 2 Jun 2022 15:53:14 +0200 Subject: ASoC: tlv320aic32x4: Use modern ASoC DAI format terminology As part of moving to remove the old style defines for the bus clocks update the tlv320aic32x4 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown Link: https://lore.kernel.org/r/20220602135316.3554400-19-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic32x4.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 8f42fd7bc053..a8e6adf62ac8 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -615,15 +615,14 @@ static int aic32x4_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) u8 iface_reg_2 = 0; u8 iface_reg_3 = 0; - /* set master/slave audio interface */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: iface_reg_1 |= AIC32X4_BCLKMASTER | AIC32X4_WCLKMASTER; break; - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBC_CFC: break; default: - printk(KERN_ERR "aic32x4: invalid DAI master/slave interface\n"); + printk(KERN_ERR "aic32x4: invalid clock provider\n"); return -EINVAL; } -- cgit v1.2.3 From 83a5f86903fbaf9c47c13975eb6f2fbd16d7f865 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 2 Jun 2022 15:53:15 +0200 Subject: ASoC: tlv320aic33: Use modern ASoC DAI format terminology As part of moving to remove the old style defines for the bus clocks update the tlv320aic33 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown Link: https://lore.kernel.org/r/20220602135316.3554400-20-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index d53037b1509d..610e41bbf388 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1253,22 +1253,21 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai, iface_areg = snd_soc_component_read(component, AIC3X_ASD_INTF_CTRLA) & 0x3f; iface_breg = snd_soc_component_read(component, AIC3X_ASD_INTF_CTRLB) & 0x3f; - /* set master/slave audio interface */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: aic3x->master = 1; iface_areg |= BIT_CLK_MASTER | WORD_CLK_MASTER; break; - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBC_CFC: aic3x->master = 0; iface_areg &= ~(BIT_CLK_MASTER | WORD_CLK_MASTER); break; - case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBP_CFC: aic3x->master = 1; iface_areg |= BIT_CLK_MASTER; iface_areg &= ~WORD_CLK_MASTER; break; - case SND_SOC_DAIFMT_CBS_CFM: + case SND_SOC_DAIFMT_CBC_CFP: aic3x->master = 1; iface_areg |= WORD_CLK_MASTER; iface_areg &= ~BIT_CLK_MASTER; -- cgit v1.2.3 From 894bf75bb1f6c274cdd877879d9215abd6ed4b1b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 2 Jun 2022 15:53:16 +0200 Subject: ASoC: tlv320dac3x: Use modern ASoC DAI format terminology As part of moving to remove the old style defines for the bus clocks update the tlv320dac3x driver to use more modern terminology for clocking. Signed-off-by: Mark Brown Link: https://lore.kernel.org/r/20220602135316.3554400-21-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320dac33.c | 12 +++++------- 1 file changed, 5 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 66f1d1cd6cf0..371026eb8f41 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -1317,16 +1317,14 @@ static int dac33_set_dai_fmt(struct snd_soc_dai *codec_dai, aictrl_a = dac33_read_reg_cache(component, DAC33_SER_AUDIOIF_CTRL_A); aictrl_b = dac33_read_reg_cache(component, DAC33_SER_AUDIOIF_CTRL_B); - /* set master/slave audio interface */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - /* Codec Master */ + + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: aictrl_a |= (DAC33_MSBCLK | DAC33_MSWCLK); break; - case SND_SOC_DAIFMT_CBS_CFS: - /* Codec Slave */ + case SND_SOC_DAIFMT_CBC_CFC: if (dac33->fifo_mode) { - dev_err(component->dev, "FIFO mode requires master mode\n"); + dev_err(component->dev, "FIFO mode requires provider mode\n"); return -EINVAL; } else aictrl_a &= ~(DAC33_MSBCLK | DAC33_MSWCLK); -- cgit v1.2.3 From 612c4695e312c753a8b06f6b052cea3d8338e3c3 Mon Sep 17 00:00:00 2001 From: Jiaxin Yu Date: Mon, 23 May 2022 21:28:39 +0800 Subject: ASoC: mediatek: mt6366: support for mt6366 codec Mt6366 is a new version of mt6358, and they are same about audio part. So we can reuse the driver of mt6358 that add a new compatible string inside of the mt6358 driver. Signed-off-by: Jiaxin Yu Link: https://lore.kernel.org/r/20220523132858.22166-2-jiaxin.yu@mediatek.com Signed-off-by: Mark Brown --- sound/soc/codecs/mt6358.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/mt6358.c b/sound/soc/codecs/mt6358.c index 60b209efe52d..93f35e8d26fc 100644 --- a/sound/soc/codecs/mt6358.c +++ b/sound/soc/codecs/mt6358.c @@ -2479,6 +2479,7 @@ static int mt6358_platform_driver_probe(struct platform_device *pdev) static const struct of_device_id mt6358_of_match[] = { {.compatible = "mediatek,mt6358-sound",}, + {.compatible = "mediatek,mt6366-sound",}, {} }; MODULE_DEVICE_TABLE(of, mt6358_of_match); -- cgit v1.2.3 From 58949aa35c0f74a98b03864817354d85f452a51c Mon Sep 17 00:00:00 2001 From: Jiaxin Yu Date: Mon, 23 May 2022 21:28:41 +0800 Subject: ASoC: mediatek: mt8186: support audsys clock control Add mt8186 audio cg control. Audio clock gates are registered to CCF for reference count and clock parent management. Signed-off-by: Jiaxin Yu Reviewed-by: AngeloGioacchino Del Regno Link: https://lore.kernel.org/r/20220523132858.22166-4-jiaxin.yu@mediatek.com Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8186/mt8186-audsys-clk.c | 150 ++++++++++++++++++++++++ sound/soc/mediatek/mt8186/mt8186-audsys-clk.h | 15 +++ sound/soc/mediatek/mt8186/mt8186-audsys-clkid.h | 45 +++++++ 3 files changed, 210 insertions(+) create mode 100644 sound/soc/mediatek/mt8186/mt8186-audsys-clk.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-audsys-clk.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-audsys-clkid.h (limited to 'sound') diff --git a/sound/soc/mediatek/mt8186/mt8186-audsys-clk.c b/sound/soc/mediatek/mt8186/mt8186-audsys-clk.c new file mode 100644 index 000000000000..578969ca91c8 --- /dev/null +++ b/sound/soc/mediatek/mt8186/mt8186-audsys-clk.c @@ -0,0 +1,150 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// mt8186-audsys-clk.h -- Mediatek 8186 audsys clock control +// +// Copyright (c) 2022 MediaTek Inc. +// Author: Jiaxin Yu + +#include +#include +#include +#include "mt8186-afe-common.h" +#include "mt8186-audsys-clk.h" +#include "mt8186-audsys-clkid.h" +#include "mt8186-reg.h" + +struct afe_gate { + int id; + const char *name; + const char *parent_name; + int reg; + u8 bit; + const struct clk_ops *ops; + unsigned long flags; + u8 cg_flags; +}; + +#define GATE_AFE_FLAGS(_id, _name, _parent, _reg, _bit, _flags, _cgflags) {\ + .id = _id, \ + .name = _name, \ + .parent_name = _parent, \ + .reg = _reg, \ + .bit = _bit, \ + .flags = _flags, \ + .cg_flags = _cgflags, \ + } + +#define GATE_AFE(_id, _name, _parent, _reg, _bit) \ + GATE_AFE_FLAGS(_id, _name, _parent, _reg, _bit, \ + CLK_SET_RATE_PARENT, CLK_GATE_SET_TO_DISABLE) + +#define GATE_AUD0(_id, _name, _parent, _bit) \ + GATE_AFE(_id, _name, _parent, AUDIO_TOP_CON0, _bit) + +#define GATE_AUD1(_id, _name, _parent, _bit) \ + GATE_AFE(_id, _name, _parent, AUDIO_TOP_CON1, _bit) + +#define GATE_AUD2(_id, _name, _parent, _bit) \ + GATE_AFE(_id, _name, _parent, AUDIO_TOP_CON2, _bit) + +static const struct afe_gate aud_clks[CLK_AUD_NR_CLK] = { + /* AUD0 */ + GATE_AUD0(CLK_AUD_AFE, "aud_afe_clk", "top_audio", 2), + GATE_AUD0(CLK_AUD_22M, "aud_apll22m_clk", "top_aud_engen1", 8), + GATE_AUD0(CLK_AUD_24M, "aud_apll24m_clk", "top_aud_engen2", 9), + GATE_AUD0(CLK_AUD_APLL2_TUNER, "aud_apll2_tuner_clk", "top_aud_engen2", 18), + GATE_AUD0(CLK_AUD_APLL_TUNER, "aud_apll_tuner_clk", "top_aud_engen1", 19), + GATE_AUD0(CLK_AUD_TDM, "aud_tdm_clk", "top_aud_1", 20), + GATE_AUD0(CLK_AUD_ADC, "aud_adc_clk", "top_audio", 24), + GATE_AUD0(CLK_AUD_DAC, "aud_dac_clk", "top_audio", 25), + GATE_AUD0(CLK_AUD_DAC_PREDIS, "aud_dac_predis_clk", "top_audio", 26), + GATE_AUD0(CLK_AUD_TML, "aud_tml_clk", "top_audio", 27), + GATE_AUD0(CLK_AUD_NLE, "aud_nle_clk", "top_audio", 28), + + /* AUD1 */ + GATE_AUD1(CLK_AUD_I2S1_BCLK, "aud_i2s1_bclk", "top_audio", 4), + GATE_AUD1(CLK_AUD_I2S2_BCLK, "aud_i2s2_bclk", "top_audio", 5), + GATE_AUD1(CLK_AUD_I2S3_BCLK, "aud_i2s3_bclk", "top_audio", 6), + GATE_AUD1(CLK_AUD_I2S4_BCLK, "aud_i2s4_bclk", "top_audio", 7), + GATE_AUD1(CLK_AUD_CONNSYS_I2S_ASRC, "aud_connsys_i2s_asrc", "top_audio", 12), + GATE_AUD1(CLK_AUD_GENERAL1_ASRC, "aud_general1_asrc", "top_audio", 13), + GATE_AUD1(CLK_AUD_GENERAL2_ASRC, "aud_general2_asrc", "top_audio", 14), + GATE_AUD1(CLK_AUD_DAC_HIRES, "aud_dac_hires_clk", "top_audio_h", 15), + GATE_AUD1(CLK_AUD_ADC_HIRES, "aud_adc_hires_clk", "top_audio_h", 16), + GATE_AUD1(CLK_AUD_ADC_HIRES_TML, "aud_adc_hires_tml", "top_audio_h", 17), + GATE_AUD1(CLK_AUD_ADDA6_ADC, "aud_adda6_adc", "top_audio", 20), + GATE_AUD1(CLK_AUD_ADDA6_ADC_HIRES, "aud_adda6_adc_hires", "top_audio_h", 21), + GATE_AUD1(CLK_AUD_3RD_DAC, "aud_3rd_dac", "top_audio", 28), + GATE_AUD1(CLK_AUD_3RD_DAC_PREDIS, "aud_3rd_dac_predis", "top_audio", 29), + GATE_AUD1(CLK_AUD_3RD_DAC_TML, "aud_3rd_dac_tml", "top_audio", 30), + GATE_AUD1(CLK_AUD_3RD_DAC_HIRES, "aud_3rd_dac_hires", "top_audio_h", 31), + + /* AUD2 */ + GATE_AUD2(CLK_AUD_ETDM_IN1_BCLK, "aud_etdm_in1_bclk", "top_audio", 23), + GATE_AUD2(CLK_AUD_ETDM_OUT1_BCLK, "aud_etdm_out1_bclk", "top_audio", 24), +}; + +int mt8186_audsys_clk_register(struct mtk_base_afe *afe) +{ + struct mt8186_afe_private *afe_priv = afe->platform_priv; + struct clk *clk; + struct clk_lookup *cl; + int i; + + afe_priv->lookup = devm_kcalloc(afe->dev, CLK_AUD_NR_CLK, + sizeof(*afe_priv->lookup), + GFP_KERNEL); + + if (!afe_priv->lookup) + return -ENOMEM; + + for (i = 0; i < ARRAY_SIZE(aud_clks); i++) { + const struct afe_gate *gate = &aud_clks[i]; + + clk = clk_register_gate(afe->dev, gate->name, gate->parent_name, + gate->flags, afe->base_addr + gate->reg, + gate->bit, gate->cg_flags, NULL); + + if (IS_ERR(clk)) { + dev_err(afe->dev, "Failed to register clk %s: %ld\n", + gate->name, PTR_ERR(clk)); + continue; + } + + /* add clk_lookup for devm_clk_get(SND_SOC_DAPM_CLOCK_SUPPLY) */ + cl = kzalloc(sizeof(*cl), GFP_KERNEL); + if (!cl) + return -ENOMEM; + + cl->clk = clk; + cl->con_id = gate->name; + cl->dev_id = dev_name(afe->dev); + clkdev_add(cl); + + afe_priv->lookup[i] = cl; + } + + return 0; +} + +void mt8186_audsys_clk_unregister(struct mtk_base_afe *afe) +{ + struct mt8186_afe_private *afe_priv = afe->platform_priv; + struct clk *clk; + struct clk_lookup *cl; + int i; + + if (!afe_priv) + return; + + for (i = 0; i < CLK_AUD_NR_CLK; i++) { + cl = afe_priv->lookup[i]; + if (!cl) + continue; + + clk = cl->clk; + clk_unregister_gate(clk); + + clkdev_drop(cl); + } +} diff --git a/sound/soc/mediatek/mt8186/mt8186-audsys-clk.h b/sound/soc/mediatek/mt8186/mt8186-audsys-clk.h new file mode 100644 index 000000000000..b8d6a06e11e8 --- /dev/null +++ b/sound/soc/mediatek/mt8186/mt8186-audsys-clk.h @@ -0,0 +1,15 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * mt8186-audsys-clk.h -- Mediatek 8186 audsys clock definition + * + * Copyright (c) 2022 MediaTek Inc. + * Author: Trevor Wu + */ + +#ifndef _MT8186_AUDSYS_CLK_H_ +#define _MT8186_AUDSYS_CLK_H_ + +int mt8186_audsys_clk_register(struct mtk_base_afe *afe); +void mt8186_audsys_clk_unregister(struct mtk_base_afe *afe); + +#endif diff --git a/sound/soc/mediatek/mt8186/mt8186-audsys-clkid.h b/sound/soc/mediatek/mt8186/mt8186-audsys-clkid.h new file mode 100644 index 000000000000..3ce5937c1823 --- /dev/null +++ b/sound/soc/mediatek/mt8186/mt8186-audsys-clkid.h @@ -0,0 +1,45 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * mt8186-audsys-clkid.h -- Mediatek 8186 audsys clock id definition + * + * Copyright (c) 2022 MediaTek Inc. + * Author: Jiaxin Yu + */ + +#ifndef _MT8186_AUDSYS_CLKID_H_ +#define _MT8186_AUDSYS_CLKID_H_ + +enum{ + CLK_AUD_AFE, + CLK_AUD_22M, + CLK_AUD_24M, + CLK_AUD_APLL2_TUNER, + CLK_AUD_APLL_TUNER, + CLK_AUD_TDM, + CLK_AUD_ADC, + CLK_AUD_DAC, + CLK_AUD_DAC_PREDIS, + CLK_AUD_TML, + CLK_AUD_NLE, + CLK_AUD_I2S1_BCLK, + CLK_AUD_I2S2_BCLK, + CLK_AUD_I2S3_BCLK, + CLK_AUD_I2S4_BCLK, + CLK_AUD_CONNSYS_I2S_ASRC, + CLK_AUD_GENERAL1_ASRC, + CLK_AUD_GENERAL2_ASRC, + CLK_AUD_DAC_HIRES, + CLK_AUD_ADC_HIRES, + CLK_AUD_ADC_HIRES_TML, + CLK_AUD_ADDA6_ADC, + CLK_AUD_ADDA6_ADC_HIRES, + CLK_AUD_3RD_DAC, + CLK_AUD_3RD_DAC_PREDIS, + CLK_AUD_3RD_DAC_TML, + CLK_AUD_3RD_DAC_HIRES, + CLK_AUD_ETDM_IN1_BCLK, + CLK_AUD_ETDM_OUT1_BCLK, + CLK_AUD_NR_CLK, +}; + +#endif -- cgit v1.2.3 From b65c466220b336f5044c1be75ebc771d087ee7ca Mon Sep 17 00:00:00 2001 From: Jiaxin Yu Date: Mon, 23 May 2022 21:28:42 +0800 Subject: ASoC: mediatek: mt8186: support adda in platform driver Add mt8186 adda dai driver. Signed-off-by: Jiaxin Yu Reviewed-by: AngeloGioacchino Del Regno Link: https://lore.kernel.org/r/20220523132858.22166-5-jiaxin.yu@mediatek.com Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8186/mt8186-dai-adda.c | 873 ++++++++++++++++++++++++++++ 1 file changed, 873 insertions(+) create mode 100644 sound/soc/mediatek/mt8186/mt8186-dai-adda.c (limited to 'sound') diff --git a/sound/soc/mediatek/mt8186/mt8186-dai-adda.c b/sound/soc/mediatek/mt8186/mt8186-dai-adda.c new file mode 100644 index 000000000000..c66861fd197d --- /dev/null +++ b/sound/soc/mediatek/mt8186/mt8186-dai-adda.c @@ -0,0 +1,873 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// MediaTek ALSA SoC Audio DAI ADDA Control +// +// Copyright (c) 2022 MediaTek Inc. +// Author: Jiaxin Yu + +#include +#include +#include "mt8186-afe-clk.h" +#include "mt8186-afe-common.h" +#include "mt8186-afe-gpio.h" +#include "mt8186-interconnection.h" + +enum { + UL_IIR_SW = 0, + UL_IIR_5HZ, + UL_IIR_10HZ, + UL_IIR_25HZ, + UL_IIR_50HZ, + UL_IIR_75HZ, +}; + +enum { + AUDIO_SDM_LEVEL_MUTE = 0, + AUDIO_SDM_LEVEL_NORMAL = 0x1d, + /* if you change level normal */ + /* you need to change formula of hp impedance and dc trim too */ +}; + +enum { + AUDIO_SDM_2ND = 0, + AUDIO_SDM_3RD, +}; + +enum { + DELAY_DATA_MISO1 = 0, + DELAY_DATA_MISO2, +}; + +enum { + MTK_AFE_ADDA_DL_RATE_8K = 0, + MTK_AFE_ADDA_DL_RATE_11K = 1, + MTK_AFE_ADDA_DL_RATE_12K = 2, + MTK_AFE_ADDA_DL_RATE_16K = 3, + MTK_AFE_ADDA_DL_RATE_22K = 4, + MTK_AFE_ADDA_DL_RATE_24K = 5, + MTK_AFE_ADDA_DL_RATE_32K = 6, + MTK_AFE_ADDA_DL_RATE_44K = 7, + MTK_AFE_ADDA_DL_RATE_48K = 8, + MTK_AFE_ADDA_DL_RATE_96K = 9, + MTK_AFE_ADDA_DL_RATE_192K = 10, +}; + +enum { + MTK_AFE_ADDA_UL_RATE_8K = 0, + MTK_AFE_ADDA_UL_RATE_16K = 1, + MTK_AFE_ADDA_UL_RATE_32K = 2, + MTK_AFE_ADDA_UL_RATE_48K = 3, + MTK_AFE_ADDA_UL_RATE_96K = 4, + MTK_AFE_ADDA_UL_RATE_192K = 5, + MTK_AFE_ADDA_UL_RATE_48K_HD = 6, +}; + +#define SDM_AUTO_RESET_THRESHOLD 0x190000 + +struct mtk_afe_adda_priv { + int dl_rate; + int ul_rate; +}; + +static struct mtk_afe_adda_priv *get_adda_priv_by_name(struct mtk_base_afe *afe, + const char *name) +{ + struct mt8186_afe_private *afe_priv = afe->platform_priv; + int dai_id; + + if (strncmp(name, "aud_dac_hires_clk", 7) == 0 || + strncmp(name, "aud_adc_hires_clk", 7) == 0) + dai_id = MT8186_DAI_ADDA; + else + return NULL; + + return afe_priv->dai_priv[dai_id]; +} + +static unsigned int adda_dl_rate_transform(struct mtk_base_afe *afe, + unsigned int rate) +{ + switch (rate) { + case 8000: + return MTK_AFE_ADDA_DL_RATE_8K; + case 11025: + return MTK_AFE_ADDA_DL_RATE_11K; + case 12000: + return MTK_AFE_ADDA_DL_RATE_12K; + case 16000: + return MTK_AFE_ADDA_DL_RATE_16K; + case 22050: + return MTK_AFE_ADDA_DL_RATE_22K; + case 24000: + return MTK_AFE_ADDA_DL_RATE_24K; + case 32000: + return MTK_AFE_ADDA_DL_RATE_32K; + case 44100: + return MTK_AFE_ADDA_DL_RATE_44K; + case 48000: + return MTK_AFE_ADDA_DL_RATE_48K; + case 96000: + return MTK_AFE_ADDA_DL_RATE_96K; + case 192000: + return MTK_AFE_ADDA_DL_RATE_192K; + default: + dev_info(afe->dev, "%s(), rate %d invalid, use 48kHz!!!\n", + __func__, rate); + } + + return MTK_AFE_ADDA_DL_RATE_48K; +} + +static unsigned int adda_ul_rate_transform(struct mtk_base_afe *afe, + unsigned int rate) +{ + switch (rate) { + case 8000: + return MTK_AFE_ADDA_UL_RATE_8K; + case 16000: + return MTK_AFE_ADDA_UL_RATE_16K; + case 32000: + return MTK_AFE_ADDA_UL_RATE_32K; + case 48000: + return MTK_AFE_ADDA_UL_RATE_48K; + case 96000: + return MTK_AFE_ADDA_UL_RATE_96K; + case 192000: + return MTK_AFE_ADDA_UL_RATE_192K; + default: + dev_info(afe->dev, "%s(), rate %d invalid, use 48kHz!!!\n", + __func__, rate); + } + + return MTK_AFE_ADDA_UL_RATE_48K; +} + +/* dai component */ +static const struct snd_kcontrol_new mtk_adda_dl_ch1_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("DL1_CH1 Switch", AFE_CONN3, I_DL1_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL12_CH1 Switch", AFE_CONN3, I_DL12_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL2_CH1 Switch", AFE_CONN3, I_DL2_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL3_CH1 Switch", AFE_CONN3, I_DL3_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL4_CH1 Switch", AFE_CONN3_1, I_DL4_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL5_CH1 Switch", AFE_CONN3_1, I_DL5_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL6_CH1 Switch", AFE_CONN3_1, I_DL6_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL8_CH1 Switch", AFE_CONN3_1, I_DL8_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("ADDA_UL_CH2 Switch", AFE_CONN3, + I_ADDA_UL_CH2, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("ADDA_UL_CH1 Switch", AFE_CONN3, + I_ADDA_UL_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("GAIN1_OUT_CH1 Switch", AFE_CONN3, + I_GAIN1_OUT_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("PCM_1_CAP_CH1 Switch", AFE_CONN3, + I_PCM_1_CAP_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("PCM_2_CAP_CH1 Switch", AFE_CONN3, + I_PCM_2_CAP_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("SRC_1_OUT_CH1 Switch", AFE_CONN3_1, + I_SRC_1_OUT_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("SRC_2_OUT_CH1 Switch", AFE_CONN3_1, + I_SRC_2_OUT_CH1, 1, 0), +}; + +static const struct snd_kcontrol_new mtk_adda_dl_ch2_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("DL1_CH1 Switch", AFE_CONN4, I_DL1_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL1_CH2 Switch", AFE_CONN4, I_DL1_CH2, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL12_CH2 Switch", AFE_CONN4, I_DL12_CH2, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL2_CH1 Switch", AFE_CONN4, I_DL2_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL2_CH2 Switch", AFE_CONN4, I_DL2_CH2, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL3_CH1 Switch", AFE_CONN4, I_DL3_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL3_CH2 Switch", AFE_CONN4, I_DL3_CH2, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL4_CH2 Switch", AFE_CONN4_1, I_DL4_CH2, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL5_CH2 Switch", AFE_CONN4_1, I_DL5_CH2, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL6_CH2 Switch", AFE_CONN4_1, I_DL6_CH2, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL8_CH2 Switch", AFE_CONN4_1, I_DL8_CH2, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("ADDA_UL_CH2 Switch", AFE_CONN4, + I_ADDA_UL_CH2, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("ADDA_UL_CH1 Switch", AFE_CONN4, + I_ADDA_UL_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("GAIN1_OUT_CH2 Switch", AFE_CONN4, + I_GAIN1_OUT_CH2, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("PCM_1_CAP_CH2 Switch", AFE_CONN4, + I_PCM_1_CAP_CH2, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("PCM_2_CAP_CH2 Switch", AFE_CONN4, + I_PCM_2_CAP_CH2, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("SRC_1_OUT_CH2 Switch", AFE_CONN4_1, + I_SRC_1_OUT_CH2, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("SRC_2_OUT_CH2 Switch", AFE_CONN4_1, + I_SRC_2_OUT_CH2, 1, 0), +}; + +enum { + SUPPLY_SEQ_ADDA_AFE_ON, + SUPPLY_SEQ_ADDA_DL_ON, + SUPPLY_SEQ_ADDA_AUD_PAD_TOP, + SUPPLY_SEQ_ADDA_MTKAIF_CFG, + SUPPLY_SEQ_ADDA_FIFO, + SUPPLY_SEQ_ADDA_AP_DMIC, + SUPPLY_SEQ_ADDA_UL_ON, +}; + +static int mtk_adda_ul_src_dmic(struct mtk_base_afe *afe, int id) +{ + unsigned int reg; + + switch (id) { + case MT8186_DAI_ADDA: + case MT8186_DAI_AP_DMIC: + reg = AFE_ADDA_UL_SRC_CON0; + break; + default: + return -EINVAL; + } + + /* dmic mode, 3.25M*/ + regmap_update_bits(afe->regmap, reg, + DIGMIC_3P25M_1P625M_SEL_MASK_SFT, 0); + regmap_update_bits(afe->regmap, reg, + DMIC_LOW_POWER_CTL_MASK_SFT, 0); + + /* turn on dmic, ch1, ch2 */ + regmap_update_bits(afe->regmap, reg, + UL_SDM_3_LEVEL_MASK_SFT, + BIT(UL_SDM_3_LEVEL_SFT)); + regmap_update_bits(afe->regmap, reg, + UL_MODE_3P25M_CH1_CTL_MASK_SFT, + BIT(UL_MODE_3P25M_CH1_CTL_SFT)); + regmap_update_bits(afe->regmap, reg, + UL_MODE_3P25M_CH2_CTL_MASK_SFT, + BIT(UL_MODE_3P25M_CH2_CTL_SFT)); + + return 0; +} + +static int mtk_adda_ul_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_component *cmpnt = snd_soc_dapm_to_component(w->dapm); + struct mtk_base_afe *afe = snd_soc_component_get_drvdata(cmpnt); + struct mt8186_afe_private *afe_priv = afe->platform_priv; + int mtkaif_dmic = afe_priv->mtkaif_dmic; + + dev_dbg(afe->dev, "%s(), name %s, event 0x%x, mtkaif_dmic %d\n", + __func__, w->name, event, mtkaif_dmic); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + mt8186_afe_gpio_request(afe->dev, true, MT8186_DAI_ADDA, 1); + + /* update setting to dmic */ + if (mtkaif_dmic) { + /* mtkaif_rxif_data_mode = 1, dmic */ + regmap_update_bits(afe->regmap, AFE_ADDA_MTKAIF_RX_CFG0, + 0x1, 0x1); + + /* dmic mode, 3.25M*/ + regmap_update_bits(afe->regmap, AFE_ADDA_MTKAIF_RX_CFG0, + MTKAIF_RXIF_VOICE_MODE_MASK_SFT, + 0x0); + mtk_adda_ul_src_dmic(afe, MT8186_DAI_ADDA); + } + break; + case SND_SOC_DAPM_POST_PMD: + /* should delayed 1/fs(smallest is 8k) = 125us before afe off */ + usleep_range(125, 135); + mt8186_afe_gpio_request(afe->dev, false, MT8186_DAI_ADDA, 1); + + /* reset dmic */ + afe_priv->mtkaif_dmic = 0; + break; + default: + break; + } + + return 0; +} + +static int mtk_adda_pad_top_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_component *cmpnt = snd_soc_dapm_to_component(w->dapm); + struct mtk_base_afe *afe = snd_soc_component_get_drvdata(cmpnt); + struct mt8186_afe_private *afe_priv = afe->platform_priv; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (afe_priv->mtkaif_protocol == MTKAIF_PROTOCOL_2_CLK_P2) + regmap_write(afe->regmap, AFE_AUD_PAD_TOP, 0x39); + else if (afe_priv->mtkaif_protocol == MTKAIF_PROTOCOL_2) + regmap_write(afe->regmap, AFE_AUD_PAD_TOP, 0x31); + else + regmap_write(afe->regmap, AFE_AUD_PAD_TOP, 0x31); + break; + default: + break; + } + + return 0; +} + +static int mtk_adda_mtkaif_cfg_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_component *cmpnt = snd_soc_dapm_to_component(w->dapm); + struct mtk_base_afe *afe = snd_soc_component_get_drvdata(cmpnt); + struct mt8186_afe_private *afe_priv = afe->platform_priv; + int delay_data; + int delay_cycle; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (afe_priv->mtkaif_protocol == MTKAIF_PROTOCOL_2_CLK_P2) { + /* set protocol 2 */ + regmap_write(afe->regmap, AFE_ADDA_MTKAIF_CFG0, 0x10000); + /* mtkaif_rxif_clkinv_adc inverse */ + regmap_update_bits(afe->regmap, AFE_ADDA_MTKAIF_CFG0, + MTKAIF_RXIF_CLKINV_ADC_MASK_SFT, + BIT(MTKAIF_RXIF_CLKINV_ADC_SFT)); + + if (strcmp(w->name, "ADDA_MTKAIF_CFG") == 0) { + if (afe_priv->mtkaif_chosen_phase[0] < 0 && + afe_priv->mtkaif_chosen_phase[1] < 0) { + dev_err(afe->dev, + "%s(), calib fail mtkaif_chosen_phase[0/1]:%d/%d\n", + __func__, + afe_priv->mtkaif_chosen_phase[0], + afe_priv->mtkaif_chosen_phase[1]); + break; + } + + if (afe_priv->mtkaif_chosen_phase[0] < 0 || + afe_priv->mtkaif_chosen_phase[1] < 0) { + dev_err(afe->dev, + "%s(), skip dealy setting mtkaif_chosen_phase[0/1]:%d/%d\n", + __func__, + afe_priv->mtkaif_chosen_phase[0], + afe_priv->mtkaif_chosen_phase[1]); + break; + } + } + + /* set delay for ch12 */ + if (afe_priv->mtkaif_phase_cycle[0] >= + afe_priv->mtkaif_phase_cycle[1]) { + delay_data = DELAY_DATA_MISO1; + delay_cycle = afe_priv->mtkaif_phase_cycle[0] - + afe_priv->mtkaif_phase_cycle[1]; + } else { + delay_data = DELAY_DATA_MISO2; + delay_cycle = afe_priv->mtkaif_phase_cycle[1] - + afe_priv->mtkaif_phase_cycle[0]; + } + + regmap_update_bits(afe->regmap, + AFE_ADDA_MTKAIF_RX_CFG2, + MTKAIF_RXIF_DELAY_DATA_MASK_SFT, + delay_data << + MTKAIF_RXIF_DELAY_DATA_SFT); + + regmap_update_bits(afe->regmap, + AFE_ADDA_MTKAIF_RX_CFG2, + MTKAIF_RXIF_DELAY_CYCLE_MASK_SFT, + delay_cycle << + MTKAIF_RXIF_DELAY_CYCLE_SFT); + + } else if (afe_priv->mtkaif_protocol == MTKAIF_PROTOCOL_2) { + regmap_write(afe->regmap, AFE_ADDA_MTKAIF_CFG0, 0x10000); + } else { + regmap_write(afe->regmap, AFE_ADDA_MTKAIF_CFG0, 0); + } + + break; + default: + break; + } + + return 0; +} + +static int mtk_adda_dl_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_component *cmpnt = snd_soc_dapm_to_component(w->dapm); + struct mtk_base_afe *afe = snd_soc_component_get_drvdata(cmpnt); + + dev_dbg(afe->dev, "%s(), name %s, event 0x%x\n", + __func__, w->name, event); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + mt8186_afe_gpio_request(afe->dev, true, MT8186_DAI_ADDA, 0); + break; + case SND_SOC_DAPM_POST_PMD: + /* should delayed 1/fs(smallest is 8k) = 125us before afe off */ + usleep_range(125, 135); + mt8186_afe_gpio_request(afe->dev, false, MT8186_DAI_ADDA, 0); + break; + default: + break; + } + + return 0; +} + +static int mt8186_adda_dmic_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct mtk_base_afe *afe = snd_soc_component_get_drvdata(cmpnt); + struct mt8186_afe_private *afe_priv = afe->platform_priv; + + ucontrol->value.integer.value[0] = afe_priv->mtkaif_dmic; + + return 0; +} + +static int mt8186_adda_dmic_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct mtk_base_afe *afe = snd_soc_component_get_drvdata(cmpnt); + struct mt8186_afe_private *afe_priv = afe->platform_priv; + int dmic_on; + + dmic_on = ucontrol->value.integer.value[0]; + + dev_dbg(afe->dev, "%s(), kcontrol name %s, dmic_on %d\n", + __func__, kcontrol->id.name, dmic_on); + + if (afe_priv->mtkaif_dmic == dmic_on) + return 0; + + afe_priv->mtkaif_dmic = dmic_on; + + return 1; +} + +static const struct snd_kcontrol_new mtk_adda_controls[] = { + SOC_SINGLE("ADDA_DL_GAIN", AFE_ADDA_DL_SRC2_CON1, + DL_2_GAIN_CTL_PRE_SFT, DL_2_GAIN_CTL_PRE_MASK, 0), + SOC_SINGLE_BOOL_EXT("MTKAIF_DMIC Switch", 0, + mt8186_adda_dmic_get, mt8186_adda_dmic_set), +}; + +/* ADDA UL MUX */ +enum { + ADDA_UL_MUX_MTKAIF = 0, + ADDA_UL_MUX_AP_DMIC, + ADDA_UL_MUX_MASK = 0x1, +}; + +static const char * const adda_ul_mux_map[] = { + "MTKAIF", "AP_DMIC" +}; + +static int adda_ul_map_value[] = { + ADDA_UL_MUX_MTKAIF, + ADDA_UL_MUX_AP_DMIC, +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(adda_ul_mux_map_enum, + SND_SOC_NOPM, + 0, + ADDA_UL_MUX_MASK, + adda_ul_mux_map, + adda_ul_map_value); + +static const struct snd_kcontrol_new adda_ul_mux_control = + SOC_DAPM_ENUM("ADDA_UL_MUX Select", adda_ul_mux_map_enum); + +static const struct snd_soc_dapm_widget mtk_dai_adda_widgets[] = { + /* inter-connections */ + SND_SOC_DAPM_MIXER("ADDA_DL_CH1", SND_SOC_NOPM, 0, 0, + mtk_adda_dl_ch1_mix, + ARRAY_SIZE(mtk_adda_dl_ch1_mix)), + SND_SOC_DAPM_MIXER("ADDA_DL_CH2", SND_SOC_NOPM, 0, 0, + mtk_adda_dl_ch2_mix, + ARRAY_SIZE(mtk_adda_dl_ch2_mix)), + + SND_SOC_DAPM_SUPPLY_S("ADDA Enable", SUPPLY_SEQ_ADDA_AFE_ON, + AFE_ADDA_UL_DL_CON0, ADDA_AFE_ON_SFT, 0, + NULL, 0), + + SND_SOC_DAPM_SUPPLY_S("ADDA Playback Enable", SUPPLY_SEQ_ADDA_DL_ON, + AFE_ADDA_DL_SRC2_CON0, + DL_2_SRC_ON_CTL_PRE_SFT, 0, + mtk_adda_dl_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_SUPPLY_S("ADDA Capture Enable", SUPPLY_SEQ_ADDA_UL_ON, + AFE_ADDA_UL_SRC_CON0, + UL_SRC_ON_CTL_SFT, 0, + mtk_adda_ul_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_SUPPLY_S("AUD_PAD_TOP", SUPPLY_SEQ_ADDA_AUD_PAD_TOP, + 0, 0, 0, + mtk_adda_pad_top_event, + SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_SUPPLY_S("ADDA_MTKAIF_CFG", SUPPLY_SEQ_ADDA_MTKAIF_CFG, + SND_SOC_NOPM, 0, 0, + mtk_adda_mtkaif_cfg_event, + SND_SOC_DAPM_PRE_PMU), + + SND_SOC_DAPM_SUPPLY_S("AP_DMIC_EN", SUPPLY_SEQ_ADDA_AP_DMIC, + AFE_ADDA_UL_SRC_CON0, + UL_AP_DMIC_ON_SFT, 0, + NULL, 0), + + SND_SOC_DAPM_SUPPLY_S("ADDA_FIFO", SUPPLY_SEQ_ADDA_FIFO, + AFE_ADDA_UL_DL_CON0, + AFE_ADDA_FIFO_AUTO_RST_SFT, 1, + NULL, 0), + + SND_SOC_DAPM_MUX("ADDA_UL_Mux", SND_SOC_NOPM, 0, 0, + &adda_ul_mux_control), + + SND_SOC_DAPM_INPUT("AP_DMIC_INPUT"), + + /* clock */ + SND_SOC_DAPM_CLOCK_SUPPLY("top_mux_audio_h"), + + SND_SOC_DAPM_CLOCK_SUPPLY("aud_dac_clk"), + SND_SOC_DAPM_CLOCK_SUPPLY("aud_dac_hires_clk"), + SND_SOC_DAPM_CLOCK_SUPPLY("aud_dac_predis_clk"), + + SND_SOC_DAPM_CLOCK_SUPPLY("aud_adc_clk"), + SND_SOC_DAPM_CLOCK_SUPPLY("aud_adc_hires_clk"), +}; + +#define HIRES_THRESHOLD 48000 +static int mtk_afe_dac_hires_connect(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_dapm_widget *w = source; + struct snd_soc_component *cmpnt = snd_soc_dapm_to_component(w->dapm); + struct mtk_base_afe *afe = snd_soc_component_get_drvdata(cmpnt); + struct mtk_afe_adda_priv *adda_priv; + + adda_priv = get_adda_priv_by_name(afe, w->name); + + if (!adda_priv) { + dev_err(afe->dev, "%s(), adda_priv == NULL", __func__); + return 0; + } + + return (adda_priv->dl_rate > HIRES_THRESHOLD) ? 1 : 0; +} + +static int mtk_afe_adc_hires_connect(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_dapm_widget *w = source; + struct snd_soc_component *cmpnt = snd_soc_dapm_to_component(w->dapm); + struct mtk_base_afe *afe = snd_soc_component_get_drvdata(cmpnt); + struct mtk_afe_adda_priv *adda_priv; + + adda_priv = get_adda_priv_by_name(afe, w->name); + + if (!adda_priv) { + dev_err(afe->dev, "%s(), adda_priv == NULL", __func__); + return 0; + } + + return (adda_priv->ul_rate > HIRES_THRESHOLD) ? 1 : 0; +} + +static const struct snd_soc_dapm_route mtk_dai_adda_routes[] = { + /* playback */ + {"ADDA_DL_CH1", "DL1_CH1 Switch", "DL1"}, + {"ADDA_DL_CH2", "DL1_CH1 Switch", "DL1"}, + {"ADDA_DL_CH2", "DL1_CH2 Switch", "DL1"}, + + {"ADDA_DL_CH1", "DL12_CH1 Switch", "DL12"}, + {"ADDA_DL_CH2", "DL12_CH2 Switch", "DL12"}, + + {"ADDA_DL_CH1", "DL6_CH1 Switch", "DL6"}, + {"ADDA_DL_CH2", "DL6_CH2 Switch", "DL6"}, + + {"ADDA_DL_CH1", "DL8_CH1 Switch", "DL8"}, + {"ADDA_DL_CH2", "DL8_CH2 Switch", "DL8"}, + + {"ADDA_DL_CH1", "DL2_CH1 Switch", "DL2"}, + {"ADDA_DL_CH2", "DL2_CH1 Switch", "DL2"}, + {"ADDA_DL_CH2", "DL2_CH2 Switch", "DL2"}, + + {"ADDA_DL_CH1", "DL3_CH1 Switch", "DL3"}, + {"ADDA_DL_CH2", "DL3_CH1 Switch", "DL3"}, + {"ADDA_DL_CH2", "DL3_CH2 Switch", "DL3"}, + + {"ADDA_DL_CH1", "DL4_CH1 Switch", "DL4"}, + {"ADDA_DL_CH2", "DL4_CH2 Switch", "DL4"}, + + {"ADDA_DL_CH1", "DL5_CH1 Switch", "DL5"}, + {"ADDA_DL_CH2", "DL5_CH2 Switch", "DL5"}, + + {"ADDA Playback", NULL, "ADDA_DL_CH1"}, + {"ADDA Playback", NULL, "ADDA_DL_CH2"}, + + {"ADDA Playback", NULL, "ADDA Enable"}, + {"ADDA Playback", NULL, "ADDA Playback Enable"}, + + /* capture */ + {"ADDA_UL_Mux", "MTKAIF", "ADDA Capture"}, + {"ADDA_UL_Mux", "AP_DMIC", "AP DMIC Capture"}, + + {"ADDA Capture", NULL, "ADDA Enable"}, + {"ADDA Capture", NULL, "ADDA Capture Enable"}, + {"ADDA Capture", NULL, "AUD_PAD_TOP"}, + {"ADDA Capture", NULL, "ADDA_MTKAIF_CFG"}, + + {"AP DMIC Capture", NULL, "ADDA Enable"}, + {"AP DMIC Capture", NULL, "ADDA Capture Enable"}, + {"AP DMIC Capture", NULL, "ADDA_FIFO"}, + {"AP DMIC Capture", NULL, "AP_DMIC_EN"}, + + {"AP DMIC Capture", NULL, "AP_DMIC_INPUT"}, + + /* clk */ + {"ADDA Playback", NULL, "aud_dac_clk"}, + {"ADDA Playback", NULL, "aud_dac_predis_clk"}, + {"ADDA Playback", NULL, "aud_dac_hires_clk", mtk_afe_dac_hires_connect}, + + {"ADDA Capture Enable", NULL, "aud_adc_clk"}, + {"ADDA Capture Enable", NULL, "aud_adc_hires_clk", + mtk_afe_adc_hires_connect}, + + /* hires source from apll1 */ + {"top_mux_audio_h", NULL, APLL2_W_NAME}, + + {"aud_dac_hires_clk", NULL, "top_mux_audio_h"}, + {"aud_adc_hires_clk", NULL, "top_mux_audio_h"}, +}; + +/* dai ops */ +static int mtk_dai_adda_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); + struct mt8186_afe_private *afe_priv = afe->platform_priv; + unsigned int rate = params_rate(params); + int id = dai->id; + struct mtk_afe_adda_priv *adda_priv = afe_priv->dai_priv[id]; + + dev_dbg(afe->dev, "%s(), id %d, stream %d, rate %d\n", + __func__, id, substream->stream, rate); + + if (!adda_priv) { + dev_err(afe->dev, "%s(), adda_priv == NULL", __func__); + return -EINVAL; + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + unsigned int dl_src2_con0; + unsigned int dl_src2_con1; + + adda_priv->dl_rate = rate; + + /* set sampling rate */ + dl_src2_con0 = adda_dl_rate_transform(afe, rate) << + DL_2_INPUT_MODE_CTL_SFT; + + /* set output mode, UP_SAMPLING_RATE_X8 */ + dl_src2_con0 |= (0x3 << DL_2_OUTPUT_SEL_CTL_SFT); + + /* turn off mute function */ + dl_src2_con0 |= BIT(DL_2_MUTE_CH2_OFF_CTL_PRE_SFT); + dl_src2_con0 |= BIT(DL_2_MUTE_CH1_OFF_CTL_PRE_SFT); + + /* set voice input data if input sample rate is 8k or 16k */ + if (rate == 8000 || rate == 16000) + dl_src2_con0 |= BIT(DL_2_VOICE_MODE_CTL_PRE_SFT); + + /* SA suggest apply -0.3db to audio/speech path */ + dl_src2_con1 = MTK_AFE_ADDA_DL_GAIN_NORMAL << + DL_2_GAIN_CTL_PRE_SFT; + + /* turn on down-link gain */ + dl_src2_con0 |= BIT(DL_2_GAIN_ON_CTL_PRE_SFT); + + if (id == MT8186_DAI_ADDA) { + /* clean predistortion */ + regmap_write(afe->regmap, AFE_ADDA_PREDIS_CON0, 0); + regmap_write(afe->regmap, AFE_ADDA_PREDIS_CON1, 0); + + regmap_write(afe->regmap, + AFE_ADDA_DL_SRC2_CON0, dl_src2_con0); + regmap_write(afe->regmap, + AFE_ADDA_DL_SRC2_CON1, dl_src2_con1); + + /* set sdm gain */ + regmap_update_bits(afe->regmap, + AFE_ADDA_DL_SDM_DCCOMP_CON, + ATTGAIN_CTL_MASK_SFT, + AUDIO_SDM_LEVEL_NORMAL << + ATTGAIN_CTL_SFT); + + /* Use new 2nd sdm */ + regmap_update_bits(afe->regmap, + AFE_ADDA_DL_SDM_DITHER_CON, + AFE_DL_SDM_DITHER_64TAP_EN_MASK_SFT, + BIT(AFE_DL_SDM_DITHER_64TAP_EN_SFT)); + regmap_update_bits(afe->regmap, + AFE_ADDA_DL_SDM_AUTO_RESET_CON, + AFE_DL_USE_NEW_2ND_SDM_MASK_SFT, + BIT(AFE_DL_USE_NEW_2ND_SDM_SFT)); + regmap_update_bits(afe->regmap, + AFE_ADDA_DL_SDM_DCCOMP_CON, + USE_3RD_SDM_MASK_SFT, + AUDIO_SDM_2ND << USE_3RD_SDM_SFT); + + /* sdm auto reset */ + regmap_write(afe->regmap, + AFE_ADDA_DL_SDM_AUTO_RESET_CON, + SDM_AUTO_RESET_THRESHOLD); + regmap_update_bits(afe->regmap, + AFE_ADDA_DL_SDM_AUTO_RESET_CON, + SDM_AUTO_RESET_TEST_ON_MASK_SFT, + BIT(SDM_AUTO_RESET_TEST_ON_SFT)); + } + } else { + unsigned int ul_src_con0 = 0; + unsigned int voice_mode = adda_ul_rate_transform(afe, rate); + + adda_priv->ul_rate = rate; + ul_src_con0 |= (voice_mode << 17) & (0x7 << 17); + + /* enable iir */ + ul_src_con0 |= (1 << UL_IIR_ON_TMP_CTL_SFT) & + UL_IIR_ON_TMP_CTL_MASK_SFT; + ul_src_con0 |= (UL_IIR_SW << UL_IIRMODE_CTL_SFT) & + UL_IIRMODE_CTL_MASK_SFT; + switch (id) { + case MT8186_DAI_ADDA: + case MT8186_DAI_AP_DMIC: + /* 35Hz @ 48k */ + regmap_write(afe->regmap, + AFE_ADDA_IIR_COEF_02_01, 0); + regmap_write(afe->regmap, + AFE_ADDA_IIR_COEF_04_03, 0x3fb8); + regmap_write(afe->regmap, + AFE_ADDA_IIR_COEF_06_05, 0x3fb80000); + regmap_write(afe->regmap, + AFE_ADDA_IIR_COEF_08_07, 0x3fb80000); + regmap_write(afe->regmap, + AFE_ADDA_IIR_COEF_10_09, 0xc048); + + regmap_write(afe->regmap, + AFE_ADDA_UL_SRC_CON0, ul_src_con0); + + /* Using Internal ADC */ + regmap_update_bits(afe->regmap, AFE_ADDA_TOP_CON0, BIT(0), 0); + + /* mtkaif_rxif_data_mode = 0, amic */ + regmap_update_bits(afe->regmap, AFE_ADDA_MTKAIF_RX_CFG0, BIT(0), 0); + break; + default: + break; + } + + /* ap dmic */ + switch (id) { + case MT8186_DAI_AP_DMIC: + mtk_adda_ul_src_dmic(afe, id); + break; + default: + break; + } + } + + return 0; +} + +static const struct snd_soc_dai_ops mtk_dai_adda_ops = { + .hw_params = mtk_dai_adda_hw_params, +}; + +/* dai driver */ +#define MTK_ADDA_PLAYBACK_RATES (SNDRV_PCM_RATE_8000_48000 |\ + SNDRV_PCM_RATE_96000 |\ + SNDRV_PCM_RATE_192000) + +#define MTK_ADDA_CAPTURE_RATES (SNDRV_PCM_RATE_8000 |\ + SNDRV_PCM_RATE_16000 |\ + SNDRV_PCM_RATE_32000 |\ + SNDRV_PCM_RATE_48000 |\ + SNDRV_PCM_RATE_96000 |\ + SNDRV_PCM_RATE_192000) + +#define MTK_ADDA_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver mtk_dai_adda_driver[] = { + { + .name = "ADDA", + .id = MT8186_DAI_ADDA, + .playback = { + .stream_name = "ADDA Playback", + .channels_min = 1, + .channels_max = 2, + .rates = MTK_ADDA_PLAYBACK_RATES, + .formats = MTK_ADDA_FORMATS, + }, + .capture = { + .stream_name = "ADDA Capture", + .channels_min = 1, + .channels_max = 2, + .rates = MTK_ADDA_CAPTURE_RATES, + .formats = MTK_ADDA_FORMATS, + }, + .ops = &mtk_dai_adda_ops, + }, + { + .name = "AP_DMIC", + .id = MT8186_DAI_AP_DMIC, + .capture = { + .stream_name = "AP DMIC Capture", + .channels_min = 1, + .channels_max = 2, + .rates = MTK_ADDA_CAPTURE_RATES, + .formats = MTK_ADDA_FORMATS, + }, + .ops = &mtk_dai_adda_ops, + }, +}; + +int mt8186_dai_adda_register(struct mtk_base_afe *afe) +{ + struct mtk_base_afe_dai *dai; + struct mt8186_afe_private *afe_priv = afe->platform_priv; + int ret; + + dai = devm_kzalloc(afe->dev, sizeof(*dai), GFP_KERNEL); + if (!dai) + return -ENOMEM; + + list_add(&dai->list, &afe->sub_dais); + + dai->dai_drivers = mtk_dai_adda_driver; + dai->num_dai_drivers = ARRAY_SIZE(mtk_dai_adda_driver); + + dai->controls = mtk_adda_controls; + dai->num_controls = ARRAY_SIZE(mtk_adda_controls); + dai->dapm_widgets = mtk_dai_adda_widgets; + dai->num_dapm_widgets = ARRAY_SIZE(mtk_dai_adda_widgets); + dai->dapm_routes = mtk_dai_adda_routes; + dai->num_dapm_routes = ARRAY_SIZE(mtk_dai_adda_routes); + + /* set dai priv */ + ret = mt8186_dai_set_priv(afe, MT8186_DAI_ADDA, + sizeof(struct mtk_afe_adda_priv), NULL); + if (ret) + return ret; + + /* ap dmic priv share with adda */ + afe_priv->dai_priv[MT8186_DAI_AP_DMIC] = + afe_priv->dai_priv[MT8186_DAI_ADDA]; + + return 0; +} -- cgit v1.2.3 From 55cac93d271166a2aa431d453bf31fdcb19bd5e6 Mon Sep 17 00:00:00 2001 From: Jiaxin Yu Date: Mon, 23 May 2022 21:28:43 +0800 Subject: ASoC: mediatek: mt8186: support hostless in platform driver Add mt8186 hostless dai driver. Signed-off-by: Jiaxin Yu Reviewed-by: AngeloGioacchino Del Regno Link: https://lore.kernel.org/r/20220523132858.22166-6-jiaxin.yu@mediatek.com Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8186/mt8186-dai-hostless.c | 298 ++++++++++++++++++++++++ 1 file changed, 298 insertions(+) create mode 100644 sound/soc/mediatek/mt8186/mt8186-dai-hostless.c (limited to 'sound') diff --git a/sound/soc/mediatek/mt8186/mt8186-dai-hostless.c b/sound/soc/mediatek/mt8186/mt8186-dai-hostless.c new file mode 100644 index 000000000000..bf0d83840cf4 --- /dev/null +++ b/sound/soc/mediatek/mt8186/mt8186-dai-hostless.c @@ -0,0 +1,298 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// MediaTek ALSA SoC Audio DAI Hostless Control +// +// Copyright (c) 2022 MediaTek Inc. +// Author: Jiaxin Yu + +#include "mt8186-afe-common.h" + +static const struct snd_pcm_hardware mt8186_hostless_hardware = { + .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP_VALID), + .period_bytes_min = 256, + .period_bytes_max = 4 * 48 * 1024, + .periods_min = 2, + .periods_max = 256, + .buffer_bytes_max = 4 * 48 * 1024, + .fifo_size = 0, +}; + +/* dai component */ +static const struct snd_soc_dapm_route mtk_dai_hostless_routes[] = { + /* Hostless ADDA Loopback */ + {"ADDA_DL_CH1", "ADDA_UL_CH1 Switch", "Hostless LPBK DL"}, + {"ADDA_DL_CH1", "ADDA_UL_CH2 Switch", "Hostless LPBK DL"}, + {"ADDA_DL_CH2", "ADDA_UL_CH1 Switch", "Hostless LPBK DL"}, + {"ADDA_DL_CH2", "ADDA_UL_CH2 Switch", "Hostless LPBK DL"}, + {"I2S1_CH1", "ADDA_UL_CH1 Switch", "Hostless LPBK DL"}, + {"I2S1_CH2", "ADDA_UL_CH2 Switch", "Hostless LPBK DL"}, + {"I2S3_CH1", "ADDA_UL_CH1 Switch", "Hostless LPBK DL"}, + {"I2S3_CH1", "ADDA_UL_CH2 Switch", "Hostless LPBK DL"}, + {"I2S3_CH2", "ADDA_UL_CH1 Switch", "Hostless LPBK DL"}, + {"I2S3_CH2", "ADDA_UL_CH2 Switch", "Hostless LPBK DL"}, + {"Hostless LPBK UL", NULL, "ADDA_UL_Mux"}, + + /* Hostelss FM */ + /* connsys_i2s to hw gain 1*/ + {"Hostless FM UL", NULL, "Connsys I2S"}, + + {"HW_GAIN1_IN_CH1", "CONNSYS_I2S_CH1 Switch", "Hostless FM DL"}, + {"HW_GAIN1_IN_CH2", "CONNSYS_I2S_CH2 Switch", "Hostless FM DL"}, + /* hw gain to adda dl */ + {"Hostless FM UL", NULL, "HW Gain 1 Out"}, + + {"ADDA_DL_CH1", "GAIN1_OUT_CH1 Switch", "Hostless FM DL"}, + {"ADDA_DL_CH2", "GAIN1_OUT_CH2 Switch", "Hostless FM DL"}, + /* hw gain to i2s3 */ + {"I2S3_CH1", "GAIN1_OUT_CH1 Switch", "Hostless FM DL"}, + {"I2S3_CH2", "GAIN1_OUT_CH2 Switch", "Hostless FM DL"}, + /* hw gain to i2s1 */ + {"I2S1_CH1", "GAIN1_OUT_CH1 Switch", "Hostless FM DL"}, + {"I2S1_CH2", "GAIN1_OUT_CH2 Switch", "Hostless FM DL"}, + + /* Hostless_SRC */ + {"ADDA_DL_CH1", "SRC_1_OUT_CH1 Switch", "Hostless_SRC_1_DL"}, + {"ADDA_DL_CH2", "SRC_1_OUT_CH2 Switch", "Hostless_SRC_1_DL"}, + {"I2S1_CH1", "SRC_1_OUT_CH1 Switch", "Hostless_SRC_1_DL"}, + {"I2S1_CH2", "SRC_1_OUT_CH2 Switch", "Hostless_SRC_1_DL"}, + {"I2S3_CH1", "SRC_1_OUT_CH1 Switch", "Hostless_SRC_1_DL"}, + {"I2S3_CH2", "SRC_1_OUT_CH2 Switch", "Hostless_SRC_1_DL"}, + {"Hostless_SRC_1_UL", NULL, "HW_SRC_1_Out"}, + + /* Hostless_SRC_bargein */ + {"HW_SRC_1_IN_CH1", "I2S0_CH1 Switch", "Hostless_SRC_Bargein_DL"}, + {"HW_SRC_1_IN_CH2", "I2S0_CH2 Switch", "Hostless_SRC_Bargein_DL"}, + {"Hostless_SRC_Bargein_UL", NULL, "I2S0"}, + + /* Hostless AAudio */ + {"Hostless HW Gain AAudio In", NULL, "HW Gain 2 In"}, + {"Hostless SRC AAudio UL", NULL, "HW Gain 2 Out"}, + {"HW_SRC_2_IN_CH1", "HW_GAIN2_OUT_CH1 Switch", "Hostless SRC AAudio DL"}, + {"HW_SRC_2_IN_CH2", "HW_GAIN2_OUT_CH2 Switch", "Hostless SRC AAudio DL"}, +}; + +/* dai ops */ +static int mtk_dai_hostless_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); + struct snd_pcm_runtime *runtime = substream->runtime; + int ret; + + snd_soc_set_runtime_hwparams(substream, &mt8186_hostless_hardware); + + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) { + dev_err(afe->dev, "snd_pcm_hw_constraint_integer failed\n"); + return ret; + } + + return 0; +} + +static const struct snd_soc_dai_ops mtk_dai_hostless_ops = { + .startup = mtk_dai_hostless_startup, +}; + +/* dai driver */ +#define MTK_HOSTLESS_RATES (SNDRV_PCM_RATE_8000_48000 |\ + SNDRV_PCM_RATE_88200 |\ + SNDRV_PCM_RATE_96000 |\ + SNDRV_PCM_RATE_176400 |\ + SNDRV_PCM_RATE_192000) + +#define MTK_HOSTLESS_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver mtk_dai_hostless_driver[] = { + { + .name = "Hostless LPBK DAI", + .id = MT8186_DAI_HOSTLESS_LPBK, + .playback = { + .stream_name = "Hostless LPBK DL", + .channels_min = 1, + .channels_max = 2, + .rates = MTK_HOSTLESS_RATES, + .formats = MTK_HOSTLESS_FORMATS, + }, + .capture = { + .stream_name = "Hostless LPBK UL", + .channels_min = 1, + .channels_max = 2, + .rates = MTK_HOSTLESS_RATES, + .formats = MTK_HOSTLESS_FORMATS, + }, + .ops = &mtk_dai_hostless_ops, + }, + { + .name = "Hostless FM DAI", + .id = MT8186_DAI_HOSTLESS_FM, + .playback = { + .stream_name = "Hostless FM DL", + .channels_min = 1, + .channels_max = 2, + .rates = MTK_HOSTLESS_RATES, + .formats = MTK_HOSTLESS_FORMATS, + }, + .capture = { + .stream_name = "Hostless FM UL", + .channels_min = 1, + .channels_max = 2, + .rates = MTK_HOSTLESS_RATES, + .formats = MTK_HOSTLESS_FORMATS, + }, + .ops = &mtk_dai_hostless_ops, + }, + { + .name = "Hostless_SRC_1_DAI", + .id = MT8186_DAI_HOSTLESS_SRC_1, + .playback = { + .stream_name = "Hostless_SRC_1_DL", + .channels_min = 1, + .channels_max = 2, + .rates = MTK_HOSTLESS_RATES, + .formats = MTK_HOSTLESS_FORMATS, + }, + .capture = { + .stream_name = "Hostless_SRC_1_UL", + .channels_min = 1, + .channels_max = 2, + .rates = MTK_HOSTLESS_RATES, + .formats = MTK_HOSTLESS_FORMATS, + }, + .ops = &mtk_dai_hostless_ops, + }, + { + .name = "Hostless_SRC_Bargein_DAI", + .id = MT8186_DAI_HOSTLESS_SRC_BARGEIN, + .playback = { + .stream_name = "Hostless_SRC_Bargein_DL", + .channels_min = 1, + .channels_max = 2, + .rates = MTK_HOSTLESS_RATES, + .formats = MTK_HOSTLESS_FORMATS, + }, + .capture = { + .stream_name = "Hostless_SRC_Bargein_UL", + .channels_min = 1, + .channels_max = 2, + .rates = MTK_HOSTLESS_RATES, + .formats = MTK_HOSTLESS_FORMATS, + }, + .ops = &mtk_dai_hostless_ops, + }, + /* BE dai */ + { + .name = "Hostless_UL1 DAI", + .id = MT8186_DAI_HOSTLESS_UL1, + .capture = { + .stream_name = "Hostless_UL1 UL", + .channels_min = 1, + .channels_max = 4, + .rates = MTK_HOSTLESS_RATES, + .formats = MTK_HOSTLESS_FORMATS, + }, + .ops = &mtk_dai_hostless_ops, + }, + { + .name = "Hostless_UL2 DAI", + .id = MT8186_DAI_HOSTLESS_UL2, + .capture = { + .stream_name = "Hostless_UL2 UL", + .channels_min = 1, + .channels_max = 4, + .rates = MTK_HOSTLESS_RATES, + .formats = MTK_HOSTLESS_FORMATS, + }, + .ops = &mtk_dai_hostless_ops, + }, + { + .name = "Hostless_UL3 DAI", + .id = MT8186_DAI_HOSTLESS_UL3, + .capture = { + .stream_name = "Hostless_UL3 UL", + .channels_min = 1, + .channels_max = 2, + .rates = MTK_HOSTLESS_RATES, + .formats = MTK_HOSTLESS_FORMATS, + }, + .ops = &mtk_dai_hostless_ops, + }, + { + .name = "Hostless_UL5 DAI", + .id = MT8186_DAI_HOSTLESS_UL5, + .capture = { + .stream_name = "Hostless_UL5 UL", + .channels_min = 1, + .channels_max = 12, + .rates = MTK_HOSTLESS_RATES, + .formats = MTK_HOSTLESS_FORMATS, + }, + .ops = &mtk_dai_hostless_ops, + }, + { + .name = "Hostless_UL6 DAI", + .id = MT8186_DAI_HOSTLESS_UL6, + .capture = { + .stream_name = "Hostless_UL6 UL", + .channels_min = 1, + .channels_max = 2, + .rates = MTK_HOSTLESS_RATES, + .formats = MTK_HOSTLESS_FORMATS, + }, + .ops = &mtk_dai_hostless_ops, + }, + { + .name = "Hostless HW Gain AAudio DAI", + .id = MT8186_DAI_HOSTLESS_HW_GAIN_AAUDIO, + .capture = { + .stream_name = "Hostless HW Gain AAudio In", + .channels_min = 1, + .channels_max = 2, + .rates = MTK_HOSTLESS_RATES, + .formats = MTK_HOSTLESS_FORMATS, + }, + .ops = &mtk_dai_hostless_ops, + }, + { + .name = "Hostless SRC AAudio DAI", + .id = MT8186_DAI_HOSTLESS_SRC_AAUDIO, + .playback = { + .stream_name = "Hostless SRC AAudio DL", + .channels_min = 1, + .channels_max = 2, + .rates = MTK_HOSTLESS_RATES, + .formats = MTK_HOSTLESS_FORMATS, + }, + .capture = { + .stream_name = "Hostless SRC AAudio UL", + .channels_min = 1, + .channels_max = 2, + .rates = MTK_HOSTLESS_RATES, + .formats = MTK_HOSTLESS_FORMATS, + }, + .ops = &mtk_dai_hostless_ops, + }, +}; + +int mt8186_dai_hostless_register(struct mtk_base_afe *afe) +{ + struct mtk_base_afe_dai *dai; + + dai = devm_kzalloc(afe->dev, sizeof(*dai), GFP_KERNEL); + if (!dai) + return -ENOMEM; + + list_add(&dai->list, &afe->sub_dais); + + dai->dai_drivers = mtk_dai_hostless_driver; + dai->num_dai_drivers = ARRAY_SIZE(mtk_dai_hostless_driver); + + dai->dapm_routes = mtk_dai_hostless_routes; + dai->num_dapm_routes = ARRAY_SIZE(mtk_dai_hostless_routes); + + return 0; +} -- cgit v1.2.3 From 2567ccae9105cbc881828f2ea09954c1b5fd975d Mon Sep 17 00:00:00 2001 From: Jiaxin Yu Date: Mon, 23 May 2022 21:28:44 +0800 Subject: ASoC: mediatek: mt8186: support hw gain in platform driver Add mt8186 hw gain dai driver. Signed-off-by: Jiaxin Yu Reviewed-by: AngeloGioacchino Del Regno Link: https://lore.kernel.org/r/20220523132858.22166-7-jiaxin.yu@mediatek.com Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8186/mt8186-dai-hw-gain.c | 236 +++++++++++++++++++++++++ 1 file changed, 236 insertions(+) create mode 100644 sound/soc/mediatek/mt8186/mt8186-dai-hw-gain.c (limited to 'sound') diff --git a/sound/soc/mediatek/mt8186/mt8186-dai-hw-gain.c b/sound/soc/mediatek/mt8186/mt8186-dai-hw-gain.c new file mode 100644 index 000000000000..33edd6cbde12 --- /dev/null +++ b/sound/soc/mediatek/mt8186/mt8186-dai-hw-gain.c @@ -0,0 +1,236 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// MediaTek ALSA SoC Audio DAI HW Gain Control +// +// Copyright (c) 2022 MediaTek Inc. +// Author: Jiaxin Yu + +#include +#include "mt8186-afe-common.h" +#include "mt8186-interconnection.h" + +#define HW_GAIN_1_EN_W_NAME "HW GAIN 1 Enable" +#define HW_GAIN_2_EN_W_NAME "HW GAIN 2 Enable" + +/* dai component */ +static const struct snd_kcontrol_new mtk_hw_gain1_in_ch1_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("CONNSYS_I2S_CH1 Switch", AFE_CONN13_1, + I_CONNSYS_I2S_CH1, 1, 0), +}; + +static const struct snd_kcontrol_new mtk_hw_gain1_in_ch2_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("CONNSYS_I2S_CH2 Switch", AFE_CONN14_1, + I_CONNSYS_I2S_CH2, 1, 0), +}; + +static const struct snd_kcontrol_new mtk_hw_gain2_in_ch1_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("ADDA_UL_CH1 Switch", AFE_CONN15, + I_ADDA_UL_CH1, 1, 0), +}; + +static const struct snd_kcontrol_new mtk_hw_gain2_in_ch2_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("ADDA_UL_CH2 Switch", AFE_CONN16, + I_ADDA_UL_CH2, 1, 0), +}; + +static int mtk_hw_gain_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_component *cmpnt = snd_soc_dapm_to_component(w->dapm); + struct mtk_base_afe *afe = snd_soc_component_get_drvdata(cmpnt); + unsigned int gain_cur; + unsigned int gain_con1; + + dev_dbg(cmpnt->dev, "%s(), name %s, event 0x%x\n", + __func__, w->name, event); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (strcmp(w->name, HW_GAIN_1_EN_W_NAME) == 0) { + gain_cur = AFE_GAIN1_CUR; + gain_con1 = AFE_GAIN1_CON1; + } else { + gain_cur = AFE_GAIN2_CUR; + gain_con1 = AFE_GAIN2_CON1; + } + + /* let hw gain ramp up, set cur gain to 0 */ + regmap_update_bits(afe->regmap, gain_cur, AFE_GAIN1_CUR_MASK_SFT, 0); + + /* set target gain to 0 */ + regmap_update_bits(afe->regmap, gain_con1, GAIN1_TARGET_MASK_SFT, 0); + break; + default: + break; + } + + return 0; +} + +static const struct snd_soc_dapm_widget mtk_dai_hw_gain_widgets[] = { + /* inter-connections */ + SND_SOC_DAPM_MIXER("HW_GAIN1_IN_CH1", SND_SOC_NOPM, 0, 0, + mtk_hw_gain1_in_ch1_mix, + ARRAY_SIZE(mtk_hw_gain1_in_ch1_mix)), + SND_SOC_DAPM_MIXER("HW_GAIN1_IN_CH2", SND_SOC_NOPM, 0, 0, + mtk_hw_gain1_in_ch2_mix, + ARRAY_SIZE(mtk_hw_gain1_in_ch2_mix)), + SND_SOC_DAPM_MIXER("HW_GAIN2_IN_CH1", SND_SOC_NOPM, 0, 0, + mtk_hw_gain2_in_ch1_mix, + ARRAY_SIZE(mtk_hw_gain2_in_ch1_mix)), + SND_SOC_DAPM_MIXER("HW_GAIN2_IN_CH2", SND_SOC_NOPM, 0, 0, + mtk_hw_gain2_in_ch2_mix, + ARRAY_SIZE(mtk_hw_gain2_in_ch2_mix)), + + SND_SOC_DAPM_SUPPLY(HW_GAIN_1_EN_W_NAME, + AFE_GAIN1_CON0, GAIN1_ON_SFT, 0, + mtk_hw_gain_event, + SND_SOC_DAPM_PRE_PMU), + + SND_SOC_DAPM_SUPPLY(HW_GAIN_2_EN_W_NAME, + AFE_GAIN2_CON0, GAIN2_ON_SFT, 0, + mtk_hw_gain_event, + SND_SOC_DAPM_PRE_PMU), + + SND_SOC_DAPM_INPUT("HW Gain 1 Out Endpoint"), + SND_SOC_DAPM_INPUT("HW Gain 2 Out Endpoint"), + SND_SOC_DAPM_OUTPUT("HW Gain 1 In Endpoint"), +}; + +static const struct snd_soc_dapm_route mtk_dai_hw_gain_routes[] = { + {"HW Gain 1 In", NULL, "HW_GAIN1_IN_CH1"}, + {"HW Gain 1 In", NULL, "HW_GAIN1_IN_CH2"}, + {"HW Gain 2 In", NULL, "HW_GAIN2_IN_CH1"}, + {"HW Gain 2 In", NULL, "HW_GAIN2_IN_CH2"}, + + {"HW Gain 1 In", NULL, HW_GAIN_1_EN_W_NAME}, + {"HW Gain 1 Out", NULL, HW_GAIN_1_EN_W_NAME}, + {"HW Gain 2 In", NULL, HW_GAIN_2_EN_W_NAME}, + {"HW Gain 2 Out", NULL, HW_GAIN_2_EN_W_NAME}, + + {"HW Gain 1 In Endpoint", NULL, "HW Gain 1 In"}, + {"HW Gain 1 Out", NULL, "HW Gain 1 Out Endpoint"}, + {"HW Gain 2 Out", NULL, "HW Gain 2 Out Endpoint"}, +}; + +static const struct snd_kcontrol_new mtk_hw_gain_controls[] = { + SOC_SINGLE("HW Gain 1 Volume", AFE_GAIN1_CON1, + GAIN1_TARGET_SFT, GAIN1_TARGET_MASK, 0), + SOC_SINGLE("HW Gain 2 Volume", AFE_GAIN2_CON1, + GAIN2_TARGET_SFT, GAIN2_TARGET_MASK, 0), +}; + +/* dai ops */ +static int mtk_dai_gain_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); + unsigned int rate = params_rate(params); + unsigned int rate_reg = mt8186_rate_transform(afe->dev, rate, dai->id); + + dev_dbg(afe->dev, "%s(), id %d, stream %d, rate %d\n", + __func__, dai->id, substream->stream, rate); + + /* rate */ + regmap_update_bits(afe->regmap, + dai->id == MT8186_DAI_HW_GAIN_1 ? + AFE_GAIN1_CON0 : AFE_GAIN2_CON0, + GAIN1_MODE_MASK_SFT, + rate_reg << GAIN1_MODE_SFT); + + /* sample per step */ + regmap_update_bits(afe->regmap, + dai->id == MT8186_DAI_HW_GAIN_1 ? + AFE_GAIN1_CON0 : AFE_GAIN2_CON0, + GAIN1_SAMPLE_PER_STEP_MASK_SFT, + (dai->id == MT8186_DAI_HW_GAIN_1 ? 0x40 : 0x0) << + GAIN1_SAMPLE_PER_STEP_SFT); + + return 0; +} + +static const struct snd_soc_dai_ops mtk_dai_gain_ops = { + .hw_params = mtk_dai_gain_hw_params, +}; + +/* dai driver */ +#define MTK_HW_GAIN_RATES (SNDRV_PCM_RATE_8000_48000 |\ + SNDRV_PCM_RATE_88200 |\ + SNDRV_PCM_RATE_96000 |\ + SNDRV_PCM_RATE_176400 |\ + SNDRV_PCM_RATE_192000) + +#define MTK_HW_GAIN_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver mtk_dai_gain_driver[] = { + { + .name = "HW Gain 1", + .id = MT8186_DAI_HW_GAIN_1, + .playback = { + .stream_name = "HW Gain 1 In", + .channels_min = 1, + .channels_max = 2, + .rates = MTK_HW_GAIN_RATES, + .formats = MTK_HW_GAIN_FORMATS, + }, + .capture = { + .stream_name = "HW Gain 1 Out", + .channels_min = 1, + .channels_max = 2, + .rates = MTK_HW_GAIN_RATES, + .formats = MTK_HW_GAIN_FORMATS, + }, + .ops = &mtk_dai_gain_ops, + .symmetric_rate = 1, + .symmetric_channels = 1, + .symmetric_sample_bits = 1, + }, + { + .name = "HW Gain 2", + .id = MT8186_DAI_HW_GAIN_2, + .playback = { + .stream_name = "HW Gain 2 In", + .channels_min = 1, + .channels_max = 2, + .rates = MTK_HW_GAIN_RATES, + .formats = MTK_HW_GAIN_FORMATS, + }, + .capture = { + .stream_name = "HW Gain 2 Out", + .channels_min = 1, + .channels_max = 2, + .rates = MTK_HW_GAIN_RATES, + .formats = MTK_HW_GAIN_FORMATS, + }, + .ops = &mtk_dai_gain_ops, + .symmetric_rate = 1, + .symmetric_channels = 1, + .symmetric_sample_bits = 1, + }, +}; + +int mt8186_dai_hw_gain_register(struct mtk_base_afe *afe) +{ + struct mtk_base_afe_dai *dai; + + dai = devm_kzalloc(afe->dev, sizeof(*dai), GFP_KERNEL); + if (!dai) + return -ENOMEM; + + list_add(&dai->list, &afe->sub_dais); + + dai->dai_drivers = mtk_dai_gain_driver; + dai->num_dai_drivers = ARRAY_SIZE(mtk_dai_gain_driver); + + dai->controls = mtk_hw_gain_controls; + dai->num_controls = ARRAY_SIZE(mtk_hw_gain_controls); + dai->dapm_widgets = mtk_dai_hw_gain_widgets; + dai->num_dapm_widgets = ARRAY_SIZE(mtk_dai_hw_gain_widgets); + dai->dapm_routes = mtk_dai_hw_gain_routes; + dai->num_dapm_routes = ARRAY_SIZE(mtk_dai_hw_gain_routes); + return 0; +} -- cgit v1.2.3 From 2907d261276e09bd84fdc8bad35930a046a99d4d Mon Sep 17 00:00:00 2001 From: Jiaxin Yu Date: Mon, 23 May 2022 21:28:45 +0800 Subject: ASoC: mediatek: mt8186: support i2s in platform driver Add mt8186 i2s dai driver. Signed-off-by: Jiaxin Yu Reviewed-by: AngeloGioacchino Del Regno Link: https://lore.kernel.org/r/20220523132858.22166-8-jiaxin.yu@mediatek.com Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8186/mt8186-dai-i2s.c | 1286 ++++++++++++++++++++++++++++ 1 file changed, 1286 insertions(+) create mode 100644 sound/soc/mediatek/mt8186/mt8186-dai-i2s.c (limited to 'sound') diff --git a/sound/soc/mediatek/mt8186/mt8186-dai-i2s.c b/sound/soc/mediatek/mt8186/mt8186-dai-i2s.c new file mode 100644 index 000000000000..5c1290b950e8 --- /dev/null +++ b/sound/soc/mediatek/mt8186/mt8186-dai-i2s.c @@ -0,0 +1,1286 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// MediaTek ALSA SoC Audio DAI I2S Control +// +// Copyright (c) 2022 MediaTek Inc. +// Author: Jiaxin Yu + +#include +#include +#include +#include "mt8186-afe-clk.h" +#include "mt8186-afe-common.h" +#include "mt8186-afe-gpio.h" +#include "mt8186-interconnection.h" + +enum { + I2S_FMT_EIAJ = 0, + I2S_FMT_I2S = 1, +}; + +enum { + I2S_WLEN_16_BIT = 0, + I2S_WLEN_32_BIT = 1, +}; + +enum { + I2S_HD_NORMAL = 0, + I2S_HD_LOW_JITTER = 1, +}; + +enum { + I2S1_SEL_O28_O29 = 0, + I2S1_SEL_O03_O04 = 1, +}; + +enum { + I2S_IN_PAD_CONNSYS = 0, + I2S_IN_PAD_IO_MUX = 1, +}; + +struct mtk_afe_i2s_priv { + int id; + int rate; /* for determine which apll to use */ + int low_jitter_en; + int master; /* only i2s0 has slave mode*/ + + const char *share_property_name; + int share_i2s_id; + + int mclk_id; + int mclk_rate; + int mclk_apll; +}; + +static unsigned int get_i2s_wlen(snd_pcm_format_t format) +{ + return snd_pcm_format_physical_width(format) <= 16 ? + I2S_WLEN_16_BIT : I2S_WLEN_32_BIT; +} + +#define MTK_AFE_I2S0_KCONTROL_NAME "I2S0_HD_Mux" +#define MTK_AFE_I2S1_KCONTROL_NAME "I2S1_HD_Mux" +#define MTK_AFE_I2S2_KCONTROL_NAME "I2S2_HD_Mux" +#define MTK_AFE_I2S3_KCONTROL_NAME "I2S3_HD_Mux" +#define MTK_AFE_I2S0_SRC_KCONTROL_NAME "I2S0_SRC_Mux" + +#define I2S0_HD_EN_W_NAME "I2S0_HD_EN" +#define I2S1_HD_EN_W_NAME "I2S1_HD_EN" +#define I2S2_HD_EN_W_NAME "I2S2_HD_EN" +#define I2S3_HD_EN_W_NAME "I2S3_HD_EN" + +#define I2S0_MCLK_EN_W_NAME "I2S0_MCLK_EN" +#define I2S1_MCLK_EN_W_NAME "I2S1_MCLK_EN" +#define I2S2_MCLK_EN_W_NAME "I2S2_MCLK_EN" +#define I2S3_MCLK_EN_W_NAME "I2S3_MCLK_EN" + +static int get_i2s_id_by_name(struct mtk_base_afe *afe, + const char *name) +{ + if (strncmp(name, "I2S0", 4) == 0) + return MT8186_DAI_I2S_0; + else if (strncmp(name, "I2S1", 4) == 0) + return MT8186_DAI_I2S_1; + else if (strncmp(name, "I2S2", 4) == 0) + return MT8186_DAI_I2S_2; + else if (strncmp(name, "I2S3", 4) == 0) + return MT8186_DAI_I2S_3; + + return -EINVAL; +} + +static struct mtk_afe_i2s_priv *get_i2s_priv_by_name(struct mtk_base_afe *afe, + const char *name) +{ + struct mt8186_afe_private *afe_priv = afe->platform_priv; + int dai_id = get_i2s_id_by_name(afe, name); + + if (dai_id < 0) + return NULL; + + return afe_priv->dai_priv[dai_id]; +} + +/* low jitter control */ +static const char * const mt8186_i2s_hd_str[] = { + "Normal", "Low_Jitter" +}; + +static const struct soc_enum mt8186_i2s_enum[] = { + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(mt8186_i2s_hd_str), + mt8186_i2s_hd_str), +}; + +static int mt8186_i2s_hd_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct mtk_base_afe *afe = snd_soc_component_get_drvdata(cmpnt); + struct mtk_afe_i2s_priv *i2s_priv; + + i2s_priv = get_i2s_priv_by_name(afe, kcontrol->id.name); + + if (!i2s_priv) { + dev_err(afe->dev, "%s(), i2s_priv == NULL", __func__); + return -EINVAL; + } + + ucontrol->value.integer.value[0] = i2s_priv->low_jitter_en; + + return 0; +} + +static int mt8186_i2s_hd_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct mtk_base_afe *afe = snd_soc_component_get_drvdata(cmpnt); + struct mtk_afe_i2s_priv *i2s_priv; + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + int hd_en; + + if (ucontrol->value.enumerated.item[0] >= e->items) + return -EINVAL; + + hd_en = ucontrol->value.integer.value[0]; + + dev_dbg(afe->dev, "%s(), kcontrol name %s, hd_en %d\n", + __func__, kcontrol->id.name, hd_en); + + i2s_priv = get_i2s_priv_by_name(afe, kcontrol->id.name); + + if (!i2s_priv) { + dev_err(afe->dev, "%s(), i2s_priv == NULL", __func__); + return -EINVAL; + } + + if (i2s_priv->low_jitter_en == hd_en) + return 0; + + i2s_priv->low_jitter_en = hd_en; + + return 1; +} + +static const struct snd_kcontrol_new mtk_dai_i2s_controls[] = { + SOC_ENUM_EXT(MTK_AFE_I2S0_KCONTROL_NAME, mt8186_i2s_enum[0], + mt8186_i2s_hd_get, mt8186_i2s_hd_set), + SOC_ENUM_EXT(MTK_AFE_I2S1_KCONTROL_NAME, mt8186_i2s_enum[0], + mt8186_i2s_hd_get, mt8186_i2s_hd_set), + SOC_ENUM_EXT(MTK_AFE_I2S2_KCONTROL_NAME, mt8186_i2s_enum[0], + mt8186_i2s_hd_get, mt8186_i2s_hd_set), + SOC_ENUM_EXT(MTK_AFE_I2S3_KCONTROL_NAME, mt8186_i2s_enum[0], + mt8186_i2s_hd_get, mt8186_i2s_hd_set), +}; + +/* dai component */ +/* i2s virtual mux to output widget */ +static const char * const i2s_mux_map[] = { + "Normal", "Dummy_Widget", +}; + +static int i2s_mux_map_value[] = { + 0, 1, +}; + +static SOC_VALUE_ENUM_SINGLE_AUTODISABLE_DECL(i2s_mux_map_enum, + SND_SOC_NOPM, + 0, + 1, + i2s_mux_map, + i2s_mux_map_value); + +static const struct snd_kcontrol_new i2s0_in_mux_control = + SOC_DAPM_ENUM("I2S0 In Select", i2s_mux_map_enum); + +static const struct snd_kcontrol_new i2s1_out_mux_control = + SOC_DAPM_ENUM("I2S1 Out Select", i2s_mux_map_enum); + +static const struct snd_kcontrol_new i2s2_in_mux_control = + SOC_DAPM_ENUM("I2S2 In Select", i2s_mux_map_enum); + +static const struct snd_kcontrol_new i2s3_out_mux_control = + SOC_DAPM_ENUM("I2S3 Out Select", i2s_mux_map_enum); + +/* i2s in lpbk */ +static const char * const i2s_lpbk_mux_map[] = { + "Normal", "Lpbk", +}; + +static int i2s_lpbk_mux_map_value[] = { + 0, 1, +}; + +static SOC_VALUE_ENUM_SINGLE_AUTODISABLE_DECL(i2s0_lpbk_mux_map_enum, + AFE_I2S_CON, + I2S_LOOPBACK_SFT, + 1, + i2s_lpbk_mux_map, + i2s_lpbk_mux_map_value); + +static const struct snd_kcontrol_new i2s0_lpbk_mux_control = + SOC_DAPM_ENUM("I2S Lpbk Select", i2s0_lpbk_mux_map_enum); + +static SOC_VALUE_ENUM_SINGLE_AUTODISABLE_DECL(i2s2_lpbk_mux_map_enum, + AFE_I2S_CON2, + I2S3_LOOPBACK_SFT, + 1, + i2s_lpbk_mux_map, + i2s_lpbk_mux_map_value); + +static const struct snd_kcontrol_new i2s2_lpbk_mux_control = + SOC_DAPM_ENUM("I2S Lpbk Select", i2s2_lpbk_mux_map_enum); + +/* interconnection */ +static const struct snd_kcontrol_new mtk_i2s3_ch1_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("DL1_CH1 Switch", AFE_CONN0, + I_DL1_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL2_CH1 Switch", AFE_CONN0, + I_DL2_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL3_CH1 Switch", AFE_CONN0, + I_DL3_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL12_CH1 Switch", AFE_CONN0, + I_DL12_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL12_CH3 Switch", AFE_CONN0, + I_DL12_CH3, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL6_CH1 Switch", AFE_CONN0_1, + I_DL6_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL4_CH1 Switch", AFE_CONN0_1, + I_DL4_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL5_CH1 Switch", AFE_CONN0_1, + I_DL5_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL8_CH1 Switch", AFE_CONN0_1, + I_DL8_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("GAIN1_OUT_CH1 Switch", AFE_CONN0, + I_GAIN1_OUT_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("ADDA_UL_CH1 Switch", AFE_CONN0, + I_ADDA_UL_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("ADDA_UL_CH2 Switch", AFE_CONN0, + I_ADDA_UL_CH2, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("ADDA_UL_CH3 Switch", AFE_CONN0, + I_ADDA_UL_CH3, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("PCM_1_CAP_CH1 Switch", AFE_CONN0, + I_PCM_1_CAP_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("SRC_1_OUT_CH1 Switch", AFE_CONN0_1, + I_SRC_1_OUT_CH1, 1, 0), +}; + +static const struct snd_kcontrol_new mtk_i2s3_ch2_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("DL1_CH2 Switch", AFE_CONN1, + I_DL1_CH2, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL2_CH2 Switch", AFE_CONN1, + I_DL2_CH2, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL3_CH2 Switch", AFE_CONN1, + I_DL3_CH2, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL12_CH2 Switch", AFE_CONN1, + I_DL12_CH2, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL12_CH4 Switch", AFE_CONN1, + I_DL12_CH4, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL6_CH2 Switch", AFE_CONN1_1, + I_DL6_CH2, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL4_CH2 Switch", AFE_CONN1_1, + I_DL4_CH2, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL5_CH2 Switch", AFE_CONN1_1, + I_DL5_CH2, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL8_CH2 Switch", AFE_CONN1_1, + I_DL8_CH2, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("GAIN1_OUT_CH2 Switch", AFE_CONN1, + I_GAIN1_OUT_CH2, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("ADDA_UL_CH1 Switch", AFE_CONN1, + I_ADDA_UL_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("ADDA_UL_CH2 Switch", AFE_CONN1, + I_ADDA_UL_CH2, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("ADDA_UL_CH3 Switch", AFE_CONN1, + I_ADDA_UL_CH3, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("PCM_1_CAP_CH2 Switch", AFE_CONN1, + I_PCM_1_CAP_CH2, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("PCM_2_CAP_CH2 Switch", AFE_CONN1, + I_PCM_2_CAP_CH2, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("SRC_1_OUT_CH2 Switch", AFE_CONN1_1, + I_SRC_1_OUT_CH2, 1, 0), +}; + +static const struct snd_kcontrol_new mtk_i2s1_ch1_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("DL1_CH1 Switch", AFE_CONN28, + I_DL1_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL2_CH1 Switch", AFE_CONN28, + I_DL2_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL3_CH1 Switch", AFE_CONN28, + I_DL3_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL12_CH1 Switch", AFE_CONN28, + I_DL12_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL12_CH3 Switch", AFE_CONN28, + I_DL12_CH3, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL6_CH1 Switch", AFE_CONN28_1, + I_DL6_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL4_CH1 Switch", AFE_CONN28_1, + I_DL4_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL5_CH1 Switch", AFE_CONN28_1, + I_DL5_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL8_CH1 Switch", AFE_CONN28_1, + I_DL8_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("GAIN1_OUT_CH1 Switch", AFE_CONN28, + I_GAIN1_OUT_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("ADDA_UL_CH1 Switch", AFE_CONN28, + I_ADDA_UL_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("PCM_1_CAP_CH1 Switch", AFE_CONN28, + I_PCM_1_CAP_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("SRC_1_OUT_CH1 Switch", AFE_CONN28_1, + I_SRC_1_OUT_CH1, 1, 0), +}; + +static const struct snd_kcontrol_new mtk_i2s1_ch2_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("DL1_CH2 Switch", AFE_CONN29, + I_DL1_CH2, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL2_CH2 Switch", AFE_CONN29, + I_DL2_CH2, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL3_CH2 Switch", AFE_CONN29, + I_DL3_CH2, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL12_CH2 Switch", AFE_CONN29, + I_DL12_CH2, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL12_CH4 Switch", AFE_CONN29, + I_DL12_CH4, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL6_CH2 Switch", AFE_CONN29_1, + I_DL6_CH2, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL4_CH2 Switch", AFE_CONN29_1, + I_DL4_CH2, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL5_CH2 Switch", AFE_CONN29_1, + I_DL5_CH2, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL8_CH2 Switch", AFE_CONN29_1, + I_DL8_CH2, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("GAIN1_OUT_CH2 Switch", AFE_CONN29, + I_GAIN1_OUT_CH2, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("ADDA_UL_CH2 Switch", AFE_CONN29, + I_ADDA_UL_CH2, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("PCM_1_CAP_CH2 Switch", AFE_CONN29, + I_PCM_1_CAP_CH2, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("PCM_2_CAP_CH2 Switch", AFE_CONN29, + I_PCM_2_CAP_CH2, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("SRC_1_OUT_CH2 Switch", AFE_CONN29_1, + I_SRC_1_OUT_CH2, 1, 0), +}; + +enum { + SUPPLY_SEQ_APLL, + SUPPLY_SEQ_I2S_MCLK_EN, + SUPPLY_SEQ_I2S_HD_EN, + SUPPLY_SEQ_I2S_EN, +}; + +static int mtk_i2s_en_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_component *cmpnt = snd_soc_dapm_to_component(w->dapm); + struct mtk_base_afe *afe = snd_soc_component_get_drvdata(cmpnt); + struct mtk_afe_i2s_priv *i2s_priv; + + i2s_priv = get_i2s_priv_by_name(afe, w->name); + + if (!i2s_priv) { + dev_err(afe->dev, "%s(), i2s_priv == NULL", __func__); + return -EINVAL; + } + + dev_dbg(cmpnt->dev, "%s(), name %s, event 0x%x\n", + __func__, w->name, event); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + mt8186_afe_gpio_request(afe->dev, true, i2s_priv->id, 0); + break; + case SND_SOC_DAPM_POST_PMD: + mt8186_afe_gpio_request(afe->dev, false, i2s_priv->id, 0); + break; + default: + break; + } + + return 0; +} + +static int mtk_apll_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_component *cmpnt = snd_soc_dapm_to_component(w->dapm); + struct mtk_base_afe *afe = snd_soc_component_get_drvdata(cmpnt); + + dev_dbg(cmpnt->dev, "%s(), name %s, event 0x%x\n", + __func__, w->name, event); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (strcmp(w->name, APLL1_W_NAME) == 0) + mt8186_apll1_enable(afe); + else + mt8186_apll2_enable(afe); + break; + case SND_SOC_DAPM_POST_PMD: + if (strcmp(w->name, APLL1_W_NAME) == 0) + mt8186_apll1_disable(afe); + else + mt8186_apll2_disable(afe); + break; + default: + break; + } + + return 0; +} + +static int mtk_mclk_en_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_component *cmpnt = snd_soc_dapm_to_component(w->dapm); + struct mtk_base_afe *afe = snd_soc_component_get_drvdata(cmpnt); + struct mtk_afe_i2s_priv *i2s_priv; + + dev_dbg(cmpnt->dev, "%s(), name %s, event 0x%x\n", + __func__, w->name, event); + + i2s_priv = get_i2s_priv_by_name(afe, w->name); + + if (!i2s_priv) { + dev_err(afe->dev, "%s(), i2s_priv == NULL", __func__); + return -EINVAL; + } + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + mt8186_mck_enable(afe, i2s_priv->mclk_id, i2s_priv->mclk_rate); + break; + case SND_SOC_DAPM_POST_PMD: + i2s_priv->mclk_rate = 0; + mt8186_mck_disable(afe, i2s_priv->mclk_id); + break; + default: + break; + } + + return 0; +} + +static const struct snd_soc_dapm_widget mtk_dai_i2s_widgets[] = { + SND_SOC_DAPM_INPUT("CONNSYS"), + + SND_SOC_DAPM_MIXER("I2S1_CH1", SND_SOC_NOPM, 0, 0, + mtk_i2s1_ch1_mix, + ARRAY_SIZE(mtk_i2s1_ch1_mix)), + SND_SOC_DAPM_MIXER("I2S1_CH2", SND_SOC_NOPM, 0, 0, + mtk_i2s1_ch2_mix, + ARRAY_SIZE(mtk_i2s1_ch2_mix)), + + SND_SOC_DAPM_MIXER("I2S3_CH1", SND_SOC_NOPM, 0, 0, + mtk_i2s3_ch1_mix, + ARRAY_SIZE(mtk_i2s3_ch1_mix)), + SND_SOC_DAPM_MIXER("I2S3_CH2", SND_SOC_NOPM, 0, 0, + mtk_i2s3_ch2_mix, + ARRAY_SIZE(mtk_i2s3_ch2_mix)), + + /* i2s en*/ + SND_SOC_DAPM_SUPPLY_S("I2S0_EN", SUPPLY_SEQ_I2S_EN, + AFE_I2S_CON, I2S_EN_SFT, 0, + mtk_i2s_en_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY_S("I2S1_EN", SUPPLY_SEQ_I2S_EN, + AFE_I2S_CON1, I2S_EN_SFT, 0, + mtk_i2s_en_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY_S("I2S2_EN", SUPPLY_SEQ_I2S_EN, + AFE_I2S_CON2, I2S_EN_SFT, 0, + mtk_i2s_en_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY_S("I2S3_EN", SUPPLY_SEQ_I2S_EN, + AFE_I2S_CON3, I2S_EN_SFT, 0, + mtk_i2s_en_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + /* i2s hd en */ + SND_SOC_DAPM_SUPPLY_S(I2S0_HD_EN_W_NAME, SUPPLY_SEQ_I2S_HD_EN, + AFE_I2S_CON, I2S1_HD_EN_SFT, 0, NULL, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY_S(I2S1_HD_EN_W_NAME, SUPPLY_SEQ_I2S_HD_EN, + AFE_I2S_CON1, I2S2_HD_EN_SFT, 0, NULL, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY_S(I2S2_HD_EN_W_NAME, SUPPLY_SEQ_I2S_HD_EN, + AFE_I2S_CON2, I2S3_HD_EN_SFT, 0, NULL, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY_S(I2S3_HD_EN_W_NAME, SUPPLY_SEQ_I2S_HD_EN, + AFE_I2S_CON3, I2S4_HD_EN_SFT, 0, NULL, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + /* i2s mclk en */ + SND_SOC_DAPM_SUPPLY_S(I2S0_MCLK_EN_W_NAME, SUPPLY_SEQ_I2S_MCLK_EN, + SND_SOC_NOPM, 0, 0, + mtk_mclk_en_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY_S(I2S1_MCLK_EN_W_NAME, SUPPLY_SEQ_I2S_MCLK_EN, + SND_SOC_NOPM, 0, 0, + mtk_mclk_en_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY_S(I2S2_MCLK_EN_W_NAME, SUPPLY_SEQ_I2S_MCLK_EN, + SND_SOC_NOPM, 0, 0, + mtk_mclk_en_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY_S(I2S3_MCLK_EN_W_NAME, SUPPLY_SEQ_I2S_MCLK_EN, + SND_SOC_NOPM, 0, 0, + mtk_mclk_en_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + /* apll */ + SND_SOC_DAPM_SUPPLY_S(APLL1_W_NAME, SUPPLY_SEQ_APLL, + SND_SOC_NOPM, 0, 0, + mtk_apll_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY_S(APLL2_W_NAME, SUPPLY_SEQ_APLL, + SND_SOC_NOPM, 0, 0, + mtk_apll_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + /* allow i2s on without codec on */ + SND_SOC_DAPM_OUTPUT("I2S_DUMMY_OUT"), + SND_SOC_DAPM_MUX("I2S1_Out_Mux", + SND_SOC_NOPM, 0, 0, &i2s1_out_mux_control), + SND_SOC_DAPM_MUX("I2S3_Out_Mux", + SND_SOC_NOPM, 0, 0, &i2s3_out_mux_control), + SND_SOC_DAPM_INPUT("I2S_DUMMY_IN"), + SND_SOC_DAPM_MUX("I2S0_In_Mux", + SND_SOC_NOPM, 0, 0, &i2s0_in_mux_control), + SND_SOC_DAPM_MUX("I2S2_In_Mux", + SND_SOC_NOPM, 0, 0, &i2s2_in_mux_control), + + /* i2s in lpbk */ + SND_SOC_DAPM_MUX("I2S0_Lpbk_Mux", + SND_SOC_NOPM, 0, 0, &i2s0_lpbk_mux_control), + SND_SOC_DAPM_MUX("I2S2_Lpbk_Mux", + SND_SOC_NOPM, 0, 0, &i2s2_lpbk_mux_control), +}; + +static int mtk_afe_i2s_share_connect(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_dapm_widget *w = sink; + struct snd_soc_component *cmpnt = snd_soc_dapm_to_component(w->dapm); + struct mtk_base_afe *afe = snd_soc_component_get_drvdata(cmpnt); + struct mtk_afe_i2s_priv *i2s_priv; + + i2s_priv = get_i2s_priv_by_name(afe, sink->name); + + if (!i2s_priv) { + dev_err(afe->dev, "%s(), i2s_priv == NULL", __func__); + return 0; + } + + if (i2s_priv->share_i2s_id < 0) + return 0; + + return i2s_priv->share_i2s_id == get_i2s_id_by_name(afe, source->name); +} + +static int mtk_afe_i2s_hd_connect(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_dapm_widget *w = sink; + struct snd_soc_component *cmpnt = snd_soc_dapm_to_component(w->dapm); + struct mtk_base_afe *afe = snd_soc_component_get_drvdata(cmpnt); + struct mtk_afe_i2s_priv *i2s_priv; + + i2s_priv = get_i2s_priv_by_name(afe, sink->name); + + if (!i2s_priv) { + dev_err(afe->dev, "%s(), i2s_priv == NULL", __func__); + return 0; + } + + if (get_i2s_id_by_name(afe, sink->name) == + get_i2s_id_by_name(afe, source->name)) + return i2s_priv->low_jitter_en; + + /* check if share i2s need hd en */ + if (i2s_priv->share_i2s_id < 0) + return 0; + + if (i2s_priv->share_i2s_id == get_i2s_id_by_name(afe, source->name)) + return i2s_priv->low_jitter_en; + + return 0; +} + +static int mtk_afe_i2s_apll_connect(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_dapm_widget *w = sink; + struct snd_soc_component *cmpnt = snd_soc_dapm_to_component(w->dapm); + struct mtk_base_afe *afe = snd_soc_component_get_drvdata(cmpnt); + struct mtk_afe_i2s_priv *i2s_priv; + int cur_apll; + int i2s_need_apll; + + i2s_priv = get_i2s_priv_by_name(afe, w->name); + + if (!i2s_priv) { + dev_err(afe->dev, "%s(), i2s_priv == NULL", __func__); + return 0; + } + + /* which apll */ + cur_apll = mt8186_get_apll_by_name(afe, source->name); + + /* choose APLL from i2s rate */ + i2s_need_apll = mt8186_get_apll_by_rate(afe, i2s_priv->rate); + + return (i2s_need_apll == cur_apll) ? 1 : 0; +} + +static int mtk_afe_i2s_mclk_connect(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_dapm_widget *w = sink; + struct snd_soc_component *cmpnt = snd_soc_dapm_to_component(w->dapm); + struct mtk_base_afe *afe = snd_soc_component_get_drvdata(cmpnt); + struct mtk_afe_i2s_priv *i2s_priv; + + i2s_priv = get_i2s_priv_by_name(afe, sink->name); + + if (!i2s_priv) { + dev_err(afe->dev, "%s(), i2s_priv == NULL", __func__); + return 0; + } + + if (get_i2s_id_by_name(afe, sink->name) == + get_i2s_id_by_name(afe, source->name)) + return (i2s_priv->mclk_rate > 0) ? 1 : 0; + + /* check if share i2s need mclk */ + if (i2s_priv->share_i2s_id < 0) + return 0; + + if (i2s_priv->share_i2s_id == get_i2s_id_by_name(afe, source->name)) + return (i2s_priv->mclk_rate > 0) ? 1 : 0; + + return 0; +} + +static int mtk_afe_mclk_apll_connect(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_dapm_widget *w = sink; + struct snd_soc_component *cmpnt = snd_soc_dapm_to_component(w->dapm); + struct mtk_base_afe *afe = snd_soc_component_get_drvdata(cmpnt); + struct mtk_afe_i2s_priv *i2s_priv; + int cur_apll; + + i2s_priv = get_i2s_priv_by_name(afe, w->name); + + if (!i2s_priv) { + dev_err(afe->dev, "%s(), i2s_priv == NULL", __func__); + return 0; + } + + /* which apll */ + cur_apll = mt8186_get_apll_by_name(afe, source->name); + + return (i2s_priv->mclk_apll == cur_apll) ? 1 : 0; +} + +static const struct snd_soc_dapm_route mtk_dai_i2s_routes[] = { + {"Connsys I2S", NULL, "CONNSYS"}, + + /* i2s0 */ + {"I2S0", NULL, "I2S0_EN"}, + {"I2S0", NULL, "I2S1_EN", mtk_afe_i2s_share_connect}, + {"I2S0", NULL, "I2S2_EN", mtk_afe_i2s_share_connect}, + {"I2S0", NULL, "I2S3_EN", mtk_afe_i2s_share_connect}, + + {"I2S0", NULL, I2S0_HD_EN_W_NAME, mtk_afe_i2s_hd_connect}, + {"I2S0", NULL, I2S1_HD_EN_W_NAME, mtk_afe_i2s_hd_connect}, + {"I2S0", NULL, I2S2_HD_EN_W_NAME, mtk_afe_i2s_hd_connect}, + {"I2S0", NULL, I2S3_HD_EN_W_NAME, mtk_afe_i2s_hd_connect}, + {I2S0_HD_EN_W_NAME, NULL, APLL1_W_NAME, mtk_afe_i2s_apll_connect}, + {I2S0_HD_EN_W_NAME, NULL, APLL2_W_NAME, mtk_afe_i2s_apll_connect}, + + {"I2S0", NULL, I2S0_MCLK_EN_W_NAME, mtk_afe_i2s_mclk_connect}, + {"I2S0", NULL, I2S1_MCLK_EN_W_NAME, mtk_afe_i2s_mclk_connect}, + {"I2S0", NULL, I2S2_MCLK_EN_W_NAME, mtk_afe_i2s_mclk_connect}, + {"I2S0", NULL, I2S3_MCLK_EN_W_NAME, mtk_afe_i2s_mclk_connect}, + {I2S0_MCLK_EN_W_NAME, NULL, APLL1_W_NAME, mtk_afe_mclk_apll_connect}, + {I2S0_MCLK_EN_W_NAME, NULL, APLL2_W_NAME, mtk_afe_mclk_apll_connect}, + + /* i2s1 */ + {"I2S1_CH1", "DL1_CH1 Switch", "DL1"}, + {"I2S1_CH2", "DL1_CH2 Switch", "DL1"}, + + {"I2S1_CH1", "DL2_CH1 Switch", "DL2"}, + {"I2S1_CH2", "DL2_CH2 Switch", "DL2"}, + + {"I2S1_CH1", "DL3_CH1 Switch", "DL3"}, + {"I2S1_CH2", "DL3_CH2 Switch", "DL3"}, + + {"I2S1_CH1", "DL12_CH1 Switch", "DL12"}, + {"I2S1_CH2", "DL12_CH2 Switch", "DL12"}, + + {"I2S1_CH1", "DL12_CH3 Switch", "DL12"}, + {"I2S1_CH2", "DL12_CH4 Switch", "DL12"}, + + {"I2S1_CH1", "DL6_CH1 Switch", "DL6"}, + {"I2S1_CH2", "DL6_CH2 Switch", "DL6"}, + + {"I2S1_CH1", "DL4_CH1 Switch", "DL4"}, + {"I2S1_CH2", "DL4_CH2 Switch", "DL4"}, + + {"I2S1_CH1", "DL5_CH1 Switch", "DL5"}, + {"I2S1_CH2", "DL5_CH2 Switch", "DL5"}, + + {"I2S1_CH1", "DL8_CH1 Switch", "DL8"}, + {"I2S1_CH2", "DL8_CH2 Switch", "DL8"}, + + {"I2S1", NULL, "I2S1_CH1"}, + {"I2S1", NULL, "I2S1_CH2"}, + + {"I2S1", NULL, "I2S0_EN", mtk_afe_i2s_share_connect}, + {"I2S1", NULL, "I2S1_EN"}, + {"I2S1", NULL, "I2S2_EN", mtk_afe_i2s_share_connect}, + {"I2S1", NULL, "I2S3_EN", mtk_afe_i2s_share_connect}, + + {"I2S1", NULL, I2S0_HD_EN_W_NAME, mtk_afe_i2s_hd_connect}, + {"I2S1", NULL, I2S1_HD_EN_W_NAME, mtk_afe_i2s_hd_connect}, + {"I2S1", NULL, I2S2_HD_EN_W_NAME, mtk_afe_i2s_hd_connect}, + {"I2S1", NULL, I2S3_HD_EN_W_NAME, mtk_afe_i2s_hd_connect}, + {I2S1_HD_EN_W_NAME, NULL, APLL1_W_NAME, mtk_afe_i2s_apll_connect}, + {I2S1_HD_EN_W_NAME, NULL, APLL2_W_NAME, mtk_afe_i2s_apll_connect}, + + {"I2S1", NULL, I2S0_MCLK_EN_W_NAME, mtk_afe_i2s_mclk_connect}, + {"I2S1", NULL, I2S1_MCLK_EN_W_NAME, mtk_afe_i2s_mclk_connect}, + {"I2S1", NULL, I2S2_MCLK_EN_W_NAME, mtk_afe_i2s_mclk_connect}, + {"I2S1", NULL, I2S3_MCLK_EN_W_NAME, mtk_afe_i2s_mclk_connect}, + {I2S1_MCLK_EN_W_NAME, NULL, APLL1_W_NAME, mtk_afe_mclk_apll_connect}, + {I2S1_MCLK_EN_W_NAME, NULL, APLL2_W_NAME, mtk_afe_mclk_apll_connect}, + + /* i2s2 */ + {"I2S2", NULL, "I2S0_EN", mtk_afe_i2s_share_connect}, + {"I2S2", NULL, "I2S1_EN", mtk_afe_i2s_share_connect}, + {"I2S2", NULL, "I2S2_EN"}, + {"I2S2", NULL, "I2S3_EN", mtk_afe_i2s_share_connect}, + + {"I2S2", NULL, I2S0_HD_EN_W_NAME, mtk_afe_i2s_hd_connect}, + {"I2S2", NULL, I2S1_HD_EN_W_NAME, mtk_afe_i2s_hd_connect}, + {"I2S2", NULL, I2S2_HD_EN_W_NAME, mtk_afe_i2s_hd_connect}, + {"I2S2", NULL, I2S3_HD_EN_W_NAME, mtk_afe_i2s_hd_connect}, + {I2S2_HD_EN_W_NAME, NULL, APLL1_W_NAME, mtk_afe_i2s_apll_connect}, + {I2S2_HD_EN_W_NAME, NULL, APLL2_W_NAME, mtk_afe_i2s_apll_connect}, + + {"I2S2", NULL, I2S0_MCLK_EN_W_NAME, mtk_afe_i2s_mclk_connect}, + {"I2S2", NULL, I2S1_MCLK_EN_W_NAME, mtk_afe_i2s_mclk_connect}, + {"I2S2", NULL, I2S2_MCLK_EN_W_NAME, mtk_afe_i2s_mclk_connect}, + {"I2S2", NULL, I2S3_MCLK_EN_W_NAME, mtk_afe_i2s_mclk_connect}, + {I2S2_MCLK_EN_W_NAME, NULL, APLL1_W_NAME, mtk_afe_mclk_apll_connect}, + {I2S2_MCLK_EN_W_NAME, NULL, APLL2_W_NAME, mtk_afe_mclk_apll_connect}, + + /* i2s3 */ + {"I2S3_CH1", "DL1_CH1 Switch", "DL1"}, + {"I2S3_CH2", "DL1_CH2 Switch", "DL1"}, + + {"I2S3_CH1", "DL2_CH1 Switch", "DL2"}, + {"I2S3_CH2", "DL2_CH2 Switch", "DL2"}, + + {"I2S3_CH1", "DL3_CH1 Switch", "DL3"}, + {"I2S3_CH2", "DL3_CH2 Switch", "DL3"}, + + {"I2S3_CH1", "DL12_CH1 Switch", "DL12"}, + {"I2S3_CH2", "DL12_CH2 Switch", "DL12"}, + + {"I2S3_CH1", "DL12_CH3 Switch", "DL12"}, + {"I2S3_CH2", "DL12_CH4 Switch", "DL12"}, + + {"I2S3_CH1", "DL6_CH1 Switch", "DL6"}, + {"I2S3_CH2", "DL6_CH2 Switch", "DL6"}, + + {"I2S3_CH1", "DL4_CH1 Switch", "DL4"}, + {"I2S3_CH2", "DL4_CH2 Switch", "DL4"}, + + {"I2S3_CH1", "DL5_CH1 Switch", "DL5"}, + {"I2S3_CH2", "DL5_CH2 Switch", "DL5"}, + + {"I2S3_CH1", "DL8_CH1 Switch", "DL8"}, + {"I2S3_CH2", "DL8_CH2 Switch", "DL8"}, + + {"I2S3", NULL, "I2S3_CH1"}, + {"I2S3", NULL, "I2S3_CH2"}, + + {"I2S3", NULL, "I2S0_EN", mtk_afe_i2s_share_connect}, + {"I2S3", NULL, "I2S1_EN", mtk_afe_i2s_share_connect}, + {"I2S3", NULL, "I2S2_EN", mtk_afe_i2s_share_connect}, + {"I2S3", NULL, "I2S3_EN"}, + + {"I2S3", NULL, I2S0_HD_EN_W_NAME, mtk_afe_i2s_hd_connect}, + {"I2S3", NULL, I2S1_HD_EN_W_NAME, mtk_afe_i2s_hd_connect}, + {"I2S3", NULL, I2S2_HD_EN_W_NAME, mtk_afe_i2s_hd_connect}, + {"I2S3", NULL, I2S3_HD_EN_W_NAME, mtk_afe_i2s_hd_connect}, + {I2S3_HD_EN_W_NAME, NULL, APLL1_W_NAME, mtk_afe_i2s_apll_connect}, + {I2S3_HD_EN_W_NAME, NULL, APLL2_W_NAME, mtk_afe_i2s_apll_connect}, + + {"I2S3", NULL, I2S0_MCLK_EN_W_NAME, mtk_afe_i2s_mclk_connect}, + {"I2S3", NULL, I2S1_MCLK_EN_W_NAME, mtk_afe_i2s_mclk_connect}, + {"I2S3", NULL, I2S2_MCLK_EN_W_NAME, mtk_afe_i2s_mclk_connect}, + {"I2S3", NULL, I2S3_MCLK_EN_W_NAME, mtk_afe_i2s_mclk_connect}, + {I2S3_MCLK_EN_W_NAME, NULL, APLL1_W_NAME, mtk_afe_mclk_apll_connect}, + {I2S3_MCLK_EN_W_NAME, NULL, APLL2_W_NAME, mtk_afe_mclk_apll_connect}, + + /* allow i2s on without codec on */ + {"I2S0", NULL, "I2S0_In_Mux"}, + {"I2S0_In_Mux", "Dummy_Widget", "I2S_DUMMY_IN"}, + + {"I2S1_Out_Mux", "Dummy_Widget", "I2S1"}, + {"I2S_DUMMY_OUT", NULL, "I2S1_Out_Mux"}, + + {"I2S2", NULL, "I2S2_In_Mux"}, + {"I2S2_In_Mux", "Dummy_Widget", "I2S_DUMMY_IN"}, + + {"I2S3_Out_Mux", "Dummy_Widget", "I2S3"}, + {"I2S_DUMMY_OUT", NULL, "I2S3_Out_Mux"}, + + /* i2s in lpbk */ + {"I2S0_Lpbk_Mux", "Lpbk", "I2S3"}, + {"I2S2_Lpbk_Mux", "Lpbk", "I2S1"}, + {"I2S0", NULL, "I2S0_Lpbk_Mux"}, + {"I2S2", NULL, "I2S2_Lpbk_Mux"}, +}; + +/* dai ops */ +static int mtk_dai_connsys_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); + unsigned int rate = params_rate(params); + unsigned int rate_reg = mt8186_rate_transform(afe->dev, + rate, dai->id); + unsigned int i2s_con = 0; + + dev_dbg(afe->dev, "%s(), id %d, stream %d, rate %d\n", + __func__, dai->id, substream->stream, rate); + + /* non-inverse, i2s mode, slave, 16bits, from connsys */ + i2s_con |= 0 << INV_PAD_CTRL_SFT; + i2s_con |= I2S_FMT_I2S << I2S_FMT_SFT; + i2s_con |= 1 << I2S_SRC_SFT; + i2s_con |= get_i2s_wlen(SNDRV_PCM_FORMAT_S16_LE) << I2S_WLEN_SFT; + i2s_con |= 0 << I2SIN_PAD_SEL_SFT; + regmap_write(afe->regmap, AFE_CONNSYS_I2S_CON, i2s_con); + + /* use asrc */ + regmap_update_bits(afe->regmap, AFE_CONNSYS_I2S_CON, + I2S_BYPSRC_MASK_SFT, 0); + + /* slave mode, set i2s for asrc */ + regmap_update_bits(afe->regmap, AFE_CONNSYS_I2S_CON, + I2S_MODE_MASK_SFT, rate_reg << I2S_MODE_SFT); + + if (rate == 44100) + regmap_write(afe->regmap, AFE_ASRC_2CH_CON3, 0x1b9000); + else if (rate == 32000) + regmap_write(afe->regmap, AFE_ASRC_2CH_CON3, 0x140000); + else + regmap_write(afe->regmap, AFE_ASRC_2CH_CON3, 0x1e0000); + + /* Calibration setting */ + regmap_write(afe->regmap, AFE_ASRC_2CH_CON4, 0x140000); + regmap_write(afe->regmap, AFE_ASRC_2CH_CON9, 0x36000); + regmap_write(afe->regmap, AFE_ASRC_2CH_CON10, 0x2fc00); + regmap_write(afe->regmap, AFE_ASRC_2CH_CON6, 0x7ef4); + regmap_write(afe->regmap, AFE_ASRC_2CH_CON5, 0xff5986); + + /* 0:Stereo 1:Mono */ + regmap_update_bits(afe->regmap, AFE_ASRC_2CH_CON2, + CHSET_IS_MONO_MASK_SFT, 0); + + return 0; +} + +static int mtk_dai_connsys_i2s_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); + struct mt8186_afe_private *afe_priv = afe->platform_priv; + + dev_dbg(afe->dev, "%s(), cmd %d, stream %d\n", + __func__, cmd, substream->stream); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + /* i2s enable */ + regmap_update_bits(afe->regmap, + AFE_CONNSYS_I2S_CON, + I2S_EN_MASK_SFT, + BIT(I2S_EN_SFT)); + + /* calibrator enable */ + regmap_update_bits(afe->regmap, + AFE_ASRC_2CH_CON5, + CALI_EN_MASK_SFT, + BIT(CALI_EN_SFT)); + + /* asrc enable */ + regmap_update_bits(afe->regmap, + AFE_ASRC_2CH_CON0, + CON0_CHSET_STR_CLR_MASK_SFT, + BIT(CON0_CHSET_STR_CLR_SFT)); + regmap_update_bits(afe->regmap, + AFE_ASRC_2CH_CON0, + CON0_ASM_ON_MASK_SFT, + BIT(CON0_ASM_ON_SFT)); + + afe_priv->dai_on[dai->id] = true; + return 0; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + regmap_update_bits(afe->regmap, AFE_ASRC_2CH_CON0, + CON0_ASM_ON_MASK_SFT, 0); + regmap_update_bits(afe->regmap, AFE_ASRC_2CH_CON5, + CALI_EN_MASK_SFT, 0); + + /* i2s disable */ + regmap_update_bits(afe->regmap, AFE_CONNSYS_I2S_CON, + I2S_EN_MASK_SFT, 0); + + /* bypass asrc */ + regmap_update_bits(afe->regmap, AFE_CONNSYS_I2S_CON, + I2S_BYPSRC_MASK_SFT, BIT(I2S_BYPSRC_SFT)); + + afe_priv->dai_on[dai->id] = false; + return 0; + default: + return -EINVAL; + } + return 0; +} + +static const struct snd_soc_dai_ops mtk_dai_connsys_i2s_ops = { + .hw_params = mtk_dai_connsys_i2s_hw_params, + .trigger = mtk_dai_connsys_i2s_trigger, +}; + +/* i2s */ +static int mtk_dai_i2s_config(struct mtk_base_afe *afe, + struct snd_pcm_hw_params *params, + int i2s_id) +{ + struct mt8186_afe_private *afe_priv = afe->platform_priv; + struct mtk_afe_i2s_priv *i2s_priv = afe_priv->dai_priv[i2s_id]; + + unsigned int rate = params_rate(params); + unsigned int rate_reg = mt8186_rate_transform(afe->dev, + rate, i2s_id); + snd_pcm_format_t format = params_format(params); + unsigned int i2s_con = 0; + int ret; + + dev_dbg(afe->dev, "%s(), id %d, rate %d, format %d\n", + __func__, i2s_id, rate, format); + + if (!i2s_priv) { + dev_err(afe->dev, "%s(), i2s_priv == NULL", __func__); + return -EINVAL; + } + + i2s_priv->rate = rate; + + switch (i2s_id) { + case MT8186_DAI_I2S_0: + i2s_con = I2S_IN_PAD_IO_MUX << I2SIN_PAD_SEL_SFT; + i2s_con |= rate_reg << I2S_OUT_MODE_SFT; + i2s_con |= I2S_FMT_I2S << I2S_FMT_SFT; + i2s_con |= get_i2s_wlen(format) << I2S_WLEN_SFT; + regmap_update_bits(afe->regmap, AFE_I2S_CON, + 0xffffeffa, i2s_con); + break; + case MT8186_DAI_I2S_1: + i2s_con = I2S1_SEL_O28_O29 << I2S2_SEL_O03_O04_SFT; + i2s_con |= rate_reg << I2S2_OUT_MODE_SFT; + i2s_con |= I2S_FMT_I2S << I2S2_FMT_SFT; + i2s_con |= get_i2s_wlen(format) << I2S2_WLEN_SFT; + regmap_update_bits(afe->regmap, AFE_I2S_CON1, + 0xffffeffa, i2s_con); + break; + case MT8186_DAI_I2S_2: + i2s_con = 8 << I2S3_UPDATE_WORD_SFT; + i2s_con |= rate_reg << I2S3_OUT_MODE_SFT; + i2s_con |= I2S_FMT_I2S << I2S3_FMT_SFT; + i2s_con |= get_i2s_wlen(format) << I2S3_WLEN_SFT; + regmap_update_bits(afe->regmap, AFE_I2S_CON2, + 0xffffeffa, i2s_con); + break; + case MT8186_DAI_I2S_3: + i2s_con = rate_reg << I2S4_OUT_MODE_SFT; + i2s_con |= I2S_FMT_I2S << I2S4_FMT_SFT; + i2s_con |= get_i2s_wlen(format) << I2S4_WLEN_SFT; + regmap_update_bits(afe->regmap, AFE_I2S_CON3, + 0xffffeffa, i2s_con); + break; + default: + dev_err(afe->dev, "%s(), id %d not support\n", + __func__, i2s_id); + return -EINVAL; + } + + /* set share i2s */ + if (i2s_priv && i2s_priv->share_i2s_id >= 0) { + ret = mtk_dai_i2s_config(afe, params, i2s_priv->share_i2s_id); + if (ret) + return ret; + } + + return 0; +} + +static int mtk_dai_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); + + return mtk_dai_i2s_config(afe, params, dai->id); +} + +static int mtk_dai_i2s_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct mtk_base_afe *afe = dev_get_drvdata(dai->dev); + struct mt8186_afe_private *afe_priv = afe->platform_priv; + struct mtk_afe_i2s_priv *i2s_priv = afe_priv->dai_priv[dai->id]; + int apll; + int apll_rate; + + if (!i2s_priv) { + dev_err(afe->dev, "%s(), i2s_priv == NULL", __func__); + return -EINVAL; + } + + if (dir != SND_SOC_CLOCK_OUT) { + dev_err(afe->dev, "%s(), dir != SND_SOC_CLOCK_OUT", __func__); + return -EINVAL; + } + + dev_dbg(afe->dev, "%s(), freq %d\n", __func__, freq); + + apll = mt8186_get_apll_by_rate(afe, freq); + apll_rate = mt8186_get_apll_rate(afe, apll); + + if (freq > apll_rate) { + dev_err(afe->dev, "%s(), freq > apll rate", __func__); + return -EINVAL; + } + + if (apll_rate % freq != 0) { + dev_err(afe->dev, "%s(), APLL cannot generate freq Hz", __func__); + return -EINVAL; + } + + i2s_priv->mclk_rate = freq; + i2s_priv->mclk_apll = apll; + + if (i2s_priv->share_i2s_id > 0) { + struct mtk_afe_i2s_priv *share_i2s_priv; + + share_i2s_priv = afe_priv->dai_priv[i2s_priv->share_i2s_id]; + if (!share_i2s_priv) { + dev_err(afe->dev, "%s(), share_i2s_priv == NULL", __func__); + return -EINVAL; + } + + share_i2s_priv->mclk_rate = i2s_priv->mclk_rate; + share_i2s_priv->mclk_apll = i2s_priv->mclk_apll; + } + + return 0; +} + +static const struct snd_soc_dai_ops mtk_dai_i2s_ops = { + .hw_params = mtk_dai_i2s_hw_params, + .set_sysclk = mtk_dai_i2s_set_sysclk, +}; + +/* dai driver */ +#define MTK_CONNSYS_I2S_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) + +#define MTK_I2S_RATES (SNDRV_PCM_RATE_8000_48000 |\ + SNDRV_PCM_RATE_88200 |\ + SNDRV_PCM_RATE_96000 |\ + SNDRV_PCM_RATE_176400 |\ + SNDRV_PCM_RATE_192000) + +#define MTK_I2S_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver mtk_dai_i2s_driver[] = { + { + .name = "CONNSYS_I2S", + .id = MT8186_DAI_CONNSYS_I2S, + .capture = { + .stream_name = "Connsys I2S", + .channels_min = 1, + .channels_max = 2, + .rates = MTK_CONNSYS_I2S_RATES, + .formats = MTK_I2S_FORMATS, + }, + .ops = &mtk_dai_connsys_i2s_ops, + }, + { + .name = "I2S0", + .id = MT8186_DAI_I2S_0, + .capture = { + .stream_name = "I2S0", + .channels_min = 1, + .channels_max = 2, + .rates = MTK_I2S_RATES, + .formats = MTK_I2S_FORMATS, + }, + .ops = &mtk_dai_i2s_ops, + }, + { + .name = "I2S1", + .id = MT8186_DAI_I2S_1, + .playback = { + .stream_name = "I2S1", + .channels_min = 1, + .channels_max = 2, + .rates = MTK_I2S_RATES, + .formats = MTK_I2S_FORMATS, + }, + .ops = &mtk_dai_i2s_ops, + }, + { + .name = "I2S2", + .id = MT8186_DAI_I2S_2, + .capture = { + .stream_name = "I2S2", + .channels_min = 1, + .channels_max = 2, + .rates = MTK_I2S_RATES, + .formats = MTK_I2S_FORMATS, + }, + .ops = &mtk_dai_i2s_ops, + }, + { + .name = "I2S3", + .id = MT8186_DAI_I2S_3, + .playback = { + .stream_name = "I2S3", + .channels_min = 1, + .channels_max = 2, + .rates = MTK_I2S_RATES, + .formats = MTK_I2S_FORMATS, + }, + .ops = &mtk_dai_i2s_ops, + } +}; + +/* this enum is merely for mtk_afe_i2s_priv declare */ +enum { + DAI_I2S0 = 0, + DAI_I2S1, + DAI_I2S2, + DAI_I2S3, + DAI_I2S_NUM, +}; + +static const struct mtk_afe_i2s_priv mt8186_i2s_priv[DAI_I2S_NUM] = { + [DAI_I2S0] = { + .id = MT8186_DAI_I2S_0, + .mclk_id = MT8186_I2S0_MCK, + .share_property_name = "i2s0-share", + .share_i2s_id = -1, + }, + [DAI_I2S1] = { + .id = MT8186_DAI_I2S_1, + .mclk_id = MT8186_I2S1_MCK, + .share_property_name = "i2s1-share", + .share_i2s_id = -1, + }, + [DAI_I2S2] = { + .id = MT8186_DAI_I2S_2, + .mclk_id = MT8186_I2S2_MCK, + .share_property_name = "i2s2-share", + .share_i2s_id = -1, + }, + [DAI_I2S3] = { + .id = MT8186_DAI_I2S_3, + /* clock gate naming is hf_faud_i2s4_m_ck*/ + .mclk_id = MT8186_I2S4_MCK, + .share_property_name = "i2s3-share", + .share_i2s_id = -1, + } +}; + +static int mt8186_dai_i2s_get_share(struct mtk_base_afe *afe) +{ + struct mt8186_afe_private *afe_priv = afe->platform_priv; + const struct device_node *of_node = afe->dev->of_node; + const char *of_str; + const char *property_name; + struct mtk_afe_i2s_priv *i2s_priv; + int i; + + for (i = 0; i < DAI_I2S_NUM; i++) { + i2s_priv = afe_priv->dai_priv[mt8186_i2s_priv[i].id]; + property_name = mt8186_i2s_priv[i].share_property_name; + if (of_property_read_string(of_node, property_name, &of_str)) + continue; + i2s_priv->share_i2s_id = get_i2s_id_by_name(afe, of_str); + } + + return 0; +} + +static int mt8186_dai_i2s_set_priv(struct mtk_base_afe *afe) +{ + int i; + int ret; + + for (i = 0; i < DAI_I2S_NUM; i++) { + ret = mt8186_dai_set_priv(afe, mt8186_i2s_priv[i].id, + sizeof(struct mtk_afe_i2s_priv), + &mt8186_i2s_priv[i]); + if (ret) + return ret; + } + + return 0; +} + +int mt8186_dai_i2s_register(struct mtk_base_afe *afe) +{ + struct mtk_base_afe_dai *dai; + int ret; + + dai = devm_kzalloc(afe->dev, sizeof(*dai), GFP_KERNEL); + if (!dai) + return -ENOMEM; + + list_add(&dai->list, &afe->sub_dais); + + dai->dai_drivers = mtk_dai_i2s_driver; + dai->num_dai_drivers = ARRAY_SIZE(mtk_dai_i2s_driver); + + dai->controls = mtk_dai_i2s_controls; + dai->num_controls = ARRAY_SIZE(mtk_dai_i2s_controls); + dai->dapm_widgets = mtk_dai_i2s_widgets; + dai->num_dapm_widgets = ARRAY_SIZE(mtk_dai_i2s_widgets); + dai->dapm_routes = mtk_dai_i2s_routes; + dai->num_dapm_routes = ARRAY_SIZE(mtk_dai_i2s_routes); + + /* set all dai i2s private data */ + ret = mt8186_dai_i2s_set_priv(afe); + if (ret) + return ret; + + /* parse share i2s */ + ret = mt8186_dai_i2s_get_share(afe); + if (ret) + return ret; + + return 0; +} -- cgit v1.2.3 From 920508f9fe2fc90f19916d74f4c23088030d32e0 Mon Sep 17 00:00:00 2001 From: Jiaxin Yu Date: Mon, 23 May 2022 21:28:46 +0800 Subject: ASoC: mediatek: mt8186: support pcm in platform driver Add mt8186 pcm dai driver. Signed-off-by: Jiaxin Yu Reviewed-by: AngeloGioacchino Del Regno Link: https://lore.kernel.org/r/20220523132858.22166-9-jiaxin.yu@mediatek.com Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8186/mt8186-dai-pcm.c | 423 +++++++++++++++++++++++++++++ 1 file changed, 423 insertions(+) create mode 100644 sound/soc/mediatek/mt8186/mt8186-dai-pcm.c (limited to 'sound') diff --git a/sound/soc/mediatek/mt8186/mt8186-dai-pcm.c b/sound/soc/mediatek/mt8186/mt8186-dai-pcm.c new file mode 100644 index 000000000000..0b0032ecfe6d --- /dev/null +++ b/sound/soc/mediatek/mt8186/mt8186-dai-pcm.c @@ -0,0 +1,423 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// MediaTek ALSA SoC Audio DAI I2S Control +// +// Copyright (c) 2022 MediaTek Inc. +// Author: Jiaxin Yu + +#include +#include +#include "mt8186-afe-common.h" +#include "mt8186-afe-gpio.h" +#include "mt8186-interconnection.h" + +struct mtk_afe_pcm_priv { + unsigned int id; + unsigned int fmt; + unsigned int bck_invert; + unsigned int lck_invert; +}; + +enum aud_tx_lch_rpt { + AUD_TX_LCH_RPT_NO_REPEAT = 0, + AUD_TX_LCH_RPT_REPEAT = 1 +}; + +enum aud_vbt_16k_mode { + AUD_VBT_16K_MODE_DISABLE = 0, + AUD_VBT_16K_MODE_ENABLE = 1 +}; + +enum aud_ext_modem { + AUD_EXT_MODEM_SELECT_INTERNAL = 0, + AUD_EXT_MODEM_SELECT_EXTERNAL = 1 +}; + +enum aud_pcm_sync_type { + /* bck sync length = 1 */ + AUD_PCM_ONE_BCK_CYCLE_SYNC = 0, + /* bck sync length = PCM_INTF_CON1[9:13] */ + AUD_PCM_EXTENDED_BCK_CYCLE_SYNC = 1 +}; + +enum aud_bt_mode { + AUD_BT_MODE_DUAL_MIC_ON_TX = 0, + AUD_BT_MODE_SINGLE_MIC_ON_TX = 1 +}; + +enum aud_pcm_afifo_src { + /* slave mode & external modem uses different crystal */ + AUD_PCM_AFIFO_ASRC = 0, + /* slave mode & external modem uses the same crystal */ + AUD_PCM_AFIFO_AFIFO = 1 +}; + +enum aud_pcm_clock_source { + AUD_PCM_CLOCK_MASTER_MODE = 0, + AUD_PCM_CLOCK_SLAVE_MODE = 1 +}; + +enum aud_pcm_wlen { + AUD_PCM_WLEN_PCM_32_BCK_CYCLES = 0, + AUD_PCM_WLEN_PCM_64_BCK_CYCLES = 1 +}; + +enum aud_pcm_24bit { + AUD_PCM_24BIT_PCM_16_BITS = 0, + AUD_PCM_24BIT_PCM_24_BITS = 1 +}; + +enum aud_pcm_mode { + AUD_PCM_MODE_PCM_MODE_8K = 0, + AUD_PCM_MODE_PCM_MODE_16K = 1, + AUD_PCM_MODE_PCM_MODE_32K = 2, + AUD_PCM_MODE_PCM_MODE_48K = 3, +}; + +enum aud_pcm_fmt { + AUD_PCM_FMT_I2S = 0, + AUD_PCM_FMT_EIAJ = 1, + AUD_PCM_FMT_PCM_MODE_A = 2, + AUD_PCM_FMT_PCM_MODE_B = 3 +}; + +enum aud_bclk_out_inv { + AUD_BCLK_OUT_INV_NO_INVERSE = 0, + AUD_BCLK_OUT_INV_INVERSE = 1 +}; + +enum aud_lrclk_out_inv { + AUD_LRCLK_OUT_INV_NO_INVERSE = 0, + AUD_LRCLK_OUT_INV_INVERSE = 1 +}; + +enum aud_pcm_en { + AUD_PCM_EN_DISABLE = 0, + AUD_PCM_EN_ENABLE = 1 +}; + +/* dai component */ +static const struct snd_kcontrol_new mtk_pcm_1_playback_ch1_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("ADDA_UL_CH1 Switch", AFE_CONN7, + I_ADDA_UL_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL2_CH1 Switch", AFE_CONN7, + I_DL2_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL4_CH1 Switch", AFE_CONN7_1, + I_DL4_CH1, 1, 0), +}; + +static const struct snd_kcontrol_new mtk_pcm_1_playback_ch2_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("ADDA_UL_CH2 Switch", AFE_CONN8, + I_ADDA_UL_CH2, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL2_CH2 Switch", AFE_CONN8, + I_DL2_CH2, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL4_CH2 Switch", AFE_CONN8_1, + I_DL4_CH2, 1, 0), +}; + +static int mtk_pcm_en_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_component *cmpnt = snd_soc_dapm_to_component(w->dapm); + struct mtk_base_afe *afe = snd_soc_component_get_drvdata(cmpnt); + + dev_dbg(afe->dev, "%s(), name %s, event 0x%x\n", + __func__, w->name, event); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + mt8186_afe_gpio_request(afe->dev, true, MT8186_DAI_PCM, 0); + break; + case SND_SOC_DAPM_POST_PMD: + mt8186_afe_gpio_request(afe->dev, false, MT8186_DAI_PCM, 0); + break; + } + + return 0; +} + +/* pcm in/out lpbk */ +static const char * const pcm_lpbk_mux_map[] = { + "Normal", "Lpbk", +}; + +static int pcm_lpbk_mux_map_value[] = { + 0, 1, +}; + +static SOC_VALUE_ENUM_SINGLE_AUTODISABLE_DECL(pcm_in_lpbk_mux_map_enum, + PCM_INTF_CON1, + PCM_I2S_PCM_LOOPBACK_SFT, + 1, + pcm_lpbk_mux_map, + pcm_lpbk_mux_map_value); + +static const struct snd_kcontrol_new pcm_in_lpbk_mux_control = + SOC_DAPM_ENUM("PCM In Lpbk Select", pcm_in_lpbk_mux_map_enum); + +static SOC_VALUE_ENUM_SINGLE_AUTODISABLE_DECL(pcm_out_lpbk_mux_map_enum, + PCM_INTF_CON1, + PCM_I2S_PCM_LOOPBACK_SFT, + 1, + pcm_lpbk_mux_map, + pcm_lpbk_mux_map_value); + +static const struct snd_kcontrol_new pcm_out_lpbk_mux_control = + SOC_DAPM_ENUM("PCM Out Lpbk Select", pcm_out_lpbk_mux_map_enum); + +static const struct snd_soc_dapm_widget mtk_dai_pcm_widgets[] = { + /* inter-connections */ + SND_SOC_DAPM_MIXER("PCM_1_PB_CH1", SND_SOC_NOPM, 0, 0, + mtk_pcm_1_playback_ch1_mix, + ARRAY_SIZE(mtk_pcm_1_playback_ch1_mix)), + SND_SOC_DAPM_MIXER("PCM_1_PB_CH2", SND_SOC_NOPM, 0, 0, + mtk_pcm_1_playback_ch2_mix, + ARRAY_SIZE(mtk_pcm_1_playback_ch2_mix)), + + SND_SOC_DAPM_SUPPLY("PCM_1_EN", + PCM_INTF_CON1, PCM_EN_SFT, 0, + mtk_pcm_en_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + /* pcm in lpbk */ + SND_SOC_DAPM_MUX("PCM_In_Lpbk_Mux", + SND_SOC_NOPM, 0, 0, &pcm_in_lpbk_mux_control), + + /* pcm out lpbk */ + SND_SOC_DAPM_MUX("PCM_Out_Lpbk_Mux", + SND_SOC_NOPM, 0, 0, &pcm_out_lpbk_mux_control), +}; + +static const struct snd_soc_dapm_route mtk_dai_pcm_routes[] = { + {"PCM 1 Playback", NULL, "PCM_1_PB_CH1"}, + {"PCM 1 Playback", NULL, "PCM_1_PB_CH2"}, + + {"PCM 1 Playback", NULL, "PCM_1_EN"}, + {"PCM 1 Capture", NULL, "PCM_1_EN"}, + + {"PCM_1_PB_CH1", "DL2_CH1 Switch", "DL2"}, + {"PCM_1_PB_CH2", "DL2_CH2 Switch", "DL2"}, + + {"PCM_1_PB_CH1", "DL4_CH1 Switch", "DL4"}, + {"PCM_1_PB_CH2", "DL4_CH2 Switch", "DL4"}, + + /* pcm out lpbk */ + {"PCM_Out_Lpbk_Mux", "Lpbk", "PCM 1 Playback"}, + {"I2S0", NULL, "PCM_Out_Lpbk_Mux"}, + + /* pcm in lpbk */ + {"PCM_In_Lpbk_Mux", "Lpbk", "PCM 1 Capture"}, + {"I2S3", NULL, "PCM_In_Lpbk_Mux"}, +}; + +/* dai ops */ +static int mtk_dai_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); + struct mt8186_afe_private *afe_priv = afe->platform_priv; + int pcm_id = dai->id; + struct mtk_afe_pcm_priv *pcm_priv = afe_priv->dai_priv[pcm_id]; + unsigned int rate = params_rate(params); + unsigned int rate_reg = mt8186_rate_transform(afe->dev, rate, dai->id); + snd_pcm_format_t format = params_format(params); + unsigned int data_width = + snd_pcm_format_width(format); + unsigned int wlen_width = + snd_pcm_format_physical_width(format); + unsigned int pcm_con = 0; + + dev_dbg(afe->dev, "%s(), id %d, stream %d, widget active p %d, c %d\n", + __func__, dai->id, substream->stream, dai->playback_widget->active, + dai->capture_widget->active); + dev_dbg(afe->dev, "%s(), rate %d, rate_reg %d, data_width %d, wlen_width %d\n", + __func__, rate, rate_reg, data_width, wlen_width); + + if (dai->playback_widget->active || dai->capture_widget->active) + return 0; + + switch (dai->id) { + case MT8186_DAI_PCM: + pcm_con |= AUD_TX_LCH_RPT_NO_REPEAT << PCM_TX_LCH_RPT_SFT; + pcm_con |= AUD_VBT_16K_MODE_DISABLE << PCM_VBT_16K_MODE_SFT; + pcm_con |= AUD_EXT_MODEM_SELECT_EXTERNAL << PCM_EXT_MODEM_SFT; + pcm_con |= AUD_PCM_ONE_BCK_CYCLE_SYNC << PCM_SYNC_TYPE_SFT; + pcm_con |= AUD_BT_MODE_DUAL_MIC_ON_TX << PCM_BT_MODE_SFT; + pcm_con |= AUD_PCM_AFIFO_AFIFO << PCM_BYP_ASRC_SFT; + pcm_con |= AUD_PCM_CLOCK_MASTER_MODE << PCM_SLAVE_SFT; + pcm_con |= 0 << PCM_SYNC_LENGTH_SFT; + + /* sampling rate */ + pcm_con |= rate_reg << PCM_MODE_SFT; + + /* format */ + pcm_con |= pcm_priv->fmt << PCM_FMT_SFT; + + /* 24bit data width */ + if (data_width > 16) + pcm_con |= AUD_PCM_24BIT_PCM_24_BITS << PCM_24BIT_SFT; + else + pcm_con |= AUD_PCM_24BIT_PCM_16_BITS << PCM_24BIT_SFT; + + /* wlen width*/ + if (wlen_width > 16) + pcm_con |= AUD_PCM_WLEN_PCM_64_BCK_CYCLES << PCM_WLEN_SFT; + else + pcm_con |= AUD_PCM_WLEN_PCM_32_BCK_CYCLES << PCM_WLEN_SFT; + + /* clock invert */ + pcm_con |= pcm_priv->lck_invert << PCM_SYNC_OUT_INV_SFT; + pcm_con |= pcm_priv->bck_invert << PCM_BCLK_OUT_INV_SFT; + + regmap_update_bits(afe->regmap, PCM_INTF_CON1, 0xfffffffe, pcm_con); + break; + default: + dev_err(afe->dev, "%s(), id %d not support\n", __func__, dai->id); + return -EINVAL; + } + + return 0; +} + +static int mtk_dai_pcm_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); + struct mt8186_afe_private *afe_priv = afe->platform_priv; + struct mtk_afe_pcm_priv *pcm_priv = afe_priv->dai_priv[dai->id]; + + if (!pcm_priv) { + dev_err(afe->dev, "%s(), tdm_priv == NULL", __func__); + return -EINVAL; + } + + /* DAI mode*/ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + pcm_priv->fmt = AUD_PCM_FMT_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + pcm_priv->fmt = AUD_PCM_FMT_EIAJ; + break; + case SND_SOC_DAIFMT_DSP_A: + pcm_priv->fmt = AUD_PCM_FMT_PCM_MODE_A; + break; + case SND_SOC_DAIFMT_DSP_B: + pcm_priv->fmt = AUD_PCM_FMT_PCM_MODE_B; + break; + default: + pcm_priv->fmt = AUD_PCM_FMT_I2S; + } + + /* DAI clock inversion*/ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + pcm_priv->bck_invert = AUD_BCLK_OUT_INV_NO_INVERSE; + pcm_priv->lck_invert = AUD_LRCLK_OUT_INV_NO_INVERSE; + break; + case SND_SOC_DAIFMT_NB_IF: + pcm_priv->bck_invert = AUD_BCLK_OUT_INV_NO_INVERSE; + pcm_priv->lck_invert = AUD_BCLK_OUT_INV_INVERSE; + break; + case SND_SOC_DAIFMT_IB_NF: + pcm_priv->bck_invert = AUD_BCLK_OUT_INV_INVERSE; + pcm_priv->lck_invert = AUD_LRCLK_OUT_INV_NO_INVERSE; + break; + case SND_SOC_DAIFMT_IB_IF: + pcm_priv->bck_invert = AUD_BCLK_OUT_INV_INVERSE; + pcm_priv->lck_invert = AUD_BCLK_OUT_INV_INVERSE; + break; + default: + pcm_priv->bck_invert = AUD_BCLK_OUT_INV_NO_INVERSE; + pcm_priv->lck_invert = AUD_LRCLK_OUT_INV_NO_INVERSE; + break; + } + + return 0; +} + +static const struct snd_soc_dai_ops mtk_dai_pcm_ops = { + .hw_params = mtk_dai_pcm_hw_params, + .set_fmt = mtk_dai_pcm_set_fmt, +}; + +/* dai driver */ +#define MTK_PCM_RATES (SNDRV_PCM_RATE_8000 |\ + SNDRV_PCM_RATE_16000 |\ + SNDRV_PCM_RATE_32000 |\ + SNDRV_PCM_RATE_48000) + +#define MTK_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver mtk_dai_pcm_driver[] = { + { + .name = "PCM 1", + .id = MT8186_DAI_PCM, + .playback = { + .stream_name = "PCM 1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = MTK_PCM_RATES, + .formats = MTK_PCM_FORMATS, + }, + .capture = { + .stream_name = "PCM 1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = MTK_PCM_RATES, + .formats = MTK_PCM_FORMATS, + }, + .ops = &mtk_dai_pcm_ops, + .symmetric_rate = 1, + .symmetric_sample_bits = 1, + }, +}; + +static struct mtk_afe_pcm_priv *init_pcm_priv_data(struct mtk_base_afe *afe) +{ + struct mtk_afe_pcm_priv *pcm_priv; + + pcm_priv = devm_kzalloc(afe->dev, sizeof(struct mtk_afe_pcm_priv), + GFP_KERNEL); + if (!pcm_priv) + return NULL; + + pcm_priv->id = MT8186_DAI_PCM; + pcm_priv->fmt = AUD_PCM_FMT_I2S; + pcm_priv->bck_invert = AUD_BCLK_OUT_INV_NO_INVERSE; + pcm_priv->lck_invert = AUD_LRCLK_OUT_INV_NO_INVERSE; + + return pcm_priv; +} + +int mt8186_dai_pcm_register(struct mtk_base_afe *afe) +{ + struct mt8186_afe_private *afe_priv = afe->platform_priv; + struct mtk_afe_pcm_priv *pcm_priv; + struct mtk_base_afe_dai *dai; + + dai = devm_kzalloc(afe->dev, sizeof(*dai), GFP_KERNEL); + if (!dai) + return -ENOMEM; + + list_add(&dai->list, &afe->sub_dais); + + dai->dai_drivers = mtk_dai_pcm_driver; + dai->num_dai_drivers = ARRAY_SIZE(mtk_dai_pcm_driver); + + dai->dapm_widgets = mtk_dai_pcm_widgets; + dai->num_dapm_widgets = ARRAY_SIZE(mtk_dai_pcm_widgets); + dai->dapm_routes = mtk_dai_pcm_routes; + dai->num_dapm_routes = ARRAY_SIZE(mtk_dai_pcm_routes); + + pcm_priv = init_pcm_priv_data(afe); + if (!pcm_priv) + return -ENOMEM; + + afe_priv->dai_priv[MT8186_DAI_PCM] = pcm_priv; + + return 0; +} -- cgit v1.2.3 From e118015db7bd0dad1744221d0fe18333ebf9c622 Mon Sep 17 00:00:00 2001 From: Jiaxin Yu Date: Mon, 23 May 2022 21:28:47 +0800 Subject: ASoC: mediatek: mt8186: support src in platform driver Add mt8186 src dai driver Signed-off-by: Jiaxin Yu Link: https://lore.kernel.org/r/20220523132858.22166-10-jiaxin.yu@mediatek.com Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8186/mt8186-dai-src.c | 695 +++++++++++++++++++++++++++++ 1 file changed, 695 insertions(+) create mode 100644 sound/soc/mediatek/mt8186/mt8186-dai-src.c (limited to 'sound') diff --git a/sound/soc/mediatek/mt8186/mt8186-dai-src.c b/sound/soc/mediatek/mt8186/mt8186-dai-src.c new file mode 100644 index 000000000000..67989ffd67ca --- /dev/null +++ b/sound/soc/mediatek/mt8186/mt8186-dai-src.c @@ -0,0 +1,695 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// MediaTek ALSA SoC Audio DAI SRC Control +// +// Copyright (c) 2022 MediaTek Inc. +// Author: Jiaxin Yu + +#include +#include "mt8186-afe-common.h" +#include "mt8186-interconnection.h" + +struct mtk_afe_src_priv { + int dl_rate; + int ul_rate; +}; + +static const unsigned int src_iir_coeff_32_to_16[] = { + 0x0dbae6, 0xff9b0a, 0x0dbae6, 0x05e488, 0xe072b9, 0x000002, + 0x0dbae6, 0x000f3b, 0x0dbae6, 0x06a537, 0xe17d79, 0x000002, + 0x0dbae6, 0x01246a, 0x0dbae6, 0x087261, 0xe306be, 0x000002, + 0x0dbae6, 0x03437d, 0x0dbae6, 0x0bc16f, 0xe57c87, 0x000002, + 0x0dbae6, 0x072981, 0x0dbae6, 0x111dd3, 0xe94f2a, 0x000002, + 0x0dbae6, 0x0dc4a6, 0x0dbae6, 0x188611, 0xee85a0, 0x000002, + 0x0dbae6, 0x168b9a, 0x0dbae6, 0x200e8f, 0xf3ccf1, 0x000002, + 0x000000, 0x1b75cb, 0x1b75cb, 0x2374a2, 0x000000, 0x000001 +}; + +static const unsigned int src_iir_coeff_44_to_16[] = { + 0x09ae28, 0xf7d97d, 0x09ae28, 0x212a3d, 0xe0ac3a, 0x000002, + 0x09ae28, 0xf8525a, 0x09ae28, 0x216d72, 0xe234be, 0x000002, + 0x09ae28, 0xf980f5, 0x09ae28, 0x22a057, 0xe45a81, 0x000002, + 0x09ae28, 0xfc0a08, 0x09ae28, 0x24d3bd, 0xe7752d, 0x000002, + 0x09ae28, 0x016162, 0x09ae28, 0x27da01, 0xeb6ea8, 0x000002, + 0x09ae28, 0x0b67df, 0x09ae28, 0x2aca4a, 0xef34c4, 0x000002, + 0x000000, 0x135c50, 0x135c50, 0x2c1079, 0x000000, 0x000001 +}; + +static const unsigned int src_iir_coeff_44_to_32[] = { + 0x096966, 0x0c4d35, 0x096966, 0xedee81, 0xf05070, 0x000003, + 0x12d2cc, 0x193910, 0x12d2cc, 0xddbf4f, 0xe21e1d, 0x000002, + 0x12d2cc, 0x1a9e60, 0x12d2cc, 0xe18916, 0xe470fd, 0x000002, + 0x12d2cc, 0x1d06e0, 0x12d2cc, 0xe8a4a6, 0xe87b24, 0x000002, + 0x12d2cc, 0x207578, 0x12d2cc, 0xf4fe62, 0xef5917, 0x000002, + 0x12d2cc, 0x24055f, 0x12d2cc, 0x05ee2b, 0xf8b502, 0x000002, + 0x000000, 0x25a599, 0x25a599, 0x0fabe2, 0x000000, 0x000001 +}; + +static const unsigned int src_iir_coeff_48_to_16[] = { + 0x0296a4, 0xfd69dd, 0x0296a4, 0x209439, 0xe01ff9, 0x000002, + 0x0f4ff3, 0xf0d6d4, 0x0f4ff3, 0x209bc9, 0xe076c3, 0x000002, + 0x0e8490, 0xf1fe63, 0x0e8490, 0x20cfd6, 0xe12124, 0x000002, + 0x14852f, 0xed794a, 0x14852f, 0x21503d, 0xe28b32, 0x000002, + 0x136222, 0xf17677, 0x136222, 0x225be1, 0xe56964, 0x000002, + 0x0a8d85, 0xfc4a97, 0x0a8d85, 0x24310c, 0xea6952, 0x000002, + 0x05eff5, 0x043455, 0x05eff5, 0x4ced8f, 0xe134d6, 0x000001, + 0x000000, 0x3aebe6, 0x3aebe6, 0x04f3b0, 0x000000, 0x000004 +}; + +static const unsigned int src_iir_coeff_48_to_32[] = { + 0x10c1b8, 0x10a7df, 0x10c1b8, 0xe7514e, 0xe0b41f, 0x000002, + 0x10c1b8, 0x116257, 0x10c1b8, 0xe9402f, 0xe25aaa, 0x000002, + 0x10c1b8, 0x130c89, 0x10c1b8, 0xed3cc3, 0xe4dddb, 0x000002, + 0x10c1b8, 0x1600dd, 0x10c1b8, 0xf48000, 0xe90c55, 0x000002, + 0x10c1b8, 0x1a672e, 0x10c1b8, 0x00494c, 0xefa807, 0x000002, + 0x10c1b8, 0x1f38e6, 0x10c1b8, 0x0ee076, 0xf7c5f3, 0x000002, + 0x000000, 0x218370, 0x218370, 0x168b40, 0x000000, 0x000001 +}; + +static const unsigned int src_iir_coeff_48_to_44[] = { + 0x0bf71c, 0x170f3f, 0x0bf71c, 0xe3a4c8, 0xf096cb, 0x000003, + 0x0bf71c, 0x17395e, 0x0bf71c, 0xe58085, 0xf210c8, 0x000003, + 0x0bf71c, 0x1782bd, 0x0bf71c, 0xe95ef6, 0xf4c899, 0x000003, + 0x0bf71c, 0x17cd97, 0x0bf71c, 0xf1608a, 0xfa3b18, 0x000003, + 0x000000, 0x2fdc6f, 0x2fdc6f, 0xf15663, 0x000000, 0x000001 +}; + +static const unsigned int src_iir_coeff_96_to_16[] = { + 0x0805a1, 0xf21ae3, 0x0805a1, 0x3840bb, 0xe02a2e, 0x000002, + 0x0d5dd8, 0xe8f259, 0x0d5dd8, 0x1c0af6, 0xf04700, 0x000003, + 0x0bb422, 0xec08d9, 0x0bb422, 0x1bfccc, 0xf09216, 0x000003, + 0x08fde6, 0xf108be, 0x08fde6, 0x1bf096, 0xf10ae0, 0x000003, + 0x0ae311, 0xeeeda3, 0x0ae311, 0x37c646, 0xe385f5, 0x000002, + 0x044089, 0xfa7242, 0x044089, 0x37a785, 0xe56526, 0x000002, + 0x00c75c, 0xffb947, 0x00c75c, 0x378ba3, 0xe72c5f, 0x000002, + 0x000000, 0x0ef76e, 0x0ef76e, 0x377fda, 0x000000, 0x000001, +}; + +static const unsigned int src_iir_coeff_96_to_44[] = { + 0x08b543, 0xfd80f4, 0x08b543, 0x0e2332, 0xe06ed0, 0x000002, + 0x1b6038, 0xf90e7e, 0x1b6038, 0x0ec1ac, 0xe16f66, 0x000002, + 0x188478, 0xfbb921, 0x188478, 0x105859, 0xe2e596, 0x000002, + 0x13eff3, 0xffa707, 0x13eff3, 0x13455c, 0xe533b7, 0x000002, + 0x0dc239, 0x03d458, 0x0dc239, 0x17f120, 0xe8b617, 0x000002, + 0x0745f1, 0x05d790, 0x0745f1, 0x1e3d75, 0xed5f18, 0x000002, + 0x05641f, 0x085e2b, 0x05641f, 0x48efd0, 0xe3e9c8, 0x000001, + 0x000000, 0x28f632, 0x28f632, 0x273905, 0x000000, 0x000001, +}; + +static unsigned int mtk_get_src_freq_mode(struct mtk_base_afe *afe, int rate) +{ + switch (rate) { + case 8000: + return 0x50000; + case 11025: + return 0x6e400; + case 12000: + return 0x78000; + case 16000: + return 0xa0000; + case 22050: + return 0xdc800; + case 24000: + return 0xf0000; + case 32000: + return 0x140000; + case 44100: + return 0x1b9000; + case 48000: + return 0x1e0000; + case 88200: + return 0x372000; + case 96000: + return 0x3c0000; + case 176400: + return 0x6e4000; + case 192000: + return 0x780000; + default: + dev_err(afe->dev, "%s(), rate %d invalid!!!\n", + __func__, rate); + return 0; + } +} + +static const unsigned int *get_iir_coeff(unsigned int rate_in, + unsigned int rate_out, + unsigned int *param_num) +{ + if (rate_in == 32000 && rate_out == 16000) { + *param_num = ARRAY_SIZE(src_iir_coeff_32_to_16); + return src_iir_coeff_32_to_16; + } else if (rate_in == 44100 && rate_out == 16000) { + *param_num = ARRAY_SIZE(src_iir_coeff_44_to_16); + return src_iir_coeff_44_to_16; + } else if (rate_in == 44100 && rate_out == 32000) { + *param_num = ARRAY_SIZE(src_iir_coeff_44_to_32); + return src_iir_coeff_44_to_32; + } else if ((rate_in == 48000 && rate_out == 16000) || + (rate_in == 96000 && rate_out == 32000)) { + *param_num = ARRAY_SIZE(src_iir_coeff_48_to_16); + return src_iir_coeff_48_to_16; + } else if (rate_in == 48000 && rate_out == 32000) { + *param_num = ARRAY_SIZE(src_iir_coeff_48_to_32); + return src_iir_coeff_48_to_32; + } else if (rate_in == 48000 && rate_out == 44100) { + *param_num = ARRAY_SIZE(src_iir_coeff_48_to_44); + return src_iir_coeff_48_to_44; + } else if (rate_in == 96000 && rate_out == 16000) { + *param_num = ARRAY_SIZE(src_iir_coeff_96_to_16); + return src_iir_coeff_96_to_16; + } else if ((rate_in == 96000 && rate_out == 44100) || + (rate_in == 48000 && rate_out == 22050)) { + *param_num = ARRAY_SIZE(src_iir_coeff_96_to_44); + return src_iir_coeff_96_to_44; + } + + *param_num = 0; + return NULL; +} + +static int mtk_set_src_1_param(struct mtk_base_afe *afe, int id) +{ + struct mt8186_afe_private *afe_priv = afe->platform_priv; + struct mtk_afe_src_priv *src_priv = afe_priv->dai_priv[id]; + unsigned int iir_coeff_num; + unsigned int iir_stage; + int rate_in = src_priv->dl_rate; + int rate_out = src_priv->ul_rate; + unsigned int out_freq_mode = mtk_get_src_freq_mode(afe, rate_out); + unsigned int in_freq_mode = mtk_get_src_freq_mode(afe, rate_in); + + /* set out freq mode */ + regmap_update_bits(afe->regmap, AFE_GENERAL1_ASRC_2CH_CON3, + G_SRC_ASM_FREQ_4_MASK_SFT, + out_freq_mode << G_SRC_ASM_FREQ_4_SFT); + + /* set in freq mode */ + regmap_update_bits(afe->regmap, AFE_GENERAL1_ASRC_2CH_CON4, + G_SRC_ASM_FREQ_5_MASK_SFT, + in_freq_mode << G_SRC_ASM_FREQ_5_SFT); + + regmap_write(afe->regmap, AFE_GENERAL1_ASRC_2CH_CON5, 0x3f5986); + regmap_write(afe->regmap, AFE_GENERAL1_ASRC_2CH_CON5, 0x3f5987); + regmap_write(afe->regmap, AFE_GENERAL1_ASRC_2CH_CON6, 0x1fbd); + regmap_write(afe->regmap, AFE_GENERAL1_ASRC_2CH_CON2, 0); + + /* set iir if in_rate > out_rate */ + if (rate_in > rate_out) { + int i; + const unsigned int *iir_coeff = get_iir_coeff(rate_in, rate_out, + &iir_coeff_num); + + if (iir_coeff_num == 0 || !iir_coeff) { + dev_err(afe->dev, "%s(), iir coeff error, num %d, coeff %p\n", + __func__, iir_coeff_num, iir_coeff); + return -EINVAL; + } + + /* COEFF_SRAM_CTRL */ + regmap_update_bits(afe->regmap, AFE_GENERAL1_ASRC_2CH_CON0, + G_SRC_COEFF_SRAM_CTRL_MASK_SFT, + BIT(G_SRC_COEFF_SRAM_CTRL_SFT)); + /* Clear coeff history to r/w coeff from the first position */ + regmap_update_bits(afe->regmap, AFE_GENERAL1_ASRC_2CH_CON13, + G_SRC_COEFF_SRAM_ADR_MASK_SFT, 0); + /* Write SRC coeff, should not read the reg during write */ + for (i = 0; i < iir_coeff_num; i++) + regmap_write(afe->regmap, AFE_GENERAL1_ASRC_2CH_CON12, + iir_coeff[i]); + /* disable sram access */ + regmap_update_bits(afe->regmap, AFE_GENERAL1_ASRC_2CH_CON0, + G_SRC_COEFF_SRAM_CTRL_MASK_SFT, 0); + /* CHSET_IIR_STAGE */ + iir_stage = (iir_coeff_num / 6) - 1; + regmap_update_bits(afe->regmap, AFE_GENERAL1_ASRC_2CH_CON2, + G_SRC_CHSET_IIR_STAGE_MASK_SFT, + iir_stage << G_SRC_CHSET_IIR_STAGE_SFT); + /* CHSET_IIR_EN */ + regmap_update_bits(afe->regmap, AFE_GENERAL1_ASRC_2CH_CON2, + G_SRC_CHSET_IIR_EN_MASK_SFT, + BIT(G_SRC_CHSET_IIR_EN_SFT)); + } else { + /* CHSET_IIR_EN off */ + regmap_update_bits(afe->regmap, AFE_GENERAL1_ASRC_2CH_CON2, + G_SRC_CHSET_IIR_EN_MASK_SFT, 0); + } + + return 0; +} + +static int mtk_set_src_2_param(struct mtk_base_afe *afe, int id) +{ + struct mt8186_afe_private *afe_priv = afe->platform_priv; + struct mtk_afe_src_priv *src_priv = afe_priv->dai_priv[id]; + unsigned int iir_coeff_num; + unsigned int iir_stage; + int rate_in = src_priv->dl_rate; + int rate_out = src_priv->ul_rate; + unsigned int out_freq_mode = mtk_get_src_freq_mode(afe, rate_out); + unsigned int in_freq_mode = mtk_get_src_freq_mode(afe, rate_in); + + /* set out freq mode */ + regmap_update_bits(afe->regmap, AFE_GENERAL2_ASRC_2CH_CON3, + G_SRC_ASM_FREQ_4_MASK_SFT, + out_freq_mode << G_SRC_ASM_FREQ_4_SFT); + + /* set in freq mode */ + regmap_update_bits(afe->regmap, AFE_GENERAL2_ASRC_2CH_CON4, + G_SRC_ASM_FREQ_5_MASK_SFT, + in_freq_mode << G_SRC_ASM_FREQ_5_SFT); + + regmap_write(afe->regmap, AFE_GENERAL2_ASRC_2CH_CON5, 0x3f5986); + regmap_write(afe->regmap, AFE_GENERAL2_ASRC_2CH_CON5, 0x3f5987); + regmap_write(afe->regmap, AFE_GENERAL2_ASRC_2CH_CON6, 0x1fbd); + regmap_write(afe->regmap, AFE_GENERAL2_ASRC_2CH_CON2, 0); + + /* set iir if in_rate > out_rate */ + if (rate_in > rate_out) { + int i; + const unsigned int *iir_coeff = get_iir_coeff(rate_in, rate_out, + &iir_coeff_num); + + if (iir_coeff_num == 0 || !iir_coeff) { + dev_err(afe->dev, "%s(), iir coeff error, num %d, coeff %p\n", + __func__, iir_coeff_num, iir_coeff); + return -EINVAL; + } + + /* COEFF_SRAM_CTRL */ + regmap_update_bits(afe->regmap, AFE_GENERAL2_ASRC_2CH_CON0, + G_SRC_COEFF_SRAM_CTRL_MASK_SFT, + BIT(G_SRC_COEFF_SRAM_CTRL_SFT)); + /* Clear coeff history to r/w coeff from the first position */ + regmap_update_bits(afe->regmap, AFE_GENERAL2_ASRC_2CH_CON13, + G_SRC_COEFF_SRAM_ADR_MASK_SFT, 0); + /* Write SRC coeff, should not read the reg during write */ + for (i = 0; i < iir_coeff_num; i++) + regmap_write(afe->regmap, AFE_GENERAL2_ASRC_2CH_CON12, + iir_coeff[i]); + /* disable sram access */ + regmap_update_bits(afe->regmap, AFE_GENERAL2_ASRC_2CH_CON0, + G_SRC_COEFF_SRAM_CTRL_MASK_SFT, 0); + /* CHSET_IIR_STAGE */ + iir_stage = (iir_coeff_num / 6) - 1; + regmap_update_bits(afe->regmap, AFE_GENERAL2_ASRC_2CH_CON2, + G_SRC_CHSET_IIR_STAGE_MASK_SFT, + iir_stage << G_SRC_CHSET_IIR_STAGE_SFT); + /* CHSET_IIR_EN */ + regmap_update_bits(afe->regmap, AFE_GENERAL2_ASRC_2CH_CON2, + G_SRC_CHSET_IIR_EN_MASK_SFT, + BIT(G_SRC_CHSET_IIR_EN_SFT)); + } else { + /* CHSET_IIR_EN off */ + regmap_update_bits(afe->regmap, AFE_GENERAL2_ASRC_2CH_CON2, + G_SRC_CHSET_IIR_EN_MASK_SFT, 0); + } + + return 0; +} + +#define HW_SRC_1_EN_W_NAME "HW_SRC_1_Enable" +#define HW_SRC_2_EN_W_NAME "HW_SRC_2_Enable" + +static int mtk_hw_src_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_component *cmpnt = snd_soc_dapm_to_component(w->dapm); + struct mtk_base_afe *afe = snd_soc_component_get_drvdata(cmpnt); + struct mt8186_afe_private *afe_priv = afe->platform_priv; + int id; + struct mtk_afe_src_priv *src_priv; + unsigned int reg; + + if (strcmp(w->name, HW_SRC_1_EN_W_NAME) == 0) + id = MT8186_DAI_SRC_1; + else + id = MT8186_DAI_SRC_2; + + src_priv = afe_priv->dai_priv[id]; + + dev_dbg(afe->dev, + "%s(), name %s, event 0x%x, id %d, src_priv %p, dl_rate %d, ul_rate %d\n", + __func__, w->name, event, id, src_priv, + src_priv->dl_rate, src_priv->ul_rate); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (id == MT8186_DAI_SRC_1) + mtk_set_src_1_param(afe, id); + else + mtk_set_src_2_param(afe, id); + break; + case SND_SOC_DAPM_POST_PMU: + reg = (id == MT8186_DAI_SRC_1) ? + AFE_GENERAL1_ASRC_2CH_CON0 : AFE_GENERAL2_ASRC_2CH_CON0; + /* ASM_ON */ + regmap_update_bits(afe->regmap, reg, + G_SRC_ASM_ON_MASK_SFT, + BIT(G_SRC_ASM_ON_SFT)); + /* CHSET_ON */ + regmap_update_bits(afe->regmap, reg, + G_SRC_CHSET_ON_MASK_SFT, + BIT(G_SRC_CHSET_ON_SFT)); + /* CHSET_STR_CLR */ + regmap_update_bits(afe->regmap, reg, + G_SRC_CHSET_STR_CLR_MASK_SFT, + BIT(G_SRC_CHSET_STR_CLR_SFT)); + break; + case SND_SOC_DAPM_PRE_PMD: + reg = (id == MT8186_DAI_SRC_1) ? + AFE_GENERAL1_ASRC_2CH_CON0 : AFE_GENERAL2_ASRC_2CH_CON0; + /* ASM_OFF */ + regmap_update_bits(afe->regmap, reg, G_SRC_ASM_ON_MASK_SFT, 0); + /* CHSET_OFF */ + regmap_update_bits(afe->regmap, reg, G_SRC_CHSET_ON_MASK_SFT, 0); + /* CHSET_STR_CLR */ + regmap_update_bits(afe->regmap, reg, G_SRC_CHSET_STR_CLR_MASK_SFT, 0); + break; + default: + break; + } + + return 0; +} + +/* dai component */ +static const struct snd_kcontrol_new mtk_hw_src_1_in_ch1_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("DL1_CH1 Switch", AFE_CONN40, + I_DL1_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL2_CH1 Switch", AFE_CONN40, + I_DL2_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL3_CH1 Switch", AFE_CONN40, + I_DL3_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL4_CH1 Switch", AFE_CONN40_1, + I_DL4_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL6_CH1 Switch", AFE_CONN40_1, + I_DL6_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I2S0_CH1 Switch", AFE_CONN40, + I_I2S0_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL5_CH1 Switch", AFE_CONN40_1, + I_DL5_CH1, 1, 0), +}; + +static const struct snd_kcontrol_new mtk_hw_src_1_in_ch2_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("DL1_CH2 Switch", AFE_CONN41, + I_DL1_CH2, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL2_CH2 Switch", AFE_CONN41, + I_DL2_CH2, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL3_CH2 Switch", AFE_CONN41, + I_DL3_CH2, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL4_CH2 Switch", AFE_CONN41_1, + I_DL4_CH2, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL6_CH2 Switch", AFE_CONN41_1, + I_DL6_CH2, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I2S0_CH2 Switch", AFE_CONN41, + I_I2S0_CH2, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL5_CH2 Switch", AFE_CONN41_1, + I_DL5_CH2, 1, 0), +}; + +static const struct snd_kcontrol_new mtk_hw_src_2_in_ch1_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("DL1_CH1 Switch", AFE_CONN42, + I_DL1_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL2_CH1 Switch", AFE_CONN42, + I_DL2_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL3_CH1 Switch", AFE_CONN42, + I_DL3_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL4_CH1 Switch", AFE_CONN42, + I_DL4_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL5_CH1 Switch", AFE_CONN42_1, + I_DL5_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL6_CH1 Switch", AFE_CONN42_1, + I_DL6_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("HW_GAIN2_OUT_CH1 Switch", AFE_CONN42, + I_GAIN2_OUT_CH1, 1, 0), +}; + +static const struct snd_kcontrol_new mtk_hw_src_2_in_ch2_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("DL1_CH2 Switch", AFE_CONN43, + I_DL1_CH2, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL2_CH2 Switch", AFE_CONN43, + I_DL2_CH2, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL3_CH2 Switch", AFE_CONN43, + I_DL3_CH2, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL4_CH2 Switch", AFE_CONN43, + I_DL4_CH2, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL5_CH2 Switch", AFE_CONN43_1, + I_DL5_CH2, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("DL6_CH2 Switch", AFE_CONN43_1, + I_DL6_CH2, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("HW_GAIN2_OUT_CH2 Switch", AFE_CONN43, + I_GAIN2_OUT_CH2, 1, 0), +}; + +static const struct snd_soc_dapm_widget mtk_dai_src_widgets[] = { + /* inter-connections */ + SND_SOC_DAPM_MIXER("HW_SRC_1_IN_CH1", SND_SOC_NOPM, 0, 0, + mtk_hw_src_1_in_ch1_mix, + ARRAY_SIZE(mtk_hw_src_1_in_ch1_mix)), + SND_SOC_DAPM_MIXER("HW_SRC_1_IN_CH2", SND_SOC_NOPM, 0, 0, + mtk_hw_src_1_in_ch2_mix, + ARRAY_SIZE(mtk_hw_src_1_in_ch2_mix)), + SND_SOC_DAPM_MIXER("HW_SRC_2_IN_CH1", SND_SOC_NOPM, 0, 0, + mtk_hw_src_2_in_ch1_mix, + ARRAY_SIZE(mtk_hw_src_2_in_ch1_mix)), + SND_SOC_DAPM_MIXER("HW_SRC_2_IN_CH2", SND_SOC_NOPM, 0, 0, + mtk_hw_src_2_in_ch2_mix, + ARRAY_SIZE(mtk_hw_src_2_in_ch2_mix)), + + SND_SOC_DAPM_SUPPLY(HW_SRC_1_EN_W_NAME, + GENERAL_ASRC_EN_ON, GENERAL1_ASRC_EN_ON_SFT, 0, + mtk_hw_src_event, + SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD), + + SND_SOC_DAPM_SUPPLY(HW_SRC_2_EN_W_NAME, + GENERAL_ASRC_EN_ON, GENERAL2_ASRC_EN_ON_SFT, 0, + mtk_hw_src_event, + SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD), + + SND_SOC_DAPM_INPUT("HW SRC 1 Out Endpoint"), + SND_SOC_DAPM_INPUT("HW SRC 2 Out Endpoint"), + SND_SOC_DAPM_OUTPUT("HW SRC 1 In Endpoint"), + SND_SOC_DAPM_OUTPUT("HW SRC 2 In Endpoint"), +}; + +static int mtk_afe_src_en_connect(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_dapm_widget *w = source; + struct snd_soc_component *cmpnt = snd_soc_dapm_to_component(w->dapm); + struct mtk_base_afe *afe = snd_soc_component_get_drvdata(cmpnt); + struct mt8186_afe_private *afe_priv = afe->platform_priv; + struct mtk_afe_src_priv *src_priv; + + if (strcmp(w->name, HW_SRC_1_EN_W_NAME) == 0) + src_priv = afe_priv->dai_priv[MT8186_DAI_SRC_1]; + else + src_priv = afe_priv->dai_priv[MT8186_DAI_SRC_2]; + + dev_dbg(afe->dev, + "%s(), source %s, sink %s, dl_rate %d, ul_rate %d\n", + __func__, source->name, sink->name, + src_priv->dl_rate, src_priv->ul_rate); + + return (src_priv->dl_rate > 0 && src_priv->ul_rate > 0) ? 1 : 0; +} + +static const struct snd_soc_dapm_route mtk_dai_src_routes[] = { + {"HW_SRC_1_IN_CH1", "DL1_CH1 Switch", "DL1"}, + {"HW_SRC_1_IN_CH2", "DL1_CH2 Switch", "DL1"}, + {"HW_SRC_2_IN_CH1", "DL1_CH1 Switch", "DL1"}, + {"HW_SRC_2_IN_CH2", "DL1_CH2 Switch", "DL1"}, + {"HW_SRC_1_IN_CH1", "DL2_CH1 Switch", "DL2"}, + {"HW_SRC_1_IN_CH2", "DL2_CH2 Switch", "DL2"}, + {"HW_SRC_2_IN_CH1", "DL2_CH1 Switch", "DL2"}, + {"HW_SRC_2_IN_CH2", "DL2_CH2 Switch", "DL2"}, + {"HW_SRC_1_IN_CH1", "DL3_CH1 Switch", "DL3"}, + {"HW_SRC_1_IN_CH2", "DL3_CH2 Switch", "DL3"}, + {"HW_SRC_2_IN_CH1", "DL3_CH1 Switch", "DL3"}, + {"HW_SRC_2_IN_CH2", "DL3_CH2 Switch", "DL3"}, + {"HW_SRC_1_IN_CH1", "DL6_CH1 Switch", "DL6"}, + {"HW_SRC_1_IN_CH2", "DL6_CH2 Switch", "DL6"}, + {"HW_SRC_2_IN_CH1", "DL6_CH1 Switch", "DL6"}, + {"HW_SRC_2_IN_CH2", "DL6_CH2 Switch", "DL6"}, + {"HW_SRC_1_IN_CH1", "DL5_CH1 Switch", "DL5"}, + {"HW_SRC_1_IN_CH2", "DL5_CH2 Switch", "DL5"}, + {"HW_SRC_2_IN_CH1", "DL5_CH1 Switch", "DL5"}, + {"HW_SRC_2_IN_CH2", "DL5_CH2 Switch", "DL5"}, + {"HW_SRC_1_IN_CH1", "DL4_CH1 Switch", "DL4"}, + {"HW_SRC_1_IN_CH2", "DL4_CH2 Switch", "DL4"}, + {"HW_SRC_2_IN_CH1", "DL4_CH1 Switch", "DL4"}, + {"HW_SRC_2_IN_CH2", "DL4_CH2 Switch", "DL4"}, + + {"HW_SRC_1_In", NULL, "HW_SRC_1_IN_CH1"}, + {"HW_SRC_1_In", NULL, "HW_SRC_1_IN_CH2"}, + + {"HW_SRC_2_In", NULL, "HW_SRC_2_IN_CH1"}, + {"HW_SRC_2_In", NULL, "HW_SRC_2_IN_CH2"}, + + {"HW_SRC_1_In", NULL, HW_SRC_1_EN_W_NAME, mtk_afe_src_en_connect}, + {"HW_SRC_1_Out", NULL, HW_SRC_1_EN_W_NAME, mtk_afe_src_en_connect}, + {"HW_SRC_2_In", NULL, HW_SRC_2_EN_W_NAME, mtk_afe_src_en_connect}, + {"HW_SRC_2_Out", NULL, HW_SRC_2_EN_W_NAME, mtk_afe_src_en_connect}, + + {"HW SRC 1 In Endpoint", NULL, "HW_SRC_1_In"}, + {"HW SRC 2 In Endpoint", NULL, "HW_SRC_2_In"}, + {"HW_SRC_1_Out", NULL, "HW SRC 1 Out Endpoint"}, + {"HW_SRC_2_Out", NULL, "HW SRC 2 Out Endpoint"}, +}; + +/* dai ops */ +static int mtk_dai_src_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); + struct mt8186_afe_private *afe_priv = afe->platform_priv; + int id = dai->id; + struct mtk_afe_src_priv *src_priv = afe_priv->dai_priv[id]; + unsigned int sft, mask; + unsigned int rate = params_rate(params); + unsigned int rate_reg = mt8186_rate_transform(afe->dev, rate, id); + + dev_dbg(afe->dev, "%s(), id %d, stream %d, rate %d\n", + __func__, id, substream->stream, rate); + + /* rate */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + src_priv->dl_rate = rate; + if (id == MT8186_DAI_SRC_1) { + sft = GENERAL1_ASRCIN_MODE_SFT; + mask = GENERAL1_ASRCIN_MODE_MASK; + } else { + sft = GENERAL2_ASRCIN_MODE_SFT; + mask = GENERAL2_ASRCIN_MODE_MASK; + } + } else { + src_priv->ul_rate = rate; + if (id == MT8186_DAI_SRC_1) { + sft = GENERAL1_ASRCOUT_MODE_SFT; + mask = GENERAL1_ASRCOUT_MODE_MASK; + } else { + sft = GENERAL2_ASRCOUT_MODE_SFT; + mask = GENERAL2_ASRCOUT_MODE_MASK; + } + } + + regmap_update_bits(afe->regmap, GENERAL_ASRC_MODE, mask << sft, rate_reg << sft); + + return 0; +} + +static int mtk_dai_src_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); + struct mt8186_afe_private *afe_priv = afe->platform_priv; + int id = dai->id; + struct mtk_afe_src_priv *src_priv = afe_priv->dai_priv[id]; + + dev_dbg(afe->dev, "%s(), id %d, stream %d\n", + __func__, id, substream->stream); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + src_priv->dl_rate = 0; + else + src_priv->ul_rate = 0; + + return 0; +} + +static const struct snd_soc_dai_ops mtk_dai_src_ops = { + .hw_params = mtk_dai_src_hw_params, + .hw_free = mtk_dai_src_hw_free, +}; + +/* dai driver */ +#define MTK_SRC_RATES (SNDRV_PCM_RATE_8000_48000 |\ + SNDRV_PCM_RATE_88200 |\ + SNDRV_PCM_RATE_96000 |\ + SNDRV_PCM_RATE_176400 |\ + SNDRV_PCM_RATE_192000) + +#define MTK_SRC_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver mtk_dai_src_driver[] = { + { + .name = "HW_SRC_1", + .id = MT8186_DAI_SRC_1, + .playback = { + .stream_name = "HW_SRC_1_In", + .channels_min = 1, + .channels_max = 2, + .rates = MTK_SRC_RATES, + .formats = MTK_SRC_FORMATS, + }, + .capture = { + .stream_name = "HW_SRC_1_Out", + .channels_min = 1, + .channels_max = 2, + .rates = MTK_SRC_RATES, + .formats = MTK_SRC_FORMATS, + }, + .ops = &mtk_dai_src_ops, + }, + { + .name = "HW_SRC_2", + .id = MT8186_DAI_SRC_2, + .playback = { + .stream_name = "HW_SRC_2_In", + .channels_min = 1, + .channels_max = 2, + .rates = MTK_SRC_RATES, + .formats = MTK_SRC_FORMATS, + }, + .capture = { + .stream_name = "HW_SRC_2_Out", + .channels_min = 1, + .channels_max = 2, + .rates = MTK_SRC_RATES, + .formats = MTK_SRC_FORMATS, + }, + .ops = &mtk_dai_src_ops, + }, +}; + +int mt8186_dai_src_register(struct mtk_base_afe *afe) +{ + struct mtk_base_afe_dai *dai; + int ret; + + dai = devm_kzalloc(afe->dev, sizeof(*dai), GFP_KERNEL); + if (!dai) + return -ENOMEM; + + list_add(&dai->list, &afe->sub_dais); + + dai->dai_drivers = mtk_dai_src_driver; + dai->num_dai_drivers = ARRAY_SIZE(mtk_dai_src_driver); + + dai->dapm_widgets = mtk_dai_src_widgets; + dai->num_dapm_widgets = ARRAY_SIZE(mtk_dai_src_widgets); + dai->dapm_routes = mtk_dai_src_routes; + dai->num_dapm_routes = ARRAY_SIZE(mtk_dai_src_routes); + + /* set dai priv */ + ret = mt8186_dai_set_priv(afe, MT8186_DAI_SRC_1, + sizeof(struct mtk_afe_src_priv), NULL); + if (ret) + return ret; + + ret = mt8186_dai_set_priv(afe, MT8186_DAI_SRC_2, + sizeof(struct mtk_afe_src_priv), NULL); + if (ret) + return ret; + + return 0; +} -- cgit v1.2.3 From ae92dcbee8b6a6f63198a2a6fea0fc9f6a0fe07b Mon Sep 17 00:00:00 2001 From: Jiaxin Yu Date: Mon, 23 May 2022 21:28:48 +0800 Subject: ASoC: mediatek: mt8186: support tdm in platform driver Add mt8186 tdm dai driver. Signed-off-by: Jiaxin Yu Link: https://lore.kernel.org/r/20220523132858.22166-11-jiaxin.yu@mediatek.com Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8186/mt8186-dai-tdm.c | 698 +++++++++++++++++++++++++++++ 1 file changed, 698 insertions(+) create mode 100644 sound/soc/mediatek/mt8186/mt8186-dai-tdm.c (limited to 'sound') diff --git a/sound/soc/mediatek/mt8186/mt8186-dai-tdm.c b/sound/soc/mediatek/mt8186/mt8186-dai-tdm.c new file mode 100644 index 000000000000..dfff209b60da --- /dev/null +++ b/sound/soc/mediatek/mt8186/mt8186-dai-tdm.c @@ -0,0 +1,698 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// MediaTek ALSA SoC Audio DAI TDM Control +// +// Copyright (c) 2022 MediaTek Inc. +// Author: Jiaxin Yu + +#include +#include + +#include "mt8186-afe-clk.h" +#include "mt8186-afe-common.h" +#include "mt8186-afe-gpio.h" +#include "mt8186-interconnection.h" + +#define TDM_HD_EN_W_NAME "TDM_HD_EN" +#define TDM_MCLK_EN_W_NAME "TDM_MCLK_EN" +#define MTK_AFE_TDM_KCONTROL_NAME "TDM_HD_Mux" + +struct mtk_afe_tdm_priv { + unsigned int id; + unsigned int rate; /* for determine which apll to use */ + unsigned int bck_invert; + unsigned int lck_invert; + unsigned int lrck_width; + unsigned int mclk_id; + unsigned int mclk_multiple; /* according to sample rate */ + unsigned int mclk_rate; + unsigned int mclk_apll; + unsigned int tdm_mode; + unsigned int data_mode; + unsigned int slave_mode; + unsigned int low_jitter_en; +}; + +enum { + TDM_IN_I2S = 0, + TDM_IN_LJ = 1, + TDM_IN_RJ = 2, + TDM_IN_DSP_A = 4, + TDM_IN_DSP_B = 5, +}; + +enum { + TDM_DATA_ONE_PIN = 0, + TDM_DATA_MULTI_PIN, +}; + +enum { + TDM_BCK_NON_INV = 0, + TDM_BCK_INV = 1, +}; + +enum { + TDM_LCK_NON_INV = 0, + TDM_LCK_INV = 1, +}; + +static unsigned int get_tdm_lrck_width(snd_pcm_format_t format, + unsigned int mode) +{ + if (mode == TDM_IN_DSP_A || mode == TDM_IN_DSP_B) + return 0; + + return snd_pcm_format_physical_width(format) - 1; +} + +static unsigned int get_tdm_ch_fixup(unsigned int channels) +{ + if (channels > 4) + return 8; + else if (channels > 2) + return 4; + + return 2; +} + +static unsigned int get_tdm_ch_per_sdata(unsigned int mode, + unsigned int channels) +{ + if (mode == TDM_IN_DSP_A || mode == TDM_IN_DSP_B) + return get_tdm_ch_fixup(channels); + + return 2; +} + +enum { + SUPPLY_SEQ_APLL, + SUPPLY_SEQ_TDM_MCK_EN, + SUPPLY_SEQ_TDM_HD_EN, + SUPPLY_SEQ_TDM_EN, +}; + +static int get_tdm_id_by_name(const char *name) +{ + return MT8186_DAI_TDM_IN; +} + +static int mtk_tdm_en_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_component *cmpnt = snd_soc_dapm_to_component(w->dapm); + struct mtk_base_afe *afe = snd_soc_component_get_drvdata(cmpnt); + struct mt8186_afe_private *afe_priv = afe->platform_priv; + int dai_id = get_tdm_id_by_name(w->name); + struct mtk_afe_tdm_priv *tdm_priv = afe_priv->dai_priv[dai_id]; + + if (!tdm_priv) { + dev_err(afe->dev, "%s(), tdm_priv == NULL", __func__); + return -EINVAL; + } + + dev_dbg(cmpnt->dev, "%s(), name %s, event 0x%x\n", + __func__, w->name, event); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + mt8186_afe_gpio_request(afe->dev, true, tdm_priv->id, 0); + break; + case SND_SOC_DAPM_POST_PMD: + mt8186_afe_gpio_request(afe->dev, false, tdm_priv->id, 0); + break; + default: + break; + } + + return 0; +} + +static int mtk_tdm_mck_en_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_component *cmpnt = snd_soc_dapm_to_component(w->dapm); + struct mtk_base_afe *afe = snd_soc_component_get_drvdata(cmpnt); + struct mt8186_afe_private *afe_priv = afe->platform_priv; + int dai_id = get_tdm_id_by_name(w->name); + struct mtk_afe_tdm_priv *tdm_priv = afe_priv->dai_priv[dai_id]; + + if (!tdm_priv) { + dev_err(afe->dev, "%s(), tdm_priv == NULL", __func__); + return -EINVAL; + } + + dev_dbg(cmpnt->dev, "%s(), name %s, event 0x%x, dai_id %d\n", + __func__, w->name, event, dai_id); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + mt8186_mck_enable(afe, tdm_priv->mclk_id, tdm_priv->mclk_rate); + break; + case SND_SOC_DAPM_POST_PMD: + tdm_priv->mclk_rate = 0; + mt8186_mck_disable(afe, tdm_priv->mclk_id); + break; + default: + break; + } + + return 0; +} + +/* dai component */ +/* tdm virtual mux to output widget */ +static const char * const tdm_mux_map[] = { + "Normal", "Dummy_Widget", +}; + +static int tdm_mux_map_value[] = { + 0, 1, +}; + +static SOC_VALUE_ENUM_SINGLE_AUTODISABLE_DECL(tdm_mux_map_enum, + SND_SOC_NOPM, + 0, + 1, + tdm_mux_map, + tdm_mux_map_value); + +static const struct snd_kcontrol_new tdm_in_mux_control = + SOC_DAPM_ENUM("TDM In Select", tdm_mux_map_enum); + +static const struct snd_soc_dapm_widget mtk_dai_tdm_widgets[] = { + SND_SOC_DAPM_CLOCK_SUPPLY("aud_tdm_clk"), + + SND_SOC_DAPM_SUPPLY_S("TDM_EN", SUPPLY_SEQ_TDM_EN, + ETDM_IN1_CON0, ETDM_IN1_CON0_REG_ETDM_IN_EN_SFT, + 0, mtk_tdm_en_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + /* tdm hd en */ + SND_SOC_DAPM_SUPPLY_S(TDM_HD_EN_W_NAME, SUPPLY_SEQ_TDM_HD_EN, + ETDM_IN1_CON2, ETDM_IN1_CON2_REG_CLOCK_SOURCE_SEL_SFT, + 0, NULL, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_SUPPLY_S(TDM_MCLK_EN_W_NAME, SUPPLY_SEQ_TDM_MCK_EN, + SND_SOC_NOPM, 0, 0, + mtk_tdm_mck_en_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_INPUT("TDM_DUMMY_IN"), + + SND_SOC_DAPM_MUX("TDM_In_Mux", + SND_SOC_NOPM, 0, 0, &tdm_in_mux_control), +}; + +static int mtk_afe_tdm_mclk_connect(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_dapm_widget *w = sink; + struct snd_soc_component *cmpnt = snd_soc_dapm_to_component(w->dapm); + struct mtk_base_afe *afe = snd_soc_component_get_drvdata(cmpnt); + struct mt8186_afe_private *afe_priv = afe->platform_priv; + int dai_id = get_tdm_id_by_name(w->name); + struct mtk_afe_tdm_priv *tdm_priv = afe_priv->dai_priv[dai_id]; + + if (!tdm_priv) { + dev_err(afe->dev, "%s(), tdm_priv == NULL", __func__); + return 0; + } + + return (tdm_priv->mclk_rate > 0) ? 1 : 0; +} + +static int mtk_afe_tdm_mclk_apll_connect(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_dapm_widget *w = sink; + struct snd_soc_component *cmpnt = snd_soc_dapm_to_component(w->dapm); + struct mtk_base_afe *afe = snd_soc_component_get_drvdata(cmpnt); + struct mt8186_afe_private *afe_priv = afe->platform_priv; + int dai_id = get_tdm_id_by_name(w->name); + struct mtk_afe_tdm_priv *tdm_priv = afe_priv->dai_priv[dai_id]; + int cur_apll; + + /* which apll */ + cur_apll = mt8186_get_apll_by_name(afe, source->name); + + return (tdm_priv->mclk_apll == cur_apll) ? 1 : 0; +} + +static int mtk_afe_tdm_hd_connect(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_dapm_widget *w = sink; + struct snd_soc_component *cmpnt = snd_soc_dapm_to_component(w->dapm); + struct mtk_base_afe *afe = snd_soc_component_get_drvdata(cmpnt); + struct mt8186_afe_private *afe_priv = afe->platform_priv; + int dai_id = get_tdm_id_by_name(w->name); + struct mtk_afe_tdm_priv *tdm_priv = afe_priv->dai_priv[dai_id]; + + if (!tdm_priv) { + dev_err(afe->dev, "%s(), tdm_priv == NULL", __func__); + return 0; + } + + return tdm_priv->low_jitter_en; +} + +static int mtk_afe_tdm_apll_connect(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_dapm_widget *w = sink; + struct snd_soc_component *cmpnt = snd_soc_dapm_to_component(w->dapm); + struct mtk_base_afe *afe = snd_soc_component_get_drvdata(cmpnt); + struct mt8186_afe_private *afe_priv = afe->platform_priv; + int dai_id = get_tdm_id_by_name(w->name); + struct mtk_afe_tdm_priv *tdm_priv = afe_priv->dai_priv[dai_id]; + int cur_apll; + int tdm_need_apll; + + if (!tdm_priv) { + dev_err(afe->dev, "%s(), tdm_priv == NULL", __func__); + return 0; + } + + /* which apll */ + cur_apll = mt8186_get_apll_by_name(afe, source->name); + + /* choose APLL from tdm rate */ + tdm_need_apll = mt8186_get_apll_by_rate(afe, tdm_priv->rate); + + return (tdm_need_apll == cur_apll) ? 1 : 0; +} + +/* low jitter control */ +static const char * const mt8186_tdm_hd_str[] = { + "Normal", "Low_Jitter" +}; + +static const struct soc_enum mt8186_tdm_enum[] = { + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(mt8186_tdm_hd_str), + mt8186_tdm_hd_str), +}; + +static int mt8186_tdm_hd_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct mtk_base_afe *afe = snd_soc_component_get_drvdata(cmpnt); + struct mt8186_afe_private *afe_priv = afe->platform_priv; + int dai_id = get_tdm_id_by_name(kcontrol->id.name); + struct mtk_afe_tdm_priv *tdm_priv = afe_priv->dai_priv[dai_id]; + + if (!tdm_priv) { + dev_err(afe->dev, "%s(), tdm_priv == NULL", __func__); + return -EINVAL; + } + + ucontrol->value.integer.value[0] = tdm_priv->low_jitter_en; + + return 0; +} + +static int mt8186_tdm_hd_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct mtk_base_afe *afe = snd_soc_component_get_drvdata(cmpnt); + struct mt8186_afe_private *afe_priv = afe->platform_priv; + int dai_id = get_tdm_id_by_name(kcontrol->id.name); + struct mtk_afe_tdm_priv *tdm_priv = afe_priv->dai_priv[dai_id]; + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + int hd_en; + + if (ucontrol->value.enumerated.item[0] >= e->items) + return -EINVAL; + + hd_en = ucontrol->value.integer.value[0]; + + dev_dbg(afe->dev, "%s(), kcontrol name %s, hd_en %d\n", + __func__, kcontrol->id.name, hd_en); + + if (!tdm_priv) { + dev_err(afe->dev, "%s(), tdm_priv == NULL", __func__); + return -EINVAL; + } + + if (tdm_priv->low_jitter_en == hd_en) + return 0; + + tdm_priv->low_jitter_en = hd_en; + + return 1; +} + +static const struct snd_kcontrol_new mtk_dai_tdm_controls[] = { + SOC_ENUM_EXT(MTK_AFE_TDM_KCONTROL_NAME, mt8186_tdm_enum[0], + mt8186_tdm_hd_get, mt8186_tdm_hd_set), +}; + +static const struct snd_soc_dapm_route mtk_dai_tdm_routes[] = { + {"TDM IN", NULL, "aud_tdm_clk"}, + {"TDM IN", NULL, "TDM_EN"}, + {"TDM IN", NULL, TDM_HD_EN_W_NAME, mtk_afe_tdm_hd_connect}, + {TDM_HD_EN_W_NAME, NULL, APLL1_W_NAME, mtk_afe_tdm_apll_connect}, + {TDM_HD_EN_W_NAME, NULL, APLL2_W_NAME, mtk_afe_tdm_apll_connect}, + + {"TDM IN", NULL, TDM_MCLK_EN_W_NAME, mtk_afe_tdm_mclk_connect}, + {TDM_MCLK_EN_W_NAME, NULL, APLL1_W_NAME, mtk_afe_tdm_mclk_apll_connect}, + {TDM_MCLK_EN_W_NAME, NULL, APLL2_W_NAME, mtk_afe_tdm_mclk_apll_connect}, + + /* allow tdm on without codec on */ + {"TDM IN", NULL, "TDM_In_Mux"}, + {"TDM_In_Mux", "Dummy_Widget", "TDM_DUMMY_IN"}, +}; + +/* dai ops */ +static int mtk_dai_tdm_cal_mclk(struct mtk_base_afe *afe, + struct mtk_afe_tdm_priv *tdm_priv, + int freq) +{ + int apll; + int apll_rate; + + apll = mt8186_get_apll_by_rate(afe, freq); + apll_rate = mt8186_get_apll_rate(afe, apll); + + if (!freq || freq > apll_rate) { + dev_err(afe->dev, + "%s(), freq(%d Hz) invalid\n", __func__, freq); + return -EINVAL; + } + + if (apll_rate % freq != 0) { + dev_err(afe->dev, + "%s(), APLL cannot generate %d Hz", __func__, freq); + return -EINVAL; + } + + tdm_priv->mclk_rate = freq; + tdm_priv->mclk_apll = apll; + + return 0; +} + +static int mtk_dai_tdm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); + struct mt8186_afe_private *afe_priv = afe->platform_priv; + int tdm_id = dai->id; + struct mtk_afe_tdm_priv *tdm_priv = afe_priv->dai_priv[tdm_id]; + unsigned int tdm_mode = tdm_priv->tdm_mode; + unsigned int data_mode = tdm_priv->data_mode; + unsigned int rate = params_rate(params); + unsigned int channels = params_channels(params); + snd_pcm_format_t format = params_format(params); + unsigned int bit_width = + snd_pcm_format_physical_width(format); + unsigned int tdm_channels = (data_mode == TDM_DATA_ONE_PIN) ? + get_tdm_ch_per_sdata(tdm_mode, channels) : 2; + unsigned int lrck_width = + get_tdm_lrck_width(format, tdm_mode); + unsigned int tdm_con = 0; + bool slave_mode = tdm_priv->slave_mode; + bool lrck_inv = tdm_priv->lck_invert; + bool bck_inv = tdm_priv->bck_invert; + unsigned int tran_rate; + unsigned int tran_relatch_rate; + + if (!tdm_priv) { + dev_err(afe->dev, "%s(), tdm_priv == NULL", __func__); + return -EINVAL; + } + + tdm_priv->rate = rate; + + tran_rate = mt8186_rate_transform(afe->dev, rate, dai->id); + tran_relatch_rate = mt8186_tdm_relatch_rate_transform(afe->dev, rate); + + /* calculate mclk_rate, if not set explicitly */ + if (!tdm_priv->mclk_rate) { + tdm_priv->mclk_rate = rate * tdm_priv->mclk_multiple; + mtk_dai_tdm_cal_mclk(afe, + tdm_priv, + tdm_priv->mclk_rate); + } + + /* ETDM_IN1_CON0 */ + tdm_con |= slave_mode << ETDM_IN1_CON0_REG_SLAVE_MODE_SFT; + tdm_con |= tdm_mode << ETDM_IN1_CON0_REG_FMT_SFT; + tdm_con |= (bit_width - 1) << ETDM_IN1_CON0_REG_BIT_LENGTH_SFT; + tdm_con |= (bit_width - 1) << ETDM_IN1_CON0_REG_WORD_LENGTH_SFT; + tdm_con |= (tdm_channels - 1) << ETDM_IN1_CON0_REG_CH_NUM_SFT; + /* need to disable sync mode otherwise this may cause latch data error */ + tdm_con |= 0 << ETDM_IN1_CON0_REG_SYNC_MODE_SFT; + /* relatch 1x en clock fix to h26m */ + tdm_con |= 0 << ETDM_IN1_CON0_REG_RELATCH_1X_EN_SEL_DOMAIN_SFT; + regmap_update_bits(afe->regmap, ETDM_IN1_CON0, ETDM_IN_CON0_CTRL_MASK, tdm_con); + + /* ETDM_IN1_CON1 */ + tdm_con = 0; + tdm_con |= 0 << ETDM_IN1_CON1_REG_LRCK_AUTO_MODE_SFT; + tdm_con |= 1 << ETDM_IN1_CON1_PINMUX_MCLK_CTRL_OE_SFT; + tdm_con |= (lrck_width - 1) << ETDM_IN1_CON1_REG_LRCK_WIDTH_SFT; + regmap_update_bits(afe->regmap, ETDM_IN1_CON1, ETDM_IN_CON1_CTRL_MASK, tdm_con); + + /* ETDM_IN1_CON3 */ + tdm_con = 0; + tdm_con = ETDM_IN_CON3_FS(tran_rate); + regmap_update_bits(afe->regmap, ETDM_IN1_CON3, ETDM_IN_CON3_CTRL_MASK, tdm_con); + + /* ETDM_IN1_CON4 */ + tdm_con = 0; + tdm_con = ETDM_IN_CON4_FS(tran_relatch_rate); + if (slave_mode) { + if (lrck_inv) + tdm_con |= ETDM_IN_CON4_CON0_SLAVE_LRCK_INV; + if (bck_inv) + tdm_con |= ETDM_IN_CON4_CON0_SLAVE_BCK_INV; + } else { + if (lrck_inv) + tdm_con |= ETDM_IN_CON4_CON0_MASTER_LRCK_INV; + if (bck_inv) + tdm_con |= ETDM_IN_CON4_CON0_MASTER_BCK_INV; + } + regmap_update_bits(afe->regmap, ETDM_IN1_CON4, ETDM_IN_CON4_CTRL_MASK, tdm_con); + + /* ETDM_IN1_CON2 */ + tdm_con = 0; + if (data_mode == TDM_DATA_MULTI_PIN) { + tdm_con |= ETDM_IN_CON2_MULTI_IP_2CH_MODE; + tdm_con |= ETDM_IN_CON2_MULTI_IP_CH(channels); + } + regmap_update_bits(afe->regmap, ETDM_IN1_CON2, ETDM_IN_CON2_CTRL_MASK, tdm_con); + + /* ETDM_IN1_CON8 */ + tdm_con = 0; + if (slave_mode) { + tdm_con |= 1 << ETDM_IN1_CON8_REG_ETDM_USE_AFIFO_SFT; + tdm_con |= 0 << ETDM_IN1_CON8_REG_AFIFO_CLOCK_DOMAIN_SEL_SFT; + tdm_con |= ETDM_IN_CON8_FS(tran_relatch_rate); + } else { + tdm_con |= 0 << ETDM_IN1_CON8_REG_ETDM_USE_AFIFO_SFT; + } + regmap_update_bits(afe->regmap, ETDM_IN1_CON8, ETDM_IN_CON8_CTRL_MASK, tdm_con); + + return 0; +} + +static int mtk_dai_tdm_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct mtk_base_afe *afe = dev_get_drvdata(dai->dev); + struct mt8186_afe_private *afe_priv = afe->platform_priv; + struct mtk_afe_tdm_priv *tdm_priv = afe_priv->dai_priv[dai->id]; + + if (!tdm_priv) { + dev_err(afe->dev, "%s(), tdm_priv == NULL", __func__); + return -EINVAL; + } + + if (dir != SND_SOC_CLOCK_IN) { + dev_err(afe->dev, "%s(), dir != SND_SOC_CLOCK_OUT", __func__); + return -EINVAL; + } + + dev_dbg(afe->dev, "%s(), freq %d\n", __func__, freq); + + return mtk_dai_tdm_cal_mclk(afe, tdm_priv, freq); +} + +static int mtk_dai_tdm_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct mtk_base_afe *afe = dev_get_drvdata(dai->dev); + struct mt8186_afe_private *afe_priv = afe->platform_priv; + struct mtk_afe_tdm_priv *tdm_priv = afe_priv->dai_priv[dai->id]; + + if (!tdm_priv) { + dev_err(afe->dev, "%s(), tdm_priv == NULL", __func__); + return -EINVAL; + } + + /* DAI mode*/ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + tdm_priv->tdm_mode = TDM_IN_I2S; + tdm_priv->data_mode = TDM_DATA_MULTI_PIN; + break; + case SND_SOC_DAIFMT_LEFT_J: + tdm_priv->tdm_mode = TDM_IN_LJ; + tdm_priv->data_mode = TDM_DATA_MULTI_PIN; + break; + case SND_SOC_DAIFMT_RIGHT_J: + tdm_priv->tdm_mode = TDM_IN_RJ; + tdm_priv->data_mode = TDM_DATA_MULTI_PIN; + break; + case SND_SOC_DAIFMT_DSP_A: + tdm_priv->tdm_mode = TDM_IN_DSP_A; + tdm_priv->data_mode = TDM_DATA_ONE_PIN; + break; + case SND_SOC_DAIFMT_DSP_B: + tdm_priv->tdm_mode = TDM_IN_DSP_B; + tdm_priv->data_mode = TDM_DATA_ONE_PIN; + break; + default: + dev_err(afe->dev, "%s(), invalid DAIFMT_FORMAT_MASK", __func__); + return -EINVAL; + } + + /* DAI clock inversion*/ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + tdm_priv->bck_invert = TDM_BCK_NON_INV; + tdm_priv->lck_invert = TDM_LCK_NON_INV; + break; + case SND_SOC_DAIFMT_NB_IF: + tdm_priv->bck_invert = TDM_BCK_NON_INV; + tdm_priv->lck_invert = TDM_LCK_INV; + break; + case SND_SOC_DAIFMT_IB_NF: + tdm_priv->bck_invert = TDM_BCK_INV; + tdm_priv->lck_invert = TDM_LCK_NON_INV; + break; + case SND_SOC_DAIFMT_IB_IF: + tdm_priv->bck_invert = TDM_BCK_INV; + tdm_priv->lck_invert = TDM_LCK_INV; + break; + default: + dev_err(afe->dev, "%s(), invalid DAIFMT_INV_MASK", __func__); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: + tdm_priv->slave_mode = false; + break; + case SND_SOC_DAIFMT_CBC_CFC: + tdm_priv->slave_mode = true; + break; + default: + dev_err(afe->dev, "%s(), invalid DAIFMT_CLOCK_PROVIDER_MASK", + __func__); + return -EINVAL; + } + + return 0; +} + +static int mtk_dai_tdm_set_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, + unsigned int rx_mask, + int slots, + int slot_width) +{ + struct mtk_base_afe *afe = dev_get_drvdata(dai->dev); + struct mt8186_afe_private *afe_priv = afe->platform_priv; + struct mtk_afe_tdm_priv *tdm_priv = afe_priv->dai_priv[dai->id]; + + dev_dbg(dai->dev, "%s %d slot_width %d\n", __func__, dai->id, slot_width); + + tdm_priv->lrck_width = slot_width; + + return 0; +} + +static const struct snd_soc_dai_ops mtk_dai_tdm_ops = { + .hw_params = mtk_dai_tdm_hw_params, + .set_sysclk = mtk_dai_tdm_set_sysclk, + .set_fmt = mtk_dai_tdm_set_fmt, + .set_tdm_slot = mtk_dai_tdm_set_tdm_slot, +}; + +/* dai driver */ +#define MTK_TDM_RATES (SNDRV_PCM_RATE_8000_48000 |\ + SNDRV_PCM_RATE_88200 |\ + SNDRV_PCM_RATE_96000 |\ + SNDRV_PCM_RATE_176400 |\ + SNDRV_PCM_RATE_192000) + +#define MTK_TDM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver mtk_dai_tdm_driver[] = { + { + .name = "TDM IN", + .id = MT8186_DAI_TDM_IN, + .capture = { + .stream_name = "TDM IN", + .channels_min = 2, + .channels_max = 8, + .rates = MTK_TDM_RATES, + .formats = MTK_TDM_FORMATS, + }, + .ops = &mtk_dai_tdm_ops, + }, +}; + +static struct mtk_afe_tdm_priv *init_tdm_priv_data(struct mtk_base_afe *afe) +{ + struct mtk_afe_tdm_priv *tdm_priv; + + tdm_priv = devm_kzalloc(afe->dev, sizeof(struct mtk_afe_tdm_priv), + GFP_KERNEL); + if (!tdm_priv) + return NULL; + + tdm_priv->mclk_multiple = 512; + tdm_priv->mclk_id = MT8186_TDM_MCK; + tdm_priv->id = MT8186_DAI_TDM_IN; + + return tdm_priv; +} + +int mt8186_dai_tdm_register(struct mtk_base_afe *afe) +{ + struct mt8186_afe_private *afe_priv = afe->platform_priv; + struct mtk_afe_tdm_priv *tdm_priv; + struct mtk_base_afe_dai *dai; + + dai = devm_kzalloc(afe->dev, sizeof(*dai), GFP_KERNEL); + if (!dai) + return -ENOMEM; + + list_add(&dai->list, &afe->sub_dais); + + dai->dai_drivers = mtk_dai_tdm_driver; + dai->num_dai_drivers = ARRAY_SIZE(mtk_dai_tdm_driver); + + dai->controls = mtk_dai_tdm_controls; + dai->num_controls = ARRAY_SIZE(mtk_dai_tdm_controls); + dai->dapm_widgets = mtk_dai_tdm_widgets; + dai->num_dapm_widgets = ARRAY_SIZE(mtk_dai_tdm_widgets); + dai->dapm_routes = mtk_dai_tdm_routes; + dai->num_dapm_routes = ARRAY_SIZE(mtk_dai_tdm_routes); + + tdm_priv = init_tdm_priv_data(afe); + if (!tdm_priv) + return -ENOMEM; + + afe_priv->dai_priv[MT8186_DAI_TDM_IN] = tdm_priv; + + return 0; +} -- cgit v1.2.3 From 55b423d5623ccd6785429431c2cf5f3e073b73ba Mon Sep 17 00:00:00 2001 From: Jiaxin Yu Date: Mon, 23 May 2022 21:28:49 +0800 Subject: ASoC: mediatek: mt8186: support audio clock control in platform driver Add audio clock control with CCF interface. Signed-off-by: Jiaxin Yu Link: https://lore.kernel.org/r/20220523132858.22166-12-jiaxin.yu@mediatek.com Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8186/mt8186-afe-clk.c | 651 +++++++++++++++++++++++++++++ sound/soc/mediatek/mt8186/mt8186-afe-clk.h | 106 +++++ 2 files changed, 757 insertions(+) create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-clk.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-clk.h (limited to 'sound') diff --git a/sound/soc/mediatek/mt8186/mt8186-afe-clk.c b/sound/soc/mediatek/mt8186/mt8186-afe-clk.c new file mode 100644 index 000000000000..0275f66ddc18 --- /dev/null +++ b/sound/soc/mediatek/mt8186/mt8186-afe-clk.c @@ -0,0 +1,651 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// mt8186-afe-clk.c -- Mediatek 8186 afe clock ctrl +// +// Copyright (c) 2022 MediaTek Inc. +// Author: Jiaxin Yu + +#include +#include +#include + +#include "mt8186-afe-common.h" +#include "mt8186-afe-clk.h" +#include "mt8186-audsys-clk.h" + +static DEFINE_MUTEX(mutex_request_dram); + +static const char *aud_clks[CLK_NUM] = { + [CLK_AFE] = "aud_afe_clk", + [CLK_DAC] = "aud_dac_clk", + [CLK_DAC_PREDIS] = "aud_dac_predis_clk", + [CLK_ADC] = "aud_adc_clk", + [CLK_TML] = "aud_tml_clk", + [CLK_APLL22M] = "aud_apll22m_clk", + [CLK_APLL24M] = "aud_apll24m_clk", + [CLK_APLL1_TUNER] = "aud_apll_tuner_clk", + [CLK_APLL2_TUNER] = "aud_apll2_tuner_clk", + [CLK_TDM] = "aud_tdm_clk", + [CLK_NLE] = "aud_nle_clk", + [CLK_DAC_HIRES] = "aud_dac_hires_clk", + [CLK_ADC_HIRES] = "aud_adc_hires_clk", + [CLK_I2S1_BCLK] = "aud_i2s1_bclk", + [CLK_I2S2_BCLK] = "aud_i2s2_bclk", + [CLK_I2S3_BCLK] = "aud_i2s3_bclk", + [CLK_I2S4_BCLK] = "aud_i2s4_bclk", + [CLK_CONNSYS_I2S_ASRC] = "aud_connsys_i2s_asrc", + [CLK_GENERAL1_ASRC] = "aud_general1_asrc", + [CLK_GENERAL2_ASRC] = "aud_general2_asrc", + [CLK_ADC_HIRES_TML] = "aud_adc_hires_tml", + [CLK_ADDA6_ADC] = "aud_adda6_adc", + [CLK_ADDA6_ADC_HIRES] = "aud_adda6_adc_hires", + [CLK_3RD_DAC] = "aud_3rd_dac", + [CLK_3RD_DAC_PREDIS] = "aud_3rd_dac_predis", + [CLK_3RD_DAC_TML] = "aud_3rd_dac_tml", + [CLK_3RD_DAC_HIRES] = "aud_3rd_dac_hires", + [CLK_ETDM_IN1_BCLK] = "aud_etdm_in1_bclk", + [CLK_ETDM_OUT1_BCLK] = "aud_etdm_out1_bclk", + [CLK_INFRA_SYS_AUDIO] = "aud_infra_clk", + [CLK_INFRA_AUDIO_26M] = "mtkaif_26m_clk", + [CLK_MUX_AUDIO] = "top_mux_audio", + [CLK_MUX_AUDIOINTBUS] = "top_mux_audio_int", + [CLK_TOP_MAINPLL_D2_D4] = "top_mainpll_d2_d4", + [CLK_TOP_MUX_AUD_1] = "top_mux_aud_1", + [CLK_TOP_APLL1_CK] = "top_apll1_ck", + [CLK_TOP_MUX_AUD_2] = "top_mux_aud_2", + [CLK_TOP_APLL2_CK] = "top_apll2_ck", + [CLK_TOP_MUX_AUD_ENG1] = "top_mux_aud_eng1", + [CLK_TOP_APLL1_D8] = "top_apll1_d8", + [CLK_TOP_MUX_AUD_ENG2] = "top_mux_aud_eng2", + [CLK_TOP_APLL2_D8] = "top_apll2_d8", + [CLK_TOP_MUX_AUDIO_H] = "top_mux_audio_h", + [CLK_TOP_I2S0_M_SEL] = "top_i2s0_m_sel", + [CLK_TOP_I2S1_M_SEL] = "top_i2s1_m_sel", + [CLK_TOP_I2S2_M_SEL] = "top_i2s2_m_sel", + [CLK_TOP_I2S4_M_SEL] = "top_i2s4_m_sel", + [CLK_TOP_TDM_M_SEL] = "top_tdm_m_sel", + [CLK_TOP_APLL12_DIV0] = "top_apll12_div0", + [CLK_TOP_APLL12_DIV1] = "top_apll12_div1", + [CLK_TOP_APLL12_DIV2] = "top_apll12_div2", + [CLK_TOP_APLL12_DIV4] = "top_apll12_div4", + [CLK_TOP_APLL12_DIV_TDM] = "top_apll12_div_tdm", + [CLK_CLK26M] = "top_clk26m_clk", +}; + +int mt8186_set_audio_int_bus_parent(struct mtk_base_afe *afe, + int clk_id) +{ + struct mt8186_afe_private *afe_priv = afe->platform_priv; + int ret; + + ret = clk_set_parent(afe_priv->clk[CLK_MUX_AUDIOINTBUS], + afe_priv->clk[clk_id]); + if (ret) { + dev_err(afe->dev, "%s clk_set_parent %s-%s fail %d\n", + __func__, aud_clks[CLK_MUX_AUDIOINTBUS], + aud_clks[clk_id], ret); + return ret; + } + + return 0; +} + +static int apll1_mux_setting(struct mtk_base_afe *afe, bool enable) +{ + struct mt8186_afe_private *afe_priv = afe->platform_priv; + int ret; + + if (enable) { + ret = clk_prepare_enable(afe_priv->clk[CLK_TOP_MUX_AUD_1]); + if (ret) { + dev_err(afe->dev, "%s clk_prepare_enable %s fail %d\n", + __func__, aud_clks[CLK_TOP_MUX_AUD_1], ret); + return ret; + } + ret = clk_set_parent(afe_priv->clk[CLK_TOP_MUX_AUD_1], + afe_priv->clk[CLK_TOP_APLL1_CK]); + if (ret) { + dev_err(afe->dev, "%s clk_set_parent %s-%s fail %d\n", + __func__, aud_clks[CLK_TOP_MUX_AUD_1], + aud_clks[CLK_TOP_APLL1_CK], ret); + return ret; + } + + /* 180.6336 / 8 = 22.5792MHz */ + ret = clk_prepare_enable(afe_priv->clk[CLK_TOP_MUX_AUD_ENG1]); + if (ret) { + dev_err(afe->dev, "%s clk_prepare_enable %s fail %d\n", + __func__, aud_clks[CLK_TOP_MUX_AUD_ENG1], ret); + return ret; + } + ret = clk_set_parent(afe_priv->clk[CLK_TOP_MUX_AUD_ENG1], + afe_priv->clk[CLK_TOP_APLL1_D8]); + if (ret) { + dev_err(afe->dev, "%s clk_set_parent %s-%s fail %d\n", + __func__, aud_clks[CLK_TOP_MUX_AUD_ENG1], + aud_clks[CLK_TOP_APLL1_D8], ret); + return ret; + } + } else { + ret = clk_set_parent(afe_priv->clk[CLK_TOP_MUX_AUD_ENG1], + afe_priv->clk[CLK_CLK26M]); + if (ret) { + dev_err(afe->dev, "%s clk_set_parent %s-%s fail %d\n", + __func__, aud_clks[CLK_TOP_MUX_AUD_ENG1], + aud_clks[CLK_CLK26M], ret); + return ret; + } + clk_disable_unprepare(afe_priv->clk[CLK_TOP_MUX_AUD_ENG1]); + + ret = clk_set_parent(afe_priv->clk[CLK_TOP_MUX_AUD_1], + afe_priv->clk[CLK_CLK26M]); + if (ret) { + dev_err(afe->dev, "%s clk_set_parent %s-%s fail %d\n", + __func__, aud_clks[CLK_TOP_MUX_AUD_1], + aud_clks[CLK_CLK26M], ret); + return ret; + } + clk_disable_unprepare(afe_priv->clk[CLK_TOP_MUX_AUD_1]); + } + + return 0; +} + +static int apll2_mux_setting(struct mtk_base_afe *afe, bool enable) +{ + struct mt8186_afe_private *afe_priv = afe->platform_priv; + int ret; + + if (enable) { + ret = clk_prepare_enable(afe_priv->clk[CLK_TOP_MUX_AUD_2]); + if (ret) { + dev_err(afe->dev, "%s clk_prepare_enable %s fail %d\n", + __func__, aud_clks[CLK_TOP_MUX_AUD_2], ret); + return ret; + } + ret = clk_set_parent(afe_priv->clk[CLK_TOP_MUX_AUD_2], + afe_priv->clk[CLK_TOP_APLL2_CK]); + if (ret) { + dev_err(afe->dev, "%s clk_set_parent %s-%s fail %d\n", + __func__, aud_clks[CLK_TOP_MUX_AUD_2], + aud_clks[CLK_TOP_APLL2_CK], ret); + return ret; + } + + /* 196.608 / 8 = 24.576MHz */ + ret = clk_prepare_enable(afe_priv->clk[CLK_TOP_MUX_AUD_ENG2]); + if (ret) { + dev_err(afe->dev, "%s clk_prepare_enable %s fail %d\n", + __func__, aud_clks[CLK_TOP_MUX_AUD_ENG2], ret); + return ret; + } + ret = clk_set_parent(afe_priv->clk[CLK_TOP_MUX_AUD_ENG2], + afe_priv->clk[CLK_TOP_APLL2_D8]); + if (ret) { + dev_err(afe->dev, "%s clk_set_parent %s-%s fail %d\n", + __func__, aud_clks[CLK_TOP_MUX_AUD_ENG2], + aud_clks[CLK_TOP_APLL2_D8], ret); + return ret; + } + } else { + ret = clk_set_parent(afe_priv->clk[CLK_TOP_MUX_AUD_ENG2], + afe_priv->clk[CLK_CLK26M]); + if (ret) { + dev_err(afe->dev, "%s clk_set_parent %s-%s fail %d\n", + __func__, aud_clks[CLK_TOP_MUX_AUD_ENG2], + aud_clks[CLK_CLK26M], ret); + return ret; + } + clk_disable_unprepare(afe_priv->clk[CLK_TOP_MUX_AUD_ENG2]); + + ret = clk_set_parent(afe_priv->clk[CLK_TOP_MUX_AUD_2], + afe_priv->clk[CLK_CLK26M]); + if (ret) { + dev_err(afe->dev, "%s clk_set_parent %s-%s fail %d\n", + __func__, aud_clks[CLK_TOP_MUX_AUD_2], + aud_clks[CLK_CLK26M], ret); + return ret; + } + clk_disable_unprepare(afe_priv->clk[CLK_TOP_MUX_AUD_2]); + } + + return 0; +} + +int mt8186_afe_enable_cgs(struct mtk_base_afe *afe) +{ + struct mt8186_afe_private *afe_priv = afe->platform_priv; + int ret = 0; + int i; + + for (i = CLK_I2S1_BCLK; i <= CLK_ETDM_OUT1_BCLK; i++) { + ret = clk_prepare_enable(afe_priv->clk[i]); + if (ret) { + dev_err(afe->dev, "%s clk_prepare_enable %s fail %d\n", + __func__, aud_clks[i], ret); + return ret; + } + } + + return 0; +} + +void mt8186_afe_disable_cgs(struct mtk_base_afe *afe) +{ + struct mt8186_afe_private *afe_priv = afe->platform_priv; + int i; + + for (i = CLK_I2S1_BCLK; i <= CLK_ETDM_OUT1_BCLK; i++) + clk_disable_unprepare(afe_priv->clk[i]); +} + +int mt8186_afe_enable_clock(struct mtk_base_afe *afe) +{ + struct mt8186_afe_private *afe_priv = afe->platform_priv; + int ret = 0; + + ret = clk_prepare_enable(afe_priv->clk[CLK_INFRA_SYS_AUDIO]); + if (ret) { + dev_err(afe->dev, "%s clk_prepare_enable %s fail %d\n", + __func__, aud_clks[CLK_INFRA_SYS_AUDIO], ret); + goto clk_infra_sys_audio_err; + } + + ret = clk_prepare_enable(afe_priv->clk[CLK_INFRA_AUDIO_26M]); + if (ret) { + dev_err(afe->dev, "%s clk_prepare_enable %s fail %d\n", + __func__, aud_clks[CLK_INFRA_AUDIO_26M], ret); + goto clk_infra_audio_26m_err; + } + + ret = clk_prepare_enable(afe_priv->clk[CLK_MUX_AUDIO]); + if (ret) { + dev_err(afe->dev, "%s clk_prepare_enable %s fail %d\n", + __func__, aud_clks[CLK_MUX_AUDIO], ret); + goto clk_mux_audio_err; + } + ret = clk_set_parent(afe_priv->clk[CLK_MUX_AUDIO], + afe_priv->clk[CLK_CLK26M]); + if (ret) { + dev_err(afe->dev, "%s clk_set_parent %s-%s fail %d\n", + __func__, aud_clks[CLK_MUX_AUDIO], + aud_clks[CLK_CLK26M], ret); + goto clk_mux_audio_err; + } + + ret = clk_prepare_enable(afe_priv->clk[CLK_MUX_AUDIOINTBUS]); + if (ret) { + dev_err(afe->dev, "%s clk_prepare_enable %s fail %d\n", + __func__, aud_clks[CLK_MUX_AUDIOINTBUS], ret); + goto clk_mux_audio_intbus_err; + } + ret = mt8186_set_audio_int_bus_parent(afe, + CLK_TOP_MAINPLL_D2_D4); + if (ret) + goto clk_mux_audio_intbus_parent_err; + + ret = clk_set_parent(afe_priv->clk[CLK_TOP_MUX_AUDIO_H], + afe_priv->clk[CLK_TOP_APLL2_CK]); + if (ret) { + dev_err(afe->dev, "%s clk_set_parent %s-%s fail %d\n", + __func__, aud_clks[CLK_TOP_MUX_AUDIO_H], + aud_clks[CLK_TOP_APLL2_CK], ret); + goto clk_mux_audio_h_parent_err; + } + + ret = clk_prepare_enable(afe_priv->clk[CLK_AFE]); + if (ret) { + dev_err(afe->dev, "%s clk_prepare_enable %s fail %d\n", + __func__, aud_clks[CLK_AFE], ret); + goto clk_afe_err; + } + + return 0; + +clk_afe_err: + clk_disable_unprepare(afe_priv->clk[CLK_AFE]); +clk_mux_audio_h_parent_err: +clk_mux_audio_intbus_parent_err: + mt8186_set_audio_int_bus_parent(afe, CLK_CLK26M); +clk_mux_audio_intbus_err: + clk_disable_unprepare(afe_priv->clk[CLK_MUX_AUDIOINTBUS]); +clk_mux_audio_err: + clk_disable_unprepare(afe_priv->clk[CLK_MUX_AUDIO]); +clk_infra_sys_audio_err: + clk_disable_unprepare(afe_priv->clk[CLK_INFRA_SYS_AUDIO]); +clk_infra_audio_26m_err: + clk_disable_unprepare(afe_priv->clk[CLK_INFRA_AUDIO_26M]); + + return ret; +} + +void mt8186_afe_disable_clock(struct mtk_base_afe *afe) +{ + struct mt8186_afe_private *afe_priv = afe->platform_priv; + + clk_disable_unprepare(afe_priv->clk[CLK_AFE]); + mt8186_set_audio_int_bus_parent(afe, CLK_CLK26M); + clk_disable_unprepare(afe_priv->clk[CLK_MUX_AUDIOINTBUS]); + clk_disable_unprepare(afe_priv->clk[CLK_MUX_AUDIO]); + clk_disable_unprepare(afe_priv->clk[CLK_INFRA_AUDIO_26M]); + clk_disable_unprepare(afe_priv->clk[CLK_INFRA_SYS_AUDIO]); +} + +int mt8186_afe_suspend_clock(struct mtk_base_afe *afe) +{ + struct mt8186_afe_private *afe_priv = afe->platform_priv; + int ret; + + /* set audio int bus to 26M */ + ret = clk_prepare_enable(afe_priv->clk[CLK_MUX_AUDIOINTBUS]); + if (ret) { + dev_info(afe->dev, "%s clk_prepare_enable %s fail %d\n", + __func__, aud_clks[CLK_MUX_AUDIOINTBUS], ret); + goto clk_mux_audio_intbus_err; + } + ret = mt8186_set_audio_int_bus_parent(afe, CLK_CLK26M); + if (ret) + goto clk_mux_audio_intbus_parent_err; + + clk_disable_unprepare(afe_priv->clk[CLK_MUX_AUDIOINTBUS]); + + return 0; + +clk_mux_audio_intbus_parent_err: + mt8186_set_audio_int_bus_parent(afe, CLK_TOP_MAINPLL_D2_D4); +clk_mux_audio_intbus_err: + clk_disable_unprepare(afe_priv->clk[CLK_MUX_AUDIOINTBUS]); + return ret; +} + +int mt8186_afe_resume_clock(struct mtk_base_afe *afe) +{ + struct mt8186_afe_private *afe_priv = afe->platform_priv; + int ret; + + /* set audio int bus to normal working clock */ + ret = clk_prepare_enable(afe_priv->clk[CLK_MUX_AUDIOINTBUS]); + if (ret) { + dev_info(afe->dev, "%s clk_prepare_enable %s fail %d\n", + __func__, aud_clks[CLK_MUX_AUDIOINTBUS], ret); + goto clk_mux_audio_intbus_err; + } + ret = mt8186_set_audio_int_bus_parent(afe, + CLK_TOP_MAINPLL_D2_D4); + if (ret) + goto clk_mux_audio_intbus_parent_err; + + clk_disable_unprepare(afe_priv->clk[CLK_MUX_AUDIOINTBUS]); + + return 0; + +clk_mux_audio_intbus_parent_err: + mt8186_set_audio_int_bus_parent(afe, CLK_CLK26M); +clk_mux_audio_intbus_err: + clk_disable_unprepare(afe_priv->clk[CLK_MUX_AUDIOINTBUS]); + return ret; +} + +int mt8186_apll1_enable(struct mtk_base_afe *afe) +{ + struct mt8186_afe_private *afe_priv = afe->platform_priv; + int ret; + + /* setting for APLL */ + apll1_mux_setting(afe, true); + + ret = clk_prepare_enable(afe_priv->clk[CLK_APLL22M]); + if (ret) { + dev_err(afe->dev, "%s clk_prepare_enable %s fail %d\n", + __func__, aud_clks[CLK_APLL22M], ret); + goto err_clk_apll22m; + } + + ret = clk_prepare_enable(afe_priv->clk[CLK_APLL1_TUNER]); + if (ret) { + dev_err(afe->dev, "%s clk_prepare_enable %s fail %d\n", + __func__, aud_clks[CLK_APLL1_TUNER], ret); + goto err_clk_apll1_tuner; + } + + regmap_update_bits(afe->regmap, AFE_APLL1_TUNER_CFG, 0xfff7, 0x832); + regmap_update_bits(afe->regmap, AFE_APLL1_TUNER_CFG, 0x1, 0x1); + + regmap_update_bits(afe->regmap, AFE_HD_ENGEN_ENABLE, + AFE_22M_ON_MASK_SFT, BIT(AFE_22M_ON_SFT)); + + return 0; + +err_clk_apll1_tuner: + clk_disable_unprepare(afe_priv->clk[CLK_APLL1_TUNER]); +err_clk_apll22m: + clk_disable_unprepare(afe_priv->clk[CLK_APLL22M]); + + return ret; +} + +void mt8186_apll1_disable(struct mtk_base_afe *afe) +{ + struct mt8186_afe_private *afe_priv = afe->platform_priv; + + regmap_update_bits(afe->regmap, AFE_HD_ENGEN_ENABLE, + AFE_22M_ON_MASK_SFT, 0); + + regmap_update_bits(afe->regmap, AFE_APLL1_TUNER_CFG, 0x1, 0); + + clk_disable_unprepare(afe_priv->clk[CLK_APLL1_TUNER]); + clk_disable_unprepare(afe_priv->clk[CLK_APLL22M]); + + apll1_mux_setting(afe, false); +} + +int mt8186_apll2_enable(struct mtk_base_afe *afe) +{ + struct mt8186_afe_private *afe_priv = afe->platform_priv; + int ret; + + /* setting for APLL */ + apll2_mux_setting(afe, true); + + ret = clk_prepare_enable(afe_priv->clk[CLK_APLL24M]); + if (ret) { + dev_err(afe->dev, "%s clk_prepare_enable %s fail %d\n", + __func__, aud_clks[CLK_APLL24M], ret); + goto err_clk_apll24m; + } + + ret = clk_prepare_enable(afe_priv->clk[CLK_APLL2_TUNER]); + if (ret) { + dev_err(afe->dev, "%s clk_prepare_enable %s fail %d\n", + __func__, aud_clks[CLK_APLL2_TUNER], ret); + goto err_clk_apll2_tuner; + } + + regmap_update_bits(afe->regmap, AFE_APLL2_TUNER_CFG, 0xfff7, 0x634); + regmap_update_bits(afe->regmap, AFE_APLL2_TUNER_CFG, 0x1, 0x1); + + regmap_update_bits(afe->regmap, AFE_HD_ENGEN_ENABLE, + AFE_24M_ON_MASK_SFT, BIT(AFE_24M_ON_SFT)); + + return 0; + +err_clk_apll2_tuner: + clk_disable_unprepare(afe_priv->clk[CLK_APLL2_TUNER]); +err_clk_apll24m: + clk_disable_unprepare(afe_priv->clk[CLK_APLL24M]); + + return ret; +} + +void mt8186_apll2_disable(struct mtk_base_afe *afe) +{ + struct mt8186_afe_private *afe_priv = afe->platform_priv; + + regmap_update_bits(afe->regmap, AFE_HD_ENGEN_ENABLE, + AFE_24M_ON_MASK_SFT, 0); + + regmap_update_bits(afe->regmap, AFE_APLL2_TUNER_CFG, 0x1, 0); + + clk_disable_unprepare(afe_priv->clk[CLK_APLL2_TUNER]); + clk_disable_unprepare(afe_priv->clk[CLK_APLL24M]); + + apll2_mux_setting(afe, false); +} + +int mt8186_get_apll_rate(struct mtk_base_afe *afe, int apll) +{ + return (apll == MT8186_APLL1) ? 180633600 : 196608000; +} + +int mt8186_get_apll_by_rate(struct mtk_base_afe *afe, int rate) +{ + return ((rate % 8000) == 0) ? MT8186_APLL2 : MT8186_APLL1; +} + +int mt8186_get_apll_by_name(struct mtk_base_afe *afe, const char *name) +{ + if (strcmp(name, APLL1_W_NAME) == 0) + return MT8186_APLL1; + + return MT8186_APLL2; +} + +/* mck */ +struct mt8186_mck_div { + u32 m_sel_id; + u32 div_clk_id; +}; + +static const struct mt8186_mck_div mck_div[MT8186_MCK_NUM] = { + [MT8186_I2S0_MCK] = { + .m_sel_id = CLK_TOP_I2S0_M_SEL, + .div_clk_id = CLK_TOP_APLL12_DIV0, + }, + [MT8186_I2S1_MCK] = { + .m_sel_id = CLK_TOP_I2S1_M_SEL, + .div_clk_id = CLK_TOP_APLL12_DIV1, + }, + [MT8186_I2S2_MCK] = { + .m_sel_id = CLK_TOP_I2S2_M_SEL, + .div_clk_id = CLK_TOP_APLL12_DIV2, + }, + [MT8186_I2S4_MCK] = { + .m_sel_id = CLK_TOP_I2S4_M_SEL, + .div_clk_id = CLK_TOP_APLL12_DIV4, + }, + [MT8186_TDM_MCK] = { + .m_sel_id = CLK_TOP_TDM_M_SEL, + .div_clk_id = CLK_TOP_APLL12_DIV_TDM, + }, +}; + +int mt8186_mck_enable(struct mtk_base_afe *afe, int mck_id, int rate) +{ + struct mt8186_afe_private *afe_priv = afe->platform_priv; + int apll = mt8186_get_apll_by_rate(afe, rate); + int apll_clk_id = apll == MT8186_APLL1 ? + CLK_TOP_MUX_AUD_1 : CLK_TOP_MUX_AUD_2; + int m_sel_id = mck_div[mck_id].m_sel_id; + int div_clk_id = mck_div[mck_id].div_clk_id; + int ret; + + /* select apll */ + if (m_sel_id >= 0) { + ret = clk_prepare_enable(afe_priv->clk[m_sel_id]); + if (ret) { + dev_err(afe->dev, "%s(), clk_prepare_enable %s fail %d\n", + __func__, aud_clks[m_sel_id], ret); + return ret; + } + ret = clk_set_parent(afe_priv->clk[m_sel_id], + afe_priv->clk[apll_clk_id]); + if (ret) { + dev_err(afe->dev, "%s(), clk_set_parent %s-%s fail %d\n", + __func__, aud_clks[m_sel_id], + aud_clks[apll_clk_id], ret); + return ret; + } + } + + /* enable div, set rate */ + ret = clk_prepare_enable(afe_priv->clk[div_clk_id]); + if (ret) { + dev_err(afe->dev, "%s(), clk_prepare_enable %s fail %d\n", + __func__, aud_clks[div_clk_id], ret); + return ret; + } + ret = clk_set_rate(afe_priv->clk[div_clk_id], rate); + if (ret) { + dev_err(afe->dev, "%s(), clk_set_rate %s, rate %d, fail %d\n", + __func__, aud_clks[div_clk_id], rate, ret); + return ret; + } + + return 0; +} + +void mt8186_mck_disable(struct mtk_base_afe *afe, int mck_id) +{ + struct mt8186_afe_private *afe_priv = afe->platform_priv; + int m_sel_id = mck_div[mck_id].m_sel_id; + int div_clk_id = mck_div[mck_id].div_clk_id; + + clk_disable_unprepare(afe_priv->clk[div_clk_id]); + if (m_sel_id >= 0) + clk_disable_unprepare(afe_priv->clk[m_sel_id]); +} + +int mt8186_init_clock(struct mtk_base_afe *afe) +{ + struct mt8186_afe_private *afe_priv = afe->platform_priv; + struct device_node *of_node = afe->dev->of_node; + int i = 0; + + mt8186_audsys_clk_register(afe); + + afe_priv->clk = devm_kcalloc(afe->dev, CLK_NUM, sizeof(*afe_priv->clk), + GFP_KERNEL); + if (!afe_priv->clk) + return -ENOMEM; + + for (i = 0; i < CLK_NUM; i++) { + afe_priv->clk[i] = devm_clk_get(afe->dev, aud_clks[i]); + if (IS_ERR(afe_priv->clk[i])) { + dev_err(afe->dev, "%s devm_clk_get %s fail, ret %ld\n", + __func__, + aud_clks[i], PTR_ERR(afe_priv->clk[i])); + afe_priv->clk[i] = NULL; + } + } + + afe_priv->apmixedsys = syscon_regmap_lookup_by_phandle(of_node, + "mediatek,apmixedsys"); + if (IS_ERR(afe_priv->apmixedsys)) { + dev_err(afe->dev, "%s() Cannot find apmixedsys controller: %ld\n", + __func__, PTR_ERR(afe_priv->apmixedsys)); + return PTR_ERR(afe_priv->apmixedsys); + } + + afe_priv->topckgen = syscon_regmap_lookup_by_phandle(of_node, + "mediatek,topckgen"); + if (IS_ERR(afe_priv->topckgen)) { + dev_err(afe->dev, "%s() Cannot find topckgen controller: %ld\n", + __func__, PTR_ERR(afe_priv->topckgen)); + return PTR_ERR(afe_priv->topckgen); + } + + afe_priv->infracfg = syscon_regmap_lookup_by_phandle(of_node, + "mediatek,infracfg"); + if (IS_ERR(afe_priv->infracfg)) { + dev_err(afe->dev, "%s() Cannot find infracfg: %ld\n", + __func__, PTR_ERR(afe_priv->infracfg)); + return PTR_ERR(afe_priv->infracfg); + } + + return 0; +} + +void mt8186_deinit_clock(struct mtk_base_afe *afe) +{ + mt8186_audsys_clk_unregister(afe); +} diff --git a/sound/soc/mediatek/mt8186/mt8186-afe-clk.h b/sound/soc/mediatek/mt8186/mt8186-afe-clk.h new file mode 100644 index 000000000000..c539557d7c78 --- /dev/null +++ b/sound/soc/mediatek/mt8186/mt8186-afe-clk.h @@ -0,0 +1,106 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * mt8186-afe-clk.h -- Mediatek 8186 afe clock ctrl definition + * + * Copyright (c) 2022 MediaTek Inc. + * Author: Jiaxin Yu + */ + +#ifndef _MT8186_AFE_CLOCK_CTRL_H_ +#define _MT8186_AFE_CLOCK_CTRL_H_ + +#define PERI_BUS_DCM_CTRL 0x74 + +/* APLL */ +#define APLL1_W_NAME "APLL1" +#define APLL2_W_NAME "APLL2" +enum { + MT8186_APLL1 = 0, + MT8186_APLL2, +}; + +enum { + CLK_AFE = 0, + CLK_DAC, + CLK_DAC_PREDIS, + CLK_ADC, + CLK_TML, + CLK_APLL22M, + CLK_APLL24M, + CLK_APLL1_TUNER, + CLK_APLL2_TUNER, + CLK_TDM, + CLK_NLE, + CLK_DAC_HIRES, + CLK_ADC_HIRES, + CLK_I2S1_BCLK, + CLK_I2S2_BCLK, + CLK_I2S3_BCLK, + CLK_I2S4_BCLK, + CLK_CONNSYS_I2S_ASRC, + CLK_GENERAL1_ASRC, + CLK_GENERAL2_ASRC, + CLK_ADC_HIRES_TML, + CLK_ADDA6_ADC, + CLK_ADDA6_ADC_HIRES, + CLK_3RD_DAC, + CLK_3RD_DAC_PREDIS, + CLK_3RD_DAC_TML, + CLK_3RD_DAC_HIRES, + CLK_ETDM_IN1_BCLK, + CLK_ETDM_OUT1_BCLK, + CLK_INFRA_SYS_AUDIO, + CLK_INFRA_AUDIO_26M, + CLK_MUX_AUDIO, + CLK_MUX_AUDIOINTBUS, + CLK_TOP_MAINPLL_D2_D4, + /* apll related mux */ + CLK_TOP_MUX_AUD_1, + CLK_TOP_APLL1_CK, + CLK_TOP_MUX_AUD_2, + CLK_TOP_APLL2_CK, + CLK_TOP_MUX_AUD_ENG1, + CLK_TOP_APLL1_D8, + CLK_TOP_MUX_AUD_ENG2, + CLK_TOP_APLL2_D8, + CLK_TOP_MUX_AUDIO_H, + CLK_TOP_I2S0_M_SEL, + CLK_TOP_I2S1_M_SEL, + CLK_TOP_I2S2_M_SEL, + CLK_TOP_I2S4_M_SEL, + CLK_TOP_TDM_M_SEL, + CLK_TOP_APLL12_DIV0, + CLK_TOP_APLL12_DIV1, + CLK_TOP_APLL12_DIV2, + CLK_TOP_APLL12_DIV4, + CLK_TOP_APLL12_DIV_TDM, + CLK_CLK26M, + CLK_NUM +}; + +struct mtk_base_afe; +int mt8186_set_audio_int_bus_parent(struct mtk_base_afe *afe, int clk_id); +int mt8186_init_clock(struct mtk_base_afe *afe); +void mt8186_deinit_clock(struct mtk_base_afe *afe); +int mt8186_afe_enable_cgs(struct mtk_base_afe *afe); +void mt8186_afe_disable_cgs(struct mtk_base_afe *afe); +int mt8186_afe_enable_clock(struct mtk_base_afe *afe); +void mt8186_afe_disable_clock(struct mtk_base_afe *afe); +int mt8186_afe_suspend_clock(struct mtk_base_afe *afe); +int mt8186_afe_resume_clock(struct mtk_base_afe *afe); + +int mt8186_apll1_enable(struct mtk_base_afe *afe); +void mt8186_apll1_disable(struct mtk_base_afe *afe); + +int mt8186_apll2_enable(struct mtk_base_afe *afe); +void mt8186_apll2_disable(struct mtk_base_afe *afe); + +int mt8186_get_apll_rate(struct mtk_base_afe *afe, int apll); +int mt8186_get_apll_by_rate(struct mtk_base_afe *afe, int rate); +int mt8186_get_apll_by_name(struct mtk_base_afe *afe, const char *name); + +/* these will be replaced by using CCF */ +int mt8186_mck_enable(struct mtk_base_afe *afe, int mck_id, int rate); +void mt8186_mck_disable(struct mtk_base_afe *afe, int mck_id); + +#endif -- cgit v1.2.3 From cfa9a966f12a91a269e50f1c3237c006ffe2ee9a Mon Sep 17 00:00:00 2001 From: Jiaxin Yu Date: Mon, 23 May 2022 21:28:50 +0800 Subject: ASoC: mediatek: mt8186: support gpio control in platform driver Add gpio control for all audio interface separately. Signed-off-by: Jiaxin Yu Link: https://lore.kernel.org/r/20220523132858.22166-13-jiaxin.yu@mediatek.com Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8186/mt8186-afe-gpio.c | 244 ++++++++++++++++++++++++++++ sound/soc/mediatek/mt8186/mt8186-afe-gpio.h | 19 +++ 2 files changed, 263 insertions(+) create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-gpio.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-gpio.h (limited to 'sound') diff --git a/sound/soc/mediatek/mt8186/mt8186-afe-gpio.c b/sound/soc/mediatek/mt8186/mt8186-afe-gpio.c new file mode 100644 index 000000000000..5ba28095b7da --- /dev/null +++ b/sound/soc/mediatek/mt8186/mt8186-afe-gpio.c @@ -0,0 +1,244 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// mt8186-afe-gpio.c -- Mediatek 8186 afe gpio ctrl +// +// Copyright (c) 2022 MediaTek Inc. +// Author: Jiaxin Yu + +#include +#include + +#include "mt8186-afe-common.h" +#include "mt8186-afe-gpio.h" + +struct pinctrl *aud_pinctrl; + +enum mt8186_afe_gpio { + MT8186_AFE_GPIO_CLK_MOSI_OFF, + MT8186_AFE_GPIO_CLK_MOSI_ON, + MT8186_AFE_GPIO_CLK_MISO_OFF, + MT8186_AFE_GPIO_CLK_MISO_ON, + MT8186_AFE_GPIO_DAT_MISO_OFF, + MT8186_AFE_GPIO_DAT_MISO_ON, + MT8186_AFE_GPIO_DAT_MOSI_OFF, + MT8186_AFE_GPIO_DAT_MOSI_ON, + MT8186_AFE_GPIO_I2S0_OFF, + MT8186_AFE_GPIO_I2S0_ON, + MT8186_AFE_GPIO_I2S1_OFF, + MT8186_AFE_GPIO_I2S1_ON, + MT8186_AFE_GPIO_I2S2_OFF, + MT8186_AFE_GPIO_I2S2_ON, + MT8186_AFE_GPIO_I2S3_OFF, + MT8186_AFE_GPIO_I2S3_ON, + MT8186_AFE_GPIO_TDM_OFF, + MT8186_AFE_GPIO_TDM_ON, + MT8186_AFE_GPIO_PCM_OFF, + MT8186_AFE_GPIO_PCM_ON, + MT8186_AFE_GPIO_GPIO_NUM +}; + +struct audio_gpio_attr { + const char *name; + bool gpio_prepare; + struct pinctrl_state *gpioctrl; +}; + +static struct audio_gpio_attr aud_gpios[MT8186_AFE_GPIO_GPIO_NUM] = { + [MT8186_AFE_GPIO_CLK_MOSI_OFF] = {"aud_clk_mosi_off", false, NULL}, + [MT8186_AFE_GPIO_CLK_MOSI_ON] = {"aud_clk_mosi_on", false, NULL}, + [MT8186_AFE_GPIO_CLK_MISO_OFF] = {"aud_clk_miso_off", false, NULL}, + [MT8186_AFE_GPIO_CLK_MISO_ON] = {"aud_clk_miso_on", false, NULL}, + [MT8186_AFE_GPIO_DAT_MISO_OFF] = {"aud_dat_miso_off", false, NULL}, + [MT8186_AFE_GPIO_DAT_MISO_ON] = {"aud_dat_miso_on", false, NULL}, + [MT8186_AFE_GPIO_DAT_MOSI_OFF] = {"aud_dat_mosi_off", false, NULL}, + [MT8186_AFE_GPIO_DAT_MOSI_ON] = {"aud_dat_mosi_on", false, NULL}, + [MT8186_AFE_GPIO_I2S0_OFF] = {"aud_gpio_i2s0_off", false, NULL}, + [MT8186_AFE_GPIO_I2S0_ON] = {"aud_gpio_i2s0_on", false, NULL}, + [MT8186_AFE_GPIO_I2S1_OFF] = {"aud_gpio_i2s1_off", false, NULL}, + [MT8186_AFE_GPIO_I2S1_ON] = {"aud_gpio_i2s1_on", false, NULL}, + [MT8186_AFE_GPIO_I2S2_OFF] = {"aud_gpio_i2s2_off", false, NULL}, + [MT8186_AFE_GPIO_I2S2_ON] = {"aud_gpio_i2s2_on", false, NULL}, + [MT8186_AFE_GPIO_I2S3_OFF] = {"aud_gpio_i2s3_off", false, NULL}, + [MT8186_AFE_GPIO_I2S3_ON] = {"aud_gpio_i2s3_on", false, NULL}, + [MT8186_AFE_GPIO_TDM_OFF] = {"aud_gpio_tdm_off", false, NULL}, + [MT8186_AFE_GPIO_TDM_ON] = {"aud_gpio_tdm_on", false, NULL}, + [MT8186_AFE_GPIO_PCM_OFF] = {"aud_gpio_pcm_off", false, NULL}, + [MT8186_AFE_GPIO_PCM_ON] = {"aud_gpio_pcm_on", false, NULL}, +}; + +static DEFINE_MUTEX(gpio_request_mutex); + +int mt8186_afe_gpio_init(struct device *dev) +{ + int i, j, ret; + + aud_pinctrl = devm_pinctrl_get(dev); + if (IS_ERR(aud_pinctrl)) { + ret = PTR_ERR(aud_pinctrl); + dev_err(dev, "%s(), ret %d, cannot get aud_pinctrl!\n", + __func__, ret); + return ret; + } + + for (i = 0; i < ARRAY_SIZE(aud_gpios); i++) { + aud_gpios[i].gpioctrl = pinctrl_lookup_state(aud_pinctrl, + aud_gpios[i].name); + if (IS_ERR(aud_gpios[i].gpioctrl)) { + ret = PTR_ERR(aud_gpios[i].gpioctrl); + dev_info(dev, "%s(), pinctrl_lookup_state %s fail, ret %d\n", + __func__, aud_gpios[i].name, ret); + } else { + aud_gpios[i].gpio_prepare = true; + } + } + + /* gpio status init */ + for (i = MT8186_DAI_ADDA; i <= MT8186_DAI_TDM_IN; i++) { + for (j = 0; j <= 1; j++) + mt8186_afe_gpio_request(dev, false, i, j); + } + + return 0; +} +EXPORT_SYMBOL_GPL(mt8186_afe_gpio_init); + +static int mt8186_afe_gpio_select(struct device *dev, + enum mt8186_afe_gpio type) +{ + int ret = 0; + + if (type < 0 || type >= MT8186_AFE_GPIO_GPIO_NUM) { + dev_err(dev, "%s(), error, invalid gpio type %d\n", + __func__, type); + return -EINVAL; + } + + if (!aud_gpios[type].gpio_prepare) { + dev_err(dev, "%s(), error, gpio type %d not prepared\n", + __func__, type); + return -EIO; + } + + ret = pinctrl_select_state(aud_pinctrl, + aud_gpios[type].gpioctrl); + if (ret) { + dev_err(dev, "%s(), error, can not set gpio type %d\n", + __func__, type); + return ret; + } + + return 0; +} + +static int mt8186_afe_gpio_adda_dl(struct device *dev, bool enable) +{ + int ret; + + if (enable) { + ret = mt8186_afe_gpio_select(dev, MT8186_AFE_GPIO_CLK_MOSI_ON); + if (ret) { + dev_err(dev, "%s(), MOSI CLK ON slect fail!\n", __func__); + return ret; + } + + ret = mt8186_afe_gpio_select(dev, MT8186_AFE_GPIO_DAT_MOSI_ON); + if (ret) { + dev_err(dev, "%s(), MOSI DAT ON slect fail!\n", __func__); + return ret; + } + } else { + ret = mt8186_afe_gpio_select(dev, MT8186_AFE_GPIO_DAT_MOSI_OFF); + if (ret) { + dev_err(dev, "%s(), MOSI DAT OFF slect fail!\n", __func__); + return ret; + } + + ret = mt8186_afe_gpio_select(dev, MT8186_AFE_GPIO_CLK_MOSI_OFF); + if (ret) { + dev_err(dev, "%s(), MOSI CLK ON slect fail!\n", __func__); + return ret; + } + } + + return 0; +} + +static int mt8186_afe_gpio_adda_ul(struct device *dev, bool enable) +{ + int ret; + + if (enable) { + ret = mt8186_afe_gpio_select(dev, MT8186_AFE_GPIO_CLK_MISO_ON); + if (ret) { + dev_err(dev, "%s(), MISO CLK ON slect fail!\n", __func__); + return ret; + } + + ret = mt8186_afe_gpio_select(dev, MT8186_AFE_GPIO_DAT_MISO_ON); + if (ret) { + dev_err(dev, "%s(), MISO DAT ON slect fail!\n", __func__); + return ret; + } + } else { + ret = mt8186_afe_gpio_select(dev, MT8186_AFE_GPIO_DAT_MISO_OFF); + if (ret) { + dev_err(dev, "%s(), MISO DAT OFF slect fail!\n", __func__); + return ret; + } + + ret = mt8186_afe_gpio_select(dev, MT8186_AFE_GPIO_CLK_MISO_OFF); + if (ret) { + dev_err(dev, "%s(), MISO CLK OFF slect fail!\n", __func__); + return ret; + } + } + + return 0; +} + +int mt8186_afe_gpio_request(struct device *dev, bool enable, + int dai, int uplink) +{ + enum mt8186_afe_gpio sel; + int ret = -EINVAL; + + mutex_lock(&gpio_request_mutex); + + switch (dai) { + case MT8186_DAI_ADDA: + if (uplink) + ret = mt8186_afe_gpio_adda_ul(dev, enable); + else + ret = mt8186_afe_gpio_adda_dl(dev, enable); + goto unlock; + case MT8186_DAI_I2S_0: + sel = enable ? MT8186_AFE_GPIO_I2S0_ON : MT8186_AFE_GPIO_I2S0_OFF; + break; + case MT8186_DAI_I2S_1: + sel = enable ? MT8186_AFE_GPIO_I2S1_ON : MT8186_AFE_GPIO_I2S1_OFF; + break; + case MT8186_DAI_I2S_2: + sel = enable ? MT8186_AFE_GPIO_I2S2_ON : MT8186_AFE_GPIO_I2S2_OFF; + break; + case MT8186_DAI_I2S_3: + sel = enable ? MT8186_AFE_GPIO_I2S3_ON : MT8186_AFE_GPIO_I2S3_OFF; + break; + case MT8186_DAI_TDM_IN: + sel = enable ? MT8186_AFE_GPIO_TDM_ON : MT8186_AFE_GPIO_TDM_OFF; + break; + case MT8186_DAI_PCM: + sel = enable ? MT8186_AFE_GPIO_PCM_ON : MT8186_AFE_GPIO_PCM_OFF; + break; + default: + mutex_unlock(&gpio_request_mutex); + dev_err(dev, "%s(), invalid dai %d\n", __func__, dai); + goto unlock; + } + + ret = mt8186_afe_gpio_select(dev, sel); + +unlock: + mutex_unlock(&gpio_request_mutex); + + return ret; +} diff --git a/sound/soc/mediatek/mt8186/mt8186-afe-gpio.h b/sound/soc/mediatek/mt8186/mt8186-afe-gpio.h new file mode 100644 index 000000000000..1ddc27838eb1 --- /dev/null +++ b/sound/soc/mediatek/mt8186/mt8186-afe-gpio.h @@ -0,0 +1,19 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * mt6833-afe-gpio.h -- Mediatek 6833 afe gpio ctrl definition + * + * Copyright (c) 2022 MediaTek Inc. + * Author: Jiaxin Yu + */ + +#ifndef _MT8186_AFE_GPIO_H_ +#define _MT8186_AFE_GPIO_H_ + +struct mtk_base_afe; + +int mt8186_afe_gpio_init(struct device *dev); + +int mt8186_afe_gpio_request(struct device *dev, bool enable, + int dai, int uplink); + +#endif -- cgit v1.2.3 From 80d8cad2e9ce21517d50c7084c12a59d38a778f7 Mon Sep 17 00:00:00 2001 From: Jiaxin Yu Date: Mon, 23 May 2022 21:28:51 +0800 Subject: ASoC: mediatek: mt8186: add misc driver and register definitions Add mt8186 platform misc driver and data tables/register definitions files. Signed-off-by: Jiaxin Yu Link: https://lore.kernel.org/r/20220523132858.22166-14-jiaxin.yu@mediatek.com Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8186/mt8186-interconnection.h | 69 + sound/soc/mediatek/mt8186/mt8186-misc-control.c | 252 ++ sound/soc/mediatek/mt8186/mt8186-reg.h | 2913 ++++++++++++++++++++ 3 files changed, 3234 insertions(+) create mode 100644 sound/soc/mediatek/mt8186/mt8186-interconnection.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-misc-control.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-reg.h (limited to 'sound') diff --git a/sound/soc/mediatek/mt8186/mt8186-interconnection.h b/sound/soc/mediatek/mt8186/mt8186-interconnection.h new file mode 100644 index 000000000000..5b188d93ebd3 --- /dev/null +++ b/sound/soc/mediatek/mt8186/mt8186-interconnection.h @@ -0,0 +1,69 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * Mediatek MT8186 audio driver interconnection definition + * + * Copyright (c) 2022 MediaTek Inc. + * Author: Jiaxin Yu + */ + +#ifndef _MT8186_INTERCONNECTION_H_ +#define _MT8186_INTERCONNECTION_H_ + +/* in port define */ +#define I_I2S0_CH1 0 +#define I_I2S0_CH2 1 +#define I_ADDA_UL_CH1 3 +#define I_ADDA_UL_CH2 4 +#define I_DL1_CH1 5 +#define I_DL1_CH2 6 +#define I_DL2_CH1 7 +#define I_DL2_CH2 8 +#define I_PCM_1_CAP_CH1 9 +#define I_GAIN1_OUT_CH1 10 +#define I_GAIN1_OUT_CH2 11 +#define I_GAIN2_OUT_CH1 12 +#define I_GAIN2_OUT_CH2 13 +#define I_PCM_2_CAP_CH1 14 +#define I_ADDA_UL_CH3 17 +#define I_ADDA_UL_CH4 18 +#define I_DL12_CH1 19 +#define I_DL12_CH2 20 +#define I_DL12_CH3 5 +#define I_DL12_CH4 6 +#define I_PCM_2_CAP_CH2 21 +#define I_PCM_1_CAP_CH2 22 +#define I_DL3_CH1 23 +#define I_DL3_CH2 24 +#define I_I2S2_CH1 25 +#define I_I2S2_CH2 26 +#define I_I2S2_CH3 27 +#define I_I2S2_CH4 28 + +/* in port define >= 32 */ +#define I_32_OFFSET 32 +#define I_CONNSYS_I2S_CH1 (34 - I_32_OFFSET) +#define I_CONNSYS_I2S_CH2 (35 - I_32_OFFSET) +#define I_SRC_1_OUT_CH1 (36 - I_32_OFFSET) +#define I_SRC_1_OUT_CH2 (37 - I_32_OFFSET) +#define I_SRC_2_OUT_CH1 (38 - I_32_OFFSET) +#define I_SRC_2_OUT_CH2 (39 - I_32_OFFSET) +#define I_DL4_CH1 (40 - I_32_OFFSET) +#define I_DL4_CH2 (41 - I_32_OFFSET) +#define I_DL5_CH1 (42 - I_32_OFFSET) +#define I_DL5_CH2 (43 - I_32_OFFSET) +#define I_DL6_CH1 (44 - I_32_OFFSET) +#define I_DL6_CH2 (45 - I_32_OFFSET) +#define I_DL7_CH1 (46 - I_32_OFFSET) +#define I_DL7_CH2 (47 - I_32_OFFSET) +#define I_DL8_CH1 (48 - I_32_OFFSET) +#define I_DL8_CH2 (49 - I_32_OFFSET) +#define I_TDM_IN_CH1 (56 - I_32_OFFSET) +#define I_TDM_IN_CH2 (57 - I_32_OFFSET) +#define I_TDM_IN_CH3 (58 - I_32_OFFSET) +#define I_TDM_IN_CH4 (59 - I_32_OFFSET) +#define I_TDM_IN_CH5 (60 - I_32_OFFSET) +#define I_TDM_IN_CH6 (61 - I_32_OFFSET) +#define I_TDM_IN_CH7 (62 - I_32_OFFSET) +#define I_TDM_IN_CH8 (63 - I_32_OFFSET) + +#endif diff --git a/sound/soc/mediatek/mt8186/mt8186-misc-control.c b/sound/soc/mediatek/mt8186/mt8186-misc-control.c new file mode 100644 index 000000000000..2317de8c44c0 --- /dev/null +++ b/sound/soc/mediatek/mt8186/mt8186-misc-control.c @@ -0,0 +1,252 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// MediaTek ALSA SoC Audio Misc Control +// +// Copyright (c) 2022 MediaTek Inc. +// Author: Jiaxin Yu + +#include +#include +#include +#include +#include + +#include "../common/mtk-afe-fe-dai.h" +#include "../common/mtk-afe-platform-driver.h" +#include "mt8186-afe-common.h" + +static const char * const mt8186_sgen_mode_str[] = { + "I0I1", "I2", "I3I4", "I5I6", + "I7I8", "I9I22", "I10I11", "I12I13", + "I14I21", "I15I16", "I17I18", "I19I20", + "I23I24", "I25I26", "I27I28", "I33", + "I34I35", "I36I37", "I38I39", "I40I41", + "I42I43", "I44I45", "I46I47", "I48I49", + "I56I57", "I58I59", "I60I61", "I62I63", + "O0O1", "O2", "O3O4", "O5O6", + "O7O8", "O9O10", "O11", "O12", + "O13O14", "O15O16", "O17O18", "O19O20", + "O21O22", "O23O24", "O25", "O28O29", + "O34", "O35", "O32O33", "O36O37", + "O38O39", "O30O31", "O40O41", "O42O43", + "O44O45", "O46O47", "O48O49", "O50O51", + "O58O59", "O60O61", "O62O63", "O64O65", + "O66O67", "O68O69", "O26O27", "OFF", +}; + +static const int mt8186_sgen_mode_idx[] = { + 0, 2, 4, 6, + 8, 22, 10, 12, + 14, -1, 18, 20, + 24, 26, 28, 33, + 34, 36, 38, 40, + 42, 44, 46, 48, + 56, 58, 60, 62, + 128, 130, 132, 134, + 135, 138, 139, 140, + 142, 144, 166, 148, + 150, 152, 153, 156, + 162, 163, 160, 164, + 166, -1, 168, 170, + 172, 174, 176, 178, + 186, 188, 190, 192, + 194, 196, -1, -1, +}; + +static const char * const mt8186_sgen_rate_str[] = { + "8K", "11K", "12K", "16K", + "22K", "24K", "32K", "44K", + "48K", "88k", "96k", "176k", + "192k" +}; + +static const int mt8186_sgen_rate_idx[] = { + 0, 1, 2, 4, + 5, 6, 8, 9, + 10, 11, 12, 13, + 14 +}; + +/* this order must match reg bit amp_div_ch1/2 */ +static const char * const mt8186_sgen_amp_str[] = { + "1/128", "1/64", "1/32", "1/16", "1/8", "1/4", "1/2", "1" }; + +static int mt8186_sgen_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct mtk_base_afe *afe = snd_soc_component_get_drvdata(cmpnt); + struct mt8186_afe_private *afe_priv = afe->platform_priv; + + ucontrol->value.integer.value[0] = afe_priv->sgen_mode; + + return 0; +} + +static int mt8186_sgen_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct mtk_base_afe *afe = snd_soc_component_get_drvdata(cmpnt); + struct mt8186_afe_private *afe_priv = afe->platform_priv; + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + int mode; + int mode_idx; + + if (ucontrol->value.enumerated.item[0] >= e->items) + return -EINVAL; + + mode = ucontrol->value.integer.value[0]; + mode_idx = mt8186_sgen_mode_idx[mode]; + + dev_dbg(afe->dev, "%s(), mode %d, mode_idx %d\n", + __func__, mode, mode_idx); + + if (mode == afe_priv->sgen_mode) + return 0; + + if (mode_idx >= 0) { + regmap_update_bits(afe->regmap, AFE_SINEGEN_CON2, + INNER_LOOP_BACK_MODE_MASK_SFT, + mode_idx << INNER_LOOP_BACK_MODE_SFT); + regmap_update_bits(afe->regmap, AFE_SINEGEN_CON0, + DAC_EN_MASK_SFT, BIT(DAC_EN_SFT)); + } else { + /* disable sgen */ + regmap_update_bits(afe->regmap, AFE_SINEGEN_CON0, + DAC_EN_MASK_SFT, 0); + regmap_update_bits(afe->regmap, AFE_SINEGEN_CON2, + INNER_LOOP_BACK_MODE_MASK_SFT, + 0x3f << INNER_LOOP_BACK_MODE_SFT); + } + + afe_priv->sgen_mode = mode; + + return 1; +} + +static int mt8186_sgen_rate_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct mtk_base_afe *afe = snd_soc_component_get_drvdata(cmpnt); + struct mt8186_afe_private *afe_priv = afe->platform_priv; + + ucontrol->value.integer.value[0] = afe_priv->sgen_rate; + + return 0; +} + +static int mt8186_sgen_rate_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct mtk_base_afe *afe = snd_soc_component_get_drvdata(cmpnt); + struct mt8186_afe_private *afe_priv = afe->platform_priv; + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + int rate; + + if (ucontrol->value.enumerated.item[0] >= e->items) + return -EINVAL; + + rate = ucontrol->value.integer.value[0]; + + dev_dbg(afe->dev, "%s(), rate %d\n", __func__, rate); + + if (rate == afe_priv->sgen_rate) + return 0; + + regmap_update_bits(afe->regmap, AFE_SINEGEN_CON0, + SINE_MODE_CH1_MASK_SFT, + mt8186_sgen_rate_idx[rate] << SINE_MODE_CH1_SFT); + + regmap_update_bits(afe->regmap, AFE_SINEGEN_CON0, + SINE_MODE_CH2_MASK_SFT, + mt8186_sgen_rate_idx[rate] << SINE_MODE_CH2_SFT); + + afe_priv->sgen_rate = rate; + + return 1; +} + +static int mt8186_sgen_amplitude_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct mtk_base_afe *afe = snd_soc_component_get_drvdata(cmpnt); + struct mt8186_afe_private *afe_priv = afe->platform_priv; + + ucontrol->value.integer.value[0] = afe_priv->sgen_amplitude; + return 0; +} + +static int mt8186_sgen_amplitude_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct mtk_base_afe *afe = snd_soc_component_get_drvdata(cmpnt); + struct mt8186_afe_private *afe_priv = afe->platform_priv; + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + int amplitude; + + if (ucontrol->value.enumerated.item[0] >= e->items) + return -EINVAL; + + amplitude = ucontrol->value.integer.value[0]; + if (amplitude > AMP_DIV_CH1_MASK) { + dev_err(afe->dev, "%s(), amplitude %d invalid\n", + __func__, amplitude); + return -EINVAL; + } + + dev_dbg(afe->dev, "%s(), amplitude %d\n", __func__, amplitude); + + if (amplitude == afe_priv->sgen_amplitude) + return 0; + + regmap_update_bits(afe->regmap, AFE_SINEGEN_CON0, + AMP_DIV_CH1_MASK_SFT, + amplitude << AMP_DIV_CH1_SFT); + regmap_update_bits(afe->regmap, AFE_SINEGEN_CON0, + AMP_DIV_CH2_MASK_SFT, + amplitude << AMP_DIV_CH2_SFT); + + afe_priv->sgen_amplitude = amplitude; + + return 1; +} + +static const struct soc_enum mt8186_afe_sgen_enum[] = { + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(mt8186_sgen_mode_str), + mt8186_sgen_mode_str), + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(mt8186_sgen_rate_str), + mt8186_sgen_rate_str), + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(mt8186_sgen_amp_str), + mt8186_sgen_amp_str), +}; + +static const struct snd_kcontrol_new mt8186_afe_sgen_controls[] = { + SOC_ENUM_EXT("Audio_SineGen_Switch", mt8186_afe_sgen_enum[0], + mt8186_sgen_get, mt8186_sgen_set), + SOC_ENUM_EXT("Audio_SineGen_SampleRate", mt8186_afe_sgen_enum[1], + mt8186_sgen_rate_get, mt8186_sgen_rate_set), + SOC_ENUM_EXT("Audio_SineGen_Amplitude", mt8186_afe_sgen_enum[2], + mt8186_sgen_amplitude_get, mt8186_sgen_amplitude_set), + SOC_SINGLE("Audio_SineGen_Mute_Ch1", AFE_SINEGEN_CON0, + MUTE_SW_CH1_MASK_SFT, MUTE_SW_CH1_MASK, 0), + SOC_SINGLE("Audio_SineGen_Mute_Ch2", AFE_SINEGEN_CON0, + MUTE_SW_CH2_MASK_SFT, MUTE_SW_CH2_MASK, 0), + SOC_SINGLE("Audio_SineGen_Freq_Div_Ch1", AFE_SINEGEN_CON0, + FREQ_DIV_CH1_SFT, FREQ_DIV_CH1_MASK, 0), + SOC_SINGLE("Audio_SineGen_Freq_Div_Ch2", AFE_SINEGEN_CON0, + FREQ_DIV_CH2_SFT, FREQ_DIV_CH2_MASK, 0), +}; + +int mt8186_add_misc_control(struct snd_soc_component *component) +{ + snd_soc_add_component_controls(component, + mt8186_afe_sgen_controls, + ARRAY_SIZE(mt8186_afe_sgen_controls)); + + return 0; +} diff --git a/sound/soc/mediatek/mt8186/mt8186-reg.h b/sound/soc/mediatek/mt8186/mt8186-reg.h new file mode 100644 index 000000000000..53c3eb7283d8 --- /dev/null +++ b/sound/soc/mediatek/mt8186/mt8186-reg.h @@ -0,0 +1,2913 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * mt8186-reg.h -- Mediatek 8186 audio driver reg definition + * + * Copyright (c) 2022 MediaTek Inc. + * Author: Jiaxin Yu + */ + +#ifndef _MT8186_REG_H_ +#define _MT8186_REG_H_ + +/* reg bit enum */ +enum { + MT8186_MEMIF_PBUF_SIZE_32_BYTES, + MT8186_MEMIF_PBUF_SIZE_64_BYTES, + MT8186_MEMIF_PBUF_SIZE_128_BYTES, + MT8186_MEMIF_PBUF_SIZE_256_BYTES, + MT8186_MEMIF_PBUF_SIZE_NUM, +}; + +/***************************************************************************** + * R E G I S T E R D E F I N I T I O N + *****************************************************************************/ +/* AUDIO_TOP_CON0 */ +#define RESERVED_SFT 31 +#define RESERVED_MASK_SFT BIT(31) +#define AHB_IDLE_EN_INT_SFT 30 +#define AHB_IDLE_EN_INT_MASK_SFT BIT(30) +#define AHB_IDLE_EN_EXT_SFT 29 +#define AHB_IDLE_EN_EXT_MASK_SFT BIT(29) +#define PDN_NLE_SFT 28 +#define PDN_NLE_MASK_SFT BIT(28) +#define PDN_TML_SFT 27 +#define PDN_TML_MASK_SFT BIT(27) +#define PDN_DAC_PREDIS_SFT 26 +#define PDN_DAC_PREDIS_MASK_SFT BIT(26) +#define PDN_DAC_SFT 25 +#define PDN_DAC_MASK_SFT BIT(25) +#define PDN_ADC_SFT 24 +#define PDN_ADC_MASK_SFT BIT(24) +#define PDN_TDM_CK_SFT 20 +#define PDN_TDM_CK_MASK_SFT BIT(20) +#define PDN_APLL_TUNER_SFT 19 +#define PDN_APLL_TUNER_MASK_SFT BIT(19) +#define PDN_APLL2_TUNER_SFT 18 +#define PDN_APLL2_TUNER_MASK_SFT BIT(18) +#define APB3_SEL_SFT 14 +#define APB3_SEL_MASK_SFT BIT(14) +#define APB_R2T_SFT 13 +#define APB_R2T_MASK_SFT BIT(13) +#define APB_W2T_SFT 12 +#define APB_W2T_MASK_SFT BIT(12) +#define PDN_24M_SFT 9 +#define PDN_24M_MASK_SFT BIT(9) +#define PDN_22M_SFT 8 +#define PDN_22M_MASK_SFT BIT(8) +#define PDN_AFE_SFT 2 +#define PDN_AFE_MASK_SFT BIT(2) + +/* AUDIO_TOP_CON1 */ +#define PDN_3RD_DAC_HIRES_SFT 31 +#define PDN_3RD_DAC_HIRES_MASK_SFT BIT(31) +#define PDN_3RD_DAC_TML_SFT 30 +#define PDN_3RD_DAC_TML_MASK_SFT BIT(30) +#define PDN_3RD_DAC_PREDIS_SFT 29 +#define PDN_3RD_DAC_PREDIS_MASK_SFT BIT(29) +#define PDN_3RD_DAC_SFT 28 +#define PDN_3RD_DAC_MASK_SFT BIT(28) +#define I2S_SOFT_RST5_SFT 22 +#define I2S_SOFT_RST5_MASK_SFT BIT(22) +#define PDN_ADDA6_ADC_HIRES_SFT 21 +#define PDN_ADDA6_ADC_HIRES_MASK_SFT BIT(21) +#define PDN_ADDA6_ADC_SFT 20 +#define PDN_ADDA6_ADC_MASK_SFT BIT(20) +#define PDN_ADC_HIRES_TML_SFT 17 +#define PDN_ADC_HIRES_TML_MASK_SFT BIT(17) +#define PDN_ADC_HIRES_SFT 16 +#define PDN_ADC_HIRES_MASK_SFT BIT(16) +#define PDN_DAC_HIRES_SFT 15 +#define PDN_DAC_HIRES_MASK_SFT BIT(15) +#define PDN_GENERAL2_ASRC_SFT 14 +#define PDN_GENERAL2_ASRC_MASK_SFT BIT(14) +#define PDN_GENERAL1_ASRC_SFT 13 +#define PDN_GENERAL1_ASRC_MASK_SFT BIT(13) +#define PDN_CONNSYS_I2S_ASRC_SFT 12 +#define PDN_CONNSYS_I2S_ASRC_MASK_SFT BIT(12) +#define I2S4_BCLK_SW_CG_SFT 7 +#define I2S4_BCLK_SW_CG_MASK_SFT BIT(7) +#define I2S3_BCLK_SW_CG_SFT 6 +#define I2S3_BCLK_SW_CG_MASK_SFT BIT(6) +#define I2S2_BCLK_SW_CG_SFT 5 +#define I2S2_BCLK_SW_CG_MASK_SFT BIT(5) +#define I2S1_BCLK_SW_CG_SFT 4 +#define I2S1_BCLK_SW_CG_MASK_SFT BIT(4) +#define I2S_SOFT_RST2_SFT 2 +#define I2S_SOFT_RST2_MASK_SFT BIT(2) +#define I2S_SOFT_RST_SFT 1 +#define I2S_SOFT_RST_MASK_SFT BIT(1) + +/* AUDIO_TOP_CON3 */ +#define BUSY_SFT 31 +#define BUSY_MASK_SFT BIT(31) +#define OS_DISABLE_SFT 30 +#define OS_DISABLE_MASK_SFT BIT(30) +#define CG_DISABLE_SFT 29 +#define CG_DISABLE_MASK_SFT BIT(29) +#define CLEAR_FLAG_SFT 0 +#define CLEAR_FLAG_MASK_SFT BIT(0) + +/* AFE_DAC_CON0 */ +#define VUL12_ON_SFT 31 +#define VUL12_ON_MASK_SFT BIT(31) +#define MOD_DAI_ON_SFT 30 +#define MOD_DAI_ON_MASK_SFT BIT(30) +#define DAI_ON_SFT 29 +#define DAI_ON_MASK_SFT BIT(29) +#define DAI2_ON_SFT 28 +#define DAI2_ON_MASK_SFT BIT(28) +#define VUL6_ON_SFT 23 +#define VUL6_ON_MASK_SFT BIT(23) +#define VUL5_ON_SFT 22 +#define VUL5_ON_MASK_SFT BIT(22) +#define VUL4_ON_SFT 21 +#define VUL4_ON_MASK_SFT BIT(21) +#define VUL3_ON_SFT 20 +#define VUL3_ON_MASK_SFT BIT(20) +#define VUL2_ON_SFT 19 +#define VUL2_ON_MASK_SFT BIT(19) +#define VUL_ON_SFT 18 +#define VUL_ON_MASK_SFT BIT(18) +#define AWB2_ON_SFT 17 +#define AWB2_ON_MASK_SFT BIT(17) +#define AWB_ON_SFT 16 +#define AWB_ON_MASK_SFT BIT(16) +#define DL12_ON_SFT 15 +#define DL12_ON_MASK_SFT BIT(15) +#define DL8_ON_SFT 11 +#define DL8_ON_MASK_SFT BIT(11) +#define DL7_ON_SFT 10 +#define DL7_ON_MASK_SFT BIT(10) +#define DL6_ON_SFT 9 +#define DL6_ON_MASK_SFT BIT(9) +#define DL5_ON_SFT 8 +#define DL5_ON_MASK_SFT BIT(8) +#define DL4_ON_SFT 7 +#define DL4_ON_MASK_SFT BIT(7) +#define DL3_ON_SFT 6 +#define DL3_ON_MASK_SFT BIT(6) +#define DL2_ON_SFT 5 +#define DL2_ON_MASK_SFT BIT(5) +#define DL1_ON_SFT 4 +#define DL1_ON_MASK_SFT BIT(4) +#define AUDIO_AFE_ON_SFT 0 +#define AUDIO_AFE_ON_MASK_SFT BIT(0) + +/* AFE_DAC_MON */ +#define AFE_ON_RETM_SFT 0 +#define AFE_ON_RETM_MASK_SFT BIT(0) + +/* AFE_I2S_CON */ +#define BCK_NEG_EG_LATCH_SFT 30 +#define BCK_NEG_EG_LATCH_MASK_SFT BIT(30) +#define BCK_INV_SFT 29 +#define BCK_INV_MASK_SFT BIT(29) +#define I2SIN_PAD_SEL_SFT 28 +#define I2SIN_PAD_SEL_MASK_SFT BIT(28) +#define I2S_LOOPBACK_SFT 20 +#define I2S_LOOPBACK_MASK_SFT BIT(20) +#define I2S_ONOFF_NOT_RESET_CK_ENABLE_SFT 17 +#define I2S_ONOFF_NOT_RESET_CK_ENABLE_MASK_SFT BIT(17) +#define I2S1_HD_EN_SFT 12 +#define I2S1_HD_EN_MASK_SFT BIT(12) +#define I2S_OUT_MODE_SFT 8 +#define I2S_OUT_MODE_MASK_SFT GENMASK(11, 8) +#define INV_PAD_CTRL_SFT 7 +#define INV_PAD_CTRL_MASK_SFT BIT(7) +#define I2S_BYPSRC_SFT 6 +#define I2S_BYPSRC_MASK_SFT BIT(6) +#define INV_LRCK_SFT 5 +#define INV_LRCK_MASK_SFT BIT(5) +#define I2S_FMT_SFT 3 +#define I2S_FMT_MASK_SFT BIT(3) +#define I2S_SRC_SFT 2 +#define I2S_SRC_MASK_SFT BIT(2) +#define I2S_WLEN_SFT 1 +#define I2S_WLEN_MASK_SFT BIT(1) +#define I2S_EN_SFT 0 +#define I2S_EN_MASK_SFT BIT(0) + +/* AFE_I2S_CON1 */ +#define I2S2_LR_SWAP_SFT 31 +#define I2S2_LR_SWAP_MASK_SFT BIT(31) +#define I2S2_SEL_O19_O20_SFT 18 +#define I2S2_SEL_O19_O20_MASK_SFT BIT(18) +#define I2S_ONOFF_NOT_RESET_CK_ENABLE_SFT 17 +#define I2S_ONOFF_NOT_RESET_CK_ENABLE_MASK_SFT BIT(17) +#define I2S2_SEL_O03_O04_SFT 16 +#define I2S2_SEL_O03_O04_MASK_SFT BIT(16) +#define I2S2_HD_EN_SFT 12 +#define I2S2_HD_EN_MASK_SFT BIT(12) +#define I2S2_OUT_MODE_SFT 8 +#define I2S2_OUT_MODE_MASK_SFT GENMASK(11, 8) +#define INV_LRCK_SFT 5 +#define INV_LRCK_MASK_SFT BIT(5) +#define I2S2_FMT_SFT 3 +#define I2S2_FMT_MASK_SFT BIT(3) +#define I2S2_WLEN_SFT 1 +#define I2S2_WLEN_MASK_SFT BIT(1) +#define I2S2_EN_SFT 0 +#define I2S2_EN_MASK_SFT BIT(0) + +/* AFE_I2S_CON2 */ +#define I2S3_LR_SWAP_SFT 31 +#define I2S3_LR_SWAP_MASK_SFT BIT(31) +#define I2S3_UPDATE_WORD_SFT 24 +#define I2S3_UPDATE_WORD_MASK_SFT GENMASK(28, 24) +#define I2S3_BCK_INV_SFT 23 +#define I2S3_BCK_INV_MASK_SFT BIT(23) +#define I2S3_FPGA_BIT_TEST_SFT 22 +#define I2S3_FPGA_BIT_TEST_MASK_SFT BIT(22) +#define I2S3_FPGA_BIT_SFT 21 +#define I2S3_FPGA_BIT_MASK_SFT BIT(21) +#define I2S3_LOOPBACK_SFT 20 +#define I2S3_LOOPBACK_MASK_SFT BIT(20) +#define I2S_ONOFF_NOT_RESET_CK_ENABLE_SFT 17 +#define I2S_ONOFF_NOT_RESET_CK_ENABLE_MASK_SFT BIT(17) +#define I2S3_HD_EN_SFT 12 +#define I2S3_HD_EN_MASK_SFT BIT(12) +#define I2S3_OUT_MODE_SFT 8 +#define I2S3_OUT_MODE_MASK_SFT GENMASK(11, 8) +#define I2S3_FMT_SFT 3 +#define I2S3_FMT_MASK_SFT BIT(3) +#define I2S3_WLEN_SFT 1 +#define I2S3_WLEN_MASK_SFT BIT(1) +#define I2S3_EN_SFT 0 +#define I2S3_EN_MASK_SFT BIT(0) + +/* AFE_I2S_CON3 */ +#define I2S4_LR_SWAP_SFT 31 +#define I2S4_LR_SWAP_MASK_SFT BIT(31) +#define I2S_ONOFF_NOT_RESET_CK_ENABLE_SFT 17 +#define I2S_ONOFF_NOT_RESET_CK_ENABLE_MASK_SFT BIT(17) +#define I2S4_HD_EN_SFT 12 +#define I2S4_HD_EN_MASK_SFT BIT(12) +#define I2S4_OUT_MODE_SFT 8 +#define I2S4_OUT_MODE_MASK_SFT GENMASK(11, 8) +#define INV_LRCK_SFT 5 +#define INV_LRCK_MASK_SFT BIT(5) +#define I2S4_FMT_SFT 3 +#define I2S4_FMT_MASK_SFT BIT(3) +#define I2S4_WLEN_SFT 1 +#define I2S4_WLEN_MASK_SFT BIT(1) +#define I2S4_EN_SFT 0 +#define I2S4_EN_MASK_SFT BIT(0) + +/* AFE_I2S_CON4 */ +#define I2S_LOOPBACK_SFT 20 +#define I2S_LOOPBACK_MASK 0x1 +#define I2S_LOOPBACK_MASK_SFT BIT(20) +#define I2S_ONOFF_NOT_RESET_CK_ENABLE_SFT 17 +#define I2S_ONOFF_NOT_RESET_CK_ENABLE_MASK 0x1 +#define I2S_ONOFF_NOT_RESET_CK_ENABLE_MASK_SFT BIT(17) +#define INV_LRCK_SFT 5 +#define INV_LRCK_MASK 0x1 +#define INV_LRCK_MASK_SFT BIT(5) + +/* AFE_CONNSYS_I2S_CON */ +#define BCK_NEG_EG_LATCH_SFT 30 +#define BCK_NEG_EG_LATCH_MASK_SFT BIT(30) +#define BCK_INV_SFT 29 +#define BCK_INV_MASK_SFT BIT(29) +#define I2SIN_PAD_SEL_SFT 28 +#define I2SIN_PAD_SEL_MASK_SFT BIT(28) +#define I2S_LOOPBACK_SFT 20 +#define I2S_LOOPBACK_MASK_SFT BIT(20) +#define I2S_ONOFF_NOT_RESET_CK_ENABLE_SFT 17 +#define I2S_ONOFF_NOT_RESET_CK_ENABLE_MASK_SFT BIT(17) +#define I2S_MODE_SFT 8 +#define I2S_MODE_MASK_SFT GENMASK(11, 8) +#define INV_PAD_CTRL_SFT 7 +#define INV_PAD_CTRL_MASK_SFT BIT(7) +#define I2S_BYPSRC_SFT 6 +#define I2S_BYPSRC_MASK_SFT BIT(6) +#define INV_LRCK_SFT 5 +#define INV_LRCK_MASK_SFT BIT(5) +#define I2S_FMT_SFT 3 +#define I2S_FMT_MASK_SFT BIT(3) +#define I2S_SRC_SFT 2 +#define I2S_SRC_MASK_SFT BIT(2) +#define I2S_WLEN_SFT 1 +#define I2S_WLEN_MASK_SFT BIT(1) +#define I2S_EN_SFT 0 +#define I2S_EN_MASK_SFT BIT(0) + +/* AFE_ASRC_2CH_CON2 */ +#define CHSET_O16BIT_SFT 19 +#define CHSET_O16BIT_MASK_SFT BIT(19) +#define CHSET_CLR_IIR_HISTORY_SFT 17 +#define CHSET_CLR_IIR_HISTORY_MASK_SFT BIT(17) +#define CHSET_IS_MONO_SFT 16 +#define CHSET_IS_MONO_MASK_SFT BIT(16) +#define CHSET_IIR_EN_SFT 11 +#define CHSET_IIR_EN_MASK_SFT BIT(11) +#define CHSET_IIR_STAGE_SFT 8 +#define CHSET_IIR_STAGE_MASK_SFT GENMASK(10, 8) +#define CHSET_STR_CLR_SFT 5 +#define CHSET_STR_CLR_MASK_SFT BIT(5) +#define CHSET_ON_SFT 2 +#define CHSET_ON_MASK_SFT BIT(2) +#define COEFF_SRAM_CTRL_SFT 1 +#define COEFF_SRAM_CTRL_MASK_SFT BIT(1) +#define ASM_ON_SFT 0 +#define ASM_ON_MASK_SFT BIT(0) + +/* AFE_GAIN1_CON0 */ +#define GAIN1_SAMPLE_PER_STEP_SFT 8 +#define GAIN1_SAMPLE_PER_STEP_MASK_SFT GENMASK(15, 8) +#define GAIN1_MODE_SFT 4 +#define GAIN1_MODE_MASK_SFT GENMASK(7, 4) +#define GAIN1_ON_SFT 0 +#define GAIN1_ON_MASK_SFT BIT(0) + +/* AFE_GAIN1_CON1 */ +#define GAIN1_TARGET_SFT 0 +#define GAIN1_TARGET_MASK 0xfffffff +#define GAIN1_TARGET_MASK_SFT GENMASK(27, 0) + +/* AFE_GAIN2_CON0 */ +#define GAIN2_SAMPLE_PER_STEP_SFT 8 +#define GAIN2_SAMPLE_PER_STEP_MASK_SFT GENMASK(15, 8) +#define GAIN2_MODE_SFT 4 +#define GAIN2_MODE_MASK_SFT GENMASK(7, 4) +#define GAIN2_ON_SFT 0 +#define GAIN2_ON_MASK_SFT BIT(0) + +/* AFE_GAIN2_CON1 */ +#define GAIN2_TARGET_SFT 0 +#define GAIN2_TARGET_MASK 0xfffffff +#define GAIN2_TARGET_MASK_SFT GENMASK(27, 0) + +/* AFE_GAIN1_CUR */ +#define AFE_GAIN1_CUR_SFT 0 +#define AFE_GAIN1_CUR_MASK_SFT GENMASK(27, 0) + +/* AFE_GAIN2_CUR */ +#define AFE_GAIN2_CUR_SFT 0 +#define AFE_GAIN2_CUR_MASK_SFT GENMASK(27, 0) + +/* PCM_INTF_CON1 */ +#define PCM_FIX_VALUE_SEL_SFT 31 +#define PCM_FIX_VALUE_SEL_MASK_SFT BIT(31) +#define PCM_BUFFER_LOOPBACK_SFT 30 +#define PCM_BUFFER_LOOPBACK_MASK_SFT BIT(30) +#define PCM_PARALLEL_LOOPBACK_SFT 29 +#define PCM_PARALLEL_LOOPBACK_MASK_SFT BIT(29) +#define PCM_SERIAL_LOOPBACK_SFT 28 +#define PCM_SERIAL_LOOPBACK_MASK_SFT BIT(28) +#define PCM_DAI_PCM_LOOPBACK_SFT 27 +#define PCM_DAI_PCM_LOOPBACK_MASK_SFT BIT(27) +#define PCM_I2S_PCM_LOOPBACK_SFT 26 +#define PCM_I2S_PCM_LOOPBACK_MASK_SFT BIT(26) +#define PCM_SYNC_DELSEL_SFT 25 +#define PCM_SYNC_DELSEL_MASK_SFT BIT(25) +#define PCM_TX_LR_SWAP_SFT 24 +#define PCM_TX_LR_SWAP_MASK_SFT BIT(24) +#define PCM_SYNC_OUT_INV_SFT 23 +#define PCM_SYNC_OUT_INV_MASK_SFT BIT(23) +#define PCM_BCLK_OUT_INV_SFT 22 +#define PCM_BCLK_OUT_INV_MASK_SFT BIT(22) +#define PCM_SYNC_IN_INV_SFT 21 +#define PCM_SYNC_IN_INV_MASK_SFT BIT(21) +#define PCM_BCLK_IN_INV_SFT 20 +#define PCM_BCLK_IN_INV_MASK_SFT BIT(20) +#define PCM_TX_LCH_RPT_SFT 19 +#define PCM_TX_LCH_RPT_MASK_SFT BIT(19) +#define PCM_VBT_16K_MODE_SFT 18 +#define PCM_VBT_16K_MODE_MASK_SFT BIT(18) +#define PCM_EXT_MODEM_SFT 17 +#define PCM_EXT_MODEM_MASK_SFT BIT(17) +#define PCM_24BIT_SFT 16 +#define PCM_24BIT_MASK_SFT BIT(16) +#define PCM_WLEN_SFT 14 +#define PCM_WLEN_MASK_SFT GENMASK(15, 14) +#define PCM_SYNC_LENGTH_SFT 9 +#define PCM_SYNC_LENGTH_MASK_SFT GENMASK(13, 9) +#define PCM_SYNC_TYPE_SFT 8 +#define PCM_SYNC_TYPE_MASK_SFT BIT(8) +#define PCM_BT_MODE_SFT 7 +#define PCM_BT_MODE_MASK_SFT BIT(7) +#define PCM_BYP_ASRC_SFT 6 +#define PCM_BYP_ASRC_MASK_SFT BIT(6) +#define PCM_SLAVE_SFT 5 +#define PCM_SLAVE_MASK_SFT BIT(5) +#define PCM_MODE_SFT 3 +#define PCM_MODE_MASK_SFT GENMASK(4, 3) +#define PCM_FMT_SFT 1 +#define PCM_FMT_MASK_SFT GENMASK(2, 1) +#define PCM_EN_SFT 0 +#define PCM_EN_MASK_SFT BIT(0) + +/* PCM_INTF_CON2 */ +#define PCM1_TX_FIFO_OV_SFT 31 +#define PCM1_TX_FIFO_OV_MASK_SFT BIT(31) +#define PCM1_RX_FIFO_OV_SFT 30 +#define PCM1_RX_FIFO_OV_MASK_SFT BIT(30) +#define PCM2_TX_FIFO_OV_SFT 29 +#define PCM2_TX_FIFO_OV_MASK_SFT BIT(29) +#define PCM2_RX_FIFO_OV_SFT 28 +#define PCM2_RX_FIFO_OV_MASK_SFT BIT(28) +#define PCM1_SYNC_GLITCH_SFT 27 +#define PCM1_SYNC_GLITCH_MASK_SFT BIT(27) +#define PCM2_SYNC_GLITCH_SFT 26 +#define PCM2_SYNC_GLITCH_MASK_SFT BIT(26) +#define TX3_RCH_DBG_MODE_SFT 17 +#define TX3_RCH_DBG_MODE_MASK_SFT BIT(17) +#define PCM1_PCM2_LOOPBACK_SFT 16 +#define PCM1_PCM2_LOOPBACK_MASK_SFT BIT(16) +#define DAI_PCM_LOOPBACK_CH_SFT 14 +#define DAI_PCM_LOOPBACK_CH_MASK_SFT GENMASK(15, 14) +#define I2S_PCM_LOOPBACK_CH_SFT 12 +#define I2S_PCM_LOOPBACK_CH_MASK_SFT GENMASK(13, 12) +#define TX_FIX_VALUE_SFT 0 +#define TX_FIX_VALUE_MASK_SFT GENMASK(7, 0) + +/* PCM2_INTF_CON */ +#define PCM2_TX_FIX_VALUE_SFT 24 +#define PCM2_TX_FIX_VALUE_MASK_SFT GENMASK(31, 24) +#define PCM2_FIX_VALUE_SEL_SFT 23 +#define PCM2_FIX_VALUE_SEL_MASK_SFT BIT(23) +#define PCM2_BUFFER_LOOPBACK_SFT 22 +#define PCM2_BUFFER_LOOPBACK_MASK_SFT BIT(22) +#define PCM2_PARALLEL_LOOPBACK_SFT 21 +#define PCM2_PARALLEL_LOOPBACK_MASK_SFT BIT(21) +#define PCM2_SERIAL_LOOPBACK_SFT 20 +#define PCM2_SERIAL_LOOPBACK_MASK_SFT BIT(20) +#define PCM2_DAI_PCM_LOOPBACK_SFT 19 +#define PCM2_DAI_PCM_LOOPBACK_MASK_SFT BIT(19) +#define PCM2_I2S_PCM_LOOPBACK_SFT 18 +#define PCM2_I2S_PCM_LOOPBACK_MASK_SFT BIT(18) +#define PCM2_SYNC_DELSEL_SFT 17 +#define PCM2_SYNC_DELSEL_MASK_SFT BIT(17) +#define PCM2_TX_LR_SWAP_SFT 16 +#define PCM2_TX_LR_SWAP_MASK_SFT BIT(16) +#define PCM2_SYNC_IN_INV_SFT 15 +#define PCM2_SYNC_IN_INV_MASK_SFT BIT(15) +#define PCM2_BCLK_IN_INV_SFT 14 +#define PCM2_BCLK_IN_INV_MASK_SFT BIT(14) +#define PCM2_TX_LCH_RPT_SFT 13 +#define PCM2_TX_LCH_RPT_MASK_SFT BIT(13) +#define PCM2_VBT_16K_MODE_SFT 12 +#define PCM2_VBT_16K_MODE_MASK_SFT BIT(12) +#define PCM2_LOOPBACK_CH_SEL_SFT 10 +#define PCM2_LOOPBACK_CH_SEL_MASK_SFT GENMASK(11, 10) +#define PCM2_TX2_BT_MODE_SFT 8 +#define PCM2_TX2_BT_MODE_MASK_SFT BIT(8) +#define PCM2_BT_MODE_SFT 7 +#define PCM2_BT_MODE_MASK_SFT BIT(7) +#define PCM2_AFIFO_SFT 6 +#define PCM2_AFIFO_MASK_SFT BIT(6) +#define PCM2_WLEN_SFT 5 +#define PCM2_WLEN_MASK_SFT BIT(5) +#define PCM2_MODE_SFT 3 +#define PCM2_MODE_MASK_SFT GENMASK(4, 3) +#define PCM2_FMT_SFT 1 +#define PCM2_FMT_MASK_SFT GENMASK(2, 1) +#define PCM2_EN_SFT 0 +#define PCM2_EN_MASK_SFT BIT(0) + +// AFE_CM1_CON +#define CHANNEL_MERGE0_DEBUG_MODE_SFT (31) +#define CHANNEL_MERGE0_DEBUG_MODE_MASK_SFT BIT(31) +#define VUL3_BYPASS_CM_SFT (30) +#define VUL3_BYPASS_CM_MASK (0x1) +#define VUL3_BYPASS_CM_MASK_SFT BIT(30) +#define CM1_DEBUG_MODE_SEL_SFT (29) +#define CM1_DEBUG_MODE_SEL_MASK_SFT BIT(29) +#define CHANNEL_MERGE0_UPDATE_CNT_SFT (16) +#define CHANNEL_MERGE0_UPDATE_CNT_MASK_SFT GENMASK(28, 16) +#define CM1_FS_SELECT_SFT (8) +#define CM1_FS_SELECT_MASK_SFT GENMASK(12, 8) +#define CHANNEL_MERGE0_CHNUM_SFT (3) +#define CHANNEL_MERGE0_CHNUM_MASK_SFT GENMASK(7, 3) +#define CHANNEL_MERGE0_BYTE_SWAP_SFT (1) +#define CHANNEL_MERGE0_BYTE_SWAP_MASK_SFT BIT(1) +#define CHANNEL_MERGE0_EN_SFT (0) +#define CHANNEL_MERGE0_EN_MASK_SFT BIT(0) + +/* AFE_ADDA_MTKAIF_CFG0 */ +#define MTKAIF_RXIF_CLKINV_ADC_SFT 31 +#define MTKAIF_RXIF_CLKINV_ADC_MASK_SFT BIT(31) +#define MTKAIF_RXIF_BYPASS_SRC_SFT 17 +#define MTKAIF_RXIF_BYPASS_SRC_MASK_SFT BIT(17) +#define MTKAIF_RXIF_PROTOCOL2_SFT 16 +#define MTKAIF_RXIF_PROTOCOL2_MASK_SFT BIT(16) +#define MTKAIF_TXIF_BYPASS_SRC_SFT 5 +#define MTKAIF_TXIF_BYPASS_SRC_MASK_SFT BIT(5) +#define MTKAIF_TXIF_PROTOCOL2_SFT 4 +#define MTKAIF_TXIF_PROTOCOL2_MASK_SFT BIT(4) +#define MTKAIF_TXIF_8TO5_SFT 2 +#define MTKAIF_TXIF_8TO5_MASK_SFT BIT(2) +#define MTKAIF_RXIF_8TO5_SFT 1 +#define MTKAIF_RXIF_8TO5_MASK_SFT BIT(1) +#define MTKAIF_IF_LOOPBACK1_SFT 0 +#define MTKAIF_IF_LOOPBACK1_MASK_SFT BIT(0) + +/* AFE_ADDA_MTKAIF_RX_CFG2 */ +#define MTKAIF_RXIF_DETECT_ON_PROTOCOL2_SFT 16 +#define MTKAIF_RXIF_DETECT_ON_PROTOCOL2_MASK_SFT BIT(16) +#define MTKAIF_RXIF_DELAY_CYCLE_SFT 12 +#define MTKAIF_RXIF_DELAY_CYCLE_MASK_SFT GENMASK(15, 12) +#define MTKAIF_RXIF_DELAY_DATA_SFT 8 +#define MTKAIF_RXIF_DELAY_DATA_MASK 0x1 +#define MTKAIF_RXIF_DELAY_DATA_MASK_SFT BIT(8) +#define MTKAIF_RXIF_FIFO_RSP_PROTOCOL2_SFT 4 +#define MTKAIF_RXIF_FIFO_RSP_PROTOCOL2_MASK_SFT GENMASK(6, 4) + +/* AFE_ADDA_DL_SRC2_CON0 */ +#define DL_2_INPUT_MODE_CTL_SFT 28 +#define DL_2_INPUT_MODE_CTL_MASK_SFT GENMASK(31, 28) +#define DL_2_CH1_SATURATION_EN_CTL_SFT 27 +#define DL_2_CH1_SATURATION_EN_CTL_MASK_SFT BIT(27) +#define DL_2_CH2_SATURATION_EN_CTL_SFT 26 +#define DL_2_CH2_SATURATION_EN_CTL_MASK_SFT BIT(26) +#define DL_2_OUTPUT_SEL_CTL_SFT 24 +#define DL_2_OUTPUT_SEL_CTL_MASK_SFT GENMASK(25, 24) +#define DL_2_FADEIN_0START_EN_SFT 16 +#define DL_2_FADEIN_0START_EN_MASK_SFT GENMASK(17, 16) +#define DL_DISABLE_HW_CG_CTL_SFT 15 +#define DL_DISABLE_HW_CG_CTL_MASK_SFT BIT(15) +#define C_DATA_EN_SEL_CTL_PRE_SFT 14 +#define C_DATA_EN_SEL_CTL_PRE_MASK_SFT BIT(14) +#define DL_2_SIDE_TONE_ON_CTL_PRE_SFT 13 +#define DL_2_SIDE_TONE_ON_CTL_PRE_MASK_SFT BIT(13) +#define DL_2_MUTE_CH1_OFF_CTL_PRE_SFT 12 +#define DL_2_MUTE_CH1_OFF_CTL_PRE_MASK_SFT BIT(12) +#define DL_2_MUTE_CH2_OFF_CTL_PRE_SFT 11 +#define DL_2_MUTE_CH2_OFF_CTL_PRE_MASK_SFT BIT(11) +#define DL2_ARAMPSP_CTL_PRE_SFT 9 +#define DL2_ARAMPSP_CTL_PRE_MASK_SFT GENMASK(10, 9) +#define DL_2_IIRMODE_CTL_PRE_SFT 6 +#define DL_2_IIRMODE_CTL_PRE_MASK_SFT GENMASK(8, 6) +#define DL_2_VOICE_MODE_CTL_PRE_SFT 5 +#define DL_2_VOICE_MODE_CTL_PRE_MASK_SFT BIT(5) +#define D2_2_MUTE_CH1_ON_CTL_PRE_SFT 4 +#define D2_2_MUTE_CH1_ON_CTL_PRE_MASK_SFT BIT(4) +#define D2_2_MUTE_CH2_ON_CTL_PRE_SFT 3 +#define D2_2_MUTE_CH2_ON_CTL_PRE_MASK_SFT BIT(3) +#define DL_2_IIR_ON_CTL_PRE_SFT 2 +#define DL_2_IIR_ON_CTL_PRE_MASK_SFT BIT(2) +#define DL_2_GAIN_ON_CTL_PRE_SFT 1 +#define DL_2_GAIN_ON_CTL_PRE_MASK_SFT BIT(1) +#define DL_2_SRC_ON_CTL_PRE_SFT 0 +#define DL_2_SRC_ON_CTL_PRE_MASK_SFT BIT(0) + +/* AFE_ADDA_DL_SRC2_CON1 */ +#define DL_2_GAIN_CTL_PRE_SFT 16 +#define DL_2_GAIN_CTL_PRE_MASK 0xffff +#define DL_2_GAIN_CTL_PRE_MASK_SFT GENMASK(31, 16) +#define DL_2_GAIN_MODE_CTL_SFT 0 +#define DL_2_GAIN_MODE_CTL_MASK_SFT BIT(0) + +/* AFE_ADDA_UL_SRC_CON0 */ +#define ULCF_CFG_EN_CTL_SFT 31 +#define ULCF_CFG_EN_CTL_MASK_SFT BIT(31) +#define UL_DMIC_PHASE_SEL_CH1_SFT 27 +#define UL_DMIC_PHASE_SEL_CH1_MASK_SFT GENMASK(29, 27) +#define UL_DMIC_PHASE_SEL_CH2_SFT 24 +#define UL_DMIC_PHASE_SEL_CH2_MASK_SFT GENMASK(26, 24) +#define UL_MODE_3P25M_CH2_CTL_SFT 22 +#define UL_MODE_3P25M_CH2_CTL_MASK_SFT BIT(22) +#define UL_MODE_3P25M_CH1_CTL_SFT 21 +#define UL_MODE_3P25M_CH1_CTL_MASK_SFT BIT(21) +#define UL_VOICE_MODE_CH1_CH2_CTL_SFT 17 +#define UL_VOICE_MODE_CH1_CH2_CTL_MASK_SFT GENMASK(19, 17) +#define UL_AP_DMIC_ON_SFT 16 +#define UL_AP_DMIC_ON_MASK_SFT BIT(16) +#define DMIC_LOW_POWER_CTL_SFT 14 +#define DMIC_LOW_POWER_CTL_MASK_SFT GENMASK(15, 14) +#define UL_DISABLE_HW_CG_CTL_SFT 12 +#define UL_DISABLE_HW_CG_CTL_MASK_SFT BIT(12) +#define UL_IIR_ON_TMP_CTL_SFT 10 +#define UL_IIR_ON_TMP_CTL_MASK_SFT BIT(10) +#define UL_IIRMODE_CTL_SFT 7 +#define UL_IIRMODE_CTL_MASK_SFT GENMASK(9, 7) +#define DIGMIC_4P33M_SEL_SFT 6 +#define DIGMIC_4P33M_SEL_MASK_SFT BIT(6) +#define DIGMIC_3P25M_1P625M_SEL_SFT 5 +#define DIGMIC_3P25M_1P625M_SEL_MASK_SFT BIT(5) +#define UL_LOOP_BACK_MODE_SFT 2 +#define UL_LOOP_BACK_MODE_MASK_SFT BIT(2) +#define UL_SDM_3_LEVEL_SFT 1 +#define UL_SDM_3_LEVEL_MASK_SFT BIT(1) +#define UL_SRC_ON_CTL_SFT 0 +#define UL_SRC_ON_CTL_MASK_SFT BIT(0) + +/* AFE_ADDA_UL_SRC_CON1 */ +#define C_DAC_EN_CTL_SFT 27 +#define C_DAC_EN_CTL_MASK_SFT BIT(27) +#define C_MUTE_SW_CTL_SFT 26 +#define C_MUTE_SW_CTL_MASK_SFT BIT(26) +#define ASDM_SRC_SEL_CTL_SFT 25 +#define ASDM_SRC_SEL_CTL_MASK_SFT BIT(25) +#define C_AMP_DIV_CH2_CTL_SFT 21 +#define C_AMP_DIV_CH2_CTL_MASK_SFT GENMASK(23, 21) +#define C_FREQ_DIV_CH2_CTL_SFT 16 +#define C_FREQ_DIV_CH2_CTL_MASK_SFT GENMASK(20, 16) +#define C_SINE_MODE_CH2_CTL_SFT 12 +#define C_SINE_MODE_CH2_CTL_MASK_SFT GENMASK(15, 12) +#define C_AMP_DIV_CH1_CTL_SFT 9 +#define C_AMP_DIV_CH1_CTL_MASK_SFT GENMASK(11, 9) +#define C_FREQ_DIV_CH1_CTL_SFT 4 +#define C_FREQ_DIV_CH1_CTL_MASK_SFT GENMASK(8, 4) +#define C_SINE_MODE_CH1_CTL_SFT 0 +#define C_SINE_MODE_CH1_CTL_MASK_SFT GENMASK(3, 0) + +/* AFE_ADDA_TOP_CON0 */ +#define C_LOOP_BACK_MODE_CTL_SFT 12 +#define C_LOOP_BACK_MODE_CTL_MASK_SFT GENMASK(15, 12) +#define ADDA_UL_GAIN_MODE_SFT 8 +#define ADDA_UL_GAIN_MODE_MASK_SFT GENMASK(9, 8) +#define C_EXT_ADC_CTL_SFT 0 +#define C_EXT_ADC_CTL_MASK_SFT BIT(0) + +/* AFE_ADDA_UL_DL_CON0 */ +#define AFE_ADDA_UL_LR_SWAP_SFT 31 +#define AFE_ADDA_UL_LR_SWAP_MASK_SFT BIT(31) +#define AFE_ADDA_CKDIV_RST_SFT 30 +#define AFE_ADDA_CKDIV_RST_MASK_SFT BIT(30) +#define AFE_ADDA_FIFO_AUTO_RST_SFT 29 +#define AFE_ADDA_FIFO_AUTO_RST_MASK_SFT BIT(29) +#define AFE_ADDA_UL_FIFO_DIGMIC_TESTIN_SFT 21 +#define AFE_ADDA_UL_FIFO_DIGMIC_TESTIN_MASK_SFT GENMASK(22, 21) +#define AFE_ADDA_UL_FIFO_DIGMIC_WDATA_TESTEN_SFT 20 +#define AFE_ADDA_UL_FIFO_DIGMIC_WDATA_TESTEN_MASK_SFT BIT(20) +#define AFE_ADDA6_UL_LR_SWAP_SFT 15 +#define AFE_ADDA6_UL_LR_SWAP_MASK_SFT BIT(15) +#define AFE_ADDA6_CKDIV_RST_SFT 14 +#define AFE_ADDA6_CKDIV_RST_MASK_SFT BIT(14) +#define AFE_ADDA6_FIFO_AUTO_RST_SFT 13 +#define AFE_ADDA6_FIFO_AUTO_RST_MASK_SFT BIT(13) +#define AFE_ADDA6_UL_FIFO_DIGMIC_TESTIN_SFT 5 +#define AFE_ADDA6_UL_FIFO_DIGMIC_TESTIN_MASK_SFT GENMASK(6, 5) +#define AFE_ADDA6_UL_FIFO_DIGMIC_WDATA_TESTEN_SFT 4 +#define AFE_ADDA6_UL_FIFO_DIGMIC_WDATA_TESTEN_MASK_SFT BIT(4) +#define ADDA_AFE_ON_SFT 0 +#define ADDA_AFE_ON_MASK_SFT BIT(0) + +/* AFE_SIDETONE_CON0 */ +#define R_RDY_SFT 30 +#define R_RDY_MASK_SFT BIT(30) +#define W_RDY_SFT 29 +#define W_RDY_MASK_SFT BIT(29) +#define R_W_EN_SFT 25 +#define R_W_EN_MASK_SFT BIT(25) +#define R_W_SEL_SFT 24 +#define R_W_SEL_MASK_SFT BIT(24) +#define SEL_CH2_SFT 23 +#define SEL_CH2_MASK_SFT BIT(23) +#define SIDE_TONE_COEFFICIENT_ADDR_SFT 16 +#define SIDE_TONE_COEFFICIENT_ADDR_MASK_SFT GENMASK(20, 16) +#define SIDE_TONE_COEFFICIENT_SFT 0 +#define SIDE_TONE_COEFFICIENT_MASK_SFT GENMASK(15, 0) + +/* AFE_SIDETONE_COEFF */ +#define SIDE_TONE_COEFF_SFT 0 +#define SIDE_TONE_COEFF_MASK_SFT GENMASK(15, 0) + +/* AFE_SIDETONE_CON1 */ +#define STF_BYPASS_MODE_SFT 31 +#define STF_BYPASS_MODE_MASK_SFT BIT(31) +#define STF_BYPASS_MODE_O28_O29_SFT 30 +#define STF_BYPASS_MODE_O28_O29_MASK_SFT BIT(30) +#define STF_BYPASS_MODE_I2S4_SFT 29 +#define STF_BYPASS_MODE_I2S4_MASK_SFT BIT(29) +#define STF_BYPASS_MODE_DL3_SFT 27 +#define STF_BYPASS_MODE_DL3_MASK_SFT BIT(27) +#define STF_BYPASS_MODE_I2S7_SFT 26 +#define STF_BYPASS_MODE_I2S7_MASK_SFT BIT(26) +#define STF_BYPASS_MODE_I2S9_SFT 25 +#define STF_BYPASS_MODE_I2S9_MASK_SFT BIT(25) +#define STF_O19O20_OUT_EN_SEL_SFT 13 +#define STF_O19O20_OUT_EN_SEL_MASK_SFT BIT(13) +#define STF_SOURCE_FROM_O19O20_SFT 12 +#define STF_SOURCE_FROM_O19O20_MASK_SFT BIT(12) +#define SIDE_TONE_ON_SFT 8 +#define SIDE_TONE_ON_MASK_SFT BIT(8) +#define SIDE_TONE_HALF_TAP_NUM_SFT 0 +#define SIDE_TONE_HALF_TAP_NUM_MASK_SFT GENMASK(5, 0) + +/* AFE_SIDETONE_GAIN */ +#define POSITIVE_GAIN_SFT 16 +#define POSITIVE_GAIN_MASK_SFT GENMASK(18, 16) +#define SIDE_TONE_GAIN_SFT 0 +#define SIDE_TONE_GAIN_MASK_SFT GENMASK(15, 0) + +/* AFE_ADDA_DL_SDM_DCCOMP_CON */ +#define USE_3RD_SDM_SFT 28 +#define USE_3RD_SDM_MASK_SFT BIT(28) +#define DL_FIFO_START_POINT_SFT 24 +#define DL_FIFO_START_POINT_MASK_SFT GENMASK(26, 24) +#define DL_FIFO_SWAP_SFT 20 +#define DL_FIFO_SWAP_MASK_SFT BIT(20) +#define C_AUDSDM1ORDSELECT_CTL_SFT 19 +#define C_AUDSDM1ORDSELECT_CTL_MASK_SFT BIT(19) +#define C_SDM7BITSEL_CTL_SFT 18 +#define C_SDM7BITSEL_CTL_MASK_SFT BIT(18) +#define GAIN_AT_SDM_RST_PRE_CTL_SFT 15 +#define GAIN_AT_SDM_RST_PRE_CTL_MASK_SFT BIT(15) +#define DL_DCM_AUTO_IDLE_EN_SFT 14 +#define DL_DCM_AUTO_IDLE_EN_MASK_SFT BIT(14) +#define AFE_DL_SRC_DCM_EN_SFT 13 +#define AFE_DL_SRC_DCM_EN_MASK_SFT BIT(13) +#define AFE_DL_POST_SRC_DCM_EN_SFT 12 +#define AFE_DL_POST_SRC_DCM_EN_MASK_SFT BIT(12) +#define AUD_SDM_MONO_SFT 9 +#define AUD_SDM_MONO_MASK_SFT BIT(9) +#define AUD_DC_COMP_EN_SFT 8 +#define AUD_DC_COMP_EN_MASK_SFT BIT(8) +#define ATTGAIN_CTL_SFT 0 +#define ATTGAIN_CTL_MASK_SFT GENMASK(5, 0) + +/* AFE_SINEGEN_CON0 */ +#define DAC_EN_SFT 26 +#define DAC_EN_MASK 0x1 +#define DAC_EN_MASK_SFT BIT(26) +#define MUTE_SW_CH2_SFT 25 +#define MUTE_SW_CH2_MASK 0x1 +#define MUTE_SW_CH2_MASK_SFT BIT(25) +#define MUTE_SW_CH1_SFT 24 +#define MUTE_SW_CH1_MASK 0x1 +#define MUTE_SW_CH1_MASK_SFT BIT(24) +#define SINE_MODE_CH2_SFT 20 +#define SINE_MODE_CH2_MASK 0xf +#define SINE_MODE_CH2_MASK_SFT GENMASK(23, 20) +#define AMP_DIV_CH2_SFT 17 +#define AMP_DIV_CH2_MASK 0x7 +#define AMP_DIV_CH2_MASK_SFT GENMASK(19, 17) +#define FREQ_DIV_CH2_SFT 12 +#define FREQ_DIV_CH2_MASK 0x1f +#define FREQ_DIV_CH2_MASK_SFT GENMASK(16, 12) +#define SINE_MODE_CH1_SFT 8 +#define SINE_MODE_CH1_MASK 0xf +#define SINE_MODE_CH1_MASK_SFT GENMASK(11, 8) +#define AMP_DIV_CH1_SFT 5 +#define AMP_DIV_CH1_MASK 0x7 +#define AMP_DIV_CH1_MASK_SFT GENMASK(7, 5) +#define FREQ_DIV_CH1_SFT 0 +#define FREQ_DIV_CH1_MASK 0x1f +#define FREQ_DIV_CH1_MASK_SFT GENMASK(4, 0) + +/* AFE_SINEGEN_CON2 */ +#define INNER_LOOP_BACK_MODE_SFT 0 +#define INNER_LOOP_BACK_MODE_MASK_SFT GENMASK(7, 0) + +/* AFE_HD_ENGEN_ENABLE */ +#define AFE_24M_ON_SFT 1 +#define AFE_24M_ON_MASK_SFT BIT(1) +#define AFE_22M_ON_SFT 0 +#define AFE_22M_ON_MASK_SFT BIT(0) + +/* AFE_ADDA_DL_NLE_FIFO_MON */ +#define DL_NLE_FIFO_WBIN_SFT 8 +#define DL_NLE_FIFO_WBIN_MASK_SFT GENMASK(11, 8) +#define DL_NLE_FIFO_RBIN_SFT 4 +#define DL_NLE_FIFO_RBIN_MASK_SFT GENMASK(7, 4) +#define DL_NLE_FIFO_RDACTIVE_SFT 3 +#define DL_NLE_FIFO_RDACTIVE_MASK_SFT BIT(3) +#define DL_NLE_FIFO_STARTRD_SFT 2 +#define DL_NLE_FIFO_STARTRD_MASK_SFT BIT(2) +#define DL_NLE_FIFO_RD_EMPTY_SFT 1 +#define DL_NLE_FIFO_RD_EMPTY_MASK_SFT BIT(1) +#define DL_NLE_FIFO_WR_FULL_SFT 0 +#define DL_NLE_FIFO_WR_FULL_MASK_SFT BIT(0) + +/* AFE_DL1_CON0 */ +#define DL1_MODE_SFT 24 +#define DL1_MODE_MASK 0xf +#define DL1_MODE_MASK_SFT GENMASK(27, 24) +#define DL1_MINLEN_SFT 20 +#define DL1_MINLEN_MASK 0xf +#define DL1_MINLEN_MASK_SFT GENMASK(23, 20) +#define DL1_MAXLEN_SFT 16 +#define DL1_MAXLEN_MASK 0xf +#define DL1_MAXLEN_MASK_SFT GENMASK(19, 16) +#define DL1_SW_CLEAR_BUF_EMPTY_SFT 15 +#define DL1_SW_CLEAR_BUF_EMPTY_MASK 0x1 +#define DL1_SW_CLEAR_BUF_EMPTY_MASK_SFT BIT(15) +#define DL1_PBUF_SIZE_SFT 12 +#define DL1_PBUF_SIZE_MASK 0x3 +#define DL1_PBUF_SIZE_MASK_SFT GENMASK(13, 12) +#define DL1_MONO_SFT 8 +#define DL1_MONO_MASK 0x1 +#define DL1_MONO_MASK_SFT BIT(8) +#define DL1_NORMAL_MODE_SFT 5 +#define DL1_NORMAL_MODE_MASK 0x1 +#define DL1_NORMAL_MODE_MASK_SFT BIT(5) +#define DL1_HALIGN_SFT 4 +#define DL1_HALIGN_MASK 0x1 +#define DL1_HALIGN_MASK_SFT BIT(4) +#define DL1_HD_MODE_SFT 0 +#define DL1_HD_MODE_MASK 0x3 +#define DL1_HD_MODE_MASK_SFT GENMASK(1, 0) + +/* AFE_DL2_CON0 */ +#define DL2_MODE_SFT 24 +#define DL2_MODE_MASK 0xf +#define DL2_MODE_MASK_SFT GENMASK(27, 24) +#define DL2_MINLEN_SFT 20 +#define DL2_MINLEN_MASK 0xf +#define DL2_MINLEN_MASK_SFT GENMASK(23, 20) +#define DL2_MAXLEN_SFT 16 +#define DL2_MAXLEN_MASK 0xf +#define DL2_MAXLEN_MASK_SFT GENMASK(19, 16) +#define DL2_SW_CLEAR_BUF_EMPTY_SFT 15 +#define DL2_SW_CLEAR_BUF_EMPTY_MASK 0x1 +#define DL2_SW_CLEAR_BUF_EMPTY_MASK_SFT BIT(15) +#define DL2_PBUF_SIZE_SFT 12 +#define DL2_PBUF_SIZE_MASK 0x3 +#define DL2_PBUF_SIZE_MASK_SFT GENMASK(13, 12) +#define DL2_MONO_SFT 8 +#define DL2_MONO_MASK 0x1 +#define DL2_MONO_MASK_SFT BIT(8) +#define DL2_NORMAL_MODE_SFT 5 +#define DL2_NORMAL_MODE_MASK 0x1 +#define DL2_NORMAL_MODE_MASK_SFT BIT(5) +#define DL2_HALIGN_SFT 4 +#define DL2_HALIGN_MASK 0x1 +#define DL2_HALIGN_MASK_SFT BIT(4) +#define DL2_HD_MODE_SFT 0 +#define DL2_HD_MODE_MASK 0x3 +#define DL2_HD_MODE_MASK_SFT GENMASK(1, 0) + +/* AFE_DL3_CON0 */ +#define DL3_MODE_SFT 24 +#define DL3_MODE_MASK 0xf +#define DL3_MODE_MASK_SFT GENMASK(27, 24) +#define DL3_MINLEN_SFT 20 +#define DL3_MINLEN_MASK 0xf +#define DL3_MINLEN_MASK_SFT GENMASK(23, 20) +#define DL3_MAXLEN_SFT 16 +#define DL3_MAXLEN_MASK 0xf +#define DL3_MAXLEN_MASK_SFT GENMASK(19, 16) +#define DL3_SW_CLEAR_BUF_EMPTY_SFT 15 +#define DL3_SW_CLEAR_BUF_EMPTY_MASK 0x1 +#define DL3_SW_CLEAR_BUF_EMPTY_MASK_SFT BIT(15) +#define DL3_PBUF_SIZE_SFT 12 +#define DL3_PBUF_SIZE_MASK 0x3 +#define DL3_PBUF_SIZE_MASK_SFT GENMASK(13, 12) +#define DL3_MONO_SFT 8 +#define DL3_MONO_MASK 0x1 +#define DL3_MONO_MASK_SFT BIT(8) +#define DL3_NORMAL_MODE_SFT 5 +#define DL3_NORMAL_MODE_MASK 0x1 +#define DL3_NORMAL_MODE_MASK_SFT BIT(5) +#define DL3_HALIGN_SFT 4 +#define DL3_HALIGN_MASK 0x1 +#define DL3_HALIGN_MASK_SFT BIT(4) +#define DL3_HD_MODE_SFT 0 +#define DL3_HD_MODE_MASK 0x3 +#define DL3_HD_MODE_MASK_SFT GENMASK(1, 0) + +/* AFE_DL4_CON0 */ +#define DL4_MODE_SFT 24 +#define DL4_MODE_MASK 0xf +#define DL4_MODE_MASK_SFT GENMASK(27, 24) +#define DL4_MINLEN_SFT 20 +#define DL4_MINLEN_MASK 0xf +#define DL4_MINLEN_MASK_SFT GENMASK(23, 20) +#define DL4_MAXLEN_SFT 16 +#define DL4_MAXLEN_MASK 0xf +#define DL4_MAXLEN_MASK_SFT GENMASK(19, 16) +#define DL4_SW_CLEAR_BUF_EMPTY_SFT 15 +#define DL4_SW_CLEAR_BUF_EMPTY_MASK 0x1 +#define DL4_SW_CLEAR_BUF_EMPTY_MASK_SFT BIT(15) +#define DL4_PBUF_SIZE_SFT 12 +#define DL4_PBUF_SIZE_MASK 0x3 +#define DL4_PBUF_SIZE_MASK_SFT GENMASK(13, 12) +#define DL4_MONO_SFT 8 +#define DL4_MONO_MASK 0x1 +#define DL4_MONO_MASK_SFT BIT(8) +#define DL4_NORMAL_MODE_SFT 5 +#define DL4_NORMAL_MODE_MASK 0x1 +#define DL4_NORMAL_MODE_MASK_SFT BIT(5) +#define DL4_HALIGN_SFT 4 +#define DL4_HALIGN_MASK 0x1 +#define DL4_HALIGN_MASK_SFT BIT(4) +#define DL4_HD_MODE_SFT 0 +#define DL4_HD_MODE_MASK 0x3 +#define DL4_HD_MODE_MASK_SFT GENMASK(1, 0) + +/* AFE_DL5_CON0 */ +#define DL5_MODE_SFT 24 +#define DL5_MODE_MASK 0xf +#define DL5_MODE_MASK_SFT GENMASK(27, 24) +#define DL5_MINLEN_SFT 20 +#define DL5_MINLEN_MASK 0xf +#define DL5_MINLEN_MASK_SFT GENMASK(23, 20) +#define DL5_MAXLEN_SFT 16 +#define DL5_MAXLEN_MASK 0xf +#define DL5_MAXLEN_MASK_SFT GENMASK(19, 16) +#define DL5_SW_CLEAR_BUF_EMPTY_SFT 15 +#define DL5_SW_CLEAR_BUF_EMPTY_MASK 0x1 +#define DL5_SW_CLEAR_BUF_EMPTY_MASK_SFT BIT(15) +#define DL5_PBUF_SIZE_SFT 12 +#define DL5_PBUF_SIZE_MASK 0x3 +#define DL5_PBUF_SIZE_MASK_SFT GENMASK(13, 12) +#define DL5_MONO_SFT 8 +#define DL5_MONO_MASK 0x1 +#define DL5_MONO_MASK_SFT BIT(8) +#define DL5_NORMAL_MODE_SFT 5 +#define DL5_NORMAL_MODE_MASK 0x1 +#define DL5_NORMAL_MODE_MASK_SFT BIT(5) +#define DL5_HALIGN_SFT 4 +#define DL5_HALIGN_MASK 0x1 +#define DL5_HALIGN_MASK_SFT BIT(4) +#define DL5_HD_MODE_SFT 0 +#define DL5_HD_MODE_MASK 0x3 +#define DL5_HD_MODE_MASK_SFT GENMASK(1, 0) + +/* AFE_DL6_CON0 */ +#define DL6_MODE_SFT 24 +#define DL6_MODE_MASK 0xf +#define DL6_MODE_MASK_SFT GENMASK(27, 24) +#define DL6_MINLEN_SFT 20 +#define DL6_MINLEN_MASK 0xf +#define DL6_MINLEN_MASK_SFT GENMASK(23, 20) +#define DL6_MAXLEN_SFT 16 +#define DL6_MAXLEN_MASK 0xf +#define DL6_MAXLEN_MASK_SFT GENMASK(19, 16) +#define DL6_SW_CLEAR_BUF_EMPTY_SFT 15 +#define DL6_SW_CLEAR_BUF_EMPTY_MASK 0x1 +#define DL6_SW_CLEAR_BUF_EMPTY_MASK_SFT BIT(15) +#define DL6_PBUF_SIZE_SFT 12 +#define DL6_PBUF_SIZE_MASK 0x3 +#define DL6_PBUF_SIZE_MASK_SFT GENMASK(13, 12) +#define DL6_MONO_SFT 8 +#define DL6_MONO_MASK 0x1 +#define DL6_MONO_MASK_SFT BIT(8) +#define DL6_NORMAL_MODE_SFT 5 +#define DL6_NORMAL_MODE_MASK 0x1 +#define DL6_NORMAL_MODE_MASK_SFT BIT(5) +#define DL6_HALIGN_SFT 4 +#define DL6_HALIGN_MASK 0x1 +#define DL6_HALIGN_MASK_SFT BIT(4) +#define DL6_HD_MODE_SFT 0 +#define DL6_HD_MODE_MASK 0x3 +#define DL6_HD_MODE_MASK_SFT GENMASK(1, 0) + +/* AFE_DL7_CON0 */ +#define DL7_MODE_SFT 24 +#define DL7_MODE_MASK 0xf +#define DL7_MODE_MASK_SFT GENMASK(27, 24) +#define DL7_MINLEN_SFT 20 +#define DL7_MINLEN_MASK 0xf +#define DL7_MINLEN_MASK_SFT GENMASK(23, 20) +#define DL7_MAXLEN_SFT 16 +#define DL7_MAXLEN_MASK 0xf +#define DL7_MAXLEN_MASK_SFT GENMASK(19, 16) +#define DL7_SW_CLEAR_BUF_EMPTY_SFT 15 +#define DL7_SW_CLEAR_BUF_EMPTY_MASK 0x1 +#define DL7_SW_CLEAR_BUF_EMPTY_MASK_SFT BIT(15) +#define DL7_PBUF_SIZE_SFT 12 +#define DL7_PBUF_SIZE_MASK 0x3 +#define DL7_PBUF_SIZE_MASK_SFT GENMASK(13, 12) +#define DL7_MONO_SFT 8 +#define DL7_MONO_MASK 0x1 +#define DL7_MONO_MASK_SFT BIT(8) +#define DL7_NORMAL_MODE_SFT 5 +#define DL7_NORMAL_MODE_MASK 0x1 +#define DL7_NORMAL_MODE_MASK_SFT BIT(5) +#define DL7_HALIGN_SFT 4 +#define DL7_HALIGN_MASK 0x1 +#define DL7_HALIGN_MASK_SFT BIT(4) +#define DL7_HD_MODE_SFT 0 +#define DL7_HD_MODE_MASK 0x3 +#define DL7_HD_MODE_MASK_SFT GENMASK(1, 0) + +/* AFE_DL8_CON0 */ +#define DL8_MODE_SFT 24 +#define DL8_MODE_MASK 0xf +#define DL8_MODE_MASK_SFT GENMASK(27, 24) +#define DL8_MINLEN_SFT 20 +#define DL8_MINLEN_MASK 0xf +#define DL8_MINLEN_MASK_SFT GENMASK(23, 20) +#define DL8_MAXLEN_SFT 16 +#define DL8_MAXLEN_MASK 0xf +#define DL8_MAXLEN_MASK_SFT GENMASK(19, 16) +#define DL8_SW_CLEAR_BUF_EMPTY_SFT 15 +#define DL8_SW_CLEAR_BUF_EMPTY_MASK 0x1 +#define DL8_SW_CLEAR_BUF_EMPTY_MASK_SFT BIT(15) +#define DL8_PBUF_SIZE_SFT 12 +#define DL8_PBUF_SIZE_MASK 0x3 +#define DL8_PBUF_SIZE_MASK_SFT GENMASK(13, 12) +#define DL8_MONO_SFT 8 +#define DL8_MONO_MASK 0x1 +#define DL8_MONO_MASK_SFT BIT(8) +#define DL8_NORMAL_MODE_SFT 5 +#define DL8_NORMAL_MODE_MASK 0x1 +#define DL8_NORMAL_MODE_MASK_SFT BIT(5) +#define DL8_HALIGN_SFT 4 +#define DL8_HALIGN_MASK 0x1 +#define DL8_HALIGN_MASK_SFT BIT(4) +#define DL8_HD_MODE_SFT 0 +#define DL8_HD_MODE_MASK 0x3 +#define DL8_HD_MODE_MASK_SFT GENMASK(1, 0) + +/* AFE_DL12_CON0 */ +#define DL12_MODE_SFT 24 +#define DL12_MODE_MASK 0xf +#define DL12_MODE_MASK_SFT GENMASK(27, 24) +#define DL12_MINLEN_SFT 20 +#define DL12_MINLEN_MASK 0xf +#define DL12_MINLEN_MASK_SFT GENMASK(23, 20) +#define DL12_MAXLEN_SFT 16 +#define DL12_MAXLEN_MASK 0xf +#define DL12_MAXLEN_MASK_SFT GENMASK(19, 16) +#define DL12_SW_CLEAR_BUF_EMPTY_SFT 15 +#define DL12_SW_CLEAR_BUF_EMPTY_MASK 0x1 +#define DL12_SW_CLEAR_BUF_EMPTY_MASK_SFT BIT(15) +#define DL12_PBUF_SIZE_SFT 12 +#define DL12_PBUF_SIZE_MASK 0x3 +#define DL12_PBUF_SIZE_MASK_SFT GENMASK(13, 12) +#define DL12_4CH_EN_SFT 11 +#define DL12_4CH_EN_MASK 0x1 +#define DL12_4CH_EN_MASK_SFT BIT(11) +#define DL12_MONO_SFT 8 +#define DL12_MONO_MASK 0x1 +#define DL12_MONO_MASK_SFT BIT(8) +#define DL12_NORMAL_MODE_SFT 5 +#define DL12_NORMAL_MODE_MASK 0x1 +#define DL12_NORMAL_MODE_MASK_SFT BIT(5) +#define DL12_HALIGN_SFT 4 +#define DL12_HALIGN_MASK 0x1 +#define DL12_HALIGN_MASK_SFT BIT(4) +#define DL12_HD_MODE_SFT 0 +#define DL12_HD_MODE_MASK 0x3 +#define DL12_HD_MODE_MASK_SFT GENMASK(1, 0) + +/* AFE_AWB_CON0 */ +#define AWB_MODE_SFT 24 +#define AWB_MODE_MASK 0xf +#define AWB_MODE_MASK_SFT GENMASK(27, 24) +#define AWB_SW_CLEAR_BUF_FULL_SFT 15 +#define AWB_SW_CLEAR_BUF_FULL_MASK 0x1 +#define AWB_SW_CLEAR_BUF_FULL_MASK_SFT BIT(15) +#define AWB_R_MONO_SFT 9 +#define AWB_R_MONO_MASK 0x1 +#define AWB_R_MONO_MASK_SFT BIT(9) +#define AWB_MONO_SFT 8 +#define AWB_MONO_MASK 0x1 +#define AWB_MONO_MASK_SFT BIT(8) +#define AWB_WR_SIGN_SFT 6 +#define AWB_WR_SIGN_MASK 0x1 +#define AWB_WR_SIGN_MASK_SFT BIT(6) +#define AWB_NORMAL_MODE_SFT 5 +#define AWB_NORMAL_MODE_MASK 0x1 +#define AWB_NORMAL_MODE_MASK_SFT BIT(5) +#define AWB_HALIGN_SFT 4 +#define AWB_HALIGN_MASK 0x1 +#define AWB_HALIGN_MASK_SFT BIT(4) +#define AWB_HD_MODE_SFT 0 +#define AWB_HD_MODE_MASK 0x3 +#define AWB_HD_MODE_MASK_SFT GENMASK(1, 0) + +/* AFE_AWB2_CON0 */ +#define AWB2_MODE_SFT 24 +#define AWB2_MODE_MASK 0xf +#define AWB2_MODE_MASK_SFT GENMASK(27, 24) +#define AWB2_SW_CLEAR_BUF_FULL_SFT 15 +#define AWB2_SW_CLEAR_BUF_FULL_MASK 0x1 +#define AWB2_SW_CLEAR_BUF_FULL_MASK_SFT BIT(15) +#define AWB2_R_MONO_SFT 9 +#define AWB2_R_MONO_MASK 0x1 +#define AWB2_R_MONO_MASK_SFT BIT(9) +#define AWB2_MONO_SFT 8 +#define AWB2_MONO_MASK 0x1 +#define AWB2_MONO_MASK_SFT BIT(8) +#define AWB2_WR_SIGN_SFT 6 +#define AWB2_WR_SIGN_MASK 0x1 +#define AWB2_WR_SIGN_MASK_SFT BIT(6) +#define AWB2_NORMAL_MODE_SFT 5 +#define AWB2_NORMAL_MODE_MASK 0x1 +#define AWB2_NORMAL_MODE_MASK_SFT BIT(5) +#define AWB2_HALIGN_SFT 4 +#define AWB2_HALIGN_MASK 0x1 +#define AWB2_HALIGN_MASK_SFT BIT(4) +#define AWB2_HD_MODE_SFT 0 +#define AWB2_HD_MODE_MASK 0x3 +#define AWB2_HD_MODE_MASK_SFT GENMASK(1, 0) + +/* AFE_VUL_CON0 */ +#define VUL_MODE_SFT 24 +#define VUL_MODE_MASK 0xf +#define VUL_MODE_MASK_SFT GENMASK(27, 24) +#define VUL_SW_CLEAR_BUF_FULL_SFT 15 +#define VUL_SW_CLEAR_BUF_FULL_MASK 0x1 +#define VUL_SW_CLEAR_BUF_FULL_MASK_SFT BIT(15) +#define VUL_R_MONO_SFT 9 +#define VUL_R_MONO_MASK 0x1 +#define VUL_R_MONO_MASK_SFT BIT(9) +#define VUL_MONO_SFT 8 +#define VUL_MONO_MASK 0x1 +#define VUL_MONO_MASK_SFT BIT(8) +#define VUL_WR_SIGN_SFT 6 +#define VUL_WR_SIGN_MASK 0x1 +#define VUL_WR_SIGN_MASK_SFT BIT(6) +#define VUL_NORMAL_MODE_SFT 5 +#define VUL_NORMAL_MODE_MASK 0x1 +#define VUL_NORMAL_MODE_MASK_SFT BIT(5) +#define VUL_HALIGN_SFT 4 +#define VUL_HALIGN_MASK 0x1 +#define VUL_HALIGN_MASK_SFT BIT(4) +#define VUL_HD_MODE_SFT 0 +#define VUL_HD_MODE_MASK 0x3 +#define VUL_HD_MODE_MASK_SFT GENMASK(1, 0) + +/* AFE_VUL12_CON0 */ +#define VUL12_MODE_SFT 24 +#define VUL12_MODE_MASK 0xf +#define VUL12_MODE_MASK_SFT GENMASK(27, 24) +#define VUL12_SW_CLEAR_BUF_FULL_SFT 15 +#define VUL12_SW_CLEAR_BUF_FULL_MASK 0x1 +#define VUL12_SW_CLEAR_BUF_FULL_MASK_SFT BIT(15) +#define VUL12_4CH_EN_SFT 11 +#define VUL12_4CH_EN_MASK 0x1 +#define VUL12_4CH_EN_MASK_SFT BIT(11) +#define VUL12_R_MONO_SFT 9 +#define VUL12_R_MONO_MASK 0x1 +#define VUL12_R_MONO_MASK_SFT BIT(9) +#define VUL12_MONO_SFT 8 +#define VUL12_MONO_MASK 0x1 +#define VUL12_MONO_MASK_SFT BIT(8) +#define VUL12_WR_SIGN_SFT 6 +#define VUL12_WR_SIGN_MASK 0x1 +#define VUL12_WR_SIGN_MASK_SFT BIT(6) +#define VUL12_NORMAL_MODE_SFT 5 +#define VUL12_NORMAL_MODE_MASK 0x1 +#define VUL12_NORMAL_MODE_MASK_SFT BIT(5) +#define VUL12_HALIGN_SFT 4 +#define VUL12_HALIGN_MASK 0x1 +#define VUL12_HALIGN_MASK_SFT BIT(4) +#define VUL12_HD_MODE_SFT 0 +#define VUL12_HD_MODE_MASK 0x3 +#define VUL12_HD_MODE_MASK_SFT GENMASK(1, 0) + +/* AFE_VUL2_CON0 */ +#define VUL2_MODE_SFT 24 +#define VUL2_MODE_MASK 0xf +#define VUL2_MODE_MASK_SFT GENMASK(27, 24) +#define VUL2_SW_CLEAR_BUF_FULL_SFT 15 +#define VUL2_SW_CLEAR_BUF_FULL_MASK 0x1 +#define VUL2_SW_CLEAR_BUF_FULL_MASK_SFT BIT(15) +#define VUL2_R_MONO_SFT 9 +#define VUL2_R_MONO_MASK 0x1 +#define VUL2_R_MONO_MASK_SFT BIT(9) +#define VUL2_MONO_SFT 8 +#define VUL2_MONO_MASK 0x1 +#define VUL2_MONO_MASK_SFT BIT(8) +#define VUL2_WR_SIGN_SFT 6 +#define VUL2_WR_SIGN_MASK 0x1 +#define VUL2_WR_SIGN_MASK_SFT BIT(6) +#define VUL2_NORMAL_MODE_SFT 5 +#define VUL2_NORMAL_MODE_MASK 0x1 +#define VUL2_NORMAL_MODE_MASK_SFT BIT(5) +#define VUL2_HALIGN_SFT 4 +#define VUL2_HALIGN_MASK 0x1 +#define VUL2_HALIGN_MASK_SFT BIT(4) +#define VUL2_HD_MODE_SFT 0 +#define VUL2_HD_MODE_MASK 0x3 +#define VUL2_HD_MODE_MASK_SFT GENMASK(1, 0) + +/* AFE_VUL3_CON0 */ +#define VUL3_MODE_SFT 24 +#define VUL3_MODE_MASK 0xf +#define VUL3_MODE_MASK_SFT GENMASK(27, 24) +#define VUL3_SW_CLEAR_BUF_FULL_SFT 15 +#define VUL3_SW_CLEAR_BUF_FULL_MASK 0x1 +#define VUL3_SW_CLEAR_BUF_FULL_MASK_SFT BIT(15) +#define VUL3_R_MONO_SFT 9 +#define VUL3_R_MONO_MASK 0x1 +#define VUL3_R_MONO_MASK_SFT BIT(9) +#define VUL3_MONO_SFT 8 +#define VUL3_MONO_MASK 0x1 +#define VUL3_MONO_MASK_SFT BIT(8) +#define VUL3_WR_SIGN_SFT 6 +#define VUL3_WR_SIGN_MASK 0x1 +#define VUL3_WR_SIGN_MASK_SFT BIT(6) +#define VUL3_NORMAL_MODE_SFT 5 +#define VUL3_NORMAL_MODE_MASK 0x1 +#define VUL3_NORMAL_MODE_MASK_SFT BIT(5) +#define VUL3_HALIGN_SFT 4 +#define VUL3_HALIGN_MASK 0x1 +#define VUL3_HALIGN_MASK_SFT BIT(4) +#define VUL3_HD_MODE_SFT 0 +#define VUL3_HD_MODE_MASK 0x3 +#define VUL3_HD_MODE_MASK_SFT GENMASK(1, 0) + +/* AFE_VUL4_CON0 */ +#define VUL4_MODE_SFT 24 +#define VUL4_MODE_MASK 0xf +#define VUL4_MODE_MASK_SFT GENMASK(27, 24) +#define VUL4_SW_CLEAR_BUF_FULL_SFT 15 +#define VUL4_SW_CLEAR_BUF_FULL_MASK 0x1 +#define VUL4_SW_CLEAR_BUF_FULL_MASK_SFT BIT(15) +#define VUL4_R_MONO_SFT 9 +#define VUL4_R_MONO_MASK 0x1 +#define VUL4_R_MONO_MASK_SFT BIT(9) +#define VUL4_MONO_SFT 8 +#define VUL4_MONO_MASK 0x1 +#define VUL4_MONO_MASK_SFT BIT(8) +#define VUL4_WR_SIGN_SFT 6 +#define VUL4_WR_SIGN_MASK 0x1 +#define VUL4_WR_SIGN_MASK_SFT BIT(6) +#define VUL4_NORMAL_MODE_SFT 5 +#define VUL4_NORMAL_MODE_MASK 0x1 +#define VUL4_NORMAL_MODE_MASK_SFT BIT(5) +#define VUL4_HALIGN_SFT 4 +#define VUL4_HALIGN_MASK 0x1 +#define VUL4_HALIGN_MASK_SFT BIT(4) +#define VUL4_HD_MODE_SFT 0 +#define VUL4_HD_MODE_MASK 0x3 +#define VUL4_HD_MODE_MASK_SFT GENMASK(1, 0) + +/* AFE_VUL5_CON0 */ +#define VUL5_MODE_SFT 24 +#define VUL5_MODE_MASK 0xf +#define VUL5_MODE_MASK_SFT GENMASK(27, 24) +#define VUL5_SW_CLEAR_BUF_FULL_SFT 15 +#define VUL5_SW_CLEAR_BUF_FULL_MASK 0x1 +#define VUL5_SW_CLEAR_BUF_FULL_MASK_SFT BIT(15) +#define VUL5_R_MONO_SFT 9 +#define VUL5_R_MONO_MASK 0x1 +#define VUL5_R_MONO_MASK_SFT BIT(9) +#define VUL5_MONO_SFT 8 +#define VUL5_MONO_MASK 0x1 +#define VUL5_MONO_MASK_SFT BIT(8) +#define VUL5_WR_SIGN_SFT 6 +#define VUL5_WR_SIGN_MASK 0x1 +#define VUL5_WR_SIGN_MASK_SFT BIT(6) +#define VUL5_NORMAL_MODE_SFT 5 +#define VUL5_NORMAL_MODE_MASK 0x1 +#define VUL5_NORMAL_MODE_MASK_SFT BIT(5) +#define VUL5_HALIGN_SFT 4 +#define VUL5_HALIGN_MASK 0x1 +#define VUL5_HALIGN_MASK_SFT BIT(4) +#define VUL5_HD_MODE_SFT 0 +#define VUL5_HD_MODE_MASK 0x3 +#define VUL5_HD_MODE_MASK_SFT GENMASK(1, 0) + +/* AFE_VUL6_CON0 */ +#define VUL6_MODE_SFT 24 +#define VUL6_MODE_MASK 0xf +#define VUL6_MODE_MASK_SFT GENMASK(27, 24) +#define VUL6_SW_CLEAR_BUF_FULL_SFT 15 +#define VUL6_SW_CLEAR_BUF_FULL_MASK 0x1 +#define VUL6_SW_CLEAR_BUF_FULL_MASK_SFT BIT(15) +#define VUL6_R_MONO_SFT 9 +#define VUL6_R_MONO_MASK 0x1 +#define VUL6_R_MONO_MASK_SFT BIT(9) +#define VUL6_MONO_SFT 8 +#define VUL6_MONO_MASK 0x1 +#define VUL6_MONO_MASK_SFT BIT(8) +#define VUL6_WR_SIGN_SFT 6 +#define VUL6_WR_SIGN_MASK 0x1 +#define VUL6_WR_SIGN_MASK_SFT BIT(6) +#define VUL6_NORMAL_MODE_SFT 5 +#define VUL6_NORMAL_MODE_MASK 0x1 +#define VUL6_NORMAL_MODE_MASK_SFT BIT(5) +#define VUL6_HALIGN_SFT 4 +#define VUL6_HALIGN_MASK 0x1 +#define VUL6_HALIGN_MASK_SFT BIT(4) +#define VUL6_HD_MODE_SFT 0 +#define VUL6_HD_MODE_MASK 0x3 +#define VUL6_HD_MODE_MASK_SFT GENMASK(1, 0) + +/* AFE_DAI_CON0 */ +#define DAI_MODE_SFT 24 +#define DAI_MODE_MASK 0x3 +#define DAI_MODE_MASK_SFT GENMASK(25, 24) +#define DAI_SW_CLEAR_BUF_FULL_SFT 15 +#define DAI_SW_CLEAR_BUF_FULL_MASK 0x1 +#define DAI_SW_CLEAR_BUF_FULL_MASK_SFT BIT(15) +#define DAI_DUPLICATE_WR_SFT 10 +#define DAI_DUPLICATE_WR_MASK 0x1 +#define DAI_DUPLICATE_WR_MASK_SFT BIT(10) +#define DAI_MONO_SFT 8 +#define DAI_MONO_MASK 0x1 +#define DAI_MONO_MASK_SFT BIT(8) +#define DAI_WR_SIGN_SFT 6 +#define DAI_WR_SIGN_MASK 0x1 +#define DAI_WR_SIGN_MASK_SFT BIT(6) +#define DAI_NORMAL_MODE_SFT 5 +#define DAI_NORMAL_MODE_MASK 0x1 +#define DAI_NORMAL_MODE_MASK_SFT BIT(5) +#define DAI_HALIGN_SFT 4 +#define DAI_HALIGN_MASK 0x1 +#define DAI_HALIGN_MASK_SFT BIT(4) +#define DAI_HD_MODE_SFT 0 +#define DAI_HD_MODE_MASK 0x3 +#define DAI_HD_MODE_MASK_SFT GENMASK(1, 0) + +/* AFE_MOD_DAI_CON0 */ +#define MOD_DAI_MODE_SFT 24 +#define MOD_DAI_MODE_MASK 0x3 +#define MOD_DAI_MODE_MASK_SFT GENMASK(25, 24) +#define MOD_DAI_SW_CLEAR_BUF_FULL_SFT 15 +#define MOD_DAI_SW_CLEAR_BUF_FULL_MASK 0x1 +#define MOD_DAI_SW_CLEAR_BUF_FULL_MASK_SFT BIT(15) +#define MOD_DAI_DUPLICATE_WR_SFT 10 +#define MOD_DAI_DUPLICATE_WR_MASK 0x1 +#define MOD_DAI_DUPLICATE_WR_MASK_SFT BIT(10) +#define MOD_DAI_MONO_SFT 8 +#define MOD_DAI_MONO_MASK 0x1 +#define MOD_DAI_MONO_MASK_SFT BIT(8) +#define MOD_DAI_WR_SIGN_SFT 6 +#define MOD_DAI_WR_SIGN_MASK 0x1 +#define MOD_DAI_WR_SIGN_MASK_SFT BIT(6) +#define MOD_DAI_NORMAL_MODE_SFT 5 +#define MOD_DAI_NORMAL_MODE_MASK 0x1 +#define MOD_DAI_NORMAL_MODE_MASK_SFT BIT(5) +#define MOD_DAI_HALIGN_SFT 4 +#define MOD_DAI_HALIGN_MASK 0x1 +#define MOD_DAI_HALIGN_MASK_SFT BIT(4) +#define MOD_DAI_HD_MODE_SFT 0 +#define MOD_DAI_HD_MODE_MASK 0x3 +#define MOD_DAI_HD_MODE_MASK_SFT GENMASK(1, 0) + +/* AFE_DAI2_CON0 */ +#define DAI2_MODE_SFT 24 +#define DAI2_MODE_MASK 0xf +#define DAI2_MODE_MASK_SFT GENMASK(27, 24) +#define DAI2_SW_CLEAR_BUF_FULL_SFT 15 +#define DAI2_SW_CLEAR_BUF_FULL_MASK 0x1 +#define DAI2_SW_CLEAR_BUF_FULL_MASK_SFT BIT(15) +#define DAI2_DUPLICATE_WR_SFT 10 +#define DAI2_DUPLICATE_WR_MASK 0x1 +#define DAI2_DUPLICATE_WR_MASK_SFT BIT(10) +#define DAI2_MONO_SFT 8 +#define DAI2_MONO_MASK 0x1 +#define DAI2_MONO_MASK_SFT BIT(8) +#define DAI2_WR_SIGN_SFT 6 +#define DAI2_WR_SIGN_MASK 0x1 +#define DAI2_WR_SIGN_MASK_SFT BIT(6) +#define DAI2_NORMAL_MODE_SFT 5 +#define DAI2_NORMAL_MODE_MASK 0x1 +#define DAI2_NORMAL_MODE_MASK_SFT BIT(5) +#define DAI2_HALIGN_SFT 4 +#define DAI2_HALIGN_MASK 0x1 +#define DAI2_HALIGN_MASK_SFT BIT(4) +#define DAI2_HD_MODE_SFT 0 +#define DAI2_HD_MODE_MASK 0x3 +#define DAI2_HD_MODE_MASK_SFT GENMASK(1, 0) + +/* AFE_MEMIF_CON0 */ +#define CPU_COMPACT_MODE_SFT 2 +#define CPU_COMPACT_MODE_MASK_SFT BIT(2) +#define CPU_HD_ALIGN_SFT 1 +#define CPU_HD_ALIGN_MASK_SFT BIT(1) +#define SYSRAM_SIGN_SFT 0 +#define SYSRAM_SIGN_MASK_SFT BIT(0) + +/* AFE_IRQ_MCU_CON0 */ +#define IRQ31_MCU_ON_SFT 31 +#define IRQ31_MCU_ON_MASK 0x1 +#define IRQ31_MCU_ON_MASK_SFT BIT(31) +#define IRQ26_MCU_ON_SFT 26 +#define IRQ26_MCU_ON_MASK 0x1 +#define IRQ26_MCU_ON_MASK_SFT BIT(26) +#define IRQ25_MCU_ON_SFT 25 +#define IRQ25_MCU_ON_MASK 0x1 +#define IRQ25_MCU_ON_MASK_SFT BIT(25) +#define IRQ24_MCU_ON_SFT 24 +#define IRQ24_MCU_ON_MASK 0x1 +#define IRQ24_MCU_ON_MASK_SFT BIT(24) +#define IRQ23_MCU_ON_SFT 23 +#define IRQ23_MCU_ON_MASK 0x1 +#define IRQ23_MCU_ON_MASK_SFT BIT(23) +#define IRQ22_MCU_ON_SFT 22 +#define IRQ22_MCU_ON_MASK 0x1 +#define IRQ22_MCU_ON_MASK_SFT BIT(22) +#define IRQ21_MCU_ON_SFT 21 +#define IRQ21_MCU_ON_MASK 0x1 +#define IRQ21_MCU_ON_MASK_SFT BIT(21) +#define IRQ20_MCU_ON_SFT 20 +#define IRQ20_MCU_ON_MASK 0x1 +#define IRQ20_MCU_ON_MASK_SFT BIT(20) +#define IRQ19_MCU_ON_SFT 19 +#define IRQ19_MCU_ON_MASK 0x1 +#define IRQ19_MCU_ON_MASK_SFT BIT(19) +#define IRQ18_MCU_ON_SFT 18 +#define IRQ18_MCU_ON_MASK 0x1 +#define IRQ18_MCU_ON_MASK_SFT BIT(18) +#define IRQ17_MCU_ON_SFT 17 +#define IRQ17_MCU_ON_MASK 0x1 +#define IRQ17_MCU_ON_MASK_SFT BIT(17) +#define IRQ16_MCU_ON_SFT 16 +#define IRQ16_MCU_ON_MASK 0x1 +#define IRQ16_MCU_ON_MASK_SFT BIT(16) +#define IRQ15_MCU_ON_SFT 15 +#define IRQ15_MCU_ON_MASK 0x1 +#define IRQ15_MCU_ON_MASK_SFT BIT(15) +#define IRQ14_MCU_ON_SFT 14 +#define IRQ14_MCU_ON_MASK 0x1 +#define IRQ14_MCU_ON_MASK_SFT BIT(14) +#define IRQ13_MCU_ON_SFT 13 +#define IRQ13_MCU_ON_MASK 0x1 +#define IRQ13_MCU_ON_MASK_SFT BIT(13) +#define IRQ12_MCU_ON_SFT 12 +#define IRQ12_MCU_ON_MASK 0x1 +#define IRQ12_MCU_ON_MASK_SFT BIT(12) +#define IRQ11_MCU_ON_SFT 11 +#define IRQ11_MCU_ON_MASK 0x1 +#define IRQ11_MCU_ON_MASK_SFT BIT(11) +#define IRQ10_MCU_ON_SFT 10 +#define IRQ10_MCU_ON_MASK 0x1 +#define IRQ10_MCU_ON_MASK_SFT BIT(10) +#define IRQ9_MCU_ON_SFT 9 +#define IRQ9_MCU_ON_MASK 0x1 +#define IRQ9_MCU_ON_MASK_SFT BIT(9) +#define IRQ8_MCU_ON_SFT 8 +#define IRQ8_MCU_ON_MASK 0x1 +#define IRQ8_MCU_ON_MASK_SFT BIT(8) +#define IRQ7_MCU_ON_SFT 7 +#define IRQ7_MCU_ON_MASK 0x1 +#define IRQ7_MCU_ON_MASK_SFT BIT(7) +#define IRQ6_MCU_ON_SFT 6 +#define IRQ6_MCU_ON_MASK 0x1 +#define IRQ6_MCU_ON_MASK_SFT BIT(6) +#define IRQ5_MCU_ON_SFT 5 +#define IRQ5_MCU_ON_MASK 0x1 +#define IRQ5_MCU_ON_MASK_SFT BIT(5) +#define IRQ4_MCU_ON_SFT 4 +#define IRQ4_MCU_ON_MASK 0x1 +#define IRQ4_MCU_ON_MASK_SFT BIT(4) +#define IRQ3_MCU_ON_SFT 3 +#define IRQ3_MCU_ON_MASK 0x1 +#define IRQ3_MCU_ON_MASK_SFT BIT(3) +#define IRQ2_MCU_ON_SFT 2 +#define IRQ2_MCU_ON_MASK 0x1 +#define IRQ2_MCU_ON_MASK_SFT BIT(2) +#define IRQ1_MCU_ON_SFT 1 +#define IRQ1_MCU_ON_MASK 0x1 +#define IRQ1_MCU_ON_MASK_SFT BIT(1) +#define IRQ0_MCU_ON_SFT 0 +#define IRQ0_MCU_ON_MASK 0x1 +#define IRQ0_MCU_ON_MASK_SFT BIT(0) + +/* AFE_IRQ_MCU_CON1 */ +#define IRQ7_MCU_MODE_SFT 28 +#define IRQ7_MCU_MODE_MASK 0xf +#define IRQ7_MCU_MODE_MASK_SFT GENMASK(31, 28) +#define IRQ6_MCU_MODE_SFT 24 +#define IRQ6_MCU_MODE_MASK 0xf +#define IRQ6_MCU_MODE_MASK_SFT GENMASK(27, 24) +#define IRQ5_MCU_MODE_SFT 20 +#define IRQ5_MCU_MODE_MASK 0xf +#define IRQ5_MCU_MODE_MASK_SFT GENMASK(23, 20) +#define IRQ4_MCU_MODE_SFT 16 +#define IRQ4_MCU_MODE_MASK 0xf +#define IRQ4_MCU_MODE_MASK_SFT GENMASK(19, 16) +#define IRQ3_MCU_MODE_SFT 12 +#define IRQ3_MCU_MODE_MASK 0xf +#define IRQ3_MCU_MODE_MASK_SFT GENMASK(15, 12) +#define IRQ2_MCU_MODE_SFT 8 +#define IRQ2_MCU_MODE_MASK 0xf +#define IRQ2_MCU_MODE_MASK_SFT GENMASK(11, 8) +#define IRQ1_MCU_MODE_SFT 4 +#define IRQ1_MCU_MODE_MASK 0xf +#define IRQ1_MCU_MODE_MASK_SFT GENMASK(7, 4) +#define IRQ0_MCU_MODE_SFT 0 +#define IRQ0_MCU_MODE_MASK 0xf +#define IRQ0_MCU_MODE_MASK_SFT GENMASK(3, 0) + +/* AFE_IRQ_MCU_CON2 */ +#define IRQ15_MCU_MODE_SFT 28 +#define IRQ15_MCU_MODE_MASK 0xf +#define IRQ15_MCU_MODE_MASK_SFT GENMASK(31, 28) +#define IRQ14_MCU_MODE_SFT 24 +#define IRQ14_MCU_MODE_MASK 0xf +#define IRQ14_MCU_MODE_MASK_SFT GENMASK(27, 24) +#define IRQ13_MCU_MODE_SFT 20 +#define IRQ13_MCU_MODE_MASK 0xf +#define IRQ13_MCU_MODE_MASK_SFT GENMASK(23, 20) +#define IRQ12_MCU_MODE_SFT 16 +#define IRQ12_MCU_MODE_MASK 0xf +#define IRQ12_MCU_MODE_MASK_SFT GENMASK(19, 16) +#define IRQ11_MCU_MODE_SFT 12 +#define IRQ11_MCU_MODE_MASK 0xf +#define IRQ11_MCU_MODE_MASK_SFT GENMASK(15, 12) +#define IRQ10_MCU_MODE_SFT 8 +#define IRQ10_MCU_MODE_MASK 0xf +#define IRQ10_MCU_MODE_MASK_SFT GENMASK(11, 8) +#define IRQ9_MCU_MODE_SFT 4 +#define IRQ9_MCU_MODE_MASK 0xf +#define IRQ9_MCU_MODE_MASK_SFT GENMASK(7, 4) +#define IRQ8_MCU_MODE_SFT 0 +#define IRQ8_MCU_MODE_MASK 0xf +#define IRQ8_MCU_MODE_MASK_SFT GENMASK(3, 0) + +/* AFE_IRQ_MCU_CON3 */ +#define IRQ23_MCU_MODE_SFT 28 +#define IRQ23_MCU_MODE_MASK 0xf +#define IRQ23_MCU_MODE_MASK_SFT GENMASK(31, 28) +#define IRQ22_MCU_MODE_SFT 24 +#define IRQ22_MCU_MODE_MASK 0xf +#define IRQ22_MCU_MODE_MASK_SFT GENMASK(27, 24) +#define IRQ21_MCU_MODE_SFT 20 +#define IRQ21_MCU_MODE_MASK 0xf +#define IRQ21_MCU_MODE_MASK_SFT GENMASK(23, 20) +#define IRQ20_MCU_MODE_SFT 16 +#define IRQ20_MCU_MODE_MASK 0xf +#define IRQ20_MCU_MODE_MASK_SFT GENMASK(19, 16) +#define IRQ19_MCU_MODE_SFT 12 +#define IRQ19_MCU_MODE_MASK 0xf +#define IRQ19_MCU_MODE_MASK_SFT GENMASK(15, 12) +#define IRQ18_MCU_MODE_SFT 8 +#define IRQ18_MCU_MODE_MASK 0xf +#define IRQ18_MCU_MODE_MASK_SFT GENMASK(11, 8) +#define IRQ17_MCU_MODE_SFT 4 +#define IRQ17_MCU_MODE_MASK 0xf +#define IRQ17_MCU_MODE_MASK_SFT GENMASK(7, 4) +#define IRQ16_MCU_MODE_SFT 0 +#define IRQ16_MCU_MODE_MASK 0xf +#define IRQ16_MCU_MODE_MASK_SFT GENMASK(3, 0) + +/* AFE_IRQ_MCU_CON4 */ +#define IRQ26_MCU_MODE_SFT 8 +#define IRQ26_MCU_MODE_MASK 0xf +#define IRQ26_MCU_MODE_MASK_SFT GENMASK(11, 8) +#define IRQ25_MCU_MODE_SFT 4 +#define IRQ25_MCU_MODE_MASK 0xf +#define IRQ25_MCU_MODE_MASK_SFT GENMASK(7, 4) +#define IRQ24_MCU_MODE_SFT 0 +#define IRQ24_MCU_MODE_MASK 0xf +#define IRQ24_MCU_MODE_MASK_SFT GENMASK(3, 0) + +/* AFE_IRQ_MCU_CLR */ +#define IRQ31_MCU_CLR_SFT 31 +#define IRQ31_MCU_CLR_MASK_SFT BIT(31) +#define IRQ26_MCU_CLR_SFT 26 +#define IRQ26_MCU_CLR_MASK_SFT BIT(26) +#define IRQ25_MCU_CLR_SFT 25 +#define IRQ25_MCU_CLR_MASK_SFT BIT(25) +#define IRQ24_MCU_CLR_SFT 24 +#define IRQ24_MCU_CLR_MASK_SFT BIT(24) +#define IRQ23_MCU_CLR_SFT 23 +#define IRQ23_MCU_CLR_MASK_SFT BIT(23) +#define IRQ22_MCU_CLR_SFT 22 +#define IRQ22_MCU_CLR_MASK_SFT BIT(22) +#define IRQ21_MCU_CLR_SFT 21 +#define IRQ21_MCU_CLR_MASK_SFT BIT(21) +#define IRQ20_MCU_CLR_SFT 20 +#define IRQ20_MCU_CLR_MASK_SFT BIT(20) +#define IRQ19_MCU_CLR_SFT 19 +#define IRQ19_MCU_CLR_MASK_SFT BIT(19) +#define IRQ18_MCU_CLR_SFT 18 +#define IRQ18_MCU_CLR_MASK_SFT BIT(18) +#define IRQ17_MCU_CLR_SFT 17 +#define IRQ17_MCU_CLR_MASK_SFT BIT(17) +#define IRQ16_MCU_CLR_SFT 16 +#define IRQ16_MCU_CLR_MASK_SFT BIT(16) +#define IRQ15_MCU_CLR_SFT 15 +#define IRQ15_MCU_CLR_MASK_SFT BIT(15) +#define IRQ14_MCU_CLR_SFT 14 +#define IRQ14_MCU_CLR_MASK_SFT BIT(14) +#define IRQ13_MCU_CLR_SFT 13 +#define IRQ13_MCU_CLR_MASK_SFT BIT(13) +#define IRQ12_MCU_CLR_SFT 12 +#define IRQ12_MCU_CLR_MASK_SFT BIT(12) +#define IRQ11_MCU_CLR_SFT 11 +#define IRQ11_MCU_CLR_MASK_SFT BIT(11) +#define IRQ10_MCU_CLR_SFT 10 +#define IRQ10_MCU_CLR_MASK_SFT BIT(10) +#define IRQ9_MCU_CLR_SFT 9 +#define IRQ9_MCU_CLR_MASK_SFT BIT(9) +#define IRQ8_MCU_CLR_SFT 8 +#define IRQ8_MCU_CLR_MASK_SFT BIT(8) +#define IRQ7_MCU_CLR_SFT 7 +#define IRQ7_MCU_CLR_MASK_SFT BIT(7) +#define IRQ6_MCU_CLR_SFT 6 +#define IRQ6_MCU_CLR_MASK_SFT BIT(6) +#define IRQ5_MCU_CLR_SFT 5 +#define IRQ5_MCU_CLR_MASK_SFT BIT(5) +#define IRQ4_MCU_CLR_SFT 4 +#define IRQ4_MCU_CLR_MASK_SFT BIT(4) +#define IRQ3_MCU_CLR_SFT 3 +#define IRQ3_MCU_CLR_MASK_SFT BIT(3) +#define IRQ2_MCU_CLR_SFT 2 +#define IRQ2_MCU_CLR_MASK_SFT BIT(2) +#define IRQ1_MCU_CLR_SFT 1 +#define IRQ1_MCU_CLR_MASK_SFT BIT(1) +#define IRQ0_MCU_CLR_SFT 0 +#define IRQ0_MCU_CLR_MASK_SFT BIT(0) + +/* AFE_IRQ_MCU_EN */ +#define IRQ31_MCU_EN_SFT 31 +#define IRQ30_MCU_EN_SFT 30 +#define IRQ29_MCU_EN_SFT 29 +#define IRQ28_MCU_EN_SFT 28 +#define IRQ27_MCU_EN_SFT 27 +#define IRQ26_MCU_EN_SFT 26 +#define IRQ25_MCU_EN_SFT 25 +#define IRQ24_MCU_EN_SFT 24 +#define IRQ23_MCU_EN_SFT 23 +#define IRQ22_MCU_EN_SFT 22 +#define IRQ21_MCU_EN_SFT 21 +#define IRQ20_MCU_EN_SFT 20 +#define IRQ19_MCU_EN_SFT 19 +#define IRQ18_MCU_EN_SFT 18 +#define IRQ17_MCU_EN_SFT 17 +#define IRQ16_MCU_EN_SFT 16 +#define IRQ15_MCU_EN_SFT 15 +#define IRQ14_MCU_EN_SFT 14 +#define IRQ13_MCU_EN_SFT 13 +#define IRQ12_MCU_EN_SFT 12 +#define IRQ11_MCU_EN_SFT 11 +#define IRQ10_MCU_EN_SFT 10 +#define IRQ9_MCU_EN_SFT 9 +#define IRQ8_MCU_EN_SFT 8 +#define IRQ7_MCU_EN_SFT 7 +#define IRQ6_MCU_EN_SFT 6 +#define IRQ5_MCU_EN_SFT 5 +#define IRQ4_MCU_EN_SFT 4 +#define IRQ3_MCU_EN_SFT 3 +#define IRQ2_MCU_EN_SFT 2 +#define IRQ1_MCU_EN_SFT 1 +#define IRQ0_MCU_EN_SFT 0 + +/* AFE_IRQ_MCU_SCP_EN */ +#define IRQ31_MCU_SCP_EN_SFT 31 +#define IRQ30_MCU_SCP_EN_SFT 30 +#define IRQ29_MCU_SCP_EN_SFT 29 +#define IRQ28_MCU_SCP_EN_SFT 28 +#define IRQ27_MCU_SCP_EN_SFT 27 +#define IRQ26_MCU_SCP_EN_SFT 26 +#define IRQ25_MCU_SCP_EN_SFT 25 +#define IRQ24_MCU_SCP_EN_SFT 24 +#define IRQ23_MCU_SCP_EN_SFT 23 +#define IRQ22_MCU_SCP_EN_SFT 22 +#define IRQ21_MCU_SCP_EN_SFT 21 +#define IRQ20_MCU_SCP_EN_SFT 20 +#define IRQ19_MCU_SCP_EN_SFT 19 +#define IRQ18_MCU_SCP_EN_SFT 18 +#define IRQ17_MCU_SCP_EN_SFT 17 +#define IRQ16_MCU_SCP_EN_SFT 16 +#define IRQ15_MCU_SCP_EN_SFT 15 +#define IRQ14_MCU_SCP_EN_SFT 14 +#define IRQ13_MCU_SCP_EN_SFT 13 +#define IRQ12_MCU_SCP_EN_SFT 12 +#define IRQ11_MCU_SCP_EN_SFT 11 +#define IRQ10_MCU_SCP_EN_SFT 10 +#define IRQ9_MCU_SCP_EN_SFT 9 +#define IRQ8_MCU_SCP_EN_SFT 8 +#define IRQ7_MCU_SCP_EN_SFT 7 +#define IRQ6_MCU_SCP_EN_SFT 6 +#define IRQ5_MCU_SCP_EN_SFT 5 +#define IRQ4_MCU_SCP_EN_SFT 4 +#define IRQ3_MCU_SCP_EN_SFT 3 +#define IRQ2_MCU_SCP_EN_SFT 2 +#define IRQ1_MCU_SCP_EN_SFT 1 +#define IRQ0_MCU_SCP_EN_SFT 0 + +/* AFE_IRQ_MCU_DSP_EN */ +#define IRQ31_MCU_DSP_EN_SFT 31 +#define IRQ30_MCU_DSP_EN_SFT 30 +#define IRQ29_MCU_DSP_EN_SFT 29 +#define IRQ28_MCU_DSP_EN_SFT 28 +#define IRQ27_MCU_DSP_EN_SFT 27 +#define IRQ26_MCU_DSP_EN_SFT 26 +#define IRQ25_MCU_DSP_EN_SFT 25 +#define IRQ24_MCU_DSP_EN_SFT 24 +#define IRQ23_MCU_DSP_EN_SFT 23 +#define IRQ22_MCU_DSP_EN_SFT 22 +#define IRQ21_MCU_DSP_EN_SFT 21 +#define IRQ20_MCU_DSP_EN_SFT 20 +#define IRQ19_MCU_DSP_EN_SFT 19 +#define IRQ18_MCU_DSP_EN_SFT 18 +#define IRQ17_MCU_DSP_EN_SFT 17 +#define IRQ16_MCU_DSP_EN_SFT 16 +#define IRQ15_MCU_DSP_EN_SFT 15 +#define IRQ14_MCU_DSP_EN_SFT 14 +#define IRQ13_MCU_DSP_EN_SFT 13 +#define IRQ12_MCU_DSP_EN_SFT 12 +#define IRQ11_MCU_DSP_EN_SFT 11 +#define IRQ10_MCU_DSP_EN_SFT 10 +#define IRQ9_MCU_DSP_EN_SFT 9 +#define IRQ8_MCU_DSP_EN_SFT 8 +#define IRQ7_MCU_DSP_EN_SFT 7 +#define IRQ6_MCU_DSP_EN_SFT 6 +#define IRQ5_MCU_DSP_EN_SFT 5 +#define IRQ4_MCU_DSP_EN_SFT 4 +#define IRQ3_MCU_DSP_EN_SFT 3 +#define IRQ2_MCU_DSP_EN_SFT 2 +#define IRQ1_MCU_DSP_EN_SFT 1 +#define IRQ0_MCU_DSP_EN_SFT 0 + +/* AFE_AUD_PAD_TOP */ +#define AUD_PAD_TOP_MON_SFT 15 +#define AUD_PAD_TOP_MON_MASK_SFT GENMASK(31, 15) +#define AUD_PAD_TOP_FIFO_RSP_SFT 4 +#define AUD_PAD_TOP_FIFO_RSP_MASK_SFT GENMASK(7, 4) +#define RG_RX_PROTOCOL2_SFT 3 +#define RG_RX_PROTOCOL2_MASK_SFT BIT(3) +#define RESERVDED_01_SFT 1 +#define RESERVDED_01_MASK_SFT GENMASK(2, 1) +#define RG_RX_FIFO_ON_SFT 0 +#define RG_RX_FIFO_ON_MASK_SFT BIT(0) + +/* AFE_ADDA_MTKAIF_SYNCWORD_CFG */ +#define RG_ADDA6_MTKAIF_RX_SYNC_WORD2_DISABLE_SFT 23 +#define RG_ADDA6_MTKAIF_RX_SYNC_WORD2_DISABLE_MASK_SFT BIT(23) + +/* AFE_ADDA_MTKAIF_RX_CFG0 */ +#define MTKAIF_RXIF_VOICE_MODE_SFT 20 +#define MTKAIF_RXIF_VOICE_MODE_MASK_SFT GENMASK(23, 20) +#define MTKAIF_RXIF_DETECT_ON_SFT 16 +#define MTKAIF_RXIF_DETECT_ON_MASK_SFT BIT(16) +#define MTKAIF_RXIF_DATA_BIT_SFT 8 +#define MTKAIF_RXIF_DATA_BIT_MASK_SFT GENMASK(10, 8) +#define MTKAIF_RXIF_FIFO_RSP_SFT 4 +#define MTKAIF_RXIF_FIFO_RSP_MASK_SFT GENMASK(6, 4) +#define MTKAIF_RXIF_DATA_MODE_SFT 0 +#define MTKAIF_RXIF_DATA_MODE_MASK_SFT BIT(0) + +/* GENERAL_ASRC_MODE */ +#define GENERAL2_ASRCOUT_MODE_SFT 12 +#define GENERAL2_ASRCOUT_MODE_MASK 0xf +#define GENERAL2_ASRCOUT_MODE_MASK_SFT GENMASK(15, 12) +#define GENERAL2_ASRCIN_MODE_SFT 8 +#define GENERAL2_ASRCIN_MODE_MASK 0xf +#define GENERAL2_ASRCIN_MODE_MASK_SFT GENMASK(11, 8) +#define GENERAL1_ASRCOUT_MODE_SFT 4 +#define GENERAL1_ASRCOUT_MODE_MASK 0xf +#define GENERAL1_ASRCOUT_MODE_MASK_SFT GENMASK(7, 4) +#define GENERAL1_ASRCIN_MODE_SFT 0 +#define GENERAL1_ASRCIN_MODE_MASK 0xf +#define GENERAL1_ASRCIN_MODE_MASK_SFT GENMASK(3, 0) + +/* GENERAL_ASRC_EN_ON */ +#define GENERAL2_ASRC_EN_ON_SFT 1 +#define GENERAL2_ASRC_EN_ON_MASK_SFT BIT(1) +#define GENERAL1_ASRC_EN_ON_SFT 0 +#define GENERAL1_ASRC_EN_ON_MASK_SFT BIT(0) + +/* AFE_GENERAL1_ASRC_2CH_CON0 */ +#define G_SRC_CHSET_STR_CLR_SFT 4 +#define G_SRC_CHSET_STR_CLR_MASK_SFT BIT(4) +#define G_SRC_CHSET_ON_SFT 2 +#define G_SRC_CHSET_ON_MASK_SFT BIT(2) +#define G_SRC_COEFF_SRAM_CTRL_SFT 1 +#define G_SRC_COEFF_SRAM_CTRL_MASK_SFT BIT(1) +#define G_SRC_ASM_ON_SFT 0 +#define G_SRC_ASM_ON_MASK_SFT BIT(0) + +/* AFE_GENERAL1_ASRC_2CH_CON3 */ +#define G_SRC_ASM_FREQ_4_SFT 0 +#define G_SRC_ASM_FREQ_4_MASK_SFT GENMASK(23, 0) + +/* AFE_GENERAL1_ASRC_2CH_CON4 */ +#define G_SRC_ASM_FREQ_5_SFT 0 +#define G_SRC_ASM_FREQ_5_MASK_SFT GENMASK(23, 0) + +/* AFE_GENERAL1_ASRC_2CH_CON13 */ +#define G_SRC_COEFF_SRAM_ADR_SFT 0 +#define G_SRC_COEFF_SRAM_ADR_MASK_SFT GENMASK(5, 0) + +/* AFE_GENERAL1_ASRC_2CH_CON2 */ +#define G_SRC_CHSET_O16BIT_SFT 19 +#define G_SRC_CHSET_O16BIT_MASK_SFT BIT(19) +#define G_SRC_CHSET_CLR_IIR_HISTORY_SFT 17 +#define G_SRC_CHSET_CLR_IIR_HISTORY_MASK_SFT BIT(17) +#define G_SRC_CHSET_IS_MONO_SFT 16 +#define G_SRC_CHSET_IS_MONO_MASK_SFT BIT(16) +#define G_SRC_CHSET_IIR_EN_SFT 11 +#define G_SRC_CHSET_IIR_EN_MASK_SFT BIT(11) +#define G_SRC_CHSET_IIR_STAGE_SFT 8 +#define G_SRC_CHSET_IIR_STAGE_MASK_SFT GENMASK(10, 8) +#define G_SRC_CHSET_STR_CLR_RU_SFT 5 +#define G_SRC_CHSET_STR_CLR_RU_MASK_SFT BIT(5) +#define G_SRC_CHSET_ON_SFT 2 +#define G_SRC_CHSET_ON_MASK_SFT BIT(2) +#define G_SRC_COEFF_SRAM_CTRL_SFT 1 +#define G_SRC_COEFF_SRAM_CTRL_MASK_SFT BIT(1) +#define G_SRC_ASM_ON_SFT 0 +#define G_SRC_ASM_ON_MASK_SFT BIT(0) + +/* AFE_ADDA_DL_SDM_DITHER_CON */ +#define AFE_DL_SDM_DITHER_64TAP_EN_SFT 20 +#define AFE_DL_SDM_DITHER_64TAP_EN_MASK_SFT BIT(20) +#define AFE_DL_SDM_DITHER_EN_SFT 16 +#define AFE_DL_SDM_DITHER_EN_MASK_SFT BIT(16) +#define AFE_DL_SDM_DITHER_GAIN_SFT 0 +#define AFE_DL_SDM_DITHER_GAIN_MASK_SFT GENMASK(7, 0) + +/* AFE_ADDA_DL_SDM_AUTO_RESET_CON */ +#define SDM_AUTO_RESET_TEST_ON_SFT 31 +#define SDM_AUTO_RESET_TEST_ON_MASK_SFT BIT(31) +#define AFE_DL_USE_NEW_2ND_SDM_SFT 28 +#define AFE_DL_USE_NEW_2ND_SDM_MASK_SFT BIT(28) +#define SDM_AUTO_RESET_COUNT_TH_SFT 0 +#define SDM_AUTO_RESET_COUNT_TH_MASK_SFT GENMASK(23, 0) + +/* AFE_ASRC_2CH_CON0 */ +#define CON0_CHSET_STR_CLR_SFT 4 +#define CON0_CHSET_STR_CLR_MASK_SFT BIT(4) +#define CON0_ASM_ON_SFT 0 +#define CON0_ASM_ON_MASK_SFT BIT(0) + +/* AFE_ASRC_2CH_CON5 */ +#define CALI_EN_SFT 0 +#define CALI_EN_MASK_SFT BIT(0) + +/* FPGA_CFG4 */ +#define IRQ_COUNTER_SFT 3 +#define IRQ_COUNTER_MASK_SFT GENMASK(31, 3) +#define IRQ_CLK_COUNTER_CLEAN_SFT 2 +#define IRQ_CLK_COUNTER_CLEAN_MASK_SFT BIT(2) +#define IRQ_CLK_COUNTER_PAUSE_SFT 1 +#define IRQ_CLK_COUNTER_PAUSE_MASK_SFT BIT(1) +#define IRQ_CLK_COUNTER_ON_SFT 0 +#define IRQ_CLK_COUNTER_ON_MASK_SFT BIT(0) + +/* FPGA_CFG5 */ +#define WR_MSTR_ON_SFT 16 +#define WR_MSTR_ON_MASK_SFT GENMASK(28, 16) +#define WR_AG_SEL_SFT 0 +#define WR_AG_SEL_MASK_SFT GENMASK(12, 0) + +/* FPGA_CFG6 */ +#define WR_MSTR_REQ_REAL_SFT 16 +#define WR_MSTR_REQ_REAL_MASK_SFT GENMASK(28, 16) +#define WR_MSTR_REQ_IN_SFT 0 +#define WR_MSTR_REQ_IN_MASK_SFT GENMASK(12, 0) + +/* FPGA_CFG7 */ +#define MEM1_WDATA_MON0_SFT 0 +#define MEM1_WDATA_MON0_MASK_SFT GENMASK(31, 0) + +/* FPGA_CFG8 */ +#define MEM1_WDATA_MON1_SFT 0 +#define MEM1_WDATA_MON1_MASK_SFT GENMASK(31, 0) + +/* FPGA_CFG9 */ +#define MEM_WE_SFT 31 +#define MEM_WE_MASK_SFT BIT(31) +#define AFE_HREADY_SFT 30 +#define AFE_HREADY_MASK_SFT BIT(30) +#define MEM_WR_REQ_SFT 29 +#define MEM_WR_REQ_MASK_SFT BIT(29) +#define WR_AG_REG_MON_SFT 16 +#define WR_AG_REG_MON_MASK_SFT GENMASK(28, 16) +#define HCLK_CK_SFT 15 +#define HCLK_CK_MASK_SFT BIT(15) +#define MEM_RD_REQ_SFT 14 +#define MEM_RD_REQ_MASK_SFT BIT(14) +#define RD_AG_REQ_MON_SFT 0 +#define RD_AG_REQ_MON_MASK_SFT GENMASK(13, 0) + +/* FPGA_CFG10 */ +#define MEM_BYTE_0_SFT 0 +#define MEM_BYTE_0_MASK_SFT GENMASK(31, 0) + +/* FPGA_CFG11 */ +#define MEM_BYTE_1_SFT 0 +#define MEM_BYTE_1_MASK_SFT GENMASK(31, 0) + +/* FPGA_CFG12 */ +#define RDATA_CNT_SFT 30 +#define RDATA_CNT_MASK_SFT GENMASK(31, 30) +#define MS2_HREADY_SFT 29 +#define MS2_HREADY_MASK_SFT BIT(29) +#define MS1_HREADY_SFT 28 +#define MS1_HREADY_MASK_SFT BIT(28) +#define AG_SEL_SFT 0 +#define AG_SEL_MASK_SFT GENMASK(25, 0) + +/* FPGA_CFG13 */ +#define AFE_ST_SFT 27 +#define AFE_ST_MASK_SFT GENMASK(31, 27) +#define AG_IN_SERVICE_SFT 0 +#define AG_IN_SERVICE_MASK_SFT GENMASK(25, 0) + +/* ETDM_IN1_CON0 */ +#define ETDM_IN1_CON0_REG_ETDM_IN_EN_SFT 0 +#define ETDM_IN1_CON0_REG_ETDM_IN_EN_MASK_SFT BIT(0) +#define ETDM_IN1_CON0_REG_SYNC_MODE_SFT 1 +#define ETDM_IN1_CON0_REG_SYNC_MODE_MASK_SFT BIT(1) +#define ETDM_IN1_CON0_REG_LSB_FIRST_SFT 3 +#define ETDM_IN1_CON0_REG_LSB_FIRST_MASK_SFT BIT(3) +#define ETDM_IN1_CON0_REG_SOFT_RST_SFT 4 +#define ETDM_IN1_CON0_REG_SOFT_RST_MASK_SFT BIT(4) +#define ETDM_IN1_CON0_REG_SLAVE_MODE_SFT 5 +#define ETDM_IN1_CON0_REG_SLAVE_MODE_MASK_SFT BIT(5) +#define ETDM_IN1_CON0_REG_FMT_SFT 6 +#define ETDM_IN1_CON0_REG_FMT_MASK_SFT GENMASK(8, 6) +#define ETDM_IN1_CON0_REG_LRCK_EDGE_SEL_SFT 10 +#define ETDM_IN1_CON0_REG_LRCK_EDGE_SEL_MASK_SFT BIT(10) +#define ETDM_IN1_CON0_REG_BIT_LENGTH_SFT 11 +#define ETDM_IN1_CON0_REG_BIT_LENGTH_MASK_SFT GENMASK(15, 11) +#define ETDM_IN1_CON0_REG_WORD_LENGTH_SFT 16 +#define ETDM_IN1_CON0_REG_WORD_LENGTH_MASK_SFT GENMASK(20, 16) +#define ETDM_IN1_CON0_REG_CH_NUM_SFT 23 +#define ETDM_IN1_CON0_REG_CH_NUM_MASK_SFT GENMASK(27, 23) +#define ETDM_IN1_CON0_REG_RELATCH_1X_EN_SEL_DOMAIN_SFT 28 +#define ETDM_IN1_CON0_REG_RELATCH_1X_EN_SEL_DOMAIN_MASK_SFT GENMASK(31, 28) +#define ETDM_IN1_CON0_REG_VALID_TOGETHER_SFT 31 +#define ETDM_IN1_CON0_REG_VALID_TOGETHER_MASK_SFT BIT(31) +#define ETDM_IN_CON0_CTRL_MASK 0x1f9ff9e2 + +/* ETDM_IN1_CON1 */ +#define ETDM_IN1_CON1_REG_INITIAL_COUNT_SFT 0 +#define ETDM_IN1_CON1_REG_INITIAL_COUNT_MASK_SFT GENMASK(4, 0) +#define ETDM_IN1_CON1_REG_INITIAL_POINT_SFT 5 +#define ETDM_IN1_CON1_REG_INITIAL_POINT_MASK_SFT GENMASK(9, 5) +#define ETDM_IN1_CON1_REG_LRCK_AUTO_OFF_SFT 10 +#define ETDM_IN1_CON1_REG_LRCK_AUTO_OFF_MASK_SFT BIT(10) +#define ETDM_IN1_CON1_REG_BCK_AUTO_OFF_SFT 11 +#define ETDM_IN1_CON1_REG_BCK_AUTO_OFF_MASK_SFT BIT(11) +#define ETDM_IN1_CON1_REG_INITIAL_LRCK_SFT 13 +#define ETDM_IN1_CON1_REG_INITIAL_LRCK_MASK_SFT BIT(13) +#define ETDM_IN1_CON1_REG_LRCK_RESET_SFT 15 +#define ETDM_IN1_CON1_REG_LRCK_RESET_MASK_SFT BIT(15) +#define ETDM_IN1_CON1_PINMUX_MCLK_CTRL_OE_SFT 16 +#define ETDM_IN1_CON1_PINMUX_MCLK_CTRL_OE_MASK_SFT BIT(16) +#define ETDM_IN1_CON1_REG_OUTPUT_CR_EN_SFT 18 +#define ETDM_IN1_CON1_REG_OUTPUT_CR_EN_MASK_SFT BIT(18) +#define ETDM_IN1_CON1_REG_LR_ALIGN_SFT 19 +#define ETDM_IN1_CON1_REG_LR_ALIGN_MASK_SFT BIT(19) +#define ETDM_IN1_CON1_REG_LRCK_WIDTH_SFT 20 +#define ETDM_IN1_CON1_REG_LRCK_WIDTH_MASK_SFT GENMASK(29, 20) +#define ETDM_IN1_CON1_REG_DIRECT_INPUT_MASTER_BCK_SFT 30 +#define ETDM_IN1_CON1_REG_DIRECT_INPUT_MASTER_BCK_MASK_SFT BIT(30) +#define ETDM_IN1_CON1_REG_LRCK_AUTO_MODE_SFT 31 +#define ETDM_IN1_CON1_REG_LRCK_AUTO_MODE_MASK_SFT BIT(31) +#define ETDM_IN_CON1_CTRL_MASK 0xbff10000 + +/* ETDM_IN1_CON2 */ +#define ETDM_IN1_CON2_REG_UPDATE_POINT_SFT 0 +#define ETDM_IN1_CON2_REG_UPDATE_POINT_MASK_SFT GENMASK(4, 0) +#define ETDM_IN1_CON2_REG_UPDATE_GAP_SFT 5 +#define ETDM_IN1_CON2_REG_UPDATE_GAP_MASK_SFT GENMASK(9, 5) +#define ETDM_IN1_CON2_REG_CLOCK_SOURCE_SEL_SFT 10 +#define ETDM_IN1_CON2_REG_CLOCK_SOURCE_SEL_MASK_SFT GENMASK(12, 10) +#define ETDM_IN1_CON2_REG_AGENT_USE_ETDM_BCK_SFT 13 +#define ETDM_IN1_CON2_REG_AGENT_USE_ETDM_BCK_MASK_SFT BIT(13) +#define ETDM_IN1_CON2_REG_CK_EN_SEL_AUTO_SFT 14 +#define ETDM_IN1_CON2_REG_CK_EN_SEL_AUTO_MASK_SFT BIT(14) +#define ETDM_IN1_CON2_REG_MULTI_IP_ONE_DATA_CH_NUM_SFT 15 +#define ETDM_IN1_CON2_REG_MULTI_IP_ONE_DATA_CH_NUM_MASK_SFT GENMASK(19, 15) +#define ETDM_IN1_CON2_REG_MASK_AUTO_SFT 20 +#define ETDM_IN1_CON2_REG_MASK_AUTO_MASK_SFT BIT(20) +#define ETDM_IN1_CON2_REG_MASK_NUM_SFT 21 +#define ETDM_IN1_CON2_REG_MASK_NUM_MASK_SFT GENMASK(25, 21) +#define ETDM_IN1_CON2_REG_UPDATE_POINT_AUTO_SFT 26 +#define ETDM_IN1_CON2_REG_UPDATE_POINT_AUTO_MASK_SFT BIT(26) +#define ETDM_IN1_CON2_REG_SDATA_DELAY_0P5T_EN_SFT 27 +#define ETDM_IN1_CON2_REG_SDATA_DELAY_0P5T_EN_MASK_SFT BIT(27) +#define ETDM_IN1_CON2_REG_SDATA_DELAY_BCK_INV_SFT 28 +#define ETDM_IN1_CON2_REG_SDATA_DELAY_BCK_INV_MASK_SFT BIT(28) +#define ETDM_IN1_CON2_REG_LRCK_DELAY_0P5T_EN_SFT 29 +#define ETDM_IN1_CON2_REG_LRCK_DELAY_0P5T_EN_MASK_SFT BIT(29) +#define ETDM_IN1_CON2_REG_LRCK_DELAY_BCK_INV_SFT 30 +#define ETDM_IN1_CON2_REG_LRCK_DELAY_BCK_INV_MASK_SFT BIT(30) +#define ETDM_IN1_CON2_REG_MULTI_IP_MODE_SFT 31 +#define ETDM_IN1_CON2_REG_MULTI_IP_MODE_MASK_SFT BIT(31) +#define ETDM_IN_CON2_CTRL_MASK 0x800f8000 +#define ETDM_IN_CON2_MULTI_IP_CH(x) (((x) - 1) << 15) +#define ETDM_IN_CON2_MULTI_IP_2CH_MODE BIT(31) + +/* ETDM_IN1_CON3 */ +#define ETDM_IN1_CON3_REG_DISABLE_OUT_0_SFT 0 +#define ETDM_IN1_CON3_REG_DISABLE_OUT_0_MASK_SFT BIT(0) +#define ETDM_IN1_CON3_REG_DISABLE_OUT_1_SFT 1 +#define ETDM_IN1_CON3_REG_DISABLE_OUT_1_MASK_SFT BIT(1) +#define ETDM_IN1_CON3_REG_DISABLE_OUT_2_SFT 2 +#define ETDM_IN1_CON3_REG_DISABLE_OUT_2_MASK_SFT BIT(2) +#define ETDM_IN1_CON3_REG_DISABLE_OUT_3_SFT 3 +#define ETDM_IN1_CON3_REG_DISABLE_OUT_3_MASK_SFT BIT(3) +#define ETDM_IN1_CON3_REG_DISABLE_OUT_4_SFT 4 +#define ETDM_IN1_CON3_REG_DISABLE_OUT_4_MASK_SFT BIT(4) +#define ETDM_IN1_CON3_REG_DISABLE_OUT_5_SFT 5 +#define ETDM_IN1_CON3_REG_DISABLE_OUT_5_MASK_SFT BIT(5) +#define ETDM_IN1_CON3_REG_DISABLE_OUT_6_SFT 6 +#define ETDM_IN1_CON3_REG_DISABLE_OUT_6_MASK_SFT BIT(6) +#define ETDM_IN1_CON3_REG_DISABLE_OUT_7_SFT 7 +#define ETDM_IN1_CON3_REG_DISABLE_OUT_7_MASK_SFT BIT(7) +#define ETDM_IN1_CON3_REG_DISABLE_OUT_8_SFT 8 +#define ETDM_IN1_CON3_REG_DISABLE_OUT_8_MASK_SFT BIT(8) +#define ETDM_IN1_CON3_REG_DISABLE_OUT_9_SFT 9 +#define ETDM_IN1_CON3_REG_DISABLE_OUT_9_MASK_SFT BIT(9) +#define ETDM_IN1_CON3_REG_DISABLE_OUT_10_SFT 10 +#define ETDM_IN1_CON3_REG_DISABLE_OUT_10_MASK_SFT BIT(10) +#define ETDM_IN1_CON3_REG_DISABLE_OUT_11_SFT 11 +#define ETDM_IN1_CON3_REG_DISABLE_OUT_11_MASK_SFT BIT(11) +#define ETDM_IN1_CON3_REG_DISABLE_OUT_12_SFT 12 +#define ETDM_IN1_CON3_REG_DISABLE_OUT_12_MASK_SFT BIT(12) +#define ETDM_IN1_CON3_REG_DISABLE_OUT_13_SFT 13 +#define ETDM_IN1_CON3_REG_DISABLE_OUT_13_MASK_SFT BIT(13) +#define ETDM_IN1_CON3_REG_DISABLE_OUT_14_SFT 14 +#define ETDM_IN1_CON3_REG_DISABLE_OUT_14_MASK_SFT BIT(14) +#define ETDM_IN1_CON3_REG_DISABLE_OUT_15_SFT 15 +#define ETDM_IN1_CON3_REG_DISABLE_OUT_15_MASK_SFT BIT(15) +#define ETDM_IN1_CON3_REG_RJ_DATA_RIGHT_ALIGN_SFT 16 +#define ETDM_IN1_CON3_REG_RJ_DATA_RIGHT_ALIGN_MASK_SFT BIT(16) +#define ETDM_IN1_CON3_REG_MONITOR_SEL_SFT 17 +#define ETDM_IN1_CON3_REG_MONITOR_SEL_MASK_SFT GENMASK(18, 17) +#define ETDM_IN1_CON3_REG_CNT_UPPER_LIMIT_SFT 19 +#define ETDM_IN1_CON3_REG_CNT_UPPER_LIMIT_MASK_SFT GENMASK(24, 19) +#define ETDM_IN1_CON3_REG_COMPACT_SAMPLE_END_DIS_SFT 25 +#define ETDM_IN1_CON3_REG_COMPACT_SAMPLE_END_DIS_MASK_SFT BIT(25) +#define ETDM_IN1_CON3_REG_FS_TIMING_SEL_SFT 26 +#define ETDM_IN1_CON3_REG_FS_TIMING_SEL_MASK_SFT GENMASK(30, 26) +#define ETDM_IN1_CON3_REG_SAMPLE_END_MODE_SFT 31 +#define ETDM_IN1_CON3_REG_SAMPLE_END_MODE_MASK_SFT BIT(31) +#define ETDM_IN_CON3_CTRL_MASK (0x7c000000) +#define ETDM_IN_CON3_FS(x) (((x) & 0x1f) << 26) + +/* ETDM_IN1_CON4 */ +#define ETDM_IN1_CON4_REG_DSD_MODE_SFT 0 +#define ETDM_IN1_CON4_REG_DSD_MODE_MASK_SFT GENMASK(5, 0) +#define ETDM_IN1_CON4_REG_DSD_REPACK_AUTO_MODE_SFT 8 +#define ETDM_IN1_CON4_REG_DSD_REPACK_AUTO_MODE_MASK_SFT BIT(8) +#define ETDM_IN1_CON4_REG_REPACK_WORD_LENGTH_SFT 9 +#define ETDM_IN1_CON4_REG_REPACK_WORD_LENGTH_MASK_SFT GENMASK(10, 9) +#define ETDM_IN1_CON4_REG_ASYNC_RESET_SFT 11 +#define ETDM_IN1_CON4_REG_ASYNC_RESET_MASK_SFT BIT(11) +#define ETDM_IN1_CON4_REG_DSD_CHNUM_SFT 12 +#define ETDM_IN1_CON4_REG_DSD_CHNUM_MASK_SFT GENMASK(15, 12) +#define ETDM_IN1_CON4_REG_SLAVE_BCK_INV_SFT 16 +#define ETDM_IN1_CON4_REG_SLAVE_BCK_INV_MASK_SFT BIT(16) +#define ETDM_IN1_CON4_REG_SLAVE_LRCK_INV_SFT 17 +#define ETDM_IN1_CON4_REG_SLAVE_LRCK_INV_MASK_SFT BIT(17) +#define ETDM_IN1_CON4_REG_MASTER_BCK_INV_SFT 18 +#define ETDM_IN1_CON4_REG_MASTER_BCK_INV_MASK_SFT BIT(18) +#define ETDM_IN1_CON4_REG_MASTER_LRCK_INV_SFT 19 +#define ETDM_IN1_CON4_REG_MASTER_LRCK_INV_MASK_SFT BIT(19) +#define ETDM_IN1_CON4_REG_RELATCH_1X_EN_SEL_SFT 20 +#define ETDM_IN1_CON4_REG_RELATCH_1X_EN_SEL_MASK_SFT GENMASK(24, 20) +#define ETDM_IN1_CON4_REG_SAMPLE_END_POINT_SFT 25 +#define ETDM_IN1_CON4_REG_SAMPLE_END_POINT_MASK_SFT GENMASK(29, 25) +#define ETDM_IN1_CON4_REG_WAIT_LAST_SAMPLE_SFT 30 +#define ETDM_IN1_CON4_REG_WAIT_LAST_SAMPLE_MASK_SFT BIT(30) +#define ETDM_IN1_CON4_REG_MASTER_BCK_FORCE_ON_SFT 31 +#define ETDM_IN1_CON4_REG_MASTER_BCK_FORCE_ON_MASK_SFT BIT(31) +#define ETDM_IN_CON4_CTRL_MASK 0x1ff0000 +#define ETDM_IN_CON4_FS(x) (((x) & 0x1f) << 20) +#define ETDM_IN_CON4_CON0_MASTER_LRCK_INV BIT(19) +#define ETDM_IN_CON4_CON0_MASTER_BCK_INV BIT(18) +#define ETDM_IN_CON4_CON0_SLAVE_LRCK_INV BIT(17) +#define ETDM_IN_CON4_CON0_SLAVE_BCK_INV BIT(16) + +/* ETDM_IN1_CON5 */ +#define ETDM_IN1_CON5_REG_ODD_FLAG_EN_0_SFT 0 +#define ETDM_IN1_CON5_REG_ODD_FLAG_EN_0_MASK_SFT BIT(0) +#define ETDM_IN1_CON5_REG_ODD_FLAG_EN_1_SFT 1 +#define ETDM_IN1_CON5_REG_ODD_FLAG_EN_1_MASK_SFT BIT(1) +#define ETDM_IN1_CON5_REG_ODD_FLAG_EN_2_SFT 2 +#define ETDM_IN1_CON5_REG_ODD_FLAG_EN_2_MASK_SFT BIT(2) +#define ETDM_IN1_CON5_REG_ODD_FLAG_EN_3_SFT 3 +#define ETDM_IN1_CON5_REG_ODD_FLAG_EN_3_MASK_SFT BIT(3) +#define ETDM_IN1_CON5_REG_ODD_FLAG_EN_4_SFT 4 +#define ETDM_IN1_CON5_REG_ODD_FLAG_EN_4_MASK_SFT BIT(4) +#define ETDM_IN1_CON5_REG_ODD_FLAG_EN_5_SFT 5 +#define ETDM_IN1_CON5_REG_ODD_FLAG_EN_5_MASK_SFT BIT(5) +#define ETDM_IN1_CON5_REG_ODD_FLAG_EN_6_SFT 6 +#define ETDM_IN1_CON5_REG_ODD_FLAG_EN_6_MASK_SFT BIT(6) +#define ETDM_IN1_CON5_REG_ODD_FLAG_EN_7_SFT 7 +#define ETDM_IN1_CON5_REG_ODD_FLAG_EN_7_MASK_SFT BIT(7) +#define ETDM_IN1_CON5_REG_ODD_FLAG_EN_8_SFT 8 +#define ETDM_IN1_CON5_REG_ODD_FLAG_EN_8_MASK_SFT BIT(8) +#define ETDM_IN1_CON5_REG_ODD_FLAG_EN_9_SFT 9 +#define ETDM_IN1_CON5_REG_ODD_FLAG_EN_9_MASK_SFT BIT(9) +#define ETDM_IN1_CON5_REG_ODD_FLAG_EN_10_SFT 10 +#define ETDM_IN1_CON5_REG_ODD_FLAG_EN_10_MASK_SFT BIT(10) +#define ETDM_IN1_CON5_REG_ODD_FLAG_EN_11_SFT 11 +#define ETDM_IN1_CON5_REG_ODD_FLAG_EN_11_MASK_SFT BIT(11) +#define ETDM_IN1_CON5_REG_ODD_FLAG_EN_12_SFT 12 +#define ETDM_IN1_CON5_REG_ODD_FLAG_EN_12_MASK_SFT BIT(12) +#define ETDM_IN1_CON5_REG_ODD_FLAG_EN_13_SFT 13 +#define ETDM_IN1_CON5_REG_ODD_FLAG_EN_13_MASK_SFT BIT(13) +#define ETDM_IN1_CON5_REG_ODD_FLAG_EN_14_SFT 14 +#define ETDM_IN1_CON5_REG_ODD_FLAG_EN_14_MASK_SFT BIT(14) +#define ETDM_IN1_CON5_REG_ODD_FLAG_EN_15_SFT 15 +#define ETDM_IN1_CON5_REG_ODD_FLAG_EN_15_MASK_SFT BIT(15) +#define ETDM_IN1_CON5_REG_LR_SWAP_0_SFT 16 +#define ETDM_IN1_CON5_REG_LR_SWAP_0_MASK_SFT BIT(16) +#define ETDM_IN1_CON5_REG_LR_SWAP_1_SFT 17 +#define ETDM_IN1_CON5_REG_LR_SWAP_1_MASK_SFT BIT(17) +#define ETDM_IN1_CON5_REG_LR_SWAP_2_SFT 18 +#define ETDM_IN1_CON5_REG_LR_SWAP_2_MASK_SFT BIT(18) +#define ETDM_IN1_CON5_REG_LR_SWAP_3_SFT 19 +#define ETDM_IN1_CON5_REG_LR_SWAP_3_MASK_SFT BIT(19) +#define ETDM_IN1_CON5_REG_LR_SWAP_4_SFT 20 +#define ETDM_IN1_CON5_REG_LR_SWAP_4_MASK_SFT BIT(20) +#define ETDM_IN1_CON5_REG_LR_SWAP_5_SFT 21 +#define ETDM_IN1_CON5_REG_LR_SWAP_5_MASK_SFT BIT(21) +#define ETDM_IN1_CON5_REG_LR_SWAP_6_SFT 22 +#define ETDM_IN1_CON5_REG_LR_SWAP_6_MASK_SFT BIT(22) +#define ETDM_IN1_CON5_REG_LR_SWAP_7_SFT 23 +#define ETDM_IN1_CON5_REG_LR_SWAP_7_MASK_SFT BIT(23) +#define ETDM_IN1_CON5_REG_LR_SWAP_8_SFT 24 +#define ETDM_IN1_CON5_REG_LR_SWAP_8_MASK_SFT BIT(24) +#define ETDM_IN1_CON5_REG_LR_SWAP_9_SFT 25 +#define ETDM_IN1_CON5_REG_LR_SWAP_9_MASK_SFT BIT(25) +#define ETDM_IN1_CON5_REG_LR_SWAP_10_SFT 26 +#define ETDM_IN1_CON5_REG_LR_SWAP_10_MASK_SFT BIT(26) +#define ETDM_IN1_CON5_REG_LR_SWAP_11_SFT 27 +#define ETDM_IN1_CON5_REG_LR_SWAP_11_MASK_SFT BIT(27) +#define ETDM_IN1_CON5_REG_LR_SWAP_12_SFT 28 +#define ETDM_IN1_CON5_REG_LR_SWAP_12_MASK_SFT BIT(28) +#define ETDM_IN1_CON5_REG_LR_SWAP_13_SFT 29 +#define ETDM_IN1_CON5_REG_LR_SWAP_13_MASK_SFT BIT(29) +#define ETDM_IN1_CON5_REG_LR_SWAP_14_SFT 30 +#define ETDM_IN1_CON5_REG_LR_SWAP_14_MASK_SFT BIT(30) +#define ETDM_IN1_CON5_REG_LR_SWAP_15_SFT 31 +#define ETDM_IN1_CON5_REG_LR_SWAP_15_MASK_SFT BIT(31) + +/* ETDM_IN1_CON6 */ +#define ETDM_IN1_CON6_LCH_DATA_REG_SFT 0 +#define ETDM_IN1_CON6_LCH_DATA_REG_MASK_SFT GENMASK(31, 0) + +/* ETDM_IN1_CON7 */ +#define ETDM_IN1_CON7_RCH_DATA_REG_SFT 0 +#define ETDM_IN1_CON7_RCH_DATA_REG_MASK_SFT GENMASK(31, 0) + +/* ETDM_IN1_CON8 */ +#define ETDM_IN1_CON8_REG_AFIFO_THRESHOLD_SFT 29 +#define ETDM_IN1_CON8_REG_AFIFO_THRESHOLD_MASK_SFT GENMASK(30, 29) +#define ETDM_IN1_CON8_REG_CK_EN_SEL_MANUAL_SFT 16 +#define ETDM_IN1_CON8_REG_CK_EN_SEL_MANUAL_MASK_SFT GENMASK(25, 16) +#define ETDM_IN1_CON8_REG_AFIFO_SW_RESET_SFT 15 +#define ETDM_IN1_CON8_REG_AFIFO_SW_RESET_MASK_SFT BIT(15) +#define ETDM_IN1_CON8_REG_AFIFO_RESET_SEL_SFT 14 +#define ETDM_IN1_CON8_REG_AFIFO_RESET_SEL_MASK_SFT BIT(14) +#define ETDM_IN1_CON8_REG_AFIFO_AUTO_RESET_DIS_SFT 9 +#define ETDM_IN1_CON8_REG_AFIFO_AUTO_RESET_DIS_MASK_SFT BIT(9) +#define ETDM_IN1_CON8_REG_ETDM_USE_AFIFO_SFT 8 +#define ETDM_IN1_CON8_REG_ETDM_USE_AFIFO_MASK_SFT BIT(8) +#define ETDM_IN1_CON8_REG_AFIFO_CLOCK_DOMAIN_SEL_SFT 5 +#define ETDM_IN1_CON8_REG_AFIFO_CLOCK_DOMAIN_SEL_MASK_SFT GENMASK(7, 5) +#define ETDM_IN1_CON8_REG_AFIFO_MODE_SFT 0 +#define ETDM_IN1_CON8_REG_AFIFO_MODE_MASK_SFT GENMASK(4, 0) +#define ETDM_IN_CON8_FS(x) (((x) & 0x1f) << 0) +#define ETDM_IN_CON8_CTRL_MASK 0x13f + +#define AUDIO_TOP_CON0 0x0000 +#define AUDIO_TOP_CON1 0x0004 +#define AUDIO_TOP_CON2 0x0008 +#define AUDIO_TOP_CON3 0x000c +#define AFE_DAC_CON0 0x0010 +#define AFE_I2S_CON 0x0018 +#define AFE_CONN0 0x0020 +#define AFE_CONN1 0x0024 +#define AFE_CONN2 0x0028 +#define AFE_CONN3 0x002c +#define AFE_CONN4 0x0030 +#define AFE_I2S_CON1 0x0034 +#define AFE_I2S_CON2 0x0038 +#define AFE_I2S_CON3 0x0040 +#define AFE_CONN5 0x0044 +#define AFE_CONN_24BIT 0x0048 +#define AFE_DL1_CON0 0x004c +#define AFE_DL1_BASE_MSB 0x0050 +#define AFE_DL1_BASE 0x0054 +#define AFE_DL1_CUR_MSB 0x0058 +#define AFE_DL1_CUR 0x005c +#define AFE_DL1_END_MSB 0x0060 +#define AFE_DL1_END 0x0064 +#define AFE_DL2_CON0 0x0068 +#define AFE_DL2_BASE_MSB 0x006c +#define AFE_DL2_BASE 0x0070 +#define AFE_DL2_CUR_MSB 0x0074 +#define AFE_DL2_CUR 0x0078 +#define AFE_DL2_END_MSB 0x007c +#define AFE_DL2_END 0x0080 +#define AFE_DL3_CON0 0x0084 +#define AFE_DL3_BASE_MSB 0x0088 +#define AFE_DL3_BASE 0x008c +#define AFE_DL3_CUR_MSB 0x0090 +#define AFE_DL3_CUR 0x0094 +#define AFE_DL3_END_MSB 0x0098 +#define AFE_DL3_END 0x009c +#define AFE_CONN6 0x00bc +#define AFE_DL4_CON0 0x00cc +#define AFE_DL4_BASE_MSB 0x00d0 +#define AFE_DL4_BASE 0x00d4 +#define AFE_DL4_CUR_MSB 0x00d8 +#define AFE_DL4_CUR 0x00dc +#define AFE_DL4_END_MSB 0x00e0 +#define AFE_DL4_END 0x00e4 +#define AFE_DL12_CON0 0x00e8 +#define AFE_DL12_BASE_MSB 0x00ec +#define AFE_DL12_BASE 0x00f0 +#define AFE_DL12_CUR_MSB 0x00f4 +#define AFE_DL12_CUR 0x00f8 +#define AFE_DL12_END_MSB 0x00fc +#define AFE_DL12_END 0x0100 +#define AFE_ADDA_DL_SRC2_CON0 0x0108 +#define AFE_ADDA_DL_SRC2_CON1 0x010c +#define AFE_ADDA_UL_SRC_CON0 0x0114 +#define AFE_ADDA_UL_SRC_CON1 0x0118 +#define AFE_ADDA_TOP_CON0 0x0120 +#define AFE_ADDA_UL_DL_CON0 0x0124 +#define AFE_ADDA_SRC_DEBUG 0x012c +#define AFE_ADDA_SRC_DEBUG_MON0 0x0130 +#define AFE_ADDA_SRC_DEBUG_MON1 0x0134 +#define AFE_ADDA_UL_SRC_MON0 0x0148 +#define AFE_ADDA_UL_SRC_MON1 0x014c +#define AFE_SECURE_CON0 0x0150 +#define AFE_SRAM_BOUND 0x0154 +#define AFE_SECURE_CON1 0x0158 +#define AFE_SECURE_CONN0 0x015c +#define AFE_VUL_CON0 0x0170 +#define AFE_VUL_BASE_MSB 0x0174 +#define AFE_VUL_BASE 0x0178 +#define AFE_VUL_CUR_MSB 0x017c +#define AFE_VUL_CUR 0x0180 +#define AFE_VUL_END_MSB 0x0184 +#define AFE_VUL_END 0x0188 +#define AFE_SIDETONE_DEBUG 0x01d0 +#define AFE_SIDETONE_MON 0x01d4 +#define AFE_SINEGEN_CON2 0x01dc +#define AFE_SIDETONE_CON0 0x01e0 +#define AFE_SIDETONE_COEFF 0x01e4 +#define AFE_SIDETONE_CON1 0x01e8 +#define AFE_SIDETONE_GAIN 0x01ec +#define AFE_SINEGEN_CON0 0x01f0 +#define AFE_TOP_CON0 0x0200 +#define AFE_VUL2_CON0 0x020c +#define AFE_VUL2_BASE_MSB 0x0210 +#define AFE_VUL2_BASE 0x0214 +#define AFE_VUL2_CUR_MSB 0x0218 +#define AFE_VUL2_CUR 0x021c +#define AFE_VUL2_END_MSB 0x0220 +#define AFE_VUL2_END 0x0224 +#define AFE_VUL3_CON0 0x0228 +#define AFE_VUL3_BASE_MSB 0x022c +#define AFE_VUL3_BASE 0x0230 +#define AFE_VUL3_CUR_MSB 0x0234 +#define AFE_VUL3_CUR 0x0238 +#define AFE_VUL3_END_MSB 0x023c +#define AFE_VUL3_END 0x0240 +#define AFE_BUSY 0x0244 +#define AFE_BUS_CFG 0x0250 +#define AFE_ADDA_PREDIS_CON0 0x0260 +#define AFE_ADDA_PREDIS_CON1 0x0264 +#define AFE_I2S_MON 0x027c +#define AFE_ADDA_IIR_COEF_02_01 0x0290 +#define AFE_ADDA_IIR_COEF_04_03 0x0294 +#define AFE_ADDA_IIR_COEF_06_05 0x0298 +#define AFE_ADDA_IIR_COEF_08_07 0x029c +#define AFE_ADDA_IIR_COEF_10_09 0x02a0 +#define AFE_IRQ_MCU_CON1 0x02e4 +#define AFE_IRQ_MCU_CON2 0x02e8 +#define AFE_DAC_MON 0x02ec +#define AFE_IRQ_MCU_CON3 0x02f0 +#define AFE_IRQ_MCU_CON4 0x02f4 +#define AFE_IRQ_MCU_CNT0 0x0300 +#define AFE_IRQ_MCU_CNT6 0x0304 +#define AFE_IRQ_MCU_CNT8 0x0308 +#define AFE_IRQ_MCU_DSP2_EN 0x030c +#define AFE_IRQ0_MCU_CNT_MON 0x0310 +#define AFE_IRQ6_MCU_CNT_MON 0x0314 +#define AFE_VUL4_CON0 0x0358 +#define AFE_VUL4_BASE_MSB 0x035c +#define AFE_VUL4_BASE 0x0360 +#define AFE_VUL4_CUR_MSB 0x0364 +#define AFE_VUL4_CUR 0x0368 +#define AFE_VUL4_END_MSB 0x036c +#define AFE_VUL4_END 0x0370 +#define AFE_VUL12_CON0 0x0374 +#define AFE_VUL12_BASE_MSB 0x0378 +#define AFE_VUL12_BASE 0x037c +#define AFE_VUL12_CUR_MSB 0x0380 +#define AFE_VUL12_CUR 0x0384 +#define AFE_VUL12_END_MSB 0x0388 +#define AFE_VUL12_END 0x038c +#define AFE_IRQ3_MCU_CNT_MON 0x0398 +#define AFE_IRQ4_MCU_CNT_MON 0x039c +#define AFE_IRQ_MCU_CON0 0x03a0 +#define AFE_IRQ_MCU_STATUS 0x03a4 +#define AFE_IRQ_MCU_CLR 0x03a8 +#define AFE_IRQ_MCU_CNT1 0x03ac +#define AFE_IRQ_MCU_CNT2 0x03b0 +#define AFE_IRQ_MCU_EN 0x03b4 +#define AFE_IRQ_MCU_MON2 0x03b8 +#define AFE_IRQ_MCU_CNT5 0x03bc +#define AFE_IRQ1_MCU_CNT_MON 0x03c0 +#define AFE_IRQ2_MCU_CNT_MON 0x03c4 +#define AFE_IRQ5_MCU_CNT_MON 0x03cc +#define AFE_IRQ_MCU_DSP_EN 0x03d0 +#define AFE_IRQ_MCU_SCP_EN 0x03d4 +#define AFE_IRQ_MCU_CNT7 0x03dc +#define AFE_IRQ7_MCU_CNT_MON 0x03e0 +#define AFE_IRQ_MCU_CNT3 0x03e4 +#define AFE_IRQ_MCU_CNT4 0x03e8 +#define AFE_IRQ_MCU_CNT11 0x03ec +#define AFE_APLL1_TUNER_CFG 0x03f0 +#define AFE_APLL2_TUNER_CFG 0x03f4 +#define AFE_IRQ_MCU_MISS_CLR 0x03f8 +#define AFE_CONN33 0x0408 +#define AFE_IRQ_MCU_CNT12 0x040c +#define AFE_GAIN1_CON0 0x0410 +#define AFE_GAIN1_CON1 0x0414 +#define AFE_GAIN1_CON2 0x0418 +#define AFE_GAIN1_CON3 0x041c +#define AFE_CONN7 0x0420 +#define AFE_GAIN1_CUR 0x0424 +#define AFE_GAIN2_CON0 0x0428 +#define AFE_GAIN2_CON1 0x042c +#define AFE_GAIN2_CON2 0x0430 +#define AFE_GAIN2_CON3 0x0434 +#define AFE_CONN8 0x0438 +#define AFE_GAIN2_CUR 0x043c +#define AFE_CONN9 0x0440 +#define AFE_CONN10 0x0444 +#define AFE_CONN11 0x0448 +#define AFE_CONN12 0x044c +#define AFE_CONN13 0x0450 +#define AFE_CONN14 0x0454 +#define AFE_CONN15 0x0458 +#define AFE_CONN16 0x045c +#define AFE_CONN17 0x0460 +#define AFE_CONN18 0x0464 +#define AFE_CONN19 0x0468 +#define AFE_CONN20 0x046c +#define AFE_CONN21 0x0470 +#define AFE_CONN22 0x0474 +#define AFE_CONN23 0x0478 +#define AFE_CONN24 0x047c +#define AFE_CONN_RS 0x0494 +#define AFE_CONN_DI 0x0498 +#define AFE_CONN25 0x04b0 +#define AFE_CONN26 0x04b4 +#define AFE_CONN27 0x04b8 +#define AFE_CONN28 0x04bc +#define AFE_CONN29 0x04c0 +#define AFE_CONN30 0x04c4 +#define AFE_CONN31 0x04c8 +#define AFE_CONN32 0x04cc +#define AFE_SRAM_DELSEL_CON1 0x04f4 +#define AFE_CONN56 0x0500 +#define AFE_CONN57 0x0504 +#define AFE_CONN58 0x0508 +#define AFE_CONN59 0x050c +#define AFE_CONN56_1 0x0510 +#define AFE_CONN57_1 0x0514 +#define AFE_CONN58_1 0x0518 +#define AFE_CONN59_1 0x051c +#define PCM_INTF_CON1 0x0530 +#define PCM_INTF_CON2 0x0538 +#define PCM2_INTF_CON 0x053c +#define AFE_CM1_CON 0x0550 +#define AFE_CONN34 0x0580 +#define FPGA_CFG0 0x05b0 +#define FPGA_CFG1 0x05b4 +#define FPGA_CFG2 0x05c0 +#define FPGA_CFG3 0x05c4 +#define AUDIO_TOP_DBG_CON 0x05c8 +#define AUDIO_TOP_DBG_MON0 0x05cc +#define AUDIO_TOP_DBG_MON1 0x05d0 +#define AFE_IRQ8_MCU_CNT_MON 0x05e4 +#define AFE_IRQ11_MCU_CNT_MON 0x05e8 +#define AFE_IRQ12_MCU_CNT_MON 0x05ec +#define AFE_IRQ_MCU_CNT9 0x0600 +#define AFE_IRQ_MCU_CNT10 0x0604 +#define AFE_IRQ_MCU_CNT13 0x0608 +#define AFE_IRQ_MCU_CNT14 0x060c +#define AFE_IRQ_MCU_CNT15 0x0610 +#define AFE_IRQ_MCU_CNT16 0x0614 +#define AFE_IRQ_MCU_CNT17 0x0618 +#define AFE_IRQ_MCU_CNT18 0x061c +#define AFE_IRQ_MCU_CNT19 0x0620 +#define AFE_IRQ_MCU_CNT20 0x0624 +#define AFE_IRQ_MCU_CNT21 0x0628 +#define AFE_IRQ_MCU_CNT22 0x062c +#define AFE_IRQ_MCU_CNT23 0x0630 +#define AFE_IRQ_MCU_CNT24 0x0634 +#define AFE_IRQ_MCU_CNT25 0x0638 +#define AFE_IRQ_MCU_CNT26 0x063c +#define AFE_IRQ9_MCU_CNT_MON 0x0660 +#define AFE_IRQ10_MCU_CNT_MON 0x0664 +#define AFE_IRQ13_MCU_CNT_MON 0x0668 +#define AFE_IRQ14_MCU_CNT_MON 0x066c +#define AFE_IRQ15_MCU_CNT_MON 0x0670 +#define AFE_IRQ16_MCU_CNT_MON 0x0674 +#define AFE_IRQ17_MCU_CNT_MON 0x0678 +#define AFE_IRQ18_MCU_CNT_MON 0x067c +#define AFE_IRQ19_MCU_CNT_MON 0x0680 +#define AFE_IRQ20_MCU_CNT_MON 0x0684 +#define AFE_IRQ21_MCU_CNT_MON 0x0688 +#define AFE_IRQ22_MCU_CNT_MON 0x068c +#define AFE_IRQ23_MCU_CNT_MON 0x0690 +#define AFE_IRQ24_MCU_CNT_MON 0x0694 +#define AFE_IRQ25_MCU_CNT_MON 0x0698 +#define AFE_IRQ26_MCU_CNT_MON 0x069c +#define AFE_IRQ31_MCU_CNT_MON 0x06a0 +#define AFE_GENERAL_REG0 0x0800 +#define AFE_GENERAL_REG1 0x0804 +#define AFE_GENERAL_REG2 0x0808 +#define AFE_GENERAL_REG3 0x080c +#define AFE_GENERAL_REG4 0x0810 +#define AFE_GENERAL_REG5 0x0814 +#define AFE_GENERAL_REG6 0x0818 +#define AFE_GENERAL_REG7 0x081c +#define AFE_GENERAL_REG8 0x0820 +#define AFE_GENERAL_REG9 0x0824 +#define AFE_GENERAL_REG10 0x0828 +#define AFE_GENERAL_REG11 0x082c +#define AFE_GENERAL_REG12 0x0830 +#define AFE_GENERAL_REG13 0x0834 +#define AFE_GENERAL_REG14 0x0838 +#define AFE_GENERAL_REG15 0x083c +#define AFE_CBIP_CFG0 0x0840 +#define AFE_CBIP_MON0 0x0844 +#define AFE_CBIP_SLV_MUX_MON0 0x0848 +#define AFE_CBIP_SLV_DECODER_MON0 0x084c +#define AFE_ADDA6_MTKAIF_MON0 0x0854 +#define AFE_ADDA6_MTKAIF_MON1 0x0858 +#define AFE_AWB_CON0 0x085c +#define AFE_AWB_BASE_MSB 0x0860 +#define AFE_AWB_BASE 0x0864 +#define AFE_AWB_CUR_MSB 0x0868 +#define AFE_AWB_CUR 0x086c +#define AFE_AWB_END_MSB 0x0870 +#define AFE_AWB_END 0x0874 +#define AFE_AWB2_CON0 0x0878 +#define AFE_AWB2_BASE_MSB 0x087c +#define AFE_AWB2_BASE 0x0880 +#define AFE_AWB2_CUR_MSB 0x0884 +#define AFE_AWB2_CUR 0x0888 +#define AFE_AWB2_END_MSB 0x088c +#define AFE_AWB2_END 0x0890 +#define AFE_DAI_CON0 0x0894 +#define AFE_DAI_BASE_MSB 0x0898 +#define AFE_DAI_BASE 0x089c +#define AFE_DAI_CUR_MSB 0x08a0 +#define AFE_DAI_CUR 0x08a4 +#define AFE_DAI_END_MSB 0x08a8 +#define AFE_DAI_END 0x08ac +#define AFE_DAI2_CON0 0x08b0 +#define AFE_DAI2_BASE_MSB 0x08b4 +#define AFE_DAI2_BASE 0x08b8 +#define AFE_DAI2_CUR_MSB 0x08bc +#define AFE_DAI2_CUR 0x08c0 +#define AFE_DAI2_END_MSB 0x08c4 +#define AFE_DAI2_END 0x08c8 +#define AFE_MEMIF_CON0 0x08cc +#define AFE_CONN0_1 0x0900 +#define AFE_CONN1_1 0x0904 +#define AFE_CONN2_1 0x0908 +#define AFE_CONN3_1 0x090c +#define AFE_CONN4_1 0x0910 +#define AFE_CONN5_1 0x0914 +#define AFE_CONN6_1 0x0918 +#define AFE_CONN7_1 0x091c +#define AFE_CONN8_1 0x0920 +#define AFE_CONN9_1 0x0924 +#define AFE_CONN10_1 0x0928 +#define AFE_CONN11_1 0x092c +#define AFE_CONN12_1 0x0930 +#define AFE_CONN13_1 0x0934 +#define AFE_CONN14_1 0x0938 +#define AFE_CONN15_1 0x093c +#define AFE_CONN16_1 0x0940 +#define AFE_CONN17_1 0x0944 +#define AFE_CONN18_1 0x0948 +#define AFE_CONN19_1 0x094c +#define AFE_CONN20_1 0x0950 +#define AFE_CONN21_1 0x0954 +#define AFE_CONN22_1 0x0958 +#define AFE_CONN23_1 0x095c +#define AFE_CONN24_1 0x0960 +#define AFE_CONN25_1 0x0964 +#define AFE_CONN26_1 0x0968 +#define AFE_CONN27_1 0x096c +#define AFE_CONN28_1 0x0970 +#define AFE_CONN29_1 0x0974 +#define AFE_CONN30_1 0x0978 +#define AFE_CONN31_1 0x097c +#define AFE_CONN32_1 0x0980 +#define AFE_CONN33_1 0x0984 +#define AFE_CONN34_1 0x0988 +#define AFE_CONN_RS_1 0x098c +#define AFE_CONN_DI_1 0x0990 +#define AFE_CONN_24BIT_1 0x0994 +#define AFE_CONN_REG 0x0998 +#define AFE_CONN35 0x09a0 +#define AFE_CONN36 0x09a4 +#define AFE_CONN37 0x09a8 +#define AFE_CONN38 0x09ac +#define AFE_CONN35_1 0x09b0 +#define AFE_CONN36_1 0x09b4 +#define AFE_CONN37_1 0x09b8 +#define AFE_CONN38_1 0x09bc +#define AFE_CONN39 0x09c0 +#define AFE_CONN40 0x09c4 +#define AFE_CONN41 0x09c8 +#define AFE_CONN42 0x09cc +#define AFE_CONN39_1 0x09e0 +#define AFE_CONN40_1 0x09e4 +#define AFE_CONN41_1 0x09e8 +#define AFE_CONN42_1 0x09ec +#define AFE_I2S_CON4 0x09f8 +#define AFE_CONN60 0x0a64 +#define AFE_CONN61 0x0a68 +#define AFE_CONN62 0x0a6c +#define AFE_CONN63 0x0a70 +#define AFE_CONN64 0x0a74 +#define AFE_CONN65 0x0a78 +#define AFE_CONN66 0x0a7c +#define AFE_ADDA6_TOP_CON0 0x0a80 +#define AFE_ADDA6_UL_SRC_CON0 0x0a84 +#define AFE_ADDA6_UL_SRC_CON1 0x0a88 +#define AFE_ADDA6_SRC_DEBUG 0x0a8c +#define AFE_ADDA6_SRC_DEBUG_MON0 0x0a90 +#define AFE_ADDA6_ULCF_CFG_02_01 0x0aa0 +#define AFE_ADDA6_ULCF_CFG_04_03 0x0aa4 +#define AFE_ADDA6_ULCF_CFG_06_05 0x0aa8 +#define AFE_ADDA6_ULCF_CFG_08_07 0x0aac +#define AFE_ADDA6_ULCF_CFG_10_09 0x0ab0 +#define AFE_ADDA6_ULCF_CFG_12_11 0x0ab4 +#define AFE_ADDA6_ULCF_CFG_14_13 0x0ab8 +#define AFE_ADDA6_ULCF_CFG_16_15 0x0abc +#define AFE_ADDA6_ULCF_CFG_18_17 0x0ac0 +#define AFE_ADDA6_ULCF_CFG_20_19 0x0ac4 +#define AFE_ADDA6_ULCF_CFG_22_21 0x0ac8 +#define AFE_ADDA6_ULCF_CFG_24_23 0x0acc +#define AFE_ADDA6_ULCF_CFG_26_25 0x0ad0 +#define AFE_ADDA6_ULCF_CFG_28_27 0x0ad4 +#define AFE_ADDA6_ULCF_CFG_30_29 0x0ad8 +#define AFE_ADD6A_UL_SRC_MON0 0x0ae4 +#define AFE_ADDA6_UL_SRC_MON1 0x0ae8 +#define AFE_CONN43 0x0af8 +#define AFE_CONN43_1 0x0afc +#define AFE_MOD_DAI_CON0 0x0b00 +#define AFE_MOD_DAI_BASE_MSB 0x0b04 +#define AFE_MOD_DAI_BASE 0x0b08 +#define AFE_MOD_DAI_CUR_MSB 0x0b0c +#define AFE_MOD_DAI_CUR 0x0b10 +#define AFE_MOD_DAI_END_MSB 0x0b14 +#define AFE_MOD_DAI_END 0x0b18 +#define AFE_AWB_RCH_MON 0x0b70 +#define AFE_AWB_LCH_MON 0x0b74 +#define AFE_VUL_RCH_MON 0x0b78 +#define AFE_VUL_LCH_MON 0x0b7c +#define AFE_VUL12_RCH_MON 0x0b80 +#define AFE_VUL12_LCH_MON 0x0b84 +#define AFE_VUL2_RCH_MON 0x0b88 +#define AFE_VUL2_LCH_MON 0x0b8c +#define AFE_DAI_DATA_MON 0x0b90 +#define AFE_MOD_DAI_DATA_MON 0x0b94 +#define AFE_DAI2_DATA_MON 0x0b98 +#define AFE_AWB2_RCH_MON 0x0b9c +#define AFE_AWB2_LCH_MON 0x0ba0 +#define AFE_VUL3_RCH_MON 0x0ba4 +#define AFE_VUL3_LCH_MON 0x0ba8 +#define AFE_VUL4_RCH_MON 0x0bac +#define AFE_VUL4_LCH_MON 0x0bb0 +#define AFE_VUL5_RCH_MON 0x0bb4 +#define AFE_VUL5_LCH_MON 0x0bb8 +#define AFE_VUL6_RCH_MON 0x0bbc +#define AFE_VUL6_LCH_MON 0x0bc0 +#define AFE_DL1_RCH_MON 0x0bc4 +#define AFE_DL1_LCH_MON 0x0bc8 +#define AFE_DL2_RCH_MON 0x0bcc +#define AFE_DL2_LCH_MON 0x0bd0 +#define AFE_DL12_RCH1_MON 0x0bd4 +#define AFE_DL12_LCH1_MON 0x0bd8 +#define AFE_DL12_RCH2_MON 0x0bdc +#define AFE_DL12_LCH2_MON 0x0be0 +#define AFE_DL3_RCH_MON 0x0be4 +#define AFE_DL3_LCH_MON 0x0be8 +#define AFE_DL4_RCH_MON 0x0bec +#define AFE_DL4_LCH_MON 0x0bf0 +#define AFE_DL5_RCH_MON 0x0bf4 +#define AFE_DL5_LCH_MON 0x0bf8 +#define AFE_DL6_RCH_MON 0x0bfc +#define AFE_DL6_LCH_MON 0x0c00 +#define AFE_DL7_RCH_MON 0x0c04 +#define AFE_DL7_LCH_MON 0x0c08 +#define AFE_DL8_RCH_MON 0x0c0c +#define AFE_DL8_LCH_MON 0x0c10 +#define AFE_VUL5_CON0 0x0c14 +#define AFE_VUL5_BASE_MSB 0x0c18 +#define AFE_VUL5_BASE 0x0c1c +#define AFE_VUL5_CUR_MSB 0x0c20 +#define AFE_VUL5_CUR 0x0c24 +#define AFE_VUL5_END_MSB 0x0c28 +#define AFE_VUL5_END 0x0c2c +#define AFE_VUL6_CON0 0x0c30 +#define AFE_VUL6_BASE_MSB 0x0c34 +#define AFE_VUL6_BASE 0x0c38 +#define AFE_VUL6_CUR_MSB 0x0c3c +#define AFE_VUL6_CUR 0x0c40 +#define AFE_VUL6_END_MSB 0x0c44 +#define AFE_VUL6_END 0x0c48 +#define AFE_ADDA_DL_SDM_DCCOMP_CON 0x0c50 +#define AFE_ADDA_DL_SDM_TEST 0x0c54 +#define AFE_ADDA_DL_DC_COMP_CFG0 0x0c58 +#define AFE_ADDA_DL_DC_COMP_CFG1 0x0c5c +#define AFE_ADDA_DL_SDM_FIFO_MON 0x0c60 +#define AFE_ADDA_DL_SRC_LCH_MON 0x0c64 +#define AFE_ADDA_DL_SRC_RCH_MON 0x0c68 +#define AFE_ADDA_DL_SDM_OUT_MON 0x0c6c +#define AFE_ADDA_DL_SDM_DITHER_CON 0x0c70 +#define AFE_ADDA_DL_SDM_AUTO_RESET_CON 0x0c74 +#define AFE_CONNSYS_I2S_CON 0x0c78 +#define AFE_CONNSYS_I2S_MON 0x0c7c +#define AFE_ASRC_2CH_CON0 0x0c80 +#define AFE_ASRC_2CH_CON1 0x0c84 +#define AFE_ASRC_2CH_CON2 0x0c88 +#define AFE_ASRC_2CH_CON3 0x0c8c +#define AFE_ASRC_2CH_CON4 0x0c90 +#define AFE_ASRC_2CH_CON5 0x0c94 +#define AFE_ASRC_2CH_CON6 0x0c98 +#define AFE_ASRC_2CH_CON7 0x0c9c +#define AFE_ASRC_2CH_CON8 0x0ca0 +#define AFE_ASRC_2CH_CON9 0x0ca4 +#define AFE_ASRC_2CH_CON10 0x0ca8 +#define AFE_ASRC_2CH_CON12 0x0cb0 +#define AFE_ASRC_2CH_CON13 0x0cb4 +#define AFE_ADDA6_IIR_COEF_02_01 0x0ce0 +#define AFE_ADDA6_IIR_COEF_04_03 0x0ce4 +#define AFE_ADDA6_IIR_COEF_06_05 0x0ce8 +#define AFE_ADDA6_IIR_COEF_08_07 0x0cec +#define AFE_ADDA6_IIR_COEF_10_09 0x0cf0 +#define AFE_CONN67 0x0cf4 +#define AFE_CONN68 0x0cf8 +#define AFE_CONN69 0x0cfc +#define AFE_SE_PROT_SIDEBAND 0x0d38 +#define AFE_SE_DOMAIN_SIDEBAND0 0x0d3c +#define AFE_ADDA_PREDIS_CON2 0x0d40 +#define AFE_ADDA_PREDIS_CON3 0x0d44 +#define AFE_SE_DOMAIN_SIDEBAND1 0x0d54 +#define AFE_SE_DOMAIN_SIDEBAND2 0x0d58 +#define AFE_SE_DOMAIN_SIDEBAND3 0x0d5c +#define AFE_CONN44 0x0d70 +#define AFE_CONN45 0x0d74 +#define AFE_CONN46 0x0d78 +#define AFE_CONN47 0x0d7c +#define AFE_CONN44_1 0x0d80 +#define AFE_CONN45_1 0x0d84 +#define AFE_CONN46_1 0x0d88 +#define AFE_CONN47_1 0x0d8c +#define AFE_HD_ENGEN_ENABLE 0x0dd0 +#define AFE_ADDA_DL_NLE_FIFO_MON 0x0dfc +#define AFE_ADDA_MTKAIF_CFG0 0x0e00 +#define AFE_CONN67_1 0x0e04 +#define AFE_CONN68_1 0x0e08 +#define AFE_CONN69_1 0x0e0c +#define AFE_ADDA_MTKAIF_SYNCWORD_CFG 0x0e14 +#define AFE_ADDA_MTKAIF_RX_CFG0 0x0e20 +#define AFE_ADDA_MTKAIF_RX_CFG1 0x0e24 +#define AFE_ADDA_MTKAIF_RX_CFG2 0x0e28 +#define AFE_ADDA_MTKAIF_MON0 0x0e34 +#define AFE_ADDA_MTKAIF_MON1 0x0e38 +#define AFE_AUD_PAD_TOP 0x0e40 +#define AFE_DL_NLE_R_CFG0 0x0e44 +#define AFE_DL_NLE_R_CFG1 0x0e48 +#define AFE_DL_NLE_L_CFG0 0x0e4c +#define AFE_DL_NLE_L_CFG1 0x0e50 +#define AFE_DL_NLE_R_MON0 0x0e54 +#define AFE_DL_NLE_R_MON1 0x0e58 +#define AFE_DL_NLE_R_MON2 0x0e5c +#define AFE_DL_NLE_L_MON0 0x0e60 +#define AFE_DL_NLE_L_MON1 0x0e64 +#define AFE_DL_NLE_L_MON2 0x0e68 +#define AFE_DL_NLE_GAIN_CFG0 0x0e6c +#define AFE_ADDA6_MTKAIF_CFG0 0x0e70 +#define AFE_ADDA6_MTKAIF_RX_CFG0 0x0e74 +#define AFE_ADDA6_MTKAIF_RX_CFG1 0x0e78 +#define AFE_ADDA6_MTKAIF_RX_CFG2 0x0e7c +#define AFE_GENERAL1_ASRC_2CH_CON0 0x0e80 +#define AFE_GENERAL1_ASRC_2CH_CON1 0x0e84 +#define AFE_GENERAL1_ASRC_2CH_CON2 0x0e88 +#define AFE_GENERAL1_ASRC_2CH_CON3 0x0e8c +#define AFE_GENERAL1_ASRC_2CH_CON4 0x0e90 +#define AFE_GENERAL1_ASRC_2CH_CON5 0x0e94 +#define AFE_GENERAL1_ASRC_2CH_CON6 0x0e98 +#define AFE_GENERAL1_ASRC_2CH_CON7 0x0e9c +#define AFE_GENERAL1_ASRC_2CH_CON8 0x0ea0 +#define AFE_GENERAL1_ASRC_2CH_CON9 0x0ea4 +#define AFE_GENERAL1_ASRC_2CH_CON10 0x0ea8 +#define AFE_GENERAL1_ASRC_2CH_CON12 0x0eb0 +#define AFE_GENERAL1_ASRC_2CH_CON13 0x0eb4 +#define GENERAL_ASRC_MODE 0x0eb8 +#define GENERAL_ASRC_EN_ON 0x0ebc +#define AFE_CONN48 0x0ec0 +#define AFE_CONN49 0x0ec4 +#define AFE_CONN50 0x0ec8 +#define AFE_CONN51 0x0ecc +#define AFE_CONN52 0x0ed0 +#define AFE_CONN53 0x0ed4 +#define AFE_CONN54 0x0ed8 +#define AFE_CONN55 0x0edc +#define AFE_CONN48_1 0x0ee0 +#define AFE_CONN49_1 0x0ee4 +#define AFE_CONN50_1 0x0ee8 +#define AFE_CONN51_1 0x0eec +#define AFE_CONN52_1 0x0ef0 +#define AFE_CONN53_1 0x0ef4 +#define AFE_CONN54_1 0x0ef8 +#define AFE_CONN55_1 0x0efc +#define AFE_GENERAL2_ASRC_2CH_CON0 0x0f00 +#define AFE_GENERAL2_ASRC_2CH_CON1 0x0f04 +#define AFE_GENERAL2_ASRC_2CH_CON2 0x0f08 +#define AFE_GENERAL2_ASRC_2CH_CON3 0x0f0c +#define AFE_GENERAL2_ASRC_2CH_CON4 0x0f10 +#define AFE_GENERAL2_ASRC_2CH_CON5 0x0f14 +#define AFE_GENERAL2_ASRC_2CH_CON6 0x0f18 +#define AFE_GENERAL2_ASRC_2CH_CON7 0x0f1c +#define AFE_GENERAL2_ASRC_2CH_CON8 0x0f20 +#define AFE_GENERAL2_ASRC_2CH_CON9 0x0f24 +#define AFE_GENERAL2_ASRC_2CH_CON10 0x0f28 +#define AFE_GENERAL2_ASRC_2CH_CON12 0x0f30 +#define AFE_GENERAL2_ASRC_2CH_CON13 0x0f34 +#define AFE_DL5_CON0 0x0f4c +#define AFE_DL5_BASE_MSB 0x0f50 +#define AFE_DL5_BASE 0x0f54 +#define AFE_DL5_CUR_MSB 0x0f58 +#define AFE_DL5_CUR 0x0f5c +#define AFE_DL5_END_MSB 0x0f60 +#define AFE_DL5_END 0x0f64 +#define AFE_DL6_CON0 0x0f68 +#define AFE_DL6_BASE_MSB 0x0f6c +#define AFE_DL6_BASE 0x0f70 +#define AFE_DL6_CUR_MSB 0x0f74 +#define AFE_DL6_CUR 0x0f78 +#define AFE_DL6_END_MSB 0x0f7c +#define AFE_DL6_END 0x0f80 +#define AFE_DL7_CON0 0x0f84 +#define AFE_DL7_BASE_MSB 0x0f88 +#define AFE_DL7_BASE 0x0f8c +#define AFE_DL7_CUR_MSB 0x0f90 +#define AFE_DL7_CUR 0x0f94 +#define AFE_DL7_END_MSB 0x0f98 +#define AFE_DL7_END 0x0f9c +#define AFE_DL8_CON0 0x0fa0 +#define AFE_DL8_BASE_MSB 0x0fa4 +#define AFE_DL8_BASE 0x0fa8 +#define AFE_DL8_CUR_MSB 0x0fac +#define AFE_DL8_CUR 0x0fb0 +#define AFE_DL8_END_MSB 0x0fb4 +#define AFE_DL8_END 0x0fb8 +#define AFE_SE_SECURE_CON 0x1004 +#define AFE_PROT_SIDEBAND_MON 0x1008 +#define AFE_DOMAIN_SIDEBAND0_MON 0x100c +#define AFE_DOMAIN_SIDEBAND1_MON 0x1010 +#define AFE_DOMAIN_SIDEBAND2_MON 0x1014 +#define AFE_DOMAIN_SIDEBAND3_MON 0x1018 +#define AFE_SECURE_MASK_CONN0 0x1020 +#define AFE_SECURE_MASK_CONN1 0x1024 +#define AFE_SECURE_MASK_CONN2 0x1028 +#define AFE_SECURE_MASK_CONN3 0x102c +#define AFE_SECURE_MASK_CONN4 0x1030 +#define AFE_SECURE_MASK_CONN5 0x1034 +#define AFE_SECURE_MASK_CONN6 0x1038 +#define AFE_SECURE_MASK_CONN7 0x103c +#define AFE_SECURE_MASK_CONN8 0x1040 +#define AFE_SECURE_MASK_CONN9 0x1044 +#define AFE_SECURE_MASK_CONN10 0x1048 +#define AFE_SECURE_MASK_CONN11 0x104c +#define AFE_SECURE_MASK_CONN12 0x1050 +#define AFE_SECURE_MASK_CONN13 0x1054 +#define AFE_SECURE_MASK_CONN14 0x1058 +#define AFE_SECURE_MASK_CONN15 0x105c +#define AFE_SECURE_MASK_CONN16 0x1060 +#define AFE_SECURE_MASK_CONN17 0x1064 +#define AFE_SECURE_MASK_CONN18 0x1068 +#define AFE_SECURE_MASK_CONN19 0x106c +#define AFE_SECURE_MASK_CONN20 0x1070 +#define AFE_SECURE_MASK_CONN21 0x1074 +#define AFE_SECURE_MASK_CONN22 0x1078 +#define AFE_SECURE_MASK_CONN23 0x107c +#define AFE_SECURE_MASK_CONN24 0x1080 +#define AFE_SECURE_MASK_CONN25 0x1084 +#define AFE_SECURE_MASK_CONN26 0x1088 +#define AFE_SECURE_MASK_CONN27 0x108c +#define AFE_SECURE_MASK_CONN28 0x1090 +#define AFE_SECURE_MASK_CONN29 0x1094 +#define AFE_SECURE_MASK_CONN30 0x1098 +#define AFE_SECURE_MASK_CONN31 0x109c +#define AFE_SECURE_MASK_CONN32 0x10a0 +#define AFE_SECURE_MASK_CONN33 0x10a4 +#define AFE_SECURE_MASK_CONN34 0x10a8 +#define AFE_SECURE_MASK_CONN35 0x10ac +#define AFE_SECURE_MASK_CONN36 0x10b0 +#define AFE_SECURE_MASK_CONN37 0x10b4 +#define AFE_SECURE_MASK_CONN38 0x10b8 +#define AFE_SECURE_MASK_CONN39 0x10bc +#define AFE_SECURE_MASK_CONN40 0x10c0 +#define AFE_SECURE_MASK_CONN41 0x10c4 +#define AFE_SECURE_MASK_CONN42 0x10c8 +#define AFE_SECURE_MASK_CONN43 0x10cc +#define AFE_SECURE_MASK_CONN44 0x10d0 +#define AFE_SECURE_MASK_CONN45 0x10d4 +#define AFE_SECURE_MASK_CONN46 0x10d8 +#define AFE_SECURE_MASK_CONN47 0x10dc +#define AFE_SECURE_MASK_CONN48 0x10e0 +#define AFE_SECURE_MASK_CONN49 0x10e4 +#define AFE_SECURE_MASK_CONN50 0x10e8 +#define AFE_SECURE_MASK_CONN51 0x10ec +#define AFE_SECURE_MASK_CONN52 0x10f0 +#define AFE_SECURE_MASK_CONN53 0x10f4 +#define AFE_SECURE_MASK_CONN54 0x10f8 +#define AFE_SECURE_MASK_CONN55 0x10fc +#define AFE_SECURE_MASK_CONN56 0x1100 +#define AFE_SECURE_MASK_CONN57 0x1104 +#define AFE_SECURE_MASK_CONN0_1 0x1108 +#define AFE_SECURE_MASK_CONN1_1 0x110c +#define AFE_SECURE_MASK_CONN2_1 0x1110 +#define AFE_SECURE_MASK_CONN3_1 0x1114 +#define AFE_SECURE_MASK_CONN4_1 0x1118 +#define AFE_SECURE_MASK_CONN5_1 0x111c +#define AFE_SECURE_MASK_CONN6_1 0x1120 +#define AFE_SECURE_MASK_CONN7_1 0x1124 +#define AFE_SECURE_MASK_CONN8_1 0x1128 +#define AFE_SECURE_MASK_CONN9_1 0x112c +#define AFE_SECURE_MASK_CONN10_1 0x1130 +#define AFE_SECURE_MASK_CONN11_1 0x1134 +#define AFE_SECURE_MASK_CONN12_1 0x1138 +#define AFE_SECURE_MASK_CONN13_1 0x113c +#define AFE_SECURE_MASK_CONN14_1 0x1140 +#define AFE_SECURE_MASK_CONN15_1 0x1144 +#define AFE_SECURE_MASK_CONN16_1 0x1148 +#define AFE_SECURE_MASK_CONN17_1 0x114c +#define AFE_SECURE_MASK_CONN18_1 0x1150 +#define AFE_SECURE_MASK_CONN19_1 0x1154 +#define AFE_SECURE_MASK_CONN20_1 0x1158 +#define AFE_SECURE_MASK_CONN21_1 0x115c +#define AFE_SECURE_MASK_CONN22_1 0x1160 +#define AFE_SECURE_MASK_CONN23_1 0x1164 +#define AFE_SECURE_MASK_CONN24_1 0x1168 +#define AFE_SECURE_MASK_CONN25_1 0x116c +#define AFE_SECURE_MASK_CONN26_1 0x1170 +#define AFE_SECURE_MASK_CONN27_1 0x1174 +#define AFE_SECURE_MASK_CONN28_1 0x1178 +#define AFE_SECURE_MASK_CONN29_1 0x117c +#define AFE_SECURE_MASK_CONN30_1 0x1180 +#define AFE_SECURE_MASK_CONN31_1 0x1184 +#define AFE_SECURE_MASK_CONN32_1 0x1188 +#define AFE_SECURE_MASK_CONN33_1 0x118c +#define AFE_SECURE_MASK_CONN34_1 0x1190 +#define AFE_SECURE_MASK_CONN35_1 0x1194 +#define AFE_SECURE_MASK_CONN36_1 0x1198 +#define AFE_SECURE_MASK_CONN37_1 0x119c +#define AFE_SECURE_MASK_CONN38_1 0x11a0 +#define AFE_SECURE_MASK_CONN39_1 0x11a4 +#define AFE_SECURE_MASK_CONN40_1 0x11a8 +#define AFE_SECURE_MASK_CONN41_1 0x11ac +#define AFE_SECURE_MASK_CONN42_1 0x11b0 +#define AFE_SECURE_MASK_CONN43_1 0x11b4 +#define AFE_SECURE_MASK_CONN44_1 0x11b8 +#define AFE_SECURE_MASK_CONN45_1 0x11bc +#define AFE_SECURE_MASK_CONN46_1 0x11c0 +#define AFE_SECURE_MASK_CONN47_1 0x11c4 +#define AFE_SECURE_MASK_CONN48_1 0x11c8 +#define AFE_SECURE_MASK_CONN49_1 0x11cc +#define AFE_SECURE_MASK_CONN50_1 0x11d0 +#define AFE_SECURE_MASK_CONN51_1 0x11d4 +#define AFE_SECURE_MASK_CONN52_1 0x11d8 +#define AFE_SECURE_MASK_CONN53_1 0x11dc +#define AFE_SECURE_MASK_CONN54_1 0x11e0 +#define AFE_SECURE_MASK_CONN55_1 0x11e4 +#define AFE_SECURE_MASK_CONN56_1 0x11e8 +#define AFE_CONN60_1 0x11f0 +#define AFE_CONN61_1 0x11f4 +#define AFE_CONN62_1 0x11f8 +#define AFE_CONN63_1 0x11fc +#define AFE_CONN64_1 0x1220 +#define AFE_CONN65_1 0x1224 +#define AFE_CONN66_1 0x1228 +#define FPGA_CFG4 0x1230 +#define FPGA_CFG5 0x1234 +#define FPGA_CFG6 0x1238 +#define FPGA_CFG7 0x123c +#define FPGA_CFG8 0x1240 +#define FPGA_CFG9 0x1244 +#define FPGA_CFG10 0x1248 +#define FPGA_CFG11 0x124c +#define FPGA_CFG12 0x1250 +#define FPGA_CFG13 0x1254 +#define ETDM_IN1_CON0 0x1430 +#define ETDM_IN1_CON1 0x1434 +#define ETDM_IN1_CON2 0x1438 +#define ETDM_IN1_CON3 0x143c +#define ETDM_IN1_CON4 0x1440 +#define ETDM_IN1_CON5 0x1444 +#define ETDM_IN1_CON6 0x1448 +#define ETDM_IN1_CON7 0x144c +#define ETDM_IN1_CON8 0x1450 +#define ETDM_OUT1_CON0 0x1454 +#define ETDM_OUT1_CON1 0x1458 +#define ETDM_OUT1_CON2 0x145c +#define ETDM_OUT1_CON3 0x1460 +#define ETDM_OUT1_CON4 0x1464 +#define ETDM_OUT1_CON5 0x1468 +#define ETDM_OUT1_CON6 0x146c +#define ETDM_OUT1_CON7 0x1470 +#define ETDM_OUT1_CON8 0x1474 +#define ETDM_IN1_MON 0x1478 +#define ETDM_OUT1_MON 0x147c +#define ETDM_0_3_COWORK_CON0 0x18b0 +#define ETDM_0_3_COWORK_CON1 0x18b4 +#define ETDM_0_3_COWORK_CON3 0x18bc + +#define AFE_MAX_REGISTER ETDM_0_3_COWORK_CON3 + +#define AFE_IRQ_STATUS_BITS 0x87FFFFFF +#define AFE_IRQ_CNT_SHIFT 0 +#define AFE_IRQ_CNT_MASK 0x3ffff +#endif -- cgit v1.2.3 From 11fe58c4450a8108b498d2f849976ba2686820fc Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Mon, 6 Jun 2022 15:46:20 -0500 Subject: ASoC: SOF: Intel: add MeteorLake machines MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add support for MeteorLake (MTL) machines support, starting with mockup devices. Reviewed-by: Péter Ujfalusi Reviewed-by: Rander Wang Reviewed-by: Ranjani Sridharan Signed-off-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220606204622.144424-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/common/Makefile | 1 + sound/soc/intel/common/soc-acpi-intel-mtl-match.c | 41 +++++++++++++++++++++++ 2 files changed, 42 insertions(+) create mode 100644 sound/soc/intel/common/soc-acpi-intel-mtl-match.c (limited to 'sound') diff --git a/sound/soc/intel/common/Makefile b/sound/soc/intel/common/Makefile index fef0b2d1de68..8ca8f872ec80 100644 --- a/sound/soc/intel/common/Makefile +++ b/sound/soc/intel/common/Makefile @@ -9,6 +9,7 @@ snd-soc-acpi-intel-match-objs := soc-acpi-intel-byt-match.o soc-acpi-intel-cht-m soc-acpi-intel-cml-match.o soc-acpi-intel-icl-match.o \ soc-acpi-intel-tgl-match.o soc-acpi-intel-ehl-match.o \ soc-acpi-intel-jsl-match.o soc-acpi-intel-adl-match.o \ + soc-acpi-intel-mtl-match.o \ soc-acpi-intel-hda-match.o \ soc-acpi-intel-sdw-mockup-match.o diff --git a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c new file mode 100644 index 000000000000..cc594b27e03b --- /dev/null +++ b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c @@ -0,0 +1,41 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * soc-acpi-intel-mtl-match.c - tables and support for MTL ACPI enumeration. + * + * Copyright (c) 2022, Intel Corporation. + * + */ + +#include +#include +#include "soc-acpi-intel-sdw-mockup-match.h" + +struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_machines[] = { + {}, +}; +EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_mtl_machines); + +/* this table is used when there is no I2S codec present */ +struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_sdw_machines[] = { + /* mockup tests need to be first */ + { + .link_mask = GENMASK(3, 0), + .links = sdw_mockup_headset_2amps_mic, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-mtl-rt711-rt1308-rt715.tplg", + }, + { + .link_mask = BIT(0) | BIT(1) | BIT(3), + .links = sdw_mockup_headset_1amp_mic, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-mtl-rt711-rt1308-mono-rt715.tplg", + }, + { + .link_mask = GENMASK(2, 0), + .links = sdw_mockup_mic_headset_1amp, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-mtl-rt715-rt711-rt1308-mono.tplg", + }, + {}, +}; +EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_mtl_sdw_machines); -- cgit v1.2.3 From 93693dcf2a4d7ab6a355f80c14653cd9c27a1422 Mon Sep 17 00:00:00 2001 From: Yong Zhi Date: Mon, 6 Jun 2022 15:46:21 -0500 Subject: ASoC: Intel: boards: rename RTL1019 compatible driver to rt1019p Use rt1019p for rt1015p.c compatible codec and reserve the name rt1019 for 10EC1019 matched driver in sof_realtek_common. Reviewed-by: Bard Liao Signed-off-by: Yong Zhi Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220606204622.144424-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_realtek_common.c | 24 ++++++++++++------------ sound/soc/intel/boards/sof_realtek_common.h | 6 +++--- sound/soc/intel/boards/sof_rt5682.c | 2 +- 3 files changed, 16 insertions(+), 16 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/sof_realtek_common.c b/sound/soc/intel/boards/sof_realtek_common.c index 2ab568c1d40b..b9643ca2e2f2 100644 --- a/sound/soc/intel/boards/sof_realtek_common.c +++ b/sound/soc/intel/boards/sof_realtek_common.c @@ -463,26 +463,26 @@ EXPORT_SYMBOL_NS(sof_rt1308_dai_link, SND_SOC_INTEL_SOF_REALTEK_COMMON); * 2-amp Configuration for RT1019 */ -static const struct snd_soc_dapm_route rt1019_dapm_routes[] = { +static const struct snd_soc_dapm_route rt1019p_dapm_routes[] = { /* speaker */ { "Left Spk", NULL, "Speaker" }, { "Right Spk", NULL, "Speaker" }, }; -static struct snd_soc_dai_link_component rt1019_components[] = { +static struct snd_soc_dai_link_component rt1019p_components[] = { { - .name = RT1019_DEV0_NAME, - .dai_name = RT1019_CODEC_DAI, + .name = RT1019P_DEV0_NAME, + .dai_name = RT1019P_CODEC_DAI, }, }; -static int rt1019_init(struct snd_soc_pcm_runtime *rtd) +static int rt1019p_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; int ret; - ret = snd_soc_dapm_add_routes(&card->dapm, rt1019_dapm_routes, - ARRAY_SIZE(rt1019_dapm_routes)); + ret = snd_soc_dapm_add_routes(&card->dapm, rt1019p_dapm_routes, + ARRAY_SIZE(rt1019p_dapm_routes)); if (ret) { dev_err(rtd->dev, "Speaker map addition failed: %d\n", ret); return ret; @@ -490,13 +490,13 @@ static int rt1019_init(struct snd_soc_pcm_runtime *rtd) return ret; } -void sof_rt1019_dai_link(struct snd_soc_dai_link *link) +void sof_rt1019p_dai_link(struct snd_soc_dai_link *link) { - link->codecs = rt1019_components; - link->num_codecs = ARRAY_SIZE(rt1019_components); - link->init = rt1019_init; + link->codecs = rt1019p_components; + link->num_codecs = ARRAY_SIZE(rt1019p_components); + link->init = rt1019p_init; } -EXPORT_SYMBOL_NS(sof_rt1019_dai_link, SND_SOC_INTEL_SOF_REALTEK_COMMON); +EXPORT_SYMBOL_NS(sof_rt1019p_dai_link, SND_SOC_INTEL_SOF_REALTEK_COMMON); MODULE_DESCRIPTION("ASoC Intel SOF Realtek helpers"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/boards/sof_realtek_common.h b/sound/soc/intel/boards/sof_realtek_common.h index ec3eea633e04..778443421090 100644 --- a/sound/soc/intel/boards/sof_realtek_common.h +++ b/sound/soc/intel/boards/sof_realtek_common.h @@ -39,9 +39,9 @@ void sof_rt1015_codec_conf(struct snd_soc_card *card); #define RT1308_DEV0_NAME "i2c-10EC1308:00" void sof_rt1308_dai_link(struct snd_soc_dai_link *link); -#define RT1019_CODEC_DAI "HiFi" -#define RT1019_DEV0_NAME "RTL1019:00" +#define RT1019P_CODEC_DAI "HiFi" +#define RT1019P_DEV0_NAME "RTL1019:00" -void sof_rt1019_dai_link(struct snd_soc_dai_link *link); +void sof_rt1019p_dai_link(struct snd_soc_dai_link *link); #endif /* __SOF_REALTEK_COMMON_H */ diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 5d67a2c87a1d..f28dae64587e 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -735,7 +735,7 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, } else if (sof_rt5682_quirk & SOF_RT1015P_SPEAKER_AMP_PRESENT) { sof_rt1015p_dai_link(&links[id]); } else if (sof_rt5682_quirk & SOF_RT1019_SPEAKER_AMP_PRESENT) { - sof_rt1019_dai_link(&links[id]); + sof_rt1019p_dai_link(&links[id]); } else if (sof_rt5682_quirk & SOF_MAX98373_SPEAKER_AMP_PRESENT) { links[id].codecs = max_98373_components; -- cgit v1.2.3 From 8208dd75eb468d1bb90aef52f385e5b3486bb737 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 6 Jun 2022 15:46:22 -0500 Subject: ASoC: Intel: sof_sdw: allow HDaudio/HDMI disable For tests, it's rather common to disable the HDaudio links and codecs in the build. Since we already get a codec_mask parameter indicating that there are no codecs detected, it's straightforward to skip the HDMI dailink creation and create a card. Note that when disabling HDMI, a modified topology without HDMI pipelines needs to be provided as well. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Rander Wang Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20220606204622.144424-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/Kconfig | 1 - sound/soc/intel/boards/sof_sdw.c | 24 +++++++++++++++--------- 2 files changed, 15 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index f3873b5bea87..4b4c1e1e4808 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -660,7 +660,6 @@ config SND_SOC_INTEL_SOUNDWIRE_SOF_MACH depends on MFD_INTEL_LPSS || COMPILE_TEST depends on SND_SOC_INTEL_USER_FRIENDLY_LONG_NAMES || COMPILE_TEST depends on SOUNDWIRE - depends on SND_HDA_CODEC_HDMI && SND_SOC_SOF_HDA_AUDIO_CODEC select SND_SOC_MAX98373_I2C select SND_SOC_MAX98373_SDW select SND_SOC_RT700_SDW diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 1f00679b4240..f871daa5cb33 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -1127,10 +1127,14 @@ static int sof_card_dai_links_create(struct device *dev, for (i = 0; i < ARRAY_SIZE(codec_info_list); i++) codec_info_list[i].amp_num = 0; - if (sof_sdw_quirk & SOF_SDW_TGL_HDMI) - hdmi_num = SOF_TGL_HDMI_COUNT; - else - hdmi_num = SOF_PRE_TGL_HDMI_COUNT; + if (mach_params->codec_mask & IDISP_CODEC_MASK) { + ctx->idisp_codec = true; + + if (sof_sdw_quirk & SOF_SDW_TGL_HDMI) + hdmi_num = SOF_TGL_HDMI_COUNT; + else + hdmi_num = SOF_PRE_TGL_HDMI_COUNT; + } ssp_mask = SOF_SSP_GET_PORT(sof_sdw_quirk); /* @@ -1150,9 +1154,6 @@ static int sof_card_dai_links_create(struct device *dev, return ret; } - if (mach_params->codec_mask & IDISP_CODEC_MASK) - ctx->idisp_codec = true; - /* enable dmic01 & dmic16k */ dmic_num = (sof_sdw_quirk & SOF_SDW_PCH_DMIC || mach_params->dmic_num) ? 2 : 0; comp_num += dmic_num; @@ -1375,7 +1376,9 @@ HDMI: static int sof_sdw_card_late_probe(struct snd_soc_card *card) { - int i, ret; + struct mc_private *ctx = snd_soc_card_get_drvdata(card); + int ret = 0; + int i; for (i = 0; i < ARRAY_SIZE(codec_info_list); i++) { if (!codec_info_list[i].late_probe) @@ -1386,7 +1389,10 @@ static int sof_sdw_card_late_probe(struct snd_soc_card *card) return ret; } - return sof_sdw_hdmi_card_late_probe(card); + if (ctx->idisp_codec) + ret = sof_sdw_hdmi_card_late_probe(card); + + return ret; } /* SoC card */ -- cgit v1.2.3 From b585692fc937dc8f9d494078b5ffe2aafe31ec18 Mon Sep 17 00:00:00 2001 From: Ajit Kumar Pandey Date: Mon, 6 Jun 2022 16:02:08 -0500 Subject: ASoC: SOF: amd: Add SOF pm ops callback for Renoir Add SOF PM ops callback in renoir dsp ops to power off and on ACP IP block during system suspend and resume on Renoir platform. Signed-off-by: Ajit Kumar Pandey Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220606210212.146626-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/amd/acp-dsp-offset.h | 2 ++ sound/soc/sof/amd/acp.c | 36 ++++++++++++++++++++++++++++++++++++ sound/soc/sof/amd/acp.h | 4 ++++ sound/soc/sof/amd/pci-rn.c | 4 ++++ sound/soc/sof/amd/renoir.c | 4 ++++ 5 files changed, 50 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/amd/acp-dsp-offset.h b/sound/soc/sof/amd/acp-dsp-offset.h index 40fbf11facba..56cefd4a84fc 100644 --- a/sound/soc/sof/amd/acp-dsp-offset.h +++ b/sound/soc/sof/amd/acp-dsp-offset.h @@ -46,12 +46,14 @@ #define ACPAXI2AXI_ATU_BASE_ADDR_GRP_8 0xC3C #define ACPAXI2AXI_ATU_CTRL 0xC40 #define ACP_SOFT_RESET 0x1000 +#define ACP_CONTROL 0x1004 #define ACP_I2S_PIN_CONFIG 0x1400 /* Registers from ACP_PGFSM block */ #define ACP_PGFSM_CONTROL 0x141C #define ACP_PGFSM_STATUS 0x1420 +#define ACP_CLKMUX_SEL 0x1424 /* Registers from ACP_INTR block */ #define ACP_EXTERNAL_INTR_ENB 0x1800 diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index 0c272573df97..c40d2900dd36 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -413,10 +413,46 @@ static int acp_init(struct snd_sof_dev *sdev) dev_err(sdev->dev, "ACP power on failed\n"); return ret; } + + snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_CONTROL, 0x01); /* Reset */ return acp_reset(sdev); } +int amd_sof_acp_suspend(struct snd_sof_dev *sdev, u32 target_state) +{ + int ret; + + ret = acp_reset(sdev); + if (ret) { + dev_err(sdev->dev, "ACP Reset failed\n"); + return ret; + } + + snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_CONTROL, 0x00); + + return 0; +} +EXPORT_SYMBOL_NS(amd_sof_acp_suspend, SND_SOC_SOF_AMD_COMMON); + +int amd_sof_acp_resume(struct snd_sof_dev *sdev) +{ + int ret; + + ret = acp_init(sdev); + if (ret) { + dev_err(sdev->dev, "ACP Init failed\n"); + return ret; + } + + snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_CLKMUX_SEL, 0x03); + + ret = acp_memory_init(sdev); + + return ret; +} +EXPORT_SYMBOL_NS(amd_sof_acp_resume, SND_SOC_SOF_AMD_COMMON); + int amd_sof_acp_probe(struct snd_sof_dev *sdev) { struct pci_dev *pci = to_pci_dev(sdev->dev); diff --git a/sound/soc/sof/amd/acp.h b/sound/soc/sof/amd/acp.h index 291b44c54bcc..4c42b8fd6abf 100644 --- a/sound/soc/sof/amd/acp.h +++ b/sound/soc/sof/amd/acp.h @@ -216,6 +216,10 @@ int acp_sof_trace_init(struct snd_sof_dev *sdev, struct snd_dma_buffer *dmab, struct sof_ipc_dma_trace_params_ext *dtrace_params); int acp_sof_trace_release(struct snd_sof_dev *sdev); +/* PM Callbacks */ +int amd_sof_acp_suspend(struct snd_sof_dev *sdev, u32 target_state); +int amd_sof_acp_resume(struct snd_sof_dev *sdev); + struct sof_amd_acp_desc { unsigned int host_bridge_id; }; diff --git a/sound/soc/sof/amd/pci-rn.c b/sound/soc/sof/amd/pci-rn.c index d5d9bcc2c997..3a7fed25a226 100644 --- a/sound/soc/sof/amd/pci-rn.c +++ b/sound/soc/sof/amd/pci-rn.c @@ -49,6 +49,7 @@ static const struct sof_amd_acp_desc renoir_chip_info = { static const struct sof_dev_desc renoir_desc = { .machines = snd_soc_acpi_amd_sof_machines, + .use_acpi_target_states = true, .resindex_lpe_base = 0, .resindex_pcicfg_base = -1, .resindex_imr_base = -1, @@ -166,6 +167,9 @@ static struct pci_driver snd_sof_pci_amd_rn_driver = { .id_table = rn_pci_ids, .probe = acp_pci_rn_probe, .remove = acp_pci_rn_remove, + .driver = { + .pm = &sof_pci_pm, + }, }; module_pci_driver(snd_sof_pci_amd_rn_driver); diff --git a/sound/soc/sof/amd/renoir.c b/sound/soc/sof/amd/renoir.c index 70190365328c..9261c8bc2236 100644 --- a/sound/soc/sof/amd/renoir.c +++ b/sound/soc/sof/amd/renoir.c @@ -173,6 +173,10 @@ struct snd_sof_dsp_ops sof_renoir_ops = { /* Trace Logger */ .trace_init = acp_sof_trace_init, .trace_release = acp_sof_trace_release, + + /* PM */ + .suspend = amd_sof_acp_suspend, + .resume = amd_sof_acp_resume, }; EXPORT_SYMBOL(sof_renoir_ops); -- cgit v1.2.3 From e53b20598f394e37951d6355f1c88ae01165b53f Mon Sep 17 00:00:00 2001 From: YC Hung Date: Mon, 6 Jun 2022 16:02:09 -0500 Subject: ASoC: SOF: mediatek: revise mt8195 clock sequence MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit clock enable : enable and set audio_h selection as 26M. Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Signed-off-by: YC Hung Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220606210212.146626-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/mediatek/mt8195/mt8195-clk.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/mediatek/mt8195/mt8195-clk.c b/sound/soc/sof/mediatek/mt8195/mt8195-clk.c index 6bcb4b9b00fb..9ef08e43aa38 100644 --- a/sound/soc/sof/mediatek/mt8195/mt8195-clk.c +++ b/sound/soc/sof/mediatek/mt8195/mt8195-clk.c @@ -132,6 +132,13 @@ static int adsp_default_clk_init(struct snd_sof_dev *sdev, bool enable) return ret; } + ret = clk_set_parent(priv->clk[CLK_TOP_AUDIO_H], + priv->clk[CLK_TOP_CLK26M]); + if (ret) { + dev_err(dev, "set audio_h_sel failed %d\n", ret); + return ret; + } + ret = adsp_enable_all_clock(sdev); if (ret) { dev_err(dev, "failed to adsp_enable_clock: %d\n", ret); -- cgit v1.2.3 From fd43dcbb859c85831a05e37287e1c5395f54aba8 Mon Sep 17 00:00:00 2001 From: YC Hung Date: Mon, 6 Jun 2022 16:02:10 -0500 Subject: ASoC: SOF: mediatek: Add shared_size for mediatek common chip information Add shared_size for mediatek common chip information which is used for audio and trace dma. Reviewed-by: Curtis Malainey Reviewed-by: Ranjani Sridharan Signed-off-by: YC Hung Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220606210212.146626-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/mediatek/adsp_helper.h | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/sof/mediatek/adsp_helper.h b/sound/soc/sof/mediatek/adsp_helper.h index 4ab998756bbc..d41e904e6614 100644 --- a/sound/soc/sof/mediatek/adsp_helper.h +++ b/sound/soc/sof/mediatek/adsp_helper.h @@ -20,6 +20,7 @@ struct mtk_adsp_chip_info { u32 sramsize; u32 dramsize; u32 cfgregsize; + u32 shared_size; void __iomem *va_sram; /* corresponding to pa_sram */ void __iomem *va_dram; /* corresponding to pa_dram */ void __iomem *va_cfgreg; -- cgit v1.2.3 From 0bf4276cc7883d65e594926c1159d4c0712d02e7 Mon Sep 17 00:00:00 2001 From: YC Hung Date: Mon, 6 Jun 2022 16:02:11 -0500 Subject: ASoC: SOF: mediatek: mt8195 modify dram type as non-cache Modify dram as non-cache memory type to avoid wrong access between host and dsp side and get the size of shared dma from device tree. Reviewed-by: Curtis Malainey Reviewed-by: Ranjani Sridharan Signed-off-by: YC Hung Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220606210212.146626-5-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/mediatek/mt8195/mt8195.c | 37 +++++++++++++++++++--------------- 1 file changed, 21 insertions(+), 16 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/mediatek/mt8195/mt8195.c b/sound/soc/sof/mediatek/mt8195/mt8195.c index 30111ab23bf5..64d1b5a4e31b 100644 --- a/sound/soc/sof/mediatek/mt8195/mt8195.c +++ b/sound/soc/sof/mediatek/mt8195/mt8195.c @@ -145,6 +145,14 @@ static int platform_parse_resource(struct platform_device *pdev, void *data) dev_dbg(dev, "DMA %pR\n", &res); + adsp->pa_shared_dram = (phys_addr_t)res.start; + adsp->shared_size = resource_size(&res); + if (adsp->pa_shared_dram & DRAM_REMAP_MASK) { + dev_err(dev, "adsp shared dma memory(%#x) is not 4K-aligned\n", + (u32)adsp->pa_shared_dram); + return -EINVAL; + } + ret = of_reserved_mem_device_init(dev); if (ret) { dev_err(dev, "of_reserved_mem_device_init failed\n"); @@ -273,23 +281,18 @@ static int adsp_shared_base_ioremap(struct platform_device *pdev, void *data) { struct device *dev = &pdev->dev; struct mtk_adsp_chip_info *adsp = data; - u32 shared_size; /* remap shared-dram base to be non-cachable */ - shared_size = TOTAL_SIZE_SHARED_DRAM_FROM_TAIL; - adsp->pa_shared_dram = adsp->pa_dram + adsp->dramsize - shared_size; - if (adsp->va_dram) { - adsp->shared_dram = adsp->va_dram + DSP_DRAM_SIZE - shared_size; - } else { - adsp->shared_dram = devm_ioremap(dev, adsp->pa_shared_dram, - shared_size); - if (!adsp->shared_dram) { - dev_err(dev, "ioremap failed for shared DRAM\n"); - return -ENOMEM; - } + adsp->shared_dram = devm_ioremap(dev, adsp->pa_shared_dram, + adsp->shared_size); + if (!adsp->shared_dram) { + dev_err(dev, "failed to ioremap base %pa size %#x\n", + adsp->shared_dram, adsp->shared_size); + return -ENOMEM; } + dev_dbg(dev, "shared-dram vbase=%p, phy addr :%pa, size=%#x\n", - adsp->shared_dram, &adsp->pa_shared_dram, shared_size); + adsp->shared_dram, &adsp->pa_shared_dram, adsp->shared_size); return 0; } @@ -361,9 +364,11 @@ static int mt8195_dsp_probe(struct snd_sof_dev *sdev) goto err_adsp_sram_power_off; } - sdev->bar[SOF_FW_BLK_TYPE_SRAM] = devm_ioremap_wc(sdev->dev, - priv->adsp->pa_dram, - priv->adsp->dramsize); + priv->adsp->va_sram = sdev->bar[SOF_FW_BLK_TYPE_IRAM]; + + sdev->bar[SOF_FW_BLK_TYPE_SRAM] = devm_ioremap(sdev->dev, + priv->adsp->pa_dram, + priv->adsp->dramsize); if (!sdev->bar[SOF_FW_BLK_TYPE_SRAM]) { dev_err(sdev->dev, "failed to ioremap base %pa size %#x\n", &priv->adsp->pa_dram, priv->adsp->dramsize); -- cgit v1.2.3 From 078f28fee5aa417169d8e8906815c684beddbe74 Mon Sep 17 00:00:00 2001 From: YC Hung Date: Mon, 6 Jun 2022 16:02:12 -0500 Subject: ASoC: SOF: mediatek: mt8195 suspend check dsp idle MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit During suspend flow, sof_suspend will be called and the pm_ops->ctx_save callback notifies DSP of the upcoming power down. Upon receipt of the ctx_save IPC, the DSP will start the D3 transition. Before the DSP enter idle, an interrupt is generated to notify the host of the power state change. Since the host and DSP are two different processors, there could be a race condition, which can be avoided by polling with 1s timeout and 500us intervals Reviewed-by: Péter Ujfalusi Signed-off-by: YC Hung Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220606210212.146626-6-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/mediatek/mt8195/mt8195.c | 13 +++++++++++++ sound/soc/sof/mediatek/mt8195/mt8195.h | 5 +++++ 2 files changed, 18 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/mediatek/mt8195/mt8195.c b/sound/soc/sof/mediatek/mt8195/mt8195.c index 64d1b5a4e31b..9c146015cd1b 100644 --- a/sound/soc/sof/mediatek/mt8195/mt8195.c +++ b/sound/soc/sof/mediatek/mt8195/mt8195.c @@ -443,6 +443,19 @@ static int mt8195_dsp_suspend(struct snd_sof_dev *sdev, u32 target_state) { struct platform_device *pdev = container_of(sdev->dev, struct platform_device, dev); int ret; + u32 reset_sw, dbg_pc; + + /* wait dsp enter idle, timeout is 1 second */ + ret = snd_sof_dsp_read_poll_timeout(sdev, DSP_REG_BAR, + DSP_RESET_SW, reset_sw, + ((reset_sw & ADSP_PWAIT) == ADSP_PWAIT), + SUSPEND_DSP_IDLE_POLL_INTERVAL_US, + SUSPEND_DSP_IDLE_TIMEOUT_US); + if (ret < 0) { + dbg_pc = snd_sof_dsp_read(sdev, DSP_REG_BAR, DSP_PDEBUGPC); + dev_warn(sdev->dev, "dsp not idle, powering off anyway : swrest %#x, pc %#x, ret %d\n", + reset_sw, dbg_pc, ret); + } /* stall and reset dsp */ sof_hifixdsp_shutdown(sdev); diff --git a/sound/soc/sof/mediatek/mt8195/mt8195.h b/sound/soc/sof/mediatek/mt8195/mt8195.h index 929424182357..7ffd523f936c 100644 --- a/sound/soc/sof/mediatek/mt8195/mt8195.h +++ b/sound/soc/sof/mediatek/mt8195/mt8195.h @@ -34,6 +34,7 @@ struct snd_sof_dev; #define ADSP_DRESET_SW BIT(1) #define ADSP_RUNSTALL BIT(3) #define STATVECTOR_SEL BIT(4) +#define ADSP_PWAIT BIT(16) #define DSP_PFAULTBUS 0x0028 #define DSP_PFAULTINFO 0x002c #define DSP_GPR00 0x0030 @@ -153,6 +154,10 @@ struct snd_sof_dev; #define DRAM_REMAP_SHIFT 12 #define DRAM_REMAP_MASK (BIT(DRAM_REMAP_SHIFT) - 1) +/* suspend dsp idle check interval and timeout */ +#define SUSPEND_DSP_IDLE_TIMEOUT_US 1000000 /* timeout to wait dsp idle, 1 sec */ +#define SUSPEND_DSP_IDLE_POLL_INTERVAL_US 500 /* 0.5 msec */ + void sof_hifixdsp_boot_sequence(struct snd_sof_dev *sdev, u32 boot_addr); void sof_hifixdsp_shutdown(struct snd_sof_dev *sdev); #endif -- cgit v1.2.3 From aa0d5f095093610e7274591d41a28381f60467a8 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Uwe=20Kleine-K=C3=B6nig?= Date: Sun, 5 Jun 2022 17:39:04 +0200 Subject: ASoC: Intel: broadwell: Make broadwell_disable_jack() return void MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit broadwell_disable_jack() returns zero unconditionally. Letting it return void instead makes it easier to see in the callers that there is no error to handle. This is a preparation for making platform remove callbacks return void. Signed-off-by: Uwe Kleine-König Acked-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220605153904.26921-1-u.kleine-koenig@pengutronix.de Signed-off-by: Mark Brown --- sound/soc/intel/boards/broadwell.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c index c30a9dca6801..b29d77dfb281 100644 --- a/sound/soc/intel/boards/broadwell.c +++ b/sound/soc/intel/boards/broadwell.c @@ -227,7 +227,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = { }, }; -static int broadwell_disable_jack(struct snd_soc_card *card) +static void broadwell_disable_jack(struct snd_soc_card *card) { struct snd_soc_component *component; @@ -239,13 +239,13 @@ static int broadwell_disable_jack(struct snd_soc_card *card) break; } } - - return 0; } static int broadwell_suspend(struct snd_soc_card *card) { - return broadwell_disable_jack(card); + broadwell_disable_jack(card); + + return 0; } static int broadwell_resume(struct snd_soc_card *card){ @@ -315,7 +315,9 @@ static int broadwell_audio_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); - return broadwell_disable_jack(card); + broadwell_disable_jack(card); + + return 0; } static struct platform_driver broadwell_audio = { -- cgit v1.2.3 From e6f08af6340eaf88e9eeff71bd4533eee9a04119 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Uwe=20Kleine-K=C3=B6nig?= Date: Sun, 5 Jun 2022 17:35:37 +0200 Subject: ASoC: simple-card-utils: Make asoc_simple_clean_reference() return void MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit asoc_simple_clean_reference() returns zero unconditionally. Letting it return void instead makes it easier to see in the caller that there is no error to handle. This is a preparation for making platform remove callbacks return void. Signed-off-by: Uwe Kleine-König Link: https://lore.kernel.org/r/20220605153537.26591-1-u.kleine-koenig@pengutronix.de Signed-off-by: Mark Brown --- sound/soc/generic/simple-card-utils.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index fa080f166345..0beda9739ebe 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -609,7 +609,7 @@ void asoc_simple_canonicalize_cpu(struct snd_soc_dai_link_component *cpus, } EXPORT_SYMBOL_GPL(asoc_simple_canonicalize_cpu); -int asoc_simple_clean_reference(struct snd_soc_card *card) +void asoc_simple_clean_reference(struct snd_soc_card *card) { struct snd_soc_dai_link *dai_link; struct snd_soc_dai_link_component *cpu; @@ -622,7 +622,6 @@ int asoc_simple_clean_reference(struct snd_soc_card *card) for_each_link_codecs(dai_link, j, codec) of_node_put(codec->of_node); } - return 0; } EXPORT_SYMBOL_GPL(asoc_simple_clean_reference); @@ -877,7 +876,9 @@ int asoc_simple_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); - return asoc_simple_clean_reference(card); + asoc_simple_clean_reference(card); + + return 0; } EXPORT_SYMBOL_GPL(asoc_simple_remove); -- cgit v1.2.3 From efe2178d1a32492f99e7f1f2568eea5c88a85729 Mon Sep 17 00:00:00 2001 From: Miaoqian Lin Date: Fri, 3 Jun 2022 16:42:41 +0400 Subject: ASoC: mediatek: mt8173-rt5650: Fix refcount leak in mt8173_rt5650_dev_probe of_parse_phandle() returns a node pointer with refcount incremented, we should use of_node_put() on it when not need anymore. Fix refcount leak in some error paths. Fixes: 0f83f9296d5c ("ASoC: mediatek: Add machine driver for ALC5650 codec") Signed-off-by: Miaoqian Lin Link: https://lore.kernel.org/r/20220603124243.31358-1-linmq006@gmail.com Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650.c b/sound/soc/mediatek/mt8173/mt8173-rt5650.c index d1c94acb4516..e05f2b0231fe 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650.c @@ -280,7 +280,8 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) if (!mt8173_rt5650_dais[DAI_LINK_CODEC_I2S].codecs[0].of_node) { dev_err(&pdev->dev, "Property 'audio-codec' missing or invalid\n"); - return -EINVAL; + ret = -EINVAL; + goto put_platform_node; } mt8173_rt5650_dais[DAI_LINK_CODEC_I2S].codecs[1].of_node = mt8173_rt5650_dais[DAI_LINK_CODEC_I2S].codecs[0].of_node; @@ -293,7 +294,7 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) dev_err(&pdev->dev, "%s codec_capture_dai name fail %d\n", __func__, ret); - return ret; + goto put_platform_node; } mt8173_rt5650_dais[DAI_LINK_CODEC_I2S].codecs[1].dai_name = codec_capture_dai; @@ -315,12 +316,14 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) if (!mt8173_rt5650_dais[DAI_LINK_HDMI_I2S].codecs->of_node) { dev_err(&pdev->dev, "Property 'audio-codec' missing or invalid\n"); - return -EINVAL; + ret = -EINVAL; + goto put_platform_node; } card->dev = &pdev->dev; ret = devm_snd_soc_register_card(&pdev->dev, card); +put_platform_node: of_node_put(platform_node); return ret; } -- cgit v1.2.3 From f9e9bdd5bb180325256e3bdfeb9c4c6526133478 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 6 Jun 2022 15:37:46 -0500 Subject: ASoC: Realtek/Maxim SoundWire codecs: disable pm_runtime on remove When binding/unbinding codec drivers, the following warnings are thrown: [ 107.266879] rt715-sdca sdw:3:025d:0714:01: Unbalanced pm_runtime_enable! [ 306.879700] rt711-sdca sdw:0:025d:0711:01: Unbalanced pm_runtime_enable! Add a remove callback for all Realtek/Maxim SoundWire codecs and remove this warning. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Rander Wang Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20220606203752.144159-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/max98373-sdw.c | 12 +++++++++++- sound/soc/codecs/rt1308-sdw.c | 11 +++++++++++ sound/soc/codecs/rt1316-sdw.c | 11 +++++++++++ sound/soc/codecs/rt5682-sdw.c | 5 ++++- sound/soc/codecs/rt700-sdw.c | 6 +++++- sound/soc/codecs/rt711-sdca-sdw.c | 6 +++++- sound/soc/codecs/rt711-sdw.c | 6 +++++- sound/soc/codecs/rt715-sdca-sdw.c | 12 ++++++++++++ sound/soc/codecs/rt715-sdw.c | 12 ++++++++++++ 9 files changed, 76 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/max98373-sdw.c b/sound/soc/codecs/max98373-sdw.c index f47e956d4f55..97b64477dde6 100644 --- a/sound/soc/codecs/max98373-sdw.c +++ b/sound/soc/codecs/max98373-sdw.c @@ -862,6 +862,16 @@ static int max98373_sdw_probe(struct sdw_slave *slave, return max98373_init(slave, regmap); } +static int max98373_sdw_remove(struct sdw_slave *slave) +{ + struct max98373_priv *max98373 = dev_get_drvdata(&slave->dev); + + if (max98373->first_hw_init) + pm_runtime_disable(&slave->dev); + + return 0; +} + #if defined(CONFIG_OF) static const struct of_device_id max98373_of_match[] = { { .compatible = "maxim,max98373", }, @@ -893,7 +903,7 @@ static struct sdw_driver max98373_sdw_driver = { .pm = &max98373_pm, }, .probe = max98373_sdw_probe, - .remove = NULL, + .remove = max98373_sdw_remove, .ops = &max98373_slave_ops, .id_table = max98373_id, }; diff --git a/sound/soc/codecs/rt1308-sdw.c b/sound/soc/codecs/rt1308-sdw.c index 1c11b42dd76e..72f673f278ee 100644 --- a/sound/soc/codecs/rt1308-sdw.c +++ b/sound/soc/codecs/rt1308-sdw.c @@ -691,6 +691,16 @@ static int rt1308_sdw_probe(struct sdw_slave *slave, return 0; } +static int rt1308_sdw_remove(struct sdw_slave *slave) +{ + struct rt1308_sdw_priv *rt1308 = dev_get_drvdata(&slave->dev); + + if (rt1308->first_hw_init) + pm_runtime_disable(&slave->dev); + + return 0; +} + static const struct sdw_device_id rt1308_id[] = { SDW_SLAVE_ENTRY_EXT(0x025d, 0x1308, 0x2, 0, 0), {}, @@ -750,6 +760,7 @@ static struct sdw_driver rt1308_sdw_driver = { .pm = &rt1308_pm, }, .probe = rt1308_sdw_probe, + .remove = rt1308_sdw_remove, .ops = &rt1308_slave_ops, .id_table = rt1308_id, }; diff --git a/sound/soc/codecs/rt1316-sdw.c b/sound/soc/codecs/rt1316-sdw.c index 60baa9ff1907..2d6b5f9d4d77 100644 --- a/sound/soc/codecs/rt1316-sdw.c +++ b/sound/soc/codecs/rt1316-sdw.c @@ -676,6 +676,16 @@ static int rt1316_sdw_probe(struct sdw_slave *slave, return rt1316_sdw_init(&slave->dev, regmap, slave); } +static int rt1316_sdw_remove(struct sdw_slave *slave) +{ + struct rt1316_sdw_priv *rt1316 = dev_get_drvdata(&slave->dev); + + if (rt1316->first_hw_init) + pm_runtime_disable(&slave->dev); + + return 0; +} + static const struct sdw_device_id rt1316_id[] = { SDW_SLAVE_ENTRY_EXT(0x025d, 0x1316, 0x3, 0x1, 0), {}, @@ -735,6 +745,7 @@ static struct sdw_driver rt1316_sdw_driver = { .pm = &rt1316_pm, }, .probe = rt1316_sdw_probe, + .remove = rt1316_sdw_remove, .ops = &rt1316_slave_ops, .id_table = rt1316_id, }; diff --git a/sound/soc/codecs/rt5682-sdw.c b/sound/soc/codecs/rt5682-sdw.c index 248257a2e4e0..f04e18c32489 100644 --- a/sound/soc/codecs/rt5682-sdw.c +++ b/sound/soc/codecs/rt5682-sdw.c @@ -719,9 +719,12 @@ static int rt5682_sdw_remove(struct sdw_slave *slave) { struct rt5682_priv *rt5682 = dev_get_drvdata(&slave->dev); - if (rt5682 && rt5682->hw_init) + if (rt5682->hw_init) cancel_delayed_work_sync(&rt5682->jack_detect_work); + if (rt5682->first_hw_init) + pm_runtime_disable(&slave->dev); + return 0; } diff --git a/sound/soc/codecs/rt700-sdw.c b/sound/soc/codecs/rt700-sdw.c index bda594899664..f7439e40ca8b 100644 --- a/sound/soc/codecs/rt700-sdw.c +++ b/sound/soc/codecs/rt700-sdw.c @@ -13,6 +13,7 @@ #include #include #include +#include #include #include #include "rt700.h" @@ -463,11 +464,14 @@ static int rt700_sdw_remove(struct sdw_slave *slave) { struct rt700_priv *rt700 = dev_get_drvdata(&slave->dev); - if (rt700 && rt700->hw_init) { + if (rt700->hw_init) { cancel_delayed_work_sync(&rt700->jack_detect_work); cancel_delayed_work_sync(&rt700->jack_btn_check_work); } + if (rt700->first_hw_init) + pm_runtime_disable(&slave->dev); + return 0; } diff --git a/sound/soc/codecs/rt711-sdca-sdw.c b/sound/soc/codecs/rt711-sdca-sdw.c index aaf5af153d3f..c722a2b0041f 100644 --- a/sound/soc/codecs/rt711-sdca-sdw.c +++ b/sound/soc/codecs/rt711-sdca-sdw.c @@ -11,6 +11,7 @@ #include #include #include +#include #include "rt711-sdca.h" #include "rt711-sdca-sdw.h" @@ -364,11 +365,14 @@ static int rt711_sdca_sdw_remove(struct sdw_slave *slave) { struct rt711_sdca_priv *rt711 = dev_get_drvdata(&slave->dev); - if (rt711 && rt711->hw_init) { + if (rt711->hw_init) { cancel_delayed_work_sync(&rt711->jack_detect_work); cancel_delayed_work_sync(&rt711->jack_btn_check_work); } + if (rt711->first_hw_init) + pm_runtime_disable(&slave->dev); + return 0; } diff --git a/sound/soc/codecs/rt711-sdw.c b/sound/soc/codecs/rt711-sdw.c index bda2cc9439c9..f49c94baa37c 100644 --- a/sound/soc/codecs/rt711-sdw.c +++ b/sound/soc/codecs/rt711-sdw.c @@ -13,6 +13,7 @@ #include #include #include +#include #include #include #include "rt711.h" @@ -464,12 +465,15 @@ static int rt711_sdw_remove(struct sdw_slave *slave) { struct rt711_priv *rt711 = dev_get_drvdata(&slave->dev); - if (rt711 && rt711->hw_init) { + if (rt711->hw_init) { cancel_delayed_work_sync(&rt711->jack_detect_work); cancel_delayed_work_sync(&rt711->jack_btn_check_work); cancel_work_sync(&rt711->calibration_work); } + if (rt711->first_hw_init) + pm_runtime_disable(&slave->dev); + return 0; } diff --git a/sound/soc/codecs/rt715-sdca-sdw.c b/sound/soc/codecs/rt715-sdca-sdw.c index 0ecd2948f7aa..13e731d16675 100644 --- a/sound/soc/codecs/rt715-sdca-sdw.c +++ b/sound/soc/codecs/rt715-sdca-sdw.c @@ -13,6 +13,7 @@ #include #include #include +#include #include #include #include "rt715-sdca.h" @@ -193,6 +194,16 @@ static int rt715_sdca_sdw_probe(struct sdw_slave *slave, return rt715_sdca_init(&slave->dev, mbq_regmap, regmap, slave); } +static int rt715_sdca_sdw_remove(struct sdw_slave *slave) +{ + struct rt715_sdca_priv *rt715 = dev_get_drvdata(&slave->dev); + + if (rt715->first_hw_init) + pm_runtime_disable(&slave->dev); + + return 0; +} + static const struct sdw_device_id rt715_sdca_id[] = { SDW_SLAVE_ENTRY_EXT(0x025d, 0x715, 0x3, 0x1, 0), SDW_SLAVE_ENTRY_EXT(0x025d, 0x714, 0x3, 0x1, 0), @@ -267,6 +278,7 @@ static struct sdw_driver rt715_sdw_driver = { .pm = &rt715_pm, }, .probe = rt715_sdca_sdw_probe, + .remove = rt715_sdca_sdw_remove, .ops = &rt715_sdca_slave_ops, .id_table = rt715_sdca_id, }; diff --git a/sound/soc/codecs/rt715-sdw.c b/sound/soc/codecs/rt715-sdw.c index a7b21b03c08b..b047bf87a100 100644 --- a/sound/soc/codecs/rt715-sdw.c +++ b/sound/soc/codecs/rt715-sdw.c @@ -14,6 +14,7 @@ #include #include #include +#include #include #include #include @@ -514,6 +515,16 @@ static int rt715_sdw_probe(struct sdw_slave *slave, return 0; } +static int rt715_sdw_remove(struct sdw_slave *slave) +{ + struct rt715_priv *rt715 = dev_get_drvdata(&slave->dev); + + if (rt715->first_hw_init) + pm_runtime_disable(&slave->dev); + + return 0; +} + static const struct sdw_device_id rt715_id[] = { SDW_SLAVE_ENTRY_EXT(0x025d, 0x714, 0x2, 0, 0), SDW_SLAVE_ENTRY_EXT(0x025d, 0x715, 0x2, 0, 0), @@ -575,6 +586,7 @@ static struct sdw_driver rt715_sdw_driver = { .pm = &rt715_pm, }, .probe = rt715_sdw_probe, + .remove = rt715_sdw_remove, .ops = &rt715_slave_ops, .id_table = rt715_id, }; -- cgit v1.2.3 From 716c2e7e1608a89423ec84398b99ff2fa855d161 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 6 Jun 2022 15:37:47 -0500 Subject: ASoC: rt711-sdca-sdw: fix calibrate mutex initialization In codec driver bind/unbind test, the following warning is thrown: DEBUG_LOCKS_WARN_ON(lock->magic != lock) ... [ 699.182495] rt711_sdca_jack_init+0x1b/0x1d0 [snd_soc_rt711_sdca] [ 699.182498] rt711_sdca_set_jack_detect+0x3b/0x90 [snd_soc_rt711_sdca] [ 699.182500] snd_soc_component_set_jack+0x24/0x50 [snd_soc_core] A quick check in the code shows that the 'calibrate_mutex' used by this driver are not initialized at probe time. Moving the initialization to the probe removes the issue. BugLink: https://github.com/thesofproject/linux/issues/3644 Signed-off-by: Pierre-Louis Bossart Reviewed-by: Rander Wang Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20220606203752.144159-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt711-sdca-sdw.c | 3 +++ sound/soc/codecs/rt711-sdca.c | 2 +- 2 files changed, 4 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt711-sdca-sdw.c b/sound/soc/codecs/rt711-sdca-sdw.c index c722a2b0041f..a085b2f530aa 100644 --- a/sound/soc/codecs/rt711-sdca-sdw.c +++ b/sound/soc/codecs/rt711-sdca-sdw.c @@ -373,6 +373,9 @@ static int rt711_sdca_sdw_remove(struct sdw_slave *slave) if (rt711->first_hw_init) pm_runtime_disable(&slave->dev); + mutex_destroy(&rt711->calibrate_mutex); + mutex_destroy(&rt711->disable_irq_lock); + return 0; } diff --git a/sound/soc/codecs/rt711-sdca.c b/sound/soc/codecs/rt711-sdca.c index 57629c18db38..af73bcb4560a 100644 --- a/sound/soc/codecs/rt711-sdca.c +++ b/sound/soc/codecs/rt711-sdca.c @@ -1412,6 +1412,7 @@ int rt711_sdca_init(struct device *dev, struct regmap *regmap, rt711->regmap = regmap; rt711->mbq_regmap = mbq_regmap; + mutex_init(&rt711->calibrate_mutex); mutex_init(&rt711->disable_irq_lock); /* @@ -1550,7 +1551,6 @@ int rt711_sdca_io_init(struct device *dev, struct sdw_slave *slave) rt711_sdca_jack_detect_handler); INIT_DELAYED_WORK(&rt711->jack_btn_check_work, rt711_sdca_btn_check_handler); - mutex_init(&rt711->calibrate_mutex); } /* calibration */ -- cgit v1.2.3 From 768ad6d80db2dbbb1bfbb5e616d701a0b560f12a Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 6 Jun 2022 15:37:48 -0500 Subject: ASoC: Intel: sof_sdw: handle errors on card registration If the card registration fails, typically because of deferred probes, the device properties added for headset codecs are not removed, which leads to kernel oopses in driver bind/unbind tests. We already clean-up the device properties when the card is removed, this code can be moved as a helper and called upon card registration errors. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Rander Wang Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20220606203752.144159-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 51 +++++++++++++++++++++++----------------- 1 file changed, 29 insertions(+), 22 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 1f00679b4240..ad826ad82d51 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -1398,6 +1398,33 @@ static struct snd_soc_card card_sof_sdw = { .late_probe = sof_sdw_card_late_probe, }; +static void mc_dailink_exit_loop(struct snd_soc_card *card) +{ + struct snd_soc_dai_link *link; + int ret; + int i, j; + + for (i = 0; i < ARRAY_SIZE(codec_info_list); i++) { + if (!codec_info_list[i].exit) + continue; + /* + * We don't need to call .exit function if there is no matched + * dai link found. + */ + for_each_card_prelinks(card, j, link) { + if (!strcmp(link->codecs[0].dai_name, + codec_info_list[i].dai_name)) { + ret = codec_info_list[i].exit(card, link); + if (ret) + dev_warn(card->dev, + "codec exit failed %d\n", + ret); + break; + } + } + } +} + static int mc_probe(struct platform_device *pdev) { struct snd_soc_card *card = &card_sof_sdw; @@ -1462,6 +1489,7 @@ static int mc_probe(struct platform_device *pdev) ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { dev_err(card->dev, "snd_soc_register_card failed %d\n", ret); + mc_dailink_exit_loop(card); return ret; } @@ -1473,29 +1501,8 @@ static int mc_probe(struct platform_device *pdev) static int mc_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); - struct snd_soc_dai_link *link; - int ret; - int i, j; - for (i = 0; i < ARRAY_SIZE(codec_info_list); i++) { - if (!codec_info_list[i].exit) - continue; - /* - * We don't need to call .exit function if there is no matched - * dai link found. - */ - for_each_card_prelinks(card, j, link) { - if (!strcmp(link->codecs[0].dai_name, - codec_info_list[i].dai_name)) { - ret = codec_info_list[i].exit(card, link); - if (ret) - dev_warn(&pdev->dev, - "codec exit failed %d\n", - ret); - break; - } - } - } + mc_dailink_exit_loop(card); return 0; } -- cgit v1.2.3 From 74d40901ebad7c466a95b1ae3c6891f1ba09786f Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 6 Jun 2022 15:37:49 -0500 Subject: ASoC: rt711: fix calibrate mutex initialization Follow the same flow as rt711-sdca and initialize all mutexes at probe time. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Rander Wang Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20220606203752.144159-5-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt711-sdw.c | 3 +++ sound/soc/codecs/rt711.c | 2 +- 2 files changed, 4 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt711-sdw.c b/sound/soc/codecs/rt711-sdw.c index f49c94baa37c..4fe68bcf2a7c 100644 --- a/sound/soc/codecs/rt711-sdw.c +++ b/sound/soc/codecs/rt711-sdw.c @@ -474,6 +474,9 @@ static int rt711_sdw_remove(struct sdw_slave *slave) if (rt711->first_hw_init) pm_runtime_disable(&slave->dev); + mutex_destroy(&rt711->calibrate_mutex); + mutex_destroy(&rt711->disable_irq_lock); + return 0; } diff --git a/sound/soc/codecs/rt711.c b/sound/soc/codecs/rt711.c index 9838fb4d5b9c..1e35ba433a7e 100644 --- a/sound/soc/codecs/rt711.c +++ b/sound/soc/codecs/rt711.c @@ -1204,6 +1204,7 @@ int rt711_init(struct device *dev, struct regmap *sdw_regmap, rt711->sdw_regmap = sdw_regmap; rt711->regmap = regmap; + mutex_init(&rt711->calibrate_mutex); mutex_init(&rt711->disable_irq_lock); /* @@ -1318,7 +1319,6 @@ int rt711_io_init(struct device *dev, struct sdw_slave *slave) rt711_jack_detect_handler); INIT_DELAYED_WORK(&rt711->jack_btn_check_work, rt711_btn_check_handler); - mutex_init(&rt711->calibrate_mutex); INIT_WORK(&rt711->calibration_work, rt711_calibration_work); schedule_work(&rt711->calibration_work); } -- cgit v1.2.3 From 05ba4c00fa9cb077a0dd91f5e6056951a787f63c Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 6 Jun 2022 15:37:50 -0500 Subject: ASoC: rt7*-sdw: harden jack_detect_handler Realtek headset codec drivers typically check if the card is instantiated before proceeding with the jack detection. The rt700, rt711 and rt711-sdca are however missing a check on the card pointer, which can lead to NULL dereferences encountered in driver bind/unbind tests. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Rander Wang Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20220606203752.144159-6-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt700.c | 2 +- sound/soc/codecs/rt711-sdca.c | 2 +- sound/soc/codecs/rt711.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt700.c b/sound/soc/codecs/rt700.c index af32295fa9b9..4a99d5f4706f 100644 --- a/sound/soc/codecs/rt700.c +++ b/sound/soc/codecs/rt700.c @@ -162,7 +162,7 @@ static void rt700_jack_detect_handler(struct work_struct *work) if (!rt700->hs_jack) return; - if (!rt700->component->card->instantiated) + if (!rt700->component->card || !rt700->component->card->instantiated) return; reg = RT700_VERB_GET_PIN_SENSE | RT700_HP_OUT; diff --git a/sound/soc/codecs/rt711-sdca.c b/sound/soc/codecs/rt711-sdca.c index af73bcb4560a..93b36f05cb56 100644 --- a/sound/soc/codecs/rt711-sdca.c +++ b/sound/soc/codecs/rt711-sdca.c @@ -294,7 +294,7 @@ static void rt711_sdca_jack_detect_handler(struct work_struct *work) if (!rt711->hs_jack) return; - if (!rt711->component->card->instantiated) + if (!rt711->component->card || !rt711->component->card->instantiated) return; /* SDW_SCP_SDCA_INT_SDCA_0 is used for jack detection */ diff --git a/sound/soc/codecs/rt711.c b/sound/soc/codecs/rt711.c index 1e35ba433a7e..2f445b27305a 100644 --- a/sound/soc/codecs/rt711.c +++ b/sound/soc/codecs/rt711.c @@ -242,7 +242,7 @@ static void rt711_jack_detect_handler(struct work_struct *work) if (!rt711->hs_jack) return; - if (!rt711->component->card->instantiated) + if (!rt711->component->card || !rt711->component->card->instantiated) return; if (pm_runtime_status_suspended(rt711->slave->dev.parent)) { -- cgit v1.2.3 From a49267a3bd102e3991514e884aac89cc0d0b5f35 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 6 Jun 2022 15:37:51 -0500 Subject: ASoC: codecs: rt700/rt711/rt711-sdca: initialize workqueues in probe The workqueues are initialized in the io_init functions, which isn't quite right. In some tests, this leads to warnings throw from __queue_delayed_work() WARN_ON_FUNCTION_MISMATCH(timer->function, delayed_work_timer_fn); Move all the initializations to the probe functions. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Rander Wang Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20220606203752.144159-7-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt700.c | 12 +++++------- sound/soc/codecs/rt711-sdca.c | 10 +++------- sound/soc/codecs/rt711.c | 12 +++++------- 3 files changed, 13 insertions(+), 21 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt700.c b/sound/soc/codecs/rt700.c index 4a99d5f4706f..7a6cf3434591 100644 --- a/sound/soc/codecs/rt700.c +++ b/sound/soc/codecs/rt700.c @@ -1115,6 +1115,11 @@ int rt700_init(struct device *dev, struct regmap *sdw_regmap, mutex_init(&rt700->disable_irq_lock); + INIT_DELAYED_WORK(&rt700->jack_detect_work, + rt700_jack_detect_handler); + INIT_DELAYED_WORK(&rt700->jack_btn_check_work, + rt700_btn_check_handler); + /* * Mark hw_init to false * HW init will be performed when device reports present @@ -1209,13 +1214,6 @@ int rt700_io_init(struct device *dev, struct sdw_slave *slave) /* Finish Initial Settings, set power to D3 */ regmap_write(rt700->regmap, RT700_SET_AUDIO_POWER_STATE, AC_PWRST_D3); - if (!rt700->first_hw_init) { - INIT_DELAYED_WORK(&rt700->jack_detect_work, - rt700_jack_detect_handler); - INIT_DELAYED_WORK(&rt700->jack_btn_check_work, - rt700_btn_check_handler); - } - /* * if set_jack callback occurred early than io_init, * we set up the jack detection function now diff --git a/sound/soc/codecs/rt711-sdca.c b/sound/soc/codecs/rt711-sdca.c index 93b36f05cb56..2b3b77577d1f 100644 --- a/sound/soc/codecs/rt711-sdca.c +++ b/sound/soc/codecs/rt711-sdca.c @@ -1415,6 +1415,9 @@ int rt711_sdca_init(struct device *dev, struct regmap *regmap, mutex_init(&rt711->calibrate_mutex); mutex_init(&rt711->disable_irq_lock); + INIT_DELAYED_WORK(&rt711->jack_detect_work, rt711_sdca_jack_detect_handler); + INIT_DELAYED_WORK(&rt711->jack_btn_check_work, rt711_sdca_btn_check_handler); + /* * Mark hw_init to false * HW init will be performed when device reports present @@ -1546,13 +1549,6 @@ int rt711_sdca_io_init(struct device *dev, struct sdw_slave *slave) rt711_sdca_index_update_bits(rt711, RT711_VENDOR_HDA_CTL, RT711_PUSH_BTN_INT_CTL0, 0x20, 0x00); - if (!rt711->first_hw_init) { - INIT_DELAYED_WORK(&rt711->jack_detect_work, - rt711_sdca_jack_detect_handler); - INIT_DELAYED_WORK(&rt711->jack_btn_check_work, - rt711_sdca_btn_check_handler); - } - /* calibration */ ret = rt711_sdca_calibration(rt711); if (ret < 0) diff --git a/sound/soc/codecs/rt711.c b/sound/soc/codecs/rt711.c index 2f445b27305a..5709a6bbe8fc 100644 --- a/sound/soc/codecs/rt711.c +++ b/sound/soc/codecs/rt711.c @@ -1207,6 +1207,10 @@ int rt711_init(struct device *dev, struct regmap *sdw_regmap, mutex_init(&rt711->calibrate_mutex); mutex_init(&rt711->disable_irq_lock); + INIT_DELAYED_WORK(&rt711->jack_detect_work, rt711_jack_detect_handler); + INIT_DELAYED_WORK(&rt711->jack_btn_check_work, rt711_btn_check_handler); + INIT_WORK(&rt711->calibration_work, rt711_calibration_work); + /* * Mark hw_init to false * HW init will be performed when device reports present @@ -1314,14 +1318,8 @@ int rt711_io_init(struct device *dev, struct sdw_slave *slave) if (rt711->first_hw_init) rt711_calibration(rt711); - else { - INIT_DELAYED_WORK(&rt711->jack_detect_work, - rt711_jack_detect_handler); - INIT_DELAYED_WORK(&rt711->jack_btn_check_work, - rt711_btn_check_handler); - INIT_WORK(&rt711->calibration_work, rt711_calibration_work); + else schedule_work(&rt711->calibration_work); - } /* * if set_jack callback occurred early than io_init, -- cgit v1.2.3 From e02b99e9b79ff272e8c299a3ee53bdb194ca885e Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 6 Jun 2022 15:37:52 -0500 Subject: ASoC: codecs: rt700/rt711/rt711-sdca: resume bus/codec in .set_jack_detect The .set_jack_detect() codec component callback is invoked during card registration, which happens when the machine driver is probed. The issue is that this callback can race with the bus suspend/resume, and IO timeouts can happen. This can be reproduced very easily if the machine driver is 'blacklisted' and manually probed after the bus suspends. The bus and codec need to be re-initialized using pm_runtime helpers. Previous contributions tried to make sure accesses to the bus during the .set_jack_detect() component callback only happen when the bus is active. This was done by changing the regcache status on a component remove. This is however a layering violation, the regcache status should only be modified on device probe, suspend and resume. The component probe/remove should not modify how the device regcache is handled. This solution also didn't handle all the possible race conditions, and the RT700 headset codec was not handled. This patch tries to resume the codec device before handling the jack initializations. In case the codec has not yet been initialized, pm_runtime may not be enabled yet, so we don't squelch the -EACCES error code and only stop the jack information. When the codec reports as attached, the jack initialization will proceed as usual. BugLink: https://github.com/thesofproject/linux/issues/3643 Fixes: 7ad4d237e7c4a ('ASoC: rt711-sdca: Add RT711 SDCA vendor-specific driver') Fixes: 899b12542b089 ('ASoC: rt711: add snd_soc_component remove callback') Signed-off-by: Pierre-Louis Bossart Reviewed-by: Rander Wang Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20220606203752.144159-8-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt700.c | 16 +++++++++++++--- sound/soc/codecs/rt711-sdca.c | 26 ++++++++++++++------------ sound/soc/codecs/rt711.c | 24 +++++++++++++----------- 3 files changed, 40 insertions(+), 26 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt700.c b/sound/soc/codecs/rt700.c index 7a6cf3434591..9bceeeb830b1 100644 --- a/sound/soc/codecs/rt700.c +++ b/sound/soc/codecs/rt700.c @@ -315,17 +315,27 @@ static int rt700_set_jack_detect(struct snd_soc_component *component, struct snd_soc_jack *hs_jack, void *data) { struct rt700_priv *rt700 = snd_soc_component_get_drvdata(component); + int ret; rt700->hs_jack = hs_jack; - if (!rt700->hw_init) { - dev_dbg(&rt700->slave->dev, - "%s hw_init not ready yet\n", __func__); + ret = pm_runtime_resume_and_get(component->dev); + if (ret < 0) { + if (ret != -EACCES) { + dev_err(component->dev, "%s: failed to resume %d\n", __func__, ret); + return ret; + } + + /* pm_runtime not enabled yet */ + dev_dbg(component->dev, "%s: skipping jack init for now\n", __func__); return 0; } rt700_jack_init(rt700); + pm_runtime_mark_last_busy(component->dev); + pm_runtime_put_autosuspend(component->dev); + return 0; } diff --git a/sound/soc/codecs/rt711-sdca.c b/sound/soc/codecs/rt711-sdca.c index 2b3b77577d1f..dfe3c9299ebd 100644 --- a/sound/soc/codecs/rt711-sdca.c +++ b/sound/soc/codecs/rt711-sdca.c @@ -487,16 +487,27 @@ static int rt711_sdca_set_jack_detect(struct snd_soc_component *component, struct snd_soc_jack *hs_jack, void *data) { struct rt711_sdca_priv *rt711 = snd_soc_component_get_drvdata(component); + int ret; rt711->hs_jack = hs_jack; - if (!rt711->hw_init) { - dev_dbg(&rt711->slave->dev, - "%s hw_init not ready yet\n", __func__); + ret = pm_runtime_resume_and_get(component->dev); + if (ret < 0) { + if (ret != -EACCES) { + dev_err(component->dev, "%s: failed to resume %d\n", __func__, ret); + return ret; + } + + /* pm_runtime not enabled yet */ + dev_dbg(component->dev, "%s: skipping jack init for now\n", __func__); return 0; } rt711_sdca_jack_init(rt711); + + pm_runtime_mark_last_busy(component->dev); + pm_runtime_put_autosuspend(component->dev); + return 0; } @@ -1190,14 +1201,6 @@ static int rt711_sdca_probe(struct snd_soc_component *component) return 0; } -static void rt711_sdca_remove(struct snd_soc_component *component) -{ - struct rt711_sdca_priv *rt711 = snd_soc_component_get_drvdata(component); - - regcache_cache_only(rt711->regmap, true); - regcache_cache_only(rt711->mbq_regmap, true); -} - static const struct snd_soc_component_driver soc_sdca_dev_rt711 = { .probe = rt711_sdca_probe, .controls = rt711_sdca_snd_controls, @@ -1207,7 +1210,6 @@ static const struct snd_soc_component_driver soc_sdca_dev_rt711 = { .dapm_routes = rt711_sdca_audio_map, .num_dapm_routes = ARRAY_SIZE(rt711_sdca_audio_map), .set_jack = rt711_sdca_set_jack_detect, - .remove = rt711_sdca_remove, .endianness = 1, }; diff --git a/sound/soc/codecs/rt711.c b/sound/soc/codecs/rt711.c index 5709a6bbe8fc..9df800abfc2d 100644 --- a/sound/soc/codecs/rt711.c +++ b/sound/soc/codecs/rt711.c @@ -457,17 +457,27 @@ static int rt711_set_jack_detect(struct snd_soc_component *component, struct snd_soc_jack *hs_jack, void *data) { struct rt711_priv *rt711 = snd_soc_component_get_drvdata(component); + int ret; rt711->hs_jack = hs_jack; - if (!rt711->hw_init) { - dev_dbg(&rt711->slave->dev, - "%s hw_init not ready yet\n", __func__); + ret = pm_runtime_resume_and_get(component->dev); + if (ret < 0) { + if (ret != -EACCES) { + dev_err(component->dev, "%s: failed to resume %d\n", __func__, ret); + return ret; + } + + /* pm_runtime not enabled yet */ + dev_dbg(component->dev, "%s: skipping jack init for now\n", __func__); return 0; } rt711_jack_init(rt711); + pm_runtime_mark_last_busy(component->dev); + pm_runtime_put_autosuspend(component->dev); + return 0; } @@ -932,13 +942,6 @@ static int rt711_probe(struct snd_soc_component *component) return 0; } -static void rt711_remove(struct snd_soc_component *component) -{ - struct rt711_priv *rt711 = snd_soc_component_get_drvdata(component); - - regcache_cache_only(rt711->regmap, true); -} - static const struct snd_soc_component_driver soc_codec_dev_rt711 = { .probe = rt711_probe, .set_bias_level = rt711_set_bias_level, @@ -949,7 +952,6 @@ static const struct snd_soc_component_driver soc_codec_dev_rt711 = { .dapm_routes = rt711_audio_map, .num_dapm_routes = ARRAY_SIZE(rt711_audio_map), .set_jack = rt711_set_jack_detect, - .remove = rt711_remove, .endianness = 1, }; -- cgit v1.2.3 From b09654e39c89a86680528345f3a95b832236ee82 Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Wed, 8 Jun 2022 09:23:38 +0100 Subject: ASoC: mediatek: mt8186: Fix a handful of spelling mistakes There are several spelling mistakes in dev_err messages. Fix them. Signed-off-by: Colin Ian King Link: https://lore.kernel.org/r/20220608082338.2083456-1-colin.i.king@gmail.com Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8186/mt8186-afe-gpio.c | 8 ++++---- sound/soc/mediatek/mt8186/mt8186-dai-adda.c | 2 +- 2 files changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/mediatek/mt8186/mt8186-afe-gpio.c b/sound/soc/mediatek/mt8186/mt8186-afe-gpio.c index 5ba28095b7da..255ffba637d3 100644 --- a/sound/soc/mediatek/mt8186/mt8186-afe-gpio.c +++ b/sound/soc/mediatek/mt8186/mt8186-afe-gpio.c @@ -137,25 +137,25 @@ static int mt8186_afe_gpio_adda_dl(struct device *dev, bool enable) if (enable) { ret = mt8186_afe_gpio_select(dev, MT8186_AFE_GPIO_CLK_MOSI_ON); if (ret) { - dev_err(dev, "%s(), MOSI CLK ON slect fail!\n", __func__); + dev_err(dev, "%s(), MOSI CLK ON select fail!\n", __func__); return ret; } ret = mt8186_afe_gpio_select(dev, MT8186_AFE_GPIO_DAT_MOSI_ON); if (ret) { - dev_err(dev, "%s(), MOSI DAT ON slect fail!\n", __func__); + dev_err(dev, "%s(), MOSI DAT ON select fail!\n", __func__); return ret; } } else { ret = mt8186_afe_gpio_select(dev, MT8186_AFE_GPIO_DAT_MOSI_OFF); if (ret) { - dev_err(dev, "%s(), MOSI DAT OFF slect fail!\n", __func__); + dev_err(dev, "%s(), MOSI DAT OFF select fail!\n", __func__); return ret; } ret = mt8186_afe_gpio_select(dev, MT8186_AFE_GPIO_CLK_MOSI_OFF); if (ret) { - dev_err(dev, "%s(), MOSI CLK ON slect fail!\n", __func__); + dev_err(dev, "%s(), MOSI CLK ON select fail!\n", __func__); return ret; } } diff --git a/sound/soc/mediatek/mt8186/mt8186-dai-adda.c b/sound/soc/mediatek/mt8186/mt8186-dai-adda.c index c66861fd197d..db71b032770d 100644 --- a/sound/soc/mediatek/mt8186/mt8186-dai-adda.c +++ b/sound/soc/mediatek/mt8186/mt8186-dai-adda.c @@ -341,7 +341,7 @@ static int mtk_adda_mtkaif_cfg_event(struct snd_soc_dapm_widget *w, if (afe_priv->mtkaif_chosen_phase[0] < 0 || afe_priv->mtkaif_chosen_phase[1] < 0) { dev_err(afe->dev, - "%s(), skip dealy setting mtkaif_chosen_phase[0/1]:%d/%d\n", + "%s(), skip delay setting mtkaif_chosen_phase[0/1]:%d/%d\n", __func__, afe_priv->mtkaif_chosen_phase[0], afe_priv->mtkaif_chosen_phase[1]); -- cgit v1.2.3 From ec3ad554b956d5dbefa1962c419f164ba223e6b3 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 8 Jun 2022 02:09:16 +0000 Subject: ASoC: ak4613: cares Simple-Audio-Card case for TDM Renesas is the only user of ak4613 on upstream for now, and commit f28dbaa958fbd8 ("ASoC: ak4613: add TDM256 support") added TDM256 support. Renesas tested part of it, because of board connection. It was assuming ak4613 is probed via Audio-Graph-Card, but it might be probed via Simple-Audio-Card either. It will indicates WARNING in such case. This patch fixup it. Reported-by: Geert Uytterhoeven Signed-off-by: Kuninori Morimoto Tested-by: Geert Uytterhoeven Link: https://lore.kernel.org/r/87h74v29f7.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/codecs/ak4613.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c index 55e773f92122..93606e5afd8f 100644 --- a/sound/soc/codecs/ak4613.c +++ b/sound/soc/codecs/ak4613.c @@ -868,10 +868,12 @@ static void ak4613_parse_of(struct ak4613_priv *priv, /* * connected STDI + * TDM support is assuming it is probed via Audio-Graph-Card style here. + * Default is SDTIx1 if it was probed via Simple-Audio-Card for now. */ sdti_num = of_graph_get_endpoint_count(np); - if (WARN_ON((sdti_num > 3) || (sdti_num < 1))) - return; + if ((sdti_num >= SDTx_MAX) || (sdti_num < 1)) + sdti_num = 1; AK4613_CONFIG_SDTI_set(priv, sdti_num); } -- cgit v1.2.3 From ff87d619ac180444db297f043962a5c325ded47b Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Thu, 19 May 2022 20:36:48 +0800 Subject: ASoC: fsl_sai: Enable MCTL_MCLK_EN bit for master mode On i.MX8MM, the MCTL_MCLK_EN bit it is not only the gate for MCLK output to PAD, but also the gate bit between root clock and SAI module, So it is need to be enabled for master mode, otherwise there is no bclk generated. Signed-off-by: Shengjiu Wang Link: https://lore.kernel.org/r/1652963808-14515-1-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index b65c9c7cf54a..b4dd3122c45e 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -437,6 +437,12 @@ static int fsl_sai_set_bclk(struct snd_soc_dai *dai, bool tx, u32 freq) FSL_SAI_CR2_DIV_MASK | FSL_SAI_CR2_BYP, savediv / 2 - 1); + if (sai->soc_data->max_register >= FSL_SAI_MCTL) { + /* SAI is in master mode at this point, so enable MCLK */ + regmap_update_bits(sai->regmap, FSL_SAI_MCTL, + FSL_SAI_MCTL_MCLK_EN, FSL_SAI_MCTL_MCLK_EN); + } + return 0; } -- cgit v1.2.3 From 537b4a0c8b9490d762e70c0ecec38144c83d0c37 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 9 Jun 2022 11:59:47 +0300 Subject: ASoC: SOF: Intel: hda-dsp: Expose hda_dsp_core_power_up() The hda_dsp_core_power_up() needs to be exposed so that it can be used in hda-loader.c to correct the boot flow. The first step must not unstall the core, it should only power up the core(s). Add sanity check for the core_mask while exposing it to be safe. Complements: 2a68ff846164 ("ASoC: SOF: Intel: hda: Revisit IMR boot sequence") Signed-off-by: Peter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/20220609085949.29062-2-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-dsp.c | 10 +++++++++- sound/soc/sof/intel/hda.h | 1 + 2 files changed, 10 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index 000ea906670c..e24eea725acb 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -181,12 +181,20 @@ int hda_dsp_core_run(struct snd_sof_dev *sdev, unsigned int core_mask) * Power Management. */ -static int hda_dsp_core_power_up(struct snd_sof_dev *sdev, unsigned int core_mask) +int hda_dsp_core_power_up(struct snd_sof_dev *sdev, unsigned int core_mask) { + struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata; + const struct sof_intel_dsp_desc *chip = hda->desc; unsigned int cpa; u32 adspcs; int ret; + /* restrict core_mask to host managed cores mask */ + core_mask &= chip->host_managed_cores_mask; + /* return if core_mask is not valid */ + if (!core_mask) + return 0; + /* update bits */ snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, HDA_DSP_REG_ADSPCS, HDA_DSP_ADSPCS_SPA_MASK(core_mask), diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 3e0f7b0c586a..0f57ef5d9b8e 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -497,6 +497,7 @@ struct sof_intel_hda_stream { */ int hda_dsp_probe(struct snd_sof_dev *sdev); int hda_dsp_remove(struct snd_sof_dev *sdev); +int hda_dsp_core_power_up(struct snd_sof_dev *sdev, unsigned int core_mask); int hda_dsp_core_run(struct snd_sof_dev *sdev, unsigned int core_mask); int hda_dsp_enable_core(struct snd_sof_dev *sdev, unsigned int core_mask); int hda_dsp_core_reset_power_down(struct snd_sof_dev *sdev, -- cgit v1.2.3 From fcb3c775f7073410965ce9414ddb2a1f339c502b Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 9 Jun 2022 11:59:48 +0300 Subject: ASoC: SOF: Intel: hda-loader: Make sure that the fw load sequence is followed The hda_dsp_enable_core() is powering up _and_ unstall the core in one call while the first step of the firmware loading must not unstall the core. The core can be unstalled only after the set cpb_cfp and the configuration of the IPC register for the ROM_CONTROL message. Complements: 2a68ff846164 ("ASoC: SOF: Intel: hda: Revisit IMR boot sequence") Signed-off-by: Peter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/20220609085949.29062-3-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-loader.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c index 64290125d7cd..103e62bcfa82 100644 --- a/sound/soc/sof/intel/hda-loader.c +++ b/sound/soc/sof/intel/hda-loader.c @@ -110,7 +110,7 @@ static int cl_dsp_init(struct snd_sof_dev *sdev, int stream_tag, bool imr_boot) int ret; /* step 1: power up corex */ - ret = hda_dsp_enable_core(sdev, chip->host_managed_cores_mask); + ret = hda_dsp_core_power_up(sdev, chip->host_managed_cores_mask); if (ret < 0) { if (hda->boot_iteration == HDA_FW_BOOT_ATTEMPTS) dev_err(sdev->dev, "error: dsp core 0/1 power up failed\n"); -- cgit v1.2.3 From 4643e10a17e549467420aaeeb35c9b3480716618 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 9 Jun 2022 11:59:49 +0300 Subject: ASoC: SOF: Intel: hda-loader: Clarify the cl_dsp_init() flow Update the comment for the cl_dsp_init() to clarify what is done by the function and use the chip->init_core_mask instead of BIT(0) when unstalling/running the init core. Complements: 2a68ff846164 ("ASoC: SOF: Intel: hda: Revisit IMR boot sequence") Signed-off-by: Peter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/20220609085949.29062-4-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-loader.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c index 103e62bcfa82..d3ec5996a9a3 100644 --- a/sound/soc/sof/intel/hda-loader.c +++ b/sound/soc/sof/intel/hda-loader.c @@ -95,9 +95,9 @@ out_put: } /* - * first boot sequence has some extra steps. core 0 waits for power - * status on core 1, so power up core 1 also momentarily, keep it in - * reset/stall and then turn it off + * first boot sequence has some extra steps. + * power on all host managed cores and only unstall/run the boot core to boot the + * DSP then turn off all non boot cores (if any) is powered on. */ static int cl_dsp_init(struct snd_sof_dev *sdev, int stream_tag, bool imr_boot) { @@ -127,7 +127,7 @@ static int cl_dsp_init(struct snd_sof_dev *sdev, int stream_tag, bool imr_boot) snd_sof_dsp_write(sdev, HDA_DSP_BAR, chip->ipc_req, ipc_hdr); /* step 3: unset core 0 reset state & unstall/run core 0 */ - ret = hda_dsp_core_run(sdev, BIT(0)); + ret = hda_dsp_core_run(sdev, chip->init_core_mask); if (ret < 0) { if (hda->boot_iteration == HDA_FW_BOOT_ATTEMPTS) dev_err(sdev->dev, -- cgit v1.2.3 From 142d456204cf4dabe18be59e043d806440f609d4 Mon Sep 17 00:00:00 2001 From: Minghao Chi Date: Mon, 6 Jun 2022 03:37:05 +0000 Subject: ASoC: imx-audmux: remove unnecessary check of clk_disable_unprepare/clk_prepare_enable Because clk_disable_unprepare/clk_prepare_enable already checked NULL clock parameter, so the additional checks are unnecessary, just remove them. Reported-by: Zeal Robot Signed-off-by: Minghao Chi Acked-by: Shengjiu Wang Link: https://lore.kernel.org/r/20220606033705.291048-1-chi.minghao@zte.com.cn Signed-off-by: Mark Brown --- sound/soc/fsl/imx-audmux.c | 22 ++++++++-------------- 1 file changed, 8 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index a8e5e0f57faf..50b71e5d4589 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -62,17 +62,14 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf, uintptr_t port = (uintptr_t)file->private_data; u32 pdcr, ptcr; - if (audmux_clk) { - ret = clk_prepare_enable(audmux_clk); - if (ret) - return ret; - } + ret = clk_prepare_enable(audmux_clk); + if (ret) + return ret; ptcr = readl(audmux_base + IMX_AUDMUX_V2_PTCR(port)); pdcr = readl(audmux_base + IMX_AUDMUX_V2_PDCR(port)); - if (audmux_clk) - clk_disable_unprepare(audmux_clk); + clk_disable_unprepare(audmux_clk); buf = kmalloc(PAGE_SIZE, GFP_KERNEL); if (!buf) @@ -209,17 +206,14 @@ int imx_audmux_v2_configure_port(unsigned int port, unsigned int ptcr, if (!audmux_base) return -ENOSYS; - if (audmux_clk) { - ret = clk_prepare_enable(audmux_clk); - if (ret) - return ret; - } + ret = clk_prepare_enable(audmux_clk); + if (ret) + return ret; writel(ptcr, audmux_base + IMX_AUDMUX_V2_PTCR(port)); writel(pdcr, audmux_base + IMX_AUDMUX_V2_PDCR(port)); - if (audmux_clk) - clk_disable_unprepare(audmux_clk); + clk_disable_unprepare(audmux_clk); return 0; } -- cgit v1.2.3 From 5702b838dd9a8be634f9c6bdfd769422c26e9162 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 10 Jun 2022 11:47:35 +0300 Subject: ASoC: SOF: ipc3-topology: Move and correct size checks in sof_ipc3_control_load_bytes() Move the size checks prior to allocating memory as these checks do not need the data to be allocated and in case of an error we would not need to free the allocation. The max size must not be less than the size of struct sof_ipc_ctrl_data + struct sof_abi_hdr as the ABI header needs to be present under all circumstances. The check was incorrectly used or between the two size checks. Fixes: b5cee8feb1d4 ("ASoC: SOF: topology: Make control parsing IPC agnostic") Signed-off-by: Peter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20220610084735.19397-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc3-topology.c | 23 +++++++++++------------ 1 file changed, 11 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/ipc3-topology.c b/sound/soc/sof/ipc3-topology.c index 043554d7cb4a..10740c55294d 100644 --- a/sound/soc/sof/ipc3-topology.c +++ b/sound/soc/sof/ipc3-topology.c @@ -1577,24 +1577,23 @@ static int sof_ipc3_control_load_bytes(struct snd_sof_dev *sdev, struct snd_sof_ struct sof_ipc_ctrl_data *cdata; int ret; - scontrol->ipc_control_data = kzalloc(scontrol->max_size, GFP_KERNEL); - if (!scontrol->ipc_control_data) - return -ENOMEM; - - if (scontrol->max_size < sizeof(*cdata) || - scontrol->max_size < sizeof(struct sof_abi_hdr)) { - ret = -EINVAL; - goto err; + if (scontrol->max_size < (sizeof(*cdata) + sizeof(struct sof_abi_hdr))) { + dev_err(sdev->dev, "%s: insufficient size for a bytes control: %zu.\n", + __func__, scontrol->max_size); + return -EINVAL; } - /* init the get/put bytes data */ if (scontrol->priv_size > scontrol->max_size - sizeof(*cdata)) { - dev_err(sdev->dev, "err: bytes data size %zu exceeds max %zu.\n", + dev_err(sdev->dev, + "%s: bytes data size %zu exceeds max %zu.\n", __func__, scontrol->priv_size, scontrol->max_size - sizeof(*cdata)); - ret = -EINVAL; - goto err; + return -EINVAL; } + scontrol->ipc_control_data = kzalloc(scontrol->max_size, GFP_KERNEL); + if (!scontrol->ipc_control_data) + return -ENOMEM; + scontrol->size = sizeof(struct sof_ipc_ctrl_data) + scontrol->priv_size; cdata = scontrol->ipc_control_data; -- cgit v1.2.3 From 03f69725749f453b9a4d454a92805f8eb5f095c2 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 10 Jun 2022 11:35:44 +0300 Subject: ASoC: SOF: make ctx_store and ctx_restore as optional Commit 657774acd00f ("ASoC: SOF: Make sof_suspend/resume IPC agnostic") did not marked ctx_store and ctx_restore as Optional. Fixes: 657774acd00f ("ASoC: SOF: Make sof_suspend/resume IPC agnostic") Signed-off-by: Peter Ujfalusi Reviewed-by: Ranjani Sridharan Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220610083549.16773-2-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/sof-priv.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index 9d7f53ff9c70..58bcb8d6f72b 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -376,8 +376,8 @@ struct sof_ipc_fw_tracing_ops { /** * struct sof_ipc_pm_ops - IPC-specific PM ops - * @ctx_save: Function pointer for context save - * @ctx_restore: Function pointer for context restore + * @ctx_save: Optional function pointer for context save + * @ctx_restore: Optional function pointer for context restore */ struct sof_ipc_pm_ops { int (*ctx_save)(struct snd_sof_dev *sdev); -- cgit v1.2.3 From b41252d8820c7009078c3d401a807a9da899075f Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 10 Jun 2022 11:35:45 +0300 Subject: ASoC: SOF: sof_ipc_pm_ops: Add support for DSP core power management Add a new ops for handling DSP core power state which can be used to tell the DSP to turn on/off a core (or to inform it that a core is going to be turned on/off if the core is host managed). Signed-off-by: Peter Ujfalusi Reviewed-by: Ranjani Sridharan Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220610083549.16773-3-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/sof-priv.h | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index 58bcb8d6f72b..0544eb6a2322 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -378,10 +378,12 @@ struct sof_ipc_fw_tracing_ops { * struct sof_ipc_pm_ops - IPC-specific PM ops * @ctx_save: Optional function pointer for context save * @ctx_restore: Optional function pointer for context restore + * @set_core_state: Optional function pointer for turning on/off a DSP core */ struct sof_ipc_pm_ops { int (*ctx_save)(struct snd_sof_dev *sdev); int (*ctx_restore)(struct snd_sof_dev *sdev); + int (*set_core_state)(struct snd_sof_dev *sdev, int core_idx, bool on); }; /** -- cgit v1.2.3 From 0a047dafefafbccc931fab2d187ce75c302088d5 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 10 Jun 2022 11:35:46 +0300 Subject: ASoC: SOF: ipc3: Add set_core_state pm_ops implementation IPC3 uses sof_ipc_pm_core_config message (SOF_IPC_GLB_PM_MSG | SOF_IPC_PM_CORE_ENABLE) to enable/disable cores managed by the DSP. The core state is set via a single bitfield, if the bit is 1 the core should be on, if it is 0 then it is off. Signed-off-by: Peter Ujfalusi Reviewed-by: Ranjani Sridharan Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220610083549.16773-4-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc3.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/ipc3.c b/sound/soc/sof/ipc3.c index dff5feaad370..ba81a6c490e9 100644 --- a/sound/soc/sof/ipc3.c +++ b/sound/soc/sof/ipc3.c @@ -1037,6 +1037,23 @@ static void sof_ipc3_rx_msg(struct snd_sof_dev *sdev) ipc3_log_header(sdev->dev, "ipc rx done", hdr.cmd); } +static int sof_ipc3_set_core_state(struct snd_sof_dev *sdev, int core_idx, bool on) +{ + struct sof_ipc_pm_core_config core_cfg = { + .hdr.size = sizeof(core_cfg), + .hdr.cmd = SOF_IPC_GLB_PM_MSG | SOF_IPC_PM_CORE_ENABLE, + }; + struct sof_ipc_reply reply; + + if (on) + core_cfg.enable_mask = sdev->enabled_cores_mask | BIT(core_idx); + else + core_cfg.enable_mask = sdev->enabled_cores_mask & ~BIT(core_idx); + + return sof_ipc3_tx_msg(sdev, &core_cfg, sizeof(core_cfg), + &reply, sizeof(reply), false); +} + static int sof_ipc3_ctx_ipc(struct snd_sof_dev *sdev, int cmd) { struct sof_ipc_pm_ctx pm_ctx = { @@ -1063,6 +1080,7 @@ static int sof_ipc3_ctx_restore(struct snd_sof_dev *sdev) static const struct sof_ipc_pm_ops ipc3_pm_ops = { .ctx_save = sof_ipc3_ctx_save, .ctx_restore = sof_ipc3_ctx_restore, + .set_core_state = sof_ipc3_set_core_state, }; const struct sof_ipc_ops ipc3_ops = { -- cgit v1.2.3 From bd3df9ff25b32b66630c283bb2e065e8bb822e72 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 10 Jun 2022 11:35:47 +0300 Subject: ASoC: SOF: ipc4: Add set_core_state pm_ops implementation IPC4 uses the SET_DX message to enable/disable cores managed by the DSP. The dx_state.core_mask indicates which core is going to change state, the dx_state.dx_mask is to power on (1) or off (0) the core. In the dx_mask only those bits (cores) checked which bit is set in the core_mask, other bits (cores) ignored. Signed-off-by: Peter Ujfalusi Reviewed-by: Ranjani Sridharan Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220610083549.16773-5-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4.c | 26 ++++++++++++++++++++++++++ 1 file changed, 26 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/ipc4.c b/sound/soc/sof/ipc4.c index 658802c86685..b2cb92745ec6 100644 --- a/sound/soc/sof/ipc4.c +++ b/sound/soc/sof/ipc4.c @@ -597,10 +597,36 @@ static void sof_ipc4_rx_msg(struct snd_sof_dev *sdev) } } +static int sof_ipc4_set_core_state(struct snd_sof_dev *sdev, int core_idx, bool on) +{ + struct sof_ipc4_dx_state_info dx_state; + struct sof_ipc4_msg msg; + + dx_state.core_mask = BIT(core_idx); + if (on) + dx_state.dx_mask = BIT(core_idx); + else + dx_state.dx_mask = 0; + + msg.primary = SOF_IPC4_MSG_TYPE_SET(SOF_IPC4_MOD_SET_DX); + msg.primary |= SOF_IPC4_MSG_DIR(SOF_IPC4_MSG_REQUEST); + msg.primary |= SOF_IPC4_MSG_TARGET(SOF_IPC4_MODULE_MSG); + msg.extension = 0; + msg.data_ptr = &dx_state; + msg.data_size = sizeof(dx_state); + + return sof_ipc4_tx_msg(sdev, &msg, msg.data_size, NULL, 0, false); +} + +static const struct sof_ipc_pm_ops ipc4_pm_ops = { + .set_core_state = sof_ipc4_set_core_state, +}; + const struct sof_ipc_ops ipc4_ops = { .tx_msg = sof_ipc4_tx_msg, .rx_msg = sof_ipc4_rx_msg, .set_get_data = sof_ipc4_set_get_data, .get_reply = sof_ipc4_get_reply, + .pm = &ipc4_pm_ops, .fw_loader = &ipc4_loader_ops, }; -- cgit v1.2.3 From 7a5677407300e8ba6af95e66f4e8cfe23059f4a7 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 10 Jun 2022 11:35:48 +0300 Subject: ASoC: SOF: Intel: Switch to use the generic pm_ops.set_core_state Instead of craft and send an IPC(3) message in hda_dsp_core_get(), tgl_dsp_core_get() and tgl_dsp_core_put(), use the generic ops for handling the IPC dependent implementation of core power on/off. Signed-off-by: Peter Ujfalusi Reviewed-by: Ranjani Sridharan Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220610083549.16773-6-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-dsp.c | 15 ++++++--------- sound/soc/sof/intel/tgl.c | 30 ++++++++++-------------------- 2 files changed, 16 insertions(+), 29 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index 000ea906670c..3a70f441a8d5 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -932,13 +932,7 @@ void hda_dsp_d0i3_work(struct work_struct *work) int hda_dsp_core_get(struct snd_sof_dev *sdev, int core) { - struct sof_ipc_pm_core_config pm_core_config = { - .hdr = { - .cmd = SOF_IPC_GLB_PM_MSG | SOF_IPC_PM_CORE_ENABLE, - .size = sizeof(pm_core_config), - }, - .enable_mask = sdev->enabled_cores_mask | BIT(core), - }; + const struct sof_ipc_pm_ops *pm_ops = sdev->ipc->ops->pm; int ret, ret1; /* power up core */ @@ -953,9 +947,12 @@ int hda_dsp_core_get(struct snd_sof_dev *sdev, int core) if (sdev->fw_state != SOF_FW_BOOT_COMPLETE || core == SOF_DSP_PRIMARY_CORE) return 0; + /* No need to continue the set_core_state ops is not available */ + if (!pm_ops->set_core_state) + return 0; + /* Now notify DSP for secondary cores */ - ret = sof_ipc_tx_message(sdev->ipc, &pm_core_config, sizeof(pm_core_config), - &pm_core_config, sizeof(pm_core_config)); + ret = pm_ops->set_core_state(sdev, core, true); if (ret < 0) { dev_err(sdev->dev, "failed to enable secondary core '%d' failed with %d\n", core, ret); diff --git a/sound/soc/sof/intel/tgl.c b/sound/soc/sof/intel/tgl.c index 1ddc492f1b13..dcad7c382de6 100644 --- a/sound/soc/sof/intel/tgl.c +++ b/sound/soc/sof/intel/tgl.c @@ -24,40 +24,30 @@ static const struct snd_sof_debugfs_map tgl_dsp_debugfs[] = { static int tgl_dsp_core_get(struct snd_sof_dev *sdev, int core) { - struct sof_ipc_pm_core_config pm_core_config = { - .hdr = { - .cmd = SOF_IPC_GLB_PM_MSG | SOF_IPC_PM_CORE_ENABLE, - .size = sizeof(pm_core_config), - }, - .enable_mask = sdev->enabled_cores_mask | BIT(core), - }; + const struct sof_ipc_pm_ops *pm_ops = sdev->ipc->ops->pm; /* power up primary core if not already powered up and return */ if (core == SOF_DSP_PRIMARY_CORE) return hda_dsp_enable_core(sdev, BIT(core)); - /* notify DSP for secondary cores */ - return sof_ipc_tx_message(sdev->ipc, &pm_core_config, sizeof(pm_core_config), - &pm_core_config, sizeof(pm_core_config)); + if (pm_ops->set_core_state) + return pm_ops->set_core_state(sdev, core, true); + + return 0; } static int tgl_dsp_core_put(struct snd_sof_dev *sdev, int core) { - struct sof_ipc_pm_core_config pm_core_config = { - .hdr = { - .cmd = SOF_IPC_GLB_PM_MSG | SOF_IPC_PM_CORE_ENABLE, - .size = sizeof(pm_core_config), - }, - .enable_mask = sdev->enabled_cores_mask & ~BIT(core), - }; + const struct sof_ipc_pm_ops *pm_ops = sdev->ipc->ops->pm; /* power down primary core and return */ if (core == SOF_DSP_PRIMARY_CORE) return hda_dsp_core_reset_power_down(sdev, BIT(core)); - /* notify DSP for secondary cores */ - return sof_ipc_tx_message(sdev->ipc, &pm_core_config, sizeof(pm_core_config), - &pm_core_config, sizeof(pm_core_config)); + if (pm_ops->set_core_state) + return pm_ops->set_core_state(sdev, core, false); + + return 0; } /* Tigerlake ops */ -- cgit v1.2.3 From 63b9069653a710b08d5fd174ac05d43711356541 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 10 Jun 2022 11:35:49 +0300 Subject: ASoC: SOF: ipc4: implement pm ctx_save callback Use the context save callback to power down the primary core which is used by the firmware as an indication that the DSP is going to be turned off. The IMR boot setup is done in response to the primary core power down. Signed-off-by: Peter Ujfalusi Reviewed-by: Ranjani Sridharan Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220610083549.16773-7-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/ipc4.c b/sound/soc/sof/ipc4.c index b2cb92745ec6..5dd22f6a0605 100644 --- a/sound/soc/sof/ipc4.c +++ b/sound/soc/sof/ipc4.c @@ -618,7 +618,22 @@ static int sof_ipc4_set_core_state(struct snd_sof_dev *sdev, int core_idx, bool return sof_ipc4_tx_msg(sdev, &msg, msg.data_size, NULL, 0, false); } +/* + * The context save callback is used to send a message to the firmware notifying + * it that the primary core is going to be turned off, which is used as an + * indication to prepare for a full power down, thus preparing for IMR boot + * (when supported) + * + * Note: in IPC4 there is no message used to restore context, thus no context + * restore callback is implemented + */ +static int sof_ipc4_ctx_save(struct snd_sof_dev *sdev) +{ + return sof_ipc4_set_core_state(sdev, SOF_DSP_PRIMARY_CORE, false); +} + static const struct sof_ipc_pm_ops ipc4_pm_ops = { + .ctx_save = sof_ipc4_ctx_save, .set_core_state = sof_ipc4_set_core_state, }; -- cgit v1.2.3 From 135786c32ed057068bec56f67a54064cfc845bde Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 10 Jun 2022 11:01:17 +0300 Subject: ASoC: SOF: ipc3-dtrace: Introduce SOF_DTRACE_INITIALIZING state With the new state we can make sure we are not missing the first host_offset update. In case the dtrace is small, the DMA copy will be fast and depending on the moonphase it might be done before we set the sdev->dtrace_state to SOF_DTRACE_ENABLED. The DMA will start the copy as soon as the host starts the DMA. Set the dtrace to enabled before we let the DMA to run in order to avoid missing the position update. The new state is needed to cover architectures where the host side snd_sof_dma_trace_trigger() is a NOP and the dtrace in the firmware is ready as soon as the IPC message has been processed. Signed-off-by: Peter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20220610080119.30880-2-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc3-dtrace.c | 20 ++++++++++++++++---- 1 file changed, 16 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/ipc3-dtrace.c b/sound/soc/sof/ipc3-dtrace.c index b4e1343f9138..9292ff7ce1e8 100644 --- a/sound/soc/sof/ipc3-dtrace.c +++ b/sound/soc/sof/ipc3-dtrace.c @@ -18,6 +18,7 @@ enum sof_dtrace_state { SOF_DTRACE_DISABLED, SOF_DTRACE_STOPPED, + SOF_DTRACE_INITIALIZING, SOF_DTRACE_ENABLED, }; @@ -32,6 +33,15 @@ struct sof_dtrace_priv { enum sof_dtrace_state dtrace_state; }; +static bool trace_pos_update_expected(struct sof_dtrace_priv *priv) +{ + if (priv->dtrace_state == SOF_DTRACE_ENABLED || + priv->dtrace_state == SOF_DTRACE_INITIALIZING) + return true; + + return false; +} + static int trace_filter_append_elem(struct snd_sof_dev *sdev, u32 key, u32 value, struct sof_ipc_trace_filter_elem *elem_list, int capacity, int *counter) @@ -274,7 +284,7 @@ static size_t sof_wait_dtrace_avail(struct snd_sof_dev *sdev, loff_t pos, if (ret) return ret; - if (priv->dtrace_state != SOF_DTRACE_ENABLED && priv->dtrace_draining) { + if (priv->dtrace_draining && !trace_pos_update_expected(priv)) { /* * tracing has ended and all traces have been * read by client, return EOF @@ -445,6 +455,7 @@ static int ipc3_dtrace_enable(struct snd_sof_dev *sdev) dev_dbg(sdev->dev, "%s: stream_tag: %d\n", __func__, params.stream_tag); /* send IPC to the DSP */ + priv->dtrace_state = SOF_DTRACE_INITIALIZING; ret = sof_ipc_tx_message(sdev->ipc, ¶ms, sizeof(params), &ipc_reply, sizeof(ipc_reply)); if (ret < 0) { dev_err(sdev->dev, "can't set params for DMA for trace %d\n", ret); @@ -452,17 +463,18 @@ static int ipc3_dtrace_enable(struct snd_sof_dev *sdev) } start: + priv->dtrace_state = SOF_DTRACE_ENABLED; + ret = sof_dtrace_host_trigger(sdev, SNDRV_PCM_TRIGGER_START); if (ret < 0) { dev_err(sdev->dev, "Host dtrace trigger start failed: %d\n", ret); goto trace_release; } - priv->dtrace_state = SOF_DTRACE_ENABLED; - return 0; trace_release: + priv->dtrace_state = SOF_DTRACE_DISABLED; sof_dtrace_host_release(sdev); return ret; } @@ -546,7 +558,7 @@ int ipc3_dtrace_posn_update(struct snd_sof_dev *sdev, if (!sdev->fw_trace_is_supported) return 0; - if (priv->dtrace_state == SOF_DTRACE_ENABLED && + if (trace_pos_update_expected(priv) && priv->host_offset != posn->host_offset) { priv->host_offset = posn->host_offset; wake_up(&priv->trace_sleep); -- cgit v1.2.3 From b66f9e703f0bee4e1aa7010299914b7b2009b4e0 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 10 Jun 2022 11:01:18 +0300 Subject: ASoC: SOF: ipc3-dtrace: Add helper function to update the sdev->host_offset We are using the READ_ONCE() on the debugfs read path for accessing sdev->host_offset, but the set is not atomic or protected in any way. Add a small helper to do the host_offset update and be really paranoid about the a possible race in update Signed-off-by: Peter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20220610080119.30880-3-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc3-dtrace.c | 23 ++++++++++++++++++----- 1 file changed, 18 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/ipc3-dtrace.c b/sound/soc/sof/ipc3-dtrace.c index 9292ff7ce1e8..1f4d7a98c8fc 100644 --- a/sound/soc/sof/ipc3-dtrace.c +++ b/sound/soc/sof/ipc3-dtrace.c @@ -252,6 +252,21 @@ static int debugfs_create_trace_filter(struct snd_sof_dev *sdev) return 0; } +static bool sof_dtrace_set_host_offset(struct sof_dtrace_priv *priv, u32 new_offset) +{ + u32 host_offset = READ_ONCE(priv->host_offset); + + if (host_offset != new_offset) { + /* This is a bit paranoid and unlikely that it is needed */ + u32 ret = cmpxchg(&priv->host_offset, host_offset, new_offset); + + if (ret == host_offset) + return true; + } + + return false; +} + static size_t sof_dtrace_avail(struct snd_sof_dev *sdev, loff_t pos, size_t buffer_size) { @@ -368,7 +383,7 @@ static int dfsentry_dtrace_release(struct inode *inode, struct file *file) /* avoid duplicate traces at next open */ if (priv->dtrace_state != SOF_DTRACE_ENABLED) - priv->host_offset = 0; + sof_dtrace_set_host_offset(priv, 0); return 0; } @@ -444,7 +459,7 @@ static int ipc3_dtrace_enable(struct snd_sof_dev *sdev) params.buffer.pages = priv->dma_trace_pages; params.stream_tag = 0; - priv->host_offset = 0; + sof_dtrace_set_host_offset(priv, 0); priv->dtrace_draining = false; ret = sof_dtrace_host_init(sdev, &priv->dmatb, ¶ms); @@ -559,10 +574,8 @@ int ipc3_dtrace_posn_update(struct snd_sof_dev *sdev, return 0; if (trace_pos_update_expected(priv) && - priv->host_offset != posn->host_offset) { - priv->host_offset = posn->host_offset; + sof_dtrace_set_host_offset(priv, posn->host_offset)) wake_up(&priv->trace_sleep); - } if (posn->overflow != 0) dev_err(sdev->dev, -- cgit v1.2.3 From 1e90de2c9a40d7d0af5c7b0a6e2d362ffba94772 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 10 Jun 2022 11:01:19 +0300 Subject: ASoC: SOF: ipc3-dtrace: Return from dtrace_read if there is no new data available If no new trace data is available then return immediately, there is no need to continue with the execution of the trace_read() function. Signed-off-by: Peter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20220610080119.30880-4-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc3-dtrace.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/ipc3-dtrace.c b/sound/soc/sof/ipc3-dtrace.c index 1f4d7a98c8fc..f59931d818c1 100644 --- a/sound/soc/sof/ipc3-dtrace.c +++ b/sound/soc/sof/ipc3-dtrace.c @@ -353,6 +353,10 @@ static ssize_t dfsentry_dtrace_read(struct file *file, char __user *buffer, return -EIO; } + /* no new trace data */ + if (!avail) + return 0; + /* make sure count is <= avail */ if (count > avail) count = avail; -- cgit v1.2.3 From 90e891551fb4949daeb3df20d43e7da838ef89a3 Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Wed, 8 Jun 2022 20:26:22 -0700 Subject: ASoC: SOF: IPC4: Introduce topology ops MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Introduce the topology ops for IPC4. Set the widget_ops and token_list for parsing the scheduler type widget. Support for other widget types will be added in the follow up patches. Co-developed-by: Rander Wang Signed-off-by: Rander Wang Co-developed-by: Bard Liao Signed-off-by: Bard Liao Signed-off-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Paul Olaru Link: https://lore.kernel.org/r/20220609032643.916882-3-ranjani.sridharan@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/Makefile | 2 +- sound/soc/sof/ipc4-priv.h | 1 + sound/soc/sof/ipc4-topology.c | 102 ++++++++++++++++++++++++++++++++++++++++++ sound/soc/sof/ipc4-topology.h | 30 +++++++++++++ sound/soc/sof/ipc4.c | 1 + 5 files changed, 135 insertions(+), 1 deletion(-) create mode 100644 sound/soc/sof/ipc4-topology.c create mode 100644 sound/soc/sof/ipc4-topology.h (limited to 'sound') diff --git a/sound/soc/sof/Makefile b/sound/soc/sof/Makefile index 92b5e83601be..73524fadb3ce 100644 --- a/sound/soc/sof/Makefile +++ b/sound/soc/sof/Makefile @@ -4,7 +4,7 @@ snd-sof-objs := core.o ops.o loader.o ipc.o pcm.o pm.o debug.o topology.o\ control.o trace.o iomem-utils.o sof-audio.o stream-ipc.o\ ipc3-topology.o ipc3-control.o ipc3.o ipc3-pcm.o ipc3-loader.o\ ipc3-dtrace.o\ - ipc4.o ipc4-loader.o + ipc4.o ipc4-loader.o ipc4-topology.o ifneq ($(CONFIG_SND_SOC_SOF_CLIENT),) snd-sof-objs += sof-client.o endif diff --git a/sound/soc/sof/ipc4-priv.h b/sound/soc/sof/ipc4-priv.h index 2b71d5675933..5388b888fefa 100644 --- a/sound/soc/sof/ipc4-priv.h +++ b/sound/soc/sof/ipc4-priv.h @@ -40,5 +40,6 @@ struct sof_ipc4_fw_module { }; extern const struct sof_ipc_fw_loader_ops ipc4_loader_ops; +extern const struct sof_ipc_tplg_ops ipc4_tplg_ops; #endif diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c new file mode 100644 index 000000000000..bccf576c8edd --- /dev/null +++ b/sound/soc/sof/ipc4-topology.c @@ -0,0 +1,102 @@ +// SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) +// +// This file is provided under a dual BSD/GPLv2 license. When using or +// redistributing this file, you may do so under either license. +// +// Copyright(c) 2022 Intel Corporation. All rights reserved. +// +// +#include +#include +#include +#include "sof-priv.h" +#include "sof-audio.h" +#include "ipc4-priv.h" +#include "ipc4-topology.h" +#include "ops.h" + +static const struct sof_topology_token ipc4_sched_tokens[] = { + {SOF_TKN_SCHED_LP_MODE, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc4_pipeline, lp_mode)} +}; + +static const struct sof_topology_token pipeline_tokens[] = { + {SOF_TKN_SCHED_DYNAMIC_PIPELINE, SND_SOC_TPLG_TUPLE_TYPE_BOOL, get_token_u16, + offsetof(struct snd_sof_widget, dynamic_pipeline_widget)}, +}; + +static const struct sof_token_info ipc4_token_list[SOF_TOKEN_COUNT] = { + [SOF_PIPELINE_TOKENS] = {"Pipeline tokens", pipeline_tokens, ARRAY_SIZE(pipeline_tokens)}, + [SOF_SCHED_TOKENS] = {"Scheduler tokens", ipc4_sched_tokens, + ARRAY_SIZE(ipc4_sched_tokens)}, +}; + +static void sof_ipc4_widget_free_comp(struct snd_sof_widget *swidget) +{ + kfree(swidget->private); +} + +static int sof_ipc4_widget_setup_comp_pipeline(struct snd_sof_widget *swidget) +{ + struct snd_soc_component *scomp = swidget->scomp; + struct sof_ipc4_pipeline *pipeline; + int ret; + + pipeline = kzalloc(sizeof(*pipeline), GFP_KERNEL); + if (!pipeline) + return -ENOMEM; + + ret = sof_update_ipc_object(scomp, pipeline, SOF_SCHED_TOKENS, swidget->tuples, + swidget->num_tuples, sizeof(*pipeline), 1); + if (ret) { + dev_err(scomp->dev, "parsing scheduler tokens failed\n"); + goto err; + } + + /* parse one set of pipeline tokens */ + ret = sof_update_ipc_object(scomp, swidget, SOF_PIPELINE_TOKENS, swidget->tuples, + swidget->num_tuples, sizeof(*swidget), 1); + if (ret) { + dev_err(scomp->dev, "parsing pipeline tokens failed\n"); + goto err; + } + + /* TODO: Get priority from topology */ + pipeline->priority = 0; + + dev_dbg(scomp->dev, "pipeline '%s': id %d pri %d lp mode %d\n", + swidget->widget->name, swidget->pipeline_id, + pipeline->priority, pipeline->lp_mode); + + swidget->private = pipeline; + + pipeline->msg.primary = SOF_IPC4_GLB_PIPE_PRIORITY(pipeline->priority); + pipeline->msg.primary |= SOF_IPC4_GLB_PIPE_INSTANCE_ID(swidget->pipeline_id); + pipeline->msg.primary |= SOF_IPC4_MSG_TYPE_SET(SOF_IPC4_GLB_CREATE_PIPELINE); + pipeline->msg.primary |= SOF_IPC4_MSG_DIR(SOF_IPC4_MSG_REQUEST); + pipeline->msg.primary |= SOF_IPC4_MSG_TARGET(SOF_IPC4_FW_GEN_MSG); + + pipeline->msg.extension = pipeline->lp_mode; + pipeline->state = SOF_IPC4_PIPE_UNINITIALIZED; + + return 0; +err: + kfree(pipeline); + return ret; +} + +static enum sof_tokens pipeline_token_list[] = { + SOF_SCHED_TOKENS, + SOF_PIPELINE_TOKENS, +}; + +static const struct sof_ipc_tplg_widget_ops tplg_ipc4_widget_ops[SND_SOC_DAPM_TYPE_COUNT] = { + [snd_soc_dapm_scheduler] = {sof_ipc4_widget_setup_comp_pipeline, sof_ipc4_widget_free_comp, + pipeline_token_list, ARRAY_SIZE(pipeline_token_list), NULL, + NULL, NULL}, +}; + +const struct sof_ipc_tplg_ops ipc4_tplg_ops = { + .widget = tplg_ipc4_widget_ops, + .token_list = ipc4_token_list, +}; diff --git a/sound/soc/sof/ipc4-topology.h b/sound/soc/sof/ipc4-topology.h new file mode 100644 index 000000000000..0e9be2b2d8a1 --- /dev/null +++ b/sound/soc/sof/ipc4-topology.h @@ -0,0 +1,30 @@ +/* SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) */ +/* + * This file is provided under a dual BSD/GPLv2 license. When using or + * redistributing this file, you may do so under either license. + * + * Copyright(c) 2022 Intel Corporation. All rights reserved. + */ + +#ifndef __INCLUDE_SOUND_SOF_IPC4_TOPOLOGY_H__ +#define __INCLUDE_SOUND_SOF_IPC4_TOPOLOGY_H__ + +#include + +/** + * struct sof_ipc4_pipeline - pipeline config data + * @priority: Priority of this pipeline + * @lp_mode: Low power mode + * @mem_usage: Memory usage + * @state: Pipeline state + * @msg: message structure for pipeline + */ +struct sof_ipc4_pipeline { + uint32_t priority; + uint32_t lp_mode; + uint32_t mem_usage; + int state; + struct sof_ipc4_msg msg; +}; + +#endif diff --git a/sound/soc/sof/ipc4.c b/sound/soc/sof/ipc4.c index 658802c86685..be677a33882d 100644 --- a/sound/soc/sof/ipc4.c +++ b/sound/soc/sof/ipc4.c @@ -603,4 +603,5 @@ const struct sof_ipc_ops ipc4_ops = { .set_get_data = sof_ipc4_set_get_data, .get_reply = sof_ipc4_get_reply, .fw_loader = &ipc4_loader_ops, + .tplg = &ipc4_tplg_ops, }; -- cgit v1.2.3 From 2cabd02b60901f4ceda4daf8c194905259797702 Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Wed, 8 Jun 2022 20:26:23 -0700 Subject: ASoC: SOF: ipc4-topology: Add support for parsing AIF_IN/AIF_OUT widgets MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add support for parsing AIF_IN/AIF_OUT type widgets in IPC4. Add all the new required token ID's for parsing these widgets to the list of tokens in enum sof_tokens and the definitions of the token arrays corresponding to each of the token ID's. Also, upgrade the sof_widget_parse_tokens() function in the common topology parser to be able to parse multiple sets of tokens for the audio format and copier gateway config tokens. Co-developed-by: Rander Wang Signed-off-by: Rander Wang Co-developed-by: Bard Liao Signed-off-by: Bard Liao Signed-off-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Paul Olaru Link: https://lore.kernel.org/r/20220609032643.916882-4-ranjani.sridharan@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-topology.c | 370 ++++++++++++++++++++++++++++++++++++++++++ sound/soc/sof/ipc4-topology.h | 83 ++++++++++ sound/soc/sof/sof-audio.h | 7 + sound/soc/sof/topology.c | 69 ++++++-- 4 files changed, 511 insertions(+), 18 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index bccf576c8edd..559148f5644c 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -25,17 +25,370 @@ static const struct sof_topology_token pipeline_tokens[] = { offsetof(struct snd_sof_widget, dynamic_pipeline_widget)}, }; +static const struct sof_topology_token ipc4_comp_tokens[] = { + {SOF_TKN_COMP_CPC, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc4_base_module_cfg, cpc)}, + {SOF_TKN_COMP_IS_PAGES, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc4_base_module_cfg, is_pages)}, +}; + +static const struct sof_topology_token ipc4_audio_format_buffer_size_tokens[] = { + {SOF_TKN_CAVS_AUDIO_FORMAT_IBS, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc4_base_module_cfg, ibs)}, + {SOF_TKN_CAVS_AUDIO_FORMAT_OBS, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc4_base_module_cfg, obs)}, +}; + +static const struct sof_topology_token ipc4_in_audio_format_tokens[] = { + {SOF_TKN_CAVS_AUDIO_FORMAT_IN_RATE, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc4_audio_format, sampling_frequency)}, + {SOF_TKN_CAVS_AUDIO_FORMAT_IN_BIT_DEPTH, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc4_audio_format, bit_depth)}, + {SOF_TKN_CAVS_AUDIO_FORMAT_IN_CH_MAP, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc4_audio_format, ch_map)}, + {SOF_TKN_CAVS_AUDIO_FORMAT_IN_CH_CFG, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc4_audio_format, ch_cfg)}, + {SOF_TKN_CAVS_AUDIO_FORMAT_IN_INTERLEAVING_STYLE, SND_SOC_TPLG_TUPLE_TYPE_WORD, + get_token_u32, offsetof(struct sof_ipc4_audio_format, interleaving_style)}, + {SOF_TKN_CAVS_AUDIO_FORMAT_IN_FMT_CFG, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc4_audio_format, fmt_cfg)}, +}; + +static const struct sof_topology_token ipc4_out_audio_format_tokens[] = { + {SOF_TKN_CAVS_AUDIO_FORMAT_OUT_RATE, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc4_audio_format, sampling_frequency)}, + {SOF_TKN_CAVS_AUDIO_FORMAT_OUT_BIT_DEPTH, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc4_audio_format, bit_depth)}, + {SOF_TKN_CAVS_AUDIO_FORMAT_OUT_CH_MAP, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc4_audio_format, ch_map)}, + {SOF_TKN_CAVS_AUDIO_FORMAT_OUT_CH_CFG, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc4_audio_format, ch_cfg)}, + {SOF_TKN_CAVS_AUDIO_FORMAT_OUT_INTERLEAVING_STYLE, SND_SOC_TPLG_TUPLE_TYPE_WORD, + get_token_u32, offsetof(struct sof_ipc4_audio_format, interleaving_style)}, + {SOF_TKN_CAVS_AUDIO_FORMAT_OUT_FMT_CFG, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc4_audio_format, fmt_cfg)}, +}; + +static const struct sof_topology_token ipc4_copier_gateway_cfg_tokens[] = { + {SOF_TKN_CAVS_AUDIO_FORMAT_DMA_BUFFER_SIZE, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, 0}, +}; + +static const struct sof_topology_token ipc4_copier_tokens[] = { + {SOF_TKN_INTEL_COPIER_NODE_TYPE, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, 0}, +}; + +static const struct sof_topology_token ipc4_audio_fmt_num_tokens[] = { + {SOF_TKN_COMP_NUM_AUDIO_FORMATS, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + 0}, +}; + +/* Component extended tokens */ +static const struct sof_topology_token comp_ext_tokens[] = { + {SOF_TKN_COMP_UUID, SND_SOC_TPLG_TUPLE_TYPE_UUID, get_token_uuid, + offsetof(struct snd_sof_widget, uuid)}, +}; + static const struct sof_token_info ipc4_token_list[SOF_TOKEN_COUNT] = { [SOF_PIPELINE_TOKENS] = {"Pipeline tokens", pipeline_tokens, ARRAY_SIZE(pipeline_tokens)}, [SOF_SCHED_TOKENS] = {"Scheduler tokens", ipc4_sched_tokens, ARRAY_SIZE(ipc4_sched_tokens)}, + [SOF_COMP_EXT_TOKENS] = {"Comp extended tokens", comp_ext_tokens, + ARRAY_SIZE(comp_ext_tokens)}, + [SOF_COMP_TOKENS] = {"IPC4 Component tokens", + ipc4_comp_tokens, ARRAY_SIZE(ipc4_comp_tokens)}, + [SOF_IN_AUDIO_FORMAT_TOKENS] = {"IPC4 Input Audio format tokens", + ipc4_in_audio_format_tokens, ARRAY_SIZE(ipc4_in_audio_format_tokens)}, + [SOF_OUT_AUDIO_FORMAT_TOKENS] = {"IPC4 Output Audio format tokens", + ipc4_out_audio_format_tokens, ARRAY_SIZE(ipc4_out_audio_format_tokens)}, + [SOF_AUDIO_FORMAT_BUFFER_SIZE_TOKENS] = {"IPC4 Audio format buffer size tokens", + ipc4_audio_format_buffer_size_tokens, + ARRAY_SIZE(ipc4_audio_format_buffer_size_tokens)}, + [SOF_COPIER_GATEWAY_CFG_TOKENS] = {"IPC4 Copier gateway config tokens", + ipc4_copier_gateway_cfg_tokens, ARRAY_SIZE(ipc4_copier_gateway_cfg_tokens)}, + [SOF_COPIER_TOKENS] = {"IPC4 Copier tokens", ipc4_copier_tokens, + ARRAY_SIZE(ipc4_copier_tokens)}, + [SOF_AUDIO_FMT_NUM_TOKENS] = {"IPC4 Audio format number tokens", + ipc4_audio_fmt_num_tokens, ARRAY_SIZE(ipc4_audio_fmt_num_tokens)}, }; +static void sof_ipc4_dbg_audio_format(struct device *dev, + struct sof_ipc4_audio_format *format, + size_t object_size, int num_format) +{ + struct sof_ipc4_audio_format *fmt; + void *ptr = format; + int i; + + for (i = 0; i < num_format; i++, ptr = (u8 *)ptr + object_size) { + fmt = ptr; + dev_dbg(dev, + " #%d: %uKHz, %ubit (ch_map %#x ch_cfg %u interleaving_style %u fmt_cfg %#x)\n", + i, fmt->sampling_frequency, fmt->bit_depth, fmt->ch_map, + fmt->ch_cfg, fmt->interleaving_style, fmt->fmt_cfg); + } +} + +/** + * sof_ipc4_get_audio_fmt - get available audio formats from swidget->tuples + * @scomp: pointer to pointer to SOC component + * @swidget: pointer to struct snd_sof_widget containing tuples + * @available_fmt: pointer to struct sof_ipc4_available_audio_format being filling in + * @has_out_format: true if available_fmt contains output format + * + * Return: 0 if successful + */ +static int sof_ipc4_get_audio_fmt(struct snd_soc_component *scomp, + struct snd_sof_widget *swidget, + struct sof_ipc4_available_audio_format *available_fmt, + bool has_out_format) +{ + struct sof_ipc4_base_module_cfg *base_config; + struct sof_ipc4_audio_format *out_format; + int audio_fmt_num = 0; + int ret, i; + + ret = sof_update_ipc_object(scomp, &audio_fmt_num, + SOF_AUDIO_FMT_NUM_TOKENS, swidget->tuples, + swidget->num_tuples, sizeof(audio_fmt_num), 1); + if (ret || audio_fmt_num <= 0) { + dev_err(scomp->dev, "Invalid number of audio formats: %d\n", audio_fmt_num); + return -EINVAL; + } + available_fmt->audio_fmt_num = audio_fmt_num; + + dev_dbg(scomp->dev, "Number of audio formats: %d\n", available_fmt->audio_fmt_num); + + base_config = kcalloc(available_fmt->audio_fmt_num, sizeof(*base_config), GFP_KERNEL); + if (!base_config) + return -ENOMEM; + + /* set cpc and is_pages for all base_cfg */ + for (i = 0; i < available_fmt->audio_fmt_num; i++) { + ret = sof_update_ipc_object(scomp, &base_config[i], + SOF_COMP_TOKENS, swidget->tuples, + swidget->num_tuples, sizeof(*base_config), 1); + if (ret) { + dev_err(scomp->dev, "parse comp tokens failed %d\n", ret); + goto err_in; + } + } + + /* copy the ibs/obs for each base_cfg */ + ret = sof_update_ipc_object(scomp, base_config, + SOF_AUDIO_FORMAT_BUFFER_SIZE_TOKENS, swidget->tuples, + swidget->num_tuples, sizeof(*base_config), + available_fmt->audio_fmt_num); + if (ret) { + dev_err(scomp->dev, "parse buffer size tokens failed %d\n", ret); + goto err_in; + } + + for (i = 0; i < available_fmt->audio_fmt_num; i++) + dev_dbg(scomp->dev, "%d: ibs: %d obs: %d cpc: %d is_pages: %d\n", i, + base_config[i].ibs, base_config[i].obs, + base_config[i].cpc, base_config[i].is_pages); + + ret = sof_update_ipc_object(scomp, &base_config->audio_fmt, + SOF_IN_AUDIO_FORMAT_TOKENS, swidget->tuples, + swidget->num_tuples, sizeof(*base_config), + available_fmt->audio_fmt_num); + if (ret) { + dev_err(scomp->dev, "parse base_config audio_fmt tokens failed %d\n", ret); + goto err_in; + } + + dev_dbg(scomp->dev, "Get input audio formats for %s\n", swidget->widget->name); + sof_ipc4_dbg_audio_format(scomp->dev, &base_config->audio_fmt, + sizeof(*base_config), + available_fmt->audio_fmt_num); + + available_fmt->base_config = base_config; + + if (!has_out_format) + return 0; + + out_format = kcalloc(available_fmt->audio_fmt_num, sizeof(*out_format), GFP_KERNEL); + if (!out_format) { + ret = -ENOMEM; + goto err_in; + } + + ret = sof_update_ipc_object(scomp, out_format, + SOF_OUT_AUDIO_FORMAT_TOKENS, swidget->tuples, + swidget->num_tuples, sizeof(*out_format), + available_fmt->audio_fmt_num); + + if (ret) { + dev_err(scomp->dev, "parse output audio_fmt tokens failed\n"); + goto err_out; + } + + available_fmt->out_audio_fmt = out_format; + dev_dbg(scomp->dev, "Get output audio formats for %s\n", swidget->widget->name); + sof_ipc4_dbg_audio_format(scomp->dev, out_format, sizeof(*out_format), + available_fmt->audio_fmt_num); + + return 0; + +err_out: + kfree(out_format); +err_in: + kfree(base_config); + + return ret; +} + static void sof_ipc4_widget_free_comp(struct snd_sof_widget *swidget) { kfree(swidget->private); } +static int sof_ipc4_widget_set_module_info(struct snd_sof_widget *swidget) +{ + struct snd_soc_component *scomp = swidget->scomp; + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); + struct sof_ipc4_fw_data *ipc4_data = sdev->private; + struct sof_ipc4_fw_module *fw_modules = ipc4_data->fw_modules; + int i; + + if (!fw_modules) { + dev_err(sdev->dev, "no fw_module information\n"); + return -EINVAL; + } + + /* set module info */ + for (i = 0; i < ipc4_data->num_fw_modules; i++) { + if (guid_equal(&swidget->uuid, &fw_modules[i].man4_module_entry.uuid)) { + swidget->module_info = &fw_modules[i]; + return 0; + } + } + + dev_err(sdev->dev, "failed to find module info for widget %s with UUID %pUL\n", + swidget->widget->name, &swidget->uuid); + return -EINVAL; +} + +static int sof_ipc4_widget_setup_msg(struct snd_sof_widget *swidget, struct sof_ipc4_msg *msg) +{ + struct sof_ipc4_fw_module *fw_module; + int ret; + + ret = sof_ipc4_widget_set_module_info(swidget); + if (ret) + return ret; + + fw_module = swidget->module_info; + + msg->primary = fw_module->man4_module_entry.id; + msg->primary |= SOF_IPC4_MSG_TYPE_SET(SOF_IPC4_MOD_INIT_INSTANCE); + msg->primary |= SOF_IPC4_MSG_DIR(SOF_IPC4_MSG_REQUEST); + msg->primary |= SOF_IPC4_MSG_TARGET(SOF_IPC4_MODULE_MSG); + + msg->extension = SOF_IPC4_MOD_EXT_PPL_ID(swidget->pipeline_id); + msg->extension |= SOF_IPC4_MOD_EXT_CORE_ID(swidget->core); + + return 0; +} + +static int sof_ipc4_widget_setup_pcm(struct snd_sof_widget *swidget) +{ + struct sof_ipc4_available_audio_format *available_fmt; + struct snd_soc_component *scomp = swidget->scomp; + struct sof_ipc4_copier *ipc4_copier; + int node_type = 0; + int ret, i; + + ipc4_copier = kzalloc(sizeof(*ipc4_copier), GFP_KERNEL); + if (!ipc4_copier) + return -ENOMEM; + + swidget->private = ipc4_copier; + available_fmt = &ipc4_copier->available_fmt; + + dev_dbg(scomp->dev, "Updating IPC structure for %s\n", swidget->widget->name); + + ret = sof_ipc4_get_audio_fmt(scomp, swidget, available_fmt, true); + if (ret) + goto free_copier; + + available_fmt->dma_buffer_size = kcalloc(available_fmt->audio_fmt_num, sizeof(u32), + GFP_KERNEL); + if (!available_fmt->dma_buffer_size) { + ret = -ENOMEM; + goto free_copier; + } + + ret = sof_update_ipc_object(scomp, available_fmt->dma_buffer_size, + SOF_COPIER_GATEWAY_CFG_TOKENS, swidget->tuples, + swidget->num_tuples, sizeof(u32), + available_fmt->audio_fmt_num); + if (ret) { + dev_err(scomp->dev, "Failed to parse dma buffer size in audio format for %s\n", + swidget->widget->name); + goto err; + } + + dev_dbg(scomp->dev, "dma buffer size:\n"); + for (i = 0; i < available_fmt->audio_fmt_num; i++) + dev_dbg(scomp->dev, "%d: %u\n", i, + available_fmt->dma_buffer_size[i]); + + ret = sof_update_ipc_object(scomp, &node_type, + SOF_COPIER_TOKENS, swidget->tuples, + swidget->num_tuples, sizeof(node_type), 1); + + if (ret) { + dev_err(scomp->dev, "parse host copier node type token failed %d\n", + ret); + goto err; + } + dev_dbg(scomp->dev, "host copier '%s' node_type %u\n", swidget->widget->name, node_type); + + ipc4_copier->data.gtw_cfg.node_id = SOF_IPC4_NODE_TYPE(node_type); + ipc4_copier->gtw_attr = kzalloc(sizeof(*ipc4_copier->gtw_attr), GFP_KERNEL); + if (!ipc4_copier->gtw_attr) { + ret = -ENOMEM; + goto err; + } + + ipc4_copier->copier_config = (uint32_t *)ipc4_copier->gtw_attr; + ipc4_copier->data.gtw_cfg.config_length = + sizeof(struct sof_ipc4_gtw_attributes) >> 2; + + /* set up module info and message header */ + ret = sof_ipc4_widget_setup_msg(swidget, &ipc4_copier->msg); + if (ret) + goto free_gtw_attr; + + return 0; + +free_gtw_attr: + kfree(ipc4_copier->gtw_attr); +err: + kfree(available_fmt->dma_buffer_size); +free_copier: + kfree(ipc4_copier); + return ret; +} + +static void sof_ipc4_widget_free_comp_pcm(struct snd_sof_widget *swidget) +{ + struct sof_ipc4_copier *ipc4_copier = swidget->private; + struct sof_ipc4_available_audio_format *available_fmt; + + if (!ipc4_copier) + return; + + available_fmt = &ipc4_copier->available_fmt; + kfree(available_fmt->dma_buffer_size); + kfree(available_fmt->base_config); + kfree(available_fmt->out_audio_fmt); + kfree(ipc4_copier->gtw_attr); + kfree(ipc4_copier); + swidget->private = NULL; +} + static int sof_ipc4_widget_setup_comp_pipeline(struct snd_sof_widget *swidget) { struct snd_soc_component *scomp = swidget->scomp; @@ -85,12 +438,29 @@ err: return ret; } +static enum sof_tokens host_token_list[] = { + SOF_COMP_TOKENS, + SOF_AUDIO_FMT_NUM_TOKENS, + SOF_AUDIO_FORMAT_BUFFER_SIZE_TOKENS, + SOF_IN_AUDIO_FORMAT_TOKENS, + SOF_OUT_AUDIO_FORMAT_TOKENS, + SOF_COPIER_GATEWAY_CFG_TOKENS, + SOF_COPIER_TOKENS, + SOF_COMP_EXT_TOKENS, +}; + static enum sof_tokens pipeline_token_list[] = { SOF_SCHED_TOKENS, SOF_PIPELINE_TOKENS, }; static const struct sof_ipc_tplg_widget_ops tplg_ipc4_widget_ops[SND_SOC_DAPM_TYPE_COUNT] = { + [snd_soc_dapm_aif_in] = {sof_ipc4_widget_setup_pcm, sof_ipc4_widget_free_comp_pcm, + host_token_list, ARRAY_SIZE(host_token_list), NULL, + NULL, NULL}, + [snd_soc_dapm_aif_out] = {sof_ipc4_widget_setup_pcm, sof_ipc4_widget_free_comp_pcm, + host_token_list, ARRAY_SIZE(host_token_list), NULL, + NULL, NULL}, [snd_soc_dapm_scheduler] = {sof_ipc4_widget_setup_comp_pipeline, sof_ipc4_widget_free_comp, pipeline_token_list, ARRAY_SIZE(pipeline_token_list), NULL, NULL, NULL}, diff --git a/sound/soc/sof/ipc4-topology.h b/sound/soc/sof/ipc4-topology.h index 0e9be2b2d8a1..f4f62dda63a3 100644 --- a/sound/soc/sof/ipc4-topology.h +++ b/sound/soc/sof/ipc4-topology.h @@ -11,6 +11,8 @@ #include +#define SOF_IPC4_NODE_TYPE(x) ((x) << 8) + /** * struct sof_ipc4_pipeline - pipeline config data * @priority: Priority of this pipeline @@ -27,4 +29,85 @@ struct sof_ipc4_pipeline { struct sof_ipc4_msg msg; }; +/** + * struct sof_ipc4_available_audio_format - Available audio formats + * @base_config: Available base config + * @out_audio_fmt: Available output audio format + * @ref_audio_fmt: Reference audio format to match runtime audio format + * @dma_buffer_size: Available Gateway DMA buffer size (in bytes) + * @audio_fmt_num: Number of available audio formats + */ +struct sof_ipc4_available_audio_format { + struct sof_ipc4_base_module_cfg *base_config; + struct sof_ipc4_audio_format *out_audio_fmt; + struct sof_ipc4_audio_format *ref_audio_fmt; + u32 *dma_buffer_size; + int audio_fmt_num; +}; + +/** + * struct sof_copier_gateway_cfg - IPC gateway configuration + * @node_id: ID of Gateway Node + * @dma_buffer_size: Preferred Gateway DMA buffer size (in bytes) + * @config_length: Length of gateway node configuration blob specified in #config_data + * config_data: Gateway node configuration blob + */ +struct sof_copier_gateway_cfg { + uint32_t node_id; + uint32_t dma_buffer_size; + uint32_t config_length; + uint32_t config_data[]; +}; + +/** + * struct sof_ipc4_copier_data - IPC data for copier + * @base_config: Base configuration including input audio format + * @out_format: Output audio format + * @copier_feature_mask: Copier feature mask + * @gtw_cfg: Gateway configuration + */ +struct sof_ipc4_copier_data { + struct sof_ipc4_base_module_cfg base_config; + struct sof_ipc4_audio_format out_format; + uint32_t copier_feature_mask; + struct sof_copier_gateway_cfg gtw_cfg; +}; + +/** + * struct sof_ipc4_gtw_attributes: Gateway attributes + * @lp_buffer_alloc: Gateway data requested in low power memory + * @alloc_from_reg_file: Gateway data requested in register file memory + * @rsvd: reserved for future use + */ +struct sof_ipc4_gtw_attributes { + uint32_t lp_buffer_alloc : 1; + uint32_t alloc_from_reg_file : 1; + uint32_t rsvd : 30; +}; + +/** + * struct sof_ipc4_copier - copier config data + * @data: IPC copier data + * @copier_config: Copier + blob + * @ipc_config_size: Size of copier_config + * @available_fmt: Available audio format + * @frame_fmt: frame format + * @msg: message structure for copier + * @gtw_attr: Gateway attributes for copier blob + * @dai_type: DAI type + * @dai_index: DAI index + */ +struct sof_ipc4_copier { + struct sof_ipc4_copier_data data; + u32 *copier_config; + uint32_t ipc_config_size; + void *ipc_config_data; + struct sof_ipc4_available_audio_format available_fmt; + u32 frame_fmt; + struct sof_ipc4_msg msg; + struct sof_ipc4_gtw_attributes *gtw_attr; + u32 dai_type; + int dai_index; +}; + #endif diff --git a/sound/soc/sof/sof-audio.h b/sound/soc/sof/sof-audio.h index 27cc5fb642e5..c38b4bdd685a 100644 --- a/sound/soc/sof/sof-audio.h +++ b/sound/soc/sof/sof-audio.h @@ -225,6 +225,13 @@ enum sof_tokens { SOF_AFE_TOKENS, SOF_CORE_TOKENS, SOF_COMP_EXT_TOKENS, + SOF_IN_AUDIO_FORMAT_TOKENS, + SOF_OUT_AUDIO_FORMAT_TOKENS, + SOF_AUDIO_FORMAT_BUFFER_SIZE_TOKENS, + SOF_COPIER_GATEWAY_CFG_TOKENS, + SOF_COPIER_TOKENS, + SOF_AUDIO_FMT_NUM_TOKENS, + SOF_COPIER_FORMAT_TOKENS, /* this should be the last */ SOF_TOKEN_COUNT, diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index b1fcab7ce48e..606dbca94246 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -1141,6 +1141,21 @@ static int spcm_bind(struct snd_soc_component *scomp, struct snd_sof_pcm *spcm, return 0; } +static int sof_get_token_value(u32 token_id, struct snd_sof_tuple *tuples, int num_tuples) +{ + int i; + + if (!tuples) + return -EINVAL; + + for (i = 0; i < num_tuples; i++) { + if (tuples[i].token == token_id) + return tuples[i].value.v; + } + + return -EINVAL; +} + static int sof_widget_parse_tokens(struct snd_soc_component *scomp, struct snd_sof_widget *swidget, struct snd_soc_tplg_dapm_widget *tw, enum sof_tokens *object_token_list, int count) @@ -1168,6 +1183,8 @@ static int sof_widget_parse_tokens(struct snd_soc_component *scomp, struct snd_s /* parse token list for widget */ for (i = 0; i < count; i++) { + int num_sets = 1; + if (object_token_list[i] >= SOF_TOKEN_COUNT) { dev_err(scomp->dev, "Invalid token id %d for widget %s\n", object_token_list[i], swidget->widget->name); @@ -1175,8 +1192,9 @@ static int sof_widget_parse_tokens(struct snd_soc_component *scomp, struct snd_s goto err; } - /* parse and save UUID in swidget */ - if (object_token_list[i] == SOF_COMP_EXT_TOKENS) { + switch (object_token_list[i]) { + case SOF_COMP_EXT_TOKENS: + /* parse and save UUID in swidget */ ret = sof_parse_tokens(scomp, swidget, token_list[object_token_list[i]].tokens, token_list[object_token_list[i]].count, @@ -1189,11 +1207,41 @@ static int sof_widget_parse_tokens(struct snd_soc_component *scomp, struct snd_s } continue; + case SOF_IN_AUDIO_FORMAT_TOKENS: + case SOF_OUT_AUDIO_FORMAT_TOKENS: + case SOF_COPIER_GATEWAY_CFG_TOKENS: + case SOF_AUDIO_FORMAT_BUFFER_SIZE_TOKENS: + num_sets = sof_get_token_value(SOF_TKN_COMP_NUM_AUDIO_FORMATS, + swidget->tuples, swidget->num_tuples); + + if (num_sets < 0) { + dev_err(sdev->dev, "Invalid audio format count for %s\n", + swidget->widget->name); + ret = num_sets; + goto err; + } + + if (num_sets > 1) { + struct snd_sof_tuple *new_tuples; + + num_tuples += token_list[object_token_list[i]].count * num_sets; + new_tuples = krealloc(swidget->tuples, + sizeof(*new_tuples) * num_tuples, GFP_KERNEL); + if (!new_tuples) { + ret = -ENOMEM; + goto err; + } + + swidget->tuples = new_tuples; + } + break; + default: + break; } /* copy one set of tuples per token ID into swidget->tuples */ ret = sof_copy_tuples(sdev, private->array, le32_to_cpu(private->size), - object_token_list[i], 1, swidget->tuples, + object_token_list[i], num_sets, swidget->tuples, num_tuples, &swidget->num_tuples); if (ret < 0) { dev_err(scomp->dev, "Failed parsing %s for widget %s err: %d\n", @@ -1208,21 +1256,6 @@ err: return ret; } -static int sof_get_token_value(u32 token_id, struct snd_sof_tuple *tuples, int num_tuples) -{ - int i; - - if (!tuples) - return -EINVAL; - - for (i = 0; i < num_tuples; i++) { - if (tuples[i].token == token_id) - return tuples[i].value.v; - } - - return -EINVAL; -} - /* external widget init - used for any driver specific init */ static int sof_widget_ready(struct snd_soc_component *scomp, int index, struct snd_soc_dapm_widget *w, -- cgit v1.2.3 From abfb536bd116d3148e92bf38255fc0989ca9b7d4 Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Wed, 8 Jun 2022 20:26:24 -0700 Subject: ASoC: SOF: ipc4-topology: Add support for parsing DAI_IN/DAI_OUT widgets MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add support for parsing and setting up the IPC structure for DAI_IN/DAI_OUT type widgets in IPC4. Co-developed-by: Rander Wang Signed-off-by: Rander Wang Co-developed-by: Bard Liao Signed-off-by: Bard Liao Signed-off-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/20220609032643.916882-5-ranjani.sridharan@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-topology.c | 135 ++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 135 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index 559148f5644c..5bb80306794b 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -82,6 +82,13 @@ static const struct sof_topology_token ipc4_audio_fmt_num_tokens[] = { 0}, }; +static const struct sof_topology_token dai_tokens[] = { + {SOF_TKN_DAI_TYPE, SND_SOC_TPLG_TUPLE_TYPE_STRING, get_token_dai_type, + offsetof(struct sof_ipc4_copier, dai_type)}, + {SOF_TKN_DAI_INDEX, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc4_copier, dai_index)}, +}; + /* Component extended tokens */ static const struct sof_topology_token comp_ext_tokens[] = { {SOF_TKN_COMP_UUID, SND_SOC_TPLG_TUPLE_TYPE_UUID, get_token_uuid, @@ -89,6 +96,7 @@ static const struct sof_topology_token comp_ext_tokens[] = { }; static const struct sof_token_info ipc4_token_list[SOF_TOKEN_COUNT] = { + [SOF_DAI_TOKENS] = {"DAI tokens", dai_tokens, ARRAY_SIZE(dai_tokens)}, [SOF_PIPELINE_TOKENS] = {"Pipeline tokens", pipeline_tokens, ARRAY_SIZE(pipeline_tokens)}, [SOF_SCHED_TOKENS] = {"Scheduler tokens", ipc4_sched_tokens, ARRAY_SIZE(ipc4_sched_tokens)}, @@ -389,6 +397,117 @@ static void sof_ipc4_widget_free_comp_pcm(struct snd_sof_widget *swidget) swidget->private = NULL; } +static int sof_ipc4_widget_setup_comp_dai(struct snd_sof_widget *swidget) +{ + struct sof_ipc4_available_audio_format *available_fmt; + struct snd_soc_component *scomp = swidget->scomp; + struct snd_sof_dai *dai = swidget->private; + struct sof_ipc4_copier *ipc4_copier; + int node_type = 0; + int ret, i; + + ipc4_copier = kzalloc(sizeof(*ipc4_copier), GFP_KERNEL); + if (!ipc4_copier) + return -ENOMEM; + + available_fmt = &ipc4_copier->available_fmt; + + dev_dbg(scomp->dev, "Updating IPC structure for %s\n", swidget->widget->name); + + ret = sof_ipc4_get_audio_fmt(scomp, swidget, available_fmt, true); + if (ret) + goto free_copier; + + available_fmt->dma_buffer_size = kcalloc(available_fmt->audio_fmt_num, sizeof(u32), + GFP_KERNEL); + if (!available_fmt->dma_buffer_size) { + ret = -ENOMEM; + goto free_copier; + } + + ret = sof_update_ipc_object(scomp, available_fmt->dma_buffer_size, + SOF_COPIER_GATEWAY_CFG_TOKENS, swidget->tuples, + swidget->num_tuples, sizeof(u32), + available_fmt->audio_fmt_num); + if (ret) { + dev_err(scomp->dev, "Failed to parse dma buffer size in audio format for %s\n", + swidget->widget->name); + goto err; + } + + for (i = 0; i < available_fmt->audio_fmt_num; i++) + dev_dbg(scomp->dev, "%d: dma buffer size: %u\n", i, + available_fmt->dma_buffer_size[i]); + + ret = sof_update_ipc_object(scomp, &node_type, + SOF_COPIER_TOKENS, swidget->tuples, + swidget->num_tuples, sizeof(node_type), 1); + if (ret) { + dev_err(scomp->dev, "parse dai node type failed %d\n", ret); + goto err; + } + + ret = sof_update_ipc_object(scomp, ipc4_copier, + SOF_DAI_TOKENS, swidget->tuples, + swidget->num_tuples, sizeof(u32), 1); + if (ret) { + dev_err(scomp->dev, "parse dai copier node token failed %d\n", ret); + goto err; + } + + dev_dbg(scomp->dev, "dai %s node_type %u dai_type %u dai_index %d\n", swidget->widget->name, + node_type, ipc4_copier->dai_type, ipc4_copier->dai_index); + + ipc4_copier->data.gtw_cfg.node_id = SOF_IPC4_NODE_TYPE(node_type); + ipc4_copier->gtw_attr = kzalloc(sizeof(*ipc4_copier->gtw_attr), GFP_KERNEL); + if (!ipc4_copier->gtw_attr) { + ret = -ENOMEM; + goto err; + } + + ipc4_copier->copier_config = (uint32_t *)ipc4_copier->gtw_attr; + ipc4_copier->data.gtw_cfg.config_length = sizeof(struct sof_ipc4_gtw_attributes) >> 2; + + dai->scomp = scomp; + dai->private = ipc4_copier; + + /* set up module info and message header */ + ret = sof_ipc4_widget_setup_msg(swidget, &ipc4_copier->msg); + if (ret) + goto free_copier_config; + + return 0; + +free_copier_config: + kfree(ipc4_copier->copier_config); +err: + kfree(available_fmt->dma_buffer_size); +free_copier: + kfree(ipc4_copier); + return ret; +} + +static void sof_ipc4_widget_free_comp_dai(struct snd_sof_widget *swidget) +{ + struct sof_ipc4_available_audio_format *available_fmt; + struct snd_sof_dai *dai = swidget->private; + struct sof_ipc4_copier *ipc4_copier; + + if (!dai) + return; + + ipc4_copier = dai->private; + available_fmt = &ipc4_copier->available_fmt; + + kfree(available_fmt->dma_buffer_size); + kfree(available_fmt->base_config); + kfree(available_fmt->out_audio_fmt); + kfree(ipc4_copier->copier_config); + kfree(dai->private); + kfree(dai); + swidget->private = NULL; +} + static int sof_ipc4_widget_setup_comp_pipeline(struct snd_sof_widget *swidget) { struct snd_soc_component *scomp = swidget->scomp; @@ -454,6 +573,18 @@ static enum sof_tokens pipeline_token_list[] = { SOF_PIPELINE_TOKENS, }; +static enum sof_tokens dai_token_list[] = { + SOF_COMP_TOKENS, + SOF_AUDIO_FMT_NUM_TOKENS, + SOF_AUDIO_FORMAT_BUFFER_SIZE_TOKENS, + SOF_IN_AUDIO_FORMAT_TOKENS, + SOF_OUT_AUDIO_FORMAT_TOKENS, + SOF_COPIER_GATEWAY_CFG_TOKENS, + SOF_COPIER_TOKENS, + SOF_DAI_TOKENS, + SOF_COMP_EXT_TOKENS, +}; + static const struct sof_ipc_tplg_widget_ops tplg_ipc4_widget_ops[SND_SOC_DAPM_TYPE_COUNT] = { [snd_soc_dapm_aif_in] = {sof_ipc4_widget_setup_pcm, sof_ipc4_widget_free_comp_pcm, host_token_list, ARRAY_SIZE(host_token_list), NULL, @@ -461,6 +592,10 @@ static const struct sof_ipc_tplg_widget_ops tplg_ipc4_widget_ops[SND_SOC_DAPM_TY [snd_soc_dapm_aif_out] = {sof_ipc4_widget_setup_pcm, sof_ipc4_widget_free_comp_pcm, host_token_list, ARRAY_SIZE(host_token_list), NULL, NULL, NULL}, + [snd_soc_dapm_dai_in] = {sof_ipc4_widget_setup_comp_dai, sof_ipc4_widget_free_comp_dai, + dai_token_list, ARRAY_SIZE(dai_token_list), NULL, NULL, NULL}, + [snd_soc_dapm_dai_out] = {sof_ipc4_widget_setup_comp_dai, sof_ipc4_widget_free_comp_dai, + dai_token_list, ARRAY_SIZE(dai_token_list), NULL, NULL, NULL}, [snd_soc_dapm_scheduler] = {sof_ipc4_widget_setup_comp_pipeline, sof_ipc4_widget_free_comp, pipeline_token_list, ARRAY_SIZE(pipeline_token_list), NULL, NULL, NULL}, -- cgit v1.2.3 From 904c48c40c66c524df90fb660bdbc514ed802e67 Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Wed, 8 Jun 2022 20:26:25 -0700 Subject: ASoC: SOF: ipc4-topology: Add prepare op for AIF type widgets MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Define the prepare op for the AIF type widgets for IPC4. The prepare op is responsible for choosing the input/output audio formats for these widgets based on the runtime PCM params, assigning the instance ID and updating the total memory usage for the pipelines these widgets belong to. Co-developed-by: Rander Wang Signed-off-by: Rander Wang Co-developed-by: Bard Liao Signed-off-by: Bard Liao Signed-off-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220609032643.916882-6-ranjani.sridharan@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-topology.c | 290 +++++++++++++++++++++++++++++++++++++++++- sound/soc/sof/ipc4-topology.h | 18 +++ 2 files changed, 306 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index 5bb80306794b..1a73c16f1624 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -557,6 +557,290 @@ err: return ret; } +static void +sof_ipc4_update_pipeline_mem_usage(struct snd_sof_dev *sdev, struct snd_sof_widget *swidget, + struct sof_ipc4_base_module_cfg *base_config) +{ + struct sof_ipc4_fw_module *fw_module = swidget->module_info; + struct snd_sof_widget *pipe_widget; + struct sof_ipc4_pipeline *pipeline; + int task_mem, queue_mem; + int ibs, bss, total; + + ibs = base_config->ibs; + bss = base_config->is_pages; + + task_mem = SOF_IPC4_PIPELINE_OBJECT_SIZE; + task_mem += SOF_IPC4_MODULE_INSTANCE_LIST_ITEM_SIZE + bss; + + if (fw_module->man4_module_entry.type & SOF_IPC4_MODULE_LL) { + task_mem += SOF_IPC4_FW_ROUNDUP(SOF_IPC4_LL_TASK_OBJECT_SIZE); + task_mem += SOF_IPC4_FW_MAX_QUEUE_COUNT * SOF_IPC4_MODULE_INSTANCE_LIST_ITEM_SIZE; + task_mem += SOF_IPC4_LL_TASK_LIST_ITEM_SIZE; + } else { + task_mem += SOF_IPC4_FW_ROUNDUP(SOF_IPC4_DP_TASK_OBJECT_SIZE); + task_mem += SOF_IPC4_DP_TASK_LIST_SIZE; + } + + ibs = SOF_IPC4_FW_ROUNDUP(ibs); + queue_mem = SOF_IPC4_FW_MAX_QUEUE_COUNT * (SOF_IPC4_DATA_QUEUE_OBJECT_SIZE + ibs); + + total = SOF_IPC4_FW_PAGE(task_mem + queue_mem); + + pipe_widget = swidget->pipe_widget; + pipeline = pipe_widget->private; + pipeline->mem_usage += total; +} + +static int sof_ipc4_widget_assign_instance_id(struct snd_sof_dev *sdev, + struct snd_sof_widget *swidget) +{ + struct sof_ipc4_fw_module *fw_module = swidget->module_info; + int max_instances = fw_module->man4_module_entry.instance_max_count; + + swidget->instance_id = ida_alloc_max(&fw_module->m_ida, max_instances, GFP_KERNEL); + if (swidget->instance_id < 0) { + dev_err(sdev->dev, "failed to assign instance id for widget %s", + swidget->widget->name); + return swidget->instance_id; + } + + return 0; +} + +static int sof_ipc4_init_audio_fmt(struct snd_sof_dev *sdev, + struct snd_sof_widget *swidget, + struct sof_ipc4_base_module_cfg *base_config, + struct sof_ipc4_audio_format *out_format, + struct snd_pcm_hw_params *params, + struct sof_ipc4_available_audio_format *available_fmt, + size_t object_offset) +{ + void *ptr = available_fmt->ref_audio_fmt; + u32 valid_bits; + u32 channels; + u32 rate; + int sample_valid_bits; + int i; + + if (!ptr) { + dev_err(sdev->dev, "no reference formats for %s\n", swidget->widget->name); + return -EINVAL; + } + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + sample_valid_bits = 16; + break; + case SNDRV_PCM_FORMAT_S24_LE: + sample_valid_bits = 24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + sample_valid_bits = 32; + break; + default: + dev_err(sdev->dev, "invalid pcm frame format %d\n", params_format(params)); + return -EINVAL; + } + + if (!available_fmt->audio_fmt_num) { + dev_err(sdev->dev, "no formats available for %s\n", swidget->widget->name); + return -EINVAL; + } + + /* + * Search supported audio formats to match rate, channels ,and + * sample_valid_bytes from runtime params + */ + for (i = 0; i < available_fmt->audio_fmt_num; i++, ptr = (u8 *)ptr + object_offset) { + struct sof_ipc4_audio_format *fmt = ptr; + + rate = fmt->sampling_frequency; + channels = SOF_IPC4_AUDIO_FORMAT_CFG_CHANNELS_COUNT(fmt->fmt_cfg); + valid_bits = SOF_IPC4_AUDIO_FORMAT_CFG_V_BIT_DEPTH(fmt->fmt_cfg); + if (params_rate(params) == rate && params_channels(params) == channels && + sample_valid_bits == valid_bits) { + dev_dbg(sdev->dev, "%s: matching audio format index for %uHz, %ubit, %u channels: %d\n", + __func__, rate, valid_bits, channels, i); + + /* copy ibs/obs and input format */ + memcpy(base_config, &available_fmt->base_config[i], + sizeof(struct sof_ipc4_base_module_cfg)); + + /* copy output format */ + if (out_format) + memcpy(out_format, &available_fmt->out_audio_fmt[i], + sizeof(struct sof_ipc4_audio_format)); + break; + } + } + + if (i == available_fmt->audio_fmt_num) { + dev_err(sdev->dev, "%s: Unsupported audio format: %uHz, %ubit, %u channels\n", + __func__, params_rate(params), sample_valid_bits, params_channels(params)); + return -EINVAL; + } + + dev_dbg(sdev->dev, "Init input audio formats for %s\n", swidget->widget->name); + sof_ipc4_dbg_audio_format(sdev->dev, &base_config->audio_fmt, + sizeof(*base_config), 1); + if (out_format) { + dev_dbg(sdev->dev, "Init output audio formats for %s\n", swidget->widget->name); + sof_ipc4_dbg_audio_format(sdev->dev, out_format, + sizeof(*out_format), 1); + } + + /* Return the index of the matched format */ + return i; +} + +static void sof_ipc4_unprepare_copier_module(struct snd_sof_widget *swidget) +{ + struct sof_ipc4_fw_module *fw_module = swidget->module_info; + struct sof_ipc4_copier *ipc4_copier = NULL; + struct snd_sof_widget *pipe_widget; + struct sof_ipc4_pipeline *pipeline; + + /* reset pipeline memory usage */ + pipe_widget = swidget->pipe_widget; + pipeline = pipe_widget->private; + pipeline->mem_usage = 0; + + if (WIDGET_IS_AIF(swidget->id)) + ipc4_copier = swidget->private; + + if (ipc4_copier) { + kfree(ipc4_copier->ipc_config_data); + ipc4_copier->ipc_config_data = NULL; + ipc4_copier->ipc_config_size = 0; + } + + ida_free(&fw_module->m_ida, swidget->instance_id); +} + +static int +sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, + struct snd_pcm_hw_params *fe_params, + struct snd_sof_platform_stream_params *platform_params, + struct snd_pcm_hw_params *pipeline_params, int dir) +{ + struct sof_ipc4_available_audio_format *available_fmt; + struct snd_soc_component *scomp = swidget->scomp; + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); + struct sof_ipc4_copier_data *copier_data; + struct snd_pcm_hw_params *ref_params; + struct sof_ipc4_copier *ipc4_copier; + struct snd_mask *fmt; + int out_sample_valid_bits; + size_t ref_audio_fmt_size; + void **ipc_config_data; + int *ipc_config_size; + u32 **data; + int ipc_size, ret; + + dev_dbg(sdev->dev, "%s: copier %s, type %d", __func__, swidget->widget->name, swidget->id); + + switch (swidget->id) { + case snd_soc_dapm_aif_in: + case snd_soc_dapm_aif_out: + { + struct sof_ipc4_gtw_attributes *gtw_attr; + struct snd_sof_widget *pipe_widget; + struct sof_ipc4_pipeline *pipeline; + + pipe_widget = swidget->pipe_widget; + pipeline = pipe_widget->private; + ipc4_copier = (struct sof_ipc4_copier *)swidget->private; + gtw_attr = ipc4_copier->gtw_attr; + copier_data = &ipc4_copier->data; + available_fmt = &ipc4_copier->available_fmt; + + /* + * base_config->audio_fmt and out_audio_fmt represent the input and output audio + * formats. Use the input format as the reference to match pcm params for playback + * and the output format as reference for capture. + */ + if (dir == SNDRV_PCM_STREAM_PLAYBACK) { + available_fmt->ref_audio_fmt = &available_fmt->base_config->audio_fmt; + ref_audio_fmt_size = sizeof(struct sof_ipc4_base_module_cfg); + } else { + available_fmt->ref_audio_fmt = available_fmt->out_audio_fmt; + ref_audio_fmt_size = sizeof(struct sof_ipc4_audio_format); + } + copier_data->gtw_cfg.node_id &= ~SOF_IPC4_NODE_INDEX_MASK; + copier_data->gtw_cfg.node_id |= + SOF_IPC4_NODE_INDEX(platform_params->stream_tag - 1); + + /* set gateway attributes */ + gtw_attr->lp_buffer_alloc = pipeline->lp_mode; + ref_params = fe_params; + break; + } + default: + dev_err(sdev->dev, "unsupported type %d for copier %s", + swidget->id, swidget->widget->name); + return -EINVAL; + } + + /* set input and output audio formats */ + ret = sof_ipc4_init_audio_fmt(sdev, swidget, &copier_data->base_config, + &copier_data->out_format, ref_params, + available_fmt, ref_audio_fmt_size); + if (ret < 0) + return ret; + + /* modify the input params for the next widget */ + fmt = hw_param_mask(pipeline_params, SNDRV_PCM_HW_PARAM_FORMAT); + out_sample_valid_bits = + SOF_IPC4_AUDIO_FORMAT_CFG_V_BIT_DEPTH(copier_data->out_format.fmt_cfg); + snd_mask_none(fmt); + switch (out_sample_valid_bits) { + case 16: + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE); + break; + case 24: + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); + break; + case 32: + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S32_LE); + break; + default: + dev_err(sdev->dev, "invalid sample frame format %d\n", + params_format(pipeline_params)); + return -EINVAL; + } + + /* set the gateway dma_buffer_size using the matched ID returned above */ + copier_data->gtw_cfg.dma_buffer_size = available_fmt->dma_buffer_size[ret]; + + data = &ipc4_copier->copier_config; + ipc_config_size = &ipc4_copier->ipc_config_size; + ipc_config_data = &ipc4_copier->ipc_config_data; + + /* config_length is DWORD based */ + ipc_size = sizeof(*copier_data) + copier_data->gtw_cfg.config_length * 4; + + dev_dbg(sdev->dev, "copier %s, IPC size is %d", swidget->widget->name, ipc_size); + + *ipc_config_data = kzalloc(ipc_size, GFP_KERNEL); + if (!*ipc_config_data) + return -ENOMEM; + + *ipc_config_size = ipc_size; + + /* copy IPC data */ + memcpy(*ipc_config_data, (void *)copier_data, sizeof(*copier_data)); + if (copier_data->gtw_cfg.config_length) + memcpy(*ipc_config_data + sizeof(*copier_data), + *data, copier_data->gtw_cfg.config_length * 4); + + /* update pipeline memory usage */ + sof_ipc4_update_pipeline_mem_usage(sdev, swidget, &copier_data->base_config); + + /* assign instance ID */ + return sof_ipc4_widget_assign_instance_id(sdev, swidget); +} + static enum sof_tokens host_token_list[] = { SOF_COMP_TOKENS, SOF_AUDIO_FMT_NUM_TOKENS, @@ -588,10 +872,12 @@ static enum sof_tokens dai_token_list[] = { static const struct sof_ipc_tplg_widget_ops tplg_ipc4_widget_ops[SND_SOC_DAPM_TYPE_COUNT] = { [snd_soc_dapm_aif_in] = {sof_ipc4_widget_setup_pcm, sof_ipc4_widget_free_comp_pcm, host_token_list, ARRAY_SIZE(host_token_list), NULL, - NULL, NULL}, + sof_ipc4_prepare_copier_module, + sof_ipc4_unprepare_copier_module}, [snd_soc_dapm_aif_out] = {sof_ipc4_widget_setup_pcm, sof_ipc4_widget_free_comp_pcm, host_token_list, ARRAY_SIZE(host_token_list), NULL, - NULL, NULL}, + sof_ipc4_prepare_copier_module, + sof_ipc4_unprepare_copier_module}, [snd_soc_dapm_dai_in] = {sof_ipc4_widget_setup_comp_dai, sof_ipc4_widget_free_comp_dai, dai_token_list, ARRAY_SIZE(dai_token_list), NULL, NULL, NULL}, [snd_soc_dapm_dai_out] = {sof_ipc4_widget_setup_comp_dai, sof_ipc4_widget_free_comp_dai, diff --git a/sound/soc/sof/ipc4-topology.h b/sound/soc/sof/ipc4-topology.h index f4f62dda63a3..5f4c463f329e 100644 --- a/sound/soc/sof/ipc4-topology.h +++ b/sound/soc/sof/ipc4-topology.h @@ -11,6 +11,24 @@ #include +#define SOF_IPC4_FW_PAGE_SIZE BIT(12) +#define SOF_IPC4_FW_PAGE(x) ((((x) + BIT(12) - 1) & ~(BIT(12) - 1)) >> 12) +#define SOF_IPC4_FW_ROUNDUP(x) (((x) + BIT(6) - 1) & (~(BIT(6) - 1))) + +#define SOF_IPC4_MODULE_LL BIT(5) +#define SOF_IPC4_MODULE_INSTANCE_LIST_ITEM_SIZE 12 +#define SOF_IPC4_PIPELINE_OBJECT_SIZE 448 +#define SOF_IPC4_DATA_QUEUE_OBJECT_SIZE 128 +#define SOF_IPC4_LL_TASK_OBJECT_SIZE 72 +#define SOF_IPC4_DP_TASK_OBJECT_SIZE 104 +#define SOF_IPC4_DP_TASK_LIST_SIZE (12 + 8) +#define SOF_IPC4_LL_TASK_LIST_ITEM_SIZE 12 +#define SOF_IPC4_FW_MAX_PAGE_COUNT 20 +#define SOF_IPC4_FW_MAX_QUEUE_COUNT 8 + +/* Node index and mask applicable for host copier */ +#define SOF_IPC4_NODE_INDEX_MASK 0xFF +#define SOF_IPC4_NODE_INDEX(x) ((x) & SOF_IPC4_NODE_INDEX_MASK) #define SOF_IPC4_NODE_TYPE(x) ((x) << 8) /** -- cgit v1.2.3 From acf525942077213e9bc00eee8a73af360ab2fc08 Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Wed, 8 Jun 2022 20:26:26 -0700 Subject: ASoC: SOF: ipc4-topology: Add prepare op for DAI type widgets MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Define the prepare op for the DAI type widgets for IPC4. The prepare op is responsible for choosing the input/output audio formats for these widgets based on the runtime PCM params, assigning the instance ID and updating the total memory usage for the pipelines these widgets belong to. Co-developed-by: Rander Wang Signed-off-by: Rander Wang Co-developed-by: Bard Liao Signed-off-by: Bard Liao Signed-off-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/20220609032643.916882-7-ranjani.sridharan@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-topology.c | 43 ++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 40 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index 1a73c16f1624..1bc5ff0154c5 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -706,8 +706,13 @@ static void sof_ipc4_unprepare_copier_module(struct snd_sof_widget *swidget) pipeline = pipe_widget->private; pipeline->mem_usage = 0; - if (WIDGET_IS_AIF(swidget->id)) + if (WIDGET_IS_AIF(swidget->id)) { ipc4_copier = swidget->private; + } else if (WIDGET_IS_DAI(swidget->id)) { + struct snd_sof_dai *dai = swidget->private; + + ipc4_copier = dai->private; + } if (ipc4_copier) { kfree(ipc4_copier->ipc_config_data); @@ -776,6 +781,34 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, ref_params = fe_params; break; } + case snd_soc_dapm_dai_in: + case snd_soc_dapm_dai_out: + { + struct snd_sof_dai *dai = swidget->private; + + ipc4_copier = (struct sof_ipc4_copier *)dai->private; + copier_data = &ipc4_copier->data; + available_fmt = &ipc4_copier->available_fmt; + if (dir == SNDRV_PCM_STREAM_CAPTURE) { + available_fmt->ref_audio_fmt = available_fmt->out_audio_fmt; + ref_audio_fmt_size = sizeof(struct sof_ipc4_audio_format); + + /* + * modify the input params for the dai copier as it only supports + * 32-bit always + */ + fmt = hw_param_mask(pipeline_params, SNDRV_PCM_HW_PARAM_FORMAT); + snd_mask_none(fmt); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S32_LE); + } else { + available_fmt->ref_audio_fmt = &available_fmt->base_config->audio_fmt; + ref_audio_fmt_size = sizeof(struct sof_ipc4_base_module_cfg); + } + + ref_params = pipeline_params; + + break; + } default: dev_err(sdev->dev, "unsupported type %d for copier %s", swidget->id, swidget->widget->name); @@ -879,9 +912,13 @@ static const struct sof_ipc_tplg_widget_ops tplg_ipc4_widget_ops[SND_SOC_DAPM_TY sof_ipc4_prepare_copier_module, sof_ipc4_unprepare_copier_module}, [snd_soc_dapm_dai_in] = {sof_ipc4_widget_setup_comp_dai, sof_ipc4_widget_free_comp_dai, - dai_token_list, ARRAY_SIZE(dai_token_list), NULL, NULL, NULL}, + dai_token_list, ARRAY_SIZE(dai_token_list), NULL, + sof_ipc4_prepare_copier_module, + sof_ipc4_unprepare_copier_module}, [snd_soc_dapm_dai_out] = {sof_ipc4_widget_setup_comp_dai, sof_ipc4_widget_free_comp_dai, - dai_token_list, ARRAY_SIZE(dai_token_list), NULL, NULL, NULL}, + dai_token_list, ARRAY_SIZE(dai_token_list), NULL, + sof_ipc4_prepare_copier_module, + sof_ipc4_unprepare_copier_module}, [snd_soc_dapm_scheduler] = {sof_ipc4_widget_setup_comp_pipeline, sof_ipc4_widget_free_comp, pipeline_token_list, ARRAY_SIZE(pipeline_token_list), NULL, NULL, NULL}, -- cgit v1.2.3 From 4f838ab2081260119677df3ba94dbbd4f8cb7183 Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Wed, 8 Jun 2022 20:26:27 -0700 Subject: ASoC: SOF: ipc4-topology: Add support for parsing and preparing pga widgets MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add support for parsing and preparing pga type widgets. Define the token ID's and the associated token arrays needed to parse these widgets. Co-developed-by: Rander Wang Signed-off-by: Rander Wang Co-developed-by: Bard Liao Signed-off-by: Bard Liao Signed-off-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Paul Olaru Link: https://lore.kernel.org/r/20220609032643.916882-8-ranjani.sridharan@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-topology.c | 113 ++++++++++++++++++++++++++++++++++++++++++ sound/soc/sof/ipc4-topology.h | 60 ++++++++++++++++++++++ sound/soc/sof/sof-audio.h | 1 + 3 files changed, 174 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index 1bc5ff0154c5..30549573bd34 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -95,6 +95,16 @@ static const struct sof_topology_token comp_ext_tokens[] = { offsetof(struct snd_sof_widget, uuid)}, }; +static const struct sof_topology_token gain_tokens[] = { + {SOF_TKN_GAIN_RAMP_TYPE, SND_SOC_TPLG_TUPLE_TYPE_WORD, + get_token_u32, offsetof(struct sof_ipc4_gain_data, curve_type)}, + {SOF_TKN_GAIN_RAMP_DURATION, + SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc4_gain_data, curve_duration)}, + {SOF_TKN_GAIN_VAL, SND_SOC_TPLG_TUPLE_TYPE_WORD, + get_token_u32, offsetof(struct sof_ipc4_gain_data, init_val)}, +}; + static const struct sof_token_info ipc4_token_list[SOF_TOKEN_COUNT] = { [SOF_DAI_TOKENS] = {"DAI tokens", dai_tokens, ARRAY_SIZE(dai_tokens)}, [SOF_PIPELINE_TOKENS] = {"Pipeline tokens", pipeline_tokens, ARRAY_SIZE(pipeline_tokens)}, @@ -117,6 +127,7 @@ static const struct sof_token_info ipc4_token_list[SOF_TOKEN_COUNT] = { ARRAY_SIZE(ipc4_copier_tokens)}, [SOF_AUDIO_FMT_NUM_TOKENS] = {"IPC4 Audio format number tokens", ipc4_audio_fmt_num_tokens, ARRAY_SIZE(ipc4_audio_fmt_num_tokens)}, + [SOF_GAIN_TOKENS] = {"Gain tokens", gain_tokens, ARRAY_SIZE(gain_tokens)}, }; static void sof_ipc4_dbg_audio_format(struct device *dev, @@ -557,6 +568,62 @@ err: return ret; } +static int sof_ipc4_widget_setup_comp_pga(struct snd_sof_widget *swidget) +{ + struct snd_soc_component *scomp = swidget->scomp; + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); + struct sof_ipc4_fw_module *fw_module; + struct snd_sof_control *scontrol; + struct sof_ipc4_gain *gain; + int ret; + + gain = kzalloc(sizeof(*gain), GFP_KERNEL); + if (!gain) + return -ENOMEM; + + swidget->private = gain; + + gain->data.channels = SOF_IPC4_GAIN_ALL_CHANNELS_MASK; + gain->data.init_val = SOF_IPC4_VOL_ZERO_DB; + + /* The out_audio_fmt in topology is ignored as it is not required to be sent to the FW */ + ret = sof_ipc4_get_audio_fmt(scomp, swidget, &gain->available_fmt, false); + if (ret) + goto err; + + ret = sof_update_ipc_object(scomp, &gain->data, SOF_GAIN_TOKENS, swidget->tuples, + swidget->num_tuples, sizeof(gain->data), 1); + if (ret) { + dev_err(scomp->dev, "Parsing gain tokens failed\n"); + goto err; + } + + dev_dbg(scomp->dev, + "pga widget %s: ramp type: %d, ramp duration %d, initial gain value: %#x, cpc %d\n", + swidget->widget->name, gain->data.curve_type, gain->data.curve_duration, + gain->data.init_val, gain->base_config.cpc); + + ret = sof_ipc4_widget_setup_msg(swidget, &gain->msg); + if (ret) + goto err; + + fw_module = swidget->module_info; + + /* update module ID for all kcontrols for this widget */ + list_for_each_entry(scontrol, &sdev->kcontrol_list, list) + if (scontrol->comp_id == swidget->comp_id) { + struct sof_ipc4_control_data *cdata = scontrol->ipc_control_data; + struct sof_ipc4_msg *msg = &cdata->msg; + + msg->primary |= fw_module->man4_module_entry.id; + } + + return 0; +err: + kfree(gain); + return ret; +} + static void sof_ipc4_update_pipeline_mem_usage(struct snd_sof_dev *sdev, struct snd_sof_widget *swidget, struct sof_ipc4_base_module_cfg *base_config) @@ -874,6 +941,39 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, return sof_ipc4_widget_assign_instance_id(sdev, swidget); } +static void sof_ipc4_unprepare_generic_module(struct snd_sof_widget *swidget) +{ + struct sof_ipc4_fw_module *fw_module = swidget->module_info; + + ida_free(&fw_module->m_ida, swidget->instance_id); +} + +static int sof_ipc4_prepare_gain_module(struct snd_sof_widget *swidget, + struct snd_pcm_hw_params *fe_params, + struct snd_sof_platform_stream_params *platform_params, + struct snd_pcm_hw_params *pipeline_params, int dir) +{ + struct snd_soc_component *scomp = swidget->scomp; + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); + struct sof_ipc4_gain *gain = swidget->private; + int ret; + + gain->available_fmt.ref_audio_fmt = &gain->available_fmt.base_config->audio_fmt; + + /* output format is not required to be sent to the FW for gain */ + ret = sof_ipc4_init_audio_fmt(sdev, swidget, &gain->base_config, + NULL, pipeline_params, &gain->available_fmt, + sizeof(gain->base_config)); + if (ret < 0) + return ret; + + /* update pipeline memory usage */ + sof_ipc4_update_pipeline_mem_usage(sdev, swidget, &gain->base_config); + + /* assign instance ID */ + return sof_ipc4_widget_assign_instance_id(sdev, swidget); +} + static enum sof_tokens host_token_list[] = { SOF_COMP_TOKENS, SOF_AUDIO_FMT_NUM_TOKENS, @@ -902,6 +1002,15 @@ static enum sof_tokens dai_token_list[] = { SOF_COMP_EXT_TOKENS, }; +static enum sof_tokens pga_token_list[] = { + SOF_COMP_TOKENS, + SOF_GAIN_TOKENS, + SOF_AUDIO_FMT_NUM_TOKENS, + SOF_AUDIO_FORMAT_BUFFER_SIZE_TOKENS, + SOF_IN_AUDIO_FORMAT_TOKENS, + SOF_COMP_EXT_TOKENS, +}; + static const struct sof_ipc_tplg_widget_ops tplg_ipc4_widget_ops[SND_SOC_DAPM_TYPE_COUNT] = { [snd_soc_dapm_aif_in] = {sof_ipc4_widget_setup_pcm, sof_ipc4_widget_free_comp_pcm, host_token_list, ARRAY_SIZE(host_token_list), NULL, @@ -922,6 +1031,10 @@ static const struct sof_ipc_tplg_widget_ops tplg_ipc4_widget_ops[SND_SOC_DAPM_TY [snd_soc_dapm_scheduler] = {sof_ipc4_widget_setup_comp_pipeline, sof_ipc4_widget_free_comp, pipeline_token_list, ARRAY_SIZE(pipeline_token_list), NULL, NULL, NULL}, + [snd_soc_dapm_pga] = {sof_ipc4_widget_setup_comp_pga, sof_ipc4_widget_free_comp, + pga_token_list, ARRAY_SIZE(pga_token_list), NULL, + sof_ipc4_prepare_gain_module, + sof_ipc4_unprepare_generic_module}, }; const struct sof_ipc_tplg_ops ipc4_tplg_ops = { diff --git a/sound/soc/sof/ipc4-topology.h b/sound/soc/sof/ipc4-topology.h index 5f4c463f329e..060123826db4 100644 --- a/sound/soc/sof/ipc4-topology.h +++ b/sound/soc/sof/ipc4-topology.h @@ -31,6 +31,9 @@ #define SOF_IPC4_NODE_INDEX(x) ((x) & SOF_IPC4_NODE_INDEX_MASK) #define SOF_IPC4_NODE_TYPE(x) ((x) << 8) +#define SOF_IPC4_GAIN_ALL_CHANNELS_MASK 0xffffffff +#define SOF_IPC4_VOL_ZERO_DB 0x7fffffff + /** * struct sof_ipc4_pipeline - pipeline config data * @priority: Priority of this pipeline @@ -128,4 +131,61 @@ struct sof_ipc4_copier { int dai_index; }; +/** + * struct sof_ipc4_ctrl_value_chan: generic channel mapped value data + * @channel: Channel ID + * @value: gain value + */ +struct sof_ipc4_ctrl_value_chan { + u32 channel; + u32 value; +}; + +/** + * struct sof_ipc4_control_data - IPC data for kcontrol IO + * @msg: message structure for kcontrol IO + * @index: pipeline ID + * @chanv: channel ID and value array used by volume type controls + * @data: data for binary kcontrols + */ +struct sof_ipc4_control_data { + struct sof_ipc4_msg msg; + int index; + + union { + struct sof_ipc4_ctrl_value_chan chanv[0]; + struct sof_abi_hdr data[0]; + }; +}; + +/** + * struct sof_ipc4_gain_data - IPC gain blob + * @channels: Channels + * @init_val: Initial value + * @curve_type: Curve type + * @reserved: reserved for future use + * @curve_duration: Curve duration + */ +struct sof_ipc4_gain_data { + uint32_t channels; + uint32_t init_val; + uint32_t curve_type; + uint32_t reserved; + uint32_t curve_duration; +} __aligned(8); + +/** + * struct sof_ipc4_gain - gain config data + * @base_config: IPC base config data + * @data: IPC gain blob + * @available_fmt: Available audio format + * @msg: message structure for gain + */ +struct sof_ipc4_gain { + struct sof_ipc4_base_module_cfg base_config; + struct sof_ipc4_gain_data data; + struct sof_ipc4_available_audio_format available_fmt; + struct sof_ipc4_msg msg; +}; + #endif diff --git a/sound/soc/sof/sof-audio.h b/sound/soc/sof/sof-audio.h index c38b4bdd685a..d896da1192c5 100644 --- a/sound/soc/sof/sof-audio.h +++ b/sound/soc/sof/sof-audio.h @@ -232,6 +232,7 @@ enum sof_tokens { SOF_COPIER_TOKENS, SOF_AUDIO_FMT_NUM_TOKENS, SOF_COPIER_FORMAT_TOKENS, + SOF_GAIN_TOKENS, /* this should be the last */ SOF_TOKEN_COUNT, -- cgit v1.2.3 From 4d4ba014ac4b3772ed39c15cd2ceacbb071c26f6 Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Wed, 8 Jun 2022 20:26:28 -0700 Subject: ASoC: SOF: ipc4-topology: Add support for parsing mixer widgets MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add support for parsing and preparing mixer type widgets. Define the token ID's and the associated token arrays needed to parse these widgets. Co-developed-by: Rander Wang Signed-off-by: Rander Wang Co-developed-by: Bard Liao Signed-off-by: Bard Liao Signed-off-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Paul Olaru Link: https://lore.kernel.org/r/20220609032643.916882-9-ranjani.sridharan@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-topology.c | 68 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/sof/ipc4-topology.h | 12 ++++++++ 2 files changed, 80 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index 30549573bd34..35457fe4edd9 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -624,6 +624,35 @@ err: return ret; } +static int sof_ipc4_widget_setup_comp_mixer(struct snd_sof_widget *swidget) +{ + struct snd_soc_component *scomp = swidget->scomp; + struct sof_ipc4_mixer *mixer; + int ret; + + dev_dbg(scomp->dev, "Updating IPC structure for %s\n", swidget->widget->name); + + mixer = kzalloc(sizeof(*mixer), GFP_KERNEL); + if (!mixer) + return -ENOMEM; + + swidget->private = mixer; + + /* The out_audio_fmt in topology is ignored as it is not required to be sent to the FW */ + ret = sof_ipc4_get_audio_fmt(scomp, swidget, &mixer->available_fmt, false); + if (ret) + goto err; + + ret = sof_ipc4_widget_setup_msg(swidget, &mixer->msg); + if (ret) + goto err; + + return 0; +err: + kfree(mixer); + return ret; +} + static void sof_ipc4_update_pipeline_mem_usage(struct snd_sof_dev *sdev, struct snd_sof_widget *swidget, struct sof_ipc4_base_module_cfg *base_config) @@ -974,6 +1003,33 @@ static int sof_ipc4_prepare_gain_module(struct snd_sof_widget *swidget, return sof_ipc4_widget_assign_instance_id(sdev, swidget); } +static int sof_ipc4_prepare_mixer_module(struct snd_sof_widget *swidget, + struct snd_pcm_hw_params *fe_params, + struct snd_sof_platform_stream_params *platform_params, + struct snd_pcm_hw_params *pipeline_params, int dir) +{ + struct snd_soc_component *scomp = swidget->scomp; + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); + struct sof_ipc4_mixer *mixer = swidget->private; + int ret; + + /* only 32bit is supported by mixer */ + mixer->available_fmt.ref_audio_fmt = &mixer->available_fmt.base_config->audio_fmt; + + /* output format is not required to be sent to the FW for mixer */ + ret = sof_ipc4_init_audio_fmt(sdev, swidget, &mixer->base_config, + NULL, pipeline_params, &mixer->available_fmt, + sizeof(mixer->base_config)); + if (ret < 0) + return ret; + + /* update pipeline memory usage */ + sof_ipc4_update_pipeline_mem_usage(sdev, swidget, &mixer->base_config); + + /* assign instance ID */ + return sof_ipc4_widget_assign_instance_id(sdev, swidget); +} + static enum sof_tokens host_token_list[] = { SOF_COMP_TOKENS, SOF_AUDIO_FMT_NUM_TOKENS, @@ -1011,6 +1067,14 @@ static enum sof_tokens pga_token_list[] = { SOF_COMP_EXT_TOKENS, }; +static enum sof_tokens mixer_token_list[] = { + SOF_COMP_TOKENS, + SOF_AUDIO_FMT_NUM_TOKENS, + SOF_IN_AUDIO_FORMAT_TOKENS, + SOF_AUDIO_FORMAT_BUFFER_SIZE_TOKENS, + SOF_COMP_EXT_TOKENS, +}; + static const struct sof_ipc_tplg_widget_ops tplg_ipc4_widget_ops[SND_SOC_DAPM_TYPE_COUNT] = { [snd_soc_dapm_aif_in] = {sof_ipc4_widget_setup_pcm, sof_ipc4_widget_free_comp_pcm, host_token_list, ARRAY_SIZE(host_token_list), NULL, @@ -1035,6 +1099,10 @@ static const struct sof_ipc_tplg_widget_ops tplg_ipc4_widget_ops[SND_SOC_DAPM_TY pga_token_list, ARRAY_SIZE(pga_token_list), NULL, sof_ipc4_prepare_gain_module, sof_ipc4_unprepare_generic_module}, + [snd_soc_dapm_mixer] = {sof_ipc4_widget_setup_comp_mixer, sof_ipc4_widget_free_comp, + mixer_token_list, ARRAY_SIZE(mixer_token_list), + NULL, sof_ipc4_prepare_mixer_module, + sof_ipc4_unprepare_generic_module}, }; const struct sof_ipc_tplg_ops ipc4_tplg_ops = { diff --git a/sound/soc/sof/ipc4-topology.h b/sound/soc/sof/ipc4-topology.h index 060123826db4..eebf46b24430 100644 --- a/sound/soc/sof/ipc4-topology.h +++ b/sound/soc/sof/ipc4-topology.h @@ -188,4 +188,16 @@ struct sof_ipc4_gain { struct sof_ipc4_msg msg; }; +/** + * struct sof_ipc4_mixer - mixer config data + * @base_config: IPC base config data + * @available_fmt: Available audio format + * @msg: IPC4 message struct containing header and data info + */ +struct sof_ipc4_mixer { + struct sof_ipc4_base_module_cfg base_config; + struct sof_ipc4_available_audio_format available_fmt; + struct sof_ipc4_msg msg; +}; + #endif -- cgit v1.2.3 From d97964f870786389f4c399a507ffa5d1ebf2a9e4 Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Wed, 8 Jun 2022 20:26:29 -0700 Subject: ASoC: SOF: ipc4-topology: Add control_setup op MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Define the control_setup op for IPC4 topology IPC ops to handle the volume kcontrol types. Support for other kcontrol types will be added in the follow up patches. Co-developed-by: Rander Wang Signed-off-by: Rander Wang Co-developed-by: Bard Liao Signed-off-by: Bard Liao Signed-off-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/20220609032643.916882-10-ranjani.sridharan@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-topology.c | 49 +++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 49 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index 35457fe4edd9..0c36b7cb6e79 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -15,6 +15,8 @@ #include "ipc4-topology.h" #include "ops.h" +#define SOF_IPC4_GAIN_PARAM_ID 0 + static const struct sof_topology_token ipc4_sched_tokens[] = { {SOF_TKN_SCHED_LP_MODE, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, offsetof(struct sof_ipc4_pipeline, lp_mode)} @@ -1030,6 +1032,52 @@ static int sof_ipc4_prepare_mixer_module(struct snd_sof_widget *swidget, return sof_ipc4_widget_assign_instance_id(sdev, swidget); } +static int sof_ipc4_control_load_volume(struct snd_sof_dev *sdev, struct snd_sof_control *scontrol) +{ + struct sof_ipc4_control_data *control_data; + struct sof_ipc4_msg *msg; + int i; + + scontrol->size = struct_size(control_data, chanv, scontrol->num_channels); + + /* scontrol->ipc_control_data will be freed in sof_control_unload */ + scontrol->ipc_control_data = kzalloc(scontrol->size, GFP_KERNEL); + if (!scontrol->ipc_control_data) + return -ENOMEM; + + control_data = scontrol->ipc_control_data; + control_data->index = scontrol->index; + + msg = &control_data->msg; + msg->primary = SOF_IPC4_MSG_TYPE_SET(SOF_IPC4_MOD_LARGE_CONFIG_SET); + msg->primary |= SOF_IPC4_MSG_DIR(SOF_IPC4_MSG_REQUEST); + msg->primary |= SOF_IPC4_MSG_TARGET(SOF_IPC4_MODULE_MSG); + + msg->extension = SOF_IPC4_MOD_EXT_MSG_PARAM_ID(SOF_IPC4_GAIN_PARAM_ID); + + /* set default volume values to 0dB in control */ + for (i = 0; i < scontrol->num_channels; i++) { + control_data->chanv[i].channel = i; + control_data->chanv[i].value = SOF_IPC4_VOL_ZERO_DB; + } + + return 0; +} + +static int sof_ipc4_control_setup(struct snd_sof_dev *sdev, struct snd_sof_control *scontrol) +{ + switch (scontrol->info_type) { + case SND_SOC_TPLG_CTL_VOLSW: + case SND_SOC_TPLG_CTL_VOLSW_SX: + case SND_SOC_TPLG_CTL_VOLSW_XR_SX: + return sof_ipc4_control_load_volume(sdev, scontrol); + default: + break; + } + + return 0; +} + static enum sof_tokens host_token_list[] = { SOF_COMP_TOKENS, SOF_AUDIO_FMT_NUM_TOKENS, @@ -1108,4 +1156,5 @@ static const struct sof_ipc_tplg_widget_ops tplg_ipc4_widget_ops[SND_SOC_DAPM_TY const struct sof_ipc_tplg_ops ipc4_tplg_ops = { .widget = tplg_ipc4_widget_ops, .token_list = ipc4_token_list, + .control_setup = sof_ipc4_control_setup, }; -- cgit v1.2.3 From 955e84fc0b6df6cfb95ee6f569be809af49d8287 Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Wed, 8 Jun 2022 20:26:30 -0700 Subject: ASoC: SOF: ipc4-topology: Add control IO ops MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Define the kcontrol IO ops for volume type controls for IPC4. Support for other kcontrol types will be added later. Co-developed-by: Rander Wang Signed-off-by: Rander Wang Co-developed-by: Bard Liao Signed-off-by: Bard Liao Signed-off-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/20220609032643.916882-11-ranjani.sridharan@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/Makefile | 2 +- sound/soc/sof/ipc4-control.c | 216 ++++++++++++++++++++++++++++++++++++++++++ sound/soc/sof/ipc4-priv.h | 1 + sound/soc/sof/ipc4-topology.c | 1 + 4 files changed, 219 insertions(+), 1 deletion(-) create mode 100644 sound/soc/sof/ipc4-control.c (limited to 'sound') diff --git a/sound/soc/sof/Makefile b/sound/soc/sof/Makefile index 73524fadb3ce..1e15937f2bde 100644 --- a/sound/soc/sof/Makefile +++ b/sound/soc/sof/Makefile @@ -4,7 +4,7 @@ snd-sof-objs := core.o ops.o loader.o ipc.o pcm.o pm.o debug.o topology.o\ control.o trace.o iomem-utils.o sof-audio.o stream-ipc.o\ ipc3-topology.o ipc3-control.o ipc3.o ipc3-pcm.o ipc3-loader.o\ ipc3-dtrace.o\ - ipc4.o ipc4-loader.o ipc4-topology.o + ipc4.o ipc4-loader.o ipc4-topology.o ipc4-control.o ifneq ($(CONFIG_SND_SOC_SOF_CLIENT),) snd-sof-objs += sof-client.o endif diff --git a/sound/soc/sof/ipc4-control.c b/sound/soc/sof/ipc4-control.c new file mode 100644 index 000000000000..95ee121dd3cf --- /dev/null +++ b/sound/soc/sof/ipc4-control.c @@ -0,0 +1,216 @@ +// SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) +// +// This file is provided under a dual BSD/GPLv2 license. When using or +// redistributing this file, you may do so under either license. +// +// Copyright(c) 2022 Intel Corporation. All rights reserved. +// +// + +#include "sof-priv.h" +#include "sof-audio.h" +#include "ipc4-priv.h" +#include "ipc4-topology.h" + +static int sof_ipc4_set_get_kcontrol_data(struct snd_sof_control *scontrol, bool set) +{ + struct sof_ipc4_control_data *cdata = scontrol->ipc_control_data; + struct snd_soc_component *scomp = scontrol->scomp; + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); + const struct sof_ipc_ops *iops = sdev->ipc->ops; + struct sof_ipc4_msg *msg = &cdata->msg; + struct snd_sof_widget *swidget; + bool widget_found = false; + + /* find widget associated with the control */ + list_for_each_entry(swidget, &sdev->widget_list, list) { + if (swidget->comp_id == scontrol->comp_id) { + widget_found = true; + break; + } + } + + if (!widget_found) { + dev_err(scomp->dev, "Failed to find widget for kcontrol %s\n", scontrol->name); + return -ENOENT; + } + + /* + * Volatile controls should always be part of static pipelines and the widget use_count + * would always be > 0 in this case. For the others, just return the cached value if the + * widget is not set up. + */ + if (!swidget->use_count) + return 0; + + msg->primary &= ~SOF_IPC4_MOD_INSTANCE_MASK; + msg->primary |= SOF_IPC4_MOD_INSTANCE(swidget->instance_id); + + return iops->set_get_data(sdev, msg, msg->data_size, set); +} + +static int +sof_ipc4_set_volume_data(struct snd_sof_dev *sdev, struct snd_sof_widget *swidget, + struct snd_sof_control *scontrol) +{ + struct sof_ipc4_control_data *cdata = scontrol->ipc_control_data; + struct sof_ipc4_gain *gain = swidget->private; + struct sof_ipc4_msg *msg = &cdata->msg; + struct sof_ipc4_gain_data data; + bool all_channels_equal = true; + u32 value; + int ret, i; + + /* check if all channel values are equal */ + value = cdata->chanv[0].value; + for (i = 1; i < scontrol->num_channels; i++) { + if (cdata->chanv[i].value != value) { + all_channels_equal = false; + break; + } + } + + /* + * notify DSP with a single IPC message if all channel values are equal. Otherwise send + * a separate IPC for each channel. + */ + for (i = 0; i < scontrol->num_channels; i++) { + if (all_channels_equal) { + data.channels = SOF_IPC4_GAIN_ALL_CHANNELS_MASK; + data.init_val = cdata->chanv[0].value; + } else { + data.channels = cdata->chanv[i].channel; + data.init_val = cdata->chanv[i].value; + } + + /* set curve type and duration from topology */ + data.curve_duration = gain->data.curve_duration; + data.curve_type = gain->data.curve_type; + + msg->data_ptr = &data; + msg->data_size = sizeof(data); + + ret = sof_ipc4_set_get_kcontrol_data(scontrol, true); + msg->data_ptr = NULL; + msg->data_size = 0; + if (ret < 0) { + dev_err(sdev->dev, "Failed to set volume update for %s\n", + scontrol->name); + return ret; + } + + if (all_channels_equal) + break; + } + + return 0; +} + +static bool sof_ipc4_volume_put(struct snd_sof_control *scontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct sof_ipc4_control_data *cdata = scontrol->ipc_control_data; + struct snd_soc_component *scomp = scontrol->scomp; + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); + unsigned int channels = scontrol->num_channels; + struct snd_sof_widget *swidget; + bool widget_found = false; + bool change = false; + unsigned int i; + int ret; + + /* update each channel */ + for (i = 0; i < channels; i++) { + u32 value = mixer_to_ipc(ucontrol->value.integer.value[i], + scontrol->volume_table, scontrol->max + 1); + + change = change || (value != cdata->chanv[i].value); + cdata->chanv[i].channel = i; + cdata->chanv[i].value = value; + } + + if (!pm_runtime_active(scomp->dev)) + return change; + + /* find widget associated with the control */ + list_for_each_entry(swidget, &sdev->widget_list, list) { + if (swidget->comp_id == scontrol->comp_id) { + widget_found = true; + break; + } + } + + if (!widget_found) { + dev_err(scomp->dev, "Failed to find widget for kcontrol %s\n", scontrol->name); + return -ENOENT; + } + + ret = sof_ipc4_set_volume_data(sdev, swidget, scontrol); + if (ret < 0) + return false; + + return change; +} + +static int sof_ipc4_volume_get(struct snd_sof_control *scontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct sof_ipc4_control_data *cdata = scontrol->ipc_control_data; + unsigned int channels = scontrol->num_channels; + unsigned int i; + + for (i = 0; i < channels; i++) + ucontrol->value.integer.value[i] = ipc_to_mixer(cdata->chanv[i].value, + scontrol->volume_table, + scontrol->max + 1); + + return 0; +} + +/* set up all controls for the widget */ +static int sof_ipc4_widget_kcontrol_setup(struct snd_sof_dev *sdev, struct snd_sof_widget *swidget) +{ + struct snd_sof_control *scontrol; + int ret; + + list_for_each_entry(scontrol, &sdev->kcontrol_list, list) + if (scontrol->comp_id == swidget->comp_id) { + ret = sof_ipc4_set_volume_data(sdev, swidget, scontrol); + if (ret < 0) { + dev_err(sdev->dev, "%s: kcontrol %d set up failed for widget %s\n", + __func__, scontrol->comp_id, swidget->widget->name); + return ret; + } + } + + return 0; +} + +static int +sof_ipc4_set_up_volume_table(struct snd_sof_control *scontrol, int tlv[SOF_TLV_ITEMS], int size) +{ + int i; + + /* init the volume table */ + scontrol->volume_table = kcalloc(size, sizeof(u32), GFP_KERNEL); + if (!scontrol->volume_table) + return -ENOMEM; + + /* populate the volume table */ + for (i = 0; i < size ; i++) { + u32 val = vol_compute_gain(i, tlv); + u64 q31val = ((u64)val) << 15; /* Can be over Q1.31, need to saturate */ + + scontrol->volume_table[i] = q31val > SOF_IPC4_VOL_ZERO_DB ? + SOF_IPC4_VOL_ZERO_DB : q31val; + } + + return 0; +} + +const struct sof_ipc_tplg_control_ops tplg_ipc4_control_ops = { + .volume_put = sof_ipc4_volume_put, + .volume_get = sof_ipc4_volume_get, + .widget_kcontrol_setup = sof_ipc4_widget_kcontrol_setup, + .set_up_volume_table = sof_ipc4_set_up_volume_table, +}; diff --git a/sound/soc/sof/ipc4-priv.h b/sound/soc/sof/ipc4-priv.h index 5388b888fefa..d0b110811aeb 100644 --- a/sound/soc/sof/ipc4-priv.h +++ b/sound/soc/sof/ipc4-priv.h @@ -41,5 +41,6 @@ struct sof_ipc4_fw_module { extern const struct sof_ipc_fw_loader_ops ipc4_loader_ops; extern const struct sof_ipc_tplg_ops ipc4_tplg_ops; +extern const struct sof_ipc_tplg_control_ops tplg_ipc4_control_ops; #endif diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index 0c36b7cb6e79..3cebd6fe7cd1 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -1157,4 +1157,5 @@ const struct sof_ipc_tplg_ops ipc4_tplg_ops = { .widget = tplg_ipc4_widget_ops, .token_list = ipc4_token_list, .control_setup = sof_ipc4_control_setup, + .control = &tplg_ipc4_control_ops, }; -- cgit v1.2.3 From e75e5db8f8ac5b9d4e8968060822bed4671f22ec Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Wed, 8 Jun 2022 20:26:31 -0700 Subject: ASoC: SOF: IPC4: Add pcm ops MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Define and set the PCM ops for IPC4. Co-developed-by: Rander Wang Signed-off-by: Rander Wang Co-developed-by: Bard Liao Signed-off-by: Bard Liao Reviewed-by: Péter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Yaochun Hung Signed-off-by: Ranjani Sridharan Link: https://lore.kernel.org/r/20220609032643.916882-12-ranjani.sridharan@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/Makefile | 2 +- sound/soc/sof/ipc4-pcm.c | 229 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/sof/ipc4-priv.h | 1 + sound/soc/sof/ipc4.c | 1 + 4 files changed, 232 insertions(+), 1 deletion(-) create mode 100644 sound/soc/sof/ipc4-pcm.c (limited to 'sound') diff --git a/sound/soc/sof/Makefile b/sound/soc/sof/Makefile index 1e15937f2bde..2fa8088707a8 100644 --- a/sound/soc/sof/Makefile +++ b/sound/soc/sof/Makefile @@ -4,7 +4,7 @@ snd-sof-objs := core.o ops.o loader.o ipc.o pcm.o pm.o debug.o topology.o\ control.o trace.o iomem-utils.o sof-audio.o stream-ipc.o\ ipc3-topology.o ipc3-control.o ipc3.o ipc3-pcm.o ipc3-loader.o\ ipc3-dtrace.o\ - ipc4.o ipc4-loader.o ipc4-topology.o ipc4-control.o + ipc4.o ipc4-loader.o ipc4-topology.o ipc4-control.o ipc4-pcm.o ifneq ($(CONFIG_SND_SOC_SOF_CLIENT),) snd-sof-objs += sof-client.o endif diff --git a/sound/soc/sof/ipc4-pcm.c b/sound/soc/sof/ipc4-pcm.c new file mode 100644 index 000000000000..7a56fba8f1d9 --- /dev/null +++ b/sound/soc/sof/ipc4-pcm.c @@ -0,0 +1,229 @@ +// SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) +// +// This file is provided under a dual BSD/GPLv2 license. When using or +// redistributing this file, you may do so under either license. +// +// Copyright(c) 2022 Intel Corporation. All rights reserved. +// + +#include +#include +#include "sof-audio.h" +#include "sof-priv.h" +#include "ipc4-priv.h" +#include "ipc4-topology.h" + +static int sof_ipc4_set_pipeline_state(struct snd_sof_dev *sdev, u32 id, u32 state) +{ + struct sof_ipc4_msg msg = {{ 0 }}; + u32 primary; + + dev_dbg(sdev->dev, "ipc4 set pipeline %d state %d", id, state); + + primary = state; + primary |= SOF_IPC4_GLB_PIPE_STATE_ID(id); + primary |= SOF_IPC4_MSG_TYPE_SET(SOF_IPC4_GLB_SET_PIPELINE_STATE); + primary |= SOF_IPC4_MSG_DIR(SOF_IPC4_MSG_REQUEST); + primary |= SOF_IPC4_MSG_TARGET(SOF_IPC4_FW_GEN_MSG); + + msg.primary = primary; + + return sof_ipc_tx_message(sdev->ipc, &msg, 0, NULL, 0); +} + +static int sof_ipc4_trigger_pipelines(struct snd_soc_component *component, + struct snd_pcm_substream *substream, int state) +{ + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component); + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_sof_widget *pipeline_widget; + struct snd_soc_dapm_widget_list *list; + struct snd_soc_dapm_widget *widget; + struct sof_ipc4_pipeline *pipeline; + struct snd_sof_widget *swidget; + struct snd_sof_pcm *spcm; + int ret = 0; + int num_widgets; + + spcm = snd_sof_find_spcm_dai(component, rtd); + if (!spcm) + return -EINVAL; + + list = spcm->stream[substream->stream].list; + + for_each_dapm_widgets(list, num_widgets, widget) { + swidget = widget->dobj.private; + + if (!swidget) + continue; + + /* + * set pipeline state for both FE and BE pipelines for RUNNING state. + * For PAUSE/RESET, set the pipeline state only for the FE pipeline. + */ + switch (state) { + case SOF_IPC4_PIPE_PAUSED: + case SOF_IPC4_PIPE_RESET: + if (!WIDGET_IS_AIF(swidget->id)) + continue; + break; + default: + break; + } + + /* find pipeline widget for the pipeline that this widget belongs to */ + pipeline_widget = swidget->pipe_widget; + pipeline = (struct sof_ipc4_pipeline *)pipeline_widget->private; + + if (pipeline->state == state) + continue; + + /* first set the pipeline to PAUSED state */ + if (pipeline->state != SOF_IPC4_PIPE_PAUSED) { + ret = sof_ipc4_set_pipeline_state(sdev, swidget->pipeline_id, + SOF_IPC4_PIPE_PAUSED); + if (ret < 0) { + dev_err(sdev->dev, "failed to pause pipeline %d\n", + swidget->pipeline_id); + return ret; + } + } + + pipeline->state = SOF_IPC4_PIPE_PAUSED; + + if (pipeline->state == state) + continue; + + /* then set the final state */ + ret = sof_ipc4_set_pipeline_state(sdev, swidget->pipeline_id, state); + if (ret < 0) { + dev_err(sdev->dev, "failed to set state %d for pipeline %d\n", + state, swidget->pipeline_id); + break; + } + + pipeline->state = state; + } + + return ret; +} + +static int sof_ipc4_pcm_trigger(struct snd_soc_component *component, + struct snd_pcm_substream *substream, int cmd) +{ + int state; + + /* determine the pipeline state */ + switch (cmd) { + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + state = SOF_IPC4_PIPE_PAUSED; + break; + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_START: + state = SOF_IPC4_PIPE_RUNNING; + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_STOP: + state = SOF_IPC4_PIPE_PAUSED; + break; + default: + dev_err(component->dev, "%s: unhandled trigger cmd %d\n", __func__, cmd); + return -EINVAL; + } + + /* set the pipeline state */ + return sof_ipc4_trigger_pipelines(component, substream, state); +} + +static int sof_ipc4_pcm_hw_free(struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + return sof_ipc4_trigger_pipelines(component, substream, SOF_IPC4_PIPE_RESET); +} + +static void ipc4_ssp_dai_config_pcm_params_match(struct snd_sof_dev *sdev, const char *link_name, + struct snd_pcm_hw_params *params) +{ + struct snd_sof_dai_link *slink; + struct snd_sof_dai *dai; + bool dai_link_found = false; + int i; + + list_for_each_entry(slink, &sdev->dai_link_list, list) { + if (!strcmp(slink->link->name, link_name)) { + dai_link_found = true; + break; + } + } + + if (!dai_link_found) + return; + + for (i = 0; i < slink->num_hw_configs; i++) { + struct snd_soc_tplg_hw_config *hw_config = &slink->hw_configs[i]; + + if (params_rate(params) == le32_to_cpu(hw_config->fsync_rate)) { + /* set current config for all DAI's with matching name */ + list_for_each_entry(dai, &sdev->dai_list, list) + if (!strcmp(slink->link->name, dai->name)) + dai->current_config = le32_to_cpu(hw_config->id); + break; + } + } +} + +static int sof_ipc4_pcm_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_component *component = snd_soc_rtdcom_lookup(rtd, SOF_AUDIO_PCM_DRV_NAME); + struct snd_sof_dai *dai = snd_sof_find_dai(component, rtd->dai_link->name); + struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component); + struct sof_ipc4_copier *ipc4_copier; + struct snd_soc_dpcm *dpcm; + + if (!dai) { + dev_err(component->dev, "%s: No DAI found with name %s\n", __func__, + rtd->dai_link->name); + return -EINVAL; + } + + ipc4_copier = dai->private; + if (!ipc4_copier) { + dev_err(component->dev, "%s: No private data found for DAI %s\n", + __func__, rtd->dai_link->name); + return -EINVAL; + } + + /* always set BE format to 32-bits for both playback and capture */ + snd_mask_none(fmt); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S32_LE); + + /* + * Set trigger order for capture to SND_SOC_DPCM_TRIGGER_PRE. This is required + * to ensure that the BE DAI pipeline gets stopped/suspended before the FE DAI + * pipeline gets triggered and the pipeline widgets are freed. + */ + for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_CAPTURE, dpcm) { + struct snd_soc_pcm_runtime *fe = dpcm->fe; + + fe->dai_link->trigger[SNDRV_PCM_STREAM_CAPTURE] = SND_SOC_DPCM_TRIGGER_PRE; + } + + switch (ipc4_copier->dai_type) { + case SOF_DAI_INTEL_SSP: + ipc4_ssp_dai_config_pcm_params_match(sdev, (char *)rtd->dai_link->name, params); + break; + default: + break; + } + + return 0; +} + +const struct sof_ipc_pcm_ops ipc4_pcm_ops = { + .trigger = sof_ipc4_pcm_trigger, + .hw_free = sof_ipc4_pcm_hw_free, + .dai_link_fixup = sof_ipc4_pcm_dai_link_fixup, +}; diff --git a/sound/soc/sof/ipc4-priv.h b/sound/soc/sof/ipc4-priv.h index d0b110811aeb..e4381a74516c 100644 --- a/sound/soc/sof/ipc4-priv.h +++ b/sound/soc/sof/ipc4-priv.h @@ -42,5 +42,6 @@ struct sof_ipc4_fw_module { extern const struct sof_ipc_fw_loader_ops ipc4_loader_ops; extern const struct sof_ipc_tplg_ops ipc4_tplg_ops; extern const struct sof_ipc_tplg_control_ops tplg_ipc4_control_ops; +extern const struct sof_ipc_pcm_ops ipc4_pcm_ops; #endif diff --git a/sound/soc/sof/ipc4.c b/sound/soc/sof/ipc4.c index be677a33882d..700069e759c4 100644 --- a/sound/soc/sof/ipc4.c +++ b/sound/soc/sof/ipc4.c @@ -604,4 +604,5 @@ const struct sof_ipc_ops ipc4_ops = { .get_reply = sof_ipc4_get_reply, .fw_loader = &ipc4_loader_ops, .tplg = &ipc4_tplg_ops, + .pcm = &ipc4_pcm_ops, }; -- cgit v1.2.3 From 6e9257a13c75b2e4fc33477f9de4912fdfae81e1 Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Wed, 8 Jun 2022 20:26:32 -0700 Subject: ASoC: SOF: ipc4-topology: Add widget_setup/widget_free ops MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Define and set the widget_setup/widget_free ops for IPC4. Co-developed-by: Rander Wang Signed-off-by: Rander Wang Co-developed-by: Bard Liao Signed-off-by: Bard Liao Signed-off-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Link: https://lore.kernel.org/r/20220609032643.916882-13-ranjani.sridharan@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-topology.c | 123 ++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 123 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index 3cebd6fe7cd1..44f65b8b526a 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -1078,6 +1078,127 @@ static int sof_ipc4_control_setup(struct snd_sof_dev *sdev, struct snd_sof_contr return 0; } +static int sof_ipc4_widget_setup(struct snd_sof_dev *sdev, struct snd_sof_widget *swidget) +{ + struct snd_sof_widget *pipe_widget = swidget->pipe_widget; + struct sof_ipc4_pipeline *pipeline; + struct sof_ipc4_msg *msg; + void *ipc_data = NULL; + u32 ipc_size = 0; + int ret; + + dev_dbg(sdev->dev, "Create widget %s instance %d - pipe %d - core %d\n", + swidget->widget->name, swidget->instance_id, swidget->pipeline_id, swidget->core); + + switch (swidget->id) { + case snd_soc_dapm_scheduler: + pipeline = swidget->private; + + dev_dbg(sdev->dev, "pipeline: %d memory pages: %d\n", swidget->pipeline_id, + pipeline->mem_usage); + + msg = &pipeline->msg; + msg->primary |= pipeline->mem_usage; + break; + case snd_soc_dapm_aif_in: + case snd_soc_dapm_aif_out: + { + struct sof_ipc4_copier *ipc4_copier = swidget->private; + + ipc_size = ipc4_copier->ipc_config_size; + ipc_data = ipc4_copier->ipc_config_data; + + msg = &ipc4_copier->msg; + break; + } + case snd_soc_dapm_dai_in: + case snd_soc_dapm_dai_out: + { + struct snd_sof_dai *dai = swidget->private; + struct sof_ipc4_copier *ipc4_copier = dai->private; + + ipc_size = ipc4_copier->ipc_config_size; + ipc_data = ipc4_copier->ipc_config_data; + + msg = &ipc4_copier->msg; + break; + } + case snd_soc_dapm_pga: + { + struct sof_ipc4_gain *gain = swidget->private; + + ipc_size = sizeof(struct sof_ipc4_base_module_cfg) + + sizeof(struct sof_ipc4_gain_data); + ipc_data = gain; + + msg = &gain->msg; + break; + } + case snd_soc_dapm_mixer: + { + struct sof_ipc4_mixer *mixer = swidget->private; + + ipc_size = sizeof(mixer->base_config); + ipc_data = &mixer->base_config; + + msg = &mixer->msg; + break; + } + default: + dev_err(sdev->dev, "widget type %d not supported", swidget->id); + return -EINVAL; + } + + if (swidget->id != snd_soc_dapm_scheduler) { + pipeline = pipe_widget->private; + msg->primary &= ~SOF_IPC4_MOD_INSTANCE_MASK; + msg->primary |= SOF_IPC4_MOD_INSTANCE(swidget->instance_id); + + msg->extension &= ~SOF_IPC4_MOD_EXT_PARAM_SIZE_MASK; + msg->extension |= ipc_size >> 2; + msg->extension &= ~SOF_IPC4_MOD_EXT_DOMAIN_MASK; + msg->extension |= SOF_IPC4_MOD_EXT_DOMAIN(pipeline->lp_mode); + } + + msg->data_size = ipc_size; + msg->data_ptr = ipc_data; + + ret = sof_ipc_tx_message(sdev->ipc, msg, ipc_size, NULL, 0); + if (ret < 0) + dev_err(sdev->dev, "failed to create module %s\n", swidget->widget->name); + + return ret; +} + +static int sof_ipc4_widget_free(struct snd_sof_dev *sdev, struct snd_sof_widget *swidget) +{ + int ret = 0; + + /* freeing a pipeline frees all the widgets associated with it */ + if (swidget->id == snd_soc_dapm_scheduler) { + struct sof_ipc4_pipeline *pipeline = swidget->private; + struct sof_ipc4_msg msg = {{ 0 }}; + u32 header; + + header = SOF_IPC4_GLB_PIPE_INSTANCE_ID(swidget->pipeline_id); + header |= SOF_IPC4_MSG_TYPE_SET(SOF_IPC4_GLB_DELETE_PIPELINE); + header |= SOF_IPC4_MSG_DIR(SOF_IPC4_MSG_REQUEST); + header |= SOF_IPC4_MSG_TARGET(SOF_IPC4_FW_GEN_MSG); + + msg.primary = header; + + ret = sof_ipc_tx_message(sdev->ipc, &msg, 0, NULL, 0); + if (ret < 0) + dev_err(sdev->dev, "failed to free pipeline widget %s\n", + swidget->widget->name); + + pipeline->mem_usage = 0; + pipeline->state = SOF_IPC4_PIPE_UNINITIALIZED; + } + + return ret; +} + static enum sof_tokens host_token_list[] = { SOF_COMP_TOKENS, SOF_AUDIO_FMT_NUM_TOKENS, @@ -1158,4 +1279,6 @@ const struct sof_ipc_tplg_ops ipc4_tplg_ops = { .token_list = ipc4_token_list, .control_setup = sof_ipc4_control_setup, .control = &tplg_ipc4_control_ops, + .widget_setup = sof_ipc4_widget_setup, + .widget_free = sof_ipc4_widget_free, }; -- cgit v1.2.3 From 3acd527089463742a3dd95e274d53c2fdd834716 Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Wed, 8 Jun 2022 20:26:33 -0700 Subject: ASoC: SOF: ipc4-topology: Add route_setup/route_free ops MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Define and set the route_setup/route_free ops for IPC4. Co-developed-by: Rander Wang Signed-off-by: Rander Wang Co-developed-by: Bard Liao Signed-off-by: Bard Liao Signed-off-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Link: https://lore.kernel.org/r/20220609032643.916882-14-ranjani.sridharan@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-topology.c | 76 +++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 76 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index 44f65b8b526a..f5067d630f2d 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -1199,6 +1199,80 @@ static int sof_ipc4_widget_free(struct snd_sof_dev *sdev, struct snd_sof_widget return ret; } +static int sof_ipc4_route_setup(struct snd_sof_dev *sdev, struct snd_sof_route *sroute) +{ + struct snd_sof_widget *src_widget = sroute->src_widget; + struct snd_sof_widget *sink_widget = sroute->sink_widget; + struct sof_ipc4_fw_module *src_fw_module = src_widget->module_info; + struct sof_ipc4_fw_module *sink_fw_module = sink_widget->module_info; + struct sof_ipc4_msg msg = {{ 0 }}; + u32 header, extension; + int src_queue = 0; + int dst_queue = 0; + int ret; + + dev_dbg(sdev->dev, "%s: bind %s -> %s\n", __func__, + src_widget->widget->name, sink_widget->widget->name); + + header = src_fw_module->man4_module_entry.id; + header |= SOF_IPC4_MOD_INSTANCE(src_widget->instance_id); + header |= SOF_IPC4_MSG_TYPE_SET(SOF_IPC4_MOD_BIND); + header |= SOF_IPC4_MSG_DIR(SOF_IPC4_MSG_REQUEST); + header |= SOF_IPC4_MSG_TARGET(SOF_IPC4_MODULE_MSG); + + extension = sink_fw_module->man4_module_entry.id; + extension |= SOF_IPC4_MOD_EXT_DST_MOD_INSTANCE(sink_widget->instance_id); + extension |= SOF_IPC4_MOD_EXT_DST_MOD_QUEUE_ID(dst_queue); + extension |= SOF_IPC4_MOD_EXT_SRC_MOD_QUEUE_ID(src_queue); + + msg.primary = header; + msg.extension = extension; + + ret = sof_ipc_tx_message(sdev->ipc, &msg, 0, NULL, 0); + if (ret < 0) + dev_err(sdev->dev, "%s: failed to bind modules %s -> %s\n", + __func__, src_widget->widget->name, sink_widget->widget->name); + + return ret; +} + +static int sof_ipc4_route_free(struct snd_sof_dev *sdev, struct snd_sof_route *sroute) +{ + struct snd_sof_widget *src_widget = sroute->src_widget; + struct snd_sof_widget *sink_widget = sroute->sink_widget; + struct sof_ipc4_fw_module *src_fw_module = src_widget->module_info; + struct sof_ipc4_fw_module *sink_fw_module = sink_widget->module_info; + struct sof_ipc4_msg msg = {{ 0 }}; + u32 header, extension; + int src_queue = 0; + int dst_queue = 0; + int ret; + + dev_dbg(sdev->dev, "%s: unbind modules %s -> %s\n", __func__, + src_widget->widget->name, sink_widget->widget->name); + + header = src_fw_module->man4_module_entry.id; + header |= SOF_IPC4_MOD_INSTANCE(src_widget->instance_id); + header |= SOF_IPC4_MSG_TYPE_SET(SOF_IPC4_MOD_UNBIND); + header |= SOF_IPC4_MSG_DIR(SOF_IPC4_MSG_REQUEST); + header |= SOF_IPC4_MSG_TARGET(SOF_IPC4_MODULE_MSG); + + extension = sink_fw_module->man4_module_entry.id; + extension |= SOF_IPC4_MOD_EXT_DST_MOD_INSTANCE(sink_widget->instance_id); + extension |= SOF_IPC4_MOD_EXT_DST_MOD_QUEUE_ID(dst_queue); + extension |= SOF_IPC4_MOD_EXT_SRC_MOD_QUEUE_ID(src_queue); + + msg.primary = header; + msg.extension = extension; + + ret = sof_ipc_tx_message(sdev->ipc, &msg, 0, NULL, 0); + if (ret < 0) + dev_err(sdev->dev, "failed to unbind modules %s -> %s\n", + src_widget->widget->name, sink_widget->widget->name); + + return ret; +} + static enum sof_tokens host_token_list[] = { SOF_COMP_TOKENS, SOF_AUDIO_FMT_NUM_TOKENS, @@ -1281,4 +1355,6 @@ const struct sof_ipc_tplg_ops ipc4_tplg_ops = { .control = &tplg_ipc4_control_ops, .widget_setup = sof_ipc4_widget_setup, .widget_free = sof_ipc4_widget_free, + .route_setup = sof_ipc4_route_setup, + .route_free = sof_ipc4_route_free, }; -- cgit v1.2.3 From acf48a1f76b887f6a63f3c91eedac80b38341c05 Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Wed, 8 Jun 2022 20:26:34 -0700 Subject: ASoC: SOF: ipc4-topology: Add the dai_config op MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Define and set the dai_config op for IPC4. Co-developed-by: Rander Wang Signed-off-by: Rander Wang Co-developed-by: Bard Liao Signed-off-by: Bard Liao Signed-off-by: Ranjani Sridharan Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220609032643.916882-15-ranjani.sridharan@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-topology.c | 45 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/sof/ipc4-topology.h | 2 +- 2 files changed, 46 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index f5067d630f2d..9615034f8c70 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -1273,6 +1273,50 @@ static int sof_ipc4_route_free(struct snd_sof_dev *sdev, struct snd_sof_route *s return ret; } +static int sof_ipc4_dai_config(struct snd_sof_dev *sdev, struct snd_sof_widget *swidget, + unsigned int flags, struct snd_sof_dai_config_data *data) +{ + struct snd_sof_widget *pipe_widget = swidget->pipe_widget; + struct sof_ipc4_pipeline *pipeline = pipe_widget->private; + struct snd_sof_dai *dai = swidget->private; + struct sof_ipc4_gtw_attributes *gtw_attr; + struct sof_ipc4_copier_data *copier_data; + struct sof_ipc4_copier *ipc4_copier; + + if (!dai || !dai->private) { + dev_err(sdev->dev, "Invalid DAI or DAI private data for %s\n", + swidget->widget->name); + return -EINVAL; + } + + ipc4_copier = (struct sof_ipc4_copier *)dai->private; + copier_data = &ipc4_copier->data; + + if (!data) + return 0; + + switch (ipc4_copier->dai_type) { + case SOF_DAI_INTEL_HDA: + gtw_attr = ipc4_copier->gtw_attr; + gtw_attr->lp_buffer_alloc = pipeline->lp_mode; + fallthrough; + case SOF_DAI_INTEL_ALH: + copier_data->gtw_cfg.node_id &= ~SOF_IPC4_NODE_INDEX_MASK; + copier_data->gtw_cfg.node_id |= SOF_IPC4_NODE_INDEX(data->dai_data); + break; + case SOF_DAI_INTEL_DMIC: + case SOF_DAI_INTEL_SSP: + /* nothing to do for SSP/DMIC */ + break; + default: + dev_err(sdev->dev, "%s: unsupported dai type %d\n", __func__, + ipc4_copier->dai_type); + return -EINVAL; + } + + return 0; +} + static enum sof_tokens host_token_list[] = { SOF_COMP_TOKENS, SOF_AUDIO_FMT_NUM_TOKENS, @@ -1357,4 +1401,5 @@ const struct sof_ipc_tplg_ops ipc4_tplg_ops = { .widget_free = sof_ipc4_widget_free, .route_setup = sof_ipc4_route_setup, .route_free = sof_ipc4_route_free, + .dai_config = sof_ipc4_dai_config, }; diff --git a/sound/soc/sof/ipc4-topology.h b/sound/soc/sof/ipc4-topology.h index eebf46b24430..0cadf04efa6a 100644 --- a/sound/soc/sof/ipc4-topology.h +++ b/sound/soc/sof/ipc4-topology.h @@ -26,7 +26,7 @@ #define SOF_IPC4_FW_MAX_PAGE_COUNT 20 #define SOF_IPC4_FW_MAX_QUEUE_COUNT 8 -/* Node index and mask applicable for host copier */ +/* Node index and mask applicable for host copier and ALH/HDA type DAI copiers */ #define SOF_IPC4_NODE_INDEX_MASK 0xFF #define SOF_IPC4_NODE_INDEX(x) ((x) & SOF_IPC4_NODE_INDEX_MASK) #define SOF_IPC4_NODE_TYPE(x) ((x) << 8) -- cgit v1.2.3 From d0c0d5bf944b13b4e293746eb655f1c2caf67231 Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Wed, 8 Jun 2022 20:26:35 -0700 Subject: ASoC: SOF: ipc4-pcm: Expose sof_ipc4_set_pipeline_state() MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Expose the sof_ipc4_set_pipeline_state() function as it will be used in the IPC4-specific BE DAI driver ops. Signed-off-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Reviewed-by: Péter Ujfalusi Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20220609032643.916882-16-ranjani.sridharan@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-pcm.c | 3 ++- sound/soc/sof/ipc4-priv.h | 2 ++ 2 files changed, 4 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sof/ipc4-pcm.c b/sound/soc/sof/ipc4-pcm.c index 7a56fba8f1d9..6a702f9dc065 100644 --- a/sound/soc/sof/ipc4-pcm.c +++ b/sound/soc/sof/ipc4-pcm.c @@ -13,7 +13,7 @@ #include "ipc4-priv.h" #include "ipc4-topology.h" -static int sof_ipc4_set_pipeline_state(struct snd_sof_dev *sdev, u32 id, u32 state) +int sof_ipc4_set_pipeline_state(struct snd_sof_dev *sdev, u32 id, u32 state) { struct sof_ipc4_msg msg = {{ 0 }}; u32 primary; @@ -30,6 +30,7 @@ static int sof_ipc4_set_pipeline_state(struct snd_sof_dev *sdev, u32 id, u32 sta return sof_ipc_tx_message(sdev->ipc, &msg, 0, NULL, 0); } +EXPORT_SYMBOL(sof_ipc4_set_pipeline_state); static int sof_ipc4_trigger_pipelines(struct snd_soc_component *component, struct snd_pcm_substream *substream, int state) diff --git a/sound/soc/sof/ipc4-priv.h b/sound/soc/sof/ipc4-priv.h index e4381a74516c..8dddceaf5eb3 100644 --- a/sound/soc/sof/ipc4-priv.h +++ b/sound/soc/sof/ipc4-priv.h @@ -44,4 +44,6 @@ extern const struct sof_ipc_tplg_ops ipc4_tplg_ops; extern const struct sof_ipc_tplg_control_ops tplg_ipc4_control_ops; extern const struct sof_ipc_pcm_ops ipc4_pcm_ops; +int sof_ipc4_set_pipeline_state(struct snd_sof_dev *sdev, u32 id, u32 state); + #endif -- cgit v1.2.3 From 4c30004a7c6920c66a08c1aa16481c28202eefd0 Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Wed, 8 Jun 2022 20:26:36 -0700 Subject: ASoC: SOF: IPC4: set the BE DAI ops MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add BE DAI drv ops for IPC4 for DMIC, SSP and HDA type DAI's. Signed-off-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Reviewed-by: Péter Ujfalusi Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20220609032643.916882-17-ranjani.sridharan@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-dai.c | 173 +++++++++++++++++++++++++++++++++++++++++- 1 file changed, 170 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index 9823230d2ef4..5423667002e5 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -10,6 +10,10 @@ #include #include +#include +#include +#include "../ipc4-priv.h" +#include "../ipc4-topology.h" #include "../sof-priv.h" #include "../sof-audio.h" #include "hda.h" @@ -369,8 +373,7 @@ static int hda_dai_config_pause_push_ipc(struct snd_soc_dapm_widget *w) return ret; } -static int ipc3_hda_dai_prepare(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) +static int hda_dai_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct hdac_ext_stream *hext_stream = snd_soc_dai_get_dma_data(dai, substream); @@ -438,6 +441,91 @@ static int ipc3_hda_dai_trigger(struct snd_pcm_substream *substream, return 0; } +/* + * In contrast to IPC3, the dai trigger in IPC4 mixes pipeline state changes + * (over IPC channel) and DMA state change (direct host register changes). + */ +static int ipc4_hda_dai_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct hdac_ext_stream *hext_stream = snd_soc_dai_get_dma_data(dai, substream); + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(dai->component); + struct snd_soc_pcm_runtime *rtd; + struct snd_sof_widget *swidget; + struct snd_soc_dapm_widget *w; + struct snd_soc_dai *codec_dai; + struct hdac_stream *hstream; + struct snd_soc_dai *cpu_dai; + int ret; + + dev_dbg(dai->dev, "%s: cmd=%d dai %s direction %d\n", __func__, cmd, + dai->name, substream->stream); + + hstream = substream->runtime->private_data; + rtd = asoc_substream_to_rtd(substream); + cpu_dai = asoc_rtd_to_cpu(rtd, 0); + codec_dai = asoc_rtd_to_codec(rtd, 0); + + w = snd_soc_dai_get_widget(dai, substream->stream); + swidget = w->dobj.private; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + snd_hdac_ext_link_stream_start(hext_stream); + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_STOP: + { + struct snd_sof_widget *pipe_widget = swidget->pipe_widget; + struct sof_ipc4_pipeline *pipeline = pipe_widget->private; + + ret = sof_ipc4_set_pipeline_state(sdev, swidget->pipeline_id, + SOF_IPC4_PIPE_PAUSED); + if (ret < 0) + return ret; + + pipeline->state = SOF_IPC4_PIPE_PAUSED; + + snd_hdac_ext_link_stream_clear(hext_stream); + + ret = sof_ipc4_set_pipeline_state(sdev, swidget->pipeline_id, + SOF_IPC4_PIPE_RESET); + if (ret < 0) + return ret; + + pipeline->state = SOF_IPC4_PIPE_RESET; + + ret = hda_link_dma_cleanup(substream, hstream, cpu_dai, codec_dai, false); + if (ret < 0) { + dev_err(sdev->dev, "%s: failed to clean up link DMA\n", __func__); + return ret; + } + break; + } + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + { + struct snd_sof_widget *pipe_widget = swidget->pipe_widget; + struct sof_ipc4_pipeline *pipeline = pipe_widget->private; + + ret = sof_ipc4_set_pipeline_state(sdev, swidget->pipeline_id, + SOF_IPC4_PIPE_PAUSED); + if (ret < 0) + return ret; + + pipeline->state = SOF_IPC4_PIPE_PAUSED; + + snd_hdac_ext_link_stream_clear(hext_stream); + break; + } + default: + dev_err(sdev->dev, "%s: unknown trigger command %d\n", __func__, cmd); + return -EINVAL; + } + + return 0; +} + static int hda_dai_hw_free(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -454,7 +542,7 @@ static const struct snd_soc_dai_ops ipc3_hda_dai_ops = { .hw_params = hda_dai_hw_params, .hw_free = hda_dai_hw_free, .trigger = ipc3_hda_dai_trigger, - .prepare = ipc3_hda_dai_prepare, + .prepare = hda_dai_prepare, }; static int hda_dai_suspend(struct hdac_bus *bus) @@ -497,6 +585,14 @@ static int hda_dai_suspend(struct hdac_bus *bus) return 0; } + +static const struct snd_soc_dai_ops ipc4_hda_dai_ops = { + .hw_params = hda_dai_hw_params, + .hw_free = hda_dai_hw_free, + .trigger = ipc4_hda_dai_trigger, + .prepare = hda_dai_prepare, +}; + #endif /* only one flag used so far to harden hw_params/hw_free/trigger/prepare */ @@ -608,6 +704,59 @@ static const struct snd_soc_dai_ops ipc3_ssp_dai_ops = { .shutdown = ssp_dai_shutdown, }; +static int ipc4_be_dai_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct snd_sof_widget *pipe_widget; + struct sof_ipc4_pipeline *pipeline; + struct snd_sof_widget *swidget; + struct snd_soc_dapm_widget *w; + struct snd_sof_dev *sdev; + int ret; + + w = snd_soc_dai_get_widget(dai, substream->stream); + swidget = w->dobj.private; + pipe_widget = swidget->pipe_widget; + pipeline = pipe_widget->private; + sdev = snd_soc_component_get_drvdata(swidget->scomp); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_STOP: + ret = sof_ipc4_set_pipeline_state(sdev, swidget->pipeline_id, + SOF_IPC4_PIPE_PAUSED); + if (ret < 0) + return ret; + pipeline->state = SOF_IPC4_PIPE_PAUSED; + + ret = sof_ipc4_set_pipeline_state(sdev, swidget->pipeline_id, + SOF_IPC4_PIPE_RESET); + if (ret < 0) + return ret; + pipeline->state = SOF_IPC4_PIPE_RESET; + break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ret = sof_ipc4_set_pipeline_state(sdev, swidget->pipeline_id, + SOF_IPC4_PIPE_PAUSED); + if (ret < 0) + return ret; + pipeline->state = SOF_IPC4_PIPE_PAUSED; + break; + default: + break; + } + + return 0; +} + +static const struct snd_soc_dai_ops ipc4_dmic_dai_ops = { + .trigger = ipc4_be_dai_trigger, +}; + +static const struct snd_soc_dai_ops ipc4_ssp_dai_ops = { + .trigger = ipc4_be_dai_trigger, +}; + void hda_set_dai_drv_ops(struct snd_sof_dev *sdev, struct snd_sof_dsp_ops *ops) { int i; @@ -624,6 +773,24 @@ void hda_set_dai_drv_ops(struct snd_sof_dev *sdev, struct snd_sof_dsp_ops *ops) strstr(ops->drv[i].name, "Analog") || strstr(ops->drv[i].name, "Digital")) ops->drv[i].ops = &ipc3_hda_dai_ops; +#endif + } + break; + case SOF_INTEL_IPC4: + for (i = 0; i < ops->num_drv; i++) { + if (strstr(ops->drv[i].name, "DMIC")) { + ops->drv[i].ops = &ipc4_dmic_dai_ops; + continue; + } + if (strstr(ops->drv[i].name, "SSP")) { + ops->drv[i].ops = &ipc4_ssp_dai_ops; + continue; + } +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) + if (strstr(ops->drv[i].name, "iDisp") || + strstr(ops->drv[i].name, "Analog") || + strstr(ops->drv[i].name, "Digital")) + ops->drv[i].ops = &ipc4_hda_dai_ops; #endif } break; -- cgit v1.2.3 From bc433fd76faefb8484f5bc653d846043822a2d35 Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Wed, 8 Jun 2022 20:26:37 -0700 Subject: ASoC: SOF: Add ops_free MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add the ops_free callback in struct sof_dev_desc. Signed-off-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Reviewed-by: Péter Ujfalusi Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20220609032643.916882-18-ranjani.sridharan@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/core.c | 7 ++++++- sound/soc/sof/ops.h | 6 ++++++ 2 files changed, 12 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index 53719c04658f..c99b5e6c026c 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -189,7 +189,7 @@ static int sof_probe_continue(struct snd_sof_dev *sdev) ret = snd_sof_probe(sdev); if (ret < 0) { dev_err(sdev->dev, "error: failed to probe DSP %d\n", ret); - return ret; + goto probe_err; } sof_set_fw_state(sdev, SOF_FW_BOOT_PREPARE); @@ -317,6 +317,8 @@ dbg_err: snd_sof_free_debug(sdev); dsp_err: snd_sof_remove(sdev); +probe_err: + sof_ops_free(sdev); /* all resources freed, update state to match */ sof_set_fw_state(sdev, SOF_FW_BOOT_NOT_STARTED); @@ -374,6 +376,7 @@ int snd_sof_device_probe(struct device *dev, struct snd_sof_pdata *plat_data) !sof_ops(sdev)->block_read || !sof_ops(sdev)->block_write || !sof_ops(sdev)->send_msg || !sof_ops(sdev)->load_firmware || !sof_ops(sdev)->ipc_msg_data) { + sof_ops_free(sdev); dev_err(dev, "error: missing mandatory ops\n"); return -EINVAL; } @@ -457,6 +460,8 @@ int snd_sof_device_remove(struct device *dev) snd_sof_remove(sdev); } + sof_ops_free(sdev); + /* release firmware */ snd_sof_fw_unload(sdev); diff --git a/sound/soc/sof/ops.h b/sound/soc/sof/ops.h index b79ae4f66eba..55d43adb6a29 100644 --- a/sound/soc/sof/ops.h +++ b/sound/soc/sof/ops.h @@ -29,6 +29,12 @@ static inline int sof_ops_init(struct snd_sof_dev *sdev) return 0; } +static inline void sof_ops_free(struct snd_sof_dev *sdev) +{ + if (sdev->pdata->desc->ops_free) + sdev->pdata->desc->ops_free(sdev); +} + /* Mandatory operations are verified during probing */ /* init */ -- cgit v1.2.3 From 1da51943725f29000ae4d2be3b3b4bf8309d99a2 Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Wed, 8 Jun 2022 20:26:38 -0700 Subject: ASoC: SOF: Intel: hda: init NHLT for IPC4 MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Init and save the BIOS NHLT as part of the IPC4 FW data. Add a kernel module param to override the BIOS NHLT with the NHLT from the topology. Also, add the ops_free callback for all HDA platforms to free the NHLT. Co-developed-by: Jaska Uimonen Signed-off-by: Jaska Uimonen Signed-off-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Reviewed-by: Péter Ujfalusi Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20220609032643.916882-19-ranjani.sridharan@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-dai.c | 28 ++++++++++++++++++++++++++++ sound/soc/sof/intel/hda.h | 1 + sound/soc/sof/intel/pci-apl.c | 1 + sound/soc/sof/intel/pci-cnl.c | 1 + sound/soc/sof/intel/pci-icl.c | 1 + sound/soc/sof/intel/pci-tgl.c | 1 + sound/soc/sof/ipc4-priv.h | 2 ++ 7 files changed, 35 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index 5423667002e5..228079a52c3d 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -10,6 +10,7 @@ #include #include +#include #include #include #include "../ipc4-priv.h" @@ -18,6 +19,14 @@ #include "../sof-audio.h" #include "hda.h" +/* + * The default method is to fetch NHLT from BIOS. With this parameter set + * it is possible to override that with NHLT in the SOF topology manifest. + */ +static bool hda_use_tplg_nhlt; +module_param_named(sof_use_tplg_nhlt, hda_use_tplg_nhlt, bool, 0444); +MODULE_PARM_DESC(sof_use_tplg_nhlt, "SOF topology nhlt override"); + #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) struct hda_pipe_params { @@ -777,6 +786,9 @@ void hda_set_dai_drv_ops(struct snd_sof_dev *sdev, struct snd_sof_dsp_ops *ops) } break; case SOF_INTEL_IPC4: + { + struct sof_ipc4_fw_data *ipc4_data = sdev->private; + for (i = 0; i < ops->num_drv; i++) { if (strstr(ops->drv[i].name, "DMIC")) { ops->drv[i].ops = &ipc4_dmic_dai_ops; @@ -793,12 +805,28 @@ void hda_set_dai_drv_ops(struct snd_sof_dev *sdev, struct snd_sof_dsp_ops *ops) ops->drv[i].ops = &ipc4_hda_dai_ops; #endif } + + if (!hda_use_tplg_nhlt) + ipc4_data->nhlt = intel_nhlt_init(sdev->dev); + break; + } default: break; } } +void hda_ops_free(struct snd_sof_dev *sdev) +{ + if (sdev->pdata->ipc_type == SOF_INTEL_IPC4) { + struct sof_ipc4_fw_data *ipc4_data = sdev->private; + + if (!hda_use_tplg_nhlt) + intel_nhlt_free(ipc4_data->nhlt); + } +} +EXPORT_SYMBOL_NS(hda_ops_free, SND_SOC_SOF_INTEL_HDA_COMMON); + /* * common dai driver for skl+ platforms. * some products who use this DAI array only physically have a subset of diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 3e0f7b0c586a..59181468e05e 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -763,6 +763,7 @@ int hda_ctrl_dai_widget_free(struct snd_soc_dapm_widget *w, unsigned int quirk_f extern int sof_hda_position_quirk; void hda_set_dai_drv_ops(struct snd_sof_dev *sdev, struct snd_sof_dsp_ops *ops); +void hda_ops_free(struct snd_sof_dev *sdev); /* IPC4 */ irqreturn_t cnl_ipc4_irq_thread(int irq, void *context); diff --git a/sound/soc/sof/intel/pci-apl.c b/sound/soc/sof/intel/pci-apl.c index 2de3658eb817..998e219011f0 100644 --- a/sound/soc/sof/intel/pci-apl.c +++ b/sound/soc/sof/intel/pci-apl.c @@ -44,6 +44,7 @@ static const struct sof_dev_desc bxt_desc = { .nocodec_tplg_filename = "sof-apl-nocodec.tplg", .ops = &sof_apl_ops, .ops_init = sof_apl_ops_init, + .ops_free = hda_ops_free, }; static const struct sof_dev_desc glk_desc = { diff --git a/sound/soc/sof/intel/pci-cnl.c b/sound/soc/sof/intel/pci-cnl.c index 87e587aef9c9..c797356f7028 100644 --- a/sound/soc/sof/intel/pci-cnl.c +++ b/sound/soc/sof/intel/pci-cnl.c @@ -73,6 +73,7 @@ static const struct sof_dev_desc cfl_desc = { .nocodec_tplg_filename = "sof-cnl-nocodec.tplg", .ops = &sof_cnl_ops, .ops_init = sof_cnl_ops_init, + .ops_free = hda_ops_free, }; static const struct sof_dev_desc cml_desc = { diff --git a/sound/soc/sof/intel/pci-icl.c b/sound/soc/sof/intel/pci-icl.c index 1c7f16ce531e..48f24f8ace26 100644 --- a/sound/soc/sof/intel/pci-icl.c +++ b/sound/soc/sof/intel/pci-icl.c @@ -45,6 +45,7 @@ static const struct sof_dev_desc icl_desc = { .nocodec_tplg_filename = "sof-icl-nocodec.tplg", .ops = &sof_icl_ops, .ops_init = sof_icl_ops_init, + .ops_free = hda_ops_free, }; static const struct sof_dev_desc jsl_desc = { diff --git a/sound/soc/sof/intel/pci-tgl.c b/sound/soc/sof/intel/pci-tgl.c index 58a9bd92a237..ccc44ba3ad94 100644 --- a/sound/soc/sof/intel/pci-tgl.c +++ b/sound/soc/sof/intel/pci-tgl.c @@ -73,6 +73,7 @@ static const struct sof_dev_desc tglh_desc = { .nocodec_tplg_filename = "sof-tgl-nocodec.tplg", .ops = &sof_tgl_ops, .ops_init = sof_tgl_ops_init, + .ops_free = hda_ops_free, }; static const struct sof_dev_desc ehl_desc = { diff --git a/sound/soc/sof/ipc4-priv.h b/sound/soc/sof/ipc4-priv.h index 8dddceaf5eb3..9492fe1796c2 100644 --- a/sound/soc/sof/ipc4-priv.h +++ b/sound/soc/sof/ipc4-priv.h @@ -18,11 +18,13 @@ * @manifest_fw_hdr_offset: FW header offset in the manifest * @num_fw_modules : Number of modules in base FW * @fw_modules: Array of base FW modules + * @nhlt: NHLT table either from the BIOS or the topology manifest */ struct sof_ipc4_fw_data { u32 manifest_fw_hdr_offset; int num_fw_modules; void *fw_modules; + void *nhlt; }; /** -- cgit v1.2.3 From 323aa1f093e6113f78a8ae808c6c097663d8cb4c Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Wed, 8 Jun 2022 20:26:40 -0700 Subject: ASoC: SOF: Add a new IPC op for parsing topology manifest MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add a new topology IPC op, parse_manifest. Define and set the op for IPC4 and IPC4. Co-developed-by: Jaska Uimonen Signed-off-by: Jaska Uimonen Signed-off-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Reviewed-by: Péter Ujfalusi Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20220609032643.916882-21-ranjani.sridharan@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc3-topology.c | 48 +++++++++++++++++++++++++++++++++ sound/soc/sof/ipc4-topology.c | 63 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/sof/sof-audio.h | 3 +++ sound/soc/sof/topology.c | 45 +++---------------------------- 4 files changed, 118 insertions(+), 41 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/ipc3-topology.c b/sound/soc/sof/ipc3-topology.c index 043554d7cb4a..a91d7df3f07e 100644 --- a/sound/soc/sof/ipc3-topology.c +++ b/sound/soc/sof/ipc3-topology.c @@ -17,6 +17,9 @@ /* Full volume for default values */ #define VOL_ZERO_DB BIT(VOLUME_FWL) +/* size of tplg ABI in bytes */ +#define SOF_IPC3_TPLG_ABI_SIZE 3 + struct sof_widget_data { int ctrl_type; int ipc_cmd; @@ -2303,6 +2306,50 @@ static int sof_ipc3_dai_get_clk(struct snd_sof_dev *sdev, struct snd_sof_dai *da return -EINVAL; } +static int sof_ipc3_parse_manifest(struct snd_soc_component *scomp, int index, + struct snd_soc_tplg_manifest *man) +{ + u32 size = le32_to_cpu(man->priv.size); + u32 abi_version; + + /* backward compatible with tplg without ABI info */ + if (!size) { + dev_dbg(scomp->dev, "No topology ABI info\n"); + return 0; + } + + if (size != SOF_IPC3_TPLG_ABI_SIZE) { + dev_err(scomp->dev, "%s: Invalid topology ABI size: %u\n", + __func__, size); + return -EINVAL; + } + + dev_info(scomp->dev, + "Topology: ABI %d:%d:%d Kernel ABI %hhu:%hhu:%hhu\n", + man->priv.data[0], man->priv.data[1], man->priv.data[2], + SOF_ABI_MAJOR, SOF_ABI_MINOR, SOF_ABI_PATCH); + + abi_version = SOF_ABI_VER(man->priv.data[0], man->priv.data[1], man->priv.data[2]); + + if (SOF_ABI_VERSION_INCOMPATIBLE(SOF_ABI_VERSION, abi_version)) { + dev_err(scomp->dev, "%s: Incompatible topology ABI version\n", __func__); + return -EINVAL; + } + + if (SOF_ABI_VERSION_MINOR(abi_version) > SOF_ABI_MINOR) { + if (!IS_ENABLED(CONFIG_SND_SOC_SOF_STRICT_ABI_CHECKS)) { + dev_warn(scomp->dev, "%s: Topology ABI is more recent than kernel\n", + __func__); + } else { + dev_err(scomp->dev, "%s: Topology ABI is more recent than kernel\n", + __func__); + return -EINVAL; + } + } + + return 0; +} + /* token list for each topology object */ static enum sof_tokens host_token_list[] = { SOF_CORE_TOKENS, @@ -2413,4 +2460,5 @@ const struct sof_ipc_tplg_ops ipc3_tplg_ops = { .dai_get_clk = sof_ipc3_dai_get_clk, .set_up_all_pipelines = sof_ipc3_set_up_all_pipelines, .tear_down_all_pipelines = sof_ipc3_tear_down_all_pipelines, + .parse_manifest = sof_ipc3_parse_manifest, }; diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index 9615034f8c70..27ad48990383 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -16,6 +16,7 @@ #include "ops.h" #define SOF_IPC4_GAIN_PARAM_ID 0 +#define SOF_IPC4_TPLG_ABI_SIZE 6 static const struct sof_topology_token ipc4_sched_tokens[] = { {SOF_TKN_SCHED_LP_MODE, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, @@ -1317,6 +1318,67 @@ static int sof_ipc4_dai_config(struct snd_sof_dev *sdev, struct snd_sof_widget * return 0; } +static int sof_ipc4_parse_manifest(struct snd_soc_component *scomp, int index, + struct snd_soc_tplg_manifest *man) +{ + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); + struct sof_ipc4_fw_data *ipc4_data = sdev->private; + struct sof_manifest_tlv *manifest_tlv; + struct sof_manifest *manifest; + u32 size = le32_to_cpu(man->priv.size); + u8 *man_ptr = man->priv.data; + u32 len_check; + int i; + + if (!size || size < SOF_IPC4_TPLG_ABI_SIZE) { + dev_err(scomp->dev, "%s: Invalid topology ABI size: %u\n", + __func__, size); + return -EINVAL; + } + + manifest = (struct sof_manifest *)man_ptr; + + dev_info(scomp->dev, + "Topology: ABI %d:%d:%d Kernel ABI %u:%u:%u\n", + le16_to_cpu(manifest->abi_major), le16_to_cpu(manifest->abi_minor), + le16_to_cpu(manifest->abi_patch), + SOF_ABI_MAJOR, SOF_ABI_MINOR, SOF_ABI_PATCH); + + /* TODO: Add ABI compatibility check */ + + /* no more data after the ABI version */ + if (size <= SOF_IPC4_TPLG_ABI_SIZE) + return 0; + + manifest_tlv = manifest->items; + len_check = sizeof(struct sof_manifest); + for (i = 0; i < le16_to_cpu(manifest->count); i++) { + len_check += sizeof(struct sof_manifest_tlv) + le32_to_cpu(manifest_tlv->size); + if (len_check > size) + return -EINVAL; + + switch (le32_to_cpu(manifest_tlv->type)) { + case SOF_MANIFEST_DATA_TYPE_NHLT: + /* no NHLT in BIOS, so use the one from topology manifest */ + if (ipc4_data->nhlt) + break; + ipc4_data->nhlt = devm_kmemdup(sdev->dev, manifest_tlv->data, + le32_to_cpu(manifest_tlv->size), GFP_KERNEL); + if (!ipc4_data->nhlt) + return -ENOMEM; + break; + default: + dev_warn(scomp->dev, "Skipping unknown manifest data type %d\n", + manifest_tlv->type); + break; + } + man_ptr += sizeof(struct sof_manifest_tlv) + le32_to_cpu(manifest_tlv->size); + manifest_tlv = (struct sof_manifest_tlv *)man_ptr; + } + + return 0; +} + static enum sof_tokens host_token_list[] = { SOF_COMP_TOKENS, SOF_AUDIO_FMT_NUM_TOKENS, @@ -1402,4 +1464,5 @@ const struct sof_ipc_tplg_ops ipc4_tplg_ops = { .route_setup = sof_ipc4_route_setup, .route_free = sof_ipc4_route_free, .dai_config = sof_ipc4_dai_config, + .parse_manifest = sof_ipc4_parse_manifest, }; diff --git a/sound/soc/sof/sof-audio.h b/sound/soc/sof/sof-audio.h index d896da1192c5..79486266081f 100644 --- a/sound/soc/sof/sof-audio.h +++ b/sound/soc/sof/sof-audio.h @@ -168,6 +168,7 @@ struct sof_ipc_tplg_widget_ops { * @dai_get_clk: Function pointer for getting the DAI clock setting * @set_up_all_pipelines: Function pointer for setting up all topology pipelines * @tear_down_all_pipelines: Function pointer for tearing down all topology pipelines + * @parse_manifest: Optional function pointer for ipc4 specific parsing of topology manifest */ struct sof_ipc_tplg_ops { const struct sof_ipc_tplg_widget_ops *widget; @@ -185,6 +186,8 @@ struct sof_ipc_tplg_ops { int (*dai_get_clk)(struct snd_sof_dev *sdev, struct snd_sof_dai *dai, int clk_type); int (*set_up_all_pipelines)(struct snd_sof_dev *sdev, bool verify); int (*tear_down_all_pipelines)(struct snd_sof_dev *sdev, bool verify); + int (*parse_manifest)(struct snd_soc_component *scomp, int index, + struct snd_soc_tplg_manifest *man); }; /** struct snd_sof_tuple - Tuple info diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 606dbca94246..1893c590f2f0 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -36,9 +36,6 @@ #define TLV_STEP 1 #define TLV_MUTE 2 -/* size of tplg abi in byte */ -#define SOF_TPLG_ABI_SIZE 3 - /** * sof_update_ipc_object - Parse multiple sets of tokens within the token array associated with the * token ID. @@ -2020,45 +2017,11 @@ static int sof_complete(struct snd_soc_component *scomp) static int sof_manifest(struct snd_soc_component *scomp, int index, struct snd_soc_tplg_manifest *man) { - u32 size; - u32 abi_version; - - size = le32_to_cpu(man->priv.size); - - /* backward compatible with tplg without ABI info */ - if (!size) { - dev_dbg(scomp->dev, "No topology ABI info\n"); - return 0; - } - - if (size != SOF_TPLG_ABI_SIZE) { - dev_err(scomp->dev, "error: invalid topology ABI size\n"); - return -EINVAL; - } - - dev_info(scomp->dev, - "Topology: ABI %d:%d:%d Kernel ABI %d:%d:%d\n", - man->priv.data[0], man->priv.data[1], - man->priv.data[2], SOF_ABI_MAJOR, SOF_ABI_MINOR, - SOF_ABI_PATCH); - - abi_version = SOF_ABI_VER(man->priv.data[0], - man->priv.data[1], - man->priv.data[2]); - - if (SOF_ABI_VERSION_INCOMPATIBLE(SOF_ABI_VERSION, abi_version)) { - dev_err(scomp->dev, "error: incompatible topology ABI version\n"); - return -EINVAL; - } + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); + const struct sof_ipc_tplg_ops *ipc_tplg_ops = sdev->ipc->ops->tplg; - if (SOF_ABI_VERSION_MINOR(abi_version) > SOF_ABI_MINOR) { - if (!IS_ENABLED(CONFIG_SND_SOC_SOF_STRICT_ABI_CHECKS)) { - dev_warn(scomp->dev, "warn: topology ABI is more recent than kernel\n"); - } else { - dev_err(scomp->dev, "error: topology ABI is more recent than kernel\n"); - return -EINVAL; - } - } + if (ipc_tplg_ops->parse_manifest) + return ipc_tplg_ops->parse_manifest(scomp, index, man); return 0; } -- cgit v1.2.3 From aa84ffb721587d134702a1932f2c8793e8709df4 Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Wed, 8 Jun 2022 20:26:41 -0700 Subject: ASoC: SOF: ipc4-topology: Add support for SSP/DMIC DAI's MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The copier config for SSP and DMIC type DAI copiers needs to be parsed and matched with the runtime hw_config from the NHLT table. Along with this, also add the change to set the node_id for these copier types. Co-developed-by: Jaska Uimonen Signed-off-by: Jaska Uimonen Reviewed-by: Pierre-Louis Bossart Reviewed-by: Péter Ujfalusi Reviewed-by: Bard Liao Signed-off-by: Ranjani Sridharan Link: https://lore.kernel.org/r/20220609032643.916882-22-ranjani.sridharan@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-topology.c | 148 +++++++++++++++++++++++++++++++++++++++--- sound/soc/sof/ipc4-topology.h | 6 ++ 2 files changed, 146 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index 27ad48990383..9f055c187b72 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -9,6 +9,7 @@ #include #include #include +#include #include "sof-priv.h" #include "sof-audio.h" #include "ipc4-priv.h" @@ -473,14 +474,30 @@ static int sof_ipc4_widget_setup_comp_dai(struct snd_sof_widget *swidget) node_type, ipc4_copier->dai_type, ipc4_copier->dai_index); ipc4_copier->data.gtw_cfg.node_id = SOF_IPC4_NODE_TYPE(node_type); - ipc4_copier->gtw_attr = kzalloc(sizeof(*ipc4_copier->gtw_attr), GFP_KERNEL); - if (!ipc4_copier->gtw_attr) { - ret = -ENOMEM; - goto err; - } - ipc4_copier->copier_config = (uint32_t *)ipc4_copier->gtw_attr; - ipc4_copier->data.gtw_cfg.config_length = sizeof(struct sof_ipc4_gtw_attributes) >> 2; + switch (ipc4_copier->dai_type) { + case SOF_DAI_INTEL_SSP: + /* set SSP DAI index as the node_id */ + ipc4_copier->data.gtw_cfg.node_id |= + SOF_IPC4_NODE_INDEX_INTEL_SSP(ipc4_copier->dai_index); + break; + case SOF_DAI_INTEL_DMIC: + /* set DMIC DAI index as the node_id */ + ipc4_copier->data.gtw_cfg.node_id |= + SOF_IPC4_NODE_INDEX_INTEL_DMIC(ipc4_copier->dai_index); + break; + default: + ipc4_copier->gtw_attr = kzalloc(sizeof(*ipc4_copier->gtw_attr), GFP_KERNEL); + if (!ipc4_copier->gtw_attr) { + ret = -ENOMEM; + goto err; + } + + ipc4_copier->copier_config = (uint32_t *)ipc4_copier->gtw_attr; + ipc4_copier->data.gtw_cfg.config_length = + sizeof(struct sof_ipc4_gtw_attributes) >> 2; + break; + } dai->scomp = scomp; dai->private = ipc4_copier; @@ -516,7 +533,9 @@ static void sof_ipc4_widget_free_comp_dai(struct snd_sof_widget *swidget) kfree(available_fmt->dma_buffer_size); kfree(available_fmt->base_config); kfree(available_fmt->out_audio_fmt); - kfree(ipc4_copier->copier_config); + if (ipc4_copier->dai_type != SOF_DAI_INTEL_SSP && + ipc4_copier->dai_type != SOF_DAI_INTEL_DMIC) + kfree(ipc4_copier->copier_config); kfree(dai->private); kfree(dai); swidget->private = NULL; @@ -822,6 +841,112 @@ static void sof_ipc4_unprepare_copier_module(struct snd_sof_widget *swidget) ida_free(&fw_module->m_ida, swidget->instance_id); } +#if IS_ENABLED(CONFIG_ACPI) && IS_ENABLED(CONFIG_SND_INTEL_NHLT) +static int snd_sof_get_hw_config_params(struct snd_sof_dev *sdev, struct snd_sof_dai *dai, + int *sample_rate, int *channel_count, int *bit_depth) +{ + struct snd_soc_tplg_hw_config *hw_config; + struct snd_sof_dai_link *slink; + bool dai_link_found = false; + bool hw_cfg_found = false; + int i; + + /* get current hw_config from link */ + list_for_each_entry(slink, &sdev->dai_link_list, list) { + if (!strcmp(slink->link->name, dai->name)) { + dai_link_found = true; + break; + } + } + + if (!dai_link_found) { + dev_err(sdev->dev, "%s: no DAI link found for DAI %s\n", __func__, dai->name); + return -EINVAL; + } + + for (i = 0; i < slink->num_hw_configs; i++) { + hw_config = &slink->hw_configs[i]; + if (dai->current_config == le32_to_cpu(hw_config->id)) { + hw_cfg_found = true; + break; + } + } + + if (!hw_cfg_found) { + dev_err(sdev->dev, "%s: no matching hw_config found for DAI %s\n", __func__, + dai->name); + return -EINVAL; + } + + *bit_depth = le32_to_cpu(hw_config->tdm_slot_width); + *channel_count = le32_to_cpu(hw_config->tdm_slots); + *sample_rate = le32_to_cpu(hw_config->fsync_rate); + + dev_dbg(sdev->dev, "%s: sample rate: %d sample width: %d channels: %d\n", + __func__, *sample_rate, *bit_depth, *channel_count); + + return 0; +} + +static int snd_sof_get_nhlt_endpoint_data(struct snd_sof_dev *sdev, struct snd_sof_dai *dai, + struct snd_pcm_hw_params *params, u32 dai_index, + u32 linktype, u8 dir, u32 **dst, u32 *len) +{ + struct sof_ipc4_fw_data *ipc4_data = sdev->private; + struct nhlt_specific_cfg *cfg; + int sample_rate, channel_count; + int bit_depth, ret; + u32 nhlt_type; + + /* convert to NHLT type */ + switch (linktype) { + case SOF_DAI_INTEL_DMIC: + nhlt_type = NHLT_LINK_DMIC; + bit_depth = params_width(params); + channel_count = params_channels(params); + sample_rate = params_rate(params); + break; + case SOF_DAI_INTEL_SSP: + nhlt_type = NHLT_LINK_SSP; + ret = snd_sof_get_hw_config_params(sdev, dai, &sample_rate, &channel_count, + &bit_depth); + if (ret < 0) + return ret; + break; + default: + return 0; + } + + dev_dbg(sdev->dev, "%s: dai index %d nhlt type %d direction %d\n", + __func__, dai_index, nhlt_type, dir); + + /* find NHLT blob with matching params */ + cfg = intel_nhlt_get_endpoint_blob(sdev->dev, ipc4_data->nhlt, dai_index, nhlt_type, + bit_depth, bit_depth, channel_count, sample_rate, + dir, 0); + + if (!cfg) { + dev_err(sdev->dev, + "no matching blob for sample rate: %d sample width: %d channels: %d\n", + sample_rate, bit_depth, channel_count); + return -EINVAL; + } + + /* config length should be in dwords */ + *len = cfg->size >> 2; + *dst = (u32 *)cfg->caps; + + return 0; +} +#else +static int snd_sof_get_nhlt_endpoint_data(struct snd_sof_dev *sdev, struct snd_sof_dai *dai, + struct snd_pcm_hw_params *params, u32 dai_index, + u32 linktype, u8 dir, u32 **dst, u32 *len) +{ + return 0; +} +#endif + static int sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, struct snd_pcm_hw_params *fe_params, @@ -906,6 +1031,13 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, ref_params = pipeline_params; + ret = snd_sof_get_nhlt_endpoint_data(sdev, dai, fe_params, ipc4_copier->dai_index, + ipc4_copier->dai_type, dir, + &ipc4_copier->copier_config, + &copier_data->gtw_cfg.config_length); + if (ret < 0) + return ret; + break; } default: diff --git a/sound/soc/sof/ipc4-topology.h b/sound/soc/sof/ipc4-topology.h index 0cadf04efa6a..64d836f05bad 100644 --- a/sound/soc/sof/ipc4-topology.h +++ b/sound/soc/sof/ipc4-topology.h @@ -31,6 +31,12 @@ #define SOF_IPC4_NODE_INDEX(x) ((x) & SOF_IPC4_NODE_INDEX_MASK) #define SOF_IPC4_NODE_TYPE(x) ((x) << 8) +/* Node ID for SSP type DAI copiers */ +#define SOF_IPC4_NODE_INDEX_INTEL_SSP(x) (((x) & 0xf) << 4) + +/* Node ID for DMIC type DAI copiers */ +#define SOF_IPC4_NODE_INDEX_INTEL_DMIC(x) (((x) & 0x7) << 5) + #define SOF_IPC4_GAIN_ALL_CHANNELS_MASK 0xffffffff #define SOF_IPC4_VOL_ZERO_DB 0x7fffffff -- cgit v1.2.3 From 9e2b5d33fec938ea2518735f2b66313cab89bb61 Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Wed, 8 Jun 2022 20:26:42 -0700 Subject: AsoC: SOF: ipc4-topology: Add dai_get_clk op MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Define and set the dai_get_clk_op for IPC4. Signed-off-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Reviewed-by: Péter Ujfalusi Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20220609032643.916882-23-ranjani.sridharan@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-topology.c | 58 +++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 58 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index 9f055c187b72..d5cb08ec1af1 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -1511,6 +1511,63 @@ static int sof_ipc4_parse_manifest(struct snd_soc_component *scomp, int index, return 0; } +static int sof_ipc4_dai_get_clk(struct snd_sof_dev *sdev, struct snd_sof_dai *dai, int clk_type) +{ + struct sof_ipc4_copier *ipc4_copier = dai->private; + struct snd_soc_tplg_hw_config *hw_config; + struct snd_sof_dai_link *slink; + bool dai_link_found = false; + bool hw_cfg_found = false; + int i; + + if (!ipc4_copier) + return 0; + + list_for_each_entry(slink, &sdev->dai_link_list, list) { + if (!strcmp(slink->link->name, dai->name)) { + dai_link_found = true; + break; + } + } + + if (!dai_link_found) { + dev_err(sdev->dev, "no DAI link found for DAI %s\n", dai->name); + return -EINVAL; + } + + for (i = 0; i < slink->num_hw_configs; i++) { + hw_config = &slink->hw_configs[i]; + if (dai->current_config == le32_to_cpu(hw_config->id)) { + hw_cfg_found = true; + break; + } + } + + if (!hw_cfg_found) { + dev_err(sdev->dev, "no matching hw_config found for DAI %s\n", dai->name); + return -EINVAL; + } + + switch (ipc4_copier->dai_type) { + case SOF_DAI_INTEL_SSP: + switch (clk_type) { + case SOF_DAI_CLK_INTEL_SSP_MCLK: + return le32_to_cpu(hw_config->mclk_rate); + case SOF_DAI_CLK_INTEL_SSP_BCLK: + return le32_to_cpu(hw_config->bclk_rate); + default: + dev_err(sdev->dev, "Invalid clk type for SSP %d\n", clk_type); + break; + } + break; + default: + dev_err(sdev->dev, "DAI type %d not supported yet!\n", ipc4_copier->dai_type); + break; + } + + return -EINVAL; +} + static enum sof_tokens host_token_list[] = { SOF_COMP_TOKENS, SOF_AUDIO_FMT_NUM_TOKENS, @@ -1597,4 +1654,5 @@ const struct sof_ipc_tplg_ops ipc4_tplg_ops = { .route_free = sof_ipc4_route_free, .dai_config = sof_ipc4_dai_config, .parse_manifest = sof_ipc4_parse_manifest, + .dai_get_clk = sof_ipc4_dai_get_clk, }; -- cgit v1.2.3 From a45a4d4390b7a562f8edc3518ba6cd2ad17be5bc Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Wed, 8 Jun 2022 20:26:43 -0700 Subject: ASoC: SOF: IPC4: add sdw blob Add IPC4 SoundWire blob. It includes a common IPC4 gateway and a multiple ALH configuration struct which is used for storing the aggregated SoundWire stream information. Signed-off-by: Bard Liao Reviewed-by: Rander Wang Reviewed-by: Pierre-Louis Bossart Signed-off-by: Ranjani Sridharan Link: https://lore.kernel.org/r/20220609032643.916882-24-ranjani.sridharan@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-topology.c | 44 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/sof/ipc4-topology.h | 25 ++++++++++++++++++++++++ 2 files changed, 69 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index d5cb08ec1af1..cb0f0823b8eb 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -476,6 +476,20 @@ static int sof_ipc4_widget_setup_comp_dai(struct snd_sof_widget *swidget) ipc4_copier->data.gtw_cfg.node_id = SOF_IPC4_NODE_TYPE(node_type); switch (ipc4_copier->dai_type) { + case SOF_DAI_INTEL_ALH: + { + struct sof_ipc4_alh_configuration_blob *blob; + + blob = kzalloc(sizeof(*blob), GFP_KERNEL); + if (!blob) { + ret = -ENOMEM; + goto err; + } + + ipc4_copier->copier_config = (uint32_t *)blob; + ipc4_copier->data.gtw_cfg.config_length = sizeof(*blob) >> 2; + break; + } case SOF_DAI_INTEL_SSP: /* set SSP DAI index as the node_id */ ipc4_copier->data.gtw_cfg.node_id |= @@ -1053,6 +1067,36 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, if (ret < 0) return ret; + switch (swidget->id) { + case snd_soc_dapm_dai_in: + case snd_soc_dapm_dai_out: + { + /* + * Only SOF_DAI_INTEL_ALH needs copier_data to set blob. + * That's why only ALH dai's blob is set after sof_ipc4_init_audio_fmt + */ + if (ipc4_copier->dai_type == SOF_DAI_INTEL_ALH) { + struct sof_ipc4_alh_configuration_blob *blob; + u32 ch_map; + int i; + + blob = (struct sof_ipc4_alh_configuration_blob *)ipc4_copier->copier_config; + /* TODO: add aggregation mode support */ + blob->alh_cfg.count = 1; + blob->alh_cfg.mapping[0].alh_id = copier_data->gtw_cfg.node_id; + blob->gw_attr.lp_buffer_alloc = 0; + + /* Get channel_mask from ch_map */ + ch_map = copier_data->base_config.audio_fmt.ch_map; + for (i = 0; ch_map; i++) { + if ((ch_map & 0xf) != 0xf) + blob->alh_cfg.mapping[0].channel_mask |= BIT(i); + ch_map >>= 4; + } + } + } + } + /* modify the input params for the next widget */ fmt = hw_param_mask(pipeline_params, SNDRV_PCM_HW_PARAM_FORMAT); out_sample_valid_bits = diff --git a/sound/soc/sof/ipc4-topology.h b/sound/soc/sof/ipc4-topology.h index 64d836f05bad..1a9c0627bae9 100644 --- a/sound/soc/sof/ipc4-topology.h +++ b/sound/soc/sof/ipc4-topology.h @@ -40,6 +40,8 @@ #define SOF_IPC4_GAIN_ALL_CHANNELS_MASK 0xffffffff #define SOF_IPC4_VOL_ZERO_DB 0x7fffffff +#define ALH_MAX_NUMBER_OF_GTW 16 + /** * struct sof_ipc4_pipeline - pipeline config data * @priority: Priority of this pipeline @@ -112,6 +114,29 @@ struct sof_ipc4_gtw_attributes { uint32_t rsvd : 30; }; +/** struct sof_ipc4_alh_multi_gtw_cfg: ALH gateway cfg data + * @count: Number of streams (valid items in mapping array) + * @alh_id: ALH stream id of a single ALH stream aggregated + * @channel_mask: Channel mask + * @mapping: ALH streams + */ +struct sof_ipc4_alh_multi_gtw_cfg { + uint32_t count; + struct { + uint32_t alh_id; + uint32_t channel_mask; + } mapping[ALH_MAX_NUMBER_OF_GTW]; +} __packed; + +/** struct sof_ipc4_alh_configuration_blob: ALH blob + * @gw_attr: Gateway attributes + * @alh_cfg: ALH configuration data + */ +struct sof_ipc4_alh_configuration_blob { + struct sof_ipc4_gtw_attributes gw_attr; + struct sof_ipc4_alh_multi_gtw_cfg alh_cfg; +}; + /** * struct sof_ipc4_copier - copier config data * @data: IPC copier data -- cgit v1.2.3 From 5babb012c847beb6c8c7108fd78f650b7a2c6054 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Thu, 9 Jun 2022 12:19:00 +0100 Subject: ASoC: codecs: msm8916-wcd-digital: move gains from SX_TLV to S8_TLV move all the digital gains form using SX_TLV to S8_TLV, these gains are actually 8 bit gains with 7th signed bit and ranges from -84dB to +40dB rest of the Qualcomm wcd codecs uses these properly. Fixes: ef8a4757a6db ("ASoC: msm8916-wcd-digital: Add sidetone support") Fixes: 150db8c5afa1 ("ASoC: codecs: Add msm8916-wcd digital codec") Signed-off-by: Srinivas Kandagatla Link: https://lore.kernel.org/r/20220609111901.318047-2-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/msm8916-wcd-digital.c | 46 +++++++++++++++++----------------- 1 file changed, 23 insertions(+), 23 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/msm8916-wcd-digital.c b/sound/soc/codecs/msm8916-wcd-digital.c index 20a07c92b2fc..098a58990f07 100644 --- a/sound/soc/codecs/msm8916-wcd-digital.c +++ b/sound/soc/codecs/msm8916-wcd-digital.c @@ -328,8 +328,8 @@ static const struct snd_kcontrol_new rx1_mix2_inp1_mux = SOC_DAPM_ENUM( static const struct snd_kcontrol_new rx2_mix2_inp1_mux = SOC_DAPM_ENUM( "RX2 MIX2 INP1 Mux", rx2_mix2_inp1_chain_enum); -/* Digital Gain control -38.4 dB to +38.4 dB in 0.3 dB steps */ -static const DECLARE_TLV_DB_SCALE(digital_gain, -3840, 30, 0); +/* Digital Gain control -84 dB to +40 dB in 1 dB steps */ +static const DECLARE_TLV_DB_SCALE(digital_gain, -8400, 100, -8400); /* Cutoff Freq for High Pass Filter at -3dB */ static const char * const hpf_cutoff_text[] = { @@ -510,15 +510,15 @@ static int wcd_iir_filter_info(struct snd_kcontrol *kcontrol, static const struct snd_kcontrol_new msm8916_wcd_digital_snd_controls[] = { SOC_SINGLE_S8_TLV("RX1 Digital Volume", LPASS_CDC_RX1_VOL_CTL_B2_CTL, - -128, 127, digital_gain), + -84, 40, digital_gain), SOC_SINGLE_S8_TLV("RX2 Digital Volume", LPASS_CDC_RX2_VOL_CTL_B2_CTL, - -128, 127, digital_gain), + -84, 40, digital_gain), SOC_SINGLE_S8_TLV("RX3 Digital Volume", LPASS_CDC_RX3_VOL_CTL_B2_CTL, - -128, 127, digital_gain), + -84, 40, digital_gain), SOC_SINGLE_S8_TLV("TX1 Digital Volume", LPASS_CDC_TX1_VOL_CTL_GAIN, - -128, 127, digital_gain), + -84, 40, digital_gain), SOC_SINGLE_S8_TLV("TX2 Digital Volume", LPASS_CDC_TX2_VOL_CTL_GAIN, - -128, 127, digital_gain), + -84, 40, digital_gain), SOC_ENUM("TX1 HPF Cutoff", tx1_hpf_cutoff_enum), SOC_ENUM("TX2 HPF Cutoff", tx2_hpf_cutoff_enum), SOC_SINGLE("TX1 HPF Switch", LPASS_CDC_TX1_MUX_CTL, 3, 1, 0), @@ -553,22 +553,22 @@ static const struct snd_kcontrol_new msm8916_wcd_digital_snd_controls[] = { WCD_IIR_FILTER_CTL("IIR2 Band3", IIR2, BAND3), WCD_IIR_FILTER_CTL("IIR2 Band4", IIR2, BAND4), WCD_IIR_FILTER_CTL("IIR2 Band5", IIR2, BAND5), - SOC_SINGLE_SX_TLV("IIR1 INP1 Volume", LPASS_CDC_IIR1_GAIN_B1_CTL, - 0, -84, 40, digital_gain), - SOC_SINGLE_SX_TLV("IIR1 INP2 Volume", LPASS_CDC_IIR1_GAIN_B2_CTL, - 0, -84, 40, digital_gain), - SOC_SINGLE_SX_TLV("IIR1 INP3 Volume", LPASS_CDC_IIR1_GAIN_B3_CTL, - 0, -84, 40, digital_gain), - SOC_SINGLE_SX_TLV("IIR1 INP4 Volume", LPASS_CDC_IIR1_GAIN_B4_CTL, - 0, -84, 40, digital_gain), - SOC_SINGLE_SX_TLV("IIR2 INP1 Volume", LPASS_CDC_IIR2_GAIN_B1_CTL, - 0, -84, 40, digital_gain), - SOC_SINGLE_SX_TLV("IIR2 INP2 Volume", LPASS_CDC_IIR2_GAIN_B2_CTL, - 0, -84, 40, digital_gain), - SOC_SINGLE_SX_TLV("IIR2 INP3 Volume", LPASS_CDC_IIR2_GAIN_B3_CTL, - 0, -84, 40, digital_gain), - SOC_SINGLE_SX_TLV("IIR2 INP4 Volume", LPASS_CDC_IIR2_GAIN_B4_CTL, - 0, -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("IIR1 INP1 Volume", LPASS_CDC_IIR1_GAIN_B1_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("IIR1 INP2 Volume", LPASS_CDC_IIR1_GAIN_B2_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("IIR1 INP3 Volume", LPASS_CDC_IIR1_GAIN_B3_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("IIR1 INP4 Volume", LPASS_CDC_IIR1_GAIN_B4_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("IIR2 INP1 Volume", LPASS_CDC_IIR2_GAIN_B1_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("IIR2 INP2 Volume", LPASS_CDC_IIR2_GAIN_B2_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("IIR2 INP3 Volume", LPASS_CDC_IIR2_GAIN_B3_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("IIR2 INP4 Volume", LPASS_CDC_IIR2_GAIN_B4_CTL, + -84, 40, digital_gain), }; -- cgit v1.2.3 From 2fbe0953732e06b471cdedbf6f615b84235580d8 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Thu, 9 Jun 2022 12:19:01 +0100 Subject: ASoC: codecs: wcd9335: move gains from SX_TLV to S8_TLV move all the digital gains form using SX_TLV to S8_TLV, these gains are actually 8 bit gains with 7th signed bit and ranges from -84dB to +40dB rest of the Qualcomm wcd codecs uses these properly. Fixes: 8c4f021d806a ("ASoC: wcd9335: add basic controls") Signed-off-by: Srinivas Kandagatla Link: https://lore.kernel.org/r/20220609111901.318047-3-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/wcd9335.c | 81 +++++++++++++++++++++------------------------- 1 file changed, 36 insertions(+), 45 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index e1b693048084..4a982770dbab 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -2253,51 +2253,42 @@ static int wcd9335_rx_hph_mode_put(struct snd_kcontrol *kc, static const struct snd_kcontrol_new wcd9335_snd_controls[] = { /* -84dB min - 40dB max */ - SOC_SINGLE_SX_TLV("RX0 Digital Volume", WCD9335_CDC_RX0_RX_VOL_CTL, - 0, -84, 40, digital_gain), - SOC_SINGLE_SX_TLV("RX1 Digital Volume", WCD9335_CDC_RX1_RX_VOL_CTL, - 0, -84, 40, digital_gain), - SOC_SINGLE_SX_TLV("RX2 Digital Volume", WCD9335_CDC_RX2_RX_VOL_CTL, - 0, -84, 40, digital_gain), - SOC_SINGLE_SX_TLV("RX3 Digital Volume", WCD9335_CDC_RX3_RX_VOL_CTL, - 0, -84, 40, digital_gain), - SOC_SINGLE_SX_TLV("RX4 Digital Volume", WCD9335_CDC_RX4_RX_VOL_CTL, - 0, -84, 40, digital_gain), - SOC_SINGLE_SX_TLV("RX5 Digital Volume", WCD9335_CDC_RX5_RX_VOL_CTL, - 0, -84, 40, digital_gain), - SOC_SINGLE_SX_TLV("RX6 Digital Volume", WCD9335_CDC_RX6_RX_VOL_CTL, - 0, -84, 40, digital_gain), - SOC_SINGLE_SX_TLV("RX7 Digital Volume", WCD9335_CDC_RX7_RX_VOL_CTL, - 0, -84, 40, digital_gain), - SOC_SINGLE_SX_TLV("RX8 Digital Volume", WCD9335_CDC_RX8_RX_VOL_CTL, - 0, -84, 40, digital_gain), - SOC_SINGLE_SX_TLV("RX0 Mix Digital Volume", - WCD9335_CDC_RX0_RX_VOL_MIX_CTL, - 0, -84, 40, digital_gain), - SOC_SINGLE_SX_TLV("RX1 Mix Digital Volume", - WCD9335_CDC_RX1_RX_VOL_MIX_CTL, - 0, -84, 40, digital_gain), - SOC_SINGLE_SX_TLV("RX2 Mix Digital Volume", - WCD9335_CDC_RX2_RX_VOL_MIX_CTL, - 0, -84, 40, digital_gain), - SOC_SINGLE_SX_TLV("RX3 Mix Digital Volume", - WCD9335_CDC_RX3_RX_VOL_MIX_CTL, - 0, -84, 40, digital_gain), - SOC_SINGLE_SX_TLV("RX4 Mix Digital Volume", - WCD9335_CDC_RX4_RX_VOL_MIX_CTL, - 0, -84, 40, digital_gain), - SOC_SINGLE_SX_TLV("RX5 Mix Digital Volume", - WCD9335_CDC_RX5_RX_VOL_MIX_CTL, - 0, -84, 40, digital_gain), - SOC_SINGLE_SX_TLV("RX6 Mix Digital Volume", - WCD9335_CDC_RX6_RX_VOL_MIX_CTL, - 0, -84, 40, digital_gain), - SOC_SINGLE_SX_TLV("RX7 Mix Digital Volume", - WCD9335_CDC_RX7_RX_VOL_MIX_CTL, - 0, -84, 40, digital_gain), - SOC_SINGLE_SX_TLV("RX8 Mix Digital Volume", - WCD9335_CDC_RX8_RX_VOL_MIX_CTL, - 0, -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("RX0 Digital Volume", WCD9335_CDC_RX0_RX_VOL_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("RX1 Digital Volume", WCD9335_CDC_RX1_RX_VOL_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("RX2 Digital Volume", WCD9335_CDC_RX2_RX_VOL_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("RX3 Digital Volume", WCD9335_CDC_RX3_RX_VOL_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("RX4 Digital Volume", WCD9335_CDC_RX4_RX_VOL_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("RX5 Digital Volume", WCD9335_CDC_RX5_RX_VOL_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("RX6 Digital Volume", WCD9335_CDC_RX6_RX_VOL_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("RX7 Digital Volume", WCD9335_CDC_RX7_RX_VOL_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("RX8 Digital Volume", WCD9335_CDC_RX8_RX_VOL_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("RX0 Mix Digital Volume", WCD9335_CDC_RX0_RX_VOL_MIX_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("RX1 Mix Digital Volume", WCD9335_CDC_RX1_RX_VOL_MIX_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("RX2 Mix Digital Volume", WCD9335_CDC_RX2_RX_VOL_MIX_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("RX3 Mix Digital Volume", WCD9335_CDC_RX3_RX_VOL_MIX_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("RX4 Mix Digital Volume", WCD9335_CDC_RX4_RX_VOL_MIX_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("RX5 Mix Digital Volume", WCD9335_CDC_RX5_RX_VOL_MIX_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("RX6 Mix Digital Volume", WCD9335_CDC_RX6_RX_VOL_MIX_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("RX7 Mix Digital Volume", WCD9335_CDC_RX7_RX_VOL_MIX_CTL, + -84, 40, digital_gain), + SOC_SINGLE_S8_TLV("RX8 Mix Digital Volume", WCD9335_CDC_RX8_RX_VOL_MIX_CTL, + -84, 40, digital_gain), SOC_ENUM("RX INT0_1 HPF cut off", cf_int0_1_enum), SOC_ENUM("RX INT0_2 HPF cut off", cf_int0_2_enum), SOC_ENUM("RX INT1_1 HPF cut off", cf_int1_1_enum), -- cgit v1.2.3 From a43b4394bb35391b74486a788be6634ed91e221a Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Thu, 9 Jun 2022 15:35:31 +0200 Subject: ASoC: codecs: rt274: Always init jack_detect_work MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Improves readability by making sure the work is always initialized. Signed-off-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Link: https://lore.kernel.org/r/20220609133541.3984886-2-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt274.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt274.c b/sound/soc/codecs/rt274.c index ab093bdb5552..a5615e94ec7d 100644 --- a/sound/soc/codecs/rt274.c +++ b/sound/soc/codecs/rt274.c @@ -980,14 +980,11 @@ static int rt274_probe(struct snd_soc_component *component) struct rt274_priv *rt274 = snd_soc_component_get_drvdata(component); rt274->component = component; + INIT_DELAYED_WORK(&rt274->jack_detect_work, rt274_jack_detect_work); - if (rt274->i2c->irq) { - INIT_DELAYED_WORK(&rt274->jack_detect_work, - rt274_jack_detect_work); + if (rt274->i2c->irq) schedule_delayed_work(&rt274->jack_detect_work, - msecs_to_jiffies(1250)); - } - + msecs_to_jiffies(1250)); return 0; } -- cgit v1.2.3 From 3082afe097cc5d794c28a629f3492a0133ee4891 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Thu, 9 Jun 2022 15:35:32 +0200 Subject: ASoC: codecs: rt286: Reorganize jack detect handling MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Clean up in order to use and expose .set_jack callback. Signed-off-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Link: https://lore.kernel.org/r/20220609133541.3984886-3-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt286.c | 17 ++++++----------- sound/soc/codecs/rt286.h | 2 -- sound/soc/intel/boards/broadwell.c | 6 +++--- sound/soc/intel/boards/skl_rt286.c | 2 +- 4 files changed, 10 insertions(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index ad8ea1fa7c23..0534a073ee69 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -311,7 +311,8 @@ static void rt286_jack_detect_work(struct work_struct *work) SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); } -int rt286_mic_detect(struct snd_soc_component *component, struct snd_soc_jack *jack) +static int rt286_mic_detect(struct snd_soc_component *component, + struct snd_soc_jack *jack, void *data) { struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); struct rt286_priv *rt286 = snd_soc_component_get_drvdata(component); @@ -335,7 +336,6 @@ int rt286_mic_detect(struct snd_soc_component *component, struct snd_soc_jack *j return 0; } -EXPORT_SYMBOL_GPL(rt286_mic_detect); static int is_mclk_mode(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) @@ -947,17 +947,11 @@ static int rt286_probe(struct snd_soc_component *component) struct rt286_priv *rt286 = snd_soc_component_get_drvdata(component); rt286->component = component; + INIT_DELAYED_WORK(&rt286->jack_detect_work, rt286_jack_detect_work); - if (rt286->i2c->irq) { - regmap_update_bits(rt286->regmap, - RT286_IRQ_CTRL, 0x2, 0x2); - - INIT_DELAYED_WORK(&rt286->jack_detect_work, - rt286_jack_detect_work); + if (rt286->i2c->irq) schedule_delayed_work(&rt286->jack_detect_work, - msecs_to_jiffies(1250)); - } - + msecs_to_jiffies(50)); return 0; } @@ -1055,6 +1049,7 @@ static const struct snd_soc_component_driver soc_component_dev_rt286 = { .suspend = rt286_suspend, .resume = rt286_resume, .set_bias_level = rt286_set_bias_level, + .set_jack = rt286_mic_detect, .controls = rt286_snd_controls, .num_controls = ARRAY_SIZE(rt286_snd_controls), .dapm_widgets = rt286_dapm_widgets, diff --git a/sound/soc/codecs/rt286.h b/sound/soc/codecs/rt286.h index f27a4e71d5b6..4b7a3bd6043d 100644 --- a/sound/soc/codecs/rt286.h +++ b/sound/soc/codecs/rt286.h @@ -196,7 +196,5 @@ enum { RT286_AIFS, }; -int rt286_mic_detect(struct snd_soc_component *component, struct snd_soc_jack *jack); - #endif /* __RT286_H__ */ diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c index b29d77dfb281..48bf3241b3e6 100644 --- a/sound/soc/intel/boards/broadwell.c +++ b/sound/soc/intel/boards/broadwell.c @@ -75,7 +75,7 @@ static int broadwell_rt286_codec_init(struct snd_soc_pcm_runtime *rtd) if (ret) return ret; - rt286_mic_detect(component, &broadwell_headset); + snd_soc_component_set_jack(component, &broadwell_headset, NULL); return 0; } @@ -235,7 +235,7 @@ static void broadwell_disable_jack(struct snd_soc_card *card) if (!strcmp(component->name, "i2c-INT343A:00")) { dev_dbg(component->dev, "disabling jack detect before going to suspend.\n"); - rt286_mic_detect(component, NULL); + snd_soc_component_set_jack(component, NULL, NULL); break; } } @@ -255,7 +255,7 @@ static int broadwell_resume(struct snd_soc_card *card){ if (!strcmp(component->name, "i2c-INT343A:00")) { dev_dbg(component->dev, "enabling jack detect for resume.\n"); - rt286_mic_detect(component, &broadwell_headset); + snd_soc_component_set_jack(component, &broadwell_headset, NULL); break; } } diff --git a/sound/soc/intel/boards/skl_rt286.c b/sound/soc/intel/boards/skl_rt286.c index e9f9520dcea4..4f3d655e2bfa 100644 --- a/sound/soc/intel/boards/skl_rt286.c +++ b/sound/soc/intel/boards/skl_rt286.c @@ -133,7 +133,7 @@ static int skylake_rt286_codec_init(struct snd_soc_pcm_runtime *rtd) if (ret) return ret; - rt286_mic_detect(component, &skylake_headset); + snd_soc_component_set_jack(component, &skylake_headset, NULL); snd_soc_dapm_ignore_suspend(&rtd->card->dapm, "SoC DMIC"); -- cgit v1.2.3 From 1eb73102da280b28bc3899f694e673bf3e4d0afd Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Thu, 9 Jun 2022 15:35:33 +0200 Subject: ASoC: codecs: rt298: Reorganize jack detect handling MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Clean up in order to use and expose .set_jack callback. Signed-off-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Link: https://lore.kernel.org/r/20220609133541.3984886-4-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt298.c | 17 ++++++----------- sound/soc/codecs/rt298.h | 2 -- sound/soc/intel/boards/bxt_rt298.c | 2 +- 3 files changed, 7 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c index c291786dc82d..1a27e5e63289 100644 --- a/sound/soc/codecs/rt298.c +++ b/sound/soc/codecs/rt298.c @@ -326,7 +326,8 @@ static void rt298_jack_detect_work(struct work_struct *work) SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); } -int rt298_mic_detect(struct snd_soc_component *component, struct snd_soc_jack *jack) +static int rt298_mic_detect(struct snd_soc_component *component, + struct snd_soc_jack *jack, void *data) { struct rt298_priv *rt298 = snd_soc_component_get_drvdata(component); struct snd_soc_dapm_context *dapm; @@ -358,7 +359,6 @@ int rt298_mic_detect(struct snd_soc_component *component, struct snd_soc_jack *j return 0; } -EXPORT_SYMBOL_GPL(rt298_mic_detect); static int is_mclk_mode(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) @@ -1011,17 +1011,11 @@ static int rt298_probe(struct snd_soc_component *component) struct rt298_priv *rt298 = snd_soc_component_get_drvdata(component); rt298->component = component; + INIT_DELAYED_WORK(&rt298->jack_detect_work, rt298_jack_detect_work); - if (rt298->i2c->irq) { - regmap_update_bits(rt298->regmap, - RT298_IRQ_CTRL, 0x2, 0x2); - - INIT_DELAYED_WORK(&rt298->jack_detect_work, - rt298_jack_detect_work); + if (rt298->i2c->irq) schedule_delayed_work(&rt298->jack_detect_work, - msecs_to_jiffies(1250)); - } - + msecs_to_jiffies(1250)); return 0; } @@ -1120,6 +1114,7 @@ static const struct snd_soc_component_driver soc_component_dev_rt298 = { .suspend = rt298_suspend, .resume = rt298_resume, .set_bias_level = rt298_set_bias_level, + .set_jack = rt298_mic_detect, .controls = rt298_snd_controls, .num_controls = ARRAY_SIZE(rt298_snd_controls), .dapm_widgets = rt298_dapm_widgets, diff --git a/sound/soc/codecs/rt298.h b/sound/soc/codecs/rt298.h index ed2b8fd87f4c..f1be9c135401 100644 --- a/sound/soc/codecs/rt298.h +++ b/sound/soc/codecs/rt298.h @@ -207,7 +207,5 @@ enum { RT298_AIFS, }; -int rt298_mic_detect(struct snd_soc_component *component, struct snd_soc_jack *jack); - #endif /* __RT298_H__ */ diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c index 75995d17597d..4bd93c3ba377 100644 --- a/sound/soc/intel/boards/bxt_rt298.c +++ b/sound/soc/intel/boards/bxt_rt298.c @@ -176,7 +176,7 @@ static int broxton_rt298_codec_init(struct snd_soc_pcm_runtime *rtd) if (ret) return ret; - rt298_mic_detect(component, &broxton_headset); + snd_soc_component_set_jack(component, &broxton_headset, NULL); snd_soc_dapm_ignore_suspend(&rtd->card->dapm, "SoC DMIC"); -- cgit v1.2.3 From df4d27b19b892f464685ea45fa6132dd1a2b6864 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Martin=20Povi=C5=A1er?= Date: Mon, 6 Jun 2022 21:19:09 +0200 Subject: ASoC: Introduce 'fixup_controls' card method MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The new method is called just before the card is registered, providing an opportune time for machine-level drivers to do some final controls amending: deactivating individual controls or obtaining control references for later use. Some controls can be created by DAPM after 'late_probe' has been called, hence the need for this new method. Signed-off-by: Martin Povišer Link: https://lore.kernel.org/r/20220606191910.16580-5-povik+lin@cutebit.org Signed-off-by: Mark Brown --- sound/soc/soc-card.c | 6 ++++++ sound/soc/soc-core.c | 1 + 2 files changed, 7 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-card.c b/sound/soc/soc-card.c index 4158f5aacfd3..285ab4c9c716 100644 --- a/sound/soc/soc-card.c +++ b/sound/soc/soc-card.c @@ -197,6 +197,12 @@ int snd_soc_card_late_probe(struct snd_soc_card *card) return 0; } +void snd_soc_card_fixup_controls(struct snd_soc_card *card) +{ + if (card->fixup_controls) + card->fixup_controls(card); +} + int snd_soc_card_remove(struct snd_soc_card *card) { int ret = 0; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 227540851ded..57f7105c12b7 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2066,6 +2066,7 @@ static int snd_soc_bind_card(struct snd_soc_card *card) goto probe_end; snd_soc_dapm_new_widgets(card); + snd_soc_card_fixup_controls(card); ret = snd_card_register(card->snd_card); if (ret < 0) { -- cgit v1.2.3 From 145cb4e7a9ee12326f99948d8980ad258462b6c4 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 10 Jun 2022 11:04:21 +0300 Subject: ASoC: SOF: debug: Clarify the IPC timeout handling path The dmesg log message of "Firmware exception" causes lots of confusion as the snd_sof_handle_fw_exception() is only called in case of an IPC tx timeout, where such a message does not make much sense. To not limit the snd_sof_handle_fw_exception() handler to just one error case, add a parameter to allow the caller to specify a meaningful message to be printed. Signed-off-by: Peter Ujfalusi Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Reviewed-by: Yaochun Hung Link: https://lore.kernel.org/r/20220610080421.31453-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/debug.c | 5 ++--- sound/soc/sof/ipc3.c | 2 +- sound/soc/sof/sof-priv.h | 2 +- 3 files changed, 4 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/debug.c b/sound/soc/sof/debug.c index cf1271eb29b2..c5d797e97c02 100644 --- a/sound/soc/sof/debug.c +++ b/sound/soc/sof/debug.c @@ -428,7 +428,7 @@ static void snd_sof_ipc_dump(struct snd_sof_dev *sdev) } } -void snd_sof_handle_fw_exception(struct snd_sof_dev *sdev) +void snd_sof_handle_fw_exception(struct snd_sof_dev *sdev, const char *msg) { if (IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_RETAIN_DSP_CONTEXT) || sof_debug_check_flag(SOF_DBG_RETAIN_CTX)) { @@ -441,8 +441,7 @@ void snd_sof_handle_fw_exception(struct snd_sof_dev *sdev) /* dump vital information to the logs */ snd_sof_ipc_dump(sdev); - snd_sof_dsp_dbg_dump(sdev, "Firmware exception", - SOF_DBG_DUMP_REGS | SOF_DBG_DUMP_MBOX); + snd_sof_dsp_dbg_dump(sdev, msg, SOF_DBG_DUMP_REGS | SOF_DBG_DUMP_MBOX); sof_fw_trace_fw_crashed(sdev); } EXPORT_SYMBOL(snd_sof_handle_fw_exception); diff --git a/sound/soc/sof/ipc3.c b/sound/soc/sof/ipc3.c index dff5feaad370..ef8019e009b7 100644 --- a/sound/soc/sof/ipc3.c +++ b/sound/soc/sof/ipc3.c @@ -290,7 +290,7 @@ static int ipc3_wait_tx_done(struct snd_sof_ipc *ipc, void *reply_data) dev_err(sdev->dev, "ipc tx timed out for %#x (msg/reply size: %d/%zu)\n", hdr->cmd, hdr->size, msg->reply_size); - snd_sof_handle_fw_exception(ipc->sdev); + snd_sof_handle_fw_exception(ipc->sdev, "IPC timeout"); ret = -ETIMEDOUT; } else { ret = msg->reply_error; diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index 9d7f53ff9c70..32c152528f1d 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -655,7 +655,7 @@ void sof_print_oops_and_stack(struct snd_sof_dev *sdev, const char *level, u32 panic_code, u32 tracep_code, void *oops, struct sof_ipc_panic_info *panic_info, void *stack, size_t stack_words); -void snd_sof_handle_fw_exception(struct snd_sof_dev *sdev); +void snd_sof_handle_fw_exception(struct snd_sof_dev *sdev, const char *msg); int snd_sof_dbg_memory_info_init(struct snd_sof_dev *sdev); int snd_sof_debugfs_add_region_item_iomem(struct snd_sof_dev *sdev, enum snd_sof_fw_blk_type blk_type, u32 offset, size_t size, -- cgit v1.2.3 From c7b6c95c3ef37d7a0b28e62391bccfefdabd7a18 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 10 Jun 2022 10:12:45 +0300 Subject: ASoC: SOF: ipc3-dtrace: use pm_runtime_resume_and_get() MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Use pm_runtime_resume_and_get() to replace the pm_runtime_get_sync() and pm_runtime_put_noidle() pattern. No functional changes. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Péter Ujfalusi Reviewed-by: Bard Liao Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20220610071245.26576-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc3-dtrace.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/ipc3-dtrace.c b/sound/soc/sof/ipc3-dtrace.c index b4e1343f9138..45bf9c5dc412 100644 --- a/sound/soc/sof/ipc3-dtrace.c +++ b/sound/soc/sof/ipc3-dtrace.c @@ -157,9 +157,8 @@ static int ipc3_trace_update_filter(struct snd_sof_dev *sdev, int num_elems, msg->elem_cnt = num_elems; memcpy(&msg->elems[0], elems, num_elems * sizeof(*elems)); - ret = pm_runtime_get_sync(sdev->dev); + ret = pm_runtime_resume_and_get(sdev->dev); if (ret < 0 && ret != -EACCES) { - pm_runtime_put_noidle(sdev->dev); dev_err(sdev->dev, "enabling device failed: %d\n", ret); goto error; } -- cgit v1.2.3 From 46c80e72c16adff20f61240f887c4842e80cb6ec Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Fri, 10 Jun 2022 14:42:57 +0200 Subject: ASoC: Intel: avs: Fix parsing UUIDs in topology MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Use correct type for parsing UUIDs, this eliminates warning present, when compiling with W=1. Fixes: 34ae2cd53673 ("ASoC: Intel: avs: Add topology parsing infrastructure") Reported-by: Pierre-Louis Bossart Signed-off-by: Amadeusz Sławiński Reviewed-by: Cezary Rojewski Link: https://lore.kernel.org/r/20220610124257.4160658-1-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/topology.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/avs/topology.c b/sound/soc/intel/avs/topology.c index 0d11cc8aab0b..6a06fe387d13 100644 --- a/sound/soc/intel/avs/topology.c +++ b/sound/soc/intel/avs/topology.c @@ -128,10 +128,10 @@ struct avs_tplg_token_parser { static int avs_parse_uuid_token(struct snd_soc_component *comp, void *elem, void *object, u32 offset) { - struct snd_soc_tplg_vendor_value_elem *tuple = elem; + struct snd_soc_tplg_vendor_uuid_elem *tuple = elem; guid_t *val = (guid_t *)((u8 *)object + offset); - guid_copy((guid_t *)val, (const guid_t *)&tuple->value); + guid_copy((guid_t *)val, (const guid_t *)&tuple->uuid); return 0; } -- cgit v1.2.3 From 6548c884a595391fab172faeae39e2b329b848f3 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Fri, 10 Jun 2022 15:48:18 +0100 Subject: ASoC: qdsp6: q6apm-dai: unprepare stream if its already prepared prepare callback can be called multiple times, so unprepare the stream if its already prepared. Without this DSP is not happy to setting the params on a already prepared graph. Fixes: 9b4fe0f1cd79 ("ASoC: qdsp6: audioreach: add q6apm-dai support") Reported-by: Srinivasa Rao Mandadapu Signed-off-by: Srinivas Kandagatla Link: https://lore.kernel.org/r/20220610144818.511797-1-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6apm-dai.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/soc/qcom/qdsp6/q6apm-dai.c b/sound/soc/qcom/qdsp6/q6apm-dai.c index 19c4a90ec1ea..ee59ef36b85a 100644 --- a/sound/soc/qcom/qdsp6/q6apm-dai.c +++ b/sound/soc/qcom/qdsp6/q6apm-dai.c @@ -147,6 +147,12 @@ static int q6apm_dai_prepare(struct snd_soc_component *component, cfg.num_channels = runtime->channels; cfg.bit_width = prtd->bits_per_sample; + if (prtd->state) { + /* clear the previous setup if any */ + q6apm_graph_stop(prtd->graph); + q6apm_unmap_memory_regions(prtd->graph, substream->stream); + } + prtd->pcm_count = snd_pcm_lib_period_bytes(substream); prtd->pos = 0; /* rate and channels are sent to audio driver */ -- cgit v1.2.3 From 7263fc6c71c3a88c17a1ce3565b7b6f378d13878 Mon Sep 17 00:00:00 2001 From: Yassine Oudjana Date: Mon, 6 Jun 2022 19:22:26 +0400 Subject: ASoC: wcd9335: Remove RX channel from old list before adding it to a new one Currently in slim_rx_mux_put, an RX channel gets added to a new list even if it is already in one. This can mess up links and make either it, the new list head, or both, get linked to the wrong entries. This can cause an entry to link to itself which in turn ends up making list_for_each_entry in other functions loop infinitely. To avoid issues, always remove the RX channel from any list it's in before adding it to a new list. Signed-off-by: Yassine Oudjana Link: https://lore.kernel.org/r/20220606152226.149164-1-y.oudjana@protonmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/wcd9335.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index 617a36a89dfe..597420679505 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -1289,9 +1289,12 @@ static int slim_rx_mux_put(struct snd_kcontrol *kc, wcd->rx_port_value[port_id] = ucontrol->value.enumerated.item[0]; + /* Remove channel from any list it's in before adding it to a new one */ + list_del_init(&wcd->rx_chs[port_id].list); + switch (wcd->rx_port_value[port_id]) { case 0: - list_del_init(&wcd->rx_chs[port_id].list); + /* Channel already removed from lists. Nothing to do here */ break; case 1: list_add_tail(&wcd->rx_chs[port_id].list, -- cgit v1.2.3 From 6bda28a2f7113b1c49eb05155ace02b75bccae7b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 3 Jun 2022 14:46:09 +0200 Subject: ASoC: wcd9335: Fix spurious event generation The slimbus mux put operation unconditionally reports a change in value which means that spurious events are generated. Fix this by exiting early in that case. Signed-off-by: Mark Brown Link: https://lore.kernel.org/r/20220603124609.4024666-1-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/wcd9335.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index 597420679505..d9f135200688 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -1287,6 +1287,9 @@ static int slim_rx_mux_put(struct snd_kcontrol *kc, struct snd_soc_dapm_update *update = NULL; u32 port_id = w->shift; + if (wcd->rx_port_value[port_id] == ucontrol->value.enumerated.item[0]) + return 0; + wcd->rx_port_value[port_id] = ucontrol->value.enumerated.item[0]; /* Remove channel from any list it's in before adding it to a new one */ -- cgit v1.2.3 From 9f1c8677724a0e6a6ac7a74d2b0192a584df859d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 2 Jun 2022 12:30:29 +0200 Subject: ASoC: hdmi-codec: Update to modern DAI terminology As part of retiring the old defines used to specify DAI formats update the hdmi_codec driver to use the modern names, including the variables in the struct hdmi_codec_daifmt exported to the DRM drivers. In updating this I did note that the only use of this information in DRM drivers is to reject clock provider settings, thinking about what this hardware is doing I rather suspect that there might not be any hardware out there which needs the configuration so it may be worth considering just having hdmi-codec support only clock consumer. Signed-off-by: Mark Brown Link: https://lore.kernel.org/r/20220602103029.3498791-1-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/hdmi-codec.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index b773466619b2..7d1e351f863a 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -606,18 +606,18 @@ static int hdmi_codec_i2s_set_fmt(struct snd_soc_dai *dai, /* Reset daifmt */ memset(cf, 0, sizeof(*cf)); - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - cf->bit_clk_master = 1; - cf->frame_clk_master = 1; + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: + cf->bit_clk_provider = 1; + cf->frame_clk_provider = 1; break; - case SND_SOC_DAIFMT_CBS_CFM: - cf->frame_clk_master = 1; + case SND_SOC_DAIFMT_CBC_CFP: + cf->frame_clk_provider = 1; break; - case SND_SOC_DAIFMT_CBM_CFS: - cf->bit_clk_master = 1; + case SND_SOC_DAIFMT_CBP_CFC: + cf->bit_clk_provider = 1; break; - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBC_CFC: break; default: return -EINVAL; -- cgit v1.2.3 From 65c1c99d96f160e3fead8c6ec67b669cbe62320f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 3 Jun 2022 14:25:26 +0200 Subject: ASoC: wcd938x: Fix event generation for some controls Currently wcd938x_*_put() unconditionally report that the value of the control changed, resulting in spurious events being generated. Return 0 in that case instead as we should. There is still an issue in the compander control which is a bit more complex. Signed-off-by: Mark Brown Reported-by: kernel test robot Link: https://lore.kernel.org/r/20220603122526.3914942-1-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/wcd938x.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c index c1b61b997f69..781ae569be29 100644 --- a/sound/soc/codecs/wcd938x.c +++ b/sound/soc/codecs/wcd938x.c @@ -2519,6 +2519,9 @@ static int wcd938x_tx_mode_put(struct snd_kcontrol *kcontrol, struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; int path = e->shift_l; + if (wcd938x->tx_mode[path] == ucontrol->value.enumerated.item[0]) + return 0; + wcd938x->tx_mode[path] = ucontrol->value.enumerated.item[0]; return 1; @@ -2541,6 +2544,9 @@ static int wcd938x_rx_hph_mode_put(struct snd_kcontrol *kcontrol, struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct wcd938x_priv *wcd938x = snd_soc_component_get_drvdata(component); + if (wcd938x->hph_mode == ucontrol->value.enumerated.item[0]) + return 0; + wcd938x->hph_mode = ucontrol->value.enumerated.item[0]; return 1; @@ -2632,6 +2638,9 @@ static int wcd938x_ldoh_put(struct snd_kcontrol *kcontrol, struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct wcd938x_priv *wcd938x = snd_soc_component_get_drvdata(component); + if (wcd938x->ldoh == ucontrol->value.integer.value[0]) + return 0; + wcd938x->ldoh = ucontrol->value.integer.value[0]; return 1; @@ -2654,6 +2663,9 @@ static int wcd938x_bcs_put(struct snd_kcontrol *kcontrol, struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct wcd938x_priv *wcd938x = snd_soc_component_get_drvdata(component); + if (wcd938x->bcs_dis == ucontrol->value.integer.value[0]) + return 0; + wcd938x->bcs_dis = ucontrol->value.integer.value[0]; return 1; -- cgit v1.2.3 From da440af07fc3dd2b5a5138671eba51991dd1fac8 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sun, 12 Jun 2022 17:56:52 +0200 Subject: ASoC: Intel: bytcr_wm5102: Fix GPIO related probe-ordering problem The "wlf,spkvdd-ena" GPIO needed by the bytcr_wm5102 driver is made available through a gpio-lookup table. This gpio-lookup table is registered by drivers/mfd/arizona-spi.c, which may get probed after the bytcr_wm5102 driver. If the gpio-lookup table has not registered yet then the gpiod_get() will return -ENOENT. Treat -ENOENT as -EPROBE_DEFER to still keep things working in this case. Signed-off-by: Hans de Goede Acked-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220612155652.107310-1-hdegoede@redhat.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_wm5102.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/bytcr_wm5102.c b/sound/soc/intel/boards/bytcr_wm5102.c index 00384c6fbcaa..330c0ace1638 100644 --- a/sound/soc/intel/boards/bytcr_wm5102.c +++ b/sound/soc/intel/boards/bytcr_wm5102.c @@ -421,8 +421,17 @@ static int snd_byt_wm5102_mc_probe(struct platform_device *pdev) priv->spkvdd_en_gpio = gpiod_get(codec_dev, "wlf,spkvdd-ena", GPIOD_OUT_LOW); put_device(codec_dev); - if (IS_ERR(priv->spkvdd_en_gpio)) - return dev_err_probe(dev, PTR_ERR(priv->spkvdd_en_gpio), "getting spkvdd-GPIO\n"); + if (IS_ERR(priv->spkvdd_en_gpio)) { + ret = PTR_ERR(priv->spkvdd_en_gpio); + /* + * The spkvdd gpio-lookup is registered by: drivers/mfd/arizona-spi.c, + * so -ENOENT means that arizona-spi hasn't probed yet. + */ + if (ret == -ENOENT) + ret = -EPROBE_DEFER; + + return dev_err_probe(dev, ret, "getting spkvdd-GPIO\n"); + } /* override platform name, if required */ byt_wm5102_card.dev = dev; -- cgit v1.2.3 From 18489174e4fb98ad07fd387973a39e755ac01dee Mon Sep 17 00:00:00 2001 From: Yong Zhi Date: Fri, 10 Jun 2022 16:44:15 -0500 Subject: ASoC: intel: sof_sdw: add RT711 SDCA card for MTL platform Enable on-board rt711 based sound card for MTL RVP. Reviewed-by: Bard Liao Signed-off-by: Yong Zhi Signed-off-by: Uday M Bhat Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220610214415.42942-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 8 ++++++ sound/soc/intel/common/soc-acpi-intel-mtl-match.c | 31 +++++++++++++++++++++++ 2 files changed, 39 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index f871daa5cb33..aae89afd4d38 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -315,6 +315,14 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { RT711_JD2 | SOF_SDW_FOUR_SPK), }, + /* MeteorLake devices */ + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_FAMILY, "Intel_mtlrvp"), + }, + .driver_data = (void *)(RT711_JD1 | SOF_SDW_TGL_HDMI), + }, {} }; diff --git a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c index cc594b27e03b..74d3b82f8d35 100644 --- a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c @@ -15,6 +15,31 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_machines[] = { }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_mtl_machines); +static const struct snd_soc_acpi_endpoint single_endpoint = { + .num = 0, + .aggregated = 0, + .group_position = 0, + .group_id = 0, +}; + +static const struct snd_soc_acpi_adr_device rt711_sdca_0_adr[] = { + { + .adr = 0x000030025D071101ull, + .num_endpoints = 1, + .endpoints = &single_endpoint, + .name_prefix = "rt711" + } +}; + +static const struct snd_soc_acpi_link_adr mtl_rvp[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(rt711_sdca_0_adr), + .adr_d = rt711_sdca_0_adr, + }, + {} +}; + /* this table is used when there is no I2S codec present */ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_sdw_machines[] = { /* mockup tests need to be first */ @@ -36,6 +61,12 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_sdw_machines[] = { .drv_name = "sof_sdw", .sof_tplg_filename = "sof-mtl-rt715-rt711-rt1308-mono.tplg", }, + { + .link_mask = BIT(0), + .links = mtl_rvp, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-mtl-rt711.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_mtl_sdw_machines); -- cgit v1.2.3 From beb89d1d49e9ae1188356d6e37581e5f0b5f62b4 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Mon, 13 Jun 2022 17:15:51 +0100 Subject: ASoC: sun8i-codec: Partial revert to fix clock specifiers Recent updates accidentally updated the clock producer/consumer specifiers on this device as part of refactoring the CPU side of the DAI links. However, this device sits on the CODEC side and shouldn't have been updated. Partially revert the changes keeping the switch to the new clock terminology but going back to the CODEC defines. Fixes: 7cc3965fde74 ("ASoC: sunxi: Update to use set_fmt_new callback") Reported-by: Samuel Holland Signed-off-by: Charles Keepax Reviewed-by: Samuel Holland Tested-by: Samuel Holland Link: https://lore.kernel.org/r/20220613161552.481337-1-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/sunxi/sun8i-codec.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/sunxi/sun8i-codec.c b/sound/soc/sunxi/sun8i-codec.c index 90d74a2d53f3..f797c535f298 100644 --- a/sound/soc/sunxi/sun8i-codec.c +++ b/sound/soc/sunxi/sun8i-codec.c @@ -287,10 +287,10 @@ static int sun8i_codec_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) /* clock masters */ switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { - case SND_SOC_DAIFMT_BP_FP: /* Codec slave, DAI master */ + case SND_SOC_DAIFMT_CBC_CFC: /* Codec slave, DAI master */ value = 0x1; break; - case SND_SOC_DAIFMT_BC_FC: /* Codec Master, DAI slave */ + case SND_SOC_DAIFMT_CBP_CFP: /* Codec Master, DAI slave */ value = 0x0; break; default: -- cgit v1.2.3 From 845a215558647acd4290dd773b9c0de62c123335 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Mon, 13 Jun 2022 17:15:52 +0100 Subject: ASoC: mediatek: mt8186: Use new direct clock defines Update this driver to the new direct clock producer/consumer defines. It appears this driver was added with the inversion taken account of but still uses the CODEC defines so no inversion of the producer/consumer is necessary. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220613161552.481337-2-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8186/mt8186-dai-tdm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/mediatek/mt8186/mt8186-dai-tdm.c b/sound/soc/mediatek/mt8186/mt8186-dai-tdm.c index dfff209b60da..c6ead7c252f0 100644 --- a/sound/soc/mediatek/mt8186/mt8186-dai-tdm.c +++ b/sound/soc/mediatek/mt8186/mt8186-dai-tdm.c @@ -585,10 +585,10 @@ static int mtk_dai_tdm_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) } switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { - case SND_SOC_DAIFMT_CBP_CFP: + case SND_SOC_DAIFMT_BP_FP: tdm_priv->slave_mode = false; break; - case SND_SOC_DAIFMT_CBC_CFC: + case SND_SOC_DAIFMT_BC_FC: tdm_priv->slave_mode = true; break; default: -- cgit v1.2.3 From 519d1130b66e000ce363ad82c0d61ae36a5392dc Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 10 Jun 2022 16:45:04 -0500 Subject: ASoC: SOF: Intel: hda-dai: enhance debug messages MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The same message was added twice for dai and link_dma, remove the latter one and add dai name and direction to better understand problematic sequences. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Link: https://lore.kernel.org/r/20220610214504.42974-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-dai.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index 228079a52c3d..70721defca46 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -270,7 +270,6 @@ static int hda_link_dma_trigger(struct snd_pcm_substream *substream, int cmd) struct hdac_ext_stream *hext_stream = snd_soc_dai_get_dma_data(cpu_dai, substream); int ret; - dev_dbg(cpu_dai->dev, "%s: cmd=%d\n", __func__, cmd); if (!hext_stream) return 0; @@ -420,13 +419,15 @@ static int ipc3_hda_dai_trigger(struct snd_pcm_substream *substream, struct snd_soc_dapm_widget *w; int ret; + dev_dbg(dai->dev, "%s: cmd=%d dai %s direction %d\n", __func__, cmd, + dai->name, substream->stream); + ret = hda_link_dma_trigger(substream, cmd); if (ret < 0) return ret; w = snd_soc_dai_get_widget(dai, substream->stream); - dev_dbg(dai->dev, "%s: cmd=%d\n", __func__, cmd); switch (cmd) { case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: -- cgit v1.2.3 From 047c69a3a9b19f29e021c77a7e9ce79230a342ed Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Fri, 10 Jun 2022 13:47:22 +0800 Subject: ASoC: fsl_mqs: Add support for i.MX93 platform Add i.MX93 compatible string and specific soc data Signed-off-by: Shengjiu Wang Link: https://lore.kernel.org/r/1654840042-7069-2-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_mqs.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_mqs.c b/sound/soc/fsl/fsl_mqs.c index c9c11914a78e..bb25c58e335f 100644 --- a/sound/soc/fsl/fsl_mqs.c +++ b/sound/soc/fsl/fsl_mqs.c @@ -338,9 +338,23 @@ static const struct fsl_mqs_soc_data fsl_mqs_imx6sx_data = { .div_shift = IMX6SX_GPR2_MQS_CLK_DIV_SHIFT, }; +static const struct fsl_mqs_soc_data fsl_mqs_imx93_data = { + .use_gpr = true, + .ctrl_off = 0x20, + .en_mask = BIT(1), + .en_shift = 1, + .rst_mask = BIT(2), + .rst_shift = 2, + .osr_mask = BIT(3), + .osr_shift = 3, + .div_mask = GENMASK(15, 8), + .div_shift = 8, +}; + static const struct of_device_id fsl_mqs_dt_ids[] = { { .compatible = "fsl,imx8qm-mqs", .data = &fsl_mqs_imx8qm_data }, { .compatible = "fsl,imx6sx-mqs", .data = &fsl_mqs_imx6sx_data }, + { .compatible = "fsl,imx93-mqs", .data = &fsl_mqs_imx93_data }, {} }; MODULE_DEVICE_TABLE(of, fsl_mqs_dt_ids); -- cgit v1.2.3 From f7309dbe628d5c8653d5f3649ef05a65c9b88daf Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 10 Jun 2022 16:46:01 -0500 Subject: ASoC: SOF: reduce default verbosity of IPC logs MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit We currently log the initiation of an IPC as well at its success. [ 3906.106987] kernel: sof-audio-pci-intel-tgl 0000:00:1f.3: ipc tx: 0x80010000: GLB_DAI_MSG: CONFIG [ 3906.107189] kernel: sof-audio-pci-intel-tgl 0000:00:1f.3: ipc tx succeeded: 0x80010000: GLB_DAI_MSG: CONFIG This is overkill in most cases, we already have a message thrown in case of errors and have tracepoints enabled to check for IPC duration. The only case where this might be useful is to check if there is an interleaved IPC RX. Add a flag and only print those logs if enabled. In addition, the DMA_POSITION_UPDATE for traces brings limited information in most cases and pollutes the logs for no good reason. [ 3906.322256] kernel: sof-audio-pci-intel-tgl 0000:00:1f.3: ipc rx: 0x90020000: GLB_TRACE_MSG: DMA_POSITION [ 3906.322308] kernel: sof-audio-pci-intel-tgl 0000:00:1f.3: ipc rx done: 0x90020000: GLB_TRACE_MSG: DMA_POSITION [ 3906.822261] kernel: sof-audio-pci-intel-tgl 0000:00:1f.3: ipc rx: 0x90020000: GLB_TRACE_MSG: DMA_POSITION [ 3906.822319] kernel: sof-audio-pci-intel-tgl 0000:00:1f.3: ipc rx done: 0x90020000: GLB_TRACE_MSG: DMA_POSITION [ 3907.822261] kernel: sof-audio-pci-intel-tgl 0000:00:1f.3: ipc rx: 0x90020000: GLB_TRACE_MSG: DMA_POSITION [ 3907.822319] kernel: sof-audio-pci-intel-tgl 0000:00:1f.3: ipc rx done: 0x90020000: GLB_TRACE_MSG: DMA_POSITION [ 3908.822251] kernel: sof-audio-pci-intel-tgl 0000:00:1f.3: ipc rx: 0x90020000: GLB_TRACE_MSG: DMA_POSITION [ 3908.822309] kernel: sof-audio-pci-intel-tgl 0000:00:1f.3: ipc rx done: 0x90020000: GLB_TRACE_MSG: DMA_POSITION This information is only helpful when debugging the trace support, not when using the trace. Add a flag to only print DMA position update logs if enabled. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Link: https://lore.kernel.org/r/20220610214601.43005-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc3.c | 5 ++++- sound/soc/sof/sof-priv.h | 6 ++++++ 2 files changed, 10 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sof/ipc3.c b/sound/soc/sof/ipc3.c index 0df57e7e83ac..1fb132b477bf 100644 --- a/sound/soc/sof/ipc3.c +++ b/sound/soc/sof/ipc3.c @@ -147,6 +147,8 @@ static void ipc3_log_header(struct device *dev, u8 *text, u32 cmd) case SOF_IPC_TRACE_DMA_PARAMS: str2 = "DMA_PARAMS"; break; case SOF_IPC_TRACE_DMA_POSITION: + if (!sof_debug_check_flag(SOF_DBG_PRINT_DMA_POSITION_UPDATE_LOGS)) + return; str2 = "DMA_POSITION"; break; case SOF_IPC_TRACE_DMA_PARAMS_EXT: str2 = "DMA_PARAMS_EXT"; break; @@ -299,7 +301,8 @@ static int ipc3_wait_tx_done(struct snd_sof_ipc *ipc, void *reply_data) "ipc tx error for %#x (msg/reply size: %d/%zu): %d\n", hdr->cmd, hdr->size, msg->reply_size, ret); } else { - ipc3_log_header(sdev->dev, "ipc tx succeeded", hdr->cmd); + if (sof_debug_check_flag(SOF_DBG_PRINT_IPC_SUCCESS_LOGS)) + ipc3_log_header(sdev->dev, "ipc tx succeeded", hdr->cmd); if (msg->reply_size) /* copy the data returned from DSP */ memcpy(reply_data, msg->reply_data, diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index bd637153c08f..52396f38dcec 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -37,6 +37,12 @@ #define SOF_DBG_IGNORE_D3_PERSISTENT BIT(7) /* ignore the DSP D3 persistent capability * and always download firmware upon D3 exit */ +#define SOF_DBG_PRINT_DMA_POSITION_UPDATE_LOGS BIT(8) /* print DMA position updates + * in dmesg logs + */ +#define SOF_DBG_PRINT_IPC_SUCCESS_LOGS BIT(9) /* print IPC success + * in dmesg logs + */ /* Flag definitions used for controlling the DSP dump behavior */ #define SOF_DBG_DUMP_REGS BIT(0) -- cgit v1.2.3 From 689614ce48b0310b50d8d6c9a64f8a98cfc6f195 Mon Sep 17 00:00:00 2001 From: Ajit Kumar Pandey Date: Tue, 14 Jun 2022 10:52:51 +0300 Subject: ASoC: SOF: topology: add code to parse config params for ACPDMIC dai MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add sof_ipc_dai_acpdmic_params and tokens to parse dmic channels and rate params from topology file Signed-off-by: Ajit Kumar Pandey Reviewed-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Reviewed-by: Pierre-Louis Bossart Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20220614075251.21499-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc3-pcm.c | 8 ++++---- sound/soc/sof/ipc3-topology.c | 25 +++++++++++++++++++------ sound/soc/sof/sof-audio.h | 1 + sound/soc/sof/topology.c | 4 ++++ 4 files changed, 28 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/ipc3-pcm.c b/sound/soc/sof/ipc3-pcm.c index c8774a891d6f..b97e63d3724a 100644 --- a/sound/soc/sof/ipc3-pcm.c +++ b/sound/soc/sof/ipc3-pcm.c @@ -344,10 +344,10 @@ static int sof_ipc3_pcm_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, channels->min, channels->max); break; case SOF_DAI_AMD_DMIC: - rate->min = private->dai_config->acpdmic.fsync_rate; - rate->max = private->dai_config->acpdmic.fsync_rate; - channels->min = private->dai_config->acpdmic.tdm_slots; - channels->max = private->dai_config->acpdmic.tdm_slots; + rate->min = private->dai_config->acpdmic.pdm_rate; + rate->max = private->dai_config->acpdmic.pdm_rate; + channels->min = private->dai_config->acpdmic.pdm_ch; + channels->max = private->dai_config->acpdmic.pdm_ch; dev_dbg(component->dev, "AMD_DMIC rate_min: %d rate_max: %d\n", rate->min, rate->max); diff --git a/sound/soc/sof/ipc3-topology.c b/sound/soc/sof/ipc3-topology.c index a91d7df3f07e..5ee1537f9c2d 100644 --- a/sound/soc/sof/ipc3-topology.c +++ b/sound/soc/sof/ipc3-topology.c @@ -266,6 +266,16 @@ static const struct sof_topology_token afe_tokens[] = { offsetof(struct sof_ipc_dai_mtk_afe_params, format)}, }; +/* ACPDMIC */ +static const struct sof_topology_token acpdmic_tokens[] = { + {SOF_TKN_AMD_ACPDMIC_RATE, + SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc_dai_acpdmic_params, pdm_rate)}, + {SOF_TKN_AMD_ACPDMIC_CH, + SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc_dai_acpdmic_params, pdm_ch)}, +}; + /* Core tokens */ static const struct sof_topology_token core_tokens[] = { {SOF_TKN_COMP_CORE_ID, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, @@ -300,6 +310,7 @@ static const struct sof_token_info ipc3_token_list[SOF_TOKEN_COUNT] = { [SOF_ESAI_TOKENS] = {"ESAI tokens", esai_tokens, ARRAY_SIZE(esai_tokens)}, [SOF_SAI_TOKENS] = {"SAI tokens", sai_tokens, ARRAY_SIZE(sai_tokens)}, [SOF_AFE_TOKENS] = {"AFE tokens", afe_tokens, ARRAY_SIZE(afe_tokens)}, + [SOF_ACPDMIC_TOKENS] = {"ACPDMIC tokens", acpdmic_tokens, ARRAY_SIZE(acpdmic_tokens)}, }; /** @@ -1120,20 +1131,22 @@ static int sof_link_acp_dmic_load(struct snd_soc_component *scomp, struct snd_so struct snd_soc_tplg_hw_config *hw_config = slink->hw_configs; struct sof_dai_private_data *private = dai->private; u32 size = sizeof(*config); + int ret; /* handle master/slave and inverted clocks */ sof_dai_set_format(hw_config, config); - /* init IPC */ - memset(&config->acpdmic, 0, sizeof(config->acpdmic)); config->hdr.size = size; - config->acpdmic.fsync_rate = le32_to_cpu(hw_config->fsync_rate); - config->acpdmic.tdm_slots = le32_to_cpu(hw_config->tdm_slots); + /* parse the required set of ACPDMIC tokens based on num_hw_cfgs */ + ret = sof_update_ipc_object(scomp, &config->acpdmic, SOF_ACPDMIC_TOKENS, slink->tuples, + slink->num_tuples, size, slink->num_hw_configs); + if (ret < 0) + return ret; dev_info(scomp->dev, "ACP_DMIC config ACP%d channel %d rate %d\n", - config->dai_index, config->acpdmic.tdm_slots, - config->acpdmic.fsync_rate); + config->dai_index, config->acpdmic.pdm_ch, + config->acpdmic.pdm_rate); dai->number_configs = 1; dai->current_config = 0; diff --git a/sound/soc/sof/sof-audio.h b/sound/soc/sof/sof-audio.h index 79486266081f..4284ea2f3a1f 100644 --- a/sound/soc/sof/sof-audio.h +++ b/sound/soc/sof/sof-audio.h @@ -236,6 +236,7 @@ enum sof_tokens { SOF_AUDIO_FMT_NUM_TOKENS, SOF_COPIER_FORMAT_TOKENS, SOF_GAIN_TOKENS, + SOF_ACPDMIC_TOKENS, /* this should be the last */ SOF_TOKEN_COUNT, diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 1893c590f2f0..7e54eb1bf77b 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -1739,6 +1739,10 @@ static int sof_link_load(struct snd_soc_component *scomp, int index, struct snd_ token_id = SOF_AFE_TOKENS; num_tuples += token_list[SOF_AFE_TOKENS].count; break; + case SOF_DAI_AMD_DMIC: + token_id = SOF_ACPDMIC_TOKENS; + num_tuples += token_list[SOF_ACPDMIC_TOKENS].count; + break; default: break; } -- cgit v1.2.3 From 7ed1f83bb4f05fe460984ae49e98d1c1be38fb5f Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 14 Jun 2022 10:56:17 +0300 Subject: ASoC: SOF: Compile and runtime IPC version selection The new IPC4 version is only supported by Intel platforms, iMX, AMD and MediaTek only uses the standard SOF IPC. There is no need for these platforms to build kernel support for IPC4 as it is just dead code for them. SND_SOC_SOF_IPC3 and SND_SOC_SOF_INTEL_IPC4 is introduced to allow compile time selection and exclusion of IPC implementations. To avoid randconfig failures add also support for runtime selection of the IPC ops in ipc.c based on sdev->pdata->ipc_type Signed-off-by: Peter Ujfalusi Reviewed-by: Bard Liao Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/20220614075618.28605-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/Kconfig | 7 +++++++ sound/soc/sof/Makefile | 16 ++++++++++++---- sound/soc/sof/amd/Kconfig | 1 + sound/soc/sof/imx/Kconfig | 1 + sound/soc/sof/intel/Kconfig | 11 +++++++++++ sound/soc/sof/ipc.c | 24 ++++++++++++++++++------ sound/soc/sof/mediatek/Kconfig | 1 + 7 files changed, 51 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/Kconfig b/sound/soc/sof/Kconfig index 4542868cd730..e90f173d067c 100644 --- a/sound/soc/sof/Kconfig +++ b/sound/soc/sof/Kconfig @@ -252,6 +252,13 @@ config SND_SOC_SOF_PROBE_WORK_QUEUE When selected, the probe is handled in two steps, for example to avoid lockdeps if request_module is used in the probe. +# Supported IPC versions +config SND_SOC_SOF_IPC3 + bool + +config SND_SOC_SOF_INTEL_IPC4 + bool + source "sound/soc/sof/amd/Kconfig" source "sound/soc/sof/imx/Kconfig" source "sound/soc/sof/intel/Kconfig" diff --git a/sound/soc/sof/Makefile b/sound/soc/sof/Makefile index 2fa8088707a8..9a74ed116ed9 100644 --- a/sound/soc/sof/Makefile +++ b/sound/soc/sof/Makefile @@ -1,10 +1,18 @@ # SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) snd-sof-objs := core.o ops.o loader.o ipc.o pcm.o pm.o debug.o topology.o\ - control.o trace.o iomem-utils.o sof-audio.o stream-ipc.o\ - ipc3-topology.o ipc3-control.o ipc3.o ipc3-pcm.o ipc3-loader.o\ - ipc3-dtrace.o\ - ipc4.o ipc4-loader.o ipc4-topology.o ipc4-control.o ipc4-pcm.o + control.o trace.o iomem-utils.o sof-audio.o stream-ipc.o + +# IPC implementations +ifneq ($(CONFIG_SND_SOC_SOF_IPC3),) +snd-sof-objs += ipc3.o ipc3-loader.o ipc3-topology.o ipc3-control.o ipc3-pcm.o\ + ipc3-dtrace.o +endif +ifneq ($(CONFIG_SND_SOC_SOF_INTEL_IPC4),) +snd-sof-objs += ipc4.o ipc4-loader.o ipc4-topology.o ipc4-control.o ipc4-pcm.o +endif + +# SOF client support ifneq ($(CONFIG_SND_SOC_SOF_CLIENT),) snd-sof-objs += sof-client.o endif diff --git a/sound/soc/sof/amd/Kconfig b/sound/soc/sof/amd/Kconfig index 085232e04582..190c85d57047 100644 --- a/sound/soc/sof/amd/Kconfig +++ b/sound/soc/sof/amd/Kconfig @@ -17,6 +17,7 @@ if SND_SOC_SOF_AMD_TOPLEVEL config SND_SOC_SOF_AMD_COMMON tristate select SND_SOC_SOF + select SND_SOC_SOF_IPC3 select SND_SOC_SOF_PCI_DEV select SND_AMD_ACP_CONFIG select SND_SOC_ACPI if ACPI diff --git a/sound/soc/sof/imx/Kconfig b/sound/soc/sof/imx/Kconfig index 9b8d5bb1e449..cc6e695f913a 100644 --- a/sound/soc/sof/imx/Kconfig +++ b/sound/soc/sof/imx/Kconfig @@ -15,6 +15,7 @@ config SND_SOC_SOF_IMX_COMMON tristate select SND_SOC_SOF_OF_DEV select SND_SOC_SOF + select SND_SOC_SOF_IPC3 select SND_SOC_SOF_XTENSA select SND_SOC_SOF_COMPRESS help diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index 0def2aa5581d..80cdc3788bbe 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -40,6 +40,7 @@ if SND_SOC_SOF_ACPI config SND_SOC_SOF_BAYTRAIL tristate "SOF support for Baytrail, Braswell and Cherrytrail" default SND_SOC_SOF_ACPI + select SND_SOC_SOF_IPC3 select SND_SOC_SOF_INTEL_COMMON select SND_SOC_SOF_INTEL_ATOM_HIFI_EP select SND_SOC_SOF_ACPI_DEV @@ -60,6 +61,7 @@ config SND_SOC_SOF_BAYTRAIL config SND_SOC_SOF_BROADWELL tristate "SOF support for Broadwell" default SND_SOC_SOF_ACPI + select SND_SOC_SOF_IPC3 select SND_SOC_SOF_INTEL_COMMON select SND_SOC_SOF_INTEL_HIFI_EP_IPC select SND_SOC_SOF_ACPI_DEV @@ -85,6 +87,7 @@ config SND_SOC_SOF_MERRIFIELD tristate "SOF support for Tangier/Merrifield" default SND_SOC_SOF_PCI select SND_SOC_SOF_PCI_DEV + select SND_SOC_SOF_IPC3 select SND_SOC_SOF_INTEL_ATOM_HIFI_EP help This adds support for Sound Open Firmware for Intel(R) platforms @@ -95,6 +98,8 @@ config SND_SOC_SOF_MERRIFIELD config SND_SOC_SOF_INTEL_APL tristate select SND_SOC_SOF_HDA_COMMON + select SND_SOC_SOF_IPC3 + select SND_SOC_SOF_INTEL_IPC4 config SND_SOC_SOF_APOLLOLAKE tristate "SOF support for Apollolake" @@ -120,6 +125,8 @@ config SND_SOC_SOF_INTEL_CNL tristate select SND_SOC_SOF_HDA_COMMON select SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE + select SND_SOC_SOF_IPC3 + select SND_SOC_SOF_INTEL_IPC4 config SND_SOC_SOF_CANNONLAKE tristate "SOF support for Cannonlake" @@ -154,6 +161,8 @@ config SND_SOC_SOF_INTEL_ICL tristate select SND_SOC_SOF_HDA_COMMON select SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE + select SND_SOC_SOF_IPC3 + select SND_SOC_SOF_INTEL_IPC4 config SND_SOC_SOF_ICELAKE tristate "SOF support for Icelake" @@ -179,6 +188,8 @@ config SND_SOC_SOF_INTEL_TGL tristate select SND_SOC_SOF_HDA_COMMON select SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE + select SND_SOC_SOF_IPC3 + select SND_SOC_SOF_INTEL_IPC4 config SND_SOC_SOF_TIGERLAKE tristate "SOF support for Tigerlake" diff --git a/sound/soc/sof/ipc.c b/sound/soc/sof/ipc.c index c5aef5fc056b..6ed3f9b6a0c4 100644 --- a/sound/soc/sof/ipc.c +++ b/sound/soc/sof/ipc.c @@ -155,12 +155,22 @@ struct snd_sof_ipc *snd_sof_ipc_init(struct snd_sof_dev *sdev) init_waitqueue_head(&msg->waitq); - /* - * Use IPC3 ops as it is the only available version now. With the addition of new IPC - * versions, this will need to be modified to use the selected version at runtime. - */ - ipc->ops = &ipc3_ops; - ops = ipc->ops; + switch (sdev->pdata->ipc_type) { +#if defined(CONFIG_SND_SOC_SOF_IPC3) + case SOF_IPC: + ops = &ipc3_ops; + break; +#endif +#if defined(CONFIG_SND_SOC_SOF_INTEL_IPC4) + case SOF_INTEL_IPC4: + ops = &ipc4_ops; + break; +#endif + default: + dev_err(sdev->dev, "Not supported IPC version: %d\n", + sdev->pdata->ipc_type); + return NULL; + } /* check for mandatory ops */ if (!ops->tx_msg || !ops->rx_msg || !ops->set_get_data || !ops->get_reply) { @@ -190,6 +200,8 @@ struct snd_sof_ipc *snd_sof_ipc_init(struct snd_sof_dev *sdev) return NULL; } + ipc->ops = ops; + return ipc; } EXPORT_SYMBOL(snd_sof_ipc_init); diff --git a/sound/soc/sof/mediatek/Kconfig b/sound/soc/sof/mediatek/Kconfig index a149dd1b3f44..4a2eddf6009a 100644 --- a/sound/soc/sof/mediatek/Kconfig +++ b/sound/soc/sof/mediatek/Kconfig @@ -15,6 +15,7 @@ config SND_SOC_SOF_MTK_COMMON tristate select SND_SOC_SOF_OF_DEV select SND_SOC_SOF + select SND_SOC_SOF_IPC3 select SND_SOC_SOF_XTENSA select SND_SOC_SOF_COMPRESS help -- cgit v1.2.3 From 30ac49841386f933339817771ec315a34a4c0edd Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 3 Jun 2022 13:25:08 +0200 Subject: ASoC: ops: Don't modify the driver's plaform_max when reading state Currently snd_soc_info_volsw() will set a platform_max based on the limit the control has if one is not already set. This isn't really great, we shouldn't be modifying the passed in driver data especially in a path like this which may not ever be executed or where we may execute other callbacks before this one. Instead make this function leave the data unchanged, and clarify things a bit by referring to max rather than platform_max within the function. platform_max is now applied as a limit after working out the natural maximum value for the control. This means that platform_max is no longer treated as a direct register value for controls were min is non-zero. The put() callbacks already validate on this basis, and there do not appear to be any in tree users that would be affected. Signed-off-by: Mark Brown Link: https://lore.kernel.org/r/20220603112508.3856519-1-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/soc-ops.c | 13 +++++++------ 1 file changed, 7 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 184910ed2d7b..b624ed79ade3 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -176,20 +176,21 @@ int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, { struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; - int platform_max; + int max; - if (!mc->platform_max) - mc->platform_max = mc->max; - platform_max = mc->platform_max; + max = uinfo->value.integer.max = mc->max - mc->min; + if (mc->platform_max && mc->platform_max < max) + max = mc->platform_max; - if (platform_max == 1 && !strstr(kcontrol->id.name, " Volume")) + if (max == 1 && !strstr(kcontrol->id.name, " Volume")) uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; else uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = snd_soc_volsw_is_stereo(mc) ? 2 : 1; uinfo->value.integer.min = 0; - uinfo->value.integer.max = platform_max - mc->min; + uinfo->value.integer.max = max; + return 0; } EXPORT_SYMBOL_GPL(snd_soc_info_volsw); -- cgit v1.2.3 From a150345aa758492e05d2934f318ce7c2566b1cfe Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 14 Jun 2022 17:26:30 +0800 Subject: ASoC: SOF: ipc4-topology: add SoundWire/ALH aggregation support Some SoundWire hardware topologies rely on different amplifiers or capture devices connected on different links. These devices need to be 'aggregated', remain synchronized and be handled as a single logical device. In the IPC3 solution, the aggregation for amplifiers was handled by a firmware 'demux' component. In the IPC4 solution, the demux component is not needed, the gateway component can handle multiple ALH/DMA transfers at the same time. This change makes the topology slightly more complicated in that only one ALH DAI will be connected in the topology with the gateway. The other DAIs that are part of the 'aggregated' dailink are not shown in the DAPM graph as connected to the gateway, but they will however be activated thanks to a feature in soc-dapm.c where events are forwarded to all DAIs in the dailink (see soc_dapm_stream_event). The topology also sets the same stream name for all widgets, dais and dailinks, so a search for the stream name helps identify cases where SoundWire/ALH aggregation is needed. Signed-off-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/20220614092630.20144-1-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-topology.c | 68 +++++++++++++++++++++++++++++++++++++++---- sound/soc/sof/ipc4-topology.h | 11 +++++++ 2 files changed, 74 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index cb0f0823b8eb..3c949298e007 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -19,6 +19,8 @@ #define SOF_IPC4_GAIN_PARAM_ID 0 #define SOF_IPC4_TPLG_ABI_SIZE 6 +static DEFINE_IDA(alh_group_ida); + static const struct sof_topology_token ipc4_sched_tokens[] = { {SOF_TKN_SCHED_LP_MODE, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, offsetof(struct sof_ipc4_pipeline, lp_mode)} @@ -478,7 +480,9 @@ static int sof_ipc4_widget_setup_comp_dai(struct snd_sof_widget *swidget) switch (ipc4_copier->dai_type) { case SOF_DAI_INTEL_ALH: { + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); struct sof_ipc4_alh_configuration_blob *blob; + struct snd_sof_widget *w; blob = kzalloc(sizeof(*blob), GFP_KERNEL); if (!blob) { @@ -486,6 +490,14 @@ static int sof_ipc4_widget_setup_comp_dai(struct snd_sof_widget *swidget) goto err; } + list_for_each_entry(w, &sdev->widget_list, list) { + if (w->widget->sname && + strcmp(w->widget->sname, swidget->widget->sname)) + continue; + + blob->alh_cfg.count++; + } + ipc4_copier->copier_config = (uint32_t *)blob; ipc4_copier->data.gtw_cfg.config_length = sizeof(*blob) >> 2; break; @@ -844,6 +856,17 @@ static void sof_ipc4_unprepare_copier_module(struct snd_sof_widget *swidget) struct snd_sof_dai *dai = swidget->private; ipc4_copier = dai->private; + if (ipc4_copier->dai_type == SOF_DAI_INTEL_ALH) { + struct sof_ipc4_alh_configuration_blob *blob; + unsigned int group_id; + + blob = (struct sof_ipc4_alh_configuration_blob *)ipc4_copier->copier_config; + if (blob->alh_cfg.count > 1) { + group_id = SOF_IPC4_NODE_INDEX(ipc4_copier->data.gtw_cfg.node_id) - + ALH_MULTI_GTW_BASE; + ida_free(&alh_group_ida, group_id); + } + } } if (ipc4_copier) { @@ -973,6 +996,7 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, struct sof_ipc4_copier_data *copier_data; struct snd_pcm_hw_params *ref_params; struct sof_ipc4_copier *ipc4_copier; + struct snd_sof_dai *dai; struct snd_mask *fmt; int out_sample_valid_bits; size_t ref_audio_fmt_size; @@ -1022,7 +1046,7 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, case snd_soc_dapm_dai_in: case snd_soc_dapm_dai_out: { - struct snd_sof_dai *dai = swidget->private; + dai = swidget->private; ipc4_copier = (struct sof_ipc4_copier *)dai->private; copier_data = &ipc4_copier->data; @@ -1077,22 +1101,56 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, */ if (ipc4_copier->dai_type == SOF_DAI_INTEL_ALH) { struct sof_ipc4_alh_configuration_blob *blob; + struct sof_ipc4_copier_data *alh_data; + struct sof_ipc4_copier *alh_copier; + struct snd_sof_widget *w; + u32 ch_mask = 0; u32 ch_map; int i; blob = (struct sof_ipc4_alh_configuration_blob *)ipc4_copier->copier_config; - /* TODO: add aggregation mode support */ - blob->alh_cfg.count = 1; - blob->alh_cfg.mapping[0].alh_id = copier_data->gtw_cfg.node_id; + blob->gw_attr.lp_buffer_alloc = 0; /* Get channel_mask from ch_map */ ch_map = copier_data->base_config.audio_fmt.ch_map; for (i = 0; ch_map; i++) { if ((ch_map & 0xf) != 0xf) - blob->alh_cfg.mapping[0].channel_mask |= BIT(i); + ch_mask |= BIT(i); ch_map >>= 4; } + + /* + * Set each gtw_cfg.node_id to blob->alh_cfg.mapping[] + * for all widgets with the same stream name + */ + i = 0; + list_for_each_entry(w, &sdev->widget_list, list) { + if (w->widget->sname && + strcmp(w->widget->sname, swidget->widget->sname)) + continue; + + dai = w->private; + alh_copier = (struct sof_ipc4_copier *)dai->private; + alh_data = &alh_copier->data; + blob->alh_cfg.mapping[i].alh_id = alh_data->gtw_cfg.node_id; + blob->alh_cfg.mapping[i].channel_mask = ch_mask; + i++; + } + if (blob->alh_cfg.count > 1) { + int group_id; + + group_id = ida_alloc_max(&alh_group_ida, ALH_MULTI_GTW_COUNT, + GFP_KERNEL); + + if (group_id < 0) + return group_id; + + /* add multi-gateway base */ + group_id += ALH_MULTI_GTW_BASE; + copier_data->gtw_cfg.node_id &= ~SOF_IPC4_NODE_INDEX_MASK; + copier_data->gtw_cfg.node_id |= SOF_IPC4_NODE_INDEX(group_id); + } } } } diff --git a/sound/soc/sof/ipc4-topology.h b/sound/soc/sof/ipc4-topology.h index 1a9c0627bae9..3bc2fe38c733 100644 --- a/sound/soc/sof/ipc4-topology.h +++ b/sound/soc/sof/ipc4-topology.h @@ -42,6 +42,17 @@ #define ALH_MAX_NUMBER_OF_GTW 16 +/* + * The base of multi-gateways. Multi-gateways addressing starts from + * ALH_MULTI_GTW_BASE and there are ALH_MULTI_GTW_COUNT multi-sources + * and ALH_MULTI_GTW_COUNT multi-sinks available. + * Addressing is continuous from ALH_MULTI_GTW_BASE to + * ALH_MULTI_GTW_BASE + ALH_MULTI_GTW_COUNT - 1. + */ +#define ALH_MULTI_GTW_BASE 0x50 +/* A magic number from FW */ +#define ALH_MULTI_GTW_COUNT 8 + /** * struct sof_ipc4_pipeline - pipeline config data * @priority: Priority of this pipeline -- cgit v1.2.3 From aa2a4b897132169fbc6d32932644b95875cf9c7f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 4 Jun 2022 11:54:07 +0100 Subject: ASoC: ops: Fix boolean/integer detection for simple controls The standard snd_soc_info_volsw() detects if a control is a volume control and needs to be reported as an integer even if it only has two values by looking for the string " Volume" in the control name. This results in false positives if the control has a name like "HP Volume Ramp Switch" since any " Volume" is matched, not just a trailing one. Fix this by making sure that we only match at the end of the control name. Signed-off-by: Mark Brown Link: https://lore.kernel.org/r/20220604105407.4055294-1-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/soc-ops.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index b624ed79ade3..c22d87581f6f 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -176,13 +176,21 @@ int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, { struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; + const char *vol_string = NULL; int max; max = uinfo->value.integer.max = mc->max - mc->min; if (mc->platform_max && mc->platform_max < max) max = mc->platform_max; - if (max == 1 && !strstr(kcontrol->id.name, " Volume")) + /* Even two value controls ending in Volume should always be integer */ + if (max == 1) { + vol_string = strstr(kcontrol->id.name, " Volume"); + if (vol_string && strcmp(vol_string, " Volume")) + vol_string = NULL; + } + + if (!vol_string) uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; else uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; -- cgit v1.2.3 From d919630fe77904931277e663c902582ea6f4e4cf Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 14 Jun 2022 14:10:22 +0100 Subject: ASoC: cs35l45: Add endianness flag in snd_soc_component_driver The endianness flag is used on the CODEC side to specify an ambivalence to endian, typically because it is lost over the hardware link. This device receives audio over an I2S DAI and as such should have endianness applied. Fixes: 0d463d016000 ("ASoC: cs35l45: Add driver for Cirrus Logic CS35L45 Smart Amp") Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220614131022.778057-1-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs35l45.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/cs35l45.c b/sound/soc/codecs/cs35l45.c index c94edfce4b72..d15b3b77c7eb 100644 --- a/sound/soc/codecs/cs35l45.c +++ b/sound/soc/codecs/cs35l45.c @@ -500,6 +500,8 @@ static const struct snd_soc_component_driver cs35l45_component = { .num_controls = ARRAY_SIZE(cs35l45_controls), .name = "cs35l45", + + .endianness = 1, }; static int __maybe_unused cs35l45_runtime_suspend(struct device *dev) -- cgit v1.2.3 From c27e1efb61c545f36c450ef60862df9251d239a4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 10 Jun 2022 08:45:37 +0200 Subject: ALSA: control: Use xarray for faster lookups The control elements are managed in a single linked list and we traverse the whole list for matching each numid or ctl id per every inquiry of a control element. This is OK-ish for a small number of elements but obviously it doesn't scale. Especially the matching with the ctl id takes time because it checks each field of the snd_ctl_id element, e.g. the name string is matched with strcmp(). This patch adds the hash tables with Xarray for improving the lookup speed of a control element. There are two xarray tables added to the card; one for numid and another for ctl id. For the numid, we use the numid as the index, while for the ctl id, we calculate a hash key. The lookup is done via a single xa_load() execution. As long as the given control element is found on the Xarray table, that's fine, we can give back a quick lookup result. The problem is when no entry hits on the table, and for this case, we have a slight optimization. Namely, the driver checks whether we had a collision on Xarray table, and do a fallback search (linear lookup of the full entries) only if a hash key collision happened beforehand. So, in theory, the inquiry for a non-existing element might take still time even with this patch in a worst case, but this must be pretty rare. The feature is enabled via CONFIG_SND_CTL_FAST_LOOKUP, which is turned on as default. For simplicity, the option can be turned off only when CONFIG_EXPERT is set ("You are expert? Then you manage 1000 knobs"). Link: https://lore.kernel.org/r/20211028130027.18764-1-tiwai@suse.de Link: https://lore.kernel.org/r/20220609180504.775-1-tiwai@suse.de Link: https://lore.kernel.org/all/cover.1653813866.git.quic_rbankapu@quicinc.com/ Link: https://lore.kernel.org/r/20220610064537.18660-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/Kconfig | 10 +++ sound/core/control.c | 180 ++++++++++++++++++++++++++++++++++++++++++--------- sound/core/init.c | 4 ++ 3 files changed, 162 insertions(+), 32 deletions(-) (limited to 'sound') diff --git a/sound/core/Kconfig b/sound/core/Kconfig index dd7b40734723..25b2434e4556 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -154,6 +154,16 @@ config SND_VERBOSE_PRINTK You don't need this unless you're debugging ALSA. +config SND_CTL_FAST_LOOKUP + bool "Fast lookup of control elements" if EXPERT + default y + select XARRAY_MULTI + help + This option enables the faster lookup of control elements. + It will consume more memory because of the additional Xarray. + If you want to choose the memory footprint over the performance + inevitably, turn this off. + config SND_DEBUG bool "Debug" help diff --git a/sound/core/control.c b/sound/core/control.c index a25c0d64d104..6a8fd9933f06 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -364,6 +364,93 @@ static int snd_ctl_find_hole(struct snd_card *card, unsigned int count) return 0; } +/* check whether the given id is contained in the given kctl */ +static bool elem_id_matches(const struct snd_kcontrol *kctl, + const struct snd_ctl_elem_id *id) +{ + return kctl->id.iface == id->iface && + kctl->id.device == id->device && + kctl->id.subdevice == id->subdevice && + !strncmp(kctl->id.name, id->name, sizeof(kctl->id.name)) && + kctl->id.index <= id->index && + kctl->id.index + kctl->count > id->index; +} + +#ifdef CONFIG_SND_CTL_FAST_LOOKUP +/* Compute a hash key for the corresponding ctl id + * It's for the name lookup, hence the numid is excluded. + * The hash key is bound in LONG_MAX to be used for Xarray key. + */ +#define MULTIPLIER 37 +static unsigned long get_ctl_id_hash(const struct snd_ctl_elem_id *id) +{ + unsigned long h; + const unsigned char *p; + + h = id->iface; + h = MULTIPLIER * h + id->device; + h = MULTIPLIER * h + id->subdevice; + for (p = id->name; *p; p++) + h = MULTIPLIER * h + *p; + h = MULTIPLIER * h + id->index; + h &= LONG_MAX; + return h; +} + +/* add hash entries to numid and ctl xarray tables */ +static void add_hash_entries(struct snd_card *card, + struct snd_kcontrol *kcontrol) +{ + struct snd_ctl_elem_id id = kcontrol->id; + int i; + + xa_store_range(&card->ctl_numids, kcontrol->id.numid, + kcontrol->id.numid + kcontrol->count - 1, + kcontrol, GFP_KERNEL); + + for (i = 0; i < kcontrol->count; i++) { + id.index = kcontrol->id.index + i; + if (xa_insert(&card->ctl_hash, get_ctl_id_hash(&id), + kcontrol, GFP_KERNEL)) { + /* skip hash for this entry, noting we had collision */ + card->ctl_hash_collision = true; + dev_dbg(card->dev, "ctl_hash collision %d:%s:%d\n", + id.iface, id.name, id.index); + } + } +} + +/* remove hash entries that have been added */ +static void remove_hash_entries(struct snd_card *card, + struct snd_kcontrol *kcontrol) +{ + struct snd_ctl_elem_id id = kcontrol->id; + struct snd_kcontrol *matched; + unsigned long h; + int i; + + for (i = 0; i < kcontrol->count; i++) { + xa_erase(&card->ctl_numids, id.numid); + h = get_ctl_id_hash(&id); + matched = xa_load(&card->ctl_hash, h); + if (matched && (matched == kcontrol || + elem_id_matches(matched, &id))) + xa_erase(&card->ctl_hash, h); + id.index++; + id.numid++; + } +} +#else /* CONFIG_SND_CTL_FAST_LOOKUP */ +static inline void add_hash_entries(struct snd_card *card, + struct snd_kcontrol *kcontrol) +{ +} +static inline void remove_hash_entries(struct snd_card *card, + struct snd_kcontrol *kcontrol) +{ +} +#endif /* CONFIG_SND_CTL_FAST_LOOKUP */ + enum snd_ctl_add_mode { CTL_ADD_EXCLUSIVE, CTL_REPLACE, CTL_ADD_ON_REPLACE, }; @@ -408,6 +495,8 @@ static int __snd_ctl_add_replace(struct snd_card *card, kcontrol->id.numid = card->last_numid + 1; card->last_numid += kcontrol->count; + add_hash_entries(card, kcontrol); + for (idx = 0; idx < kcontrol->count; idx++) snd_ctl_notify_one(card, SNDRV_CTL_EVENT_MASK_ADD, kcontrol, idx); @@ -479,6 +568,26 @@ int snd_ctl_replace(struct snd_card *card, struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL(snd_ctl_replace); +static int __snd_ctl_remove(struct snd_card *card, + struct snd_kcontrol *kcontrol, + bool remove_hash) +{ + unsigned int idx; + + if (snd_BUG_ON(!card || !kcontrol)) + return -EINVAL; + list_del(&kcontrol->list); + + if (remove_hash) + remove_hash_entries(card, kcontrol); + + card->controls_count -= kcontrol->count; + for (idx = 0; idx < kcontrol->count; idx++) + snd_ctl_notify_one(card, SNDRV_CTL_EVENT_MASK_REMOVE, kcontrol, idx); + snd_ctl_free_one(kcontrol); + return 0; +} + /** * snd_ctl_remove - remove the control from the card and release it * @card: the card instance @@ -492,16 +601,7 @@ EXPORT_SYMBOL(snd_ctl_replace); */ int snd_ctl_remove(struct snd_card *card, struct snd_kcontrol *kcontrol) { - unsigned int idx; - - if (snd_BUG_ON(!card || !kcontrol)) - return -EINVAL; - list_del(&kcontrol->list); - card->controls_count -= kcontrol->count; - for (idx = 0; idx < kcontrol->count; idx++) - snd_ctl_notify_one(card, SNDRV_CTL_EVENT_MASK_REMOVE, kcontrol, idx); - snd_ctl_free_one(kcontrol); - return 0; + return __snd_ctl_remove(card, kcontrol, true); } EXPORT_SYMBOL(snd_ctl_remove); @@ -642,14 +742,30 @@ int snd_ctl_rename_id(struct snd_card *card, struct snd_ctl_elem_id *src_id, up_write(&card->controls_rwsem); return -ENOENT; } + remove_hash_entries(card, kctl); kctl->id = *dst_id; kctl->id.numid = card->last_numid + 1; card->last_numid += kctl->count; + add_hash_entries(card, kctl); up_write(&card->controls_rwsem); return 0; } EXPORT_SYMBOL(snd_ctl_rename_id); +#ifndef CONFIG_SND_CTL_FAST_LOOKUP +static struct snd_kcontrol * +snd_ctl_find_numid_slow(struct snd_card *card, unsigned int numid) +{ + struct snd_kcontrol *kctl; + + list_for_each_entry(kctl, &card->controls, list) { + if (kctl->id.numid <= numid && kctl->id.numid + kctl->count > numid) + return kctl; + } + return NULL; +} +#endif /* !CONFIG_SND_CTL_FAST_LOOKUP */ + /** * snd_ctl_find_numid - find the control instance with the given number-id * @card: the card instance @@ -665,15 +781,13 @@ EXPORT_SYMBOL(snd_ctl_rename_id); */ struct snd_kcontrol *snd_ctl_find_numid(struct snd_card *card, unsigned int numid) { - struct snd_kcontrol *kctl; - if (snd_BUG_ON(!card || !numid)) return NULL; - list_for_each_entry(kctl, &card->controls, list) { - if (kctl->id.numid <= numid && kctl->id.numid + kctl->count > numid) - return kctl; - } - return NULL; +#ifdef CONFIG_SND_CTL_FAST_LOOKUP + return xa_load(&card->ctl_numids, numid); +#else + return snd_ctl_find_numid_slow(card, numid); +#endif } EXPORT_SYMBOL(snd_ctl_find_numid); @@ -699,21 +813,18 @@ struct snd_kcontrol *snd_ctl_find_id(struct snd_card *card, return NULL; if (id->numid != 0) return snd_ctl_find_numid(card, id->numid); - list_for_each_entry(kctl, &card->controls, list) { - if (kctl->id.iface != id->iface) - continue; - if (kctl->id.device != id->device) - continue; - if (kctl->id.subdevice != id->subdevice) - continue; - if (strncmp(kctl->id.name, id->name, sizeof(kctl->id.name))) - continue; - if (kctl->id.index > id->index) - continue; - if (kctl->id.index + kctl->count <= id->index) - continue; +#ifdef CONFIG_SND_CTL_FAST_LOOKUP + kctl = xa_load(&card->ctl_hash, get_ctl_id_hash(id)); + if (kctl && elem_id_matches(kctl, id)) return kctl; - } + if (!card->ctl_hash_collision) + return NULL; /* we can rely on only hash table */ +#endif + /* no matching in hash table - try all as the last resort */ + list_for_each_entry(kctl, &card->controls, list) + if (elem_id_matches(kctl, id)) + return kctl; + return NULL; } EXPORT_SYMBOL(snd_ctl_find_id); @@ -2195,8 +2306,13 @@ static int snd_ctl_dev_free(struct snd_device *device) down_write(&card->controls_rwsem); while (!list_empty(&card->controls)) { control = snd_kcontrol(card->controls.next); - snd_ctl_remove(card, control); + __snd_ctl_remove(card, control, false); } + +#ifdef CONFIG_SND_CTL_FAST_LOOKUP + xa_destroy(&card->ctl_numids); + xa_destroy(&card->ctl_hash); +#endif up_write(&card->controls_rwsem); put_device(&card->ctl_dev); return 0; diff --git a/sound/core/init.c b/sound/core/init.c index 726a8353201f..1870aee7b64f 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -310,6 +310,10 @@ static int snd_card_init(struct snd_card *card, struct device *parent, rwlock_init(&card->ctl_files_rwlock); INIT_LIST_HEAD(&card->controls); INIT_LIST_HEAD(&card->ctl_files); +#ifdef CONFIG_SND_CTL_FAST_LOOKUP + xa_init(&card->ctl_numids); + xa_init(&card->ctl_hash); +#endif spin_lock_init(&card->files_lock); INIT_LIST_HEAD(&card->files_list); mutex_init(&card->memory_mutex); -- cgit v1.2.3 From 2c7463d070c4e49ff850322d4f0321e97b0b6740 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 9 Jun 2022 14:02:16 +0200 Subject: ASoC: topology: Drop superfluous check of CONFIG_SND_CTL_VALIDATION The compiler must be clever enough to optimize out for the no-op when CONFIG_SND_CTL_VALIDATION is disabled. Let's drop the superfluous check. Link: https://lore.kernel.org/r/20220609120219.3937-2-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/soc/soc-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 3f9d314fba16..b101db85446f 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -535,7 +535,7 @@ static int soc_tplg_kcontrol_bind_io(struct snd_soc_tplg_ctl_hdr *hdr, * return an -EINVAL error and prevent the card from * being configured. */ - if (IS_ENABLED(CONFIG_SND_CTL_VALIDATION) && sbe->max > 512) + if (sbe->max > 512) k->access |= SNDRV_CTL_ELEM_ACCESS_SKIP_CHECK; ext_ops = tplg->bytes_ext_ops; -- cgit v1.2.3 From 1b7ec5143c34f167266fa21245d99bacb4db4aa6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 9 Jun 2022 14:02:17 +0200 Subject: ALSA: control: Rename CONFIG_SND_CTL_VALIDATION to CONFIG_SND_CTL_DEBUG The purpose of CONFIG_SND_CTL_VALIDATION is rather to enable the debugging feature for the control API. The validation is only a part of it. Let's rename it to be more explicit and intuitive. While we're at it, let's advertise, give more comment to recommend this feature for development in the kconfig help text. Link: https://lore.kernel.org/r/20220609120219.3937-3-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/Kconfig | 17 +++++++++++------ sound/core/control.c | 4 ++-- 2 files changed, 13 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/core/Kconfig b/sound/core/Kconfig index 25b2434e4556..5289bb29131b 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -188,14 +188,19 @@ config SND_PCM_XRUN_DEBUG sound clicking when system is loaded, it may help to determine the process or driver which causes the scheduling gaps. -config SND_CTL_VALIDATION - bool "Perform sanity-checks for each control element access" +config SND_CTL_DEBUG + bool "Enable debugging feature for control API" depends on SND_DEBUG help - Say Y to enable the additional validation of each control element - access, including sanity-checks like whether the values returned - from the driver are in the proper ranges or the check of the invalid - access at out-of-array areas. + Say Y to enable the debugging feature for ALSA control API. + It performs the additional sanity-checks for each control element + read access, such as whether the values returned from the driver + are in the proper ranges or the check of the invalid access at + out-of-array areas. The error is printed when the driver gives + such unexpected values. + When you develop a driver that deals with control elements, it's + strongly recommended to try this one once and verify whether you see + any relevant errors or not. config SND_JACK_INJECTION_DEBUG bool "Sound jack injection interface via debugfs" diff --git a/sound/core/control.c b/sound/core/control.c index 6a8fd9933f06..1401522ce552 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -966,7 +966,7 @@ static const unsigned int value_sizes[] = { [SNDRV_CTL_ELEM_TYPE_INTEGER64] = sizeof(long long), }; -#ifdef CONFIG_SND_CTL_VALIDATION +#ifdef CONFIG_SND_CTL_DEBUG /* fill the remaining snd_ctl_elem_value data with the given pattern */ static void fill_remaining_elem_value(struct snd_ctl_elem_value *control, struct snd_ctl_elem_info *info, @@ -1188,7 +1188,7 @@ static int snd_ctl_elem_read(struct snd_card *card, snd_ctl_build_ioff(&control->id, kctl, index_offset); -#ifdef CONFIG_SND_CTL_VALIDATION +#ifdef CONFIG_SND_CTL_DEBUG /* info is needed only for validation */ memset(&info, 0, sizeof(info)); info.id = control->id; -- cgit v1.2.3 From 4e54316ad2485dedf8570fc2afa6fa6ce32db207 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 9 Jun 2022 14:02:18 +0200 Subject: ALSA: control: Drop superfluous ifdef CONFIG_SND_CTL_DEBUG Compilers should be smart enough to optimize out the dead functions, so we don't need to define ugly dummy functions with ifdef. Link: https://lore.kernel.org/r/20220609120219.3937-4-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/control.c | 16 ---------------- 1 file changed, 16 deletions(-) (limited to 'sound') diff --git a/sound/core/control.c b/sound/core/control.c index 1401522ce552..559398891eb9 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -966,7 +966,6 @@ static const unsigned int value_sizes[] = { [SNDRV_CTL_ELEM_TYPE_INTEGER64] = sizeof(long long), }; -#ifdef CONFIG_SND_CTL_DEBUG /* fill the remaining snd_ctl_elem_value data with the given pattern */ static void fill_remaining_elem_value(struct snd_ctl_elem_value *control, struct snd_ctl_elem_info *info, @@ -1078,21 +1077,6 @@ static int sanity_check_elem_value(struct snd_card *card, return ret; } -#else -static inline void fill_remaining_elem_value(struct snd_ctl_elem_value *control, - struct snd_ctl_elem_info *info, - u32 pattern) -{ -} - -static inline int sanity_check_elem_value(struct snd_card *card, - struct snd_ctl_elem_value *control, - struct snd_ctl_elem_info *info, - u32 pattern) -{ - return 0; -} -#endif static int __snd_ctl_elem_info(struct snd_card *card, struct snd_kcontrol *kctl, -- cgit v1.2.3 From f5e829f92a494a0b66d309497bab4e9d10d4ce3e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 9 Jun 2022 14:02:19 +0200 Subject: ALSA: control: Add input validation This patch adds a new feature to enable the validation of input data to control elements in the ALSA core side. When CONFIG_SND_CTL_INPUT_VALIDATION is set, ALSA core verifies whether the each input value via control API is in the defined ranges, also checks whether it's aligned to the defined steps. If an invalid value is detected, ALSA core returns -EINVAL error immediately without passing further to the driver's callback. So this is a kind of hardening for (badly written) drivers that have no proper error checks, at the cost of a slight performance overhead. Technically seen, this reuses a part of the existing validation code for CONFIG_SND_CTL_DEBUG case with a slight modification to suppress error prints for the input validation. Link: https://lore.kernel.org/r/20220609120219.3937-5-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/Kconfig | 10 ++++++++ sound/core/control.c | 69 +++++++++++++++++++++++++++++++++++++--------------- 2 files changed, 59 insertions(+), 20 deletions(-) (limited to 'sound') diff --git a/sound/core/Kconfig b/sound/core/Kconfig index 5289bb29131b..12990d9a4dff 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -188,6 +188,16 @@ config SND_PCM_XRUN_DEBUG sound clicking when system is loaded, it may help to determine the process or driver which causes the scheduling gaps. +config SND_CTL_INPUT_VALIDATION + bool "Validate input data to control API" + help + Say Y to enable the additional validation for the input data to + each control element, including the value range checks. + An error is returned from ALSA core for invalid inputs without + passing to the driver. This is a kind of hardening for drivers + that have no proper error checks, at the cost of a slight + performance overhead. + config SND_CTL_DEBUG bool "Enable debugging feature for control API" depends on SND_DEBUG diff --git a/sound/core/control.c b/sound/core/control.c index 559398891eb9..fa04a9233155 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -982,7 +982,7 @@ static void fill_remaining_elem_value(struct snd_ctl_elem_value *control, static int sanity_check_int_value(struct snd_card *card, const struct snd_ctl_elem_value *control, const struct snd_ctl_elem_info *info, - int i) + int i, bool print_error) { long long lval, lmin, lmax, lstep; u64 rem; @@ -1016,21 +1016,23 @@ static int sanity_check_int_value(struct snd_card *card, } if (lval < lmin || lval > lmax) { - dev_err(card->dev, - "control %i:%i:%i:%s:%i: value out of range %lld (%lld/%lld) at count %i\n", - control->id.iface, control->id.device, - control->id.subdevice, control->id.name, - control->id.index, lval, lmin, lmax, i); + if (print_error) + dev_err(card->dev, + "control %i:%i:%i:%s:%i: value out of range %lld (%lld/%lld) at count %i\n", + control->id.iface, control->id.device, + control->id.subdevice, control->id.name, + control->id.index, lval, lmin, lmax, i); return -EINVAL; } if (lstep) { div64_u64_rem(lval, lstep, &rem); if (rem) { - dev_err(card->dev, - "control %i:%i:%i:%s:%i: unaligned value %lld (step %lld) at count %i\n", - control->id.iface, control->id.device, - control->id.subdevice, control->id.name, - control->id.index, lval, lstep, i); + if (print_error) + dev_err(card->dev, + "control %i:%i:%i:%s:%i: unaligned value %lld (step %lld) at count %i\n", + control->id.iface, control->id.device, + control->id.subdevice, control->id.name, + control->id.index, lval, lstep, i); return -EINVAL; } } @@ -1038,15 +1040,13 @@ static int sanity_check_int_value(struct snd_card *card, return 0; } -/* perform sanity checks to the given snd_ctl_elem_value object */ -static int sanity_check_elem_value(struct snd_card *card, - const struct snd_ctl_elem_value *control, - const struct snd_ctl_elem_info *info, - u32 pattern) +/* check whether the all input values are valid for the given elem value */ +static int sanity_check_input_values(struct snd_card *card, + const struct snd_ctl_elem_value *control, + const struct snd_ctl_elem_info *info, + bool print_error) { - size_t offset; - int i, ret = 0; - u32 *p; + int i, ret; switch (info->type) { case SNDRV_CTL_ELEM_TYPE_BOOLEAN: @@ -1054,7 +1054,8 @@ static int sanity_check_elem_value(struct snd_card *card, case SNDRV_CTL_ELEM_TYPE_INTEGER64: case SNDRV_CTL_ELEM_TYPE_ENUMERATED: for (i = 0; i < info->count; i++) { - ret = sanity_check_int_value(card, control, info, i); + ret = sanity_check_int_value(card, control, info, i, + print_error); if (ret < 0) return ret; } @@ -1063,6 +1064,23 @@ static int sanity_check_elem_value(struct snd_card *card, break; } + return 0; +} + +/* perform sanity checks to the given snd_ctl_elem_value object */ +static int sanity_check_elem_value(struct snd_card *card, + const struct snd_ctl_elem_value *control, + const struct snd_ctl_elem_info *info, + u32 pattern) +{ + size_t offset; + int ret; + u32 *p; + + ret = sanity_check_input_values(card, control, info, true); + if (ret < 0) + return ret; + /* check whether the remaining area kept untouched */ offset = value_sizes[info->type] * info->count; offset = DIV_ROUND_UP(offset, sizeof(u32)); @@ -1249,6 +1267,17 @@ static int snd_ctl_elem_write(struct snd_card *card, struct snd_ctl_file *file, snd_ctl_build_ioff(&control->id, kctl, index_offset); result = snd_power_ref_and_wait(card); + /* validate input values */ + if (IS_ENABLED(CONFIG_SND_CTL_INPUT_VALIDATION) && !result) { + struct snd_ctl_elem_info info; + + memset(&info, 0, sizeof(info)); + info.id = control->id; + result = __snd_ctl_elem_info(card, kctl, &info, NULL); + if (!result) + result = sanity_check_input_values(card, control, &info, + false); + } if (!result) result = kctl->put(kctl, control); snd_power_unref(card); -- cgit v1.2.3 From 5983a8a4a4dc13b5f192212a5e744eb303cd65c2 Mon Sep 17 00:00:00 2001 From: Sameer Pujar Date: Wed, 15 Jun 2022 13:34:37 +0530 Subject: ASoC: tegra: Fix clock DAI format on Tegra210 I2S reset failures are seen on Tegra210 and later platforms. This indicates absence of I2S bit clock, which is required to perform the reset operation. Following failures are seen with I2S based tests on Tegra210 and later: tegra210-i2s 2901100.i2s: timeout: failed to reset I2S for playback tegra210-i2s 2901100.i2s: ASoC: PRE_PMU: I2S2 RX event failed: -110 tegra210-i2s 2901100.i2s: timeout: failed to reset I2S for capture tegra210-i2s 2901100.i2s: ASoC: PRE_PMU: I2S2 TX event failed: -110 The commit d92ad6633fa7 ("ASoC: tegra: Update to use set_fmt_new callback") regressed I2S functionality on Tegra platforms. Basically it flipped clock provider and consumer DAI formats. This configures Tegra I2S in consumer mode by default now and there is none to provide bit clock during loopback tests. The external codec based tests also fail because both Tegra I2S and codec I2S get configured in consumer mode. ASoC core flips the DAI format before calling set_fmt() for CPU DAIs. This is negated in above commit. Fix this by swapping SND_SOC_DAIFMT_BC_FC and SND_SOC_DAIFMT_BP_FP switch cases. Fixes: d92ad6633fa7 ("ASoC: tegra: Update to use set_fmt_new callback") Signed-off-by: Sameer Pujar Cc: Charles Keepax Link: https://lore.kernel.org/r/1655280277-4701-1-git-send-email-spujar@nvidia.com Signed-off-by: Mark Brown --- sound/soc/tegra/tegra210_i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra210_i2s.c b/sound/soc/tegra/tegra210_i2s.c index a28945895466..01c76ba36e1a 100644 --- a/sound/soc/tegra/tegra210_i2s.c +++ b/sound/soc/tegra/tegra210_i2s.c @@ -215,10 +215,10 @@ static int tegra210_i2s_set_fmt(struct snd_soc_dai *dai, mask = I2S_CTRL_MASTER_EN_MASK; switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { - case SND_SOC_DAIFMT_BP_FP: + case SND_SOC_DAIFMT_BC_FC: val = 0; break; - case SND_SOC_DAIFMT_BC_FC: + case SND_SOC_DAIFMT_BP_FP: val = I2S_CTRL_MASTER_EN; break; default: -- cgit v1.2.3 From 4edf738d4c7989c315e37d4d61e34c94557b6ed2 Mon Sep 17 00:00:00 2001 From: Sameer Pujar Date: Wed, 15 Jun 2022 10:08:34 +0530 Subject: ASoC: tegra: Fix MBDRC bypass mode check MBDRC supports different modes of operation. There is no configuration required for bypass mode. The hw_params() call does not filter bypass mode correctly and it leads to following Smatch static checker warning: sound/soc/tegra/tegra210_mbdrc.c:778 tegra210_mbdrc_hw_params() warn: bitwise AND condition is false here Fix this condition by using proper mode mask and just return for bypass mode. Reported-by: Dan Carpenter Fixes: 7358a803c778 ("ASoC: tegra: Add Tegra210 based OPE driver") Signed-off-by: Sameer Pujar Link: https://lore.kernel.org/r/1655267914-24702-1-git-send-email-spujar@nvidia.com Signed-off-by: Mark Brown --- sound/soc/tegra/tegra210_mbdrc.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra210_mbdrc.c b/sound/soc/tegra/tegra210_mbdrc.c index 7d9da33a9951..d786daa6aba6 100644 --- a/sound/soc/tegra/tegra210_mbdrc.c +++ b/sound/soc/tegra/tegra210_mbdrc.c @@ -775,7 +775,9 @@ int tegra210_mbdrc_hw_params(struct snd_soc_component *cmpnt) regmap_read(ope->mbdrc_regmap, TEGRA210_MBDRC_CFG, &val); - if (val & TEGRA210_MBDRC_CFG_MBDRC_MODE_BYPASS) + val &= TEGRA210_MBDRC_CFG_MBDRC_MODE_MASK; + + if (val == TEGRA210_MBDRC_CFG_MBDRC_MODE_BYPASS) return 0; for (i = 0; i < MBDRC_NUM_BAND; i++) { -- cgit v1.2.3 From ab222a4aaecfafece1516c775143e1cb9eb31612 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Wed, 15 Jun 2022 16:43:47 +0800 Subject: ASoC: SOC: Intel: introduce cl_init callback MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The code loader init sequences are different between versions of Intel platforms. Have a cl_init callback allows us to reuse the common code. No function changed. Signed-off-by: Bard Liao Reviewed-by: Péter Ujfalusi Reviewed-by: Rander Wang Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/20220615084348.3489-2-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/apl.c | 1 + sound/soc/sof/intel/cnl.c | 2 ++ sound/soc/sof/intel/hda-loader.c | 15 ++++++++++++--- sound/soc/sof/intel/hda.h | 1 + sound/soc/sof/intel/icl.c | 1 + sound/soc/sof/intel/shim.h | 1 + sound/soc/sof/intel/tgl.c | 4 ++++ 7 files changed, 22 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/intel/apl.c b/sound/soc/sof/intel/apl.c index 0cea280a6d2d..084c245a9522 100644 --- a/sound/soc/sof/intel/apl.c +++ b/sound/soc/sof/intel/apl.c @@ -101,6 +101,7 @@ const struct sof_intel_dsp_desc apl_chip_info = { .ssp_base_offset = APL_SSP_BASE_OFFSET, .quirks = SOF_INTEL_PROCEN_FMT_QUIRK, .check_ipc_irq = hda_dsp_check_ipc_irq, + .cl_init = cl_dsp_init, .hw_ip_version = SOF_INTEL_CAVS_1_5_PLUS, }; EXPORT_SYMBOL_NS(apl_chip_info, SND_SOC_SOF_INTEL_HDA_COMMON); diff --git a/sound/soc/sof/intel/cnl.c b/sound/soc/sof/intel/cnl.c index cd6e5f8a5eb4..ccf46fcd6c9a 100644 --- a/sound/soc/sof/intel/cnl.c +++ b/sound/soc/sof/intel/cnl.c @@ -401,6 +401,7 @@ const struct sof_intel_dsp_desc cnl_chip_info = { .sdw_alh_base = SDW_ALH_BASE, .check_sdw_irq = hda_common_check_sdw_irq, .check_ipc_irq = hda_dsp_check_ipc_irq, + .cl_init = cl_dsp_init, .hw_ip_version = SOF_INTEL_CAVS_1_8, }; EXPORT_SYMBOL_NS(cnl_chip_info, SND_SOC_SOF_INTEL_HDA_COMMON); @@ -430,6 +431,7 @@ const struct sof_intel_dsp_desc jsl_chip_info = { .sdw_alh_base = SDW_ALH_BASE, .check_sdw_irq = hda_common_check_sdw_irq, .check_ipc_irq = hda_dsp_check_ipc_irq, + .cl_init = cl_dsp_init, .hw_ip_version = SOF_INTEL_CAVS_2_0, }; EXPORT_SYMBOL_NS(jsl_chip_info, SND_SOC_SOF_INTEL_HDA_COMMON); diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c index d3ec5996a9a3..9e99f376f2b3 100644 --- a/sound/soc/sof/intel/hda-loader.c +++ b/sound/soc/sof/intel/hda-loader.c @@ -99,7 +99,7 @@ out_put: * power on all host managed cores and only unstall/run the boot core to boot the * DSP then turn off all non boot cores (if any) is powered on. */ -static int cl_dsp_init(struct snd_sof_dev *sdev, int stream_tag, bool imr_boot) +int cl_dsp_init(struct snd_sof_dev *sdev, int stream_tag, bool imr_boot) { struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata; const struct sof_intel_dsp_desc *chip = hda->desc; @@ -369,9 +369,15 @@ int hda_dsp_cl_boot_firmware_iccmax(struct snd_sof_dev *sdev) static int hda_dsp_boot_imr(struct snd_sof_dev *sdev) { + const struct sof_intel_dsp_desc *chip_info; int ret; - ret = cl_dsp_init(sdev, 0, true); + chip_info = get_chip_info(sdev->pdata); + if (chip_info->cl_init) + ret = chip_info->cl_init(sdev, 0, true); + else + ret = -EINVAL; + if (!ret) hda_sdw_process_wakeen(sdev); @@ -430,7 +436,10 @@ int hda_dsp_cl_boot_firmware(struct snd_sof_dev *sdev) "Attempting iteration %d of Core En/ROM load...\n", i); hda->boot_iteration = i + 1; - ret = cl_dsp_init(sdev, hext_stream->hstream.stream_tag, false); + if (chip_info->cl_init) + ret = chip_info->cl_init(sdev, hext_stream->hstream.stream_tag, false); + else + ret = -EINVAL; /* don't retry anymore if successful */ if (!ret) diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index f4e4cd7d7406..8b7f3c07d478 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -602,6 +602,7 @@ struct hdac_ext_stream *hda_cl_stream_prepare(struct snd_sof_dev *sdev, unsigned int direction); int hda_cl_cleanup(struct snd_sof_dev *sdev, struct snd_dma_buffer *dmab, struct hdac_ext_stream *hext_stream); +int cl_dsp_init(struct snd_sof_dev *sdev, int stream_tag, bool imr_boot); #define HDA_CL_STREAM_FORMAT 0x40 /* pre and post fw run ops */ diff --git a/sound/soc/sof/intel/icl.c b/sound/soc/sof/intel/icl.c index f19517dffd62..4e37b7fe0627 100644 --- a/sound/soc/sof/intel/icl.c +++ b/sound/soc/sof/intel/icl.c @@ -152,6 +152,7 @@ const struct sof_intel_dsp_desc icl_chip_info = { .sdw_alh_base = SDW_ALH_BASE, .check_sdw_irq = hda_common_check_sdw_irq, .check_ipc_irq = hda_dsp_check_ipc_irq, + .cl_init = cl_dsp_init, .hw_ip_version = SOF_INTEL_CAVS_2_0, }; EXPORT_SYMBOL_NS(icl_chip_info, SND_SOC_SOF_INTEL_HDA_COMMON); diff --git a/sound/soc/sof/intel/shim.h b/sound/soc/sof/intel/shim.h index 1fd7b485d821..371991fa474f 100644 --- a/sound/soc/sof/intel/shim.h +++ b/sound/soc/sof/intel/shim.h @@ -185,6 +185,7 @@ struct sof_intel_dsp_desc { enum sof_intel_hw_ip_version hw_ip_version; bool (*check_sdw_irq)(struct snd_sof_dev *sdev); bool (*check_ipc_irq)(struct snd_sof_dev *sdev); + int (*cl_init)(struct snd_sof_dev *sdev, int stream_tag, bool imr_boot); }; extern struct snd_sof_dsp_ops sof_tng_ops; diff --git a/sound/soc/sof/intel/tgl.c b/sound/soc/sof/intel/tgl.c index dcad7c382de6..6dfb4786c782 100644 --- a/sound/soc/sof/intel/tgl.c +++ b/sound/soc/sof/intel/tgl.c @@ -127,6 +127,7 @@ const struct sof_intel_dsp_desc tgl_chip_info = { .sdw_alh_base = SDW_ALH_BASE, .check_sdw_irq = hda_common_check_sdw_irq, .check_ipc_irq = hda_dsp_check_ipc_irq, + .cl_init = cl_dsp_init, .hw_ip_version = SOF_INTEL_CAVS_2_5, }; EXPORT_SYMBOL_NS(tgl_chip_info, SND_SOC_SOF_INTEL_HDA_COMMON); @@ -149,6 +150,7 @@ const struct sof_intel_dsp_desc tglh_chip_info = { .sdw_alh_base = SDW_ALH_BASE, .check_sdw_irq = hda_common_check_sdw_irq, .check_ipc_irq = hda_dsp_check_ipc_irq, + .cl_init = cl_dsp_init, .hw_ip_version = SOF_INTEL_CAVS_2_5, }; EXPORT_SYMBOL_NS(tglh_chip_info, SND_SOC_SOF_INTEL_HDA_COMMON); @@ -171,6 +173,7 @@ const struct sof_intel_dsp_desc ehl_chip_info = { .sdw_alh_base = SDW_ALH_BASE, .check_sdw_irq = hda_common_check_sdw_irq, .check_ipc_irq = hda_dsp_check_ipc_irq, + .cl_init = cl_dsp_init, .hw_ip_version = SOF_INTEL_CAVS_2_5, }; EXPORT_SYMBOL_NS(ehl_chip_info, SND_SOC_SOF_INTEL_HDA_COMMON); @@ -193,6 +196,7 @@ const struct sof_intel_dsp_desc adls_chip_info = { .sdw_alh_base = SDW_ALH_BASE, .check_sdw_irq = hda_common_check_sdw_irq, .check_ipc_irq = hda_dsp_check_ipc_irq, + .cl_init = cl_dsp_init, .hw_ip_version = SOF_INTEL_CAVS_2_5, }; EXPORT_SYMBOL_NS(adls_chip_info, SND_SOC_SOF_INTEL_HDA_COMMON); -- cgit v1.2.3 From 064520e8aeaa2569f6504a50a37ac801b73656bc Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Wed, 15 Jun 2022 16:43:48 +0800 Subject: ASoC: SOF: Intel: Add support for MeteorLake (MTL) MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add platform abstraction for the Meteor Lake platform. This platform has significant differences compared to the TGL/ADL generation: it relies on new hardware using the code name 'ACE' and only supports the INTEL_IPC4 protocol and firmware architecture based on the Zephyr RTOS Co-developed-by: Ranjani Sridharan Signed-off-by: Ranjani Sridharan Signed-off-by: Bard Liao Reviewed-by: Péter Ujfalusi Reviewed-by: Rander Wang Link: https://lore.kernel.org/r/20220615084348.3489-3-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/Kconfig | 16 + sound/soc/sof/intel/Makefile | 4 +- sound/soc/sof/intel/hda.h | 3 + sound/soc/sof/intel/mtl.c | 800 ++++++++++++++++++++++++++++++++++++++++++ sound/soc/sof/intel/mtl.h | 76 ++++ sound/soc/sof/intel/pci-mtl.c | 71 ++++ sound/soc/sof/intel/shim.h | 1 + 7 files changed, 970 insertions(+), 1 deletion(-) create mode 100644 sound/soc/sof/intel/mtl.c create mode 100644 sound/soc/sof/intel/mtl.h create mode 100644 sound/soc/sof/intel/pci-mtl.c (limited to 'sound') diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index 80cdc3788bbe..3f54678e810b 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -221,6 +221,22 @@ config SND_SOC_SOF_ALDERLAKE Say Y if you have such a device. If unsure select "N". +config SND_SOC_SOF_INTEL_MTL + tristate + select SND_SOC_SOF_HDA_COMMON + select SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE + select SND_SOC_SOF_INTEL_IPC4 + +config SND_SOC_SOF_METEORLAKE + tristate "SOF support for Meteorlake" + default SND_SOC_SOF_PCI + select SND_SOC_SOF_INTEL_MTL + help + This adds support for Sound Open Firmware for Intel(R) platforms + using the Meteorlake processors. + Say Y if you have such a device. + If unsure select "N". + config SND_SOC_SOF_HDA_COMMON tristate select SND_SOC_SOF_INTEL_COMMON diff --git a/sound/soc/sof/intel/Makefile b/sound/soc/sof/intel/Makefile index b9d51dc39ffa..a079159bb2f0 100644 --- a/sound/soc/sof/intel/Makefile +++ b/sound/soc/sof/intel/Makefile @@ -6,7 +6,7 @@ snd-sof-acpi-intel-bdw-objs := bdw.o snd-sof-intel-hda-common-objs := hda.o hda-loader.o hda-stream.o hda-trace.o \ hda-dsp.o hda-ipc.o hda-ctrl.o hda-pcm.o \ hda-dai.o hda-bus.o \ - apl.o cnl.o tgl.o icl.o hda-common-ops.o + apl.o cnl.o tgl.o icl.o mtl.o hda-common-ops.o snd-sof-intel-hda-common-$(CONFIG_SND_SOC_SOF_HDA_PROBES) += hda-probes.o snd-sof-intel-hda-objs := hda-codec.o @@ -24,9 +24,11 @@ snd-sof-pci-intel-apl-objs := pci-apl.o snd-sof-pci-intel-cnl-objs := pci-cnl.o snd-sof-pci-intel-icl-objs := pci-icl.o snd-sof-pci-intel-tgl-objs := pci-tgl.o +snd-sof-pci-intel-mtl-objs := pci-mtl.o obj-$(CONFIG_SND_SOC_SOF_MERRIFIELD) += snd-sof-pci-intel-tng.o obj-$(CONFIG_SND_SOC_SOF_INTEL_APL) += snd-sof-pci-intel-apl.o obj-$(CONFIG_SND_SOC_SOF_INTEL_CNL) += snd-sof-pci-intel-cnl.o obj-$(CONFIG_SND_SOC_SOF_INTEL_ICL) += snd-sof-pci-intel-icl.o obj-$(CONFIG_SND_SOC_SOF_INTEL_TGL) += snd-sof-pci-intel-tgl.o +obj-$(CONFIG_SND_SOC_SOF_INTEL_MTL) += snd-sof-pci-intel-mtl.o diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 8b7f3c07d478..a3118499e34f 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -714,6 +714,8 @@ extern struct snd_sof_dsp_ops sof_tgl_ops; int sof_tgl_ops_init(struct snd_sof_dev *sdev); extern struct snd_sof_dsp_ops sof_icl_ops; int sof_icl_ops_init(struct snd_sof_dev *sdev); +extern struct snd_sof_dsp_ops sof_mtl_ops; +int sof_mtl_ops_init(struct snd_sof_dev *sdev); extern const struct sof_intel_dsp_desc apl_chip_info; extern const struct sof_intel_dsp_desc cnl_chip_info; @@ -723,6 +725,7 @@ extern const struct sof_intel_dsp_desc tglh_chip_info; extern const struct sof_intel_dsp_desc ehl_chip_info; extern const struct sof_intel_dsp_desc jsl_chip_info; extern const struct sof_intel_dsp_desc adls_chip_info; +extern const struct sof_intel_dsp_desc mtl_chip_info; /* Probes support */ #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_PROBES) diff --git a/sound/soc/sof/intel/mtl.c b/sound/soc/sof/intel/mtl.c new file mode 100644 index 000000000000..37be77beb415 --- /dev/null +++ b/sound/soc/sof/intel/mtl.c @@ -0,0 +1,800 @@ +// SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) +// +// Copyright(c) 2022 Intel Corporation. All rights reserved. +// +// Authors: Ranjani Sridharan +// + +/* + * Hardware interface for audio DSP on Meteorlake. + */ + +#include +#include +#include "../ipc4-priv.h" +#include "../ops.h" +#include "hda.h" +#include "hda-ipc.h" +#include "../sof-audio.h" +#include "mtl.h" + +static const struct snd_sof_debugfs_map mtl_dsp_debugfs[] = { + {"hda", HDA_DSP_HDA_BAR, 0, 0x4000, SOF_DEBUGFS_ACCESS_ALWAYS}, + {"pp", HDA_DSP_PP_BAR, 0, 0x1000, SOF_DEBUGFS_ACCESS_ALWAYS}, + {"dsp", HDA_DSP_BAR, 0, 0x10000, SOF_DEBUGFS_ACCESS_ALWAYS}, +}; + +static void mtl_ipc_host_done(struct snd_sof_dev *sdev) +{ + /* + * clear busy interrupt to tell dsp controller this interrupt has been accepted, + * not trigger it again + */ + snd_sof_dsp_update_bits_forced(sdev, HDA_DSP_BAR, MTL_DSP_REG_HFIPCXTDR, + MTL_DSP_REG_HFIPCXTDR_BUSY, MTL_DSP_REG_HFIPCXTDR_BUSY); + /* + * clear busy bit to ack dsp the msg has been processed and send reply msg to dsp + */ + snd_sof_dsp_update_bits_forced(sdev, HDA_DSP_BAR, MTL_DSP_REG_HFIPCXTDA, + MTL_DSP_REG_HFIPCXTDA_BUSY, 0); +} + +static void mtl_ipc_dsp_done(struct snd_sof_dev *sdev) +{ + /* + * set DONE bit - tell DSP we have received the reply msg from DSP, and processed it, + * don't send more reply to host + */ + snd_sof_dsp_update_bits_forced(sdev, HDA_DSP_BAR, MTL_DSP_REG_HFIPCXIDA, + MTL_DSP_REG_HFIPCXIDA_DONE, MTL_DSP_REG_HFIPCXIDA_DONE); + + /* unmask Done interrupt */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, MTL_DSP_REG_HFIPCXCTL, + MTL_DSP_REG_HFIPCXCTL_DONE, MTL_DSP_REG_HFIPCXCTL_DONE); +} + +/* Check if an IPC IRQ occurred */ +static bool mtl_dsp_check_ipc_irq(struct snd_sof_dev *sdev) +{ + u32 irq_status; + u32 hfintipptr; + + /* read Interrupt IP Pointer */ + hfintipptr = snd_sof_dsp_read(sdev, HDA_DSP_BAR, MTL_HFINTIPPTR) & MTL_HFINTIPPTR_PTR_MASK; + irq_status = snd_sof_dsp_read(sdev, HDA_DSP_BAR, hfintipptr + MTL_DSP_IRQSTS); + + dev_vdbg(sdev->dev, "irq handler: irq_status:0x%x\n", irq_status); + + if (irq_status != U32_MAX && (irq_status & MTL_DSP_IRQSTS_IPC)) + return true; + + return false; +} + +/* Check if an SDW IRQ occurred */ +static bool mtl_dsp_check_sdw_irq(struct snd_sof_dev *sdev) +{ + u32 irq_status; + u32 hfintipptr; + + /* read Interrupt IP Pointer */ + hfintipptr = snd_sof_dsp_read(sdev, HDA_DSP_BAR, MTL_HFINTIPPTR) & MTL_HFINTIPPTR_PTR_MASK; + irq_status = snd_sof_dsp_read(sdev, HDA_DSP_BAR, hfintipptr + MTL_DSP_IRQSTS); + + if (irq_status != U32_MAX && (irq_status & MTL_DSP_IRQSTS_SDW)) + return true; + + return false; +} + +static int mtl_ipc_send_msg(struct snd_sof_dev *sdev, struct snd_sof_ipc_msg *msg) +{ + struct sof_ipc4_msg *msg_data = msg->msg_data; + + /* send the message via mailbox */ + if (msg_data->data_size) + sof_mailbox_write(sdev, sdev->host_box.offset, msg_data->data_ptr, + msg_data->data_size); + + snd_sof_dsp_write(sdev, HDA_DSP_BAR, MTL_DSP_REG_HFIPCXIDDY, + msg_data->extension); + snd_sof_dsp_write(sdev, HDA_DSP_BAR, MTL_DSP_REG_HFIPCXIDR, + msg_data->primary | MTL_DSP_REG_HFIPCXIDR_BUSY); + + return 0; +} + +static void mtl_enable_ipc_interrupts(struct snd_sof_dev *sdev) +{ + struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata; + const struct sof_intel_dsp_desc *chip = hda->desc; + + /* enable IPC DONE and BUSY interrupts */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, chip->ipc_ctl, + MTL_DSP_REG_HFIPCXCTL_BUSY | MTL_DSP_REG_HFIPCXCTL_DONE, + MTL_DSP_REG_HFIPCXCTL_BUSY | MTL_DSP_REG_HFIPCXCTL_DONE); +} + +static void mtl_disable_ipc_interrupts(struct snd_sof_dev *sdev) +{ + struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata; + const struct sof_intel_dsp_desc *chip = hda->desc; + + /* disable IPC DONE and BUSY interrupts */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, chip->ipc_ctl, + MTL_DSP_REG_HFIPCXCTL_BUSY | MTL_DSP_REG_HFIPCXCTL_DONE, 0); +} + +static int mtl_enable_interrupts(struct snd_sof_dev *sdev) +{ + u32 hfintipptr; + u32 irqinten; + u32 host_ipc; + u32 hipcie; + int ret; + + /* read Interrupt IP Pointer */ + hfintipptr = snd_sof_dsp_read(sdev, HDA_DSP_BAR, MTL_HFINTIPPTR) & MTL_HFINTIPPTR_PTR_MASK; + + /* Enable Host IPC and SOUNDWIRE */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, hfintipptr, + MTL_IRQ_INTEN_L_HOST_IPC_MASK | MTL_IRQ_INTEN_L_SOUNDWIRE_MASK, + MTL_IRQ_INTEN_L_HOST_IPC_MASK | MTL_IRQ_INTEN_L_SOUNDWIRE_MASK); + + /* check if operation was successful */ + host_ipc = MTL_IRQ_INTEN_L_HOST_IPC_MASK | MTL_IRQ_INTEN_L_SOUNDWIRE_MASK; + irqinten = snd_sof_dsp_read(sdev, HDA_DSP_BAR, hfintipptr); + ret = snd_sof_dsp_read_poll_timeout(sdev, HDA_DSP_BAR, hfintipptr, irqinten, + (irqinten & host_ipc) == host_ipc, + HDA_DSP_REG_POLL_INTERVAL_US, HDA_DSP_RESET_TIMEOUT_US); + if (ret < 0) { + dev_err(sdev->dev, "failed to enable Host IPC and/or SOUNDWIRE\n"); + return ret; + } + + /* Set Host IPC interrupt enable */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, MTL_DSP_REG_HfHIPCIE, + MTL_DSP_REG_HfHIPCIE_IE_MASK, MTL_DSP_REG_HfHIPCIE_IE_MASK); + + /* check if operation was successful */ + host_ipc = MTL_DSP_REG_HfHIPCIE_IE_MASK; + hipcie = snd_sof_dsp_read(sdev, HDA_DSP_BAR, MTL_DSP_REG_HfHIPCIE); + ret = snd_sof_dsp_read_poll_timeout(sdev, HDA_DSP_BAR, MTL_DSP_REG_HfHIPCIE, hipcie, + (hipcie & host_ipc) == host_ipc, + HDA_DSP_REG_POLL_INTERVAL_US, HDA_DSP_RESET_TIMEOUT_US); + if (ret < 0) { + dev_err(sdev->dev, "failed to set Host IPC interrupt enable\n"); + return ret; + } + + snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, MTL_DSP_REG_HfSNDWIE, + MTL_DSP_REG_HfSNDWIE_IE_MASK, MTL_DSP_REG_HfSNDWIE_IE_MASK); + host_ipc = MTL_DSP_REG_HfSNDWIE_IE_MASK; + hipcie = snd_sof_dsp_read(sdev, HDA_DSP_BAR, MTL_DSP_REG_HfSNDWIE); + ret = snd_sof_dsp_read_poll_timeout(sdev, HDA_DSP_BAR, MTL_DSP_REG_HfSNDWIE, hipcie, + (hipcie & host_ipc) == host_ipc, + HDA_DSP_REG_POLL_INTERVAL_US, HDA_DSP_RESET_TIMEOUT_US); + if (ret < 0) + dev_err(sdev->dev, "failed to set SoundWire IPC interrupt enable\n"); + + return ret; +} + +static int mtl_disable_interrupts(struct snd_sof_dev *sdev) +{ + u32 hfintipptr; + u32 irqinten; + u32 host_ipc; + u32 hipcie; + int ret1; + int ret; + + /* read Interrupt IP Pointer */ + hfintipptr = snd_sof_dsp_read(sdev, HDA_DSP_BAR, MTL_HFINTIPPTR) & MTL_HFINTIPPTR_PTR_MASK; + + /* Disable Host IPC and SOUNDWIRE */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, hfintipptr, + MTL_IRQ_INTEN_L_HOST_IPC_MASK | MTL_IRQ_INTEN_L_SOUNDWIRE_MASK, 0); + + /* check if operation was successful */ + host_ipc = MTL_IRQ_INTEN_L_HOST_IPC_MASK | MTL_IRQ_INTEN_L_SOUNDWIRE_MASK; + irqinten = snd_sof_dsp_read(sdev, HDA_DSP_BAR, hfintipptr); + ret = snd_sof_dsp_read_poll_timeout(sdev, HDA_DSP_BAR, hfintipptr, irqinten, + (irqinten & host_ipc) == 0, + HDA_DSP_REG_POLL_INTERVAL_US, HDA_DSP_RESET_TIMEOUT_US); + /* Continue to disable other interrupts when error happens */ + if (ret < 0) + dev_err(sdev->dev, "failed to disable Host IPC and SoundWire\n"); + + /* Set Host IPC interrupt disable */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, MTL_DSP_REG_HfHIPCIE, + MTL_DSP_REG_HfHIPCIE_IE_MASK, 0); + + /* check if operation was successful */ + host_ipc = MTL_DSP_REG_HfHIPCIE_IE_MASK; + hipcie = snd_sof_dsp_read(sdev, HDA_DSP_BAR, MTL_DSP_REG_HfHIPCIE); + ret1 = snd_sof_dsp_read_poll_timeout(sdev, HDA_DSP_BAR, MTL_DSP_REG_HfHIPCIE, hipcie, + (hipcie & host_ipc) == 0, + HDA_DSP_REG_POLL_INTERVAL_US, + HDA_DSP_RESET_TIMEOUT_US); + if (ret1 < 0) { + dev_err(sdev->dev, "failed to set Host IPC interrupt disable\n"); + if (!ret) + ret = ret1; + } + + /* Set SoundWire IPC interrupt disable */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, MTL_DSP_REG_HfSNDWIE, + MTL_DSP_REG_HfSNDWIE_IE_MASK, 0); + host_ipc = MTL_DSP_REG_HfSNDWIE_IE_MASK; + hipcie = snd_sof_dsp_read(sdev, HDA_DSP_BAR, MTL_DSP_REG_HfSNDWIE); + ret1 = snd_sof_dsp_read_poll_timeout(sdev, HDA_DSP_BAR, MTL_DSP_REG_HfSNDWIE, hipcie, + (hipcie & host_ipc) == 0, + HDA_DSP_REG_POLL_INTERVAL_US, + HDA_DSP_RESET_TIMEOUT_US); + if (ret1 < 0) { + dev_err(sdev->dev, "failed to set SoundWire IPC interrupt disable\n"); + if (!ret) + ret = ret1; + } + + return ret; +} + +/* pre fw run operations */ +static int mtl_dsp_pre_fw_run(struct snd_sof_dev *sdev) +{ + u32 dsphfpwrsts; + u32 dsphfdsscs; + u32 cpa; + u32 pgs; + int ret; + + /* Set the DSP subsystem power on */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, MTL_HFDSSCS, + MTL_HFDSSCS_SPA_MASK, MTL_HFDSSCS_SPA_MASK); + + /* Wait for unstable CPA read (1 then 0 then 1) just after setting SPA bit */ + usleep_range(1000, 1010); + + /* poll with timeout to check if operation successful */ + cpa = MTL_HFDSSCS_CPA_MASK; + dsphfdsscs = snd_sof_dsp_read(sdev, HDA_DSP_BAR, MTL_HFDSSCS); + ret = snd_sof_dsp_read_poll_timeout(sdev, HDA_DSP_BAR, MTL_HFDSSCS, dsphfdsscs, + (dsphfdsscs & cpa) == cpa, HDA_DSP_REG_POLL_INTERVAL_US, + HDA_DSP_RESET_TIMEOUT_US); + if (ret < 0) { + dev_err(sdev->dev, "failed to enable DSP subsystem\n"); + return ret; + } + + /* Power up gated-DSP-0 domain in order to access the DSP shim register block. */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, MTL_HFPWRCTL, + MTL_HFPWRCTL_WPDSPHPXPG, MTL_HFPWRCTL_WPDSPHPXPG); + + usleep_range(1000, 1010); + + /* poll with timeout to check if operation successful */ + pgs = MTL_HFPWRSTS_DSPHPXPGS_MASK; + dsphfpwrsts = snd_sof_dsp_read(sdev, HDA_DSP_BAR, MTL_HFPWRSTS); + ret = snd_sof_dsp_read_poll_timeout(sdev, HDA_DSP_BAR, MTL_HFPWRSTS, dsphfpwrsts, + (dsphfpwrsts & pgs) == pgs, + HDA_DSP_REG_POLL_INTERVAL_US, + HDA_DSP_RESET_TIMEOUT_US); + if (ret < 0) + dev_err(sdev->dev, "failed to power up gated DSP domain\n"); + + /* make sure SoundWire is not power-gated */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, MTL_HFPWRCTL, + MTL_HfPWRCTL_WPIOXPG(1), MTL_HfPWRCTL_WPIOXPG(1)); + return ret; +} + +static int mtl_dsp_post_fw_run(struct snd_sof_dev *sdev) +{ + int ret; + + if (sdev->first_boot) { + struct sof_intel_hda_dev *hdev = sdev->pdata->hw_pdata; + + ret = hda_sdw_startup(sdev); + if (ret < 0) { + dev_err(sdev->dev, "could not startup SoundWire links\n"); + return ret; + } + + /* Check if IMR boot is usable */ + if (!sof_debug_check_flag(SOF_DBG_IGNORE_D3_PERSISTENT)) + hdev->imrboot_supported = true; + } + + hda_sdw_int_enable(sdev, true); + return 0; +} + +static void mtl_dsp_dump(struct snd_sof_dev *sdev, u32 flags) +{ + char *level = (flags & SOF_DBG_DUMP_OPTIONAL) ? KERN_DEBUG : KERN_ERR; + u32 romdbgsts; + u32 romdbgerr; + u32 fwsts; + u32 fwlec; + + fwsts = snd_sof_dsp_read(sdev, HDA_DSP_BAR, MTL_DSP_ROM_STS); + fwlec = snd_sof_dsp_read(sdev, HDA_DSP_BAR, MTL_DSP_ROM_ERROR); + romdbgsts = snd_sof_dsp_read(sdev, HDA_DSP_BAR, MTL_DSP_REG_HFFLGPXQWY); + romdbgerr = snd_sof_dsp_read(sdev, HDA_DSP_BAR, MTL_DSP_REG_HFFLGPXQWY_ERROR); + + dev_err(sdev->dev, "ROM status: %#x, ROM error: %#x\n", fwsts, fwlec); + dev_err(sdev->dev, "ROM debug status: %#x, ROM debug error: %#x\n", romdbgsts, + romdbgerr); + romdbgsts = snd_sof_dsp_read(sdev, HDA_DSP_BAR, MTL_DSP_REG_HFFLGPXQWY + 0x8 * 3); + dev_printk(level, sdev->dev, "ROM feature bit%s enabled\n", + romdbgsts & BIT(24) ? "" : " not"); +} + +static bool mtl_dsp_primary_core_is_enabled(struct snd_sof_dev *sdev) +{ + int val; + + val = snd_sof_dsp_read(sdev, HDA_DSP_BAR, MTL_DSP2CXCTL_PRIMARY_CORE); + if (val != U32_MAX && val & MTL_DSP2CXCTL_PRIMARY_CORE_CPA_MASK) + return true; + + return false; +} + +static int mtl_dsp_core_power_up(struct snd_sof_dev *sdev, int core) +{ + unsigned int cpa; + u32 dspcxctl; + int ret; + + /* Only the primary core can be powered up by the host */ + if (core != SOF_DSP_PRIMARY_CORE || mtl_dsp_primary_core_is_enabled(sdev)) + return 0; + + /* Program the owner of the IP & shim registers (10: Host CPU) */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, MTL_DSP2CXCTL_PRIMARY_CORE, + MTL_DSP2CXCTL_PRIMARY_CORE_OSEL, + 0x2 << MTL_DSP2CXCTL_PRIMARY_CORE_OSEL_SHIFT); + + /* enable SPA bit */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, MTL_DSP2CXCTL_PRIMARY_CORE, + MTL_DSP2CXCTL_PRIMARY_CORE_SPA_MASK, + MTL_DSP2CXCTL_PRIMARY_CORE_SPA_MASK); + + /* Wait for unstable CPA read (1 then 0 then 1) just after setting SPA bit */ + usleep_range(1000, 1010); + + /* poll with timeout to check if operation successful */ + cpa = MTL_DSP2CXCTL_PRIMARY_CORE_CPA_MASK; + ret = snd_sof_dsp_read_poll_timeout(sdev, HDA_DSP_BAR, MTL_DSP2CXCTL_PRIMARY_CORE, dspcxctl, + (dspcxctl & cpa) == cpa, HDA_DSP_REG_POLL_INTERVAL_US, + HDA_DSP_RESET_TIMEOUT_US); + if (ret < 0) { + dev_err(sdev->dev, "%s: timeout on MTL_DSP2CXCTL_PRIMARY_CORE read\n", + __func__); + return ret; + } + + /* did core power up ? */ + dspcxctl = snd_sof_dsp_read(sdev, HDA_DSP_BAR, MTL_DSP2CXCTL_PRIMARY_CORE); + if ((dspcxctl & MTL_DSP2CXCTL_PRIMARY_CORE_CPA_MASK) + != MTL_DSP2CXCTL_PRIMARY_CORE_CPA_MASK) { + dev_err(sdev->dev, "power up core failed core %d adspcs %#x\n", + core, dspcxctl); + ret = -EIO; + } + + return ret; +} + +static int mtl_dsp_core_power_down(struct snd_sof_dev *sdev, int core) +{ + u32 dspcxctl; + int ret; + + /* Only the primary core can be powered down by the host */ + if (core != SOF_DSP_PRIMARY_CORE || !mtl_dsp_primary_core_is_enabled(sdev)) + return 0; + + /* disable SPA bit */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, MTL_DSP2CXCTL_PRIMARY_CORE, + MTL_DSP2CXCTL_PRIMARY_CORE_SPA_MASK, 0); + + /* Wait for unstable CPA read (1 then 0 then 1) just after setting SPA bit */ + usleep_range(1000, 1010); + + ret = snd_sof_dsp_read_poll_timeout(sdev, HDA_DSP_BAR, MTL_DSP2CXCTL_PRIMARY_CORE, dspcxctl, + !(dspcxctl & MTL_DSP2CXCTL_PRIMARY_CORE_CPA_MASK), + HDA_DSP_REG_POLL_INTERVAL_US, + HDA_DSP_PD_TIMEOUT * USEC_PER_MSEC); + if (ret < 0) + dev_err(sdev->dev, "failed to power down primary core\n"); + + return ret; +} + +static int mtl_dsp_cl_init(struct snd_sof_dev *sdev, int stream_tag, bool imr_boot) +{ + struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata; + const struct sof_intel_dsp_desc *chip = hda->desc; + unsigned int status; + u32 ipc_hdr; + int ret; + + /* step 1: purge FW request */ + ipc_hdr = chip->ipc_req_mask | HDA_DSP_ROM_IPC_CONTROL; + if (!imr_boot) + ipc_hdr |= HDA_DSP_ROM_IPC_PURGE_FW | ((stream_tag - 1) << 9); + + snd_sof_dsp_write(sdev, HDA_DSP_BAR, chip->ipc_req, ipc_hdr); + + /* step 2: power up primary core */ + ret = mtl_dsp_core_power_up(sdev, SOF_DSP_PRIMARY_CORE); + if (ret < 0) { + if (hda->boot_iteration == HDA_FW_BOOT_ATTEMPTS) + dev_err(sdev->dev, "dsp core 0/1 power up failed\n"); + goto err; + } + + dev_dbg(sdev->dev, "Primary core power up successful\n"); + + /* step 3: wait for IPC DONE bit from ROM */ + ret = snd_sof_dsp_read_poll_timeout(sdev, HDA_DSP_BAR, chip->ipc_ack, status, + ((status & chip->ipc_ack_mask) == chip->ipc_ack_mask), + HDA_DSP_REG_POLL_INTERVAL_US, MTL_DSP_PURGE_TIMEOUT_US); + if (ret < 0) { + if (hda->boot_iteration == HDA_FW_BOOT_ATTEMPTS) + dev_err(sdev->dev, "timeout waiting for purge IPC done\n"); + goto err; + } + + /* set DONE bit to clear the reply IPC message */ + snd_sof_dsp_update_bits_forced(sdev, HDA_DSP_BAR, chip->ipc_ack, chip->ipc_ack_mask, + chip->ipc_ack_mask); + + /* step 4: enable interrupts */ + ret = mtl_enable_interrupts(sdev); + if (ret < 0) { + if (hda->boot_iteration == HDA_FW_BOOT_ATTEMPTS) + dev_err(sdev->dev, "%s: failed to enable interrupts\n", __func__); + goto err; + } + + mtl_enable_ipc_interrupts(sdev); + + /* + * ACE workaround: don't wait for ROM INIT. + * The platform cannot catch ROM_INIT_DONE because of a very short + * timing window. Follow the recommendations and skip this part. + */ + + return 0; + +err: + snd_sof_dsp_dbg_dump(sdev, "MTL DSP init fail", 0); + mtl_dsp_core_power_down(sdev, SOF_DSP_PRIMARY_CORE); + return ret; +} + +static irqreturn_t mtl_ipc_irq_thread(int irq, void *context) +{ + struct sof_ipc4_msg notification_data = {{ 0 }}; + struct snd_sof_dev *sdev = context; + bool ipc_irq = false; + u32 hipcida; + u32 hipctdr; + + hipcida = snd_sof_dsp_read(sdev, HDA_DSP_BAR, MTL_DSP_REG_HFIPCXIDA); + + /* reply message from DSP */ + if (hipcida & MTL_DSP_REG_HFIPCXIDA_DONE) { + /* DSP received the message */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, MTL_DSP_REG_HFIPCXCTL, + MTL_DSP_REG_HFIPCXCTL_DONE, 0); + + mtl_ipc_dsp_done(sdev); + + ipc_irq = true; + } + + hipctdr = snd_sof_dsp_read(sdev, HDA_DSP_BAR, MTL_DSP_REG_HFIPCXTDR); + if (hipctdr & MTL_DSP_REG_HFIPCXTDR_BUSY) { + /* Message from DSP (reply or notification) */ + u32 extension = snd_sof_dsp_read(sdev, HDA_DSP_BAR, MTL_DSP_REG_HFIPCXTDDY); + u32 primary = hipctdr & MTL_DSP_REG_HFIPCXTDR_MSG_MASK; + + /* + * ACE fw sends a new fw ipc message to host to + * notify the status of the last host ipc message + */ + if (primary & SOF_IPC4_MSG_DIR_MASK) { + /* Reply received */ + struct sof_ipc4_msg *data = sdev->ipc->msg.reply_data; + + data->primary = primary; + data->extension = extension; + + spin_lock_irq(&sdev->ipc_lock); + + snd_sof_ipc_get_reply(sdev); + snd_sof_ipc_reply(sdev, data->primary); + + spin_unlock_irq(&sdev->ipc_lock); + } else { + /* Notification received */ + notification_data.primary = primary; + notification_data.extension = extension; + + sdev->ipc->msg.rx_data = ¬ification_data; + snd_sof_ipc_msgs_rx(sdev); + sdev->ipc->msg.rx_data = NULL; + } + + mtl_ipc_host_done(sdev); + + ipc_irq = true; + } + + if (!ipc_irq) { + /* This interrupt is not shared so no need to return IRQ_NONE. */ + dev_dbg_ratelimited(sdev->dev, "%s nothing to do in IPC IRQ thread\n", + __func__); + } + + return IRQ_HANDLED; +} + +static int mtl_dsp_ipc_get_mailbox_offset(struct snd_sof_dev *sdev) +{ + return MTL_DSP_MBOX_UPLINK_OFFSET; +} + +static int mtl_dsp_ipc_get_window_offset(struct snd_sof_dev *sdev, u32 id) +{ + return MTL_SRAM_WINDOW_OFFSET(id); +} + +static int mtl_suspend(struct snd_sof_dev *sdev, bool runtime_suspend) +{ + struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata; + const struct sof_intel_dsp_desc *chip = hda->desc; +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) + struct hdac_bus *bus = sof_to_bus(sdev); +#endif + u32 dsphfdsscs; + u32 cpa; + int ret; + int i; + + mtl_disable_ipc_interrupts(sdev); + ret = mtl_disable_interrupts(sdev); + if (ret) + return ret; + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) + hda_codec_jack_wake_enable(sdev, runtime_suspend); + /* power down all hda link */ + snd_hdac_ext_bus_link_power_down_all(bus); +#endif + snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, MTL_HFPWRCTL, + MTL_HFPWRCTL_WPDSPHPXPG, 0); + + /* Set the DSP subsystem power down */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, MTL_HFDSSCS, + MTL_HFDSSCS_SPA_MASK, 0); + + /* Wait for unstable CPA read (1 then 0 then 1) just after setting SPA bit */ + usleep_range(1000, 1010); + + /* poll with timeout to check if operation successful */ + cpa = MTL_HFDSSCS_CPA_MASK; + dsphfdsscs = snd_sof_dsp_read(sdev, HDA_DSP_BAR, MTL_HFDSSCS); + ret = snd_sof_dsp_read_poll_timeout(sdev, HDA_DSP_BAR, MTL_HFDSSCS, dsphfdsscs, + (dsphfdsscs & cpa) == 0, HDA_DSP_REG_POLL_INTERVAL_US, + HDA_DSP_RESET_TIMEOUT_US); + if (ret < 0) + dev_err(sdev->dev, "failed to disable DSP subsystem\n"); + + /* reset ref counts for all cores */ + for (i = 0; i < chip->cores_num; i++) + sdev->dsp_core_ref_count[i] = 0; + + /* TODO: need to reset controller? */ + + /* display codec can be powered off after link reset */ + hda_codec_i915_display_power(sdev, false); + + return 0; +} + +static int mtl_dsp_suspend(struct snd_sof_dev *sdev, u32 target_state) +{ + const struct sof_dsp_power_state target_dsp_state = { + .state = target_state, + .substate = target_state == SOF_DSP_PM_D0 ? + SOF_HDA_DSP_PM_D0I3 : 0, + }; + int ret; + + ret = mtl_suspend(sdev, false); + if (ret < 0) + return ret; + + return snd_sof_dsp_set_power_state(sdev, &target_dsp_state); +} + +static int mtl_dsp_runtime_suspend(struct snd_sof_dev *sdev) +{ + const struct sof_dsp_power_state target_state = { + .state = SOF_DSP_PM_D3, + }; + int ret; + + ret = mtl_suspend(sdev, true); + if (ret < 0) + return ret; + + return snd_sof_dsp_set_power_state(sdev, &target_state); +} + +static int mtl_resume(struct snd_sof_dev *sdev, bool runtime_resume) +{ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) + struct hdac_bus *bus = sof_to_bus(sdev); + struct hdac_ext_link *hlink = NULL; +#endif + + /* display codec must be powered before link reset */ + hda_codec_i915_display_power(sdev, true); + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) + /* check jack status */ + if (runtime_resume) { + hda_codec_jack_wake_enable(sdev, false); + if (sdev->system_suspend_target == SOF_SUSPEND_NONE) + hda_codec_jack_check(sdev); + } + + /* turn off the links that were off before suspend */ + list_for_each_entry(hlink, &bus->hlink_list, list) { + if (!hlink->ref_count) + snd_hdac_ext_bus_link_power_down(hlink); + } + + /* check dma status and clean up CORB/RIRB buffers */ + if (!bus->cmd_dma_state) + snd_hdac_bus_stop_cmd_io(bus); +#endif + + return 0; +} + +static int mtl_dsp_resume(struct snd_sof_dev *sdev) +{ + const struct sof_dsp_power_state target_state = { + .state = SOF_DSP_PM_D0, + .substate = SOF_HDA_DSP_PM_D0I0, + }; + int ret; + + ret = mtl_resume(sdev, false); + if (ret < 0) + return ret; + + return snd_sof_dsp_set_power_state(sdev, &target_state); +} + +static int mtl_dsp_runtime_resume(struct snd_sof_dev *sdev) +{ + const struct sof_dsp_power_state target_state = { + .state = SOF_DSP_PM_D0, + }; + int ret; + + ret = mtl_resume(sdev, true); + if (ret < 0) + return ret; + + return snd_sof_dsp_set_power_state(sdev, &target_state); +} + +static void mtl_ipc_dump(struct snd_sof_dev *sdev) +{ + u32 hipcctl; + u32 hipcida; + u32 hipctdr; + + /* read IPC status */ + hipcida = snd_sof_dsp_read(sdev, HDA_DSP_BAR, MTL_DSP_REG_HFIPCXIDA); + hipcctl = snd_sof_dsp_read(sdev, HDA_DSP_BAR, MTL_DSP_REG_HFIPCXCTL); + hipctdr = snd_sof_dsp_read(sdev, HDA_DSP_BAR, MTL_DSP_REG_HFIPCXTDR); + + /* dump the IPC regs */ + /* TODO: parse the raw msg */ + dev_err(sdev->dev, + "error: host status 0x%8.8x dsp status 0x%8.8x mask 0x%8.8x\n", + hipcida, hipctdr, hipcctl); +} + +/* Meteorlake ops */ +struct snd_sof_dsp_ops sof_mtl_ops; +EXPORT_SYMBOL_NS(sof_mtl_ops, SND_SOC_SOF_INTEL_HDA_COMMON); + +int sof_mtl_ops_init(struct snd_sof_dev *sdev) +{ + struct sof_ipc4_fw_data *ipc4_data; + + /* common defaults */ + memcpy(&sof_mtl_ops, &sof_hda_common_ops, sizeof(struct snd_sof_dsp_ops)); + + /* shutdown */ + sof_mtl_ops.shutdown = hda_dsp_shutdown; + + /* doorbell */ + sof_mtl_ops.irq_thread = mtl_ipc_irq_thread; + + /* ipc */ + sof_mtl_ops.send_msg = mtl_ipc_send_msg; + sof_mtl_ops.get_mailbox_offset = mtl_dsp_ipc_get_mailbox_offset; + sof_mtl_ops.get_window_offset = mtl_dsp_ipc_get_window_offset; + + /* debug */ + sof_mtl_ops.debug_map = mtl_dsp_debugfs; + sof_mtl_ops.debug_map_count = ARRAY_SIZE(mtl_dsp_debugfs); + sof_mtl_ops.dbg_dump = mtl_dsp_dump; + sof_mtl_ops.ipc_dump = mtl_ipc_dump; + + /* pre/post fw run */ + sof_mtl_ops.pre_fw_run = mtl_dsp_pre_fw_run; + sof_mtl_ops.post_fw_run = mtl_dsp_post_fw_run; + + /* parse platform specific extended manifest */ + sof_mtl_ops.parse_platform_ext_manifest = NULL; + + /* dsp core get/put */ + /* TODO: add core_get and core_put */ + + /* PM */ + sof_mtl_ops.suspend = mtl_dsp_suspend; + sof_mtl_ops.resume = mtl_dsp_resume; + sof_mtl_ops.runtime_suspend = mtl_dsp_runtime_suspend; + sof_mtl_ops.runtime_resume = mtl_dsp_runtime_resume; + + sdev->private = devm_kzalloc(sdev->dev, sizeof(struct sof_ipc4_fw_data), GFP_KERNEL); + if (!sdev->private) + return -ENOMEM; + + ipc4_data = sdev->private; + ipc4_data->manifest_fw_hdr_offset = SOF_MAN4_FW_HDR_OFFSET; + + /* set DAI ops */ + hda_set_dai_drv_ops(sdev, &sof_mtl_ops); + + return 0; +}; +EXPORT_SYMBOL_NS(sof_mtl_ops_init, SND_SOC_SOF_INTEL_HDA_COMMON); + +const struct sof_intel_dsp_desc mtl_chip_info = { + .cores_num = 3, + .init_core_mask = BIT(0), + .host_managed_cores_mask = BIT(0), + .ipc_req = MTL_DSP_REG_HFIPCXIDR, + .ipc_req_mask = MTL_DSP_REG_HFIPCXIDR_BUSY, + .ipc_ack = MTL_DSP_REG_HFIPCXIDA, + .ipc_ack_mask = MTL_DSP_REG_HFIPCXIDA_DONE, + .ipc_ctl = MTL_DSP_REG_HFIPCXCTL, + .rom_status_reg = MTL_DSP_ROM_STS, + .rom_init_timeout = 300, + .ssp_count = ICL_SSP_COUNT, + .ssp_base_offset = CNL_SSP_BASE_OFFSET, + .sdw_shim_base = SDW_SHIM_BASE_ACE, + .sdw_alh_base = SDW_ALH_BASE_ACE, + .check_sdw_irq = mtl_dsp_check_sdw_irq, + .check_ipc_irq = mtl_dsp_check_ipc_irq, + .cl_init = mtl_dsp_cl_init, + .hw_ip_version = SOF_INTEL_ACE_1_0, +}; +EXPORT_SYMBOL_NS(mtl_chip_info, SND_SOC_SOF_INTEL_HDA_COMMON); diff --git a/sound/soc/sof/intel/mtl.h b/sound/soc/sof/intel/mtl.h new file mode 100644 index 000000000000..788bf0e3ea87 --- /dev/null +++ b/sound/soc/sof/intel/mtl.h @@ -0,0 +1,76 @@ +/* SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) */ +/* + * This file is provided under a dual BSD/GPLv2 license. When using or + * redistributing this file, you may do so under either license. + * + * Copyright(c) 2020-2022 Intel Corporation. All rights reserved. + */ + +/* DSP Registers */ +#define MTL_HFDSSCS 0x1000 +#define MTL_HFDSSCS_SPA_MASK BIT(16) +#define MTL_HFDSSCS_CPA_MASK BIT(24) +#define MTL_HFSNDWIE 0x114C +#define MTL_HFPWRCTL 0x1D18 +#define MTL_HfPWRCTL_WPIOXPG(x) BIT((x) + 8) +#define MTL_HFPWRCTL_WPDSPHPXPG BIT(0) +#define MTL_HFPWRSTS 0x1D1C +#define MTL_HFPWRSTS_DSPHPXPGS_MASK BIT(0) +#define MTL_HFINTIPPTR 0x1108 +#define MTL_IRQ_INTEN_L_HOST_IPC_MASK BIT(0) +#define MTL_IRQ_INTEN_L_SOUNDWIRE_MASK BIT(6) +#define MTL_HFINTIPPTR_PTR_MASK GENMASK(20, 0) + +#define MTL_DSP2CXCAP_PRIMARY_CORE 0x178D00 +#define MTL_DSP2CXCTL_PRIMARY_CORE 0x178D04 +#define MTL_DSP2CXCTL_PRIMARY_CORE_SPA_MASK BIT(0) +#define MTL_DSP2CXCTL_PRIMARY_CORE_CPA_MASK BIT(8) +#define MTL_DSP2CXCTL_PRIMARY_CORE_OSEL GENMASK(25, 24) +#define MTL_DSP2CXCTL_PRIMARY_CORE_OSEL_SHIFT 24 + +/* IPC Registers */ +#define MTL_DSP_REG_HFIPCXTDR 0x73200 +#define MTL_DSP_REG_HFIPCXTDR_BUSY BIT(31) +#define MTL_DSP_REG_HFIPCXTDR_MSG_MASK GENMASK(30, 0) +#define MTL_DSP_REG_HFIPCXTDA 0x73204 +#define MTL_DSP_REG_HFIPCXTDA_BUSY BIT(31) +#define MTL_DSP_REG_HFIPCXIDR 0x73210 +#define MTL_DSP_REG_HFIPCXIDR_BUSY BIT(31) +#define MTL_DSP_REG_HFIPCXIDR_MSG_MASK GENMASK(30, 0) +#define MTL_DSP_REG_HFIPCXIDA 0x73214 +#define MTL_DSP_REG_HFIPCXIDA_DONE BIT(31) +#define MTL_DSP_REG_HFIPCXIDA_MSG_MASK GENMASK(30, 0) +#define MTL_DSP_REG_HFIPCXCTL 0x73228 +#define MTL_DSP_REG_HFIPCXCTL_BUSY BIT(0) +#define MTL_DSP_REG_HFIPCXCTL_DONE BIT(1) +#define MTL_DSP_REG_HFIPCXTDDY 0x73300 +#define MTL_DSP_REG_HFIPCXIDDY 0x73380 +#define MTL_DSP_REG_HfHIPCIE 0x1140 +#define MTL_DSP_REG_HfHIPCIE_IE_MASK BIT(0) +#define MTL_DSP_REG_HfSNDWIE 0x114C +#define MTL_DSP_REG_HfSNDWIE_IE_MASK GENMASK(3, 0) + +#define MTL_DSP_IRQSTS 0x20 +#define MTL_DSP_IRQSTS_IPC BIT(0) +#define MTL_DSP_IRQSTS_SDW BIT(6) + +#define MTL_DSP_PURGE_TIMEOUT_US 20000000 /* 20s */ +#define MTL_DSP_REG_POLL_INTERVAL_US 10 /* 10 us */ + +/* Memory windows */ +#define MTL_SRAM_WINDOW_OFFSET(x) (0x180000 + 0x8000 * (x)) + +#define MTL_DSP_MBOX_UPLINK_OFFSET (MTL_SRAM_WINDOW_OFFSET(0) + 0x1000) +#define MTL_DSP_MBOX_UPLINK_SIZE 0x1000 +#define MTL_DSP_MBOX_DOWNLINK_OFFSET MTL_SRAM_WINDOW_OFFSET(1) +#define MTL_DSP_MBOX_DOWNLINK_SIZE 0x1000 + +/* FW registers */ +#define MTL_DSP_ROM_STS MTL_SRAM_WINDOW_OFFSET(0) /* ROM status */ +#define MTL_DSP_ROM_ERROR (MTL_SRAM_WINDOW_OFFSET(0) + 0x4) /* ROM error code */ + +#define MTL_DSP_REG_HFFLGPXQWY 0x163200 /* ROM debug status */ +#define MTL_DSP_REG_HFFLGPXQWY_ERROR 0x163204 /* ROM debug error code */ +#define MTL_DSP_REG_HfIMRIS1 0x162088 +#define MTL_DSP_REG_HfIMRIS1_IU_MASK BIT(0) + diff --git a/sound/soc/sof/intel/pci-mtl.c b/sound/soc/sof/intel/pci-mtl.c new file mode 100644 index 000000000000..899b00d53d64 --- /dev/null +++ b/sound/soc/sof/intel/pci-mtl.c @@ -0,0 +1,71 @@ +// SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) +// +// This file is provided under a dual BSD/GPLv2 license. When using or +// redistributing this file, you may do so under either license. +// +// Copyright(c) 2018-2022 Intel Corporation. All rights reserved. +// +// Author: Ranjani Sridharan +// + +#include +#include +#include +#include +#include +#include "../ops.h" +#include "../sof-pci-dev.h" + +/* platform specific devices */ +#include "hda.h" +#include "mtl.h" + +static const struct sof_dev_desc mtl_desc = { + .use_acpi_target_states = true, + .machines = snd_soc_acpi_intel_mtl_machines, + .alt_machines = snd_soc_acpi_intel_mtl_sdw_machines, + .resindex_lpe_base = 0, + .resindex_pcicfg_base = -1, + .resindex_imr_base = -1, + .irqindex_host_ipc = -1, + .chip_info = &mtl_chip_info, + .ipc_supported_mask = BIT(SOF_INTEL_IPC4), + .ipc_default = SOF_INTEL_IPC4, + .default_fw_path = { + [SOF_INTEL_IPC4] = "intel/sof-ipc4/mtl", + }, + .default_tplg_path = { + [SOF_INTEL_IPC4] = "intel/sof-ace-tplg", + }, + .default_fw_filename = { + [SOF_INTEL_IPC4] = "dsp_basefw.bin", + }, + .nocodec_tplg_filename = "sof-mtl-nocodec.tplg", + .ops = &sof_mtl_ops, + .ops_init = sof_mtl_ops_init, +}; + +/* PCI IDs */ +static const struct pci_device_id sof_pci_ids[] = { + { PCI_DEVICE(0x8086, 0x7E28), /* MTL */ + .driver_data = (unsigned long)&mtl_desc}, + { 0, } +}; +MODULE_DEVICE_TABLE(pci, sof_pci_ids); + +/* pci_driver definition */ +static struct pci_driver snd_sof_pci_intel_mtl_driver = { + .name = "sof-audio-pci-intel-mtl", + .id_table = sof_pci_ids, + .probe = hda_pci_intel_probe, + .remove = sof_pci_remove, + .shutdown = sof_pci_shutdown, + .driver = { + .pm = &sof_pci_pm, + }, +}; +module_pci_driver(snd_sof_pci_intel_mtl_driver); + +MODULE_LICENSE("Dual BSD/GPL"); +MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_HDA_COMMON); +MODULE_IMPORT_NS(SND_SOC_SOF_PCI_DEV); diff --git a/sound/soc/sof/intel/shim.h b/sound/soc/sof/intel/shim.h index 371991fa474f..638159bee864 100644 --- a/sound/soc/sof/intel/shim.h +++ b/sound/soc/sof/intel/shim.h @@ -20,6 +20,7 @@ enum sof_intel_hw_ip_version { SOF_INTEL_CAVS_1_8, /* CannonLake, CometLake, CoffeeLake */ SOF_INTEL_CAVS_2_0, /* IceLake, JasperLake */ SOF_INTEL_CAVS_2_5, /* TigerLake, AlderLake */ + SOF_INTEL_ACE_1_0, /* MeteorLake */ }; /* -- cgit v1.2.3 From b23662406b1b225847b964e4549a5718c45f20d6 Mon Sep 17 00:00:00 2001 From: Li Chen Date: Sun, 22 May 2022 20:27:59 -0700 Subject: ASoC: sunxi: Use {regmap/regmap_field}_{set/clear}_bits helpers Appropriately change calls to {regmap/regmap_field}_update_bits() with {regmap/regmap_field}_set_bits() and {regmap/regmap_field}_clear_bits() for improved readability. Signed-off-by: Li Chen Link: https://lore.kernel.org/r/180eef50e96.cb7c34db60740.8898768158778553647@zohomail.com Signed-off-by: Mark Brown --- sound/soc/sunxi/sun4i-codec.c | 78 +++++++++++++++++-------------------------- 1 file changed, 30 insertions(+), 48 deletions(-) (limited to 'sound') diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index 60712f24ade5..53e3f43816cc 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -250,37 +250,33 @@ struct sun4i_codec { static void sun4i_codec_start_playback(struct sun4i_codec *scodec) { /* Flush TX FIFO */ - regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, - BIT(SUN4I_CODEC_DAC_FIFOC_FIFO_FLUSH), - BIT(SUN4I_CODEC_DAC_FIFOC_FIFO_FLUSH)); + regmap_set_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, + BIT(SUN4I_CODEC_DAC_FIFOC_FIFO_FLUSH)); /* Enable DAC DRQ */ - regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, - BIT(SUN4I_CODEC_DAC_FIFOC_DAC_DRQ_EN), - BIT(SUN4I_CODEC_DAC_FIFOC_DAC_DRQ_EN)); + regmap_set_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, + BIT(SUN4I_CODEC_DAC_FIFOC_DAC_DRQ_EN)); } static void sun4i_codec_stop_playback(struct sun4i_codec *scodec) { /* Disable DAC DRQ */ - regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, - BIT(SUN4I_CODEC_DAC_FIFOC_DAC_DRQ_EN), - 0); + regmap_clear_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, + BIT(SUN4I_CODEC_DAC_FIFOC_DAC_DRQ_EN)); } static void sun4i_codec_start_capture(struct sun4i_codec *scodec) { /* Enable ADC DRQ */ - regmap_field_update_bits(scodec->reg_adc_fifoc, - BIT(SUN4I_CODEC_ADC_FIFOC_ADC_DRQ_EN), - BIT(SUN4I_CODEC_ADC_FIFOC_ADC_DRQ_EN)); + regmap_field_set_bits(scodec->reg_adc_fifoc, + BIT(SUN4I_CODEC_ADC_FIFOC_ADC_DRQ_EN)); } static void sun4i_codec_stop_capture(struct sun4i_codec *scodec) { /* Disable ADC DRQ */ - regmap_field_update_bits(scodec->reg_adc_fifoc, - BIT(SUN4I_CODEC_ADC_FIFOC_ADC_DRQ_EN), 0); + regmap_field_clear_bits(scodec->reg_adc_fifoc, + BIT(SUN4I_CODEC_ADC_FIFOC_ADC_DRQ_EN)); } static int sun4i_codec_trigger(struct snd_pcm_substream *substream, int cmd, @@ -323,8 +319,7 @@ static int sun4i_codec_prepare_capture(struct snd_pcm_substream *substream, /* Flush RX FIFO */ - regmap_field_update_bits(scodec->reg_adc_fifoc, - BIT(SUN4I_CODEC_ADC_FIFOC_FIFO_FLUSH), + regmap_field_set_bits(scodec->reg_adc_fifoc, BIT(SUN4I_CODEC_ADC_FIFOC_FIFO_FLUSH)); @@ -365,8 +360,7 @@ static int sun4i_codec_prepare_playback(struct snd_pcm_substream *substream, u32 val; /* Flush the TX FIFO */ - regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, - BIT(SUN4I_CODEC_DAC_FIFOC_FIFO_FLUSH), + regmap_set_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, BIT(SUN4I_CODEC_DAC_FIFOC_FIFO_FLUSH)); /* Set TX FIFO Empty Trigger Level */ @@ -386,9 +380,8 @@ static int sun4i_codec_prepare_playback(struct snd_pcm_substream *substream, val); /* Send zeros when we have an underrun */ - regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, - BIT(SUN4I_CODEC_DAC_FIFOC_SEND_LASAT), - 0); + regmap_clear_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, + BIT(SUN4I_CODEC_DAC_FIFOC_SEND_LASAT)); return 0; }; @@ -485,33 +478,27 @@ static int sun4i_codec_hw_params_capture(struct sun4i_codec *scodec, /* Set the number of channels we want to use */ if (params_channels(params) == 1) - regmap_field_update_bits(scodec->reg_adc_fifoc, - BIT(SUN4I_CODEC_ADC_FIFOC_MONO_EN), + regmap_field_set_bits(scodec->reg_adc_fifoc, BIT(SUN4I_CODEC_ADC_FIFOC_MONO_EN)); else - regmap_field_update_bits(scodec->reg_adc_fifoc, - BIT(SUN4I_CODEC_ADC_FIFOC_MONO_EN), - 0); + regmap_field_clear_bits(scodec->reg_adc_fifoc, + BIT(SUN4I_CODEC_ADC_FIFOC_MONO_EN)); /* Set the number of sample bits to either 16 or 24 bits */ if (hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS)->min == 32) { - regmap_field_update_bits(scodec->reg_adc_fifoc, - BIT(SUN4I_CODEC_ADC_FIFOC_RX_SAMPLE_BITS), + regmap_field_set_bits(scodec->reg_adc_fifoc, BIT(SUN4I_CODEC_ADC_FIFOC_RX_SAMPLE_BITS)); - regmap_field_update_bits(scodec->reg_adc_fifoc, - BIT(SUN4I_CODEC_ADC_FIFOC_RX_FIFO_MODE), - 0); + regmap_field_clear_bits(scodec->reg_adc_fifoc, + BIT(SUN4I_CODEC_ADC_FIFOC_RX_FIFO_MODE)); scodec->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; } else { - regmap_field_update_bits(scodec->reg_adc_fifoc, - BIT(SUN4I_CODEC_ADC_FIFOC_RX_SAMPLE_BITS), - 0); + regmap_field_clear_bits(scodec->reg_adc_fifoc, + BIT(SUN4I_CODEC_ADC_FIFOC_RX_SAMPLE_BITS)); /* Fill most significant bits with valid data MSB */ - regmap_field_update_bits(scodec->reg_adc_fifoc, - BIT(SUN4I_CODEC_ADC_FIFOC_RX_FIFO_MODE), + regmap_field_set_bits(scodec->reg_adc_fifoc, BIT(SUN4I_CODEC_ADC_FIFOC_RX_FIFO_MODE)); scodec->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; @@ -543,24 +530,20 @@ static int sun4i_codec_hw_params_playback(struct sun4i_codec *scodec, /* Set the number of sample bits to either 16 or 24 bits */ if (hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS)->min == 32) { - regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, - BIT(SUN4I_CODEC_DAC_FIFOC_TX_SAMPLE_BITS), + regmap_set_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, BIT(SUN4I_CODEC_DAC_FIFOC_TX_SAMPLE_BITS)); /* Set TX FIFO mode to padding the LSBs with 0 */ - regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, - BIT(SUN4I_CODEC_DAC_FIFOC_TX_FIFO_MODE), - 0); + regmap_clear_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, + BIT(SUN4I_CODEC_DAC_FIFOC_TX_FIFO_MODE)); scodec->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; } else { - regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, - BIT(SUN4I_CODEC_DAC_FIFOC_TX_SAMPLE_BITS), - 0); + regmap_clear_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, + BIT(SUN4I_CODEC_DAC_FIFOC_TX_SAMPLE_BITS)); /* Set TX FIFO mode to repeat the MSB */ - regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, - BIT(SUN4I_CODEC_DAC_FIFOC_TX_FIFO_MODE), + regmap_set_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, BIT(SUN4I_CODEC_DAC_FIFOC_TX_FIFO_MODE)); scodec->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; @@ -624,8 +607,7 @@ static int sun4i_codec_startup(struct snd_pcm_substream *substream, * Stop issuing DRQ when we have room for less than 16 samples * in our TX FIFO */ - regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, - 3 << SUN4I_CODEC_DAC_FIFOC_DRQ_CLR_CNT, + regmap_set_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, 3 << SUN4I_CODEC_DAC_FIFOC_DRQ_CLR_CNT); return clk_prepare_enable(scodec->clk_module); -- cgit v1.2.3 From 62257638170eee07926c9df5a4c9059ec69a3734 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Wed, 15 Jun 2022 11:19:44 +0300 Subject: ASoC: SOF: mediatek: Fix error code in probe This should return PTR_ERR() instead of IS_ERR(). Fixes: e0100bfd383c ("ASoC: SOF: mediatek: Add mt8186 ipc support") Signed-off-by: Dan Carpenter Acked-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/YqmWIK8sTj578OJP@kili Signed-off-by: Mark Brown --- sound/soc/sof/mediatek/mt8186/mt8186.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sof/mediatek/mt8186/mt8186.c b/sound/soc/sof/mediatek/mt8186/mt8186.c index 3333a0634e29..e006532caf2f 100644 --- a/sound/soc/sof/mediatek/mt8186/mt8186.c +++ b/sound/soc/sof/mediatek/mt8186/mt8186.c @@ -392,7 +392,7 @@ static int mt8186_dsp_probe(struct snd_sof_dev *sdev) PLATFORM_DEVID_NONE, pdev, sizeof(*pdev)); if (IS_ERR(priv->ipc_dev)) { - ret = IS_ERR(priv->ipc_dev); + ret = PTR_ERR(priv->ipc_dev); dev_err(sdev->dev, "failed to create mtk-adsp-ipc device\n"); goto err_adsp_off; } -- cgit v1.2.3 From 7acf970a6fbb3c10bb5979d0dc3ed42b161daf15 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Thu, 16 Jun 2022 07:31:09 +0300 Subject: ASoC: SOF: ipc4-topology: Fix error code in sof_ipc4_volume_put() The sof_ipc4_volume_put() function returns type bool so returning -ENOENT means returning true. Return false instead. Fixes: 955e84fc0b6d ("ASoC: SOF: ipc4-topology: Add control IO ops") Signed-off-by: Dan Carpenter Acked-by: Peter Ujfalusi Link: https://lore.kernel.org/r/YqqyDU5BhOzpRjco@kili Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-control.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sof/ipc4-control.c b/sound/soc/sof/ipc4-control.c index 95ee121dd3cf..0d5a578c3496 100644 --- a/sound/soc/sof/ipc4-control.c +++ b/sound/soc/sof/ipc4-control.c @@ -142,7 +142,7 @@ static bool sof_ipc4_volume_put(struct snd_sof_control *scontrol, if (!widget_found) { dev_err(scomp->dev, "Failed to find widget for kcontrol %s\n", scontrol->name); - return -ENOENT; + return false; } ret = sof_ipc4_set_volume_data(sdev, swidget, scontrol); -- cgit v1.2.3 From 1ec0c91f6d6b21703c17d5e89f32d52feac5887e Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Tue, 14 Jun 2022 19:38:09 +0100 Subject: ASoC: Intel: Skylake: remove redundant re-assignments to pointer array There are two occurrences where the pointer array is being assigned a value that is never read, the pointer gets updated in the next iteration of a loop. These assignments are redundant and can be removed. Cleans up clang scan-build warnings: sound/soc/intel/skylake/skl-topology.c:2953:3: warning: Value stored to 'array' is never read [deadcode.DeadStores] sound/soc/intel/skylake/skl-topology.c:3602:3: warning: Value stored to 'array' is never read [deadcode.DeadStores] Signed-off-by: Colin Ian King Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/20220614183809.163531-1-colin.i.king@gmail.com Signed-off-by: Mark Brown --- sound/soc/intel/skylake/skl-topology.c | 6 ------ 1 file changed, 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 9bdf020a2b64..e06eac592da1 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -2950,9 +2950,6 @@ static int skl_tplg_get_pvt_data(struct snd_soc_tplg_dapm_widget *tplg_w, block_size = ret; off += array->size; - array = (struct snd_soc_tplg_vendor_array *) - (tplg_w->priv.data + off); - data = (tplg_w->priv.data + off); if (block_type == SKL_TYPE_TUPLE) { @@ -3599,9 +3596,6 @@ static int skl_tplg_get_manifest_data(struct snd_soc_tplg_manifest *manifest, block_size = ret; off += array->size; - array = (struct snd_soc_tplg_vendor_array *) - (manifest->priv.data + off); - data = (manifest->priv.data + off); if (block_type == SKL_TYPE_TUPLE) { -- cgit v1.2.3 From 2964e31cdda03fdff3b7c2f4f043e788e607987f Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 16 Jun 2022 08:49:10 +0300 Subject: ASoC: SOF: Intel: IPC4: enable IMR boot IPC4 based firmwares have unconditional support for IMR boot. Signed-off-by: Peter Ujfalusi Reviewed-by: Ranjani Sridharan Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220616054910.16690-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-loader.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c index 9e99f376f2b3..bca9dc5917f4 100644 --- a/sound/soc/sof/intel/hda-loader.c +++ b/sound/soc/sof/intel/hda-loader.c @@ -538,7 +538,8 @@ int hda_dsp_post_fw_run(struct snd_sof_dev *sdev) /* Check if IMR boot is usable */ if (!sof_debug_check_flag(SOF_DBG_IGNORE_D3_PERSISTENT) && - sdev->fw_ready.flags & SOF_IPC_INFO_D3_PERSISTENT) + (sdev->fw_ready.flags & SOF_IPC_INFO_D3_PERSISTENT || + sdev->pdata->ipc_type == SOF_INTEL_IPC4)) hdev->imrboot_supported = true; } -- cgit v1.2.3 From 6639990dbb25257eeb3df4d03e38e7d26c2484ab Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 16 Jun 2022 15:18:16 -0500 Subject: ASoC: SOF: pm: add explicit behavior for ACPI S1 and S2 MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The existing code only deals with S0 and S3, let's start adding S1 and S2. No functional change. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Link: https://lore.kernel.org/r/20220616201818.130802-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/pm.c | 12 +++++++++++- 1 file changed, 11 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sof/pm.c b/sound/soc/sof/pm.c index 18eb327a57f0..239f39a5166a 100644 --- a/sound/soc/sof/pm.c +++ b/sound/soc/sof/pm.c @@ -335,8 +335,18 @@ int snd_sof_prepare(struct device *dev) return 0; #if defined(CONFIG_ACPI) - if (acpi_target_system_state() == ACPI_STATE_S0) + switch (acpi_target_system_state()) { + case ACPI_STATE_S0: sdev->system_suspend_target = SOF_SUSPEND_S0IX; + break; + case ACPI_STATE_S1: + case ACPI_STATE_S2: + case ACPI_STATE_S3: + sdev->system_suspend_target = SOF_SUSPEND_S3; + break; + default: + break; + } #endif return 0; -- cgit v1.2.3 From 7a5974e035a6d496797547e4b469bc88938343c2 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 16 Jun 2022 15:18:17 -0500 Subject: ASoC: SOF: pm: add definitions for S4 and S5 states MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit We currently don't have a means to differentiate between S3, S4 and S5. Add definitions so that we have select different code paths depending on the target state in follow-up patches. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Link: https://lore.kernel.org/r/20220616201818.130802-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/pm.c | 9 +++++++++ sound/soc/sof/sof-priv.h | 2 ++ 2 files changed, 11 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/pm.c b/sound/soc/sof/pm.c index 239f39a5166a..df740be645e8 100644 --- a/sound/soc/sof/pm.c +++ b/sound/soc/sof/pm.c @@ -23,6 +23,9 @@ static u32 snd_sof_dsp_power_target(struct snd_sof_dev *sdev) u32 target_dsp_state; switch (sdev->system_suspend_target) { + case SOF_SUSPEND_S5: + case SOF_SUSPEND_S4: + /* DSP should be in D3 if the system is suspending to S3+ */ case SOF_SUSPEND_S3: /* DSP should be in D3 if the system is suspending to S3 */ target_dsp_state = SOF_DSP_PM_D3; @@ -344,6 +347,12 @@ int snd_sof_prepare(struct device *dev) case ACPI_STATE_S3: sdev->system_suspend_target = SOF_SUSPEND_S3; break; + case ACPI_STATE_S4: + sdev->system_suspend_target = SOF_SUSPEND_S4; + break; + case ACPI_STATE_S5: + sdev->system_suspend_target = SOF_SUSPEND_S5; + break; default: break; } diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index 9d7f53ff9c70..f0f3d72c0da7 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -85,6 +85,8 @@ enum sof_system_suspend_state { SOF_SUSPEND_NONE = 0, SOF_SUSPEND_S0IX, SOF_SUSPEND_S3, + SOF_SUSPEND_S4, + SOF_SUSPEND_S5, }; enum sof_dfsentry_type { -- cgit v1.2.3 From 58ecb11eab44dd5d64e35664ac4d62fecb6328f4 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 16 Jun 2022 15:18:18 -0500 Subject: ASoC: SOF: Intel: disable IMR boot when resuming from ACPI S4 and S5 states MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The IMR was assumed to be preserved when suspending to S4 and S5 states, but community reports invalidate that assumption, the hardware seems to be powered off and the IMR memory content cleared. Make sure regular boot with firmware download is used for S4 and S5. BugLink: https://github.com/thesofproject/sof/issues/5892 Fixes: 5fb5f51185126 ("ASoC: SOF: Intel: hda-loader: add IMR restore support") Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Link: https://lore.kernel.org/r/20220616201818.130802-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-loader.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c index d3ec5996a9a3..145d483bd129 100644 --- a/sound/soc/sof/intel/hda-loader.c +++ b/sound/soc/sof/intel/hda-loader.c @@ -389,7 +389,8 @@ int hda_dsp_cl_boot_firmware(struct snd_sof_dev *sdev) struct snd_dma_buffer dmab; int ret, ret1, i; - if (hda->imrboot_supported && !sdev->first_boot) { + if (sdev->system_suspend_target < SOF_SUSPEND_S4 && + hda->imrboot_supported && !sdev->first_boot) { dev_dbg(sdev->dev, "IMR restore supported, booting from IMR directly\n"); hda->boot_iteration = 0; ret = hda_dsp_boot_imr(sdev); -- cgit v1.2.3 From a37a9224d0500f0cf5bf13cb225163c21b29e0f6 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 16 Jun 2022 15:19:53 -0500 Subject: ASoC: SOF: Intel: hda: Fix compressed stream position tracking Commit 288fad2f71fa ("ASoC: SOF: Intel: hda: add quirks for HDAudio DMA position information") modified the PCM path only, but left the compressed data patch using an obsolete option. Move the functionality in a helper that can be called for both PCM and compressed data. Reviewed-by: Ranjani Sridharan Fixes: 288fad2f71fa ("ASoC: SOF: Intel: hda: add quirks for HDAudio DMA position information") Signed-off-by: Peter Ujfalusi Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220616201953.130876-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-pcm.c | 74 +------------------------------ sound/soc/sof/intel/hda-stream.c | 94 ++++++++++++++++++++++++++++++++++++++-- sound/soc/sof/intel/hda.h | 3 ++ 3 files changed, 94 insertions(+), 77 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/intel/hda-pcm.c b/sound/soc/sof/intel/hda-pcm.c index dc1f743730c0..6888e0a4665d 100644 --- a/sound/soc/sof/intel/hda-pcm.c +++ b/sound/soc/sof/intel/hda-pcm.c @@ -192,79 +192,7 @@ snd_pcm_uframes_t hda_dsp_pcm_pointer(struct snd_sof_dev *sdev, goto found; } - switch (sof_hda_position_quirk) { - case SOF_HDA_POSITION_QUIRK_USE_SKYLAKE_LEGACY: - /* - * This legacy code, inherited from the Skylake driver, - * mixes DPIB registers and DPIB DDR updates and - * does not seem to follow any known hardware recommendations. - * It's not clear e.g. why there is a different flow - * for capture and playback, the only information that matters is - * what traffic class is used, and on all SOF-enabled platforms - * only VC0 is supported so the work-around was likely not necessary - * and quite possibly wrong. - */ - - /* DPIB/posbuf position mode: - * For Playback, Use DPIB register from HDA space which - * reflects the actual data transferred. - * For Capture, Use the position buffer for pointer, as DPIB - * is not accurate enough, its update may be completed - * earlier than the data written to DDR. - */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - pos = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, - AZX_REG_VS_SDXDPIB_XBASE + - (AZX_REG_VS_SDXDPIB_XINTERVAL * - hstream->index)); - } else { - /* - * For capture stream, we need more workaround to fix the - * position incorrect issue: - * - * 1. Wait at least 20us before reading position buffer after - * the interrupt generated(IOC), to make sure position update - * happens on frame boundary i.e. 20.833uSec for 48KHz. - * 2. Perform a dummy Read to DPIB register to flush DMA - * position value. - * 3. Read the DMA Position from posbuf. Now the readback - * value should be >= period boundary. - */ - usleep_range(20, 21); - snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, - AZX_REG_VS_SDXDPIB_XBASE + - (AZX_REG_VS_SDXDPIB_XINTERVAL * - hstream->index)); - pos = snd_hdac_stream_get_pos_posbuf(hstream); - } - break; - case SOF_HDA_POSITION_QUIRK_USE_DPIB_REGISTERS: - /* - * In case VC1 traffic is disabled this is the recommended option - */ - pos = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, - AZX_REG_VS_SDXDPIB_XBASE + - (AZX_REG_VS_SDXDPIB_XINTERVAL * - hstream->index)); - break; - case SOF_HDA_POSITION_QUIRK_USE_DPIB_DDR_UPDATE: - /* - * This is the recommended option when VC1 is enabled. - * While this isn't needed for SOF platforms it's added for - * consistency and debug. - */ - pos = snd_hdac_stream_get_pos_posbuf(hstream); - break; - default: - dev_err_once(sdev->dev, "hda_position_quirk value %d not supported\n", - sof_hda_position_quirk); - pos = 0; - break; - } - - if (pos >= hstream->bufsize) - pos = 0; - + pos = hda_dsp_stream_get_position(hstream, substream->stream, true); found: pos = bytes_to_frames(substream->runtime, pos); diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index daeb64c495e4..d95ae17e81cc 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -707,12 +707,13 @@ bool hda_dsp_check_stream_irq(struct snd_sof_dev *sdev) } static void -hda_dsp_set_bytes_transferred(struct hdac_stream *hstream, u64 buffer_size) +hda_dsp_compr_bytes_transferred(struct hdac_stream *hstream, int direction) { + u64 buffer_size = hstream->bufsize; u64 prev_pos, pos, num_bytes; div64_u64_rem(hstream->curr_pos, buffer_size, &prev_pos); - pos = snd_hdac_stream_get_pos_posbuf(hstream); + pos = hda_dsp_stream_get_position(hstream, direction, false); if (pos < prev_pos) num_bytes = (buffer_size - prev_pos) + pos; @@ -748,8 +749,7 @@ static bool hda_dsp_stream_check(struct hdac_bus *bus, u32 status) if (s->substream && sof_hda->no_ipc_position) { snd_sof_pcm_period_elapsed(s->substream); } else if (s->cstream) { - hda_dsp_set_bytes_transferred(s, - s->cstream->runtime->buffer_size); + hda_dsp_compr_bytes_transferred(s, s->cstream->direction); snd_compr_fragment_elapsed(s->cstream); } } @@ -1009,3 +1009,89 @@ void hda_dsp_stream_free(struct snd_sof_dev *sdev) devm_kfree(sdev->dev, hda_stream); } } + +snd_pcm_uframes_t hda_dsp_stream_get_position(struct hdac_stream *hstream, + int direction, bool can_sleep) +{ + struct hdac_ext_stream *hext_stream = stream_to_hdac_ext_stream(hstream); + struct sof_intel_hda_stream *hda_stream = hstream_to_sof_hda_stream(hext_stream); + struct snd_sof_dev *sdev = hda_stream->sdev; + snd_pcm_uframes_t pos; + + switch (sof_hda_position_quirk) { + case SOF_HDA_POSITION_QUIRK_USE_SKYLAKE_LEGACY: + /* + * This legacy code, inherited from the Skylake driver, + * mixes DPIB registers and DPIB DDR updates and + * does not seem to follow any known hardware recommendations. + * It's not clear e.g. why there is a different flow + * for capture and playback, the only information that matters is + * what traffic class is used, and on all SOF-enabled platforms + * only VC0 is supported so the work-around was likely not necessary + * and quite possibly wrong. + */ + + /* DPIB/posbuf position mode: + * For Playback, Use DPIB register from HDA space which + * reflects the actual data transferred. + * For Capture, Use the position buffer for pointer, as DPIB + * is not accurate enough, its update may be completed + * earlier than the data written to DDR. + */ + if (direction == SNDRV_PCM_STREAM_PLAYBACK) { + pos = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, + AZX_REG_VS_SDXDPIB_XBASE + + (AZX_REG_VS_SDXDPIB_XINTERVAL * + hstream->index)); + } else { + /* + * For capture stream, we need more workaround to fix the + * position incorrect issue: + * + * 1. Wait at least 20us before reading position buffer after + * the interrupt generated(IOC), to make sure position update + * happens on frame boundary i.e. 20.833uSec for 48KHz. + * 2. Perform a dummy Read to DPIB register to flush DMA + * position value. + * 3. Read the DMA Position from posbuf. Now the readback + * value should be >= period boundary. + */ + if (can_sleep) + usleep_range(20, 21); + + snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, + AZX_REG_VS_SDXDPIB_XBASE + + (AZX_REG_VS_SDXDPIB_XINTERVAL * + hstream->index)); + pos = snd_hdac_stream_get_pos_posbuf(hstream); + } + break; + case SOF_HDA_POSITION_QUIRK_USE_DPIB_REGISTERS: + /* + * In case VC1 traffic is disabled this is the recommended option + */ + pos = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, + AZX_REG_VS_SDXDPIB_XBASE + + (AZX_REG_VS_SDXDPIB_XINTERVAL * + hstream->index)); + break; + case SOF_HDA_POSITION_QUIRK_USE_DPIB_DDR_UPDATE: + /* + * This is the recommended option when VC1 is enabled. + * While this isn't needed for SOF platforms it's added for + * consistency and debug. + */ + pos = snd_hdac_stream_get_pos_posbuf(hstream); + break; + default: + dev_err_once(sdev->dev, "hda_position_quirk value %d not supported\n", + sof_hda_position_quirk); + pos = 0; + break; + } + + if (pos >= hstream->bufsize) + pos = 0; + + return pos; +} diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 0f57ef5d9b8e..06476ffe96d7 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -565,6 +565,9 @@ int hda_dsp_stream_setup_bdl(struct snd_sof_dev *sdev, bool hda_dsp_check_ipc_irq(struct snd_sof_dev *sdev); bool hda_dsp_check_stream_irq(struct snd_sof_dev *sdev); +snd_pcm_uframes_t hda_dsp_stream_get_position(struct hdac_stream *hstream, + int direction, bool can_sleep); + struct hdac_ext_stream * hda_dsp_stream_get(struct snd_sof_dev *sdev, int direction, u32 flags); int hda_dsp_stream_put(struct snd_sof_dev *sdev, int direction, int stream_tag); -- cgit v1.2.3 From e1ab67be68e900a6585277ca442ca7f67dffb3bd Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 16 Jun 2022 17:08:01 -0500 Subject: ASoC: cs4270: update kernel-doc Remove warning sound/soc/codecs/cs4270.c:672: warning: Excess function parameter 'id' description in 'cs4270_i2c_probe' Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Kai Vehmanen Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/20220616220802.136282-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs4270.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 531f63b01554..97d26b9e8f7f 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -663,7 +663,6 @@ static int cs4270_i2c_remove(struct i2c_client *i2c_client) /** * cs4270_i2c_probe - initialize the I2C interface of the CS4270 * @i2c_client: the I2C client object - * @id: the I2C device ID (ignored) * * This function is called whenever the I2C subsystem finds a device that * matches the device ID given via a prior call to i2c_add_driver(). -- cgit v1.2.3 From 7c619b306285588725573d975fd44607d13438cf Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 16 Jun 2022 17:08:02 -0500 Subject: ASoC: sunxi: sun4i-i2s: update kernel-doc Remove warnings sound/soc/sunxi/sun4i-i2s.c:205: warning: Function parameter or member 'num_din_pins' not described in 'sun4i_i2s_quirks' sound/soc/sunxi/sun4i-i2s.c:205: warning: Function parameter or member 'num_dout_pins' not described in 'sun4i_i2s_quirks' Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Kai Vehmanen Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/20220616220802.136282-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sunxi/sun4i-i2s.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index f58aa6406a87..5be33d07361b 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -161,6 +161,8 @@ struct sun4i_i2s; * @field_clkdiv_mclk_en: regmap field to enable mclk output. * @field_fmt_wss: regmap field to set word select size. * @field_fmt_sr: regmap field to set sample resolution. + * @num_din_pins: input pins + * @num_dout_pins: output pins (currently set but unused) * @bclk_dividers: bit clock dividers array * @num_bclk_dividers: number of bit clock dividers * @mclk_dividers: mclk dividers array -- cgit v1.2.3 From e33ea0685a219543f3e23d88186bc6c3259b3ff4 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 16 Jun 2022 16:40:43 -0500 Subject: ASoC: Intel: skl_nau88l25_max98357a: remap jack pins MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The card did not map jack pins to controls, which prevents PulseAudio/PipeWire from dealing with jack detection. It's likely that jack detection was only tested with the CRAS server and extensions of UCM. Suggested-by: Jaroslav Kysela Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20220616214055.134943-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 19 ++++++++++++++++--- 1 file changed, 16 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/skl_nau88l25_max98357a.c b/sound/soc/intel/boards/skl_nau88l25_max98357a.c index 8e2d03e36079..8dceb0b02581 100644 --- a/sound/soc/intel/boards/skl_nau88l25_max98357a.c +++ b/sound/soc/intel/boards/skl_nau88l25_max98357a.c @@ -97,6 +97,17 @@ static const struct snd_soc_dapm_widget skylake_widgets[] = { SND_SOC_DAPM_POST_PMD), }; +static struct snd_soc_jack_pin jack_pins[] = { + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, +}; + static const struct snd_soc_dapm_route skylake_map[] = { /* HP jack connectors - unknown if we have jack detection */ { "Headphone Jack", NULL, "HPOL" }, @@ -163,9 +174,11 @@ static int skylake_nau8825_codec_init(struct snd_soc_pcm_runtime *rtd) * Headset buttons map to the google Reference headset. * These can be configured by userspace. */ - ret = snd_soc_card_jack_new(&skylake_audio_card, "Headset Jack", - SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | - SND_JACK_BTN_2 | SND_JACK_BTN_3, &skylake_headset); + ret = snd_soc_card_jack_new_pins(&skylake_audio_card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3, &skylake_headset, + jack_pins, + ARRAY_SIZE(jack_pins)); if (ret) { dev_err(rtd->dev, "Headset Jack creation failed %d\n", ret); return ret; -- cgit v1.2.3 From 4864ef4a67edfbf802ba36c921c5e9f66e1530bf Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 16 Jun 2022 16:40:44 -0500 Subject: ASoC: Intel: skl_nau88l25_ssm4567: remap jack pins MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The card did not map jack pins to controls, which prevents PulseAudio/PipeWire from dealing with jack detection. It's likely that jack detection was only tested with the CRAS server and extensions of UCM. Suggested-by: Jaroslav Kysela Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20220616214055.134943-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 19 ++++++++++++++++--- 1 file changed, 16 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c index 501f0bbfc404..62c0d46d0086 100644 --- a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c +++ b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c @@ -101,6 +101,17 @@ static const struct snd_soc_dapm_widget skylake_widgets[] = { SND_SOC_DAPM_POST_PMD), }; +static struct snd_soc_jack_pin jack_pins[] = { + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, +}; + static const struct snd_soc_dapm_route skylake_map[] = { /* HP jack connectors - unknown if we have jack detection */ {"Headphone Jack", NULL, "HPOL"}, @@ -182,9 +193,11 @@ static int skylake_nau8825_codec_init(struct snd_soc_pcm_runtime *rtd) * 4 buttons here map to the google Reference headset * The use of these buttons can be decided by the user space. */ - ret = snd_soc_card_jack_new(&skylake_audio_card, "Headset Jack", - SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | - SND_JACK_BTN_2 | SND_JACK_BTN_3, &skylake_headset); + ret = snd_soc_card_jack_new_pins(&skylake_audio_card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3, &skylake_headset, + jack_pins, + ARRAY_SIZE(jack_pins)); if (ret) { dev_err(rtd->dev, "Headset Jack creation failed %d\n", ret); return ret; -- cgit v1.2.3 From decdbf3dd7ec3e3522548f50e22d81558d151118 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 16 Jun 2022 16:40:45 -0500 Subject: ASoC: Intel: kbl_rt5663_max98927: remap jack pins MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The card did not map jack pins to controls, which prevents PulseAudio/PipeWire from dealing with jack detection. It's likely that jack detection was only tested with the CRAS server and extensions of UCM. Suggested-by: Jaroslav Kysela Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20220616214055.134943-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/kbl_rt5663_max98927.c | 21 +++++++++++++++++---- 1 file changed, 17 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index 8d37b2676a81..2d4224c5b152 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -206,6 +206,17 @@ static const struct snd_soc_dapm_widget kabylake_5663_widgets[] = { SND_SOC_DAPM_POST_PMD), }; +static struct snd_soc_jack_pin jack_pins[] = { + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, +}; + static const struct snd_soc_dapm_route kabylake_5663_map[] = { { "Headphone Jack", NULL, "Platform Clock" }, { "Headphone Jack", NULL, "HPOL" }, @@ -271,10 +282,12 @@ static int kabylake_rt5663_codec_init(struct snd_soc_pcm_runtime *rtd) * Headset buttons map to the google Reference headset. * These can be configured by userspace. */ - ret = snd_soc_card_jack_new(kabylake_audio_card, "Headset Jack", - SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | - SND_JACK_BTN_2 | SND_JACK_BTN_3, - &ctx->kabylake_headset); + ret = snd_soc_card_jack_new_pins(kabylake_audio_card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3, + &ctx->kabylake_headset, + jack_pins, + ARRAY_SIZE(jack_pins)); if (ret) { dev_err(rtd->dev, "Headset Jack creation failed %d\n", ret); return ret; -- cgit v1.2.3 From c2065d43ae8546668f8f187138eda8a18f7625fd Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 16 Jun 2022 16:40:46 -0500 Subject: ASoC: Intel: kbl_da7219_max98357a: remap jack pins MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The card did not map jack pins to controls, which prevents PulseAudio/PipeWire from dealing with jack detection. It's likely that jack detection was only tested with the CRAS server and extensions of UCM. Suggested-by: Jaroslav Kysela Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20220616214055.134943-5-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/kbl_da7219_max98357a.c | 21 +++++++++++++++++---- 1 file changed, 17 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/kbl_da7219_max98357a.c b/sound/soc/intel/boards/kbl_da7219_max98357a.c index ceabed85e9da..329457e3e3a2 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98357a.c +++ b/sound/soc/intel/boards/kbl_da7219_max98357a.c @@ -99,6 +99,17 @@ static const struct snd_soc_dapm_widget kabylake_widgets[] = { SND_SOC_DAPM_POST_PMD), }; +static struct snd_soc_jack_pin jack_pins[] = { + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, +}; + static const struct snd_soc_dapm_route kabylake_map[] = { { "Headphone Jack", NULL, "HPL" }, { "Headphone Jack", NULL, "HPR" }, @@ -179,10 +190,12 @@ static int kabylake_da7219_codec_init(struct snd_soc_pcm_runtime *rtd) * Headset buttons map to the google Reference headset. * These can be configured by userspace. */ - ret = snd_soc_card_jack_new(kabylake_audio_card, "Headset Jack", - SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | - SND_JACK_BTN_2 | SND_JACK_BTN_3 | SND_JACK_LINEOUT, - &ctx->kabylake_headset); + ret = snd_soc_card_jack_new_pins(kabylake_audio_card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3 | SND_JACK_LINEOUT, + &ctx->kabylake_headset, + jack_pins, + ARRAY_SIZE(jack_pins)); if (ret) { dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret); return ret; -- cgit v1.2.3 From b9f53b9fc14e26ef3b3c33160afb094ad7ae192b Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 16 Jun 2022 16:40:47 -0500 Subject: ASoC: Intel: kbl_da7219_max98927: remap jack pins MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The card did not map jack pins to controls, which prevents PulseAudio/PipeWire from dealing with jack detection. It's likely that jack detection was only tested with the CRAS server and extensions of UCM. Suggested-by: Jaroslav Kysela Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20220616214055.134943-6-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/kbl_da7219_max98927.c | 21 +++++++++++++++++---- 1 file changed, 17 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c index 703ccff634b0..362579f25835 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98927.c +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -119,6 +119,17 @@ static const struct snd_soc_dapm_widget kabylake_widgets[] = { SND_SOC_DAPM_POST_PMD), }; +static struct snd_soc_jack_pin jack_pins[] = { + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, +}; + static const struct snd_soc_dapm_route kabylake_map[] = { /* speaker */ { "Left Spk", NULL, "Left BE_OUT" }, @@ -354,10 +365,12 @@ static int kabylake_da7219_codec_init(struct snd_soc_pcm_runtime *rtd) * Headset buttons map to the google Reference headset. * These can be configured by userspace. */ - ret = snd_soc_card_jack_new(kabylake_audio_card, "Headset Jack", - SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | - SND_JACK_BTN_2 | SND_JACK_BTN_3 | SND_JACK_LINEOUT, - &ctx->kabylake_headset); + ret = snd_soc_card_jack_new_pins(kabylake_audio_card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3 | SND_JACK_LINEOUT, + &ctx->kabylake_headset, + jack_pins, + ARRAY_SIZE(jack_pins)); if (ret) { dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret); return ret; -- cgit v1.2.3 From c0703be996c343b4d1036b6ba258133d88b7932b Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 16 Jun 2022 16:40:48 -0500 Subject: ASoC: Intel: kbl_rt5663_rt5514_max98927: remap jack pins MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The card did not map jack pins to controls, which prevents PulseAudio/PipeWire from dealing with jack detection. It's likely that jack detection was only tested with the CRAS server and extensions of UCM. Suggested-by: Jaroslav Kysela Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20220616214055.134943-7-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 21 +++++++++++++++++---- 1 file changed, 17 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 564c70a0fbc8..2c79fca57b19 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -145,6 +145,17 @@ static const struct snd_soc_dapm_widget kabylake_widgets[] = { }; +static struct snd_soc_jack_pin jack_pins[] = { + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, +}; + static const struct snd_soc_dapm_route kabylake_map[] = { /* Headphones */ { "Headphone Jack", NULL, "Platform Clock" }, @@ -228,10 +239,12 @@ static int kabylake_rt5663_codec_init(struct snd_soc_pcm_runtime *rtd) * Headset buttons map to the google Reference headset. * These can be configured by userspace. */ - ret = snd_soc_card_jack_new(&kabylake_audio_card, "Headset Jack", - SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | - SND_JACK_BTN_2 | SND_JACK_BTN_3, - &ctx->kabylake_headset); + ret = snd_soc_card_jack_new_pins(&kabylake_audio_card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3, + &ctx->kabylake_headset, + jack_pins, + ARRAY_SIZE(jack_pins)); if (ret) { dev_err(rtd->dev, "Headset Jack creation failed %d\n", ret); return ret; -- cgit v1.2.3 From bbdd4ea2190b4712c0cd9989a5e402c7f99fc122 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 16 Jun 2022 16:40:49 -0500 Subject: ASoC: Intel: bxt_da7219_max98357a: remap jack pins MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The card did not map jack pins to controls, which prevents PulseAudio/PipeWire from dealing with jack detection. It's likely that jack detection was only tested with the CRAS server and extensions of UCM. Suggested-by: Jaroslav Kysela Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20220616214055.134943-8-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/bxt_da7219_max98357a.c | 21 +++++++++++++++++---- 1 file changed, 17 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index d98376da425a..7c6c95e99ade 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -186,6 +186,17 @@ static const struct snd_soc_dapm_route gemini_map[] = { {"ssp2 Rx", NULL, "Capture"}, }; +static struct snd_soc_jack_pin jack_pins[] = { + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, +}; + static int broxton_ssp_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { @@ -231,10 +242,12 @@ static int broxton_da7219_codec_init(struct snd_soc_pcm_runtime *rtd) * Headset buttons map to the google Reference headset. * These can be configured by userspace. */ - ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", - SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | - SND_JACK_BTN_2 | SND_JACK_BTN_3 | SND_JACK_LINEOUT, - &broxton_headset); + ret = snd_soc_card_jack_new_pins(rtd->card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3 | SND_JACK_LINEOUT, + &broxton_headset, + jack_pins, + ARRAY_SIZE(jack_pins)); if (ret) { dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret); return ret; -- cgit v1.2.3 From 4c3a68e9026ad7d3aa61278ce5702407d91d5dd9 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 16 Jun 2022 16:40:50 -0500 Subject: ASoC: Intel: glk_rt5682_max98357a: remap jack pins MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The card did not map jack pins to controls, which prevents PulseAudio/PipeWire from dealing with jack detection. It's likely that jack detection was only tested with the CRAS server and extensions of UCM. Suggested-by: Jaroslav Kysela Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20220616214055.134943-9-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/glk_rt5682_max98357a.c | 21 +++++++++++++++++---- 1 file changed, 17 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/glk_rt5682_max98357a.c b/sound/soc/intel/boards/glk_rt5682_max98357a.c index 170164baae7d..cf0f89db3e20 100644 --- a/sound/soc/intel/boards/glk_rt5682_max98357a.c +++ b/sound/soc/intel/boards/glk_rt5682_max98357a.c @@ -78,6 +78,17 @@ static const struct snd_soc_dapm_widget geminilake_widgets[] = { SND_SOC_DAPM_SPK("HDMI3", NULL), }; +static struct snd_soc_jack_pin jack_pins[] = { + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, +}; + static const struct snd_soc_dapm_route geminilake_map[] = { /* HP jack connectors - unknown if we have jack detection */ { "Headphone Jack", NULL, "HPOL" }, @@ -173,10 +184,12 @@ static int geminilake_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) * Headset buttons map to the google Reference headset. * These can be configured by userspace. */ - ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", - SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | - SND_JACK_BTN_2 | SND_JACK_BTN_3 | SND_JACK_LINEOUT, - &ctx->geminilake_headset); + ret = snd_soc_card_jack_new_pins(rtd->card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3 | SND_JACK_LINEOUT, + &ctx->geminilake_headset, + jack_pins, + ARRAY_SIZE(jack_pins)); if (ret) { dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret); return ret; -- cgit v1.2.3 From 77a036e8b074a679c0177f61c9d3b8e942673141 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 16 Jun 2022 16:40:51 -0500 Subject: ASoC: Intel: cml_rt1011_rt5682: remap jack pins MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The card did not map jack pins to controls, which prevents PulseAudio/PipeWire from dealing with jack detection. It's likely that jack detection was only tested with the CRAS server and extensions of UCM. Suggested-by: Jaroslav Kysela Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20220616214055.134943-10-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/cml_rt1011_rt5682.c | 23 ++++++++++++++++++----- 1 file changed, 18 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/cml_rt1011_rt5682.c b/sound/soc/intel/boards/cml_rt1011_rt5682.c index a99f74a15b5f..20da83d9eece 100644 --- a/sound/soc/intel/boards/cml_rt1011_rt5682.c +++ b/sound/soc/intel/boards/cml_rt1011_rt5682.c @@ -121,6 +121,17 @@ static const struct snd_soc_dapm_route cml_rt1011_tt_map[] = { {"TR Ext Spk", NULL, "TR SPO" }, }; +static struct snd_soc_jack_pin jack_pins[] = { + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, +}; + static int cml_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) { struct card_private *ctx = snd_soc_card_get_drvdata(rtd->card); @@ -137,11 +148,13 @@ static int cml_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) * Headset buttons map to the google Reference headset. * These can be configured by userspace. */ - ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", - SND_JACK_HEADSET | SND_JACK_BTN_0 | - SND_JACK_BTN_1 | SND_JACK_BTN_2 | - SND_JACK_BTN_3, - &ctx->headset); + ret = snd_soc_card_jack_new_pins(rtd->card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | + SND_JACK_BTN_1 | SND_JACK_BTN_2 | + SND_JACK_BTN_3, + &ctx->headset, + jack_pins, + ARRAY_SIZE(jack_pins)); if (ret) { dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret); return ret; -- cgit v1.2.3 From 7459c8940a506280908f8b5e9e4227784a0b6569 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 16 Jun 2022 16:40:52 -0500 Subject: ASoC: Intel: sof_cs42l42: remap jack pins MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The card did not map jack pins to controls, which prevents PulseAudio/PipeWire from dealing with jack detection. It's likely that jack detection was only tested with the CRAS server and extensions of UCM. Suggested-by: Jaroslav Kysela Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20220616214055.134943-11-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_cs42l42.c | 23 ++++++++++++++++++----- 1 file changed, 18 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/sof_cs42l42.c b/sound/soc/intel/boards/sof_cs42l42.c index 6a979c333bc5..a1a14d6d7c23 100644 --- a/sound/soc/intel/boards/sof_cs42l42.c +++ b/sound/soc/intel/boards/sof_cs42l42.c @@ -52,6 +52,17 @@ enum { LINK_HDMI = 4, }; +static struct snd_soc_jack_pin jack_pins[] = { + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, +}; + /* Default: SSP2 */ static unsigned long sof_cs42l42_quirk = SOF_CS42L42_SSP_CODEC(2); @@ -98,11 +109,13 @@ static int sof_cs42l42_init(struct snd_soc_pcm_runtime *rtd) * Headset buttons map to the google Reference headset. * These can be configured by userspace. */ - ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", - SND_JACK_HEADSET | SND_JACK_BTN_0 | - SND_JACK_BTN_1 | SND_JACK_BTN_2 | - SND_JACK_BTN_3, - jack); + ret = snd_soc_card_jack_new_pins(rtd->card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | + SND_JACK_BTN_1 | SND_JACK_BTN_2 | + SND_JACK_BTN_3, + jack, + jack_pins, + ARRAY_SIZE(jack_pins)); if (ret) { dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret); return ret; -- cgit v1.2.3 From 2913bb1f6830251416659dbb04c392bbc9592f14 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 16 Jun 2022 16:40:53 -0500 Subject: ASoC: Intel: sof_da7219_max98373: remap jack pins MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The card did not map jack pins to controls, which prevents PulseAudio/PipeWire from dealing with jack detection. It's likely that jack detection was only tested with the CRAS server and extensions of UCM. Suggested-by: Jaroslav Kysela Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20220616214055.134943-12-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_da7219_max98373.c | 23 ++++++++++++++++++----- 1 file changed, 18 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/sof_da7219_max98373.c b/sound/soc/intel/boards/sof_da7219_max98373.c index a83f30b687cf..34cf849a8344 100644 --- a/sound/soc/intel/boards/sof_da7219_max98373.c +++ b/sound/soc/intel/boards/sof_da7219_max98373.c @@ -135,6 +135,17 @@ static const struct snd_soc_dapm_route max98360a_map[] = { {"DMic", NULL, "SoC DMIC"}, }; +static struct snd_soc_jack_pin jack_pins[] = { + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, +}; + static struct snd_soc_jack headset; static int da7219_codec_init(struct snd_soc_pcm_runtime *rtd) @@ -156,11 +167,13 @@ static int da7219_codec_init(struct snd_soc_pcm_runtime *rtd) * Headset buttons map to the google Reference headset. * These can be configured by userspace. */ - ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", - SND_JACK_HEADSET | SND_JACK_BTN_0 | - SND_JACK_BTN_1 | SND_JACK_BTN_2 | - SND_JACK_BTN_3 | SND_JACK_LINEOUT, - &headset); + ret = snd_soc_card_jack_new_pins(rtd->card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | + SND_JACK_BTN_1 | SND_JACK_BTN_2 | + SND_JACK_BTN_3 | SND_JACK_LINEOUT, + &headset, + jack_pins, + ARRAY_SIZE(jack_pins)); if (ret) { dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret); return ret; -- cgit v1.2.3 From 2a172d2f06c155ea7c9b34f47febdfe9b9bbe1c2 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 16 Jun 2022 16:40:54 -0500 Subject: ASoC: Intel: sof_nau8825: remap jack pins MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The card did not map jack pins to controls, which prevents PulseAudio/PipeWire from dealing with jack detection. It's likely that jack detection was only tested with the CRAS server and extensions of UCM. Suggested-by: Jaroslav Kysela Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20220616214055.134943-13-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_nau8825.c | 23 ++++++++++++++++++----- 1 file changed, 18 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/sof_nau8825.c b/sound/soc/intel/boards/sof_nau8825.c index 97dcd204a246..f49700eb721b 100644 --- a/sound/soc/intel/boards/sof_nau8825.c +++ b/sound/soc/intel/boards/sof_nau8825.c @@ -81,6 +81,17 @@ static int sof_hdmi_init(struct snd_soc_pcm_runtime *rtd) return 0; } +static struct snd_soc_jack_pin jack_pins[] = { + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, +}; + static int sof_nau8825_codec_init(struct snd_soc_pcm_runtime *rtd) { struct sof_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); @@ -93,11 +104,13 @@ static int sof_nau8825_codec_init(struct snd_soc_pcm_runtime *rtd) * Headset buttons map to the google Reference headset. * These can be configured by userspace. */ - ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", - SND_JACK_HEADSET | SND_JACK_BTN_0 | - SND_JACK_BTN_1 | SND_JACK_BTN_2 | - SND_JACK_BTN_3, - &ctx->sof_headset); + ret = snd_soc_card_jack_new_pins(rtd->card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | + SND_JACK_BTN_1 | SND_JACK_BTN_2 | + SND_JACK_BTN_3, + &ctx->sof_headset, + jack_pins, + ARRAY_SIZE(jack_pins)); if (ret) { dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret); return ret; -- cgit v1.2.3 From c3ce12b27e562bf3a255bc9f3096dacea2194dd8 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 16 Jun 2022 16:40:55 -0500 Subject: ASoC: Intel: sof_rt5682: remap jack pins MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The card did not map jack pins to controls, which prevents PulseAudio/PipeWire from dealing with jack detection. It's likely that jack detection was only tested with the CRAS server and extensions of UCM. Suggested-by: Jaroslav Kysela Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20220616214055.134943-14-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_rt5682.c | 23 ++++++++++++++++++----- 1 file changed, 18 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index f28dae64587e..a24fb71d5ff3 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -248,6 +248,17 @@ static int sof_hdmi_init(struct snd_soc_pcm_runtime *rtd) return 0; } +static struct snd_soc_jack_pin jack_pins[] = { + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, +}; + static int sof_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) { struct sof_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); @@ -295,11 +306,13 @@ static int sof_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) * Headset buttons map to the google Reference headset. * These can be configured by userspace. */ - ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", - SND_JACK_HEADSET | SND_JACK_BTN_0 | - SND_JACK_BTN_1 | SND_JACK_BTN_2 | - SND_JACK_BTN_3, - &ctx->sof_headset); + ret = snd_soc_card_jack_new_pins(rtd->card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | + SND_JACK_BTN_1 | SND_JACK_BTN_2 | + SND_JACK_BTN_3, + &ctx->sof_headset, + jack_pins, + ARRAY_SIZE(jack_pins)); if (ret) { dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret); return ret; -- cgit v1.2.3 From 6d5e37b0f343af70a7e824641f264fb140bbead5 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 16 Jun 2022 16:53:39 -0500 Subject: ASoC: SOF: Intel: hda-dsp: report error on power-up/down MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit dev_dbg() is not good-enough since the flow returns an error. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20220616215351.135643-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-dsp.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index 263ad455e283..2afaee91b982 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -743,7 +743,7 @@ int hda_dsp_resume(struct snd_sof_dev *sdev) if (hlink->ref_count) { ret = snd_hdac_ext_bus_link_power_up(hlink); if (ret < 0) { - dev_dbg(sdev->dev, + dev_err(sdev->dev, "error %d in %s: failed to power up links", ret, __func__); return ret; @@ -871,7 +871,7 @@ int hda_dsp_suspend(struct snd_sof_dev *sdev, u32 target_state) /* no link can be powered in s0ix state */ ret = snd_hdac_ext_bus_link_power_down_all(bus); if (ret < 0) { - dev_dbg(sdev->dev, + dev_err(sdev->dev, "error %d in %s: failed to power down links", ret, __func__); return ret; -- cgit v1.2.3 From 3abc88730a0e45247296a561a12e811b5d2d2236 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 16 Jun 2022 16:53:40 -0500 Subject: ASoC: SOF: Intel: hda-stream: report error on stream not opened MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit We report -ENODEV but only use dev_dbg, this is inconsistent. dev_err() makes sense here. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20220616215351.135643-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-stream.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index daeb64c495e4..d3403b7a2b2e 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -271,7 +271,7 @@ int hda_dsp_stream_put(struct snd_sof_dev *sdev, int direction, int stream_tag) HDA_VS_INTEL_EM2_L1SEN, HDA_VS_INTEL_EM2_L1SEN); if (!found) { - dev_dbg(sdev->dev, "%s: stream_tag %d not opened!\n", + dev_err(sdev->dev, "%s: stream_tag %d not opened!\n", __func__, stream_tag); return -ENODEV; } -- cgit v1.2.3 From 18701bb1370cb6b34a8f3ad820045930138083dc Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 16 Jun 2022 16:53:41 -0500 Subject: ASoC: SOF: Intel: hda-dai: remove use of __func__ in dev_dbg MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The module and function information can be added with 'modprobe foo dyndbg=+pmf' Suggested-by: Greg KH Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20220616215351.135643-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-dai.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index 70721defca46..ed74a1f264e8 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -393,7 +393,7 @@ static int hda_dai_prepare(struct snd_pcm_substream *substream, struct snd_soc_d if (hext_stream && hext_stream->link_prepared) return 0; - dev_dbg(sdev->dev, "%s: prepare stream dir %d\n", __func__, substream->stream); + dev_dbg(sdev->dev, "prepare stream dir %d\n", substream->stream); ret = hda_link_dma_prepare(substream); if (ret < 0) @@ -419,7 +419,7 @@ static int ipc3_hda_dai_trigger(struct snd_pcm_substream *substream, struct snd_soc_dapm_widget *w; int ret; - dev_dbg(dai->dev, "%s: cmd=%d dai %s direction %d\n", __func__, cmd, + dev_dbg(dai->dev, "cmd=%d dai %s direction %d\n", cmd, dai->name, substream->stream); ret = hda_link_dma_trigger(substream, cmd); @@ -468,7 +468,7 @@ static int ipc4_hda_dai_trigger(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai; int ret; - dev_dbg(dai->dev, "%s: cmd=%d dai %s direction %d\n", __func__, cmd, + dev_dbg(dai->dev, "cmd=%d dai %s direction %d\n", cmd, dai->name, substream->stream); hstream = substream->runtime->private_data; -- cgit v1.2.3 From 8bf064f8e439d9b92a023a54adc657f920f4e1a8 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 16 Jun 2022 16:53:42 -0500 Subject: ASoC: SOF: Intel: hda-stream: remove use of __func__ in dev_dbg MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The module and function information can be added with 'modprobe foo dyndbg=+pmf' Suggested-by: Greg KH Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20220616215351.135643-5-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-stream.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index d3403b7a2b2e..a4d51f855e56 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -116,13 +116,13 @@ int hda_dsp_stream_setup_bdl(struct snd_sof_dev *sdev, int remain, ioc; period_bytes = hstream->period_bytes; - dev_dbg(sdev->dev, "%s: period_bytes:0x%x\n", __func__, period_bytes); + dev_dbg(sdev->dev, "period_bytes:0x%x\n", period_bytes); if (!period_bytes) period_bytes = hstream->bufsize; periods = hstream->bufsize / period_bytes; - dev_dbg(sdev->dev, "%s: periods:%d\n", __func__, periods); + dev_dbg(sdev->dev, "periods:%d\n", periods); remain = hstream->bufsize % period_bytes; if (remain) -- cgit v1.2.3 From b837870fe17f21cf80b15d143c9ea0bc6b342741 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 16 Jun 2022 16:53:43 -0500 Subject: ASoC: SOF: Intel: mtl: remove use of __func__ in dev_dbg MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The module and function information can be added with 'modprobe foo dyndbg=+pmf' Suggested-by: Greg KH Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20220616215351.135643-6-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/mtl.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/intel/mtl.c b/sound/soc/sof/intel/mtl.c index 37be77beb415..3a043589c12b 100644 --- a/sound/soc/sof/intel/mtl.c +++ b/sound/soc/sof/intel/mtl.c @@ -540,8 +540,7 @@ static irqreturn_t mtl_ipc_irq_thread(int irq, void *context) if (!ipc_irq) { /* This interrupt is not shared so no need to return IRQ_NONE. */ - dev_dbg_ratelimited(sdev->dev, "%s nothing to do in IPC IRQ thread\n", - __func__); + dev_dbg_ratelimited(sdev->dev, "nothing to do in IPC IRQ thread\n"); } return IRQ_HANDLED; -- cgit v1.2.3 From 9fd8fcd03451ea3f04af9a419748248d3fa8fb59 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 16 Jun 2022 16:53:44 -0500 Subject: ASoC: SOF: ipc3-dtrace: remove use of __func__ in dev_dbg MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The module and function information can be added with 'modprobe foo dyndbg=+pmf' Suggested-by: Greg KH Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20220616215351.135643-7-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc3-dtrace.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/ipc3-dtrace.c b/sound/soc/sof/ipc3-dtrace.c index ecca6dceaad2..b815b0244d9e 100644 --- a/sound/soc/sof/ipc3-dtrace.c +++ b/sound/soc/sof/ipc3-dtrace.c @@ -470,7 +470,7 @@ static int ipc3_dtrace_enable(struct snd_sof_dev *sdev) dev_err(sdev->dev, "Host dtrace init failed: %d\n", ret); return ret; } - dev_dbg(sdev->dev, "%s: stream_tag: %d\n", __func__, params.stream_tag); + dev_dbg(sdev->dev, "stream_tag: %d\n", params.stream_tag); /* send IPC to the DSP */ priv->dtrace_state = SOF_DTRACE_INITIALIZING; @@ -544,8 +544,7 @@ static int ipc3_dtrace_init(struct snd_sof_dev *sdev) goto table_err; priv->dma_trace_pages = ret; - dev_dbg(sdev->dev, "%s: dma_trace_pages: %d\n", __func__, - priv->dma_trace_pages); + dev_dbg(sdev->dev, "dma_trace_pages: %d\n", priv->dma_trace_pages); if (sdev->first_boot) { ret = debugfs_create_dtrace(sdev); -- cgit v1.2.3 From e16809a74f09b2c2e066b3d7cf1d87be2a75911e Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 16 Jun 2022 16:53:45 -0500 Subject: ASoC: SOF: ipc3-loader: remove use of __func__ in dev_dbg MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The module and function information can be added with 'modprobe foo dyndbg=+pmf' Suggested-by: Greg KH Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20220616215351.135643-8-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc3-loader.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/ipc3-loader.c b/sound/soc/sof/ipc3-loader.c index f3c741b49519..c573e7593808 100644 --- a/sound/soc/sof/ipc3-loader.c +++ b/sound/soc/sof/ipc3-loader.c @@ -75,13 +75,12 @@ static int ipc3_fw_ext_man_get_config_data(struct snd_sof_dev *sdev, elems_size = config->hdr.size - sizeof(struct sof_ext_man_elem_header); elems_counter = elems_size / sizeof(struct sof_config_elem); - dev_dbg(sdev->dev, "%s can hold up to %d config elements\n", - __func__, elems_counter); + dev_dbg(sdev->dev, "manifest can hold up to %d config elements\n", elems_counter); for (i = 0; i < elems_counter; ++i) { elem = &config->elems[i]; - dev_dbg(sdev->dev, "%s get index %d token %d val %d\n", - __func__, i, elem->token, elem->value); + dev_dbg(sdev->dev, "get index %d token %d val %d\n", + i, elem->token, elem->value); switch (elem->token) { case SOF_EXT_MAN_CONFIG_EMPTY: /* unused memory space is zero filled - mapped to EMPTY elements */ @@ -323,10 +322,10 @@ static int sof_ipc3_load_fw_to_dsp(struct snd_sof_dev *sdev) header = (struct snd_sof_fw_header *)(fw->data + plat_data->fw_offset); load_module = sof_ops(sdev)->load_module; if (!load_module) { - dev_dbg(sdev->dev, "%s: Using generic module loading\n", __func__); + dev_dbg(sdev->dev, "Using generic module loading\n"); load_module = sof_ipc3_parse_module_memcpy; } else { - dev_dbg(sdev->dev, "%s: Using custom module loading\n", __func__); + dev_dbg(sdev->dev, "Using custom module loading\n"); } /* parse each module */ -- cgit v1.2.3 From f132dc020270976fe83c86f8c826890873023980 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 16 Jun 2022 16:53:46 -0500 Subject: ASoC: SOF: ipc3-topology: remove use of __func__ in dev_dbg MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The module and function information can be added with 'modprobe foo dyndbg=+pmf' Suggested-by: Greg KH Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20220616215351.135643-9-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc3-topology.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/ipc3-topology.c b/sound/soc/sof/ipc3-topology.c index 28d3c1414572..99b62fe7a95c 100644 --- a/sound/soc/sof/ipc3-topology.c +++ b/sound/soc/sof/ipc3-topology.c @@ -1452,8 +1452,8 @@ static int sof_ipc3_widget_setup_comp_dai(struct snd_sof_widget *swidget) if (ret < 0) goto free; - dev_dbg(scomp->dev, "%s dai %s: type %d index %d\n", - __func__, swidget->widget->name, comp_dai->type, comp_dai->dai_index); + dev_dbg(scomp->dev, "dai %s: type %d index %d\n", + swidget->widget->name, comp_dai->type, comp_dai->dai_index); sof_dbg_comp_config(scomp, &comp_dai->config); /* now update DAI config */ -- cgit v1.2.3 From 3809264b53906b8b666b93831ecc23a00e119b68 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 16 Jun 2022 16:53:47 -0500 Subject: ASoC: SOF: ipc4-topology remove use of __func__ in dev_dbg MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The module and function information can be added with 'modprobe foo dyndbg=+pmf' Suggested-by: Greg KH Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20220616215351.135643-10-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-topology.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index 3c949298e007..34f805431f2e 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -804,8 +804,8 @@ static int sof_ipc4_init_audio_fmt(struct snd_sof_dev *sdev, valid_bits = SOF_IPC4_AUDIO_FORMAT_CFG_V_BIT_DEPTH(fmt->fmt_cfg); if (params_rate(params) == rate && params_channels(params) == channels && sample_valid_bits == valid_bits) { - dev_dbg(sdev->dev, "%s: matching audio format index for %uHz, %ubit, %u channels: %d\n", - __func__, rate, valid_bits, channels, i); + dev_dbg(sdev->dev, "matching audio format index for %uHz, %ubit, %u channels: %d\n", + rate, valid_bits, channels, i); /* copy ibs/obs and input format */ memcpy(base_config, &available_fmt->base_config[i], @@ -919,8 +919,8 @@ static int snd_sof_get_hw_config_params(struct snd_sof_dev *sdev, struct snd_sof *channel_count = le32_to_cpu(hw_config->tdm_slots); *sample_rate = le32_to_cpu(hw_config->fsync_rate); - dev_dbg(sdev->dev, "%s: sample rate: %d sample width: %d channels: %d\n", - __func__, *sample_rate, *bit_depth, *channel_count); + dev_dbg(sdev->dev, "sample rate: %d sample width: %d channels: %d\n", + *sample_rate, *bit_depth, *channel_count); return 0; } @@ -954,8 +954,8 @@ static int snd_sof_get_nhlt_endpoint_data(struct snd_sof_dev *sdev, struct snd_s return 0; } - dev_dbg(sdev->dev, "%s: dai index %d nhlt type %d direction %d\n", - __func__, dai_index, nhlt_type, dir); + dev_dbg(sdev->dev, "dai index %d nhlt type %d direction %d\n", + dai_index, nhlt_type, dir); /* find NHLT blob with matching params */ cfg = intel_nhlt_get_endpoint_blob(sdev->dev, ipc4_data->nhlt, dai_index, nhlt_type, @@ -1005,7 +1005,7 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, u32 **data; int ipc_size, ret; - dev_dbg(sdev->dev, "%s: copier %s, type %d", __func__, swidget->widget->name, swidget->id); + dev_dbg(sdev->dev, "copier %s, type %d", swidget->widget->name, swidget->id); switch (swidget->id) { case snd_soc_dapm_aif_in: @@ -1446,7 +1446,7 @@ static int sof_ipc4_route_setup(struct snd_sof_dev *sdev, struct snd_sof_route * int dst_queue = 0; int ret; - dev_dbg(sdev->dev, "%s: bind %s -> %s\n", __func__, + dev_dbg(sdev->dev, "bind %s -> %s\n", src_widget->widget->name, sink_widget->widget->name); header = src_fw_module->man4_module_entry.id; @@ -1483,7 +1483,7 @@ static int sof_ipc4_route_free(struct snd_sof_dev *sdev, struct snd_sof_route *s int dst_queue = 0; int ret; - dev_dbg(sdev->dev, "%s: unbind modules %s -> %s\n", __func__, + dev_dbg(sdev->dev, "unbind modules %s -> %s\n", src_widget->widget->name, sink_widget->widget->name); header = src_fw_module->man4_module_entry.id; -- cgit v1.2.3 From 298e3aba1b56d19dcb70e10ffe93057d1ddd18f6 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 16 Jun 2022 16:53:48 -0500 Subject: ASoC: SOF: sof-client: remove use of __func__ in dev_dbg MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The module and function information can be added with 'modprobe foo dyndbg=+pmf' Suggested-by: Greg KH Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20220616215351.135643-11-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/sof-client.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/sof-client.c b/sound/soc/sof/sof-client.c index 16cca666bb85..125aa2137195 100644 --- a/sound/soc/sof/sof-client.c +++ b/sound/soc/sof/sof-client.c @@ -383,8 +383,8 @@ void sof_client_ipc_rx_dispatcher(struct snd_sof_dev *sdev, void *msg_buf) msg_type = SOF_IPC4_NOTIFICATION_TYPE_GET(msg->primary); } else { - dev_dbg_once(sdev->dev, "%s: Not supported IPC version: %d\n", - __func__, sdev->pdata->ipc_type); + dev_dbg_once(sdev->dev, "Not supported IPC version: %d\n", + sdev->pdata->ipc_type); return; } -- cgit v1.2.3 From b3ec3eb2baaad057631ab7e09c38ab3ad5c7a42b Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 16 Jun 2022 16:53:49 -0500 Subject: ASoC: SOF: ipc4: remove use of __func__ in dev_dbg MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The module and function information can be added with 'modprobe foo dyndbg=+pmf' Suggested-by: Greg KH Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20220616215351.135643-12-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sof/ipc4.c b/sound/soc/sof/ipc4.c index a8dea5abf265..432b812bdf9c 100644 --- a/sound/soc/sof/ipc4.c +++ b/sound/soc/sof/ipc4.c @@ -574,7 +574,7 @@ static void sof_ipc4_rx_msg(struct snd_sof_dev *sdev) data_size = sizeof(struct sof_ipc4_notify_resource_data); break; default: - dev_dbg(sdev->dev, "%s: Unhandled DSP message: %#x|%#x\n", __func__, + dev_dbg(sdev->dev, "Unhandled DSP message: %#x|%#x\n", ipc4_msg->primary, ipc4_msg->extension); break; } -- cgit v1.2.3 From 46bc6bc3a6a3af5306e8e3320a083cf3c32350d4 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 16 Jun 2022 16:53:50 -0500 Subject: ASoC: Intel: boards: hda: remove use of __func__ in dev_dbg MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The module and function information can be added with 'modprobe foo dyndbg=+pmf' Suggested-by: Greg KH Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20220616215351.135643-13-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/hda_dsp_common.c | 4 ++-- sound/soc/intel/boards/skl_hda_dsp_generic.c | 4 ++-- 2 files changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/hda_dsp_common.c b/sound/soc/intel/boards/hda_dsp_common.c index 5c31ddc0884a..83c7dfbccd9d 100644 --- a/sound/soc/intel/boards/hda_dsp_common.c +++ b/sound/soc/intel/boards/hda_dsp_common.c @@ -62,8 +62,8 @@ int hda_dsp_hdmi_build_controls(struct snd_soc_card *card, hpcm->pcm = spcm; hpcm->device = spcm->device; dev_dbg(card->dev, - "%s: mapping HDMI converter %d to PCM %d (%p)\n", - __func__, i, hpcm->device, spcm); + "mapping HDMI converter %d to PCM %d (%p)\n", + i, hpcm->device, spcm); } else { hpcm->pcm = NULL; hpcm->device = SNDRV_PCM_INVALID_DEVICE; diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c index f4b4eeca3e03..81144efb4b44 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_generic.c +++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c @@ -75,7 +75,7 @@ skl_hda_add_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link *link) struct skl_hda_private *ctx = snd_soc_card_get_drvdata(card); int ret = 0; - dev_dbg(card->dev, "%s: dai link name - %s\n", __func__, link->name); + dev_dbg(card->dev, "dai link name - %s\n", link->name); link->platforms->name = ctx->platform_name; link->nonatomic = 1; @@ -203,7 +203,7 @@ static int skl_hda_audio_probe(struct platform_device *pdev) struct skl_hda_private *ctx; int ret; - dev_dbg(&pdev->dev, "%s: entry\n", __func__); + dev_dbg(&pdev->dev, "entry\n"); ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) -- cgit v1.2.3 From d2d19cb6ed13eb54dd6c958f3808a23820c3ebba Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 16 Jun 2022 16:53:51 -0500 Subject: ASoC: Intel: boards: sof_sdw: remove use of __func__ in dev_dbg MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The module and function information can be added with 'modprobe foo dyndbg=+pmf' Suggested-by: Greg KH Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20220616215351.135643-14-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index d23846572543..0c47d76a79e2 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -1447,7 +1447,7 @@ static int mc_probe(struct platform_device *pdev) int amp_num = 0, i; int ret; - dev_dbg(&pdev->dev, "Entry %s\n", __func__); + dev_dbg(&pdev->dev, "Entry\n"); ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) -- cgit v1.2.3 From 7adadfb06b9839fa7d9de0cde7ad57a3be3665f0 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 16 Jun 2022 18:35:21 +0300 Subject: ASoC: twl4030: Drop legacy, non DT boot support Legacy or non DT boot is no longer possible on systems where the tw4030/5030 is used. Drop the support for handling legacy pdata and replace it with a local board_params struct to allow further cleanups on the mfd side. Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20220616153521.29701-1-peter.ujfalusi@gmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 101 ++++++++++++++++++++++++--------------------- 1 file changed, 54 insertions(+), 47 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 0ba3546ef870..87b58017094b 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -34,6 +34,14 @@ #define TWL4030_CACHEREGNUM (TWL4030_REG_MISC_SET_2 + 1) +struct twl4030_board_params { + unsigned int digimic_delay; /* in ms */ + unsigned int ramp_delay_value; + unsigned int offset_cncl_path; + unsigned int hs_extmute:1; + int hs_extmute_gpio; +}; + /* codec private data */ struct twl4030_priv { unsigned int codec_powered; @@ -58,7 +66,7 @@ struct twl4030_priv { u8 carkitl_enabled, carkitr_enabled; u8 ctl_cache[TWL4030_REG_PRECKR_CTL - TWL4030_REG_EAR_CTL + 1]; - struct twl4030_codec_data *pdata; + struct twl4030_board_params *board_params; }; static void tw4030_init_ctl_cache(struct twl4030_priv *twl4030) @@ -193,73 +201,71 @@ static void twl4030_codec_enable(struct snd_soc_component *component, int enable udelay(10); } -static void twl4030_setup_pdata_of(struct twl4030_codec_data *pdata, - struct device_node *node) +static void +twl4030_get_board_param_values(struct twl4030_board_params *board_params, + struct device_node *node) { int value; - of_property_read_u32(node, "ti,digimic_delay", - &pdata->digimic_delay); - of_property_read_u32(node, "ti,ramp_delay_value", - &pdata->ramp_delay_value); - of_property_read_u32(node, "ti,offset_cncl_path", - &pdata->offset_cncl_path); + of_property_read_u32(node, "ti,digimic_delay", &board_params->digimic_delay); + of_property_read_u32(node, "ti,ramp_delay_value", &board_params->ramp_delay_value); + of_property_read_u32(node, "ti,offset_cncl_path", &board_params->offset_cncl_path); if (!of_property_read_u32(node, "ti,hs_extmute", &value)) - pdata->hs_extmute = value; + board_params->hs_extmute = value; - pdata->hs_extmute_gpio = of_get_named_gpio(node, - "ti,hs_extmute_gpio", 0); - if (gpio_is_valid(pdata->hs_extmute_gpio)) - pdata->hs_extmute = 1; + board_params->hs_extmute_gpio = of_get_named_gpio(node, "ti,hs_extmute_gpio", 0); + if (gpio_is_valid(board_params->hs_extmute_gpio)) + board_params->hs_extmute = 1; } -static struct twl4030_codec_data *twl4030_get_pdata(struct snd_soc_component *component) +static struct twl4030_board_params* +twl4030_get_board_params(struct snd_soc_component *component) { - struct twl4030_codec_data *pdata = dev_get_platdata(component->dev); + struct twl4030_board_params *board_params = NULL; struct device_node *twl4030_codec_node = NULL; twl4030_codec_node = of_get_child_by_name(component->dev->parent->of_node, "codec"); - if (!pdata && twl4030_codec_node) { - pdata = devm_kzalloc(component->dev, - sizeof(struct twl4030_codec_data), - GFP_KERNEL); - if (!pdata) { + if (twl4030_codec_node) { + board_params = devm_kzalloc(component->dev, + sizeof(struct twl4030_board_params), + GFP_KERNEL); + if (!board_params) { of_node_put(twl4030_codec_node); return NULL; } - twl4030_setup_pdata_of(pdata, twl4030_codec_node); + twl4030_get_board_param_values(board_params, twl4030_codec_node); of_node_put(twl4030_codec_node); } - return pdata; + return board_params; } static void twl4030_init_chip(struct snd_soc_component *component) { - struct twl4030_codec_data *pdata; + struct twl4030_board_params *board_params; struct twl4030_priv *twl4030 = snd_soc_component_get_drvdata(component); u8 reg, byte; int i = 0; - pdata = twl4030_get_pdata(component); + board_params = twl4030_get_board_params(component); - if (pdata && pdata->hs_extmute) { - if (gpio_is_valid(pdata->hs_extmute_gpio)) { + if (board_params && board_params->hs_extmute) { + if (gpio_is_valid(board_params->hs_extmute_gpio)) { int ret; - if (!pdata->hs_extmute_gpio) + if (!board_params->hs_extmute_gpio) dev_warn(component->dev, "Extmute GPIO is 0 is this correct?\n"); - ret = gpio_request_one(pdata->hs_extmute_gpio, + ret = gpio_request_one(board_params->hs_extmute_gpio, GPIOF_OUT_INIT_LOW, "hs_extmute"); if (ret) { dev_err(component->dev, "Failed to get hs_extmute GPIO\n"); - pdata->hs_extmute_gpio = -1; + board_params->hs_extmute_gpio = -1; } } else { u8 pin_mux; @@ -290,14 +296,14 @@ static void twl4030_init_chip(struct snd_soc_component *component) twl4030_write(component, TWL4030_REG_ARXR2_APGA_CTL, 0x32); /* Machine dependent setup */ - if (!pdata) + if (!board_params) return; - twl4030->pdata = pdata; + twl4030->board_params = board_params; reg = twl4030_read(component, TWL4030_REG_HS_POPN_SET); reg &= ~TWL4030_RAMP_DELAY; - reg |= (pdata->ramp_delay_value << 2); + reg |= (board_params->ramp_delay_value << 2); twl4030_write(component, TWL4030_REG_HS_POPN_SET, reg); /* initiate offset cancellation */ @@ -305,7 +311,7 @@ static void twl4030_init_chip(struct snd_soc_component *component) reg = twl4030_read(component, TWL4030_REG_ANAMICL); reg &= ~TWL4030_OFFSET_CNCL_SEL; - reg |= pdata->offset_cncl_path; + reg |= board_params->offset_cncl_path; twl4030_write(component, TWL4030_REG_ANAMICL, reg | TWL4030_CNCL_OFFSET_START); @@ -692,7 +698,7 @@ static void headset_ramp(struct snd_soc_component *component, int ramp) { unsigned char hs_gain, hs_pop; struct twl4030_priv *twl4030 = snd_soc_component_get_drvdata(component); - struct twl4030_codec_data *pdata = twl4030->pdata; + struct twl4030_board_params *board_params = twl4030->board_params; /* Base values for ramp delay calculation: 2^19 - 2^26 */ unsigned int ramp_base[] = {524288, 1048576, 2097152, 4194304, 8388608, 16777216, 33554432, 67108864}; @@ -705,9 +711,9 @@ static void headset_ramp(struct snd_soc_component *component, int ramp) /* Enable external mute control, this dramatically reduces * the pop-noise */ - if (pdata && pdata->hs_extmute) { - if (gpio_is_valid(pdata->hs_extmute_gpio)) { - gpio_set_value(pdata->hs_extmute_gpio, 1); + if (board_params && board_params->hs_extmute) { + if (gpio_is_valid(board_params->hs_extmute_gpio)) { + gpio_set_value(board_params->hs_extmute_gpio, 1); } else { hs_pop |= TWL4030_EXTMUTE; twl4030_write(component, TWL4030_REG_HS_POPN_SET, hs_pop); @@ -741,9 +747,9 @@ static void headset_ramp(struct snd_soc_component *component, int ramp) } /* Disable external mute */ - if (pdata && pdata->hs_extmute) { - if (gpio_is_valid(pdata->hs_extmute_gpio)) { - gpio_set_value(pdata->hs_extmute_gpio, 0); + if (board_params && board_params->hs_extmute) { + if (gpio_is_valid(board_params->hs_extmute_gpio)) { + gpio_set_value(board_params->hs_extmute_gpio, 0); } else { hs_pop &= ~TWL4030_EXTMUTE; twl4030_write(component, TWL4030_REG_HS_POPN_SET, hs_pop); @@ -806,10 +812,10 @@ static int digimic_event(struct snd_soc_dapm_widget *w, { struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); struct twl4030_priv *twl4030 = snd_soc_component_get_drvdata(component); - struct twl4030_codec_data *pdata = twl4030->pdata; + struct twl4030_board_params *board_params = twl4030->board_params; - if (pdata && pdata->digimic_delay) - twl4030_wait_ms(pdata->digimic_delay); + if (board_params && board_params->digimic_delay) + twl4030_wait_ms(board_params->digimic_delay); return 0; } @@ -2168,10 +2174,11 @@ static int twl4030_soc_probe(struct snd_soc_component *component) static void twl4030_soc_remove(struct snd_soc_component *component) { struct twl4030_priv *twl4030 = snd_soc_component_get_drvdata(component); - struct twl4030_codec_data *pdata = twl4030->pdata; + struct twl4030_board_params *board_params = twl4030->board_params; - if (pdata && pdata->hs_extmute && gpio_is_valid(pdata->hs_extmute_gpio)) - gpio_free(pdata->hs_extmute_gpio); + if (board_params && board_params->hs_extmute && + gpio_is_valid(board_params->hs_extmute_gpio)) + gpio_free(board_params->hs_extmute_gpio); } static const struct snd_soc_component_driver soc_component_dev_twl4030 = { -- cgit v1.2.3 From 442302003bd2b151e12d52b0af9a7dac131bf931 Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Fri, 17 Jun 2022 16:36:06 +0100 Subject: ASoC: ops: Fix integer detection for when max possible values > 1 The standard snd_soc_info_volsw() allows a two value control to be defined as an integer control only if the control name ends in "Volume". It achieves this by creating a substring if it contains " Volume", and ensuring this exists at the end of the name. The volume substring is then used to decide whether the type is a SNDRV_CTL_ELEM_TYPE_INTEGER or SNDRV_CTL_ELEM_TYPE_BOOLEAN. However this volume substring is only computed for a two value control. This means for controls where there are more than two possible values, the substring is never created, so in this case the substring remains NULL, and the condition yields SNDRV_CTL_ELEM_TYPE_BOOLEAN, even though there are more than 2 possible values. If there are more than 2 possible values for the control, then it should always be an integer control. Fixes: aa2a4b897132 ("ASoC: ops: Fix boolean/integer detection for simple controls") Signed-off-by: Stefan Binding Reviewed-by: Charles Keepax Link: https://lore.kernel.org/r/20220617153606.2619457-1-sbinding@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/soc-ops.c | 15 +++++++-------- 1 file changed, 7 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index c22d87581f6f..bd88de056358 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -183,17 +183,16 @@ int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, if (mc->platform_max && mc->platform_max < max) max = mc->platform_max; - /* Even two value controls ending in Volume should always be integer */ if (max == 1) { + /* Even two value controls ending in Volume should always be integer */ vol_string = strstr(kcontrol->id.name, " Volume"); - if (vol_string && strcmp(vol_string, " Volume")) - vol_string = NULL; - } - - if (!vol_string) - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - else + if (vol_string && !strcmp(vol_string, " Volume")) + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + else + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + } else { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + } uinfo->count = snd_soc_volsw_is_stereo(mc) ? 2 : 1; uinfo->value.integer.min = 0; -- cgit v1.2.3 From 6c9e9046e1ff356bda66661213735d33c6cfea53 Mon Sep 17 00:00:00 2001 From: Fei Shao Date: Fri, 17 Jun 2022 19:10:04 +0800 Subject: ASoC: mediatek: mt8186: Fix mutex double unlock in GPIO request The lockdep mechanism revealed an unbalanced unlocking on MT8186: [ 2.993966] WARNING: bad unlock balance detected! [ 2.993978] ------------------------------------- [ 2.993983] kworker/u16:1/10 is trying to release lock (gpio_request_mutex) at: [ 2.993994] [] mt8186_afe_gpio_request+0xf8/0x210 [ 2.994012] but there are no more locks to release! The cause is that the mutex will be double unlocked if dai is unknown during GPIO selection, and this patch fixes it. Fixes: cfa9a966f12a ("ASoC: mediatek: mt8186: support gpio control in platform driver") Signed-off-by: Fei Shao Link: https://lore.kernel.org/r/20220617111003.2014395-1-fshao@chromium.org Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8186/mt8186-afe-gpio.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/mediatek/mt8186/mt8186-afe-gpio.c b/sound/soc/mediatek/mt8186/mt8186-afe-gpio.c index 255ffba637d3..274c0c8ec2f2 100644 --- a/sound/soc/mediatek/mt8186/mt8186-afe-gpio.c +++ b/sound/soc/mediatek/mt8186/mt8186-afe-gpio.c @@ -230,7 +230,6 @@ int mt8186_afe_gpio_request(struct device *dev, bool enable, sel = enable ? MT8186_AFE_GPIO_PCM_ON : MT8186_AFE_GPIO_PCM_OFF; break; default: - mutex_unlock(&gpio_request_mutex); dev_err(dev, "%s(), invalid dai %d\n", __func__, dai); goto unlock; } -- cgit v1.2.3 From 4ea3bfd13a2484b5f1c19f60b1dc7494f261f0a4 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 16 Jun 2022 16:08:24 -0500 Subject: ASoC: SOF: pcm: use pm_resume_and_get() on component probe MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Before initiating IPC and/or bus transactions when loading the topology during a component probe, which happens on card registration/creation, make sure the device for the SOF driver is pm_runtime active. The SOF probe is not necessarily followed by the component probe, such a timing assumption can be broken in driver bind/unbind tests. This can be artifially shown if the module for the machine driver is 'blacklisted' and the SOF device becomes pm_runtime_suspended before manually calling modprobe to register the card. In an initial experiment, pm_resume_and_get() was called from soc-component.c, since the current ASoC component model is arguably missing dependencies between component status and device status. However this approach proved too invasive and breaks all existing HDMI playback solutions on Intel platforms. While this will result in duplication of code, generating pm_runtime transitions only if strictly required for a given component makes more sense overall. This patch adds the pm_runtime resume transition for SOF only. BugLink: https://github.com/thesofproject/linux/issues/3651 Signed-off-by: Pierre-Louis Bossart Reviewed-by: Rander Wang Reviewed-by: Ranjani Sridharan Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Link: https://lore.kernel.org/r/20220616210825.132093-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/pcm.c | 11 +++++++++++ 1 file changed, 11 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c index a76d0b5b2ad9..27504abc5385 100644 --- a/sound/soc/sof/pcm.c +++ b/sound/soc/sof/pcm.c @@ -604,6 +604,14 @@ static int sof_pcm_probe(struct snd_soc_component *component) const char *tplg_filename; int ret; + /* + * make sure the device is pm_runtime_active before loading the + * topology and initiating IPC or bus transactions + */ + ret = pm_runtime_resume_and_get(component->dev); + if (ret < 0 && ret != -EACCES) + return ret; + /* load the default topology */ sdev->component = component; @@ -621,6 +629,9 @@ static int sof_pcm_probe(struct snd_soc_component *component) return ret; } + pm_runtime_mark_last_busy(component->dev); + pm_runtime_put_autosuspend(component->dev); + return ret; } -- cgit v1.2.3 From 011e397f5c9c96e533d4a244af84e74c9caefb83 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 16 Jun 2022 16:08:25 -0500 Subject: ASoC: codecs: soundwire: call pm_runtime_resume() in component probe MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Make sure that the bus and codecs are pm_runtime active when the card is registered/created. This avoid timeouts when accessing registers. BugLink: https://github.com/thesofproject/linux/issues/3651 BugLink: https://github.com/thesofproject/linux/issues/3650 Signed-off-by: Pierre-Louis Bossart Reviewed-by: Rander Wang Reviewed-by: Ranjani Sridharan Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Link: https://lore.kernel.org/r/20220616210825.132093-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/max98373.c | 14 +++++++++++++- sound/soc/codecs/rt1308-sdw.c | 12 ++++++++++++ sound/soc/codecs/rt1316-sdw.c | 12 ++++++++++++ sound/soc/codecs/rt700.c | 5 +++++ sound/soc/codecs/rt711-sdca.c | 5 +++++ sound/soc/codecs/rt711.c | 5 +++++ sound/soc/codecs/rt715-sdca.c | 12 ++++++++++++ sound/soc/codecs/rt715.c | 12 ++++++++++++ 8 files changed, 76 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index e14fe98349a5..1517c47afbf1 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -5,6 +5,7 @@ #include #include #include +#include #include #include #include @@ -440,8 +441,19 @@ const struct snd_soc_component_driver soc_codec_dev_max98373 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_max98373); +static int max98373_sdw_probe(struct snd_soc_component *component) +{ + int ret; + + ret = pm_runtime_resume(component->dev); + if (ret < 0 && ret != -EACCES) + return ret; + + return 0; +} + const struct snd_soc_component_driver soc_codec_dev_max98373_sdw = { - .probe = NULL, + .probe = max98373_sdw_probe, .controls = max98373_snd_controls, .num_controls = ARRAY_SIZE(max98373_snd_controls), .dapm_widgets = max98373_dapm_widgets, diff --git a/sound/soc/codecs/rt1308-sdw.c b/sound/soc/codecs/rt1308-sdw.c index 72f673f278ee..0be6e72ff5a9 100644 --- a/sound/soc/codecs/rt1308-sdw.c +++ b/sound/soc/codecs/rt1308-sdw.c @@ -608,7 +608,19 @@ static const struct sdw_slave_ops rt1308_slave_ops = { .bus_config = rt1308_bus_config, }; +static int rt1308_sdw_component_probe(struct snd_soc_component *component) +{ + int ret; + + ret = pm_runtime_resume(component->dev); + if (ret < 0 && ret != -EACCES) + return ret; + + return 0; +} + static const struct snd_soc_component_driver soc_component_sdw_rt1308 = { + .probe = rt1308_sdw_component_probe, .controls = rt1308_snd_controls, .num_controls = ARRAY_SIZE(rt1308_snd_controls), .dapm_widgets = rt1308_dapm_widgets, diff --git a/sound/soc/codecs/rt1316-sdw.c b/sound/soc/codecs/rt1316-sdw.c index 2d6b5f9d4d77..e53396606a1c 100644 --- a/sound/soc/codecs/rt1316-sdw.c +++ b/sound/soc/codecs/rt1316-sdw.c @@ -590,7 +590,19 @@ static struct sdw_slave_ops rt1316_slave_ops = { .update_status = rt1316_update_status, }; +static int rt1316_sdw_component_probe(struct snd_soc_component *component) +{ + int ret; + + ret = pm_runtime_resume(component->dev); + if (ret < 0 && ret != -EACCES) + return ret; + + return 0; +} + static const struct snd_soc_component_driver soc_component_sdw_rt1316 = { + .probe = rt1316_sdw_component_probe, .controls = rt1316_snd_controls, .num_controls = ARRAY_SIZE(rt1316_snd_controls), .dapm_widgets = rt1316_dapm_widgets, diff --git a/sound/soc/codecs/rt700.c b/sound/soc/codecs/rt700.c index 9bceeeb830b1..055c3ae974d8 100644 --- a/sound/soc/codecs/rt700.c +++ b/sound/soc/codecs/rt700.c @@ -818,9 +818,14 @@ static const struct snd_soc_dapm_route rt700_audio_map[] = { static int rt700_probe(struct snd_soc_component *component) { struct rt700_priv *rt700 = snd_soc_component_get_drvdata(component); + int ret; rt700->component = component; + ret = pm_runtime_resume(component->dev); + if (ret < 0 && ret != -EACCES) + return ret; + return 0; } diff --git a/sound/soc/codecs/rt711-sdca.c b/sound/soc/codecs/rt711-sdca.c index dfe3c9299ebd..9d226b1cb7e9 100644 --- a/sound/soc/codecs/rt711-sdca.c +++ b/sound/soc/codecs/rt711-sdca.c @@ -1194,10 +1194,15 @@ static int rt711_sdca_parse_dt(struct rt711_sdca_priv *rt711, struct device *dev static int rt711_sdca_probe(struct snd_soc_component *component) { struct rt711_sdca_priv *rt711 = snd_soc_component_get_drvdata(component); + int ret; rt711_sdca_parse_dt(rt711, &rt711->slave->dev); rt711->component = component; + ret = pm_runtime_resume(component->dev); + if (ret < 0 && ret != -EACCES) + return ret; + return 0; } diff --git a/sound/soc/codecs/rt711.c b/sound/soc/codecs/rt711.c index 9df800abfc2d..1bf618089194 100644 --- a/sound/soc/codecs/rt711.c +++ b/sound/soc/codecs/rt711.c @@ -935,10 +935,15 @@ static int rt711_parse_dt(struct rt711_priv *rt711, struct device *dev) static int rt711_probe(struct snd_soc_component *component) { struct rt711_priv *rt711 = snd_soc_component_get_drvdata(component); + int ret; rt711_parse_dt(rt711, &rt711->slave->dev); rt711->component = component; + ret = pm_runtime_resume(component->dev); + if (ret < 0 && ret != -EACCES) + return ret; + return 0; } diff --git a/sound/soc/codecs/rt715-sdca.c b/sound/soc/codecs/rt715-sdca.c index 5857d0866307..ce8bbc76199a 100644 --- a/sound/soc/codecs/rt715-sdca.c +++ b/sound/soc/codecs/rt715-sdca.c @@ -758,7 +758,19 @@ static const struct snd_soc_dapm_route rt715_sdca_audio_map[] = { {"ADC 25 Mux", "DMIC4", "DMIC4"}, }; +static int rt715_sdca_probe(struct snd_soc_component *component) +{ + int ret; + + ret = pm_runtime_resume(component->dev); + if (ret < 0 && ret != -EACCES) + return ret; + + return 0; +} + static const struct snd_soc_component_driver soc_codec_dev_rt715_sdca = { + .probe = rt715_sdca_probe, .controls = rt715_sdca_snd_controls, .num_controls = ARRAY_SIZE(rt715_sdca_snd_controls), .dapm_widgets = rt715_sdca_dapm_widgets, diff --git a/sound/soc/codecs/rt715.c b/sound/soc/codecs/rt715.c index 418e006b19ef..e93240521c74 100644 --- a/sound/soc/codecs/rt715.c +++ b/sound/soc/codecs/rt715.c @@ -737,7 +737,19 @@ static int rt715_set_bias_level(struct snd_soc_component *component, return 0; } +static int rt715_probe(struct snd_soc_component *component) +{ + int ret; + + ret = pm_runtime_resume(component->dev); + if (ret < 0 && ret != -EACCES) + return ret; + + return 0; +} + static const struct snd_soc_component_driver soc_codec_dev_rt715 = { + .probe = rt715_probe, .set_bias_level = rt715_set_bias_level, .controls = rt715_snd_controls, .num_controls = ARRAY_SIZE(rt715_snd_controls), -- cgit v1.2.3 From cd76175a2b204911a3cddef36b99e56945b6938c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 17 Jun 2022 16:40:47 +0200 Subject: ALSA: rawmidi: Make internal functions local static __snd_rawmidi_transmit_peek() and __snd_rawmidi_transmit_ack() are never called from the outside. Let's make them local static and unexport them. Link: https://lore.kernel.org/r/20220617144051.18985-2-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/rawmidi.c | 13 ++++++------- 1 file changed, 6 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index befa9809ff00..82e8f656bbb2 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -1258,7 +1258,7 @@ int snd_rawmidi_transmit_empty(struct snd_rawmidi_substream *substream) } EXPORT_SYMBOL(snd_rawmidi_transmit_empty); -/** +/* * __snd_rawmidi_transmit_peek - copy data from the internal buffer * @substream: the rawmidi substream * @buffer: the buffer pointer @@ -1266,8 +1266,8 @@ EXPORT_SYMBOL(snd_rawmidi_transmit_empty); * * This is a variant of snd_rawmidi_transmit_peek() without spinlock. */ -int __snd_rawmidi_transmit_peek(struct snd_rawmidi_substream *substream, - unsigned char *buffer, int count) +static int __snd_rawmidi_transmit_peek(struct snd_rawmidi_substream *substream, + unsigned char *buffer, int count) { int result, count1; struct snd_rawmidi_runtime *runtime = substream->runtime; @@ -1304,7 +1304,6 @@ int __snd_rawmidi_transmit_peek(struct snd_rawmidi_substream *substream, __skip: return result; } -EXPORT_SYMBOL(__snd_rawmidi_transmit_peek); /** * snd_rawmidi_transmit_peek - copy data from the internal buffer @@ -1334,14 +1333,15 @@ int snd_rawmidi_transmit_peek(struct snd_rawmidi_substream *substream, } EXPORT_SYMBOL(snd_rawmidi_transmit_peek); -/** +/* * __snd_rawmidi_transmit_ack - acknowledge the transmission * @substream: the rawmidi substream * @count: the transferred count * * This is a variant of __snd_rawmidi_transmit_ack() without spinlock. */ -int __snd_rawmidi_transmit_ack(struct snd_rawmidi_substream *substream, int count) +static int __snd_rawmidi_transmit_ack(struct snd_rawmidi_substream *substream, + int count) { struct snd_rawmidi_runtime *runtime = substream->runtime; @@ -1361,7 +1361,6 @@ int __snd_rawmidi_transmit_ack(struct snd_rawmidi_substream *substream, int coun } return count; } -EXPORT_SYMBOL(__snd_rawmidi_transmit_ack); /** * snd_rawmidi_transmit_ack - acknowledge the transmission -- cgit v1.2.3 From f1d40433352e5d4babd59c0dd50b5f9414073ddb Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 17 Jun 2022 16:40:48 +0200 Subject: ALSA: rawmidi: Move lock to snd_rawmidi_substream Having a lock in snd_rawmidi_runtime can be a problem especially when a substream is accessed from the outside, as the runtime creation might be racy with the external calls. As a first step for hardening, move the spinlock from snd_rawmidi_runtime to snd_rawmidi_substream. This patch just replaces the lock calls, no real functional change is put yet. Link: https://lore.kernel.org/r/20220617144051.18985-3-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/rawmidi.c | 131 +++++++++++++++++++++++++-------------------------- 1 file changed, 64 insertions(+), 67 deletions(-) (limited to 'sound') diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 82e8f656bbb2..0a00f37d8c42 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -102,13 +102,12 @@ static inline bool __snd_rawmidi_ready(struct snd_rawmidi_runtime *runtime) static bool snd_rawmidi_ready(struct snd_rawmidi_substream *substream) { - struct snd_rawmidi_runtime *runtime = substream->runtime; unsigned long flags; bool ready; - spin_lock_irqsave(&runtime->lock, flags); - ready = __snd_rawmidi_ready(runtime); - spin_unlock_irqrestore(&runtime->lock, flags); + spin_lock_irqsave(&substream->lock, flags); + ready = __snd_rawmidi_ready(substream->runtime); + spin_unlock_irqrestore(&substream->lock, flags); return ready; } @@ -130,7 +129,7 @@ static void snd_rawmidi_input_event_work(struct work_struct *work) runtime->event(runtime->substream); } -/* buffer refcount management: call with runtime->lock held */ +/* buffer refcount management: call with substream->lock held */ static inline void snd_rawmidi_buffer_ref(struct snd_rawmidi_runtime *runtime) { runtime->buffer_ref++; @@ -149,7 +148,6 @@ static int snd_rawmidi_runtime_create(struct snd_rawmidi_substream *substream) if (!runtime) return -ENOMEM; runtime->substream = substream; - spin_lock_init(&runtime->lock); init_waitqueue_head(&runtime->sleep); INIT_WORK(&runtime->event_work, snd_rawmidi_input_event_work); runtime->event = NULL; @@ -203,20 +201,20 @@ static void __reset_runtime_ptrs(struct snd_rawmidi_runtime *runtime, runtime->avail = is_input ? 0 : runtime->buffer_size; } -static void reset_runtime_ptrs(struct snd_rawmidi_runtime *runtime, +static void reset_runtime_ptrs(struct snd_rawmidi_substream *substream, bool is_input) { unsigned long flags; - spin_lock_irqsave(&runtime->lock, flags); - __reset_runtime_ptrs(runtime, is_input); - spin_unlock_irqrestore(&runtime->lock, flags); + spin_lock_irqsave(&substream->lock, flags); + __reset_runtime_ptrs(substream->runtime, is_input); + spin_unlock_irqrestore(&substream->lock, flags); } int snd_rawmidi_drop_output(struct snd_rawmidi_substream *substream) { snd_rawmidi_output_trigger(substream, 0); - reset_runtime_ptrs(substream->runtime, false); + reset_runtime_ptrs(substream, false); return 0; } EXPORT_SYMBOL(snd_rawmidi_drop_output); @@ -256,7 +254,7 @@ EXPORT_SYMBOL(snd_rawmidi_drain_output); int snd_rawmidi_drain_input(struct snd_rawmidi_substream *substream) { snd_rawmidi_input_trigger(substream, 0); - reset_runtime_ptrs(substream->runtime, true); + reset_runtime_ptrs(substream, true); return 0; } EXPORT_SYMBOL(snd_rawmidi_drain_input); @@ -676,10 +674,11 @@ static int snd_rawmidi_info_select_user(struct snd_card *card, return 0; } -static int resize_runtime_buffer(struct snd_rawmidi_runtime *runtime, +static int resize_runtime_buffer(struct snd_rawmidi_substream *substream, struct snd_rawmidi_params *params, bool is_input) { + struct snd_rawmidi_runtime *runtime = substream->runtime; char *newbuf, *oldbuf; unsigned int framing = params->mode & SNDRV_RAWMIDI_MODE_FRAMING_MASK; @@ -693,9 +692,9 @@ static int resize_runtime_buffer(struct snd_rawmidi_runtime *runtime, newbuf = kvzalloc(params->buffer_size, GFP_KERNEL); if (!newbuf) return -ENOMEM; - spin_lock_irq(&runtime->lock); + spin_lock_irq(&substream->lock); if (runtime->buffer_ref) { - spin_unlock_irq(&runtime->lock); + spin_unlock_irq(&substream->lock); kvfree(newbuf); return -EBUSY; } @@ -703,7 +702,7 @@ static int resize_runtime_buffer(struct snd_rawmidi_runtime *runtime, runtime->buffer = newbuf; runtime->buffer_size = params->buffer_size; __reset_runtime_ptrs(runtime, is_input); - spin_unlock_irq(&runtime->lock); + spin_unlock_irq(&substream->lock); kvfree(oldbuf); } runtime->avail_min = params->avail_min; @@ -717,7 +716,7 @@ int snd_rawmidi_output_params(struct snd_rawmidi_substream *substream, return -EBUSY; snd_rawmidi_drain_output(substream); substream->active_sensing = !params->no_active_sensing; - return resize_runtime_buffer(substream->runtime, params, false); + return resize_runtime_buffer(substream, params, false); } EXPORT_SYMBOL(snd_rawmidi_output_params); @@ -735,7 +734,7 @@ int snd_rawmidi_input_params(struct snd_rawmidi_substream *substream, if (framing > SNDRV_RAWMIDI_MODE_FRAMING_TSTAMP) return -EINVAL; snd_rawmidi_drain_input(substream); - err = resize_runtime_buffer(substream->runtime, params, true); + err = resize_runtime_buffer(substream, params, true); if (err < 0) return err; @@ -752,9 +751,9 @@ static int snd_rawmidi_output_status(struct snd_rawmidi_substream *substream, memset(status, 0, sizeof(*status)); status->stream = SNDRV_RAWMIDI_STREAM_OUTPUT; - spin_lock_irq(&runtime->lock); + spin_lock_irq(&substream->lock); status->avail = runtime->avail; - spin_unlock_irq(&runtime->lock); + spin_unlock_irq(&substream->lock); return 0; } @@ -765,11 +764,11 @@ static int snd_rawmidi_input_status(struct snd_rawmidi_substream *substream, memset(status, 0, sizeof(*status)); status->stream = SNDRV_RAWMIDI_STREAM_INPUT; - spin_lock_irq(&runtime->lock); + spin_lock_irq(&substream->lock); status->avail = runtime->avail; status->xruns = runtime->xruns; runtime->xruns = 0; - spin_unlock_irq(&runtime->lock); + spin_unlock_irq(&substream->lock); return 0; } @@ -1074,7 +1073,7 @@ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream, return -EINVAL; } - spin_lock_irqsave(&runtime->lock, flags); + spin_lock_irqsave(&substream->lock, flags); if (substream->framing == SNDRV_RAWMIDI_MODE_FRAMING_TSTAMP) { result = receive_with_tstamp_framing(substream, buffer, count, &ts64); } else if (count == 1) { /* special case, faster code */ @@ -1121,7 +1120,7 @@ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream, else if (__snd_rawmidi_ready(runtime)) wake_up(&runtime->sleep); } - spin_unlock_irqrestore(&runtime->lock, flags); + spin_unlock_irqrestore(&substream->lock, flags); return result; } EXPORT_SYMBOL(snd_rawmidi_receive); @@ -1136,7 +1135,7 @@ static long snd_rawmidi_kernel_read1(struct snd_rawmidi_substream *substream, unsigned long appl_ptr; int err = 0; - spin_lock_irqsave(&runtime->lock, flags); + spin_lock_irqsave(&substream->lock, flags); snd_rawmidi_buffer_ref(runtime); while (count > 0 && runtime->avail) { count1 = runtime->buffer_size - runtime->appl_ptr; @@ -1154,11 +1153,11 @@ static long snd_rawmidi_kernel_read1(struct snd_rawmidi_substream *substream, if (kernelbuf) memcpy(kernelbuf + result, runtime->buffer + appl_ptr, count1); if (userbuf) { - spin_unlock_irqrestore(&runtime->lock, flags); + spin_unlock_irqrestore(&substream->lock, flags); if (copy_to_user(userbuf + result, runtime->buffer + appl_ptr, count1)) err = -EFAULT; - spin_lock_irqsave(&runtime->lock, flags); + spin_lock_irqsave(&substream->lock, flags); if (err) goto out; } @@ -1167,7 +1166,7 @@ static long snd_rawmidi_kernel_read1(struct snd_rawmidi_substream *substream, } out: snd_rawmidi_buffer_unref(runtime); - spin_unlock_irqrestore(&runtime->lock, flags); + spin_unlock_irqrestore(&substream->lock, flags); return result > 0 ? result : err; } @@ -1196,31 +1195,31 @@ static ssize_t snd_rawmidi_read(struct file *file, char __user *buf, size_t coun snd_rawmidi_input_trigger(substream, 1); result = 0; while (count > 0) { - spin_lock_irq(&runtime->lock); + spin_lock_irq(&substream->lock); while (!__snd_rawmidi_ready(runtime)) { wait_queue_entry_t wait; if ((file->f_flags & O_NONBLOCK) != 0 || result > 0) { - spin_unlock_irq(&runtime->lock); + spin_unlock_irq(&substream->lock); return result > 0 ? result : -EAGAIN; } init_waitqueue_entry(&wait, current); add_wait_queue(&runtime->sleep, &wait); set_current_state(TASK_INTERRUPTIBLE); - spin_unlock_irq(&runtime->lock); + spin_unlock_irq(&substream->lock); schedule(); remove_wait_queue(&runtime->sleep, &wait); if (rfile->rmidi->card->shutdown) return -ENODEV; if (signal_pending(current)) return result > 0 ? result : -ERESTARTSYS; - spin_lock_irq(&runtime->lock); + spin_lock_irq(&substream->lock); if (!runtime->avail) { - spin_unlock_irq(&runtime->lock); + spin_unlock_irq(&substream->lock); return result > 0 ? result : -EIO; } } - spin_unlock_irq(&runtime->lock); + spin_unlock_irq(&substream->lock); count1 = snd_rawmidi_kernel_read1(substream, (unsigned char __user *)buf, NULL/*kernelbuf*/, @@ -1251,9 +1250,9 @@ int snd_rawmidi_transmit_empty(struct snd_rawmidi_substream *substream) "snd_rawmidi_transmit_empty: output is not active!!!\n"); return 1; } - spin_lock_irqsave(&runtime->lock, flags); + spin_lock_irqsave(&substream->lock, flags); result = runtime->avail >= runtime->buffer_size; - spin_unlock_irqrestore(&runtime->lock, flags); + spin_unlock_irqrestore(&substream->lock, flags); return result; } EXPORT_SYMBOL(snd_rawmidi_transmit_empty); @@ -1322,13 +1321,12 @@ static int __snd_rawmidi_transmit_peek(struct snd_rawmidi_substream *substream, int snd_rawmidi_transmit_peek(struct snd_rawmidi_substream *substream, unsigned char *buffer, int count) { - struct snd_rawmidi_runtime *runtime = substream->runtime; int result; unsigned long flags; - spin_lock_irqsave(&runtime->lock, flags); + spin_lock_irqsave(&substream->lock, flags); result = __snd_rawmidi_transmit_peek(substream, buffer, count); - spin_unlock_irqrestore(&runtime->lock, flags); + spin_unlock_irqrestore(&substream->lock, flags); return result; } EXPORT_SYMBOL(snd_rawmidi_transmit_peek); @@ -1375,13 +1373,12 @@ static int __snd_rawmidi_transmit_ack(struct snd_rawmidi_substream *substream, */ int snd_rawmidi_transmit_ack(struct snd_rawmidi_substream *substream, int count) { - struct snd_rawmidi_runtime *runtime = substream->runtime; int result; unsigned long flags; - spin_lock_irqsave(&runtime->lock, flags); + spin_lock_irqsave(&substream->lock, flags); result = __snd_rawmidi_transmit_ack(substream, count); - spin_unlock_irqrestore(&runtime->lock, flags); + spin_unlock_irqrestore(&substream->lock, flags); return result; } EXPORT_SYMBOL(snd_rawmidi_transmit_ack); @@ -1399,11 +1396,10 @@ EXPORT_SYMBOL(snd_rawmidi_transmit_ack); int snd_rawmidi_transmit(struct snd_rawmidi_substream *substream, unsigned char *buffer, int count) { - struct snd_rawmidi_runtime *runtime = substream->runtime; int result; unsigned long flags; - spin_lock_irqsave(&runtime->lock, flags); + spin_lock_irqsave(&substream->lock, flags); if (!substream->opened) result = -EBADFD; else { @@ -1413,7 +1409,7 @@ int snd_rawmidi_transmit(struct snd_rawmidi_substream *substream, else result = __snd_rawmidi_transmit_ack(substream, count); } - spin_unlock_irqrestore(&runtime->lock, flags); + spin_unlock_irqrestore(&substream->lock, flags); return result; } EXPORT_SYMBOL(snd_rawmidi_transmit); @@ -1430,12 +1426,12 @@ int snd_rawmidi_proceed(struct snd_rawmidi_substream *substream) unsigned long flags; int count = 0; - spin_lock_irqsave(&runtime->lock, flags); + spin_lock_irqsave(&substream->lock, flags); if (runtime->avail < runtime->buffer_size) { count = runtime->buffer_size - runtime->avail; __snd_rawmidi_transmit_ack(substream, count); } - spin_unlock_irqrestore(&runtime->lock, flags); + spin_unlock_irqrestore(&substream->lock, flags); return count; } EXPORT_SYMBOL(snd_rawmidi_proceed); @@ -1456,10 +1452,10 @@ static long snd_rawmidi_kernel_write1(struct snd_rawmidi_substream *substream, return -EINVAL; result = 0; - spin_lock_irqsave(&runtime->lock, flags); + spin_lock_irqsave(&substream->lock, flags); if (substream->append) { if ((long)runtime->avail < count) { - spin_unlock_irqrestore(&runtime->lock, flags); + spin_unlock_irqrestore(&substream->lock, flags); return -EAGAIN; } } @@ -1481,14 +1477,14 @@ static long snd_rawmidi_kernel_write1(struct snd_rawmidi_substream *substream, memcpy(runtime->buffer + appl_ptr, kernelbuf + result, count1); else if (userbuf) { - spin_unlock_irqrestore(&runtime->lock, flags); + spin_unlock_irqrestore(&substream->lock, flags); if (copy_from_user(runtime->buffer + appl_ptr, userbuf + result, count1)) { - spin_lock_irqsave(&runtime->lock, flags); + spin_lock_irqsave(&substream->lock, flags); result = result > 0 ? result : -EFAULT; goto __end; } - spin_lock_irqsave(&runtime->lock, flags); + spin_lock_irqsave(&substream->lock, flags); } result += count1; count -= count1; @@ -1496,7 +1492,7 @@ static long snd_rawmidi_kernel_write1(struct snd_rawmidi_substream *substream, __end: count1 = runtime->avail < runtime->buffer_size; snd_rawmidi_buffer_unref(runtime); - spin_unlock_irqrestore(&runtime->lock, flags); + spin_unlock_irqrestore(&substream->lock, flags); if (count1) snd_rawmidi_output_trigger(substream, 1); return result; @@ -1526,31 +1522,31 @@ static ssize_t snd_rawmidi_write(struct file *file, const char __user *buf, return -EIO; result = 0; while (count > 0) { - spin_lock_irq(&runtime->lock); + spin_lock_irq(&substream->lock); while (!snd_rawmidi_ready_append(substream, count)) { wait_queue_entry_t wait; if (file->f_flags & O_NONBLOCK) { - spin_unlock_irq(&runtime->lock); + spin_unlock_irq(&substream->lock); return result > 0 ? result : -EAGAIN; } init_waitqueue_entry(&wait, current); add_wait_queue(&runtime->sleep, &wait); set_current_state(TASK_INTERRUPTIBLE); - spin_unlock_irq(&runtime->lock); + spin_unlock_irq(&substream->lock); timeout = schedule_timeout(30 * HZ); remove_wait_queue(&runtime->sleep, &wait); if (rfile->rmidi->card->shutdown) return -ENODEV; if (signal_pending(current)) return result > 0 ? result : -ERESTARTSYS; - spin_lock_irq(&runtime->lock); + spin_lock_irq(&substream->lock); if (!runtime->avail && !timeout) { - spin_unlock_irq(&runtime->lock); + spin_unlock_irq(&substream->lock); return result > 0 ? result : -EIO; } } - spin_unlock_irq(&runtime->lock); + spin_unlock_irq(&substream->lock); count1 = snd_rawmidi_kernel_write1(substream, buf, NULL, count); if (count1 < 0) return result > 0 ? result : count1; @@ -1561,7 +1557,7 @@ static ssize_t snd_rawmidi_write(struct file *file, const char __user *buf, count -= count1; } if (file->f_flags & O_DSYNC) { - spin_lock_irq(&runtime->lock); + spin_lock_irq(&substream->lock); while (runtime->avail != runtime->buffer_size) { wait_queue_entry_t wait; unsigned int last_avail = runtime->avail; @@ -1569,16 +1565,16 @@ static ssize_t snd_rawmidi_write(struct file *file, const char __user *buf, init_waitqueue_entry(&wait, current); add_wait_queue(&runtime->sleep, &wait); set_current_state(TASK_INTERRUPTIBLE); - spin_unlock_irq(&runtime->lock); + spin_unlock_irq(&substream->lock); timeout = schedule_timeout(30 * HZ); remove_wait_queue(&runtime->sleep, &wait); if (signal_pending(current)) return result > 0 ? result : -ERESTARTSYS; if (runtime->avail == last_avail && !timeout) return result > 0 ? result : -EIO; - spin_lock_irq(&runtime->lock); + spin_lock_irq(&substream->lock); } - spin_unlock_irq(&runtime->lock); + spin_unlock_irq(&substream->lock); } return result; } @@ -1649,10 +1645,10 @@ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, " Owner PID : %d\n", pid_vnr(substream->pid)); runtime = substream->runtime; - spin_lock_irq(&runtime->lock); + spin_lock_irq(&substream->lock); buffer_size = runtime->buffer_size; avail = runtime->avail; - spin_unlock_irq(&runtime->lock); + spin_unlock_irq(&substream->lock); snd_iprintf(buffer, " Mode : %s\n" " Buffer size : %lu\n" @@ -1676,11 +1672,11 @@ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, " Owner PID : %d\n", pid_vnr(substream->pid)); runtime = substream->runtime; - spin_lock_irq(&runtime->lock); + spin_lock_irq(&substream->lock); buffer_size = runtime->buffer_size; avail = runtime->avail; xruns = runtime->xruns; - spin_unlock_irq(&runtime->lock); + spin_unlock_irq(&substream->lock); snd_iprintf(buffer, " Buffer size : %lu\n" " Avail : %lu\n" @@ -1732,6 +1728,7 @@ static int snd_rawmidi_alloc_substreams(struct snd_rawmidi *rmidi, substream->number = idx; substream->rmidi = rmidi; substream->pstr = stream; + spin_lock_init(&substream->lock); list_add_tail(&substream->list, &stream->substreams); stream->substream_count++; } -- cgit v1.2.3 From 94b98194b62e3fe3f27129d8e4b1f3fd7c5e972b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 17 Jun 2022 16:40:49 +0200 Subject: ALSA: rawmidi: Take open_mutex around parameter changes The input/output parameter changes are pretty intrusive, possibly involving with the buffer resizing operation. Hence those should be performed exclusively; otherwise some ugly race could happen. This patch puts the existing open_mutex for snd_rawmidi_input_params() and *_output_params() for protecting the concurrent calls. Since those are exported, it's also meant for hardening from the external calls, too. Link: https://lore.kernel.org/r/20220617144051.18985-4-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/rawmidi.c | 39 +++++++++++++++++++++++++-------------- 1 file changed, 25 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 0a00f37d8c42..7fd6b369d46f 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -712,11 +712,19 @@ static int resize_runtime_buffer(struct snd_rawmidi_substream *substream, int snd_rawmidi_output_params(struct snd_rawmidi_substream *substream, struct snd_rawmidi_params *params) { - if (substream->append && substream->use_count > 1) - return -EBUSY; + int err; + snd_rawmidi_drain_output(substream); - substream->active_sensing = !params->no_active_sensing; - return resize_runtime_buffer(substream, params, false); + mutex_lock(&substream->rmidi->open_mutex); + if (substream->append && substream->use_count > 1) + err = -EBUSY; + else + err = resize_runtime_buffer(substream, params, false); + + if (!err) + substream->active_sensing = !params->no_active_sensing; + mutex_unlock(&substream->rmidi->open_mutex); + return err; } EXPORT_SYMBOL(snd_rawmidi_output_params); @@ -727,19 +735,22 @@ int snd_rawmidi_input_params(struct snd_rawmidi_substream *substream, unsigned int clock_type = params->mode & SNDRV_RAWMIDI_MODE_CLOCK_MASK; int err; + snd_rawmidi_drain_input(substream); + mutex_lock(&substream->rmidi->open_mutex); if (framing == SNDRV_RAWMIDI_MODE_FRAMING_NONE && clock_type != SNDRV_RAWMIDI_MODE_CLOCK_NONE) - return -EINVAL; + err = -EINVAL; else if (clock_type > SNDRV_RAWMIDI_MODE_CLOCK_MONOTONIC_RAW) - return -EINVAL; - if (framing > SNDRV_RAWMIDI_MODE_FRAMING_TSTAMP) - return -EINVAL; - snd_rawmidi_drain_input(substream); - err = resize_runtime_buffer(substream, params, true); - if (err < 0) - return err; + err = -EINVAL; + else if (framing > SNDRV_RAWMIDI_MODE_FRAMING_TSTAMP) + err = -EINVAL; + else + err = resize_runtime_buffer(substream, params, true); - substream->framing = framing; - substream->clock_type = clock_type; + if (!err) { + substream->framing = framing; + substream->clock_type = clock_type; + } + mutex_unlock(&substream->rmidi->open_mutex); return 0; } EXPORT_SYMBOL(snd_rawmidi_input_params); -- cgit v1.2.3 From 463a20fd3481de33c2746f050b4e3f2e6db8017f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 17 Jun 2022 16:40:50 +0200 Subject: ALSA: rawmidi: Check stream state at exported functions The rawmidi interface provides some exported functions to be called from outside, and currently there is no state check for those calls whether the stream is properly opened and running. Although such an invalid call shouldn't happen, but who knows. This patch adds the proper rawmidi stream state checks with spinlocks for avoiding unexpected accesses when such exported functions are called in an invalid state. After this patch, with the substream->opened and substream->runtime are always tied and guaranteed to be set under substream->lock. Link: https://lore.kernel.org/r/20220617144051.18985-5-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/rawmidi.c | 56 +++++++++++++++++++++++++++++++++++----------------- 1 file changed, 38 insertions(+), 18 deletions(-) (limited to 'sound') diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 7fd6b369d46f..889fa4747dad 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -207,7 +207,8 @@ static void reset_runtime_ptrs(struct snd_rawmidi_substream *substream, unsigned long flags; spin_lock_irqsave(&substream->lock, flags); - __reset_runtime_ptrs(substream->runtime, is_input); + if (substream->opened && substream->runtime) + __reset_runtime_ptrs(substream->runtime, is_input); spin_unlock_irqrestore(&substream->lock, flags); } @@ -309,12 +310,14 @@ static int open_substream(struct snd_rawmidi *rmidi, snd_rawmidi_runtime_free(substream); return err; } + spin_lock_irq(&substream->lock); substream->opened = 1; substream->active_sensing = 0; if (mode & SNDRV_RAWMIDI_LFLG_APPEND) substream->append = 1; substream->pid = get_pid(task_pid(current)); rmidi->streams[substream->stream].substream_opened++; + spin_unlock_irq(&substream->lock); } substream->use_count++; return 0; @@ -520,12 +523,14 @@ static void close_substream(struct snd_rawmidi *rmidi, snd_rawmidi_output_trigger(substream, 0); } } + spin_lock_irq(&substream->lock); + substream->opened = 0; + substream->append = 0; + spin_unlock_irq(&substream->lock); substream->ops->close(substream); if (substream->runtime->private_free) substream->runtime->private_free(substream); snd_rawmidi_runtime_free(substream); - substream->opened = 0; - substream->append = 0; put_pid(substream->pid); substream->pid = NULL; rmidi->streams[substream->stream].substream_opened--; @@ -1074,17 +1079,21 @@ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream, unsigned long flags; struct timespec64 ts64 = get_framing_tstamp(substream); int result = 0, count1; - struct snd_rawmidi_runtime *runtime = substream->runtime; + struct snd_rawmidi_runtime *runtime; - if (!substream->opened) - return -EBADFD; - if (runtime->buffer == NULL) { + spin_lock_irqsave(&substream->lock, flags); + if (!substream->opened) { + result = -EBADFD; + goto unlock; + } + runtime = substream->runtime; + if (!runtime || !runtime->buffer) { rmidi_dbg(substream->rmidi, "snd_rawmidi_receive: input is not active!!!\n"); - return -EINVAL; + result = -EINVAL; + goto unlock; } - spin_lock_irqsave(&substream->lock, flags); if (substream->framing == SNDRV_RAWMIDI_MODE_FRAMING_TSTAMP) { result = receive_with_tstamp_framing(substream, buffer, count, &ts64); } else if (count == 1) { /* special case, faster code */ @@ -1131,6 +1140,7 @@ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream, else if (__snd_rawmidi_ready(runtime)) wake_up(&runtime->sleep); } + unlock: spin_unlock_irqrestore(&substream->lock, flags); return result; } @@ -1252,17 +1262,19 @@ static ssize_t snd_rawmidi_read(struct file *file, char __user *buf, size_t coun */ int snd_rawmidi_transmit_empty(struct snd_rawmidi_substream *substream) { - struct snd_rawmidi_runtime *runtime = substream->runtime; + struct snd_rawmidi_runtime *runtime; int result; unsigned long flags; - if (runtime->buffer == NULL) { + spin_lock_irqsave(&substream->lock, flags); + runtime = substream->runtime; + if (!substream->opened || !runtime || !runtime->buffer) { rmidi_dbg(substream->rmidi, "snd_rawmidi_transmit_empty: output is not active!!!\n"); - return 1; + result = 1; + } else { + result = runtime->avail >= runtime->buffer_size; } - spin_lock_irqsave(&substream->lock, flags); - result = runtime->avail >= runtime->buffer_size; spin_unlock_irqrestore(&substream->lock, flags); return result; } @@ -1336,7 +1348,10 @@ int snd_rawmidi_transmit_peek(struct snd_rawmidi_substream *substream, unsigned long flags; spin_lock_irqsave(&substream->lock, flags); - result = __snd_rawmidi_transmit_peek(substream, buffer, count); + if (!substream->opened || !substream->runtime) + result = -EBADFD; + else + result = __snd_rawmidi_transmit_peek(substream, buffer, count); spin_unlock_irqrestore(&substream->lock, flags); return result; } @@ -1388,7 +1403,10 @@ int snd_rawmidi_transmit_ack(struct snd_rawmidi_substream *substream, int count) unsigned long flags; spin_lock_irqsave(&substream->lock, flags); - result = __snd_rawmidi_transmit_ack(substream, count); + if (!substream->opened || !substream->runtime) + result = -EBADFD; + else + result = __snd_rawmidi_transmit_ack(substream, count); spin_unlock_irqrestore(&substream->lock, flags); return result; } @@ -1433,12 +1451,14 @@ EXPORT_SYMBOL(snd_rawmidi_transmit); */ int snd_rawmidi_proceed(struct snd_rawmidi_substream *substream) { - struct snd_rawmidi_runtime *runtime = substream->runtime; + struct snd_rawmidi_runtime *runtime; unsigned long flags; int count = 0; spin_lock_irqsave(&substream->lock, flags); - if (runtime->avail < runtime->buffer_size) { + runtime = substream->runtime; + if (substream->opened && runtime && + runtime->avail < runtime->buffer_size) { count = runtime->buffer_size - runtime->avail; __snd_rawmidi_transmit_ack(substream, count); } -- cgit v1.2.3 From 3809db6430bf6a725d234e6eec9a6f6be6b8c1ea Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 17 Jun 2022 16:40:51 +0200 Subject: ALSA: rawmidi: Take buffer refcount while draining output Although snd_rawmidi_drain_output() may take some long time, it has no protection and intrusive operations like the buffer resize may happen meanwhile. For making the operation a bit more robust, this patch takes the buffer refcount for blocking the buffer resize. Also, as this function is exported, in theory, it might be called asynchronously from the stream open/close state. For avoiding the missing refcount, now the close call checks the buffer refcount, too. Link: https://lore.kernel.org/r/20220617144051.18985-6-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/rawmidi.c | 45 +++++++++++++++++++++++++++++++++++++++++---- 1 file changed, 41 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 889fa4747dad..6963d5a487b3 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -140,6 +140,23 @@ static inline void snd_rawmidi_buffer_unref(struct snd_rawmidi_runtime *runtime) runtime->buffer_ref--; } +static void snd_rawmidi_buffer_ref_sync(struct snd_rawmidi_substream *substream) +{ + int loop = HZ; + + spin_lock_irq(&substream->lock); + while (substream->runtime->buffer_ref) { + spin_unlock_irq(&substream->lock); + if (!--loop) { + rmidi_err(substream->rmidi, "Buffer ref sync timeout\n"); + return; + } + schedule_timeout_uninterruptible(1); + spin_lock_irq(&substream->lock); + } + spin_unlock_irq(&substream->lock); +} + static int snd_rawmidi_runtime_create(struct snd_rawmidi_substream *substream) { struct snd_rawmidi_runtime *runtime; @@ -222,15 +239,27 @@ EXPORT_SYMBOL(snd_rawmidi_drop_output); int snd_rawmidi_drain_output(struct snd_rawmidi_substream *substream) { - int err; + int err = 0; long timeout; - struct snd_rawmidi_runtime *runtime = substream->runtime; + struct snd_rawmidi_runtime *runtime; + + spin_lock_irq(&substream->lock); + runtime = substream->runtime; + if (!substream->opened || !runtime || !runtime->buffer) { + err = -EINVAL; + } else { + snd_rawmidi_buffer_ref(runtime); + runtime->drain = 1; + } + spin_unlock_irq(&substream->lock); + if (err < 0) + return err; - err = 0; - runtime->drain = 1; timeout = wait_event_interruptible_timeout(runtime->sleep, (runtime->avail >= runtime->buffer_size), 10*HZ); + + spin_lock_irq(&substream->lock); if (signal_pending(current)) err = -ERESTARTSYS; if (runtime->avail < runtime->buffer_size && !timeout) { @@ -240,6 +269,8 @@ int snd_rawmidi_drain_output(struct snd_rawmidi_substream *substream) err = -EIO; } runtime->drain = 0; + spin_unlock_irq(&substream->lock); + if (err != -ERESTARTSYS) { /* we need wait a while to make sure that Tx FIFOs are empty */ if (substream->ops->drain) @@ -248,6 +279,11 @@ int snd_rawmidi_drain_output(struct snd_rawmidi_substream *substream) msleep(50); snd_rawmidi_drop_output(substream); } + + spin_lock_irq(&substream->lock); + snd_rawmidi_buffer_unref(runtime); + spin_unlock_irq(&substream->lock); + return err; } EXPORT_SYMBOL(snd_rawmidi_drain_output); @@ -522,6 +558,7 @@ static void close_substream(struct snd_rawmidi *rmidi, if (snd_rawmidi_drain_output(substream) == -ERESTARTSYS) snd_rawmidi_output_trigger(substream, 0); } + snd_rawmidi_buffer_ref_sync(substream); } spin_lock_irq(&substream->lock); substream->opened = 0; -- cgit v1.2.3 From 44f362c2cc6dd0c5e3cb499c4fb4ed45b63a6196 Mon Sep 17 00:00:00 2001 From: Judy Hsiao Date: Wed, 15 Jun 2022 04:56:43 +0000 Subject: ASoC: rockchip: i2s: switch BCLK to GPIO We discoverd that the state of BCLK on, LRCLK off and SD_MODE on may cause the speaker melting issue. Removing LRCLK while BCLK is present can cause unexpected output behavior including a large DC output voltage as described in the Max98357a datasheet. In order to: 1. prevent BCLK from turning on by other component. 2. keep BCLK and LRCLK being present at the same time This patch switches BCLK to GPIO func before LRCLK output, and configures BCLK func back during LRCLK is output. Without this fix, BCLK is turned on 11 ms earlier than LRCK by the da7219. With this fix, BCLK is turned on only 0.4 ms earlier than LRCK by the rockchip codec. Signed-off-by: Judy Hsiao Link: https://lore.kernel.org/r/20220615045643.3137287-1-judyhsiao@chromium.org Signed-off-by: Mark Brown --- sound/soc/rockchip/rockchip_i2s.c | 160 ++++++++++++++++++++++++++++++-------- 1 file changed, 129 insertions(+), 31 deletions(-) (limited to 'sound') diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 4ce5d2579387..99a128a666fb 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -13,6 +13,7 @@ #include #include #include +#include #include #include #include @@ -54,8 +55,40 @@ struct rk_i2s_dev { const struct rk_i2s_pins *pins; unsigned int bclk_ratio; spinlock_t lock; /* tx/rx lock */ + struct pinctrl *pinctrl; + struct pinctrl_state *bclk_on; + struct pinctrl_state *bclk_off; }; +static int i2s_pinctrl_select_bclk_on(struct rk_i2s_dev *i2s) +{ + int ret = 0; + + if (!IS_ERR(i2s->pinctrl) && !IS_ERR_OR_NULL(i2s->bclk_on)) + ret = pinctrl_select_state(i2s->pinctrl, + i2s->bclk_on); + + if (ret) + dev_err(i2s->dev, "bclk enable failed %d\n", ret); + + return ret; +} + +static int i2s_pinctrl_select_bclk_off(struct rk_i2s_dev *i2s) +{ + + int ret = 0; + + if (!IS_ERR(i2s->pinctrl) && !IS_ERR_OR_NULL(i2s->bclk_off)) + ret = pinctrl_select_state(i2s->pinctrl, + i2s->bclk_off); + + if (ret) + dev_err(i2s->dev, "bclk disable failed %d\n", ret); + + return ret; +} + static int i2s_runtime_suspend(struct device *dev) { struct rk_i2s_dev *i2s = dev_get_drvdata(dev); @@ -92,38 +125,49 @@ static inline struct rk_i2s_dev *to_info(struct snd_soc_dai *dai) return snd_soc_dai_get_drvdata(dai); } -static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) +static int rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) { unsigned int val = 0; int retry = 10; + int ret = 0; spin_lock(&i2s->lock); if (on) { - regmap_update_bits(i2s->regmap, I2S_DMACR, - I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_ENABLE); + ret = regmap_update_bits(i2s->regmap, I2S_DMACR, + I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_ENABLE); + if (ret < 0) + goto end; - regmap_update_bits(i2s->regmap, I2S_XFER, - I2S_XFER_TXS_START | I2S_XFER_RXS_START, - I2S_XFER_TXS_START | I2S_XFER_RXS_START); + ret = regmap_update_bits(i2s->regmap, I2S_XFER, + I2S_XFER_TXS_START | I2S_XFER_RXS_START, + I2S_XFER_TXS_START | I2S_XFER_RXS_START); + if (ret < 0) + goto end; i2s->tx_start = true; } else { i2s->tx_start = false; - regmap_update_bits(i2s->regmap, I2S_DMACR, - I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_DISABLE); + ret = regmap_update_bits(i2s->regmap, I2S_DMACR, + I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_DISABLE); + if (ret < 0) + goto end; if (!i2s->rx_start) { - regmap_update_bits(i2s->regmap, I2S_XFER, - I2S_XFER_TXS_START | - I2S_XFER_RXS_START, - I2S_XFER_TXS_STOP | - I2S_XFER_RXS_STOP); + ret = regmap_update_bits(i2s->regmap, I2S_XFER, + I2S_XFER_TXS_START | + I2S_XFER_RXS_START, + I2S_XFER_TXS_STOP | + I2S_XFER_RXS_STOP); + if (ret < 0) + goto end; udelay(150); - regmap_update_bits(i2s->regmap, I2S_CLR, - I2S_CLR_TXC | I2S_CLR_RXC, - I2S_CLR_TXC | I2S_CLR_RXC); + ret = regmap_update_bits(i2s->regmap, I2S_CLR, + I2S_CLR_TXC | I2S_CLR_RXC, + I2S_CLR_TXC | I2S_CLR_RXC); + if (ret < 0) + goto end; regmap_read(i2s->regmap, I2S_CLR, &val); @@ -138,44 +182,57 @@ static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) } } } +end: spin_unlock(&i2s->lock); + if (ret < 0) + dev_err(i2s->dev, "lrclk update failed\n"); + + return ret; } -static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) +static int rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) { unsigned int val = 0; int retry = 10; + int ret = 0; spin_lock(&i2s->lock); if (on) { - regmap_update_bits(i2s->regmap, I2S_DMACR, + ret = regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_ENABLE); + if (ret < 0) + goto end; - regmap_update_bits(i2s->regmap, I2S_XFER, + ret = regmap_update_bits(i2s->regmap, I2S_XFER, I2S_XFER_TXS_START | I2S_XFER_RXS_START, I2S_XFER_TXS_START | I2S_XFER_RXS_START); + if (ret < 0) + goto end; i2s->rx_start = true; } else { i2s->rx_start = false; - regmap_update_bits(i2s->regmap, I2S_DMACR, + ret = regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_DISABLE); + if (ret < 0) + goto end; if (!i2s->tx_start) { - regmap_update_bits(i2s->regmap, I2S_XFER, + ret = regmap_update_bits(i2s->regmap, I2S_XFER, I2S_XFER_TXS_START | I2S_XFER_RXS_START, I2S_XFER_TXS_STOP | I2S_XFER_RXS_STOP); - + if (ret < 0) + goto end; udelay(150); - regmap_update_bits(i2s->regmap, I2S_CLR, + ret = regmap_update_bits(i2s->regmap, I2S_CLR, I2S_CLR_TXC | I2S_CLR_RXC, I2S_CLR_TXC | I2S_CLR_RXC); - + if (ret < 0) + goto end; regmap_read(i2s->regmap, I2S_CLR, &val); - /* Should wait for clear operation to finish */ while (val) { regmap_read(i2s->regmap, I2S_CLR, &val); @@ -187,7 +244,12 @@ static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) } } } +end: spin_unlock(&i2s->lock); + if (ret < 0) + dev_err(i2s->dev, "lrclk update failed\n"); + + return ret; } static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, @@ -425,17 +487,26 @@ static int rockchip_i2s_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - rockchip_snd_rxctrl(i2s, 1); + ret = rockchip_snd_rxctrl(i2s, 1); else - rockchip_snd_txctrl(i2s, 1); + ret = rockchip_snd_txctrl(i2s, 1); + /* Do not turn on bclk if lrclk open fails. */ + if (ret < 0) + return ret; + i2s_pinctrl_select_bclk_on(i2s); break; case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - rockchip_snd_rxctrl(i2s, 0); - else - rockchip_snd_txctrl(i2s, 0); + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (!i2s->tx_start) + i2s_pinctrl_select_bclk_off(i2s); + ret = rockchip_snd_rxctrl(i2s, 0); + } else { + if (!i2s->rx_start) + i2s_pinctrl_select_bclk_off(i2s); + ret = rockchip_snd_txctrl(i2s, 0); + } break; default: ret = -EINVAL; @@ -736,6 +807,33 @@ static int rockchip_i2s_probe(struct platform_device *pdev) } i2s->bclk_ratio = 64; + i2s->pinctrl = devm_pinctrl_get(&pdev->dev); + if (IS_ERR(i2s->pinctrl)) + dev_err(&pdev->dev, "failed to find i2s pinctrl\n"); + + i2s->bclk_on = pinctrl_lookup_state(i2s->pinctrl, + "bclk_on"); + if (IS_ERR_OR_NULL(i2s->bclk_on)) + dev_err(&pdev->dev, "failed to find i2s default state\n"); + else + dev_dbg(&pdev->dev, "find i2s bclk state\n"); + + i2s->bclk_off = pinctrl_lookup_state(i2s->pinctrl, + "bclk_off"); + if (IS_ERR_OR_NULL(i2s->bclk_off)) + dev_err(&pdev->dev, "failed to find i2s gpio state\n"); + else + dev_dbg(&pdev->dev, "find i2s bclk_off state\n"); + + i2s_pinctrl_select_bclk_off(i2s); + + i2s->playback_dma_data.addr = res->start + I2S_TXDR; + i2s->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + i2s->playback_dma_data.maxburst = 4; + + i2s->capture_dma_data.addr = res->start + I2S_RXDR; + i2s->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + i2s->capture_dma_data.maxburst = 4; dev_set_drvdata(&pdev->dev, i2s); -- cgit v1.2.3 From 289a3ec0b5b9a2de6fc75633aa81f017792ecc99 Mon Sep 17 00:00:00 2001 From: Sascha Hauer Date: Fri, 17 Jun 2022 14:01:33 +0200 Subject: ASoC: fsl_asrc_dma: Use dmaengine_terminate_async() dmaengine_terminate_all() is deprecated and should no longer be used. Use dmaengine_terminate_async() instead. This involves no functional change since both functions do the same. After dmaengine_terminate_async() dmaengine_synchronize() must be called to make sure the channel has really stopped before the underlying memory is freed. This is done implicitly by dma_release_channel() called from the .hw_free hook. Signed-off-by: Sascha Hauer Link: https://lore.kernel.org/r/20220617120133.4011846-1-s.hauer@pengutronix.de Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_asrc_dma.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c index aaf7993935b7..33eabb96340e 100644 --- a/sound/soc/fsl/fsl_asrc_dma.c +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -114,8 +114,8 @@ static int fsl_asrc_dma_trigger(struct snd_soc_component *component, case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - dmaengine_terminate_all(pair->dma_chan[OUT]); - dmaengine_terminate_all(pair->dma_chan[IN]); + dmaengine_terminate_async(pair->dma_chan[OUT]); + dmaengine_terminate_async(pair->dma_chan[IN]); break; default: return -EINVAL; -- cgit v1.2.3 From 81d74ddae83fbd85c9006835f36c362114127a7a Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 21 Jun 2022 11:20:38 +0100 Subject: ASoC: wm_adsp: Fix event for preloader The preloader controls on ADSP should return a value of 1 if the preloader value was changed, update to correct this. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220621102041.1713504-1-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 6d7fd88243aa..a7784ac15dde 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -997,7 +997,7 @@ int wm_adsp2_preloader_put(struct snd_kcontrol *kcontrol, snd_soc_dapm_sync(dapm); } - return 0; + return 1; } EXPORT_SYMBOL_GPL(wm_adsp2_preloader_put); -- cgit v1.2.3 From 630cc5983740d784a1a6458f9dc2112c43fe0931 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 21 Jun 2022 11:20:39 +0100 Subject: ASoC: wm5110: Fix DRE control The DRE controls on wm5110 should return a value of 1 if the DRE state is actually changed, update to fix this. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220621102041.1713504-2-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/wm5110.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 4973ba1ed779..4ab7a672f8de 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -413,6 +413,7 @@ static int wm5110_put_dre(struct snd_kcontrol *kcontrol, unsigned int rnew = (!!ucontrol->value.integer.value[1]) << mc->rshift; unsigned int lold, rold; unsigned int lena, rena; + bool change = false; int ret; snd_soc_dapm_mutex_lock(dapm); @@ -440,8 +441,8 @@ static int wm5110_put_dre(struct snd_kcontrol *kcontrol, goto err; } - ret = regmap_update_bits(arizona->regmap, ARIZONA_DRE_ENABLE, - mask, lnew | rnew); + ret = regmap_update_bits_check(arizona->regmap, ARIZONA_DRE_ENABLE, + mask, lnew | rnew, &change); if (ret) { dev_err(arizona->dev, "Failed to set DRE: %d\n", ret); goto err; @@ -454,6 +455,9 @@ static int wm5110_put_dre(struct snd_kcontrol *kcontrol, if (!rnew && rold) wm5110_clear_pga_volume(arizona, mc->rshift); + if (change) + ret = 1; + err: snd_soc_dapm_mutex_unlock(dapm); -- cgit v1.2.3 From 87912e97a1678d62877aab353ecfd201bc92b372 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 21 Jun 2022 11:20:40 +0100 Subject: ASoC: cs35l41: Correct some control names Various boolean controls on cs35l41 are missing the required "Switch" in the name, add these. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220621102041.1713504-3-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs35l41.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs35l41.c b/sound/soc/codecs/cs35l41.c index 3e68a07a3c8e..71ab2a5d1c55 100644 --- a/sound/soc/codecs/cs35l41.c +++ b/sound/soc/codecs/cs35l41.c @@ -333,7 +333,7 @@ static const struct snd_kcontrol_new cs35l41_aud_controls[] = { SOC_SINGLE("HW Noise Gate Enable", CS35L41_NG_CFG, 8, 63, 0), SOC_SINGLE("HW Noise Gate Delay", CS35L41_NG_CFG, 4, 7, 0), SOC_SINGLE("HW Noise Gate Threshold", CS35L41_NG_CFG, 0, 7, 0), - SOC_SINGLE("Aux Noise Gate CH1 Enable", + SOC_SINGLE("Aux Noise Gate CH1 Switch", CS35L41_MIXER_NGATE_CH1_CFG, 16, 1, 0), SOC_SINGLE("Aux Noise Gate CH1 Entry Delay", CS35L41_MIXER_NGATE_CH1_CFG, 8, 15, 0), @@ -341,15 +341,15 @@ static const struct snd_kcontrol_new cs35l41_aud_controls[] = { CS35L41_MIXER_NGATE_CH1_CFG, 0, 7, 0), SOC_SINGLE("Aux Noise Gate CH2 Entry Delay", CS35L41_MIXER_NGATE_CH2_CFG, 8, 15, 0), - SOC_SINGLE("Aux Noise Gate CH2 Enable", + SOC_SINGLE("Aux Noise Gate CH2 Switch", CS35L41_MIXER_NGATE_CH2_CFG, 16, 1, 0), SOC_SINGLE("Aux Noise Gate CH2 Threshold", CS35L41_MIXER_NGATE_CH2_CFG, 0, 7, 0), - SOC_SINGLE("SCLK Force", CS35L41_SP_FORMAT, CS35L41_SCLK_FRC_SHIFT, 1, 0), - SOC_SINGLE("LRCLK Force", CS35L41_SP_FORMAT, CS35L41_LRCLK_FRC_SHIFT, 1, 0), - SOC_SINGLE("Invert Class D", CS35L41_AMP_DIG_VOL_CTRL, + SOC_SINGLE("SCLK Force Switch", CS35L41_SP_FORMAT, CS35L41_SCLK_FRC_SHIFT, 1, 0), + SOC_SINGLE("LRCLK Force Switch", CS35L41_SP_FORMAT, CS35L41_LRCLK_FRC_SHIFT, 1, 0), + SOC_SINGLE("Invert Class D Switch", CS35L41_AMP_DIG_VOL_CTRL, CS35L41_AMP_INV_PCM_SHIFT, 1, 0), - SOC_SINGLE("Amp Gain ZC", CS35L41_AMP_GAIN_CTRL, + SOC_SINGLE("Amp Gain ZC Switch", CS35L41_AMP_GAIN_CTRL, CS35L41_AMP_GAIN_ZC_SHIFT, 1, 0), WM_ADSP2_PRELOAD_SWITCH("DSP1", 1), WM_ADSP_FW_CONTROL("DSP1", 0), -- cgit v1.2.3 From f69a10f84cb5ff0b1c6aef0e19e866bbe53ec7ea Mon Sep 17 00:00:00 2001 From: Shuming Fan Date: Tue, 21 Jun 2022 17:07:19 +0800 Subject: ASoC: rt711-sdca: fix kernel NULL pointer dereference when IO error The initial settings will be written before the codec probe function. But, the rt711->component doesn't be assigned yet. If IO error happened during initial settings operations, it will cause the kernel panic. This patch changed component->dev to slave->dev to fix this issue. Signed-off-by: Shuming Fan Link: https://lore.kernel.org/r/20220621090719.30558-1-shumingf@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt711-sdca.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt711-sdca.c b/sound/soc/codecs/rt711-sdca.c index dfe3c9299ebd..5ad53bbc8528 100644 --- a/sound/soc/codecs/rt711-sdca.c +++ b/sound/soc/codecs/rt711-sdca.c @@ -34,7 +34,7 @@ static int rt711_sdca_index_write(struct rt711_sdca_priv *rt711, ret = regmap_write(regmap, addr, value); if (ret < 0) - dev_err(rt711->component->dev, + dev_err(&rt711->slave->dev, "Failed to set private value: %06x <= %04x ret=%d\n", addr, value, ret); @@ -50,7 +50,7 @@ static int rt711_sdca_index_read(struct rt711_sdca_priv *rt711, ret = regmap_read(regmap, addr, value); if (ret < 0) - dev_err(rt711->component->dev, + dev_err(&rt711->slave->dev, "Failed to get private value: %06x => %04x ret=%d\n", addr, *value, ret); -- cgit v1.2.3 From 8c77cf26a82e751ce827614664faf40243058d5d Mon Sep 17 00:00:00 2001 From: Judy Hsiao Date: Sun, 19 Jun 2022 09:53:22 +0000 Subject: ASoC: rockchip: i2s: switch BCLK to GPIO We discoverd that the state of BCLK on, LRCLK off and SD_MODE on may cause the speaker melting issue. Removing LRCLK while BCLK is present can cause unexpected output behavior including a large DC output voltage as described in the Max98357a datasheet. In order to: 1. prevent BCLK from turning on by other component. 2. keep BCLK and LRCLK being present at the same time This patch switches BCLK to GPIO func before LRCLK output, and configures BCLK func back during LRCLK is output. Without this fix, BCLK is turned on 11 ms earlier than LRCK by the da7219. With this fix, BCLK is turned on only 0.4 ms earlier than LRCK by the rockchip codec. Signed-off-by: Judy Hsiao Reviewed-by: Brian Norris Link: https://lore.kernel.org/r/20220619095324.492678-2-judyhsiao@chromium.org Signed-off-by: Mark Brown --- sound/soc/rockchip/rockchip_i2s.c | 169 +++++++++++++++++++++++++++----------- 1 file changed, 123 insertions(+), 46 deletions(-) (limited to 'sound') diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 47a3971a9ce1..aa7d9984022a 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -54,8 +54,38 @@ struct rk_i2s_dev { const struct rk_i2s_pins *pins; unsigned int bclk_ratio; spinlock_t lock; /* tx/rx lock */ + struct pinctrl *pinctrl; + struct pinctrl_state *bclk_on; + struct pinctrl_state *bclk_off; }; +static int i2s_pinctrl_select_bclk_on(struct rk_i2s_dev *i2s) +{ + int ret = 0; + + if (!IS_ERR(i2s->pinctrl) && !IS_ERR_OR_NULL(i2s->bclk_on)) + ret = pinctrl_select_state(i2s->pinctrl, i2s->bclk_on); + + if (ret) + dev_err(i2s->dev, "bclk enable failed %d\n", ret); + + return ret; +} + +static int i2s_pinctrl_select_bclk_off(struct rk_i2s_dev *i2s) +{ + + int ret = 0; + + if (!IS_ERR(i2s->pinctrl) && !IS_ERR_OR_NULL(i2s->bclk_off)) + ret = pinctrl_select_state(i2s->pinctrl, i2s->bclk_off); + + if (ret) + dev_err(i2s->dev, "bclk disable failed %d\n", ret); + + return ret; +} + static int i2s_runtime_suspend(struct device *dev) { struct rk_i2s_dev *i2s = dev_get_drvdata(dev); @@ -92,39 +122,46 @@ static inline struct rk_i2s_dev *to_info(struct snd_soc_dai *dai) return snd_soc_dai_get_drvdata(dai); } -static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) +static int rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) { unsigned int val = 0; int retry = 10; + int ret = 0; spin_lock(&i2s->lock); if (on) { - regmap_update_bits(i2s->regmap, I2S_DMACR, - I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_ENABLE); - - regmap_update_bits(i2s->regmap, I2S_XFER, - I2S_XFER_TXS_START | I2S_XFER_RXS_START, - I2S_XFER_TXS_START | I2S_XFER_RXS_START); - + ret = regmap_update_bits(i2s->regmap, I2S_DMACR, + I2S_DMACR_TDE_ENABLE, + I2S_DMACR_TDE_ENABLE); + if (ret < 0) + goto end; + ret = regmap_update_bits(i2s->regmap, I2S_XFER, + I2S_XFER_TXS_START | I2S_XFER_RXS_START, + I2S_XFER_TXS_START | I2S_XFER_RXS_START); + if (ret < 0) + goto end; i2s->tx_start = true; } else { i2s->tx_start = false; - regmap_update_bits(i2s->regmap, I2S_DMACR, - I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_DISABLE); + ret = regmap_update_bits(i2s->regmap, I2S_DMACR, + I2S_DMACR_TDE_ENABLE, + I2S_DMACR_TDE_DISABLE); + if (ret < 0) + goto end; if (!i2s->rx_start) { - regmap_update_bits(i2s->regmap, I2S_XFER, - I2S_XFER_TXS_START | - I2S_XFER_RXS_START, - I2S_XFER_TXS_STOP | - I2S_XFER_RXS_STOP); - + ret = regmap_update_bits(i2s->regmap, I2S_XFER, + I2S_XFER_TXS_START | I2S_XFER_RXS_START, + I2S_XFER_TXS_STOP | I2S_XFER_RXS_STOP); + if (ret < 0) + goto end; udelay(150); - regmap_update_bits(i2s->regmap, I2S_CLR, - I2S_CLR_TXC | I2S_CLR_RXC, - I2S_CLR_TXC | I2S_CLR_RXC); - + ret = regmap_update_bits(i2s->regmap, I2S_CLR, + I2S_CLR_TXC | I2S_CLR_RXC, + I2S_CLR_TXC | I2S_CLR_RXC); + if (ret < 0) + goto end; regmap_read(i2s->regmap, I2S_CLR, &val); /* Should wait for clear operation to finish */ @@ -138,42 +175,55 @@ static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) } } } +end: spin_unlock(&i2s->lock); + if (ret < 0) + dev_err(i2s->dev, "lrclk update failed\n"); + + return ret; } -static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) +static int rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) { unsigned int val = 0; int retry = 10; + int ret = 0; spin_lock(&i2s->lock); if (on) { - regmap_update_bits(i2s->regmap, I2S_DMACR, - I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_ENABLE); - - regmap_update_bits(i2s->regmap, I2S_XFER, - I2S_XFER_TXS_START | I2S_XFER_RXS_START, - I2S_XFER_TXS_START | I2S_XFER_RXS_START); - + ret = regmap_update_bits(i2s->regmap, I2S_DMACR, + I2S_DMACR_RDE_ENABLE, + I2S_DMACR_RDE_ENABLE); + if (ret < 0) + goto end; + + ret = regmap_update_bits(i2s->regmap, I2S_XFER, + I2S_XFER_TXS_START | I2S_XFER_RXS_START, + I2S_XFER_TXS_START | I2S_XFER_RXS_START); + if (ret < 0) + goto end; i2s->rx_start = true; } else { i2s->rx_start = false; - regmap_update_bits(i2s->regmap, I2S_DMACR, - I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_DISABLE); + ret = regmap_update_bits(i2s->regmap, I2S_DMACR, + I2S_DMACR_RDE_ENABLE, + I2S_DMACR_RDE_DISABLE); + if (ret < 0) + goto end; if (!i2s->tx_start) { - regmap_update_bits(i2s->regmap, I2S_XFER, - I2S_XFER_TXS_START | - I2S_XFER_RXS_START, - I2S_XFER_TXS_STOP | - I2S_XFER_RXS_STOP); - + ret = regmap_update_bits(i2s->regmap, I2S_XFER, + I2S_XFER_TXS_START | I2S_XFER_RXS_START, + I2S_XFER_TXS_STOP | I2S_XFER_RXS_STOP); + if (ret < 0) + goto end; udelay(150); - regmap_update_bits(i2s->regmap, I2S_CLR, - I2S_CLR_TXC | I2S_CLR_RXC, - I2S_CLR_TXC | I2S_CLR_RXC); - + ret = regmap_update_bits(i2s->regmap, I2S_CLR, + I2S_CLR_TXC | I2S_CLR_RXC, + I2S_CLR_TXC | I2S_CLR_RXC); + if (ret < 0) + goto end; regmap_read(i2s->regmap, I2S_CLR, &val); /* Should wait for clear operation to finish */ @@ -187,7 +237,12 @@ static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) } } } +end: spin_unlock(&i2s->lock); + if (ret < 0) + dev_err(i2s->dev, "lrclk update failed\n"); + + return ret; } static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, @@ -425,17 +480,25 @@ static int rockchip_i2s_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - rockchip_snd_rxctrl(i2s, 1); + ret = rockchip_snd_rxctrl(i2s, 1); else - rockchip_snd_txctrl(i2s, 1); + ret = rockchip_snd_txctrl(i2s, 1); + if (ret < 0) + return ret; + i2s_pinctrl_select_bclk_on(i2s); break; case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - rockchip_snd_rxctrl(i2s, 0); - else - rockchip_snd_txctrl(i2s, 0); + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (!i2s->tx_start) + i2s_pinctrl_select_bclk_off(i2s); + ret = rockchip_snd_rxctrl(i2s, 0); + } else { + if (!i2s->rx_start) + i2s_pinctrl_select_bclk_off(i2s); + ret = rockchip_snd_txctrl(i2s, 0); + } break; default: ret = -EINVAL; @@ -736,6 +799,20 @@ static int rockchip_i2s_probe(struct platform_device *pdev) } i2s->bclk_ratio = 64; + i2s->pinctrl = devm_pinctrl_get(&pdev->dev); + if (IS_ERR(i2s->pinctrl)) + dev_err(&pdev->dev, "failed to find i2s pinctrl\n"); + + i2s->bclk_on = pinctrl_lookup_state(i2s->pinctrl, "bclk_on"); + if (!IS_ERR_OR_NULL(i2s->bclk_on)) { + i2s->bclk_off = pinctrl_lookup_state(i2s->pinctrl, "bclk_off"); + if (IS_ERR_OR_NULL(i2s->bclk_off)) { + dev_err(&pdev->dev, "failed to find i2s bclk_off\n"); + goto err_clk; + } + } + + i2s_pinctrl_select_bclk_off(i2s); dev_set_drvdata(&pdev->dev, i2s); -- cgit v1.2.3 From acaeb8c62fd1b2b57be1523b8d5b1d64a1a9dc38 Mon Sep 17 00:00:00 2001 From: Tinghan Shen Date: Wed, 22 Jun 2022 14:22:45 +0800 Subject: ASoC: SOF: mediatek: Align mt8186 clock names with dt-bindings Align clock names in mt8186 dsp driver with dt-bindings. Signed-off-by: Tinghan Shen Link: https://lore.kernel.org/r/20220622062245.21021-5-tinghan.shen@mediatek.com Signed-off-by: Mark Brown --- sound/soc/sof/mediatek/mt8186/mt8186-clk.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/mediatek/mt8186/mt8186-clk.c b/sound/soc/sof/mediatek/mt8186/mt8186-clk.c index 22220fd50b62..2df3b7ae1c6f 100644 --- a/sound/soc/sof/mediatek/mt8186/mt8186-clk.c +++ b/sound/soc/sof/mediatek/mt8186/mt8186-clk.c @@ -18,8 +18,8 @@ #include "mt8186-clk.h" static const char *adsp_clks[ADSP_CLK_MAX] = { - [CLK_TOP_AUDIODSP] = "audiodsp_sel", - [CLK_TOP_ADSP_BUS] = "adsp_bus_sel", + [CLK_TOP_AUDIODSP] = "audiodsp", + [CLK_TOP_ADSP_BUS] = "adsp_bus", }; int mt8186_adsp_init_clock(struct snd_sof_dev *sdev) -- cgit v1.2.3 From 1892a991886ace2c3450bec801df2cf4028a803a Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Uwe=20Kleine-K=C3=B6nig?= Date: Tue, 21 Jun 2022 16:58:34 +0200 Subject: ASoC: core: Make snd_soc_unregister_card() return void MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The function snd_soc_unregister_card() returned 0 unconditionally and most callers don't care to check the return value. Make it return void and adapt the callers that didn't ignore the return value before. This is a preparation for making platform remove callbacks return void. Signed-off-by: Uwe Kleine-König Link: https://lore.kernel.org/r/20220621145834.198519-1-u.kleine-koenig@pengutronix.de Signed-off-by: Mark Brown --- sound/soc/atmel/mikroe-proto.c | 4 +++- sound/soc/fsl/pcm030-audio-fabric.c | 5 ++--- sound/soc/soc-core.c | 4 +--- sound/soc/soc-topology-test.c | 4 +--- 4 files changed, 7 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/atmel/mikroe-proto.c b/sound/soc/atmel/mikroe-proto.c index ce46d8a0b7e4..954460719aa3 100644 --- a/sound/soc/atmel/mikroe-proto.c +++ b/sound/soc/atmel/mikroe-proto.c @@ -157,7 +157,9 @@ put_codec_node: static int snd_proto_remove(struct platform_device *pdev) { - return snd_soc_unregister_card(&snd_proto); + snd_soc_unregister_card(&snd_proto); + + return 0; } static const struct of_device_id snd_proto_of_match[] = { diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c index 83b4a22bf15a..e4c805acc349 100644 --- a/sound/soc/fsl/pcm030-audio-fabric.c +++ b/sound/soc/fsl/pcm030-audio-fabric.c @@ -113,12 +113,11 @@ static int pcm030_fabric_probe(struct platform_device *op) static int pcm030_fabric_remove(struct platform_device *op) { struct pcm030_audio_data *pdata = platform_get_drvdata(op); - int ret; - ret = snd_soc_unregister_card(pdata->card); + snd_soc_unregister_card(pdata->card); platform_device_unregister(pdata->codec_device); - return ret; + return 0; } static const struct of_device_id pcm030_audio_match[] = { diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 57f7105c12b7..30f0da711ca9 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2319,14 +2319,12 @@ EXPORT_SYMBOL_GPL(snd_soc_register_card); * @card: Card to unregister * */ -int snd_soc_unregister_card(struct snd_soc_card *card) +void snd_soc_unregister_card(struct snd_soc_card *card) { mutex_lock(&client_mutex); snd_soc_unbind_card(card, true); mutex_unlock(&client_mutex); dev_dbg(card->dev, "ASoC: Unregistered card '%s'\n", card->name); - - return 0; } EXPORT_SYMBOL_GPL(snd_soc_unregister_card); diff --git a/sound/soc/soc-topology-test.c b/sound/soc/soc-topology-test.c index ae3968161509..225d74355974 100644 --- a/sound/soc/soc-topology-test.c +++ b/sound/soc/soc-topology-test.c @@ -271,9 +271,7 @@ static void snd_soc_tplg_test_load_with_null_comp(struct kunit *test) KUNIT_EXPECT_EQ(test, 0, ret); /* cleanup */ - ret = snd_soc_unregister_card(&kunit_comp->card); - KUNIT_EXPECT_EQ(test, 0, ret); - + snd_soc_unregister_card(&kunit_comp->card); snd_soc_unregister_component(test_dev); } -- cgit v1.2.3 From 0deb003933052ac1a44b5f94b927484be6e34f86 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Uwe=20Kleine-K=C3=B6nig?= Date: Wed, 22 Jun 2022 08:17:39 +0200 Subject: ASoC: amd: acp: Fix error handling in .remove() MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Even in the presence of problems (here: rn_acp_deinit() might fail), it's important to unregister all resources acquired during .probe() because even if .remove() returns an error code, the device is removed. As .remove() is only called after .probe() returned success, platdata must be valid, so the first check in .remove() can just be dropped. This is a preparation for making platform remove callbacks return void. Signed-off-by: Uwe Kleine-König Link: https://lore.kernel.org/r/20220622061739.225966-1-u.kleine-koenig@pengutronix.de Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp-renoir.c | 10 ++-------- 1 file changed, 2 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/amd/acp/acp-renoir.c b/sound/soc/amd/acp/acp-renoir.c index 75c9229ece97..8375c00ff4c3 100644 --- a/sound/soc/amd/acp/acp-renoir.c +++ b/sound/soc/amd/acp/acp-renoir.c @@ -307,16 +307,10 @@ static int renoir_audio_remove(struct platform_device *pdev) int ret; chip = dev_get_platdata(&pdev->dev); - if (!chip || !chip->base) { - dev_err(&pdev->dev, "ACP chip data is NULL\n"); - return -ENODEV; - } ret = rn_acp_deinit(chip->base); - if (ret) { - dev_err(&pdev->dev, "ACP de-init Failed\n"); - return -EINVAL; - } + if (ret) + dev_err(&pdev->dev, "ACP de-init Failed (%pe)\n", ERR_PTR(ret)); acp_platform_unregister(dev); return 0; -- cgit v1.2.3 From c3b5fd7fbb698496461f280728b456d9927f22af Mon Sep 17 00:00:00 2001 From: Chen-Yu Tsai Date: Wed, 22 Jun 2022 02:57:47 +0800 Subject: ASoC: rockchip: i2s: Fix crash on missing pinctrl Commit 44f362c2cc6d ("ASoC: rockchip: i2s: switch BCLK to GPIO") added pinctrl lookups, but did not skip the lookup if there was no pinctrl device tied to the I2S controller. As a result, the lookup was done on an invalid pointer in such cases, causing a kernel panic. Only do the subsequent pinctrl state lookups and switch if a pinctrl device was found. i2s_pinctrl_select_bclk_{on,off} already guard against missing pinctrl device or pinctrl state, so those two functions aren't touched. Fixes: 44f362c2cc6d ("ASoC: rockchip: i2s: switch BCLK to GPIO") Signed-off-by: Chen-Yu Tsai Link: https://lore.kernel.org/r/20220621185747.2782-1-wens@kernel.org Signed-off-by: Mark Brown --- sound/soc/rockchip/rockchip_i2s.c | 29 ++++++++++++++--------------- 1 file changed, 14 insertions(+), 15 deletions(-) (limited to 'sound') diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index be051e48b97b..9fa8ffd712ea 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -808,24 +808,23 @@ static int rockchip_i2s_probe(struct platform_device *pdev) i2s->bclk_ratio = 64; i2s->pinctrl = devm_pinctrl_get(&pdev->dev); - if (IS_ERR(i2s->pinctrl)) + if (IS_ERR(i2s->pinctrl)) { dev_err(&pdev->dev, "failed to find i2s pinctrl\n"); + } else { + i2s->bclk_on = pinctrl_lookup_state(i2s->pinctrl, "bclk_on"); + if (IS_ERR_OR_NULL(i2s->bclk_on)) + dev_err(&pdev->dev, "failed to find i2s default state\n"); + else + dev_dbg(&pdev->dev, "find i2s bclk state\n"); - i2s->bclk_on = pinctrl_lookup_state(i2s->pinctrl, - "bclk_on"); - if (IS_ERR_OR_NULL(i2s->bclk_on)) - dev_err(&pdev->dev, "failed to find i2s default state\n"); - else - dev_dbg(&pdev->dev, "find i2s bclk state\n"); - - i2s->bclk_off = pinctrl_lookup_state(i2s->pinctrl, - "bclk_off"); - if (IS_ERR_OR_NULL(i2s->bclk_off)) - dev_err(&pdev->dev, "failed to find i2s gpio state\n"); - else - dev_dbg(&pdev->dev, "find i2s bclk_off state\n"); + i2s->bclk_off = pinctrl_lookup_state(i2s->pinctrl, "bclk_off"); + if (IS_ERR_OR_NULL(i2s->bclk_off)) + dev_err(&pdev->dev, "failed to find i2s gpio state\n"); + else + dev_dbg(&pdev->dev, "find i2s bclk_off state\n"); - i2s_pinctrl_select_bclk_off(i2s); + i2s_pinctrl_select_bclk_off(i2s); + } i2s->playback_dma_data.addr = res->start + I2S_TXDR; i2s->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; -- cgit v1.2.3 From 8b99e24de3fae72ff5ef38832b94b1e41059eeed Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Mon, 20 Jun 2022 12:13:46 +0200 Subject: ASoC: Intel: Rename haswell source file to hsw_rt5640 MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Rename source file to drop any ambiguity. Signed-off-by: Cezary Rojewski Reviewed-by: Amadeusz Sławiński Tested-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220620101402.2684366-2-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/Makefile | 2 +- sound/soc/intel/boards/haswell.c | 202 ------------------------------------ sound/soc/intel/boards/hsw_rt5640.c | 202 ++++++++++++++++++++++++++++++++++++ 3 files changed, 203 insertions(+), 203 deletions(-) delete mode 100644 sound/soc/intel/boards/haswell.c create mode 100644 sound/soc/intel/boards/hsw_rt5640.c (limited to 'sound') diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 40c0c3d1c500..e479546a3d4b 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -1,5 +1,5 @@ # SPDX-License-Identifier: GPL-2.0-only -snd-soc-sst-haswell-objs := haswell.o +snd-soc-sst-haswell-objs := hsw_rt5640.o snd-soc-sst-bdw-rt5650-mach-objs := bdw-rt5650.o snd-soc-sst-bdw-rt5677-mach-objs := bdw-rt5677.o snd-soc-sst-broadwell-objs := broadwell.o diff --git a/sound/soc/intel/boards/haswell.c b/sound/soc/intel/boards/haswell.c deleted file mode 100644 index aa61e101f793..000000000000 --- a/sound/soc/intel/boards/haswell.c +++ /dev/null @@ -1,202 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-only -/* - * Intel Haswell Lynxpoint SST Audio - * - * Copyright (C) 2013, Intel Corporation. All rights reserved. - */ - -#include -#include -#include -#include -#include -#include -#include - -#include "../../codecs/rt5640.h" - -/* Haswell ULT platforms have a Headphone and Mic jack */ -static const struct snd_soc_dapm_widget haswell_widgets[] = { - SND_SOC_DAPM_HP("Headphones", NULL), - SND_SOC_DAPM_MIC("Mic", NULL), -}; - -static const struct snd_soc_dapm_route haswell_rt5640_map[] = { - - {"Headphones", NULL, "HPOR"}, - {"Headphones", NULL, "HPOL"}, - {"IN2P", NULL, "Mic"}, - - /* CODEC BE connections */ - {"SSP0 CODEC IN", NULL, "AIF1 Capture"}, - {"AIF1 Playback", NULL, "SSP0 CODEC OUT"}, -}; - -static int haswell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) -{ - struct snd_interval *rate = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_RATE); - struct snd_interval *channels = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_CHANNELS); - - /* The ADSP will covert the FE rate to 48k, stereo */ - rate->min = rate->max = 48000; - channels->min = channels->max = 2; - - /* set SSP0 to 16 bit */ - params_set_format(params, SNDRV_PCM_FORMAT_S16_LE); - return 0; -} - -static int haswell_rt5640_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); - struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); - int ret; - - ret = snd_soc_dai_set_sysclk(codec_dai, RT5640_SCLK_S_MCLK, 12288000, - SND_SOC_CLOCK_IN); - - if (ret < 0) { - dev_err(rtd->dev, "can't set codec sysclk configuration\n"); - return ret; - } - - /* set correct codec filter for DAI format and clock config */ - snd_soc_component_update_bits(codec_dai->component, 0x83, 0xffff, 0x8000); - - return ret; -} - -static const struct snd_soc_ops haswell_rt5640_ops = { - .hw_params = haswell_rt5640_hw_params, -}; - -SND_SOC_DAILINK_DEF(dummy, - DAILINK_COMP_ARRAY(COMP_DUMMY())); - -SND_SOC_DAILINK_DEF(system, - DAILINK_COMP_ARRAY(COMP_CPU("System Pin"))); - -SND_SOC_DAILINK_DEF(offload0, - DAILINK_COMP_ARRAY(COMP_CPU("Offload0 Pin"))); - -SND_SOC_DAILINK_DEF(offload1, - DAILINK_COMP_ARRAY(COMP_CPU("Offload1 Pin"))); - -SND_SOC_DAILINK_DEF(loopback, - DAILINK_COMP_ARRAY(COMP_CPU("Loopback Pin"))); - -SND_SOC_DAILINK_DEF(codec, - DAILINK_COMP_ARRAY(COMP_CODEC("i2c-INT33CA:00", "rt5640-aif1"))); - -SND_SOC_DAILINK_DEF(platform, - DAILINK_COMP_ARRAY(COMP_PLATFORM("haswell-pcm-audio"))); - -SND_SOC_DAILINK_DEF(ssp0_port, - DAILINK_COMP_ARRAY(COMP_CPU("ssp0-port"))); - -static struct snd_soc_dai_link haswell_rt5640_dais[] = { - /* Front End DAI links */ - { - .name = "System", - .stream_name = "System Playback/Capture", - .nonatomic = 1, - .dynamic = 1, - .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_playback = 1, - .dpcm_capture = 1, - SND_SOC_DAILINK_REG(system, dummy, platform), - }, - { - .name = "Offload0", - .stream_name = "Offload0 Playback", - .nonatomic = 1, - .dynamic = 1, - .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_playback = 1, - SND_SOC_DAILINK_REG(offload0, dummy, platform), - }, - { - .name = "Offload1", - .stream_name = "Offload1 Playback", - .nonatomic = 1, - .dynamic = 1, - .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_playback = 1, - SND_SOC_DAILINK_REG(offload1, dummy, platform), - }, - { - .name = "Loopback", - .stream_name = "Loopback", - .nonatomic = 1, - .dynamic = 1, - .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_capture = 1, - SND_SOC_DAILINK_REG(loopback, dummy, platform), - }, - - /* Back End DAI links */ - { - /* SSP0 - Codec */ - .name = "Codec", - .id = 0, - .no_pcm = 1, - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBC_CFC, - .ignore_pmdown_time = 1, - .be_hw_params_fixup = haswell_ssp0_fixup, - .ops = &haswell_rt5640_ops, - .dpcm_playback = 1, - .dpcm_capture = 1, - SND_SOC_DAILINK_REG(ssp0_port, codec, platform), - }, -}; - -/* audio machine driver for Haswell Lynxpoint DSP + RT5640 */ -static struct snd_soc_card haswell_rt5640 = { - .name = "haswell-rt5640", - .owner = THIS_MODULE, - .dai_link = haswell_rt5640_dais, - .num_links = ARRAY_SIZE(haswell_rt5640_dais), - .dapm_widgets = haswell_widgets, - .num_dapm_widgets = ARRAY_SIZE(haswell_widgets), - .dapm_routes = haswell_rt5640_map, - .num_dapm_routes = ARRAY_SIZE(haswell_rt5640_map), - .fully_routed = true, -}; - -static int haswell_audio_probe(struct platform_device *pdev) -{ - struct snd_soc_acpi_mach *mach; - int ret; - - haswell_rt5640.dev = &pdev->dev; - - /* override platform name, if required */ - mach = pdev->dev.platform_data; - ret = snd_soc_fixup_dai_links_platform_name(&haswell_rt5640, - mach->mach_params.platform); - if (ret) - return ret; - - return devm_snd_soc_register_card(&pdev->dev, &haswell_rt5640); -} - -static struct platform_driver haswell_audio = { - .probe = haswell_audio_probe, - .driver = { - .name = "haswell-audio", - .pm = &snd_soc_pm_ops, - }, -}; - -module_platform_driver(haswell_audio) - -/* Module information */ -MODULE_AUTHOR("Liam Girdwood, Xingchao Wang"); -MODULE_DESCRIPTION("Intel SST Audio for Haswell Lynxpoint"); -MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:haswell-audio"); diff --git a/sound/soc/intel/boards/hsw_rt5640.c b/sound/soc/intel/boards/hsw_rt5640.c new file mode 100644 index 000000000000..aa61e101f793 --- /dev/null +++ b/sound/soc/intel/boards/hsw_rt5640.c @@ -0,0 +1,202 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * Intel Haswell Lynxpoint SST Audio + * + * Copyright (C) 2013, Intel Corporation. All rights reserved. + */ + +#include +#include +#include +#include +#include +#include +#include + +#include "../../codecs/rt5640.h" + +/* Haswell ULT platforms have a Headphone and Mic jack */ +static const struct snd_soc_dapm_widget haswell_widgets[] = { + SND_SOC_DAPM_HP("Headphones", NULL), + SND_SOC_DAPM_MIC("Mic", NULL), +}; + +static const struct snd_soc_dapm_route haswell_rt5640_map[] = { + + {"Headphones", NULL, "HPOR"}, + {"Headphones", NULL, "HPOL"}, + {"IN2P", NULL, "Mic"}, + + /* CODEC BE connections */ + {"SSP0 CODEC IN", NULL, "AIF1 Capture"}, + {"AIF1 Playback", NULL, "SSP0 CODEC OUT"}, +}; + +static int haswell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + /* The ADSP will covert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP0 to 16 bit */ + params_set_format(params, SNDRV_PCM_FORMAT_S16_LE); + return 0; +} + +static int haswell_rt5640_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, RT5640_SCLK_S_MCLK, 12288000, + SND_SOC_CLOCK_IN); + + if (ret < 0) { + dev_err(rtd->dev, "can't set codec sysclk configuration\n"); + return ret; + } + + /* set correct codec filter for DAI format and clock config */ + snd_soc_component_update_bits(codec_dai->component, 0x83, 0xffff, 0x8000); + + return ret; +} + +static const struct snd_soc_ops haswell_rt5640_ops = { + .hw_params = haswell_rt5640_hw_params, +}; + +SND_SOC_DAILINK_DEF(dummy, + DAILINK_COMP_ARRAY(COMP_DUMMY())); + +SND_SOC_DAILINK_DEF(system, + DAILINK_COMP_ARRAY(COMP_CPU("System Pin"))); + +SND_SOC_DAILINK_DEF(offload0, + DAILINK_COMP_ARRAY(COMP_CPU("Offload0 Pin"))); + +SND_SOC_DAILINK_DEF(offload1, + DAILINK_COMP_ARRAY(COMP_CPU("Offload1 Pin"))); + +SND_SOC_DAILINK_DEF(loopback, + DAILINK_COMP_ARRAY(COMP_CPU("Loopback Pin"))); + +SND_SOC_DAILINK_DEF(codec, + DAILINK_COMP_ARRAY(COMP_CODEC("i2c-INT33CA:00", "rt5640-aif1"))); + +SND_SOC_DAILINK_DEF(platform, + DAILINK_COMP_ARRAY(COMP_PLATFORM("haswell-pcm-audio"))); + +SND_SOC_DAILINK_DEF(ssp0_port, + DAILINK_COMP_ARRAY(COMP_CPU("ssp0-port"))); + +static struct snd_soc_dai_link haswell_rt5640_dais[] = { + /* Front End DAI links */ + { + .name = "System", + .stream_name = "System Playback/Capture", + .nonatomic = 1, + .dynamic = 1, + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + .dpcm_capture = 1, + SND_SOC_DAILINK_REG(system, dummy, platform), + }, + { + .name = "Offload0", + .stream_name = "Offload0 Playback", + .nonatomic = 1, + .dynamic = 1, + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + SND_SOC_DAILINK_REG(offload0, dummy, platform), + }, + { + .name = "Offload1", + .stream_name = "Offload1 Playback", + .nonatomic = 1, + .dynamic = 1, + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + SND_SOC_DAILINK_REG(offload1, dummy, platform), + }, + { + .name = "Loopback", + .stream_name = "Loopback", + .nonatomic = 1, + .dynamic = 1, + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_capture = 1, + SND_SOC_DAILINK_REG(loopback, dummy, platform), + }, + + /* Back End DAI links */ + { + /* SSP0 - Codec */ + .name = "Codec", + .id = 0, + .no_pcm = 1, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBC_CFC, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = haswell_ssp0_fixup, + .ops = &haswell_rt5640_ops, + .dpcm_playback = 1, + .dpcm_capture = 1, + SND_SOC_DAILINK_REG(ssp0_port, codec, platform), + }, +}; + +/* audio machine driver for Haswell Lynxpoint DSP + RT5640 */ +static struct snd_soc_card haswell_rt5640 = { + .name = "haswell-rt5640", + .owner = THIS_MODULE, + .dai_link = haswell_rt5640_dais, + .num_links = ARRAY_SIZE(haswell_rt5640_dais), + .dapm_widgets = haswell_widgets, + .num_dapm_widgets = ARRAY_SIZE(haswell_widgets), + .dapm_routes = haswell_rt5640_map, + .num_dapm_routes = ARRAY_SIZE(haswell_rt5640_map), + .fully_routed = true, +}; + +static int haswell_audio_probe(struct platform_device *pdev) +{ + struct snd_soc_acpi_mach *mach; + int ret; + + haswell_rt5640.dev = &pdev->dev; + + /* override platform name, if required */ + mach = pdev->dev.platform_data; + ret = snd_soc_fixup_dai_links_platform_name(&haswell_rt5640, + mach->mach_params.platform); + if (ret) + return ret; + + return devm_snd_soc_register_card(&pdev->dev, &haswell_rt5640); +} + +static struct platform_driver haswell_audio = { + .probe = haswell_audio_probe, + .driver = { + .name = "haswell-audio", + .pm = &snd_soc_pm_ops, + }, +}; + +module_platform_driver(haswell_audio) + +/* Module information */ +MODULE_AUTHOR("Liam Girdwood, Xingchao Wang"); +MODULE_DESCRIPTION("Intel SST Audio for Haswell Lynxpoint"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:haswell-audio"); -- cgit v1.2.3 From 675002b6ca9132445e340bd106297d584e44fc9a Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Mon, 20 Jun 2022 12:13:47 +0200 Subject: ASoC: Intel: hsw_rt5640: Reword prefixes of all driver members MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Replace ambiguous 'haswell_rt5640_' prefixes in favour of 'card_', 'link_' and other similar strings to clearly state which object given member implements behavior for. Signed-off-by: Cezary Rojewski Reviewed-by: Amadeusz Sławiński Tested-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220620101402.2684366-3-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/hsw_rt5640.c | 46 ++++++++++++++++++------------------- 1 file changed, 23 insertions(+), 23 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/hsw_rt5640.c b/sound/soc/intel/boards/hsw_rt5640.c index aa61e101f793..b51ce8d0ca22 100644 --- a/sound/soc/intel/boards/hsw_rt5640.c +++ b/sound/soc/intel/boards/hsw_rt5640.c @@ -16,12 +16,12 @@ #include "../../codecs/rt5640.h" /* Haswell ULT platforms have a Headphone and Mic jack */ -static const struct snd_soc_dapm_widget haswell_widgets[] = { +static const struct snd_soc_dapm_widget card_widgets[] = { SND_SOC_DAPM_HP("Headphones", NULL), SND_SOC_DAPM_MIC("Mic", NULL), }; -static const struct snd_soc_dapm_route haswell_rt5640_map[] = { +static const struct snd_soc_dapm_route card_routes[] = { {"Headphones", NULL, "HPOR"}, {"Headphones", NULL, "HPOL"}, @@ -32,7 +32,7 @@ static const struct snd_soc_dapm_route haswell_rt5640_map[] = { {"AIF1 Playback", NULL, "SSP0 CODEC OUT"}, }; -static int haswell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, +static int codec_link_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { struct snd_interval *rate = hw_param_interval(params, @@ -49,7 +49,7 @@ static int haswell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } -static int haswell_rt5640_hw_params(struct snd_pcm_substream *substream, +static int codec_link_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); @@ -70,8 +70,8 @@ static int haswell_rt5640_hw_params(struct snd_pcm_substream *substream, return ret; } -static const struct snd_soc_ops haswell_rt5640_ops = { - .hw_params = haswell_rt5640_hw_params, +static const struct snd_soc_ops codec_link_ops = { + .hw_params = codec_link_hw_params, }; SND_SOC_DAILINK_DEF(dummy, @@ -98,7 +98,7 @@ SND_SOC_DAILINK_DEF(platform, SND_SOC_DAILINK_DEF(ssp0_port, DAILINK_COMP_ARRAY(COMP_CPU("ssp0-port"))); -static struct snd_soc_dai_link haswell_rt5640_dais[] = { +static struct snd_soc_dai_link card_dai_links[] = { /* Front End DAI links */ { .name = "System", @@ -147,8 +147,8 @@ static struct snd_soc_dai_link haswell_rt5640_dais[] = { .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBC_CFC, .ignore_pmdown_time = 1, - .be_hw_params_fixup = haswell_ssp0_fixup, - .ops = &haswell_rt5640_ops, + .be_hw_params_fixup = codec_link_hw_params_fixup, + .ops = &codec_link_ops, .dpcm_playback = 1, .dpcm_capture = 1, SND_SOC_DAILINK_REG(ssp0_port, codec, platform), @@ -156,44 +156,44 @@ static struct snd_soc_dai_link haswell_rt5640_dais[] = { }; /* audio machine driver for Haswell Lynxpoint DSP + RT5640 */ -static struct snd_soc_card haswell_rt5640 = { +static struct snd_soc_card hsw_rt5640_card = { .name = "haswell-rt5640", .owner = THIS_MODULE, - .dai_link = haswell_rt5640_dais, - .num_links = ARRAY_SIZE(haswell_rt5640_dais), - .dapm_widgets = haswell_widgets, - .num_dapm_widgets = ARRAY_SIZE(haswell_widgets), - .dapm_routes = haswell_rt5640_map, - .num_dapm_routes = ARRAY_SIZE(haswell_rt5640_map), + .dai_link = card_dai_links, + .num_links = ARRAY_SIZE(card_dai_links), + .dapm_widgets = card_widgets, + .num_dapm_widgets = ARRAY_SIZE(card_widgets), + .dapm_routes = card_routes, + .num_dapm_routes = ARRAY_SIZE(card_routes), .fully_routed = true, }; -static int haswell_audio_probe(struct platform_device *pdev) +static int hsw_rt5640_probe(struct platform_device *pdev) { struct snd_soc_acpi_mach *mach; int ret; - haswell_rt5640.dev = &pdev->dev; + hsw_rt5640_card.dev = &pdev->dev; /* override platform name, if required */ mach = pdev->dev.platform_data; - ret = snd_soc_fixup_dai_links_platform_name(&haswell_rt5640, + ret = snd_soc_fixup_dai_links_platform_name(&hsw_rt5640_card, mach->mach_params.platform); if (ret) return ret; - return devm_snd_soc_register_card(&pdev->dev, &haswell_rt5640); + return devm_snd_soc_register_card(&pdev->dev, &hsw_rt5640_card); } -static struct platform_driver haswell_audio = { - .probe = haswell_audio_probe, +static struct platform_driver hsw_rt5640_driver = { + .probe = hsw_rt5640_probe, .driver = { .name = "haswell-audio", .pm = &snd_soc_pm_ops, }, }; -module_platform_driver(haswell_audio) +module_platform_driver(hsw_rt5640_driver) /* Module information */ MODULE_AUTHOR("Liam Girdwood, Xingchao Wang"); -- cgit v1.2.3 From a69615e81709da0ff1f035886e8b3faf6125cd22 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Mon, 20 Jun 2022 12:13:48 +0200 Subject: ASoC: Intel: hsw_rt5640: Reword driver name MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Align with other Intel boards naming convention and let the name explicitly state which components are being connected. Signed-off-by: Cezary Rojewski Reviewed-by: Amadeusz Sławiński Tested-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220620101402.2684366-4-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/hsw_rt5640.c | 4 ++-- sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c | 4 ++-- 2 files changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/hsw_rt5640.c b/sound/soc/intel/boards/hsw_rt5640.c index b51ce8d0ca22..a096453bf1df 100644 --- a/sound/soc/intel/boards/hsw_rt5640.c +++ b/sound/soc/intel/boards/hsw_rt5640.c @@ -188,7 +188,7 @@ static int hsw_rt5640_probe(struct platform_device *pdev) static struct platform_driver hsw_rt5640_driver = { .probe = hsw_rt5640_probe, .driver = { - .name = "haswell-audio", + .name = "hsw_rt5640", .pm = &snd_soc_pm_ops, }, }; @@ -199,4 +199,4 @@ module_platform_driver(hsw_rt5640_driver) MODULE_AUTHOR("Liam Girdwood, Xingchao Wang"); MODULE_DESCRIPTION("Intel SST Audio for Haswell Lynxpoint"); MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:haswell-audio"); +MODULE_ALIAS("platform:hsw_rt5640"); diff --git a/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c b/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c index 0441df97b260..4e00f8f6c521 100644 --- a/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c @@ -12,7 +12,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_haswell_machines[] = { { .id = "INT33CA", - .drv_name = "haswell-audio", + .drv_name = "hsw_rt5640", .fw_filename = "intel/IntcSST1.bin", .sof_tplg_filename = "sof-hsw.tplg", }, @@ -41,7 +41,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_broadwell_machines[] = { }, { .id = "INT33CA", - .drv_name = "haswell-audio", + .drv_name = "hsw_rt5640", .fw_filename = "intel/IntcSST2.bin", .sof_tplg_filename = "sof-bdw-rt5640.tplg", }, -- cgit v1.2.3 From 5b66dde4ada531c1a2417d8daf68004067932a19 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Mon, 20 Jun 2022 12:13:49 +0200 Subject: ASoC: Intel: hsw_rt5640: Update code indentation MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Make use of 100 character limit and modify indentation so code is easier to read. While at it, sort includes in alphabetical order. Signed-off-by: Cezary Rojewski Reviewed-by: Amadeusz Sławiński Tested-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220620101402.2684366-5-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/hsw_rt5640.c | 59 +++++++++++-------------------------- 1 file changed, 18 insertions(+), 41 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/hsw_rt5640.c b/sound/soc/intel/boards/hsw_rt5640.c index a096453bf1df..da31b011b495 100644 --- a/sound/soc/intel/boards/hsw_rt5640.c +++ b/sound/soc/intel/boards/hsw_rt5640.c @@ -9,10 +9,9 @@ #include #include #include +#include #include #include -#include - #include "../../codecs/rt5640.h" /* Haswell ULT platforms have a Headphone and Mic jack */ @@ -22,7 +21,6 @@ static const struct snd_soc_dapm_widget card_widgets[] = { }; static const struct snd_soc_dapm_route card_routes[] = { - {"Headphones", NULL, "HPOR"}, {"Headphones", NULL, "HPOL"}, {"IN2P", NULL, "Mic"}, @@ -33,32 +31,28 @@ static const struct snd_soc_dapm_route card_routes[] = { }; static int codec_link_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params) { - struct snd_interval *rate = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_RATE); - struct snd_interval *channels = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_interval *channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_interval *rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); /* The ADSP will covert the FE rate to 48k, stereo */ rate->min = rate->max = 48000; channels->min = channels->max = 2; - /* set SSP0 to 16 bit */ params_set_format(params, SNDRV_PCM_FORMAT_S16_LE); + return 0; } static int codec_link_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret; - ret = snd_soc_dai_set_sysclk(codec_dai, RT5640_SCLK_S_MCLK, 12288000, - SND_SOC_CLOCK_IN); - + ret = snd_soc_dai_set_sysclk(codec_dai, RT5640_SCLK_S_MCLK, 12288000, SND_SOC_CLOCK_IN); if (ret < 0) { dev_err(rtd->dev, "can't set codec sysclk configuration\n"); return ret; @@ -74,29 +68,15 @@ static const struct snd_soc_ops codec_link_ops = { .hw_params = codec_link_hw_params, }; -SND_SOC_DAILINK_DEF(dummy, - DAILINK_COMP_ARRAY(COMP_DUMMY())); - -SND_SOC_DAILINK_DEF(system, - DAILINK_COMP_ARRAY(COMP_CPU("System Pin"))); +SND_SOC_DAILINK_DEF(system, DAILINK_COMP_ARRAY(COMP_CPU("System Pin"))); +SND_SOC_DAILINK_DEF(offload0, DAILINK_COMP_ARRAY(COMP_CPU("Offload0 Pin"))); +SND_SOC_DAILINK_DEF(offload1, DAILINK_COMP_ARRAY(COMP_CPU("Offload1 Pin"))); +SND_SOC_DAILINK_DEF(loopback, DAILINK_COMP_ARRAY(COMP_CPU("Loopback Pin"))); -SND_SOC_DAILINK_DEF(offload0, - DAILINK_COMP_ARRAY(COMP_CPU("Offload0 Pin"))); - -SND_SOC_DAILINK_DEF(offload1, - DAILINK_COMP_ARRAY(COMP_CPU("Offload1 Pin"))); - -SND_SOC_DAILINK_DEF(loopback, - DAILINK_COMP_ARRAY(COMP_CPU("Loopback Pin"))); - -SND_SOC_DAILINK_DEF(codec, - DAILINK_COMP_ARRAY(COMP_CODEC("i2c-INT33CA:00", "rt5640-aif1"))); - -SND_SOC_DAILINK_DEF(platform, - DAILINK_COMP_ARRAY(COMP_PLATFORM("haswell-pcm-audio"))); - -SND_SOC_DAILINK_DEF(ssp0_port, - DAILINK_COMP_ARRAY(COMP_CPU("ssp0-port"))); +SND_SOC_DAILINK_DEF(dummy, DAILINK_COMP_ARRAY(COMP_DUMMY())); +SND_SOC_DAILINK_DEF(codec, DAILINK_COMP_ARRAY(COMP_CODEC("i2c-INT33CA:00", "rt5640-aif1"))); +SND_SOC_DAILINK_DEF(platform, DAILINK_COMP_ARRAY(COMP_PLATFORM("haswell-pcm-audio"))); +SND_SOC_DAILINK_DEF(ssp0_port, DAILINK_COMP_ARRAY(COMP_CPU("ssp0-port"))); static struct snd_soc_dai_link card_dai_links[] = { /* Front End DAI links */ @@ -137,15 +117,13 @@ static struct snd_soc_dai_link card_dai_links[] = { .dpcm_capture = 1, SND_SOC_DAILINK_REG(loopback, dummy, platform), }, - /* Back End DAI links */ { /* SSP0 - Codec */ .name = "Codec", .id = 0, .no_pcm = 1, - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBC_CFC, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBC_CFC, .ignore_pmdown_time = 1, .be_hw_params_fixup = codec_link_hw_params_fixup, .ops = &codec_link_ops, @@ -174,11 +152,10 @@ static int hsw_rt5640_probe(struct platform_device *pdev) int ret; hsw_rt5640_card.dev = &pdev->dev; - /* override platform name, if required */ mach = pdev->dev.platform_data; - ret = snd_soc_fixup_dai_links_platform_name(&hsw_rt5640_card, - mach->mach_params.platform); + + ret = snd_soc_fixup_dai_links_platform_name(&hsw_rt5640_card, mach->mach_params.platform); if (ret) return ret; -- cgit v1.2.3 From 2c53debbbf04eb40854fa33813514828fa455783 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Mon, 20 Jun 2022 12:13:50 +0200 Subject: ASoC: Intel: hsw_rt5640: Update file comments MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Drop redundant and update valuable comments within the file to increase readability. This patch also revisits module information and kconfig help strings. Signed-off-by: Cezary Rojewski Reviewed-by: Amadeusz Sławiński Tested-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220620101402.2684366-6-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/Kconfig | 2 +- sound/soc/intel/boards/hsw_rt5640.c | 16 ++++++---------- 2 files changed, 7 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 4b4c1e1e4808..817b4c04bf6a 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -41,7 +41,7 @@ config SND_SOC_INTEL_SOF_CIRRUS_COMMON if SND_SOC_INTEL_CATPT config SND_SOC_INTEL_HASWELL_MACH - tristate "Haswell Lynxpoint" + tristate "Haswell with RT5640 I2S codec" depends on I2C depends on I2C_DESIGNWARE_PLATFORM || COMPILE_TEST depends on X86_INTEL_LPSS || COMPILE_TEST diff --git a/sound/soc/intel/boards/hsw_rt5640.c b/sound/soc/intel/boards/hsw_rt5640.c index da31b011b495..f843ba5f4718 100644 --- a/sound/soc/intel/boards/hsw_rt5640.c +++ b/sound/soc/intel/boards/hsw_rt5640.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only /* - * Intel Haswell Lynxpoint SST Audio + * Sound card driver for Intel Haswell Lynx Point with Realtek 5640 * * Copyright (C) 2013, Intel Corporation. All rights reserved. */ @@ -14,7 +14,6 @@ #include #include "../../codecs/rt5640.h" -/* Haswell ULT platforms have a Headphone and Mic jack */ static const struct snd_soc_dapm_widget card_widgets[] = { SND_SOC_DAPM_HP("Headphones", NULL), SND_SOC_DAPM_MIC("Mic", NULL), @@ -36,10 +35,10 @@ static int codec_link_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_interval *channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_interval *rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); - /* The ADSP will covert the FE rate to 48k, stereo */ + /* The ADSP will convert the FE rate to 48k, stereo. */ rate->min = rate->max = 48000; channels->min = channels->max = 2; - /* set SSP0 to 16 bit */ + /* Set SSP0 to 16 bit. */ params_set_format(params, SNDRV_PCM_FORMAT_S16_LE); return 0; @@ -58,7 +57,7 @@ static int codec_link_hw_params(struct snd_pcm_substream *substream, return ret; } - /* set correct codec filter for DAI format and clock config */ + /* Set correct codec filter for DAI format and clock config. */ snd_soc_component_update_bits(codec_dai->component, 0x83, 0xffff, 0x8000); return ret; @@ -133,7 +132,6 @@ static struct snd_soc_dai_link card_dai_links[] = { }, }; -/* audio machine driver for Haswell Lynxpoint DSP + RT5640 */ static struct snd_soc_card hsw_rt5640_card = { .name = "haswell-rt5640", .owner = THIS_MODULE, @@ -152,7 +150,6 @@ static int hsw_rt5640_probe(struct platform_device *pdev) int ret; hsw_rt5640_card.dev = &pdev->dev; - /* override platform name, if required */ mach = pdev->dev.platform_data; ret = snd_soc_fixup_dai_links_platform_name(&hsw_rt5640_card, mach->mach_params.platform); @@ -172,8 +169,7 @@ static struct platform_driver hsw_rt5640_driver = { module_platform_driver(hsw_rt5640_driver) -/* Module information */ MODULE_AUTHOR("Liam Girdwood, Xingchao Wang"); -MODULE_DESCRIPTION("Intel SST Audio for Haswell Lynxpoint"); -MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("Sound card driver for Intel Haswell Lynx Point with Realtek 5640"); +MODULE_LICENSE("GPL"); MODULE_ALIAS("platform:hsw_rt5640"); -- cgit v1.2.3 From 0439f262a9b39734c1440733850969f0342c50c3 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Mon, 20 Jun 2022 12:13:51 +0200 Subject: ASoC: Intel: hsw_rt5640: Improve probe() function quality MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Declare local 'dev' and make use of it plus dev_get_platdata() to improve code readability. Signed-off-by: Cezary Rojewski Reviewed-by: Amadeusz Sławiński Tested-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220620101402.2684366-7-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/hsw_rt5640.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/hsw_rt5640.c b/sound/soc/intel/boards/hsw_rt5640.c index f843ba5f4718..273c8e274d25 100644 --- a/sound/soc/intel/boards/hsw_rt5640.c +++ b/sound/soc/intel/boards/hsw_rt5640.c @@ -147,16 +147,17 @@ static struct snd_soc_card hsw_rt5640_card = { static int hsw_rt5640_probe(struct platform_device *pdev) { struct snd_soc_acpi_mach *mach; + struct device *dev = &pdev->dev; int ret; - hsw_rt5640_card.dev = &pdev->dev; - mach = pdev->dev.platform_data; + hsw_rt5640_card.dev = dev; + mach = dev_get_platdata(dev); ret = snd_soc_fixup_dai_links_platform_name(&hsw_rt5640_card, mach->mach_params.platform); if (ret) return ret; - return devm_snd_soc_register_card(&pdev->dev, &hsw_rt5640_card); + return devm_snd_soc_register_card(dev, &hsw_rt5640_card); } static struct platform_driver hsw_rt5640_driver = { -- cgit v1.2.3 From 6c65908251edc637b53bdeb9e79d918a8d081183 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Mon, 20 Jun 2022 12:13:52 +0200 Subject: ASoC: Intel: hsw_rt5640: Improve hw_params() debug-ability MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Print status if setting sysclk fails. Signed-off-by: Cezary Rojewski Reviewed-by: Amadeusz Sławiński Tested-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220620101402.2684366-8-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/hsw_rt5640.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/hsw_rt5640.c b/sound/soc/intel/boards/hsw_rt5640.c index 273c8e274d25..ad747363d112 100644 --- a/sound/soc/intel/boards/hsw_rt5640.c +++ b/sound/soc/intel/boards/hsw_rt5640.c @@ -53,7 +53,7 @@ static int codec_link_hw_params(struct snd_pcm_substream *substream, ret = snd_soc_dai_set_sysclk(codec_dai, RT5640_SCLK_S_MCLK, 12288000, SND_SOC_CLOCK_IN); if (ret < 0) { - dev_err(rtd->dev, "can't set codec sysclk configuration\n"); + dev_err(rtd->dev, "set codec sysclk failed: %d\n", ret); return ret; } -- cgit v1.2.3 From 6d8758f6afd91cced9c6c5571337a5fbc6955bb2 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Mon, 20 Jun 2022 12:13:53 +0200 Subject: ASoC: Intel: Rename broadwell source file to bdw_rt286 MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Rename source file to drop any ambiguity. Signed-off-by: Cezary Rojewski Reviewed-by: Amadeusz Sławiński Tested-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220620101402.2684366-9-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/Makefile | 2 +- sound/soc/intel/boards/bdw_rt286.c | 338 +++++++++++++++++++++++++++++++++++++ sound/soc/intel/boards/broadwell.c | 338 ------------------------------------- 3 files changed, 339 insertions(+), 339 deletions(-) create mode 100644 sound/soc/intel/boards/bdw_rt286.c delete mode 100644 sound/soc/intel/boards/broadwell.c (limited to 'sound') diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index e479546a3d4b..eea1e26acfda 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -2,7 +2,7 @@ snd-soc-sst-haswell-objs := hsw_rt5640.o snd-soc-sst-bdw-rt5650-mach-objs := bdw-rt5650.o snd-soc-sst-bdw-rt5677-mach-objs := bdw-rt5677.o -snd-soc-sst-broadwell-objs := broadwell.o +snd-soc-sst-broadwell-objs := bdw_rt286.o snd-soc-sst-bxt-da7219_max98357a-objs := bxt_da7219_max98357a.o snd-soc-sst-bxt-rt298-objs := bxt_rt298.o snd-soc-sst-sof-pcm512x-objs := sof_pcm512x.o diff --git a/sound/soc/intel/boards/bdw_rt286.c b/sound/soc/intel/boards/bdw_rt286.c new file mode 100644 index 000000000000..48bf3241b3e6 --- /dev/null +++ b/sound/soc/intel/boards/bdw_rt286.c @@ -0,0 +1,338 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * Intel Broadwell Wildcatpoint SST Audio + * + * Copyright (C) 2013, Intel Corporation. All rights reserved. + */ + +#include +#include +#include +#include +#include +#include +#include +#include + +#include "../../codecs/rt286.h" + +static struct snd_soc_jack broadwell_headset; +/* Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin broadwell_headset_pins[] = { + { + .pin = "Mic Jack", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, +}; + +static const struct snd_kcontrol_new broadwell_controls[] = { + SOC_DAPM_PIN_SWITCH("Speaker"), + SOC_DAPM_PIN_SWITCH("Headphone Jack"), +}; + +static const struct snd_soc_dapm_widget broadwell_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_MIC("DMIC1", NULL), + SND_SOC_DAPM_MIC("DMIC2", NULL), + SND_SOC_DAPM_LINE("Line Jack", NULL), +}; + +static const struct snd_soc_dapm_route broadwell_rt286_map[] = { + + /* speaker */ + {"Speaker", NULL, "SPOR"}, + {"Speaker", NULL, "SPOL"}, + + /* HP jack connectors - unknown if we have jack deteck */ + {"Headphone Jack", NULL, "HPO Pin"}, + + /* other jacks */ + {"MIC1", NULL, "Mic Jack"}, + {"LINE1", NULL, "Line Jack"}, + + /* digital mics */ + {"DMIC1 Pin", NULL, "DMIC1"}, + {"DMIC2 Pin", NULL, "DMIC2"}, + + /* CODEC BE connections */ + {"SSP0 CODEC IN", NULL, "AIF1 Capture"}, + {"AIF1 Playback", NULL, "SSP0 CODEC OUT"}, +}; + +static int broadwell_rt286_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; + int ret = 0; + ret = snd_soc_card_jack_new_pins(rtd->card, "Headset", + SND_JACK_HEADSET | SND_JACK_BTN_0, &broadwell_headset, + broadwell_headset_pins, ARRAY_SIZE(broadwell_headset_pins)); + if (ret) + return ret; + + snd_soc_component_set_jack(component, &broadwell_headset, NULL); + return 0; +} + + +static int broadwell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *chan = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + /* The ADSP will covert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + chan->min = chan->max = 2; + + /* set SSP0 to 16 bit */ + params_set_format(params, SNDRV_PCM_FORMAT_S16_LE); + return 0; +} + +static int broadwell_rt286_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, RT286_SCLK_S_PLL, 24000000, + SND_SOC_CLOCK_IN); + + if (ret < 0) { + dev_err(rtd->dev, "can't set codec sysclk configuration\n"); + return ret; + } + + return ret; +} + +static const struct snd_soc_ops broadwell_rt286_ops = { + .hw_params = broadwell_rt286_hw_params, +}; + +static const unsigned int channels[] = { + 2, +}; + +static const struct snd_pcm_hw_constraint_list constraints_channels = { + .count = ARRAY_SIZE(channels), + .list = channels, + .mask = 0, +}; + +static int broadwell_fe_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + /* Board supports stereo configuration only */ + runtime->hw.channels_max = 2; + return snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); +} + +static const struct snd_soc_ops broadwell_fe_ops = { + .startup = broadwell_fe_startup, +}; + +SND_SOC_DAILINK_DEF(system, + DAILINK_COMP_ARRAY(COMP_CPU("System Pin"))); + +SND_SOC_DAILINK_DEF(offload0, + DAILINK_COMP_ARRAY(COMP_CPU("Offload0 Pin"))); + +SND_SOC_DAILINK_DEF(offload1, + DAILINK_COMP_ARRAY(COMP_CPU("Offload1 Pin"))); + +SND_SOC_DAILINK_DEF(loopback, + DAILINK_COMP_ARRAY(COMP_CPU("Loopback Pin"))); + +SND_SOC_DAILINK_DEF(dummy, + DAILINK_COMP_ARRAY(COMP_DUMMY())); + +SND_SOC_DAILINK_DEF(platform, + DAILINK_COMP_ARRAY(COMP_PLATFORM("haswell-pcm-audio"))); + +SND_SOC_DAILINK_DEF(codec, + DAILINK_COMP_ARRAY(COMP_CODEC("i2c-INT343A:00", "rt286-aif1"))); + +SND_SOC_DAILINK_DEF(ssp0_port, + DAILINK_COMP_ARRAY(COMP_CPU("ssp0-port"))); + +/* broadwell digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link broadwell_rt286_dais[] = { + /* Front End DAI links */ + { + .name = "System PCM", + .stream_name = "System Playback/Capture", + .nonatomic = 1, + .dynamic = 1, + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .ops = &broadwell_fe_ops, + .dpcm_playback = 1, + .dpcm_capture = 1, + SND_SOC_DAILINK_REG(system, dummy, platform), + }, + { + .name = "Offload0", + .stream_name = "Offload0 Playback", + .nonatomic = 1, + .dynamic = 1, + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + SND_SOC_DAILINK_REG(offload0, dummy, platform), + }, + { + .name = "Offload1", + .stream_name = "Offload1 Playback", + .nonatomic = 1, + .dynamic = 1, + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + SND_SOC_DAILINK_REG(offload1, dummy, platform), + }, + { + .name = "Loopback PCM", + .stream_name = "Loopback", + .nonatomic = 1, + .dynamic = 1, + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_capture = 1, + SND_SOC_DAILINK_REG(loopback, dummy, platform), + }, + /* Back End DAI links */ + { + /* SSP0 - Codec */ + .name = "Codec", + .id = 0, + .no_pcm = 1, + .init = broadwell_rt286_codec_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBC_CFC, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = broadwell_ssp0_fixup, + .ops = &broadwell_rt286_ops, + .dpcm_playback = 1, + .dpcm_capture = 1, + SND_SOC_DAILINK_REG(ssp0_port, codec, platform), + }, +}; + +static void broadwell_disable_jack(struct snd_soc_card *card) +{ + struct snd_soc_component *component; + + for_each_card_components(card, component) { + if (!strcmp(component->name, "i2c-INT343A:00")) { + + dev_dbg(component->dev, "disabling jack detect before going to suspend.\n"); + snd_soc_component_set_jack(component, NULL, NULL); + break; + } + } +} + +static int broadwell_suspend(struct snd_soc_card *card) +{ + broadwell_disable_jack(card); + + return 0; +} + +static int broadwell_resume(struct snd_soc_card *card){ + struct snd_soc_component *component; + + for_each_card_components(card, component) { + if (!strcmp(component->name, "i2c-INT343A:00")) { + + dev_dbg(component->dev, "enabling jack detect for resume.\n"); + snd_soc_component_set_jack(component, &broadwell_headset, NULL); + break; + } + } + return 0; +} + +/* use space before codec name to simplify card ID, and simplify driver name */ +#define SOF_CARD_NAME "bdw rt286" /* card name will be 'sof-bdw rt286' */ +#define SOF_DRIVER_NAME "SOF" + +#define CARD_NAME "broadwell-rt286" +#define DRIVER_NAME NULL /* card name will be used for driver name */ + +/* broadwell audio machine driver for WPT + RT286S */ +static struct snd_soc_card broadwell_rt286 = { + .owner = THIS_MODULE, + .dai_link = broadwell_rt286_dais, + .num_links = ARRAY_SIZE(broadwell_rt286_dais), + .controls = broadwell_controls, + .num_controls = ARRAY_SIZE(broadwell_controls), + .dapm_widgets = broadwell_widgets, + .num_dapm_widgets = ARRAY_SIZE(broadwell_widgets), + .dapm_routes = broadwell_rt286_map, + .num_dapm_routes = ARRAY_SIZE(broadwell_rt286_map), + .fully_routed = true, + .suspend_pre = broadwell_suspend, + .resume_post = broadwell_resume, +}; + +static int broadwell_audio_probe(struct platform_device *pdev) +{ + struct snd_soc_acpi_mach *mach; + int ret; + + broadwell_rt286.dev = &pdev->dev; + + /* override platform name, if required */ + mach = pdev->dev.platform_data; + ret = snd_soc_fixup_dai_links_platform_name(&broadwell_rt286, + mach->mach_params.platform); + if (ret) + return ret; + + /* set card and driver name */ + if (snd_soc_acpi_sof_parent(&pdev->dev)) { + broadwell_rt286.name = SOF_CARD_NAME; + broadwell_rt286.driver_name = SOF_DRIVER_NAME; + } else { + broadwell_rt286.name = CARD_NAME; + broadwell_rt286.driver_name = DRIVER_NAME; + } + + return devm_snd_soc_register_card(&pdev->dev, &broadwell_rt286); +} + +static int broadwell_audio_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + broadwell_disable_jack(card); + + return 0; +} + +static struct platform_driver broadwell_audio = { + .probe = broadwell_audio_probe, + .remove = broadwell_audio_remove, + .driver = { + .name = "broadwell-audio", + .pm = &snd_soc_pm_ops + }, +}; + +module_platform_driver(broadwell_audio) + +/* Module information */ +MODULE_AUTHOR("Liam Girdwood, Xingchao Wang"); +MODULE_DESCRIPTION("Intel SST Audio for WPT/Broadwell"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:broadwell-audio"); diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c deleted file mode 100644 index 48bf3241b3e6..000000000000 --- a/sound/soc/intel/boards/broadwell.c +++ /dev/null @@ -1,338 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-only -/* - * Intel Broadwell Wildcatpoint SST Audio - * - * Copyright (C) 2013, Intel Corporation. All rights reserved. - */ - -#include -#include -#include -#include -#include -#include -#include -#include - -#include "../../codecs/rt286.h" - -static struct snd_soc_jack broadwell_headset; -/* Headset jack detection DAPM pins */ -static struct snd_soc_jack_pin broadwell_headset_pins[] = { - { - .pin = "Mic Jack", - .mask = SND_JACK_MICROPHONE, - }, - { - .pin = "Headphone Jack", - .mask = SND_JACK_HEADPHONE, - }, -}; - -static const struct snd_kcontrol_new broadwell_controls[] = { - SOC_DAPM_PIN_SWITCH("Speaker"), - SOC_DAPM_PIN_SWITCH("Headphone Jack"), -}; - -static const struct snd_soc_dapm_widget broadwell_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_SPK("Speaker", NULL), - SND_SOC_DAPM_MIC("Mic Jack", NULL), - SND_SOC_DAPM_MIC("DMIC1", NULL), - SND_SOC_DAPM_MIC("DMIC2", NULL), - SND_SOC_DAPM_LINE("Line Jack", NULL), -}; - -static const struct snd_soc_dapm_route broadwell_rt286_map[] = { - - /* speaker */ - {"Speaker", NULL, "SPOR"}, - {"Speaker", NULL, "SPOL"}, - - /* HP jack connectors - unknown if we have jack deteck */ - {"Headphone Jack", NULL, "HPO Pin"}, - - /* other jacks */ - {"MIC1", NULL, "Mic Jack"}, - {"LINE1", NULL, "Line Jack"}, - - /* digital mics */ - {"DMIC1 Pin", NULL, "DMIC1"}, - {"DMIC2 Pin", NULL, "DMIC2"}, - - /* CODEC BE connections */ - {"SSP0 CODEC IN", NULL, "AIF1 Capture"}, - {"AIF1 Playback", NULL, "SSP0 CODEC OUT"}, -}; - -static int broadwell_rt286_codec_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; - int ret = 0; - ret = snd_soc_card_jack_new_pins(rtd->card, "Headset", - SND_JACK_HEADSET | SND_JACK_BTN_0, &broadwell_headset, - broadwell_headset_pins, ARRAY_SIZE(broadwell_headset_pins)); - if (ret) - return ret; - - snd_soc_component_set_jack(component, &broadwell_headset, NULL); - return 0; -} - - -static int broadwell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) -{ - struct snd_interval *rate = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_RATE); - struct snd_interval *chan = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_CHANNELS); - - /* The ADSP will covert the FE rate to 48k, stereo */ - rate->min = rate->max = 48000; - chan->min = chan->max = 2; - - /* set SSP0 to 16 bit */ - params_set_format(params, SNDRV_PCM_FORMAT_S16_LE); - return 0; -} - -static int broadwell_rt286_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); - struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); - int ret; - - ret = snd_soc_dai_set_sysclk(codec_dai, RT286_SCLK_S_PLL, 24000000, - SND_SOC_CLOCK_IN); - - if (ret < 0) { - dev_err(rtd->dev, "can't set codec sysclk configuration\n"); - return ret; - } - - return ret; -} - -static const struct snd_soc_ops broadwell_rt286_ops = { - .hw_params = broadwell_rt286_hw_params, -}; - -static const unsigned int channels[] = { - 2, -}; - -static const struct snd_pcm_hw_constraint_list constraints_channels = { - .count = ARRAY_SIZE(channels), - .list = channels, - .mask = 0, -}; - -static int broadwell_fe_startup(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - - /* Board supports stereo configuration only */ - runtime->hw.channels_max = 2; - return snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_CHANNELS, - &constraints_channels); -} - -static const struct snd_soc_ops broadwell_fe_ops = { - .startup = broadwell_fe_startup, -}; - -SND_SOC_DAILINK_DEF(system, - DAILINK_COMP_ARRAY(COMP_CPU("System Pin"))); - -SND_SOC_DAILINK_DEF(offload0, - DAILINK_COMP_ARRAY(COMP_CPU("Offload0 Pin"))); - -SND_SOC_DAILINK_DEF(offload1, - DAILINK_COMP_ARRAY(COMP_CPU("Offload1 Pin"))); - -SND_SOC_DAILINK_DEF(loopback, - DAILINK_COMP_ARRAY(COMP_CPU("Loopback Pin"))); - -SND_SOC_DAILINK_DEF(dummy, - DAILINK_COMP_ARRAY(COMP_DUMMY())); - -SND_SOC_DAILINK_DEF(platform, - DAILINK_COMP_ARRAY(COMP_PLATFORM("haswell-pcm-audio"))); - -SND_SOC_DAILINK_DEF(codec, - DAILINK_COMP_ARRAY(COMP_CODEC("i2c-INT343A:00", "rt286-aif1"))); - -SND_SOC_DAILINK_DEF(ssp0_port, - DAILINK_COMP_ARRAY(COMP_CPU("ssp0-port"))); - -/* broadwell digital audio interface glue - connects codec <--> CPU */ -static struct snd_soc_dai_link broadwell_rt286_dais[] = { - /* Front End DAI links */ - { - .name = "System PCM", - .stream_name = "System Playback/Capture", - .nonatomic = 1, - .dynamic = 1, - .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .ops = &broadwell_fe_ops, - .dpcm_playback = 1, - .dpcm_capture = 1, - SND_SOC_DAILINK_REG(system, dummy, platform), - }, - { - .name = "Offload0", - .stream_name = "Offload0 Playback", - .nonatomic = 1, - .dynamic = 1, - .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_playback = 1, - SND_SOC_DAILINK_REG(offload0, dummy, platform), - }, - { - .name = "Offload1", - .stream_name = "Offload1 Playback", - .nonatomic = 1, - .dynamic = 1, - .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_playback = 1, - SND_SOC_DAILINK_REG(offload1, dummy, platform), - }, - { - .name = "Loopback PCM", - .stream_name = "Loopback", - .nonatomic = 1, - .dynamic = 1, - .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_capture = 1, - SND_SOC_DAILINK_REG(loopback, dummy, platform), - }, - /* Back End DAI links */ - { - /* SSP0 - Codec */ - .name = "Codec", - .id = 0, - .no_pcm = 1, - .init = broadwell_rt286_codec_init, - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBC_CFC, - .ignore_pmdown_time = 1, - .be_hw_params_fixup = broadwell_ssp0_fixup, - .ops = &broadwell_rt286_ops, - .dpcm_playback = 1, - .dpcm_capture = 1, - SND_SOC_DAILINK_REG(ssp0_port, codec, platform), - }, -}; - -static void broadwell_disable_jack(struct snd_soc_card *card) -{ - struct snd_soc_component *component; - - for_each_card_components(card, component) { - if (!strcmp(component->name, "i2c-INT343A:00")) { - - dev_dbg(component->dev, "disabling jack detect before going to suspend.\n"); - snd_soc_component_set_jack(component, NULL, NULL); - break; - } - } -} - -static int broadwell_suspend(struct snd_soc_card *card) -{ - broadwell_disable_jack(card); - - return 0; -} - -static int broadwell_resume(struct snd_soc_card *card){ - struct snd_soc_component *component; - - for_each_card_components(card, component) { - if (!strcmp(component->name, "i2c-INT343A:00")) { - - dev_dbg(component->dev, "enabling jack detect for resume.\n"); - snd_soc_component_set_jack(component, &broadwell_headset, NULL); - break; - } - } - return 0; -} - -/* use space before codec name to simplify card ID, and simplify driver name */ -#define SOF_CARD_NAME "bdw rt286" /* card name will be 'sof-bdw rt286' */ -#define SOF_DRIVER_NAME "SOF" - -#define CARD_NAME "broadwell-rt286" -#define DRIVER_NAME NULL /* card name will be used for driver name */ - -/* broadwell audio machine driver for WPT + RT286S */ -static struct snd_soc_card broadwell_rt286 = { - .owner = THIS_MODULE, - .dai_link = broadwell_rt286_dais, - .num_links = ARRAY_SIZE(broadwell_rt286_dais), - .controls = broadwell_controls, - .num_controls = ARRAY_SIZE(broadwell_controls), - .dapm_widgets = broadwell_widgets, - .num_dapm_widgets = ARRAY_SIZE(broadwell_widgets), - .dapm_routes = broadwell_rt286_map, - .num_dapm_routes = ARRAY_SIZE(broadwell_rt286_map), - .fully_routed = true, - .suspend_pre = broadwell_suspend, - .resume_post = broadwell_resume, -}; - -static int broadwell_audio_probe(struct platform_device *pdev) -{ - struct snd_soc_acpi_mach *mach; - int ret; - - broadwell_rt286.dev = &pdev->dev; - - /* override platform name, if required */ - mach = pdev->dev.platform_data; - ret = snd_soc_fixup_dai_links_platform_name(&broadwell_rt286, - mach->mach_params.platform); - if (ret) - return ret; - - /* set card and driver name */ - if (snd_soc_acpi_sof_parent(&pdev->dev)) { - broadwell_rt286.name = SOF_CARD_NAME; - broadwell_rt286.driver_name = SOF_DRIVER_NAME; - } else { - broadwell_rt286.name = CARD_NAME; - broadwell_rt286.driver_name = DRIVER_NAME; - } - - return devm_snd_soc_register_card(&pdev->dev, &broadwell_rt286); -} - -static int broadwell_audio_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - broadwell_disable_jack(card); - - return 0; -} - -static struct platform_driver broadwell_audio = { - .probe = broadwell_audio_probe, - .remove = broadwell_audio_remove, - .driver = { - .name = "broadwell-audio", - .pm = &snd_soc_pm_ops - }, -}; - -module_platform_driver(broadwell_audio) - -/* Module information */ -MODULE_AUTHOR("Liam Girdwood, Xingchao Wang"); -MODULE_DESCRIPTION("Intel SST Audio for WPT/Broadwell"); -MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:broadwell-audio"); -- cgit v1.2.3 From 40b5c9030a87e97c00c84403902481deadd2a57b Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Mon, 20 Jun 2022 12:13:54 +0200 Subject: ASoC: Intel: bdw_rt286: Reword prefixes of all driver members MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Replace ambiguous 'broadwell_rt286_' prefixes in favour of 'card_', 'link_' and other similar strings to clearly state which object given member implements behavior for. Signed-off-by: Cezary Rojewski Reviewed-by: Amadeusz Sławiński Tested-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220620101402.2684366-10-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/bdw_rt286.c | 102 ++++++++++++++++++------------------- 1 file changed, 51 insertions(+), 51 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/bdw_rt286.c b/sound/soc/intel/boards/bdw_rt286.c index 48bf3241b3e6..f28341ec8eb3 100644 --- a/sound/soc/intel/boards/bdw_rt286.c +++ b/sound/soc/intel/boards/bdw_rt286.c @@ -16,9 +16,9 @@ #include "../../codecs/rt286.h" -static struct snd_soc_jack broadwell_headset; +static struct snd_soc_jack card_headset; /* Headset jack detection DAPM pins */ -static struct snd_soc_jack_pin broadwell_headset_pins[] = { +static struct snd_soc_jack_pin card_headset_pins[] = { { .pin = "Mic Jack", .mask = SND_JACK_MICROPHONE, @@ -29,12 +29,12 @@ static struct snd_soc_jack_pin broadwell_headset_pins[] = { }, }; -static const struct snd_kcontrol_new broadwell_controls[] = { +static const struct snd_kcontrol_new card_controls[] = { SOC_DAPM_PIN_SWITCH("Speaker"), SOC_DAPM_PIN_SWITCH("Headphone Jack"), }; -static const struct snd_soc_dapm_widget broadwell_widgets[] = { +static const struct snd_soc_dapm_widget card_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_SPK("Speaker", NULL), SND_SOC_DAPM_MIC("Mic Jack", NULL), @@ -43,7 +43,7 @@ static const struct snd_soc_dapm_widget broadwell_widgets[] = { SND_SOC_DAPM_LINE("Line Jack", NULL), }; -static const struct snd_soc_dapm_route broadwell_rt286_map[] = { +static const struct snd_soc_dapm_route card_routes[] = { /* speaker */ {"Speaker", NULL, "SPOR"}, @@ -65,22 +65,22 @@ static const struct snd_soc_dapm_route broadwell_rt286_map[] = { {"AIF1 Playback", NULL, "SSP0 CODEC OUT"}, }; -static int broadwell_rt286_codec_init(struct snd_soc_pcm_runtime *rtd) +static int codec_link_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; int ret = 0; ret = snd_soc_card_jack_new_pins(rtd->card, "Headset", - SND_JACK_HEADSET | SND_JACK_BTN_0, &broadwell_headset, - broadwell_headset_pins, ARRAY_SIZE(broadwell_headset_pins)); + SND_JACK_HEADSET | SND_JACK_BTN_0, &card_headset, + card_headset_pins, ARRAY_SIZE(card_headset_pins)); if (ret) return ret; - snd_soc_component_set_jack(component, &broadwell_headset, NULL); + snd_soc_component_set_jack(component, &card_headset, NULL); return 0; } -static int broadwell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, +static int codec_link_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { struct snd_interval *rate = hw_param_interval(params, @@ -97,7 +97,7 @@ static int broadwell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } -static int broadwell_rt286_hw_params(struct snd_pcm_substream *substream, +static int codec_link_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); @@ -115,8 +115,8 @@ static int broadwell_rt286_hw_params(struct snd_pcm_substream *substream, return ret; } -static const struct snd_soc_ops broadwell_rt286_ops = { - .hw_params = broadwell_rt286_hw_params, +static const struct snd_soc_ops codec_link_ops = { + .hw_params = codec_link_hw_params, }; static const unsigned int channels[] = { @@ -129,7 +129,7 @@ static const struct snd_pcm_hw_constraint_list constraints_channels = { .mask = 0, }; -static int broadwell_fe_startup(struct snd_pcm_substream *substream) +static int bdw_rt286_fe_startup(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -140,8 +140,8 @@ static int broadwell_fe_startup(struct snd_pcm_substream *substream) &constraints_channels); } -static const struct snd_soc_ops broadwell_fe_ops = { - .startup = broadwell_fe_startup, +static const struct snd_soc_ops bdw_rt286_fe_ops = { + .startup = bdw_rt286_fe_startup, }; SND_SOC_DAILINK_DEF(system, @@ -169,7 +169,7 @@ SND_SOC_DAILINK_DEF(ssp0_port, DAILINK_COMP_ARRAY(COMP_CPU("ssp0-port"))); /* broadwell digital audio interface glue - connects codec <--> CPU */ -static struct snd_soc_dai_link broadwell_rt286_dais[] = { +static struct snd_soc_dai_link card_dai_links[] = { /* Front End DAI links */ { .name = "System PCM", @@ -177,7 +177,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = { .nonatomic = 1, .dynamic = 1, .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .ops = &broadwell_fe_ops, + .ops = &bdw_rt286_fe_ops, .dpcm_playback = 1, .dpcm_capture = 1, SND_SOC_DAILINK_REG(system, dummy, platform), @@ -215,19 +215,19 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = { .name = "Codec", .id = 0, .no_pcm = 1, - .init = broadwell_rt286_codec_init, + .init = codec_link_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBC_CFC, .ignore_pmdown_time = 1, - .be_hw_params_fixup = broadwell_ssp0_fixup, - .ops = &broadwell_rt286_ops, + .be_hw_params_fixup = codec_link_hw_params_fixup, + .ops = &codec_link_ops, .dpcm_playback = 1, .dpcm_capture = 1, SND_SOC_DAILINK_REG(ssp0_port, codec, platform), }, }; -static void broadwell_disable_jack(struct snd_soc_card *card) +static void bdw_rt286_disable_jack(struct snd_soc_card *card) { struct snd_soc_component *component; @@ -241,21 +241,21 @@ static void broadwell_disable_jack(struct snd_soc_card *card) } } -static int broadwell_suspend(struct snd_soc_card *card) +static int bdw_rt286_suspend(struct snd_soc_card *card) { - broadwell_disable_jack(card); + bdw_rt286_disable_jack(card); return 0; } -static int broadwell_resume(struct snd_soc_card *card){ +static int bdw_rt286_resume(struct snd_soc_card *card){ struct snd_soc_component *component; for_each_card_components(card, component) { if (!strcmp(component->name, "i2c-INT343A:00")) { dev_dbg(component->dev, "enabling jack detect for resume.\n"); - snd_soc_component_set_jack(component, &broadwell_headset, NULL); + snd_soc_component_set_jack(component, &card_headset, NULL); break; } } @@ -270,66 +270,66 @@ static int broadwell_resume(struct snd_soc_card *card){ #define DRIVER_NAME NULL /* card name will be used for driver name */ /* broadwell audio machine driver for WPT + RT286S */ -static struct snd_soc_card broadwell_rt286 = { +static struct snd_soc_card bdw_rt286_card = { .owner = THIS_MODULE, - .dai_link = broadwell_rt286_dais, - .num_links = ARRAY_SIZE(broadwell_rt286_dais), - .controls = broadwell_controls, - .num_controls = ARRAY_SIZE(broadwell_controls), - .dapm_widgets = broadwell_widgets, - .num_dapm_widgets = ARRAY_SIZE(broadwell_widgets), - .dapm_routes = broadwell_rt286_map, - .num_dapm_routes = ARRAY_SIZE(broadwell_rt286_map), + .dai_link = card_dai_links, + .num_links = ARRAY_SIZE(card_dai_links), + .controls = card_controls, + .num_controls = ARRAY_SIZE(card_controls), + .dapm_widgets = card_widgets, + .num_dapm_widgets = ARRAY_SIZE(card_widgets), + .dapm_routes = card_routes, + .num_dapm_routes = ARRAY_SIZE(card_routes), .fully_routed = true, - .suspend_pre = broadwell_suspend, - .resume_post = broadwell_resume, + .suspend_pre = bdw_rt286_suspend, + .resume_post = bdw_rt286_resume, }; -static int broadwell_audio_probe(struct platform_device *pdev) +static int bdw_rt286_probe(struct platform_device *pdev) { struct snd_soc_acpi_mach *mach; int ret; - broadwell_rt286.dev = &pdev->dev; + bdw_rt286_card.dev = &pdev->dev; /* override platform name, if required */ mach = pdev->dev.platform_data; - ret = snd_soc_fixup_dai_links_platform_name(&broadwell_rt286, + ret = snd_soc_fixup_dai_links_platform_name(&bdw_rt286_card, mach->mach_params.platform); if (ret) return ret; /* set card and driver name */ if (snd_soc_acpi_sof_parent(&pdev->dev)) { - broadwell_rt286.name = SOF_CARD_NAME; - broadwell_rt286.driver_name = SOF_DRIVER_NAME; + bdw_rt286_card.name = SOF_CARD_NAME; + bdw_rt286_card.driver_name = SOF_DRIVER_NAME; } else { - broadwell_rt286.name = CARD_NAME; - broadwell_rt286.driver_name = DRIVER_NAME; + bdw_rt286_card.name = CARD_NAME; + bdw_rt286_card.driver_name = DRIVER_NAME; } - return devm_snd_soc_register_card(&pdev->dev, &broadwell_rt286); + return devm_snd_soc_register_card(&pdev->dev, &bdw_rt286_card); } -static int broadwell_audio_remove(struct platform_device *pdev) +static int bdw_rt286_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); - broadwell_disable_jack(card); + bdw_rt286_disable_jack(card); return 0; } -static struct platform_driver broadwell_audio = { - .probe = broadwell_audio_probe, - .remove = broadwell_audio_remove, +static struct platform_driver bdw_rt286_driver = { + .probe = bdw_rt286_probe, + .remove = bdw_rt286_remove, .driver = { .name = "broadwell-audio", .pm = &snd_soc_pm_ops }, }; -module_platform_driver(broadwell_audio) +module_platform_driver(bdw_rt286_driver) /* Module information */ MODULE_AUTHOR("Liam Girdwood, Xingchao Wang"); -- cgit v1.2.3 From 86156bcbca08ee32d04ca56c57ff3fce6fc5fc4b Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Mon, 20 Jun 2022 12:13:55 +0200 Subject: ASoC: Intel: bdw_rt286: Reword driver name MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Align with other Intel boards naming convention and let the name explicitly state which components are being connected. Signed-off-by: Cezary Rojewski Reviewed-by: Amadeusz Sławiński Tested-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220620101402.2684366-11-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/bdw_rt286.c | 4 ++-- sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/bdw_rt286.c b/sound/soc/intel/boards/bdw_rt286.c index f28341ec8eb3..26ec671a5a52 100644 --- a/sound/soc/intel/boards/bdw_rt286.c +++ b/sound/soc/intel/boards/bdw_rt286.c @@ -324,7 +324,7 @@ static struct platform_driver bdw_rt286_driver = { .probe = bdw_rt286_probe, .remove = bdw_rt286_remove, .driver = { - .name = "broadwell-audio", + .name = "bdw_rt286", .pm = &snd_soc_pm_ops }, }; @@ -335,4 +335,4 @@ module_platform_driver(bdw_rt286_driver) MODULE_AUTHOR("Liam Girdwood, Xingchao Wang"); MODULE_DESCRIPTION("Intel SST Audio for WPT/Broadwell"); MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:broadwell-audio"); +MODULE_ALIAS("platform:bdw_rt286"); diff --git a/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c b/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c index 4e00f8f6c521..cbcb649604e5 100644 --- a/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c @@ -23,7 +23,7 @@ EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_haswell_machines); struct snd_soc_acpi_mach snd_soc_acpi_intel_broadwell_machines[] = { { .id = "INT343A", - .drv_name = "broadwell-audio", + .drv_name = "bdw_rt286", .fw_filename = "intel/IntcSST2.bin", .sof_tplg_filename = "sof-bdw-rt286.tplg", }, -- cgit v1.2.3 From 9de833d2dcd43c953f7869f27bffd41896adb425 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Mon, 20 Jun 2022 12:13:56 +0200 Subject: ASoC: Intel: bdw_rt286: Update code indentation MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Make use of 100 character limit and modify indentation so code is easier to read. While at it, sort includes in alphabetical order. While at it, rename local variable 'chan' to 'channels' to match hsw_rt5640 board's equivalent. Signed-off-by: Cezary Rojewski Reviewed-by: Amadeusz Sławiński Tested-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220620101402.2684366-12-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/bdw_rt286.c | 79 ++++++++++++++------------------------ 1 file changed, 28 insertions(+), 51 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/bdw_rt286.c b/sound/soc/intel/boards/bdw_rt286.c index 26ec671a5a52..6c0cd53224d5 100644 --- a/sound/soc/intel/boards/bdw_rt286.c +++ b/sound/soc/intel/boards/bdw_rt286.c @@ -8,12 +8,11 @@ #include #include #include -#include -#include #include +#include #include +#include #include - #include "../../codecs/rt286.h" static struct snd_soc_jack card_headset; @@ -44,7 +43,6 @@ static const struct snd_soc_dapm_widget card_widgets[] = { }; static const struct snd_soc_dapm_route card_routes[] = { - /* speaker */ {"Speaker", NULL, "SPOR"}, {"Speaker", NULL, "SPOL"}, @@ -69,9 +67,10 @@ static int codec_link_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; int ret = 0; - ret = snd_soc_card_jack_new_pins(rtd->card, "Headset", - SND_JACK_HEADSET | SND_JACK_BTN_0, &card_headset, - card_headset_pins, ARRAY_SIZE(card_headset_pins)); + + ret = snd_soc_card_jack_new_pins(rtd->card, "Headset", SND_JACK_HEADSET | SND_JACK_BTN_0, + &card_headset, card_headset_pins, + ARRAY_SIZE(card_headset_pins)); if (ret) return ret; @@ -79,34 +78,29 @@ static int codec_link_init(struct snd_soc_pcm_runtime *rtd) return 0; } - static int codec_link_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params) { - struct snd_interval *rate = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_RATE); - struct snd_interval *chan = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_interval *channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_interval *rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); /* The ADSP will covert the FE rate to 48k, stereo */ rate->min = rate->max = 48000; - chan->min = chan->max = 2; - + channels->min = channels->max = 2; /* set SSP0 to 16 bit */ params_set_format(params, SNDRV_PCM_FORMAT_S16_LE); + return 0; } static int codec_link_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret; - ret = snd_soc_dai_set_sysclk(codec_dai, RT286_SCLK_S_PLL, 24000000, - SND_SOC_CLOCK_IN); - + ret = snd_soc_dai_set_sysclk(codec_dai, RT286_SCLK_S_PLL, 24000000, SND_SOC_CLOCK_IN); if (ret < 0) { dev_err(rtd->dev, "can't set codec sysclk configuration\n"); return ret; @@ -135,8 +129,7 @@ static int bdw_rt286_fe_startup(struct snd_pcm_substream *substream) /* Board supports stereo configuration only */ runtime->hw.channels_max = 2; - return snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_CHANNELS, + return snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, &constraints_channels); } @@ -144,29 +137,15 @@ static const struct snd_soc_ops bdw_rt286_fe_ops = { .startup = bdw_rt286_fe_startup, }; -SND_SOC_DAILINK_DEF(system, - DAILINK_COMP_ARRAY(COMP_CPU("System Pin"))); - -SND_SOC_DAILINK_DEF(offload0, - DAILINK_COMP_ARRAY(COMP_CPU("Offload0 Pin"))); +SND_SOC_DAILINK_DEF(system, DAILINK_COMP_ARRAY(COMP_CPU("System Pin"))); +SND_SOC_DAILINK_DEF(offload0, DAILINK_COMP_ARRAY(COMP_CPU("Offload0 Pin"))); +SND_SOC_DAILINK_DEF(offload1, DAILINK_COMP_ARRAY(COMP_CPU("Offload1 Pin"))); +SND_SOC_DAILINK_DEF(loopback, DAILINK_COMP_ARRAY(COMP_CPU("Loopback Pin"))); -SND_SOC_DAILINK_DEF(offload1, - DAILINK_COMP_ARRAY(COMP_CPU("Offload1 Pin"))); - -SND_SOC_DAILINK_DEF(loopback, - DAILINK_COMP_ARRAY(COMP_CPU("Loopback Pin"))); - -SND_SOC_DAILINK_DEF(dummy, - DAILINK_COMP_ARRAY(COMP_DUMMY())); - -SND_SOC_DAILINK_DEF(platform, - DAILINK_COMP_ARRAY(COMP_PLATFORM("haswell-pcm-audio"))); - -SND_SOC_DAILINK_DEF(codec, - DAILINK_COMP_ARRAY(COMP_CODEC("i2c-INT343A:00", "rt286-aif1"))); - -SND_SOC_DAILINK_DEF(ssp0_port, - DAILINK_COMP_ARRAY(COMP_CPU("ssp0-port"))); +SND_SOC_DAILINK_DEF(dummy, DAILINK_COMP_ARRAY(COMP_DUMMY())); +SND_SOC_DAILINK_DEF(platform, DAILINK_COMP_ARRAY(COMP_PLATFORM("haswell-pcm-audio"))); +SND_SOC_DAILINK_DEF(codec, DAILINK_COMP_ARRAY(COMP_CODEC("i2c-INT343A:00", "rt286-aif1"))); +SND_SOC_DAILINK_DEF(ssp0_port, DAILINK_COMP_ARRAY(COMP_CPU("ssp0-port"))); /* broadwell digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link card_dai_links[] = { @@ -216,8 +195,7 @@ static struct snd_soc_dai_link card_dai_links[] = { .id = 0, .no_pcm = 1, .init = codec_link_init, - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBC_CFC, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBC_CFC, .ignore_pmdown_time = 1, .be_hw_params_fixup = codec_link_hw_params_fixup, .ops = &codec_link_ops, @@ -233,7 +211,6 @@ static void bdw_rt286_disable_jack(struct snd_soc_card *card) for_each_card_components(card, component) { if (!strcmp(component->name, "i2c-INT343A:00")) { - dev_dbg(component->dev, "disabling jack detect before going to suspend.\n"); snd_soc_component_set_jack(component, NULL, NULL); break; @@ -248,17 +225,18 @@ static int bdw_rt286_suspend(struct snd_soc_card *card) return 0; } -static int bdw_rt286_resume(struct snd_soc_card *card){ +static int bdw_rt286_resume(struct snd_soc_card *card) +{ struct snd_soc_component *component; for_each_card_components(card, component) { if (!strcmp(component->name, "i2c-INT343A:00")) { - dev_dbg(component->dev, "enabling jack detect for resume.\n"); snd_soc_component_set_jack(component, &card_headset, NULL); break; } } + return 0; } @@ -291,11 +269,10 @@ static int bdw_rt286_probe(struct platform_device *pdev) int ret; bdw_rt286_card.dev = &pdev->dev; - /* override platform name, if required */ mach = pdev->dev.platform_data; - ret = snd_soc_fixup_dai_links_platform_name(&bdw_rt286_card, - mach->mach_params.platform); + + ret = snd_soc_fixup_dai_links_platform_name(&bdw_rt286_card, mach->mach_params.platform); if (ret) return ret; -- cgit v1.2.3 From 128bb6fb530841348ee4d9b4234b30006c44c803 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Mon, 20 Jun 2022 12:13:57 +0200 Subject: ASoC: Intel: bdw_rt286: Update file comments MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Drop redundant and update valuable comments within the file to increase readability. This patch also revisits module information and kconfig help strings. Signed-off-by: Cezary Rojewski Reviewed-by: Amadeusz Sławiński Tested-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220620101402.2684366-13-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/Kconfig | 2 +- sound/soc/intel/boards/bdw_rt286.c | 23 +++++++---------------- 2 files changed, 8 insertions(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 817b4c04bf6a..aa12d7e3dd2f 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -85,7 +85,7 @@ config SND_SOC_INTEL_BDW_RT5677_MACH If unsure select "N". config SND_SOC_INTEL_BROADWELL_MACH - tristate "Broadwell Wildcatpoint" + tristate "Broadwell with RT286 I2S codec" depends on I2C depends on I2C_DESIGNWARE_PLATFORM || COMPILE_TEST depends on X86_INTEL_LPSS || COMPILE_TEST diff --git a/sound/soc/intel/boards/bdw_rt286.c b/sound/soc/intel/boards/bdw_rt286.c index 6c0cd53224d5..9d815c31e1f4 100644 --- a/sound/soc/intel/boards/bdw_rt286.c +++ b/sound/soc/intel/boards/bdw_rt286.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only /* - * Intel Broadwell Wildcatpoint SST Audio + * Sound card driver for Intel Broadwell Wildcat Point with Realtek 286 * * Copyright (C) 2013, Intel Corporation. All rights reserved. */ @@ -16,7 +16,7 @@ #include "../../codecs/rt286.h" static struct snd_soc_jack card_headset; -/* Headset jack detection DAPM pins */ + static struct snd_soc_jack_pin card_headset_pins[] = { { .pin = "Mic Jack", @@ -43,18 +43,14 @@ static const struct snd_soc_dapm_widget card_widgets[] = { }; static const struct snd_soc_dapm_route card_routes[] = { - /* speaker */ {"Speaker", NULL, "SPOR"}, {"Speaker", NULL, "SPOL"}, - /* HP jack connectors - unknown if we have jack deteck */ {"Headphone Jack", NULL, "HPO Pin"}, - /* other jacks */ {"MIC1", NULL, "Mic Jack"}, {"LINE1", NULL, "Line Jack"}, - /* digital mics */ {"DMIC1 Pin", NULL, "DMIC1"}, {"DMIC2 Pin", NULL, "DMIC2"}, @@ -84,10 +80,10 @@ static int codec_link_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_interval *channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_interval *rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); - /* The ADSP will covert the FE rate to 48k, stereo */ + /* The ADSP will convert the FE rate to 48kHz, stereo. */ rate->min = rate->max = 48000; channels->min = channels->max = 2; - /* set SSP0 to 16 bit */ + /* Set SSP0 to 16 bit. */ params_set_format(params, SNDRV_PCM_FORMAT_S16_LE); return 0; @@ -147,7 +143,6 @@ SND_SOC_DAILINK_DEF(platform, DAILINK_COMP_ARRAY(COMP_PLATFORM("haswell-pcm-audi SND_SOC_DAILINK_DEF(codec, DAILINK_COMP_ARRAY(COMP_CODEC("i2c-INT343A:00", "rt286-aif1"))); SND_SOC_DAILINK_DEF(ssp0_port, DAILINK_COMP_ARRAY(COMP_CPU("ssp0-port"))); -/* broadwell digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link card_dai_links[] = { /* Front End DAI links */ { @@ -240,14 +235,13 @@ static int bdw_rt286_resume(struct snd_soc_card *card) return 0; } -/* use space before codec name to simplify card ID, and simplify driver name */ +/* Use space before codec name to simplify card ID, and simplify driver name. */ #define SOF_CARD_NAME "bdw rt286" /* card name will be 'sof-bdw rt286' */ #define SOF_DRIVER_NAME "SOF" #define CARD_NAME "broadwell-rt286" #define DRIVER_NAME NULL /* card name will be used for driver name */ -/* broadwell audio machine driver for WPT + RT286S */ static struct snd_soc_card bdw_rt286_card = { .owner = THIS_MODULE, .dai_link = card_dai_links, @@ -269,14 +263,12 @@ static int bdw_rt286_probe(struct platform_device *pdev) int ret; bdw_rt286_card.dev = &pdev->dev; - /* override platform name, if required */ mach = pdev->dev.platform_data; ret = snd_soc_fixup_dai_links_platform_name(&bdw_rt286_card, mach->mach_params.platform); if (ret) return ret; - /* set card and driver name */ if (snd_soc_acpi_sof_parent(&pdev->dev)) { bdw_rt286_card.name = SOF_CARD_NAME; bdw_rt286_card.driver_name = SOF_DRIVER_NAME; @@ -308,8 +300,7 @@ static struct platform_driver bdw_rt286_driver = { module_platform_driver(bdw_rt286_driver) -/* Module information */ MODULE_AUTHOR("Liam Girdwood, Xingchao Wang"); -MODULE_DESCRIPTION("Intel SST Audio for WPT/Broadwell"); -MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("Sound card driver for Intel Broadwell Wildcat Point with Realtek 286"); +MODULE_LICENSE("GPL"); MODULE_ALIAS("platform:bdw_rt286"); -- cgit v1.2.3 From 9177203c209d9137dce52c7f0bc28e54960e5a41 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Mon, 20 Jun 2022 12:13:58 +0200 Subject: ASoC: Intel: bdw_rt286: Improve probe() function quality MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Declare local 'dev' and make use of it plus dev_get_platdata() to improve code readability. Relocate few relevant to the function macros for the exact same read too. Signed-off-by: Cezary Rojewski Reviewed-by: Amadeusz Sławiński Tested-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220620101402.2684366-14-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/bdw_rt286.c | 23 +++++++++++------------ 1 file changed, 11 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/bdw_rt286.c b/sound/soc/intel/boards/bdw_rt286.c index 9d815c31e1f4..c5737f548bef 100644 --- a/sound/soc/intel/boards/bdw_rt286.c +++ b/sound/soc/intel/boards/bdw_rt286.c @@ -235,13 +235,6 @@ static int bdw_rt286_resume(struct snd_soc_card *card) return 0; } -/* Use space before codec name to simplify card ID, and simplify driver name. */ -#define SOF_CARD_NAME "bdw rt286" /* card name will be 'sof-bdw rt286' */ -#define SOF_DRIVER_NAME "SOF" - -#define CARD_NAME "broadwell-rt286" -#define DRIVER_NAME NULL /* card name will be used for driver name */ - static struct snd_soc_card bdw_rt286_card = { .owner = THIS_MODULE, .dai_link = card_dai_links, @@ -257,27 +250,33 @@ static struct snd_soc_card bdw_rt286_card = { .resume_post = bdw_rt286_resume, }; +/* Use space before codec name to simplify card ID, and simplify driver name. */ +#define SOF_CARD_NAME "bdw rt286" /* card name will be 'sof-bdw rt286' */ +#define SOF_DRIVER_NAME "SOF" + +#define CARD_NAME "broadwell-rt286" + static int bdw_rt286_probe(struct platform_device *pdev) { struct snd_soc_acpi_mach *mach; + struct device *dev = &pdev->dev; int ret; - bdw_rt286_card.dev = &pdev->dev; - mach = pdev->dev.platform_data; + bdw_rt286_card.dev = dev; + mach = dev_get_platdata(dev); ret = snd_soc_fixup_dai_links_platform_name(&bdw_rt286_card, mach->mach_params.platform); if (ret) return ret; - if (snd_soc_acpi_sof_parent(&pdev->dev)) { + if (snd_soc_acpi_sof_parent(dev)) { bdw_rt286_card.name = SOF_CARD_NAME; bdw_rt286_card.driver_name = SOF_DRIVER_NAME; } else { bdw_rt286_card.name = CARD_NAME; - bdw_rt286_card.driver_name = DRIVER_NAME; } - return devm_snd_soc_register_card(&pdev->dev, &bdw_rt286_card); + return devm_snd_soc_register_card(dev, &bdw_rt286_card); } static int bdw_rt286_remove(struct platform_device *pdev) -- cgit v1.2.3 From 423cc2d0e8506a0ce6e3ef1806a561de1076e033 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Mon, 20 Jun 2022 12:13:59 +0200 Subject: ASoC: Intel: bdw_rt286: Improve hw_params() debug-ability MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Print status if setting sysclk fails. Signed-off-by: Cezary Rojewski Reviewed-by: Amadeusz Sławiński Tested-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220620101402.2684366-15-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/bdw_rt286.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/bdw_rt286.c b/sound/soc/intel/boards/bdw_rt286.c index c5737f548bef..8604b221b60d 100644 --- a/sound/soc/intel/boards/bdw_rt286.c +++ b/sound/soc/intel/boards/bdw_rt286.c @@ -98,7 +98,7 @@ static int codec_link_hw_params(struct snd_pcm_substream *substream, ret = snd_soc_dai_set_sysclk(codec_dai, RT286_SCLK_S_PLL, 24000000, SND_SOC_CLOCK_IN); if (ret < 0) { - dev_err(rtd->dev, "can't set codec sysclk configuration\n"); + dev_err(rtd->dev, "set codec sysclk failed: %d\n", ret); return ret; } -- cgit v1.2.3 From 8fe4709962d74a19c0c1dfc877ba600101340c62 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Mon, 20 Jun 2022 12:14:00 +0200 Subject: ASoC: Intel: bdw_rt286: Improve codec_init() quality MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Drop redundant 'ret' assignemnt, stop ignoring set_jack() return value and reword local 'component' variable to 'codec' to improve readability. Signed-off-by: Cezary Rojewski Reviewed-by: Amadeusz Sławiński Tested-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220620101402.2684366-16-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/bdw_rt286.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/bdw_rt286.c b/sound/soc/intel/boards/bdw_rt286.c index 8604b221b60d..36f984ff56c5 100644 --- a/sound/soc/intel/boards/bdw_rt286.c +++ b/sound/soc/intel/boards/bdw_rt286.c @@ -61,8 +61,8 @@ static const struct snd_soc_dapm_route card_routes[] = { static int codec_link_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; - int ret = 0; + struct snd_soc_component *codec = asoc_rtd_to_codec(rtd, 0)->component; + int ret; ret = snd_soc_card_jack_new_pins(rtd->card, "Headset", SND_JACK_HEADSET | SND_JACK_BTN_0, &card_headset, card_headset_pins, @@ -70,8 +70,7 @@ static int codec_link_init(struct snd_soc_pcm_runtime *rtd) if (ret) return ret; - snd_soc_component_set_jack(component, &card_headset, NULL); - return 0; + return snd_soc_component_set_jack(codec, &card_headset, NULL); } static int codec_link_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, -- cgit v1.2.3 From e7f68863545163ec75b6bc3cc48fe888c28e0ec6 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Mon, 20 Jun 2022 12:14:02 +0200 Subject: ASoC: Intel: bdw_rt286: Remove FE DAI ops MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit bdw_rt286_fe_ops is redundant as platform components already limit the number of channels available for the endpoint. Signed-off-by: Cezary Rojewski Reviewed-by: Amadeusz Sławiński Tested-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220620101402.2684366-18-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/bdw_rt286.c | 25 ------------------------- 1 file changed, 25 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/bdw_rt286.c b/sound/soc/intel/boards/bdw_rt286.c index 36f984ff56c5..47eaddb00936 100644 --- a/sound/soc/intel/boards/bdw_rt286.c +++ b/sound/soc/intel/boards/bdw_rt286.c @@ -108,30 +108,6 @@ static const struct snd_soc_ops codec_link_ops = { .hw_params = codec_link_hw_params, }; -static const unsigned int channels[] = { - 2, -}; - -static const struct snd_pcm_hw_constraint_list constraints_channels = { - .count = ARRAY_SIZE(channels), - .list = channels, - .mask = 0, -}; - -static int bdw_rt286_fe_startup(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - - /* Board supports stereo configuration only */ - runtime->hw.channels_max = 2; - return snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - &constraints_channels); -} - -static const struct snd_soc_ops bdw_rt286_fe_ops = { - .startup = bdw_rt286_fe_startup, -}; - SND_SOC_DAILINK_DEF(system, DAILINK_COMP_ARRAY(COMP_CPU("System Pin"))); SND_SOC_DAILINK_DEF(offload0, DAILINK_COMP_ARRAY(COMP_CPU("Offload0 Pin"))); SND_SOC_DAILINK_DEF(offload1, DAILINK_COMP_ARRAY(COMP_CPU("Offload1 Pin"))); @@ -150,7 +126,6 @@ static struct snd_soc_dai_link card_dai_links[] = { .nonatomic = 1, .dynamic = 1, .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .ops = &bdw_rt286_fe_ops, .dpcm_playback = 1, .dpcm_capture = 1, SND_SOC_DAILINK_REG(system, dummy, platform), -- cgit v1.2.3 From d5017d1323d45db14d1db3d348779264ffce9fb2 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Uwe=20Kleine-K=C3=B6nig?= Date: Wed, 22 Jun 2022 23:06:29 +0200 Subject: ASoC: topology: KUnit: Followup prototype change of snd_soc_unregister_card() MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit snd_soc_unregister_card() was recently converted to return void. Only the first instance was adapted, so convert the remaining ones now to fix building the topology test. Reported-by: kernel test robot Fixes: 1892a991886a ("ASoC: core: Make snd_soc_unregister_card() return void") Signed-off-by: Uwe Kleine-König Link: https://lore.kernel.org/r/20220622210629.286487-1-u.kleine-koenig@pengutronix.de Signed-off-by: Mark Brown --- sound/soc/soc-topology-test.c | 30 ++++++++++-------------------- 1 file changed, 10 insertions(+), 20 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-topology-test.c b/sound/soc/soc-topology-test.c index 225d74355974..51d650bb05b7 100644 --- a/sound/soc/soc-topology-test.c +++ b/sound/soc/soc-topology-test.c @@ -313,8 +313,7 @@ static void snd_soc_tplg_test_load_with_null_ops(struct kunit *test) KUNIT_EXPECT_EQ(test, 0, ret); /* cleanup */ - ret = snd_soc_unregister_card(&kunit_comp->card); - KUNIT_EXPECT_EQ(test, 0, ret); + snd_soc_unregister_card(&kunit_comp->card); snd_soc_unregister_component(test_dev); } @@ -377,8 +376,7 @@ static void snd_soc_tplg_test_load_with_null_fw(struct kunit *test) KUNIT_EXPECT_EQ(test, 0, ret); /* cleanup */ - ret = snd_soc_unregister_card(&kunit_comp->card); - KUNIT_EXPECT_EQ(test, 0, ret); + snd_soc_unregister_card(&kunit_comp->card); snd_soc_unregister_component(test_dev); } @@ -426,8 +424,7 @@ static void snd_soc_tplg_test_load_empty_tplg(struct kunit *test) KUNIT_EXPECT_EQ(test, 0, ret); /* cleanup */ - ret = snd_soc_unregister_card(&kunit_comp->card); - KUNIT_EXPECT_EQ(test, 0, ret); + snd_soc_unregister_card(&kunit_comp->card); snd_soc_unregister_component(test_dev); } @@ -482,8 +479,7 @@ static void snd_soc_tplg_test_load_empty_tplg_bad_magic(struct kunit *test) KUNIT_EXPECT_EQ(test, 0, ret); /* cleanup */ - ret = snd_soc_unregister_card(&kunit_comp->card); - KUNIT_EXPECT_EQ(test, 0, ret); + snd_soc_unregister_card(&kunit_comp->card); snd_soc_unregister_component(test_dev); } @@ -538,8 +534,7 @@ static void snd_soc_tplg_test_load_empty_tplg_bad_abi(struct kunit *test) KUNIT_EXPECT_EQ(test, 0, ret); /* cleanup */ - ret = snd_soc_unregister_card(&kunit_comp->card); - KUNIT_EXPECT_EQ(test, 0, ret); + snd_soc_unregister_card(&kunit_comp->card); snd_soc_unregister_component(test_dev); } @@ -594,8 +589,7 @@ static void snd_soc_tplg_test_load_empty_tplg_bad_size(struct kunit *test) KUNIT_EXPECT_EQ(test, 0, ret); /* cleanup */ - ret = snd_soc_unregister_card(&kunit_comp->card); - KUNIT_EXPECT_EQ(test, 0, ret); + snd_soc_unregister_card(&kunit_comp->card); snd_soc_unregister_component(test_dev); } @@ -653,8 +647,7 @@ static void snd_soc_tplg_test_load_empty_tplg_bad_payload_size(struct kunit *tes /* cleanup */ snd_soc_unregister_component(test_dev); - ret = snd_soc_unregister_card(&kunit_comp->card); - KUNIT_EXPECT_EQ(test, 0, ret); + snd_soc_unregister_card(&kunit_comp->card); } // TEST CASE @@ -702,8 +695,7 @@ static void snd_soc_tplg_test_load_pcm_tplg(struct kunit *test) snd_soc_unregister_component(test_dev); /* cleanup */ - ret = snd_soc_unregister_card(&kunit_comp->card); - KUNIT_EXPECT_EQ(test, 0, ret); + snd_soc_unregister_card(&kunit_comp->card); } // TEST CASE @@ -755,8 +747,7 @@ static void snd_soc_tplg_test_load_pcm_tplg_reload_comp(struct kunit *test) } /* cleanup */ - ret = snd_soc_unregister_card(&kunit_comp->card); - KUNIT_EXPECT_EQ(test, 0, ret); + snd_soc_unregister_card(&kunit_comp->card); } // TEST CASE @@ -804,8 +795,7 @@ static void snd_soc_tplg_test_load_pcm_tplg_reload_card(struct kunit *test) if (ret != 0 && ret != -EPROBE_DEFER) KUNIT_FAIL(test, "Failed to register card"); - ret = snd_soc_unregister_card(&kunit_comp->card); - KUNIT_EXPECT_EQ(test, 0, ret); + snd_soc_unregister_card(&kunit_comp->card); } /* cleanup */ -- cgit v1.2.3 From 4d6c2b46d81765e920007f76185a8d1fb5e41ca3 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 11:51:20 +0100 Subject: ASoC: dapm: Move stereo autodisable check Tidy up the code a little, rather than repeating the check of mc->autodisable move the stereo error check to be under the existing if for mc->autodisable. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623105120.1981154-6-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 869c76506b66..62c90e297aab 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -368,14 +368,14 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, case snd_soc_dapm_mixer_named_ctl: mc = (struct soc_mixer_control *)kcontrol->private_value; - if (mc->autodisable && snd_soc_volsw_is_stereo(mc)) - dev_warn(widget->dapm->dev, - "ASoC: Unsupported stereo autodisable control '%s'\n", - ctrl_name); - if (mc->autodisable) { struct snd_soc_dapm_widget template; + if (snd_soc_volsw_is_stereo(mc)) + dev_warn(widget->dapm->dev, + "ASoC: Unsupported stereo autodisable control '%s'\n", + ctrl_name); + name = kasprintf(GFP_KERNEL, "%s %s", ctrl_name, "Autodisable"); if (!name) { -- cgit v1.2.3 From 7f6409fd9b54b6f56444edc996cd28059f215415 Mon Sep 17 00:00:00 2001 From: Jiapeng Chong Date: Fri, 24 Jun 2022 16:27:45 +0800 Subject: ASoC: rockchip: i2s: Fix missing error code in rockchip_i2s_probe() The error code is missing in this code scenario, add the error code '-EINVAL' to the return value 'ret'. This was found by coccicheck: sound/soc/rockchip/rockchip_i2s.c:810 rockchip_i2s_probe() warn: missing error code 'ret'. Signed-off-by: Jiapeng Chong Link: https://lore.kernel.org/r/20220624082745.68367-1-jiapeng.chong@linux.alibaba.com Signed-off-by: Mark Brown --- sound/soc/rockchip/rockchip_i2s.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index d300eee9ddaa..0ed01624a2db 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -807,6 +807,7 @@ static int rockchip_i2s_probe(struct platform_device *pdev) i2s->bclk_off = pinctrl_lookup_state(i2s->pinctrl, "bclk_off"); if (IS_ERR_OR_NULL(i2s->bclk_off)) { dev_err(&pdev->dev, "failed to find i2s bclk_off\n"); + ret = -EINVAL; goto err_clk; } } -- cgit v1.2.3 From 658e95953075ca781ef8712d0a3203e485888c7f Mon Sep 17 00:00:00 2001 From: Cristian Ciocaltea Date: Wed, 22 Jun 2022 00:38:19 +0300 Subject: ASoC: cs35l41: Add support for CLSA3541 ACPI device ID Add support for the CLSA3541 ACPI device ID used on Valve's Steam Deck. The driver is fully compatible with the indicated hardware, hence no additional changes are required. Signed-off-by: Cristian Ciocaltea Acked-by: David Rhodes Link: https://lore.kernel.org/r/20220621213819.262537-1-cristian.ciocaltea@collabora.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs35l41-spi.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/cs35l41-spi.c b/sound/soc/codecs/cs35l41-spi.c index 9e19c946a66b..5c8bb24909eb 100644 --- a/sound/soc/codecs/cs35l41-spi.c +++ b/sound/soc/codecs/cs35l41-spi.c @@ -74,6 +74,7 @@ MODULE_DEVICE_TABLE(of, cs35l41_of_match); #ifdef CONFIG_ACPI static const struct acpi_device_id cs35l41_acpi_match[] = { { "CSC3541", 0 }, /* Cirrus Logic PnP ID + part ID */ + { "CLSA3541", 0 }, /* Cirrus Logic PnP ID + part ID */ {}, }; MODULE_DEVICE_TABLE(acpi, cs35l41_acpi_match); -- cgit v1.2.3 From bf2aebccddef890c4385d1ef19f9fee62d51bcc2 Mon Sep 17 00:00:00 2001 From: Francesco Dolcini Date: Fri, 24 Jun 2022 12:13:01 +0200 Subject: ASoC: sgtl5000: Fix noise on shutdown/remove Put the SGTL5000 in a silent/safe state on shutdown/remove, this is required since the SGTL5000 produces a constant noise on its output after it is configured and its clock is removed. Without this change this is happening every time the module is unbound/removed or from reboot till the clock is enabled again. The issue was experienced on both a Toradex Colibri/Apalis iMX6, but can be easily reproduced everywhere just playing something on the codec and after that removing/unbinding the driver. Fixes: 9b34e6cc3bc2 ("ASoC: Add Freescale SGTL5000 codec support") Signed-off-by: Francesco Dolcini Reviewed-by: Fabio Estevam Link: https://lore.kernel.org/r/20220624101301.441314-1-francesco.dolcini@toradex.com Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 9 +++++++++ sound/soc/codecs/sgtl5000.h | 1 + 2 files changed, 10 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 2aa48aef6a97..3363d1696ad7 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1795,6 +1795,9 @@ static int sgtl5000_i2c_remove(struct i2c_client *client) { struct sgtl5000_priv *sgtl5000 = i2c_get_clientdata(client); + regmap_write(sgtl5000->regmap, SGTL5000_CHIP_DIG_POWER, SGTL5000_DIG_POWER_DEFAULT); + regmap_write(sgtl5000->regmap, SGTL5000_CHIP_ANA_POWER, SGTL5000_ANA_POWER_DEFAULT); + clk_disable_unprepare(sgtl5000->mclk); regulator_bulk_disable(sgtl5000->num_supplies, sgtl5000->supplies); regulator_bulk_free(sgtl5000->num_supplies, sgtl5000->supplies); @@ -1802,6 +1805,11 @@ static int sgtl5000_i2c_remove(struct i2c_client *client) return 0; } +static void sgtl5000_i2c_shutdown(struct i2c_client *client) +{ + sgtl5000_i2c_remove(client); +} + static const struct i2c_device_id sgtl5000_id[] = { {"sgtl5000", 0}, {}, @@ -1822,6 +1830,7 @@ static struct i2c_driver sgtl5000_i2c_driver = { }, .probe_new = sgtl5000_i2c_probe, .remove = sgtl5000_i2c_remove, + .shutdown = sgtl5000_i2c_shutdown, .id_table = sgtl5000_id, }; diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h index 56ec5863f250..3a808c762299 100644 --- a/sound/soc/codecs/sgtl5000.h +++ b/sound/soc/codecs/sgtl5000.h @@ -80,6 +80,7 @@ /* * SGTL5000_CHIP_DIG_POWER */ +#define SGTL5000_DIG_POWER_DEFAULT 0x0000 #define SGTL5000_ADC_EN 0x0040 #define SGTL5000_DAC_EN 0x0020 #define SGTL5000_DAP_POWERUP 0x0010 -- cgit v1.2.3 From e112c42eb3b7225dd722493e9be8ce286c8a5af0 Mon Sep 17 00:00:00 2001 From: Sascha Hauer Date: Fri, 24 Jun 2022 11:26:01 +0200 Subject: ASoC: audio_graph_card2: Fix port numbers in example The example in audio-graph-card2.c has multiple nodes with the same name in it. Change the port numbers to get different names. Signed-off-by: Sascha Hauer Link: https://lore.kernel.org/r/20220624092601.2445224-1-s.hauer@pengutronix.de Signed-off-by: Mark Brown --- sound/soc/generic/audio-graph-card2.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/generic/audio-graph-card2.c b/sound/soc/generic/audio-graph-card2.c index 77ac4051b827..d34b29a49268 100644 --- a/sound/soc/generic/audio-graph-card2.c +++ b/sound/soc/generic/audio-graph-card2.c @@ -90,12 +90,12 @@ links indicates connection part of CPU side (= A). ports@0 { (X) (A) mcpu: port@0 { mcpu0_ep: endpoint { remote-endpoint = <&mcodec0_ep>; }; }; (y) port@1 { mcpu1_ep: endpoint { remote-endpoint = <&cpu1_ep>; }; }; -(y) port@1 { mcpu2_ep: endpoint { remote-endpoint = <&cpu2_ep>; }; }; +(y) port@2 { mcpu2_ep: endpoint { remote-endpoint = <&cpu2_ep>; }; }; }; ports@1 { (X) port@0 { mcodec0_ep: endpoint { remote-endpoint = <&mcpu0_ep>; }; }; -(y) port@0 { mcodec1_ep: endpoint { remote-endpoint = <&codec1_ep>; }; }; -(y) port@1 { mcodec2_ep: endpoint { remote-endpoint = <&codec2_ep>; }; }; +(y) port@1 { mcodec1_ep: endpoint { remote-endpoint = <&codec1_ep>; }; }; +(y) port@2 { mcodec2_ep: endpoint { remote-endpoint = <&codec2_ep>; }; }; }; }; }; -- cgit v1.2.3 From 703ee0557f8921c96e8c42f832b3bd69b7bfb262 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Fri, 24 Jun 2022 12:47:08 +0200 Subject: ASoC: max98396: add voltage regulators The device has up to 5 potentially independent power supplies: AVDD, DVDD, DVVDIO, VBAT and PVDD. The former 3 are mandatory for the device to function. One of VBAT and PVDD should also be made available. Regulators are enabled during probe time and will stay active except when in suspend mode. Futher, the chip needs to be informed about the presence of VBAT through a bit in register 0x20a0. Signed-off-by: Daniel Mack Link: https://lore.kernel.org/r/20220624104712.1934484-5-daniel@zonque.org Signed-off-by: Mark Brown --- sound/soc/codecs/max98396.c | 117 +++++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/max98396.h | 7 +++ 2 files changed, 123 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/max98396.c b/sound/soc/codecs/max98396.c index 56eb62bb041f..06ac637f2696 100644 --- a/sound/soc/codecs/max98396.c +++ b/sound/soc/codecs/max98396.c @@ -5,11 +5,18 @@ #include #include #include +#include #include #include #include #include "max98396.h" +static const char * const max98396_core_supplies[MAX98396_NUM_CORE_SUPPLIES] = { + "avdd", + "dvdd", + "dvddio", +}; + static struct reg_default max98396_reg[] = { {MAX98396_R2000_SW_RESET, 0x00}, {MAX98396_R2001_INT_RAW1, 0x00}, @@ -1329,6 +1336,12 @@ static int max98396_probe(struct snd_soc_component *component) regmap_write(max98396->regmap, MAX98397_R2057_PCM_RX_SRC2, 0x10); } + /* Supply control */ + regmap_update_bits(max98396->regmap, + MAX98396_R20A0_AMP_SUPPLY_CTL, + MAX98396_AMP_SUPPLY_NOVBAT, + (max98396->vbat == NULL) ? + MAX98396_AMP_SUPPLY_NOVBAT : 0); /* Enable DC blocker */ regmap_update_bits(max98396->regmap, MAX98396_R2092_AMP_DSP_CFG, 1, 1); @@ -1424,12 +1437,38 @@ static int max98396_suspend(struct device *dev) regcache_cache_only(max98396->regmap, true); regcache_mark_dirty(max98396->regmap); + regulator_bulk_disable(MAX98396_NUM_CORE_SUPPLIES, + max98396->core_supplies); + if (max98396->pvdd) + regulator_disable(max98396->pvdd); + + if (max98396->vbat) + regulator_disable(max98396->vbat); + return 0; } static int max98396_resume(struct device *dev) { struct max98396_priv *max98396 = dev_get_drvdata(dev); + int ret; + + ret = regulator_bulk_enable(MAX98396_NUM_CORE_SUPPLIES, + max98396->core_supplies); + if (ret < 0) + return ret; + + if (max98396->pvdd) { + ret = regulator_enable(max98396->pvdd); + if (ret < 0) + return ret; + } + + if (max98396->vbat) { + ret = regulator_enable(max98396->vbat); + if (ret < 0) + return ret; + } regcache_cache_only(max98396->regmap, false); max98396_reset(max98396, dev); @@ -1513,11 +1552,24 @@ static void max98396_read_device_property(struct device *dev, max98396->bypass_slot = 0; } +static void max98396_core_supplies_disable(void *priv) +{ + struct max98396_priv *max98396 = priv; + + regulator_bulk_disable(MAX98396_NUM_CORE_SUPPLIES, + max98396->core_supplies); +} + +static void max98396_supply_disable(void *r) +{ + regulator_disable((struct regulator *) r); +} + static int max98396_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct max98396_priv *max98396 = NULL; - int ret, reg; + int i, ret, reg; max98396 = devm_kzalloc(&i2c->dev, sizeof(*max98396), GFP_KERNEL); @@ -1543,6 +1595,69 @@ static int max98396_i2c_probe(struct i2c_client *i2c, return ret; } + /* Obtain regulator supplies */ + for (i = 0; i < MAX98396_NUM_CORE_SUPPLIES; i++) + max98396->core_supplies[i].supply = max98396_core_supplies[i]; + + ret = devm_regulator_bulk_get(&i2c->dev, MAX98396_NUM_CORE_SUPPLIES, + max98396->core_supplies); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to request core supplies: %d\n", ret); + return ret; + } + + max98396->vbat = devm_regulator_get_optional(&i2c->dev, "vbat"); + if (IS_ERR(max98396->vbat)) { + if (PTR_ERR(max98396->vbat) == -EPROBE_DEFER) + return -EPROBE_DEFER; + + max98396->vbat = NULL; + } + + max98396->pvdd = devm_regulator_get_optional(&i2c->dev, "pvdd"); + if (IS_ERR(max98396->pvdd)) { + if (PTR_ERR(max98396->pvdd) == -EPROBE_DEFER) + return -EPROBE_DEFER; + + max98396->pvdd = NULL; + } + + ret = regulator_bulk_enable(MAX98396_NUM_CORE_SUPPLIES, + max98396->core_supplies); + if (ret < 0) { + dev_err(&i2c->dev, "Unable to enable core supplies: %d", ret); + return ret; + } + + ret = devm_add_action_or_reset(&i2c->dev, max98396_core_supplies_disable, + max98396); + if (ret < 0) + return ret; + + if (max98396->pvdd) { + ret = regulator_enable(max98396->pvdd); + if (ret < 0) + return ret; + + ret = devm_add_action_or_reset(&i2c->dev, + max98396_supply_disable, + max98396->pvdd); + if (ret < 0) + return ret; + } + + if (max98396->vbat) { + ret = regulator_enable(max98396->vbat); + if (ret < 0) + return ret; + + ret = devm_add_action_or_reset(&i2c->dev, + max98396_supply_disable, + max98396->vbat); + if (ret < 0) + return ret; + } + /* update interleave mode info */ if (device_property_read_bool(&i2c->dev, "adi,interleave_mode")) max98396->interleave_mode = true; diff --git a/sound/soc/codecs/max98396.h b/sound/soc/codecs/max98396.h index 694411038597..8fa081f5d2d3 100644 --- a/sound/soc/codecs/max98396.h +++ b/sound/soc/codecs/max98396.h @@ -274,6 +274,9 @@ #define MAX98396_DSP_SPK_SAFE_EN_SHIFT (5) #define MAX98396_DSP_SPK_WB_FLT_EN_SHIFT (6) +/* MAX98396_R20A0_AMP_SUPPLY_CTL */ +#define MAX98396_AMP_SUPPLY_NOVBAT (0x1 << 0) + /* MAX98396_R20E0_IV_SENSE_PATH_CFG */ #define MAX98396_IV_SENSE_DCBLK_EN_MASK (0x3 << 0) #define MAX98396_IV_SENSE_DCBLK_EN_SHIFT (0) @@ -291,9 +294,13 @@ enum { CODEC_TYPE_MAX98397, }; +#define MAX98396_NUM_CORE_SUPPLIES 3 + struct max98396_priv { struct regmap *regmap; struct gpio_desc *reset_gpio; + struct regulator_bulk_data core_supplies[MAX98396_NUM_CORE_SUPPLIES]; + struct regulator *pvdd, *vbat; unsigned int v_slot; unsigned int i_slot; unsigned int bypass_slot; -- cgit v1.2.3 From a8c1dc9e8f01811b0c3fee65b9bc4773b2d00d96 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Fri, 24 Jun 2022 12:47:09 +0200 Subject: ASoC: max98396: Improve some error prints Let's log what actually failed and log at some more places. Signed-off-by: Daniel Mack Link: https://lore.kernel.org/r/20220624104712.1934484-6-daniel@zonque.org Signed-off-by: Mark Brown --- sound/soc/codecs/max98396.c | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/max98396.c b/sound/soc/codecs/max98396.c index 06ac637f2696..faa81b4bb709 100644 --- a/sound/soc/codecs/max98396.c +++ b/sound/soc/codecs/max98396.c @@ -372,7 +372,8 @@ static int max98396_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) break; default: - dev_err(component->dev, "DAI invert mode unsupported\n"); + dev_err(component->dev, "DAI invert mode %d unsupported\n", + fmt & SND_SOC_DAIFMT_INV_MASK); return -EINVAL; } @@ -391,6 +392,8 @@ static int max98396_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) format |= MAX98396_PCM_FORMAT_TDM_MODE0; break; default: + dev_err(component->dev, "DAI format %d unsupported\n", + fmt & SND_SOC_DAIFMT_FORMAT_MASK); return -EINVAL; } @@ -461,8 +464,9 @@ static int max98396_set_clock(struct snd_soc_component *component, /* BCLK configuration */ value = max98396_get_bclk_sel(blr_clk_ratio); if (!value) { - dev_err(component->dev, "format unsupported %d\n", - params_format(params)); + dev_err(component->dev, + "blr_clk_ratio %d unsupported, format %d\n", + blr_clk_ratio, params_format(params)); return -EINVAL; } @@ -647,7 +651,7 @@ static int max98396_dai_tdm_slot(struct snd_soc_dai *dai, chan_sz = MAX98396_PCM_MODE_CFG_CHANSZ_32; break; default: - dev_err(component->dev, "format unsupported %d\n", + dev_err(component->dev, "slot width %d unsupported\n", slot_width); return -EINVAL; } -- cgit v1.2.3 From c529fd620b842c926ccc8cea913886d802c30f16 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Fri, 24 Jun 2022 12:47:10 +0200 Subject: ASoC: max98396: Fix register access for PCM format settings max98396_dai_set_fmt() modifes register 2041 and touches bits in the mask 0x3a. Make sure to use the right mask for that operation. Signed-off-by: Daniel Mack Link: https://lore.kernel.org/r/20220624104712.1934484-7-daniel@zonque.org Signed-off-by: Mark Brown --- sound/soc/codecs/max98396.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/max98396.c b/sound/soc/codecs/max98396.c index faa81b4bb709..0a1d98279a3e 100644 --- a/sound/soc/codecs/max98396.c +++ b/sound/soc/codecs/max98396.c @@ -349,12 +349,15 @@ static int max98396_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_component *component = codec_dai->component; struct max98396_priv *max98396 = snd_soc_component_get_drvdata(component); - unsigned int format = 0; + unsigned int format_mask, format = 0; unsigned int bclk_pol = 0; int ret, status; int reg; bool update = false; + format_mask = MAX98396_PCM_MODE_CFG_FORMAT_MASK | + MAX98396_PCM_MODE_CFG_LRCLKEDGE; + dev_dbg(component->dev, "%s: fmt 0x%08X\n", __func__, fmt); switch (fmt & SND_SOC_DAIFMT_INV_MASK) { @@ -405,7 +408,7 @@ static int max98396_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) ret = regmap_read(max98396->regmap, MAX98396_R2041_PCM_MODE_CFG, ®); if (ret < 0) return -EINVAL; - if (format != (reg & MAX98396_PCM_BCLKEDGE_BSEL_MASK)) { + if (format != (reg & format_mask)) { update = true; } else { ret = regmap_read(max98396->regmap, @@ -422,8 +425,7 @@ static int max98396_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) regmap_update_bits(max98396->regmap, MAX98396_R2041_PCM_MODE_CFG, - MAX98396_PCM_BCLKEDGE_BSEL_MASK, - format); + format_mask, format); regmap_update_bits(max98396->regmap, MAX98396_R2042_PCM_CLK_SETUP, -- cgit v1.2.3 From f42924b49bf7935d21e2f2e98fdc9aa8dd176699 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Fri, 24 Jun 2022 12:47:11 +0200 Subject: ASoC: max98396: Implement DSP speaker monitor Allow the selection of the TDM slot that is used to send back speaker monitor data. The DT property adi,spkfb-slot-no can be used to configure this setting which defaults to 2. Signed-off-by: Daniel Mack Link: https://lore.kernel.org/r/20220624104712.1934484-8-daniel@zonque.org Signed-off-by: Mark Brown --- sound/soc/codecs/max98396.c | 8 ++++++++ sound/soc/codecs/max98396.h | 1 + 2 files changed, 9 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/max98396.c b/sound/soc/codecs/max98396.c index 0a1d98279a3e..f28831f4e74b 100644 --- a/sound/soc/codecs/max98396.c +++ b/sound/soc/codecs/max98396.c @@ -1377,6 +1377,9 @@ static int max98396_probe(struct snd_soc_component *component) regmap_write(max98396->regmap, MAX98396_R2045_PCM_TX_CTRL_2, max98396->i_slot); + regmap_write(max98396->regmap, + MAX98396_R204A_PCM_TX_CTRL_7, + max98396->spkfb_slot); if (max98396->v_slot < 8) if (max98396->device_id == CODEC_TYPE_MAX98396) @@ -1552,6 +1555,11 @@ static void max98396_read_device_property(struct device *dev, else max98396->i_slot = 1; + if (!device_property_read_u32(dev, "adi,spkfb-slot-no", &value)) + max98396->spkfb_slot = value & 0xF; + else + max98396->spkfb_slot = 2; + if (!device_property_read_u32(dev, "adi,bypass-slot-no", &value)) max98396->bypass_slot = value & 0xF; else diff --git a/sound/soc/codecs/max98396.h b/sound/soc/codecs/max98396.h index 8fa081f5d2d3..ff330ef61568 100644 --- a/sound/soc/codecs/max98396.h +++ b/sound/soc/codecs/max98396.h @@ -303,6 +303,7 @@ struct max98396_priv { struct regulator *pvdd, *vbat; unsigned int v_slot; unsigned int i_slot; + unsigned int spkfb_slot; unsigned int bypass_slot; bool interleave_mode; unsigned int ch_size; -- cgit v1.2.3 From 5f9d69986014945b826c712081678446c1f10fd7 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:51:18 +0100 Subject: ASoC: img: Migrate to new style legacy DAI naming flag Change the legacy DAI naming flag from opting in to the new scheme (non_legacy_dai_naming), to opting out of it (legacy_dai_naming). These drivers appear to be on the CPU side of the DAI link and currently uses the legacy naming, so add the new flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-5-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/img/img-i2s-in.c | 3 ++- sound/soc/img/img-i2s-out.c | 3 ++- sound/soc/img/img-parallel-out.c | 3 ++- sound/soc/img/img-spdif-in.c | 3 ++- sound/soc/img/img-spdif-out.c | 3 ++- 5 files changed, 10 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/img/img-i2s-in.c b/sound/soc/img/img-i2s-in.c index 97cab6d95b16..56bb7bbd3976 100644 --- a/sound/soc/img/img-i2s-in.c +++ b/sound/soc/img/img-i2s-in.c @@ -386,7 +386,8 @@ static int img_i2s_in_dai_probe(struct snd_soc_dai *dai) } static const struct snd_soc_component_driver img_i2s_in_component = { - .name = "img-i2s-in" + .name = "img-i2s-in", + .legacy_dai_naming = 1, }; static int img_i2s_in_dma_prepare_slave_config(struct snd_pcm_substream *st, diff --git a/sound/soc/img/img-i2s-out.c b/sound/soc/img/img-i2s-out.c index 9ec6fc528e2b..e3c6e662aa86 100644 --- a/sound/soc/img/img-i2s-out.c +++ b/sound/soc/img/img-i2s-out.c @@ -394,7 +394,8 @@ static int img_i2s_out_dai_probe(struct snd_soc_dai *dai) } static const struct snd_soc_component_driver img_i2s_out_component = { - .name = "img-i2s-out" + .name = "img-i2s-out", + .legacy_dai_naming = 1, }; static int img_i2s_out_dma_prepare_slave_config(struct snd_pcm_substream *st, diff --git a/sound/soc/img/img-parallel-out.c b/sound/soc/img/img-parallel-out.c index cd6a6a825741..08506b05e226 100644 --- a/sound/soc/img/img-parallel-out.c +++ b/sound/soc/img/img-parallel-out.c @@ -201,7 +201,8 @@ static struct snd_soc_dai_driver img_prl_out_dai = { }; static const struct snd_soc_component_driver img_prl_out_component = { - .name = "img-prl-out" + .name = "img-prl-out", + .legacy_dai_naming = 1, }; static int img_prl_out_probe(struct platform_device *pdev) diff --git a/sound/soc/img/img-spdif-in.c b/sound/soc/img/img-spdif-in.c index a79d1ccaeec0..3f1d1a7e8735 100644 --- a/sound/soc/img/img-spdif-in.c +++ b/sound/soc/img/img-spdif-in.c @@ -711,7 +711,8 @@ static struct snd_soc_dai_driver img_spdif_in_dai = { }; static const struct snd_soc_component_driver img_spdif_in_component = { - .name = "img-spdif-in" + .name = "img-spdif-in", + .legacy_dai_naming = 1, }; static int img_spdif_in_probe(struct platform_device *pdev) diff --git a/sound/soc/img/img-spdif-out.c b/sound/soc/img/img-spdif-out.c index f7062eba2611..983761d3fa7e 100644 --- a/sound/soc/img/img-spdif-out.c +++ b/sound/soc/img/img-spdif-out.c @@ -316,7 +316,8 @@ static struct snd_soc_dai_driver img_spdif_out_dai = { }; static const struct snd_soc_component_driver img_spdif_out_component = { - .name = "img-spdif-out" + .name = "img-spdif-out", + .legacy_dai_naming = 1, }; static int img_spdif_out_probe(struct platform_device *pdev) -- cgit v1.2.3 From eeb021ee8fab0baae82e3784664666fd6b826e89 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:51:19 +0100 Subject: ASoC: spear: Migrate to new style legacy DAI naming flag Change the legacy DAI naming flag from opting in to the new scheme (non_legacy_dai_naming), to opting out of it (legacy_dai_naming). These drivers appear to be on the CPU side of the DAI link and currently uses the legacy naming, so add the new flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-6-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/spear/spdif_in.c | 3 ++- sound/soc/spear/spdif_out.c | 3 ++- 2 files changed, 4 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/spear/spdif_in.c b/sound/soc/spear/spdif_in.c index 4b68d6ee75da..4ad8b1fc713a 100644 --- a/sound/soc/spear/spdif_in.c +++ b/sound/soc/spear/spdif_in.c @@ -172,7 +172,8 @@ static struct snd_soc_dai_driver spdif_in_dai = { }; static const struct snd_soc_component_driver spdif_in_component = { - .name = "spdif-in", + .name = "spdif-in", + .legacy_dai_naming = 1, }; static irqreturn_t spdif_in_irq(int irq, void *arg) diff --git a/sound/soc/spear/spdif_out.c b/sound/soc/spear/spdif_out.c index 549295a6ed50..fb107c5790ad 100644 --- a/sound/soc/spear/spdif_out.c +++ b/sound/soc/spear/spdif_out.c @@ -273,7 +273,8 @@ static struct snd_soc_dai_driver spdif_out_dai = { }; static const struct snd_soc_component_driver spdif_out_component = { - .name = "spdif-out", + .name = "spdif-out", + .legacy_dai_naming = 1, }; static int spdif_out_probe(struct platform_device *pdev) -- cgit v1.2.3 From 2bebc3b622c3c300eb3a3f603473429d8264c3b6 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:51:20 +0100 Subject: ASoC: jz4740-i2c: Migrate to new style legacy DAI naming flag Change the legacy DAI naming flag from opting in to the new scheme (non_legacy_dai_naming), to opting out of it (legacy_dai_naming). This driver appears to be on the CPU side of the DAI link and currently uses the legacy naming, so add the new flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-7-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/jz4740/jz4740-i2s.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index 446c5e061564..79afac0c5003 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -498,9 +498,10 @@ static const struct i2s_soc_info jz4780_i2s_soc_info = { }; static const struct snd_soc_component_driver jz4740_i2s_component = { - .name = "jz4740-i2s", - .suspend = jz4740_i2s_suspend, - .resume = jz4740_i2s_resume, + .name = "jz4740-i2s", + .suspend = jz4740_i2s_suspend, + .resume = jz4740_i2s_resume, + .legacy_dai_naming = 1, }; static const struct of_device_id jz4740_of_matches[] = { -- cgit v1.2.3 From fe58b58330434ffad5fa0bc97e177aa93a9a6222 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:51:21 +0100 Subject: ASoC: ep93xx: Migrate to new style legacy DAI naming flag Change the legacy DAI naming flag from opting in to the new scheme (non_legacy_dai_naming), to opting out of it (legacy_dai_naming). These drivers appear to be on the CPU side of the DAI link and currently uses the legacy naming, so add the new flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-8-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/cirrus/ep93xx-ac97.c | 3 ++- sound/soc/cirrus/ep93xx-i2s.c | 7 ++++--- 2 files changed, 6 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/cirrus/ep93xx-ac97.c b/sound/soc/cirrus/ep93xx-ac97.c index 16f9bb283b5c..37593abe6053 100644 --- a/sound/soc/cirrus/ep93xx-ac97.c +++ b/sound/soc/cirrus/ep93xx-ac97.c @@ -355,7 +355,8 @@ static struct snd_soc_dai_driver ep93xx_ac97_dai = { }; static const struct snd_soc_component_driver ep93xx_ac97_component = { - .name = "ep93xx-ac97", + .name = "ep93xx-ac97", + .legacy_dai_naming = 1, }; static int ep93xx_ac97_probe(struct platform_device *pdev) diff --git a/sound/soc/cirrus/ep93xx-i2s.c b/sound/soc/cirrus/ep93xx-i2s.c index 47959794353a..982151330c89 100644 --- a/sound/soc/cirrus/ep93xx-i2s.c +++ b/sound/soc/cirrus/ep93xx-i2s.c @@ -422,9 +422,10 @@ static struct snd_soc_dai_driver ep93xx_i2s_dai = { }; static const struct snd_soc_component_driver ep93xx_i2s_component = { - .name = "ep93xx-i2s", - .suspend = ep93xx_i2s_suspend, - .resume = ep93xx_i2s_resume, + .name = "ep93xx-i2s", + .suspend = ep93xx_i2s_suspend, + .resume = ep93xx_i2s_resume, + .legacy_dai_naming = 1, }; static int ep93xx_i2s_probe(struct platform_device *pdev) -- cgit v1.2.3 From 36f07985f81b7482dceb8e650d2ce1f0222294d1 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:51:22 +0100 Subject: ASoC: stm32: Migrate to new style legacy DAI naming flag Change the legacy DAI naming flag from opting in to the new scheme (non_legacy_dai_naming), to opting out of it (legacy_dai_naming). These drivers appear to be on the CPU side of the DAI link and currently uses the legacy naming, so add the new flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-9-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/stm/stm32_adfsdm.c | 1 + sound/soc/stm/stm32_i2s.c | 1 + sound/soc/stm/stm32_sai_sub.c | 1 + sound/soc/stm/stm32_spdifrx.c | 1 + 4 files changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/soc/stm/stm32_adfsdm.c b/sound/soc/stm/stm32_adfsdm.c index 122805160e70..04f2912e1418 100644 --- a/sound/soc/stm/stm32_adfsdm.c +++ b/sound/soc/stm/stm32_adfsdm.c @@ -149,6 +149,7 @@ static const struct snd_soc_dai_driver stm32_adfsdm_dai = { static const struct snd_soc_component_driver stm32_adfsdm_dai_component = { .name = "stm32_dfsdm_audio", + .legacy_dai_naming = 1, }; static void stm32_memcpy_32to16(void *dest, const void *src, size_t n) diff --git a/sound/soc/stm/stm32_i2s.c b/sound/soc/stm/stm32_i2s.c index 32d885f84a92..6aafe793eec4 100644 --- a/sound/soc/stm/stm32_i2s.c +++ b/sound/soc/stm/stm32_i2s.c @@ -978,6 +978,7 @@ static const struct snd_dmaengine_pcm_config stm32_i2s_pcm_config = { static const struct snd_soc_component_driver stm32_i2s_component = { .name = "stm32-i2s", + .legacy_dai_naming = 1, }; static void stm32_i2s_dai_init(struct snd_soc_pcm_stream *stream, diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 03cc6d12d18b..eb31b49e6597 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -1336,6 +1336,7 @@ static const struct snd_dmaengine_pcm_config stm32_sai_pcm_config_spdif = { static const struct snd_soc_component_driver stm32_component = { .name = "stm32-sai", + .legacy_dai_naming = 1, }; static const struct of_device_id stm32_sai_sub_ids[] = { diff --git a/sound/soc/stm/stm32_spdifrx.c b/sound/soc/stm/stm32_spdifrx.c index 6f7882c4fe6a..0f7146756717 100644 --- a/sound/soc/stm/stm32_spdifrx.c +++ b/sound/soc/stm/stm32_spdifrx.c @@ -888,6 +888,7 @@ static const struct snd_pcm_hardware stm32_spdifrx_pcm_hw = { static const struct snd_soc_component_driver stm32_spdifrx_component = { .name = "stm32-spdifrx", + .legacy_dai_naming = 1, }; static const struct snd_dmaengine_pcm_config stm32_spdifrx_pcm_config = { -- cgit v1.2.3 From b9a0db0ae5247d92f379107a9c479f881914999d Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:51:23 +0100 Subject: ASoC: bcm: Migrate to new style legacy DAI naming flag Change the legacy DAI naming flag from opting in to the new scheme (non_legacy_dai_naming), to opting out of it (legacy_dai_naming). These drivers appear to be on the CPU side of the DAI link and currently uses the legacy naming, so add the new flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-10-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/bcm/bcm2835-i2s.c | 3 ++- sound/soc/bcm/bcm63xx-i2s-whistler.c | 1 + sound/soc/bcm/cygnus-ssp.c | 7 ++++--- 3 files changed, 7 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/bcm/bcm2835-i2s.c b/sound/soc/bcm/bcm2835-i2s.c index e39c8d9f4099..f4d84774dac7 100644 --- a/sound/soc/bcm/bcm2835-i2s.c +++ b/sound/soc/bcm/bcm2835-i2s.c @@ -821,7 +821,8 @@ static const struct regmap_config bcm2835_regmap_config = { }; static const struct snd_soc_component_driver bcm2835_i2s_component = { - .name = "bcm2835-i2s-comp", + .name = "bcm2835-i2s-comp", + .legacy_dai_naming = 1, }; static int bcm2835_i2s_probe(struct platform_device *pdev) diff --git a/sound/soc/bcm/bcm63xx-i2s-whistler.c b/sound/soc/bcm/bcm63xx-i2s-whistler.c index 527caf430715..2da1384ffe91 100644 --- a/sound/soc/bcm/bcm63xx-i2s-whistler.c +++ b/sound/soc/bcm/bcm63xx-i2s-whistler.c @@ -218,6 +218,7 @@ static struct snd_soc_dai_driver bcm63xx_i2s_dai = { static const struct snd_soc_component_driver bcm63xx_i2s_component = { .name = "bcm63xx", + .legacy_dai_naming = 1, }; static int bcm63xx_i2s_dev_probe(struct platform_device *pdev) diff --git a/sound/soc/bcm/cygnus-ssp.c b/sound/soc/bcm/cygnus-ssp.c index 4bfa2d715ff4..8b7a21573070 100644 --- a/sound/soc/bcm/cygnus-ssp.c +++ b/sound/soc/bcm/cygnus-ssp.c @@ -1201,9 +1201,10 @@ static const struct snd_soc_dai_driver cygnus_spdif_dai_info = { static struct snd_soc_dai_driver cygnus_ssp_dai[CYGNUS_MAX_PORTS]; static const struct snd_soc_component_driver cygnus_ssp_component = { - .name = "cygnus-audio", - .suspend = cygnus_ssp_suspend, - .resume = cygnus_ssp_resume, + .name = "cygnus-audio", + .suspend = cygnus_ssp_suspend, + .resume = cygnus_ssp_resume, + .legacy_dai_naming = 1, }; /* -- cgit v1.2.3 From f712ff57a27090baff61f92bdb6521e8781d5e6b Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:51:24 +0100 Subject: ASoC: sh: Migrate to new style legacy DAI naming flag Change the legacy DAI naming flag from opting in to the new scheme (non_legacy_dai_naming), to opting out of it (legacy_dai_naming). These drivers appear to be on the CPU side of the DAI link and currently uses the legacy naming, so add the new flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-11-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/sh/hac.c | 3 ++- sound/soc/sh/rcar/core.c | 11 ++++++----- sound/soc/sh/rz-ssi.c | 9 +++++---- sound/soc/sh/siu_pcm.c | 17 +++++++++-------- sound/soc/sh/ssi.c | 3 ++- 5 files changed, 24 insertions(+), 19 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c index 475fc984f8c5..46d145cbaf29 100644 --- a/sound/soc/sh/hac.c +++ b/sound/soc/sh/hac.c @@ -307,7 +307,8 @@ static struct snd_soc_dai_driver sh4_hac_dai[] = { }; static const struct snd_soc_component_driver sh4_hac_component = { - .name = "sh4-hac", + .name = "sh4-hac", + .legacy_dai_naming = 1, }; static int hac_soc_platform_probe(struct platform_device *pdev) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index a4180dc5a59b..4973f94a2144 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1813,11 +1813,12 @@ int rsnd_kctrl_new(struct rsnd_mod *mod, * snd_soc_component */ static const struct snd_soc_component_driver rsnd_soc_component = { - .name = "rsnd", - .probe = rsnd_debugfs_probe, - .hw_params = rsnd_hw_params, - .hw_free = rsnd_hw_free, - .pointer = rsnd_pointer, + .name = "rsnd", + .probe = rsnd_debugfs_probe, + .hw_params = rsnd_hw_params, + .hw_free = rsnd_hw_free, + .pointer = rsnd_pointer, + .legacy_dai_naming = 1, }; static int rsnd_rdai_continuance_probe(struct rsnd_priv *priv, diff --git a/sound/soc/sh/rz-ssi.c b/sound/soc/sh/rz-ssi.c index beaf1a8d6da1..0d0594a0e4f6 100644 --- a/sound/soc/sh/rz-ssi.c +++ b/sound/soc/sh/rz-ssi.c @@ -906,10 +906,11 @@ static struct snd_soc_dai_driver rz_ssi_soc_dai[] = { }; static const struct snd_soc_component_driver rz_ssi_soc_component = { - .name = "rz-ssi", - .open = rz_ssi_pcm_open, - .pointer = rz_ssi_pcm_pointer, - .pcm_construct = rz_ssi_pcm_new, + .name = "rz-ssi", + .open = rz_ssi_pcm_open, + .pointer = rz_ssi_pcm_pointer, + .pcm_construct = rz_ssi_pcm_new, + .legacy_dai_naming = 1, }; static int rz_ssi_probe(struct platform_device *pdev) diff --git a/sound/soc/sh/siu_pcm.c b/sound/soc/sh/siu_pcm.c index 0a8a3c314a73..f15ff36e7934 100644 --- a/sound/soc/sh/siu_pcm.c +++ b/sound/soc/sh/siu_pcm.c @@ -540,13 +540,14 @@ static void siu_pcm_free(struct snd_soc_component *component, } const struct snd_soc_component_driver siu_component = { - .name = DRV_NAME, - .open = siu_pcm_open, - .close = siu_pcm_close, - .prepare = siu_pcm_prepare, - .trigger = siu_pcm_trigger, - .pointer = siu_pcm_pointer_dma, - .pcm_construct = siu_pcm_new, - .pcm_destruct = siu_pcm_free, + .name = DRV_NAME, + .open = siu_pcm_open, + .close = siu_pcm_close, + .prepare = siu_pcm_prepare, + .trigger = siu_pcm_trigger, + .pointer = siu_pcm_pointer_dma, + .pcm_construct = siu_pcm_new, + .pcm_destruct = siu_pcm_free, + .legacy_dai_naming = 1, }; EXPORT_SYMBOL_GPL(siu_component); diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c index bf7a3c69920a..96cf523c2273 100644 --- a/sound/soc/sh/ssi.c +++ b/sound/soc/sh/ssi.c @@ -377,7 +377,8 @@ static struct snd_soc_dai_driver sh4_ssi_dai[] = { }; static const struct snd_soc_component_driver sh4_ssi_component = { - .name = "sh4-ssi", + .name = "sh4-ssi", + .legacy_dai_naming = 1, }; static int sh4_soc_dai_probe(struct platform_device *pdev) -- cgit v1.2.3 From 3172582c10540d4bf1caac1c39c903793648db8f Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:51:25 +0100 Subject: ASoC: tegra: Migrate to new style legacy DAI naming flag Change the legacy DAI naming flag from opting in to the new scheme (non_legacy_dai_naming), to opting out of it (legacy_dai_naming). These drivers appear to be on the CPU side of the DAI link and currently uses the legacy naming, so add the new flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-12-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/tegra/tegra20_ac97.c | 3 ++- sound/soc/tegra/tegra20_i2s.c | 3 ++- sound/soc/tegra/tegra20_spdif.c | 1 + sound/soc/tegra/tegra30_i2s.c | 3 ++- 4 files changed, 7 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c index c454a34c15c4..e17375c6cddb 100644 --- a/sound/soc/tegra/tegra20_ac97.c +++ b/sound/soc/tegra/tegra20_ac97.c @@ -239,7 +239,8 @@ static struct snd_soc_dai_driver tegra20_ac97_dai = { }; static const struct snd_soc_component_driver tegra20_ac97_component = { - .name = DRV_NAME, + .name = DRV_NAME, + .legacy_dai_naming = 1, }; static bool tegra20_ac97_wr_rd_reg(struct device *dev, unsigned int reg) diff --git a/sound/soc/tegra/tegra20_i2s.c b/sound/soc/tegra/tegra20_i2s.c index 2e1a726602f0..fff0cd6588f5 100644 --- a/sound/soc/tegra/tegra20_i2s.c +++ b/sound/soc/tegra/tegra20_i2s.c @@ -338,7 +338,8 @@ static const struct snd_soc_dai_driver tegra20_i2s_dai_template = { }; static const struct snd_soc_component_driver tegra20_i2s_component = { - .name = DRV_NAME, + .name = DRV_NAME, + .legacy_dai_naming = 1, }; static bool tegra20_i2s_wr_rd_reg(struct device *dev, unsigned int reg) diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c index 64c2f304f254..ca7b222e07d0 100644 --- a/sound/soc/tegra/tegra20_spdif.c +++ b/sound/soc/tegra/tegra20_spdif.c @@ -264,6 +264,7 @@ static struct snd_soc_dai_driver tegra20_spdif_dai = { static const struct snd_soc_component_driver tegra20_spdif_component = { .name = "tegra20-spdif", + .legacy_dai_naming = 1, }; static bool tegra20_spdif_wr_rd_reg(struct device *dev, unsigned int reg) diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c index 3aa157c82ae2..10cd37096fb3 100644 --- a/sound/soc/tegra/tegra30_i2s.c +++ b/sound/soc/tegra/tegra30_i2s.c @@ -331,7 +331,8 @@ static const struct snd_soc_dai_driver tegra30_i2s_dai_template = { }; static const struct snd_soc_component_driver tegra30_i2s_component = { - .name = DRV_NAME, + .name = DRV_NAME, + .legacy_dai_naming = 1, }; static bool tegra30_i2s_wr_rd_reg(struct device *dev, unsigned int reg) -- cgit v1.2.3 From bf6dacb784f0efb5a225f6560d693fa71c7fda64 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:51:26 +0100 Subject: ASoC: hisilicon: Migrate to new style legacy DAI naming flag Change the legacy DAI naming flag from opting in to the new scheme (non_legacy_dai_naming), to opting out of it (legacy_dai_naming). This driver appears to be on the CPU side of the DAI link and currently uses the legacy naming, so add the new flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-13-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/hisilicon/hi6210-i2s.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/hisilicon/hi6210-i2s.c b/sound/soc/hisilicon/hi6210-i2s.c index 689ae13f34f5..27219a9e7d0d 100644 --- a/sound/soc/hisilicon/hi6210-i2s.c +++ b/sound/soc/hisilicon/hi6210-i2s.c @@ -539,6 +539,7 @@ static const struct snd_soc_dai_driver hi6210_i2s_dai_init = { static const struct snd_soc_component_driver hi6210_i2s_i2s_comp = { .name = "hi6210_i2s-i2s", + .legacy_dai_naming = 1, }; static int hi6210_i2s_probe(struct platform_device *pdev) -- cgit v1.2.3 From bd486b070b1e24b38b3d6d7e33abffe4a18e3296 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:51:27 +0100 Subject: ASoC: xilinx: Migrate to new style legacy DAI naming flag Change the legacy DAI naming flag from opting in to the new scheme (non_legacy_dai_naming), to opting out of it (legacy_dai_naming). These drivers appear to be on the CPU side of the DAI link and currently uses the legacy naming, so add the new flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-14-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/xilinx/xlnx_formatter_pcm.c | 16 ++++++++-------- sound/soc/xilinx/xlnx_i2s.c | 1 + sound/soc/xilinx/xlnx_spdif.c | 1 + 3 files changed, 10 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/xilinx/xlnx_formatter_pcm.c b/sound/soc/xilinx/xlnx_formatter_pcm.c index 5c4158069a5a..f5ac0aa312d6 100644 --- a/sound/soc/xilinx/xlnx_formatter_pcm.c +++ b/sound/soc/xilinx/xlnx_formatter_pcm.c @@ -575,14 +575,14 @@ static int xlnx_formatter_pcm_new(struct snd_soc_component *component, } static const struct snd_soc_component_driver xlnx_asoc_component = { - .name = DRV_NAME, - .set_sysclk = xlnx_formatter_set_sysclk, - .open = xlnx_formatter_pcm_open, - .close = xlnx_formatter_pcm_close, - .hw_params = xlnx_formatter_pcm_hw_params, - .trigger = xlnx_formatter_pcm_trigger, - .pointer = xlnx_formatter_pcm_pointer, - .pcm_construct = xlnx_formatter_pcm_new, + .name = DRV_NAME, + .set_sysclk = xlnx_formatter_set_sysclk, + .open = xlnx_formatter_pcm_open, + .close = xlnx_formatter_pcm_close, + .hw_params = xlnx_formatter_pcm_hw_params, + .trigger = xlnx_formatter_pcm_trigger, + .pointer = xlnx_formatter_pcm_pointer, + .pcm_construct = xlnx_formatter_pcm_new, }; static int xlnx_formatter_pcm_probe(struct platform_device *pdev) diff --git a/sound/soc/xilinx/xlnx_i2s.c b/sound/soc/xilinx/xlnx_i2s.c index 4cc6ee7c81a3..9de92d35e30e 100644 --- a/sound/soc/xilinx/xlnx_i2s.c +++ b/sound/soc/xilinx/xlnx_i2s.c @@ -158,6 +158,7 @@ static const struct snd_soc_dai_ops xlnx_i2s_dai_ops = { static const struct snd_soc_component_driver xlnx_i2s_component = { .name = DRV_NAME, + .legacy_dai_naming = 1, }; static const struct of_device_id xlnx_i2s_of_match[] = { diff --git a/sound/soc/xilinx/xlnx_spdif.c b/sound/soc/xilinx/xlnx_spdif.c index cba0e868a7d7..7342048e9875 100644 --- a/sound/soc/xilinx/xlnx_spdif.c +++ b/sound/soc/xilinx/xlnx_spdif.c @@ -226,6 +226,7 @@ static struct snd_soc_dai_driver xlnx_spdif_rx_dai = { static const struct snd_soc_component_driver xlnx_spdif_component = { .name = "xlnx-spdif", + .legacy_dai_naming = 1, }; static const struct of_device_id xlnx_spdif_of_match[] = { -- cgit v1.2.3 From f450b5dbce413b276e6c9215b40868b905c7b634 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:51:28 +0100 Subject: ASoC: sunxi: Migrate to new style legacy DAI naming flag Change the legacy DAI naming flag from opting in to the new scheme (non_legacy_dai_naming), to opting out of it (legacy_dai_naming). These drivers appear to be on the CPU side of the DAI link and currently uses the legacy naming, so add the new flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-15-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/sunxi/sun4i-codec.c | 3 ++- sound/soc/sunxi/sun4i-i2s.c | 3 ++- sound/soc/sunxi/sun4i-spdif.c | 3 ++- 3 files changed, 6 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index 53e3f43816cc..bc634962a57e 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -1234,7 +1234,8 @@ static const struct snd_soc_component_driver sun8i_a23_codec_codec = { }; static const struct snd_soc_component_driver sun4i_codec_component = { - .name = "sun4i-codec", + .name = "sun4i-codec", + .legacy_dai_naming = 1, }; #define SUN4I_CODEC_RATES SNDRV_PCM_RATE_CONTINUOUS diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index 5be33d07361b..6028871825ba 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -1125,7 +1125,8 @@ static struct snd_soc_dai_driver sun4i_i2s_dai = { }; static const struct snd_soc_component_driver sun4i_i2s_component = { - .name = "sun4i-dai", + .name = "sun4i-dai", + .legacy_dai_naming = 1, }; static bool sun4i_i2s_rd_reg(struct device *dev, unsigned int reg) diff --git a/sound/soc/sunxi/sun4i-spdif.c b/sound/soc/sunxi/sun4i-spdif.c index 17090f43150e..bcceebca915a 100644 --- a/sound/soc/sunxi/sun4i-spdif.c +++ b/sound/soc/sunxi/sun4i-spdif.c @@ -583,7 +583,8 @@ static const struct of_device_id sun4i_spdif_of_match[] = { MODULE_DEVICE_TABLE(of, sun4i_spdif_of_match); static const struct snd_soc_component_driver sun4i_spdif_component = { - .name = "sun4i-spdif", + .name = "sun4i-spdif", + .legacy_dai_naming = 1, }; static int sun4i_spdif_runtime_suspend(struct device *dev) -- cgit v1.2.3 From 725cf3bc6009b7fa156b73982eddf23c71767fbb Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:51:29 +0100 Subject: ASoC: Intel: Migrate to new style legacy DAI naming flag Change the legacy DAI naming flag from opting in to the new scheme (non_legacy_dai_naming), to opting out of it (legacy_dai_naming). These drivers appear to be on the CPU side of the DAI link and currently uses the legacy naming, so add the new flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-16-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/intel/keembay/kmb_platform.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/keembay/kmb_platform.c b/sound/soc/intel/keembay/kmb_platform.c index d10881fedc8b..b4893365d01d 100644 --- a/sound/soc/intel/keembay/kmb_platform.c +++ b/sound/soc/intel/keembay/kmb_platform.c @@ -388,15 +388,17 @@ static snd_pcm_uframes_t kmb_pcm_pointer(struct snd_soc_component *component, } static const struct snd_soc_component_driver kmb_component = { - .name = "kmb", - .pcm_construct = kmb_platform_pcm_new, - .open = kmb_pcm_open, - .trigger = kmb_pcm_trigger, - .pointer = kmb_pcm_pointer, + .name = "kmb", + .pcm_construct = kmb_platform_pcm_new, + .open = kmb_pcm_open, + .trigger = kmb_pcm_trigger, + .pointer = kmb_pcm_pointer, + .legacy_dai_naming = 1, }; static const struct snd_soc_component_driver kmb_component_dma = { - .name = "kmb", + .name = "kmb", + .legacy_dai_naming = 1, }; static int kmb_probe(struct snd_soc_dai *cpu_dai) -- cgit v1.2.3 From d8572da099247860e97b27a7fddc9d80a71b8c25 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:51:30 +0100 Subject: ASoC: meson: Migrate to new style legacy DAI naming flag Change the legacy DAI naming flag from opting in to the new scheme (non_legacy_dai_naming), to opting out of it (legacy_dai_naming). These drivers appear to be on the CPU side of the DAI link and currently uses the legacy naming, so add the new flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-17-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/meson/axg-frddr.c | 3 +++ sound/soc/meson/axg-pdm.c | 4 +++- sound/soc/meson/axg-spdifin.c | 1 + sound/soc/meson/axg-spdifout.c | 1 + sound/soc/meson/axg-toddr.c | 3 +++ 5 files changed, 11 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/meson/axg-frddr.c b/sound/soc/meson/axg-frddr.c index 37f4bb3469b5..61f9d417fd60 100644 --- a/sound/soc/meson/axg-frddr.c +++ b/sound/soc/meson/axg-frddr.c @@ -161,6 +161,7 @@ static const struct snd_soc_component_driver axg_frddr_component_drv = { .hw_free = axg_fifo_pcm_hw_free, .pointer = axg_fifo_pcm_pointer, .trigger = axg_fifo_pcm_trigger, + .legacy_dai_naming = 1, }; static const struct axg_fifo_match_data axg_frddr_match_data = { @@ -286,6 +287,7 @@ static const struct snd_soc_component_driver g12a_frddr_component_drv = { .hw_free = axg_fifo_pcm_hw_free, .pointer = axg_fifo_pcm_pointer, .trigger = axg_fifo_pcm_trigger, + .legacy_dai_naming = 1, }; static const struct axg_fifo_match_data g12a_frddr_match_data = { @@ -356,6 +358,7 @@ static const struct snd_soc_component_driver sm1_frddr_component_drv = { .hw_free = axg_fifo_pcm_hw_free, .pointer = axg_fifo_pcm_pointer, .trigger = axg_fifo_pcm_trigger, + .legacy_dai_naming = 1, }; static const struct axg_fifo_match_data sm1_frddr_match_data = { diff --git a/sound/soc/meson/axg-pdm.c b/sound/soc/meson/axg-pdm.c index 672e43a9729d..88ac58272f95 100644 --- a/sound/soc/meson/axg-pdm.c +++ b/sound/soc/meson/axg-pdm.c @@ -457,7 +457,9 @@ static struct snd_soc_dai_driver axg_pdm_dai_drv = { .remove = axg_pdm_dai_remove, }; -static const struct snd_soc_component_driver axg_pdm_component_drv = {}; +static const struct snd_soc_component_driver axg_pdm_component_drv = { + .legacy_dai_naming = 1, +}; static const struct regmap_config axg_pdm_regmap_cfg = { .reg_bits = 32, diff --git a/sound/soc/meson/axg-spdifin.c b/sound/soc/meson/axg-spdifin.c index 4ba44e0d65d9..e2cc4c4be758 100644 --- a/sound/soc/meson/axg-spdifin.c +++ b/sound/soc/meson/axg-spdifin.c @@ -390,6 +390,7 @@ static const struct snd_kcontrol_new axg_spdifin_controls[] = { static const struct snd_soc_component_driver axg_spdifin_component_drv = { .controls = axg_spdifin_controls, .num_controls = ARRAY_SIZE(axg_spdifin_controls), + .legacy_dai_naming = 1, }; static const struct regmap_config axg_spdifin_regmap_cfg = { diff --git a/sound/soc/meson/axg-spdifout.c b/sound/soc/meson/axg-spdifout.c index 3960d082e143..e8a12f15f3b4 100644 --- a/sound/soc/meson/axg-spdifout.c +++ b/sound/soc/meson/axg-spdifout.c @@ -383,6 +383,7 @@ static const struct snd_soc_component_driver axg_spdifout_component_drv = { .dapm_routes = axg_spdifout_dapm_routes, .num_dapm_routes = ARRAY_SIZE(axg_spdifout_dapm_routes), .set_bias_level = axg_spdifout_set_bias_level, + .legacy_dai_naming = 1, }; static const struct regmap_config axg_spdifout_regmap_cfg = { diff --git a/sound/soc/meson/axg-toddr.c b/sound/soc/meson/axg-toddr.c index d6adf7edea41..e9208e74e965 100644 --- a/sound/soc/meson/axg-toddr.c +++ b/sound/soc/meson/axg-toddr.c @@ -182,6 +182,7 @@ static const struct snd_soc_component_driver axg_toddr_component_drv = { .hw_free = axg_fifo_pcm_hw_free, .pointer = axg_fifo_pcm_pointer, .trigger = axg_fifo_pcm_trigger, + .legacy_dai_naming = 1, }; static const struct axg_fifo_match_data axg_toddr_match_data = { @@ -242,6 +243,7 @@ static const struct snd_soc_component_driver g12a_toddr_component_drv = { .hw_free = axg_fifo_pcm_hw_free, .pointer = axg_fifo_pcm_pointer, .trigger = axg_fifo_pcm_trigger, + .legacy_dai_naming = 1, }; static const struct axg_fifo_match_data g12a_toddr_match_data = { @@ -312,6 +314,7 @@ static const struct snd_soc_component_driver sm1_toddr_component_drv = { .hw_free = axg_fifo_pcm_hw_free, .pointer = axg_fifo_pcm_pointer, .trigger = axg_fifo_pcm_trigger, + .legacy_dai_naming = 1, }; static const struct axg_fifo_match_data sm1_toddr_match_data = { -- cgit v1.2.3 From ad483da7b0a17fdf4df0bd75b2cf29b5650ca2f7 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:51:31 +0100 Subject: ASoC: sti-uniperf: Migrate to new style legacy DAI naming flag Change the legacy DAI naming flag from opting in to the new scheme (non_legacy_dai_naming), to opting out of it (legacy_dai_naming). This driver appears to be on the CPU side of the DAI link and currently uses the legacy naming, so add the new flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-18-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/sti/sti_uniperif.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sti/sti_uniperif.c b/sound/soc/sti/sti_uniperif.c index 34668fe3909d..a4d74d1e3c24 100644 --- a/sound/soc/sti/sti_uniperif.c +++ b/sound/soc/sti/sti_uniperif.c @@ -376,7 +376,8 @@ static const struct snd_soc_dai_driver sti_uniperiph_dai_template = { static const struct snd_soc_component_driver sti_uniperiph_dai_component = { .name = "sti_cpu_dai", .suspend = sti_uniperiph_suspend, - .resume = sti_uniperiph_resume + .resume = sti_uniperiph_resume, + .legacy_dai_naming = 1, }; static int sti_uniperiph_cpu_dai_of(struct device_node *node, -- cgit v1.2.3 From 0bc1e7d1fc3c50cf1eb62cd3c8d2b73c5f6d83fe Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:51:32 +0100 Subject: ASoC: amd: Migrate to new style legacy DAI naming flag Change the legacy DAI naming flag from opting in to the new scheme (non_legacy_dai_naming), to opting out of it (legacy_dai_naming). These drivers appear to be on the CPU side of the DAI link and currently uses the legacy naming, so add the new flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-19-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp-platform.c | 15 ++++++++------- sound/soc/amd/raven/acp3x-i2s.c | 3 ++- sound/soc/amd/renoir/acp3x-pdm-dma.c | 13 +++++++------ sound/soc/amd/vangogh/acp5x-i2s.c | 1 + sound/soc/amd/yc/acp6x-pdm-dma.c | 13 +++++++------ 5 files changed, 25 insertions(+), 20 deletions(-) (limited to 'sound') diff --git a/sound/soc/amd/acp/acp-platform.c b/sound/soc/amd/acp/acp-platform.c index 65a809e2c29f..3c4fd8b80589 100644 --- a/sound/soc/amd/acp/acp-platform.c +++ b/sound/soc/amd/acp/acp-platform.c @@ -267,13 +267,14 @@ static int acp_dma_close(struct snd_soc_component *component, } static const struct snd_soc_component_driver acp_pcm_component = { - .name = DRV_NAME, - .open = acp_dma_open, - .close = acp_dma_close, - .hw_params = acp_dma_hw_params, - .pointer = acp_dma_pointer, - .mmap = acp_dma_mmap, - .pcm_construct = acp_dma_new, + .name = DRV_NAME, + .open = acp_dma_open, + .close = acp_dma_close, + .hw_params = acp_dma_hw_params, + .pointer = acp_dma_pointer, + .mmap = acp_dma_mmap, + .pcm_construct = acp_dma_new, + .legacy_dai_naming = 1, }; int acp_platform_register(struct device *dev) diff --git a/sound/soc/amd/raven/acp3x-i2s.c b/sound/soc/amd/raven/acp3x-i2s.c index de6f70d7ef36..aa38cef1776d 100644 --- a/sound/soc/amd/raven/acp3x-i2s.c +++ b/sound/soc/amd/raven/acp3x-i2s.c @@ -257,7 +257,8 @@ static const struct snd_soc_dai_ops acp3x_i2s_dai_ops = { }; static const struct snd_soc_component_driver acp3x_dai_component = { - .name = DRV_NAME, + .name = DRV_NAME, + .legacy_dai_naming = 1, }; static struct snd_soc_dai_driver acp3x_i2s_dai = { diff --git a/sound/soc/amd/renoir/acp3x-pdm-dma.c b/sound/soc/amd/renoir/acp3x-pdm-dma.c index 8c42345ee41e..7203c6488df0 100644 --- a/sound/soc/amd/renoir/acp3x-pdm-dma.c +++ b/sound/soc/amd/renoir/acp3x-pdm-dma.c @@ -363,12 +363,13 @@ static struct snd_soc_dai_driver acp_pdm_dai_driver = { }; static const struct snd_soc_component_driver acp_pdm_component = { - .name = DRV_NAME, - .open = acp_pdm_dma_open, - .close = acp_pdm_dma_close, - .hw_params = acp_pdm_dma_hw_params, - .pointer = acp_pdm_dma_pointer, - .pcm_construct = acp_pdm_dma_new, + .name = DRV_NAME, + .open = acp_pdm_dma_open, + .close = acp_pdm_dma_close, + .hw_params = acp_pdm_dma_hw_params, + .pointer = acp_pdm_dma_pointer, + .pcm_construct = acp_pdm_dma_new, + .legacy_dai_naming = 1, }; static int acp_pdm_audio_probe(struct platform_device *pdev) diff --git a/sound/soc/amd/vangogh/acp5x-i2s.c b/sound/soc/amd/vangogh/acp5x-i2s.c index 72c8c68e5933..773e96f1b4dd 100644 --- a/sound/soc/amd/vangogh/acp5x-i2s.c +++ b/sound/soc/amd/vangogh/acp5x-i2s.c @@ -345,6 +345,7 @@ static const struct snd_soc_dai_ops acp5x_i2s_dai_ops = { static const struct snd_soc_component_driver acp5x_dai_component = { .name = "acp5x-i2s", + .legacy_dai_naming = 1, }; static struct snd_soc_dai_driver acp5x_i2s_dai = { diff --git a/sound/soc/amd/yc/acp6x-pdm-dma.c b/sound/soc/amd/yc/acp6x-pdm-dma.c index 7e66393e4153..acecd6a4ec4b 100644 --- a/sound/soc/amd/yc/acp6x-pdm-dma.c +++ b/sound/soc/amd/yc/acp6x-pdm-dma.c @@ -335,12 +335,13 @@ static struct snd_soc_dai_driver acp6x_pdm_dai_driver = { }; static const struct snd_soc_component_driver acp6x_pdm_component = { - .name = DRV_NAME, - .open = acp6x_pdm_dma_open, - .close = acp6x_pdm_dma_close, - .hw_params = acp6x_pdm_dma_hw_params, - .pointer = acp6x_pdm_dma_pointer, - .pcm_construct = acp6x_pdm_dma_new, + .name = DRV_NAME, + .open = acp6x_pdm_dma_open, + .close = acp6x_pdm_dma_close, + .hw_params = acp6x_pdm_dma_hw_params, + .pointer = acp6x_pdm_dma_pointer, + .pcm_construct = acp6x_pdm_dma_new, + .legacy_dai_naming = 1, }; static int acp6x_pdm_audio_probe(struct platform_device *pdev) -- cgit v1.2.3 From 7593e00807fb62e9f5e7367fc2500428cc317ff0 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:51:33 +0100 Subject: ASoC: atmel: Migrate to new style legacy DAI naming flag Change the legacy DAI naming flag from opting in to the new scheme (non_legacy_dai_naming), to opting out of it (legacy_dai_naming). These drivers appear to be on the CPU side of the DAI link and currently uses the legacy naming, so add the new flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-20-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/atmel/atmel-classd.c | 1 + sound/soc/atmel/atmel-i2s.c | 3 ++- sound/soc/atmel/atmel-pdmic.c | 1 + sound/soc/atmel/atmel_ssc_dai.c | 7 ++++--- sound/soc/atmel/mchp-i2s-mcc.c | 3 ++- sound/soc/atmel/mchp-pdmc.c | 1 + sound/soc/atmel/mchp-spdifrx.c | 3 ++- sound/soc/atmel/mchp-spdiftx.c | 3 ++- 8 files changed, 15 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/atmel/atmel-classd.c b/sound/soc/atmel/atmel-classd.c index 74b7b2611aa7..87d6d6ed026b 100644 --- a/sound/soc/atmel/atmel-classd.c +++ b/sound/soc/atmel/atmel-classd.c @@ -458,6 +458,7 @@ static const struct snd_soc_component_driver atmel_classd_cpu_dai_component = { .num_controls = ARRAY_SIZE(atmel_classd_snd_controls), .idle_bias_on = 1, .use_pmdown_time = 1, + .legacy_dai_naming = 1, }; /* ASoC sound card */ diff --git a/sound/soc/atmel/atmel-i2s.c b/sound/soc/atmel/atmel-i2s.c index ba56d6ac7e57..425d66edbf86 100644 --- a/sound/soc/atmel/atmel-i2s.c +++ b/sound/soc/atmel/atmel-i2s.c @@ -564,7 +564,8 @@ static struct snd_soc_dai_driver atmel_i2s_dai = { }; static const struct snd_soc_component_driver atmel_i2s_component = { - .name = "atmel-i2s", + .name = "atmel-i2s", + .legacy_dai_naming = 1, }; static int atmel_i2s_sama5d2_mck_init(struct atmel_i2s_dev *dev, diff --git a/sound/soc/atmel/atmel-pdmic.c b/sound/soc/atmel/atmel-pdmic.c index ea34efac2fff..77ff12baead5 100644 --- a/sound/soc/atmel/atmel-pdmic.c +++ b/sound/soc/atmel/atmel-pdmic.c @@ -481,6 +481,7 @@ static const struct snd_soc_component_driver atmel_pdmic_cpu_dai_component = { .num_controls = ARRAY_SIZE(atmel_pdmic_snd_controls), .idle_bias_on = 1, .use_pmdown_time = 1, + .legacy_dai_naming = 1, }; /* ASoC sound card */ diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index c92905e343e7..8aae0beadcfe 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -858,9 +858,10 @@ static struct snd_soc_dai_driver atmel_ssc_dai = { }; static const struct snd_soc_component_driver atmel_ssc_component = { - .name = "atmel-ssc", - .suspend = atmel_ssc_suspend, - .resume = atmel_ssc_resume, + .name = "atmel-ssc", + .suspend = atmel_ssc_suspend, + .resume = atmel_ssc_resume, + .legacy_dai_naming = 1, }; static int asoc_ssc_init(struct device *dev) diff --git a/sound/soc/atmel/mchp-i2s-mcc.c b/sound/soc/atmel/mchp-i2s-mcc.c index 269eab56b6dd..6dfb96c576ff 100644 --- a/sound/soc/atmel/mchp-i2s-mcc.c +++ b/sound/soc/atmel/mchp-i2s-mcc.c @@ -928,7 +928,8 @@ static struct snd_soc_dai_driver mchp_i2s_mcc_dai = { }; static const struct snd_soc_component_driver mchp_i2s_mcc_component = { - .name = "mchp-i2s-mcc", + .name = "mchp-i2s-mcc", + .legacy_dai_naming = 1, }; #ifdef CONFIG_OF diff --git a/sound/soc/atmel/mchp-pdmc.c b/sound/soc/atmel/mchp-pdmc.c index b9f637059448..aba7c5cde62c 100644 --- a/sound/soc/atmel/mchp-pdmc.c +++ b/sound/soc/atmel/mchp-pdmc.c @@ -423,6 +423,7 @@ static const struct snd_soc_component_driver mchp_pdmc_dai_component = { .num_controls = ARRAY_SIZE(mchp_pdmc_snd_controls), .open = &mchp_pdmc_open, .close = &mchp_pdmc_close, + .legacy_dai_naming = 1, }; static const unsigned int mchp_pdmc_1mic[] = {1}; diff --git a/sound/soc/atmel/mchp-spdifrx.c b/sound/soc/atmel/mchp-spdifrx.c index 5fc968483f2c..0d37b78b94a0 100644 --- a/sound/soc/atmel/mchp-spdifrx.c +++ b/sound/soc/atmel/mchp-spdifrx.c @@ -846,7 +846,8 @@ static struct snd_soc_dai_driver mchp_spdifrx_dai = { }; static const struct snd_soc_component_driver mchp_spdifrx_component = { - .name = "mchp-spdifrx", + .name = "mchp-spdifrx", + .legacy_dai_naming = 1, }; static const struct of_device_id mchp_spdifrx_dt_ids[] = { diff --git a/sound/soc/atmel/mchp-spdiftx.c b/sound/soc/atmel/mchp-spdiftx.c index d24380046435..78d5bcf0819a 100644 --- a/sound/soc/atmel/mchp-spdiftx.c +++ b/sound/soc/atmel/mchp-spdiftx.c @@ -753,7 +753,8 @@ static struct snd_soc_dai_driver mchp_spdiftx_dai = { }; static const struct snd_soc_component_driver mchp_spdiftx_component = { - .name = "mchp-spdiftx", + .name = "mchp-spdiftx", + .legacy_dai_naming = 1, }; static const struct of_device_id mchp_spdiftx_dt_ids[] = { -- cgit v1.2.3 From 1e63fcc74ace9824f3529eeabbb8f1881a7d3800 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:51:34 +0100 Subject: ASoC: fsl: Migrate to new style legacy DAI naming flag Change the legacy DAI naming flag from opting in to the new scheme (non_legacy_dai_naming), to opting out of it (legacy_dai_naming). These drivers appear to be on the CPU side of the DAI link and currently uses the legacy naming, so add the new flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-21-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_aud2htx.c | 3 ++- sound/soc/fsl/fsl_easrc.c | 7 ++++--- sound/soc/fsl/fsl_esai.c | 3 ++- sound/soc/fsl/fsl_rpmsg.c | 3 ++- sound/soc/fsl/fsl_sai.c | 3 ++- sound/soc/fsl/fsl_spdif.c | 3 ++- sound/soc/fsl/fsl_ssi.c | 1 + sound/soc/fsl/fsl_xcvr.c | 3 ++- sound/soc/fsl/mpc5200_psc_i2s.c | 3 ++- 9 files changed, 19 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_aud2htx.c b/sound/soc/fsl/fsl_aud2htx.c index 422922146f2a..873295f59ad7 100644 --- a/sound/soc/fsl/fsl_aud2htx.c +++ b/sound/soc/fsl/fsl_aud2htx.c @@ -103,7 +103,8 @@ static struct snd_soc_dai_driver fsl_aud2htx_dai = { }; static const struct snd_soc_component_driver fsl_aud2htx_component = { - .name = "fsl-aud2htx", + .name = "fsl-aud2htx", + .legacy_dai_naming = 1, }; static const struct reg_default fsl_aud2htx_reg_defaults[] = { diff --git a/sound/soc/fsl/fsl_easrc.c b/sound/soc/fsl/fsl_easrc.c index be14f84796cb..ea96b0fb6b20 100644 --- a/sound/soc/fsl/fsl_easrc.c +++ b/sound/soc/fsl/fsl_easrc.c @@ -1572,9 +1572,10 @@ static struct snd_soc_dai_driver fsl_easrc_dai = { }; static const struct snd_soc_component_driver fsl_easrc_component = { - .name = "fsl-easrc-dai", - .controls = fsl_easrc_snd_controls, - .num_controls = ARRAY_SIZE(fsl_easrc_snd_controls), + .name = "fsl-easrc-dai", + .controls = fsl_easrc_snd_controls, + .num_controls = ARRAY_SIZE(fsl_easrc_snd_controls), + .legacy_dai_naming = 1, }; static const struct reg_default fsl_easrc_reg_defaults[] = { diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 75f7807df29a..5c21fc490fce 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -824,7 +824,8 @@ static struct snd_soc_dai_driver fsl_esai_dai = { }; static const struct snd_soc_component_driver fsl_esai_component = { - .name = "fsl-esai", + .name = "fsl-esai", + .legacy_dai_naming = 1, }; static const struct reg_default fsl_esai_reg_defaults[] = { diff --git a/sound/soc/fsl/fsl_rpmsg.c b/sound/soc/fsl/fsl_rpmsg.c index 19fd31250883..bf94838bdbef 100644 --- a/sound/soc/fsl/fsl_rpmsg.c +++ b/sound/soc/fsl/fsl_rpmsg.c @@ -135,7 +135,8 @@ static struct snd_soc_dai_driver fsl_rpmsg_dai = { }; static const struct snd_soc_component_driver fsl_component = { - .name = "fsl-rpmsg", + .name = "fsl-rpmsg", + .legacy_dai_naming = 1, }; static const struct fsl_rpmsg_soc_data imx7ulp_data = { diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 4f5bd9597c74..68b5b488deeb 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -767,7 +767,8 @@ static struct snd_soc_dai_driver fsl_sai_dai_template = { }; static const struct snd_soc_component_driver fsl_component = { - .name = "fsl-sai", + .name = "fsl-sai", + .legacy_dai_naming = 1, }; static struct reg_default fsl_sai_reg_defaults_ofs0[] = { diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 42d11aca38a1..0504431792cf 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -1237,7 +1237,8 @@ static struct snd_soc_dai_driver fsl_spdif_dai = { }; static const struct snd_soc_component_driver fsl_spdif_component = { - .name = "fsl-spdif", + .name = "fsl-spdif", + .legacy_dai_naming = 1, }; /* FSL SPDIF REGMAP */ diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 7dd0c48cd9ae..c9e0e31d5b34 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1182,6 +1182,7 @@ static struct snd_soc_dai_driver fsl_ssi_dai_template = { static const struct snd_soc_component_driver fsl_ssi_component = { .name = "fsl-ssi", + .legacy_dai_naming = 1, }; static struct snd_soc_dai_driver fsl_ssi_ac97_dai = { diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c index 55e640cba87d..c043efe4548d 100644 --- a/sound/soc/fsl/fsl_xcvr.c +++ b/sound/soc/fsl/fsl_xcvr.c @@ -911,7 +911,8 @@ static struct snd_soc_dai_driver fsl_xcvr_dai = { }; static const struct snd_soc_component_driver fsl_xcvr_comp = { - .name = "fsl-xcvr-dai", + .name = "fsl-xcvr-dai", + .legacy_dai_naming = 1, }; static const struct reg_default fsl_xcvr_reg_defaults[] = { diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c index 3149d59ae968..73f3e61f208a 100644 --- a/sound/soc/fsl/mpc5200_psc_i2s.c +++ b/sound/soc/fsl/mpc5200_psc_i2s.c @@ -148,7 +148,8 @@ static struct snd_soc_dai_driver psc_i2s_dai[] = {{ } }; static const struct snd_soc_component_driver psc_i2s_component = { - .name = "mpc5200-i2s", + .name = "mpc5200-i2s", + .legacy_dai_naming = 1, }; /* --------------------------------------------------------------------- -- cgit v1.2.3 From f257dea1c589fa3f558502b3ac7b1c09699a73ab Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:51:35 +0100 Subject: ASoC: xtensa: Migrate to new style legacy DAI naming flag Change the legacy DAI naming flag from opting in to the new scheme (non_legacy_dai_naming), to opting out of it (legacy_dai_naming). This driver appears to be on the CPU side of the DAI link and currently uses the legacy naming, so add the new flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-22-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/xtensa/xtfpga-i2s.c | 15 ++++++++------- 1 file changed, 8 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/xtensa/xtfpga-i2s.c b/sound/soc/xtensa/xtfpga-i2s.c index 8bd121546032..a8f156540b50 100644 --- a/sound/soc/xtensa/xtfpga-i2s.c +++ b/sound/soc/xtensa/xtfpga-i2s.c @@ -475,13 +475,14 @@ static int xtfpga_pcm_new(struct snd_soc_component *component, } static const struct snd_soc_component_driver xtfpga_i2s_component = { - .name = DRV_NAME, - .open = xtfpga_pcm_open, - .close = xtfpga_pcm_close, - .hw_params = xtfpga_pcm_hw_params, - .trigger = xtfpga_pcm_trigger, - .pointer = xtfpga_pcm_pointer, - .pcm_construct = xtfpga_pcm_new, + .name = DRV_NAME, + .open = xtfpga_pcm_open, + .close = xtfpga_pcm_close, + .hw_params = xtfpga_pcm_hw_params, + .trigger = xtfpga_pcm_trigger, + .pointer = xtfpga_pcm_pointer, + .pcm_construct = xtfpga_pcm_new, + .legacy_dai_naming = 1, }; static const struct snd_soc_dai_ops xtfpga_i2s_dai_ops = { -- cgit v1.2.3 From 9a34161a0bc90df825694195659d894e80afe7a3 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:51:36 +0100 Subject: ASoC: adi: Migrate to new style legacy DAI naming flag Change the legacy DAI naming flag from opting in to the new scheme (non_legacy_dai_naming), to opting out of it (legacy_dai_naming). These drivers appear to be on the CPU side of the DAI link and currently uses the legacy naming, so add the new flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-23-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/adi/axi-i2s.c | 1 + sound/soc/adi/axi-spdif.c | 1 + 2 files changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/adi/axi-i2s.c b/sound/soc/adi/axi-i2s.c index 1289cb4e2988..b1342351bff4 100644 --- a/sound/soc/adi/axi-i2s.c +++ b/sound/soc/adi/axi-i2s.c @@ -161,6 +161,7 @@ static struct snd_soc_dai_driver axi_i2s_dai = { static const struct snd_soc_component_driver axi_i2s_component = { .name = "axi-i2s", + .legacy_dai_naming = 1, }; static const struct regmap_config axi_i2s_regmap_config = { diff --git a/sound/soc/adi/axi-spdif.c b/sound/soc/adi/axi-spdif.c index 8d4a6cb4e5c5..51b968ea21da 100644 --- a/sound/soc/adi/axi-spdif.c +++ b/sound/soc/adi/axi-spdif.c @@ -167,6 +167,7 @@ static struct snd_soc_dai_driver axi_spdif_dai = { static const struct snd_soc_component_driver axi_spdif_component = { .name = "axi-spdif", + .legacy_dai_naming = 1, }; static const struct regmap_config axi_spdif_regmap_config = { -- cgit v1.2.3 From e740ef3d9418db78ac7a8a24071933f9146af6e4 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:51:37 +0100 Subject: ASoC: dwc: Migrate to new style legacy DAI naming flag Change the legacy DAI naming flag from opting in to the new scheme (non_legacy_dai_naming), to opting out of it (legacy_dai_naming). This driver appears to be on the CPU side of the DAI link and currently uses the legacy naming, so add the new flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-24-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/dwc/dwc-i2s.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index e794e020052e..7f7dd07c63b2 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -449,9 +449,10 @@ static int dw_i2s_resume(struct snd_soc_component *component) #endif static const struct snd_soc_component_driver dw_i2s_component = { - .name = "dw-i2s", - .suspend = dw_i2s_suspend, - .resume = dw_i2s_resume, + .name = "dw-i2s", + .suspend = dw_i2s_suspend, + .resume = dw_i2s_resume, + .legacy_dai_naming = 1, }; /* -- cgit v1.2.3 From 8135d0290a9a1f1f752bb374f93a017b2074d09b Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:51:38 +0100 Subject: ASoC: qcom: Migrate to new style legacy DAI naming flag Change the legacy DAI naming flag from opting in to the new scheme (non_legacy_dai_naming), to opting out of it (legacy_dai_naming). These drivers appear to be on the CPU side of the DAI link and currently uses the legacy naming, so add the new flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-25-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/qcom/lpass-cpu.c | 1 + sound/soc/qcom/qdsp6/q6asm-dai.c | 23 ++++++++++++----------- 2 files changed, 13 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index e6846ad2b5fa..6f1cd3800001 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -472,6 +472,7 @@ static int asoc_qcom_of_xlate_dai_name(struct snd_soc_component *component, static const struct snd_soc_component_driver lpass_cpu_comp_driver = { .name = "lpass-cpu", .of_xlate_dai_name = asoc_qcom_of_xlate_dai_name, + .legacy_dai_naming = 1, }; static bool lpass_cpu_regmap_writeable(struct device *dev, unsigned int reg) diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index b74b67720ef4..5fc8088e63c8 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -1205,17 +1205,18 @@ static const struct snd_soc_dapm_widget q6asm_dapm_widgets[] = { }; static const struct snd_soc_component_driver q6asm_fe_dai_component = { - .name = DRV_NAME, - .open = q6asm_dai_open, - .hw_params = q6asm_dai_hw_params, - .close = q6asm_dai_close, - .prepare = q6asm_dai_prepare, - .trigger = q6asm_dai_trigger, - .pointer = q6asm_dai_pointer, - .pcm_construct = q6asm_dai_pcm_new, - .compress_ops = &q6asm_dai_compress_ops, - .dapm_widgets = q6asm_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(q6asm_dapm_widgets), + .name = DRV_NAME, + .open = q6asm_dai_open, + .hw_params = q6asm_dai_hw_params, + .close = q6asm_dai_close, + .prepare = q6asm_dai_prepare, + .trigger = q6asm_dai_trigger, + .pointer = q6asm_dai_pointer, + .pcm_construct = q6asm_dai_pcm_new, + .compress_ops = &q6asm_dai_compress_ops, + .dapm_widgets = q6asm_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(q6asm_dapm_widgets), + .legacy_dai_naming = 1, }; static struct snd_soc_dai_driver q6asm_fe_dais_template[] = { -- cgit v1.2.3 From d73130ba523b88a3edb097ae3eb9f93df844b5e2 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:51:39 +0100 Subject: ASoC: test-component: Migrate to new style legacy DAI naming flag Change the legacy DAI naming flag from opting in to the new scheme (non_legacy_dai_naming), to opting out of it (legacy_dai_naming). This driver appears to be on the CPU side of the DAI link and currently uses the legacy naming, so add the new flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-26-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/generic/test-component.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/generic/test-component.c b/sound/soc/generic/test-component.c index d28712fabe3f..e2a009bc69af 100644 --- a/sound/soc/generic/test-component.c +++ b/sound/soc/generic/test-component.c @@ -564,6 +564,7 @@ static int test_driver_probe(struct platform_device *pdev) cdriv->pcm_construct = test_component_pcm_construct; cdriv->pointer = test_component_pointer; cdriv->trigger = test_component_trigger; + cdriv->legacy_dai_naming = 1; } else { cdriv->name = "test_codec"; cdriv->idle_bias_on = 1; -- cgit v1.2.3 From d48a77173534df90788075e76fa88c52b7395a1e Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:51:40 +0100 Subject: ASoC: rockchip: Migrate to new style legacy DAI naming flag Change the legacy DAI naming flag from opting in to the new scheme (non_legacy_dai_naming), to opting out of it (legacy_dai_naming). These drivers appear to be on the CPU side of the DAI link and currently uses the legacy naming, so add the new flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-27-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/rockchip/rockchip_i2s.c | 1 + sound/soc/rockchip/rockchip_i2s_tdm.c | 1 + sound/soc/rockchip/rockchip_pdm.c | 1 + sound/soc/rockchip/rockchip_spdif.c | 1 + 4 files changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 0ed01624a2db..96e7aa9eedfc 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -561,6 +561,7 @@ static struct snd_soc_dai_driver rockchip_i2s_dai = { static const struct snd_soc_component_driver rockchip_i2s_component = { .name = DRV_NAME, + .legacy_dai_naming = 1, }; static bool rockchip_i2s_wr_reg(struct device *dev, unsigned int reg) diff --git a/sound/soc/rockchip/rockchip_i2s_tdm.c b/sound/soc/rockchip/rockchip_i2s_tdm.c index 48b3ecfa58b4..2aad0f309cb6 100644 --- a/sound/soc/rockchip/rockchip_i2s_tdm.c +++ b/sound/soc/rockchip/rockchip_i2s_tdm.c @@ -1120,6 +1120,7 @@ static const struct snd_soc_dai_ops rockchip_i2s_tdm_dai_ops = { static const struct snd_soc_component_driver rockchip_i2s_tdm_component = { .name = DRV_NAME, + .legacy_dai_naming = 1, }; static bool rockchip_i2s_tdm_wr_reg(struct device *dev, unsigned int reg) diff --git a/sound/soc/rockchip/rockchip_pdm.c b/sound/soc/rockchip/rockchip_pdm.c index 64d9891b6434..6d93155411b0 100644 --- a/sound/soc/rockchip/rockchip_pdm.c +++ b/sound/soc/rockchip/rockchip_pdm.c @@ -405,6 +405,7 @@ static struct snd_soc_dai_driver rockchip_pdm_dai = { static const struct snd_soc_component_driver rockchip_pdm_component = { .name = "rockchip-pdm", + .legacy_dai_naming = 1, }; static int rockchip_pdm_runtime_suspend(struct device *dev) diff --git a/sound/soc/rockchip/rockchip_spdif.c b/sound/soc/rockchip/rockchip_spdif.c index d027ca4b1796..8bef572d3cbc 100644 --- a/sound/soc/rockchip/rockchip_spdif.c +++ b/sound/soc/rockchip/rockchip_spdif.c @@ -225,6 +225,7 @@ static struct snd_soc_dai_driver rk_spdif_dai = { static const struct snd_soc_component_driver rk_spdif_component = { .name = "rockchip-spdif", + .legacy_dai_naming = 1, }; static bool rk_spdif_wr_reg(struct device *dev, unsigned int reg) -- cgit v1.2.3 From 8e750817a1943b49d81c633f48370bce93bab98c Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:51:41 +0100 Subject: ASoC: au1x: Migrate to new style legacy DAI naming flag Change the legacy DAI naming flag from opting in to the new scheme (non_legacy_dai_naming), to opting out of it (legacy_dai_naming). These drivers appear to be on the CPU side of the DAI link and currently uses the legacy naming, so add the new flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-28-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/au1x/ac97c.c | 3 ++- sound/soc/au1x/i2sc.c | 3 ++- sound/soc/au1x/psc-ac97.c | 3 ++- sound/soc/au1x/psc-i2s.c | 3 ++- 4 files changed, 8 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/au1x/ac97c.c b/sound/soc/au1x/ac97c.c index 3b1700e665f5..b18512ca2578 100644 --- a/sound/soc/au1x/ac97c.c +++ b/sound/soc/au1x/ac97c.c @@ -223,7 +223,8 @@ static struct snd_soc_dai_driver au1xac97c_dai_driver = { }; static const struct snd_soc_component_driver au1xac97c_component = { - .name = "au1xac97c", + .name = "au1xac97c", + .legacy_dai_naming = 1, }; static int au1xac97c_drvprobe(struct platform_device *pdev) diff --git a/sound/soc/au1x/i2sc.c b/sound/soc/au1x/i2sc.c index 45bb7851e75d..b15c8baa9ee4 100644 --- a/sound/soc/au1x/i2sc.c +++ b/sound/soc/au1x/i2sc.c @@ -227,7 +227,8 @@ static struct snd_soc_dai_driver au1xi2s_dai_driver = { }; static const struct snd_soc_component_driver au1xi2s_component = { - .name = "au1xi2s", + .name = "au1xi2s", + .legacy_dai_naming = 1, }; static int au1xi2s_drvprobe(struct platform_device *pdev) diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index 05eb36991f14..b536394b9ca0 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -356,7 +356,8 @@ static const struct snd_soc_dai_driver au1xpsc_ac97_dai_template = { }; static const struct snd_soc_component_driver au1xpsc_ac97_component = { - .name = "au1xpsc-ac97", + .name = "au1xpsc-ac97", + .legacy_dai_naming = 1, }; static int au1xpsc_ac97_drvprobe(struct platform_device *pdev) diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index 530a072d7427..79b5ae4e494c 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -286,7 +286,8 @@ static const struct snd_soc_dai_driver au1xpsc_i2s_dai_template = { }; static const struct snd_soc_component_driver au1xpsc_i2s_component = { - .name = "au1xpsc-i2s", + .name = "au1xpsc-i2s", + .legacy_dai_naming = 1, }; static int au1xpsc_i2s_drvprobe(struct platform_device *pdev) -- cgit v1.2.3 From 05603f15b67a517c05ee2e2298e9accb8b7f1794 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:51:42 +0100 Subject: ASoC: pxa: Migrate to new style legacy DAI naming flag Change the legacy DAI naming flag from opting in to the new scheme (non_legacy_dai_naming), to opting out of it (legacy_dai_naming). These drivers appear to be on the CPU side of the DAI link and currently uses the legacy naming, so add the new flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-29-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/pxa/mmp-sspa.c | 9 +++++---- sound/soc/pxa/pxa-ssp.c | 21 +++++++++++---------- sound/soc/pxa/pxa2xx-i2s.c | 21 +++++++++++---------- 3 files changed, 27 insertions(+), 24 deletions(-) (limited to 'sound') diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c index 382e9d8608a3..fb5a4390443f 100644 --- a/sound/soc/pxa/mmp-sspa.c +++ b/sound/soc/pxa/mmp-sspa.c @@ -456,10 +456,11 @@ static int mmp_sspa_close(struct snd_soc_component *component, } static const struct snd_soc_component_driver mmp_sspa_component = { - .name = "mmp-sspa", - .mmap = mmp_pcm_mmap, - .open = mmp_sspa_open, - .close = mmp_sspa_close, + .name = "mmp-sspa", + .mmap = mmp_pcm_mmap, + .open = mmp_sspa_open, + .close = mmp_sspa_close, + .legacy_dai_naming = 1, }; static int asoc_mmp_sspa_probe(struct platform_device *pdev) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 0f504a9f4983..430dd446321e 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -848,16 +848,17 @@ static struct snd_soc_dai_driver pxa_ssp_dai = { }; static const struct snd_soc_component_driver pxa_ssp_component = { - .name = "pxa-ssp", - .pcm_construct = pxa2xx_soc_pcm_new, - .open = pxa2xx_soc_pcm_open, - .close = pxa2xx_soc_pcm_close, - .hw_params = pxa2xx_soc_pcm_hw_params, - .prepare = pxa2xx_soc_pcm_prepare, - .trigger = pxa2xx_soc_pcm_trigger, - .pointer = pxa2xx_soc_pcm_pointer, - .suspend = pxa_ssp_suspend, - .resume = pxa_ssp_resume, + .name = "pxa-ssp", + .pcm_construct = pxa2xx_soc_pcm_new, + .open = pxa2xx_soc_pcm_open, + .close = pxa2xx_soc_pcm_close, + .hw_params = pxa2xx_soc_pcm_hw_params, + .prepare = pxa2xx_soc_pcm_prepare, + .trigger = pxa2xx_soc_pcm_trigger, + .pointer = pxa2xx_soc_pcm_pointer, + .suspend = pxa_ssp_suspend, + .resume = pxa_ssp_resume, + .legacy_dai_naming = 1, }; #ifdef CONFIG_OF diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index ffcf44e4dc8c..3e4c70403672 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -355,16 +355,17 @@ static struct snd_soc_dai_driver pxa_i2s_dai = { }; static const struct snd_soc_component_driver pxa_i2s_component = { - .name = "pxa-i2s", - .pcm_construct = pxa2xx_soc_pcm_new, - .open = pxa2xx_soc_pcm_open, - .close = pxa2xx_soc_pcm_close, - .hw_params = pxa2xx_soc_pcm_hw_params, - .prepare = pxa2xx_soc_pcm_prepare, - .trigger = pxa2xx_soc_pcm_trigger, - .pointer = pxa2xx_soc_pcm_pointer, - .suspend = pxa2xx_soc_pcm_suspend, - .resume = pxa2xx_soc_pcm_resume, + .name = "pxa-i2s", + .pcm_construct = pxa2xx_soc_pcm_new, + .open = pxa2xx_soc_pcm_open, + .close = pxa2xx_soc_pcm_close, + .hw_params = pxa2xx_soc_pcm_hw_params, + .prepare = pxa2xx_soc_pcm_prepare, + .trigger = pxa2xx_soc_pcm_trigger, + .pointer = pxa2xx_soc_pcm_pointer, + .suspend = pxa2xx_soc_pcm_suspend, + .resume = pxa2xx_soc_pcm_resume, + .legacy_dai_naming = 1, }; static int pxa2xx_i2s_drv_probe(struct platform_device *pdev) -- cgit v1.2.3 From a718ba30038402e6daa311c566d9be39e4ab3f05 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:51:43 +0100 Subject: ASoC: sof: Migrate to new style legacy DAI naming flag Change the legacy DAI naming flag from opting in to the new scheme (non_legacy_dai_naming), to opting out of it (legacy_dai_naming). These drivers appear to be on the CPU side of the DAI link and currently uses the legacy naming, so add the new flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-30-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/sof/pcm.c | 2 ++ sound/soc/sof/sof-client-probes.c | 1 + 2 files changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c index 27504abc5385..6cb6a432be5e 100644 --- a/sound/soc/sof/pcm.c +++ b/sound/soc/sof/pcm.c @@ -682,4 +682,6 @@ void snd_sof_new_platform_drv(struct snd_sof_dev *sdev) /* increment module refcount when a pcm is opened */ pd->module_get_upon_open = 1; + + pd->legacy_dai_naming = 1; } diff --git a/sound/soc/sof/sof-client-probes.c b/sound/soc/sof/sof-client-probes.c index 34e6bd356e71..1f1ea93a7fbf 100644 --- a/sound/soc/sof/sof-client-probes.c +++ b/sound/soc/sof/sof-client-probes.c @@ -667,6 +667,7 @@ static const struct snd_soc_component_driver sof_probes_component = { .name = "sof-probes-component", .compress_ops = &sof_probes_compressed_ops, .module_get_upon_open = 1, + .legacy_dai_naming = 1, }; SND_SOC_DAILINK_DEF(dummy, DAILINK_COMP_ARRAY(COMP_DUMMY())); -- cgit v1.2.3 From 768be0d633d9ff668a7ca4ba3b8e3eebea328cb8 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:51:44 +0100 Subject: ASoC: ux500: Migrate to new style legacy DAI naming flag Change the legacy DAI naming flag from opting in to the new scheme (non_legacy_dai_naming), to opting out of it (legacy_dai_naming). This driver appears to be on the CPU side of the DAI link and currently uses the legacy naming, so add the new flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-31-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/ux500/ux500_msp_dai.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c index e48098f039d9..9d99ea6d7f30 100644 --- a/sound/soc/ux500/ux500_msp_dai.c +++ b/sound/soc/ux500/ux500_msp_dai.c @@ -729,7 +729,8 @@ static struct snd_soc_dai_driver ux500_msp_dai_drv = { }; static const struct snd_soc_component_driver ux500_msp_component = { - .name = "ux500-msp", + .name = "ux500-msp", + .legacy_dai_naming = 1, }; -- cgit v1.2.3 From 39c84e77da04f66f20fc54c6c6f49a5863bace5d Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:51:45 +0100 Subject: ASoC: ti: Migrate to new style legacy DAI naming flag Change the legacy DAI naming flag from opting in to the new scheme (non_legacy_dai_naming), to opting out of it (legacy_dai_naming). These drivers appear to be on the CPU side of the DAI link and currently uses the legacy naming, so add the new flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-32-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/ti/davinci-i2s.c | 3 ++- sound/soc/ti/davinci-mcasp.c | 3 ++- sound/soc/ti/davinci-vcif.c | 3 ++- sound/soc/ti/omap-dmic.c | 3 ++- sound/soc/ti/omap-hdmi.c | 1 + sound/soc/ti/omap-mcbsp.c | 3 ++- sound/soc/ti/omap-mcpdm.c | 7 ++++--- 7 files changed, 15 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/ti/davinci-i2s.c b/sound/soc/ti/davinci-i2s.c index fe572b720b09..e6e77a5f3c1e 100644 --- a/sound/soc/ti/davinci-i2s.c +++ b/sound/soc/ti/davinci-i2s.c @@ -640,7 +640,8 @@ static struct snd_soc_dai_driver davinci_i2s_dai = { }; static const struct snd_soc_component_driver davinci_i2s_component = { - .name = DRV_NAME, + .name = DRV_NAME, + .legacy_dai_naming = 1, }; static int davinci_i2s_probe(struct platform_device *pdev) diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index e2aab4729f3a..45ffcc7aadc9 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -1765,7 +1765,8 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = { }; static const struct snd_soc_component_driver davinci_mcasp_component = { - .name = "davinci-mcasp", + .name = "davinci-mcasp", + .legacy_dai_naming = 1, }; /* Some HW specific values and defaults. The rest is filled in from DT. */ diff --git a/sound/soc/ti/davinci-vcif.c b/sound/soc/ti/davinci-vcif.c index f810123cc407..36fa97e2b9e2 100644 --- a/sound/soc/ti/davinci-vcif.c +++ b/sound/soc/ti/davinci-vcif.c @@ -185,7 +185,8 @@ static struct snd_soc_dai_driver davinci_vcif_dai = { }; static const struct snd_soc_component_driver davinci_vcif_component = { - .name = "davinci-vcif", + .name = "davinci-vcif", + .legacy_dai_naming = 1, }; static int davinci_vcif_probe(struct platform_device *pdev) diff --git a/sound/soc/ti/omap-dmic.c b/sound/soc/ti/omap-dmic.c index f3eed20611a3..825c70a443da 100644 --- a/sound/soc/ti/omap-dmic.c +++ b/sound/soc/ti/omap-dmic.c @@ -453,7 +453,8 @@ static struct snd_soc_dai_driver omap_dmic_dai = { }; static const struct snd_soc_component_driver omap_dmic_component = { - .name = "omap-dmic", + .name = "omap-dmic", + .legacy_dai_naming = 1, }; static int asoc_dmic_probe(struct platform_device *pdev) diff --git a/sound/soc/ti/omap-hdmi.c b/sound/soc/ti/omap-hdmi.c index 3328c02f93c7..0dc0475670ff 100644 --- a/sound/soc/ti/omap-hdmi.c +++ b/sound/soc/ti/omap-hdmi.c @@ -275,6 +275,7 @@ static const struct snd_soc_dai_ops hdmi_dai_ops = { static const struct snd_soc_component_driver omap_hdmi_component = { .name = "omapdss_hdmi", + .legacy_dai_naming = 1, }; static struct snd_soc_dai_driver omap5_hdmi_dai = { diff --git a/sound/soc/ti/omap-mcbsp.c b/sound/soc/ti/omap-mcbsp.c index 58d8e200a7b9..76df0e7844f8 100644 --- a/sound/soc/ti/omap-mcbsp.c +++ b/sound/soc/ti/omap-mcbsp.c @@ -1317,7 +1317,8 @@ static struct snd_soc_dai_driver omap_mcbsp_dai = { }; static const struct snd_soc_component_driver omap_mcbsp_component = { - .name = "omap-mcbsp", + .name = "omap-mcbsp", + .legacy_dai_naming = 1, }; static struct omap_mcbsp_platform_data omap2420_pdata = { diff --git a/sound/soc/ti/omap-mcpdm.c b/sound/soc/ti/omap-mcpdm.c index fafb2998ad0d..0b18a7bfd3fd 100644 --- a/sound/soc/ti/omap-mcpdm.c +++ b/sound/soc/ti/omap-mcpdm.c @@ -524,9 +524,10 @@ static struct snd_soc_dai_driver omap_mcpdm_dai = { }; static const struct snd_soc_component_driver omap_mcpdm_component = { - .name = "omap-mcpdm", - .suspend = omap_mcpdm_suspend, - .resume = omap_mcpdm_resume, + .name = "omap-mcpdm", + .suspend = omap_mcpdm_suspend, + .resume = omap_mcpdm_resume, + .legacy_dai_naming = 1, }; void omap_mcpdm_configure_dn_offsets(struct snd_soc_pcm_runtime *rtd, -- cgit v1.2.3 From 4cc4e22843e9bec6e9083d85e8a0bfed85fe5423 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:51:46 +0100 Subject: ASoC: mxs-saif: Migrate to new style legacy DAI naming flag Change the legacy DAI naming flag from opting in to the new scheme (non_legacy_dai_naming), to opting out of it (legacy_dai_naming). This driver appears to be on the CPU side of the DAI link and currently uses the legacy naming, so add the new flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-33-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/mxs/mxs-saif.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 467b0f2ce0bb..ac761d3a01c0 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -663,7 +663,8 @@ static struct snd_soc_dai_driver mxs_saif_dai = { }; static const struct snd_soc_component_driver mxs_saif_component = { - .name = "mxs-saif", + .name = "mxs-saif", + .legacy_dai_naming = 1, }; static irqreturn_t mxs_saif_irq(int irq, void *dev_id) -- cgit v1.2.3 From f7bfa516a39a111a5d3b6473cdac20ee6075358c Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:51:47 +0100 Subject: ASoC: samsung: Migrate to new style legacy DAI naming flag Change the legacy DAI naming flag from opting in to the new scheme (non_legacy_dai_naming), to opting out of it (legacy_dai_naming). These drivers appear to be on the CPU side of the DAI link and currently uses the legacy naming, so add the new flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-34-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/samsung/i2s.c | 2 ++ sound/soc/samsung/pcm.c | 3 ++- sound/soc/samsung/s3c2412-i2s.c | 7 ++++--- sound/soc/samsung/s3c24xx-i2s.c | 7 ++++--- sound/soc/samsung/spdif.c | 7 ++++--- 5 files changed, 16 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index fdd9561c6a9f..9505200f3d11 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1143,6 +1143,8 @@ static const struct snd_soc_component_driver samsung_i2s_component = { .suspend = i2s_suspend, .resume = i2s_resume, + + .legacy_dai_naming = 1, }; #define SAMSUNG_I2S_FMTS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \ diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index c2eb3534bfcc..e859252ae5e6 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -480,7 +480,8 @@ static struct snd_soc_dai_driver s3c_pcm_dai[] = { }; static const struct snd_soc_component_driver s3c_pcm_component = { - .name = "s3c-pcm", + .name = "s3c-pcm", + .legacy_dai_naming = 1, }; static int s3c_pcm_dev_probe(struct platform_device *pdev) diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index ec1c6f9d76ac..0579a352961c 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -192,9 +192,10 @@ static struct snd_soc_dai_driver s3c2412_i2s_dai = { }; static const struct snd_soc_component_driver s3c2412_i2s_component = { - .name = "s3c2412-i2s", - .suspend = s3c2412_i2s_suspend, - .resume = s3c2412_i2s_resume, + .name = "s3c2412-i2s", + .suspend = s3c2412_i2s_suspend, + .resume = s3c2412_i2s_resume, + .legacy_dai_naming = 1, }; static int s3c2412_iis_dev_probe(struct platform_device *pdev) diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index 4082ad7cbcc1..e760fc8b4263 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -415,9 +415,10 @@ static struct snd_soc_dai_driver s3c24xx_i2s_dai = { }; static const struct snd_soc_component_driver s3c24xx_i2s_component = { - .name = "s3c24xx-i2s", - .suspend = s3c24xx_i2s_suspend, - .resume = s3c24xx_i2s_resume, + .name = "s3c24xx-i2s", + .suspend = s3c24xx_i2s_suspend, + .resume = s3c24xx_i2s_resume, + .legacy_dai_naming = 1, }; static int s3c24xx_iis_dev_probe(struct platform_device *pdev) diff --git a/sound/soc/samsung/spdif.c b/sound/soc/samsung/spdif.c index 47b6d19e43ff..7d815e237e5c 100644 --- a/sound/soc/samsung/spdif.c +++ b/sound/soc/samsung/spdif.c @@ -352,9 +352,10 @@ static struct snd_soc_dai_driver samsung_spdif_dai = { }; static const struct snd_soc_component_driver samsung_spdif_component = { - .name = "samsung-spdif", - .suspend = spdif_suspend, - .resume = spdif_resume, + .name = "samsung-spdif", + .suspend = spdif_suspend, + .resume = spdif_resume, + .legacy_dai_naming = 1, }; static int spdif_probe(struct platform_device *pdev) -- cgit v1.2.3 From 129f055a2144ab588a43c2e66d21a1f55ce54f81 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:51:48 +0100 Subject: ASoC: core: Switch core to new DAI naming flag Now all the drivers are updated to have the new legacy_dai_naming flag, update the core code so it also uses the new flag. Paving the way for the old non_legacy_dai_naming flag to be removed. It should be noted this patch will affect the CODEC drivers that don't specify the non_legacy_dai_naming flag. These drivers will update from using legacy DAI naming to the new scheme after this patch, this is being considered a fix as the intention was for all CODEC drivers to use the new scheme and all existing CODEC drivers were updated to do so before componentisation. This just corrects those devices that have snuck in since componentisation. The corrected devices are as follows: adau1372, cros_ec_codec, cs35l41, cs35l45, cx2072x, hdac_hda, jz4725/60/70, lpass-rx/tx/va/wsa-macro, max98504, max9877, mt6351/58/59, mt6660, pcm3060, rk3328, rt1308/16, rt5514, rt5677, rt700/11/15, rt9120, sdw-mockup, tlv320adc3xxx, tscs454, wcd9335/4x/8x, wsa881x Some of these devices are used in some in kernel machine drivers, however it appears all the usages use the actual DAI driver name (since snd_soc_find_dai checks both the DAI name and the DAI driver name). So it is not believed this change will break any in tree machine drivers. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-35-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 30f0da711ca9..60e21b06b1dc 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2488,7 +2488,7 @@ static int snd_soc_register_dais(struct snd_soc_component *component, for (i = 0; i < count; i++) { dai = snd_soc_register_dai(component, dai_drv + i, count == 1 && - !component->driver->non_legacy_dai_naming); + component->driver->legacy_dai_naming); if (dai == NULL) { ret = -ENOMEM; goto err; -- cgit v1.2.3 From 89836f00429b5c3dedb2e2f30262e53847b82ad0 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:51:50 +0100 Subject: ASoC: fsl: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-37-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_mqs.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_mqs.c b/sound/soc/fsl/fsl_mqs.c index bb25c58e335f..c1e2f671191b 100644 --- a/sound/soc/fsl/fsl_mqs.c +++ b/sound/soc/fsl/fsl_mqs.c @@ -155,7 +155,6 @@ static void fsl_mqs_shutdown(struct snd_pcm_substream *substream, static const struct snd_soc_component_driver soc_codec_fsl_mqs = { .idle_bias_on = 1, - .non_legacy_dai_naming = 1, }; static const struct snd_soc_dai_ops fsl_mqs_dai_ops = { -- cgit v1.2.3 From 9455e289246d8769631e6bec78c0c2ef40171b70 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:51:51 +0100 Subject: ASoC: meson: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-38-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/meson/aiu-acodec-ctrl.c | 1 - sound/soc/meson/aiu-codec-ctrl.c | 1 - sound/soc/meson/g12a-toacodec.c | 2 -- sound/soc/meson/g12a-tohdmitx.c | 1 - sound/soc/meson/t9015.c | 1 - 5 files changed, 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/meson/aiu-acodec-ctrl.c b/sound/soc/meson/aiu-acodec-ctrl.c index 3776b073a3db..d0f0ada5f4bc 100644 --- a/sound/soc/meson/aiu-acodec-ctrl.c +++ b/sound/soc/meson/aiu-acodec-ctrl.c @@ -192,7 +192,6 @@ static const struct snd_soc_component_driver aiu_acodec_ctrl_component = { .num_dapm_routes = ARRAY_SIZE(aiu_acodec_ctrl_routes), .of_xlate_dai_name = aiu_acodec_of_xlate_dai_name, .endianness = 1, - .non_legacy_dai_naming = 1, #ifdef CONFIG_DEBUG_FS .debugfs_prefix = "acodec", #endif diff --git a/sound/soc/meson/aiu-codec-ctrl.c b/sound/soc/meson/aiu-codec-ctrl.c index 286ac4983d40..84c10956c241 100644 --- a/sound/soc/meson/aiu-codec-ctrl.c +++ b/sound/soc/meson/aiu-codec-ctrl.c @@ -139,7 +139,6 @@ static const struct snd_soc_component_driver aiu_hdmi_ctrl_component = { .num_dapm_routes = ARRAY_SIZE(aiu_hdmi_ctrl_routes), .of_xlate_dai_name = aiu_hdmi_of_xlate_dai_name, .endianness = 1, - .non_legacy_dai_naming = 1, #ifdef CONFIG_DEBUG_FS .debugfs_prefix = "hdmi", #endif diff --git a/sound/soc/meson/g12a-toacodec.c b/sound/soc/meson/g12a-toacodec.c index 1dfee1396843..ddc667956cf5 100644 --- a/sound/soc/meson/g12a-toacodec.c +++ b/sound/soc/meson/g12a-toacodec.c @@ -242,7 +242,6 @@ static const struct snd_soc_component_driver g12a_toacodec_component_drv = { .dapm_routes = g12a_toacodec_routes, .num_dapm_routes = ARRAY_SIZE(g12a_toacodec_routes), .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct snd_soc_component_driver sm1_toacodec_component_drv = { @@ -254,7 +253,6 @@ static const struct snd_soc_component_driver sm1_toacodec_component_drv = { .dapm_routes = g12a_toacodec_routes, .num_dapm_routes = ARRAY_SIZE(g12a_toacodec_routes), .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config g12a_toacodec_regmap_cfg = { diff --git a/sound/soc/meson/g12a-tohdmitx.c b/sound/soc/meson/g12a-tohdmitx.c index 6c99052feafd..579a04ad4d19 100644 --- a/sound/soc/meson/g12a-tohdmitx.c +++ b/sound/soc/meson/g12a-tohdmitx.c @@ -226,7 +226,6 @@ static const struct snd_soc_component_driver g12a_tohdmitx_component_drv = { .dapm_routes = g12a_tohdmitx_routes, .num_dapm_routes = ARRAY_SIZE(g12a_tohdmitx_routes), .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config g12a_tohdmitx_regmap_cfg = { diff --git a/sound/soc/meson/t9015.c b/sound/soc/meson/t9015.c index a9b8c4e77d40..9c6b4dac6893 100644 --- a/sound/soc/meson/t9015.c +++ b/sound/soc/meson/t9015.c @@ -234,7 +234,6 @@ static const struct snd_soc_component_driver t9015_codec_driver = { .num_dapm_routes = ARRAY_SIZE(t9015_dapm_routes), .suspend_bias_off = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config t9015_regmap_config = { -- cgit v1.2.3 From 7e91c90863df7387b9e1f04d9bfc2a43c77d2a46 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:51:52 +0100 Subject: ASoC: pistachio: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-39-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/img/pistachio-internal-dac.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/img/pistachio-internal-dac.c b/sound/soc/img/pistachio-internal-dac.c index 802c0ee63aa2..e3b858643bd5 100644 --- a/sound/soc/img/pistachio-internal-dac.c +++ b/sound/soc/img/pistachio-internal-dac.c @@ -138,7 +138,6 @@ static const struct snd_soc_component_driver pistachio_internal_dac_driver = { .num_dapm_routes = ARRAY_SIZE(pistachio_internal_dac_routes), .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int pistachio_internal_dac_probe(struct platform_device *pdev) -- cgit v1.2.3 From 752044db5b54c867dadfbd0daea90f1b9ecb21f1 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:51:53 +0100 Subject: ASoC: samsung: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-40-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/samsung/aries_wm8994.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/samsung/aries_wm8994.c b/sound/soc/samsung/aries_wm8994.c index edee02d7f100..e7d52d27132e 100644 --- a/sound/soc/samsung/aries_wm8994.c +++ b/sound/soc/samsung/aries_wm8994.c @@ -432,7 +432,6 @@ static const struct snd_soc_component_driver aries_component = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static struct snd_soc_dai_driver aries_ext_dai[] = { -- cgit v1.2.3 From 0f91b4de756415382c10c502010c7536500a1632 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:51:54 +0100 Subject: ASoC: soc-utils: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-41-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/soc-utils.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index 594cb311ff30..70c380c0ac7b 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -141,7 +141,6 @@ static const struct snd_soc_component_driver dummy_codec = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; #define STUB_RATES SNDRV_PCM_RATE_8000_384000 -- cgit v1.2.3 From 4c6391f59c459e7cf8d584299d0746cb681c2cb7 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:51:55 +0100 Subject: ASoC: sunxi: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-42-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/sunxi/sun4i-codec.c | 4 ---- sound/soc/sunxi/sun8i-codec.c | 1 - 2 files changed, 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index bc634962a57e..830beb38bf15 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -881,7 +881,6 @@ static const struct snd_soc_component_driver sun4i_codec_codec = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct snd_soc_component_driver sun7i_codec_codec = { @@ -894,7 +893,6 @@ static const struct snd_soc_component_driver sun7i_codec_codec = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; /*** sun6i Codec ***/ @@ -1202,7 +1200,6 @@ static const struct snd_soc_component_driver sun6i_codec_codec = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; /* sun8i A23 codec */ @@ -1230,7 +1227,6 @@ static const struct snd_soc_component_driver sun8i_a23_codec_codec = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct snd_soc_component_driver sun4i_codec_component = { diff --git a/sound/soc/sunxi/sun8i-codec.c b/sound/soc/sunxi/sun8i-codec.c index f797c535f298..9844978d91e6 100644 --- a/sound/soc/sunxi/sun8i-codec.c +++ b/sound/soc/sunxi/sun8i-codec.c @@ -1278,7 +1278,6 @@ static const struct snd_soc_component_driver sun8i_soc_component = { .probe = sun8i_codec_component_probe, .idle_bias_on = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config sun8i_codec_regmap_config = { -- cgit v1.2.3 From 63c0ec9ebfec499d603993ea8244907bfbe39598 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:51:56 +0100 Subject: ASoC: tegra: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-43-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/tegra/tegra210_i2s.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra210_i2s.c b/sound/soc/tegra/tegra210_i2s.c index 01c76ba36e1a..39ffa4d76b59 100644 --- a/sound/soc/tegra/tegra210_i2s.c +++ b/sound/soc/tegra/tegra210_i2s.c @@ -803,7 +803,6 @@ static const struct snd_soc_component_driver tegra210_i2s_cmpnt = { .num_dapm_routes = ARRAY_SIZE(tegra210_i2s_routes), .controls = tegra210_i2s_controls, .num_controls = ARRAY_SIZE(tegra210_i2s_controls), - .non_legacy_dai_naming = 1, }; static bool tegra210_i2s_wr_reg(struct device *dev, unsigned int reg) -- cgit v1.2.3 From 485c5924f262d4aef720c508ee2ff3cb8e2e531b Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:51:57 +0100 Subject: ASoC: test-component: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-44-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/generic/test-component.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/generic/test-component.c b/sound/soc/generic/test-component.c index e2a009bc69af..98c8990596a8 100644 --- a/sound/soc/generic/test-component.c +++ b/sound/soc/generic/test-component.c @@ -569,7 +569,6 @@ static int test_driver_probe(struct platform_device *pdev) cdriv->name = "test_codec"; cdriv->idle_bias_on = 1; cdriv->endianness = 1; - cdriv->non_legacy_dai_naming = 1; } cdriv->open = test_component_open; -- cgit v1.2.3 From 7cfb102a55556f5f165a2150a6f77a5aa7257599 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:51:58 +0100 Subject: ASoC: topology: KUnit: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-45-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/soc-topology-test.c | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-topology-test.c b/sound/soc/soc-topology-test.c index 51d650bb05b7..2cd3540cec04 100644 --- a/sound/soc/soc-topology-test.c +++ b/sound/soc/soc-topology-test.c @@ -104,7 +104,6 @@ static const struct snd_soc_component_driver test_component = { .name = "sound-soc-topology-test", .probe = d_probe, .remove = d_remove, - .non_legacy_dai_naming = 1, }; /* ===== TOPOLOGY TEMPLATES ================================================= */ @@ -238,7 +237,6 @@ static int d_probe_null_comp(struct snd_soc_component *component) static const struct snd_soc_component_driver test_component_null_comp = { .name = "sound-soc-topology-test", .probe = d_probe_null_comp, - .non_legacy_dai_naming = 1, }; static void snd_soc_tplg_test_load_with_null_comp(struct kunit *test) @@ -343,7 +341,6 @@ static int d_probe_null_fw(struct snd_soc_component *component) static const struct snd_soc_component_driver test_component_null_fw = { .name = "sound-soc-topology-test", .probe = d_probe_null_fw, - .non_legacy_dai_naming = 1, }; static void snd_soc_tplg_test_load_with_null_fw(struct kunit *test) -- cgit v1.2.3 From 36e79a44b12e4ce2d8659f47dbcce42690919567 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:51:59 +0100 Subject: ASoC: uniphier: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-46-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/uniphier/evea.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/uniphier/evea.c b/sound/soc/uniphier/evea.c index 96343d19a1e0..42403ae8e31b 100644 --- a/sound/soc/uniphier/evea.c +++ b/sound/soc/uniphier/evea.c @@ -397,7 +397,6 @@ static struct snd_soc_component_driver soc_codec_evea = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static struct snd_soc_dai_driver soc_dai_evea[] = { -- cgit v1.2.3 From d9e7ddb98604de6470a0fe4f9e2434a55ca35730 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:52:00 +0100 Subject: ASoC: ad*: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-47-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/ad1836.c | 1 - sound/soc/codecs/ad193x.c | 1 - sound/soc/codecs/ad1980.c | 1 - sound/soc/codecs/ad73311.c | 1 - sound/soc/codecs/adau1373.c | 1 - sound/soc/codecs/adau1701.c | 1 - sound/soc/codecs/adau1761.c | 1 - sound/soc/codecs/adau1781.c | 1 - sound/soc/codecs/adau1977.c | 1 - sound/soc/codecs/adau7002.c | 1 - sound/soc/codecs/adau7118.c | 1 - sound/soc/codecs/adav80x.c | 1 - 12 files changed, 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index 29e1689da67f..2c64df96b5ce 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -332,7 +332,6 @@ static const struct snd_soc_component_driver soc_component_dev_ad1836 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct reg_default ad1836_reg_defaults[] = { diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index 30b98b4267e1..1d3c4d94b4ae 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -523,7 +523,6 @@ static const struct snd_soc_component_driver soc_component_dev_ad193x = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; const struct regmap_config ad193x_regmap_config = { diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 9fd2023da218..5e777d7fd5d9 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -302,7 +302,6 @@ static const struct snd_soc_component_driver soc_component_dev_ad1980 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int ad1980_probe(struct platform_device *pdev) diff --git a/sound/soc/codecs/ad73311.c b/sound/soc/codecs/ad73311.c index b98bf19f594e..f6090ac57e93 100644 --- a/sound/soc/codecs/ad73311.c +++ b/sound/soc/codecs/ad73311.c @@ -58,7 +58,6 @@ static const struct snd_soc_component_driver soc_component_dev_ad73311 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int ad73311_probe(struct platform_device *pdev) diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index a9032b5c8d78..7f832d00ab17 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -1470,7 +1470,6 @@ static const struct snd_soc_component_driver adau1373_component_driver = { .num_dapm_routes = ARRAY_SIZE(adau1373_dapm_routes), .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int adau1373_i2c_probe(struct i2c_client *client) diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index 98768e5300f0..135a7db7fcf9 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -772,7 +772,6 @@ static const struct snd_soc_component_driver adau1701_component_drv = { .set_sysclk = adau1701_set_sysclk, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config adau1701_regmap = { diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c index 8f887227981f..3ccc7acac205 100644 --- a/sound/soc/codecs/adau1761.c +++ b/sound/soc/codecs/adau1761.c @@ -930,7 +930,6 @@ static const struct snd_soc_component_driver adau1761_component_driver = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; #define ADAU1761_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | \ diff --git a/sound/soc/codecs/adau1781.c b/sound/soc/codecs/adau1781.c index 74dc3344b259..ff6be24863bf 100644 --- a/sound/soc/codecs/adau1781.c +++ b/sound/soc/codecs/adau1781.c @@ -439,7 +439,6 @@ static const struct snd_soc_component_driver adau1781_component_driver = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; #define ADAU1781_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | \ diff --git a/sound/soc/codecs/adau1977.c b/sound/soc/codecs/adau1977.c index 5fcbdf2ec313..7a9672f94fc6 100644 --- a/sound/soc/codecs/adau1977.c +++ b/sound/soc/codecs/adau1977.c @@ -876,7 +876,6 @@ static const struct snd_soc_component_driver adau1977_component_driver = { .num_dapm_routes = ARRAY_SIZE(adau1977_dapm_routes), .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int adau1977_setup_micbias(struct adau1977 *adau1977) diff --git a/sound/soc/codecs/adau7002.c b/sound/soc/codecs/adau7002.c index 0e00de6ce3fb..401bafabc8eb 100644 --- a/sound/soc/codecs/adau7002.c +++ b/sound/soc/codecs/adau7002.c @@ -91,7 +91,6 @@ static const struct snd_soc_component_driver adau7002_component_driver = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int adau7002_probe(struct platform_device *pdev) diff --git a/sound/soc/codecs/adau7118.c b/sound/soc/codecs/adau7118.c index 841229dcbca1..bbb097249887 100644 --- a/sound/soc/codecs/adau7118.c +++ b/sound/soc/codecs/adau7118.c @@ -442,7 +442,6 @@ static const struct snd_soc_component_driver adau7118_component_driver = { .num_dapm_widgets = ARRAY_SIZE(adau7118_widgets), .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static void adau7118_regulator_disable(void *data) diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index 90f3a5e9e31f..fcff35f26cec 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -842,7 +842,6 @@ static const struct snd_soc_component_driver adav80x_component_driver = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; int adav80x_bus_probe(struct device *dev, struct regmap *regmap) -- cgit v1.2.3 From 410e73a5338d72c31a32a50c1629d81d8ce6a71f Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:52:01 +0100 Subject: ASoC: ak*: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-48-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/ak4104.c | 1 - sound/soc/codecs/ak4118.c | 1 - sound/soc/codecs/ak4375.c | 1 - sound/soc/codecs/ak4458.c | 2 -- sound/soc/codecs/ak4535.c | 1 - sound/soc/codecs/ak4554.c | 1 - sound/soc/codecs/ak4613.c | 1 - sound/soc/codecs/ak4641.c | 1 - sound/soc/codecs/ak4642.c | 1 - sound/soc/codecs/ak4671.c | 1 - sound/soc/codecs/ak5386.c | 1 - sound/soc/codecs/ak5558.c | 2 -- 12 files changed, 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index dc4747c77a7a..ce99f30b4613 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -248,7 +248,6 @@ static const struct snd_soc_component_driver soc_component_device_ak4104 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config ak4104_regmap = { diff --git a/sound/soc/codecs/ak4118.c b/sound/soc/codecs/ak4118.c index 5c4a78c16733..b6d9a10bdccd 100644 --- a/sound/soc/codecs/ak4118.c +++ b/sound/soc/codecs/ak4118.c @@ -342,7 +342,6 @@ static const struct snd_soc_component_driver soc_component_drv_ak4118 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config ak4118_regmap = { diff --git a/sound/soc/codecs/ak4375.c b/sound/soc/codecs/ak4375.c index 9a7b662016b9..1ed004ba7cd2 100644 --- a/sound/soc/codecs/ak4375.c +++ b/sound/soc/codecs/ak4375.c @@ -473,7 +473,6 @@ static const struct snd_soc_component_driver soc_codec_dev_ak4375 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config ak4375_regmap = { diff --git a/sound/soc/codecs/ak4458.c b/sound/soc/codecs/ak4458.c index baa9ff5d0ce5..ea33cc83c86c 100644 --- a/sound/soc/codecs/ak4458.c +++ b/sound/soc/codecs/ak4458.c @@ -725,7 +725,6 @@ static const struct snd_soc_component_driver soc_codec_dev_ak4458 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct snd_soc_component_driver soc_codec_dev_ak4497 = { @@ -740,7 +739,6 @@ static const struct snd_soc_component_driver soc_codec_dev_ak4497 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config ak4458_regmap = { diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index cc803e730c6e..8c8c93eea704 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -402,7 +402,6 @@ static const struct snd_soc_component_driver soc_component_dev_ak4535 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int ak4535_i2c_probe(struct i2c_client *i2c) diff --git a/sound/soc/codecs/ak4554.c b/sound/soc/codecs/ak4554.c index 8e60e2b56ad6..b9607de5a191 100644 --- a/sound/soc/codecs/ak4554.c +++ b/sound/soc/codecs/ak4554.c @@ -67,7 +67,6 @@ static const struct snd_soc_component_driver soc_component_dev_ak4554 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int ak4554_soc_probe(struct platform_device *pdev) diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c index d29d5e0db168..f75c19ef3551 100644 --- a/sound/soc/codecs/ak4613.c +++ b/sound/soc/codecs/ak4613.c @@ -827,7 +827,6 @@ static const struct snd_soc_component_driver soc_component_dev_ak4613 = { .num_dapm_routes = ARRAY_SIZE(ak4613_intercon), .idle_bias_on = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static void ak4613_parse_of(struct ak4613_priv *priv, diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c index d8d9cc712d67..88851e94b045 100644 --- a/sound/soc/codecs/ak4641.c +++ b/sound/soc/codecs/ak4641.c @@ -535,7 +535,6 @@ static const struct snd_soc_component_driver soc_component_dev_ak4641 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config ak4641_regmap = { diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 3c20ff5595eb..914d5b1969f8 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -559,7 +559,6 @@ static const struct snd_soc_component_driver soc_component_dev_ak4642 = { .num_dapm_routes = ARRAY_SIZE(ak4642_intercon), .idle_bias_on = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config ak4642_regmap = { diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c index 60edcbe56014..cd76765f8cc0 100644 --- a/sound/soc/codecs/ak4671.c +++ b/sound/soc/codecs/ak4671.c @@ -616,7 +616,6 @@ static const struct snd_soc_component_driver soc_component_dev_ak4671 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config ak4671_regmap = { diff --git a/sound/soc/codecs/ak5386.c b/sound/soc/codecs/ak5386.c index c76bfff24602..0c5e00679c7d 100644 --- a/sound/soc/codecs/ak5386.c +++ b/sound/soc/codecs/ak5386.c @@ -77,7 +77,6 @@ static const struct snd_soc_component_driver soc_component_ak5386 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int ak5386_set_dai_fmt(struct snd_soc_dai *codec_dai, diff --git a/sound/soc/codecs/ak5558.c b/sound/soc/codecs/ak5558.c index c94cfde3e4a8..887d2c04d647 100644 --- a/sound/soc/codecs/ak5558.c +++ b/sound/soc/codecs/ak5558.c @@ -393,7 +393,6 @@ static const struct snd_soc_component_driver soc_codec_dev_ak5558 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct snd_soc_component_driver soc_codec_dev_ak5552 = { @@ -408,7 +407,6 @@ static const struct snd_soc_component_driver soc_codec_dev_ak5552 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config ak5558_regmap = { -- cgit v1.2.3 From 60d28b5c47c7f02bb52fc5e52a84d669b9b54dbc Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:52:02 +0100 Subject: ASoC: alc*: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-49-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/alc5623.c | 1 - sound/soc/codecs/alc5632.c | 1 - 2 files changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index 8e6235d2c544..9ef01b1dd294 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -956,7 +956,6 @@ static const struct snd_soc_component_driver soc_component_device_alc5623 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config alc5623_regmap = { diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index 641bdfddae16..a770704a4e17 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -1078,7 +1078,6 @@ static const struct snd_soc_component_driver soc_component_device_alc5632 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config alc5632_regmap = { -- cgit v1.2.3 From ff946fd98bffe5de450047f54a27492827186b75 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:52:03 +0100 Subject: ASoC: cs*: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-50-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs35l32.c | 1 - sound/soc/codecs/cs35l33.c | 1 - sound/soc/codecs/cs35l34.c | 1 - sound/soc/codecs/cs35l35.c | 1 - sound/soc/codecs/cs35l36.c | 1 - sound/soc/codecs/cs4234.c | 1 - sound/soc/codecs/cs4265.c | 1 - sound/soc/codecs/cs4270.c | 1 - sound/soc/codecs/cs4271.c | 1 - sound/soc/codecs/cs42l42.c | 1 - sound/soc/codecs/cs42l51.c | 1 - sound/soc/codecs/cs42l52.c | 1 - sound/soc/codecs/cs42l56.c | 1 - sound/soc/codecs/cs42l73.c | 1 - sound/soc/codecs/cs42xx8.c | 1 - sound/soc/codecs/cs43130.c | 1 - sound/soc/codecs/cs4341.c | 1 - sound/soc/codecs/cs4349.c | 1 - sound/soc/codecs/cs47l15.c | 1 - sound/soc/codecs/cs47l24.c | 1 - sound/soc/codecs/cs47l35.c | 1 - sound/soc/codecs/cs47l85.c | 1 - sound/soc/codecs/cs47l90.c | 1 - sound/soc/codecs/cs47l92.c | 1 - sound/soc/codecs/cs53l30.c | 1 - 25 files changed, 25 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs35l32.c b/sound/soc/codecs/cs35l32.c index badfc55bc5fa..8ff6f66be86f 100644 --- a/sound/soc/codecs/cs35l32.c +++ b/sound/soc/codecs/cs35l32.c @@ -236,7 +236,6 @@ static const struct snd_soc_component_driver soc_component_dev_cs35l32 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; /* Current and threshold powerup sequence Pg37 in datasheet */ diff --git a/sound/soc/codecs/cs35l33.c b/sound/soc/codecs/cs35l33.c index 47dc0f6d90a2..082025fa0370 100644 --- a/sound/soc/codecs/cs35l33.c +++ b/sound/soc/codecs/cs35l33.c @@ -840,7 +840,6 @@ static const struct snd_soc_component_driver soc_component_dev_cs35l33 = { .num_dapm_routes = ARRAY_SIZE(cs35l33_audio_map), .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config cs35l33_regmap = { diff --git a/sound/soc/codecs/cs35l34.c b/sound/soc/codecs/cs35l34.c index 50d509a06071..472ac982779b 100644 --- a/sound/soc/codecs/cs35l34.c +++ b/sound/soc/codecs/cs35l34.c @@ -787,7 +787,6 @@ static const struct snd_soc_component_driver soc_component_dev_cs35l34 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static struct regmap_config cs35l34_regmap = { diff --git a/sound/soc/codecs/cs35l35.c b/sound/soc/codecs/cs35l35.c index 6b70afb70a67..714a759dca21 100644 --- a/sound/soc/codecs/cs35l35.c +++ b/sound/soc/codecs/cs35l35.c @@ -1087,7 +1087,6 @@ static const struct snd_soc_component_driver soc_component_dev_cs35l35 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static struct regmap_config cs35l35_regmap = { diff --git a/sound/soc/codecs/cs35l36.c b/sound/soc/codecs/cs35l36.c index dfe85dc2cd20..4dc13e6f4874 100644 --- a/sound/soc/codecs/cs35l36.c +++ b/sound/soc/codecs/cs35l36.c @@ -1300,7 +1300,6 @@ static const struct snd_soc_component_driver soc_component_dev_cs35l36 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static struct regmap_config cs35l36_regmap = { diff --git a/sound/soc/codecs/cs4234.c b/sound/soc/codecs/cs4234.c index 881c5ba70c0e..b49a3cf21ebe 100644 --- a/sound/soc/codecs/cs4234.c +++ b/sound/soc/codecs/cs4234.c @@ -660,7 +660,6 @@ static const struct snd_soc_component_driver soc_component_cs4234 = { .controls = cs4234_snd_controls, .num_controls = ARRAY_SIZE(cs4234_snd_controls), .set_bias_level = cs4234_set_bias_level, - .non_legacy_dai_naming = 1, .idle_bias_on = 1, .suspend_bias_off = 1, .endianness = 1, diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index 86bfa8d5ec78..76c19802d5fe 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -553,7 +553,6 @@ static const struct snd_soc_component_driver soc_component_cs4265 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config cs4265_regmap = { diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 97d26b9e8f7f..ba67e43edf35 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -619,7 +619,6 @@ static const struct snd_soc_component_driver soc_component_device_cs4270 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; /* diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 7663f89ac6a2..2021cf442606 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -642,7 +642,6 @@ static const struct snd_soc_component_driver soc_component_dev_cs4271 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int cs4271_common_probe(struct device *dev, diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index 6ca74c0d46ea..d545a593a251 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -581,7 +581,6 @@ static const struct snd_soc_component_driver soc_component_dev_cs42l42 = { .num_controls = ARRAY_SIZE(cs42l42_snd_controls), .idle_bias_on = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; /* Switch to SCLK. Atomic delay after the write to allow the switch to complete. */ diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 0e933181b5db..51721edd8f53 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -600,7 +600,6 @@ static const struct snd_soc_component_driver soc_component_device_cs42l51 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static bool cs42l51_writeable_reg(struct device *dev, unsigned int reg) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 10e696406a71..90bf535fc5a5 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -1061,7 +1061,6 @@ static const struct snd_soc_component_driver soc_component_dev_cs42l52 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; /* Current and threshold powerup sequence Pg37 */ diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index 510c94265b1f..03e2540a0ba1 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -1114,7 +1114,6 @@ static const struct snd_soc_component_driver soc_component_dev_cs42l56 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config cs42l56_regmap = { diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 5a9166289f36..0a146319755a 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1256,7 +1256,6 @@ static const struct snd_soc_component_driver soc_component_dev_cs42l73 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config cs42l73_regmap = { diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c index 5d6ef660f851..d14eb2f6e1dd 100644 --- a/sound/soc/codecs/cs42xx8.c +++ b/sound/soc/codecs/cs42xx8.c @@ -497,7 +497,6 @@ static const struct snd_soc_component_driver cs42xx8_driver = { .num_dapm_routes = ARRAY_SIZE(cs42xx8_dapm_routes), .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; const struct cs42xx8_driver_data cs42448_data = { diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c index a2bce0f9f247..ca4d47cc9c91 100644 --- a/sound/soc/codecs/cs43130.c +++ b/sound/soc/codecs/cs43130.c @@ -2345,7 +2345,6 @@ static struct snd_soc_component_driver soc_component_dev_cs43130 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config cs43130_regmap = { diff --git a/sound/soc/codecs/cs4341.c b/sound/soc/codecs/cs4341.c index 8ac043f1aae0..ac1696034846 100644 --- a/sound/soc/codecs/cs4341.c +++ b/sound/soc/codecs/cs4341.c @@ -202,7 +202,6 @@ static const struct snd_soc_component_driver soc_component_cs4341 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct of_device_id __maybe_unused cs4341_dt_ids[] = { diff --git a/sound/soc/codecs/cs4349.c b/sound/soc/codecs/cs4349.c index 7069e9b54857..f7c5c2fd4304 100644 --- a/sound/soc/codecs/cs4349.c +++ b/sound/soc/codecs/cs4349.c @@ -260,7 +260,6 @@ static const struct snd_soc_component_driver soc_component_dev_cs4349 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config cs4349_regmap = { diff --git a/sound/soc/codecs/cs47l15.c b/sound/soc/codecs/cs47l15.c index 391fd7da331f..0193173b8637 100644 --- a/sound/soc/codecs/cs47l15.c +++ b/sound/soc/codecs/cs47l15.c @@ -1353,7 +1353,6 @@ static const struct snd_soc_component_driver soc_component_dev_cs47l15 = { .num_dapm_routes = ARRAY_SIZE(cs47l15_dapm_routes), .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int cs47l15_probe(struct platform_device *pdev) diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c index 6356f81aafc5..f9a2b865d717 100644 --- a/sound/soc/codecs/cs47l24.c +++ b/sound/soc/codecs/cs47l24.c @@ -1203,7 +1203,6 @@ static const struct snd_soc_component_driver soc_component_dev_cs47l24 = { .num_dapm_routes = ARRAY_SIZE(cs47l24_dapm_routes), .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int cs47l24_probe(struct platform_device *pdev) diff --git a/sound/soc/codecs/cs47l35.c b/sound/soc/codecs/cs47l35.c index db2f844b8b17..c1032d6c9143 100644 --- a/sound/soc/codecs/cs47l35.c +++ b/sound/soc/codecs/cs47l35.c @@ -1638,7 +1638,6 @@ static const struct snd_soc_component_driver soc_component_dev_cs47l35 = { .num_dapm_routes = ARRAY_SIZE(cs47l35_dapm_routes), .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int cs47l35_probe(struct platform_device *pdev) diff --git a/sound/soc/codecs/cs47l85.c b/sound/soc/codecs/cs47l85.c index d4fedc5ad516..215d8211aa59 100644 --- a/sound/soc/codecs/cs47l85.c +++ b/sound/soc/codecs/cs47l85.c @@ -2582,7 +2582,6 @@ static const struct snd_soc_component_driver soc_component_dev_cs47l85 = { .num_dapm_routes = ARRAY_SIZE(cs47l85_dapm_routes), .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int cs47l85_probe(struct platform_device *pdev) diff --git a/sound/soc/codecs/cs47l90.c b/sound/soc/codecs/cs47l90.c index 5aec937a2462..1ad6526c7871 100644 --- a/sound/soc/codecs/cs47l90.c +++ b/sound/soc/codecs/cs47l90.c @@ -2497,7 +2497,6 @@ static const struct snd_soc_component_driver soc_component_dev_cs47l90 = { .num_dapm_routes = ARRAY_SIZE(cs47l90_dapm_routes), .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int cs47l90_probe(struct platform_device *pdev) diff --git a/sound/soc/codecs/cs47l92.c b/sound/soc/codecs/cs47l92.c index a1b8dcdb9f7b..59da34b480a8 100644 --- a/sound/soc/codecs/cs47l92.c +++ b/sound/soc/codecs/cs47l92.c @@ -1958,7 +1958,6 @@ static const struct snd_soc_component_driver soc_component_dev_cs47l92 = { .num_dapm_routes = ARRAY_SIZE(cs47l92_dapm_routes), .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int cs47l92_probe(struct platform_device *pdev) diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c index 360ca2ffd506..8796d8e84b7a 100644 --- a/sound/soc/codecs/cs53l30.c +++ b/sound/soc/codecs/cs53l30.c @@ -899,7 +899,6 @@ static const struct snd_soc_component_driver cs53l30_driver = { .num_dapm_routes = ARRAY_SIZE(cs53l30_dapm_routes), .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static struct regmap_config cs53l30_regmap = { -- cgit v1.2.3 From c03a5b4c419799676013cb0c58c03e00ebe21a61 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:52:04 +0100 Subject: ASoC: da*: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-51-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/da7210.c | 1 - sound/soc/codecs/da7213.c | 1 - sound/soc/codecs/da7218.c | 1 - sound/soc/codecs/da7219.c | 1 - sound/soc/codecs/da732x.c | 1 - sound/soc/codecs/da9055.c | 1 - 6 files changed, 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 76a21976ccdd..f838466bfebf 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -1173,7 +1173,6 @@ static const struct snd_soc_component_driver soc_component_dev_da7210 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; #if IS_ENABLED(CONFIG_I2C) diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index 2e645dc60eda..544ccbcfc884 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -1922,7 +1922,6 @@ static const struct snd_soc_component_driver soc_component_dev_da7213 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config da7213_regmap_config = { diff --git a/sound/soc/codecs/da7218.c b/sound/soc/codecs/da7218.c index a5d7c350a3de..91372909d184 100644 --- a/sound/soc/codecs/da7218.c +++ b/sound/soc/codecs/da7218.c @@ -3040,7 +3040,6 @@ static const struct snd_soc_component_driver soc_component_dev_da7218 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index c18f76f370fc..50ecf30e6136 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -2647,7 +2647,6 @@ static const struct snd_soc_component_driver soc_component_dev_da7219 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index 3f1cfee10df3..2c5b0b74201c 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -1503,7 +1503,6 @@ static const struct snd_soc_component_driver soc_component_dev_da732x = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int da732x_i2c_probe(struct i2c_client *i2c) diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c index 9d8c8adc5d76..28043b4530df 100644 --- a/sound/soc/codecs/da9055.c +++ b/sound/soc/codecs/da9055.c @@ -1460,7 +1460,6 @@ static const struct snd_soc_component_driver soc_component_dev_da9055 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config da9055_regmap_config = { -- cgit v1.2.3 From f0b163b4d5a215f610bd64eb8ab8a8906e53bec6 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:52:05 +0100 Subject: ASoC: es*: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-52-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/es7134.c | 1 - sound/soc/codecs/es7241.c | 1 - sound/soc/codecs/es8316.c | 1 - sound/soc/codecs/es8328.c | 1 - 4 files changed, 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/es7134.c b/sound/soc/codecs/es7134.c index f443351677df..f5150d2f95da 100644 --- a/sound/soc/codecs/es7134.c +++ b/sound/soc/codecs/es7134.c @@ -213,7 +213,6 @@ static const struct snd_soc_component_driver es7134_component_driver = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static struct snd_soc_dai_driver es7154_dai = { diff --git a/sound/soc/codecs/es7241.c b/sound/soc/codecs/es7241.c index 0baa86241cf9..339553cfbb48 100644 --- a/sound/soc/codecs/es7241.c +++ b/sound/soc/codecs/es7241.c @@ -232,7 +232,6 @@ static const struct snd_soc_component_driver es7241_component_driver = { .num_dapm_routes = ARRAY_SIZE(es7241_dapm_routes), .idle_bias_on = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static void es7241_parse_fmt(struct device *dev, struct es7241_data *priv) diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index 4407166bb338..eb15be9095e7 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -769,7 +769,6 @@ static const struct snd_soc_component_driver soc_component_dev_es8316 = { .num_dapm_routes = ARRAY_SIZE(es8316_dapm_routes), .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_range es8316_volatile_ranges[] = { diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index dd53dfd87b04..160adc706cc6 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -844,7 +844,6 @@ static const struct snd_soc_component_driver es8328_component_driver = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; int es8328_probe(struct device *dev, struct regmap *regmap) -- cgit v1.2.3 From d2d3219ebe568fe4ee90ac748939304f7e05a8ec Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:52:06 +0100 Subject: ASoC: max*: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-53-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/max98088.c | 1 - sound/soc/codecs/max98090.c | 1 - sound/soc/codecs/max98095.c | 1 - sound/soc/codecs/max98357a.c | 1 - sound/soc/codecs/max98371.c | 1 - sound/soc/codecs/max98373.c | 2 -- sound/soc/codecs/max98390.c | 1 - sound/soc/codecs/max98396.c | 2 -- sound/soc/codecs/max9850.c | 1 - sound/soc/codecs/max98520.c | 1 - sound/soc/codecs/max9860.c | 1 - sound/soc/codecs/max9867.c | 1 - sound/soc/codecs/max98925.c | 1 - sound/soc/codecs/max98926.c | 1 - sound/soc/codecs/max98927.c | 1 - 15 files changed, 17 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 5ef2e1279ee7..08e5c606ff27 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -1734,7 +1734,6 @@ static const struct snd_soc_component_driver soc_component_dev_max98088 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct i2c_device_id max98088_i2c_id[] = { diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 72471cdb2229..142083b13ac3 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2519,7 +2519,6 @@ static const struct snd_soc_component_driver soc_component_dev_max98090 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config max98090_regmap = { diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 7bca99fa61b5..44aa58fcc23f 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -2103,7 +2103,6 @@ static const struct snd_soc_component_driver soc_component_dev_max98095 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct i2c_device_id max98095_i2c_id[] = { diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index 918812763884..2a2b286f1747 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -93,7 +93,6 @@ static const struct snd_soc_component_driver max98357a_component_driver = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct snd_soc_dai_ops max98357a_dai_ops = { diff --git a/sound/soc/codecs/max98371.c b/sound/soc/codecs/max98371.c index 800f2bca6a0f..bac9d1bcf60a 100644 --- a/sound/soc/codecs/max98371.c +++ b/sound/soc/codecs/max98371.c @@ -351,7 +351,6 @@ static const struct snd_soc_component_driver max98371_component = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config max98371_regmap = { diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index 1517c47afbf1..f90a6a7ba83b 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -437,7 +437,6 @@ const struct snd_soc_component_driver soc_codec_dev_max98373 = { .num_dapm_routes = ARRAY_SIZE(max98373_audio_map), .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; EXPORT_SYMBOL_GPL(soc_codec_dev_max98373); @@ -462,7 +461,6 @@ const struct snd_soc_component_driver soc_codec_dev_max98373_sdw = { .num_dapm_routes = ARRAY_SIZE(max98373_audio_map), .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; EXPORT_SYMBOL_GPL(soc_codec_dev_max98373_sdw); diff --git a/sound/soc/codecs/max98390.c b/sound/soc/codecs/max98390.c index d83f81d9ff4e..5c08166a8dc6 100644 --- a/sound/soc/codecs/max98390.c +++ b/sound/soc/codecs/max98390.c @@ -983,7 +983,6 @@ static const struct snd_soc_component_driver soc_codec_dev_max98390 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config max98390_regmap = { diff --git a/sound/soc/codecs/max98396.c b/sound/soc/codecs/max98396.c index 56eb62bb041f..225effede9d2 100644 --- a/sound/soc/codecs/max98396.c +++ b/sound/soc/codecs/max98396.c @@ -1453,7 +1453,6 @@ static const struct snd_soc_component_driver soc_codec_dev_max98396 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct snd_soc_component_driver soc_codec_dev_max98397 = { @@ -1467,7 +1466,6 @@ static const struct snd_soc_component_driver soc_codec_dev_max98397 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config max98396_regmap = { diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c index 9ca6fc254883..a6733396b0ca 100644 --- a/sound/soc/codecs/max9850.c +++ b/sound/soc/codecs/max9850.c @@ -296,7 +296,6 @@ static const struct snd_soc_component_driver soc_component_dev_max9850 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int max9850_i2c_probe(struct i2c_client *i2c) diff --git a/sound/soc/codecs/max98520.c b/sound/soc/codecs/max98520.c index f0f085ecab55..5edd6f90f8a7 100644 --- a/sound/soc/codecs/max98520.c +++ b/sound/soc/codecs/max98520.c @@ -657,7 +657,6 @@ static const struct snd_soc_component_driver soc_codec_dev_max98520 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config max98520_regmap = { diff --git a/sound/soc/codecs/max9860.c b/sound/soc/codecs/max9860.c index a1d0179e12c7..771b3dcd6cc3 100644 --- a/sound/soc/codecs/max9860.c +++ b/sound/soc/codecs/max9860.c @@ -537,7 +537,6 @@ static const struct snd_soc_component_driver max9860_component_driver = { .num_dapm_routes = ARRAY_SIZE(max9860_dapm_routes), .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; #ifdef CONFIG_PM diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c index eb628b7e84f5..6d2941a9dbd6 100644 --- a/sound/soc/codecs/max9867.c +++ b/sound/soc/codecs/max9867.c @@ -589,7 +589,6 @@ static const struct snd_soc_component_driver max9867_component = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static bool max9867_volatile_register(struct device *dev, unsigned int reg) diff --git a/sound/soc/codecs/max98925.c b/sound/soc/codecs/max98925.c index 63849ebcfd35..c24d9f2c8874 100644 --- a/sound/soc/codecs/max98925.c +++ b/sound/soc/codecs/max98925.c @@ -544,7 +544,6 @@ static const struct snd_soc_component_driver soc_component_dev_max98925 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config max98925_regmap = { diff --git a/sound/soc/codecs/max98926.c b/sound/soc/codecs/max98926.c index 56e0a87c7112..bffd56e240e9 100644 --- a/sound/soc/codecs/max98926.c +++ b/sound/soc/codecs/max98926.c @@ -496,7 +496,6 @@ static const struct snd_soc_component_driver soc_component_dev_max98926 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config max98926_regmap = { diff --git a/sound/soc/codecs/max98927.c b/sound/soc/codecs/max98927.c index b7cff76d7b5b..9cce7c0f0142 100644 --- a/sound/soc/codecs/max98927.c +++ b/sound/soc/codecs/max98927.c @@ -832,7 +832,6 @@ static const struct snd_soc_component_driver soc_component_dev_max98927 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config max98927_regmap = { -- cgit v1.2.3 From 736f48714c1b85b0b1f6c88af07989a5828531c9 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:52:07 +0100 Subject: ASoC: msm*: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-54-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/msm8916-wcd-analog.c | 1 - sound/soc/codecs/msm8916-wcd-digital.c | 1 - 2 files changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c index e52a559c52d6..78e543eb3c83 100644 --- a/sound/soc/codecs/msm8916-wcd-analog.c +++ b/sound/soc/codecs/msm8916-wcd-analog.c @@ -1128,7 +1128,6 @@ static const struct snd_soc_component_driver pm8916_wcd_analog = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int pm8916_wcd_analog_parse_dt(struct device *dev, diff --git a/sound/soc/codecs/msm8916-wcd-digital.c b/sound/soc/codecs/msm8916-wcd-digital.c index 098a58990f07..d490a0f18675 100644 --- a/sound/soc/codecs/msm8916-wcd-digital.c +++ b/sound/soc/codecs/msm8916-wcd-digital.c @@ -1155,7 +1155,6 @@ static const struct snd_soc_component_driver msm8916_wcd_digital = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config msm8916_codec_regmap_config = { -- cgit v1.2.3 From c2fd88f0029172679917ebc536cfdc4b8fabe168 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:52:08 +0100 Subject: ASoC: nau*: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-55-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/nau8315.c | 1 - sound/soc/codecs/nau8540.c | 1 - sound/soc/codecs/nau8810.c | 1 - sound/soc/codecs/nau8821.c | 1 - sound/soc/codecs/nau8822.c | 1 - sound/soc/codecs/nau8824.c | 1 - sound/soc/codecs/nau8825.c | 1 - 7 files changed, 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/nau8315.c b/sound/soc/codecs/nau8315.c index 2b66e3f7a8b7..ad4dce9e5080 100644 --- a/sound/soc/codecs/nau8315.c +++ b/sound/soc/codecs/nau8315.c @@ -93,7 +93,6 @@ static const struct snd_soc_component_driver nau8315_component_driver = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct snd_soc_dai_ops nau8315_dai_ops = { diff --git a/sound/soc/codecs/nau8540.c b/sound/soc/codecs/nau8540.c index 347c715e22a4..58f70a02f18a 100644 --- a/sound/soc/codecs/nau8540.c +++ b/sound/soc/codecs/nau8540.c @@ -806,7 +806,6 @@ static const struct snd_soc_component_driver nau8540_component_driver = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config nau8540_regmap_config = { diff --git a/sound/soc/codecs/nau8810.c b/sound/soc/codecs/nau8810.c index 7b3b1e4ac246..ccb512c21d74 100644 --- a/sound/soc/codecs/nau8810.c +++ b/sound/soc/codecs/nau8810.c @@ -866,7 +866,6 @@ static const struct snd_soc_component_driver nau8810_component_driver = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int nau8810_i2c_probe(struct i2c_client *i2c) diff --git a/sound/soc/codecs/nau8821.c b/sound/soc/codecs/nau8821.c index ce4e7f46bb06..6453e93678d2 100644 --- a/sound/soc/codecs/nau8821.c +++ b/sound/soc/codecs/nau8821.c @@ -1430,7 +1430,6 @@ static const struct snd_soc_component_driver nau8821_component_driver = { .dapm_routes = nau8821_dapm_routes, .num_dapm_routes = ARRAY_SIZE(nau8821_dapm_routes), .suspend_bias_off = 1, - .non_legacy_dai_naming = 1, .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, diff --git a/sound/soc/codecs/nau8822.c b/sound/soc/codecs/nau8822.c index 3907d1dd8cee..1aef281a9972 100644 --- a/sound/soc/codecs/nau8822.c +++ b/sound/soc/codecs/nau8822.c @@ -1083,7 +1083,6 @@ static const struct snd_soc_component_driver soc_component_dev_nau8822 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config nau8822_regmap_config = { diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index 2a7c93508535..ad54d70f7d8e 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -1544,7 +1544,6 @@ static const struct snd_soc_component_driver nau8824_component_driver = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct snd_soc_dai_ops nau8824_dai_ops = { diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 20e45a337b8f..907ec88c759a 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -2478,7 +2478,6 @@ static const struct snd_soc_component_driver nau8825_component_driver = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static void nau8825_reset_chip(struct regmap *regmap) -- cgit v1.2.3 From 8d4470b8d08b4aab5136cc3265eb0b05d2a1c72d Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:52:09 +0100 Subject: ASoC: pcm*: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-56-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/pcm1681.c | 1 - sound/soc/codecs/pcm1789.c | 1 - sound/soc/codecs/pcm179x.c | 1 - sound/soc/codecs/pcm186x.c | 2 -- sound/soc/codecs/pcm3008.c | 1 - sound/soc/codecs/pcm3168a.c | 1 - sound/soc/codecs/pcm5102a.c | 1 - sound/soc/codecs/pcm512x.c | 1 - 8 files changed, 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c index 20eb04c8a41a..3591f6f53901 100644 --- a/sound/soc/codecs/pcm1681.c +++ b/sound/soc/codecs/pcm1681.c @@ -290,7 +290,6 @@ static const struct snd_soc_component_driver soc_component_dev_pcm1681 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct i2c_device_id pcm1681_i2c_id[] = { diff --git a/sound/soc/codecs/pcm1789.c b/sound/soc/codecs/pcm1789.c index 35788b57e11f..3ab381e9a856 100644 --- a/sound/soc/codecs/pcm1789.c +++ b/sound/soc/codecs/pcm1789.c @@ -229,7 +229,6 @@ static const struct snd_soc_component_driver soc_component_dev_pcm1789 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; int pcm1789_common_init(struct device *dev, struct regmap *regmap) diff --git a/sound/soc/codecs/pcm179x.c b/sound/soc/codecs/pcm179x.c index ee60373d7d25..f52ff66b6e64 100644 --- a/sound/soc/codecs/pcm179x.c +++ b/sound/soc/codecs/pcm179x.c @@ -207,7 +207,6 @@ static const struct snd_soc_component_driver soc_component_dev_pcm179x = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; int pcm179x_common_init(struct device *dev, struct regmap *regmap) diff --git a/sound/soc/codecs/pcm186x.c b/sound/soc/codecs/pcm186x.c index fda9d7ee3fe6..dd21803ba13c 100644 --- a/sound/soc/codecs/pcm186x.c +++ b/sound/soc/codecs/pcm186x.c @@ -578,7 +578,6 @@ static struct snd_soc_component_driver soc_codec_dev_pcm1863 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static struct snd_soc_component_driver soc_codec_dev_pcm1865 = { @@ -593,7 +592,6 @@ static struct snd_soc_component_driver soc_codec_dev_pcm1865 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static bool pcm186x_volatile(struct device *dev, unsigned int reg) diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c index aef40ec40aa1..09c6c1326833 100644 --- a/sound/soc/codecs/pcm3008.c +++ b/sound/soc/codecs/pcm3008.c @@ -102,7 +102,6 @@ static const struct snd_soc_component_driver soc_component_dev_pcm3008 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int pcm3008_codec_probe(struct platform_device *pdev) diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index cf27f05dc46a..9d6431338fb7 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -716,7 +716,6 @@ static const struct snd_soc_component_driver pcm3168a_driver = { .num_dapm_routes = ARRAY_SIZE(pcm3168a_dapm_routes), .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; int pcm3168a_probe(struct device *dev, struct regmap *regmap) diff --git a/sound/soc/codecs/pcm5102a.c b/sound/soc/codecs/pcm5102a.c index f39f98bbc97f..3401a25341e6 100644 --- a/sound/soc/codecs/pcm5102a.c +++ b/sound/soc/codecs/pcm5102a.c @@ -28,7 +28,6 @@ static struct snd_soc_component_driver soc_component_dev_pcm5102a = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int pcm5102a_probe(struct platform_device *pdev) diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index a3ff4a07aff7..767463e82665 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -1512,7 +1512,6 @@ static const struct snd_soc_component_driver pcm512x_component_driver = { .num_dapm_routes = ARRAY_SIZE(pcm512x_dapm_routes), .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_range_cfg pcm512x_range = { -- cgit v1.2.3 From a524837ddd11bc20ec59d033d0260707cfa3cb99 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:52:10 +0100 Subject: ASoC: rt*: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-57-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt1011.c | 1 - sound/soc/codecs/rt1015.c | 1 - sound/soc/codecs/rt1015p.c | 1 - sound/soc/codecs/rt1016.c | 1 - sound/soc/codecs/rt1019.c | 1 - sound/soc/codecs/rt1305.c | 1 - sound/soc/codecs/rt1308.c | 1 - sound/soc/codecs/rt274.c | 1 - sound/soc/codecs/rt286.c | 1 - sound/soc/codecs/rt298.c | 1 - sound/soc/codecs/rt5514.c | 1 - sound/soc/codecs/rt5616.c | 1 - sound/soc/codecs/rt5631.c | 1 - sound/soc/codecs/rt5640.c | 2 -- sound/soc/codecs/rt5645.c | 1 - sound/soc/codecs/rt5651.c | 1 - sound/soc/codecs/rt5659.c | 1 - sound/soc/codecs/rt5660.c | 1 - sound/soc/codecs/rt5663.c | 1 - sound/soc/codecs/rt5665.c | 1 - sound/soc/codecs/rt5668.c | 1 - sound/soc/codecs/rt5670.c | 1 - sound/soc/codecs/rt5677.c | 1 - sound/soc/codecs/rt5682.c | 1 - sound/soc/codecs/rt5682s.c | 1 - 25 files changed, 26 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index 08dbaef84d4e..c1568216126e 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -2176,7 +2176,6 @@ static const struct snd_soc_component_driver soc_component_dev_rt1011 = { .set_pll = rt1011_set_component_pll, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config rt1011_regmap = { diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index 7a06f2654afb..57d0f1c69e46 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -1071,7 +1071,6 @@ static const struct snd_soc_component_driver soc_component_dev_rt1015 = { .set_pll = rt1015_set_component_pll, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config rt1015_regmap = { diff --git a/sound/soc/codecs/rt1015p.c b/sound/soc/codecs/rt1015p.c index 415cfb3b2f0d..06800dad8798 100644 --- a/sound/soc/codecs/rt1015p.c +++ b/sound/soc/codecs/rt1015p.c @@ -89,7 +89,6 @@ static const struct snd_soc_component_driver rt1015p_component_driver = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static struct snd_soc_dai_driver rt1015p_dai_driver = { diff --git a/sound/soc/codecs/rt1016.c b/sound/soc/codecs/rt1016.c index e31c4736627f..37eeec650f03 100644 --- a/sound/soc/codecs/rt1016.c +++ b/sound/soc/codecs/rt1016.c @@ -595,7 +595,6 @@ static const struct snd_soc_component_driver soc_component_dev_rt1016 = { .set_pll = rt1016_set_component_pll, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config rt1016_regmap = { diff --git a/sound/soc/codecs/rt1019.c b/sound/soc/codecs/rt1019.c index f3f15fbe85d0..b66bfecbb879 100644 --- a/sound/soc/codecs/rt1019.c +++ b/sound/soc/codecs/rt1019.c @@ -522,7 +522,6 @@ static const struct snd_soc_component_driver soc_component_dev_rt1019 = { .num_dapm_widgets = ARRAY_SIZE(rt1019_dapm_widgets), .dapm_routes = rt1019_dapm_routes, .num_dapm_routes = ARRAY_SIZE(rt1019_dapm_routes), - .non_legacy_dai_naming = 1, .endianness = 1, }; diff --git a/sound/soc/codecs/rt1305.c b/sound/soc/codecs/rt1305.c index 58d97e3c5087..5b39a440b6dc 100644 --- a/sound/soc/codecs/rt1305.c +++ b/sound/soc/codecs/rt1305.c @@ -946,7 +946,6 @@ static const struct snd_soc_component_driver soc_component_dev_rt1305 = { .set_pll = rt1305_set_component_pll, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config rt1305_regmap = { diff --git a/sound/soc/codecs/rt1308.c b/sound/soc/codecs/rt1308.c index eec2b1760408..d2a8e9fe3e23 100644 --- a/sound/soc/codecs/rt1308.c +++ b/sound/soc/codecs/rt1308.c @@ -765,7 +765,6 @@ static const struct snd_soc_component_driver soc_component_dev_rt1308 = { .set_pll = rt1308_set_component_pll, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config rt1308_regmap = { diff --git a/sound/soc/codecs/rt274.c b/sound/soc/codecs/rt274.c index a5615e94ec7d..6b208f9eb503 100644 --- a/sound/soc/codecs/rt274.c +++ b/sound/soc/codecs/rt274.c @@ -1072,7 +1072,6 @@ static const struct snd_soc_component_driver soc_component_dev_rt274 = { .num_dapm_routes = ARRAY_SIZE(rt274_dapm_routes), .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config rt274_regmap = { diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 0534a073ee69..b2b0b2b1e4d0 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -1058,7 +1058,6 @@ static const struct snd_soc_component_driver soc_component_dev_rt286 = { .num_dapm_routes = ARRAY_SIZE(rt286_dapm_routes), .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config rt286_regmap = { diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c index 1a27e5e63289..266a2cc55b8d 100644 --- a/sound/soc/codecs/rt298.c +++ b/sound/soc/codecs/rt298.c @@ -1123,7 +1123,6 @@ static const struct snd_soc_component_driver soc_component_dev_rt298 = { .num_dapm_routes = ARRAY_SIZE(rt298_dapm_routes), .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config rt298_regmap = { diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index be8ece4630df..b9bcf04d4dc9 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -1173,7 +1173,6 @@ static const struct snd_soc_component_driver soc_component_dev_rt5514 = { .num_dapm_routes = ARRAY_SIZE(rt5514_dapm_routes), .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config rt5514_i2c_regmap = { diff --git a/sound/soc/codecs/rt5616.c b/sound/soc/codecs/rt5616.c index 37f1bf552eff..970d6c4a358e 100644 --- a/sound/soc/codecs/rt5616.c +++ b/sound/soc/codecs/rt5616.c @@ -1304,7 +1304,6 @@ static const struct snd_soc_component_driver soc_component_dev_rt5616 = { .num_dapm_routes = ARRAY_SIZE(rt5616_dapm_routes), .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config rt5616_regmap = { diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index c941e878471c..957f6b19beec 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -1666,7 +1666,6 @@ static const struct snd_soc_component_driver soc_component_dev_rt5631 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct i2c_device_id rt5631_i2c_id[] = { diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 69c80d80ed9d..56008e4518f3 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2867,8 +2867,6 @@ static const struct snd_soc_component_driver soc_component_dev_rt5640 = { .num_dapm_routes = ARRAY_SIZE(rt5640_dapm_routes), .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, - }; static const struct regmap_config rt5640_regmap = { diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 507aba8de3cc..8635bc6567dc 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3534,7 +3534,6 @@ static const struct snd_soc_component_driver soc_component_dev_rt5645 = { .num_dapm_routes = ARRAY_SIZE(rt5645_dapm_routes), .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config rt5645_regmap = { diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index d11d201b1d03..df90af906563 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -2161,7 +2161,6 @@ static const struct snd_soc_component_driver soc_component_dev_rt5651 = { .num_dapm_routes = ARRAY_SIZE(rt5651_dapm_routes), .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config rt5651_regmap = { diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c index 6efa90f46362..5e21e3c37ab5 100644 --- a/sound/soc/codecs/rt5659.c +++ b/sound/soc/codecs/rt5659.c @@ -3801,7 +3801,6 @@ static const struct snd_soc_component_driver soc_component_dev_rt5659 = { .set_pll = rt5659_set_component_pll, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; diff --git a/sound/soc/codecs/rt5660.c b/sound/soc/codecs/rt5660.c index d5f9926625d2..341baa29fdb1 100644 --- a/sound/soc/codecs/rt5660.c +++ b/sound/soc/codecs/rt5660.c @@ -1208,7 +1208,6 @@ static const struct snd_soc_component_driver soc_component_dev_rt5660 = { .num_dapm_routes = ARRAY_SIZE(rt5660_dapm_routes), .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config rt5660_regmap = { diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index e51eed8a79ab..ca981b374b0c 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -3258,7 +3258,6 @@ static const struct snd_soc_component_driver soc_component_dev_rt5663 = { .set_jack = rt5663_set_jack_detect, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config rt5663_v2_regmap = { diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index 4a8d62e1dd2b..6e66cc218fa8 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -4617,7 +4617,6 @@ static const struct snd_soc_component_driver soc_component_dev_rt5665 = { .set_jack = rt5665_set_jack_detect, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; diff --git a/sound/soc/codecs/rt5668.c b/sound/soc/codecs/rt5668.c index 01566f036ca1..beb0951ff680 100644 --- a/sound/soc/codecs/rt5668.c +++ b/sound/soc/codecs/rt5668.c @@ -2362,7 +2362,6 @@ static const struct snd_soc_component_driver soc_component_dev_rt5668 = { .set_jack = rt5668_set_jack_detect, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config rt5668_regmap = { diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 8a97f6db04d5..60dbfa2a54f1 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -2852,7 +2852,6 @@ static const struct snd_soc_component_driver soc_component_dev_rt5670 = { .num_dapm_routes = ARRAY_SIZE(rt5670_dapm_routes), .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config rt5670_regmap = { diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 4a8c267d4fbc..31a2dd0aafb6 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -5189,7 +5189,6 @@ static const struct snd_soc_component_driver soc_component_dev_rt5677 = { .num_dapm_routes = ARRAY_SIZE(rt5677_dapm_routes), .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config rt5677_regmap_physical = { diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 2b6c6d6b9771..2df95e792900 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -3064,7 +3064,6 @@ const struct snd_soc_component_driver rt5682_soc_component_dev = { .set_jack = rt5682_set_jack_detect, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; EXPORT_SYMBOL_GPL(rt5682_soc_component_dev); diff --git a/sound/soc/codecs/rt5682s.c b/sound/soc/codecs/rt5682s.c index 4d44eddee901..eb47e7cd485a 100644 --- a/sound/soc/codecs/rt5682s.c +++ b/sound/soc/codecs/rt5682s.c @@ -2893,7 +2893,6 @@ static const struct snd_soc_component_driver rt5682s_soc_component_dev = { .set_jack = rt5682s_set_jack_detect, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int rt5682s_parse_dt(struct rt5682s_priv *rt5682s, struct device *dev) -- cgit v1.2.3 From 792a8a944e7aa3f6ae0733429ba9937d7029ee4b Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:52:11 +0100 Subject: ASoC: spdif: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-58-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/spdif_receiver.c | 1 - sound/soc/codecs/spdif_transmitter.c | 1 - 2 files changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/spdif_receiver.c b/sound/soc/codecs/spdif_receiver.c index 276db978e587..862e0b654a1c 100644 --- a/sound/soc/codecs/spdif_receiver.c +++ b/sound/soc/codecs/spdif_receiver.c @@ -43,7 +43,6 @@ static struct snd_soc_component_driver soc_codec_spdif_dir = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static struct snd_soc_dai_driver dir_stub_dai = { diff --git a/sound/soc/codecs/spdif_transmitter.c b/sound/soc/codecs/spdif_transmitter.c index 2c8cebfc6603..736518921555 100644 --- a/sound/soc/codecs/spdif_transmitter.c +++ b/sound/soc/codecs/spdif_transmitter.c @@ -43,7 +43,6 @@ static struct snd_soc_component_driver soc_codec_spdif_dit = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static struct snd_soc_dai_driver dit_stub_dai = { -- cgit v1.2.3 From a4311a5b1502f747576e5995d1b5ab04f60033f9 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:52:12 +0100 Subject: ASoC: ssm*: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-59-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/ssm2518.c | 1 - sound/soc/codecs/ssm2602.c | 1 - sound/soc/codecs/ssm4567.c | 1 - 3 files changed, 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c index 012f209e76e9..6d8847848299 100644 --- a/sound/soc/codecs/ssm2518.c +++ b/sound/soc/codecs/ssm2518.c @@ -721,7 +721,6 @@ static const struct snd_soc_component_driver ssm2518_component_driver = { .num_dapm_routes = ARRAY_SIZE(ssm2518_routes), .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config ssm2518_regmap_config = { diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 1821854ca0f3..cbbe83b85ada 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -624,7 +624,6 @@ static const struct snd_soc_component_driver soc_component_dev_ssm2602 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static bool ssm2602_register_volatile(struct device *dev, unsigned int reg) diff --git a/sound/soc/codecs/ssm4567.c b/sound/soc/codecs/ssm4567.c index b47321c597d0..4b0265617c7b 100644 --- a/sound/soc/codecs/ssm4567.c +++ b/sound/soc/codecs/ssm4567.c @@ -427,7 +427,6 @@ static const struct snd_soc_component_driver ssm4567_component_driver = { .num_dapm_routes = ARRAY_SIZE(ssm4567_routes), .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config ssm4567_regmap_config = { -- cgit v1.2.3 From 402f437b43870e65377bb97240ee3911858547cb Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:52:13 +0100 Subject: ASoC: sta*: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-60-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/sta32x.c | 1 - sound/soc/codecs/sta350.c | 1 - sound/soc/codecs/sta529.c | 1 - 3 files changed, 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 17e5077f26b0..8c86b578eba8 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -1014,7 +1014,6 @@ static const struct snd_soc_component_driver sta32x_component = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config sta32x_regmap = { diff --git a/sound/soc/codecs/sta350.c b/sound/soc/codecs/sta350.c index b2d15d20fe63..7b2c5b57d5d4 100644 --- a/sound/soc/codecs/sta350.c +++ b/sound/soc/codecs/sta350.c @@ -1057,7 +1057,6 @@ static const struct snd_soc_component_driver sta350_component = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config sta350_regmap = { diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c index d90e5512a731..313957099145 100644 --- a/sound/soc/codecs/sta529.c +++ b/sound/soc/codecs/sta529.c @@ -322,7 +322,6 @@ static const struct snd_soc_component_driver sta529_component_driver = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config sta529_regmap = { -- cgit v1.2.3 From 02bcc2be4c12763dd4c524e67973afe4d8ea6d4c Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:52:14 +0100 Subject: ASoC: tas*: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-61-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/tas2552.c | 1 - sound/soc/codecs/tas2562.c | 2 -- sound/soc/codecs/tas2764.c | 1 - sound/soc/codecs/tas2770.c | 1 - sound/soc/codecs/tas5086.c | 1 - sound/soc/codecs/tas571x.c | 1 - sound/soc/codecs/tas5720.c | 2 -- sound/soc/codecs/tas5805m.c | 1 - sound/soc/codecs/tas6424.c | 1 - 9 files changed, 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index c98a9332dcc0..bf3d8539a268 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -668,7 +668,6 @@ static const struct snd_soc_component_driver soc_component_dev_tas2552 = { .num_dapm_routes = ARRAY_SIZE(tas2552_audio_map), .idle_bias_on = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config tas2552_regmap_config = { diff --git a/sound/soc/codecs/tas2562.c b/sound/soc/codecs/tas2562.c index e62a3da16aed..dc088a1c6721 100644 --- a/sound/soc/codecs/tas2562.c +++ b/sound/soc/codecs/tas2562.c @@ -589,7 +589,6 @@ static const struct snd_soc_component_driver soc_component_dev_tas2110 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct snd_soc_dapm_widget tas2562_dapm_widgets[] = { @@ -629,7 +628,6 @@ static const struct snd_soc_component_driver soc_component_dev_tas2562 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct snd_soc_dai_ops tas2562_speaker_dai_ops = { diff --git a/sound/soc/codecs/tas2764.c b/sound/soc/codecs/tas2764.c index d395feffb30b..42f0c1e449ba 100644 --- a/sound/soc/codecs/tas2764.c +++ b/sound/soc/codecs/tas2764.c @@ -548,7 +548,6 @@ static const struct snd_soc_component_driver soc_component_driver_tas2764 = { .num_dapm_routes = ARRAY_SIZE(tas2764_audio_map), .idle_bias_on = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct reg_default tas2764_reg_defaults[] = { diff --git a/sound/soc/codecs/tas2770.c b/sound/soc/codecs/tas2770.c index f6037a148cb6..3cb634c28261 100644 --- a/sound/soc/codecs/tas2770.c +++ b/sound/soc/codecs/tas2770.c @@ -546,7 +546,6 @@ static const struct snd_soc_component_driver soc_component_driver_tas2770 = { .num_dapm_routes = ARRAY_SIZE(tas2770_audio_map), .idle_bias_on = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int tas2770_register_codec(struct tas2770_priv *tas2770) diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c index 05b57bb1aea0..a864984225bc 100644 --- a/sound/soc/codecs/tas5086.c +++ b/sound/soc/codecs/tas5086.c @@ -888,7 +888,6 @@ static const struct snd_soc_component_driver soc_component_dev_tas5086 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct i2c_device_id tas5086_i2c_id[] = { diff --git a/sound/soc/codecs/tas571x.c b/sound/soc/codecs/tas571x.c index 7b599664db20..4e7f20db57c4 100644 --- a/sound/soc/codecs/tas571x.c +++ b/sound/soc/codecs/tas571x.c @@ -756,7 +756,6 @@ static const struct snd_soc_component_driver tas571x_component = { .num_dapm_routes = ARRAY_SIZE(tas571x_dapm_routes), .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static struct snd_soc_dai_driver tas571x_dai = { diff --git a/sound/soc/codecs/tas5720.c b/sound/soc/codecs/tas5720.c index 2ee06a95f3e4..3885c0bf0b01 100644 --- a/sound/soc/codecs/tas5720.c +++ b/sound/soc/codecs/tas5720.c @@ -572,7 +572,6 @@ static const struct snd_soc_component_driver soc_component_dev_tas5720 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct snd_soc_component_driver soc_component_dev_tas5722 = { @@ -589,7 +588,6 @@ static const struct snd_soc_component_driver soc_component_dev_tas5722 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; /* PCM rates supported by the TAS5720 driver */ diff --git a/sound/soc/codecs/tas5805m.c b/sound/soc/codecs/tas5805m.c index fa0e81ec875a..b1bb614534f7 100644 --- a/sound/soc/codecs/tas5805m.c +++ b/sound/soc/codecs/tas5805m.c @@ -367,7 +367,6 @@ static const struct snd_soc_component_driver soc_codec_dev_tas5805m = { .num_dapm_routes = ARRAY_SIZE(tas5805m_audio_map), .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int tas5805m_mute(struct snd_soc_dai *dai, int mute, int direction) diff --git a/sound/soc/codecs/tas6424.c b/sound/soc/codecs/tas6424.c index 9c9a6ec4d977..63d2983c3fcf 100644 --- a/sound/soc/codecs/tas6424.c +++ b/sound/soc/codecs/tas6424.c @@ -375,7 +375,6 @@ static struct snd_soc_component_driver soc_codec_dev_tas6424 = { .num_dapm_routes = ARRAY_SIZE(tas6424_audio_map), .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct snd_soc_dai_ops tas6424_speaker_dai_ops = { -- cgit v1.2.3 From c91f7e94ce931f058543174a409bb082208cae4a Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:52:15 +0100 Subject: ASoC: tfa*: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-62-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/tfa9879.c | 1 - sound/soc/codecs/tfa989x.c | 1 - 2 files changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tfa9879.c b/sound/soc/codecs/tfa9879.c index 41a9b1b76e62..9f7902ec40db 100644 --- a/sound/soc/codecs/tfa9879.c +++ b/sound/soc/codecs/tfa9879.c @@ -235,7 +235,6 @@ static const struct snd_soc_component_driver tfa9879_component = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config tfa9879_regmap = { diff --git a/sound/soc/codecs/tfa989x.c b/sound/soc/codecs/tfa989x.c index 8ab2656de750..1c27429b9af6 100644 --- a/sound/soc/codecs/tfa989x.c +++ b/sound/soc/codecs/tfa989x.c @@ -138,7 +138,6 @@ static const struct snd_soc_component_driver tfa989x_component = { .num_dapm_routes = ARRAY_SIZE(tfa989x_dapm_routes), .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const unsigned int tfa989x_rates[] = { -- cgit v1.2.3 From 04f3d715df3a463985dc25e55a55dbd970dd77b7 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:52:16 +0100 Subject: ASoC: tlv320*: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-63-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320adcx140.c | 1 - sound/soc/codecs/tlv320aic23.c | 1 - sound/soc/codecs/tlv320aic26.c | 1 - sound/soc/codecs/tlv320aic31xx.c | 1 - sound/soc/codecs/tlv320aic32x4.c | 2 -- sound/soc/codecs/tlv320aic3x.c | 1 - sound/soc/codecs/tlv320dac33.c | 1 - 7 files changed, 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c index de5b184a701e..6618ac4a7d5c 100644 --- a/sound/soc/codecs/tlv320adcx140.c +++ b/sound/soc/codecs/tlv320adcx140.c @@ -1053,7 +1053,6 @@ static const struct snd_soc_component_driver soc_codec_driver_adcx140 = { .idle_bias_on = 0, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static struct snd_soc_dai_driver adcx140_dai_driver[] = { diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index c86ca793a2b6..c47aa4d4162d 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -586,7 +586,6 @@ static const struct snd_soc_component_driver soc_component_dev_tlv320aic23 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; int tlv320aic23_probe(struct device *dev, struct regmap *regmap) diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index f85f8061639f..8bae4b475068 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -331,7 +331,6 @@ static const struct snd_soc_component_driver aic26_soc_component_dev = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config aic26_regmap = { diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index aacee2367992..0847302121f6 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1417,7 +1417,6 @@ static const struct snd_soc_component_driver soc_codec_driver_aic31xx = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct snd_soc_dai_ops aic31xx_dai_ops = { diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index a8e6adf62ac8..4b74805cdd2e 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -1077,7 +1077,6 @@ static const struct snd_soc_component_driver soc_component_dev_aic32x4 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct snd_kcontrol_new aic32x4_tas2505_snd_controls[] = { @@ -1199,7 +1198,6 @@ static const struct snd_soc_component_driver soc_component_dev_aic32x4_tas2505 = .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int aic32x4_parse_dt(struct aic32x4_priv *aic32x4, diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 610e41bbf388..08938801daec 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1697,7 +1697,6 @@ static const struct snd_soc_component_driver soc_component_dev_aic3x = { .num_dapm_routes = ARRAY_SIZE(intercon), .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static void aic3x_configure_ocmv(struct device *dev, struct aic3x_priv *aic3x) diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 371026eb8f41..17ae3b1d96fb 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -1431,7 +1431,6 @@ static const struct snd_soc_component_driver soc_component_dev_tlv320dac33 = { .num_dapm_routes = ARRAY_SIZE(audio_map), .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; #define DAC33_RATES (SNDRV_PCM_RATE_44100 | \ -- cgit v1.2.3 From 5947b42cbe0ee580c31f7f327119e7f7c703c25c Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:52:17 +0100 Subject: ASoC: twl*: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-64-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 1 - sound/soc/codecs/twl6040.c | 1 - 2 files changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 87b58017094b..e48768233e20 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -2195,7 +2195,6 @@ static const struct snd_soc_component_driver soc_component_dev_twl4030 = { .num_dapm_routes = ARRAY_SIZE(intercon), .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int twl4030_codec_probe(struct platform_device *pdev) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index b37203336c4e..dd5ee5dc0cd7 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1153,7 +1153,6 @@ static const struct snd_soc_component_driver soc_component_dev_twl6040 = { .suspend_bias_off = 1, .idle_bias_on = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int twl6040_codec_probe(struct platform_device *pdev) -- cgit v1.2.3 From 792008f6df86f7e5f861ef80fd4d6eb444a4aa92 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:52:18 +0100 Subject: ASoC: uda*: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-65-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/uda1334.c | 1 - sound/soc/codecs/uda134x.c | 1 - sound/soc/codecs/uda1380.c | 1 - 3 files changed, 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/uda1334.c b/sound/soc/codecs/uda1334.c index 9d5ed34e5420..eace96533600 100644 --- a/sound/soc/codecs/uda1334.c +++ b/sound/soc/codecs/uda1334.c @@ -236,7 +236,6 @@ static const struct snd_soc_component_driver soc_component_dev_uda1334 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct of_device_id uda1334_of_match[] = { diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index 037833c509f7..2db3d8a60c7a 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -527,7 +527,6 @@ static const struct snd_soc_component_driver soc_component_dev_uda134x = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config uda134x_regmap_config = { diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index b5004842520b..fdaaee845176 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -736,7 +736,6 @@ static const struct snd_soc_component_driver soc_component_dev_uda1380 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int uda1380_i2c_probe(struct i2c_client *i2c) -- cgit v1.2.3 From 02004449dbe6ec05b5b64a88824939b8fe474b82 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:52:19 +0100 Subject: ASoC: wm*: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-66-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/wm0010.c | 1 - sound/soc/codecs/wm1250-ev1.c | 1 - sound/soc/codecs/wm2000.c | 1 - sound/soc/codecs/wm2200.c | 1 - sound/soc/codecs/wm5100.c | 1 - sound/soc/codecs/wm5102.c | 1 - sound/soc/codecs/wm5110.c | 1 - sound/soc/codecs/wm8350.c | 1 - sound/soc/codecs/wm8400.c | 1 - sound/soc/codecs/wm8510.c | 1 - sound/soc/codecs/wm8523.c | 1 - sound/soc/codecs/wm8524.c | 1 - sound/soc/codecs/wm8580.c | 1 - sound/soc/codecs/wm8711.c | 1 - sound/soc/codecs/wm8727.c | 1 - sound/soc/codecs/wm8728.c | 1 - sound/soc/codecs/wm8731.c | 1 - sound/soc/codecs/wm8737.c | 1 - sound/soc/codecs/wm8741.c | 1 - sound/soc/codecs/wm8750.c | 1 - sound/soc/codecs/wm8753.c | 1 - sound/soc/codecs/wm8770.c | 1 - sound/soc/codecs/wm8776.c | 1 - sound/soc/codecs/wm8782.c | 1 - sound/soc/codecs/wm8804.c | 1 - sound/soc/codecs/wm8900.c | 1 - sound/soc/codecs/wm8903.c | 1 - sound/soc/codecs/wm8904.c | 1 - sound/soc/codecs/wm8940.c | 1 - sound/soc/codecs/wm8955.c | 1 - sound/soc/codecs/wm8960.c | 1 - sound/soc/codecs/wm8961.c | 1 - sound/soc/codecs/wm8962.c | 1 - sound/soc/codecs/wm8971.c | 1 - sound/soc/codecs/wm8974.c | 1 - sound/soc/codecs/wm8978.c | 1 - sound/soc/codecs/wm8983.c | 1 - sound/soc/codecs/wm8985.c | 1 - sound/soc/codecs/wm8988.c | 1 - sound/soc/codecs/wm8990.c | 1 - sound/soc/codecs/wm8991.c | 1 - sound/soc/codecs/wm8993.c | 1 - sound/soc/codecs/wm8994.c | 1 - sound/soc/codecs/wm8995.c | 1 - sound/soc/codecs/wm8996.c | 2 -- sound/soc/codecs/wm8997.c | 1 - sound/soc/codecs/wm8998.c | 1 - sound/soc/codecs/wm9081.c | 1 - sound/soc/codecs/wm9090.c | 1 - sound/soc/codecs/wm9705.c | 1 - sound/soc/codecs/wm9712.c | 1 - sound/soc/codecs/wm9713.c | 1 - 52 files changed, 53 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index 1bef1c500c8e..034a4e858c7e 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -789,7 +789,6 @@ static const struct snd_soc_component_driver soc_component_dev_wm0010 = { .num_dapm_routes = ARRAY_SIZE(wm0010_dapm_routes), .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; #define WM0010_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) diff --git a/sound/soc/codecs/wm1250-ev1.c b/sound/soc/codecs/wm1250-ev1.c index b6366dea15a6..98343626078b 100644 --- a/sound/soc/codecs/wm1250-ev1.c +++ b/sound/soc/codecs/wm1250-ev1.c @@ -144,7 +144,6 @@ static const struct snd_soc_component_driver soc_component_dev_wm1250_ev1 = { .set_bias_level = wm1250_ev1_set_bias_level, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int wm1250_ev1_pdata(struct i2c_client *i2c) diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index ede5f2a982a6..14b4fd97488c 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -803,7 +803,6 @@ static const struct snd_soc_component_driver soc_component_dev_wm2000 = { .num_dapm_routes = ARRAY_SIZE(wm2000_audio_map), .idle_bias_on = 1, .use_pmdown_time = 1, - .non_legacy_dai_naming = 1, }; static int wm2000_i2c_probe(struct i2c_client *i2c) diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index 1cd544580c83..7b4e162a298c 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -2104,7 +2104,6 @@ static const struct snd_soc_component_driver soc_component_wm2200 = { .dapm_routes = wm2200_dapm_routes, .num_dapm_routes = ARRAY_SIZE(wm2200_dapm_routes), .endianness = 1, - .non_legacy_dai_naming = 1, }; static irqreturn_t wm2200_irq(int irq, void *data) diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index a89870918174..35a85ce6b464 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -2389,7 +2389,6 @@ static const struct snd_soc_component_driver soc_component_dev_wm5100 = { .num_dapm_routes = ARRAY_SIZE(wm5100_dapm_routes), .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config wm5100_regmap = { diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index da2f8998df87..8b1caac65c3a 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -2015,7 +2015,6 @@ static const struct snd_soc_component_driver soc_component_dev_wm5102 = { .num_dapm_routes = ARRAY_SIZE(wm5102_dapm_routes), .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int wm5102_probe(struct platform_device *pdev) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 4ab7a672f8de..f3f4a10bf0f7 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -2385,7 +2385,6 @@ static const struct snd_soc_component_driver soc_component_dev_wm5110 = { .num_dapm_routes = ARRAY_SIZE(wm5110_dapm_routes), .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int wm5110_probe(struct platform_device *pdev) diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 41504ce2a682..66bd281095e1 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1613,7 +1613,6 @@ static const struct snd_soc_component_driver soc_component_dev_wm8350 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int wm8350_probe(struct platform_device *pdev) diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index bf5e77c86aed..19ce839f6ef7 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -1322,7 +1322,6 @@ static const struct snd_soc_component_driver soc_component_dev_wm8400 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int wm8400_probe(struct platform_device *pdev) diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index c6439d25006b..e13f9780a111 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -592,7 +592,6 @@ static const struct snd_soc_component_driver soc_component_dev_wm8510 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct of_device_id wm8510_of_match[] = { diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index ba35a0221dc8..66f6371d8acf 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -422,7 +422,6 @@ static const struct snd_soc_component_driver soc_component_dev_wm8523 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct of_device_id wm8523_of_match[] = { diff --git a/sound/soc/codecs/wm8524.c b/sound/soc/codecs/wm8524.c index 81f858f6bd67..b56dcac60244 100644 --- a/sound/soc/codecs/wm8524.c +++ b/sound/soc/codecs/wm8524.c @@ -203,7 +203,6 @@ static const struct snd_soc_component_driver soc_component_dev_wm8524 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct of_device_id wm8524_of_match[] = { diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 84020195314d..ca796aa0aeb7 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -966,7 +966,6 @@ static const struct snd_soc_component_driver soc_component_dev_wm8580 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config wm8580_regmap = { diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index b68a1ebcd061..383c6796e8a3 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -378,7 +378,6 @@ static const struct snd_soc_component_driver soc_component_dev_wm8711 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct of_device_id wm8711_of_match[] = { diff --git a/sound/soc/codecs/wm8727.c b/sound/soc/codecs/wm8727.c index 1a118b75b539..d6b0a570dd87 100644 --- a/sound/soc/codecs/wm8727.c +++ b/sound/soc/codecs/wm8727.c @@ -55,7 +55,6 @@ static const struct snd_soc_component_driver soc_component_dev_wm8727 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int wm8727_probe(struct platform_device *pdev) diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index 119ff0a1bb35..a3dbdbf40723 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -221,7 +221,6 @@ static const struct snd_soc_component_driver soc_component_dev_wm8728 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct of_device_id wm8728_of_match[] = { diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 2408c4a591d5..d5ab3ba126a6 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -561,7 +561,6 @@ static const struct snd_soc_component_driver soc_component_dev_wm8731 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; int wm8731_init(struct device *dev, struct wm8731_priv *wm8731) diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c index 5778091d1c09..90b54343370c 100644 --- a/sound/soc/codecs/wm8737.c +++ b/sound/soc/codecs/wm8737.c @@ -583,7 +583,6 @@ static const struct snd_soc_component_driver soc_component_dev_wm8737 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct of_device_id wm8737_of_match[] = { diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 871e2c5421b8..c7afa4f2795d 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -528,7 +528,6 @@ static const struct snd_soc_component_driver soc_component_dev_wm8741 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct of_device_id wm8741_of_match[] = { diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 1426fc1f7c5a..2f6ee8d6639f 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -719,7 +719,6 @@ static const struct snd_soc_component_driver soc_component_dev_wm8750 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct of_device_id wm8750_of_match[] = { diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 931134d334ec..bb18f58dc670 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1492,7 +1492,6 @@ static const struct snd_soc_component_driver soc_component_dev_wm8753 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct of_device_id wm8753_of_match[] = { diff --git a/sound/soc/codecs/wm8770.c b/sound/soc/codecs/wm8770.c index 5f394065030d..e03fee8869c3 100644 --- a/sound/soc/codecs/wm8770.c +++ b/sound/soc/codecs/wm8770.c @@ -617,7 +617,6 @@ static const struct snd_soc_component_driver soc_component_dev_wm8770 = { .num_dapm_routes = ARRAY_SIZE(wm8770_intercon), .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct of_device_id wm8770_of_match[] = { diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index f164cb6744c4..936ea24621b0 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -436,7 +436,6 @@ static const struct snd_soc_component_driver soc_component_dev_wm8776 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct of_device_id wm8776_of_match[] = { diff --git a/sound/soc/codecs/wm8782.c b/sound/soc/codecs/wm8782.c index f89855c616eb..95ff4339d103 100644 --- a/sound/soc/codecs/wm8782.c +++ b/sound/soc/codecs/wm8782.c @@ -99,7 +99,6 @@ static const struct snd_soc_component_driver soc_component_dev_wm8782 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int wm8782_probe(struct platform_device *pdev) diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index 21bf0cfa1e7e..0b234bae480e 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -546,7 +546,6 @@ static const struct snd_soc_component_driver soc_component_dev_wm8804 = { .num_dapm_routes = ARRAY_SIZE(wm8804_dapm_routes), .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; const struct regmap_config wm8804_regmap_config = { diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 84a3daf0c11e..d6420df3505d 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -1214,7 +1214,6 @@ static const struct snd_soc_component_driver soc_component_dev_wm8900 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config wm8900_regmap = { diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 3c95c2aea515..54e0a7628cd5 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1893,7 +1893,6 @@ static const struct snd_soc_component_driver soc_component_dev_wm8903 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config wm8903_regmap = { diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 04bb8e392497..ca6a01a230af 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -2131,7 +2131,6 @@ static const struct snd_soc_component_driver soc_component_dev_wm8904 = { .set_bias_level = wm8904_set_bias_level, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config wm8904_regmap = { diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index 589394d420ce..8dac9fd88547 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -734,7 +734,6 @@ static const struct snd_soc_component_driver soc_component_dev_wm8940 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config wm8940_regmap = { diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index 80e3cbd704ee..05ef45672ebc 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -952,7 +952,6 @@ static const struct snd_soc_component_driver soc_component_dev_wm8955 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config wm8955_regmap = { diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 8c8f32b23083..37956516d997 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -1378,7 +1378,6 @@ static const struct snd_soc_component_driver soc_component_dev_wm8960 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config wm8960_regmap = { diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index 69eb731dbf4b..7dc6aaf65576 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -895,7 +895,6 @@ static const struct snd_soc_component_driver soc_component_dev_wm8961 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config wm8961_regmap = { diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 5cca89364280..398c448ea854 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3502,7 +3502,6 @@ static const struct snd_soc_component_driver soc_component_dev_wm8962 = { .set_pll = wm8962_set_fll, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; /* Improve power consumption for IN4 DC measurement mode */ diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 8a289b048e66..4db9248de54b 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -659,7 +659,6 @@ static const struct snd_soc_component_driver soc_component_dev_wm8971 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config wm8971_regmap = { diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index a8d7809f3f64..010a394c705c 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -682,7 +682,6 @@ static const struct snd_soc_component_driver soc_component_dev_wm8974 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int wm8974_i2c_probe(struct i2c_client *i2c) diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 141f50bfec68..a682f8020eb6 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -1005,7 +1005,6 @@ static const struct snd_soc_component_driver soc_component_dev_wm8978 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config wm8978_regmap_config = { diff --git a/sound/soc/codecs/wm8983.c b/sound/soc/codecs/wm8983.c index ae89554d47bc..50e6ac6ccbe0 100644 --- a/sound/soc/codecs/wm8983.c +++ b/sound/soc/codecs/wm8983.c @@ -987,7 +987,6 @@ static const struct snd_soc_component_driver soc_component_dev_wm8983 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config wm8983_regmap = { diff --git a/sound/soc/codecs/wm8985.c b/sound/soc/codecs/wm8985.c index cf2c32eac773..751aa6730833 100644 --- a/sound/soc/codecs/wm8985.c +++ b/sound/soc/codecs/wm8985.c @@ -1116,7 +1116,6 @@ static const struct snd_soc_component_driver soc_component_dev_wm8985 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config wm8985_regmap = { diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index 27538d6598cf..5dbdf647cd97 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -823,7 +823,6 @@ static const struct snd_soc_component_driver soc_component_dev_wm8988 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config wm8988_regmap = { diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index c9448a59c872..589af286f133 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1217,7 +1217,6 @@ static const struct snd_soc_component_driver soc_component_dev_wm8990 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int wm8990_i2c_probe(struct i2c_client *i2c) diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c index 998bc89bb7e1..30121993b7b4 100644 --- a/sound/soc/codecs/wm8991.c +++ b/sound/soc/codecs/wm8991.c @@ -1243,7 +1243,6 @@ static const struct snd_soc_component_driver soc_component_dev_wm8991 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config wm8991_regmap = { diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index f4da77ec9d6c..8db98b5a06bf 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1621,7 +1621,6 @@ static const struct snd_soc_component_driver soc_component_dev_wm8993 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int wm8993_i2c_probe(struct i2c_client *i2c) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index f117ec0c489f..d3cfd3788f2a 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -4614,7 +4614,6 @@ static const struct snd_soc_component_driver soc_component_dev_wm8994 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int wm8994_probe(struct platform_device *pdev) diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index ea9727446707..eed48bf339f2 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -2182,7 +2182,6 @@ static const struct snd_soc_component_driver soc_component_dev_wm8995 = { .num_dapm_routes = ARRAY_SIZE(wm8995_intercon), .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config wm8995_regmap = { diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index f7bb27d1c76d..17f307a31046 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -2695,8 +2695,6 @@ static const struct snd_soc_component_driver soc_component_dev_wm8996 = { .set_pll = wm8996_set_fll, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, - }; #define WM8996_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index 38ef631d1a1f..210ad662fc26 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -1105,7 +1105,6 @@ static const struct snd_soc_component_driver soc_component_dev_wm8997 = { .num_dapm_routes = ARRAY_SIZE(wm8997_dapm_routes), .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int wm8997_probe(struct platform_device *pdev) diff --git a/sound/soc/codecs/wm8998.c b/sound/soc/codecs/wm8998.c index 00b59fc9b1fe..328f1946f584 100644 --- a/sound/soc/codecs/wm8998.c +++ b/sound/soc/codecs/wm8998.c @@ -1325,7 +1325,6 @@ static const struct snd_soc_component_driver soc_component_dev_wm8998 = { .num_dapm_routes = ARRAY_SIZE(wm8998_dapm_routes), .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int wm8998_probe(struct platform_device *pdev) diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 87b58448cea7..d5151877d0fa 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1284,7 +1284,6 @@ static const struct snd_soc_component_driver soc_component_dev_wm9081 = { .num_dapm_routes = ARRAY_SIZE(wm9081_audio_paths), .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config wm9081_regmap = { diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index f7d80f1e37a8..ef3524c3f07f 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -543,7 +543,6 @@ static const struct snd_soc_component_driver soc_component_dev_wm9090 = { .suspend_bias_off = 1, .idle_bias_on = 1, .use_pmdown_time = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config wm9090_regmap = { diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index 99fe8f316624..d04902ef1d5f 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -368,7 +368,6 @@ static const struct snd_soc_component_driver soc_component_dev_wm9705 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int wm9705_probe(struct platform_device *pdev) diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 7515c9d4006e..df9b7980706b 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -692,7 +692,6 @@ static const struct snd_soc_component_driver soc_component_dev_wm9712 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int wm9712_probe(struct platform_device *pdev) diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index e0ce32dd4a81..5d2e54e06e30 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1257,7 +1257,6 @@ static const struct snd_soc_component_driver soc_component_dev_wm9713 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int wm9713_probe(struct platform_device *pdev) -- cgit v1.2.3 From 4c90eebd97c519361f32e11de991e299f5b47e3d Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:52:20 +0100 Subject: ASoC: 88pm860x: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-67-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/88pm860x-codec.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index c6043fa58c74..fc65283031cd 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -1345,7 +1345,6 @@ static const struct snd_soc_component_driver soc_component_dev_pm860x = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int pm860x_codec_probe(struct platform_device *pdev) -- cgit v1.2.3 From bb426d37dcd9a1474f785fea434875233d24e537 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:52:21 +0100 Subject: ASoC: ab8500: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-68-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/ab8500-codec.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index cbd4a92cb06c..68342917419e 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -2523,7 +2523,6 @@ static const struct snd_soc_component_driver ab8500_component_driver = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int ab8500_codec_driver_probe(struct platform_device *pdev) -- cgit v1.2.3 From 96b409c94d6766ae8faa9f07fabc3694ddb7d018 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:52:22 +0100 Subject: ASoC: ac97: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-69-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/ac97.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index 6ad9c9443b5d..cc12052e1920 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -119,7 +119,6 @@ static const struct snd_soc_component_driver soc_component_dev_ac97 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int ac97_probe(struct platform_device *pdev) -- cgit v1.2.3 From e556a108e0aab4688cb0c7b1c0517e3fab8b5eb4 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:52:23 +0100 Subject: ASoC: ads117x: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-70-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/ads117x.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/ads117x.c b/sound/soc/codecs/ads117x.c index 1d07e2699f04..44aa06e03486 100644 --- a/sound/soc/codecs/ads117x.c +++ b/sound/soc/codecs/ads117x.c @@ -62,7 +62,6 @@ static const struct snd_soc_component_driver soc_component_dev_ads117x = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int ads117x_probe(struct platform_device *pdev) -- cgit v1.2.3 From 310288271f55ae0edccd01257c9fdf460dd45e30 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:52:24 +0100 Subject: ASoC: bd28623: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-71-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/bd28623.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/bd28623.c b/sound/soc/codecs/bd28623.c index a6267cb86d86..82a94211d012 100644 --- a/sound/soc/codecs/bd28623.c +++ b/sound/soc/codecs/bd28623.c @@ -161,7 +161,6 @@ static const struct snd_soc_component_driver soc_codec_bd = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static struct snd_soc_dai_driver soc_dai_bd = { -- cgit v1.2.3 From 8c657358f685cec541d7ad3c54f899a65f56783e Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:52:25 +0100 Subject: ASoC: bt-sco: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-72-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/bt-sco.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/bt-sco.c b/sound/soc/codecs/bt-sco.c index cf17b9741bd8..4086b6a53de8 100644 --- a/sound/soc/codecs/bt-sco.c +++ b/sound/soc/codecs/bt-sco.c @@ -69,7 +69,6 @@ static const struct snd_soc_component_driver soc_component_dev_bt_sco = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int bt_sco_probe(struct platform_device *pdev) -- cgit v1.2.3 From 35c5013ce7ca3ad55974e3517273a0e42140b5e7 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:52:26 +0100 Subject: ASoC: cpcap: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-73-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cpcap.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/cpcap.c b/sound/soc/codecs/cpcap.c index ffdf8b615efa..4f9dabd9d78a 100644 --- a/sound/soc/codecs/cpcap.c +++ b/sound/soc/codecs/cpcap.c @@ -1660,7 +1660,6 @@ static struct snd_soc_component_driver soc_codec_dev_cpcap = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int cpcap_codec_probe(struct platform_device *pdev) -- cgit v1.2.3 From 73a3dca65cbe5e7de20f3453b6881acf3fb3cfbe Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:52:27 +0100 Subject: ASoC: cq93vc: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-74-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cq93vc.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c index 0aae5790222a..14403b76c724 100644 --- a/sound/soc/codecs/cq93vc.c +++ b/sound/soc/codecs/cq93vc.c @@ -126,7 +126,6 @@ static const struct snd_soc_component_driver soc_component_dev_cq93vc = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int cq93vc_platform_probe(struct platform_device *pdev) -- cgit v1.2.3 From a0b6e4048228829485a43247c12c7774531728c4 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:52:28 +0100 Subject: ASoC: cx20442: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-75-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cx20442.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c index 1af0bf5f1e2f..43c0cac0ec9e 100644 --- a/sound/soc/codecs/cx20442.c +++ b/sound/soc/codecs/cx20442.c @@ -411,7 +411,6 @@ static const struct snd_soc_component_driver cx20442_component_dev = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int cx20442_platform_probe(struct platform_device *pdev) -- cgit v1.2.3 From 4eaf75fa427262289a2bc34d3fcfbc602ebbacfa Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:52:29 +0100 Subject: ASoC: dmic: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-76-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/dmic.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/dmic.c b/sound/soc/codecs/dmic.c index d1a30ca4571a..4fd6f97e5a49 100644 --- a/sound/soc/codecs/dmic.c +++ b/sound/soc/codecs/dmic.c @@ -140,7 +140,6 @@ static const struct snd_soc_component_driver soc_dmic = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int dmic_dev_probe(struct platform_device *pdev) -- cgit v1.2.3 From 33b179e7513c30f03277f5de2a845e940a9bde9c Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:52:30 +0100 Subject: ASoC: gtm601: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-77-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/gtm601.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/gtm601.c b/sound/soc/codecs/gtm601.c index e1235e695b0f..c6b1e77ffccd 100644 --- a/sound/soc/codecs/gtm601.c +++ b/sound/soc/codecs/gtm601.c @@ -73,7 +73,6 @@ static const struct snd_soc_component_driver soc_component_dev_gtm601 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int gtm601_platform_probe(struct platform_device *pdev) -- cgit v1.2.3 From f02a7d11998eefe8c5627b93627469a0aab8d3da Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:52:31 +0100 Subject: ASoC: hdac_hdmi: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-78-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/hdac_hdmi.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 66408a98298b..cb23650ad522 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -2058,7 +2058,6 @@ static const struct snd_soc_component_driver hdmi_hda_codec = { .remove = hdmi_codec_remove, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static void hdac_hdmi_get_chmap(struct hdac_device *hdev, int pcm_idx, -- cgit v1.2.3 From f5f8019371b42c742d9777052c189e89a0745319 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:52:32 +0100 Subject: ASoC: hdmi-codec: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-79-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/hdmi-codec.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 7d1e351f863a..5679102de91f 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -977,7 +977,6 @@ static const struct snd_soc_component_driver hdmi_driver = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, .set_jack = hdmi_codec_set_jack, }; -- cgit v1.2.3 From e8f88be5c1548791397dadf2250fb5dcc9461f10 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:52:33 +0100 Subject: ASoC: ics43432: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-80-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/ics43432.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/ics43432.c b/sound/soc/codecs/ics43432.c index de4c8460ab3d..58a382254718 100644 --- a/sound/soc/codecs/ics43432.c +++ b/sound/soc/codecs/ics43432.c @@ -41,7 +41,6 @@ static const struct snd_soc_component_driver ics43432_component_driver = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int ics43432_probe(struct platform_device *pdev) -- cgit v1.2.3 From 1f1ee5ae7a8b3d30cbfe18561a4e3b7430e96c9f Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:52:34 +0100 Subject: ASoC: inno_rk3036: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-81-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/inno_rk3036.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/inno_rk3036.c b/sound/soc/codecs/inno_rk3036.c index ca0f4c1911e4..8222cde6e3b9 100644 --- a/sound/soc/codecs/inno_rk3036.c +++ b/sound/soc/codecs/inno_rk3036.c @@ -387,7 +387,6 @@ static const struct snd_soc_component_driver rk3036_codec_driver = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config rk3036_codec_regmap_config = { -- cgit v1.2.3 From 22afe04dd84a63440e69dfc7f0e670404fbce831 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:52:35 +0100 Subject: ASoC: Intel: avs: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-82-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/pcm.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/avs/pcm.c b/sound/soc/intel/avs/pcm.c index 668f533578a6..f21b0cdd3206 100644 --- a/sound/soc/intel/avs/pcm.c +++ b/sound/soc/intel/avs/pcm.c @@ -846,7 +846,6 @@ static const struct snd_soc_component_driver avs_component_driver = { .pcm_construct = avs_component_construct, .module_get_upon_open = 1, /* increment refcount when a pcm is opened */ .topology_name_prefix = "intel/avs", - .non_legacy_dai_naming = true, }; static int avs_soc_component_register(struct device *dev, const char *name, @@ -1172,7 +1171,6 @@ static const struct snd_soc_component_driver avs_hda_component_driver = { .remove_order = SND_SOC_COMP_ORDER_EARLY, .module_get_upon_open = 1, .topology_name_prefix = "intel/avs", - .non_legacy_dai_naming = true, }; int avs_hda_platform_register(struct avs_dev *adev, const char *name) -- cgit v1.2.3 From 328bd81743f0823d9604b0098c95f071e7d02805 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:52:36 +0100 Subject: ASoC: isabelle: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-83-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/isabelle.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c index 39be31e1282e..50105d72b2b7 100644 --- a/sound/soc/codecs/isabelle.c +++ b/sound/soc/codecs/isabelle.c @@ -1095,7 +1095,6 @@ static const struct snd_soc_component_driver soc_component_dev_isabelle = { .num_dapm_routes = ARRAY_SIZE(isabelle_intercon), .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config isabelle_regmap_config = { -- cgit v1.2.3 From dd213681c801fd9d26aef95f4eb563c38f4967f9 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:52:37 +0100 Subject: ASoC: jz4740: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-84-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/jz4740.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c index 081485f784e9..7c25acf6ff0d 100644 --- a/sound/soc/codecs/jz4740.c +++ b/sound/soc/codecs/jz4740.c @@ -291,8 +291,6 @@ static const struct snd_soc_component_driver soc_codec_dev_jz4740_codec = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, - }; static const struct regmap_config jz4740_codec_regmap_config = { -- cgit v1.2.3 From 191889406df931cc2e40abf0a0de141b098f0481 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:52:38 +0100 Subject: ASoC: lm49453: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-85-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/lm49453.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index c4900ada8618..a2e782cc4276 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -1399,7 +1399,6 @@ static const struct snd_soc_component_driver soc_component_dev_lm49453 = { .num_dapm_routes = ARRAY_SIZE(lm49453_audio_map), .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config lm49453_regmap_config = { -- cgit v1.2.3 From 34b89b309441f7f45f68d7ec3633ee3d50921bc8 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:52:39 +0100 Subject: ASoC: lochnagar: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-86-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/lochnagar-sc.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/lochnagar-sc.c b/sound/soc/codecs/lochnagar-sc.c index 54a8ba7ed3c2..13fbd8830b09 100644 --- a/sound/soc/codecs/lochnagar-sc.c +++ b/sound/soc/codecs/lochnagar-sc.c @@ -217,7 +217,6 @@ static const struct snd_soc_component_driver lochnagar_sc_driver = { .dapm_routes = lochnagar_sc_routes, .num_dapm_routes = ARRAY_SIZE(lochnagar_sc_routes), - .non_legacy_dai_naming = 1, .endianness = 1, }; -- cgit v1.2.3 From 139db4ad9e0b793ffd3f4f23976bf72d5e4e6703 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:52:40 +0100 Subject: ASoC: mc13783: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-87-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/mc13783.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 08517547e66c..71490f11d96a 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -727,7 +727,6 @@ static const struct snd_soc_component_driver soc_component_dev_mc13783 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int __init mc13783_codec_probe(struct platform_device *pdev) -- cgit v1.2.3 From 7e6fcd7f6223ab32bdccc5e22cdec780cde305c3 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:52:41 +0100 Subject: ASoC: ml26124: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-88-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/ml26124.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c index de8fcbdd85be..3c6ac77379cb 100644 --- a/sound/soc/codecs/ml26124.c +++ b/sound/soc/codecs/ml26124.c @@ -537,7 +537,6 @@ static const struct snd_soc_component_driver soc_component_dev_ml26124 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config ml26124_i2c_regmap = { -- cgit v1.2.3 From 2e938b8edfedb73efd07545a58fe51bb7fc48a56 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:52:42 +0100 Subject: ASoC: rk817: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-89-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/rk817_codec.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/rk817_codec.c b/sound/soc/codecs/rk817_codec.c index cce6f4e7992f..2a5b274bfc0f 100644 --- a/sound/soc/codecs/rk817_codec.c +++ b/sound/soc/codecs/rk817_codec.c @@ -444,7 +444,6 @@ static const struct snd_soc_component_driver soc_codec_dev_rk817 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, .controls = rk817_volume_controls, .num_controls = ARRAY_SIZE(rk817_volume_controls), .dapm_routes = rk817_dapm_routes, -- cgit v1.2.3 From 81ed3cb8d93936fe32b2b5c213dd56d8ecae7be8 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:52:43 +0100 Subject: ASoC: sgtl5000: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-90-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 2aa48aef6a97..0b8a377ba145 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1536,7 +1536,6 @@ static const struct snd_soc_component_driver sgtl5000_driver = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct regmap_config sgtl5000_regmap = { -- cgit v1.2.3 From 89571b892e74b9724e155774576651cd675b4110 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:52:44 +0100 Subject: ASoC: si476x: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-91-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/si476x.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c index 8bd2edf70f13..d87141ba8438 100644 --- a/sound/soc/codecs/si476x.c +++ b/sound/soc/codecs/si476x.c @@ -239,7 +239,6 @@ static const struct snd_soc_component_driver soc_component_dev_si476x = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int si476x_platform_probe(struct platform_device *pdev) -- cgit v1.2.3 From e5257aa583b6d9f80e3aaa3ed6fc68c1b1b5925a Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:52:45 +0100 Subject: ASoC: stac9766: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-92-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/stac9766.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index d99f6e466d0a..1824a71fe053 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -313,8 +313,6 @@ static const struct snd_soc_component_driver soc_component_dev_stac9766 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, - }; static int stac9766_probe(struct platform_device *pdev) -- cgit v1.2.3 From 20b1894d16547dcd99f190f5a0604a06a0c4479f Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:52:46 +0100 Subject: ASoC: sti-sas: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-93-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/sti-sas.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/sti-sas.c b/sound/soc/codecs/sti-sas.c index 10a6a112f4b4..f076878908ee 100644 --- a/sound/soc/codecs/sti-sas.c +++ b/sound/soc/codecs/sti-sas.c @@ -398,7 +398,6 @@ static struct snd_soc_component_driver sti_sas_driver = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static const struct of_device_id sti_sas_dev_match[] = { -- cgit v1.2.3 From c06fb318493a059ac2c47937761d048f9ab1b542 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:52:47 +0100 Subject: ASoC: tscs42xx: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-94-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/tscs42xx.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/tscs42xx.c b/sound/soc/codecs/tscs42xx.c index 4fb0bb01bcdc..fa0c525189c2 100644 --- a/sound/soc/codecs/tscs42xx.c +++ b/sound/soc/codecs/tscs42xx.c @@ -1358,7 +1358,6 @@ static const struct snd_soc_component_driver soc_codec_dev_tscs42xx = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static inline void init_coeff_ram_cache(struct tscs42xx *tscs42xx) -- cgit v1.2.3 From 11c8bfaacbcd6c8251f65101d5ceeb173a76b1a3 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:52:48 +0100 Subject: ASoC: wl1273: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-95-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/wl1273.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c index 02232f64110e..626278e4c923 100644 --- a/sound/soc/codecs/wl1273.c +++ b/sound/soc/codecs/wl1273.c @@ -475,7 +475,6 @@ static const struct snd_soc_component_driver soc_component_dev_wl1273 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, - .non_legacy_dai_naming = 1, }; static int wl1273_platform_probe(struct platform_device *pdev) -- cgit v1.2.3 From 4a7a283a41dad608ce32c4ed623cc2caf68150c4 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 13:52:49 +0100 Subject: ASoC: zl38060: Remove now redundant non_legacy_dai_naming flag The ASoC core has now been changed to default to the non-legacy DAI naming, as such drivers using the new scheme no longer need to specify the non_legacy_dai_naming flag. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623125250.2355471-96-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/zl38060.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/zl38060.c b/sound/soc/codecs/zl38060.c index 6cae0fb08093..c3d0a2a7c36f 100644 --- a/sound/soc/codecs/zl38060.c +++ b/sound/soc/codecs/zl38060.c @@ -385,7 +385,6 @@ static const struct snd_soc_component_driver zl38_component_dev = { .dapm_routes = zl38_dapm_routes, .num_dapm_routes = ARRAY_SIZE(zl38_dapm_routes), .endianness = 1, - .non_legacy_dai_naming = 1, }; static void chip_gpio_set(struct gpio_chip *c, unsigned int offset, int val) -- cgit v1.2.3 From 82102a24c930986aedc572f89b437cd9e4d44d7e Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 16 Jun 2022 17:04:17 -0500 Subject: ASoC: Intel: catpt: use pm_runtime_resume_and_get() The current code does not check for errors and does not release the reference on errors. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Kai Vehmanen Reviewed-by: Ranjani Sridharan Acked-by: Cezary Rojewski Reviewed-by: Cezary Rojewski Link: https://lore.kernel.org/r/20220616220427.136036-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/catpt/pcm.c | 26 ++++++++++++++++++++------ sound/soc/intel/catpt/sysfs.c | 4 +++- 2 files changed, 23 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/catpt/pcm.c b/sound/soc/intel/catpt/pcm.c index a26000cd5ceb..30ca5416c9a3 100644 --- a/sound/soc/intel/catpt/pcm.c +++ b/sound/soc/intel/catpt/pcm.c @@ -667,7 +667,9 @@ static int catpt_dai_pcm_new(struct snd_soc_pcm_runtime *rtm, if (!memcmp(&cdev->devfmt[devfmt.iface], &devfmt, sizeof(devfmt))) return 0; - pm_runtime_get_sync(cdev->dev); + ret = pm_runtime_resume_and_get(cdev->dev); + if (ret < 0 && ret != -EACCES) + return ret; ret = catpt_ipc_set_device_format(cdev, &devfmt); @@ -853,9 +855,12 @@ static int catpt_mixer_volume_get(struct snd_kcontrol *kcontrol, snd_soc_kcontrol_component(kcontrol); struct catpt_dev *cdev = dev_get_drvdata(component->dev); u32 dspvol; + int ret; int i; - pm_runtime_get_sync(cdev->dev); + ret = pm_runtime_resume_and_get(cdev->dev); + if (ret < 0 && ret != -EACCES) + return ret; for (i = 0; i < CATPT_CHANNELS_MAX; i++) { dspvol = catpt_mixer_volume(cdev, &cdev->mixer, i); @@ -876,7 +881,9 @@ static int catpt_mixer_volume_put(struct snd_kcontrol *kcontrol, struct catpt_dev *cdev = dev_get_drvdata(component->dev); int ret; - pm_runtime_get_sync(cdev->dev); + ret = pm_runtime_resume_and_get(cdev->dev); + if (ret < 0 && ret != -EACCES) + return ret; ret = catpt_set_dspvol(cdev, cdev->mixer.mixer_hw_id, ucontrol->value.integer.value); @@ -897,6 +904,7 @@ static int catpt_stream_volume_get(struct snd_kcontrol *kcontrol, struct catpt_dev *cdev = dev_get_drvdata(component->dev); long *ctlvol = (long *)kcontrol->private_value; u32 dspvol; + int ret; int i; stream = catpt_stream_find(cdev, pin_id); @@ -906,7 +914,9 @@ static int catpt_stream_volume_get(struct snd_kcontrol *kcontrol, return 0; } - pm_runtime_get_sync(cdev->dev); + ret = pm_runtime_resume_and_get(cdev->dev); + if (ret < 0 && ret != -EACCES) + return ret; for (i = 0; i < CATPT_CHANNELS_MAX; i++) { dspvol = catpt_stream_volume(cdev, stream, i); @@ -937,7 +947,9 @@ static int catpt_stream_volume_put(struct snd_kcontrol *kcontrol, return 0; } - pm_runtime_get_sync(cdev->dev); + ret = pm_runtime_resume_and_get(cdev->dev); + if (ret < 0 && ret != -EACCES) + return ret; ret = catpt_set_dspvol(cdev, stream->info.stream_hw_id, ucontrol->value.integer.value); @@ -1013,7 +1025,9 @@ static int catpt_loopback_switch_put(struct snd_kcontrol *kcontrol, return 0; } - pm_runtime_get_sync(cdev->dev); + ret = pm_runtime_resume_and_get(cdev->dev); + if (ret < 0 && ret != -EACCES) + return ret; ret = catpt_ipc_mute_loopback(cdev, stream->info.stream_hw_id, mute); diff --git a/sound/soc/intel/catpt/sysfs.c b/sound/soc/intel/catpt/sysfs.c index 9579e233a15d..1bdbcc04dc71 100644 --- a/sound/soc/intel/catpt/sysfs.c +++ b/sound/soc/intel/catpt/sysfs.c @@ -15,7 +15,9 @@ static ssize_t fw_version_show(struct device *dev, struct catpt_fw_version version; int ret; - pm_runtime_get_sync(cdev->dev); + ret = pm_runtime_resume_and_get(cdev->dev); + if (ret < 0 && ret != -EACCES) + return ret; ret = catpt_ipc_get_fw_version(cdev, &version); -- cgit v1.2.3 From 7213170a9515109322f75c08b5268d8e9cdad8e4 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 16 Jun 2022 17:04:18 -0500 Subject: ASoC: Intel: skylake: skl-pcm: use pm_runtime_resume_and_get() The current code does not check for errors and does not release the reference on errors. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Kai Vehmanen Reviewed-by: Ranjani Sridharan Acked-by: Cezary Rojewski Reviewed-by: Cezary Rojewski Link: https://lore.kernel.org/r/20220616220427.136036-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/skylake/skl-pcm.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 55f310e91b55..9d72ebd812af 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -1380,7 +1380,10 @@ static int skl_platform_soc_probe(struct snd_soc_component *component) const struct skl_dsp_ops *ops; int ret; - pm_runtime_get_sync(component->dev); + ret = pm_runtime_resume_and_get(component->dev); + if (ret < 0 && ret != -EACCES) + return ret; + if (bus->ppcap) { skl->component = component; -- cgit v1.2.3 From ddea4bbf287b6028eaa15a185d0693856956ecf2 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 16 Jun 2022 17:04:20 -0500 Subject: ASoC: wcd-mbhc-v2: use pm_runtime_resume_and_get() simplify the flow. No functionality change, except that on -EACCESS the reference count will be decreased. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Kai Vehmanen Reviewed-by: Ranjani Sridharan Reviewed-by: Srinivas Kandagatla Reviewed-by: Cezary Rojewski Link: https://lore.kernel.org/r/20220616220427.136036-5-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/wcd-mbhc-v2.c | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wcd-mbhc-v2.c b/sound/soc/codecs/wcd-mbhc-v2.c index 31009283e7d4..98baef594bf3 100644 --- a/sound/soc/codecs/wcd-mbhc-v2.c +++ b/sound/soc/codecs/wcd-mbhc-v2.c @@ -714,12 +714,11 @@ static int wcd_mbhc_initialise(struct wcd_mbhc *mbhc) struct snd_soc_component *component = mbhc->component; int ret; - ret = pm_runtime_get_sync(component->dev); + ret = pm_runtime_resume_and_get(component->dev); if (ret < 0 && ret != -EACCES) { dev_err_ratelimited(component->dev, - "pm_runtime_get_sync failed in %s, ret %d\n", + "pm_runtime_resume_and_get failed in %s, ret %d\n", __func__, ret); - pm_runtime_put_noidle(component->dev); return ret; } @@ -1097,12 +1096,11 @@ static void wcd_correct_swch_plug(struct work_struct *work) mbhc = container_of(work, struct wcd_mbhc, correct_plug_swch); component = mbhc->component; - ret = pm_runtime_get_sync(component->dev); + ret = pm_runtime_resume_and_get(component->dev); if (ret < 0 && ret != -EACCES) { dev_err_ratelimited(component->dev, - "pm_runtime_get_sync failed in %s, ret %d\n", + "pm_runtime_resume_and_get failed in %s, ret %d\n", __func__, ret); - pm_runtime_put_noidle(component->dev); return; } micbias_mv = wcd_mbhc_get_micbias(mbhc); -- cgit v1.2.3 From 9a1a28610a1c49bf93777d017aa3fe121eef944e Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 16 Jun 2022 17:04:21 -0500 Subject: ASoC: wsa881x: use pm_runtime_resume_and_get() simplify the flow. No functionality change, except that on -EACCESS the reference count will be decreased. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Kai Vehmanen Reviewed-by: Ranjani Sridharan Reviewed-by: Srinivas Kandagatla Reviewed-by: Cezary Rojewski Link: https://lore.kernel.org/r/20220616220427.136036-6-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/wsa881x.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wsa881x.c b/sound/soc/codecs/wsa881x.c index f3a56f3ce487..dc954b85a988 100644 --- a/sound/soc/codecs/wsa881x.c +++ b/sound/soc/codecs/wsa881x.c @@ -749,11 +749,9 @@ static int wsa881x_put_pa_gain(struct snd_kcontrol *kc, unsigned int mask = (1 << fls(max)) - 1; int val, ret, min_gain, max_gain; - ret = pm_runtime_get_sync(comp->dev); - if (ret < 0 && ret != -EACCES) { - pm_runtime_put_noidle(comp->dev); + ret = pm_runtime_resume_and_get(comp->dev); + if (ret < 0 && ret != -EACCES) return ret; - } max_gain = (max - ucontrol->value.integer.value[0]) & mask; /* -- cgit v1.2.3 From 8c8a13e83c29472044d733dfb1fced2ccd025d35 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 16 Jun 2022 17:04:22 -0500 Subject: ASoC: rockchip: i2s_tdm: use pm_runtime_resume_and_get() simplify the flow. No functionality change, except that on -EACCESS the reference count will be decreased. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Kai Vehmanen Reviewed-by: Ranjani Sridharan Reviewed-by: Cezary Rojewski Link: https://lore.kernel.org/r/20220616220427.136036-7-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/rockchip/rockchip_i2s_tdm.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/rockchip/rockchip_i2s_tdm.c b/sound/soc/rockchip/rockchip_i2s_tdm.c index 48b3ecfa58b4..70542a402477 100644 --- a/sound/soc/rockchip/rockchip_i2s_tdm.c +++ b/sound/soc/rockchip/rockchip_i2s_tdm.c @@ -404,11 +404,9 @@ static int rockchip_i2s_tdm_set_fmt(struct snd_soc_dai *cpu_dai, int ret; bool is_tdm = i2s_tdm->tdm_mode; - ret = pm_runtime_get_sync(cpu_dai->dev); - if (ret < 0 && ret != -EACCES) { - pm_runtime_put_noidle(cpu_dai->dev); + ret = pm_runtime_resume_and_get(cpu_dai->dev); + if (ret < 0 && ret != -EACCES) return ret; - } mask = I2S_CKR_MSS_MASK; switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { -- cgit v1.2.3 From 37cb8a58013fc6ca2febaed355f6559012699542 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 16 Jun 2022 17:04:23 -0500 Subject: ASoC: fsl: fsl_sai: use pm_runtime_resume_and_get() Simplify the flow. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Kai Vehmanen Reviewed-by: Ranjani Sridharan Reviewed-by: Cezary Rojewski Link: https://lore.kernel.org/r/20220616220427.136036-8-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 4f5bd9597c74..b6407d4d3e09 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -1141,11 +1141,9 @@ static int fsl_sai_probe(struct platform_device *pdev) goto err_pm_disable; } - ret = pm_runtime_get_sync(dev); - if (ret < 0) { - pm_runtime_put_noidle(dev); + ret = pm_runtime_resume_and_get(dev); + if (ret < 0) goto err_pm_get_sync; - } /* Get sai version */ ret = fsl_sai_check_version(dev); -- cgit v1.2.3 From 57d714535051b1baca9ffd92e79fbda1fae3177a Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 16 Jun 2022 17:04:24 -0500 Subject: ASoC: img: img-i2s-out: use pm_runtime_resume_and_get() Simplify the flow. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Kai Vehmanen Reviewed-by: Ranjani Sridharan Reviewed-by: Cezary Rojewski Link: https://lore.kernel.org/r/20220616220427.136036-9-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/img/img-i2s-out.c | 12 ++++-------- 1 file changed, 4 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/img/img-i2s-out.c b/sound/soc/img/img-i2s-out.c index 9ec6fc528e2b..50a522aca419 100644 --- a/sound/soc/img/img-i2s-out.c +++ b/sound/soc/img/img-i2s-out.c @@ -346,11 +346,9 @@ static int img_i2s_out_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) chan_control_mask = IMG_I2S_OUT_CHAN_CTL_CLKT_MASK; - ret = pm_runtime_get_sync(i2s->dev); - if (ret < 0) { - pm_runtime_put_noidle(i2s->dev); + ret = pm_runtime_resume_and_get(i2s->dev); + if (ret < 0) return ret; - } img_i2s_out_disable(i2s); @@ -482,11 +480,9 @@ static int img_i2s_out_probe(struct platform_device *pdev) if (ret) goto err_pm_disable; } - ret = pm_runtime_get_sync(&pdev->dev); - if (ret < 0) { - pm_runtime_put_noidle(&pdev->dev); + ret = pm_runtime_resume_and_get(&pdev->dev); + if (ret < 0) goto err_suspend; - } reg = IMG_I2S_OUT_CTL_FRM_SIZE_MASK; img_i2s_out_writel(i2s, reg, IMG_I2S_OUT_CTL); -- cgit v1.2.3 From 76a6f4537650e6d2211f34661de35630487c7c64 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 16 Jun 2022 17:04:25 -0500 Subject: ASoC: rockchip: pdm: use pm_runtime_resume_and_get() Simplify the flow. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Kai Vehmanen Reviewed-by: Ranjani Sridharan Reviewed-by: Cezary Rojewski Link: https://lore.kernel.org/r/20220616220427.136036-10-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/rockchip/rockchip_pdm.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/rockchip/rockchip_pdm.c b/sound/soc/rockchip/rockchip_pdm.c index 64d9891b6434..066e29b7879d 100644 --- a/sound/soc/rockchip/rockchip_pdm.c +++ b/sound/soc/rockchip/rockchip_pdm.c @@ -688,11 +688,9 @@ static int rockchip_pdm_resume(struct device *dev) struct rk_pdm_dev *pdm = dev_get_drvdata(dev); int ret; - ret = pm_runtime_get_sync(dev); - if (ret < 0) { - pm_runtime_put(dev); + ret = pm_runtime_resume_and_get(dev); + if (ret < 0) return ret; - } ret = regcache_sync(pdm->regmap); -- cgit v1.2.3 From 05b71fb2a5014d2430ff6c5678db021c67afa9ec Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 16 Jun 2022 17:04:26 -0500 Subject: ASoC: tas2552: use pm_runtime_resume_and_get() The use of pm_runtime_get_sync() is buggy with no use of put_noidle() on error. Use pm_runtime_resume_and_get() instead. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Kai Vehmanen Reviewed-by: Ranjani Sridharan Reviewed-by: Cezary Rojewski Link: https://lore.kernel.org/r/20220616220427.136036-11-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/tas2552.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index c98a9332dcc0..da744cfae611 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -581,7 +581,7 @@ static int tas2552_component_probe(struct snd_soc_component *component) gpiod_set_value(tas2552->enable_gpio, 1); - ret = pm_runtime_get_sync(component->dev); + ret = pm_runtime_resume_and_get(component->dev); if (ret < 0) { dev_err(component->dev, "Enabling device failed: %d\n", ret); -- cgit v1.2.3 From cecc81d6a5deb094bdbc6a1d7f2c014ba9b71cf8 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 16 Jun 2022 17:04:27 -0500 Subject: ASoC: ti: davinci-mcasp: use pm_runtime_resume_and_get() The use of pm_runtime_get_sync() is buggy with no use of put_noidle on error. Use pm_runtime_resume_and_get() instead. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Kai Vehmanen Reviewed-by: Ranjani Sridharan Acked-by: Peter Ujfalusi Reviewed-by: Cezary Rojewski Link: https://lore.kernel.org/r/20220616220427.136036-12-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/ti/davinci-mcasp.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index e2aab4729f3a..0201000b619f 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -2111,8 +2111,7 @@ static int davinci_mcasp_gpio_request(struct gpio_chip *chip, unsigned offset) } /* Do not change the PIN yet */ - - return pm_runtime_get_sync(mcasp->dev); + return pm_runtime_resume_and_get(mcasp->dev); } static void davinci_mcasp_gpio_free(struct gpio_chip *chip, unsigned offset) -- cgit v1.2.3 From 25ae1a04da0d32c22db0b018e5668129b91fa104 Mon Sep 17 00:00:00 2001 From: Arnaud Ferraris Date: Mon, 20 Jun 2022 22:54:51 -0500 Subject: ASoC: sun50i-codec-analog: Add support for internal bias In order to properly bias headset microphones, there should be a pull-up resistor between pins HBIAS and MIC2P. This can be an external resistor, but the codec also provides an internal 2.2K resistor which is enabled by a register. This patch enables or disables the internal bias resistor based on a device tree property. Signed-off-by: Arnaud Ferraris [Samuel: split binding and implementation; move to device probe] Signed-off-by: Samuel Holland Link: https://lore.kernel.org/r/20220621035452.60272-3-samuel@sholland.org Signed-off-by: Mark Brown --- sound/soc/sunxi/sun50i-codec-analog.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound') diff --git a/sound/soc/sunxi/sun50i-codec-analog.c b/sound/soc/sunxi/sun50i-codec-analog.c index a41e25ad0aaf..e1e5e8de0130 100644 --- a/sound/soc/sunxi/sun50i-codec-analog.c +++ b/sound/soc/sunxi/sun50i-codec-analog.c @@ -117,6 +117,7 @@ #define SUN50I_ADDA_HS_MBIAS_CTRL_MMICBIASEN 7 #define SUN50I_ADDA_JACK_MIC_CTRL 0x1d +#define SUN50I_ADDA_JACK_MIC_CTRL_INNERRESEN 6 #define SUN50I_ADDA_JACK_MIC_CTRL_HMICBIASEN 5 /* mixer controls */ @@ -507,6 +508,7 @@ static int sun50i_codec_analog_probe(struct platform_device *pdev) { struct regmap *regmap; void __iomem *base; + bool enable; base = devm_platform_ioremap_resource(pdev, 0); if (IS_ERR(base)) { @@ -520,6 +522,12 @@ static int sun50i_codec_analog_probe(struct platform_device *pdev) return PTR_ERR(regmap); } + enable = device_property_read_bool(&pdev->dev, + "allwinner,internal-bias-resistor"); + regmap_update_bits(regmap, SUN50I_ADDA_JACK_MIC_CTRL, + BIT(SUN50I_ADDA_JACK_MIC_CTRL_INNERRESEN), + enable << SUN50I_ADDA_JACK_MIC_CTRL_INNERRESEN); + return devm_snd_soc_register_component(&pdev->dev, &sun50i_codec_analog_cmpnt_drv, NULL, 0); -- cgit v1.2.3 From c111c2ddb3fdfca06bb5c7a56db7f97d6d9ea640 Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Fri, 17 Jun 2022 15:44:31 +0800 Subject: ASoC: fsl_sai: Add PDM daifmt support PDM format is used for 1-bit stream, so clear the FBT and SYWD, and the each dataline only has one channel data. Signed-off-by: Shengjiu Wang Link: https://lore.kernel.org/r/1655451877-16382-2-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 17 +++++++++++++++-- sound/soc/fsl/fsl_sai.h | 1 + 2 files changed, 16 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 4f5bd9597c74..d11ee3b6f163 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -224,6 +224,7 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, if (!sai->is_lsb_first) val_cr4 |= FSL_SAI_CR4_MF; + sai->is_pdm_mode = false; /* DAI mode */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: @@ -262,6 +263,11 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, val_cr2 |= FSL_SAI_CR2_BCP; sai->is_dsp_mode = true; break; + case SND_SOC_DAIFMT_PDM: + val_cr2 |= FSL_SAI_CR2_BCP; + val_cr4 &= ~FSL_SAI_CR4_MF; + sai->is_pdm_mode = true; + break; case SND_SOC_DAIFMT_RIGHT_J: /* To be done */ default: @@ -470,6 +476,13 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, pins = DIV_ROUND_UP(channels, slots); + /* + * PDM mode, channels are independent + * each channels are on one dataline/FIFO. + */ + if (sai->is_pdm_mode) + pins = channels; + if (!sai->is_consumer_mode) { if (sai->bclk_ratio) ret = fsl_sai_set_bclk(cpu_dai, tx, @@ -492,13 +505,13 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, } } - if (!sai->is_dsp_mode) + if (!sai->is_dsp_mode && !sai->is_pdm_mode) val_cr4 |= FSL_SAI_CR4_SYWD(slot_width); val_cr5 |= FSL_SAI_CR5_WNW(slot_width); val_cr5 |= FSL_SAI_CR5_W0W(slot_width); - if (sai->is_lsb_first) + if (sai->is_lsb_first || sai->is_pdm_mode) val_cr5 |= FSL_SAI_CR5_FBT(0); else val_cr5 |= FSL_SAI_CR5_FBT(word_width - 1); diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 1c8f5ca07f9d..bc2a86a413e1 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -259,6 +259,7 @@ struct fsl_sai { bool is_consumer_mode; bool is_lsb_first; bool is_dsp_mode; + bool is_pdm_mode; bool synchronous[2]; unsigned int mclk_id[2]; -- cgit v1.2.3 From 4665770407de8af3b24250cec2209eaf58546f8a Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Fri, 17 Jun 2022 15:44:32 +0800 Subject: ASoC: fsl_sai: Add DSD bit format support Support DSD_U8, DSD_U16_LE, DSD_U32_LE. Signed-off-by: Shengjiu Wang Link: https://lore.kernel.org/r/1655451877-16382-3-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.h | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index bc2a86a413e1..e28a49ce12ef 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -11,7 +11,10 @@ #define FSL_SAI_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE |\ - SNDRV_PCM_FMTBIT_S32_LE) + SNDRV_PCM_FMTBIT_S32_LE |\ + SNDRV_PCM_FMTBIT_DSD_U8 |\ + SNDRV_PCM_FMTBIT_DSD_U16_LE |\ + SNDRV_PCM_FMTBIT_DSD_U32_LE) /* SAI Register Map Register */ #define FSL_SAI_VERID 0x00 /* SAI Version ID Register */ -- cgit v1.2.3 From 0d11bab8ef3e5540dfba111947dbd8dcfb813150 Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Fri, 17 Jun 2022 15:44:33 +0800 Subject: ASoC: fsl_sai: Add support for more sample rates Add support for more sample rates, because PDM format bitstream has higher sample rates. for example DSD512 format, the bit clock is 22.5792MHz, if the word width is U8_LE, then the max sample rate is 2822400. Signed-off-by: Shengjiu Wang Link: https://lore.kernel.org/r/1655451877-16382-4-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index d11ee3b6f163..9d2828b55c07 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -30,7 +30,8 @@ static const unsigned int fsl_sai_rates[] = { 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000, 64000, - 88200, 96000, 176400, 192000 + 88200, 96000, 176400, 192000, 352800, + 384000, 705600, 768000, 1411200, 2822400, }; static const struct snd_pcm_hw_constraint_list fsl_sai_rate_constraints = { @@ -763,7 +764,7 @@ static struct snd_soc_dai_driver fsl_sai_dai_template = { .channels_min = 1, .channels_max = 32, .rate_min = 8000, - .rate_max = 192000, + .rate_max = 2822400, .rates = SNDRV_PCM_RATE_KNOT, .formats = FSL_SAI_FORMATS, }, @@ -772,7 +773,7 @@ static struct snd_soc_dai_driver fsl_sai_dai_template = { .channels_min = 1, .channels_max = 32, .rate_min = 8000, - .rate_max = 192000, + .rate_max = 2822400, .rates = SNDRV_PCM_RATE_KNOT, .formats = FSL_SAI_FORMATS, }, -- cgit v1.2.3 From b4ee8a913e617a2d0f19226225bc025c8640bf34 Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Fri, 17 Jun 2022 15:44:34 +0800 Subject: ASoc: fsl_sai: Add pinctrl operation for PDM and DSD With DSD format, the pinctrl is different compare with I2S format, because one dataline only has one channel data, and the codec always mux the LRCLK pin to DSD data line, and on i.MX8MQ the BCLK pin can route to codec on DSD case for the MCLK is too high. Add pinctrl operation that the pinctrl can be switched on runtime according to the I2S format or DSD format Signed-off-by: Viorel Suman Signed-off-by: Shengjiu Wang Link: https://lore.kernel.org/r/1655451877-16382-5-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 70 ++++++++++++++++++++++++++++++++++++++++++------- sound/soc/fsl/fsl_sai.h | 2 ++ 2 files changed, 63 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 9d2828b55c07..ddfe28cb7df0 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -10,6 +10,7 @@ #include #include #include +#include #include #include #include @@ -57,6 +58,31 @@ static inline bool fsl_sai_dir_is_synced(struct fsl_sai *sai, int dir) return !sai->synchronous[dir] && sai->synchronous[adir]; } +static struct pinctrl_state *fsl_sai_get_pins_state(struct fsl_sai *sai, u32 bclk) +{ + struct pinctrl_state *state = 0; + + if (sai->is_pdm_mode) { + /* DSD512@44.1kHz, DSD512@48kHz */ + if (bclk >= 22579200) + state = pinctrl_lookup_state(sai->pinctrl, "dsd512"); + + /* Get default DSD state */ + if (IS_ERR_OR_NULL(state)) + state = pinctrl_lookup_state(sai->pinctrl, "dsd"); + } else { + /* 706k32b2c, 768k32b2c, etc */ + if (bclk >= 45158400) + state = pinctrl_lookup_state(sai->pinctrl, "pcm_b2m"); + } + + /* Get default state */ + if (IS_ERR_OR_NULL(state)) + state = pinctrl_lookup_state(sai->pinctrl, "default"); + + return state; +} + static irqreturn_t fsl_sai_isr(int irq, void *devid) { struct fsl_sai *sai = (struct fsl_sai *)devid; @@ -466,7 +492,7 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, u32 slots = (channels == 1) ? 2 : channels; u32 slot_width = word_width; int adir = tx ? RX : TX; - u32 pins; + u32 pins, bclk; int ret; if (sai->slots) @@ -484,15 +510,21 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, if (sai->is_pdm_mode) pins = channels; + bclk = params_rate(params) * (sai->bclk_ratio ? sai->bclk_ratio : slots * slot_width); + + if (!IS_ERR_OR_NULL(sai->pinctrl)) { + sai->pins_state = fsl_sai_get_pins_state(sai, bclk); + if (!IS_ERR_OR_NULL(sai->pins_state)) { + ret = pinctrl_select_state(sai->pinctrl, sai->pins_state); + if (ret) { + dev_err(cpu_dai->dev, "failed to set proper pins state: %d\n", ret); + return ret; + } + } + } + if (!sai->is_consumer_mode) { - if (sai->bclk_ratio) - ret = fsl_sai_set_bclk(cpu_dai, tx, - sai->bclk_ratio * - params_rate(params)); - else - ret = fsl_sai_set_bclk(cpu_dai, tx, - slots * slot_width * - params_rate(params)); + ret = fsl_sai_set_bclk(cpu_dai, tx, bclk); if (ret) return ret; @@ -757,6 +789,23 @@ static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai) return 0; } +static int fsl_sai_dai_resume(struct snd_soc_component *component) +{ + struct fsl_sai *sai = snd_soc_component_get_drvdata(component); + struct device *dev = &sai->pdev->dev; + int ret; + + if (!IS_ERR_OR_NULL(sai->pinctrl) && !IS_ERR_OR_NULL(sai->pins_state)) { + ret = pinctrl_select_state(sai->pinctrl, sai->pins_state); + if (ret) { + dev_err(dev, "failed to set proper pins state: %d\n", ret); + return ret; + } + } + + return 0; +} + static struct snd_soc_dai_driver fsl_sai_dai_template = { .probe = fsl_sai_dai_probe, .playback = { @@ -782,6 +831,7 @@ static struct snd_soc_dai_driver fsl_sai_dai_template = { static const struct snd_soc_component_driver fsl_component = { .name = "fsl-sai", + .resume = fsl_sai_dai_resume, }; static struct reg_default fsl_sai_reg_defaults_ofs0[] = { @@ -1147,6 +1197,8 @@ static int fsl_sai_probe(struct platform_device *pdev) sai->dma_params_rx.maxburst = FSL_SAI_MAXBURST_RX; sai->dma_params_tx.maxburst = FSL_SAI_MAXBURST_TX; + sai->pinctrl = devm_pinctrl_get(&pdev->dev); + platform_set_drvdata(pdev, sai); pm_runtime_enable(dev); if (!pm_runtime_enabled(dev)) { diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index e28a49ce12ef..c0b6bc42fc3c 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -278,6 +278,8 @@ struct fsl_sai { struct fsl_sai_verid verid; struct fsl_sai_param param; struct pm_qos_request pm_qos_req; + struct pinctrl *pinctrl; + struct pinctrl_state *pins_state; }; #define TX 1 -- cgit v1.2.3 From cd640ca20095ed3b9306981f0064313a54fd4568 Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Fri, 17 Jun 2022 15:44:35 +0800 Subject: ASoC: fsl_sai: Make res a member of struct fsl_sai The resource info need to be accessed by hw_params() function for multi fifo case, the start address may be not the FIFO0. So move it to be a member of struct fsl_sai. Signed-off-by: Shengjiu Wang Link: https://lore.kernel.org/r/1655451877-16382-6-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 7 +++---- sound/soc/fsl/fsl_sai.h | 1 + 2 files changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index ddfe28cb7df0..86aa0baba848 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -1077,7 +1077,6 @@ static int fsl_sai_probe(struct platform_device *pdev) struct device *dev = &pdev->dev; struct fsl_sai *sai; struct regmap *gpr; - struct resource *res; void __iomem *base; char tmp[8]; int irq, ret, i; @@ -1092,7 +1091,7 @@ static int fsl_sai_probe(struct platform_device *pdev) sai->is_lsb_first = of_property_read_bool(np, "lsb-first"); - base = devm_platform_get_and_ioremap_resource(pdev, 0, &res); + base = devm_platform_get_and_ioremap_resource(pdev, 0, &sai->res); if (IS_ERR(base)) return PTR_ERR(base); @@ -1192,8 +1191,8 @@ static int fsl_sai_probe(struct platform_device *pdev) MCLK_DIR(index)); } - sai->dma_params_rx.addr = res->start + FSL_SAI_RDR0; - sai->dma_params_tx.addr = res->start + FSL_SAI_TDR0; + sai->dma_params_rx.addr = sai->res->start + FSL_SAI_RDR0; + sai->dma_params_tx.addr = sai->res->start + FSL_SAI_TDR0; sai->dma_params_rx.maxburst = FSL_SAI_MAXBURST_RX; sai->dma_params_tx.maxburst = FSL_SAI_MAXBURST_TX; diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index c0b6bc42fc3c..4d657edc9c9f 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -258,6 +258,7 @@ struct fsl_sai { struct regmap *regmap; struct clk *bus_clk; struct clk *mclk_clk[FSL_SAI_MCLK_MAX]; + struct resource *res; bool is_consumer_mode; bool is_lsb_first; -- cgit v1.2.3 From e3f4e5b1a3e654d518155b37c7b2084cbce9d1a7 Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Fri, 17 Jun 2022 15:44:37 +0800 Subject: ASoC: fsl_sai: Configure dataline/FIFO information from dts property The SAI has multiple successive FIFO registers, but in some use case the required dataline/FIFOs are not successive, so need get such information from dts property "fsl,dataline" fsl,dataline has 3 values for each configuration: first one means the type: I2S(1) or DSD(2), second one is dataline mask for 'rx', third one is dataline mask for 'tx'. Also set dma peripheral address and TRCE bits according to data lane. Signed-off-by: Shengjiu Wang Signed-off-by: Viorel Suman Link: https://lore.kernel.org/r/1655451877-16382-8-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 161 ++++++++++++++++++++++++++++++++++++++++++++++-- sound/soc/fsl/fsl_sai.h | 17 +++++ 2 files changed, 174 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 86aa0baba848..f5eabb0b10e8 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -487,13 +487,18 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, unsigned int ofs = sai->soc_data->reg_offset; bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; unsigned int channels = params_channels(params); + struct snd_dmaengine_dai_dma_data *dma_params; + struct fsl_sai_dl_cfg *dl_cfg = sai->dl_cfg; u32 word_width = params_width(params); + int trce_mask = 0, dl_cfg_idx = 0; + int dl_cfg_cnt = sai->dl_cfg_cnt; + u32 dl_type = FSL_SAI_DL_I2S; u32 val_cr4 = 0, val_cr5 = 0; u32 slots = (channels == 1) ? 2 : channels; u32 slot_width = word_width; int adir = tx ? RX : TX; u32 pins, bclk; - int ret; + int ret, i; if (sai->slots) slots = sai->slots; @@ -507,8 +512,22 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, * PDM mode, channels are independent * each channels are on one dataline/FIFO. */ - if (sai->is_pdm_mode) + if (sai->is_pdm_mode) { pins = channels; + dl_type = FSL_SAI_DL_PDM; + } + + for (i = 0; i < dl_cfg_cnt; i++) { + if (dl_cfg[i].type == dl_type && dl_cfg[i].pins[tx] == pins) { + dl_cfg_idx = i; + break; + } + } + + if (hweight8(dl_cfg[dl_cfg_idx].mask[tx]) < pins) { + dev_err(cpu_dai->dev, "channel not supported\n"); + return -EINVAL; + } bclk = params_rate(params) * (sai->bclk_ratio ? sai->bclk_ratio : slots * slot_width); @@ -571,13 +590,28 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, FSL_SAI_CR5_FBT_MASK, val_cr5); } - if (sai->soc_data->pins > 1) + if (hweight8(dl_cfg[dl_cfg_idx].mask[tx]) <= 1) + regmap_update_bits(sai->regmap, FSL_SAI_xCR4(tx, ofs), + FSL_SAI_CR4_FCOMB_MASK, 0); + else regmap_update_bits(sai->regmap, FSL_SAI_xCR4(tx, ofs), FSL_SAI_CR4_FCOMB_MASK, FSL_SAI_CR4_FCOMB_SOFT); + dma_params = tx ? &sai->dma_params_tx : &sai->dma_params_rx; + dma_params->addr = sai->res->start + FSL_SAI_xDR0(tx) + + dl_cfg[dl_cfg_idx].start_off[tx] * 0x4; + + /* Find a proper tcre setting */ + for (i = 0; i < sai->soc_data->pins; i++) { + trce_mask = (1 << (i + 1)) - 1; + if (hweight8(dl_cfg[dl_cfg_idx].mask[tx] & trce_mask) == pins) + break; + } + regmap_update_bits(sai->regmap, FSL_SAI_xCR3(tx, ofs), FSL_SAI_CR3_TRCE_MASK, - FSL_SAI_CR3_TRCE((1 << pins) - 1)); + FSL_SAI_CR3_TRCE((dl_cfg[dl_cfg_idx].mask[tx] & trce_mask))); + regmap_update_bits(sai->regmap, FSL_SAI_xCR4(tx, ofs), FSL_SAI_CR4_SYWD_MASK | FSL_SAI_CR4_FRSZ_MASK | FSL_SAI_CR4_CHMOD_MASK, @@ -1068,6 +1102,118 @@ static int fsl_sai_check_version(struct device *dev) return 0; } +/* + * Calculate the offset between first two datalines, don't + * different offset in one case. + */ +static unsigned int fsl_sai_calc_dl_off(unsigned long dl_mask) +{ + int fbidx, nbidx, offset; + + fbidx = find_first_bit(&dl_mask, FSL_SAI_DL_NUM); + nbidx = find_next_bit(&dl_mask, FSL_SAI_DL_NUM, fbidx + 1); + offset = nbidx - fbidx - 1; + + return (offset < 0 || offset >= (FSL_SAI_DL_NUM - 1) ? 0 : offset); +} + +/* + * read the fsl,dataline property from dts file. + * It has 3 value for each configuration, first one means the type: + * I2S(1) or PDM(2), second one is dataline mask for 'rx', third one is + * dataline mask for 'tx'. for example + * + * fsl,dataline = <1 0xff 0xff 2 0xff 0x11>, + * + * It means I2S type rx mask is 0xff, tx mask is 0xff, PDM type + * rx mask is 0xff, tx mask is 0x11 (dataline 1 and 4 enabled). + * + */ +static int fsl_sai_read_dlcfg(struct fsl_sai *sai) +{ + struct platform_device *pdev = sai->pdev; + struct device_node *np = pdev->dev.of_node; + struct device *dev = &pdev->dev; + int ret, elems, i, index, num_cfg; + char *propname = "fsl,dataline"; + struct fsl_sai_dl_cfg *cfg; + unsigned long dl_mask; + unsigned int soc_dl; + u32 rx, tx, type; + + elems = of_property_count_u32_elems(np, propname); + + if (elems <= 0) { + elems = 0; + } else if (elems % 3) { + dev_err(dev, "Number of elements must be divisible to 3.\n"); + return -EINVAL; + } + + num_cfg = elems / 3; + /* Add one more for default value */ + cfg = devm_kzalloc(&pdev->dev, (num_cfg + 1) * sizeof(*cfg), GFP_KERNEL); + if (!cfg) + return -ENOMEM; + + /* Consider default value "0 0xFF 0xFF" if property is missing */ + soc_dl = BIT(sai->soc_data->pins) - 1; + cfg[0].type = FSL_SAI_DL_DEFAULT; + cfg[0].pins[0] = sai->soc_data->pins; + cfg[0].mask[0] = soc_dl; + cfg[0].start_off[0] = 0; + cfg[0].next_off[0] = 0; + + cfg[0].pins[1] = sai->soc_data->pins; + cfg[0].mask[1] = soc_dl; + cfg[0].start_off[1] = 0; + cfg[0].next_off[1] = 0; + for (i = 1, index = 0; i < num_cfg + 1; i++) { + /* + * type of dataline + * 0 means default mode + * 1 means I2S mode + * 2 means PDM mode + */ + ret = of_property_read_u32_index(np, propname, index++, &type); + if (ret) + return -EINVAL; + + ret = of_property_read_u32_index(np, propname, index++, &rx); + if (ret) + return -EINVAL; + + ret = of_property_read_u32_index(np, propname, index++, &tx); + if (ret) + return -EINVAL; + + if ((rx & ~soc_dl) || (tx & ~soc_dl)) { + dev_err(dev, "dataline cfg[%d] setting error, mask is 0x%x\n", i, soc_dl); + return -EINVAL; + } + + rx = rx & soc_dl; + tx = tx & soc_dl; + + cfg[i].type = type; + cfg[i].pins[0] = hweight8(rx); + cfg[i].mask[0] = rx; + dl_mask = rx; + cfg[i].start_off[0] = find_first_bit(&dl_mask, FSL_SAI_DL_NUM); + cfg[i].next_off[0] = fsl_sai_calc_dl_off(rx); + + cfg[i].pins[1] = hweight8(tx); + cfg[i].mask[1] = tx; + dl_mask = tx; + cfg[i].start_off[1] = find_first_bit(&dl_mask, FSL_SAI_DL_NUM); + cfg[i].next_off[1] = fsl_sai_calc_dl_off(tx); + } + + sai->dl_cfg = cfg; + sai->dl_cfg_cnt = num_cfg + 1; + return 0; +} + static int fsl_sai_runtime_suspend(struct device *dev); static int fsl_sai_runtime_resume(struct device *dev); @@ -1134,6 +1280,13 @@ static int fsl_sai_probe(struct platform_device *pdev) else sai->mclk_clk[0] = sai->bus_clk; + /* read dataline mask for rx and tx*/ + ret = fsl_sai_read_dlcfg(sai); + if (ret < 0) { + dev_err(dev, "failed to read dlcfg %d\n", ret); + return ret; + } + irq = platform_get_irq(pdev, 0); if (irq < 0) return irq; diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 4d657edc9c9f..9bb8ced520c8 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -218,6 +218,13 @@ #define PMQOS_CPU_LATENCY BIT(0) +/* Max number of dataline */ +#define FSL_SAI_DL_NUM (8) +/* default dataline type is zero */ +#define FSL_SAI_DL_DEFAULT (0) +#define FSL_SAI_DL_I2S BIT(0) +#define FSL_SAI_DL_PDM BIT(1) + struct fsl_sai_soc_data { bool use_imx_pcm; bool use_edma; @@ -253,6 +260,14 @@ struct fsl_sai_param { u32 dataline; }; +struct fsl_sai_dl_cfg { + unsigned int type; + unsigned int pins[2]; + unsigned int mask[2]; + unsigned int start_off[2]; + unsigned int next_off[2]; +}; + struct fsl_sai { struct platform_device *pdev; struct regmap *regmap; @@ -265,6 +280,8 @@ struct fsl_sai { bool is_dsp_mode; bool is_pdm_mode; bool synchronous[2]; + struct fsl_sai_dl_cfg *dl_cfg; + unsigned int dl_cfg_cnt; unsigned int mclk_id[2]; unsigned int mclk_streams; -- cgit v1.2.3 From ccb0bbe3e93efa1c794176200785737ba65b0131 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Mon, 27 Jun 2022 10:43:35 +0100 Subject: ASoC: samsung: s3c24xx-i2s: Fix typo in DAIFMT handling The conversion of the set_fmt callback to direct clock specification included a small typo, correct the affected code. Reported-by: kernel test robot Signed-off-by: Charles Keepax Reviewed-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20220627094335.3051210-1-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/samsung/s3c24xx-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index 4082ad7cbcc1..c1a314b86b15 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -170,7 +170,7 @@ static int s3c24xx_i2s_set_fmt(struct snd_soc_dai *cpu_dai, pr_debug("hw_params r: IISMOD: %x \n", iismod); switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { - case SND_SOC_DAIFMT_BC_CFC: + case SND_SOC_DAIFMT_BC_FC: iismod |= S3C2410_IISMOD_SLAVE; break; case SND_SOC_DAIFMT_BP_FP: -- cgit v1.2.3 From 17a1ffc7bc4d5b4657d0f3fe5c01778d8fcab9a3 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Mon, 27 Jun 2022 16:34:10 +0200 Subject: ASoC: samsung: s3c-i2s-v2: Allow build for unsupported hardware There is no particular need to restrict building of S3C I2S driver to supported platforms within the C unit, because Kconfig does it. Removing such restricting #ifdef from s3c-i2s-v2 allows compile testing it on other platforms. Signed-off-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20220627143412.477226-1-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown --- sound/soc/samsung/s3c-i2s-v2.c | 11 ----------- 1 file changed, 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c index 1bec72246ed0..2b221cb0ed03 100644 --- a/sound/soc/samsung/s3c-i2s-v2.c +++ b/sound/soc/samsung/s3c-i2s-v2.c @@ -21,17 +21,6 @@ #include "regs-i2s-v2.h" #include "s3c-i2s-v2.h" -#undef S3C_IIS_V2_SUPPORTED - -#if defined(CONFIG_CPU_S3C2412) \ - || defined(CONFIG_ARCH_S3C64XX) || defined(CONFIG_CPU_S5PV210) -#define S3C_IIS_V2_SUPPORTED -#endif - -#ifndef S3C_IIS_V2_SUPPORTED -#error Unsupported CPU model -#endif - #define S3C2412_I2S_DEBUG_CON 0 static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai) -- cgit v1.2.3 From 3e4bac7cf06e46225322f264e7387efe6ddd457e Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Mon, 27 Jun 2022 16:34:11 +0200 Subject: ASoC: samsung: s3c24xx-i2s: Drop unneeded gpio.h include The module does not use anything from gpio.h header. Signed-off-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20220627143412.477226-2-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown --- sound/soc/samsung/s3c24xx-i2s.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index c1a314b86b15..44e93dc16fc3 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -12,7 +12,6 @@ #include #include #include -#include #include #include -- cgit v1.2.3 From f43ff8038e8289ca811b5b89e8cc15083dafe5c4 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Mon, 27 Jun 2022 16:34:12 +0200 Subject: ASoC: samsung: Enable compile test Allow compile testing of Samsung SoC Sound drivers. Signed-off-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20220627143412.477226-3-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown --- sound/soc/samsung/Kconfig | 20 +++++++++++++------- 1 file changed, 13 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index a2221ebb1b6a..2a61e620cd3b 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -33,7 +33,8 @@ config SND_SAMSUNG_I2S config SND_SOC_SAMSUNG_NEO1973_WM8753 tristate "Audio support for Openmoko Neo1973 Smartphones (GTA02)" - depends on MACH_NEO1973_GTA02 + depends on MACH_NEO1973_GTA02 || COMPILE_TEST + depends on SND_SOC_I2C_AND_SPI select SND_S3C24XX_I2S select SND_SOC_WM8753 select SND_SOC_BT_SCO @@ -43,7 +44,8 @@ config SND_SOC_SAMSUNG_NEO1973_WM8753 config SND_SOC_SAMSUNG_JIVE_WM8750 tristate "SoC I2S Audio support for Jive" - depends on MACH_JIVE && I2C + depends on MACH_JIVE && I2C || COMPILE_TEST && ARM + depends on SND_SOC_I2C_AND_SPI select SND_SOC_WM8750 select SND_S3C2412_SOC_I2S help @@ -69,7 +71,7 @@ config SND_SOC_SAMSUNG_SMDK_WM8994 config SND_SOC_SAMSUNG_S3C24XX_UDA134X tristate "SoC I2S Audio support UDA134X wired to a S3C24XX" - depends on ARCH_S3C24XX + depends on ARCH_S3C24XX || COMPILE_TEST select SND_S3C24XX_I2S select SND_SOC_L3 select SND_SOC_UDA134X @@ -81,21 +83,24 @@ config SND_SOC_SAMSUNG_SIMTEC config SND_SOC_SAMSUNG_SIMTEC_TLV320AIC23 tristate "SoC I2S Audio support for TLV320AIC23 on Simtec boards" - depends on ARCH_S3C24XX && I2C + depends on ARCH_S3C24XX || COMPILE_TEST + depends on I2C select SND_S3C24XX_I2S select SND_SOC_TLV320AIC23_I2C select SND_SOC_SAMSUNG_SIMTEC config SND_SOC_SAMSUNG_SIMTEC_HERMES tristate "SoC I2S Audio support for Simtec Hermes board" - depends on ARCH_S3C24XX && I2C + depends on ARCH_S3C24XX || COMPILE_TEST + depends on I2C select SND_S3C24XX_I2S select SND_SOC_TLV320AIC3X select SND_SOC_SAMSUNG_SIMTEC config SND_SOC_SAMSUNG_H1940_UDA1380 tristate "Audio support for the HP iPAQ H1940" - depends on ARCH_H1940 && I2C + depends on ARCH_H1940 || COMPILE_TEST + depends on I2C select SND_S3C24XX_I2S select SND_SOC_UDA1380 help @@ -103,7 +108,8 @@ config SND_SOC_SAMSUNG_H1940_UDA1380 config SND_SOC_SAMSUNG_RX1950_UDA1380 tristate "Audio support for the HP iPAQ RX1950" - depends on MACH_RX1950 && I2C + depends on MACH_RX1950 || COMPILE_TEST + depends on I2C select SND_S3C24XX_I2S select SND_SOC_UDA1380 help -- cgit v1.2.3 From bd10b0dafdcf0ec1677cad70101e1f97b9e28f2e Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Mon, 27 Jun 2022 16:19:00 +0200 Subject: ASoC: samsung: h1940_uda1380: include proepr GPIO consumer header h1940_uda1380 uses gpiod*/GPIOD* so it should include GPIO consumer header. Fixes: 9666e27f90b9 ("ASoC: samsung: h1940: turn into platform driver") Signed-off-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20220627141900.470469-1-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown --- sound/soc/samsung/h1940_uda1380.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c index 907266aee839..fa45a54ab18f 100644 --- a/sound/soc/samsung/h1940_uda1380.c +++ b/sound/soc/samsung/h1940_uda1380.c @@ -8,7 +8,7 @@ // Based on version from Arnaud Patard #include -#include +#include #include #include -- cgit v1.2.3 From 2a2ef688b1b03eea3a5b020d9bef50d015f619be Mon Sep 17 00:00:00 2001 From: Bryan O'Donoghue Date: Tue, 28 Jun 2022 13:04:34 +0100 Subject: ASoC: qcom: lpass: Fix apq8016 compat string to match yaml The documented yaml compat string for the apq8016 is "qcom,apq8016-lpass-cpu" not "qcom,lpass-cpu-apq8016". Looking at the other lpass compat strings the general form is "qcom,socnum-lpass-cpu". We need to fix both the driver and dts to match. Reviewed-by: Srinivas Kandagatla Reviewed-by: Bjorn Andersson Signed-off-by: Bryan O'Donoghue Link: https://lore.kernel.org/r/20220628120435.3044939-2-bryan.odonoghue@linaro.org Signed-off-by: Mark Brown --- sound/soc/qcom/lpass-apq8016.c | 1 + sound/soc/qcom/lpass-cpu.c | 5 +++++ 2 files changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/soc/qcom/lpass-apq8016.c b/sound/soc/qcom/lpass-apq8016.c index 3efa133d1c64..abaf694ee9a3 100644 --- a/sound/soc/qcom/lpass-apq8016.c +++ b/sound/soc/qcom/lpass-apq8016.c @@ -293,6 +293,7 @@ static struct lpass_variant apq8016_data = { static const struct of_device_id apq8016_lpass_cpu_device_id[] __maybe_unused = { { .compatible = "qcom,lpass-cpu-apq8016", .data = &apq8016_data }, + { .compatible = "qcom,apq8016-lpass-cpu", .data = &apq8016_data }, {} }; MODULE_DEVICE_TABLE(of, apq8016_lpass_cpu_device_id); diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index e6846ad2b5fa..53f9bf6581d3 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -1102,6 +1102,11 @@ int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev) if (!match || !match->data) return -EINVAL; + if (of_device_is_compatible(dev->of_node, "qcom,lpass-cpu-apq8016")) { + dev_warn(dev, "%s compatible is deprecated\n", + match->compatible); + } + drvdata->variant = (struct lpass_variant *)match->data; variant = drvdata->variant; -- cgit v1.2.3 From 5f78e1fb7a3ed1acc355145536ddd54f183b635d Mon Sep 17 00:00:00 2001 From: Srinivasa Rao Mandadapu Date: Mon, 27 Jun 2022 16:14:22 +0530 Subject: ASoC: qcom: Add driver support for audioreach solution Add Machine driver support for audioreach solution, which uses ADSP in SC7280 based paltforms. Signed-off-by: Srinivasa Rao Mandadapu Link: https://lore.kernel.org/r/1656326662-14524-1-git-send-email-quic_srivasam@quicinc.com Signed-off-by: Mark Brown --- sound/soc/qcom/sc7280.c | 33 +++++++++++++++++++++++++++++++++ 1 file changed, 33 insertions(+) (limited to 'sound') diff --git a/sound/soc/qcom/sc7280.c b/sound/soc/qcom/sc7280.c index 34cdb99d4ed6..da7469a6a267 100644 --- a/sound/soc/qcom/sc7280.c +++ b/sound/soc/qcom/sc7280.c @@ -19,9 +19,11 @@ #include "../codecs/rt5682s.h" #include "common.h" #include "lpass.h" +#include "qdsp6/q6afe.h" #define DEFAULT_MCLK_RATE 19200000 #define RT5682_PLL_FREQ (48000 * 512) +#define MI2S_BCLK_RATE 1536000 struct sc7280_snd_data { struct snd_soc_card card; @@ -79,6 +81,7 @@ static int sc7280_headset_init(struct snd_soc_pcm_runtime *rtd) case MI2S_PRIMARY: case LPASS_CDC_DMA_RX0: case LPASS_CDC_DMA_TX3: + case TX_CODEC_DMA_TX_3: for_each_rtd_codec_dais(rtd, i, codec_dai) { rval = snd_soc_component_set_jack(component, &pdata->hs_jack, NULL); if (rval != 0 && rval != -ENOTSUPP) { @@ -164,10 +167,14 @@ static int sc7280_init(struct snd_soc_pcm_runtime *rtd) switch (cpu_dai->id) { case MI2S_PRIMARY: case LPASS_CDC_DMA_TX3: + case TX_CODEC_DMA_TX_3: return sc7280_headset_init(rtd); case LPASS_CDC_DMA_RX0: case LPASS_CDC_DMA_VA_TX0: case MI2S_SECONDARY: + case RX_CODEC_DMA_RX_0: + case SECONDARY_MI2S_RX: + case VA_CODEC_DMA_TX_0: return 0; case LPASS_DP_RX: return sc7280_hdmi_init(rtd); @@ -195,6 +202,10 @@ static int sc7280_snd_hw_params(struct snd_pcm_substream *substream, switch (cpu_dai->id) { case LPASS_CDC_DMA_TX3: case LPASS_CDC_DMA_RX0: + case RX_CODEC_DMA_RX_0: + case SECONDARY_MI2S_RX: + case TX_CODEC_DMA_TX_3: + case VA_CODEC_DMA_TX_0: for_each_rtd_codec_dais(rtd, i, codec_dai) { sruntime = snd_soc_dai_get_stream(codec_dai, substream->stream); if (sruntime != ERR_PTR(-ENOTSUPP)) @@ -245,6 +256,9 @@ static int sc7280_snd_prepare(struct snd_pcm_substream *substream) switch (cpu_dai->id) { case LPASS_CDC_DMA_RX0: case LPASS_CDC_DMA_TX3: + case RX_CODEC_DMA_RX_0: + case TX_CODEC_DMA_TX_3: + case VA_CODEC_DMA_TX_0: return sc7280_snd_swr_prepare(substream); default: break; @@ -263,6 +277,9 @@ static int sc7280_snd_hw_free(struct snd_pcm_substream *substream) switch (cpu_dai->id) { case LPASS_CDC_DMA_RX0: case LPASS_CDC_DMA_TX3: + case RX_CODEC_DMA_RX_0: + case TX_CODEC_DMA_TX_3: + case VA_CODEC_DMA_TX_0: if (sruntime && data->stream_prepared[cpu_dai->id]) { sdw_disable_stream(sruntime); sdw_deprepare_stream(sruntime); @@ -291,6 +308,10 @@ static void sc7280_snd_shutdown(struct snd_pcm_substream *substream) SNDRV_PCM_STREAM_PLAYBACK); } break; + case SECONDARY_MI2S_RX: + snd_soc_dai_set_sysclk(cpu_dai, Q6AFE_LPASS_CLK_ID_SEC_MI2S_IBIT, + 0, SNDRV_PCM_STREAM_PLAYBACK); + break; default: break; } @@ -298,14 +319,26 @@ static void sc7280_snd_shutdown(struct snd_pcm_substream *substream) static int sc7280_snd_startup(struct snd_pcm_substream *substream) { + unsigned int fmt = SND_SOC_DAIFMT_CBS_CFS; + unsigned int codec_dai_fmt = SND_SOC_DAIFMT_CBS_CFS; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret = 0; switch (cpu_dai->id) { case MI2S_PRIMARY: ret = sc7280_rt5682_init(rtd); break; + case SECONDARY_MI2S_RX: + codec_dai_fmt |= SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_I2S; + + snd_soc_dai_set_sysclk(cpu_dai, Q6AFE_LPASS_CLK_ID_SEC_MI2S_IBIT, + MI2S_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK); + + snd_soc_dai_set_fmt(cpu_dai, fmt); + snd_soc_dai_set_fmt(codec_dai, codec_dai_fmt); + break; default: break; } -- cgit v1.2.3 From 43b8c7dc85a14f36048a27bb6c627fd49144a8d1 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Wed, 29 Jun 2022 10:06:42 +0100 Subject: ASoC: codecs: add wsa883x amplifier support This patch adds support to WSA8830/WSA8812/WSA8835 Class-D Smart Speaker Amplifier. This Amplifier is primarily interfaced with SoundWire. This patch is tested on SM8450 MTP Board. Signed-off-by: Srinivas Kandagatla Link: https://lore.kernel.org/r/20220629090644.67982-3-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wsa883x.c | 1301 ++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 1313 insertions(+) create mode 100644 sound/soc/codecs/wsa883x.c (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 5a60633a196c..ee7e028e8402 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -308,6 +308,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_WM9712 imply SND_SOC_WM9713 imply SND_SOC_WSA881X + imply SND_SOC_WSA883X imply SND_SOC_ZL38060 help Normally ASoC codec drivers are only built if a machine driver which @@ -1985,6 +1986,15 @@ config SND_SOC_WSA881X This enables support for Qualcomm WSA8810/WSA8815 Class-D Smart Speaker Amplifier. +config SND_SOC_WSA883X + tristate "WSA883X Codec" + depends on SOUNDWIRE + select REGMAP_SOUNDWIRE + tristate + help + This enables support for Qualcomm WSA8830/WSA8835 Class-D + Smart Speaker Amplifier. + config SND_SOC_ZL38060 tristate "Microsemi ZL38060 Connected Home Audio Processor" depends on SPI_MASTER diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index d32026ae326f..60354579fe5c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -338,6 +338,7 @@ snd-soc-wm9712-objs := wm9712.o snd-soc-wm9713-objs := wm9713.o snd-soc-wm-hubs-objs := wm_hubs.o snd-soc-wsa881x-objs := wsa881x.o +snd-soc-wsa883x-objs := wsa883x.o snd-soc-zl38060-objs := zl38060.o # Amp snd-soc-max9877-objs := max9877.o @@ -690,6 +691,7 @@ obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o obj-$(CONFIG_SND_SOC_WM_ADSP) += snd-soc-wm-adsp.o obj-$(CONFIG_SND_SOC_WM_HUBS) += snd-soc-wm-hubs.o obj-$(CONFIG_SND_SOC_WSA881X) += snd-soc-wsa881x.o +obj-$(CONFIG_SND_SOC_WSA883X) += snd-soc-wsa883x.o obj-$(CONFIG_SND_SOC_ZL38060) += snd-soc-zl38060.o # Amp diff --git a/sound/soc/codecs/wsa883x.c b/sound/soc/codecs/wsa883x.c new file mode 100644 index 000000000000..856709ec017e --- /dev/null +++ b/sound/soc/codecs/wsa883x.c @@ -0,0 +1,1301 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * Copyright (c) 2015-2021, The Linux Foundation. All rights reserved. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#define WSA883X_BASE 0x3000 +#define WSA883X_ANA_BG_TSADC_BASE (WSA883X_BASE + 0x00000001) +#define WSA883X_REF_CTRL (WSA883X_ANA_BG_TSADC_BASE + 0x0000) +#define WSA883X_TEST_CTL_0 (WSA883X_ANA_BG_TSADC_BASE + 0x0001) +#define WSA883X_BIAS_0 (WSA883X_ANA_BG_TSADC_BASE + 0x0002) +#define WSA883X_OP_CTL (WSA883X_ANA_BG_TSADC_BASE + 0x0003) +#define WSA883X_IREF_CTL (WSA883X_ANA_BG_TSADC_BASE + 0x0004) +#define WSA883X_ISENS_CTL (WSA883X_ANA_BG_TSADC_BASE + 0x0005) +#define WSA883X_CLK_CTL (WSA883X_ANA_BG_TSADC_BASE + 0x0006) +#define WSA883X_TEST_CTL_1 (WSA883X_ANA_BG_TSADC_BASE + 0x0007) +#define WSA883X_BIAS_1 (WSA883X_ANA_BG_TSADC_BASE + 0x0008) +#define WSA883X_ADC_CTL (WSA883X_ANA_BG_TSADC_BASE + 0x0009) +#define WSA883X_DOUT_MSB (WSA883X_ANA_BG_TSADC_BASE + 0x000A) +#define WSA883X_DOUT_LSB (WSA883X_ANA_BG_TSADC_BASE + 0x000B) +#define WSA883X_VBAT_SNS (WSA883X_ANA_BG_TSADC_BASE + 0x000C) +#define WSA883X_ITRIM_CODE (WSA883X_ANA_BG_TSADC_BASE + 0x000D) + +#define WSA883X_ANA_IVSENSE_BASE (WSA883X_BASE + 0x0000000F) +#define WSA883X_EN (WSA883X_ANA_IVSENSE_BASE + 0x0000) +#define WSA883X_OVERRIDE1 (WSA883X_ANA_IVSENSE_BASE + 0x0001) +#define WSA883X_OVERRIDE2 (WSA883X_ANA_IVSENSE_BASE + 0x0002) +#define WSA883X_VSENSE1 (WSA883X_ANA_IVSENSE_BASE + 0x0003) +#define WSA883X_ISENSE1 (WSA883X_ANA_IVSENSE_BASE + 0x0004) +#define WSA883X_ISENSE2 (WSA883X_ANA_IVSENSE_BASE + 0x0005) +#define WSA883X_ISENSE_CAL (WSA883X_ANA_IVSENSE_BASE + 0x0006) +#define WSA883X_MISC (WSA883X_ANA_IVSENSE_BASE + 0x0007) +#define WSA883X_ADC_0 (WSA883X_ANA_IVSENSE_BASE + 0x0008) +#define WSA883X_ADC_1 (WSA883X_ANA_IVSENSE_BASE + 0x0009) +#define WSA883X_ADC_2 (WSA883X_ANA_IVSENSE_BASE + 0x000A) +#define WSA883X_ADC_3 (WSA883X_ANA_IVSENSE_BASE + 0x000B) +#define WSA883X_ADC_4 (WSA883X_ANA_IVSENSE_BASE + 0x000C) +#define WSA883X_ADC_5 (WSA883X_ANA_IVSENSE_BASE + 0x000D) +#define WSA883X_ADC_6 (WSA883X_ANA_IVSENSE_BASE + 0x000E) +#define WSA883X_ADC_7 (WSA883X_ANA_IVSENSE_BASE + 0x000F) +#define WSA883X_STATUS (WSA883X_ANA_IVSENSE_BASE + 0x0010) + +#define WSA883X_ANA_SPK_TOP_BASE (WSA883X_BASE + 0x00000025) +#define WSA883X_DAC_CTRL_REG (WSA883X_ANA_SPK_TOP_BASE + 0x0000) +#define WSA883X_DAC_EN_DEBUG_REG (WSA883X_ANA_SPK_TOP_BASE + 0x0001) +#define WSA883X_DAC_OPAMP_BIAS1_REG (WSA883X_ANA_SPK_TOP_BASE + 0x0002) +#define WSA883X_DAC_OPAMP_BIAS2_REG (WSA883X_ANA_SPK_TOP_BASE + 0x0003) +#define WSA883X_DAC_VCM_CTRL_REG (WSA883X_ANA_SPK_TOP_BASE + 0x0004) +#define WSA883X_DAC_VOLTAGE_CTRL_REG (WSA883X_ANA_SPK_TOP_BASE + 0x0005) +#define WSA883X_ATEST1_REG (WSA883X_ANA_SPK_TOP_BASE + 0x0006) +#define WSA883X_ATEST2_REG (WSA883X_ANA_SPK_TOP_BASE + 0x0007) +#define WSA883X_SPKR_TOP_BIAS_REG1 (WSA883X_ANA_SPK_TOP_BASE + 0x0008) +#define WSA883X_SPKR_TOP_BIAS_REG2 (WSA883X_ANA_SPK_TOP_BASE + 0x0009) +#define WSA883X_SPKR_TOP_BIAS_REG3 (WSA883X_ANA_SPK_TOP_BASE + 0x000A) +#define WSA883X_SPKR_TOP_BIAS_REG4 (WSA883X_ANA_SPK_TOP_BASE + 0x000B) +#define WSA883X_SPKR_CLIP_DET_REG (WSA883X_ANA_SPK_TOP_BASE + 0x000C) +#define WSA883X_SPKR_DRV_LF_BLK_EN (WSA883X_ANA_SPK_TOP_BASE + 0x000D) +#define WSA883X_SPKR_DRV_LF_EN (WSA883X_ANA_SPK_TOP_BASE + 0x000E) +#define WSA883X_SPKR_DRV_LF_MASK_DCC_CTL (WSA883X_ANA_SPK_TOP_BASE + 0x000F) +#define WSA883X_SPKR_DRV_LF_MISC_CTL (WSA883X_ANA_SPK_TOP_BASE + 0x0010) +#define WSA883X_SPKR_DRV_LF_REG_GAIN (WSA883X_ANA_SPK_TOP_BASE + 0x0011) +#define WSA883X_SPKR_DRV_OS_CAL_CTL (WSA883X_ANA_SPK_TOP_BASE + 0x0012) +#define WSA883X_SPKR_DRV_OS_CAL_CTL1 (WSA883X_ANA_SPK_TOP_BASE + 0x0013) +#define WSA883X_SPKR_PWM_CLK_CTL (WSA883X_ANA_SPK_TOP_BASE + 0x0014) +#define WSA883X_SPKR_PWM_FREQ_SEL_MASK BIT(3) +#define WSA883X_SPKR_PWM_FREQ_F300KHZ 0 +#define WSA883X_SPKR_PWM_FREQ_F600KHZ 1 +#define WSA883X_SPKR_PDRV_HS_CTL (WSA883X_ANA_SPK_TOP_BASE + 0x0015) +#define WSA883X_SPKR_PDRV_LS_CTL (WSA883X_ANA_SPK_TOP_BASE + 0x0016) +#define WSA883X_SPKR_PWRSTG_DBG (WSA883X_ANA_SPK_TOP_BASE + 0x0017) +#define WSA883X_SPKR_OCP_CTL (WSA883X_ANA_SPK_TOP_BASE + 0x0018) +#define WSA883X_SPKR_BBM_CTL (WSA883X_ANA_SPK_TOP_BASE + 0x0019) +#define WSA883X_PA_STATUS0 (WSA883X_ANA_SPK_TOP_BASE + 0x001A) +#define WSA883X_PA_STATUS1 (WSA883X_ANA_SPK_TOP_BASE + 0x001B) +#define WSA883X_PA_STATUS2 (WSA883X_ANA_SPK_TOP_BASE + 0x001C) + +#define WSA883X_ANA_BOOST_BASE (WSA883X_BASE + 0x00000043) +#define WSA883X_EN_CTRL (WSA883X_ANA_BOOST_BASE + 0x0000) +#define WSA883X_CURRENT_LIMIT (WSA883X_ANA_BOOST_BASE + 0x0001) +#define WSA883X_IBIAS1 (WSA883X_ANA_BOOST_BASE + 0x0002) +#define WSA883X_IBIAS2 (WSA883X_ANA_BOOST_BASE + 0x0003) +#define WSA883X_IBIAS3 (WSA883X_ANA_BOOST_BASE + 0x0004) +#define WSA883X_LDO_PROG (WSA883X_ANA_BOOST_BASE + 0x0005) +#define WSA883X_STABILITY_CTRL1 (WSA883X_ANA_BOOST_BASE + 0x0006) +#define WSA883X_STABILITY_CTRL2 (WSA883X_ANA_BOOST_BASE + 0x0007) +#define WSA883X_PWRSTAGE_CTRL1 (WSA883X_ANA_BOOST_BASE + 0x0008) +#define WSA883X_PWRSTAGE_CTRL2 (WSA883X_ANA_BOOST_BASE + 0x0009) +#define WSA883X_BYPASS_1 (WSA883X_ANA_BOOST_BASE + 0x000A) +#define WSA883X_BYPASS_2 (WSA883X_ANA_BOOST_BASE + 0x000B) +#define WSA883X_ZX_CTRL_1 (WSA883X_ANA_BOOST_BASE + 0x000C) +#define WSA883X_ZX_CTRL_2 (WSA883X_ANA_BOOST_BASE + 0x000D) +#define WSA883X_MISC1 (WSA883X_ANA_BOOST_BASE + 0x000E) +#define WSA883X_MISC2 (WSA883X_ANA_BOOST_BASE + 0x000F) +#define WSA883X_GMAMP_SUP1 (WSA883X_ANA_BOOST_BASE + 0x0010) +#define WSA883X_PWRSTAGE_CTRL3 (WSA883X_ANA_BOOST_BASE + 0x0011) +#define WSA883X_PWRSTAGE_CTRL4 (WSA883X_ANA_BOOST_BASE + 0x0012) +#define WSA883X_TEST1 (WSA883X_ANA_BOOST_BASE + 0x0013) +#define WSA883X_SPARE1 (WSA883X_ANA_BOOST_BASE + 0x0014) +#define WSA883X_SPARE2 (WSA883X_ANA_BOOST_BASE + 0x0015) + +#define WSA883X_ANA_PON_LDOL_BASE (WSA883X_BASE + 0x00000059) +#define WSA883X_PON_CTL_0 (WSA883X_ANA_PON_LDOL_BASE + 0x0000) +#define WSA883X_PON_CLT_1 (WSA883X_ANA_PON_LDOL_BASE + 0x0001) +#define WSA883X_PON_CTL_2 (WSA883X_ANA_PON_LDOL_BASE + 0x0002) +#define WSA883X_PON_CTL_3 (WSA883X_ANA_PON_LDOL_BASE + 0x0003) +#define WSA883X_CKWD_CTL_0 (WSA883X_ANA_PON_LDOL_BASE + 0x0004) +#define WSA883X_CKWD_CTL_1 (WSA883X_ANA_PON_LDOL_BASE + 0x0005) +#define WSA883X_CKWD_CTL_2 (WSA883X_ANA_PON_LDOL_BASE + 0x0006) +#define WSA883X_CKSK_CTL_0 (WSA883X_ANA_PON_LDOL_BASE + 0x0007) +#define WSA883X_PADSW_CTL_0 (WSA883X_ANA_PON_LDOL_BASE + 0x0008) +#define WSA883X_TEST_0 (WSA883X_ANA_PON_LDOL_BASE + 0x0009) +#define WSA883X_TEST_1 (WSA883X_ANA_PON_LDOL_BASE + 0x000A) +#define WSA883X_STATUS_0 (WSA883X_ANA_PON_LDOL_BASE + 0x000B) +#define WSA883X_STATUS_1 (WSA883X_ANA_PON_LDOL_BASE + 0x000C) + +#define WSA883X_DIG_CTRL_BASE (WSA883X_BASE + 0x00000400) +#define WSA883X_CHIP_ID0 (WSA883X_DIG_CTRL_BASE + 0x0001) +#define WSA883X_CHIP_ID1 (WSA883X_DIG_CTRL_BASE + 0x0002) +#define WSA883X_CHIP_ID2 (WSA883X_DIG_CTRL_BASE + 0x0003) +#define WSA883X_CHIP_ID3 (WSA883X_DIG_CTRL_BASE + 0x0004) +#define WSA883X_BUS_ID (WSA883X_DIG_CTRL_BASE + 0x0005) +#define WSA883X_CDC_RST_CTL (WSA883X_DIG_CTRL_BASE + 0x0006) +#define WSA883X_TOP_CLK_CFG (WSA883X_DIG_CTRL_BASE + 0x0007) +#define WSA883X_CDC_PATH_MODE (WSA883X_DIG_CTRL_BASE + 0x0008) +#define WSA883X_RXD_MODE_MASK BIT(1) +#define WSA883X_RXD_MODE_NORMAL 0 +#define WSA883X_RXD_MODE_HIFI 1 +#define WSA883X_CDC_CLK_CTL (WSA883X_DIG_CTRL_BASE + 0x0009) +#define WSA883X_SWR_RESET_EN (WSA883X_DIG_CTRL_BASE + 0x000A) +#define WSA883X_RESET_CTL (WSA883X_DIG_CTRL_BASE + 0x000B) +#define WSA883X_PA_FSM_CTL (WSA883X_DIG_CTRL_BASE + 0x0010) +#define WSA883X_GLOBAL_PA_EN_MASK BIT(0) +#define WSA883X_GLOBAL_PA_ENABLE 1 +#define WSA883X_PA_FSM_TIMER0 (WSA883X_DIG_CTRL_BASE + 0x0011) +#define WSA883X_PA_FSM_TIMER1 (WSA883X_DIG_CTRL_BASE + 0x0012) +#define WSA883X_PA_FSM_STA (WSA883X_DIG_CTRL_BASE + 0x0013) +#define WSA883X_PA_FSM_ERR_COND (WSA883X_DIG_CTRL_BASE + 0x0014) +#define WSA883X_PA_FSM_MSK (WSA883X_DIG_CTRL_BASE + 0x0015) +#define WSA883X_PA_FSM_BYP (WSA883X_DIG_CTRL_BASE + 0x0016) +#define WSA883X_PA_FSM_DBG (WSA883X_DIG_CTRL_BASE + 0x0017) +#define WSA883X_TADC_VALUE_CTL (WSA883X_DIG_CTRL_BASE + 0x0020) +#define WSA883X_TEMP_DETECT_CTL (WSA883X_DIG_CTRL_BASE + 0x0021) +#define WSA883X_TEMP_MSB (WSA883X_DIG_CTRL_BASE + 0x0022) +#define WSA883X_TEMP_LSB (WSA883X_DIG_CTRL_BASE + 0x0023) +#define WSA883X_TEMP_CONFIG0 (WSA883X_DIG_CTRL_BASE + 0x0024) +#define WSA883X_TEMP_CONFIG1 (WSA883X_DIG_CTRL_BASE + 0x0025) +#define WSA883X_VBAT_ADC_FLT_CTL (WSA883X_DIG_CTRL_BASE + 0x0026) +#define WSA883X_VBAT_ADC_FLT_EN_MASK BIT(0) +#define WSA883X_VBAT_ADC_COEF_SEL_MASK GENMASK(3, 1) +#define WSA883X_VBAT_ADC_COEF_F_1DIV2 0x0 +#define WSA883X_VBAT_ADC_COEF_F_1DIV16 0x3 +#define WSA883X_VBAT_DIN_MSB (WSA883X_DIG_CTRL_BASE + 0x0027) +#define WSA883X_VBAT_DIN_LSB (WSA883X_DIG_CTRL_BASE + 0x0028) +#define WSA883X_VBAT_DOUT (WSA883X_DIG_CTRL_BASE + 0x0029) +#define WSA883X_SDM_PDM9_LSB (WSA883X_DIG_CTRL_BASE + 0x002A) +#define WSA883X_SDM_PDM9_MSB (WSA883X_DIG_CTRL_BASE + 0x002B) +#define WSA883X_CDC_RX_CTL (WSA883X_DIG_CTRL_BASE + 0x0030) +#define WSA883X_CDC_SPK_DSM_A1_0 (WSA883X_DIG_CTRL_BASE + 0x0031) +#define WSA883X_CDC_SPK_DSM_A1_1 (WSA883X_DIG_CTRL_BASE + 0x0032) +#define WSA883X_CDC_SPK_DSM_A2_0 (WSA883X_DIG_CTRL_BASE + 0x0033) +#define WSA883X_CDC_SPK_DSM_A2_1 (WSA883X_DIG_CTRL_BASE + 0x0034) +#define WSA883X_CDC_SPK_DSM_A3_0 (WSA883X_DIG_CTRL_BASE + 0x0035) +#define WSA883X_CDC_SPK_DSM_A3_1 (WSA883X_DIG_CTRL_BASE + 0x0036) +#define WSA883X_CDC_SPK_DSM_A4_0 (WSA883X_DIG_CTRL_BASE + 0x0037) +#define WSA883X_CDC_SPK_DSM_A4_1 (WSA883X_DIG_CTRL_BASE + 0x0038) +#define WSA883X_CDC_SPK_DSM_A5_0 (WSA883X_DIG_CTRL_BASE + 0x0039) +#define WSA883X_CDC_SPK_DSM_A5_1 (WSA883X_DIG_CTRL_BASE + 0x003A) +#define WSA883X_CDC_SPK_DSM_A6_0 (WSA883X_DIG_CTRL_BASE + 0x003B) +#define WSA883X_CDC_SPK_DSM_A7_0 (WSA883X_DIG_CTRL_BASE + 0x003C) +#define WSA883X_CDC_SPK_DSM_C_0 (WSA883X_DIG_CTRL_BASE + 0x003D) +#define WSA883X_CDC_SPK_DSM_C_1 (WSA883X_DIG_CTRL_BASE + 0x003E) +#define WSA883X_CDC_SPK_DSM_C_2 (WSA883X_DIG_CTRL_BASE + 0x003F) +#define WSA883X_CDC_SPK_DSM_C_3 (WSA883X_DIG_CTRL_BASE + 0x0040) +#define WSA883X_CDC_SPK_DSM_R1 (WSA883X_DIG_CTRL_BASE + 0x0041) +#define WSA883X_CDC_SPK_DSM_R2 (WSA883X_DIG_CTRL_BASE + 0x0042) +#define WSA883X_CDC_SPK_DSM_R3 (WSA883X_DIG_CTRL_BASE + 0x0043) +#define WSA883X_CDC_SPK_DSM_R4 (WSA883X_DIG_CTRL_BASE + 0x0044) +#define WSA883X_CDC_SPK_DSM_R5 (WSA883X_DIG_CTRL_BASE + 0x0045) +#define WSA883X_CDC_SPK_DSM_R6 (WSA883X_DIG_CTRL_BASE + 0x0046) +#define WSA883X_CDC_SPK_DSM_R7 (WSA883X_DIG_CTRL_BASE + 0x0047) +#define WSA883X_CDC_SPK_GAIN_PDM_0 (WSA883X_DIG_CTRL_BASE + 0x0048) +#define WSA883X_CDC_SPK_GAIN_PDM_1 (WSA883X_DIG_CTRL_BASE + 0x0049) +#define WSA883X_CDC_SPK_GAIN_PDM_2 (WSA883X_DIG_CTRL_BASE + 0x004A) +#define WSA883X_PDM_WD_CTL (WSA883X_DIG_CTRL_BASE + 0x004B) +#define WSA883X_PDM_EN_MASK BIT(0) +#define WSA883X_PDM_ENABLE BIT(0) +#define WSA883X_DEM_BYPASS_DATA0 (WSA883X_DIG_CTRL_BASE + 0x004C) +#define WSA883X_DEM_BYPASS_DATA1 (WSA883X_DIG_CTRL_BASE + 0x004D) +#define WSA883X_DEM_BYPASS_DATA2 (WSA883X_DIG_CTRL_BASE + 0x004E) +#define WSA883X_DEM_BYPASS_DATA3 (WSA883X_DIG_CTRL_BASE + 0x004F) +#define WSA883X_WAVG_CTL (WSA883X_DIG_CTRL_BASE + 0x0050) +#define WSA883X_WAVG_LRA_PER_0 (WSA883X_DIG_CTRL_BASE + 0x0051) +#define WSA883X_WAVG_LRA_PER_1 (WSA883X_DIG_CTRL_BASE + 0x0052) +#define WSA883X_WAVG_DELTA_THETA_0 (WSA883X_DIG_CTRL_BASE + 0x0053) +#define WSA883X_WAVG_DELTA_THETA_1 (WSA883X_DIG_CTRL_BASE + 0x0054) +#define WSA883X_WAVG_DIRECT_AMP_0 (WSA883X_DIG_CTRL_BASE + 0x0055) +#define WSA883X_WAVG_DIRECT_AMP_1 (WSA883X_DIG_CTRL_BASE + 0x0056) +#define WSA883X_WAVG_PTRN_AMP0_0 (WSA883X_DIG_CTRL_BASE + 0x0057) +#define WSA883X_WAVG_PTRN_AMP0_1 (WSA883X_DIG_CTRL_BASE + 0x0058) +#define WSA883X_WAVG_PTRN_AMP1_0 (WSA883X_DIG_CTRL_BASE + 0x0059) +#define WSA883X_WAVG_PTRN_AMP1_1 (WSA883X_DIG_CTRL_BASE + 0x005A) +#define WSA883X_WAVG_PTRN_AMP2_0 (WSA883X_DIG_CTRL_BASE + 0x005B) +#define WSA883X_WAVG_PTRN_AMP2_1 (WSA883X_DIG_CTRL_BASE + 0x005C) +#define WSA883X_WAVG_PTRN_AMP3_0 (WSA883X_DIG_CTRL_BASE + 0x005D) +#define WSA883X_WAVG_PTRN_AMP3_1 (WSA883X_DIG_CTRL_BASE + 0x005E) +#define WSA883X_WAVG_PTRN_AMP4_0 (WSA883X_DIG_CTRL_BASE + 0x005F) +#define WSA883X_WAVG_PTRN_AMP4_1 (WSA883X_DIG_CTRL_BASE + 0x0060) +#define WSA883X_WAVG_PTRN_AMP5_0 (WSA883X_DIG_CTRL_BASE + 0x0061) +#define WSA883X_WAVG_PTRN_AMP5_1 (WSA883X_DIG_CTRL_BASE + 0x0062) +#define WSA883X_WAVG_PTRN_AMP6_0 (WSA883X_DIG_CTRL_BASE + 0x0063) +#define WSA883X_WAVG_PTRN_AMP6_1 (WSA883X_DIG_CTRL_BASE + 0x0064) +#define WSA883X_WAVG_PTRN_AMP7_0 (WSA883X_DIG_CTRL_BASE + 0x0065) +#define WSA883X_WAVG_PTRN_AMP7_1 (WSA883X_DIG_CTRL_BASE + 0x0066) +#define WSA883X_WAVG_PER_0_1 (WSA883X_DIG_CTRL_BASE + 0x0067) +#define WSA883X_WAVG_PER_2_3 (WSA883X_DIG_CTRL_BASE + 0x0068) +#define WSA883X_WAVG_PER_4_5 (WSA883X_DIG_CTRL_BASE + 0x0069) +#define WSA883X_WAVG_PER_6_7 (WSA883X_DIG_CTRL_BASE + 0x006A) +#define WSA883X_WAVG_STA (WSA883X_DIG_CTRL_BASE + 0x006B) +#define WSA883X_DRE_CTL_0 (WSA883X_DIG_CTRL_BASE + 0x006C) +#define WSA883X_DRE_OFFSET_MASK GENMASK(2, 0) +#define WSA883X_DRE_PROG_DELAY_MASK GENMASK(7, 4) +#define WSA883X_DRE_CTL_1 (WSA883X_DIG_CTRL_BASE + 0x006D) +#define WSA883X_DRE_GAIN_EN_MASK BIT(0) +#define WSA883X_DRE_GAIN_FROM_CSR 1 +#define WSA883X_DRE_IDLE_DET_CTL (WSA883X_DIG_CTRL_BASE + 0x006E) +#define WSA883X_CLSH_CTL_0 (WSA883X_DIG_CTRL_BASE + 0x0070) +#define WSA883X_CLSH_CTL_1 (WSA883X_DIG_CTRL_BASE + 0x0071) +#define WSA883X_CLSH_V_HD_PA (WSA883X_DIG_CTRL_BASE + 0x0072) +#define WSA883X_CLSH_V_PA_MIN (WSA883X_DIG_CTRL_BASE + 0x0073) +#define WSA883X_CLSH_OVRD_VAL (WSA883X_DIG_CTRL_BASE + 0x0074) +#define WSA883X_CLSH_HARD_MAX (WSA883X_DIG_CTRL_BASE + 0x0075) +#define WSA883X_CLSH_SOFT_MAX (WSA883X_DIG_CTRL_BASE + 0x0076) +#define WSA883X_CLSH_SIG_DP (WSA883X_DIG_CTRL_BASE + 0x0077) +#define WSA883X_TAGC_CTL (WSA883X_DIG_CTRL_BASE + 0x0078) +#define WSA883X_TAGC_TIME (WSA883X_DIG_CTRL_BASE + 0x0079) +#define WSA883X_TAGC_E2E_GAIN (WSA883X_DIG_CTRL_BASE + 0x007A) +#define WSA883X_TAGC_FORCE_VAL (WSA883X_DIG_CTRL_BASE + 0x007B) +#define WSA883X_VAGC_CTL (WSA883X_DIG_CTRL_BASE + 0x007C) +#define WSA883X_VAGC_TIME (WSA883X_DIG_CTRL_BASE + 0x007D) +#define WSA883X_VAGC_ATTN_LVL_1_2 (WSA883X_DIG_CTRL_BASE + 0x007E) +#define WSA883X_VAGC_ATTN_LVL_3 (WSA883X_DIG_CTRL_BASE + 0x007F) +#define WSA883X_INTR_MODE (WSA883X_DIG_CTRL_BASE + 0x0080) +#define WSA883X_INTR_MASK0 (WSA883X_DIG_CTRL_BASE + 0x0081) +#define WSA883X_INTR_MASK1 (WSA883X_DIG_CTRL_BASE + 0x0082) +#define WSA883X_INTR_STATUS0 (WSA883X_DIG_CTRL_BASE + 0x0083) +#define WSA883X_INTR_STATUS1 (WSA883X_DIG_CTRL_BASE + 0x0084) +#define WSA883X_INTR_CLEAR0 (WSA883X_DIG_CTRL_BASE + 0x0085) +#define WSA883X_INTR_CLEAR1 (WSA883X_DIG_CTRL_BASE + 0x0086) +#define WSA883X_INTR_LEVEL0 (WSA883X_DIG_CTRL_BASE + 0x0087) +#define WSA883X_INTR_LEVEL1 (WSA883X_DIG_CTRL_BASE + 0x0088) +#define WSA883X_INTR_SET0 (WSA883X_DIG_CTRL_BASE + 0x0089) +#define WSA883X_INTR_SET1 (WSA883X_DIG_CTRL_BASE + 0x008A) +#define WSA883X_INTR_TEST0 (WSA883X_DIG_CTRL_BASE + 0x008B) +#define WSA883X_INTR_TEST1 (WSA883X_DIG_CTRL_BASE + 0x008C) +#define WSA883X_OTP_CTRL0 (WSA883X_DIG_CTRL_BASE + 0x0090) +#define WSA883X_OTP_CTRL1 (WSA883X_DIG_CTRL_BASE + 0x0091) +#define WSA883X_HDRIVE_CTL_GROUP1 (WSA883X_DIG_CTRL_BASE + 0x0092) +#define WSA883X_PIN_CTL (WSA883X_DIG_CTRL_BASE + 0x0093) +#define WSA883X_PIN_CTL_OE (WSA883X_DIG_CTRL_BASE + 0x0094) +#define WSA883X_PIN_WDATA_IOPAD (WSA883X_DIG_CTRL_BASE + 0x0095) +#define WSA883X_PIN_STATUS (WSA883X_DIG_CTRL_BASE + 0x0096) +#define WSA883X_I2C_SLAVE_CTL (WSA883X_DIG_CTRL_BASE + 0x0097) +#define WSA883X_PDM_TEST_MODE (WSA883X_DIG_CTRL_BASE + 0x00A0) +#define WSA883X_ATE_TEST_MODE (WSA883X_DIG_CTRL_BASE + 0x00A1) +#define WSA883X_DIG_DEBUG_MODE (WSA883X_DIG_CTRL_BASE + 0x00A3) +#define WSA883X_DIG_DEBUG_SEL (WSA883X_DIG_CTRL_BASE + 0x00A4) +#define WSA883X_DIG_DEBUG_EN (WSA883X_DIG_CTRL_BASE + 0x00A5) +#define WSA883X_SWR_HM_TEST0 (WSA883X_DIG_CTRL_BASE + 0x00A6) +#define WSA883X_SWR_HM_TEST1 (WSA883X_DIG_CTRL_BASE + 0x00A7) +#define WSA883X_SWR_PAD_CTL (WSA883X_DIG_CTRL_BASE + 0x00A8) +#define WSA883X_TADC_DETECT_DBG_CTL (WSA883X_DIG_CTRL_BASE + 0x00A9) +#define WSA883X_TADC_DEBUG_MSB (WSA883X_DIG_CTRL_BASE + 0x00AA) +#define WSA883X_TADC_DEBUG_LSB (WSA883X_DIG_CTRL_BASE + 0x00AB) +#define WSA883X_SAMPLE_EDGE_SEL (WSA883X_DIG_CTRL_BASE + 0x00AC) +#define WSA883X_SWR_EDGE_SEL (WSA883X_DIG_CTRL_BASE + 0x00AD) +#define WSA883X_TEST_MODE_CTL (WSA883X_DIG_CTRL_BASE + 0x00AE) +#define WSA883X_IOPAD_CTL (WSA883X_DIG_CTRL_BASE + 0x00AF) +#define WSA883X_ANA_CSR_DBG_ADD (WSA883X_DIG_CTRL_BASE + 0x00B0) +#define WSA883X_ANA_CSR_DBG_CTL (WSA883X_DIG_CTRL_BASE + 0x00B1) +#define WSA883X_SPARE_R (WSA883X_DIG_CTRL_BASE + 0x00BC) +#define WSA883X_SPARE_0 (WSA883X_DIG_CTRL_BASE + 0x00BD) +#define WSA883X_SPARE_1 (WSA883X_DIG_CTRL_BASE + 0x00BE) +#define WSA883X_SPARE_2 (WSA883X_DIG_CTRL_BASE + 0x00BF) +#define WSA883X_SCODE (WSA883X_DIG_CTRL_BASE + 0x00C0) + +#define WSA883X_DIG_TRIM_BASE (WSA883X_BASE + 0x00000500) +#define WSA883X_OTP_REG_0 (WSA883X_DIG_TRIM_BASE + 0x0080) +#define WSA883X_ID_MASK GENMASK(3, 0) +#define WSA883X_OTP_REG_1 (WSA883X_DIG_TRIM_BASE + 0x0081) +#define WSA883X_OTP_REG_2 (WSA883X_DIG_TRIM_BASE + 0x0082) +#define WSA883X_OTP_REG_3 (WSA883X_DIG_TRIM_BASE + 0x0083) +#define WSA883X_OTP_REG_4 (WSA883X_DIG_TRIM_BASE + 0x0084) +#define WSA883X_OTP_REG_5 (WSA883X_DIG_TRIM_BASE + 0x0085) +#define WSA883X_OTP_REG_6 (WSA883X_DIG_TRIM_BASE + 0x0086) +#define WSA883X_OTP_REG_7 (WSA883X_DIG_TRIM_BASE + 0x0087) +#define WSA883X_OTP_REG_8 (WSA883X_DIG_TRIM_BASE + 0x0088) +#define WSA883X_OTP_REG_9 (WSA883X_DIG_TRIM_BASE + 0x0089) +#define WSA883X_OTP_REG_10 (WSA883X_DIG_TRIM_BASE + 0x008A) +#define WSA883X_OTP_REG_11 (WSA883X_DIG_TRIM_BASE + 0x008B) +#define WSA883X_OTP_REG_12 (WSA883X_DIG_TRIM_BASE + 0x008C) +#define WSA883X_OTP_REG_13 (WSA883X_DIG_TRIM_BASE + 0x008D) +#define WSA883X_OTP_REG_14 (WSA883X_DIG_TRIM_BASE + 0x008E) +#define WSA883X_OTP_REG_15 (WSA883X_DIG_TRIM_BASE + 0x008F) +#define WSA883X_OTP_REG_16 (WSA883X_DIG_TRIM_BASE + 0x0090) +#define WSA883X_OTP_REG_17 (WSA883X_DIG_TRIM_BASE + 0x0091) +#define WSA883X_OTP_REG_18 (WSA883X_DIG_TRIM_BASE + 0x0092) +#define WSA883X_OTP_REG_19 (WSA883X_DIG_TRIM_BASE + 0x0093) +#define WSA883X_OTP_REG_20 (WSA883X_DIG_TRIM_BASE + 0x0094) +#define WSA883X_OTP_REG_21 (WSA883X_DIG_TRIM_BASE + 0x0095) +#define WSA883X_OTP_REG_22 (WSA883X_DIG_TRIM_BASE + 0x0096) +#define WSA883X_OTP_REG_23 (WSA883X_DIG_TRIM_BASE + 0x0097) +#define WSA883X_OTP_REG_24 (WSA883X_DIG_TRIM_BASE + 0x0098) +#define WSA883X_OTP_REG_25 (WSA883X_DIG_TRIM_BASE + 0x0099) +#define WSA883X_OTP_REG_26 (WSA883X_DIG_TRIM_BASE + 0x009A) +#define WSA883X_OTP_REG_27 (WSA883X_DIG_TRIM_BASE + 0x009B) +#define WSA883X_OTP_REG_28 (WSA883X_DIG_TRIM_BASE + 0x009C) +#define WSA883X_OTP_REG_29 (WSA883X_DIG_TRIM_BASE + 0x009D) +#define WSA883X_OTP_REG_30 (WSA883X_DIG_TRIM_BASE + 0x009E) +#define WSA883X_OTP_REG_31 (WSA883X_DIG_TRIM_BASE + 0x009F) +#define WSA883X_OTP_REG_32 (WSA883X_DIG_TRIM_BASE + 0x00A0) +#define WSA883X_OTP_REG_33 (WSA883X_DIG_TRIM_BASE + 0x00A1) +#define WSA883X_OTP_REG_34 (WSA883X_DIG_TRIM_BASE + 0x00A2) +#define WSA883X_OTP_REG_35 (WSA883X_DIG_TRIM_BASE + 0x00A3) +#define WSA883X_OTP_REG_63 (WSA883X_DIG_TRIM_BASE + 0x00BF) + +#define WSA883X_DIG_EMEM_BASE (WSA883X_BASE + 0x000005C0) +#define WSA883X_EMEM_0 (WSA883X_DIG_EMEM_BASE + 0x0000) +#define WSA883X_EMEM_1 (WSA883X_DIG_EMEM_BASE + 0x0001) +#define WSA883X_EMEM_2 (WSA883X_DIG_EMEM_BASE + 0x0002) +#define WSA883X_EMEM_3 (WSA883X_DIG_EMEM_BASE + 0x0003) +#define WSA883X_EMEM_4 (WSA883X_DIG_EMEM_BASE + 0x0004) +#define WSA883X_EMEM_5 (WSA883X_DIG_EMEM_BASE + 0x0005) +#define WSA883X_EMEM_6 (WSA883X_DIG_EMEM_BASE + 0x0006) +#define WSA883X_EMEM_7 (WSA883X_DIG_EMEM_BASE + 0x0007) +#define WSA883X_EMEM_8 (WSA883X_DIG_EMEM_BASE + 0x0008) +#define WSA883X_EMEM_9 (WSA883X_DIG_EMEM_BASE + 0x0009) +#define WSA883X_EMEM_10 (WSA883X_DIG_EMEM_BASE + 0x000A) +#define WSA883X_EMEM_11 (WSA883X_DIG_EMEM_BASE + 0x000B) +#define WSA883X_EMEM_12 (WSA883X_DIG_EMEM_BASE + 0x000C) +#define WSA883X_EMEM_13 (WSA883X_DIG_EMEM_BASE + 0x000D) +#define WSA883X_EMEM_14 (WSA883X_DIG_EMEM_BASE + 0x000E) +#define WSA883X_EMEM_15 (WSA883X_DIG_EMEM_BASE + 0x000F) +#define WSA883X_EMEM_16 (WSA883X_DIG_EMEM_BASE + 0x0010) +#define WSA883X_EMEM_17 (WSA883X_DIG_EMEM_BASE + 0x0011) +#define WSA883X_EMEM_18 (WSA883X_DIG_EMEM_BASE + 0x0012) +#define WSA883X_EMEM_19 (WSA883X_DIG_EMEM_BASE + 0x0013) +#define WSA883X_EMEM_20 (WSA883X_DIG_EMEM_BASE + 0x0014) +#define WSA883X_EMEM_21 (WSA883X_DIG_EMEM_BASE + 0x0015) +#define WSA883X_EMEM_22 (WSA883X_DIG_EMEM_BASE + 0x0016) +#define WSA883X_EMEM_23 (WSA883X_DIG_EMEM_BASE + 0x0017) +#define WSA883X_EMEM_24 (WSA883X_DIG_EMEM_BASE + 0x0018) +#define WSA883X_EMEM_25 (WSA883X_DIG_EMEM_BASE + 0x0019) +#define WSA883X_EMEM_26 (WSA883X_DIG_EMEM_BASE + 0x001A) +#define WSA883X_EMEM_27 (WSA883X_DIG_EMEM_BASE + 0x001B) +#define WSA883X_EMEM_28 (WSA883X_DIG_EMEM_BASE + 0x001C) +#define WSA883X_EMEM_29 (WSA883X_DIG_EMEM_BASE + 0x001D) +#define WSA883X_EMEM_30 (WSA883X_DIG_EMEM_BASE + 0x001E) +#define WSA883X_EMEM_31 (WSA883X_DIG_EMEM_BASE + 0x001F) +#define WSA883X_EMEM_32 (WSA883X_DIG_EMEM_BASE + 0x0020) +#define WSA883X_EMEM_33 (WSA883X_DIG_EMEM_BASE + 0x0021) +#define WSA883X_EMEM_34 (WSA883X_DIG_EMEM_BASE + 0x0022) +#define WSA883X_EMEM_35 (WSA883X_DIG_EMEM_BASE + 0x0023) +#define WSA883X_EMEM_36 (WSA883X_DIG_EMEM_BASE + 0x0024) +#define WSA883X_EMEM_37 (WSA883X_DIG_EMEM_BASE + 0x0025) +#define WSA883X_EMEM_38 (WSA883X_DIG_EMEM_BASE + 0x0026) +#define WSA883X_EMEM_39 (WSA883X_DIG_EMEM_BASE + 0x0027) +#define WSA883X_EMEM_40 (WSA883X_DIG_EMEM_BASE + 0x0028) +#define WSA883X_EMEM_41 (WSA883X_DIG_EMEM_BASE + 0x0029) +#define WSA883X_EMEM_42 (WSA883X_DIG_EMEM_BASE + 0x002A) +#define WSA883X_EMEM_43 (WSA883X_DIG_EMEM_BASE + 0x002B) +#define WSA883X_EMEM_44 (WSA883X_DIG_EMEM_BASE + 0x002C) +#define WSA883X_EMEM_45 (WSA883X_DIG_EMEM_BASE + 0x002D) +#define WSA883X_EMEM_46 (WSA883X_DIG_EMEM_BASE + 0x002E) +#define WSA883X_EMEM_47 (WSA883X_DIG_EMEM_BASE + 0x002F) +#define WSA883X_EMEM_48 (WSA883X_DIG_EMEM_BASE + 0x0030) +#define WSA883X_EMEM_49 (WSA883X_DIG_EMEM_BASE + 0x0031) +#define WSA883X_EMEM_50 (WSA883X_DIG_EMEM_BASE + 0x0032) +#define WSA883X_EMEM_51 (WSA883X_DIG_EMEM_BASE + 0x0033) +#define WSA883X_EMEM_52 (WSA883X_DIG_EMEM_BASE + 0x0034) +#define WSA883X_EMEM_53 (WSA883X_DIG_EMEM_BASE + 0x0035) +#define WSA883X_EMEM_54 (WSA883X_DIG_EMEM_BASE + 0x0036) +#define WSA883X_EMEM_55 (WSA883X_DIG_EMEM_BASE + 0x0037) +#define WSA883X_EMEM_56 (WSA883X_DIG_EMEM_BASE + 0x0038) +#define WSA883X_EMEM_57 (WSA883X_DIG_EMEM_BASE + 0x0039) +#define WSA883X_EMEM_58 (WSA883X_DIG_EMEM_BASE + 0x003A) +#define WSA883X_EMEM_59 (WSA883X_DIG_EMEM_BASE + 0x003B) +#define WSA883X_EMEM_60 (WSA883X_DIG_EMEM_BASE + 0x003C) +#define WSA883X_EMEM_61 (WSA883X_DIG_EMEM_BASE + 0x003D) +#define WSA883X_EMEM_62 (WSA883X_DIG_EMEM_BASE + 0x003E) +#define WSA883X_EMEM_63 (WSA883X_DIG_EMEM_BASE + 0x003F) + +#define WSA883X_NUM_REGISTERS (WSA883X_EMEM_63 + 1) +#define WSA883X_MAX_REGISTER (WSA883X_NUM_REGISTERS - 1) +#define WSA883X_PROBE_TIMEOUT 1000 + +#define WSA883X_VERSION_1_0 0 +#define WSA883X_VERSION_1_1 1 + +#define WSA883X_MAX_SWR_PORTS 4 +#define WSA883X_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_48000 |\ + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000 |\ + SNDRV_PCM_RATE_384000) +/* Fractional Rates */ +#define WSA883X_FRAC_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_88200 |\ + SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_352800) + +#define WSA883X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + +struct wsa883x_priv { + struct regmap *regmap; + struct device *dev; + struct regulator *vdd; + struct sdw_slave *slave; + struct sdw_stream_config sconfig; + struct sdw_stream_runtime *sruntime; + struct sdw_port_config port_config[WSA883X_MAX_SWR_PORTS]; + struct gpio_desc *sd_n; + bool port_prepared[WSA883X_MAX_SWR_PORTS]; + bool port_enable[WSA883X_MAX_SWR_PORTS]; + int version; + int variant; + int active_ports; + int dev_mode; + int comp_offset; +}; + +enum { + WSA8830 = 0, + WSA8835, + WSA8832, + WSA8835_V2 = 5, +}; + +enum { + COMP_OFFSET0, + COMP_OFFSET1, + COMP_OFFSET2, + COMP_OFFSET3, + COMP_OFFSET4, +}; + +enum wsa_port_ids { + WSA883X_PORT_DAC, + WSA883X_PORT_COMP, + WSA883X_PORT_BOOST, + WSA883X_PORT_VISENSE, +}; + +/* 4 ports */ +static struct sdw_dpn_prop wsa_sink_dpn_prop[WSA883X_MAX_SWR_PORTS] = { + { + /* DAC */ + .num = 1, + .type = SDW_DPN_SIMPLE, + .min_ch = 1, + .max_ch = 1, + .simple_ch_prep_sm = true, + .read_only_wordlength = true, + }, { + /* COMP */ + .num = 2, + .type = SDW_DPN_SIMPLE, + .min_ch = 1, + .max_ch = 1, + .simple_ch_prep_sm = true, + .read_only_wordlength = true, + }, { + /* BOOST */ + .num = 3, + .type = SDW_DPN_SIMPLE, + .min_ch = 1, + .max_ch = 1, + .simple_ch_prep_sm = true, + .read_only_wordlength = true, + }, { + /* VISENSE */ + .num = 4, + .type = SDW_DPN_SIMPLE, + .min_ch = 1, + .max_ch = 1, + .simple_ch_prep_sm = true, + .read_only_wordlength = true, + } +}; + +static struct sdw_port_config wsa883x_pconfig[WSA883X_MAX_SWR_PORTS] = { + { + .num = 1, + .ch_mask = 0x1, + }, { + .num = 2, + .ch_mask = 0xf, + }, { + .num = 3, + .ch_mask = 0x3, + }, { /* IV feedback */ + .num = 4, + .ch_mask = 0x3, + }, +}; + +static struct reg_default wsa883x_defaults[] = { + { WSA883X_REF_CTRL, 0xD5 }, + { WSA883X_TEST_CTL_0, 0x06 }, + { WSA883X_BIAS_0, 0xD2 }, + { WSA883X_OP_CTL, 0xE0 }, + { WSA883X_IREF_CTL, 0x57 }, + { WSA883X_ISENS_CTL, 0x47 }, + { WSA883X_CLK_CTL, 0x87 }, + { WSA883X_TEST_CTL_1, 0x00 }, + { WSA883X_BIAS_1, 0x51 }, + { WSA883X_ADC_CTL, 0x01 }, + { WSA883X_DOUT_MSB, 0x00 }, + { WSA883X_DOUT_LSB, 0x00 }, + { WSA883X_VBAT_SNS, 0x40 }, + { WSA883X_ITRIM_CODE, 0x9F }, + { WSA883X_EN, 0x20 }, + { WSA883X_OVERRIDE1, 0x00 }, + { WSA883X_OVERRIDE2, 0x08 }, + { WSA883X_VSENSE1, 0xD3 }, + { WSA883X_ISENSE1, 0xD4 }, + { WSA883X_ISENSE2, 0x20 }, + { WSA883X_ISENSE_CAL, 0x00 }, + { WSA883X_MISC, 0x08 }, + { WSA883X_ADC_0, 0x00 }, + { WSA883X_ADC_1, 0x00 }, + { WSA883X_ADC_2, 0x40 }, + { WSA883X_ADC_3, 0x80 }, + { WSA883X_ADC_4, 0x25 }, + { WSA883X_ADC_5, 0x25 }, + { WSA883X_ADC_6, 0x08 }, + { WSA883X_ADC_7, 0x81 }, + { WSA883X_STATUS, 0x00 }, + { WSA883X_DAC_CTRL_REG, 0x53 }, + { WSA883X_DAC_EN_DEBUG_REG, 0x00 }, + { WSA883X_DAC_OPAMP_BIAS1_REG, 0x48 }, + { WSA883X_DAC_OPAMP_BIAS2_REG, 0x48 }, + { WSA883X_DAC_VCM_CTRL_REG, 0x88 }, + { WSA883X_DAC_VOLTAGE_CTRL_REG, 0xA5 }, + { WSA883X_ATEST1_REG, 0x00 }, + { WSA883X_ATEST2_REG, 0x00 }, + { WSA883X_SPKR_TOP_BIAS_REG1, 0x6A }, + { WSA883X_SPKR_TOP_BIAS_REG2, 0x65 }, + { WSA883X_SPKR_TOP_BIAS_REG3, 0x55 }, + { WSA883X_SPKR_TOP_BIAS_REG4, 0xA9 }, + { WSA883X_SPKR_CLIP_DET_REG, 0x9C }, + { WSA883X_SPKR_DRV_LF_BLK_EN, 0x0F }, + { WSA883X_SPKR_DRV_LF_EN, 0x0A }, + { WSA883X_SPKR_DRV_LF_MASK_DCC_CTL, 0x00 }, + { WSA883X_SPKR_DRV_LF_MISC_CTL, 0x3A }, + { WSA883X_SPKR_DRV_LF_REG_GAIN, 0x00 }, + { WSA883X_SPKR_DRV_OS_CAL_CTL, 0x00 }, + { WSA883X_SPKR_DRV_OS_CAL_CTL1, 0x90 }, + { WSA883X_SPKR_PWM_CLK_CTL, 0x00 }, + { WSA883X_SPKR_PDRV_HS_CTL, 0x52 }, + { WSA883X_SPKR_PDRV_LS_CTL, 0x48 }, + { WSA883X_SPKR_PWRSTG_DBG, 0x08 }, + { WSA883X_SPKR_OCP_CTL, 0xE2 }, + { WSA883X_SPKR_BBM_CTL, 0x92 }, + { WSA883X_PA_STATUS0, 0x00 }, + { WSA883X_PA_STATUS1, 0x00 }, + { WSA883X_PA_STATUS2, 0x80 }, + { WSA883X_EN_CTRL, 0x44 }, + { WSA883X_CURRENT_LIMIT, 0xCC }, + { WSA883X_IBIAS1, 0x00 }, + { WSA883X_IBIAS2, 0x00 }, + { WSA883X_IBIAS3, 0x00 }, + { WSA883X_LDO_PROG, 0x02 }, + { WSA883X_STABILITY_CTRL1, 0x8E }, + { WSA883X_STABILITY_CTRL2, 0x10 }, + { WSA883X_PWRSTAGE_CTRL1, 0x06 }, + { WSA883X_PWRSTAGE_CTRL2, 0x00 }, + { WSA883X_BYPASS_1, 0x19 }, + { WSA883X_BYPASS_2, 0x13 }, + { WSA883X_ZX_CTRL_1, 0xF0 }, + { WSA883X_ZX_CTRL_2, 0x04 }, + { WSA883X_MISC1, 0x06 }, + { WSA883X_MISC2, 0xA0 }, + { WSA883X_GMAMP_SUP1, 0x82 }, + { WSA883X_PWRSTAGE_CTRL3, 0x39 }, + { WSA883X_PWRSTAGE_CTRL4, 0x5F }, + { WSA883X_TEST1, 0x00 }, + { WSA883X_SPARE1, 0x00 }, + { WSA883X_SPARE2, 0x00 }, + { WSA883X_PON_CTL_0, 0x10 }, + { WSA883X_PON_CLT_1, 0xE0 }, + { WSA883X_PON_CTL_2, 0x90 }, + { WSA883X_PON_CTL_3, 0x70 }, + { WSA883X_CKWD_CTL_0, 0x34 }, + { WSA883X_CKWD_CTL_1, 0x0F }, + { WSA883X_CKWD_CTL_2, 0x00 }, + { WSA883X_CKSK_CTL_0, 0x00 }, + { WSA883X_PADSW_CTL_0, 0x00 }, + { WSA883X_TEST_0, 0x00 }, + { WSA883X_TEST_1, 0x00 }, + { WSA883X_STATUS_0, 0x00 }, + { WSA883X_STATUS_1, 0x00 }, + { WSA883X_CHIP_ID0, 0x00 }, + { WSA883X_CHIP_ID1, 0x00 }, + { WSA883X_CHIP_ID2, 0x02 }, + { WSA883X_CHIP_ID3, 0x02 }, + { WSA883X_BUS_ID, 0x00 }, + { WSA883X_CDC_RST_CTL, 0x01 }, + { WSA883X_TOP_CLK_CFG, 0x00 }, + { WSA883X_CDC_PATH_MODE, 0x00 }, + { WSA883X_CDC_CLK_CTL, 0xFF }, + { WSA883X_SWR_RESET_EN, 0x00 }, + { WSA883X_RESET_CTL, 0x00 }, + { WSA883X_PA_FSM_CTL, 0x00 }, + { WSA883X_PA_FSM_TIMER0, 0x80 }, + { WSA883X_PA_FSM_TIMER1, 0x80 }, + { WSA883X_PA_FSM_STA, 0x00 }, + { WSA883X_PA_FSM_ERR_COND, 0x00 }, + { WSA883X_PA_FSM_MSK, 0x00 }, + { WSA883X_PA_FSM_BYP, 0x01 }, + { WSA883X_PA_FSM_DBG, 0x00 }, + { WSA883X_TADC_VALUE_CTL, 0x03 }, + { WSA883X_TEMP_DETECT_CTL, 0x01 }, + { WSA883X_TEMP_MSB, 0x00 }, + { WSA883X_TEMP_LSB, 0x00 }, + { WSA883X_TEMP_CONFIG0, 0x00 }, + { WSA883X_TEMP_CONFIG1, 0x00 }, + { WSA883X_VBAT_ADC_FLT_CTL, 0x00 }, + { WSA883X_VBAT_DIN_MSB, 0x00 }, + { WSA883X_VBAT_DIN_LSB, 0x00 }, + { WSA883X_VBAT_DOUT, 0x00 }, + { WSA883X_SDM_PDM9_LSB, 0x00 }, + { WSA883X_SDM_PDM9_MSB, 0x00 }, + { WSA883X_CDC_RX_CTL, 0xFE }, + { WSA883X_CDC_SPK_DSM_A1_0, 0x00 }, + { WSA883X_CDC_SPK_DSM_A1_1, 0x01 }, + { WSA883X_CDC_SPK_DSM_A2_0, 0x96 }, + { WSA883X_CDC_SPK_DSM_A2_1, 0x09 }, + { WSA883X_CDC_SPK_DSM_A3_0, 0xAB }, + { WSA883X_CDC_SPK_DSM_A3_1, 0x05 }, + { WSA883X_CDC_SPK_DSM_A4_0, 0x1C }, + { WSA883X_CDC_SPK_DSM_A4_1, 0x02 }, + { WSA883X_CDC_SPK_DSM_A5_0, 0x17 }, + { WSA883X_CDC_SPK_DSM_A5_1, 0x02 }, + { WSA883X_CDC_SPK_DSM_A6_0, 0xAA }, + { WSA883X_CDC_SPK_DSM_A7_0, 0xE3 }, + { WSA883X_CDC_SPK_DSM_C_0, 0x69 }, + { WSA883X_CDC_SPK_DSM_C_1, 0x54 }, + { WSA883X_CDC_SPK_DSM_C_2, 0x02 }, + { WSA883X_CDC_SPK_DSM_C_3, 0x15 }, + { WSA883X_CDC_SPK_DSM_R1, 0xA4 }, + { WSA883X_CDC_SPK_DSM_R2, 0xB5 }, + { WSA883X_CDC_SPK_DSM_R3, 0x86 }, + { WSA883X_CDC_SPK_DSM_R4, 0x85 }, + { WSA883X_CDC_SPK_DSM_R5, 0xAA }, + { WSA883X_CDC_SPK_DSM_R6, 0xE2 }, + { WSA883X_CDC_SPK_DSM_R7, 0x62 }, + { WSA883X_CDC_SPK_GAIN_PDM_0, 0x00 }, + { WSA883X_CDC_SPK_GAIN_PDM_1, 0xFC }, + { WSA883X_CDC_SPK_GAIN_PDM_2, 0x05 }, + { WSA883X_PDM_WD_CTL, 0x00 }, + { WSA883X_DEM_BYPASS_DATA0, 0x00 }, + { WSA883X_DEM_BYPASS_DATA1, 0x00 }, + { WSA883X_DEM_BYPASS_DATA2, 0x00 }, + { WSA883X_DEM_BYPASS_DATA3, 0x00 }, + { WSA883X_WAVG_CTL, 0x06 }, + { WSA883X_WAVG_LRA_PER_0, 0xD1 }, + { WSA883X_WAVG_LRA_PER_1, 0x00 }, + { WSA883X_WAVG_DELTA_THETA_0, 0xE6 }, + { WSA883X_WAVG_DELTA_THETA_1, 0x04 }, + { WSA883X_WAVG_DIRECT_AMP_0, 0x50 }, + { WSA883X_WAVG_DIRECT_AMP_1, 0x00 }, + { WSA883X_WAVG_PTRN_AMP0_0, 0x50 }, + { WSA883X_WAVG_PTRN_AMP0_1, 0x00 }, + { WSA883X_WAVG_PTRN_AMP1_0, 0x50 }, + { WSA883X_WAVG_PTRN_AMP1_1, 0x00 }, + { WSA883X_WAVG_PTRN_AMP2_0, 0x50 }, + { WSA883X_WAVG_PTRN_AMP2_1, 0x00 }, + { WSA883X_WAVG_PTRN_AMP3_0, 0x50 }, + { WSA883X_WAVG_PTRN_AMP3_1, 0x00 }, + { WSA883X_WAVG_PTRN_AMP4_0, 0x50 }, + { WSA883X_WAVG_PTRN_AMP4_1, 0x00 }, + { WSA883X_WAVG_PTRN_AMP5_0, 0x50 }, + { WSA883X_WAVG_PTRN_AMP5_1, 0x00 }, + { WSA883X_WAVG_PTRN_AMP6_0, 0x50 }, + { WSA883X_WAVG_PTRN_AMP6_1, 0x00 }, + { WSA883X_WAVG_PTRN_AMP7_0, 0x50 }, + { WSA883X_WAVG_PTRN_AMP7_1, 0x00 }, + { WSA883X_WAVG_PER_0_1, 0x88 }, + { WSA883X_WAVG_PER_2_3, 0x88 }, + { WSA883X_WAVG_PER_4_5, 0x88 }, + { WSA883X_WAVG_PER_6_7, 0x88 }, + { WSA883X_WAVG_STA, 0x00 }, + { WSA883X_DRE_CTL_0, 0x70 }, + { WSA883X_DRE_CTL_1, 0x08 }, + { WSA883X_DRE_IDLE_DET_CTL, 0x1F }, + { WSA883X_CLSH_CTL_0, 0x37 }, + { WSA883X_CLSH_CTL_1, 0x81 }, + { WSA883X_CLSH_V_HD_PA, 0x0F }, + { WSA883X_CLSH_V_PA_MIN, 0x00 }, + { WSA883X_CLSH_OVRD_VAL, 0x00 }, + { WSA883X_CLSH_HARD_MAX, 0xFF }, + { WSA883X_CLSH_SOFT_MAX, 0xF5 }, + { WSA883X_CLSH_SIG_DP, 0x00 }, + { WSA883X_TAGC_CTL, 0x10 }, + { WSA883X_TAGC_TIME, 0x20 }, + { WSA883X_TAGC_E2E_GAIN, 0x02 }, + { WSA883X_TAGC_FORCE_VAL, 0x00 }, + { WSA883X_VAGC_CTL, 0x00 }, + { WSA883X_VAGC_TIME, 0x08 }, + { WSA883X_VAGC_ATTN_LVL_1_2, 0x21 }, + { WSA883X_VAGC_ATTN_LVL_3, 0x03 }, + { WSA883X_INTR_MODE, 0x00 }, + { WSA883X_INTR_MASK0, 0x90 }, + { WSA883X_INTR_MASK1, 0x00 }, + { WSA883X_INTR_STATUS0, 0x00 }, + { WSA883X_INTR_STATUS1, 0x00 }, + { WSA883X_INTR_CLEAR0, 0x00 }, + { WSA883X_INTR_CLEAR1, 0x00 }, + { WSA883X_INTR_LEVEL0, 0x00 }, + { WSA883X_INTR_LEVEL1, 0x00 }, + { WSA883X_INTR_SET0, 0x00 }, + { WSA883X_INTR_SET1, 0x00 }, + { WSA883X_INTR_TEST0, 0x00 }, + { WSA883X_INTR_TEST1, 0x00 }, + { WSA883X_OTP_CTRL0, 0x00 }, + { WSA883X_OTP_CTRL1, 0x00 }, + { WSA883X_HDRIVE_CTL_GROUP1, 0x00 }, + { WSA883X_PIN_CTL, 0x04 }, + { WSA883X_PIN_CTL_OE, 0x00 }, + { WSA883X_PIN_WDATA_IOPAD, 0x00 }, + { WSA883X_PIN_STATUS, 0x00 }, + { WSA883X_I2C_SLAVE_CTL, 0x00 }, + { WSA883X_PDM_TEST_MODE, 0x00 }, + { WSA883X_ATE_TEST_MODE, 0x00 }, + { WSA883X_DIG_DEBUG_MODE, 0x00 }, + { WSA883X_DIG_DEBUG_SEL, 0x00 }, + { WSA883X_DIG_DEBUG_EN, 0x00 }, + { WSA883X_SWR_HM_TEST0, 0x08 }, + { WSA883X_SWR_HM_TEST1, 0x00 }, + { WSA883X_SWR_PAD_CTL, 0x37 }, + { WSA883X_TADC_DETECT_DBG_CTL, 0x00 }, + { WSA883X_TADC_DEBUG_MSB, 0x00 }, + { WSA883X_TADC_DEBUG_LSB, 0x00 }, + { WSA883X_SAMPLE_EDGE_SEL, 0x7F }, + { WSA883X_SWR_EDGE_SEL, 0x00 }, + { WSA883X_TEST_MODE_CTL, 0x04 }, + { WSA883X_IOPAD_CTL, 0x00 }, + { WSA883X_ANA_CSR_DBG_ADD, 0x00 }, + { WSA883X_ANA_CSR_DBG_CTL, 0x12 }, + { WSA883X_SPARE_R, 0x00 }, + { WSA883X_SPARE_0, 0x00 }, + { WSA883X_SPARE_1, 0x00 }, + { WSA883X_SPARE_2, 0x00 }, + { WSA883X_SCODE, 0x00 }, + { WSA883X_OTP_REG_0, 0x05 }, + { WSA883X_OTP_REG_1, 0xFF }, + { WSA883X_OTP_REG_2, 0xC0 }, + { WSA883X_OTP_REG_3, 0xFF }, + { WSA883X_OTP_REG_4, 0xC0 }, + { WSA883X_OTP_REG_5, 0xFF }, + { WSA883X_OTP_REG_6, 0xFF }, + { WSA883X_OTP_REG_7, 0xFF }, + { WSA883X_OTP_REG_8, 0xFF }, + { WSA883X_OTP_REG_9, 0xFF }, + { WSA883X_OTP_REG_10, 0xFF }, + { WSA883X_OTP_REG_11, 0xFF }, + { WSA883X_OTP_REG_12, 0xFF }, + { WSA883X_OTP_REG_13, 0xFF }, + { WSA883X_OTP_REG_14, 0xFF }, + { WSA883X_OTP_REG_15, 0xFF }, + { WSA883X_OTP_REG_16, 0xFF }, + { WSA883X_OTP_REG_17, 0xFF }, + { WSA883X_OTP_REG_18, 0xFF }, + { WSA883X_OTP_REG_19, 0xFF }, + { WSA883X_OTP_REG_20, 0xFF }, + { WSA883X_OTP_REG_21, 0xFF }, + { WSA883X_OTP_REG_22, 0xFF }, + { WSA883X_OTP_REG_23, 0xFF }, + { WSA883X_OTP_REG_24, 0x37 }, + { WSA883X_OTP_REG_25, 0x3F }, + { WSA883X_OTP_REG_26, 0x03 }, + { WSA883X_OTP_REG_27, 0x00 }, + { WSA883X_OTP_REG_28, 0x00 }, + { WSA883X_OTP_REG_29, 0x00 }, + { WSA883X_OTP_REG_30, 0x00 }, + { WSA883X_OTP_REG_31, 0x03 }, + { WSA883X_OTP_REG_32, 0x00 }, + { WSA883X_OTP_REG_33, 0xFF }, + { WSA883X_OTP_REG_34, 0x00 }, + { WSA883X_OTP_REG_35, 0x00 }, + { WSA883X_OTP_REG_63, 0x40 }, + { WSA883X_EMEM_0, 0x00 }, + { WSA883X_EMEM_1, 0x00 }, + { WSA883X_EMEM_2, 0x00 }, + { WSA883X_EMEM_3, 0x00 }, + { WSA883X_EMEM_4, 0x00 }, + { WSA883X_EMEM_5, 0x00 }, + { WSA883X_EMEM_6, 0x00 }, + { WSA883X_EMEM_7, 0x00 }, + { WSA883X_EMEM_8, 0x00 }, + { WSA883X_EMEM_9, 0x00 }, + { WSA883X_EMEM_10, 0x00 }, + { WSA883X_EMEM_11, 0x00 }, + { WSA883X_EMEM_12, 0x00 }, + { WSA883X_EMEM_13, 0x00 }, + { WSA883X_EMEM_14, 0x00 }, + { WSA883X_EMEM_15, 0x00 }, + { WSA883X_EMEM_16, 0x00 }, + { WSA883X_EMEM_17, 0x00 }, + { WSA883X_EMEM_18, 0x00 }, + { WSA883X_EMEM_19, 0x00 }, + { WSA883X_EMEM_20, 0x00 }, + { WSA883X_EMEM_21, 0x00 }, + { WSA883X_EMEM_22, 0x00 }, + { WSA883X_EMEM_23, 0x00 }, + { WSA883X_EMEM_24, 0x00 }, + { WSA883X_EMEM_25, 0x00 }, + { WSA883X_EMEM_26, 0x00 }, + { WSA883X_EMEM_27, 0x00 }, + { WSA883X_EMEM_28, 0x00 }, + { WSA883X_EMEM_29, 0x00 }, + { WSA883X_EMEM_30, 0x00 }, + { WSA883X_EMEM_31, 0x00 }, + { WSA883X_EMEM_32, 0x00 }, + { WSA883X_EMEM_33, 0x00 }, + { WSA883X_EMEM_34, 0x00 }, + { WSA883X_EMEM_35, 0x00 }, + { WSA883X_EMEM_36, 0x00 }, + { WSA883X_EMEM_37, 0x00 }, + { WSA883X_EMEM_38, 0x00 }, + { WSA883X_EMEM_39, 0x00 }, + { WSA883X_EMEM_40, 0x00 }, + { WSA883X_EMEM_41, 0x00 }, + { WSA883X_EMEM_42, 0x00 }, + { WSA883X_EMEM_43, 0x00 }, + { WSA883X_EMEM_44, 0x00 }, + { WSA883X_EMEM_45, 0x00 }, + { WSA883X_EMEM_46, 0x00 }, + { WSA883X_EMEM_47, 0x00 }, + { WSA883X_EMEM_48, 0x00 }, + { WSA883X_EMEM_49, 0x00 }, + { WSA883X_EMEM_50, 0x00 }, + { WSA883X_EMEM_51, 0x00 }, + { WSA883X_EMEM_52, 0x00 }, + { WSA883X_EMEM_53, 0x00 }, + { WSA883X_EMEM_54, 0x00 }, + { WSA883X_EMEM_55, 0x00 }, + { WSA883X_EMEM_56, 0x00 }, + { WSA883X_EMEM_57, 0x00 }, + { WSA883X_EMEM_58, 0x00 }, + { WSA883X_EMEM_59, 0x00 }, + { WSA883X_EMEM_60, 0x00 }, + { WSA883X_EMEM_61, 0x00 }, + { WSA883X_EMEM_62, 0x00 }, + { WSA883X_EMEM_63, 0x00 }, +}; + +static bool wsa883x_readonly_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case WSA883X_DOUT_MSB: + case WSA883X_DOUT_LSB: + case WSA883X_STATUS: + case WSA883X_PA_STATUS0: + case WSA883X_PA_STATUS1: + case WSA883X_PA_STATUS2: + case WSA883X_STATUS_0: + case WSA883X_STATUS_1: + case WSA883X_CHIP_ID0: + case WSA883X_CHIP_ID1: + case WSA883X_CHIP_ID2: + case WSA883X_CHIP_ID3: + case WSA883X_BUS_ID: + case WSA883X_PA_FSM_STA: + case WSA883X_PA_FSM_ERR_COND: + case WSA883X_TEMP_MSB: + case WSA883X_TEMP_LSB: + case WSA883X_VBAT_DIN_MSB: + case WSA883X_VBAT_DIN_LSB: + case WSA883X_VBAT_DOUT: + case WSA883X_SDM_PDM9_LSB: + case WSA883X_SDM_PDM9_MSB: + case WSA883X_WAVG_STA: + case WSA883X_INTR_STATUS0: + case WSA883X_INTR_STATUS1: + case WSA883X_OTP_CTRL1: + case WSA883X_PIN_STATUS: + case WSA883X_ATE_TEST_MODE: + case WSA883X_SWR_HM_TEST1: + case WSA883X_SPARE_R: + case WSA883X_OTP_REG_0: + return true; + } + return false; +} + +static bool wsa883x_writeable_register(struct device *dev, unsigned int reg) +{ + return !wsa883x_readonly_register(dev, reg); +} + +static bool wsa883x_volatile_register(struct device *dev, unsigned int reg) +{ + return wsa883x_readonly_register(dev, reg); +} + +static struct regmap_config wsa883x_regmap_config = { + .reg_bits = 32, + .val_bits = 8, + .cache_type = REGCACHE_RBTREE, + .reg_defaults = wsa883x_defaults, + .max_register = WSA883X_MAX_REGISTER, + .num_reg_defaults = ARRAY_SIZE(wsa883x_defaults), + .volatile_reg = wsa883x_volatile_register, + .writeable_reg = wsa883x_writeable_register, + .reg_format_endian = REGMAP_ENDIAN_NATIVE, + .val_format_endian = REGMAP_ENDIAN_NATIVE, + .can_multi_write = true, + .use_single_read = true, +}; + +static const struct reg_sequence reg_init[] = { + {WSA883X_PA_FSM_BYP, 0x00}, + {WSA883X_ADC_6, 0x02}, + {WSA883X_CDC_SPK_DSM_A2_0, 0x0A}, + {WSA883X_CDC_SPK_DSM_A2_1, 0x08}, + {WSA883X_CDC_SPK_DSM_A3_0, 0xF3}, + {WSA883X_CDC_SPK_DSM_A3_1, 0x07}, + {WSA883X_CDC_SPK_DSM_A4_0, 0x79}, + {WSA883X_CDC_SPK_DSM_A4_1, 0x02}, + {WSA883X_CDC_SPK_DSM_A5_0, 0x0B}, + {WSA883X_CDC_SPK_DSM_A5_1, 0x02}, + {WSA883X_CDC_SPK_DSM_A6_0, 0x8A}, + {WSA883X_CDC_SPK_DSM_A7_0, 0x9B}, + {WSA883X_CDC_SPK_DSM_C_0, 0x68}, + {WSA883X_CDC_SPK_DSM_C_1, 0x54}, + {WSA883X_CDC_SPK_DSM_C_2, 0xF2}, + {WSA883X_CDC_SPK_DSM_C_3, 0x20}, + {WSA883X_CDC_SPK_DSM_R1, 0x83}, + {WSA883X_CDC_SPK_DSM_R2, 0x7F}, + {WSA883X_CDC_SPK_DSM_R3, 0x9D}, + {WSA883X_CDC_SPK_DSM_R4, 0x82}, + {WSA883X_CDC_SPK_DSM_R5, 0x8B}, + {WSA883X_CDC_SPK_DSM_R6, 0x9B}, + {WSA883X_CDC_SPK_DSM_R7, 0x3F}, + {WSA883X_VBAT_SNS, 0x20}, + {WSA883X_DRE_CTL_0, 0x92}, + {WSA883X_DRE_IDLE_DET_CTL, 0x0F}, + {WSA883X_CURRENT_LIMIT, 0xC4}, + {WSA883X_VAGC_TIME, 0x0F}, + {WSA883X_VAGC_ATTN_LVL_1_2, 0x00}, + {WSA883X_VAGC_ATTN_LVL_3, 0x01}, + {WSA883X_VAGC_CTL, 0x01}, + {WSA883X_TAGC_CTL, 0x1A}, + {WSA883X_TAGC_TIME, 0x2C}, + {WSA883X_TEMP_CONFIG0, 0x02}, + {WSA883X_TEMP_CONFIG1, 0x02}, + {WSA883X_OTP_REG_1, 0x49}, + {WSA883X_OTP_REG_2, 0x80}, + {WSA883X_OTP_REG_3, 0xC9}, + {WSA883X_OTP_REG_4, 0x40}, + {WSA883X_TAGC_CTL, 0x1B}, + {WSA883X_ADC_2, 0x00}, + {WSA883X_ADC_7, 0x85}, + {WSA883X_ADC_7, 0x87}, + {WSA883X_CKWD_CTL_0, 0x14}, + {WSA883X_CKWD_CTL_1, 0x1B}, + {WSA883X_GMAMP_SUP1, 0xE2}, +}; + +static void wsa883x_init(struct wsa883x_priv *wsa883x) +{ + struct regmap *regmap = wsa883x->regmap; + int variant, version; + + regmap_read(regmap, WSA883X_OTP_REG_0, &variant); + wsa883x->variant = variant & WSA883X_ID_MASK; + + regmap_read(regmap, WSA883X_CHIP_ID0, &version); + wsa883x->version = version; + + switch (wsa883x->variant) { + case WSA8830: + dev_info(wsa883x->dev, "WSA883X Version 1_%d, Variant: WSA8830\n", + wsa883x->version); + break; + case WSA8835: + dev_info(wsa883x->dev, "WSA883X Version 1_%d, Variant: WSA8835\n", + wsa883x->version); + break; + case WSA8832: + dev_info(wsa883x->dev, "WSA883X Version 1_%d, Variant: WSA8832\n", + wsa883x->version); + break; + case WSA8835_V2: + dev_info(wsa883x->dev, "WSA883X Version 1_%d, Variant: WSA8835_V2\n", + wsa883x->version); + break; + default: + break; + } + + wsa883x->comp_offset = COMP_OFFSET2; + + /* Initial settings */ + regmap_multi_reg_write(regmap, reg_init, ARRAY_SIZE(reg_init)); + + if (wsa883x->variant == WSA8830 || wsa883x->variant == WSA8832) { + wsa883x->comp_offset = COMP_OFFSET3; + regmap_update_bits(regmap, WSA883X_DRE_CTL_0, + WSA883X_DRE_OFFSET_MASK, + wsa883x->comp_offset); + } +} + +static int wsa883x_update_status(struct sdw_slave *slave, + enum sdw_slave_status status) +{ + struct wsa883x_priv *wsa883x = dev_get_drvdata(&slave->dev); + + if (status == SDW_SLAVE_ATTACHED && slave->dev_num > 0) + wsa883x_init(wsa883x); + + return 0; +} + +static int wsa883x_port_prep(struct sdw_slave *slave, + struct sdw_prepare_ch *prepare_ch, + enum sdw_port_prep_ops state) +{ + struct wsa883x_priv *wsa883x = dev_get_drvdata(&slave->dev); + + if (state == SDW_OPS_PORT_POST_PREP) + wsa883x->port_prepared[prepare_ch->num - 1] = true; + else + wsa883x->port_prepared[prepare_ch->num - 1] = false; + + return 0; +} + +static struct sdw_slave_ops wsa883x_slave_ops = { + .update_status = wsa883x_update_status, + .port_prep = wsa883x_port_prep, +}; + +static int wsa883x_codec_probe(struct snd_soc_component *comp) +{ + struct wsa883x_priv *wsa883x = snd_soc_component_get_drvdata(comp); + + snd_soc_component_init_regmap(comp, wsa883x->regmap); + + return 0; +} + +static const struct snd_soc_component_driver wsa883x_component_drv = { + .name = "WSA883x", + .probe = wsa883x_codec_probe, +}; + +static int wsa883x_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct wsa883x_priv *wsa883x = dev_get_drvdata(dai->dev); + int i; + + wsa883x->active_ports = 0; + for (i = 0; i < WSA883X_MAX_SWR_PORTS; i++) { + if (!wsa883x->port_enable[i]) + continue; + + wsa883x->port_config[wsa883x->active_ports] = wsa883x_pconfig[i]; + wsa883x->active_ports++; + } + + wsa883x->sconfig.frame_rate = params_rate(params); + + return sdw_stream_add_slave(wsa883x->slave, &wsa883x->sconfig, + wsa883x->port_config, wsa883x->active_ports, + wsa883x->sruntime); +} + +static int wsa883x_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct wsa883x_priv *wsa883x = dev_get_drvdata(dai->dev); + + sdw_stream_remove_slave(wsa883x->slave, wsa883x->sruntime); + + return 0; +} + +static int wsa883x_set_sdw_stream(struct snd_soc_dai *dai, + void *stream, int direction) +{ + struct wsa883x_priv *wsa883x = dev_get_drvdata(dai->dev); + + wsa883x->sruntime = stream; + + return 0; +} + +static int wsa883x_digital_mute(struct snd_soc_dai *dai, int mute, int stream) +{ + struct snd_soc_component *component = dai->component; + + if (mute) { + snd_soc_component_write_field(component, WSA883X_DRE_CTL_1, + WSA883X_DRE_GAIN_EN_MASK, 0); + snd_soc_component_write_field(component, WSA883X_PA_FSM_CTL, + WSA883X_GLOBAL_PA_EN_MASK, 0); + + } else { + snd_soc_component_write_field(component, WSA883X_DRE_CTL_1, + WSA883X_DRE_GAIN_EN_MASK, + WSA883X_DRE_GAIN_FROM_CSR); + snd_soc_component_write_field(component, WSA883X_PA_FSM_CTL, + WSA883X_GLOBAL_PA_EN_MASK, 1); + + } + + return 0; +} + +static const struct snd_soc_dai_ops wsa883x_dai_ops = { + .hw_params = wsa883x_hw_params, + .hw_free = wsa883x_hw_free, + .mute_stream = wsa883x_digital_mute, + .set_stream = wsa883x_set_sdw_stream, +}; + +static struct snd_soc_dai_driver wsa883x_dais[] = { + { + .name = "SPKR", + .playback = { + .stream_name = "SPKR Playback", + .rates = WSA883X_RATES | WSA883X_FRAC_RATES, + .formats = WSA883X_FORMATS, + .rate_max = 8000, + .rate_min = 352800, + .channels_min = 1, + .channels_max = 1, + }, + .ops = &wsa883x_dai_ops, + }, +}; + +static int wsa883x_probe(struct sdw_slave *pdev, + const struct sdw_device_id *id) +{ + struct wsa883x_priv *wsa883x; + struct device *dev = &pdev->dev; + int ret; + + wsa883x = devm_kzalloc(&pdev->dev, sizeof(*wsa883x), GFP_KERNEL); + if (!wsa883x) + return -ENOMEM; + + wsa883x->vdd = devm_regulator_get(dev, "vdd"); + if (IS_ERR(wsa883x->vdd)) { + dev_err(dev, "No vdd regulator found\n"); + return PTR_ERR(wsa883x->vdd); + } + + ret = regulator_enable(wsa883x->vdd); + if (ret) { + dev_err(dev, "Failed to enable vdd regulator (%d)\n", ret); + return ret; + } + + wsa883x->sd_n = devm_gpiod_get_optional(&pdev->dev, "powerdown", + GPIOD_FLAGS_BIT_NONEXCLUSIVE); + if (IS_ERR(wsa883x->sd_n)) { + dev_err(&pdev->dev, "Shutdown Control GPIO not found\n"); + ret = PTR_ERR(wsa883x->sd_n); + goto err; + } + + dev_set_drvdata(&pdev->dev, wsa883x); + wsa883x->slave = pdev; + wsa883x->dev = &pdev->dev; + wsa883x->sconfig.ch_count = 1; + wsa883x->sconfig.bps = 1; + wsa883x->sconfig.direction = SDW_DATA_DIR_RX; + wsa883x->sconfig.type = SDW_STREAM_PDM; + + pdev->prop.sink_ports = GENMASK(WSA883X_MAX_SWR_PORTS, 0); + pdev->prop.sink_dpn_prop = wsa_sink_dpn_prop; + pdev->prop.scp_int1_mask = SDW_SCP_INT1_BUS_CLASH | SDW_SCP_INT1_PARITY; + gpiod_direction_output(wsa883x->sd_n, 1); + + wsa883x->regmap = devm_regmap_init_sdw(pdev, &wsa883x_regmap_config); + if (IS_ERR(wsa883x->regmap)) { + dev_err(&pdev->dev, "regmap_init failed\n"); + ret = PTR_ERR(wsa883x->regmap); + goto err; + } + pm_runtime_set_autosuspend_delay(dev, 3000); + pm_runtime_use_autosuspend(dev); + pm_runtime_mark_last_busy(dev); + pm_runtime_set_active(dev); + pm_runtime_enable(dev); + + ret = devm_snd_soc_register_component(&pdev->dev, + &wsa883x_component_drv, + wsa883x_dais, + ARRAY_SIZE(wsa883x_dais)); +err: + if (ret) + regulator_disable(wsa883x->vdd); + + return ret; + +} + +static int __maybe_unused wsa883x_runtime_suspend(struct device *dev) +{ + struct regmap *regmap = dev_get_regmap(dev, NULL); + struct wsa883x_priv *wsa883x = dev_get_drvdata(dev); + + gpiod_direction_output(wsa883x->sd_n, 0); + + regcache_cache_only(regmap, true); + regcache_mark_dirty(regmap); + + regulator_disable(wsa883x->vdd); + return 0; +} + +static int __maybe_unused wsa883x_runtime_resume(struct device *dev) +{ + struct sdw_slave *slave = dev_to_sdw_dev(dev); + struct regmap *regmap = dev_get_regmap(dev, NULL); + struct wsa883x_priv *wsa883x = dev_get_drvdata(dev); + int ret; + + ret = regulator_enable(wsa883x->vdd); + if (ret) { + dev_err(dev, "Failed to enable vdd regulator (%d)\n", ret); + return ret; + } + + gpiod_direction_output(wsa883x->sd_n, 1); + + wait_for_completion_timeout(&slave->initialization_complete, + msecs_to_jiffies(WSA883X_PROBE_TIMEOUT)); + + usleep_range(20000, 20010); + regcache_cache_only(regmap, false); + regcache_sync(regmap); + + return 0; +} + +static const struct dev_pm_ops wsa883x_pm_ops = { + SET_RUNTIME_PM_OPS(wsa883x_runtime_suspend, wsa883x_runtime_resume, NULL) +}; + +static const struct sdw_device_id wsa883x_swr_id[] = { + SDW_SLAVE_ENTRY(0x0217, 0x0202, 0), + {}, +}; + +static struct sdw_driver wsa883x_codec_driver = { + .driver = { + .name = "wsa883x-codec", + .pm = &wsa883x_pm_ops, + .suppress_bind_attrs = true, + }, + .probe = wsa883x_probe, + .ops = &wsa883x_slave_ops, + .id_table = wsa883x_swr_id, +}; + +module_sdw_driver(wsa883x_codec_driver); + +MODULE_DESCRIPTION("WSA883x codec driver"); +MODULE_LICENSE("GPL"); -- cgit v1.2.3 From 48620f17e071060092197a09663a1c1fe6207829 Mon Sep 17 00:00:00 2001 From: Judy Hsiao Date: Wed, 29 Jun 2022 08:03:45 +0000 Subject: ASoC: rockchip: i2s: Fix the debug level on missing pinctrl Use dev_dbg on missing i2s->pinctrl as the pinctrl property is optional. Fixes: 44f362c2cc6d ("ASoC: rockchip: i2s: switch BCLK to GPIO") Signed-off-by: Judy Hsiao Link: https://lore.kernel.org/r/20220629080345.2427872-1-judyhsiao@chromium.org Signed-off-by: Mark Brown --- sound/soc/rockchip/rockchip_i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 0ed01624a2db..ebd829ac09fe 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -812,7 +812,7 @@ static int rockchip_i2s_probe(struct platform_device *pdev) } } } else { - dev_err(&pdev->dev, "failed to find i2s pinctrl\n"); + dev_dbg(&pdev->dev, "failed to find i2s pinctrl\n"); } i2s_pinctrl_select_bclk_off(i2s); -- cgit v1.2.3 From d29e0a6e3631724c0b36786c6d9616b6e4ebeaa4 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 29 Jun 2022 07:06:30 +0200 Subject: ASoC: max98396: Fix TDM mode BSEL settings In TDM mode, the BSEL register value must be set according to table 5 in the datasheet. This patch adds a lookup function and uses it in max98396_dai_tdm_slot(). As the first 3 entries can also be used for non-TDM setups, the code re-uses the same table for such scenarios. Signed-off-by: Daniel Mack Link: https://lore.kernel.org/r/20220629050630.2848317-1-daniel@zonque.org Signed-off-by: Mark Brown --- sound/soc/codecs/max98396.c | 138 ++++++++++++++++++++++++++++++-------------- sound/soc/codecs/max98396.h | 2 +- 2 files changed, 96 insertions(+), 44 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/max98396.c b/sound/soc/codecs/max98396.c index f28831f4e74b..1b6f053adba2 100644 --- a/sound/soc/codecs/max98396.c +++ b/sound/soc/codecs/max98396.c @@ -438,47 +438,68 @@ static int max98396_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) return 0; } -/* BCLKs per LRCLK */ -static const int bclk_sel_table[] = { - 32, 48, 64, 96, 128, 192, 256, 384, 512, 320, +#define MAX98396_BSEL_32 0x2 +#define MAX98396_BSEL_48 0x3 +#define MAX98396_BSEL_64 0x4 +#define MAX98396_BSEL_96 0x5 +#define MAX98396_BSEL_128 0x6 +#define MAX98396_BSEL_192 0x7 +#define MAX98396_BSEL_256 0x8 +#define MAX98396_BSEL_384 0x9 +#define MAX98396_BSEL_512 0xa +#define MAX98396_BSEL_320 0xb +#define MAX98396_BSEL_250 0xc +#define MAX98396_BSEL_125 0xd + +/* Refer to table 5 in the datasheet */ +static const struct max98396_pcm_config { + int in, out, width, bsel, max_sr; +} max98396_pcm_configs[] = { + { .in = 2, .out = 4, .width = 16, .bsel = MAX98396_BSEL_32, .max_sr = 192000 }, + { .in = 2, .out = 6, .width = 24, .bsel = MAX98396_BSEL_48, .max_sr = 192000 }, + { .in = 2, .out = 8, .width = 32, .bsel = MAX98396_BSEL_64, .max_sr = 192000 }, + { .in = 3, .out = 15, .width = 32, .bsel = MAX98396_BSEL_125, .max_sr = 192000 }, + { .in = 4, .out = 8, .width = 16, .bsel = MAX98396_BSEL_64, .max_sr = 192000 }, + { .in = 4, .out = 12, .width = 24, .bsel = MAX98396_BSEL_96, .max_sr = 192000 }, + { .in = 4, .out = 16, .width = 32, .bsel = MAX98396_BSEL_128, .max_sr = 192000 }, + { .in = 5, .out = 15, .width = 24, .bsel = MAX98396_BSEL_125, .max_sr = 192000 }, + { .in = 7, .out = 15, .width = 16, .bsel = MAX98396_BSEL_125, .max_sr = 192000 }, + { .in = 2, .out = 4, .width = 16, .bsel = MAX98396_BSEL_32, .max_sr = 96000 }, + { .in = 2, .out = 6, .width = 24, .bsel = MAX98396_BSEL_48, .max_sr = 96000 }, + { .in = 2, .out = 8, .width = 32, .bsel = MAX98396_BSEL_64, .max_sr = 96000 }, + { .in = 3, .out = 15, .width = 32, .bsel = MAX98396_BSEL_125, .max_sr = 96000 }, + { .in = 4, .out = 8, .width = 16, .bsel = MAX98396_BSEL_64, .max_sr = 96000 }, + { .in = 4, .out = 12, .width = 24, .bsel = MAX98396_BSEL_96, .max_sr = 96000 }, + { .in = 4, .out = 16, .width = 32, .bsel = MAX98396_BSEL_128, .max_sr = 96000 }, + { .in = 5, .out = 15, .width = 24, .bsel = MAX98396_BSEL_125, .max_sr = 96000 }, + { .in = 7, .out = 15, .width = 16, .bsel = MAX98396_BSEL_125, .max_sr = 96000 }, + { .in = 7, .out = 31, .width = 32, .bsel = MAX98396_BSEL_250, .max_sr = 96000 }, + { .in = 8, .out = 16, .width = 16, .bsel = MAX98396_BSEL_128, .max_sr = 96000 }, + { .in = 8, .out = 24, .width = 24, .bsel = MAX98396_BSEL_192, .max_sr = 96000 }, + { .in = 8, .out = 32, .width = 32, .bsel = MAX98396_BSEL_256, .max_sr = 96000 }, + { .in = 10, .out = 31, .width = 24, .bsel = MAX98396_BSEL_250, .max_sr = 96000 }, + { .in = 15, .out = 31, .width = 16, .bsel = MAX98396_BSEL_250, .max_sr = 96000 }, + { .in = 16, .out = 32, .width = 16, .bsel = MAX98396_BSEL_256, .max_sr = 96000 }, + { .in = 7, .out = 31, .width = 32, .bsel = MAX98396_BSEL_250, .max_sr = 48000 }, + { .in = 10, .out = 31, .width = 24, .bsel = MAX98396_BSEL_250, .max_sr = 48000 }, + { .in = 10, .out = 40, .width = 32, .bsel = MAX98396_BSEL_320, .max_sr = 48000 }, + { .in = 15, .out = 31, .width = 16, .bsel = MAX98396_BSEL_250, .max_sr = 48000 }, + { .in = 16, .out = 48, .width = 24, .bsel = MAX98396_BSEL_384, .max_sr = 48000 }, + { .in = 16, .out = 64, .width = 32, .bsel = MAX98396_BSEL_512, .max_sr = 48000 }, }; -static int max98396_get_bclk_sel(int bclk) +static int max98396_pcm_config_index(int in_slots, int out_slots, int width) { int i; - /* match BCLKs per LRCLK */ - for (i = 0; i < ARRAY_SIZE(bclk_sel_table); i++) { - if (bclk_sel_table[i] == bclk) - return i + 2; - } - return 0; -} - -static int max98396_set_clock(struct snd_soc_component *component, - struct snd_pcm_hw_params *params) -{ - struct max98396_priv *max98396 = snd_soc_component_get_drvdata(component); - /* BCLK/LRCLK ratio calculation */ - int blr_clk_ratio = params_channels(params) * max98396->ch_size; - int value; - if (!max98396->tdm_mode) { - /* BCLK configuration */ - value = max98396_get_bclk_sel(blr_clk_ratio); - if (!value) { - dev_err(component->dev, - "blr_clk_ratio %d unsupported, format %d\n", - blr_clk_ratio, params_format(params)); - return -EINVAL; - } + for (i = 0; i < ARRAY_SIZE(max98396_pcm_configs); i++) { + const struct max98396_pcm_config *c = &max98396_pcm_configs[i]; - regmap_update_bits(max98396->regmap, - MAX98396_R2042_PCM_CLK_SETUP, - MAX98396_PCM_CLK_SETUP_BSEL_MASK, - value); + if (in_slots == c->in && out_slots <= c->out && width == c->width) + return i; } - return 0; + return -1; } static int max98396_dai_hw_params(struct snd_pcm_substream *substream, @@ -489,8 +510,7 @@ static int max98396_dai_hw_params(struct snd_pcm_substream *substream, struct max98396_priv *max98396 = snd_soc_component_get_drvdata(component); unsigned int sampling_rate = 0; unsigned int chan_sz = 0; - int ret, reg; - int status; + int ret, reg, status, bsel = 0; bool update = false; /* pcm mode configuration */ @@ -510,8 +530,6 @@ static int max98396_dai_hw_params(struct snd_pcm_substream *substream, goto err; } - max98396->ch_size = snd_pcm_format_width(params_format(params)); - dev_dbg(component->dev, "format supported %d", params_format(params)); @@ -559,6 +577,33 @@ static int max98396_dai_hw_params(struct snd_pcm_substream *substream, goto err; } + if (max98396->tdm_mode) { + if (params_rate(params) > max98396->tdm_max_samplerate) { + dev_err(component->dev, "TDM sample rate %d too high", + params_rate(params)); + goto err; + } + } else { + /* BCLK configuration */ + ret = max98396_pcm_config_index(params_channels(params), + params_channels(params), + snd_pcm_format_width(params_format(params))); + if (ret < 0) { + dev_err(component->dev, + "no PCM config for %d channels, format %d\n", + params_channels(params), params_format(params)); + goto err; + } + + bsel = max98396_pcm_configs[ret].bsel; + + if (params_rate(params) > max98396_pcm_configs[ret].max_sr) { + dev_err(component->dev, "sample rate %d too high", + params_rate(params)); + goto err; + } + } + ret = regmap_read(max98396->regmap, MAX98396_R210F_GLOBAL_EN, &status); if (ret < 0) goto err; @@ -604,12 +649,16 @@ static int max98396_dai_hw_params(struct snd_pcm_substream *substream, MAX98396_IVADC_SR_MASK, sampling_rate << MAX98396_IVADC_SR_SHIFT); - ret = max98396_set_clock(component, params); + if (bsel) + regmap_update_bits(max98396->regmap, + MAX98396_R2042_PCM_CLK_SETUP, + MAX98396_PCM_CLK_SETUP_BSEL_MASK, + bsel); if (status && update) max98396_global_enable_onoff(max98396->regmap, true); - return ret; + return 0; err: return -EINVAL; @@ -634,13 +683,16 @@ static int max98396_dai_tdm_slot(struct snd_soc_dai *dai, max98396->tdm_mode = true; /* BCLK configuration */ - bsel = max98396_get_bclk_sel(slots * slot_width); - if (bsel == 0) { - dev_err(component->dev, "BCLK %d not supported\n", - slots * slot_width); + ret = max98396_pcm_config_index(slots, slots, slot_width); + if (ret < 0) { + dev_err(component->dev, "no TDM config for %d slots %d bits\n", + slots, slot_width); return -EINVAL; } + bsel = max98396_pcm_configs[ret].bsel; + max98396->tdm_max_samplerate = max98396_pcm_configs[ret].max_sr; + /* Channel size configuration */ switch (slot_width) { case 16: diff --git a/sound/soc/codecs/max98396.h b/sound/soc/codecs/max98396.h index ff330ef61568..7278c779989a 100644 --- a/sound/soc/codecs/max98396.h +++ b/sound/soc/codecs/max98396.h @@ -306,8 +306,8 @@ struct max98396_priv { unsigned int spkfb_slot; unsigned int bypass_slot; bool interleave_mode; - unsigned int ch_size; bool tdm_mode; + int tdm_max_samplerate; int device_id; }; #endif -- cgit v1.2.3 From 3b13b1437dcce4469db575c60d1da4fa9ff80694 Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Tue, 28 Jun 2022 16:39:49 +0800 Subject: ASoC: fsl_micfil: change micfil default settings Previous default settings resulted in loose dynamic range and low sound level. New default configuration changes: - outgain = 2 - quality mode = VLOW0 - dc remover = bypass Signed-off-by: Irina Patru Signed-off-by: Shengjiu Wang Link: https://lore.kernel.org/r/1656405589-29850-1-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_micfil.c | 22 ++++++++++++++++++---- sound/soc/fsl/fsl_micfil.h | 9 +++++++++ 2 files changed, 27 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_micfil.c b/sound/soc/fsl/fsl_micfil.c index 7b88d52f27de..53e105a93d75 100644 --- a/sound/soc/fsl/fsl_micfil.c +++ b/sound/soc/fsl/fsl_micfil.c @@ -48,6 +48,7 @@ struct fsl_micfil { char name[32]; int irq[MICFIL_IRQ_LINES]; enum quality quality; + int dc_remover; }; struct fsl_micfil_soc_data { @@ -317,12 +318,25 @@ static const struct snd_soc_dai_ops fsl_micfil_dai_ops = { static int fsl_micfil_dai_probe(struct snd_soc_dai *cpu_dai) { struct fsl_micfil *micfil = dev_get_drvdata(cpu_dai->dev); - int ret; + struct device *dev = cpu_dai->dev; + unsigned int val = 0; + int ret, i; + + micfil->quality = QUALITY_VLOW0; - micfil->quality = QUALITY_MEDIUM; + /* set default gain to 2 */ + regmap_write(micfil->regmap, REG_MICFIL_OUT_CTRL, 0x22222222); - /* set default gain to max_gain */ - regmap_write(micfil->regmap, REG_MICFIL_OUT_CTRL, 0x77777777); + /* set DC Remover in bypass mode*/ + for (i = 0; i < MICFIL_OUTPUT_CHANNELS; i++) + val |= MICFIL_DC_BYPASS << MICFIL_DC_CHX_SHIFT(i); + ret = regmap_update_bits(micfil->regmap, REG_MICFIL_DC_CTRL, + MICFIL_DC_CTRL_CONFIG, val); + if (ret) { + dev_err(dev, "failed to set DC Remover mode bits\n"); + return ret; + } + micfil->dc_remover = MICFIL_DC_BYPASS; snd_soc_dai_init_dma_data(cpu_dai, NULL, &micfil->dma_params_rx); diff --git a/sound/soc/fsl/fsl_micfil.h b/sound/soc/fsl/fsl_micfil.h index 053caba3caf3..d60285dd07bc 100644 --- a/sound/soc/fsl/fsl_micfil.h +++ b/sound/soc/fsl/fsl_micfil.h @@ -73,6 +73,15 @@ #define MICFIL_FIFO_STAT_FIFOX_OVER(ch) BIT(ch) #define MICFIL_FIFO_STAT_FIFOX_UNDER(ch) BIT((ch) + 8) +/* MICFIL DC Remover Control Register -- REG_MICFIL_DC_CTRL */ +#define MICFIL_DC_CTRL_CONFIG GENMASK(15, 0) +#define MICFIL_DC_CHX_SHIFT(ch) ((ch) << 1) +#define MICFIL_DC_CHX(ch) GENMASK((((ch) << 1) + 1), ((ch) << 1)) +#define MICFIL_DC_CUTOFF_21HZ 0 +#define MICFIL_DC_CUTOFF_83HZ 1 +#define MICFIL_DC_CUTOFF_152Hz 2 +#define MICFIL_DC_BYPASS 3 + /* MICFIL HWVAD0 Control 1 Register -- REG_MICFIL_VAD0_CTRL1*/ #define MICFIL_VAD0_CTRL1_CHSEL GENMASK(26, 24) #define MICFIL_VAD0_CTRL1_CICOSR GENMASK(19, 16) -- cgit v1.2.3 From d6910eaa6fc71c0307e16b310a07cdb347d26d7d Mon Sep 17 00:00:00 2001 From: Judy Hsiao Date: Wed, 29 Jun 2022 08:04:21 +0000 Subject: ASoC: rockchip: i2s: Remove unwanted dma settings in rockchip_i2s_probe Remove the unwanted dma settings in rockchip_i2s_probe. Fixes: 44f362c2cc6d ("ASoC: rockchip: i2s: switch BCLK to GPIO") Signed-off-by: Judy Hsiao Link: https://lore.kernel.org/r/20220629080421.2427933-1-judyhsiao@chromium.org Signed-off-by: Mark Brown --- sound/soc/rockchip/rockchip_i2s.c | 8 -------- 1 file changed, 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index ebd829ac09fe..f783994cc16a 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -817,14 +817,6 @@ static int rockchip_i2s_probe(struct platform_device *pdev) i2s_pinctrl_select_bclk_off(i2s); - i2s->playback_dma_data.addr = res->start + I2S_TXDR; - i2s->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; - i2s->playback_dma_data.maxburst = 4; - - i2s->capture_dma_data.addr = res->start + I2S_RXDR; - i2s->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; - i2s->capture_dma_data.maxburst = 4; - dev_set_drvdata(&pdev->dev, i2s); pm_runtime_enable(&pdev->dev); -- cgit v1.2.3 From 586fb2641371cf7f23a401ab1c79b17e3ec457f4 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 22 Jun 2022 05:54:06 +0000 Subject: ASoC: soc-core.c: fixup snd_soc_of_get_dai_link_cpus() MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit commit 900dedd7e47cc3f ("ASoC: Introduce snd_soc_of_get_dai_link_cpus") adds new snd_soc_of_get_dai_link_cpus(), but it is using "codec" everywhere. It is very strange, and is issue when error case. It should call cpu instead of codec in error case. This patch tidyup it. Fixes: 900dedd7e47cc3f ("ASoC: Introduce snd_soc_of_get_dai_link_cpus") Signed-off-by: Kuninori Morimoto Reviewed-by: Martin Povišer Link: https://lore.kernel.org/r/87zgi5p7k1.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 60e21b06b1dc..89d016323888 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3424,26 +3424,26 @@ int snd_soc_of_get_dai_link_cpus(struct device *dev, struct of_phandle_args args; struct snd_soc_dai_link_component *component; char *name; - int index, num_codecs, ret; + int index, num_cpus, ret; - /* Count the number of CODECs */ + /* Count the number of CPUs */ name = "sound-dai"; - num_codecs = of_count_phandle_with_args(of_node, name, + num_cpus = of_count_phandle_with_args(of_node, name, "#sound-dai-cells"); - if (num_codecs <= 0) { - if (num_codecs == -ENOENT) + if (num_cpus <= 0) { + if (num_cpus == -ENOENT) dev_err(dev, "No 'sound-dai' property\n"); else dev_err(dev, "Bad phandle in 'sound-dai'\n"); - return num_codecs; + return num_cpus; } component = devm_kcalloc(dev, - num_codecs, sizeof(*component), + num_cpus, sizeof(*component), GFP_KERNEL); if (!component) return -ENOMEM; dai_link->cpus = component; - dai_link->num_cpus = num_codecs; + dai_link->num_cpus = num_cpus; /* Parse the list */ for_each_link_cpus(dai_link, index, component) { @@ -3459,7 +3459,7 @@ int snd_soc_of_get_dai_link_cpus(struct device *dev, } return 0; err: - snd_soc_of_put_dai_link_codecs(dai_link); + snd_soc_of_put_dai_link_cpus(dai_link); dai_link->cpus = NULL; dai_link->num_cpus = 0; return ret; -- cgit v1.2.3 From 9cc69528188a4e3eb24370f6c05a92791ac249ba Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 22 Jun 2022 05:54:13 +0000 Subject: ASoC: soc-core.c: share code for snd_soc_of_get_dai_link_cpus/codecs() ASoC has snd_soc_of_get_dai_link_cpus/codecs(), and these are almost same code. The main difference are below. for_each_link_cpus() dai_link->cpus dai_link->num_cpus for_each_link_codecs() dai_link->codecs dai_link->num_codecs Because we need to use these parameters, we can't share full-code for now, but can share some codes. This patch adds __snd_soc_of_get/put_xxx() functions, and share the code. Signed-off-by: Kuninori Morimoto Link: https://lore.kernel.org/r/87y1xpp7ju.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 140 ++++++++++++++++++++++++++------------------------- 1 file changed, 72 insertions(+), 68 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 89d016323888..e824ff1a9fc0 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3303,6 +3303,61 @@ int snd_soc_of_get_dai_name(struct device_node *of_node, } EXPORT_SYMBOL_GPL(snd_soc_of_get_dai_name); +static void __snd_soc_of_put_component(struct snd_soc_dai_link_component *component) +{ + if (component->of_node) { + of_node_put(component->of_node); + component->of_node = NULL; + } +} + +static int __snd_soc_of_get_dai_link_component_alloc( + struct device *dev, struct device_node *of_node, + struct snd_soc_dai_link_component **ret_component, + int *ret_num) +{ + struct snd_soc_dai_link_component *component; + int num; + + /* Count the number of CPUs/CODECs */ + num = of_count_phandle_with_args(of_node, "sound-dai", "#sound-dai-cells"); + if (num <= 0) { + if (num == -ENOENT) + dev_err(dev, "No 'sound-dai' property\n"); + else + dev_err(dev, "Bad phandle in 'sound-dai'\n"); + return num; + } + component = devm_kcalloc(dev, num, sizeof(*component), GFP_KERNEL); + if (!component) + return -ENOMEM; + + *ret_component = component; + *ret_num = num; + + return 0; +} + +static int __snd_soc_of_get_dai_link_component_parse( + struct device_node *of_node, + struct snd_soc_dai_link_component *component, int index) +{ + struct of_phandle_args args; + int ret; + + ret = of_parse_phandle_with_args(of_node, "sound-dai", "#sound-dai-cells", + index, &args); + if (ret) + return ret; + + ret = snd_soc_get_dai_name(&args, &component->dai_name); + if (ret < 0) + return ret; + + component->of_node = args.np; + return 0; +} + /* * snd_soc_of_put_dai_link_codecs - Dereference device nodes in the codecs array * @dai_link: DAI link @@ -3314,12 +3369,8 @@ void snd_soc_of_put_dai_link_codecs(struct snd_soc_dai_link *dai_link) struct snd_soc_dai_link_component *component; int index; - for_each_link_codecs(dai_link, index, component) { - if (!component->of_node) - break; - of_node_put(component->of_node); - component->of_node = NULL; - } + for_each_link_codecs(dai_link, index, component) + __snd_soc_of_put_component(component); } EXPORT_SYMBOL_GPL(snd_soc_of_put_dai_link_codecs); @@ -3341,41 +3392,19 @@ int snd_soc_of_get_dai_link_codecs(struct device *dev, struct device_node *of_node, struct snd_soc_dai_link *dai_link) { - struct of_phandle_args args; struct snd_soc_dai_link_component *component; - char *name; - int index, num_codecs, ret; - - /* Count the number of CODECs */ - name = "sound-dai"; - num_codecs = of_count_phandle_with_args(of_node, name, - "#sound-dai-cells"); - if (num_codecs <= 0) { - if (num_codecs == -ENOENT) - dev_err(dev, "No 'sound-dai' property\n"); - else - dev_err(dev, "Bad phandle in 'sound-dai'\n"); - return num_codecs; - } - component = devm_kcalloc(dev, - num_codecs, sizeof(*component), - GFP_KERNEL); - if (!component) - return -ENOMEM; - dai_link->codecs = component; - dai_link->num_codecs = num_codecs; + int index, ret; + + ret = __snd_soc_of_get_dai_link_component_alloc(dev, of_node, + &dai_link->codecs, &dai_link->num_codecs); + if (ret < 0) + return ret; /* Parse the list */ for_each_link_codecs(dai_link, index, component) { - ret = of_parse_phandle_with_args(of_node, name, - "#sound-dai-cells", - index, &args); + ret = __snd_soc_of_get_dai_link_component_parse(of_node, component, index); if (ret) goto err; - component->of_node = args.np; - ret = snd_soc_get_dai_name(&args, &component->dai_name); - if (ret < 0) - goto err; } return 0; err: @@ -3397,12 +3426,8 @@ void snd_soc_of_put_dai_link_cpus(struct snd_soc_dai_link *dai_link) struct snd_soc_dai_link_component *component; int index; - for_each_link_cpus(dai_link, index, component) { - if (!component->of_node) - break; - of_node_put(component->of_node); - component->of_node = NULL; - } + for_each_link_cpus(dai_link, index, component) + __snd_soc_of_put_component(component); } EXPORT_SYMBOL_GPL(snd_soc_of_put_dai_link_cpus); @@ -3421,41 +3446,20 @@ int snd_soc_of_get_dai_link_cpus(struct device *dev, struct device_node *of_node, struct snd_soc_dai_link *dai_link) { - struct of_phandle_args args; struct snd_soc_dai_link_component *component; - char *name; - int index, num_cpus, ret; + int index, ret; /* Count the number of CPUs */ - name = "sound-dai"; - num_cpus = of_count_phandle_with_args(of_node, name, - "#sound-dai-cells"); - if (num_cpus <= 0) { - if (num_cpus == -ENOENT) - dev_err(dev, "No 'sound-dai' property\n"); - else - dev_err(dev, "Bad phandle in 'sound-dai'\n"); - return num_cpus; - } - component = devm_kcalloc(dev, - num_cpus, sizeof(*component), - GFP_KERNEL); - if (!component) - return -ENOMEM; - dai_link->cpus = component; - dai_link->num_cpus = num_cpus; + ret = __snd_soc_of_get_dai_link_component_alloc(dev, of_node, + &dai_link->cpus, &dai_link->num_cpus); + if (ret < 0) + return ret; /* Parse the list */ for_each_link_cpus(dai_link, index, component) { - ret = of_parse_phandle_with_args(of_node, name, - "#sound-dai-cells", - index, &args); + ret = __snd_soc_of_get_dai_link_component_parse(of_node, component, index); if (ret) goto err; - component->of_node = args.np; - ret = snd_soc_get_dai_name(&args, &component->dai_name); - if (ret < 0) - goto err; } return 0; err: -- cgit v1.2.3 From 66348f178d5a842c8afe52c3b743fb4af24cdb2a Mon Sep 17 00:00:00 2001 From: Yassine Oudjana Date: Wed, 22 Jun 2022 20:13:21 +0400 Subject: ASoC: wcd9335: Use DT bindings instead of local DAI definitions Get DAI indices from DT bindings and remove the currently used local definitions. Signed-off-by: Yassine Oudjana Acked-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20220622161322.168017-3-y.oudjana@protonmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/wcd9335.c | 13 ++----------- 1 file changed, 2 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index 7d40a61b03b0..3554b95462e8 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -24,6 +24,8 @@ #include "wcd9335.h" #include "wcd-clsh-v2.h" +#include + #define WCD9335_RATES_MASK (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_48000 |\ SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000) @@ -203,17 +205,6 @@ enum wcd9335_sido_voltage { SIDO_VOLTAGE_NOMINAL_MV = 1100, }; -enum { - AIF1_PB = 0, - AIF1_CAP, - AIF2_PB, - AIF2_CAP, - AIF3_PB, - AIF3_CAP, - AIF4_PB, - NUM_CODEC_DAIS, -}; - enum { COMPANDER_1, /* HPH_L */ COMPANDER_2, /* HPH_R */ -- cgit v1.2.3 From d2294461b90e0c5b3bbfaaf2c8baff4fd3e2bb13 Mon Sep 17 00:00:00 2001 From: Tom Rix Date: Wed, 29 Jun 2022 14:53:45 -0400 Subject: ASoC: samsung: change gpiod_speaker_power and rx1950_audio from global to static variables sparse reports sound/soc/samsung/rx1950_uda1380.c:131:18: warning: symbol 'gpiod_speaker_power' was not declared. Should it be static? sound/soc/samsung/rx1950_uda1380.c:231:24: warning: symbol 'rx1950_audio' was not declared. Should it be static? Both gpiod_speaker_power and rx1950_audio are only used in rx1950_uda1380.c, so their storage class specifiers should be static. Fixes: 83d74e354200 ("ASoC: samsung: rx1950: turn into platform driver") Signed-off-by: Tom Rix Reviewed-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20220629185345.910406-1-trix@redhat.com Signed-off-by: Mark Brown --- sound/soc/samsung/rx1950_uda1380.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c index ff3acc94a454..abf28321f7d7 100644 --- a/sound/soc/samsung/rx1950_uda1380.c +++ b/sound/soc/samsung/rx1950_uda1380.c @@ -128,7 +128,7 @@ static int rx1950_startup(struct snd_pcm_substream *substream) &hw_rates); } -struct gpio_desc *gpiod_speaker_power; +static struct gpio_desc *gpiod_speaker_power; static int rx1950_spk_power(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) @@ -228,7 +228,7 @@ static int rx1950_probe(struct platform_device *pdev) return devm_snd_soc_register_card(dev, &rx1950_asoc); } -struct platform_driver rx1950_audio = { +static struct platform_driver rx1950_audio = { .driver = { .name = "rx1950-audio", .pm = &snd_soc_pm_ops, -- cgit v1.2.3 From d15534a6f4cff031f1233154f1e275302c03e5d4 Mon Sep 17 00:00:00 2001 From: Marek Vasut Date: Tue, 28 Jun 2022 18:58:07 +0200 Subject: ASoC: doc: Update dead links The alsa-project documentation is now part of the kernel docs, the original links are long dead, update links. Signed-off-by: Marek Vasut Cc: Mark Brown Cc: Takashi Iwai Link: https://lore.kernel.org/r/20220628165807.152191-1-marex@denx.de Signed-off-by: Mark Brown --- sound/pci/ens1370.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index 94efe347a97a..89210b2c7342 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -8,7 +8,7 @@ /* Power-Management-Code ( CONFIG_PM ) * for ens1371 only ( FIXME ) * derived from cs4281.c, atiixp.c and via82xx.c - * using http://www.alsa-project.org/~tiwai/writing-an-alsa-driver/ + * using https://www.kernel.org/doc/html/latest/sound/kernel-api/writing-an-alsa-driver.html * by Kurt J. Bosch */ -- cgit v1.2.3 From cdb09e6231433b65e31c40fbe298099db6513a7f Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Thu, 30 Jun 2022 13:36:33 +0100 Subject: ASoC: codecs: wsa883x: add control, dapm widgets and map Add controls, dapm widgets along with route. Signed-off-by: Srinivas Kandagatla Link: https://lore.kernel.org/r/20220630123633.8047-1-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/wsa883x.c | 200 +++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 200 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wsa883x.c b/sound/soc/codecs/wsa883x.c index 856709ec017e..e8f519e89213 100644 --- a/sound/soc/codecs/wsa883x.c +++ b/sound/soc/codecs/wsa883x.c @@ -473,6 +473,19 @@ enum wsa_port_ids { WSA883X_PORT_VISENSE, }; +static const char * const wsa_dev_mode_text[] = { + "Speaker", "Receiver", "Ultrasound" +}; + +enum { + SPEAKER, + RECEIVER, + ULTRASOUND, +}; + +static const struct soc_enum wsa_dev_mode_enum = + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(wsa_dev_mode_text), wsa_dev_mode_text); + /* 4 ports */ static struct sdw_dpn_prop wsa_sink_dpn_prop[WSA883X_MAX_SWR_PORTS] = { { @@ -1066,6 +1079,94 @@ static struct sdw_slave_ops wsa883x_slave_ops = { .port_prep = wsa883x_port_prep, }; +static int wsa_dev_mode_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct wsa883x_priv *wsa883x = snd_soc_component_get_drvdata(component); + + ucontrol->value.enumerated.item[0] = wsa883x->dev_mode; + + return 0; +} + +static int wsa_dev_mode_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct wsa883x_priv *wsa883x = snd_soc_component_get_drvdata(component); + + if (wsa883x->dev_mode == ucontrol->value.enumerated.item[0]) + return 0; + + wsa883x->dev_mode = ucontrol->value.enumerated.item[0]; + + return 1; +} + +static const DECLARE_TLV_DB_SCALE(pa_gain, -300, 150, -300); + +static int wsa883x_get_swr_port(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct wsa883x_priv *data = snd_soc_component_get_drvdata(comp); + struct soc_mixer_control *mixer = (struct soc_mixer_control *)kcontrol->private_value; + int portidx = mixer->reg; + + ucontrol->value.integer.value[0] = data->port_enable[portidx]; + + return 0; +} + +static int wsa883x_set_swr_port(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct wsa883x_priv *data = snd_soc_component_get_drvdata(comp); + struct soc_mixer_control *mixer = (struct soc_mixer_control *)kcontrol->private_value; + int portidx = mixer->reg; + + if (ucontrol->value.integer.value[0]) { + if (data->port_enable[portidx]) + return 0; + + data->port_enable[portidx] = true; + } else { + if (!data->port_enable[portidx]) + return 0; + + data->port_enable[portidx] = false; + } + + return 1; +} + +static int wsa883x_get_comp_offset(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct wsa883x_priv *wsa883x = snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = wsa883x->comp_offset; + + return 0; +} + +static int wsa883x_set_comp_offset(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct wsa883x_priv *wsa883x = snd_soc_component_get_drvdata(component); + + if (wsa883x->comp_offset == ucontrol->value.integer.value[0]) + return 0; + + wsa883x->comp_offset = ucontrol->value.integer.value[0]; + + return 1; +} + static int wsa883x_codec_probe(struct snd_soc_component *comp) { struct wsa883x_priv *wsa883x = snd_soc_component_get_drvdata(comp); @@ -1075,9 +1176,108 @@ static int wsa883x_codec_probe(struct snd_soc_component *comp) return 0; } +static int wsa883x_spkr_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct wsa883x_priv *wsa883x = snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + switch (wsa883x->dev_mode) { + case RECEIVER: + snd_soc_component_write_field(component, WSA883X_CDC_PATH_MODE, + WSA883X_RXD_MODE_MASK, + WSA883X_RXD_MODE_HIFI); + snd_soc_component_write_field(component, WSA883X_SPKR_PWM_CLK_CTL, + WSA883X_SPKR_PWM_FREQ_SEL_MASK, + WSA883X_SPKR_PWM_FREQ_F600KHZ); + snd_soc_component_write_field(component, WSA883X_DRE_CTL_0, + WSA883X_DRE_PROG_DELAY_MASK, 0x0); + break; + case SPEAKER: + snd_soc_component_write_field(component, WSA883X_CDC_PATH_MODE, + WSA883X_RXD_MODE_MASK, + WSA883X_RXD_MODE_NORMAL); + snd_soc_component_write_field(component, WSA883X_SPKR_PWM_CLK_CTL, + WSA883X_SPKR_PWM_FREQ_SEL_MASK, + WSA883X_SPKR_PWM_FREQ_F300KHZ); + snd_soc_component_write_field(component, WSA883X_DRE_CTL_0, + WSA883X_DRE_PROG_DELAY_MASK, 0x9); + default: + break; + } + + snd_soc_component_write_field(component, WSA883X_DRE_CTL_1, + WSA883X_DRE_GAIN_EN_MASK, + WSA883X_DRE_GAIN_FROM_CSR); + if (wsa883x->port_enable[WSA883X_PORT_COMP]) + snd_soc_component_write_field(component, WSA883X_DRE_CTL_0, + WSA883X_DRE_OFFSET_MASK, + wsa883x->comp_offset); + snd_soc_component_write_field(component, WSA883X_VBAT_ADC_FLT_CTL, + WSA883X_VBAT_ADC_COEF_SEL_MASK, + WSA883X_VBAT_ADC_COEF_F_1DIV16); + snd_soc_component_write_field(component, WSA883X_VBAT_ADC_FLT_CTL, + WSA883X_VBAT_ADC_FLT_EN_MASK, 0x1); + snd_soc_component_write_field(component, WSA883X_PDM_WD_CTL, + WSA883X_PDM_EN_MASK, + WSA883X_PDM_ENABLE); + snd_soc_component_write_field(component, WSA883X_PA_FSM_CTL, + WSA883X_GLOBAL_PA_EN_MASK, + WSA883X_GLOBAL_PA_ENABLE); + + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_component_write_field(component, WSA883X_VBAT_ADC_FLT_CTL, + WSA883X_VBAT_ADC_FLT_EN_MASK, 0x0); + snd_soc_component_write_field(component, WSA883X_VBAT_ADC_FLT_CTL, + WSA883X_VBAT_ADC_COEF_SEL_MASK, + WSA883X_VBAT_ADC_COEF_F_1DIV2); + snd_soc_component_write_field(component, WSA883X_PA_FSM_CTL, + WSA883X_GLOBAL_PA_EN_MASK, 0); + snd_soc_component_write_field(component, WSA883X_PDM_WD_CTL, + WSA883X_PDM_EN_MASK, 0); + break; + } + return 0; +} + +static const struct snd_soc_dapm_widget wsa883x_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("IN"), + SND_SOC_DAPM_SPK("SPKR", wsa883x_spkr_event), +}; + +static const struct snd_kcontrol_new wsa883x_snd_controls[] = { + SOC_SINGLE_RANGE_TLV("PA Volume", WSA883X_DRE_CTL_1, 1, + 0x0, 0x1f, 1, pa_gain), + SOC_ENUM_EXT("WSA MODE", wsa_dev_mode_enum, + wsa_dev_mode_get, wsa_dev_mode_put), + SOC_SINGLE_EXT("COMP Offset", SND_SOC_NOPM, 0, 4, 0, + wsa883x_get_comp_offset, wsa883x_set_comp_offset), + SOC_SINGLE_EXT("DAC Switch", WSA883X_PORT_DAC, 0, 1, 0, + wsa883x_get_swr_port, wsa883x_set_swr_port), + SOC_SINGLE_EXT("COMP Switch", WSA883X_PORT_COMP, 0, 1, 0, + wsa883x_get_swr_port, wsa883x_set_swr_port), + SOC_SINGLE_EXT("BOOST Switch", WSA883X_PORT_BOOST, 0, 1, 0, + wsa883x_get_swr_port, wsa883x_set_swr_port), + SOC_SINGLE_EXT("VISENSE Switch", WSA883X_PORT_VISENSE, 0, 1, 0, + wsa883x_get_swr_port, wsa883x_set_swr_port), +}; + +static const struct snd_soc_dapm_route wsa883x_audio_map[] = { + {"SPKR", NULL, "IN"}, +}; + static const struct snd_soc_component_driver wsa883x_component_drv = { .name = "WSA883x", .probe = wsa883x_codec_probe, + .controls = wsa883x_snd_controls, + .num_controls = ARRAY_SIZE(wsa883x_snd_controls), + .dapm_widgets = wsa883x_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wsa883x_dapm_widgets), + .dapm_routes = wsa883x_audio_map, + .num_dapm_routes = ARRAY_SIZE(wsa883x_audio_map), }; static int wsa883x_hw_params(struct snd_pcm_substream *substream, -- cgit v1.2.3 From 871325d800ed532ba5874257f04bb4ae75125bc4 Mon Sep 17 00:00:00 2001 From: Tom Rix Date: Wed, 29 Jun 2022 16:18:11 -0400 Subject: ASoC: samsung: change neo1973_audio from a global to static sparse reports sound/soc/samsung/neo1973_wm8753.c:347:24: warning: symbol 'neo1973_audio' was not declared. Should it be static? neo1973_audio is only used in neo1973_wm8753.c, so it's storage class specifier should be static. Fixes: e26a2abcc246 ("ASoC: samsung: neo1973: turn into platform driver") Signed-off-by: Tom Rix Reviewed-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20220629201811.2537853-1-trix@redhat.com Signed-off-by: Mark Brown --- sound/soc/samsung/neo1973_wm8753.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index c98b68567a89..e9f2334028bf 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -344,7 +344,7 @@ static int neo1973_probe(struct platform_device *pdev) return devm_snd_soc_register_card(dev, &neo1973); } -struct platform_driver neo1973_audio = { +static struct platform_driver neo1973_audio = { .driver = { .name = "neo1973-audio", .pm = &snd_soc_pm_ops, -- cgit v1.2.3 From e8010efc7b83038d1c18abe1b8d171e3c7d4ed92 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 30 Jun 2022 11:14:59 +0100 Subject: ASoC: wm_adsp: Minor clean and redundant code removal The cs_dsp core will return an error if passed a NULL cs_dsp struct so there is no need for the wm_adsp_write|read_ctl functions to manually check that. The cs_dsp core will also check the data is within bounds of the control so the additional bounds check is redundant too. Simplify things a bit by removing said code. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220630101459.3442327-1-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 25 +++++-------------------- 1 file changed, 5 insertions(+), 20 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index a7784ac15dde..cfaa45ede916 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -675,21 +675,12 @@ static void wm_adsp_control_remove(struct cs_dsp_coeff_ctl *cs_ctl) int wm_adsp_write_ctl(struct wm_adsp *dsp, const char *name, int type, unsigned int alg, void *buf, size_t len) { - struct cs_dsp_coeff_ctl *cs_ctl; + struct cs_dsp_coeff_ctl *cs_ctl = cs_dsp_get_ctl(&dsp->cs_dsp, name, type, alg); struct wm_coeff_ctl *ctl; struct snd_kcontrol *kcontrol; char ctl_name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; int ret; - cs_ctl = cs_dsp_get_ctl(&dsp->cs_dsp, name, type, alg); - if (!cs_ctl) - return -EINVAL; - - ctl = cs_ctl->priv; - - if (len > cs_ctl->len) - return -EINVAL; - ret = cs_dsp_coeff_write_ctrl(cs_ctl, 0, buf, len); if (ret) return ret; @@ -697,6 +688,8 @@ int wm_adsp_write_ctl(struct wm_adsp *dsp, const char *name, int type, if (cs_ctl->flags & WMFW_CTL_FLAG_SYS) return 0; + ctl = cs_ctl->priv; + if (dsp->component->name_prefix) snprintf(ctl_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "%s %s", dsp->component->name_prefix, ctl->name); @@ -720,16 +713,8 @@ EXPORT_SYMBOL_GPL(wm_adsp_write_ctl); int wm_adsp_read_ctl(struct wm_adsp *dsp, const char *name, int type, unsigned int alg, void *buf, size_t len) { - struct cs_dsp_coeff_ctl *cs_ctl; - - cs_ctl = cs_dsp_get_ctl(&dsp->cs_dsp, name, type, alg); - if (!cs_ctl) - return -EINVAL; - - if (len > cs_ctl->len) - return -EINVAL; - - return cs_dsp_coeff_read_ctrl(cs_ctl, 0, buf, len); + return cs_dsp_coeff_read_ctrl(cs_dsp_get_ctl(&dsp->cs_dsp, name, type, alg), + 0, buf, len); } EXPORT_SYMBOL_GPL(wm_adsp_read_ctl); -- cgit v1.2.3 From d8d6253b36f55d199590ef908712fe52bb39ee97 Mon Sep 17 00:00:00 2001 From: Li kunyu Date: Thu, 30 Jun 2022 10:03:47 +0800 Subject: ASoC: tegra: delete a semicolon extra semicolons could be deleted. Signed-off-by: Li kunyu Link: https://lore.kernel.org/r/20220630020347.7148-1-kunyu@nfschina.com Signed-off-by: Mark Brown --- sound/soc/tegra/tegra210_adx.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra210_adx.c b/sound/soc/tegra/tegra210_adx.c index 3785cade2d9a..49691d2cce50 100644 --- a/sound/soc/tegra/tegra210_adx.c +++ b/sound/soc/tegra/tegra210_adx.c @@ -191,7 +191,7 @@ static int tegra210_adx_put_byte_map(struct snd_kcontrol *kcontrol, unsigned char *bytes_map = (unsigned char *)&adx->map; int value = ucontrol->value.integer.value[0]; struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value;; + (struct soc_mixer_control *)kcontrol->private_value; if (value == bytes_map[mc->reg]) return 0; -- cgit v1.2.3 From cf6af24b54903f9f70c29b3e5b19cb72cc862d60 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Thu, 30 Jun 2022 14:00:22 +0100 Subject: ASoC: codecs: wsa881x: handle timeouts in resume path Currently we do not check if SoundWire slave initialization timeout expired before continuing to access its registers. Its possible that the registers are not accessible if timeout is expired. Handle this by returning timeout in resume path. Reported-by: Pierre-Louis Bossart Fixes: 8dd552458361 ("ASoC: codecs: wsa881x: add runtime pm support") Signed-off-by: Srinivas Kandagatla Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220630130023.9308-1-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/wsa881x.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wsa881x.c b/sound/soc/codecs/wsa881x.c index dc954b85a988..6c8b1db649b8 100644 --- a/sound/soc/codecs/wsa881x.c +++ b/sound/soc/codecs/wsa881x.c @@ -1173,11 +1173,17 @@ static int __maybe_unused wsa881x_runtime_resume(struct device *dev) struct sdw_slave *slave = dev_to_sdw_dev(dev); struct regmap *regmap = dev_get_regmap(dev, NULL); struct wsa881x_priv *wsa881x = dev_get_drvdata(dev); + unsigned long time; gpiod_direction_output(wsa881x->sd_n, 1); - wait_for_completion_timeout(&slave->initialization_complete, - msecs_to_jiffies(WSA881X_PROBE_TIMEOUT)); + time = wait_for_completion_timeout(&slave->initialization_complete, + msecs_to_jiffies(WSA881X_PROBE_TIMEOUT)); + if (!time) { + dev_err(dev, "Initialization not complete, timed out\n"); + gpiod_direction_output(wsa881x->sd_n, 0); + return -ETIMEDOUT; + } regcache_cache_only(regmap, false); regcache_sync(regmap); -- cgit v1.2.3 From 0df73e1a9f7b1152ace21b6406138f7487239128 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Thu, 30 Jun 2022 14:00:23 +0100 Subject: ASoC: codecs: wsa883x: handle timeouts in resume path Currently we do not check if SoundWire slave initialization timeout expired before continuing to access its registers. Its possible that the registers are not accessible if timeout is expired. Handle this by returning timeout in resume path. Reported-by: Pierre-Louis Bossart Fixes: 43b8c7dc85a1 ("ASoC: codecs: add wsa883x amplifier support") Signed-off-by: Srinivas Kandagatla Link: https://lore.kernel.org/r/20220630130023.9308-2-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/wsa883x.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wsa883x.c b/sound/soc/codecs/wsa883x.c index e8f519e89213..40c7d64a9c41 100644 --- a/sound/soc/codecs/wsa883x.c +++ b/sound/soc/codecs/wsa883x.c @@ -1455,6 +1455,7 @@ static int __maybe_unused wsa883x_runtime_resume(struct device *dev) struct sdw_slave *slave = dev_to_sdw_dev(dev); struct regmap *regmap = dev_get_regmap(dev, NULL); struct wsa883x_priv *wsa883x = dev_get_drvdata(dev); + unsigned long time; int ret; ret = regulator_enable(wsa883x->vdd); @@ -1465,8 +1466,14 @@ static int __maybe_unused wsa883x_runtime_resume(struct device *dev) gpiod_direction_output(wsa883x->sd_n, 1); - wait_for_completion_timeout(&slave->initialization_complete, - msecs_to_jiffies(WSA883X_PROBE_TIMEOUT)); + time = wait_for_completion_timeout(&slave->initialization_complete, + msecs_to_jiffies(WSA883X_PROBE_TIMEOUT)); + if (!time) { + dev_err(dev, "Initialization not complete, timed out\n"); + gpiod_direction_output(wsa883x->sd_n, 0); + regulator_disable(wsa883x->vdd); + return -ETIMEDOUT; + } usleep_range(20000, 20010); regcache_cache_only(regmap, false); -- cgit v1.2.3 From 68f26639dc40b5d6aca201f3e250a1538e68eae6 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Fri, 1 Jul 2022 13:55:15 +0100 Subject: ASoC: codecs: wsa883x: add missing break statement Add missing break in one of the switch statement. Reported-by: kernel test robot Fixes: cdb09e623143 ("ASoC: codecs: wsa883x: add control, dapm widgets and map") Signed-off-by: Srinivas Kandagatla Link: https://lore.kernel.org/r/20220701125515.32332-1-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/wsa883x.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/wsa883x.c b/sound/soc/codecs/wsa883x.c index 40c7d64a9c41..dcd88175b9cd 100644 --- a/sound/soc/codecs/wsa883x.c +++ b/sound/soc/codecs/wsa883x.c @@ -1204,6 +1204,7 @@ static int wsa883x_spkr_event(struct snd_soc_dapm_widget *w, WSA883X_SPKR_PWM_FREQ_F300KHZ); snd_soc_component_write_field(component, WSA883X_DRE_CTL_0, WSA883X_DRE_PROG_DELAY_MASK, 0x9); + break; default: break; } -- cgit v1.2.3 From f507c0c67dac57d2bcd5dcae4b6139b0305d8957 Mon Sep 17 00:00:00 2001 From: Liang He Date: Sat, 2 Jul 2022 10:01:09 +0800 Subject: ASoC: qcom: Fix missing of_node_put() in asoc_qcom_lpass_cpu_platform_probe() We should call of_node_put() for the reference 'dsp_of_node' returned by of_parse_phandle() which will increase the refcount. Fixes: 9bae4880acee ("ASoC: qcom: move ipq806x specific bits out of lpass driver.") Co-authored-by: Miaoqian Lin Signed-off-by: Liang He Link: https://lore.kernel.org/r/20220702020109.263980-1-windhl@126.com Signed-off-by: Mark Brown --- sound/soc/qcom/lpass-cpu.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index 263a066769d4..8a56f38dc7e8 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -1091,6 +1091,7 @@ int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev) dsp_of_node = of_parse_phandle(pdev->dev.of_node, "qcom,adsp", 0); if (dsp_of_node) { dev_err(dev, "DSP exists and holds audio resources\n"); + of_node_put(dsp_of_node); return -EBUSY; } -- cgit v1.2.3 From 1d5c7a91dfc2b7a5672a2706553e5782482d6e6f Mon Sep 17 00:00:00 2001 From: Jiapeng Chong Date: Fri, 1 Jul 2022 15:30:39 +0800 Subject: ASoC: codecs: max98088: Clean up some inconsistent indenting This was found by coccicheck: sound/soc/codecs/max98088.c:1761 max98088_i2c_probe() warn: inconsistent indenting. Signed-off-by: Jiapeng Chong Link: https://lore.kernel.org/r/20220701073039.64556-1-jiapeng.chong@linux.alibaba.com Signed-off-by: Mark Brown --- sound/soc/codecs/max98088.c | 32 ++++++++++++++++---------------- 1 file changed, 16 insertions(+), 16 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 08e5c606ff27..5435a49604cf 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -1745,18 +1745,18 @@ MODULE_DEVICE_TABLE(i2c, max98088_i2c_id); static int max98088_i2c_probe(struct i2c_client *i2c) { - struct max98088_priv *max98088; - int ret; - const struct i2c_device_id *id; + struct max98088_priv *max98088; + int ret; + const struct i2c_device_id *id; - max98088 = devm_kzalloc(&i2c->dev, sizeof(struct max98088_priv), - GFP_KERNEL); - if (max98088 == NULL) - return -ENOMEM; + max98088 = devm_kzalloc(&i2c->dev, sizeof(struct max98088_priv), + GFP_KERNEL); + if (max98088 == NULL) + return -ENOMEM; - max98088->regmap = devm_regmap_init_i2c(i2c, &max98088_regmap); - if (IS_ERR(max98088->regmap)) - return PTR_ERR(max98088->regmap); + max98088->regmap = devm_regmap_init_i2c(i2c, &max98088_regmap); + if (IS_ERR(max98088->regmap)) + return PTR_ERR(max98088->regmap); max98088->mclk = devm_clk_get(&i2c->dev, "mclk"); if (IS_ERR(max98088->mclk)) @@ -1764,14 +1764,14 @@ static int max98088_i2c_probe(struct i2c_client *i2c) return PTR_ERR(max98088->mclk); id = i2c_match_id(max98088_i2c_id, i2c); - max98088->devtype = id->driver_data; + max98088->devtype = id->driver_data; - i2c_set_clientdata(i2c, max98088); - max98088->pdata = i2c->dev.platform_data; + i2c_set_clientdata(i2c, max98088); + max98088->pdata = i2c->dev.platform_data; - ret = devm_snd_soc_register_component(&i2c->dev, - &soc_component_dev_max98088, &max98088_dai[0], 2); - return ret; + ret = devm_snd_soc_register_component(&i2c->dev, &soc_component_dev_max98088, + &max98088_dai[0], 2); + return ret; } #if defined(CONFIG_OF) -- cgit v1.2.3 From 679139ea62e3e78542cd409c2437ac1da9f31026 Mon Sep 17 00:00:00 2001 From: Yang Yingliang Date: Mon, 4 Jul 2022 15:51:34 +0800 Subject: ASoC: fsl: pcm030-audio-fabric: use platform_device_unregsiter() Replace platform_device_del/put() with platform_device_unregsiter() to simplify code. Signed-off-by: Yang Yingliang Link: https://lore.kernel.org/r/20220704075134.26230-1-yangyingliang@huawei.com Signed-off-by: Mark Brown --- sound/soc/fsl/pcm030-audio-fabric.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c index e4c805acc349..997c3e66c636 100644 --- a/sound/soc/fsl/pcm030-audio-fabric.c +++ b/sound/soc/fsl/pcm030-audio-fabric.c @@ -101,8 +101,7 @@ static int pcm030_fabric_probe(struct platform_device *op) ret = snd_soc_register_card(card); if (ret) { dev_err(&op->dev, "snd_soc_register_card() failed: %d\n", ret); - platform_device_del(pdata->codec_device); - platform_device_put(pdata->codec_device); + platform_device_unregister(pdata->codec_device); } platform_set_drvdata(op, pdata); -- cgit v1.2.3 From 3684020a82ff43a64b5a7e42564ee7e2065d3011 Mon Sep 17 00:00:00 2001 From: Zhu Ning Date: Mon, 4 Jul 2022 09:24:16 +0800 Subject: ASoC: codes: Add support for ES8316 producer mode The AMD acp-es8336 machine driver requires ES8316 run in producer mode, which is not supported previously. Signed-off-by: David Yang Signed-off-by: Zhu Ning Link: https://lore.kernel.org/r/20220704012416.3165-1-zhuning0077@gmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/es8316.c | 20 +++++++++++++++----- 1 file changed, 15 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index eb15be9095e7..de7185f73e1e 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -401,10 +401,8 @@ static int es8316_set_dai_fmt(struct snd_soc_dai *codec_dai, u8 clksw; u8 mask; - if ((fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) != SND_SOC_DAIFMT_CBC_CFC) { - dev_err(component->dev, "Codec driver only supports consumer mode\n"); - return -EINVAL; - } + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) == SND_SOC_DAIFMT_CBP_CFP) + serdata1 |= ES8316_SERDATA1_MASTER; if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) != SND_SOC_DAIFMT_I2S) { dev_err(component->dev, "Codec driver only supports I2S format\n"); @@ -464,6 +462,8 @@ static int es8316_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_component *component = dai->component; struct es8316_priv *es8316 = snd_soc_component_get_drvdata(component); u8 wordlen = 0; + u8 bclk_divider; + u16 lrck_divider; int i; /* Validate supported sample rates that are autodetected from MCLK */ @@ -477,19 +477,24 @@ static int es8316_pcm_hw_params(struct snd_pcm_substream *substream, } if (i == NR_SUPPORTED_MCLK_LRCK_RATIOS) return -EINVAL; - + lrck_divider = es8316->sysclk / params_rate(params); + bclk_divider = lrck_divider / 4; switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: wordlen = ES8316_SERDATA2_LEN_16; + bclk_divider /= 16; break; case SNDRV_PCM_FORMAT_S20_3LE: wordlen = ES8316_SERDATA2_LEN_20; + bclk_divider /= 20; break; case SNDRV_PCM_FORMAT_S24_LE: wordlen = ES8316_SERDATA2_LEN_24; + bclk_divider /= 24; break; case SNDRV_PCM_FORMAT_S32_LE: wordlen = ES8316_SERDATA2_LEN_32; + bclk_divider /= 32; break; default: return -EINVAL; @@ -499,6 +504,11 @@ static int es8316_pcm_hw_params(struct snd_pcm_substream *substream, ES8316_SERDATA2_LEN_MASK, wordlen); snd_soc_component_update_bits(component, ES8316_SERDATA_ADC, ES8316_SERDATA2_LEN_MASK, wordlen); + snd_soc_component_update_bits(component, ES8316_SERDATA1, 0x1f, bclk_divider); + snd_soc_component_update_bits(component, ES8316_CLKMGR_ADCDIV1, 0x0f, lrck_divider >> 8); + snd_soc_component_update_bits(component, ES8316_CLKMGR_ADCDIV2, 0xff, lrck_divider & 0xff); + snd_soc_component_update_bits(component, ES8316_CLKMGR_DACDIV1, 0x0f, lrck_divider >> 8); + snd_soc_component_update_bits(component, ES8316_CLKMGR_DACDIV2, 0xff, lrck_divider & 0xff); return 0; } -- cgit v1.2.3 From 978bd27c9aed13d7d739bdcdcf98cbca9106b0ec Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Mon, 4 Jul 2022 09:50:16 +0800 Subject: ASoC: fsl_micfil: Add legacy_dai_naming flag Need to add legacy_dai_naming flag otherwise there will be issue when registerring component, that cause the probe failure. Fixes: 1e63fcc74ace ("ASoC: fsl: Migrate to new style legacy DAI naming flag") Signed-off-by: Shengjiu Wang Link: https://lore.kernel.org/r/1656899417-4775-1-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_micfil.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_micfil.c b/sound/soc/fsl/fsl_micfil.c index 53e105a93d75..be9781ad8849 100644 --- a/sound/soc/fsl/fsl_micfil.c +++ b/sound/soc/fsl/fsl_micfil.c @@ -367,7 +367,7 @@ static const struct snd_soc_component_driver fsl_micfil_component = { .name = "fsl-micfil-dai", .controls = fsl_micfil_snd_controls, .num_controls = ARRAY_SIZE(fsl_micfil_snd_controls), - + .legacy_dai_naming = 1, }; /* REGMAP */ -- cgit v1.2.3 From 446499743b26958a58891a8f9a061deb5cce7c82 Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Mon, 4 Jul 2022 09:50:17 +0800 Subject: ASoC: fsl_asrc_dma: Add legacy_dai_naming flag Need to add legacy_dai_naming flag otherwise there will be issue when registerring component, that cause the probe failure. Fixes: 1e63fcc74ace ("ASoC: fsl: Migrate to new style legacy DAI naming flag") Signed-off-by: Shengjiu Wang Link: https://lore.kernel.org/r/1656899417-4775-2-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_asrc_dma.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c index 33eabb96340e..12ddf2320f2d 100644 --- a/sound/soc/fsl/fsl_asrc_dma.c +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -455,5 +455,6 @@ struct snd_soc_component_driver fsl_asrc_component = { .close = fsl_asrc_dma_shutdown, .pointer = fsl_asrc_dma_pcm_pointer, .pcm_construct = fsl_asrc_dma_pcm_new, + .legacy_dai_naming = 1, }; EXPORT_SYMBOL_GPL(fsl_asrc_component); -- cgit v1.2.3 From acf981f94edca13c85fa24dd8511cdc6bd4c98ed Mon Sep 17 00:00:00 2001 From: Jiapeng Chong Date: Fri, 1 Jul 2022 15:28:50 +0800 Subject: ASoC: tegra20_ac97: Fix missing error code in tegra20_ac97_platform_probe() The error code is missing in this code scenario, add the error code '-EINVAL' to the return value 'ret'. This was found by coccicheck: sound/soc/tegra/tegra20_ac97.c:357 tegra20_ac97_platform_probe() warn: missing error code 'ret'. Signed-off-by: Jiapeng Chong Link: https://lore.kernel.org/r/20220701072850.62408-1-jiapeng.chong@linux.alibaba.com Signed-off-by: Mark Brown --- sound/soc/tegra/tegra20_ac97.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c index e17375c6cddb..87facfbcdd11 100644 --- a/sound/soc/tegra/tegra20_ac97.c +++ b/sound/soc/tegra/tegra20_ac97.c @@ -354,6 +354,7 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) } } else { dev_err(&pdev->dev, "no codec-reset GPIO supplied\n"); + ret = -EINVAL; goto err_clk_put; } @@ -361,6 +362,7 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) "nvidia,codec-sync-gpio", 0); if (!gpio_is_valid(ac97->sync_gpio)) { dev_err(&pdev->dev, "no codec-sync GPIO supplied\n"); + ret = -EINVAL; goto err_clk_put; } -- cgit v1.2.3 From 4b8ea38fabab45ad911a32a336416062553dfe9c Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Jan=20Sch=C3=A4r?= Date: Mon, 27 Jun 2022 19:18:54 +0200 Subject: ALSA: usb-audio: Support jack detection on Dell dock MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The Dell WD15 dock has a headset and a line out port. Add support for detecting if a jack is inserted into one of these ports. For the headset jack, additionally determine if a mic is present. The WD15 contains an ALC4020 USB audio controller and ALC3263 audio codec from Realtek. It is a UAC 1 device, and UAC 1 does not support jack detection. Instead, jack detection works by sending HD Audio commands over vendor-type USB messages. I found out how it works by looking at USB captures on Windows. The audio codec is very similar to the one supported by sound/soc/codecs/rt298.c / rt298.h, some constant names and the mic detection are adapted from there. The realtek_add_jack function is adapted from build_connector_control in sound/usb/mixer.c. I tested this on a WD15 dock with the latest firmware. Signed-off-by: Jan Schär Link: https://lore.kernel.org/r/20220627171855.42338-1-jan@jschaer.ch Signed-off-by: Takashi Iwai --- sound/usb/mixer_quirks.c | 167 +++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 167 insertions(+) (limited to 'sound') diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index d35cf54cab33..66b6476994eb 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -24,6 +24,7 @@ #include #include #include +#include #include #include #include @@ -1934,6 +1935,169 @@ static int snd_soundblaster_e1_switch_create(struct usb_mixer_interface *mixer) NULL); } +/* + * Dell WD15 dock jack detection + * + * The WD15 contains an ALC4020 USB audio controller and ALC3263 audio codec + * from Realtek. It is a UAC 1 device, and UAC 1 does not support jack + * detection. Instead, jack detection works by sending HD Audio commands over + * vendor-type USB messages. + */ + +#define HDA_VERB_CMD(V, N, D) (((N) << 20) | ((V) << 8) | (D)) + +#define REALTEK_HDA_VALUE 0x0038 + +#define REALTEK_HDA_SET 62 +#define REALTEK_HDA_GET_OUT 88 +#define REALTEK_HDA_GET_IN 89 + +#define REALTEK_LINE1 0x1a +#define REALTEK_VENDOR_REGISTERS 0x20 +#define REALTEK_HP_OUT 0x21 + +#define REALTEK_CBJ_CTRL2 0x50 + +#define REALTEK_JACK_INTERRUPT_NODE 5 + +#define REALTEK_MIC_FLAG 0x100 + +static int realtek_hda_set(struct snd_usb_audio *chip, u32 cmd) +{ + struct usb_device *dev = chip->dev; + u32 buf = cpu_to_be32(cmd); + + return snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), REALTEK_HDA_SET, + USB_RECIP_DEVICE | USB_TYPE_VENDOR | USB_DIR_OUT, + REALTEK_HDA_VALUE, 0, &buf, sizeof(buf)); +} + +static int realtek_hda_get(struct snd_usb_audio *chip, u32 cmd, u32 *value) +{ + struct usb_device *dev = chip->dev; + int err; + u32 buf = cpu_to_be32(cmd); + + err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), REALTEK_HDA_GET_OUT, + USB_RECIP_DEVICE | USB_TYPE_VENDOR | USB_DIR_OUT, + REALTEK_HDA_VALUE, 0, &buf, sizeof(buf)); + if (err < 0) + return err; + err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), REALTEK_HDA_GET_IN, + USB_RECIP_DEVICE | USB_TYPE_VENDOR | USB_DIR_IN, + REALTEK_HDA_VALUE, 0, &buf, sizeof(buf)); + if (err < 0) + return err; + + *value = be32_to_cpu(buf); + return 0; +} + +static int realtek_ctl_connector_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *cval = kcontrol->private_data; + struct snd_usb_audio *chip = cval->head.mixer->chip; + u32 pv = kcontrol->private_value; + u32 node_id = pv & 0xff; + u32 sense; + u32 cbj_ctrl2; + bool presence; + int err; + + err = snd_usb_lock_shutdown(chip); + if (err < 0) + return err; + err = realtek_hda_get(chip, + HDA_VERB_CMD(AC_VERB_GET_PIN_SENSE, node_id, 0), + &sense); + if (err < 0) + goto err; + if (pv & REALTEK_MIC_FLAG) { + err = realtek_hda_set(chip, + HDA_VERB_CMD(AC_VERB_SET_COEF_INDEX, + REALTEK_VENDOR_REGISTERS, + REALTEK_CBJ_CTRL2)); + if (err < 0) + goto err; + err = realtek_hda_get(chip, + HDA_VERB_CMD(AC_VERB_GET_PROC_COEF, + REALTEK_VENDOR_REGISTERS, 0), + &cbj_ctrl2); + if (err < 0) + goto err; + } +err: + snd_usb_unlock_shutdown(chip); + if (err < 0) + return err; + + presence = sense & AC_PINSENSE_PRESENCE; + if (pv & REALTEK_MIC_FLAG) + presence = presence && (cbj_ctrl2 & 0x0070) == 0x0070; + ucontrol->value.integer.value[0] = presence; + return 0; +} + +static const struct snd_kcontrol_new realtek_connector_ctl_ro = { + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .name = "", /* will be filled later manually */ + .access = SNDRV_CTL_ELEM_ACCESS_READ, + .info = snd_ctl_boolean_mono_info, + .get = realtek_ctl_connector_get, +}; + +static int realtek_resume_jack(struct usb_mixer_elem_list *list) +{ + snd_ctl_notify(list->mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE, + &list->kctl->id); + return 0; +} + +static int realtek_add_jack(struct usb_mixer_interface *mixer, + char *name, u32 val) +{ + struct usb_mixer_elem_info *cval; + struct snd_kcontrol *kctl; + + cval = kzalloc(sizeof(*cval), GFP_KERNEL); + if (!cval) + return -ENOMEM; + snd_usb_mixer_elem_init_std(&cval->head, mixer, + REALTEK_JACK_INTERRUPT_NODE); + cval->head.resume = realtek_resume_jack; + cval->val_type = USB_MIXER_BOOLEAN; + cval->channels = 1; + cval->min = 0; + cval->max = 1; + kctl = snd_ctl_new1(&realtek_connector_ctl_ro, cval); + if (!kctl) { + kfree(cval); + return -ENOMEM; + } + kctl->private_value = val; + strscpy(kctl->id.name, name, sizeof(kctl->id.name)); + kctl->private_free = snd_usb_mixer_elem_free; + return snd_usb_mixer_add_control(&cval->head, kctl); +} + +static int dell_dock_mixer_create(struct usb_mixer_interface *mixer) +{ + int err; + + err = realtek_add_jack(mixer, "Line Out Jack", REALTEK_LINE1); + if (err < 0) + return err; + err = realtek_add_jack(mixer, "Headphone Jack", REALTEK_HP_OUT); + if (err < 0) + return err; + err = realtek_add_jack(mixer, "Headset Mic Jack", + REALTEK_HP_OUT | REALTEK_MIC_FLAG); + if (err < 0) + return err; + return 0; +} + static void dell_dock_init_vol(struct snd_usb_audio *chip, int ch, int id) { u16 buf = 0; @@ -3245,6 +3409,9 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) err = snd_soundblaster_e1_switch_create(mixer); break; case USB_ID(0x0bda, 0x4014): /* Dell WD15 dock */ + err = dell_dock_mixer_create(mixer); + if (err < 0) + break; err = dell_dock_mixer_init(mixer); break; -- cgit v1.2.3 From 2e57a3358dda20128593fff9b39b522f1bdd26c6 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Jan=20Sch=C3=A4r?= Date: Mon, 27 Jun 2022 19:18:55 +0200 Subject: ALSA: usb-audio: Turn off 'manual mode' on Dell dock MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This removes the need to power cycle the Dell WD15 dock if it has been attached to a Windows machine. The Windows driver puts the ALC4020 USB audio controller into 'manual mode', and then does all the power management and other configuration itself, by sending HD audio commands directly to the ALC3263 audio codec via vendor-type USB messages. If manual mode is off, this is all handled by the firmware, and works well enough. If manual mode is turned on, the latency of the SET INTERFACE command goes from several hundred ms to less than 1 ms (see https://bugzilla.suse.com/show_bug.cgi?id=1089467), but I'm not sure if the additional code that would be required is worth it. Funnily enough, the Windows driver tries to turn off manual mode when the dock is disconnected, which doesn't work for obvious reasons. Additionally, fix a bug in dell_dock_init_vol, which didn't work because the Control Selector was missing. Now, it properly resets the volume to 0dB. Fixes: 964af639ad69 ("ALSA: usb-audio: Initialize Dell Dock playback volumes") Signed-off-by: Jan Schär Link: https://lore.kernel.org/r/20220627171855.42338-2-jan@jschaer.ch Signed-off-by: Takashi Iwai --- sound/usb/mixer_quirks.c | 20 +++++++++++++++++++- 1 file changed, 19 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 66b6476994eb..5a45822e60e7 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -1949,9 +1949,11 @@ static int snd_soundblaster_e1_switch_create(struct usb_mixer_interface *mixer) #define REALTEK_HDA_VALUE 0x0038 #define REALTEK_HDA_SET 62 +#define REALTEK_MANUAL_MODE 72 #define REALTEK_HDA_GET_OUT 88 #define REALTEK_HDA_GET_IN 89 +#define REALTEK_AUDIO_FUNCTION_GROUP 0x01 #define REALTEK_LINE1 0x1a #define REALTEK_VENDOR_REGISTERS 0x20 #define REALTEK_HP_OUT 0x21 @@ -2084,6 +2086,21 @@ static int realtek_add_jack(struct usb_mixer_interface *mixer, static int dell_dock_mixer_create(struct usb_mixer_interface *mixer) { int err; + struct usb_device *dev = mixer->chip->dev; + + /* Power down the audio codec to avoid loud pops in the next step. */ + realtek_hda_set(mixer->chip, + HDA_VERB_CMD(AC_VERB_SET_POWER_STATE, + REALTEK_AUDIO_FUNCTION_GROUP, + AC_PWRST_D3)); + + /* + * Turn off 'manual mode' in case it was enabled. This removes the need + * to power cycle the dock after it was attached to a Windows machine. + */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), REALTEK_MANUAL_MODE, + USB_RECIP_DEVICE | USB_TYPE_VENDOR | USB_DIR_OUT, + 0, 0, NULL, 0); err = realtek_add_jack(mixer, "Line Out Jack", REALTEK_LINE1); if (err < 0) @@ -2104,7 +2121,8 @@ static void dell_dock_init_vol(struct snd_usb_audio *chip, int ch, int id) snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), UAC_SET_CUR, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT, - ch, snd_usb_ctrl_intf(chip) | (id << 8), + (UAC_FU_VOLUME << 8) | ch, + snd_usb_ctrl_intf(chip) | (id << 8), &buf, 2); } -- cgit v1.2.3 From df98a94ce9c450a1af1193e06add37e601cbf2df Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Wed, 29 Jun 2022 11:27:43 +0100 Subject: ALSA: wavefront: remove redundant assignment to pointer end Pointer end is being re-assigned the same value as it was initialized with in the previous statement. The re-assignment is redundant and can be removed. Cleans up clang scan-build warning: sound/isa/wavefront/wavefront_synth.c:582:17: warning: Value stored to 'end' during its initialization is never read Signed-off-by: Colin Ian King Link: https://lore.kernel.org/r/20220629102744.139673-1-colin.i.king@gmail.com Signed-off-by: Takashi Iwai --- sound/isa/wavefront/wavefront_synth.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/isa/wavefront/wavefront_synth.c b/sound/isa/wavefront/wavefront_synth.c index 2aaaa6807174..13ce96148fa3 100644 --- a/sound/isa/wavefront/wavefront_synth.c +++ b/sound/isa/wavefront/wavefront_synth.c @@ -581,8 +581,6 @@ demunge_buf (unsigned char *src, unsigned char *dst, unsigned int src_bytes) int i; unsigned char *end = src + src_bytes; - end = src + src_bytes; - /* NOTE: src and dst *CAN* point to the same address */ for (i = 0; src != end; i++) { -- cgit v1.2.3 From c71531007ef0fe5cd64a8aa9b86bdb53ccef1504 Mon Sep 17 00:00:00 2001 From: "Steven Rostedt (Google)" Date: Sun, 3 Jul 2022 11:06:05 -0400 Subject: tracing: ALSA: hda: Remove string manipulation out of the fast path The TRACE_EVENT() macro is broken up into various parts to be efficient. The TP_fast_assign() is just to record the event into the ring buffer, and is to be done as fast as possible as this occurs during the actual running of the code. The slower this is, the slower the code that is being traced becomes. The TP_printk() is processed when reading the tracing buffer. This is considered the slow path. Any processing that can be moved from the TP_fast_assign() to the TP_printk() should do so. For some reason, the entire string processing of the trace events hda_send_cmd, hda_get_response, and hda_unsol_event was moved from the TP_printk() into the TP_fast_assign(). On top of that, the __dynamic_array() was used with a fixed size of HDAC_MSG_MAX, which is useless as a dynamic_array as it will always allocate HDAC_MSG_MAX bytes on the ring buffer and even save that amount into the event (as it expects the size to be dynamic, which using a fixed size defeats that purpose). Instead, just save the necessary elements in the TP_fast_assign() and do the string manipulation in the slow path. The output should be the same. Signed-off-by: Steven Rostedt (Google) Link: https://lore.kernel.org/r/20220703110605.07a86fb2@rorschach.local.home Signed-off-by: Takashi Iwai --- sound/hda/trace.h | 41 ++++++++++++++++++++++++++--------------- 1 file changed, 26 insertions(+), 15 deletions(-) (limited to 'sound') diff --git a/sound/hda/trace.h b/sound/hda/trace.h index 70af6c815089..2cc493434a8f 100644 --- a/sound/hda/trace.h +++ b/sound/hda/trace.h @@ -19,37 +19,48 @@ struct hdac_codec; TRACE_EVENT(hda_send_cmd, TP_PROTO(struct hdac_bus *bus, unsigned int cmd), TP_ARGS(bus, cmd), - TP_STRUCT__entry(__dynamic_array(char, msg, HDAC_MSG_MAX)), + TP_STRUCT__entry( + __string(name, dev_name((bus)->dev)) + __field(u32, cmd) + ), TP_fast_assign( - snprintf(__get_str(msg), HDAC_MSG_MAX, - "[%s:%d] val=0x%08x", - dev_name((bus)->dev), (cmd) >> 28, cmd); + __assign_str(name, dev_name((bus)->dev)); + __entry->cmd = cmd; ), - TP_printk("%s", __get_str(msg)) + TP_printk("[%s:%d] val=0x%08x", __get_str(name), __entry->cmd >> 28, __entry->cmd) ); TRACE_EVENT(hda_get_response, TP_PROTO(struct hdac_bus *bus, unsigned int addr, unsigned int res), TP_ARGS(bus, addr, res), - TP_STRUCT__entry(__dynamic_array(char, msg, HDAC_MSG_MAX)), + TP_STRUCT__entry( + __string(name, dev_name((bus)->dev)) + __field(u32, addr) + __field(u32, res) + ), TP_fast_assign( - snprintf(__get_str(msg), HDAC_MSG_MAX, - "[%s:%d] val=0x%08x", - dev_name((bus)->dev), addr, res); + __assign_str(name, dev_name((bus)->dev)); + __entry->addr = addr; + __entry->res = res; ), - TP_printk("%s", __get_str(msg)) + TP_printk("[%s:%d] val=0x%08x", __get_str(name), __entry->addr, __entry->res) ); TRACE_EVENT(hda_unsol_event, TP_PROTO(struct hdac_bus *bus, u32 res, u32 res_ex), TP_ARGS(bus, res, res_ex), - TP_STRUCT__entry(__dynamic_array(char, msg, HDAC_MSG_MAX)), + TP_STRUCT__entry( + __string(name, dev_name((bus)->dev)) + __field(u32, res) + __field(u32, res_ex) + ), TP_fast_assign( - snprintf(__get_str(msg), HDAC_MSG_MAX, - "[%s:%d] res=0x%08x, res_ex=0x%08x", - dev_name((bus)->dev), res_ex & 0x0f, res, res_ex); + __assign_str(name, dev_name((bus)->dev)); + __entry->res = res; + __entry->res_ex = res_ex; ), - TP_printk("%s", __get_str(msg)) + TP_printk("[%s:%d] res=0x%08x, res_ex=0x%08x", __get_str(name), + __entry->res_ex & 0x0f, __entry->res, __entry->res_ex) ); DECLARE_EVENT_CLASS(hdac_stream, -- cgit v1.2.3 From 65123b899818b1adf7388b3583624e0f1d8d6858 Mon Sep 17 00:00:00 2001 From: Tom Rix Date: Mon, 4 Jul 2022 10:28:36 -0400 Subject: ALSA: hda/cs8409: change cs8409_fixups v.pins initializers to static sparse reports sound/pci/hda/patch_cs8409-tables.c:79:25: warning: symbol 'cs8409_cs42l42_pincfgs_no_dmic' was not declared. Should it be static? cs8409_cs42l42_pincfgs_no_dmic is only used by cs8409_fixups table as an initializer for the hda_fixup element v.pins. Both are defined in the patch_cs8408-table.c file but only cs8409_fixups is used externally in patch_cs8409.c. So cs8409_cs42l42_pincfgs_no_dmic should have a static storage class specifier. The other v.pins initializers in cs8409_fixups table, though declared extern in patch_cs8409.h are also only used in patch_cs8409-tables.c. So change all the v.pins initializers to static. Fixes: 9e7647b5070f ("ALSA: hda/cs8409: Move arrays of configuration to a new file") Signed-off-by: Tom Rix Link: https://lore.kernel.org/r/20220704142836.636204-1-trix@redhat.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cs8409-tables.c | 6 +++--- sound/pci/hda/patch_cs8409.h | 2 -- 2 files changed, 3 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_cs8409-tables.c b/sound/pci/hda/patch_cs8409-tables.c index 4f4cc8215917..e0d3a8be2e38 100644 --- a/sound/pci/hda/patch_cs8409-tables.c +++ b/sound/pci/hda/patch_cs8409-tables.c @@ -68,7 +68,7 @@ const struct hda_verb cs8409_cs42l42_init_verbs[] = { {} /* terminator */ }; -const struct hda_pintbl cs8409_cs42l42_pincfgs[] = { +static const struct hda_pintbl cs8409_cs42l42_pincfgs[] = { { CS8409_PIN_ASP1_TRANSMITTER_A, 0x042120f0 }, /* ASP-1-TX */ { CS8409_PIN_ASP1_RECEIVER_A, 0x04a12050 }, /* ASP-1-RX */ { CS8409_PIN_ASP2_TRANSMITTER_A, 0x901000f0 }, /* ASP-2-TX */ @@ -76,7 +76,7 @@ const struct hda_pintbl cs8409_cs42l42_pincfgs[] = { {} /* terminator */ }; -const struct hda_pintbl cs8409_cs42l42_pincfgs_no_dmic[] = { +static const struct hda_pintbl cs8409_cs42l42_pincfgs_no_dmic[] = { { CS8409_PIN_ASP1_TRANSMITTER_A, 0x042120f0 }, /* ASP-1-TX */ { CS8409_PIN_ASP1_RECEIVER_A, 0x04a12050 }, /* ASP-1-RX */ { CS8409_PIN_ASP2_TRANSMITTER_A, 0x901000f0 }, /* ASP-2-TX */ @@ -279,7 +279,7 @@ const struct hda_verb dolphin_init_verbs[] = { {} /* terminator */ }; -const struct hda_pintbl dolphin_pincfgs[] = { +static const struct hda_pintbl dolphin_pincfgs[] = { { 0x24, 0x022210f0 }, /* ASP-1-TX-A */ { 0x25, 0x010240f0 }, /* ASP-1-TX-B */ { 0x34, 0x02a21050 }, /* ASP-1-RX */ diff --git a/sound/pci/hda/patch_cs8409.h b/sound/pci/hda/patch_cs8409.h index 260388a6256c..2a8dfb4ff046 100644 --- a/sound/pci/hda/patch_cs8409.h +++ b/sound/pci/hda/patch_cs8409.h @@ -358,13 +358,11 @@ extern const struct snd_pci_quirk cs8409_fixup_tbl[]; extern const struct hda_model_fixup cs8409_models[]; extern const struct hda_fixup cs8409_fixups[]; extern const struct hda_verb cs8409_cs42l42_init_verbs[]; -extern const struct hda_pintbl cs8409_cs42l42_pincfgs[]; extern const struct cs8409_cir_param cs8409_cs42l42_hw_cfg[]; extern const struct cs8409_cir_param cs8409_cs42l42_bullseye_atn[]; extern struct sub_codec cs8409_cs42l42_codec; extern const struct hda_verb dolphin_init_verbs[]; -extern const struct hda_pintbl dolphin_pincfgs[]; extern const struct cs8409_cir_param dolphin_hw_cfg[]; extern struct sub_codec dolphin_cs42l42_0; extern struct sub_codec dolphin_cs42l42_1; -- cgit v1.2.3 From 7bad8125549cda14d9ccf97d7d76f7ef6ac9d206 Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Fri, 1 Jul 2022 17:32:36 +0800 Subject: ASoC: fsl_utils: Add function to handle PLL clock source i.MX8MQ/MN/MM/MP platforms typically have 2 AUDIO PLLs being configured to handle 8kHz and 11kHz series audio rates. Add common function in fsl_utils to handle these two PLL clock source, which are needed by CPU DAI drivers Signed-off-by: Shengjiu Wang Link: https://lore.kernel.org/r/1656667961-1799-2-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_utils.c | 69 +++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/fsl/fsl_utils.h | 7 +++++ 2 files changed, 76 insertions(+) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_utils.c b/sound/soc/fsl/fsl_utils.c index 9bab202569af..b75843e31f00 100644 --- a/sound/soc/fsl/fsl_utils.c +++ b/sound/soc/fsl/fsl_utils.c @@ -6,6 +6,8 @@ // // Copyright 2010 Freescale Semiconductor, Inc. +#include +#include #include #include #include @@ -83,6 +85,73 @@ int fsl_asoc_get_dma_channel(struct device_node *ssi_np, } EXPORT_SYMBOL(fsl_asoc_get_dma_channel); +/** + * fsl_asoc_get_pll_clocks - get two PLL clock source + * + * @dev: device pointer + * @pll8k_clk: PLL clock pointer for 8kHz + * @pll11k_clk: PLL clock pointer for 11kHz + * + * This function get two PLL clock source + */ +void fsl_asoc_get_pll_clocks(struct device *dev, struct clk **pll8k_clk, + struct clk **pll11k_clk) +{ + *pll8k_clk = devm_clk_get(dev, "pll8k"); + if (IS_ERR(*pll8k_clk)) + *pll8k_clk = NULL; + + *pll11k_clk = devm_clk_get(dev, "pll11k"); + if (IS_ERR(*pll11k_clk)) + *pll11k_clk = NULL; +} +EXPORT_SYMBOL(fsl_asoc_get_pll_clocks); + +/** + * fsl_asoc_reparent_pll_clocks - set clock parent if necessary + * + * @dev: device pointer + * @clk: root clock pointer + * @pll8k_clk: PLL clock pointer for 8kHz + * @pll11k_clk: PLL clock pointer for 11kHz + * @ratio: target requency for root clock + * + * This function set root clock parent according to the target ratio + */ +void fsl_asoc_reparent_pll_clocks(struct device *dev, struct clk *clk, + struct clk *pll8k_clk, + struct clk *pll11k_clk, u64 ratio) +{ + struct clk *p, *pll = 0, *npll = 0; + bool reparent = false; + int ret = 0; + + if (!clk || !pll8k_clk || !pll11k_clk) + return; + + p = clk; + while (p && pll8k_clk && pll11k_clk) { + struct clk *pp = clk_get_parent(p); + + if (clk_is_match(pp, pll8k_clk) || + clk_is_match(pp, pll11k_clk)) { + pll = pp; + break; + } + p = pp; + } + + npll = (do_div(ratio, 8000) ? pll11k_clk : pll8k_clk); + reparent = (pll && !clk_is_match(pll, npll)); + + if (reparent) { + ret = clk_set_parent(p, npll); + if (ret < 0) + dev_warn(dev, "failed to set parent %s: %d\n", __clk_get_name(npll), ret); + } +} +EXPORT_SYMBOL(fsl_asoc_reparent_pll_clocks); + MODULE_AUTHOR("Timur Tabi "); MODULE_DESCRIPTION("Freescale ASoC utility code"); MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/fsl/fsl_utils.h b/sound/soc/fsl/fsl_utils.h index c5dc2a14b492..4d5f3d93bc81 100644 --- a/sound/soc/fsl/fsl_utils.h +++ b/sound/soc/fsl/fsl_utils.h @@ -19,4 +19,11 @@ int fsl_asoc_get_dma_channel(struct device_node *ssi_np, const char *name, struct snd_soc_dai_link *dai, unsigned int *dma_channel_id, unsigned int *dma_id); + +void fsl_asoc_get_pll_clocks(struct device *dev, struct clk **pll8k_clk, + struct clk **pll11k_clk); + +void fsl_asoc_reparent_pll_clocks(struct device *dev, struct clk *clk, + struct clk *pll8k_clk, + struct clk *pll11k_clk, u64 ratio); #endif /* _FSL_UTILS_H */ -- cgit v1.2.3 From 34dcdebecf2f05e1b275e1da8352f8e4c1aab6f6 Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Fri, 1 Jul 2022 17:32:37 +0800 Subject: ASoC: fsl_spdif: Add support for PLL switch at runtime. i.MX8MQ/MN/MM/MP platforms typically have 2 AUDIO PLLs being configured to handle 8kHz and 11kHz series audio rates. The patch implements the functionality to select at runtime the appropriate AUDIO PLL as function of audio file rate. As the clock parent may be changed, need to probe txclk according to sample rate again. Signed-off-by: Viorel Suman Signed-off-by: Shengjiu Wang Link: https://lore.kernel.org/r/1656667961-1799-3-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/fsl_spdif.c | 48 ++++++++++++++++++++++++++++++++++++++++++----- 2 files changed, 44 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 10fa38753453..910b8747b6bd 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -59,6 +59,7 @@ config SND_SOC_FSL_SPDIF select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n select SND_SOC_IMX_PCM_FIQ if SND_IMX_SOC != n && (MXC_TZIC || MXC_AVIC) select BITREVERSE + select SND_SOC_FSL_UTILS help Say Y if you want to add Sony/Philips Digital Interface (SPDIF) support for the Freescale CPUs. diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 0504431792cf..7fc1c96929bb 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -23,6 +23,7 @@ #include #include "fsl_spdif.h" +#include "fsl_utils.h" #include "imx-pcm.h" #define FSL_SPDIF_TXFIFO_WML 0x8 @@ -114,6 +115,8 @@ struct spdif_mixer_control { * @dma_params_rx: DMA parameters for receive channel * @regcache_srpc: regcache for SRPC * @bypass: status of bypass input to output + * @pll8k_clk: PLL clock for the rate of multiply of 8kHz + * @pll11k_clk: PLL clock for the rate of multiply of 11kHz */ struct fsl_spdif_priv { const struct fsl_spdif_soc_data *soc; @@ -137,6 +140,8 @@ struct fsl_spdif_priv { /* regcache for SRPC */ u32 regcache_srpc; bool bypass; + struct clk *pll8k_clk; + struct clk *pll11k_clk; }; static struct fsl_spdif_soc_data fsl_spdif_vf610 = { @@ -480,6 +485,8 @@ static int spdif_set_rx_clksrc(struct fsl_spdif_priv *spdif_priv, return 0; } +static int fsl_spdif_probe_txclk(struct fsl_spdif_priv *spdif_priv, enum spdif_txrate index); + static int spdif_set_sample_rate(struct snd_pcm_substream *substream, int sample_rate) { @@ -528,6 +535,10 @@ static int spdif_set_sample_rate(struct snd_pcm_substream *substream, return -EINVAL; } + ret = fsl_spdif_probe_txclk(spdif_priv, rate); + if (ret) + return ret; + clk = spdif_priv->txclk_src[rate]; if (clk >= STC_TXCLK_SRC_MAX) { dev_err(&pdev->dev, "tx clock source is out of range\n"); @@ -647,6 +658,29 @@ static void fsl_spdif_shutdown(struct snd_pcm_substream *substream, } } +static int spdif_reparent_rootclk(struct fsl_spdif_priv *spdif_priv, unsigned int sample_rate) +{ + struct platform_device *pdev = spdif_priv->pdev; + struct clk *clk; + int ret; + + /* Reparent clock if required condition is true */ + if (!fsl_spdif_can_set_clk_rate(spdif_priv, STC_TXCLK_SPDIF_ROOT)) + return 0; + + /* Get root clock */ + clk = spdif_priv->txclk[STC_TXCLK_SPDIF_ROOT]; + + /* Disable clock first, for it was enabled by pm_runtime */ + clk_disable_unprepare(clk); + fsl_asoc_reparent_pll_clocks(&pdev->dev, clk, spdif_priv->pll8k_clk, + spdif_priv->pll11k_clk, sample_rate); + ret = clk_prepare_enable(clk); + if (ret) + return ret; + + return 0; +} static int fsl_spdif_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -659,6 +693,13 @@ static int fsl_spdif_hw_params(struct snd_pcm_substream *substream, int ret = 0; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + ret = spdif_reparent_rootclk(spdif_priv, sample_rate); + if (ret) { + dev_err(&pdev->dev, "%s: reparent root clk failed: %d\n", + __func__, sample_rate); + return ret; + } + ret = spdif_set_sample_rate(substream, sample_rate); if (ret) { dev_err(&pdev->dev, "%s: set sample rate failed: %d\n", @@ -1548,11 +1589,8 @@ static int fsl_spdif_probe(struct platform_device *pdev) } spdif_priv->rxclk_src = DEFAULT_RXCLK_SRC; - for (i = 0; i < SPDIF_TXRATE_MAX; i++) { - ret = fsl_spdif_probe_txclk(spdif_priv, i); - if (ret) - return ret; - } + fsl_asoc_get_pll_clocks(&pdev->dev, &spdif_priv->pll8k_clk, + &spdif_priv->pll11k_clk); /* Initial spinlock for control data */ ctrl = &spdif_priv->fsl_spdif_control; -- cgit v1.2.3 From 93f54100fbdedc22e8d88d037a8a3e32101724eb Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Fri, 1 Jul 2022 17:32:38 +0800 Subject: ASoC: fsl_micfil: Add support for PLL switch at runtime i.MX8MQ/MN/MM/MP platforms typically have 2 AUDIO PLLs being configured to handle 8kHz and 11kHz series audio rates. The patch implements the functionality to select at runtime the appropriate AUDIO PLL as function of audio file rate. Signed-off-by: Viorel Suman Signed-off-by: Shengjiu Wang Link: https://lore.kernel.org/r/1656667961-1799-4-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/fsl_micfil.c | 31 +++++++++++++++++++++++++++++++ 2 files changed, 32 insertions(+) (limited to 'sound') diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 910b8747b6bd..533937166b4a 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -81,6 +81,7 @@ config SND_SOC_FSL_MICFIL select REGMAP_MMIO select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n select SND_SOC_GENERIC_DMAENGINE_PCM + select SND_SOC_FSL_UTILS help Say Y if you want to add Pulse Density Modulation microphone interface (MICFIL) support for NXP. diff --git a/sound/soc/fsl/fsl_micfil.c b/sound/soc/fsl/fsl_micfil.c index be9781ad8849..79ef4e269bc9 100644 --- a/sound/soc/fsl/fsl_micfil.c +++ b/sound/soc/fsl/fsl_micfil.c @@ -24,6 +24,7 @@ #include #include "fsl_micfil.h" +#include "fsl_utils.h" #define MICFIL_OSR_DEFAULT 16 @@ -42,6 +43,8 @@ struct fsl_micfil { const struct fsl_micfil_soc_data *soc; struct clk *busclk; struct clk *mclk; + struct clk *pll8k_clk; + struct clk *pll11k_clk; struct snd_dmaengine_dai_dma_data dma_params_rx; struct sdma_peripheral_config sdmacfg; unsigned int dataline; @@ -264,6 +267,27 @@ static int fsl_micfil_trigger(struct snd_pcm_substream *substream, int cmd, return 0; } +static int fsl_micfil_reparent_rootclk(struct fsl_micfil *micfil, unsigned int sample_rate) +{ + struct device *dev = &micfil->pdev->dev; + u64 ratio = sample_rate; + struct clk *clk; + int ret; + + /* Get root clock */ + clk = micfil->mclk; + + /* Disable clock first, for it was enabled by pm_runtime */ + clk_disable_unprepare(clk); + fsl_asoc_reparent_pll_clocks(dev, clk, micfil->pll8k_clk, + micfil->pll11k_clk, ratio); + ret = clk_prepare_enable(clk); + if (ret) + return ret; + + return 0; +} + static int fsl_micfil_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -287,6 +311,10 @@ static int fsl_micfil_hw_params(struct snd_pcm_substream *substream, if (ret) return ret; + ret = fsl_micfil_reparent_rootclk(micfil, rate); + if (ret) + return ret; + ret = clk_set_rate(micfil->mclk, rate * clk_div * osr * 8); if (ret) return ret; @@ -591,6 +619,9 @@ static int fsl_micfil_probe(struct platform_device *pdev) return PTR_ERR(micfil->busclk); } + fsl_asoc_get_pll_clocks(&pdev->dev, &micfil->pll8k_clk, + &micfil->pll11k_clk); + /* init regmap */ regs = devm_platform_get_and_ioremap_resource(pdev, 0, &res); if (IS_ERR(regs)) -- cgit v1.2.3 From 7cb7f07d2491a3435578ab97eeeb70fadac6385c Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Fri, 1 Jul 2022 17:32:39 +0800 Subject: ASoC: fsl_sai: Add support for PLL switch at runtime i.MX8MQ/MN/MM/MP platforms typically have 2 AUDIO PLLs being configured to handle 8kHz and 11kHz series audio rates. The patch implements the functionality to select at runtime the appropriate AUDIO PLL as function of sysclk rate. Signed-off-by: Viorel Suman Signed-off-by: Shengjiu Wang Link: https://lore.kernel.org/r/1656667961-1799-5-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/fsl_sai.c | 38 ++++++++++++++++++++++++++++++++++++++ sound/soc/fsl/fsl_sai.h | 2 ++ 3 files changed, 41 insertions(+) (limited to 'sound') diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 533937166b4a..614eceda6b9e 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -19,6 +19,7 @@ config SND_SOC_FSL_SAI select REGMAP_MMIO select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n select SND_SOC_GENERIC_DMAENGINE_PCM + select SND_SOC_FSL_UTILS help Say Y if you want to add Synchronous Audio Interface (SAI) support for the Freescale CPUs. diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index a0ddaf7e9f60..974ba0780b19 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -23,6 +23,7 @@ #include #include "fsl_sai.h" +#include "fsl_utils.h" #include "imx-pcm.h" #define FSL_SAI_FLAGS (FSL_SAI_CSR_SEIE |\ @@ -220,14 +221,48 @@ static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai, return 0; } +static int fsl_sai_set_mclk_rate(struct snd_soc_dai *dai, int clk_id, unsigned int freq) +{ + struct fsl_sai *sai = snd_soc_dai_get_drvdata(dai); + int ret; + + fsl_asoc_reparent_pll_clocks(dai->dev, sai->mclk_clk[clk_id], + sai->pll8k_clk, sai->pll11k_clk, freq); + + ret = clk_set_rate(sai->mclk_clk[clk_id], freq); + if (ret < 0) + dev_err(dai->dev, "failed to set clock rate (%u): %d\n", freq, ret); + + return ret; +} + static int fsl_sai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { + struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); int ret; if (dir == SND_SOC_CLOCK_IN) return 0; + if (freq > 0 && clk_id != FSL_SAI_CLK_BUS) { + if (clk_id < 0 || clk_id >= FSL_SAI_MCLK_MAX) { + dev_err(cpu_dai->dev, "Unknown clock id: %d\n", clk_id); + return -EINVAL; + } + + if (IS_ERR_OR_NULL(sai->mclk_clk[clk_id])) { + dev_err(cpu_dai->dev, "Unassigned clock: %d\n", clk_id); + return -EINVAL; + } + + if (sai->mclk_streams == 0) { + ret = fsl_sai_set_mclk_rate(cpu_dai, clk_id, freq); + if (ret < 0) + return ret; + } + } + ret = fsl_sai_set_dai_sysclk_tr(cpu_dai, clk_id, freq, true); if (ret) { dev_err(cpu_dai->dev, "Cannot set tx sysclk: %d\n", ret); @@ -1281,6 +1316,9 @@ static int fsl_sai_probe(struct platform_device *pdev) else sai->mclk_clk[0] = sai->bus_clk; + fsl_asoc_get_pll_clocks(&pdev->dev, &sai->pll8k_clk, + &sai->pll11k_clk); + /* read dataline mask for rx and tx*/ ret = fsl_sai_read_dlcfg(sai); if (ret < 0) { diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 9bb8ced520c8..17956b5731dc 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -273,6 +273,8 @@ struct fsl_sai { struct regmap *regmap; struct clk *bus_clk; struct clk *mclk_clk[FSL_SAI_MCLK_MAX]; + struct clk *pll8k_clk; + struct clk *pll11k_clk; struct resource *res; bool is_consumer_mode; -- cgit v1.2.3 From 3eb8440d0d268437202ccbd28a3ca3212e02e57f Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Fri, 1 Jul 2022 17:11:05 +0530 Subject: ASoC: amd: add I2S MICSP instance support Add I2S MICSP instance support for Stoney variant. Signed-off-by: Vijendar Mukunda Link: https://lore.kernel.org/r/20220701114107.1105948-4-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp-pcm-dma.c | 50 +++++++++++++++++++++++++++++++++++++++++++-- sound/soc/amd/acp.h | 13 ++++++++++++ 2 files changed, 61 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 1cd2e70a57df..198358d28ea9 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -433,6 +433,7 @@ static void acp_dma_start(void __iomem *acp_mmio, u16 ch_num, bool is_circular) case I2S_TO_ACP_DMA_CH_NUM: case ACP_TO_I2S_DMA_BT_INSTANCE_CH_NUM: case I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM: + case ACP_TO_I2S_DMA_MICSP_INSTANCE_CH_NUM: dma_ctrl |= ACP_DMA_CNTL_0__DMAChIOCEn_MASK; break; default: @@ -710,6 +711,13 @@ static irqreturn_t dma_irq_handler(int irq, void *arg) acp_mmio, mmACP_EXTERNAL_INTR_STAT); } + if ((intr_flag & BIT(ACP_TO_I2S_DMA_MICSP_INSTANCE_CH_NUM)) != 0) { + valid_irq = true; + snd_pcm_period_elapsed(irq_data->play_i2s_micsp_stream); + acp_reg_write((intr_flag & BIT(ACP_TO_I2S_DMA_MICSP_INSTANCE_CH_NUM)) << 16, + acp_mmio, mmACP_EXTERNAL_INTR_STAT); + } + if ((intr_flag & BIT(ACP_TO_I2S_DMA_BT_INSTANCE_CH_NUM)) != 0) { valid_irq = true; snd_pcm_period_elapsed(irq_data->play_i2sbt_stream); @@ -807,7 +815,8 @@ static int acp_dma_open(struct snd_soc_component *component, * stream is not closed */ if (!intr_data->play_i2ssp_stream && !intr_data->capture_i2ssp_stream && - !intr_data->play_i2sbt_stream && !intr_data->capture_i2sbt_stream) + !intr_data->play_i2sbt_stream && !intr_data->capture_i2sbt_stream && + !intr_data->play_i2s_micsp_stream) acp_reg_write(1, adata->acp_mmio, mmACP_EXTERNAL_INTR_ENB); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { @@ -867,6 +876,9 @@ static int acp_dma_hw_params(struct snd_soc_component *component, case I2S_BT_INSTANCE: val |= ACP_I2S_BT_16BIT_RESOLUTION_EN; break; + case I2S_MICSP_INSTANCE: + val |= ACP_I2S_MICSP_16BIT_RESOLUTION_EN; + break; case I2S_SP_INSTANCE: default: val |= ACP_I2S_SP_16BIT_RESOLUTION_EN; @@ -876,6 +888,7 @@ static int acp_dma_hw_params(struct snd_soc_component *component, case I2S_BT_INSTANCE: val |= ACP_I2S_BT_16BIT_RESOLUTION_EN; break; + case I2S_MICSP_INSTANCE: case I2S_SP_INSTANCE: default: val |= ACP_I2S_MIC_16BIT_RESOLUTION_EN; @@ -901,6 +914,27 @@ static int acp_dma_hw_params(struct snd_soc_component *component, mmACP_I2S_BT_TRANSMIT_BYTE_CNT_LOW; adata->play_i2sbt_stream = substream; break; + case I2S_MICSP_INSTANCE: + switch (adata->asic_type) { + case CHIP_STONEY: + rtd->pte_offset = ACP_ST_PLAYBACK_PTE_OFFSET; + break; + default: + rtd->pte_offset = ACP_PLAYBACK_PTE_OFFSET; + } + rtd->ch1 = SYSRAM_TO_ACP_MICSP_INSTANCE_CH_NUM; + rtd->ch2 = ACP_TO_I2S_DMA_MICSP_INSTANCE_CH_NUM; + rtd->sram_bank = ACP_SRAM_BANK_1_ADDRESS; + rtd->destination = TO_ACP_I2S_2; + rtd->dma_dscr_idx_1 = PLAYBACK_START_DMA_DESCR_CH4; + rtd->dma_dscr_idx_2 = PLAYBACK_START_DMA_DESCR_CH5; + rtd->byte_cnt_high_reg_offset = + mmACP_I2S_MICSP_TRANSMIT_BYTE_CNT_HIGH; + rtd->byte_cnt_low_reg_offset = + mmACP_I2S_MICSP_TRANSMIT_BYTE_CNT_LOW; + + adata->play_i2s_micsp_stream = substream; + break; case I2S_SP_INSTANCE: default: switch (adata->asic_type) { @@ -939,6 +973,7 @@ static int acp_dma_hw_params(struct snd_soc_component *component, rtd->dma_curr_dscr = mmACP_DMA_CUR_DSCR_11; adata->capture_i2sbt_stream = substream; break; + case I2S_MICSP_INSTANCE: case I2S_SP_INSTANCE: default: rtd->pte_offset = ACP_CAPTURE_PTE_OFFSET; @@ -1160,6 +1195,9 @@ static int acp_dma_close(struct snd_soc_component *component, case I2S_BT_INSTANCE: adata->play_i2sbt_stream = NULL; break; + case I2S_MICSP_INSTANCE: + adata->play_i2s_micsp_stream = NULL; + break; case I2S_SP_INSTANCE: default: adata->play_i2ssp_stream = NULL; @@ -1181,6 +1219,7 @@ static int acp_dma_close(struct snd_soc_component *component, case I2S_BT_INSTANCE: adata->capture_i2sbt_stream = NULL; break; + case I2S_MICSP_INSTANCE: case I2S_SP_INSTANCE: default: adata->capture_i2ssp_stream = NULL; @@ -1197,7 +1236,8 @@ static int acp_dma_close(struct snd_soc_component *component, * another stream is also not active. */ if (!adata->play_i2ssp_stream && !adata->capture_i2ssp_stream && - !adata->play_i2sbt_stream && !adata->capture_i2sbt_stream) + !adata->play_i2sbt_stream && !adata->capture_i2sbt_stream && + !adata->play_i2s_micsp_stream) acp_reg_write(0, adata->acp_mmio, mmACP_EXTERNAL_INTR_ENB); kfree(rtd); return 0; @@ -1245,6 +1285,7 @@ static int acp_audio_probe(struct platform_device *pdev) audio_drv_data->capture_i2ssp_stream = NULL; audio_drv_data->play_i2sbt_stream = NULL; audio_drv_data->capture_i2sbt_stream = NULL; + audio_drv_data->play_i2s_micsp_stream = NULL; audio_drv_data->asic_type = *pdata; @@ -1333,6 +1374,11 @@ static int acp_pcm_resume(struct device *dev) config_acp_dma(adata->acp_mmio, rtd, adata->asic_type); } if (adata->asic_type != CHIP_CARRIZO) { + if (adata->play_i2s_micsp_stream && + adata->play_i2s_micsp_stream->runtime) { + rtd = adata->play_i2s_micsp_stream->runtime->private_data; + config_acp_dma(adata->acp_mmio, rtd, adata->asic_type); + } if (adata->play_i2sbt_stream && adata->play_i2sbt_stream->runtime) { rtd = adata->play_i2sbt_stream->runtime->private_data; diff --git a/sound/soc/amd/acp.h b/sound/soc/amd/acp.h index db80a73aa593..b29bef90f886 100644 --- a/sound/soc/amd/acp.h +++ b/sound/soc/amd/acp.h @@ -55,6 +55,7 @@ #define I2S_SP_INSTANCE 0x01 #define I2S_BT_INSTANCE 0x02 +#define I2S_MICSP_INSTANCE 0x03 #define CAP_CHANNEL0 0x00 #define CAP_CHANNEL1 0x01 @@ -85,6 +86,10 @@ #define I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM 10 #define ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM 11 +/* Playback DMA channels for I2S MICSP instance */ +#define SYSRAM_TO_ACP_MICSP_INSTANCE_CH_NUM 4 +#define ACP_TO_I2S_DMA_MICSP_INSTANCE_CH_NUM 5 + #define NUM_DSCRS_PER_CHANNEL 2 #define PLAYBACK_START_DMA_DESCR_CH12 0 @@ -108,8 +113,15 @@ #define CAPTURE_START_DMA_DESCR_CH11 14 #define CAPTURE_END_DMA_DESCR_CH11 15 +/* I2S MICSP Instance DMA Descriptors */ +#define PLAYBACK_START_DMA_DESCR_CH4 0 +#define PLAYBACK_END_DMA_DESCR_CH4 1 +#define PLAYBACK_START_DMA_DESCR_CH5 2 +#define PLAYBACK_END_DMA_DESCR_CH5 3 + #define mmACP_I2S_16BIT_RESOLUTION_EN 0x5209 #define ACP_I2S_MIC_16BIT_RESOLUTION_EN 0x01 +#define ACP_I2S_MICSP_16BIT_RESOLUTION_EN 0x01 #define ACP_I2S_SP_16BIT_RESOLUTION_EN 0x02 #define ACP_I2S_BT_16BIT_RESOLUTION_EN 0x04 #define ACP_BT_UART_PAD_SELECT_MASK 0x1 @@ -149,6 +161,7 @@ struct audio_drv_data { struct snd_pcm_substream *capture_i2ssp_stream; struct snd_pcm_substream *play_i2sbt_stream; struct snd_pcm_substream *capture_i2sbt_stream; + struct snd_pcm_substream *play_i2s_micsp_stream; void __iomem *acp_mmio; u32 asic_type; snd_pcm_sframes_t delay; -- cgit v1.2.3 From 02527c3f2300100a25524c8c020d98c7957e485e Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Fri, 1 Jul 2022 17:11:06 +0530 Subject: ASoC: amd: add Machine driver for Jadeite platform Add Machine driver for Jadeite platform which uses ES8336 codec. Signed-off-by: Vijendar Mukunda Link: https://lore.kernel.org/r/20220701114107.1105948-5-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp-es8336.c | 324 +++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 324 insertions(+) create mode 100644 sound/soc/amd/acp-es8336.c (limited to 'sound') diff --git a/sound/soc/amd/acp-es8336.c b/sound/soc/amd/acp-es8336.c new file mode 100644 index 000000000000..eec3d57092fa --- /dev/null +++ b/sound/soc/amd/acp-es8336.c @@ -0,0 +1,324 @@ +// SPDX-License-Identifier: GPL-2.0+ +/* + * Machine driver for AMD Stoney platform using ES8336 Codec + * + * Copyright 2022 Advanced Micro Devices, Inc. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "../codecs/es8316.h" +#include "acp.h" + +#define DUAL_CHANNEL 2 +#define DRV_NAME "acp2x_mach" +#define ST_JADEITE 1 +#define ES8336_PLL_FREQ (48000 * 256) + +static unsigned long acp2x_machine_id; +static struct snd_soc_jack st_jack; +struct device *codec_dev; +struct gpio_desc *gpio_pa; + +static int sof_es8316_speaker_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + if (SND_SOC_DAPM_EVENT_ON(event)) + gpiod_set_value_cansleep(gpio_pa, true); + else + gpiod_set_value_cansleep(gpio_pa, false); + + return 0; +} + +static struct snd_soc_jack_pin st_es8316_jack_pins[] = { + { + .pin = "Headphone", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, +}; + +static int st_es8336_init(struct snd_soc_pcm_runtime *rtd) +{ + int ret; + struct snd_soc_card *card; + struct snd_soc_component *codec; + + codec = asoc_rtd_to_codec(rtd, 0)->component; + card = rtd->card; + + ret = snd_soc_card_jack_new_pins(card, "Headset", SND_JACK_HEADSET | SND_JACK_BTN_0, + &st_jack, st_es8316_jack_pins, + ARRAY_SIZE(st_es8316_jack_pins)); + if (ret) { + dev_err(card->dev, "HP jack creation failed %d\n", ret); + return ret; + } + snd_jack_set_key(st_jack.jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + ret = snd_soc_component_set_jack(codec, &st_jack, NULL); + if (ret) { + dev_err(rtd->dev, "Headset Jack call-back failed: %d\n", ret); + return ret; + } + return 0; +} + +static const unsigned int st_channels[] = { + DUAL_CHANNEL, +}; + +static const unsigned int st_rates[] = { + 48000, +}; + +static const struct snd_pcm_hw_constraint_list st_constraints_rates = { + .count = ARRAY_SIZE(st_rates), + .list = st_rates, + .mask = 0, +}; + +static const struct snd_pcm_hw_constraint_list st_constraints_channels = { + .count = ARRAY_SIZE(st_channels), + .list = st_channels, + .mask = 0, +}; + +static int st_es8336_codec_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime; + struct snd_soc_pcm_runtime *rtd; + struct snd_soc_card *card; + struct acp_platform_info *machine; + struct snd_soc_dai *codec_dai; + int ret; + + runtime = substream->runtime; + rtd = asoc_substream_to_rtd(substream); + card = rtd->card; + machine = snd_soc_card_get_drvdata(card); + codec_dai = asoc_rtd_to_codec(rtd, 0); + ret = snd_soc_dai_set_sysclk(codec_dai, 0, ES8336_PLL_FREQ, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec sysclk: %d\n", ret); + return ret; + } + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &st_constraints_channels); + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &st_constraints_rates); + + machine->play_i2s_instance = I2S_MICSP_INSTANCE; + machine->cap_i2s_instance = I2S_MICSP_INSTANCE; + machine->capture_channel = CAP_CHANNEL0; + return 0; +} + +static const struct snd_soc_ops st_es8336_ops = { + .startup = st_es8336_codec_startup, +}; + +SND_SOC_DAILINK_DEF(designware1, + DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.2.auto"))); +SND_SOC_DAILINK_DEF(codec, + DAILINK_COMP_ARRAY(COMP_CODEC("i2c-ESSX8336:00", "ES8316 HiFi"))); +SND_SOC_DAILINK_DEF(platform, + DAILINK_COMP_ARRAY(COMP_PLATFORM("acp_audio_dma.1.auto"))); + +static struct snd_soc_dai_link st_dai_es8336[] = { + { + .name = "amdes8336", + .stream_name = "ES8336 HiFi Play", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBP_CFP, + .stop_dma_first = 1, + .dpcm_capture = 1, + .dpcm_playback = 1, + .init = st_es8336_init, + .ops = &st_es8336_ops, + SND_SOC_DAILINK_REG(designware1, codec, platform), + }, +}; + +static const struct snd_soc_dapm_widget st_widgets[] = { + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Internal Mic", NULL), + + SND_SOC_DAPM_SUPPLY("Speaker Power", SND_SOC_NOPM, 0, 0, + sof_es8316_speaker_power_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +}; + +static const struct snd_soc_dapm_route st_audio_route[] = { + {"Speaker", NULL, "HPOL"}, + {"Speaker", NULL, "HPOR"}, + {"Headphone", NULL, "HPOL"}, + {"Headphone", NULL, "HPOR"}, + {"MIC1", NULL, "Headset Mic"}, + {"MIC2", NULL, "Internal Mic"}, + {"Speaker", NULL, "Speaker Power"}, +}; + +static const struct snd_kcontrol_new st_mc_controls[] = { + SOC_DAPM_PIN_SWITCH("Speaker"), + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Internal Mic"), +}; + +static const struct acpi_gpio_params pa_enable_gpio = { 0, 0, false }; +static const struct acpi_gpio_mapping acpi_es8336_gpios[] = { + { "pa-enable-gpios", &pa_enable_gpio, 1 }, + { } +}; + +static int st_es8336_late_probe(struct snd_soc_card *card) +{ + struct acpi_device *adev; + int ret; + + adev = acpi_dev_get_first_match_dev("ESSX8336", NULL, -1); + if (adev) + put_device(&adev->dev); + codec_dev = acpi_get_first_physical_node(adev); + if (!codec_dev) + dev_err(card->dev, "can not find codec dev\n"); + + ret = devm_acpi_dev_add_driver_gpios(codec_dev, acpi_es8336_gpios); + + gpio_pa = gpiod_get_optional(codec_dev, "pa-enable", GPIOD_OUT_LOW); + if (IS_ERR(gpio_pa)) { + ret = dev_err_probe(card->dev, PTR_ERR(gpio_pa), + "could not get pa-enable GPIO\n"); + gpiod_put(gpio_pa); + put_device(codec_dev); + } + return 0; +} + +static struct snd_soc_card st_card = { + .name = "acpes8336", + .owner = THIS_MODULE, + .dai_link = st_dai_es8336, + .num_links = ARRAY_SIZE(st_dai_es8336), + .dapm_widgets = st_widgets, + .num_dapm_widgets = ARRAY_SIZE(st_widgets), + .dapm_routes = st_audio_route, + .num_dapm_routes = ARRAY_SIZE(st_audio_route), + .controls = st_mc_controls, + .num_controls = ARRAY_SIZE(st_mc_controls), + .late_probe = st_es8336_late_probe, +}; + +static int st_es8336_quirk_cb(const struct dmi_system_id *id) +{ + acp2x_machine_id = ST_JADEITE; + return 1; +} + +static const struct dmi_system_id st_es8336_quirk_table[] = { + { + .callback = st_es8336_quirk_cb, + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "AMD"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "Jadeite"), + }, + }, + { + .callback = st_es8336_quirk_cb, + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "IP3 Technology CO.,Ltd."), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "ASN1D"), + }, + }, + { + .callback = st_es8336_quirk_cb, + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "Standard"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "ASN10"), + }, + }, + {} +}; + +static int st_es8336_probe(struct platform_device *pdev) +{ + int ret; + struct snd_soc_card *card; + struct acp_platform_info *machine; + + machine = devm_kzalloc(&pdev->dev, sizeof(struct acp_platform_info), GFP_KERNEL); + if (!machine) + return -ENOMEM; + + dmi_check_system(st_es8336_quirk_table); + switch (acp2x_machine_id) { + case ST_JADEITE: + card = &st_card; + st_card.dev = &pdev->dev; + break; + default: + return -ENODEV; + } + + platform_set_drvdata(pdev, card); + snd_soc_card_set_drvdata(card, machine); + ret = devm_snd_soc_register_card(&pdev->dev, &st_card); + if (ret) { + return dev_err_probe(&pdev->dev, ret, + "devm_snd_soc_register_card(%s) failed\n", + card->name); + } + return 0; +} + +static int st_es8336_remove(struct platform_device *pdev) +{ + return 0; +} + +#ifdef CONFIG_ACPI +static const struct acpi_device_id st_audio_acpi_match[] = { + {"AMDI8336", 0}, + {}, +}; +MODULE_DEVICE_TABLE(acpi, st_audio_acpi_match); +#endif + +static struct platform_driver st_mach_driver = { + .driver = { + .name = "st-es8316", + .acpi_match_table = ACPI_PTR(st_audio_acpi_match), + .pm = &snd_soc_pm_ops, + }, + .probe = st_es8336_probe, + .remove = st_es8336_remove, +}; + +module_platform_driver(st_mach_driver); + +MODULE_AUTHOR("Vijendar.Mukunda@amd.com"); +MODULE_DESCRIPTION("st-es8316 audio support"); +MODULE_LICENSE("GPL v2"); -- cgit v1.2.3 From f94fa84058014f81ad526641f1b1f583ca2cf32f Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Fri, 1 Jul 2022 17:11:07 +0530 Subject: ASoC: amd: enable machine driver build for Jadeite platform Enable machine driver build for Jadeite platform using ES8336 Codec. Signed-off-by: Vijendar Mukunda Link: https://lore.kernel.org/r/20220701114107.1105948-6-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/Kconfig | 13 +++++++++++++ sound/soc/amd/Makefile | 2 ++ 2 files changed, 15 insertions(+) (limited to 'sound') diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index 1381aec23048..c373f0823462 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -23,6 +23,19 @@ config SND_SOC_AMD_CZ_RT5645_MACH help This option enables machine driver for rt5645. +config SND_SOC_AMD_ST_ES8336_MACH + tristate "AMD ST support for ES8336" + select SND_SOC_ACPI + select SND_SOC_ES8316 + depends on SND_SOC_AMD_ACP + depends on ACPI || COMPILE_TEST + depends on I2C || COMPILE_TEST + help + This option enables machine driver for Jadeite platform + using es8336 codec. + Say m if you have such a device. + If unsure select "N". + config SND_SOC_AMD_ACP3x tristate "AMD Audio Coprocessor-v3.x support" depends on X86 && PCI diff --git a/sound/soc/amd/Makefile b/sound/soc/amd/Makefile index 4b1f77930a4a..8823f6f28611 100644 --- a/sound/soc/amd/Makefile +++ b/sound/soc/amd/Makefile @@ -2,12 +2,14 @@ acp_audio_dma-objs := acp-pcm-dma.o snd-soc-acp-da7219mx98357-mach-objs := acp-da7219-max98357a.o snd-soc-acp-rt5645-mach-objs := acp-rt5645.o +snd-soc-acp-es8336-mach-objs := acp-es8336.o snd-soc-acp-rt5682-mach-objs := acp3x-rt5682-max9836.o snd-acp-config-objs := acp-config.o obj-$(CONFIG_SND_SOC_AMD_ACP) += acp_audio_dma.o obj-$(CONFIG_SND_SOC_AMD_CZ_DA7219MX98357_MACH) += snd-soc-acp-da7219mx98357-mach.o obj-$(CONFIG_SND_SOC_AMD_CZ_RT5645_MACH) += snd-soc-acp-rt5645-mach.o +obj-$(CONFIG_SND_SOC_AMD_ST_ES8336_MACH) += snd-soc-acp-es8336-mach.o obj-$(CONFIG_SND_SOC_AMD_ACP3x) += raven/ obj-$(CONFIG_SND_SOC_AMD_RV_RT5682_MACH) += snd-soc-acp-rt5682-mach.o obj-$(CONFIG_SND_SOC_AMD_RENOIR) += renoir/ -- cgit v1.2.3 From 8dbefb20b2d0fa8dbf81db161db443096120b326 Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Tue, 5 Jul 2022 18:11:34 +0800 Subject: ASoC: rt5640: Add the MICBIAS1 to the dapm routing The patch adds the MICBIAS1 to the dapm routing while the HDA header used. Signed-off-by: Oder Chiou Reported-by: Sameer Pujar Link: https://lore.kernel.org/r/20220705101134.16792-2-oder_chiou@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5640.c | 11 +++++++++++ 1 file changed, 11 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 56008e4518f3..5092856a262d 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2556,10 +2556,18 @@ static void rt5640_enable_jack_detect(struct snd_soc_component *component, queue_delayed_work(system_long_wq, &rt5640->jack_work, 0); } +static const struct snd_soc_dapm_route rt5640_hda_jack_dapm_routes[] = { + {"IN1P", NULL, "MICBIAS1"}, + {"IN2P", NULL, "MICBIAS1"}, + {"IN3P", NULL, "MICBIAS1"}, +}; + static void rt5640_enable_hda_jack_detect( struct snd_soc_component *component, struct snd_soc_jack *jack) { struct rt5640_priv *rt5640 = snd_soc_component_get_drvdata(component); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); int ret; /* Select JD1 for Mic */ @@ -2592,6 +2600,9 @@ static void rt5640_enable_hda_jack_detect( /* sync initial jack state */ queue_delayed_work(system_long_wq, &rt5640->jack_work, 0); + + snd_soc_dapm_add_routes(dapm, rt5640_hda_jack_dapm_routes, + ARRAY_SIZE(rt5640_hda_jack_dapm_routes)); } static int rt5640_set_jack(struct snd_soc_component *component, -- cgit v1.2.3 From 61c606a43b6c74556e35acc645c7a1b6a67c2af9 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Jan=20Sch=C3=A4r?= Date: Tue, 5 Jul 2022 15:57:46 +0200 Subject: ALSA: usb-audio: Add endianness annotations MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Fixes: 4b8ea38fabab ("ALSA: usb-audio: Support jack detection on Dell dock") Reported-by: kernel test robot Link: https://lore.kernel.org/r/202207051932.qUilU0am-lkp@intel.com Signed-off-by: Jan Schär Link: https://lore.kernel.org/r/20220705135746.13713-1-jan@jschaer.ch Signed-off-by: Takashi Iwai --- sound/usb/mixer_quirks.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 5a45822e60e7..c06d6dfa8139 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -1967,7 +1967,7 @@ static int snd_soundblaster_e1_switch_create(struct usb_mixer_interface *mixer) static int realtek_hda_set(struct snd_usb_audio *chip, u32 cmd) { struct usb_device *dev = chip->dev; - u32 buf = cpu_to_be32(cmd); + __be32 buf = cpu_to_be32(cmd); return snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), REALTEK_HDA_SET, USB_RECIP_DEVICE | USB_TYPE_VENDOR | USB_DIR_OUT, @@ -1978,7 +1978,7 @@ static int realtek_hda_get(struct snd_usb_audio *chip, u32 cmd, u32 *value) { struct usb_device *dev = chip->dev; int err; - u32 buf = cpu_to_be32(cmd); + __be32 buf = cpu_to_be32(cmd); err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), REALTEK_HDA_GET_OUT, USB_RECIP_DEVICE | USB_TYPE_VENDOR | USB_DIR_OUT, -- cgit v1.2.3 From eec8a5f44e4f68c64ce21d90e438e31e85b92178 Mon Sep 17 00:00:00 2001 From: Gaosheng Cui Date: Tue, 5 Jul 2022 08:53:15 +0800 Subject: ASoC: codecs: wsa883x: fix warning using-module-alias-sdw.cocci This patch adds missing MODULE_DEVICE_TABLE definition which generates correct modalias for automatic loading of this driver when it is built as an external module. Reported-by: Hulk Robot Signed-off-by: Gaosheng Cui Link: https://lore.kernel.org/r/20220705005315.663920-1-cuigaosheng1@huawei.com Signed-off-by: Mark Brown --- sound/soc/codecs/wsa883x.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wsa883x.c b/sound/soc/codecs/wsa883x.c index dcd88175b9cd..63e1d7aa6137 100644 --- a/sound/soc/codecs/wsa883x.c +++ b/sound/soc/codecs/wsa883x.c @@ -1492,6 +1492,8 @@ static const struct sdw_device_id wsa883x_swr_id[] = { {}, }; +MODULE_DEVICE_TABLE(sdw, wsa883x_swr_id); + static struct sdw_driver wsa883x_codec_driver = { .driver = { .name = "wsa883x-codec", -- cgit v1.2.3 From 275cc7f5bd6f60565672ce339505b77fd47a8157 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Uwe=20Kleine-K=C3=B6nig?= Date: Tue, 5 Jul 2022 11:26:45 +0200 Subject: ASoC: xilinx: Suppress second error message about reset failure in .remove() MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Returning an error value in a platform remove callback results in an error message being emitted by the platform core, but otherwise it doesn't make a difference. If ret is != 0, there is already an error message and another very generic doesn't add any value, so return 0 unconditionally. This is a preparation for making platform remove callbacks return void. Signed-off-by: Uwe Kleine-König Link: https://lore.kernel.org/r/20220705092645.101343-1-u.kleine-koenig@pengutronix.de Signed-off-by: Mark Brown --- sound/soc/xilinx/xlnx_formatter_pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/xilinx/xlnx_formatter_pcm.c b/sound/soc/xilinx/xlnx_formatter_pcm.c index f5ac0aa312d6..ff1fe62fea70 100644 --- a/sound/soc/xilinx/xlnx_formatter_pcm.c +++ b/sound/soc/xilinx/xlnx_formatter_pcm.c @@ -703,7 +703,7 @@ static int xlnx_formatter_pcm_remove(struct platform_device *pdev) dev_err(&pdev->dev, "audio formatter reset failed\n"); clk_disable_unprepare(adata->axi_clk); - return ret; + return 0; } static const struct of_device_id xlnx_formatter_pcm_of_match[] = { -- cgit v1.2.3 From f4ba35b79bd0104f00e8e21e400b980bfaa2f17e Mon Sep 17 00:00:00 2001 From: Lukas Bulwahn Date: Tue, 5 Jul 2022 12:32:38 +0200 Subject: ASoC: Intel: avs: correct config reference for I2S test board MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Commit e39acc4cfd92 ("ASoC: Intel: avs: Add I2S-test machine board") adds the config "SND_SOC_INTEL_AVS_MACH_I2S_TEST", but in the Makefile refers to config "SND_SOC_INTEL_AVS_MACH_i2s_TEST" (notice the uppercase and lowercase difference). Adjust the Makefile to refer to the actual existing config. Signed-off-by: Lukas Bulwahn Acked-by: Cezary Rojewski Reviewed-by: Amadeusz Sławiński Link: https://lore.kernel.org/r/20220705103238.7484-1-lukas.bulwahn@gmail.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/boards/Makefile | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/avs/boards/Makefile b/sound/soc/intel/avs/boards/Makefile index 25e8c4bb07db..bc75376d58c2 100644 --- a/sound/soc/intel/avs/boards/Makefile +++ b/sound/soc/intel/avs/boards/Makefile @@ -16,7 +16,7 @@ snd-soc-avs-ssm4567-objs := ssm4567.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_DA7219) += snd-soc-avs-da7219.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_DMIC) += snd-soc-avs-dmic.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_HDAUDIO) += snd-soc-avs-hdaudio.o -obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_i2s_TEST) += snd-soc-avs-i2s-test.o +obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_I2S_TEST) += snd-soc-avs-i2s-test.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_MAX98357A) += snd-soc-avs-max98357a.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_MAX98373) += snd-soc-avs-max98373.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_NAU8825) += snd-soc-avs-nau8825.o -- cgit v1.2.3 From b03bd215742c620812e47a9ef5f08e4e0e5f0a1a Mon Sep 17 00:00:00 2001 From: Andy Shevchenko Date: Tue, 5 Jul 2022 18:58:13 +0300 Subject: ASoC: Intel: catpt: remove duplicating driver data retrieval device_get_match_data() in ACPI case calls similar to acpi_match_device(). Hence there is no need to duplicate the call. Just assign what is in the id->driver_data. Signed-off-by: Andy Shevchenko Acked-by: Cezary Rojewski Link: https://lore.kernel.org/r/20220705155813.75917-1-andriy.shevchenko@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/catpt/device.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/catpt/device.c b/sound/soc/intel/catpt/device.c index 85a34e37316d..d48a71d2cf1e 100644 --- a/sound/soc/intel/catpt/device.c +++ b/sound/soc/intel/catpt/device.c @@ -254,14 +254,11 @@ static int catpt_acpi_probe(struct platform_device *pdev) return -ENODEV; } - spec = device_get_match_data(dev); - if (!spec) - return -ENODEV; - cdev = devm_kzalloc(dev, sizeof(*cdev), GFP_KERNEL); if (!cdev) return -ENOMEM; + spec = (const struct catpt_spec *)id->driver_data; catpt_dev_init(cdev, dev, spec); /* map DSP bar address */ -- cgit v1.2.3 From 0ff9f8b9f59208332c6707e37d5739c57c7f7bce Mon Sep 17 00:00:00 2001 From: Judy Hsiao Date: Fri, 1 Jul 2022 02:14:27 +0000 Subject: ASoC: rockchip: i2s: Fix error code when fail to read I2S_CLR Add the error code '-EBUSY' when fail to read I2S_CLR in rockchip_snd_rxctrl() and rockchip_snd_txctrl() Fixes: 44f362c2cc6d ("ASoC: rockchip: i2s: switch BCLK to GPIO") Signed-off-by: Judy Hsiao Reviewed-by: Brian Norris Link: https://lore.kernel.org/r/20220701021427.3120549-1-judyhsiao@chromium.org Signed-off-by: Mark Brown --- sound/soc/rockchip/rockchip_i2s.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 22ba1066c933..ee33c5d2e948 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -171,6 +171,7 @@ static int rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) retry--; if (!retry) { dev_warn(i2s->dev, "fail to clear\n"); + ret = -EBUSY; break; } } @@ -232,6 +233,7 @@ static int rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) retry--; if (!retry) { dev_warn(i2s->dev, "fail to clear\n"); + ret = -EBUSY; break; } } -- cgit v1.2.3 From 6dbc34d9c31e71aeb8175ce443c11b9e19e9f8ee Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Uwe=20Kleine-K=C3=B6nig?= Date: Wed, 29 Jun 2022 21:42:20 +0200 Subject: ASoC: tegra: tegra20_das: Fold header file into only user MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Since commit fcff5f99742e ("ASoC: tegra: remove unnecessary includes") the header file (which at the time was named tegra_das.h) there is only the actual driver that includes it. Just move the definitions into the driver, drop the exports and remove the completely unused function. Signed-off-by: Uwe Kleine-König Link: https://lore.kernel.org/r/20220629194224.175607-1-u.kleine-koenig@pengutronix.de Signed-off-by: Mark Brown --- sound/soc/tegra/tegra20_das.c | 110 ++++++++++++++++++++++++++++---------- sound/soc/tegra/tegra20_das.h | 120 ------------------------------------------ 2 files changed, 83 insertions(+), 147 deletions(-) delete mode 100644 sound/soc/tegra/tegra20_das.h (limited to 'sound') diff --git a/sound/soc/tegra/tegra20_das.c b/sound/soc/tegra/tegra20_das.c index 69c651274c12..d2801130a986 100644 --- a/sound/soc/tegra/tegra20_das.c +++ b/sound/soc/tegra/tegra20_das.c @@ -13,10 +13,90 @@ #include #include #include -#include "tegra20_das.h" #define DRV_NAME "tegra20-das" +/* Register TEGRA20_DAS_DAP_CTRL_SEL */ +#define TEGRA20_DAS_DAP_CTRL_SEL 0x00 +#define TEGRA20_DAS_DAP_CTRL_SEL_COUNT 5 +#define TEGRA20_DAS_DAP_CTRL_SEL_STRIDE 4 +#define TEGRA20_DAS_DAP_CTRL_SEL_DAP_MS_SEL_P 31 +#define TEGRA20_DAS_DAP_CTRL_SEL_DAP_MS_SEL_S 1 +#define TEGRA20_DAS_DAP_CTRL_SEL_DAP_SDATA1_TX_RX_P 30 +#define TEGRA20_DAS_DAP_CTRL_SEL_DAP_SDATA1_TX_RX_S 1 +#define TEGRA20_DAS_DAP_CTRL_SEL_DAP_SDATA2_TX_RX_P 29 +#define TEGRA20_DAS_DAP_CTRL_SEL_DAP_SDATA2_TX_RX_S 1 +#define TEGRA20_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL_P 0 +#define TEGRA20_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL_S 5 + +/* Values for field TEGRA20_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL */ +#define TEGRA20_DAS_DAP_SEL_DAC1 0 +#define TEGRA20_DAS_DAP_SEL_DAC2 1 +#define TEGRA20_DAS_DAP_SEL_DAC3 2 +#define TEGRA20_DAS_DAP_SEL_DAP1 16 +#define TEGRA20_DAS_DAP_SEL_DAP2 17 +#define TEGRA20_DAS_DAP_SEL_DAP3 18 +#define TEGRA20_DAS_DAP_SEL_DAP4 19 +#define TEGRA20_DAS_DAP_SEL_DAP5 20 + +/* Register TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL */ +#define TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL 0x40 +#define TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_COUNT 3 +#define TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_STRIDE 4 +#define TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA2_SEL_P 28 +#define TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA2_SEL_S 4 +#define TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA1_SEL_P 24 +#define TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA1_SEL_S 4 +#define TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_CLK_SEL_P 0 +#define TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_CLK_SEL_S 4 + +/* + * Values for: + * TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA2_SEL + * TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA1_SEL + * TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_CLK_SEL + */ +#define TEGRA20_DAS_DAC_SEL_DAP1 0 +#define TEGRA20_DAS_DAC_SEL_DAP2 1 +#define TEGRA20_DAS_DAC_SEL_DAP3 2 +#define TEGRA20_DAS_DAC_SEL_DAP4 3 +#define TEGRA20_DAS_DAC_SEL_DAP5 4 + +/* + * Names/IDs of the DACs/DAPs. + */ + +#define TEGRA20_DAS_DAP_ID_1 0 +#define TEGRA20_DAS_DAP_ID_2 1 +#define TEGRA20_DAS_DAP_ID_3 2 +#define TEGRA20_DAS_DAP_ID_4 3 +#define TEGRA20_DAS_DAP_ID_5 4 + +#define TEGRA20_DAS_DAC_ID_1 0 +#define TEGRA20_DAS_DAC_ID_2 1 +#define TEGRA20_DAS_DAC_ID_3 2 + +struct tegra20_das { + struct device *dev; + struct regmap *regmap; +}; + +/* + * Terminology: + * DAS: Digital audio switch (HW module controlled by this driver) + * DAP: Digital audio port (port/pins on Tegra device) + * DAC: Digital audio controller (e.g. I2S or AC97 controller elsewhere) + * + * The Tegra DAS is a mux/cross-bar which can connect each DAP to a specific + * DAC, or another DAP. When DAPs are connected, one must be the master and + * one the slave. Each DAC allows selection of a specific DAP for input, to + * cater for the case where N DAPs are connected to 1 DAC for broadcast + * output. + * + * This driver is dumb; no attempt is made to ensure that a valid routing + * configuration is programmed. + */ + static struct tegra20_das *das; static inline void tegra20_das_write(u32 reg, u32 val) @@ -32,7 +112,7 @@ static inline u32 tegra20_das_read(u32 reg) return val; } -int tegra20_das_connect_dap_to_dac(int dap, int dac) +static int tegra20_das_connect_dap_to_dac(int dap, int dac) { u32 addr; u32 reg; @@ -48,31 +128,8 @@ int tegra20_das_connect_dap_to_dac(int dap, int dac) return 0; } -EXPORT_SYMBOL_GPL(tegra20_das_connect_dap_to_dac); - -int tegra20_das_connect_dap_to_dap(int dap, int otherdap, int master, - int sdata1rx, int sdata2rx) -{ - u32 addr; - u32 reg; - - if (!das) - return -ENODEV; - - addr = TEGRA20_DAS_DAP_CTRL_SEL + - (dap * TEGRA20_DAS_DAP_CTRL_SEL_STRIDE); - reg = (otherdap << TEGRA20_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL_P) | - (!!sdata2rx << TEGRA20_DAS_DAP_CTRL_SEL_DAP_SDATA2_TX_RX_P) | - (!!sdata1rx << TEGRA20_DAS_DAP_CTRL_SEL_DAP_SDATA1_TX_RX_P) | - (!!master << TEGRA20_DAS_DAP_CTRL_SEL_DAP_MS_SEL_P); - - tegra20_das_write(addr, reg); - - return 0; -} -EXPORT_SYMBOL_GPL(tegra20_das_connect_dap_to_dap); -int tegra20_das_connect_dac_to_dap(int dac, int dap) +static int tegra20_das_connect_dac_to_dap(int dac, int dap) { u32 addr; u32 reg; @@ -90,7 +147,6 @@ int tegra20_das_connect_dac_to_dap(int dac, int dap) return 0; } -EXPORT_SYMBOL_GPL(tegra20_das_connect_dac_to_dap); #define LAST_REG(name) \ (TEGRA20_DAS_##name + \ diff --git a/sound/soc/tegra/tegra20_das.h b/sound/soc/tegra/tegra20_das.h deleted file mode 100644 index 18e832ded73a..000000000000 --- a/sound/soc/tegra/tegra20_das.h +++ /dev/null @@ -1,120 +0,0 @@ -/* SPDX-License-Identifier: GPL-2.0-only */ -/* - * tegra20_das.h - Definitions for Tegra20 DAS driver - * - * Author: Stephen Warren - * Copyright (C) 2010,2012 - NVIDIA, Inc. - */ - -#ifndef __TEGRA20_DAS_H__ -#define __TEGRA20_DAS_H__ - -/* Register TEGRA20_DAS_DAP_CTRL_SEL */ -#define TEGRA20_DAS_DAP_CTRL_SEL 0x00 -#define TEGRA20_DAS_DAP_CTRL_SEL_COUNT 5 -#define TEGRA20_DAS_DAP_CTRL_SEL_STRIDE 4 -#define TEGRA20_DAS_DAP_CTRL_SEL_DAP_MS_SEL_P 31 -#define TEGRA20_DAS_DAP_CTRL_SEL_DAP_MS_SEL_S 1 -#define TEGRA20_DAS_DAP_CTRL_SEL_DAP_SDATA1_TX_RX_P 30 -#define TEGRA20_DAS_DAP_CTRL_SEL_DAP_SDATA1_TX_RX_S 1 -#define TEGRA20_DAS_DAP_CTRL_SEL_DAP_SDATA2_TX_RX_P 29 -#define TEGRA20_DAS_DAP_CTRL_SEL_DAP_SDATA2_TX_RX_S 1 -#define TEGRA20_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL_P 0 -#define TEGRA20_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL_S 5 - -/* Values for field TEGRA20_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL */ -#define TEGRA20_DAS_DAP_SEL_DAC1 0 -#define TEGRA20_DAS_DAP_SEL_DAC2 1 -#define TEGRA20_DAS_DAP_SEL_DAC3 2 -#define TEGRA20_DAS_DAP_SEL_DAP1 16 -#define TEGRA20_DAS_DAP_SEL_DAP2 17 -#define TEGRA20_DAS_DAP_SEL_DAP3 18 -#define TEGRA20_DAS_DAP_SEL_DAP4 19 -#define TEGRA20_DAS_DAP_SEL_DAP5 20 - -/* Register TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL */ -#define TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL 0x40 -#define TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_COUNT 3 -#define TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_STRIDE 4 -#define TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA2_SEL_P 28 -#define TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA2_SEL_S 4 -#define TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA1_SEL_P 24 -#define TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA1_SEL_S 4 -#define TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_CLK_SEL_P 0 -#define TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_CLK_SEL_S 4 - -/* - * Values for: - * TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA2_SEL - * TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA1_SEL - * TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_CLK_SEL - */ -#define TEGRA20_DAS_DAC_SEL_DAP1 0 -#define TEGRA20_DAS_DAC_SEL_DAP2 1 -#define TEGRA20_DAS_DAC_SEL_DAP3 2 -#define TEGRA20_DAS_DAC_SEL_DAP4 3 -#define TEGRA20_DAS_DAC_SEL_DAP5 4 - -/* - * Names/IDs of the DACs/DAPs. - */ - -#define TEGRA20_DAS_DAP_ID_1 0 -#define TEGRA20_DAS_DAP_ID_2 1 -#define TEGRA20_DAS_DAP_ID_3 2 -#define TEGRA20_DAS_DAP_ID_4 3 -#define TEGRA20_DAS_DAP_ID_5 4 - -#define TEGRA20_DAS_DAC_ID_1 0 -#define TEGRA20_DAS_DAC_ID_2 1 -#define TEGRA20_DAS_DAC_ID_3 2 - -struct tegra20_das { - struct device *dev; - struct regmap *regmap; -}; - -/* - * Terminology: - * DAS: Digital audio switch (HW module controlled by this driver) - * DAP: Digital audio port (port/pins on Tegra device) - * DAC: Digital audio controller (e.g. I2S or AC97 controller elsewhere) - * - * The Tegra DAS is a mux/cross-bar which can connect each DAP to a specific - * DAC, or another DAP. When DAPs are connected, one must be the master and - * one the slave. Each DAC allows selection of a specific DAP for input, to - * cater for the case where N DAPs are connected to 1 DAC for broadcast - * output. - * - * This driver is dumb; no attempt is made to ensure that a valid routing - * configuration is programmed. - */ - -/* - * Connect a DAP to a DAC - * dap_id: DAP to connect: TEGRA20_DAS_DAP_ID_* - * dac_sel: DAC to connect to: TEGRA20_DAS_DAP_SEL_DAC* - */ -extern int tegra20_das_connect_dap_to_dac(int dap, int dac); - -/* - * Connect a DAP to another DAP - * dap_id: DAP to connect: TEGRA20_DAS_DAP_ID_* - * other_dap_sel: DAP to connect to: TEGRA20_DAS_DAP_SEL_DAP* - * master: Is this DAP the master (1) or slave (0) - * sdata1rx: Is this DAP's SDATA1 pin RX (1) or TX (0) - * sdata2rx: Is this DAP's SDATA2 pin RX (1) or TX (0) - */ -extern int tegra20_das_connect_dap_to_dap(int dap, int otherdap, - int master, int sdata1rx, - int sdata2rx); - -/* - * Connect a DAC's input to a DAP - * (DAC outputs are selected by the DAP) - * dac_id: DAC ID to connect: TEGRA20_DAS_DAC_ID_* - * dap_sel: DAP to receive input from: TEGRA20_DAS_DAC_SEL_DAP* - */ -extern int tegra20_das_connect_dac_to_dap(int dac, int dap); - -#endif -- cgit v1.2.3 From 9a99b9b26451ca2a81867ce0cd8fe18dce856a8c Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Uwe=20Kleine-K=C3=B6nig?= Date: Wed, 29 Jun 2022 21:42:21 +0200 Subject: ASoC: tegra: tegra20_das: Remove unused function tegra20_das_read MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This function is unused since commit 7203a62562dc ("ASoC: convert Tegra20 DAS driver to regmap"). Signed-off-by: Uwe Kleine-König Link: https://lore.kernel.org/r/20220629194224.175607-2-u.kleine-koenig@pengutronix.de Signed-off-by: Mark Brown --- sound/soc/tegra/tegra20_das.c | 8 -------- 1 file changed, 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra20_das.c b/sound/soc/tegra/tegra20_das.c index d2801130a986..4e23fd96c745 100644 --- a/sound/soc/tegra/tegra20_das.c +++ b/sound/soc/tegra/tegra20_das.c @@ -104,14 +104,6 @@ static inline void tegra20_das_write(u32 reg, u32 val) regmap_write(das->regmap, reg, val); } -static inline u32 tegra20_das_read(u32 reg) -{ - u32 val; - - regmap_read(das->regmap, reg, &val); - return val; -} - static int tegra20_das_connect_dap_to_dac(int dap, int dac) { u32 addr; -- cgit v1.2.3 From eefaea93235523d248cc8cadcd6be9f47b03b9d5 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Uwe=20Kleine-K=C3=B6nig?= Date: Wed, 29 Jun 2022 21:42:22 +0200 Subject: ASoC: tegra: tegra20_das: Get rid of global pointer for driver data MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This enables the driver (at least theoretically) to bind to more than one device. The remove function has nothing to do now, so it is dropped. Signed-off-by: Uwe Kleine-König Link: https://lore.kernel.org/r/20220629194224.175607-3-u.kleine-koenig@pengutronix.de Signed-off-by: Mark Brown --- sound/soc/tegra/tegra20_das.c | 66 +++++++++++++------------------------------ 1 file changed, 20 insertions(+), 46 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra20_das.c b/sound/soc/tegra/tegra20_das.c index 4e23fd96c745..8637a0cc1f5e 100644 --- a/sound/soc/tegra/tegra20_das.c +++ b/sound/soc/tegra/tegra20_das.c @@ -97,14 +97,12 @@ struct tegra20_das { * configuration is programmed. */ -static struct tegra20_das *das; - -static inline void tegra20_das_write(u32 reg, u32 val) +static inline void tegra20_das_write(struct tegra20_das *das, u32 reg, u32 val) { regmap_write(das->regmap, reg, val); } -static int tegra20_das_connect_dap_to_dac(int dap, int dac) +static int tegra20_das_connect_dap_to_dac(struct tegra20_das *das, int dap, int dac) { u32 addr; u32 reg; @@ -116,12 +114,12 @@ static int tegra20_das_connect_dap_to_dac(int dap, int dac) (dap * TEGRA20_DAS_DAP_CTRL_SEL_STRIDE); reg = dac << TEGRA20_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL_P; - tegra20_das_write(addr, reg); + tegra20_das_write(das, addr, reg); return 0; } -static int tegra20_das_connect_dac_to_dap(int dac, int dap) +static int tegra20_das_connect_dac_to_dap(struct tegra20_das *das, int dac, int dap) { u32 addr; u32 reg; @@ -135,7 +133,7 @@ static int tegra20_das_connect_dac_to_dap(int dac, int dap) dap << TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA1_SEL_P | dap << TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA2_SEL_P; - tegra20_das_write(addr, reg); + tegra20_das_write(das, addr, reg); return 0; } @@ -168,74 +166,51 @@ static const struct regmap_config tegra20_das_regmap_config = { static int tegra20_das_probe(struct platform_device *pdev) { void __iomem *regs; + struct tegra20_das *das; int ret = 0; - if (das) - return -ENODEV; - das = devm_kzalloc(&pdev->dev, sizeof(struct tegra20_das), GFP_KERNEL); - if (!das) { - ret = -ENOMEM; - goto err; - } + if (!das) + return -ENOMEM; + das->dev = &pdev->dev; regs = devm_platform_ioremap_resource(pdev, 0); - if (IS_ERR(regs)) { - ret = PTR_ERR(regs); - goto err; - } + if (IS_ERR(regs)) + return PTR_ERR(regs); das->regmap = devm_regmap_init_mmio(&pdev->dev, regs, &tegra20_das_regmap_config); if (IS_ERR(das->regmap)) { dev_err(&pdev->dev, "regmap init failed\n"); - ret = PTR_ERR(das->regmap); - goto err; + return PTR_ERR(das->regmap); } - ret = tegra20_das_connect_dap_to_dac(TEGRA20_DAS_DAP_ID_1, + ret = tegra20_das_connect_dap_to_dac(das, TEGRA20_DAS_DAP_ID_1, TEGRA20_DAS_DAP_SEL_DAC1); if (ret) { dev_err(&pdev->dev, "Can't set up DAS DAP connection\n"); - goto err; + return ret; } - ret = tegra20_das_connect_dac_to_dap(TEGRA20_DAS_DAC_ID_1, + ret = tegra20_das_connect_dac_to_dap(das, TEGRA20_DAS_DAC_ID_1, TEGRA20_DAS_DAC_SEL_DAP1); if (ret) { dev_err(&pdev->dev, "Can't set up DAS DAC connection\n"); - goto err; + return ret; } - - ret = tegra20_das_connect_dap_to_dac(TEGRA20_DAS_DAP_ID_3, + ret = tegra20_das_connect_dap_to_dac(das, TEGRA20_DAS_DAP_ID_3, TEGRA20_DAS_DAP_SEL_DAC3); if (ret) { dev_err(&pdev->dev, "Can't set up DAS DAP connection\n"); - goto err; + return ret; } - ret = tegra20_das_connect_dac_to_dap(TEGRA20_DAS_DAC_ID_3, + ret = tegra20_das_connect_dac_to_dap(das, TEGRA20_DAS_DAC_ID_3, TEGRA20_DAS_DAC_SEL_DAP3); if (ret) { dev_err(&pdev->dev, "Can't set up DAS DAC connection\n"); - goto err; + return ret; } - platform_set_drvdata(pdev, das); - - return 0; - -err: - das = NULL; - return ret; -} - -static int tegra20_das_remove(struct platform_device *pdev) -{ - if (!das) - return -ENODEV; - - das = NULL; - return 0; } @@ -246,7 +221,6 @@ static const struct of_device_id tegra20_das_of_match[] = { static struct platform_driver tegra20_das_driver = { .probe = tegra20_das_probe, - .remove = tegra20_das_remove, .driver = { .name = DRV_NAME, .of_match_table = tegra20_das_of_match, -- cgit v1.2.3 From a10a8b6661c478dac3a8c55ad41f5cb00779c6b9 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Uwe=20Kleine-K=C3=B6nig?= Date: Wed, 29 Jun 2022 21:42:23 +0200 Subject: ASoC: tegra: tegra20_das: Make helper functions return void MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit These only ever return a value != 0 if the parameter das is NULL. In the only caller however it's already asserted this isn't the case. So convert the functions to return void and simplify the caller accordingly. Signed-off-by: Uwe Kleine-König Link: https://lore.kernel.org/r/20220629194224.175607-4-u.kleine-koenig@pengutronix.de Signed-off-by: Mark Brown --- sound/soc/tegra/tegra20_das.c | 47 +++++++++---------------------------------- 1 file changed, 10 insertions(+), 37 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra20_das.c b/sound/soc/tegra/tegra20_das.c index 8637a0cc1f5e..39a6135dd0d0 100644 --- a/sound/soc/tegra/tegra20_das.c +++ b/sound/soc/tegra/tegra20_das.c @@ -102,31 +102,23 @@ static inline void tegra20_das_write(struct tegra20_das *das, u32 reg, u32 val) regmap_write(das->regmap, reg, val); } -static int tegra20_das_connect_dap_to_dac(struct tegra20_das *das, int dap, int dac) +static void tegra20_das_connect_dap_to_dac(struct tegra20_das *das, int dap, int dac) { u32 addr; u32 reg; - if (!das) - return -ENODEV; - addr = TEGRA20_DAS_DAP_CTRL_SEL + (dap * TEGRA20_DAS_DAP_CTRL_SEL_STRIDE); reg = dac << TEGRA20_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL_P; tegra20_das_write(das, addr, reg); - - return 0; } -static int tegra20_das_connect_dac_to_dap(struct tegra20_das *das, int dac, int dap) +static void tegra20_das_connect_dac_to_dap(struct tegra20_das *das, int dac, int dap) { u32 addr; u32 reg; - if (!das) - return -ENODEV; - addr = TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL + (dac * TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_STRIDE); reg = dap << TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_CLK_SEL_P | @@ -134,8 +126,6 @@ static int tegra20_das_connect_dac_to_dap(struct tegra20_das *das, int dac, int dap << TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA2_SEL_P; tegra20_das_write(das, addr, reg); - - return 0; } #define LAST_REG(name) \ @@ -167,7 +157,6 @@ static int tegra20_das_probe(struct platform_device *pdev) { void __iomem *regs; struct tegra20_das *das; - int ret = 0; das = devm_kzalloc(&pdev->dev, sizeof(struct tegra20_das), GFP_KERNEL); if (!das) @@ -186,30 +175,14 @@ static int tegra20_das_probe(struct platform_device *pdev) return PTR_ERR(das->regmap); } - ret = tegra20_das_connect_dap_to_dac(das, TEGRA20_DAS_DAP_ID_1, - TEGRA20_DAS_DAP_SEL_DAC1); - if (ret) { - dev_err(&pdev->dev, "Can't set up DAS DAP connection\n"); - return ret; - } - ret = tegra20_das_connect_dac_to_dap(das, TEGRA20_DAS_DAC_ID_1, - TEGRA20_DAS_DAC_SEL_DAP1); - if (ret) { - dev_err(&pdev->dev, "Can't set up DAS DAC connection\n"); - return ret; - } - ret = tegra20_das_connect_dap_to_dac(das, TEGRA20_DAS_DAP_ID_3, - TEGRA20_DAS_DAP_SEL_DAC3); - if (ret) { - dev_err(&pdev->dev, "Can't set up DAS DAP connection\n"); - return ret; - } - ret = tegra20_das_connect_dac_to_dap(das, TEGRA20_DAS_DAC_ID_3, - TEGRA20_DAS_DAC_SEL_DAP3); - if (ret) { - dev_err(&pdev->dev, "Can't set up DAS DAC connection\n"); - return ret; - } + tegra20_das_connect_dap_to_dac(das, TEGRA20_DAS_DAP_ID_1, + TEGRA20_DAS_DAP_SEL_DAC1); + tegra20_das_connect_dac_to_dap(das, TEGRA20_DAS_DAC_ID_1, + TEGRA20_DAS_DAC_SEL_DAP1); + tegra20_das_connect_dap_to_dac(das, TEGRA20_DAS_DAP_ID_3, + TEGRA20_DAS_DAP_SEL_DAC3); + tegra20_das_connect_dac_to_dap(das, TEGRA20_DAS_DAC_ID_3, + TEGRA20_DAS_DAC_SEL_DAP3); return 0; } -- cgit v1.2.3 From fb617612fd8e017720d7fe907b22b4bb44027948 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Uwe=20Kleine-K=C3=B6nig?= Date: Wed, 29 Jun 2022 21:42:24 +0200 Subject: ASoC: tegra: tegra20_das: Drop write-only driver data member MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The dev member of struct tegra20_das is only written once in .probe(). There is no loss of functionality if the member and the assignment go away. Signed-off-by: Uwe Kleine-König Link: https://lore.kernel.org/r/20220629194224.175607-5-u.kleine-koenig@pengutronix.de Signed-off-by: Mark Brown --- sound/soc/tegra/tegra20_das.c | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra20_das.c b/sound/soc/tegra/tegra20_das.c index 39a6135dd0d0..c620ab0c601f 100644 --- a/sound/soc/tegra/tegra20_das.c +++ b/sound/soc/tegra/tegra20_das.c @@ -77,7 +77,6 @@ #define TEGRA20_DAS_DAC_ID_3 2 struct tegra20_das { - struct device *dev; struct regmap *regmap; }; @@ -162,8 +161,6 @@ static int tegra20_das_probe(struct platform_device *pdev) if (!das) return -ENOMEM; - das->dev = &pdev->dev; - regs = devm_platform_ioremap_resource(pdev, 0); if (IS_ERR(regs)) return PTR_ERR(regs); -- cgit v1.2.3 From 0d356c186ffd6d4c3e10abb283379d09a93d2515 Mon Sep 17 00:00:00 2001 From: Andy Shevchenko Date: Tue, 5 Jul 2022 19:11:01 +0300 Subject: ASoC: SOF: Intel: bdw: remove duplicating driver data retrieval device_get_match_data() in ACPI case calls similar to acpi_match_device(). Hence there is no need to duplicate the call. Just assign what is in the id->driver_data. Signed-off-by: Andy Shevchenko Acked-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220705161102.76250-1-andriy.shevchenko@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/bdw.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/intel/bdw.c b/sound/soc/sof/intel/bdw.c index 26df780c702e..a446154f2803 100644 --- a/sound/soc/sof/intel/bdw.c +++ b/sound/soc/sof/intel/bdw.c @@ -681,11 +681,8 @@ static int sof_broadwell_probe(struct platform_device *pdev) return -ENODEV; } - desc = device_get_match_data(dev); - if (!desc) - return -ENODEV; - - return sof_acpi_probe(pdev, device_get_match_data(dev)); + desc = (const struct sof_dev_desc *)id->driver_data; + return sof_acpi_probe(pdev, desc); } /* acpi_driver definition */ -- cgit v1.2.3 From 65b6851d243ff54cbd4adfb887a8af9d04b7f286 Mon Sep 17 00:00:00 2001 From: Andy Shevchenko Date: Tue, 5 Jul 2022 19:11:02 +0300 Subject: ASoC: SOF: Intel: byt: remove duplicating driver data retrieval device_get_match_data() in ACPI case calls similar to acpi_match_device(). Hence there is no need to duplicate the call. Just assign what is in the id->driver_data. Signed-off-by: Andy Shevchenko Acked-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220705161102.76250-2-andriy.shevchenko@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/byt.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/intel/byt.c b/sound/soc/sof/intel/byt.c index 4ed8381eceda..e6dc4ff531c3 100644 --- a/sound/soc/sof/intel/byt.c +++ b/sound/soc/sof/intel/byt.c @@ -465,10 +465,7 @@ static int sof_baytrail_probe(struct platform_device *pdev) return -ENODEV; } - desc = device_get_match_data(&pdev->dev); - if (!desc) - return -ENODEV; - + desc = (const struct sof_dev_desc *)id->driver_data; if (desc == &sof_acpi_baytrail_desc && soc_intel_is_byt_cr(pdev)) desc = &sof_acpi_baytrailcr_desc; -- cgit v1.2.3 From c0fabd12a8570cb932f13d9388f3d887ad44369b Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Wed, 6 Jul 2022 17:42:55 +0800 Subject: ASoC: imx-card: Fix DSD/PDM mclk frequency The DSD/PDM rate not only DSD64/128/256/512, which are the multiple rate of 44.1kHz, but also support the multiple rate of 8kHz, so can't force all mclk frequency to be 22579200Hz, need to assign the frequency according to rate. Fixes: aa736700f42f ("ASoC: imx-card: Add imx-card machine driver") Signed-off-by: Shengjiu Wang Link: https://lore.kernel.org/r/1657100575-8261-1-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/imx-card.c | 14 +++++++++++--- 1 file changed, 11 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/imx-card.c b/sound/soc/fsl/imx-card.c index 1797d777b1b8..ccc4194dc5e7 100644 --- a/sound/soc/fsl/imx-card.c +++ b/sound/soc/fsl/imx-card.c @@ -17,6 +17,9 @@ #include "fsl_sai.h" +#define IMX_CARD_MCLK_22P5792MHZ 22579200 +#define IMX_CARD_MCLK_24P576MHZ 24576000 + enum codec_type { CODEC_DUMMY = 0, CODEC_AK5558 = 1, @@ -353,9 +356,14 @@ static int imx_aif_hw_params(struct snd_pcm_substream *substream, mclk_freq = akcodec_get_mclk_rate(substream, params, slots, slot_width); else mclk_freq = params_rate(params) * slots * slot_width; - /* Use the maximum freq from DSD512 (512*44100 = 22579200) */ - if (format_is_dsd(params)) - mclk_freq = 22579200; + + if (format_is_dsd(params)) { + /* Use the maximum freq from DSD512 (512*44100 = 22579200) */ + if (!(params_rate(params) % 11025)) + mclk_freq = IMX_CARD_MCLK_22P5792MHZ; + else + mclk_freq = IMX_CARD_MCLK_24P576MHZ; + } ret = snd_soc_dai_set_sysclk(cpu_dai, link_data->cpu_sysclk_id, mclk_freq, SND_SOC_CLOCK_OUT); -- cgit v1.2.3 From f1fd46e068f52893608469df98d4608672e3e45f Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Wed, 6 Jul 2022 08:29:52 +0200 Subject: ASoC: Intel: avs: Fix i2s_test card name initialization Update printf formatting as 'ssp_port' argument is of type 'int', not 'long int'. Fixes: e39acc4cfd92 ("ASoC: Intel: avs: Add I2S-test machine board") Reported-by: kernel test robot Signed-off-by: Cezary Rojewski Link: https://lore.kernel.org/r/20220706062952.251704-1-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/boards/i2s_test.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/avs/boards/i2s_test.c b/sound/soc/intel/avs/boards/i2s_test.c index 461b651cd331..8f0fd87bc866 100644 --- a/sound/soc/intel/avs/boards/i2s_test.c +++ b/sound/soc/intel/avs/boards/i2s_test.c @@ -127,7 +127,7 @@ static int avs_i2s_test_probe(struct platform_device *pdev) if (!card) return -ENOMEM; - card->name = devm_kasprintf(dev, GFP_KERNEL, "ssp%ld-loopback", ssp_port); + card->name = devm_kasprintf(dev, GFP_KERNEL, "ssp%d-loopback", ssp_port); if (!card->name) return -ENOMEM; -- cgit v1.2.3 From e57297fc0915e2f95de26d18ad8ab6f17c068658 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Uwe=20Kleine-K=C3=B6nig?= Date: Tue, 5 Jul 2022 08:36:13 +0200 Subject: ASoC: rsnd: Emit useful error messages in .remove() MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit If more than one call of rsnd_dai_call(remove, ...) fails the platform remove callback returns all values orred together which then makes the driver core emit a generic error message which is little helpful. Instead emit details of which call failed exactly and return 0. Note returning 0 instead of an error code doesn't make a difference in the driver core apart from the error message. This is a preparation for making platform remove callbacks return void. Signed-off-by: Uwe Kleine-König Link: https://lore.kernel.org/r/20220705063613.93770-1-u.kleine-koenig@pengutronix.de Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 15 +++++++++++---- 1 file changed, 11 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 4973f94a2144..7e380d71b0f8 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1969,19 +1969,26 @@ static int rsnd_remove(struct platform_device *pdev) rsnd_cmd_remove, rsnd_adg_remove, }; - int ret = 0, i; + int i; pm_runtime_disable(&pdev->dev); for_each_rsnd_dai(rdai, priv, i) { - ret |= rsnd_dai_call(remove, &rdai->playback, priv); - ret |= rsnd_dai_call(remove, &rdai->capture, priv); + int ret; + + ret = rsnd_dai_call(remove, &rdai->playback, priv); + if (ret) + dev_warn(&pdev->dev, "Failed to remove playback dai #%d\n", i); + + ret = rsnd_dai_call(remove, &rdai->capture, priv); + if (ret) + dev_warn(&pdev->dev, "Failed to remove capture dai #%d\n", i); } for (i = 0; i < ARRAY_SIZE(remove_func); i++) remove_func[i](priv); - return ret; + return 0; } static int __maybe_unused rsnd_suspend(struct device *dev) -- cgit v1.2.3 From ab34403db24233e603338b70deb9a84093c88397 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Thu, 7 Jul 2022 02:25:14 +0530 Subject: ASoC: amd: fix ACPI dependency compile errors and warnings Fixed ACPI dependency complie errors and warnings as listed below. All warnings (new ones prefixed by >>): sound/soc/soc-acpi.c:34:1: error: redefinition of 'snd_soc_acpi_find_machine' 34 | snd_soc_acpi_find_machine(struct snd_soc_acpi_mach *machines) | ^~~~~~~~~~~~~~~~~~~~~~~~~ In file included from sound/soc/soc-acpi.c:9: include/sound/soc-acpi.h:38:1: note: previous definition of 'snd_soc_acpi_find_machine' with type 'struct snd_soc_acpi_mach *(struct snd_soc_acpi_mach *)' 38 | snd_soc_acpi_find_machine(struct snd_soc_acpi_mach *machines) | ^~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/soc-acpi.c: In function 'snd_soc_acpi_find_package': sound/soc/soc-acpi.c:58:36: error: implicit declaration of function 'acpi_fetch_acpi_dev'; did you mean 'device_match_acpi_dev'? [-Werror=implicit-function-declaration] 58 | struct acpi_device *adev = acpi_fetch_acpi_dev(handle); | ^~~~~~~~~~~~~~~~~~~ | device_match_acpi_dev >> sound/soc/soc-acpi.c:58:36: warning: initialization of 'struct acpi_device *' from 'int' makes pointer from integer without a cast [-Wint-conversion] sound/soc/soc-acpi.c:64:25: error: invalid use of undefined type 'struct acpi_device' 64 | if (adev && adev->status.present && adev->status.functional) { | ^~ sound/soc/soc-acpi.c:64:49: error: invalid use of undefined type 'struct acpi_device' 64 | if (adev && adev->status.present && adev->status.functional) { | ^~ sound/soc/soc-acpi.c:80:26: error: implicit declaration of function 'acpi_extract_package' [-Werror=implicit-function-declaration] 80 | status = acpi_extract_package(myobj, | ^~~~~~~~~~~~~~~~~~~~ sound/soc/soc-acpi.c: At top level: sound/soc/soc-acpi.c:95:6: error: redefinition of 'snd_soc_acpi_find_package_from_hid' 95 | bool snd_soc_acpi_find_package_from_hid(const u8 hid[ACPI_ID_LEN], | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ In file included from sound/soc/soc-acpi.c:9: include/sound/soc-acpi.h:44:1: note: previous definition of 'snd_soc_acpi_find_package_from_hid' with type 'bool(const u8 *, struct snd_soc_acpi_package_context *)' {aka '_Bool(const unsigned char *, struct snd_soc_acpi_package_context *)'} 44 | snd_soc_acpi_find_package_from_hid(const u8 hid[ACPI_ID_LEN], | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/soc-acpi.c:109:27: error: redefinition of 'snd_soc_acpi_codec_list' 109 | struct snd_soc_acpi_mach *snd_soc_acpi_codec_list(void *arg) | ^~~~~~~~~~~~~~~~~~~~~~~ In file included from sound/soc/soc-acpi.c:9: include/sound/soc-acpi.h:51:41: note: previous definition of 'snd_soc_acpi_codec_list' with type 'struct snd_soc_acpi_mach *(void *)' 51 | static inline struct snd_soc_acpi_mach *snd_soc_acpi_codec_list(void *arg) | ^~~~~~~~~~~~~~~~~~~~~~~ Signed-off-by: Vijendar Mukunda Reported-by: kernel test robot Link: https://lore.kernel.org/r/20220706205515.2485601-1-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/Kconfig | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index c373f0823462..9629328c419e 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -25,10 +25,9 @@ config SND_SOC_AMD_CZ_RT5645_MACH config SND_SOC_AMD_ST_ES8336_MACH tristate "AMD ST support for ES8336" - select SND_SOC_ACPI + select SND_SOC_ACPI if ACPI select SND_SOC_ES8316 - depends on SND_SOC_AMD_ACP - depends on ACPI || COMPILE_TEST + depends on SND_SOC_AMD_ACP && ACPI depends on I2C || COMPILE_TEST help This option enables machine driver for Jadeite platform -- cgit v1.2.3 From 98356c89d44dac838dfbab02975645d828de3099 Mon Sep 17 00:00:00 2001 From: Aidan MacDonald Date: Wed, 6 Jul 2022 22:13:20 +0100 Subject: ASoC: jz4740-i2s: Remove Open Firmware dependency This driver doesn't require Open Firmware support. Remove the OF-specific includes and drop the Kconfig dependency. Signed-off-by: Aidan MacDonald Acked-by: Paul Cercueil Link: https://lore.kernel.org/r/20220706211330.120198-2-aidanmacdonald.0x0@gmail.com Signed-off-by: Mark Brown --- sound/soc/jz4740/Kconfig | 2 +- sound/soc/jz4740/jz4740-i2s.c | 3 +-- 2 files changed, 2 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/jz4740/Kconfig b/sound/soc/jz4740/Kconfig index 29144720cb62..e72f826062e9 100644 --- a/sound/soc/jz4740/Kconfig +++ b/sound/soc/jz4740/Kconfig @@ -2,7 +2,7 @@ config SND_JZ4740_SOC_I2S tristate "SoC Audio (I2S protocol) for Ingenic JZ4740 SoC" depends on MIPS || COMPILE_TEST - depends on OF && HAS_IOMEM + depends on HAS_IOMEM select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y if you want to use I2S protocol and I2S codec on Ingenic JZ4740 diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index 79afac0c5003..298ff0a83931 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -5,10 +5,9 @@ #include #include -#include -#include #include #include +#include #include #include -- cgit v1.2.3 From 8a7691010992886290b340a1ba943067c2e70f85 Mon Sep 17 00:00:00 2001 From: Aidan MacDonald Date: Wed, 6 Jul 2022 22:13:21 +0100 Subject: ASoC: jz4740-i2s: Refactor DMA channel setup It's simpler to set up the playback and capture DMA settings at driver probe time instead of during DAI probing. Signed-off-by: Aidan MacDonald Acked-by: Paul Cercueil Link: https://lore.kernel.org/r/20220706211330.120198-3-aidanmacdonald.0x0@gmail.com Signed-off-by: Mark Brown --- sound/soc/jz4740/jz4740-i2s.c | 23 +++++------------------ 1 file changed, 5 insertions(+), 18 deletions(-) (limited to 'sound') diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index 298ff0a83931..ecd8df70d39c 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -95,7 +95,6 @@ struct i2s_soc_info { struct jz4740_i2s { struct resource *mem; void __iomem *base; - dma_addr_t phys_base; struct clk *clk_aic; struct clk *clk_i2s; @@ -370,21 +369,6 @@ static int jz4740_i2s_resume(struct snd_soc_component *component) return 0; } -static void jz4740_i2s_init_pcm_config(struct jz4740_i2s *i2s) -{ - struct snd_dmaengine_dai_dma_data *dma_data; - - /* Playback */ - dma_data = &i2s->playback_dma_data; - dma_data->maxburst = 16; - dma_data->addr = i2s->phys_base + JZ_REG_AIC_FIFO; - - /* Capture */ - dma_data = &i2s->capture_dma_data; - dma_data->maxburst = 16; - dma_data->addr = i2s->phys_base + JZ_REG_AIC_FIFO; -} - static int jz4740_i2s_dai_probe(struct snd_soc_dai *dai) { struct jz4740_i2s *i2s = snd_soc_dai_get_drvdata(dai); @@ -395,7 +379,6 @@ static int jz4740_i2s_dai_probe(struct snd_soc_dai *dai) if (ret) return ret; - jz4740_i2s_init_pcm_config(i2s); snd_soc_dai_init_dma_data(dai, &i2s->playback_dma_data, &i2s->capture_dma_data); @@ -529,7 +512,11 @@ static int jz4740_i2s_dev_probe(struct platform_device *pdev) if (IS_ERR(i2s->base)) return PTR_ERR(i2s->base); - i2s->phys_base = mem->start; + i2s->playback_dma_data.maxburst = 16; + i2s->playback_dma_data.addr = mem->start + JZ_REG_AIC_FIFO; + + i2s->capture_dma_data.maxburst = 16; + i2s->capture_dma_data.addr = mem->start + JZ_REG_AIC_FIFO; i2s->clk_aic = devm_clk_get(dev, "aic"); if (IS_ERR(i2s->clk_aic)) -- cgit v1.2.3 From 050237e6b0bea0fafbf7d3d57e717c6fa1e4e819 Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Thu, 7 Jul 2022 19:20:06 +0800 Subject: ASoC: fsl_utils: Don't use plain integer as NULL pointer Fix sparse warning: sound/soc/fsl/fsl_utils.c:125:31: sparse: warning: Using plain integer as NULL pointer sound/soc/fsl/fsl_utils.c:125:42: sparse: warning: Using plain integer as NULL pointer Fixes: 7bad8125549c ("ASoC: fsl_utils: Add function to handle PLL clock source") Reported-by: kernel test robot Signed-off-by: Shengjiu Wang Link: https://lore.kernel.org/r/1657192806-10569-1-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_utils.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_utils.c b/sound/soc/fsl/fsl_utils.c index b75843e31f00..3e969c7bc1c5 100644 --- a/sound/soc/fsl/fsl_utils.c +++ b/sound/soc/fsl/fsl_utils.c @@ -122,7 +122,7 @@ void fsl_asoc_reparent_pll_clocks(struct device *dev, struct clk *clk, struct clk *pll8k_clk, struct clk *pll11k_clk, u64 ratio) { - struct clk *p, *pll = 0, *npll = 0; + struct clk *p, *pll = NULL, *npll = NULL; bool reparent = false; int ret = 0; -- cgit v1.2.3 From 817a62108dfacebd548e38451bf0e7eee023e97f Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 1 Jul 2022 05:18:14 +0000 Subject: ASoC: audio-graph-card2.c: use of_property_read_u32() for rate Audio Graph Card2 is using of_get_property(), but it should use of_property_read_u32() to getting rate. Otherwise the setting will be strange value. This patch fixup it. Fixes: c3a15c92a67b701 ("ASoC: audio-graph-card2: add Codec2Codec support") Signed-off-by: Kuninori Morimoto Link: https://lore.kernel.org/r/87h741s961.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/generic/audio-graph-card2.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/generic/audio-graph-card2.c b/sound/soc/generic/audio-graph-card2.c index d34b29a49268..8e0628e6f2a0 100644 --- a/sound/soc/generic/audio-graph-card2.c +++ b/sound/soc/generic/audio-graph-card2.c @@ -856,7 +856,7 @@ int audio_graph2_link_c2c(struct asoc_simple_priv *priv, struct device_node *port0, *port1, *ports; struct device_node *codec0_port, *codec1_port; struct device_node *ep0, *ep1; - u32 val; + u32 val = 0; int ret = -EINVAL; /* @@ -880,7 +880,8 @@ int audio_graph2_link_c2c(struct asoc_simple_priv *priv, ports = of_get_parent(port0); port1 = of_get_next_child(ports, lnk); - if (!of_get_property(ports, "rate", &val)) { + of_property_read_u32(ports, "rate", &val); + if (!val) { struct device *dev = simple_priv_to_dev(priv); dev_err(dev, "Codec2Codec needs rate settings\n"); -- cgit v1.2.3 From c2ff7f15a4ef74b8cb6d425dfa8d8b928f193a80 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 1 Jul 2022 05:18:21 +0000 Subject: ASoC: audio-graph-card2.c: make Codec2Codec settings optional Current audio-graph-card2 can use Codec2Codec, and having its original parameter (= rate) on DT is mandatory for now. But simple-card-utils.c has asoc_simple_init_for_codec2codec() to setup *default* Codec2Codec settings. This patch makes Audio Graph Card2 Codec2Codec rate settings optional. Signed-off-by: Kuninori Morimoto Link: https://lore.kernel.org/r/87fsjls95u.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- .../generic/audio-graph-card2-custom-sample.dtsi | 3 +- sound/soc/generic/audio-graph-card2.c | 36 +++++++++++++--------- 2 files changed, 23 insertions(+), 16 deletions(-) (limited to 'sound') diff --git a/sound/soc/generic/audio-graph-card2-custom-sample.dtsi b/sound/soc/generic/audio-graph-card2-custom-sample.dtsi index 8eee7b821ff7..053d987a1fec 100644 --- a/sound/soc/generic/audio-graph-card2-custom-sample.dtsi +++ b/sound/soc/generic/audio-graph-card2-custom-sample.dtsi @@ -154,11 +154,12 @@ codec2codec { ports@0 { - rate = <48000>; + /* use default settings */ c2c: port@0 { c2cf_ep: endpoint { remote-endpoint = <&codec6_ep>; }; }; port@1 { c2cb_ep: endpoint { remote-endpoint = <&codec7_ep>; }; }; }; ports@1 { + /* use original settings */ rate = <48000>; c2c_m: port@0 { c2cmf_ep: endpoint { remote-endpoint = <&mc2c0_ep>; }; }; port@1 { c2cmb_ep: endpoint { remote-endpoint = <&mc2c1_ep>; }; }; diff --git a/sound/soc/generic/audio-graph-card2.c b/sound/soc/generic/audio-graph-card2.c index 8e0628e6f2a0..510058c47a92 100644 --- a/sound/soc/generic/audio-graph-card2.c +++ b/sound/soc/generic/audio-graph-card2.c @@ -851,8 +851,6 @@ int audio_graph2_link_c2c(struct asoc_simple_priv *priv, struct link_info *li) { struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); - struct simple_dai_props *dai_props = simple_priv_to_props(priv, li->link); - struct snd_soc_pcm_stream *c2c_conf = dai_props->c2c_conf; struct device_node *port0, *port1, *ports; struct device_node *codec0_port, *codec1_port; struct device_node *ep0, *ep1; @@ -880,21 +878,30 @@ int audio_graph2_link_c2c(struct asoc_simple_priv *priv, ports = of_get_parent(port0); port1 = of_get_next_child(ports, lnk); + /* + * Card2 can use original Codec2Codec settings if DT has. + * It will use default settings if no settings on DT. + * see + * asoc_simple_init_for_codec2codec() + * + * Add more settings here if needed + */ of_property_read_u32(ports, "rate", &val); - if (!val) { - struct device *dev = simple_priv_to_dev(priv); - - dev_err(dev, "Codec2Codec needs rate settings\n"); - goto err1; + if (val) { + struct simple_dai_props *dai_props = simple_priv_to_props(priv, li->link); + struct snd_soc_pcm_stream *c2c_conf = dai_props->c2c_conf; + + c2c_conf->formats = SNDRV_PCM_FMTBIT_S32_LE; /* update ME */ + c2c_conf->rates = SNDRV_PCM_RATE_8000_384000; + c2c_conf->rate_min = + c2c_conf->rate_max = val; + c2c_conf->channels_min = + c2c_conf->channels_max = 2; /* update ME */ + + dai_link->params = c2c_conf; + dai_link->num_params = 1; } - c2c_conf->formats = SNDRV_PCM_FMTBIT_S32_LE; /* update ME */ - c2c_conf->rate_min = - c2c_conf->rate_max = val; - c2c_conf->channels_min = - c2c_conf->channels_max = 2; /* update ME */ - dai_link->params = c2c_conf; - ep0 = port_to_endpoint(port0); ep1 = port_to_endpoint(port1); @@ -923,7 +930,6 @@ err2: of_node_put(ep1); of_node_put(codec0_port); of_node_put(codec1_port); -err1: of_node_put(ports); of_node_put(port0); of_node_put(port1); -- cgit v1.2.3 From 6976ed0137d98c2ec0f11af8a01716e9f3af873d Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 1 Jul 2022 05:18:27 +0000 Subject: ASoC: audio-graph-card2.c: remove pre-alloced Codec2Codec space Because Codec2Codec settings becomes optional, we don't need to keep its parameter space when init time. This patch removes its default memory allocation from simple-card-utils.c, and allocate it at audio-graph-card2 ondemand. Signed-off-by: Kuninori Morimoto Link: https://lore.kernel.org/r/87edz5s95o.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/generic/audio-graph-card2.c | 10 +++++++--- sound/soc/generic/simple-card-utils.c | 18 +----------------- 2 files changed, 8 insertions(+), 20 deletions(-) (limited to 'sound') diff --git a/sound/soc/generic/audio-graph-card2.c b/sound/soc/generic/audio-graph-card2.c index 510058c47a92..19e31d53422a 100644 --- a/sound/soc/generic/audio-graph-card2.c +++ b/sound/soc/generic/audio-graph-card2.c @@ -888,8 +888,12 @@ int audio_graph2_link_c2c(struct asoc_simple_priv *priv, */ of_property_read_u32(ports, "rate", &val); if (val) { - struct simple_dai_props *dai_props = simple_priv_to_props(priv, li->link); - struct snd_soc_pcm_stream *c2c_conf = dai_props->c2c_conf; + struct device *dev = simple_priv_to_dev(priv); + struct snd_soc_pcm_stream *c2c_conf; + + c2c_conf = devm_kzalloc(dev, sizeof(*c2c_conf), GFP_KERNEL); + if (!c2c_conf) + goto err1; c2c_conf->formats = SNDRV_PCM_FMTBIT_S32_LE; /* update ME */ c2c_conf->rates = SNDRV_PCM_RATE_8000_384000; @@ -930,6 +934,7 @@ err2: of_node_put(ep1); of_node_put(codec0_port); of_node_put(codec1_port); +err1: of_node_put(ports); of_node_put(port0); of_node_put(port1); @@ -1093,7 +1098,6 @@ static int graph_count_c2c(struct asoc_simple_priv *priv, li->num[li->link].cpus = li->num[li->link].platforms = graph_counter(codec0); li->num[li->link].codecs = graph_counter(codec1); - li->num[li->link].c2c = 1; of_node_put(ports); of_node_put(port1); diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 7be84c7840cb..a761af6b13b6 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -746,8 +746,7 @@ int asoc_simple_init_priv(struct asoc_simple_priv *priv, struct asoc_simple_dai *dais; struct snd_soc_dai_link_component *dlcs; struct snd_soc_codec_conf *cconf = NULL; - struct snd_soc_pcm_stream *c2c_conf = NULL; - int i, dai_num = 0, dlc_num = 0, cnf_num = 0, c2c_num = 0; + int i, dai_num = 0, dlc_num = 0, cnf_num = 0; dai_props = devm_kcalloc(dev, li->link, sizeof(*dai_props), GFP_KERNEL); dai_link = devm_kcalloc(dev, li->link, sizeof(*dai_link), GFP_KERNEL); @@ -766,8 +765,6 @@ int asoc_simple_init_priv(struct asoc_simple_priv *priv, if (!li->num[i].cpus) cnf_num += li->num[i].codecs; - - c2c_num += li->num[i].c2c; } dais = devm_kcalloc(dev, dai_num, sizeof(*dais), GFP_KERNEL); @@ -781,12 +778,6 @@ int asoc_simple_init_priv(struct asoc_simple_priv *priv, return -ENOMEM; } - if (c2c_num) { - c2c_conf = devm_kcalloc(dev, c2c_num, sizeof(*c2c_conf), GFP_KERNEL); - if (!c2c_conf) - return -ENOMEM; - } - dev_dbg(dev, "link %d, dais %d, ccnf %d\n", li->link, dai_num, cnf_num); @@ -800,7 +791,6 @@ int asoc_simple_init_priv(struct asoc_simple_priv *priv, priv->dais = dais; priv->dlcs = dlcs; priv->codec_conf = cconf; - priv->c2c_conf = c2c_conf; card->dai_link = priv->dai_link; card->num_links = li->link; @@ -818,12 +808,6 @@ int asoc_simple_init_priv(struct asoc_simple_priv *priv, dlcs += li->num[i].cpus; dais += li->num[i].cpus; - - if (li->num[i].c2c) { - /* Codec2Codec */ - dai_props[i].c2c_conf = c2c_conf; - c2c_conf += li->num[i].c2c; - } } else { /* DPCM Be's CPU = dummy */ dai_props[i].cpus = -- cgit v1.2.3 From d33083f941150dc2079975d032547f6ce9a8e81b Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 1 Jul 2022 05:18:34 +0000 Subject: ASoC: audio-graph-card2-custom-sample.dtsi: add verbose explanation audio-graph-card2-custom-sample.dtsi will be used to test Audio-Graph-Card2 behavior. But it is difficult to say that it is easy to understand, because the comment/explanation are not so many. This patch add verbose explanation to it. Signed-off-by: Kuninori Morimoto Link: https://lore.kernel.org/r/87czeps95h.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- .../generic/audio-graph-card2-custom-sample.dtsi | 98 ++++++++++++++++++++-- 1 file changed, 90 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/generic/audio-graph-card2-custom-sample.dtsi b/sound/soc/generic/audio-graph-card2-custom-sample.dtsi index 053d987a1fec..fe547c18771f 100644 --- a/sound/soc/generic/audio-graph-card2-custom-sample.dtsi +++ b/sound/soc/generic/audio-graph-card2-custom-sample.dtsi @@ -17,6 +17,23 @@ * CONFIG_SND_AUDIO_GRAPH_CARD2 * CONFIG_SND_AUDIO_GRAPH_CARD2_CUSTOM_SAMPLE * CONFIG_SND_TEST_COMPONENT + * + * + * You can indicate more detail each device behavior as debug if you modify + * "compatible" on each test-component. see below + * + * test_cpu { + * - compatible = "test-cpu"; + * + compatible = "test-cpu-verbose"; + * ... + * }; + * + * test_codec { + * - compatible = "test-codec"; + * + compatible = "test-codec-verbose"; + * ... + * }; + * */ / { /* @@ -101,35 +118,74 @@ "TC OUT", "TC DAI11 Playback", "TC DAI9 Capture", "TC IN"; - links = <&cpu0 /* normal: cpu side only */ - &mcpu0 /* multi: cpu side only */ - &fe00 &fe01 &be0 /* dpcm: both FE / BE */ - &fe10 &fe11 &be1 /* dpcm-m: both FE / BE */ - &c2c /* c2c: cpu side only */ - &c2c_m /* c2c: cpu side only */ + links = < + /* + * [Normal]: cpu side only + * cpu0/codec0 + */ + &cpu0 + + /* + * [Multi-CPU/Codec]: cpu side only + * cpu1/cpu2/codec1/codec2 + */ + &mcpu0 + + /* + * [DPCM]: both FE / BE + * cpu3/cpu4/codec3 + */ + &fe00 &fe01 &be0 + + /* + * [DPCM-Multi]: both FE / BE + * cpu5/cpu6/codec4/codec5 + */ + &fe10 &fe11 &be1 + + /* + * [Codec2Codec]: cpu side only + * codec6/codec7 + */ + &c2c + + /* + * [Codec2Codec-Multi]: cpu side only + * codec8/codec9/codec10/codec11 + */ + &c2c_m >; multi { ports@0 { + /* [Multi-CPU] */ mcpu0: port@0 { mcpu0_ep: endpoint { remote-endpoint = <&mcodec0_ep>; }; }; port@1 { mcpu1_ep: endpoint { remote-endpoint = <&cpu1_ep>; }; }; port@2 { mcpu2_ep: endpoint { remote-endpoint = <&cpu2_ep>; }; }; }; + + /* [Multi-Codec] */ ports@1 { port@0 { mcodec0_ep: endpoint { remote-endpoint = <&mcpu0_ep>; }; }; port@1 { mcodec1_ep: endpoint { remote-endpoint = <&codec1_ep>; }; }; port@2 { mcodec2_ep: endpoint { remote-endpoint = <&codec2_ep>; }; }; }; + + /* [DPCM-Multi]::BE */ ports@2 { port@0 { mbe_ep: endpoint { remote-endpoint = <&be10_ep>; }; }; port@1 { mbe1_ep: endpoint { remote-endpoint = <&codec4_ep>; }; }; port@2 { mbe2_ep: endpoint { remote-endpoint = <&codec5_ep>; }; }; }; + + /* [Codec2Codec-Multi]::CPU */ ports@3 { port@0 { mc2c0_ep: endpoint { remote-endpoint = <&c2cmf_ep>; }; }; port@1 { mc2c00_ep: endpoint { remote-endpoint = <&codec8_ep>; }; }; port@2 { mc2c01_ep: endpoint { remote-endpoint = <&codec9_ep>; }; }; }; + + /* [Codec2Codec-Multi]::Codec */ ports@4 { port@0 { mc2c1_ep: endpoint { remote-endpoint = <&c2cmb_ep>; }; }; port@1 { mc2c10_ep: endpoint { remote-endpoint = <&codec10_ep>; }; }; @@ -138,26 +194,34 @@ }; dpcm { - /* FE */ ports@0 { + /* [DPCM]::FE */ fe00: port@0 { fe00_ep: endpoint { remote-endpoint = <&cpu3_ep>; }; }; fe01: port@1 { fe01_ep: endpoint { remote-endpoint = <&cpu4_ep>; }; }; + + /* [DPCM-Multi]::FE */ fe10: port@2 { fe10_ep: endpoint { remote-endpoint = <&cpu5_ep>; }; }; fe11: port@3 { fe11_ep: endpoint { remote-endpoint = <&cpu6_ep>; }; }; }; - /* BE */ + ports@1 { + /* [DPCM]::BE */ be0: port@0 { be00_ep: endpoint { remote-endpoint = <&codec3_ep>; }; }; + + /* [DPCM-Multi]::BE */ be1: port@1 { be10_ep: endpoint { remote-endpoint = <&mbe_ep>; }; }; }; }; codec2codec { + /* [Codec2Codec] */ ports@0 { /* use default settings */ c2c: port@0 { c2cf_ep: endpoint { remote-endpoint = <&codec6_ep>; }; }; port@1 { c2cb_ep: endpoint { remote-endpoint = <&codec7_ep>; }; }; }; + + /* [Codec2Codec-Multi] */ ports@1 { /* use original settings */ rate = <48000>; @@ -180,11 +244,18 @@ ports { bitclock-master; frame-master; + /* [Normal] */ cpu0: port@0 { cpu0_ep: endpoint { remote-endpoint = <&codec0_ep>; }; }; + + /* [Multi-CPU] */ port@1 { cpu1_ep: endpoint { remote-endpoint = <&mcpu1_ep>; }; }; port@2 { cpu2_ep: endpoint { remote-endpoint = <&mcpu2_ep>; }; }; + + /* [DPCM]::FE */ port@3 { cpu3_ep: endpoint { remote-endpoint = <&fe00_ep>; }; }; port@4 { cpu4_ep: endpoint { remote-endpoint = <&fe01_ep>; }; }; + + /* [DPCM-Multi]::FE */ port@5 { cpu5_ep: endpoint { remote-endpoint = <&fe10_ep>; }; }; port@6 { cpu6_ep: endpoint { remote-endpoint = <&fe11_ep>; }; }; }; @@ -207,16 +278,27 @@ */ prefix = "TC"; + /* [Normal] */ port@0 { codec0_ep: endpoint { remote-endpoint = <&cpu0_ep>; }; }; + + /* [Multi-Codec] */ port@1 { codec1_ep: endpoint { remote-endpoint = <&mcodec1_ep>; }; }; port@2 { codec2_ep: endpoint { remote-endpoint = <&mcodec2_ep>; }; }; + + /* [DPCM]::BE */ port@3 { codec3_ep: endpoint { remote-endpoint = <&be00_ep>; }; }; + + /* [DPCM-Multi]::BE */ port@4 { codec4_ep: endpoint { remote-endpoint = <&mbe1_ep>; }; }; port@5 { codec5_ep: endpoint { remote-endpoint = <&mbe2_ep>; }; }; + + /* [Codec2Codec] */ port@6 { bitclock-master; frame-master; codec6_ep: endpoint { remote-endpoint = <&c2cf_ep>; }; }; port@7 { codec7_ep: endpoint { remote-endpoint = <&c2cb_ep>; }; }; + + /* [Codec2Codec-Multi] */ port@8 { bitclock-master; frame-master; codec8_ep: endpoint { remote-endpoint = <&mc2c00_ep>; }; }; -- cgit v1.2.3 From 75d1b39067ed6699ec8a906fa9d83609bca9113b Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 1 Jul 2022 05:18:40 +0000 Subject: ASoC: simple-card-utils.c: ignore Codec2Codec setting if it already have Audio Graph Card2 setups own Codec2Codec settings, but current simple-card-utils.c will try to setup Codec2Codec default settings if needed, it will overwirtes the settings. This patch ignores default Codec2Codec settings if it already have. Signed-off-by: Kuninori Morimoto Link: https://lore.kernel.org/r/87bku9s95b.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/generic/simple-card-utils.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index a761af6b13b6..b8a3da692ee8 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -527,6 +527,10 @@ static int asoc_simple_init_for_codec2codec(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hardware hw; int i, ret, stream; + /* Do nothing if it already has Codec2Codec settings */ + if (dai_link->params) + return 0; + /* Only Codecs */ for_each_rtd_components(rtd, i, component) { if (!asoc_simple_component_is_codec(component)) -- cgit v1.2.3 From 16b7ba9c0f53032e2a9365f3de89b66426b5716c Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 1 Jul 2022 05:18:51 +0000 Subject: ASoC: simple-card-utils.c: care Codec2Codec vs DPCM:BE Current asoc_simple_init_for_codec2codec() adds default Codec2Codec settings if rtd was Codec only. But DPCM:BE also judged as Codec only, because dummy-DAI doesn't have "endianness" (which is key parameter to judge as Codec). This patch ignores setup Codec2Codec settings if it was DPCM:BE case. Signed-off-by: Kuninori Morimoto Link: https://lore.kernel.org/r/87a69ts950.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/generic/simple-card-utils.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index b8a3da692ee8..4a29e314fa95 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -531,6 +531,10 @@ static int asoc_simple_init_for_codec2codec(struct snd_soc_pcm_runtime *rtd, if (dai_link->params) return 0; + /* Do nothing if it was DPCM :: BE */ + if (dai_link->no_pcm) + return 0; + /* Only Codecs */ for_each_rtd_components(rtd, i, component) { if (!asoc_simple_component_is_codec(component)) -- cgit v1.2.3 From f460e3a9740b8c1ec5a9a034262674514bbbdcac Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 7 Jul 2022 16:46:14 -0500 Subject: ASoC: amd: acp-es8336: use static variables Sparse warnings: sound/soc/amd/acp-es8336.c:36:15: error: symbol 'codec_dev' was not declared. Should it be static? sound/soc/amd/acp-es8336.c:37:18: error: symbol 'gpio_pa' was not declared. Should it be static? Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220707214614.61081-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/amd/acp-es8336.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/amd/acp-es8336.c b/sound/soc/amd/acp-es8336.c index eec3d57092fa..4f3992532332 100644 --- a/sound/soc/amd/acp-es8336.c +++ b/sound/soc/amd/acp-es8336.c @@ -33,8 +33,8 @@ static unsigned long acp2x_machine_id; static struct snd_soc_jack st_jack; -struct device *codec_dev; -struct gpio_desc *gpio_pa; +static struct device *codec_dev; +static struct gpio_desc *gpio_pa; static int sof_es8316_speaker_power_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) -- cgit v1.2.3 From d7e5d8d24c1179b36a3cb40b3f785e23a8992acd Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Thu, 7 Jul 2022 18:56:08 +0530 Subject: ASoC: amd: remove unused header file inclusion Removed unused header file inclusion from Jadeite platform machine driver. Signed-off-by: Vijendar Mukunda Link: https://lore.kernel.org/r/20220707132613.3150931-1-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp-es8336.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/amd/acp-es8336.c b/sound/soc/amd/acp-es8336.c index 4f3992532332..54ba399fe596 100644 --- a/sound/soc/amd/acp-es8336.c +++ b/sound/soc/amd/acp-es8336.c @@ -23,7 +23,6 @@ #include #include -#include "../codecs/es8316.h" #include "acp.h" #define DUAL_CHANNEL 2 -- cgit v1.2.3 From 8d9cd3ead42a6d3bac131c4331acfa5244674fbb Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Thu, 7 Jul 2022 18:56:09 +0530 Subject: ASoC: amd: drop machine driver remove function Drop machine driver remove() function. Signed-off-by: Vijendar Mukunda Link: https://lore.kernel.org/r/20220707132613.3150931-2-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp-es8336.c | 6 ------ 1 file changed, 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/amd/acp-es8336.c b/sound/soc/amd/acp-es8336.c index 54ba399fe596..a9997627f991 100644 --- a/sound/soc/amd/acp-es8336.c +++ b/sound/soc/amd/acp-es8336.c @@ -293,11 +293,6 @@ static int st_es8336_probe(struct platform_device *pdev) return 0; } -static int st_es8336_remove(struct platform_device *pdev) -{ - return 0; -} - #ifdef CONFIG_ACPI static const struct acpi_device_id st_audio_acpi_match[] = { {"AMDI8336", 0}, @@ -313,7 +308,6 @@ static struct platform_driver st_mach_driver = { .pm = &snd_soc_pm_ops, }, .probe = st_es8336_probe, - .remove = st_es8336_remove, }; module_platform_driver(st_mach_driver); -- cgit v1.2.3 From 0de876c125188e502d2899de4bcba8d4a6b1f98c Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Thu, 7 Jul 2022 18:56:10 +0530 Subject: ASoC: amd: fix for variable set but not used warning Fix below kernel warning. >>> sound/soc/amd/acp-es8336.c:200:13: warning: variable 'ret' set but >>> not used [-Wunused-but-set-variable] Signed-off-by: Vijendar Mukunda Reported-by: kernel test robot Link: https://lore.kernel.org/r/20220707132613.3150931-3-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp-es8336.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/amd/acp-es8336.c b/sound/soc/amd/acp-es8336.c index a9997627f991..ebd4fa9f1f00 100644 --- a/sound/soc/amd/acp-es8336.c +++ b/sound/soc/amd/acp-es8336.c @@ -206,6 +206,8 @@ static int st_es8336_late_probe(struct snd_soc_card *card) dev_err(card->dev, "can not find codec dev\n"); ret = devm_acpi_dev_add_driver_gpios(codec_dev, acpi_es8336_gpios); + if (ret) + dev_warn(card->dev, "Failed to add driver gpios\n"); gpio_pa = gpiod_get_optional(codec_dev, "pa-enable", GPIOD_OUT_LOW); if (IS_ERR(gpio_pa)) { @@ -213,6 +215,7 @@ static int st_es8336_late_probe(struct snd_soc_card *card) "could not get pa-enable GPIO\n"); gpiod_put(gpio_pa); put_device(codec_dev); + return ret; } return 0; } -- cgit v1.2.3 From eae9f9ce181be4f47dcba1ee93185b71cac3f312 Mon Sep 17 00:00:00 2001 From: Raphael-Xu <13691752556@139.com> Date: Thu, 7 Jul 2022 20:33:42 +0800 Subject: ASoC: add tas2780 driver 1.update Kconfig and Makefile 2.add tas2780.c and tas2780.h Signed-off-by: Raphael-Xu <13691752556@139.com> Link: https://lore.kernel.org/r/20220707123343.2403-1-13691752556@139.com Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas2780.c | 663 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tas2780.h | 101 +++++++ 4 files changed, 774 insertions(+) create mode 100644 sound/soc/codecs/tas2780.c create mode 100644 sound/soc/codecs/tas2780.h (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index ee7e028e8402..d16b4efb88a7 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -219,6 +219,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_TAS2562 imply SND_SOC_TAS2764 imply SND_SOC_TAS2770 + imply SND_SOC_TAS2780 imply SND_SOC_TAS5086 imply SND_SOC_TAS571X imply SND_SOC_TAS5720 @@ -1535,6 +1536,13 @@ config SND_SOC_TAS2770 tristate "Texas Instruments TAS2770 speaker amplifier" depends on I2C +config SND_SOC_TAS2780 + tristate "Texas Instruments TAS2780 Mono Audio amplifier" + depends on I2C + help + Enable support for Texas Instruments TAS2780 high-efficiency + digital input mono Class-D audio power amplifiers. + config SND_SOC_TAS5086 tristate "Texas Instruments TAS5086 speaker amplifier" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 60354579fe5c..92fd441d426a 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -348,6 +348,7 @@ snd-soc-tpa6130a2-objs := tpa6130a2.o snd-soc-tas2552-objs := tas2552.o snd-soc-tas2562-objs := tas2562.o snd-soc-tas2764-objs := tas2764.o +snd-soc-tas2780-objs := tas2780.o # Mux snd-soc-simple-mux-objs := simple-mux.o @@ -592,6 +593,7 @@ obj-$(CONFIG_SND_SOC_STI_SAS) += snd-soc-sti-sas.o obj-$(CONFIG_SND_SOC_TAS2552) += snd-soc-tas2552.o obj-$(CONFIG_SND_SOC_TAS2562) += snd-soc-tas2562.o obj-$(CONFIG_SND_SOC_TAS2764) += snd-soc-tas2764.o +obj-$(CONFIG_SND_SOC_TAS2780) += snd-soc-tas2780.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TAS571X) += snd-soc-tas571x.o obj-$(CONFIG_SND_SOC_TAS5720) += snd-soc-tas5720.o diff --git a/sound/soc/codecs/tas2780.c b/sound/soc/codecs/tas2780.c new file mode 100644 index 000000000000..a6db6f0e5431 --- /dev/null +++ b/sound/soc/codecs/tas2780.c @@ -0,0 +1,663 @@ +// SPDX-License-Identifier: GPL-2.0 +// Driver for the Texas Instruments TAS2780 Mono +// Audio amplifier +// Copyright (C) 2022 Texas Instruments Inc. + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "tas2780.h" + +struct tas2780_priv { + struct snd_soc_component *component; + struct gpio_desc *reset_gpio; + struct regmap *regmap; + struct device *dev; + int v_sense_slot; + int i_sense_slot; +}; + +static void tas2780_reset(struct tas2780_priv *tas2780) +{ + int ret = 0; + + if (tas2780->reset_gpio) { + gpiod_set_value_cansleep(tas2780->reset_gpio, 0); + usleep_range(2000, 2050); + gpiod_set_value_cansleep(tas2780->reset_gpio, 1); + usleep_range(2000, 2050); + } + + snd_soc_component_write(tas2780->component, TAS2780_SW_RST, + TAS2780_RST); + if (ret) + dev_err(tas2780->dev, "%s:errCode:0x%x Reset error!\n", + __func__, ret); +} + +#ifdef CONFIG_PM +static int tas2780_codec_suspend(struct snd_soc_component *component) +{ + struct tas2780_priv *tas2780 = + snd_soc_component_get_drvdata(component); + int ret = 0; + + ret = snd_soc_component_update_bits(component, TAS2780_PWR_CTRL, + TAS2780_PWR_CTRL_MASK, TAS2780_PWR_CTRL_SHUTDOWN); + if (ret < 0) { + dev_err(tas2780->dev, "%s:errCode:0x%0x:power down error\n", + __func__, ret); + goto err; + } + ret = 0; + regcache_cache_only(tas2780->regmap, true); + regcache_mark_dirty(tas2780->regmap); +err: + return ret; +} + +static int tas2780_codec_resume(struct snd_soc_component *component) +{ + struct tas2780_priv *tas2780 = + snd_soc_component_get_drvdata(component); + int ret = 0; + + ret = snd_soc_component_update_bits(component, TAS2780_PWR_CTRL, + TAS2780_PWR_CTRL_MASK, TAS2780_PWR_CTRL_ACTIVE); + + if (ret < 0) { + dev_err(tas2780->dev, "%s:errCode:0x%0x:power down error\n", + __func__, ret); + goto err; + } + ret = 0; + regcache_cache_only(tas2780->regmap, false); + ret = regcache_sync(tas2780->regmap); +err: + return ret; +} +#endif + +static const char * const tas2780_ASI1_src[] = { + "I2C offset", "Left", "Right", "LeftRightDiv2", +}; + +static SOC_ENUM_SINGLE_DECL( + tas2780_ASI1_src_enum, TAS2780_TDM_CFG2, 4, tas2780_ASI1_src); + +static const struct snd_kcontrol_new tas2780_asi1_mux = + SOC_DAPM_ENUM("ASI1 Source", tas2780_ASI1_src_enum); + +static const struct snd_kcontrol_new isense_switch = + SOC_DAPM_SINGLE("Switch", TAS2780_PWR_CTRL, + TAS2780_ISENSE_POWER_EN, 1, 1); +static const struct snd_kcontrol_new vsense_switch = + SOC_DAPM_SINGLE("Switch", TAS2780_PWR_CTRL, + TAS2780_VSENSE_POWER_EN, 1, 1); + +static const struct snd_soc_dapm_widget tas2780_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("ASI1", "ASI1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_MUX("ASI1 Sel", SND_SOC_NOPM, 0, 0, &tas2780_asi1_mux), + SND_SOC_DAPM_SWITCH("ISENSE", TAS2780_PWR_CTRL, + TAS2780_ISENSE_POWER_EN, 1, &isense_switch), + SND_SOC_DAPM_SWITCH("VSENSE", TAS2780_PWR_CTRL, + TAS2780_VSENSE_POWER_EN, 1, &vsense_switch), + SND_SOC_DAPM_OUTPUT("OUT"), + SND_SOC_DAPM_SIGGEN("VMON"), + SND_SOC_DAPM_SIGGEN("IMON") +}; + +static const struct snd_soc_dapm_route tas2780_audio_map[] = { + {"ASI1 Sel", "I2C offset", "ASI1"}, + {"ASI1 Sel", "Left", "ASI1"}, + {"ASI1 Sel", "Right", "ASI1"}, + {"ASI1 Sel", "LeftRightDiv2", "ASI1"}, + {"OUT", NULL, "ASI1 Sel"}, + {"ISENSE", "Switch", "IMON"}, + {"VSENSE", "Switch", "VMON"}, +}; + +static int tas2780_mute(struct snd_soc_dai *dai, int mute, int direction) +{ + struct snd_soc_component *component = dai->component; + struct tas2780_priv *tas2780 = + snd_soc_component_get_drvdata(component); + int ret = 0; + + ret = snd_soc_component_update_bits(component, TAS2780_PWR_CTRL, + TAS2780_PWR_CTRL_MASK, + mute ? TAS2780_PWR_CTRL_MUTE : 0); + if (ret < 0) { + dev_err(tas2780->dev, "%s: Failed to set powercontrol\n", + __func__); + goto err; + } + ret = 0; +err: + return ret; +} + +static int tas2780_set_bitwidth(struct tas2780_priv *tas2780, int bitwidth) +{ + struct snd_soc_component *component = tas2780->component; + int sense_en; + int val; + int ret; + int slot_size; + + switch (bitwidth) { + case SNDRV_PCM_FORMAT_S16_LE: + ret = snd_soc_component_update_bits(component, + TAS2780_TDM_CFG2, + TAS2780_TDM_CFG2_RXW_MASK, + TAS2780_TDM_CFG2_RXW_16BITS); + slot_size = TAS2780_TDM_CFG2_RXS_16BITS; + break; + case SNDRV_PCM_FORMAT_S24_LE: + ret = snd_soc_component_update_bits(component, + TAS2780_TDM_CFG2, + TAS2780_TDM_CFG2_RXW_MASK, + TAS2780_TDM_CFG2_RXW_24BITS); + slot_size = TAS2780_TDM_CFG2_RXS_24BITS; + break; + case SNDRV_PCM_FORMAT_S32_LE: + ret = snd_soc_component_update_bits(component, + TAS2780_TDM_CFG2, + TAS2780_TDM_CFG2_RXW_MASK, + TAS2780_TDM_CFG2_RXW_32BITS); + slot_size = TAS2780_TDM_CFG2_RXS_32BITS; + break; + + default: + ret = -EINVAL; + } + + if (ret < 0) { + dev_err(tas2780->dev, "%s:errCode:0x%x set bitwidth error\n", + __func__, ret); + goto err; + } + + ret = snd_soc_component_update_bits(component, TAS2780_TDM_CFG2, + TAS2780_TDM_CFG2_RXS_MASK, slot_size); + if (ret < 0) { + dev_err(tas2780->dev, + "%s:errCode:0x%x set RX slot size error\n", + __func__, ret); + goto err; + } + + val = snd_soc_component_read(tas2780->component, TAS2780_PWR_CTRL); + if (val < 0) { + dev_err(tas2780->dev, "%s:errCode:0x%x read PWR_CTRL error\n", + __func__, val); + ret = val; + goto err; + } + + if (val & (1 << TAS2780_VSENSE_POWER_EN)) + sense_en = 0; + else + sense_en = TAS2780_TDM_CFG5_VSNS_ENABLE; + + ret = snd_soc_component_update_bits(tas2780->component, + TAS2780_TDM_CFG5, TAS2780_TDM_CFG5_VSNS_ENABLE, sense_en); + if (ret < 0) { + dev_err(tas2780->dev, "%s:errCode:0x%x enable vSNS error\n", + __func__, ret); + goto err; + } + + if (val & (1 << TAS2780_ISENSE_POWER_EN)) + sense_en = 0; + else + sense_en = TAS2780_TDM_CFG6_ISNS_ENABLE; + + ret = snd_soc_component_update_bits(tas2780->component, + TAS2780_TDM_CFG6, TAS2780_TDM_CFG6_ISNS_ENABLE, sense_en); + if (ret < 0) { + dev_err(tas2780->dev, "%s:errCode:0x%x enable iSNS error\n", + __func__, ret); + goto err; + } + ret = 0; +err: + return ret; +} + +static int tas2780_set_samplerate( + struct tas2780_priv *tas2780, int samplerate) +{ + struct snd_soc_component *component = tas2780->component; + int ramp_rate_val; + int ret; + + switch (samplerate) { + case 48000: + ramp_rate_val = TAS2780_TDM_CFG0_SMP_48KHZ | + TAS2780_TDM_CFG0_44_1_48KHZ; + break; + case 44100: + ramp_rate_val = TAS2780_TDM_CFG0_SMP_44_1KHZ | + TAS2780_TDM_CFG0_44_1_48KHZ; + break; + case 96000: + ramp_rate_val = TAS2780_TDM_CFG0_SMP_48KHZ | + TAS2780_TDM_CFG0_88_2_96KHZ; + break; + case 88200: + ramp_rate_val = TAS2780_TDM_CFG0_SMP_44_1KHZ | + TAS2780_TDM_CFG0_88_2_96KHZ; + break; + default: + return -EINVAL; + } + ret = snd_soc_component_update_bits(component, TAS2780_TDM_CFG0, + TAS2780_TDM_CFG0_SMP_MASK | TAS2780_TDM_CFG0_MASK, + ramp_rate_val); + if (ret < 0) { + dev_err(tas2780->dev, + "%s:errCode:0x%x Failed to set ramp_rate_val\n", + __func__, ret); + goto err; + } + ret = 0; +err: + return ret; +} + +static int tas2780_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct tas2780_priv *tas2780 = + snd_soc_component_get_drvdata(component); + int ret; + + ret = tas2780_set_bitwidth(tas2780, params_format(params)); + if (ret < 0) + return ret; + + return tas2780_set_samplerate(tas2780, params_rate(params)); +} + +static int tas2780_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_component *component = dai->component; + struct tas2780_priv *tas2780 = + snd_soc_component_get_drvdata(component); + u8 tdm_rx_start_slot = 0, asi_cfg_1 = 0; + int iface; + int ret = 0; + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + asi_cfg_1 = TAS2780_TDM_CFG1_RX_RISING; + break; + case SND_SOC_DAIFMT_IB_NF: + asi_cfg_1 = TAS2780_TDM_CFG1_RX_FALLING; + break; + default: + dev_err(tas2780->dev, "ASI format Inverse is not found\n"); + return -EINVAL; + } + + ret = snd_soc_component_update_bits(component, TAS2780_TDM_CFG1, + TAS2780_TDM_CFG1_RX_MASK, asi_cfg_1); + if (ret < 0) { + dev_err(tas2780->dev, + "%s:errCode:0x%x Failed to set asi_cfg_1\n", + __func__, ret); + goto err; + } + + if (((fmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S) + || ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) + == SND_SOC_DAIFMT_DSP_A)){ + iface = TAS2780_TDM_CFG2_SCFG_I2S; + tdm_rx_start_slot = 1; + } else { + if (((fmt & SND_SOC_DAIFMT_FORMAT_MASK) + == SND_SOC_DAIFMT_DSP_B) + || ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) + == SND_SOC_DAIFMT_LEFT_J)) { + iface = TAS2780_TDM_CFG2_SCFG_LEFT_J; + tdm_rx_start_slot = 0; + } else { + dev_err(tas2780->dev, + "%s:DAI Format is not found, fmt=0x%x\n", + __func__, fmt); + ret = -EINVAL; + goto err; + } + } + ret = snd_soc_component_update_bits(component, TAS2780_TDM_CFG1, + TAS2780_TDM_CFG1_MASK, + (tdm_rx_start_slot << TAS2780_TDM_CFG1_51_SHIFT)); + if (ret < 0) { + dev_err(tas2780->dev, + "%s:errCode:0x%x Failed to set tdm_rx_start_slot\n", + __func__, ret); + goto err; + } + + ret = snd_soc_component_update_bits(component, TAS2780_TDM_CFG2, + TAS2780_TDM_CFG2_SCFG_MASK, iface); + if (ret < 0) { + dev_err(tas2780->dev, "%s:errCode:0x%x Failed to set iface\n", + __func__, ret); + goto err; + } + ret = 0; +err: + return ret; +} + +static int tas2780_set_dai_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, + unsigned int rx_mask, + int slots, int slot_width) +{ + struct snd_soc_component *component = dai->component; + struct tas2780_priv *tas2780 = + snd_soc_component_get_drvdata(component); + int left_slot, right_slot; + int slots_cfg; + int slot_size; + int ret = 0; + + if (tx_mask == 0 || rx_mask != 0) + return -EINVAL; + + if (slots == 1) { + if (tx_mask != 1) + return -EINVAL; + left_slot = 0; + right_slot = 0; + } else { + left_slot = __ffs(tx_mask); + tx_mask &= ~(1 << left_slot); + if (tx_mask == 0) { + right_slot = left_slot; + } else { + right_slot = __ffs(tx_mask); + tx_mask &= ~(1 << right_slot); + } + } + + if (tx_mask != 0 || left_slot >= slots || right_slot >= slots) + return -EINVAL; + + slots_cfg = (right_slot << TAS2780_TDM_CFG3_RXS_SHIFT) | left_slot; + ret = snd_soc_component_write(component, TAS2780_TDM_CFG3, slots_cfg); + if (ret) { + dev_err(tas2780->dev, + "%s:errCode:0x%x Failed to set slots_cfg\n", + __func__, ret); + goto err; + } + + switch (slot_width) { + case 16: + slot_size = TAS2780_TDM_CFG2_RXS_16BITS; + break; + case 24: + slot_size = TAS2780_TDM_CFG2_RXS_24BITS; + break; + case 32: + slot_size = TAS2780_TDM_CFG2_RXS_32BITS; + break; + default: + ret = -EINVAL; + goto err; + } + + ret = snd_soc_component_update_bits(component, TAS2780_TDM_CFG2, + TAS2780_TDM_CFG2_RXS_MASK, slot_size); + if (ret < 0) { + dev_err(tas2780->dev, + "%s:errCode:0x%x Failed to set slot_size\n", + __func__, ret); + goto err; + } + + ret = snd_soc_component_update_bits(component, TAS2780_TDM_CFG5, + TAS2780_TDM_CFG5_50_MASK, tas2780->v_sense_slot); + if (ret < 0) { + dev_err(tas2780->dev, + "%s:errCode:0x%x Failed to set v_sense_slot\n", + __func__, ret); + goto err; + } + + ret = snd_soc_component_update_bits(component, TAS2780_TDM_CFG6, + TAS2780_TDM_CFG6_50_MASK, tas2780->i_sense_slot); + if (ret < 0) { + dev_err(tas2780->dev, + "%s:errCode:0x%x Failed to set i_sense_slot\n", + __func__, ret); + goto err; + } + ret = 0; +err: + return ret; +} + +static const struct snd_soc_dai_ops tas2780_dai_ops = { + .mute_stream = tas2780_mute, + .hw_params = tas2780_hw_params, + .set_fmt = tas2780_set_fmt, + .set_tdm_slot = tas2780_set_dai_tdm_slot, + .no_capture_mute = 1, +}; + +#define TAS2780_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +#define TAS2780_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_88200) + +static struct snd_soc_dai_driver tas2780_dai_driver[] = { + { + .name = "tas2780 ASI1", + .id = 0, + .playback = { + .stream_name = "ASI1 Playback", + .channels_min = 2, + .channels_max = 2, + .rates = TAS2780_RATES, + .formats = TAS2780_FORMATS, + }, + .capture = { + .stream_name = "ASI1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = TAS2780_RATES, + .formats = TAS2780_FORMATS, + }, + .ops = &tas2780_dai_ops, + .symmetric_rate = 1, + }, +}; + +static int tas2780_codec_probe(struct snd_soc_component *component) +{ + struct tas2780_priv *tas2780 = + snd_soc_component_get_drvdata(component); + int ret = 0; + + tas2780->component = component; + + tas2780_reset(tas2780); + ret = snd_soc_component_update_bits(component, + TAS2780_IC_CFG, TAS2780_IC_CFG_MASK, + TAS2780_IC_CFG_ENABLE); + if (ret < 0) + dev_err(tas2780->dev, "%s:errCode:0x%0x\n", + __func__, ret); + + return ret; +} + +static DECLARE_TLV_DB_SCALE(tas2780_digital_tlv, 1100, 50, 0); +static DECLARE_TLV_DB_SCALE(tas2780_playback_volume, -10000, 50, 0); + +static const struct snd_kcontrol_new tas2780_snd_controls[] = { + SOC_SINGLE_TLV("Speaker Volume", TAS2780_DVC, 0, + TAS2780_DVC_MAX, 1, tas2780_playback_volume), + SOC_SINGLE_TLV("Amp Gain Volume", TAS2780_CHNL_0, 0, 0x14, 0, + tas2780_digital_tlv), +}; + +static const struct snd_soc_component_driver soc_component_driver_tas2780 = { + .probe = tas2780_codec_probe, +#ifdef CONFIG_PM + .suspend = tas2780_codec_suspend, + .resume = tas2780_codec_resume, +#endif + .controls = tas2780_snd_controls, + .num_controls = ARRAY_SIZE(tas2780_snd_controls), + .dapm_widgets = tas2780_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tas2780_dapm_widgets), + .dapm_routes = tas2780_audio_map, + .num_dapm_routes = ARRAY_SIZE(tas2780_audio_map), + .idle_bias_on = 1, + .endianness = 1, +}; + +static const struct reg_default tas2780_reg_defaults[] = { + { TAS2780_PAGE, 0x00 }, + { TAS2780_SW_RST, 0x00 }, + { TAS2780_PWR_CTRL, 0x1a }, + { TAS2780_DVC, 0x00 }, + { TAS2780_CHNL_0, 0x00 }, + { TAS2780_TDM_CFG0, 0x09 }, + { TAS2780_TDM_CFG1, 0x02 }, + { TAS2780_TDM_CFG2, 0x0a }, + { TAS2780_TDM_CFG3, 0x10 }, + { TAS2780_TDM_CFG5, 0x42 }, +}; + +static const struct regmap_range_cfg tas2780_regmap_ranges[] = { + { + .range_min = 0, + .range_max = 1 * 128, + .selector_reg = TAS2780_PAGE, + .selector_mask = 0xff, + .selector_shift = 0, + .window_start = 0, + .window_len = 128, + }, +}; + +static const struct regmap_config tas2780_i2c_regmap = { + .reg_bits = 8, + .val_bits = 8, + .reg_defaults = tas2780_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(tas2780_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .ranges = tas2780_regmap_ranges, + .num_ranges = ARRAY_SIZE(tas2780_regmap_ranges), + .max_register = 1 * 128, +}; + +static int tas2780_parse_dt(struct device *dev, struct tas2780_priv *tas2780) +{ + int ret = 0; + + tas2780->reset_gpio = devm_gpiod_get_optional(tas2780->dev, "reset", + GPIOD_OUT_HIGH); + if (IS_ERR(tas2780->reset_gpio)) { + if (PTR_ERR(tas2780->reset_gpio) == -EPROBE_DEFER) { + tas2780->reset_gpio = NULL; + return -EPROBE_DEFER; + } + } + + ret = fwnode_property_read_u32(dev->fwnode, "ti,imon-slot-no", + &tas2780->i_sense_slot); + if (ret) + tas2780->i_sense_slot = 0; + + ret = fwnode_property_read_u32(dev->fwnode, "ti,vmon-slot-no", + &tas2780->v_sense_slot); + if (ret) + tas2780->v_sense_slot = 2; + + return 0; +} + +static int tas2780_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct tas2780_priv *tas2780; + int result; + + tas2780 = devm_kzalloc(&client->dev, sizeof(struct tas2780_priv), + GFP_KERNEL); + if (!tas2780) + return -ENOMEM; + tas2780->dev = &client->dev; + i2c_set_clientdata(client, tas2780); + dev_set_drvdata(&client->dev, tas2780); + + tas2780->regmap = devm_regmap_init_i2c(client, &tas2780_i2c_regmap); + if (IS_ERR(tas2780->regmap)) { + result = PTR_ERR(tas2780->regmap); + dev_err(&client->dev, "Failed to allocate register map: %d\n", + result); + return result; + } + + if (client->dev.of_node) { + result = tas2780_parse_dt(&client->dev, tas2780); + if (result) { + dev_err(tas2780->dev, + "%s: Failed to parse devicetree\n", __func__); + return result; + } + } + + return devm_snd_soc_register_component(tas2780->dev, + &soc_component_driver_tas2780, tas2780_dai_driver, + ARRAY_SIZE(tas2780_dai_driver)); +} + +static const struct i2c_device_id tas2780_i2c_id[] = { + { "tas2780", 0}, + { } +}; +MODULE_DEVICE_TABLE(i2c, tas2780_i2c_id); + +#if defined(CONFIG_OF) +static const struct of_device_id tas2780_of_match[] = { + { .compatible = "ti,tas2780" }, + {}, +}; +MODULE_DEVICE_TABLE(of, tas2780_of_match); +#endif + +static struct i2c_driver tas2780_i2c_driver = { + .driver = { + .name = "tas2780", + .of_match_table = of_match_ptr(tas2780_of_match), + }, + .probe = tas2780_i2c_probe, + .id_table = tas2780_i2c_id, +}; +module_i2c_driver(tas2780_i2c_driver); + +MODULE_AUTHOR("Raphael Xu "); +MODULE_DESCRIPTION("TAS2780 I2C Smart Amplifier driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tas2780.h b/sound/soc/codecs/tas2780.h new file mode 100644 index 000000000000..661c25df4e29 --- /dev/null +++ b/sound/soc/codecs/tas2780.h @@ -0,0 +1,101 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * TAS2780.h - ALSA SoC Texas Instruments TAS2780 Mono Audio Amplifier + * + * Copyright (C) 2020-2022 Texas Instruments Incorporated - https://www.ti.com + * + * Author: Raphael Xu + */ + +#ifndef __TAS2780_H__ +#define __TAS2780_H__ + +/* Book Control Register */ +#define TAS2780_BOOKCTL_PAGE 0 +#define TAS2780_BOOKCTL_REG 127 +#define TAS2780_REG(page, reg) ((page * 128) + reg) + +/* Page */ +#define TAS2780_PAGE TAS2780_REG(0X0, 0x00) +#define TAS2780_PAGE_PAGE_MASK 255 + +/* Software Reset */ +#define TAS2780_SW_RST TAS2780_REG(0X0, 0x01) +#define TAS2780_RST BIT(0) + +/* Power Control */ +#define TAS2780_PWR_CTRL TAS2780_REG(0X0, 0x02) +#define TAS2780_PWR_CTRL_MASK GENMASK(1, 0) +#define TAS2780_PWR_CTRL_ACTIVE 0x0 +#define TAS2780_PWR_CTRL_MUTE BIT(0) +#define TAS2780_PWR_CTRL_SHUTDOWN BIT(1) + +#define TAS2780_VSENSE_POWER_EN 3 +#define TAS2780_ISENSE_POWER_EN 4 + +/* Digital Volume Control */ +#define TAS2780_DVC TAS2780_REG(0X0, 0x1a) +#define TAS2780_DVC_MAX 0xc9 + +#define TAS2780_CHNL_0 TAS2780_REG(0X0, 0x03) + +/* TDM Configuration Reg0 */ +#define TAS2780_TDM_CFG0 TAS2780_REG(0X0, 0x08) +#define TAS2780_TDM_CFG0_SMP_MASK BIT(5) +#define TAS2780_TDM_CFG0_SMP_48KHZ 0x0 +#define TAS2780_TDM_CFG0_SMP_44_1KHZ BIT(5) +#define TAS2780_TDM_CFG0_MASK GENMASK(3, 1) +#define TAS2780_TDM_CFG0_44_1_48KHZ BIT(3) +#define TAS2780_TDM_CFG0_88_2_96KHZ (BIT(3) | BIT(1)) + +/* TDM Configuration Reg1 */ +#define TAS2780_TDM_CFG1 TAS2780_REG(0X0, 0x09) +#define TAS2780_TDM_CFG1_MASK GENMASK(5, 1) +#define TAS2780_TDM_CFG1_51_SHIFT 1 +#define TAS2780_TDM_CFG1_RX_MASK BIT(0) +#define TAS2780_TDM_CFG1_RX_RISING 0x0 +#define TAS2780_TDM_CFG1_RX_FALLING BIT(0) + +/* TDM Configuration Reg2 */ +#define TAS2780_TDM_CFG2 TAS2780_REG(0X0, 0x0a) +#define TAS2780_TDM_CFG2_RXW_MASK GENMASK(3, 2) +#define TAS2780_TDM_CFG2_RXW_16BITS 0x0 +#define TAS2780_TDM_CFG2_RXW_24BITS BIT(3) +#define TAS2780_TDM_CFG2_RXW_32BITS (BIT(3) | BIT(2)) +#define TAS2780_TDM_CFG2_RXS_MASK GENMASK(1, 0) +#define TAS2780_TDM_CFG2_RXS_16BITS 0x0 +#define TAS2780_TDM_CFG2_RXS_24BITS BIT(0) +#define TAS2780_TDM_CFG2_RXS_32BITS BIT(1) +#define TAS2780_TDM_CFG2_SCFG_MASK GENMASK(5, 4) +#define TAS2780_TDM_CFG2_SCFG_I2S 0x0 +#define TAS2780_TDM_CFG2_SCFG_LEFT_J BIT(4) +#define TAS2780_TDM_CFG2_SCFG_RIGHT_J BIT(5) + +/* TDM Configuration Reg3 */ +#define TAS2780_TDM_CFG3 TAS2780_REG(0X0, 0x0c) +#define TAS2780_TDM_CFG3_RXS_MASK GENMASK(7, 4) +#define TAS2780_TDM_CFG3_RXS_SHIFT 0x4 +#define TAS2780_TDM_CFG3_MASK GENMASK(3, 0) + +/* TDM Configuration Reg4 */ +#define TAS2780_TDM_CFG4 TAS2780_REG(0X0, 0x0d) +#define TAS2780_TDM_CFG4_TX_OFFSET_MASK GENMASK(3, 1) + +/* TDM Configuration Reg5 */ +#define TAS2780_TDM_CFG5 TAS2780_REG(0X0, 0x0e) +#define TAS2780_TDM_CFG5_VSNS_MASK BIT(6) +#define TAS2780_TDM_CFG5_VSNS_ENABLE BIT(6) +#define TAS2780_TDM_CFG5_50_MASK GENMASK(5, 0) + +/* TDM Configuration Reg6 */ +#define TAS2780_TDM_CFG6 TAS2780_REG(0X0, 0x0f) +#define TAS2780_TDM_CFG6_ISNS_MASK BIT(6) +#define TAS2780_TDM_CFG6_ISNS_ENABLE BIT(6) +#define TAS2780_TDM_CFG6_50_MASK GENMASK(5, 0) + +/* IC CFG */ +#define TAS2780_IC_CFG TAS2780_REG(0X0, 0x5c) +#define TAS2780_IC_CFG_MASK GENMASK(7, 6) +#define TAS2780_IC_CFG_ENABLE (BIT(7) | BIT(6)) + +#endif /* __TAS2780_H__ */ -- cgit v1.2.3 From 657efd9c985255960cdd90bafc382a39dc303277 Mon Sep 17 00:00:00 2001 From: Yang Li Date: Fri, 8 Jul 2022 07:25:40 +0800 Subject: ASoC: amd: Remove duplicated include in acp-es8336.c Fix following includecheck warning: ./sound/soc/amd/acp-es8336.c: linux/module.h is included more than once. Signed-off-by: Yang Li Link: https://lore.kernel.org/r/20220707232540.22589-1-yang.lee@linux.alibaba.com Signed-off-by: Mark Brown --- sound/soc/amd/acp-es8336.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/amd/acp-es8336.c b/sound/soc/amd/acp-es8336.c index ebd4fa9f1f00..d501618b78f6 100644 --- a/sound/soc/amd/acp-es8336.c +++ b/sound/soc/amd/acp-es8336.c @@ -20,7 +20,6 @@ #include #include #include -#include #include #include "acp.h" -- cgit v1.2.3 From 0ca3d2ba1dfd110bf5e0b25ebeb8f1e1587598fb Mon Sep 17 00:00:00 2001 From: David Lin Date: Fri, 8 Jul 2022 13:46:48 +0800 Subject: ASoC: nau8825: Declare 2 channels for DAI of capture stream The patch is to make driver with flexibility for more platforms support even if the internal design is just one ADC. Besides, many I2S controllers only support 2 channels. Signed-off-by: David Lin Link: https://lore.kernel.org/r/20220708054647.540621-1-CTLIN0@nuvoton.com Signed-off-by: Mark Brown --- sound/soc/codecs/nau8825.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 907ec88c759a..54ef7b0fa878 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -1440,7 +1440,7 @@ static struct snd_soc_dai_driver nau8825_dai = { .capture = { .stream_name = "Capture", .channels_min = 1, - .channels_max = 1, + .channels_max = 2, /* Only 1 channel of data */ .rates = NAU8825_RATES, .formats = NAU8825_FORMATS, }, -- cgit v1.2.3 From 1460b85daa0af45c1cd2c5e20133ce413184e3d6 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Fri, 8 Jul 2022 19:00:29 +0800 Subject: ASoC: Intel: sof_cs42l42: support BT offload audio Add the capability to machine driver of creating DAI Link for BT offload. Although BT offload always uses SSP2 port but we reserve the flexibility to assign the port number in macro. Signed-off-by: Brent Lu Acked-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220708110030.658468-2-brent.lu@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_cs42l42.c | 75 ++++++++++++++++++++++++++++++++++-- 1 file changed, 71 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/sof_cs42l42.c b/sound/soc/intel/boards/sof_cs42l42.c index a1a14d6d7c23..3d53bb420c66 100644 --- a/sound/soc/intel/boards/sof_cs42l42.c +++ b/sound/soc/intel/boards/sof_cs42l42.c @@ -41,8 +41,13 @@ #define SOF_CS42L42_DAILINK_MASK (GENMASK(24, 10)) #define SOF_CS42L42_DAILINK(link1, link2, link3, link4, link5) \ ((((link1) | ((link2) << 3) | ((link3) << 6) | ((link4) << 9) | ((link5) << 12)) << SOF_CS42L42_DAILINK_SHIFT) & SOF_CS42L42_DAILINK_MASK) -#define SOF_MAX98357A_SPEAKER_AMP_PRESENT BIT(25) -#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(26) +#define SOF_BT_OFFLOAD_PRESENT BIT(25) +#define SOF_CS42L42_SSP_BT_SHIFT 26 +#define SOF_CS42L42_SSP_BT_MASK (GENMASK(28, 26)) +#define SOF_CS42L42_SSP_BT(quirk) \ + (((quirk) << SOF_CS42L42_SSP_BT_SHIFT) & SOF_CS42L42_SSP_BT_MASK) +#define SOF_MAX98357A_SPEAKER_AMP_PRESENT BIT(29) +#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(30) enum { LINK_NONE = 0, @@ -50,6 +55,7 @@ enum { LINK_SPK = 2, LINK_DMIC = 3, LINK_HDMI = 4, + LINK_BT = 5, }; static struct snd_soc_jack_pin jack_pins[] = { @@ -290,6 +296,13 @@ static struct snd_soc_dai_link_component dmic_component[] = { } }; +static struct snd_soc_dai_link_component dummy_component[] = { + { + .name = "snd-soc-dummy", + .dai_name = "snd-soc-dummy-dai", + } +}; + static int create_spk_amp_dai_links(struct device *dev, struct snd_soc_dai_link *links, struct snd_soc_dai_link_component *cpus, @@ -479,9 +492,50 @@ devm_err: return -ENOMEM; } +static int create_bt_offload_dai_links(struct device *dev, + struct snd_soc_dai_link *links, + struct snd_soc_dai_link_component *cpus, + int *id, int ssp_bt) +{ + /* bt offload */ + if (!(sof_cs42l42_quirk & SOF_BT_OFFLOAD_PRESENT)) + return 0; + + links[*id].name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-BT", + ssp_bt); + if (!links[*id].name) + goto devm_err; + + links[*id].id = *id; + links[*id].codecs = dummy_component; + links[*id].num_codecs = ARRAY_SIZE(dummy_component); + links[*id].platforms = platform_component; + links[*id].num_platforms = ARRAY_SIZE(platform_component); + + links[*id].dpcm_playback = 1; + links[*id].dpcm_capture = 1; + links[*id].no_pcm = 1; + links[*id].cpus = &cpus[*id]; + links[*id].num_cpus = 1; + + links[*id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, + "SSP%d Pin", + ssp_bt); + if (!links[*id].cpus->dai_name) + goto devm_err; + + (*id)++; + + return 0; + +devm_err: + return -ENOMEM; +} + static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, int ssp_codec, int ssp_amp, + int ssp_bt, int dmic_be_num, int hdmi_num) { @@ -534,6 +588,14 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, goto devm_err; } break; + case LINK_BT: + ret = create_bt_offload_dai_links(dev, links, cpus, &id, ssp_bt); + if (ret < 0) { + dev_err(dev, "fail to create bt offload dai links, ret %d\n", + ret); + goto devm_err; + } + break; case LINK_NONE: /* caught here if it's not used as terminator in macro */ default: @@ -555,7 +617,7 @@ static int sof_audio_probe(struct platform_device *pdev) struct snd_soc_acpi_mach *mach; struct sof_card_private *ctx; int dmic_be_num, hdmi_num; - int ret, ssp_amp, ssp_codec; + int ret, ssp_bt, ssp_amp, ssp_codec; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) @@ -580,6 +642,9 @@ static int sof_audio_probe(struct platform_device *pdev) dev_dbg(&pdev->dev, "sof_cs42l42_quirk = %lx\n", sof_cs42l42_quirk); + ssp_bt = (sof_cs42l42_quirk & SOF_CS42L42_SSP_BT_MASK) >> + SOF_CS42L42_SSP_BT_SHIFT; + ssp_amp = (sof_cs42l42_quirk & SOF_CS42L42_SSP_AMP_MASK) >> SOF_CS42L42_SSP_AMP_SHIFT; @@ -590,9 +655,11 @@ static int sof_audio_probe(struct platform_device *pdev) if (sof_cs42l42_quirk & SOF_SPEAKER_AMP_PRESENT) sof_audio_card_cs42l42.num_links++; + if (sof_cs42l42_quirk & SOF_BT_OFFLOAD_PRESENT) + sof_audio_card_cs42l42.num_links++; dai_links = sof_card_dai_links_create(&pdev->dev, ssp_codec, ssp_amp, - dmic_be_num, hdmi_num); + ssp_bt, dmic_be_num, hdmi_num); if (!dai_links) return -ENOMEM; -- cgit v1.2.3 From cd486d37493357369ec1d8f130d93806418def84 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Fri, 8 Jul 2022 19:00:30 +0800 Subject: ASoC: Intel: sof_cs42l42: add adl_mx98360a_cs4242 board config This patch adds driver data for adl_mx98360a_cs4242 which supports two max98360a speaker amplifiers on SSP1 and cs42l42 headphone codec on SSP0 running on ADL platform. Signed-off-by: Brent Lu Acked-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220708110030.658468-3-brent.lu@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_cs42l42.c | 11 +++++++++++ sound/soc/intel/common/soc-acpi-intel-adl-match.c | 7 +++++++ 2 files changed, 18 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/boards/sof_cs42l42.c b/sound/soc/intel/boards/sof_cs42l42.c index 3d53bb420c66..85ffd065895d 100644 --- a/sound/soc/intel/boards/sof_cs42l42.c +++ b/sound/soc/intel/boards/sof_cs42l42.c @@ -700,6 +700,17 @@ static const struct platform_device_id board_ids[] = { SOF_CS42L42_SSP_AMP(1)) | SOF_CS42L42_DAILINK(LINK_HP, LINK_DMIC, LINK_HDMI, LINK_SPK, LINK_NONE), }, + { + .name = "adl_mx98360a_cs4242", + .driver_data = (kernel_ulong_t)(SOF_CS42L42_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98360A_SPEAKER_AMP_PRESENT | + SOF_CS42L42_SSP_AMP(1) | + SOF_CS42L42_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_PRESENT | + SOF_CS42L42_SSP_BT(2) | + SOF_CS42L42_DAILINK(LINK_HP, LINK_DMIC, LINK_HDMI, LINK_SPK, LINK_BT)), + }, { } }; MODULE_DEVICE_TABLE(platform, board_ids); diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c index c1385161cdc8..fea087d3fa15 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -479,6 +479,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { .drv_name = "adl_rt5682", .sof_tplg_filename = "sof-adl-rt5682.tplg", }, + { + .id = "10134242", + .drv_name = "adl_mx98360a_cs4242", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &adl_max98360a_amp, + .sof_tplg_filename = "sof-adl-max98360a-cs42l42.tplg", + }, /* place amp-only boards in the end of table */ { .id = "CSC3541", -- cgit v1.2.3 From ac2606df8a3fb4450240cf0893ff3934b5882c69 Mon Sep 17 00:00:00 2001 From: V sujith kumar Reddy Date: Thu, 7 Jul 2022 21:41:40 +0530 Subject: ASoC: amd: acp: Remove rt1019_1 codec conf from machine driver Remove rt1019_1 codec configuration which has i2c-10EC1019:01 and i2c-10EC1019:02 codec components, Now Using default i2c-10EC1019:00 and i2c-10EC1019:01 codec components. Signed-off-by: V sujith kumar Reddy Link: https://lore.kernel.org/r/20220707161142.491034-2-Vsujithkumar.Reddy@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp-mach-common.c | 18 ------------------ 1 file changed, 18 deletions(-) (limited to 'sound') diff --git a/sound/soc/amd/acp/acp-mach-common.c b/sound/soc/amd/acp/acp-mach-common.c index 7530cab24bc8..86145398fa25 100644 --- a/sound/soc/amd/acp/acp-mach-common.c +++ b/sound/soc/amd/acp/acp-mach-common.c @@ -313,9 +313,6 @@ static const struct snd_soc_ops acp_card_dmic_ops = { SND_SOC_DAILINK_DEF(rt1019, DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10EC1019:00", "rt1019-aif"), COMP_CODEC("i2c-10EC1019:01", "rt1019-aif"))); -SND_SOC_DAILINK_DEF(rt1019_1, - DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10EC1019:02", "rt1019-aif"), - COMP_CODEC("i2c-10EC1019:01", "rt1019-aif"))); static const struct snd_soc_dapm_route rt1019_map_lr[] = { { "Left Spk", NULL, "Left SPO" }, @@ -333,17 +330,6 @@ static struct snd_soc_codec_conf rt1019_conf[] = { }, }; -static struct snd_soc_codec_conf rt1019_1_conf[] = { - { - .dlc = COMP_CODEC_CONF("i2c-10EC1019:02"), - .name_prefix = "Left", - }, - { - .dlc = COMP_CODEC_CONF("i2c-10EC1019:01"), - .name_prefix = "Right", - }, -}; - static int acp_card_rt1019_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; @@ -716,10 +702,6 @@ int acp_sofdsp_dai_links_create(struct snd_soc_card *card) links[i].init = acp_card_rt1019_init; card->codec_conf = rt1019_conf; card->num_configs = ARRAY_SIZE(rt1019_conf); - links[i].codecs = rt1019_1; - links[i].num_codecs = ARRAY_SIZE(rt1019_1); - card->codec_conf = rt1019_1_conf; - card->num_configs = ARRAY_SIZE(rt1019_1_conf); } i++; } -- cgit v1.2.3 From b24484c18b1089f9dd1ef7901b05a85e315e9f41 Mon Sep 17 00:00:00 2001 From: V sujith kumar Reddy Date: Thu, 7 Jul 2022 21:41:41 +0530 Subject: ASoC: amd: acp: ACP code generic to support newer platforms ADD Generic code to support to newer platforms, add control threshold, irq control macros ,added structure for register offset differences. Signed-off-by: V sujith kumar Reddy Link: https://lore.kernel.org/r/20220707161142.491034-3-Vsujithkumar.Reddy@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp-i2s.c | 34 +++++++++++++++++++------------- sound/soc/amd/acp/acp-pdm.c | 8 ++++---- sound/soc/amd/acp/acp-platform.c | 16 +++++++++------ sound/soc/amd/acp/acp-renoir.c | 38 +++++++++++++++++++++++++----------- sound/soc/amd/acp/amd.h | 24 +++++++++++++++++------ sound/soc/amd/acp/chip_offset_byte.h | 12 +++++++----- 6 files changed, 86 insertions(+), 46 deletions(-) (limited to 'sound') diff --git a/sound/soc/amd/acp/acp-i2s.c b/sound/soc/amd/acp/acp-i2s.c index ce9aca8dd6f5..a736c00db86e 100644 --- a/sound/soc/amd/acp/acp-i2s.c +++ b/sound/soc/amd/acp/acp-i2s.c @@ -199,6 +199,7 @@ static int acp_i2s_prepare(struct snd_pcm_substream *substream, struct snd_soc_d { struct device *dev = dai->component->dev; struct acp_dev_data *adata = dev_get_drvdata(dev); + struct acp_resource *rsrc = adata->rsrc; struct acp_stream *stream = substream->runtime->private_data; u32 reg_dma_size = 0, reg_fifo_size = 0, reg_fifo_addr = 0; u32 phy_addr = 0, acp_fifo_addr = 0, ext_int_ctrl; @@ -208,7 +209,7 @@ static int acp_i2s_prepare(struct snd_pcm_substream *substream, struct snd_soc_d case I2S_SP_INSTANCE: if (dir == SNDRV_PCM_STREAM_PLAYBACK) { reg_dma_size = ACP_I2S_TX_DMA_SIZE; - acp_fifo_addr = ACP_SRAM_PTE_OFFSET + + acp_fifo_addr = rsrc->sram_pte_offset + SP_PB_FIFO_ADDR_OFFSET; reg_fifo_addr = ACP_I2S_TX_FIFOADDR; reg_fifo_size = ACP_I2S_TX_FIFOSIZE; @@ -217,7 +218,7 @@ static int acp_i2s_prepare(struct snd_pcm_substream *substream, struct snd_soc_d writel(phy_addr, adata->acp_base + ACP_I2S_TX_RINGBUFADDR); } else { reg_dma_size = ACP_I2S_RX_DMA_SIZE; - acp_fifo_addr = ACP_SRAM_PTE_OFFSET + + acp_fifo_addr = rsrc->sram_pte_offset + SP_CAPT_FIFO_ADDR_OFFSET; reg_fifo_addr = ACP_I2S_RX_FIFOADDR; reg_fifo_size = ACP_I2S_RX_FIFOSIZE; @@ -228,7 +229,7 @@ static int acp_i2s_prepare(struct snd_pcm_substream *substream, struct snd_soc_d case I2S_BT_INSTANCE: if (dir == SNDRV_PCM_STREAM_PLAYBACK) { reg_dma_size = ACP_BT_TX_DMA_SIZE; - acp_fifo_addr = ACP_SRAM_PTE_OFFSET + + acp_fifo_addr = rsrc->sram_pte_offset + BT_PB_FIFO_ADDR_OFFSET; reg_fifo_addr = ACP_BT_TX_FIFOADDR; reg_fifo_size = ACP_BT_TX_FIFOSIZE; @@ -237,7 +238,7 @@ static int acp_i2s_prepare(struct snd_pcm_substream *substream, struct snd_soc_d writel(phy_addr, adata->acp_base + ACP_BT_TX_RINGBUFADDR); } else { reg_dma_size = ACP_BT_RX_DMA_SIZE; - acp_fifo_addr = ACP_SRAM_PTE_OFFSET + + acp_fifo_addr = rsrc->sram_pte_offset + BT_CAPT_FIFO_ADDR_OFFSET; reg_fifo_addr = ACP_BT_RX_FIFOADDR; reg_fifo_size = ACP_BT_RX_FIFOSIZE; @@ -255,11 +256,13 @@ static int acp_i2s_prepare(struct snd_pcm_substream *substream, struct snd_soc_d writel(acp_fifo_addr, adata->acp_base + reg_fifo_addr); writel(FIFO_SIZE, adata->acp_base + reg_fifo_size); - ext_int_ctrl = readl(adata->acp_base + ACP_EXTERNAL_INTR_CNTL); - ext_int_ctrl |= BIT(I2S_RX_THRESHOLD) | BIT(BT_RX_THRESHOLD) - | BIT(I2S_TX_THRESHOLD) | BIT(BT_TX_THRESHOLD); + ext_int_ctrl = readl(ACP_EXTERNAL_INTR_CNTL(adata, rsrc->irqp_used)); + ext_int_ctrl |= BIT(I2S_RX_THRESHOLD(rsrc->offset)) | + BIT(BT_RX_THRESHOLD(rsrc->offset)) | + BIT(I2S_TX_THRESHOLD(rsrc->offset)) | + BIT(BT_TX_THRESHOLD(rsrc->offset)); - writel(ext_int_ctrl, adata->acp_base + ACP_EXTERNAL_INTR_CNTL); + writel(ext_int_ctrl, ACP_EXTERNAL_INTR_CNTL(adata, rsrc->irqp_used)); return 0; } @@ -268,28 +271,30 @@ static int acp_i2s_startup(struct snd_pcm_substream *substream, struct snd_soc_d { struct acp_stream *stream = substream->runtime->private_data; struct device *dev = dai->component->dev; + struct acp_dev_data *adata = dev_get_drvdata(dev); + struct acp_resource *rsrc = adata->rsrc; unsigned int dir = substream->stream; unsigned int irq_bit = 0; switch (dai->driver->id) { case I2S_SP_INSTANCE: if (dir == SNDRV_PCM_STREAM_PLAYBACK) { - irq_bit = BIT(I2S_TX_THRESHOLD); + irq_bit = BIT(I2S_TX_THRESHOLD(rsrc->offset)); stream->pte_offset = ACP_SRAM_SP_PB_PTE_OFFSET; stream->fifo_offset = SP_PB_FIFO_ADDR_OFFSET; } else { - irq_bit = BIT(I2S_RX_THRESHOLD); + irq_bit = BIT(I2S_RX_THRESHOLD(rsrc->offset)); stream->pte_offset = ACP_SRAM_SP_CP_PTE_OFFSET; stream->fifo_offset = SP_CAPT_FIFO_ADDR_OFFSET; } break; case I2S_BT_INSTANCE: if (dir == SNDRV_PCM_STREAM_PLAYBACK) { - irq_bit = BIT(BT_TX_THRESHOLD); + irq_bit = BIT(BT_TX_THRESHOLD(rsrc->offset)); stream->pte_offset = ACP_SRAM_BT_PB_PTE_OFFSET; stream->fifo_offset = BT_PB_FIFO_ADDR_OFFSET; } else { - irq_bit = BIT(BT_RX_THRESHOLD); + irq_bit = BIT(BT_RX_THRESHOLD(rsrc->offset)); stream->pte_offset = ACP_SRAM_BT_CP_PTE_OFFSET; stream->fifo_offset = BT_CAPT_FIFO_ADDR_OFFSET; } @@ -319,6 +324,7 @@ int asoc_acp_i2s_probe(struct snd_soc_dai *dai) { struct device *dev = dai->component->dev; struct acp_dev_data *adata = dev_get_drvdata(dev); + struct acp_resource *rsrc = adata->rsrc; unsigned int val; if (!adata->acp_base) { @@ -326,8 +332,8 @@ int asoc_acp_i2s_probe(struct snd_soc_dai *dai) return -EINVAL; } - val = readl(adata->acp_base + ACP_I2S_PIN_CONFIG); - if (val != I2S_MODE) { + val = readl(adata->acp_base + rsrc->i2s_pin_cfg_offset); + if (val != rsrc->i2s_mode) { dev_err(dev, "I2S Mode not supported val %x\n", val); return -EINVAL; } diff --git a/sound/soc/amd/acp/acp-pdm.c b/sound/soc/amd/acp/acp-pdm.c index 7a0b26a30051..66ec6b6a5972 100644 --- a/sound/soc/amd/acp/acp-pdm.c +++ b/sound/soc/amd/acp/acp-pdm.c @@ -160,9 +160,9 @@ static int acp_dmic_dai_startup(struct snd_pcm_substream *substream, stream->reg_offset = ACP_REGION2_OFFSET; /* Enable DMIC Interrupts */ - ext_int_ctrl = readl(adata->acp_base + ACP_EXTERNAL_INTR_CNTL); + ext_int_ctrl = readl(ACP_EXTERNAL_INTR_CNTL(adata, 0)); ext_int_ctrl |= PDM_DMA_INTR_MASK; - writel(ext_int_ctrl, adata->acp_base + ACP_EXTERNAL_INTR_CNTL); + writel(ext_int_ctrl, ACP_EXTERNAL_INTR_CNTL(adata, 0)); return 0; } @@ -175,9 +175,9 @@ static void acp_dmic_dai_shutdown(struct snd_pcm_substream *substream, u32 ext_int_ctrl; /* Disable DMIC interrupts */ - ext_int_ctrl = readl(adata->acp_base + ACP_EXTERNAL_INTR_CNTL); + ext_int_ctrl = readl(ACP_EXTERNAL_INTR_CNTL(adata, 0)); ext_int_ctrl |= ~PDM_DMA_INTR_MASK; - writel(ext_int_ctrl, adata->acp_base + ACP_EXTERNAL_INTR_CNTL); + writel(ext_int_ctrl, ACP_EXTERNAL_INTR_CNTL(adata, 0)); } const struct snd_soc_dai_ops acp_dmic_dai_ops = { diff --git a/sound/soc/amd/acp/acp-platform.c b/sound/soc/amd/acp/acp-platform.c index 3c4fd8b80589..e93c9e478cfa 100644 --- a/sound/soc/amd/acp/acp-platform.c +++ b/sound/soc/amd/acp/acp-platform.c @@ -91,6 +91,7 @@ EXPORT_SYMBOL_NS_GPL(acp_machine_select, SND_SOC_ACP_COMMON); static irqreturn_t i2s_irq_handler(int irq, void *data) { struct acp_dev_data *adata = data; + struct acp_resource *rsrc = adata->rsrc; struct acp_stream *stream; u16 i2s_flag = 0; u32 val, i; @@ -98,12 +99,13 @@ static irqreturn_t i2s_irq_handler(int irq, void *data) if (!adata) return IRQ_NONE; - val = readl(adata->acp_base + ACP_EXTERNAL_INTR_STAT); + val = readl(ACP_EXTERNAL_INTR_STAT(adata, rsrc->irqp_used)); for (i = 0; i < ACP_MAX_STREAM; i++) { stream = adata->stream[i]; if (stream && (val & stream->irq_bit)) { - writel(stream->irq_bit, adata->acp_base + ACP_EXTERNAL_INTR_STAT); + writel(stream->irq_bit, + ACP_EXTERNAL_INTR_STAT(adata, rsrc->irqp_used)); snd_pcm_period_elapsed(stream->substream); i2s_flag = 1; break; @@ -118,6 +120,7 @@ static irqreturn_t i2s_irq_handler(int irq, void *data) static void config_pte_for_stream(struct acp_dev_data *adata, struct acp_stream *stream) { + struct acp_resource *rsrc = adata->rsrc; u32 pte_reg, pte_size, reg_val; /* Use ATU base Group5 */ @@ -126,7 +129,7 @@ static void config_pte_for_stream(struct acp_dev_data *adata, struct acp_stream stream->reg_offset = 0x02000000; /* Group Enable */ - reg_val = ACP_SRAM_PTE_OFFSET; + reg_val = rsrc->sram_pte_offset; writel(reg_val | BIT(31), adata->acp_base + pte_reg); writel(PAGE_SIZE_4K_ENABLE, adata->acp_base + pte_size); } @@ -135,6 +138,7 @@ static void config_acp_dma(struct acp_dev_data *adata, int cpu_id, int size) { struct acp_stream *stream = adata->stream[cpu_id]; struct snd_pcm_substream *substream = stream->substream; + struct acp_resource *rsrc = adata->rsrc; dma_addr_t addr = substream->dma_buffer.addr; int num_pages = (PAGE_ALIGN(size) >> PAGE_SHIFT); u32 low, high, val; @@ -146,9 +150,9 @@ static void config_acp_dma(struct acp_dev_data *adata, int cpu_id, int size) /* Load the low address of page int ACP SRAM through SRBM */ low = lower_32_bits(addr); high = upper_32_bits(addr); - writel(low, adata->acp_base + ACP_SCRATCH_REG_0 + val); + writel(low, adata->acp_base + rsrc->scratch_reg_offset + val); high |= BIT(31); - writel(high, adata->acp_base + ACP_SCRATCH_REG_0 + val + 4); + writel(high, adata->acp_base + rsrc->scratch_reg_offset + val + 4); /* Move to next physically contiguous page */ val += 8; @@ -187,7 +191,7 @@ static int acp_dma_open(struct snd_soc_component *component, struct snd_pcm_subs } runtime->private_data = stream; - writel(1, adata->acp_base + ACP_EXTERNAL_INTR_ENB); + writel(1, ACP_EXTERNAL_INTR_ENB(adata)); return ret; } diff --git a/sound/soc/amd/acp/acp-renoir.c b/sound/soc/amd/acp/acp-renoir.c index 8375c00ff4c3..2a89a0d2e601 100644 --- a/sound/soc/amd/acp/acp-renoir.c +++ b/sound/soc/amd/acp/acp-renoir.c @@ -39,6 +39,17 @@ #define ACP_ERROR_MASK 0x20000000 #define ACP_EXT_INTR_STAT_CLEAR_MASK 0xFFFFFFFF +static struct acp_resource rsrc = { + .offset = 20, + .no_of_ctrls = 1, + .irqp_used = 0, + .irq_reg_offset = 0x1800, + .i2s_pin_cfg_offset = 0x1400, + .i2s_mode = 0x04, + .scratch_reg_offset = 0x12800, + .sram_pte_offset = 0x02052800, +}; + static struct snd_soc_acpi_codecs amp_rt1019 = { .num_codecs = 1, .codecs = {"10EC1019"} @@ -186,20 +197,24 @@ static int acp3x_reset(void __iomem *base) return readl_poll_timeout(base + ACP_SOFT_RESET, val, !val, DELAY_US, ACP_TIMEOUT); } -static void acp3x_enable_interrupts(void __iomem *base) +static void acp3x_enable_interrupts(struct acp_dev_data *adata) { + struct acp_resource *rsrc = adata->rsrc; u32 ext_intr_ctrl; - writel(0x01, base + ACP_EXTERNAL_INTR_ENB); - ext_intr_ctrl = readl(base + ACP_EXTERNAL_INTR_CNTL); + writel(0x01, ACP_EXTERNAL_INTR_ENB(adata)); + ext_intr_ctrl = readl(ACP_EXTERNAL_INTR_CNTL(adata, rsrc->irqp_used)); ext_intr_ctrl |= ACP_ERROR_MASK; - writel(ext_intr_ctrl, base + ACP_EXTERNAL_INTR_CNTL); + writel(ext_intr_ctrl, ACP_EXTERNAL_INTR_CNTL(adata, rsrc->irqp_used)); } -static void acp3x_disable_interrupts(void __iomem *base) +static void acp3x_disable_interrupts(struct acp_dev_data *adata) { - writel(ACP_EXT_INTR_STAT_CLEAR_MASK, base + ACP_EXTERNAL_INTR_STAT); - writel(0x00, base + ACP_EXTERNAL_INTR_ENB); + struct acp_resource *rsrc = adata->rsrc; + + writel(ACP_EXT_INTR_STAT_CLEAR_MASK, + ACP_EXTERNAL_INTR_STAT(adata, rsrc->irqp_used)); + writel(0x00, ACP_EXTERNAL_INTR_ENB(adata)); } static int rn_acp_init(void __iomem *base) @@ -218,8 +233,6 @@ static int rn_acp_init(void __iomem *base) if (ret) return ret; - acp3x_enable_interrupts(base); - return 0; } @@ -227,8 +240,6 @@ static int rn_acp_deinit(void __iomem *base) { int ret = 0; - acp3x_disable_interrupts(base); - /* Reset */ ret = acp3x_reset(base); if (ret) @@ -290,11 +301,13 @@ static int renoir_audio_probe(struct platform_device *pdev) adata->dev = dev; adata->dai_driver = acp_renoir_dai; adata->num_dai = ARRAY_SIZE(acp_renoir_dai); + adata->rsrc = &rsrc; adata->machines = snd_soc_acpi_amd_acp_machines; acp_machine_select(adata); dev_set_drvdata(dev, adata); + acp3x_enable_interrupts(adata); acp_platform_register(dev); return 0; @@ -303,11 +316,14 @@ static int renoir_audio_probe(struct platform_device *pdev) static int renoir_audio_remove(struct platform_device *pdev) { struct device *dev = &pdev->dev; + struct acp_dev_data *adata = dev_get_drvdata(dev); struct acp_chip_info *chip; int ret; chip = dev_get_platdata(&pdev->dev); + acp3x_disable_interrupts(adata); + ret = rn_acp_deinit(chip->base); if (ret) dev_err(&pdev->dev, "ACP de-init Failed (%pe)\n", ERR_PTR(ret)); diff --git a/sound/soc/amd/acp/amd.h b/sound/soc/amd/acp/amd.h index 8fd38bf4d3bd..186cb8b26175 100644 --- a/sound/soc/amd/acp/amd.h +++ b/sound/soc/amd/acp/amd.h @@ -32,13 +32,12 @@ #define ACP3x_I2STDM_REG_END 0x1242410 #define ACP3x_BT_TDM_REG_START 0x1242800 #define ACP3x_BT_TDM_REG_END 0x1242810 -#define I2S_MODE 0x04 -#define I2S_RX_THRESHOLD 27 -#define I2S_TX_THRESHOLD 28 -#define BT_TX_THRESHOLD 26 -#define BT_RX_THRESHOLD 25 -#define ACP_SRAM_PTE_OFFSET 0x02052800 +#define THRESHOLD(bit, base) ((bit) + (base)) +#define I2S_RX_THRESHOLD(base) THRESHOLD(7, base) +#define I2S_TX_THRESHOLD(base) THRESHOLD(8, base) +#define BT_TX_THRESHOLD(base) THRESHOLD(6, base) +#define BT_RX_THRESHOLD(base) THRESHOLD(5, base) #define ACP_SRAM_SP_PB_PTE_OFFSET 0x0 #define ACP_SRAM_SP_CP_PTE_OFFSET 0x100 @@ -92,6 +91,17 @@ struct acp_stream { u32 fifo_offset; }; +struct acp_resource { + int offset; + int no_of_ctrls; + int irqp_used; + u32 irq_reg_offset; + u32 i2s_pin_cfg_offset; + int i2s_mode; + u64 scratch_reg_offset; + u64 sram_pte_offset; +}; + struct acp_dev_data { char *name; struct device *dev; @@ -106,6 +116,8 @@ struct acp_dev_data { struct snd_soc_acpi_mach *machines; struct platform_device *mach_dev; + + struct acp_resource *rsrc; }; extern const struct snd_soc_dai_ops asoc_acp_cpu_dai_ops; diff --git a/sound/soc/amd/acp/chip_offset_byte.h b/sound/soc/amd/acp/chip_offset_byte.h index 88f6fa597cd6..fff7e80475ba 100644 --- a/sound/soc/amd/acp/chip_offset_byte.h +++ b/sound/soc/amd/acp/chip_offset_byte.h @@ -20,11 +20,13 @@ #define ACP_SOFT_RESET 0x1000 #define ACP_CONTROL 0x1004 -#define ACP_EXTERNAL_INTR_ENB 0x1800 -#define ACP_EXTERNAL_INTR_CNTL 0x1804 -#define ACP_EXTERNAL_INTR_STAT 0x1808 -#define ACP_I2S_PIN_CONFIG 0x1400 -#define ACP_SCRATCH_REG_0 0x12800 +#define ACP_EXTERNAL_INTR_REG_ADDR(adata, offset, ctrl) \ + (adata->acp_base + adata->rsrc->irq_reg_offset + offset + (ctrl * 0x04)) + +#define ACP_EXTERNAL_INTR_ENB(adata) ACP_EXTERNAL_INTR_REG_ADDR(adata, 0x0, 0x0) +#define ACP_EXTERNAL_INTR_CNTL(adata, ctrl) ACP_EXTERNAL_INTR_REG_ADDR(adata, 0x4, ctrl) +#define ACP_EXTERNAL_INTR_STAT(adata, ctrl) ACP_EXTERNAL_INTR_REG_ADDR(adata, \ + (0x4 + (adata->rsrc->no_of_ctrls * 0x04)), ctrl) /* Registers from ACP_AUDIO_BUFFERS block */ -- cgit v1.2.3 From e8a33a94078560df73761f6d6147a25bda07605c Mon Sep 17 00:00:00 2001 From: V sujith kumar Reddy Date: Thu, 7 Jul 2022 21:41:42 +0530 Subject: ASoC: amd: acp: Add legacy audio driver support for Rembrandt platform Add i2s and dmic support for Rembrandt platform, Add machine support for nau8825, max98360 and rt5682s,rt1019 codec in legacy driver for rembrandt platform. Here codec is in a slave mode. Signed-off-by: V sujith kumar Reddy Link: https://lore.kernel.org/r/20220707161142.491034-4-Vsujithkumar.Reddy@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/Kconfig | 11 + sound/soc/amd/acp/Makefile | 2 + sound/soc/amd/acp/acp-i2s.c | 137 +++++++++++- sound/soc/amd/acp/acp-legacy-mach.c | 32 +++ sound/soc/amd/acp/acp-mach-common.c | 86 +++++++- sound/soc/amd/acp/acp-mach.h | 6 + sound/soc/amd/acp/acp-pci.c | 6 + sound/soc/amd/acp/acp-platform.c | 16 +- sound/soc/amd/acp/acp-rembrandt.c | 401 +++++++++++++++++++++++++++++++++++ sound/soc/amd/acp/amd.h | 62 +++++- sound/soc/amd/acp/chip_offset_byte.h | 28 +++ 11 files changed, 781 insertions(+), 6 deletions(-) create mode 100644 sound/soc/amd/acp/acp-rembrandt.c (limited to 'sound') diff --git a/sound/soc/amd/acp/Kconfig b/sound/soc/amd/acp/Kconfig index 7e56d2644105..ce0037810743 100644 --- a/sound/soc/amd/acp/Kconfig +++ b/sound/soc/amd/acp/Kconfig @@ -40,6 +40,17 @@ config SND_AMD_ASOC_RENOIR help This option enables Renoir I2S support on AMD platform. +config SND_AMD_ASOC_REMBRANDT + tristate "AMD ACP ASOC Rembrandt Support" + select SND_SOC_AMD_ACP_PCM + select SND_SOC_AMD_ACP_I2S + select SND_SOC_AMD_ACP_PDM + depends on X86 && PCI + help + This option enables Rembrandt I2S support on AMD platform. + Say Y if you want to enable AUDIO on Rembrandt + If unsure select "N". + config SND_SOC_AMD_MACH_COMMON tristate depends on X86 && PCI && I2C diff --git a/sound/soc/amd/acp/Makefile b/sound/soc/amd/acp/Makefile index 657ddfadf0bb..d9abb0ee5218 100644 --- a/sound/soc/amd/acp/Makefile +++ b/sound/soc/amd/acp/Makefile @@ -12,6 +12,7 @@ snd-acp-pci-objs := acp-pci.o #platform specific driver snd-acp-renoir-objs := acp-renoir.o +snd-acp-rembrandt-objs := acp-rembrandt.o #machine specific driver snd-acp-mach-objs := acp-mach-common.o @@ -24,6 +25,7 @@ obj-$(CONFIG_SND_SOC_AMD_ACP_PDM) += snd-acp-pdm.o obj-$(CONFIG_SND_SOC_AMD_ACP_PCI) += snd-acp-pci.o obj-$(CONFIG_SND_AMD_ASOC_RENOIR) += snd-acp-renoir.o +obj-$(CONFIG_SND_AMD_ASOC_REMBRANDT) += snd-acp-rembrandt.o obj-$(CONFIG_SND_SOC_AMD_MACH_COMMON) += snd-acp-mach.o obj-$(CONFIG_SND_SOC_AMD_LEGACY_MACH) += snd-acp-legacy-mach.o diff --git a/sound/soc/amd/acp/acp-i2s.c b/sound/soc/amd/acp/acp-i2s.c index a736c00db86e..393f729ef561 100644 --- a/sound/soc/amd/acp/acp-i2s.c +++ b/sound/soc/amd/acp/acp-i2s.c @@ -30,11 +30,14 @@ static int acp_i2s_hwparams(struct snd_pcm_substream *substream, struct snd_pcm_ { struct device *dev = dai->component->dev; struct acp_dev_data *adata; + struct acp_resource *rsrc; u32 val; u32 xfer_resolution; u32 reg_val; + u32 lrclk_div_val, bclk_div_val; adata = snd_soc_dai_get_drvdata(dai); + rsrc = adata->rsrc; /* These values are as per Hardware Spec */ switch (params_format(params)) { @@ -63,6 +66,9 @@ static int acp_i2s_hwparams(struct snd_pcm_substream *substream, struct snd_pcm_ case I2S_SP_INSTANCE: reg_val = ACP_I2STDM_ITER; break; + case I2S_HS_INSTANCE: + reg_val = ACP_HSTDM_ITER; + break; default: dev_err(dev, "Invalid dai id %x\n", dai->driver->id); return -EINVAL; @@ -75,6 +81,9 @@ static int acp_i2s_hwparams(struct snd_pcm_substream *substream, struct snd_pcm_ case I2S_SP_INSTANCE: reg_val = ACP_I2STDM_IRER; break; + case I2S_HS_INSTANCE: + reg_val = ACP_HSTDM_IRER; + break; default: dev_err(dev, "Invalid dai id %x\n", dai->driver->id); return -EINVAL; @@ -86,6 +95,74 @@ static int acp_i2s_hwparams(struct snd_pcm_substream *substream, struct snd_pcm_ val = val | (xfer_resolution << 3); writel(val, adata->acp_base + reg_val); + if (rsrc->soc_mclk) { + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + switch (params_rate(params)) { + case 8000: + bclk_div_val = 768; + break; + case 16000: + bclk_div_val = 384; + break; + case 24000: + bclk_div_val = 256; + break; + case 32000: + bclk_div_val = 192; + break; + case 44100: + case 48000: + bclk_div_val = 128; + break; + case 88200: + case 96000: + bclk_div_val = 64; + break; + case 192000: + bclk_div_val = 32; + break; + default: + return -EINVAL; + } + lrclk_div_val = 32; + break; + case SNDRV_PCM_FORMAT_S32_LE: + switch (params_rate(params)) { + case 8000: + bclk_div_val = 384; + break; + case 16000: + bclk_div_val = 192; + break; + case 24000: + bclk_div_val = 128; + break; + case 32000: + bclk_div_val = 96; + break; + case 44100: + case 48000: + bclk_div_val = 64; + break; + case 88200: + case 96000: + bclk_div_val = 32; + break; + case 192000: + bclk_div_val = 16; + break; + default: + return -EINVAL; + } + lrclk_div_val = 64; + break; + default: + return -EINVAL; + } + adata->lrclk_div = lrclk_div_val; + adata->bclk_div = bclk_div_val; + } return 0; } @@ -94,6 +171,7 @@ static int acp_i2s_trigger(struct snd_pcm_substream *substream, int cmd, struct struct acp_stream *stream = substream->runtime->private_data; struct device *dev = dai->component->dev; struct acp_dev_data *adata = dev_get_drvdata(dev); + struct acp_resource *rsrc = adata->rsrc; u32 val, period_bytes, reg_val, ier_val, water_val, buf_size, buf_reg; period_bytes = frames_to_bytes(substream->runtime, substream->runtime->period_size); @@ -118,6 +196,12 @@ static int acp_i2s_trigger(struct snd_pcm_substream *substream, int cmd, struct ier_val = ACP_I2STDM_IER; buf_reg = ACP_I2S_TX_RINGBUFSIZE; break; + case I2S_HS_INSTANCE: + water_val = ACP_HS_TX_INTR_WATERMARK_SIZE; + reg_val = ACP_HSTDM_ITER; + ier_val = ACP_HSTDM_IER; + buf_reg = ACP_HS_TX_RINGBUFSIZE; + break; default: dev_err(dev, "Invalid dai id %x\n", dai->driver->id); return -EINVAL; @@ -136,6 +220,12 @@ static int acp_i2s_trigger(struct snd_pcm_substream *substream, int cmd, struct ier_val = ACP_I2STDM_IER; buf_reg = ACP_I2S_RX_RINGBUFSIZE; break; + case I2S_HS_INSTANCE: + water_val = ACP_HS_RX_INTR_WATERMARK_SIZE; + reg_val = ACP_HSTDM_IRER; + ier_val = ACP_HSTDM_IER; + buf_reg = ACP_HS_RX_RINGBUFSIZE; + break; default: dev_err(dev, "Invalid dai id %x\n", dai->driver->id); return -EINVAL; @@ -147,6 +237,8 @@ static int acp_i2s_trigger(struct snd_pcm_substream *substream, int cmd, struct val = val | BIT(0); writel(val, adata->acp_base + reg_val); writel(1, adata->acp_base + ier_val); + if (rsrc->soc_mclk) + acp_set_i2s_clk(adata, dai->driver->id); return 0; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: @@ -159,6 +251,9 @@ static int acp_i2s_trigger(struct snd_pcm_substream *substream, int cmd, struct case I2S_SP_INSTANCE: reg_val = ACP_I2STDM_ITER; break; + case I2S_HS_INSTANCE: + reg_val = ACP_HSTDM_ITER; + break; default: dev_err(dev, "Invalid dai id %x\n", dai->driver->id); return -EINVAL; @@ -172,6 +267,9 @@ static int acp_i2s_trigger(struct snd_pcm_substream *substream, int cmd, struct case I2S_SP_INSTANCE: reg_val = ACP_I2STDM_IRER; break; + case I2S_HS_INSTANCE: + reg_val = ACP_HSTDM_IRER; + break; default: dev_err(dev, "Invalid dai id %x\n", dai->driver->id); return -EINVAL; @@ -187,6 +285,9 @@ static int acp_i2s_trigger(struct snd_pcm_substream *substream, int cmd, struct if (!(readl(adata->acp_base + ACP_I2STDM_ITER) & BIT(0)) && !(readl(adata->acp_base + ACP_I2STDM_IRER) & BIT(0))) writel(0, adata->acp_base + ACP_I2STDM_IER); + if (!(readl(adata->acp_base + ACP_HSTDM_ITER) & BIT(0)) && + !(readl(adata->acp_base + ACP_HSTDM_IRER) & BIT(0))) + writel(0, adata->acp_base + ACP_HSTDM_IER); return 0; default: return -EINVAL; @@ -247,6 +348,27 @@ static int acp_i2s_prepare(struct snd_pcm_substream *substream, struct snd_soc_d writel(phy_addr, adata->acp_base + ACP_BT_RX_RINGBUFADDR); } break; + case I2S_HS_INSTANCE: + if (dir == SNDRV_PCM_STREAM_PLAYBACK) { + reg_dma_size = ACP_HS_TX_DMA_SIZE; + acp_fifo_addr = rsrc->sram_pte_offset + + HS_PB_FIFO_ADDR_OFFSET; + reg_fifo_addr = ACP_HS_TX_FIFOADDR; + reg_fifo_size = ACP_HS_TX_FIFOSIZE; + + phy_addr = I2S_HS_TX_MEM_WINDOW_START + stream->reg_offset; + writel(phy_addr, adata->acp_base + ACP_HS_TX_RINGBUFADDR); + } else { + reg_dma_size = ACP_HS_RX_DMA_SIZE; + acp_fifo_addr = rsrc->sram_pte_offset + + HS_CAPT_FIFO_ADDR_OFFSET; + reg_fifo_addr = ACP_HS_RX_FIFOADDR; + reg_fifo_size = ACP_HS_RX_FIFOSIZE; + + phy_addr = I2S_HS_RX_MEM_WINDOW_START + stream->reg_offset; + writel(phy_addr, adata->acp_base + ACP_HS_RX_RINGBUFADDR); + } + break; default: dev_err(dev, "Invalid dai id %x\n", dai->driver->id); return -EINVAL; @@ -260,7 +382,9 @@ static int acp_i2s_prepare(struct snd_pcm_substream *substream, struct snd_soc_d ext_int_ctrl |= BIT(I2S_RX_THRESHOLD(rsrc->offset)) | BIT(BT_RX_THRESHOLD(rsrc->offset)) | BIT(I2S_TX_THRESHOLD(rsrc->offset)) | - BIT(BT_TX_THRESHOLD(rsrc->offset)); + BIT(BT_TX_THRESHOLD(rsrc->offset)) | + BIT(HS_RX_THRESHOLD(rsrc->offset)) | + BIT(HS_TX_THRESHOLD(rsrc->offset)); writel(ext_int_ctrl, ACP_EXTERNAL_INTR_CNTL(adata, rsrc->irqp_used)); @@ -299,6 +423,17 @@ static int acp_i2s_startup(struct snd_pcm_substream *substream, struct snd_soc_d stream->fifo_offset = BT_CAPT_FIFO_ADDR_OFFSET; } break; + case I2S_HS_INSTANCE: + if (dir == SNDRV_PCM_STREAM_PLAYBACK) { + irq_bit = BIT(HS_TX_THRESHOLD(rsrc->offset)); + stream->pte_offset = ACP_SRAM_HS_PB_PTE_OFFSET; + stream->fifo_offset = HS_PB_FIFO_ADDR_OFFSET; + } else { + irq_bit = BIT(HS_RX_THRESHOLD(rsrc->offset)); + stream->pte_offset = ACP_SRAM_HS_CP_PTE_OFFSET; + stream->fifo_offset = HS_CAPT_FIFO_ADDR_OFFSET; + } + break; default: dev_err(dev, "Invalid dai id %x\n", dai->driver->id); return -EINVAL; diff --git a/sound/soc/amd/acp/acp-legacy-mach.c b/sound/soc/amd/acp/acp-legacy-mach.c index 7f04a048ca3a..1f4878ff7d37 100644 --- a/sound/soc/amd/acp/acp-legacy-mach.c +++ b/sound/soc/amd/acp/acp-legacy-mach.c @@ -47,6 +47,28 @@ static struct acp_card_drvdata rt5682s_rt1019_data = { .dmic_codec_id = DMIC, }; +static struct acp_card_drvdata max_nau8825_data = { + .hs_cpu_id = I2S_HS, + .amp_cpu_id = I2S_HS, + .dmic_cpu_id = DMIC, + .hs_codec_id = NAU8825, + .amp_codec_id = MAX98360A, + .dmic_codec_id = DMIC, + .soc_mclk = true, + .platform = REMBRANDT, +}; + +static struct acp_card_drvdata rt5682s_rt1019_rmb_data = { + .hs_cpu_id = I2S_HS, + .amp_cpu_id = I2S_HS, + .dmic_cpu_id = DMIC, + .hs_codec_id = RT5682S, + .amp_codec_id = RT1019, + .dmic_codec_id = DMIC, + .soc_mclk = true, + .platform = REMBRANDT, +}; + static const struct snd_kcontrol_new acp_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone Jack"), SOC_DAPM_PIN_SWITCH("Headset Mic"), @@ -112,6 +134,14 @@ static const struct platform_device_id board_ids[] = { .name = "acp3xalc5682s1019", .driver_data = (kernel_ulong_t)&rt5682s_rt1019_data, }, + { + .name = "rmb-nau8825-max", + .driver_data = (kernel_ulong_t)&max_nau8825_data, + }, + { + .name = "rmb-rt5682s-rt1019", + .driver_data = (kernel_ulong_t)&rt5682s_rt1019_rmb_data, + }, { } }; static struct platform_driver acp_asoc_audio = { @@ -130,4 +160,6 @@ MODULE_DESCRIPTION("ACP chrome audio support"); MODULE_ALIAS("platform:acp3xalc56821019"); MODULE_ALIAS("platform:acp3xalc5682sm98360"); MODULE_ALIAS("platform:acp3xalc5682s1019"); +MODULE_ALIAS("platform:rmb-nau8825-max"); +MODULE_ALIAS("platform:rmb-rt5682s-rt1019"); MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/amd/acp/acp-mach-common.c b/sound/soc/amd/acp/acp-mach-common.c index 86145398fa25..f0c49127aad1 100644 --- a/sound/soc/amd/acp/acp-mach-common.c +++ b/sound/soc/amd/acp/acp-mach-common.c @@ -545,6 +545,12 @@ static struct snd_soc_dai_link_component platform_component[] = { } }; +static struct snd_soc_dai_link_component platform_rmb_component[] = { + { + .name = "acp_asoc_rembrandt.0", + } +}; + static struct snd_soc_dai_link_component sof_component[] = { { .name = "0000:04:00.5", @@ -553,6 +559,8 @@ static struct snd_soc_dai_link_component sof_component[] = { SND_SOC_DAILINK_DEF(i2s_sp, DAILINK_COMP_ARRAY(COMP_CPU("acp-i2s-sp"))); +SND_SOC_DAILINK_DEF(i2s_hs, + DAILINK_COMP_ARRAY(COMP_CPU("acp-i2s-hs"))); SND_SOC_DAILINK_DEF(sof_sp, DAILINK_COMP_ARRAY(COMP_CPU("acp-sof-sp"))); SND_SOC_DAILINK_DEF(sof_hs, @@ -774,6 +782,40 @@ int acp_legacy_dai_links_create(struct snd_soc_card *card) i++; } + if (drv_data->hs_cpu_id == I2S_HS) { + links[i].name = "acp-headset-codec"; + links[i].id = HEADSET_BE_ID; + links[i].cpus = i2s_hs; + links[i].num_cpus = ARRAY_SIZE(i2s_hs); + if (drv_data->platform == REMBRANDT) { + links[i].platforms = platform_rmb_component; + links[i].num_platforms = ARRAY_SIZE(platform_rmb_component); + } else { + links[i].platforms = platform_component; + links[i].num_platforms = ARRAY_SIZE(platform_component); + } + links[i].dpcm_playback = 1; + links[i].dpcm_capture = 1; + if (!drv_data->hs_codec_id) { + /* Use dummy codec if codec id not specified */ + links[i].codecs = dummy_codec; + links[i].num_codecs = ARRAY_SIZE(dummy_codec); + } + if (drv_data->hs_codec_id == NAU8825) { + links[i].codecs = nau8825; + links[i].num_codecs = ARRAY_SIZE(nau8825); + links[i].init = acp_card_nau8825_init; + links[i].ops = &acp_card_nau8825_ops; + } + if (drv_data->hs_codec_id == RT5682S) { + links[i].codecs = rt5682s; + links[i].num_codecs = ARRAY_SIZE(rt5682s); + links[i].init = acp_card_rt5682s_init; + links[i].ops = &acp_card_rt5682s_ops; + } + i++; + } + if (drv_data->amp_cpu_id == I2S_SP) { links[i].name = "acp-amp-codec"; links[i].id = AMP_BE_ID; @@ -804,6 +846,41 @@ int acp_legacy_dai_links_create(struct snd_soc_card *card) i++; } + if (drv_data->amp_cpu_id == I2S_HS) { + links[i].name = "acp-amp-codec"; + links[i].id = AMP_BE_ID; + links[i].cpus = i2s_hs; + links[i].num_cpus = ARRAY_SIZE(i2s_hs); + if (drv_data->platform == REMBRANDT) { + links[i].platforms = platform_rmb_component; + links[i].num_platforms = ARRAY_SIZE(platform_rmb_component); + } else { + links[i].platforms = platform_component; + links[i].num_platforms = ARRAY_SIZE(platform_component); + } + links[i].dpcm_playback = 1; + if (!drv_data->amp_codec_id) { + /* Use dummy codec if codec id not specified */ + links[i].codecs = dummy_codec; + links[i].num_codecs = ARRAY_SIZE(dummy_codec); + } + if (drv_data->amp_codec_id == MAX98360A) { + links[i].codecs = max98360a; + links[i].num_codecs = ARRAY_SIZE(max98360a); + links[i].ops = &acp_card_maxim_ops; + links[i].init = acp_card_maxim_init; + } + if (drv_data->amp_codec_id == RT1019) { + links[i].codecs = rt1019; + links[i].num_codecs = ARRAY_SIZE(rt1019); + links[i].ops = &acp_card_rt1019_ops; + links[i].init = acp_card_rt1019_init; + card->codec_conf = rt1019_conf; + card->num_configs = ARRAY_SIZE(rt1019_conf); + } + i++; + } + if (drv_data->dmic_cpu_id == DMIC) { links[i].name = "acp-dmic-codec"; links[i].id = DMIC_BE_ID; @@ -817,8 +894,13 @@ int acp_legacy_dai_links_create(struct snd_soc_card *card) } links[i].cpus = pdm_dmic; links[i].num_cpus = ARRAY_SIZE(pdm_dmic); - links[i].platforms = platform_component; - links[i].num_platforms = ARRAY_SIZE(platform_component); + if (drv_data->platform == REMBRANDT) { + links[i].platforms = platform_rmb_component; + links[i].num_platforms = ARRAY_SIZE(platform_rmb_component); + } else { + links[i].platforms = platform_component; + links[i].num_platforms = ARRAY_SIZE(platform_component); + } links[i].ops = &acp_card_dmic_ops; links[i].dpcm_capture = 1; } diff --git a/sound/soc/amd/acp/acp-mach.h b/sound/soc/amd/acp/acp-mach.h index c95ee1c52eb1..20583ef902df 100644 --- a/sound/soc/amd/acp/acp-mach.h +++ b/sound/soc/amd/acp/acp-mach.h @@ -41,6 +41,11 @@ enum codec_endpoints { NAU8825, }; +enum platform_end_point { + RENOIR = 0, + REMBRANDT, +}; + struct acp_card_drvdata { unsigned int hs_cpu_id; unsigned int amp_cpu_id; @@ -49,6 +54,7 @@ struct acp_card_drvdata { unsigned int amp_codec_id; unsigned int dmic_codec_id; unsigned int dai_fmt; + unsigned int platform; struct clk *wclk; struct clk *bclk; bool soc_mclk; diff --git a/sound/soc/amd/acp/acp-pci.c b/sound/soc/amd/acp/acp-pci.c index c893963ee2d0..c03bcd31fc95 100644 --- a/sound/soc/amd/acp/acp-pci.c +++ b/sound/soc/amd/acp/acp-pci.c @@ -82,6 +82,12 @@ static int acp_pci_probe(struct pci_dev *pci, const struct pci_device_id *pci_id chip->name = "acp_asoc_renoir"; chip->acp_rev = ACP3X_DEV; break; + case 0x6f: + res_acp = acp3x_res; + num_res = ARRAY_SIZE(acp3x_res); + chip->name = "acp_asoc_rembrandt"; + chip->acp_rev = ACP6X_DEV; + break; default: dev_err(dev, "Unsupported device revision:0x%x\n", pci->revision); return -EINVAL; diff --git a/sound/soc/amd/acp/acp-platform.c b/sound/soc/amd/acp/acp-platform.c index e93c9e478cfa..327e17736dbd 100644 --- a/sound/soc/amd/acp/acp-platform.c +++ b/sound/soc/amd/acp/acp-platform.c @@ -94,11 +94,14 @@ static irqreturn_t i2s_irq_handler(int irq, void *data) struct acp_resource *rsrc = adata->rsrc; struct acp_stream *stream; u16 i2s_flag = 0; - u32 val, i; + u32 val, val1, i; if (!adata) return IRQ_NONE; + if (adata->rsrc->no_of_ctrls == 2) + val1 = readl(ACP_EXTERNAL_INTR_STAT(adata, (rsrc->irqp_used - 1))); + val = readl(ACP_EXTERNAL_INTR_STAT(adata, rsrc->irqp_used)); for (i = 0; i < ACP_MAX_STREAM; i++) { @@ -110,8 +113,16 @@ static irqreturn_t i2s_irq_handler(int irq, void *data) i2s_flag = 1; break; } + if (adata->rsrc->no_of_ctrls == 2) { + if (stream && (val1 & stream->irq_bit)) { + writel(stream->irq_bit, ACP_EXTERNAL_INTR_STAT(adata, + (rsrc->irqp_used - 1))); + snd_pcm_period_elapsed(stream->substream); + i2s_flag = 1; + break; + } + } } - if (i2s_flag) return IRQ_HANDLED; @@ -132,6 +143,7 @@ static void config_pte_for_stream(struct acp_dev_data *adata, struct acp_stream reg_val = rsrc->sram_pte_offset; writel(reg_val | BIT(31), adata->acp_base + pte_reg); writel(PAGE_SIZE_4K_ENABLE, adata->acp_base + pte_size); + writel(0x01, adata->acp_base + ACPAXI2AXI_ATU_CTRL); } static void config_acp_dma(struct acp_dev_data *adata, int cpu_id, int size) diff --git a/sound/soc/amd/acp/acp-rembrandt.c b/sound/soc/amd/acp/acp-rembrandt.c new file mode 100644 index 000000000000..2b57c0ca4e99 --- /dev/null +++ b/sound/soc/amd/acp/acp-rembrandt.c @@ -0,0 +1,401 @@ +// SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) +// +// This file is provided under a dual BSD/GPLv2 license. When using or +// redistributing this file, you may do so under either license. +// +// Copyright(c) 2022 Advanced Micro Devices, Inc. +// +// Authors: Ajit Kumar Pandey +// V sujith kumar Reddy +/* + * Hardware interface for Renoir ACP block + */ + +#include +#include +#include +#include +#include +#include +#include +#include + +#include "amd.h" + +#define DRV_NAME "acp_asoc_rembrandt" + +#define ACP6X_PGFSM_CONTROL 0x1024 +#define ACP6X_PGFSM_STATUS 0x1028 + +#define ACP_SOFT_RESET_SOFTRESET_AUDDONE_MASK 0x00010001 + +#define ACP_PGFSM_CNTL_POWER_ON_MASK 0x01 +#define ACP_PGFSM_CNTL_POWER_OFF_MASK 0x00 +#define ACP_PGFSM_STATUS_MASK 0x03 +#define ACP_POWERED_ON 0x00 +#define ACP_POWER_ON_IN_PROGRESS 0x01 +#define ACP_POWERED_OFF 0x02 +#define ACP_POWER_OFF_IN_PROGRESS 0x03 + +#define ACP_ERROR_MASK 0x20000000 +#define ACP_EXT_INTR_STAT_CLEAR_MASK 0xFFFFFFFF + + +static int rmb_acp_init(void __iomem *base); +static int rmb_acp_deinit(void __iomem *base); + +static struct acp_resource rsrc = { + .offset = 0, + .no_of_ctrls = 2, + .irqp_used = 1, + .soc_mclk = true, + .irq_reg_offset = 0x1a00, + .i2s_pin_cfg_offset = 0x1440, + .i2s_mode = 0x0a, + .scratch_reg_offset = 0x12800, + .sram_pte_offset = 0x03802800, +}; + +static struct snd_soc_acpi_codecs amp_rt1019 = { + .num_codecs = 1, + .codecs = {"10EC1019"} +}; + +static struct snd_soc_acpi_codecs amp_max = { + .num_codecs = 1, + .codecs = {"MX98360A"} +}; + +static struct snd_soc_acpi_mach snd_soc_acpi_amd_rmb_acp_machines[] = { + { + .id = "10508825", + .drv_name = "rmb-nau8825-max", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &_max, + }, + { + .id = "AMDI0007", + .drv_name = "rembrandt-acp", + }, + { + .id = "RTL5682", + .drv_name = "rmb-rt5682s-rt1019", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &_rt1019, + }, + {}, +}; + +static struct snd_soc_dai_driver acp_rmb_dai[] = { +{ + .name = "acp-i2s-sp", + .id = I2S_SP_INSTANCE, + .playback = { + .stream_name = "I2S SP Playback", + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 2, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 96000, + }, + .capture = { + .stream_name = "I2S SP Capture", + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 2, + .channels_max = 2, + .rate_min = 8000, + .rate_max = 48000, + }, + .ops = &asoc_acp_cpu_dai_ops, + .probe = &asoc_acp_i2s_probe, +}, +{ + .name = "acp-i2s-bt", + .id = I2S_BT_INSTANCE, + .playback = { + .stream_name = "I2S BT Playback", + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 2, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 96000, + }, + .capture = { + .stream_name = "I2S BT Capture", + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 2, + .channels_max = 2, + .rate_min = 8000, + .rate_max = 48000, + }, + .ops = &asoc_acp_cpu_dai_ops, + .probe = &asoc_acp_i2s_probe, +}, +{ + .name = "acp-i2s-hs", + .id = I2S_HS_INSTANCE, + .playback = { + .stream_name = "I2S HS Playback", + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 2, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 96000, + }, + .capture = { + .stream_name = "I2S HS Capture", + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 2, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 48000, + }, + .ops = &asoc_acp_cpu_dai_ops, + .probe = &asoc_acp_i2s_probe, +}, +{ + .name = "acp-pdm-dmic", + .id = DMIC_INSTANCE, + .capture = { + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 2, + .channels_max = 2, + .rate_min = 8000, + .rate_max = 48000, + }, + .ops = &acp_dmic_dai_ops, +}, +}; + +static int acp6x_power_on(void __iomem *base) +{ + u32 val; + int timeout; + + val = readl(base + ACP6X_PGFSM_STATUS); + + if (val == ACP_POWERED_ON) + return 0; + + if ((val & ACP_PGFSM_STATUS_MASK) != + ACP_POWER_ON_IN_PROGRESS) + writel(ACP_PGFSM_CNTL_POWER_ON_MASK, + base + ACP6X_PGFSM_CONTROL); + timeout = 0; + while (++timeout < 500) { + val = readl(base + ACP6X_PGFSM_STATUS); + if (!val) + return 0; + udelay(1); + } + return -ETIMEDOUT; +} + +static int acp6x_power_off(void __iomem *base) +{ + u32 val; + int timeout; + + writel(ACP_PGFSM_CNTL_POWER_OFF_MASK, + base + ACP6X_PGFSM_CONTROL); + timeout = 0; + while (++timeout < 500) { + val = readl(base + ACP6X_PGFSM_STATUS); + if ((val & ACP_PGFSM_STATUS_MASK) == ACP_POWERED_OFF) + return 0; + udelay(1); + } + return -ETIMEDOUT; +} + +static int acp6x_reset(void __iomem *base) +{ + u32 val; + int timeout; + + writel(1, base + ACP_SOFT_RESET); + timeout = 0; + while (++timeout < 500) { + val = readl(base + ACP_SOFT_RESET); + if (val & ACP_SOFT_RESET_SOFTRESET_AUDDONE_MASK) + break; + cpu_relax(); + } + writel(0, base + ACP_SOFT_RESET); + timeout = 0; + while (++timeout < 500) { + val = readl(base + ACP_SOFT_RESET); + if (!val) + return 0; + cpu_relax(); + } + return -ETIMEDOUT; +} + +static void acp6x_enable_interrupts(struct acp_dev_data *adata) +{ + struct acp_resource *rsrc = adata->rsrc; + u32 ext_intr_ctrl; + + writel(0x01, ACP_EXTERNAL_INTR_ENB(adata)); + ext_intr_ctrl = readl(ACP_EXTERNAL_INTR_CNTL(adata, rsrc->irqp_used)); + ext_intr_ctrl |= ACP_ERROR_MASK; + writel(ext_intr_ctrl, ACP_EXTERNAL_INTR_CNTL(adata, rsrc->irqp_used)); +} + +static void acp6x_disable_interrupts(struct acp_dev_data *adata) +{ + struct acp_resource *rsrc = adata->rsrc; + + writel(ACP_EXT_INTR_STAT_CLEAR_MASK, + ACP_EXTERNAL_INTR_STAT(adata, rsrc->irqp_used)); + writel(0x00, ACP_EXTERNAL_INTR_ENB(adata)); +} + +static int rmb_acp_init(void __iomem *base) +{ + int ret; + + /* power on */ + ret = acp6x_power_on(base); + if (ret) { + pr_err("ACP power on failed\n"); + return ret; + } + writel(0x01, base + ACP_CONTROL); + + /* Reset */ + ret = acp6x_reset(base); + if (ret) { + pr_err("ACP reset failed\n"); + return ret; + } + + return 0; +} + +static int rmb_acp_deinit(void __iomem *base) +{ + int ret = 0; + + /* Reset */ + ret = acp6x_reset(base); + if (ret) { + pr_err("ACP reset failed\n"); + return ret; + } + + writel(0x00, base + ACP_CONTROL); + + /* power off */ + ret = acp6x_power_off(base); + if (ret) { + pr_err("ACP power off failed\n"); + return ret; + } + + return 0; +} + +static int rembrandt_audio_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct acp_chip_info *chip; + struct acp_dev_data *adata; + struct resource *res; + + chip = dev_get_platdata(&pdev->dev); + if (!chip || !chip->base) { + dev_err(&pdev->dev, "ACP chip data is NULL\n"); + return -ENODEV; + } + + if (chip->acp_rev != ACP6X_DEV) { + dev_err(&pdev->dev, "Un-supported ACP Revision %d\n", chip->acp_rev); + return -ENODEV; + } + + rmb_acp_init(chip->base); + + adata = devm_kzalloc(dev, sizeof(struct acp_dev_data), GFP_KERNEL); + if (!adata) + return -ENOMEM; + + res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "acp_mem"); + if (!res) { + dev_err(&pdev->dev, "IORESOURCE_MEM FAILED\n"); + return -ENODEV; + } + + adata->acp_base = devm_ioremap(&pdev->dev, res->start, resource_size(res)); + if (!adata->acp_base) + return -ENOMEM; + + res = platform_get_resource_byname(pdev, IORESOURCE_IRQ, "acp_dai_irq"); + if (!res) { + dev_err(&pdev->dev, "IORESOURCE_IRQ FAILED\n"); + return -ENODEV; + } + + adata->i2s_irq = res->start; + adata->dev = dev; + adata->dai_driver = acp_rmb_dai; + adata->num_dai = ARRAY_SIZE(acp_rmb_dai); + adata->rsrc = &rsrc; + + adata->machines = snd_soc_acpi_amd_rmb_acp_machines; + acp_machine_select(adata); + + dev_set_drvdata(dev, adata); + acp6x_enable_interrupts(adata); + acp_platform_register(dev); + + return 0; +} + +static int rembrandt_audio_remove(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct acp_dev_data *adata = dev_get_drvdata(dev); + struct acp_chip_info *chip; + + chip = dev_get_platdata(&pdev->dev); + if (!chip || !chip->base) { + dev_err(&pdev->dev, "ACP chip data is NULL\n"); + return -ENODEV; + } + + rmb_acp_deinit(chip->base); + + acp6x_disable_interrupts(adata); + acp_platform_unregister(dev); + return 0; +} + +static struct platform_driver rembrandt_driver = { + .probe = rembrandt_audio_probe, + .remove = rembrandt_audio_remove, + .driver = { + .name = "acp_asoc_rembrandt", + }, +}; + +module_platform_driver(rembrandt_driver); + +MODULE_DESCRIPTION("AMD ACP Rembrandt Driver"); +MODULE_IMPORT_NS(SND_SOC_ACP_COMMON); +MODULE_LICENSE("Dual BSD/GPL"); +MODULE_ALIAS("platform:" DRV_NAME); diff --git a/sound/soc/amd/acp/amd.h b/sound/soc/amd/acp/amd.h index 186cb8b26175..af9603724a68 100644 --- a/sound/soc/amd/acp/amd.h +++ b/sound/soc/amd/acp/amd.h @@ -19,10 +19,12 @@ #include "chip_offset_byte.h" #define ACP3X_DEV 3 +#define ACP6X_DEV 6 #define I2S_SP_INSTANCE 0x00 #define I2S_BT_INSTANCE 0x01 #define DMIC_INSTANCE 0x02 +#define I2S_HS_INSTANCE 0x03 #define MEM_WINDOW_START 0x4080000 @@ -38,23 +40,31 @@ #define I2S_TX_THRESHOLD(base) THRESHOLD(8, base) #define BT_TX_THRESHOLD(base) THRESHOLD(6, base) #define BT_RX_THRESHOLD(base) THRESHOLD(5, base) +#define HS_TX_THRESHOLD(base) THRESHOLD(4, base) +#define HS_RX_THRESHOLD(base) THRESHOLD(3, base) #define ACP_SRAM_SP_PB_PTE_OFFSET 0x0 #define ACP_SRAM_SP_CP_PTE_OFFSET 0x100 #define ACP_SRAM_BT_PB_PTE_OFFSET 0x200 #define ACP_SRAM_BT_CP_PTE_OFFSET 0x300 #define ACP_SRAM_PDM_PTE_OFFSET 0x400 +#define ACP_SRAM_HS_PB_PTE_OFFSET 0x500 +#define ACP_SRAM_HS_CP_PTE_OFFSET 0x600 #define PAGE_SIZE_4K_ENABLE 0x2 #define I2S_SP_TX_MEM_WINDOW_START 0x4000000 #define I2S_SP_RX_MEM_WINDOW_START 0x4020000 #define I2S_BT_TX_MEM_WINDOW_START 0x4040000 #define I2S_BT_RX_MEM_WINDOW_START 0x4060000 +#define I2S_HS_TX_MEM_WINDOW_START 0x40A0000 +#define I2S_HS_RX_MEM_WINDOW_START 0x40C0000 #define SP_PB_FIFO_ADDR_OFFSET 0x500 #define SP_CAPT_FIFO_ADDR_OFFSET 0x700 #define BT_PB_FIFO_ADDR_OFFSET 0x900 #define BT_CAPT_FIFO_ADDR_OFFSET 0xB00 +#define HS_PB_FIFO_ADDR_OFFSET 0xD00 +#define HS_CAPT_FIFO_ADDR_OFFSET 0xF00 #define PLAYBACK_MIN_NUM_PERIODS 2 #define PLAYBACK_MAX_NUM_PERIODS 8 #define PLAYBACK_MAX_PERIOD_SIZE 8192 @@ -72,7 +82,7 @@ #define ACP3x_ITER_IRER_SAMP_LEN_MASK 0x38 -#define ACP_MAX_STREAM 6 +#define ACP_MAX_STREAM 8 struct acp_chip_info { char *name; /* Platform name */ @@ -95,6 +105,7 @@ struct acp_resource { int offset; int no_of_ctrls; int irqp_used; + bool soc_mclk; u32 irq_reg_offset; u32 i2s_pin_cfg_offset; int i2s_mode; @@ -117,9 +128,23 @@ struct acp_dev_data { struct snd_soc_acpi_mach *machines; struct platform_device *mach_dev; + u32 bclk_div; + u32 lrclk_div; + struct acp_resource *rsrc; }; +union acp_i2stdm_mstrclkgen { + struct { + u32 i2stdm_master_mode : 1; + u32 i2stdm_format_mode : 1; + u32 i2stdm_lrclk_div_val : 9; + u32 i2stdm_bclk_div_val : 11; + u32:10; + } bitfields, bits; + u32 u32_all; +}; + extern const struct snd_soc_dai_ops asoc_acp_cpu_dai_ops; extern const struct snd_soc_dai_ops acp_dmic_dai_ops; @@ -146,6 +171,10 @@ static inline u64 acp_get_byte_count(struct acp_dev_data *adata, int dai_id, int high = readl(adata->acp_base + ACP_I2S_TX_LINEARPOSITIONCNTR_HIGH); low = readl(adata->acp_base + ACP_I2S_TX_LINEARPOSITIONCNTR_LOW); break; + case I2S_HS_INSTANCE: + high = readl(adata->acp_base + ACP_HS_TX_LINEARPOSITIONCNTR_HIGH); + low = readl(adata->acp_base + ACP_HS_TX_LINEARPOSITIONCNTR_LOW); + break; default: dev_err(adata->dev, "Invalid dai id %x\n", dai_id); return -EINVAL; @@ -160,6 +189,10 @@ static inline u64 acp_get_byte_count(struct acp_dev_data *adata, int dai_id, int high = readl(adata->acp_base + ACP_I2S_RX_LINEARPOSITIONCNTR_HIGH); low = readl(adata->acp_base + ACP_I2S_RX_LINEARPOSITIONCNTR_LOW); break; + case I2S_HS_INSTANCE: + high = readl(adata->acp_base + ACP_HS_RX_LINEARPOSITIONCNTR_HIGH); + low = readl(adata->acp_base + ACP_HS_RX_LINEARPOSITIONCNTR_LOW); + break; case DMIC_INSTANCE: high = readl(adata->acp_base + ACP_WOV_RX_LINEARPOSITIONCNTR_HIGH); low = readl(adata->acp_base + ACP_WOV_RX_LINEARPOSITIONCNTR_LOW); @@ -175,4 +208,31 @@ static inline u64 acp_get_byte_count(struct acp_dev_data *adata, int dai_id, int return byte_count; } +static inline void acp_set_i2s_clk(struct acp_dev_data *adata, int dai_id) +{ + union acp_i2stdm_mstrclkgen mclkgen; + u32 master_reg; + + switch (dai_id) { + case I2S_SP_INSTANCE: + master_reg = ACP_I2STDM0_MSTRCLKGEN; + break; + case I2S_BT_INSTANCE: + master_reg = ACP_I2STDM1_MSTRCLKGEN; + break; + case I2S_HS_INSTANCE: + master_reg = ACP_I2STDM2_MSTRCLKGEN; + break; + default: + master_reg = ACP_I2STDM0_MSTRCLKGEN; + break; + } + + mclkgen.bits.i2stdm_master_mode = 0x1; + mclkgen.bits.i2stdm_format_mode = 0x00; + + mclkgen.bits.i2stdm_bclk_div_val = adata->bclk_div; + mclkgen.bits.i2stdm_lrclk_div_val = adata->lrclk_div; + writel(mclkgen.u32_all, adata->acp_base + master_reg); +} #endif diff --git a/sound/soc/amd/acp/chip_offset_byte.h b/sound/soc/amd/acp/chip_offset_byte.h index fff7e80475ba..ce3948e0679c 100644 --- a/sound/soc/amd/acp/chip_offset_byte.h +++ b/sound/soc/amd/acp/chip_offset_byte.h @@ -66,6 +66,24 @@ #define ACP_BT_TX_LINEARPOSITIONCNTR_HIGH 0x2084 #define ACP_BT_TX_LINEARPOSITIONCNTR_LOW 0x2088 #define ACP_BT_TX_INTR_WATERMARK_SIZE 0x208C +#define ACP_HS_RX_RINGBUFADDR 0x3A90 +#define ACP_HS_RX_RINGBUFSIZE 0x3A94 +#define ACP_HS_RX_LINKPOSITIONCNTR 0x3A98 +#define ACP_HS_RX_FIFOADDR 0x3A9C +#define ACP_HS_RX_FIFOSIZE 0x3AA0 +#define ACP_HS_RX_DMA_SIZE 0x3AA4 +#define ACP_HS_RX_LINEARPOSITIONCNTR_HIGH 0x3AA8 +#define ACP_HS_RX_LINEARPOSITIONCNTR_LOW 0x3AAC +#define ACP_HS_RX_INTR_WATERMARK_SIZE 0x3AB0 +#define ACP_HS_TX_RINGBUFADDR 0x3AB4 +#define ACP_HS_TX_RINGBUFSIZE 0x3AB8 +#define ACP_HS_TX_LINKPOSITIONCNTR 0x3ABC +#define ACP_HS_TX_FIFOADDR 0x3AC0 +#define ACP_HS_TX_FIFOSIZE 0x3AC4 +#define ACP_HS_TX_DMA_SIZE 0x3AC8 +#define ACP_HS_TX_LINEARPOSITIONCNTR_HIGH 0x3ACC +#define ACP_HS_TX_LINEARPOSITIONCNTR_LOW 0x3AD0 +#define ACP_HS_TX_INTR_WATERMARK_SIZE 0x3AD4 #define ACP_I2STDM_IER 0x2400 #define ACP_I2STDM_IRER 0x2404 @@ -81,6 +99,13 @@ #define ACP_BTTDM_ITER 0x280C #define ACP_BTTDM_TXFRMT 0x2810 +/* Registers from ACP_HS_TDM block */ +#define ACP_HSTDM_IER 0x2814 +#define ACP_HSTDM_IRER 0x2818 +#define ACP_HSTDM_RXFRMT 0x281C +#define ACP_HSTDM_ITER 0x2820 +#define ACP_HSTDM_TXFRMT 0x2824 + /* Registers from ACP_WOV_PDM block */ #define ACP_WOV_PDM_ENABLE 0x2C04 @@ -101,4 +126,7 @@ #define ACP_PDM_VAD_DYNAMIC_CLK_GATING_EN 0x2C64 #define ACP_WOV_ERROR_STATUS_REGISTER 0x2C68 +#define ACP_I2STDM0_MSTRCLKGEN 0x2414 +#define ACP_I2STDM1_MSTRCLKGEN 0x2418 +#define ACP_I2STDM2_MSTRCLKGEN 0x241C #endif -- cgit v1.2.3 From c50cea054e04769471d2f17a57fafd7c5dfe8df8 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Thu, 7 Jul 2022 14:41:42 +0200 Subject: ASoC: Intel: avs: Register HDAudio ext-bus operations With ASoC representation of HDAudio codec added, update bus initiazation to complete it. Signed-off-by: Cezary Rojewski Link: https://lore.kernel.org/r/20220707124153.1858249-2-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/Kconfig | 2 +- sound/soc/intel/avs/core.c | 3 ++- 2 files changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index e5107a3ce16a..ded903f95b67 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -216,7 +216,7 @@ config SND_SOC_INTEL_AVS depends on COMMON_CLK select SND_SOC_ACPI if ACPI select SND_SOC_TOPOLOGY - select SND_HDA + select SND_SOC_HDA select SND_HDA_EXT_CORE select SND_HDA_DSP_LOADER select SND_INTEL_DSP_CONFIG diff --git a/sound/soc/intel/avs/core.c b/sound/soc/intel/avs/core.c index 3a0997c3af2b..664f87c33e9d 100644 --- a/sound/soc/intel/avs/core.c +++ b/sound/soc/intel/avs/core.c @@ -23,6 +23,7 @@ #include #include #include +#include "../../codecs/hda.h" #include "avs.h" #include "cldma.h" @@ -356,7 +357,7 @@ static int avs_bus_init(struct avs_dev *adev, struct pci_dev *pci, const struct struct device *dev = &pci->dev; int ret; - ret = snd_hdac_ext_bus_init(&bus->core, dev, NULL, NULL); + ret = snd_hdac_ext_bus_init(&bus->core, dev, NULL, &soc_hda_ext_bus_ops); if (ret < 0) return ret; -- cgit v1.2.3 From 5f267aa4adad13f764e0b00926c349f8728fce77 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Thu, 7 Jul 2022 14:41:43 +0200 Subject: ASoC: Intel: avs: Assign I2S gateway when parsing topology For formatted port - ssp%d - descriptions to have an effect, copier module templates need to be updated with specified port value. This value is later propagated to the firmware when module instances are being instantiated. Signed-off-by: Cezary Rojewski Link: https://lore.kernel.org/r/20220707124153.1858249-3-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/topology.c | 27 +++++++++++++++++++++++++++ 1 file changed, 27 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/avs/topology.c b/sound/soc/intel/avs/topology.c index 6a06fe387d13..8a9f9fc48938 100644 --- a/sound/soc/intel/avs/topology.c +++ b/sound/soc/intel/avs/topology.c @@ -808,6 +808,30 @@ static const struct avs_tplg_token_parser pin_format_parsers[] = { }, }; +static void +assign_copier_gtw_instance(struct snd_soc_component *comp, struct avs_tplg_modcfg_ext *cfg) +{ + struct snd_soc_acpi_mach *mach; + + if (!guid_equal(&cfg->type, &AVS_COPIER_MOD_UUID)) + return; + + /* Only I2S boards assign port instance in ->i2s_link_mask. */ + switch (cfg->copier.dma_type) { + case AVS_DMA_I2S_LINK_OUTPUT: + case AVS_DMA_I2S_LINK_INPUT: + break; + default: + return; + } + + mach = dev_get_platdata(comp->card->dev); + + /* Automatic assignment only when board describes single SSP. */ + if (hweight_long(mach->mach_params.i2s_link_mask) == 1 && !cfg->copier.vindex.i2s.instance) + cfg->copier.vindex.i2s.instance = __ffs(mach->mach_params.i2s_link_mask); +} + static int avs_tplg_parse_modcfg_ext(struct snd_soc_component *comp, struct avs_tplg_modcfg_ext *cfg, struct snd_soc_tplg_vendor_array *tuples, @@ -827,6 +851,9 @@ static int avs_tplg_parse_modcfg_ext(struct snd_soc_component *comp, if (ret) return ret; + /* Update copier gateway based on board's i2s_link_mask. */ + assign_copier_gtw_instance(comp, cfg); + block_size -= esize; /* Parse trailing in/out pin formats if any. */ if (block_size) { -- cgit v1.2.3 From 8192d24cccfbd93dadefd2b7553ff15e41d0e680 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Thu, 7 Jul 2022 14:41:44 +0200 Subject: ASoC: Intel: avs: Relax DSP core transition timings To avoid any false positives when checking CPA after setting SPA, do a short wait. For stall operation, give HW more time to propagate the change before moving on. Signed-off-by: Cezary Rojewski Link: https://lore.kernel.org/r/20220707124153.1858249-4-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/dsp.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/avs/dsp.c b/sound/soc/intel/avs/dsp.c index 06d2f7af520f..b881100d3e02 100644 --- a/sound/soc/intel/avs/dsp.c +++ b/sound/soc/intel/avs/dsp.c @@ -13,6 +13,7 @@ #define AVS_ADSPCS_INTERVAL_US 500 #define AVS_ADSPCS_TIMEOUT_US 50000 +#define AVS_ADSPCS_DELAY_US 1000 int avs_dsp_core_power(struct avs_dev *adev, u32 core_mask, bool power) { @@ -26,6 +27,8 @@ int avs_dsp_core_power(struct avs_dev *adev, u32 core_mask, bool power) value = power ? mask : 0; snd_hdac_adsp_updatel(adev, AVS_ADSP_REG_ADSPCS, mask, value); + /* Delay the polling to avoid false positives. */ + usleep_range(AVS_ADSPCS_DELAY_US, 2 * AVS_ADSPCS_DELAY_US); mask = AVS_ADSPCS_CPA_MASK(core_mask); value = power ? mask : 0; @@ -82,11 +85,15 @@ int avs_dsp_core_stall(struct avs_dev *adev, u32 core_mask, bool stall) reg, (reg & mask) == value, AVS_ADSPCS_INTERVAL_US, AVS_ADSPCS_TIMEOUT_US); - if (ret) + if (ret) { dev_err(adev->dev, "core_mask %d %sstall failed: %d\n", core_mask, stall ? "" : "un", ret); + return ret; + } - return ret; + /* Give HW time to propagate the change. */ + usleep_range(AVS_ADSPCS_DELAY_US, 2 * AVS_ADSPCS_DELAY_US); + return 0; } int avs_dsp_core_enable(struct avs_dev *adev, u32 core_mask) -- cgit v1.2.3 From 3c1923a119a61534f8ce221766041ddf470c9307 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Thu, 7 Jul 2022 14:41:45 +0200 Subject: ASoC: Intel: avs: Copy only as many RX bytes as necessary There is no need to copy number of bytes specified by IPC message caller if DSP firmware returned lower number. In consequence, LARGE_CONFIG_GET handler is simplified. Signed-off-by: Cezary Rojewski Link: https://lore.kernel.org/r/20220707124153.1858249-5-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/ipc.c | 1 + sound/soc/intel/avs/messages.c | 6 ++---- 2 files changed, 3 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/avs/ipc.c b/sound/soc/intel/avs/ipc.c index d755ba8b8518..020d85c7520d 100644 --- a/sound/soc/intel/avs/ipc.c +++ b/sound/soc/intel/avs/ipc.c @@ -480,6 +480,7 @@ static int avs_dsp_do_send_msg(struct avs_dev *adev, struct avs_ipc_msg *request ret = ipc->rx.rsp.status; if (reply) { reply->header = ipc->rx.header; + reply->size = ipc->rx.size; if (reply->data && ipc->rx.size) memcpy(reply->data, ipc->rx.data, reply->size); } diff --git a/sound/soc/intel/avs/messages.c b/sound/soc/intel/avs/messages.c index 6404fce8cde4..3559fb496e0b 100644 --- a/sound/soc/intel/avs/messages.c +++ b/sound/soc/intel/avs/messages.c @@ -378,7 +378,6 @@ int avs_ipc_get_large_config(struct avs_dev *adev, u16 module_id, u8 instance_id union avs_module_msg msg = AVS_MODULE_REQUEST(LARGE_CONFIG_GET); struct avs_ipc_msg request; struct avs_ipc_msg reply = {{0}}; - size_t size; void *buf; int ret; @@ -406,15 +405,14 @@ int avs_ipc_get_large_config(struct avs_dev *adev, u16 module_id, u8 instance_id return ret; } - size = reply.rsp.ext.large_config.data_off_size; - buf = krealloc(reply.data, size, GFP_KERNEL); + buf = krealloc(reply.data, reply.size, GFP_KERNEL); if (!buf) { kfree(reply.data); return -ENOMEM; } *reply_data = buf; - *reply_size = size; + *reply_size = reply.size; return 0; } -- cgit v1.2.3 From 00566ad4ce9d394c6ebaacde12eda06eef4e5279 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Thu, 7 Jul 2022 14:41:46 +0200 Subject: ASoC: Intel: avs: Shield LARGE_CONFIG_GETs against zero payload_size Some LARGE_CONFIG_GETs are never expected to return payload of size 0. Check for such situation and collapse if met. Signed-off-by: Cezary Rojewski Link: https://lore.kernel.org/r/20220707124153.1858249-6-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/messages.c | 9 +++++++++ 1 file changed, 9 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/avs/messages.c b/sound/soc/intel/avs/messages.c index 3559fb496e0b..9cf621eaec5a 100644 --- a/sound/soc/intel/avs/messages.c +++ b/sound/soc/intel/avs/messages.c @@ -474,6 +474,9 @@ int avs_ipc_get_fw_config(struct avs_dev *adev, struct avs_fw_cfg *cfg) &payload, &payload_size); if (ret) return ret; + /* Non-zero payload expected for FIRMWARE_CONFIG. */ + if (!payload_size) + return -EREMOTEIO; while (offset < payload_size) { tlv = (struct avs_tlv *)(payload + offset); @@ -587,6 +590,9 @@ int avs_ipc_get_hw_config(struct avs_dev *adev, struct avs_hw_cfg *cfg) &payload, &payload_size); if (ret) return ret; + /* Non-zero payload expected for HARDWARE_CONFIG. */ + if (!payload_size) + return -EREMOTEIO; while (offset < payload_size) { tlv = (struct avs_tlv *)(payload + offset); @@ -670,6 +676,9 @@ int avs_ipc_get_modules_info(struct avs_dev *adev, struct avs_mods_info **info) &payload, &payload_size); if (ret) return ret; + /* Non-zero payload expected for MODULES_INFO. */ + if (!payload_size) + return -EREMOTEIO; *info = (struct avs_mods_info *)payload; return 0; -- cgit v1.2.3 From daa36bbcd78bca24db84e273bcafec9a8f81c767 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Thu, 7 Jul 2022 14:41:47 +0200 Subject: ASoC: Intel: avs: Block IPC channel on suspend To allow for driver's filesystem interfaces e.g.: debugfs, to be touched even when the device is asleep, mark IPC-channel as blocked when the device is suspended. This causes any invocation of said interfaces that do not toggle PM themselves to gracefully fail with "Operation not permitted" message. Signed-off-by: Cezary Rojewski Link: https://lore.kernel.org/r/20220707124153.1858249-7-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/core.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/intel/avs/core.c b/sound/soc/intel/avs/core.c index 664f87c33e9d..4234adeb3d1c 100644 --- a/sound/soc/intel/avs/core.c +++ b/sound/soc/intel/avs/core.c @@ -556,6 +556,7 @@ static int __maybe_unused avs_suspend_common(struct avs_dev *adev) return AVS_IPC_RET(ret); } + avs_ipc_block(adev->ipc); avs_dsp_op(adev, int_control, false); snd_hdac_ext_bus_ppcap_int_enable(bus, false); -- cgit v1.2.3 From 8544eebc78c96f1834a46b26ade3e7ebe785d10c Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Thu, 7 Jul 2022 14:41:48 +0200 Subject: ASoC: Intel: avs: Set max DMA segment size MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Apparently it is possible for code to allocate large buffers which may cause warnings as reported in [1]. This was fixed for HDA, SOF and skylake in patchset [2], fix it also for avs driver. [1] https://github.com/thesofproject/linux/issues/3430 [2] https://lore.kernel.org/all/20220215132756.31236-1-tiwai@suse.de/ Signed-off-by: Amadeusz Sławiński Signed-off-by: Cezary Rojewski Link: https://lore.kernel.org/r/20220707124153.1858249-8-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/core.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/intel/avs/core.c b/sound/soc/intel/avs/core.c index 4234adeb3d1c..6a35bf45efcb 100644 --- a/sound/soc/intel/avs/core.c +++ b/sound/soc/intel/avs/core.c @@ -446,6 +446,7 @@ static int avs_pci_probe(struct pci_dev *pci, const struct pci_device_id *id) dma_set_mask(dev, DMA_BIT_MASK(32)); dma_set_coherent_mask(dev, DMA_BIT_MASK(32)); } + dma_set_max_seg_size(dev, UINT_MAX); ret = avs_hdac_bus_init_streams(bus); if (ret < 0) { -- cgit v1.2.3 From a5bbbde2b81e41cea7fa1b38911e88da5febc2d5 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Thu, 7 Jul 2022 14:41:49 +0200 Subject: ASoC: Intel: avs: Use helper function to set up DMA MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit dma_set_mask() and dma_set_coherent_mask() can be performed with one call to dma_set_mask_and_coherent(), which slightly reduces amount of code on our side. Signed-off-by: Amadeusz Sławiński Signed-off-by: Cezary Rojewski Link: https://lore.kernel.org/r/20220707124153.1858249-9-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/core.c | 8 ++------ 1 file changed, 2 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/avs/core.c b/sound/soc/intel/avs/core.c index 6a35bf45efcb..c50c20fd681a 100644 --- a/sound/soc/intel/avs/core.c +++ b/sound/soc/intel/avs/core.c @@ -440,12 +440,8 @@ static int avs_pci_probe(struct pci_dev *pci, const struct pci_device_id *id) if (bus->mlcap) snd_hdac_ext_bus_get_ml_capabilities(bus); - if (!dma_set_mask(dev, DMA_BIT_MASK(64))) { - dma_set_coherent_mask(dev, DMA_BIT_MASK(64)); - } else { - dma_set_mask(dev, DMA_BIT_MASK(32)); - dma_set_coherent_mask(dev, DMA_BIT_MASK(32)); - } + if (!dma_set_mask_and_coherent(dev, DMA_BIT_MASK(64))) + dma_set_mask_and_coherent(dev, DMA_BIT_MASK(32)); dma_set_max_seg_size(dev, UINT_MAX); ret = avs_hdac_bus_init_streams(bus); -- cgit v1.2.3 From 79c351fb50e7e37eacf93f55f1e7056148d0d814 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Thu, 7 Jul 2022 14:41:50 +0200 Subject: ASoC: Intel: avs: Recognize FW_CFG_RESERVED If exposed by firmware, count RESERVED parameter as known one to avoid dumping noise in kernel logs. Signed-off-by: Cezary Rojewski Link: https://lore.kernel.org/r/20220707124153.1858249-10-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/messages.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/intel/avs/messages.c b/sound/soc/intel/avs/messages.c index 9cf621eaec5a..28a948cf790f 100644 --- a/sound/soc/intel/avs/messages.c +++ b/sound/soc/intel/avs/messages.c @@ -562,6 +562,7 @@ int avs_ipc_get_fw_config(struct avs_dev *adev, struct avs_fw_cfg *cfg) case AVS_FW_CFG_DMA_BUFFER_CONFIG: case AVS_FW_CFG_SCHEDULER_CONFIG: case AVS_FW_CFG_CLOCKS_CONFIG: + case AVS_FW_CFG_RESERVED: break; default: -- cgit v1.2.3 From 4b38bd16ca6d8b16c1dc2cc4aa61663193b0b893 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Thu, 7 Jul 2022 14:41:51 +0200 Subject: ASoC: Intel: avs: Replace hardcodes with SD_CTL_STREAM_RESET Improve readability of CLDMA reset operation by making use of already defined SD_CTL_STREAM_RESET. Signed-off-by: Cezary Rojewski Link: https://lore.kernel.org/r/20220707124153.1858249-11-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/cldma.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/avs/cldma.c b/sound/soc/intel/avs/cldma.c index d100c6ba4d8a..d7a9390b5e48 100644 --- a/sound/soc/intel/avs/cldma.c +++ b/sound/soc/intel/avs/cldma.c @@ -176,17 +176,17 @@ int hda_cldma_reset(struct hda_cldma *cl) return ret; } - snd_hdac_stream_updateb(cl, SD_CTL, 1, 1); - ret = snd_hdac_stream_readb_poll(cl, SD_CTL, reg, (reg & 1), AVS_CL_OP_INTERVAL_US, - AVS_CL_OP_TIMEOUT_US); + snd_hdac_stream_updateb(cl, SD_CTL, SD_CTL_STREAM_RESET, SD_CTL_STREAM_RESET); + ret = snd_hdac_stream_readb_poll(cl, SD_CTL, reg, (reg & SD_CTL_STREAM_RESET), + AVS_CL_OP_INTERVAL_US, AVS_CL_OP_TIMEOUT_US); if (ret < 0) { dev_err(cl->dev, "cldma set SRST failed: %d\n", ret); return ret; } - snd_hdac_stream_updateb(cl, SD_CTL, 1, 0); - ret = snd_hdac_stream_readb_poll(cl, SD_CTL, reg, !(reg & 1), AVS_CL_OP_INTERVAL_US, - AVS_CL_OP_TIMEOUT_US); + snd_hdac_stream_updateb(cl, SD_CTL, SD_CTL_STREAM_RESET, 0); + ret = snd_hdac_stream_readb_poll(cl, SD_CTL, reg, !(reg & SD_CTL_STREAM_RESET), + AVS_CL_OP_INTERVAL_US, AVS_CL_OP_TIMEOUT_US); if (ret < 0) { dev_err(cl->dev, "cldma unset SRST failed: %d\n", ret); return ret; -- cgit v1.2.3 From 8758ae88f0f4ade16e6a1b709eb5ea7271f62320 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Thu, 7 Jul 2022 14:41:52 +0200 Subject: ASoC: Intel: avs: Lower UNLOAD_MULTIPLE_MODULES IPC timeout Module unloading operation performs memory unmapping and the weight of the opration does not different from any other standard IPC. There is no dependency on secondary task like in module loading scenario where larger message timeout is recommended. Signed-off-by: Cezary Rojewski Link: https://lore.kernel.org/r/20220707124153.1858249-12-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/messages.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/avs/messages.c b/sound/soc/intel/avs/messages.c index 28a948cf790f..d4bcee1aabcf 100644 --- a/sound/soc/intel/avs/messages.c +++ b/sound/soc/intel/avs/messages.c @@ -59,7 +59,7 @@ int avs_ipc_unload_modules(struct avs_dev *adev, u16 *mod_ids, u32 num_mod_ids) request.data = mod_ids; request.size = sizeof(*mod_ids) * num_mod_ids; - ret = avs_dsp_send_msg_timeout(adev, &request, NULL, AVS_CL_TIMEOUT_MS); + ret = avs_dsp_send_msg(adev, &request, NULL); if (ret) avs_ipc_err(adev, &request, "unload multiple modules", ret); -- cgit v1.2.3 From f1eea11523e4394d6670f10a51356e9b7c8567a8 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Thu, 7 Jul 2022 14:41:53 +0200 Subject: ASoC: Intel: avs: Update AVS_FW_INIT_TIMEOUT_US declaration To reduce the number of places to update if timeouts would have to change, modify the constant declaration. Signed-off-by: Cezary Rojewski Link: https://lore.kernel.org/r/20220707124153.1858249-13-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/loader.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/avs/loader.c b/sound/soc/intel/avs/loader.c index 542fd44aa501..9e3f8ff33a87 100644 --- a/sound/soc/intel/avs/loader.c +++ b/sound/soc/intel/avs/loader.c @@ -27,8 +27,8 @@ #define APL_ROM_INIT_RETRIES 3 #define AVS_FW_INIT_POLLING_US 500 -#define AVS_FW_INIT_TIMEOUT_US 3000000 #define AVS_FW_INIT_TIMEOUT_MS 3000 +#define AVS_FW_INIT_TIMEOUT_US (AVS_FW_INIT_TIMEOUT_MS * 1000) #define AVS_CLDMA_START_DELAY_MS 100 -- cgit v1.2.3 From b737fd8cf196bc96e471304007c3cd9b78672069 Mon Sep 17 00:00:00 2001 From: Libin Yang Date: Fri, 8 Jul 2022 15:05:15 -0500 Subject: ASoC: SOF: ipc4-topology: check dai->private in ipc_free() MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Set the swidget->private or dai->private to NULL after kfree in the error handling in ipc_setup(). The private needs to be set NULL because if ipc_setup() returns error, ipc_free() will be called later. ipc_free() will judge the private is NULL or not to do the clearing. For dai widget, dai->private is allocated and set in dai widget ipc_setup(). So we need to check dai->private is NULL or not in the ipc_free(). Fixes: 2cabd02b6090 ("ASoC: SOF: ipc4-topology: Add support for parsing AIF_IN/AIF_OUT widgets") Fixes: abfb536bd116 ("ASoC: SOF: ipc4-topology: Add support for parsing DAI_IN/DAI_OUT widgets") Fixes: 4f838ab20812 ("ASoC: SOF: ipc4-topology: Add support for parsing and preparing pga widgets") Fixes: 4d4ba014ac4b ("ASoC: SOF: ipc4-topology: Add support for parsing mixer widgets") Reviewed-by: Péter Ujfalusi Signed-off-by: Libin Yang Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220708200516.26853-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-topology.c | 11 +++++++++++ 1 file changed, 11 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index 34f805431f2e..2d157ea79db5 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -394,6 +394,7 @@ err: kfree(available_fmt->dma_buffer_size); free_copier: kfree(ipc4_copier); + swidget->private = NULL; return ret; } @@ -541,6 +542,8 @@ err: kfree(available_fmt->dma_buffer_size); free_copier: kfree(ipc4_copier); + dai->private = NULL; + dai->scomp = NULL; return ret; } @@ -553,6 +556,12 @@ static void sof_ipc4_widget_free_comp_dai(struct snd_sof_widget *swidget) if (!dai) return; + if (!dai->private) { + kfree(dai); + swidget->private = NULL; + return; + } + ipc4_copier = dai->private; available_fmt = &ipc4_copier->available_fmt; @@ -669,6 +678,7 @@ static int sof_ipc4_widget_setup_comp_pga(struct snd_sof_widget *swidget) return 0; err: kfree(gain); + swidget->private = NULL; return ret; } @@ -698,6 +708,7 @@ static int sof_ipc4_widget_setup_comp_mixer(struct snd_sof_widget *swidget) return 0; err: kfree(mixer); + swidget->private = NULL; return ret; } -- cgit v1.2.3 From dc4fc0ae94cf87f1017f158b6fa2b7536ef29b4e Mon Sep 17 00:00:00 2001 From: Libin Yang Date: Fri, 8 Jul 2022 15:05:16 -0500 Subject: ASoC: SOF: ipc4-topology: free memories allocated in sof_ipc4_get_audio_fmt MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Free the memories allocated in sof_ipc4_get_audio_fmt in error handling and ipc_free() Fixes: 2cabd02b6090 ("ASoC: SOF: ipc4-topology: Add support for parsing AIF_IN/AIF_OUT widgets") Fixes: abfb536bd116 ("ASoC: SOF: ipc4-topology: Add support for parsing DAI_IN/DAI_OUT widgets") Fixes: 4f838ab20812 ("ASoC: SOF: ipc4-topology: Add support for parsing and preparing pga widgets") Fixes: 4d4ba014ac4b ("ASoC: SOF: ipc4-topology: Add support for parsing mixer widgets") Reviewed-by: Péter Ujfalusi Signed-off-by: Libin Yang Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220708200516.26853-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-topology.c | 48 +++++++++++++++++++++++++++++++++++++++---- 1 file changed, 44 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index 2d157ea79db5..22ea628d78d0 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -263,6 +263,16 @@ err_in: return ret; } +/* release the memory allocated in sof_ipc4_get_audio_fmt */ +static void sof_ipc4_free_audio_fmt(struct sof_ipc4_available_audio_format *available_fmt) + +{ + kfree(available_fmt->base_config); + available_fmt->base_config = NULL; + kfree(available_fmt->out_audio_fmt); + available_fmt->out_audio_fmt = NULL; +} + static void sof_ipc4_widget_free_comp(struct snd_sof_widget *swidget) { kfree(swidget->private); @@ -341,7 +351,7 @@ static int sof_ipc4_widget_setup_pcm(struct snd_sof_widget *swidget) GFP_KERNEL); if (!available_fmt->dma_buffer_size) { ret = -ENOMEM; - goto free_copier; + goto free_available_fmt; } ret = sof_update_ipc_object(scomp, available_fmt->dma_buffer_size, @@ -392,6 +402,8 @@ free_gtw_attr: kfree(ipc4_copier->gtw_attr); err: kfree(available_fmt->dma_buffer_size); +free_available_fmt: + sof_ipc4_free_audio_fmt(available_fmt); free_copier: kfree(ipc4_copier); swidget->private = NULL; @@ -440,7 +452,7 @@ static int sof_ipc4_widget_setup_comp_dai(struct snd_sof_widget *swidget) GFP_KERNEL); if (!available_fmt->dma_buffer_size) { ret = -ENOMEM; - goto free_copier; + goto free_available_fmt; } ret = sof_update_ipc_object(scomp, available_fmt->dma_buffer_size, @@ -540,6 +552,8 @@ free_copier_config: kfree(ipc4_copier->copier_config); err: kfree(available_fmt->dma_buffer_size); +free_available_fmt: + sof_ipc4_free_audio_fmt(available_fmt); free_copier: kfree(ipc4_copier); dai->private = NULL; @@ -677,11 +691,24 @@ static int sof_ipc4_widget_setup_comp_pga(struct snd_sof_widget *swidget) return 0; err: + sof_ipc4_free_audio_fmt(&gain->available_fmt); kfree(gain); swidget->private = NULL; return ret; } +static void sof_ipc4_widget_free_comp_pga(struct snd_sof_widget *swidget) +{ + struct sof_ipc4_gain *gain = swidget->private; + + if (!gain) + return; + + sof_ipc4_free_audio_fmt(&gain->available_fmt); + kfree(swidget->private); + swidget->private = NULL; +} + static int sof_ipc4_widget_setup_comp_mixer(struct snd_sof_widget *swidget) { struct snd_soc_component *scomp = swidget->scomp; @@ -707,11 +734,24 @@ static int sof_ipc4_widget_setup_comp_mixer(struct snd_sof_widget *swidget) return 0; err: + sof_ipc4_free_audio_fmt(&mixer->available_fmt); kfree(mixer); swidget->private = NULL; return ret; } +static void sof_ipc4_widget_free_comp_mixer(struct snd_sof_widget *swidget) +{ + struct sof_ipc4_mixer *mixer = swidget->private; + + if (!mixer) + return; + + sof_ipc4_free_audio_fmt(&mixer->available_fmt); + kfree(swidget->private); + swidget->private = NULL; +} + static void sof_ipc4_update_pipeline_mem_usage(struct snd_sof_dev *sdev, struct snd_sof_widget *swidget, struct sof_ipc4_base_module_cfg *base_config) @@ -1746,11 +1786,11 @@ static const struct sof_ipc_tplg_widget_ops tplg_ipc4_widget_ops[SND_SOC_DAPM_TY [snd_soc_dapm_scheduler] = {sof_ipc4_widget_setup_comp_pipeline, sof_ipc4_widget_free_comp, pipeline_token_list, ARRAY_SIZE(pipeline_token_list), NULL, NULL, NULL}, - [snd_soc_dapm_pga] = {sof_ipc4_widget_setup_comp_pga, sof_ipc4_widget_free_comp, + [snd_soc_dapm_pga] = {sof_ipc4_widget_setup_comp_pga, sof_ipc4_widget_free_comp_pga, pga_token_list, ARRAY_SIZE(pga_token_list), NULL, sof_ipc4_prepare_gain_module, sof_ipc4_unprepare_generic_module}, - [snd_soc_dapm_mixer] = {sof_ipc4_widget_setup_comp_mixer, sof_ipc4_widget_free_comp, + [snd_soc_dapm_mixer] = {sof_ipc4_widget_setup_comp_mixer, sof_ipc4_widget_free_comp_mixer, mixer_token_list, ARRAY_SIZE(mixer_token_list), NULL, sof_ipc4_prepare_mixer_module, sof_ipc4_unprepare_generic_module}, -- cgit v1.2.3 From 0fcc43e2e159f2f609686a5339093177f019ae26 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Wed, 6 Jul 2022 14:02:23 +0200 Subject: ALSA: hda: Fix null-ptr-deref when i915 fails and hdmi is denylisted If snd_hda_hdmi_codec module is denylisted and any event causes i915 enumeration to fail, is_likely_hdmi_codec() ends in null-ptr-deref. As snd_soc_hda is an ASoC-based driver, its initialization is delayed until all the necessary components appear in the system - allowing actual sound card to enumerate. snd_hda_codec_configure() gets called by the avs-driver core during probe_codecs() but the snd_hda_codec_device_new(), necessary to complete codecs initialization, happens only when codec-component of hda sound card is being probed. Denylisting snd_hda_codec_hdmi module causes snd_hda_codec_configure() to reach: codec_bind_generic() -> is_likely_hdmi_codec() which makes use of ->wcaps and at this point the it isn't initialized yet - again, requires completion of snd_hda_codec_device_new(). Signed-off-by: Cezary Rojewski Link: https://lore.kernel.org/r/20220706120230.427296-3-cezary.rojewski@intel.com Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_bind.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index c572fb5886d5..cae9a975cbcc 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -248,6 +248,13 @@ static bool is_likely_hdmi_codec(struct hda_codec *codec) { hda_nid_t nid; + /* + * For ASoC users, if snd_hda_hdmi_codec module is denylisted and any + * event causes i915 enumeration to fail, ->wcaps remains uninitialized. + */ + if (!codec->wcaps) + return true; + for_each_hda_codec_node(nid, codec) { unsigned int wcaps = get_wcaps(codec, nid); switch (get_wcaps_type(wcaps)) { -- cgit v1.2.3 From 9c76958b396a1342f08f205c350447b4bb54b26a Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Wed, 6 Jul 2022 14:02:24 +0200 Subject: ALSA: hda: Make device usage_count consistent across subsequent probing AVS HDAudio bus driver does not tie with codec drivers tighly and snd_hda_codec_device_new() can be called after codec's module reload. In such case, rpm is forbidden and invoking pm_runtime_forbid() unconditionally causes device's usage_count to become unbalanced. This is later caught by WARN_ON() found in sound/soc/hda.c. Detect such circumstance and bump the usage_count instead. Signed-off-by: Cezary Rojewski Link: https://lore.kernel.org/r/20220706120230.427296-4-cezary.rojewski@intel.com Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 7579a6982f47..018067addd86 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1045,8 +1045,14 @@ int snd_hda_codec_device_new(struct hda_bus *bus, struct snd_card *card, goto error; } +#ifdef CONFIG_PM /* PM runtime needs to be enabled later after binding codec */ - pm_runtime_forbid(&codec->core.dev); + if (codec->core.dev.power.runtime_auto) + pm_runtime_forbid(&codec->core.dev); + else + /* Keep the usage_count consistent across subsequent probing */ + pm_runtime_get_noresume(&codec->core.dev); +#endif return 0; -- cgit v1.2.3 From ebe043a3dfcade2756a41a3956365e1bfe4198cf Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Wed, 6 Jul 2022 14:02:25 +0200 Subject: ALSA: hda: Fix put_device() inconsistency in error path AVS HDAudio bus driver does not tie with codec drivers tighly. Codec device and its respective driver cleanup procedures are split and may not occur one after the other. Device cleanup is performed only on snd_hdac_ext_bus_device_remove() i.e. it's the bus driver's responsibility. If codec component probing fails, put_device() found in snd_hda_codec_device_new() may lead to page fault. Relocate it to snd_hda_codec_new() to address the problem on ASoC side while keeping status quo for snd_hda_intel. Signed-off-by: Cezary Rojewski Link: https://lore.kernel.org/r/20220706120230.427296-5-cezary.rojewski@intel.com Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 23 +++++++++++------------ 1 file changed, 11 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 018067addd86..e8a5d0e4105b 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -950,6 +950,7 @@ int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, unsigned int codec_addr, struct hda_codec **codecp) { struct hda_codec *codec; + int ret; codec = snd_hda_codec_device_init(bus, codec_addr, "hdaudioC%dD%d", card->number, codec_addr); @@ -957,7 +958,11 @@ int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, return PTR_ERR(codec); *codecp = codec; - return snd_hda_codec_device_new(bus, card, codec_addr, *codecp, true); + ret = snd_hda_codec_device_new(bus, card, codec_addr, *codecp, true); + if (ret) + put_device(hda_codec_dev(*codecp)); + + return ret; } EXPORT_SYMBOL_GPL(snd_hda_codec_new); @@ -1012,19 +1017,17 @@ int snd_hda_codec_device_new(struct hda_bus *bus, struct snd_card *card, if (codec->bus->modelname) { codec->modelname = kstrdup(codec->bus->modelname, GFP_KERNEL); - if (!codec->modelname) { - err = -ENOMEM; - goto error; - } + if (!codec->modelname) + return -ENOMEM; } fg = codec->core.afg ? codec->core.afg : codec->core.mfg; err = read_widget_caps(codec, fg); if (err < 0) - goto error; + return err; err = read_pin_defaults(codec); if (err < 0) - goto error; + return err; /* power-up all before initialization */ hda_set_power_state(codec, AC_PWRST_D0); @@ -1042,7 +1045,7 @@ int snd_hda_codec_device_new(struct hda_bus *bus, struct snd_card *card, /* ASoC features component management instead */ err = snd_device_new(card, SNDRV_DEV_CODEC, codec, &dev_ops); if (err < 0) - goto error; + return err; } #ifdef CONFIG_PM @@ -1055,10 +1058,6 @@ int snd_hda_codec_device_new(struct hda_bus *bus, struct snd_card *card, #endif return 0; - - error: - put_device(hda_codec_dev(codec)); - return err; } EXPORT_SYMBOL_GPL(snd_hda_codec_device_new); -- cgit v1.2.3 From 980b3a8790b402e959a6d773b38b771019682be1 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Wed, 6 Jul 2022 14:02:27 +0200 Subject: ALSA: hda: Fix page fault in snd_hda_codec_shutdown() If early probe of HDAudio bus driver fails e.g.: due to missing firmware file, snd_hda_codec_shutdown() ends in manipulating uninitialized codec->pcm_list_head causing page fault. Iinitialization of HDAudio codec in ASoC is split in two: - snd_hda_codec_device_init() - snd_hda_codec_device_new() snd_hda_codec_device_init() is called during probe_codecs() by HDAudio bus driver while snd_hda_codec_device_new() is called by codec-component's ->probe(). The second call will not happen until all components required by related sound card are present within the ASoC framework. With firmware failing to load during the PCI's deferred initialization i.e.: probe_work(), no platform components are ever registered. HDAudio codec enumeration is done at that point though, so the codec components became registered to ASoC framework, calling snd_hda_codec_device_init() in the process. Now, during platform reboot snd_hda_codec_shutdown() is called for every codec found on the HDAudio bus causing oops if any of them has not completed both of their initialization steps. Relocating field initialization fixes the issue. Signed-off-by: Cezary Rojewski Link: https://lore.kernel.org/r/20220706120230.427296-7-cezary.rojewski@intel.com Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 41 ++++++++++++++++++++--------------------- 1 file changed, 20 insertions(+), 21 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index e8a5d0e4105b..b1921f920513 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -931,8 +931,28 @@ snd_hda_codec_device_init(struct hda_bus *bus, unsigned int codec_addr, } codec->bus = bus; + codec->depop_delay = -1; + codec->fixup_id = HDA_FIXUP_ID_NOT_SET; + codec->core.dev.release = snd_hda_codec_dev_release; + codec->core.exec_verb = codec_exec_verb; codec->core.type = HDA_DEV_LEGACY; + mutex_init(&codec->spdif_mutex); + mutex_init(&codec->control_mutex); + snd_array_init(&codec->mixers, sizeof(struct hda_nid_item), 32); + snd_array_init(&codec->nids, sizeof(struct hda_nid_item), 32); + snd_array_init(&codec->init_pins, sizeof(struct hda_pincfg), 16); + snd_array_init(&codec->driver_pins, sizeof(struct hda_pincfg), 16); + snd_array_init(&codec->cvt_setups, sizeof(struct hda_cvt_setup), 8); + snd_array_init(&codec->spdif_out, sizeof(struct hda_spdif_out), 16); + snd_array_init(&codec->jacktbl, sizeof(struct hda_jack_tbl), 16); + snd_array_init(&codec->verbs, sizeof(struct hda_verb *), 8); + INIT_LIST_HEAD(&codec->conn_list); + INIT_LIST_HEAD(&codec->pcm_list_head); + INIT_DELAYED_WORK(&codec->jackpoll_work, hda_jackpoll_work); + refcount_set(&codec->pcm_ref, 1); + init_waitqueue_head(&codec->remove_sleep); + return codec; } EXPORT_SYMBOL_GPL(snd_hda_codec_device_init); @@ -985,29 +1005,8 @@ int snd_hda_codec_device_new(struct hda_bus *bus, struct snd_card *card, if (snd_BUG_ON(codec_addr > HDA_MAX_CODEC_ADDRESS)) return -EINVAL; - codec->core.dev.release = snd_hda_codec_dev_release; - codec->core.exec_verb = codec_exec_verb; - codec->card = card; codec->addr = codec_addr; - mutex_init(&codec->spdif_mutex); - mutex_init(&codec->control_mutex); - snd_array_init(&codec->mixers, sizeof(struct hda_nid_item), 32); - snd_array_init(&codec->nids, sizeof(struct hda_nid_item), 32); - snd_array_init(&codec->init_pins, sizeof(struct hda_pincfg), 16); - snd_array_init(&codec->driver_pins, sizeof(struct hda_pincfg), 16); - snd_array_init(&codec->cvt_setups, sizeof(struct hda_cvt_setup), 8); - snd_array_init(&codec->spdif_out, sizeof(struct hda_spdif_out), 16); - snd_array_init(&codec->jacktbl, sizeof(struct hda_jack_tbl), 16); - snd_array_init(&codec->verbs, sizeof(struct hda_verb *), 8); - INIT_LIST_HEAD(&codec->conn_list); - INIT_LIST_HEAD(&codec->pcm_list_head); - refcount_set(&codec->pcm_ref, 1); - init_waitqueue_head(&codec->remove_sleep); - - INIT_DELAYED_WORK(&codec->jackpoll_work, hda_jackpoll_work); - codec->depop_delay = -1; - codec->fixup_id = HDA_FIXUP_ID_NOT_SET; #ifdef CONFIG_PM codec->power_jiffies = jiffies; -- cgit v1.2.3 From 856282f166d7b95004e70859ca87cf33eb7f9a6d Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Wed, 6 Jul 2022 14:02:28 +0200 Subject: ALSA: hda: Reset all SIE bits in INTCTL MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Old code resets SIE for up to 8 streams using byte accessor, but register is laid out in following way: 31 GIE 30 CIE 29:x Reserved x-1:0 SIE If there is more than 8 streams, some of them may and up with enabled interrupts. To fix this just clear whole INTCTL register when disabling interrupts. Signed-off-by: Amadeusz Sławiński Signed-off-by: Cezary Rojewski Link: https://lore.kernel.org/r/20220706120230.427296-8-cezary.rojewski@intel.com Signed-off-by: Takashi Iwai --- sound/hda/hdac_controller.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index f7bd6e2db085..9a60bfdb39ba 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -474,11 +474,8 @@ static void azx_int_disable(struct hdac_bus *bus) list_for_each_entry(azx_dev, &bus->stream_list, list) snd_hdac_stream_updateb(azx_dev, SD_CTL, SD_INT_MASK, 0); - /* disable SIE for all streams */ - snd_hdac_chip_writeb(bus, INTCTL, 0); - - /* disable controller CIE and GIE */ - snd_hdac_chip_updatel(bus, INTCTL, AZX_INT_CTRL_EN | AZX_INT_GLOBAL_EN, 0); + /* disable SIE for all streams & disable controller CIE and GIE */ + snd_hdac_chip_writel(bus, INTCTL, 0); } /* clear interrupts */ -- cgit v1.2.3 From 0440741254ed27bd695d994f00358647c92ed832 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Wed, 6 Jul 2022 14:02:29 +0200 Subject: ALSA: hda: Remove unused macro definition MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit It is not used anywhere in the file, so there is no need to keep it. Signed-off-by: Amadeusz Sławiński Signed-off-by: Cezary Rojewski Link: https://lore.kernel.org/r/20220706120230.427296-9-cezary.rojewski@intel.com Signed-off-by: Takashi Iwai --- sound/hda/ext/hdac_ext_controller.c | 7 ------- 1 file changed, 7 deletions(-) (limited to 'sound') diff --git a/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c index b072392725c7..a42f66f561f5 100644 --- a/sound/hda/ext/hdac_ext_controller.c +++ b/sound/hda/ext/hdac_ext_controller.c @@ -14,13 +14,6 @@ #include #include -/* - * maximum HDAC capablities we should parse to avoid endless looping: - * currently we have 4 extended caps, so this is future proof for now. - * extend when this limit is seen meeting in real HW - */ -#define HDAC_MAX_CAPS 10 - /* * processing pipe helpers - these helpers are useful for dealing with HDA * new capability of processing pipelines -- cgit v1.2.3 From 19bb587f3ffcb9c166bac2debdc3b08fb362c0b7 Mon Sep 17 00:00:00 2001 From: Zhongjun Tan Date: Fri, 8 Jul 2022 10:46:51 +0800 Subject: ASoC: mediatek: mt8186: Remove condition with no effect Remove condition with no effect Signed-off-by: Zhongjun Tan Reviewed-by: AngeloGioacchino Del Regno Link: https://lore.kernel.org/r/20220708024651.42999-1-hbut_tan@163.com Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8186/mt8186-dai-adda.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/mediatek/mt8186/mt8186-dai-adda.c b/sound/soc/mediatek/mt8186/mt8186-dai-adda.c index db71b032770d..6be6d4f3b585 100644 --- a/sound/soc/mediatek/mt8186/mt8186-dai-adda.c +++ b/sound/soc/mediatek/mt8186/mt8186-dai-adda.c @@ -295,8 +295,6 @@ static int mtk_adda_pad_top_event(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_PRE_PMU: if (afe_priv->mtkaif_protocol == MTKAIF_PROTOCOL_2_CLK_P2) regmap_write(afe->regmap, AFE_AUD_PAD_TOP, 0x39); - else if (afe_priv->mtkaif_protocol == MTKAIF_PROTOCOL_2) - regmap_write(afe->regmap, AFE_AUD_PAD_TOP, 0x31); else regmap_write(afe->regmap, AFE_AUD_PAD_TOP, 0x31); break; -- cgit v1.2.3 From eaa27e7fe43f16fe587c3e93fd5c25ce86be3c43 Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Mon, 11 Jul 2022 10:39:50 +0800 Subject: ASoC: fsl_utils: Drop usage of __clk_get_name() Avoid build errors when CONFIG_COMMON_CLK is not set/enabled. ERROR: modpost: "__clk_get_name" [sound/soc/fsl/snd-soc-fsl-utils.ko] undefined! Fixes: 7bad8125549c ("ASoC: fsl_utils: Add function to handle PLL clock source") Reported-by: kernel test robot Signed-off-by: Shengjiu Wang Link: https://lore.kernel.org/r/1657507190-14546-1-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_utils.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_utils.c b/sound/soc/fsl/fsl_utils.c index 3e969c7bc1c5..d0fc430f7033 100644 --- a/sound/soc/fsl/fsl_utils.c +++ b/sound/soc/fsl/fsl_utils.c @@ -147,7 +147,7 @@ void fsl_asoc_reparent_pll_clocks(struct device *dev, struct clk *clk, if (reparent) { ret = clk_set_parent(p, npll); if (ret < 0) - dev_warn(dev, "failed to set parent %s: %d\n", __clk_get_name(npll), ret); + dev_warn(dev, "failed to set parent:%d\n", ret); } } EXPORT_SYMBOL(fsl_asoc_reparent_pll_clocks); -- cgit v1.2.3 From 9b6803ec1fe0f10942b9297d2d60ec46f2999323 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Thu, 7 Jul 2022 14:56:57 +0200 Subject: ASoC: codecs: rt298: Fix NULL jack in interrupt MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Set rt298->jack to passed value in mic_detect, otherwise when jack is set to NULL on next interrupt call, we may use freed pointer. Signed-off-by: Amadeusz Sławiński Reviewed-by: Cezary Rojewski Link: https://lore.kernel.org/r/20220707125701.3518263-2-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt298.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c index 266a2cc55b8d..6a615943f983 100644 --- a/sound/soc/codecs/rt298.c +++ b/sound/soc/codecs/rt298.c @@ -335,6 +335,8 @@ static int rt298_mic_detect(struct snd_soc_component *component, bool mic = false; int status = 0; + rt298->jack = jack; + /* If jack in NULL, disable HS jack */ if (!jack) { regmap_update_bits(rt298->regmap, RT298_IRQ_CTRL, 0x2, 0x0); @@ -344,7 +346,6 @@ static int rt298_mic_detect(struct snd_soc_component *component, return 0; } - rt298->jack = jack; regmap_update_bits(rt298->regmap, RT298_IRQ_CTRL, 0x2, 0x2); rt298_jack_detect(rt298, &hp, &mic); -- cgit v1.2.3 From c0c5a242bba878b4d34f7c9612fd6cd6c404d414 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Thu, 7 Jul 2022 14:56:58 +0200 Subject: ASoC: codecs: rt298: Fix jack detection MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit On our RVP platforms using rt298 with combojack we've seen issues with controls being in incorrect state after suspend/resume cycle. This is caused by codec driver not setting pins to correct state and causing codec suspend method to not be called. Which on resume caused codec registers to be in undefined state. Fix this by setting pins correctly in jack detect function. Above problem is caused by the fact that when jack == NULL code doesn't reach rt298_jack_detect() function which sets pins. Alternatively problem could be fixed by just moving rt298_jack_detect, but as rt298 codec is similar to rt286, align the code by setting pins explicitly. Signed-off-by: Amadeusz Sławiński Reviewed-by: Cezary Rojewski Link: https://lore.kernel.org/r/20220707125701.3518263-3-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt298.c | 39 ++++++++++++++++++--------------------- 1 file changed, 18 insertions(+), 21 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c index 6a615943f983..e1d94f128fd9 100644 --- a/sound/soc/codecs/rt298.c +++ b/sound/soc/codecs/rt298.c @@ -329,34 +329,31 @@ static void rt298_jack_detect_work(struct work_struct *work) static int rt298_mic_detect(struct snd_soc_component *component, struct snd_soc_jack *jack, void *data) { + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); struct rt298_priv *rt298 = snd_soc_component_get_drvdata(component); - struct snd_soc_dapm_context *dapm; - bool hp = false; - bool mic = false; - int status = 0; rt298->jack = jack; - /* If jack in NULL, disable HS jack */ - if (!jack) { + if (jack) { + /* Enable IRQ */ + if (rt298->jack->status & SND_JACK_HEADPHONE) + snd_soc_dapm_force_enable_pin(dapm, "LDO1"); + if (rt298->jack->status & SND_JACK_MICROPHONE) { + snd_soc_dapm_force_enable_pin(dapm, "HV"); + snd_soc_dapm_force_enable_pin(dapm, "VREF"); + } + regmap_update_bits(rt298->regmap, RT298_IRQ_CTRL, 0x2, 0x2); + /* Send an initial empty report */ + snd_soc_jack_report(rt298->jack, rt298->jack->status, + SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); + } else { + /* Disable IRQ */ regmap_update_bits(rt298->regmap, RT298_IRQ_CTRL, 0x2, 0x0); - dapm = snd_soc_component_get_dapm(component); + snd_soc_dapm_disable_pin(dapm, "HV"); + snd_soc_dapm_disable_pin(dapm, "VREF"); snd_soc_dapm_disable_pin(dapm, "LDO1"); - snd_soc_dapm_sync(dapm); - return 0; } - - regmap_update_bits(rt298->regmap, RT298_IRQ_CTRL, 0x2, 0x2); - - rt298_jack_detect(rt298, &hp, &mic); - if (hp) - status |= SND_JACK_HEADPHONE; - - if (mic) - status |= SND_JACK_MICROPHONE; - - snd_soc_jack_report(rt298->jack, status, - SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); + snd_soc_dapm_sync(dapm); return 0; } -- cgit v1.2.3 From c1d7ebda11aae4659b665af61d7277dd351659b9 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Thu, 7 Jul 2022 14:56:59 +0200 Subject: ASoC: codecs: rt286: Set component to NULL on remove MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Make sure that component is set to proper value, otherwise we may dereference freed component in interrupt. Signed-off-by: Amadeusz Sławiński Reviewed-by: Cezary Rojewski Link: https://lore.kernel.org/r/20220707125701.3518263-4-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt286.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index b2b0b2b1e4d0..c4f7c4c2d793 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -960,6 +960,7 @@ static void rt286_remove(struct snd_soc_component *component) struct rt286_priv *rt286 = snd_soc_component_get_drvdata(component); cancel_delayed_work_sync(&rt286->jack_detect_work); + rt286->component = NULL; } #ifdef CONFIG_PM -- cgit v1.2.3 From af3b33b9707d453a12e0cf5ac35d7b97b3524ace Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Thu, 7 Jul 2022 14:57:00 +0200 Subject: ASoC: codecs: rt298: Set component to NULL on remove MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Make sure that component is set to proper value, otherwise we may dereference freed component in interrupt. Signed-off-by: Amadeusz Sławiński Reviewed-by: Cezary Rojewski Link: https://lore.kernel.org/r/20220707125701.3518263-5-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt298.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c index e1d94f128fd9..b0b53d4f07df 100644 --- a/sound/soc/codecs/rt298.c +++ b/sound/soc/codecs/rt298.c @@ -1022,6 +1022,7 @@ static void rt298_remove(struct snd_soc_component *component) struct rt298_priv *rt298 = snd_soc_component_get_drvdata(component); cancel_delayed_work_sync(&rt298->jack_detect_work); + rt298->component = NULL; } #ifdef CONFIG_PM -- cgit v1.2.3 From b9f098aa7ae2a022dee06a8ca363e3e0e077f05a Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Thu, 7 Jul 2022 14:57:01 +0200 Subject: ASoC: codecs: rt274: Set component to NULL on remove MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Make sure that component is set to proper value, otherwise we may dereference freed component in interrupt. Signed-off-by: Amadeusz Sławiński Reviewed-by: Cezary Rojewski Link: https://lore.kernel.org/r/20220707125701.3518263-6-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt274.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/rt274.c b/sound/soc/codecs/rt274.c index 6b208f9eb503..f2c50b11e4d0 100644 --- a/sound/soc/codecs/rt274.c +++ b/sound/soc/codecs/rt274.c @@ -993,6 +993,7 @@ static void rt274_remove(struct snd_soc_component *component) struct rt274_priv *rt274 = snd_soc_component_get_drvdata(component); cancel_delayed_work_sync(&rt274->jack_detect_work); + rt274->component = NULL; } #ifdef CONFIG_PM -- cgit v1.2.3 From 375f53566cf04324825b7a0f545aeb4405963bd0 Mon Sep 17 00:00:00 2001 From: Claudiu Beznea Date: Mon, 11 Jul 2022 14:22:12 +0300 Subject: ASoC: atmel: mchp-pdmc: remove space in front of mchp_pdmc_dt_init() Remove extra space in front of mchp_pdmc_dt_init(). Signed-off-by: Claudiu Beznea Link: https://lore.kernel.org/r/20220711112212.888895-1-claudiu.beznea@microchip.com Signed-off-by: Mark Brown --- sound/soc/atmel/mchp-pdmc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/atmel/mchp-pdmc.c b/sound/soc/atmel/mchp-pdmc.c index aba7c5cde62c..44aefbd5b62c 100644 --- a/sound/soc/atmel/mchp-pdmc.c +++ b/sound/soc/atmel/mchp-pdmc.c @@ -985,7 +985,7 @@ static int mchp_pdmc_probe(struct platform_device *pdev) return -ENOMEM; dd->dev = &pdev->dev; - ret = mchp_pdmc_dt_init(dd); + ret = mchp_pdmc_dt_init(dd); if (ret < 0) return ret; -- cgit v1.2.3 From fd1c769d33872d6c7ec474c199f6a910d3555927 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 8 Jul 2022 15:07:19 -0500 Subject: ASoC: SOF: remove warning on ABI checks MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit We should only have an error when enforcing strict mapping between kernel and firmware versions. In all other cases, there is no reason to throw a warning. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Péter Ujfalusi Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/20220708200719.26961-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc3-topology.c | 13 ++++--------- sound/soc/sof/ipc3.c | 11 ++++------- 2 files changed, 8 insertions(+), 16 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/ipc3-topology.c b/sound/soc/sof/ipc3-topology.c index 99b62fe7a95c..9448d5338423 100644 --- a/sound/soc/sof/ipc3-topology.c +++ b/sound/soc/sof/ipc3-topology.c @@ -2348,15 +2348,10 @@ static int sof_ipc3_parse_manifest(struct snd_soc_component *scomp, int index, return -EINVAL; } - if (SOF_ABI_VERSION_MINOR(abi_version) > SOF_ABI_MINOR) { - if (!IS_ENABLED(CONFIG_SND_SOC_SOF_STRICT_ABI_CHECKS)) { - dev_warn(scomp->dev, "%s: Topology ABI is more recent than kernel\n", - __func__); - } else { - dev_err(scomp->dev, "%s: Topology ABI is more recent than kernel\n", - __func__); - return -EINVAL; - } + if (IS_ENABLED(CONFIG_SND_SOC_SOF_STRICT_ABI_CHECKS) && + SOF_ABI_VERSION_MINOR(abi_version) > SOF_ABI_MINOR) { + dev_err(scomp->dev, "%s: Topology ABI is more recent than kernel\n", __func__); + return -EINVAL; } return 0; diff --git a/sound/soc/sof/ipc3.c b/sound/soc/sof/ipc3.c index 1fb132b477bf..82fa320253be 100644 --- a/sound/soc/sof/ipc3.c +++ b/sound/soc/sof/ipc3.c @@ -758,13 +758,10 @@ int sof_ipc3_validate_fw_version(struct snd_sof_dev *sdev) return -EINVAL; } - if (SOF_ABI_VERSION_MINOR(v->abi_version) > SOF_ABI_MINOR) { - if (!IS_ENABLED(CONFIG_SND_SOC_SOF_STRICT_ABI_CHECKS)) { - dev_warn(sdev->dev, "FW ABI is more recent than kernel\n"); - } else { - dev_err(sdev->dev, "FW ABI is more recent than kernel\n"); - return -EINVAL; - } + if (IS_ENABLED(CONFIG_SND_SOC_SOF_STRICT_ABI_CHECKS) && + SOF_ABI_VERSION_MINOR(v->abi_version) > SOF_ABI_MINOR) { + dev_err(sdev->dev, "FW ABI is more recent than kernel\n"); + return -EINVAL; } if (ready->flags & SOF_IPC_INFO_BUILD) { -- cgit v1.2.3 From 2551b6e89936f98406bce9c1d50110e3ff443f81 Mon Sep 17 00:00:00 2001 From: Seven Lee Date: Mon, 27 Jun 2022 11:29:59 +0800 Subject: ASoC: nau8821: Add headset button detection This patch adds the function of headphone button detection, Button detection will be enabled if the device tree has a key_enable property. Signed-off-by: Seven Lee Link: https://lore.kernel.org/r/20220627032959.3442064-1-wtli@nuvoton.com Signed-off-by: Mark Brown --- sound/soc/codecs/nau8821.c | 35 +++++++++++++++++++++++++++++++++++ sound/soc/codecs/nau8821.h | 1 + 2 files changed, 36 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/nau8821.c b/sound/soc/codecs/nau8821.c index 6453e93678d2..2600be250a3c 100644 --- a/sound/soc/codecs/nau8821.c +++ b/sound/soc/codecs/nau8821.c @@ -29,6 +29,8 @@ #define NAU_FVCO_MAX 100000000 #define NAU_FVCO_MIN 90000000 +#define NAU8821_BUTTON SND_JACK_BTN_0 + /* the maximum frequency of CLK_ADC and CLK_DAC */ #define CLK_DA_AD_MAX 6144000 @@ -911,6 +913,20 @@ static void nau8821_eject_jack(struct nau8821 *nau8821) /* Recover to normal channel input */ regmap_update_bits(regmap, NAU8821_R2B_ADC_RATE, NAU8821_ADC_R_SRC_EN, 0); + if (nau8821->key_enable) { + regmap_update_bits(regmap, NAU8821_R0F_INTERRUPT_MASK, + NAU8821_IRQ_KEY_RELEASE_EN | + NAU8821_IRQ_KEY_PRESS_EN, + NAU8821_IRQ_KEY_RELEASE_EN | + NAU8821_IRQ_KEY_PRESS_EN); + regmap_update_bits(regmap, + NAU8821_R12_INTERRUPT_DIS_CTRL, + NAU8821_IRQ_KEY_RELEASE_DIS | + NAU8821_IRQ_KEY_PRESS_DIS, + NAU8821_IRQ_KEY_RELEASE_DIS | + NAU8821_IRQ_KEY_PRESS_DIS); + } + } static void nau8821_jdet_work(struct work_struct *work) @@ -940,6 +956,15 @@ static void nau8821_jdet_work(struct work_struct *work) */ regmap_update_bits(regmap, NAU8821_R2B_ADC_RATE, NAU8821_ADC_R_SRC_EN, NAU8821_ADC_R_SRC_EN); + if (nau8821->key_enable) { + regmap_update_bits(regmap, NAU8821_R0F_INTERRUPT_MASK, + NAU8821_IRQ_KEY_RELEASE_EN | + NAU8821_IRQ_KEY_PRESS_EN, 0); + regmap_update_bits(regmap, + NAU8821_R12_INTERRUPT_DIS_CTRL, + NAU8821_IRQ_KEY_RELEASE_DIS | + NAU8821_IRQ_KEY_PRESS_DIS, 0); + } } else { dev_dbg(nau8821->dev, "Headphone connected\n"); event |= SND_JACK_HEADPHONE; @@ -999,6 +1024,13 @@ static irqreturn_t nau8821_interrupt(int irq, void *data) nau8821_eject_jack(nau8821); event_mask |= SND_JACK_HEADSET; clear_irq = NAU8821_JACK_EJECT_IRQ_MASK; + } else if (active_irq & NAU8821_KEY_SHORT_PRESS_IRQ) { + event |= NAU8821_BUTTON; + event_mask |= NAU8821_BUTTON; + clear_irq = NAU8821_KEY_SHORT_PRESS_IRQ; + } else if (active_irq & NAU8821_KEY_RELEASE_IRQ) { + event_mask = NAU8821_BUTTON; + clear_irq = NAU8821_KEY_RELEASE_IRQ; } else if ((active_irq & NAU8821_JACK_INSERT_IRQ_MASK) == NAU8821_JACK_INSERT_DETECTED) { regmap_update_bits(regmap, NAU8821_R71_ANALOG_ADC_1, @@ -1489,6 +1521,7 @@ static void nau8821_print_device_properties(struct nau8821 *nau8821) nau8821->jack_eject_debounce); dev_dbg(dev, "dmic-clk-threshold: %d\n", nau8821->dmic_clk_threshold); + dev_dbg(dev, "key_enable: %d\n", nau8821->key_enable); } static int nau8821_read_device_properties(struct device *dev, @@ -1502,6 +1535,8 @@ static int nau8821_read_device_properties(struct device *dev, "nuvoton,jkdet-pull-enable"); nau8821->jkdet_pull_up = device_property_read_bool(dev, "nuvoton,jkdet-pull-up"); + nau8821->key_enable = device_property_read_bool(dev, + "nuvoton,key-enable"); ret = device_property_read_u32(dev, "nuvoton,jkdet-polarity", &nau8821->jkdet_polarity); if (ret) diff --git a/sound/soc/codecs/nau8821.h b/sound/soc/codecs/nau8821.h index a92edfeb9d3a..c44251f54d48 100644 --- a/sound/soc/codecs/nau8821.h +++ b/sound/soc/codecs/nau8821.h @@ -525,6 +525,7 @@ struct nau8821 { int jack_eject_debounce; int fs; int dmic_clk_threshold; + int key_enable; }; int nau8821_enable_jack_detect(struct snd_soc_component *component, -- cgit v1.2.3 From 642999365da3b7cd5552ec758d6e1bb6f2f465d8 Mon Sep 17 00:00:00 2001 From: Andy Shevchenko Date: Mon, 11 Jul 2022 13:01:29 +0300 Subject: ALSA: hda: cs35l41: Fix comments wrt serial-multi-instantiate reference The comments are inconsistent and point to the wrong driver name. The initially named i2c-multi-instantiate it was renamed to the serial-multi-instantiate exactly due to support of the platforms with multiple CS35L41 codecs. Fix comments accordingly. While at it, drop file names from the files. Signed-off-by: Andy Shevchenko Link: https://lore.kernel.org/r/20220711100129.37326-1-andriy.shevchenko@linux.intel.com Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l41_hda.c | 4 ++-- sound/pci/hda/cs35l41_hda_i2c.c | 7 ++++--- sound/pci/hda/cs35l41_hda_spi.c | 7 ++++--- 3 files changed, 10 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index cce27a86267f..a9923a8818a2 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -415,8 +415,8 @@ err: no_acpi_dsd: /* * Device CLSA0100 doesn't have _DSD so a gpiod_get by the label reset won't work. - * And devices created by i2c-multi-instantiate don't have their device struct pointing to - * the correct fwnode, so acpi_dev must be used here. + * And devices created by serial-multi-instantiate don't have their device struct + * pointing to the correct fwnode, so acpi_dev must be used here. * And devm functions expect that the device requesting the resource has the correct * fwnode. */ diff --git a/sound/pci/hda/cs35l41_hda_i2c.c b/sound/pci/hda/cs35l41_hda_i2c.c index e810b278fb91..ec626e0fbedc 100644 --- a/sound/pci/hda/cs35l41_hda_i2c.c +++ b/sound/pci/hda/cs35l41_hda_i2c.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0 // -// cs35l41.c -- CS35l41 HDA I2C driver +// CS35l41 HDA I2C driver // // Copyright 2021 Cirrus Logic, Inc. // @@ -16,8 +16,9 @@ static int cs35l41_hda_i2c_probe(struct i2c_client *clt, const struct i2c_device { const char *device_name; - /* Compare against the device name so it works for I2C, normal ACPI - * and for ACPI by i2c-multi-instantiate matching cases + /* + * Compare against the device name so it works for SPI, normal ACPI + * and for ACPI by serial-multi-instantiate matching cases. */ if (strstr(dev_name(&clt->dev), "CLSA0100")) device_name = "CLSA0100"; diff --git a/sound/pci/hda/cs35l41_hda_spi.c b/sound/pci/hda/cs35l41_hda_spi.c index 22e088f28438..3a1472e1bc24 100644 --- a/sound/pci/hda/cs35l41_hda_spi.c +++ b/sound/pci/hda/cs35l41_hda_spi.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0 // -// cs35l41.c -- CS35l41 HDA SPI driver +// CS35l41 HDA SPI driver // // Copyright 2021 Cirrus Logic, Inc. // @@ -16,8 +16,9 @@ static int cs35l41_hda_spi_probe(struct spi_device *spi) { const char *device_name; - /* Compare against the device name so it works for SPI, normal ACPI - * and for ACPI by spi-multi-instantiate matching cases + /* + * Compare against the device name so it works for SPI, normal ACPI + * and for ACPI by serial-multi-instantiate matching cases. */ if (strstr(dev_name(&spi->dev), "CSC3551")) device_name = "CSC3551"; -- cgit v1.2.3 From e35cd6881dd56d5ad7711d23faab668268e17555 Mon Sep 17 00:00:00 2001 From: Andy Shevchenko Date: Mon, 11 Jul 2022 12:52:16 +0300 Subject: ALSA: hda: cs35l41: Improve dev_err_probe() messaging Drop duplicate print of returned value in the messages and use pattern return dev_err_probe(...) where it's possible. Signed-off-by: Andy Shevchenko Link: https://lore.kernel.org/r/20220711095219.36915-1-andriy.shevchenko@linux.intel.com Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l41_hda.c | 8 +++----- 1 file changed, 3 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index a9923a8818a2..49b25432a9f5 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -460,10 +460,8 @@ int cs35l41_hda_probe(struct device *dev, const char *device_name, int id, int i dev_set_drvdata(dev, cs35l41); ret = cs35l41_hda_read_acpi(cs35l41, device_name, id); - if (ret) { - dev_err_probe(cs35l41->dev, ret, "Platform not supported %d\n", ret); - return ret; - } + if (ret) + return dev_err_probe(cs35l41->dev, ret, "Platform not supported\n"); if (IS_ERR(cs35l41->reset_gpio)) { ret = PTR_ERR(cs35l41->reset_gpio); @@ -471,7 +469,7 @@ int cs35l41_hda_probe(struct device *dev, const char *device_name, int id, int i if (ret == -EBUSY) { dev_info(cs35l41->dev, "Reset line busy, assuming shared reset\n"); } else { - dev_err_probe(cs35l41->dev, ret, "Failed to get reset GPIO: %d\n", ret); + dev_err_probe(cs35l41->dev, ret, "Failed to get reset GPIO\n"); goto err; } } -- cgit v1.2.3 From acacd9eefd0def5a83244d88e5483b5f38ee7287 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 12 Jul 2022 15:23:55 +0300 Subject: ASoC: SOF: Intel: cnl: Do not process IPC reply before firmware boot It is not yet clear, but it is possible to create a firmware so broken that it will send a reply message before a FW_READY message (it is not yet clear if FW_READY will arrive later). Since the reply_data is allocated only after the FW_READY message, this will lead to a NULL pointer dereference if not filtered out. The issue was reported with IPC4 firmware but the same condition is present for IPC3. Reported-by: Kai Vehmanen Signed-off-by: Peter Ujfalusi Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220712122357.31282-2-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/cnl.c | 37 ++++++++++++++++++++++++------------- 1 file changed, 24 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/intel/cnl.c b/sound/soc/sof/intel/cnl.c index ccf46fcd6c9a..a064453f0bc3 100644 --- a/sound/soc/sof/intel/cnl.c +++ b/sound/soc/sof/intel/cnl.c @@ -60,17 +60,23 @@ irqreturn_t cnl_ipc4_irq_thread(int irq, void *context) if (primary & SOF_IPC4_MSG_DIR_MASK) { /* Reply received */ - struct sof_ipc4_msg *data = sdev->ipc->msg.reply_data; + if (likely(sdev->fw_state == SOF_FW_BOOT_COMPLETE)) { + struct sof_ipc4_msg *data = sdev->ipc->msg.reply_data; - data->primary = primary; - data->extension = extension; + data->primary = primary; + data->extension = extension; - spin_lock_irq(&sdev->ipc_lock); + spin_lock_irq(&sdev->ipc_lock); - snd_sof_ipc_get_reply(sdev); - snd_sof_ipc_reply(sdev, data->primary); + snd_sof_ipc_get_reply(sdev); + snd_sof_ipc_reply(sdev, data->primary); - spin_unlock_irq(&sdev->ipc_lock); + spin_unlock_irq(&sdev->ipc_lock); + } else { + dev_dbg_ratelimited(sdev->dev, + "IPC reply before FW_READY: %#x|%#x\n", + primary, extension); + } } else { /* Notification received */ notification_data.primary = primary; @@ -124,15 +130,20 @@ irqreturn_t cnl_ipc_irq_thread(int irq, void *context) CNL_DSP_REG_HIPCCTL, CNL_DSP_REG_HIPCCTL_DONE, 0); - spin_lock_irq(&sdev->ipc_lock); + if (likely(sdev->fw_state == SOF_FW_BOOT_COMPLETE)) { + spin_lock_irq(&sdev->ipc_lock); - /* handle immediate reply from DSP core */ - hda_dsp_ipc_get_reply(sdev); - snd_sof_ipc_reply(sdev, msg); + /* handle immediate reply from DSP core */ + hda_dsp_ipc_get_reply(sdev); + snd_sof_ipc_reply(sdev, msg); - cnl_ipc_dsp_done(sdev); + cnl_ipc_dsp_done(sdev); - spin_unlock_irq(&sdev->ipc_lock); + spin_unlock_irq(&sdev->ipc_lock); + } else { + dev_dbg_ratelimited(sdev->dev, "IPC reply before FW_READY: %#x\n", + msg); + } ipc_irq = true; } -- cgit v1.2.3 From 499cc881b09c8283ab5e75b0d6d21cb427722161 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 12 Jul 2022 15:23:56 +0300 Subject: ASoC: SOF: Intel: hda-ipc: Do not process IPC reply before firmware boot It is not yet clear, but it is possible to create a firmware so broken that it will send a reply message before a FW_READY message (it is not yet clear if FW_READY will arrive later). Since the reply_data is allocated only after the FW_READY message, this will lead to a NULL pointer dereference if not filtered out. The issue was reported with IPC4 firmware but the same condition is present for IPC3. Reported-by: Kai Vehmanen Signed-off-by: Peter Ujfalusi Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220712122357.31282-3-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-ipc.c | 39 +++++++++++++++++++++++++-------------- 1 file changed, 25 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/intel/hda-ipc.c b/sound/soc/sof/intel/hda-ipc.c index f08011249955..65e688f749ea 100644 --- a/sound/soc/sof/intel/hda-ipc.c +++ b/sound/soc/sof/intel/hda-ipc.c @@ -148,17 +148,23 @@ irqreturn_t hda_dsp_ipc4_irq_thread(int irq, void *context) if (primary & SOF_IPC4_MSG_DIR_MASK) { /* Reply received */ - struct sof_ipc4_msg *data = sdev->ipc->msg.reply_data; + if (likely(sdev->fw_state == SOF_FW_BOOT_COMPLETE)) { + struct sof_ipc4_msg *data = sdev->ipc->msg.reply_data; - data->primary = primary; - data->extension = extension; + data->primary = primary; + data->extension = extension; - spin_lock_irq(&sdev->ipc_lock); + spin_lock_irq(&sdev->ipc_lock); - snd_sof_ipc_get_reply(sdev); - snd_sof_ipc_reply(sdev, data->primary); + snd_sof_ipc_get_reply(sdev); + snd_sof_ipc_reply(sdev, data->primary); - spin_unlock_irq(&sdev->ipc_lock); + spin_unlock_irq(&sdev->ipc_lock); + } else { + dev_dbg_ratelimited(sdev->dev, + "IPC reply before FW_READY: %#x|%#x\n", + primary, extension); + } } else { /* Notification received */ @@ -225,16 +231,21 @@ irqreturn_t hda_dsp_ipc_irq_thread(int irq, void *context) * place, the message might not yet be marked as expecting a * reply. */ - spin_lock_irq(&sdev->ipc_lock); + if (likely(sdev->fw_state == SOF_FW_BOOT_COMPLETE)) { + spin_lock_irq(&sdev->ipc_lock); - /* handle immediate reply from DSP core */ - hda_dsp_ipc_get_reply(sdev); - snd_sof_ipc_reply(sdev, msg); + /* handle immediate reply from DSP core */ + hda_dsp_ipc_get_reply(sdev); + snd_sof_ipc_reply(sdev, msg); - /* set the done bit */ - hda_dsp_ipc_dsp_done(sdev); + /* set the done bit */ + hda_dsp_ipc_dsp_done(sdev); - spin_unlock_irq(&sdev->ipc_lock); + spin_unlock_irq(&sdev->ipc_lock); + } else { + dev_dbg_ratelimited(sdev->dev, "IPC reply before FW_READY: %#x\n", + msg); + } ipc_irq = true; } -- cgit v1.2.3 From 1549a69b89b7e5b1b830da897529344766728a4b Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 12 Jul 2022 15:23:57 +0300 Subject: ASoC: SOF: Intel: mtl: Do not process IPC reply before firmware boot It is not yet clear, but it is possible to create a firmware so broken that it will send a reply message before a FW_READY message (it is not yet clear if FW_READY will arrive later). Since the reply_data is allocated only after the FW_READY message, this will lead to a NULL pointer dereference if not filtered out. Reported-by: Kai Vehmanen Signed-off-by: Peter Ujfalusi Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220712122357.31282-4-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/mtl.c | 20 +++++++++++++------- 1 file changed, 13 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/intel/mtl.c b/sound/soc/sof/intel/mtl.c index 3a043589c12b..a5e244ea0688 100644 --- a/sound/soc/sof/intel/mtl.c +++ b/sound/soc/sof/intel/mtl.c @@ -512,17 +512,23 @@ static irqreturn_t mtl_ipc_irq_thread(int irq, void *context) */ if (primary & SOF_IPC4_MSG_DIR_MASK) { /* Reply received */ - struct sof_ipc4_msg *data = sdev->ipc->msg.reply_data; + if (likely(sdev->fw_state == SOF_FW_BOOT_COMPLETE)) { + struct sof_ipc4_msg *data = sdev->ipc->msg.reply_data; - data->primary = primary; - data->extension = extension; + data->primary = primary; + data->extension = extension; - spin_lock_irq(&sdev->ipc_lock); + spin_lock_irq(&sdev->ipc_lock); - snd_sof_ipc_get_reply(sdev); - snd_sof_ipc_reply(sdev, data->primary); + snd_sof_ipc_get_reply(sdev); + snd_sof_ipc_reply(sdev, data->primary); - spin_unlock_irq(&sdev->ipc_lock); + spin_unlock_irq(&sdev->ipc_lock); + } else { + dev_dbg_ratelimited(sdev->dev, + "IPC reply before FW_READY: %#x|%#x\n", + primary, extension); + } } else { /* Notification received */ notification_data.primary = primary; -- cgit v1.2.3 From 57724db17a946476f11c1b1be9750bc0cf877adc Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 12 Jul 2022 15:09:35 +0300 Subject: ASoC: SOF: Intel: hda: Introduce skip_imr_boot flag Use a dedicated flag instead of directly checking the sdev->system_suspend_target to decide if we need to skip IMR boot due to too deep sleep state where the memory used for IMR booting will not retain its content. The skip_imr_boot flag will be set true during suspend if the target state is deeper than S3 and reset back to false on successful boot to re-enable IMR booting in shallower sleep states. Signed-off-by: Peter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/20220712120936.28072-2-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-dsp.c | 7 +++++++ sound/soc/sof/intel/hda-loader.c | 10 ++++++---- sound/soc/sof/intel/hda.h | 1 + 3 files changed, 14 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index 2afaee91b982..eddfd77ad90f 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -617,6 +617,13 @@ static int hda_suspend(struct snd_sof_dev *sdev, bool runtime_suspend) #endif int ret, j; + /* + * The memory used for IMR boot loses its content in deeper than S3 state + * We must not try IMR boot on next power up (as it will fail). + */ + if (sdev->system_suspend_target > SOF_SUSPEND_S3) + hda->skip_imr_boot = true; + hda_sdw_int_enable(sdev, false); /* disable IPC interrupts */ diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c index 819b3b08c655..eb22eb3f6fee 100644 --- a/sound/soc/sof/intel/hda-loader.c +++ b/sound/soc/sof/intel/hda-loader.c @@ -395,8 +395,7 @@ int hda_dsp_cl_boot_firmware(struct snd_sof_dev *sdev) struct snd_dma_buffer dmab; int ret, ret1, i; - if (sdev->system_suspend_target < SOF_SUSPEND_S4 && - hda->imrboot_supported && !sdev->first_boot) { + if (hda->imrboot_supported && !sdev->first_boot && !hda->skip_imr_boot) { dev_dbg(sdev->dev, "IMR restore supported, booting from IMR directly\n"); hda->boot_iteration = 0; ret = hda_dsp_boot_imr(sdev); @@ -480,11 +479,14 @@ int hda_dsp_cl_boot_firmware(struct snd_sof_dev *sdev) */ hda->boot_iteration = HDA_FW_BOOT_ATTEMPTS; ret = hda_cl_copy_fw(sdev, hext_stream); - if (!ret) + if (!ret) { dev_dbg(sdev->dev, "Firmware download successful, booting...\n"); - else + hda->skip_imr_boot = false; + } else { snd_sof_dsp_dbg_dump(sdev, "Firmware download failed", SOF_DBG_DUMP_PCI | SOF_DBG_DUMP_MBOX); + hda->skip_imr_boot = true; + } cleanup: /* diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index dc713c20ba1d..2b4d23af6054 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -419,6 +419,7 @@ enum sof_hda_D0_substate { /* represents DSP HDA controller frontend - i.e. host facing control */ struct sof_intel_hda_dev { bool imrboot_supported; + bool skip_imr_boot; int boot_iteration; -- cgit v1.2.3 From 4ccf0949cd364811217a0e61754ff7e52cb4f0e4 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 8 Jul 2022 15:06:41 -0500 Subject: ASoC: soc-pcm: demote warnings on non-atomic BE connection MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit When an FE, typically non-atomic, is connected to an atomic BE, we force the BE as non-atomic. There's no reason to throw a warning, this is a perfectly fine configuration and a conversion that's required by-design. This removes the unconditional warnings such as [ 12.054213] iDisp1: dpcm_be_connect: FE is nonatomic but BE is not, forcing BE as nonatomic [ 12.074693] iDisp2: dpcm_be_connect: FE is nonatomic but BE is not, forcing BE as nonatomic [ 12.096612] iDisp3: dpcm_be_connect: FE is nonatomic but BE is not, forcing BE as nonatomic [ 12.118637] iDisp4: dpcm_be_connect: FE is nonatomic but BE is not, forcing BE as nonatomic [ 12.140660] dmic01: dpcm_be_connect: FE is nonatomic but BE is not, forcing BE as nonatomic [ 12.147521] dmic16k: dpcm_be_connect: FE is nonatomic but BE is not, forcing BE as nonatomic and demotes them to dev_dbg(), as suggested in review comments. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Péter Ujfalusi Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/20220708200641.26923-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index a827cc3c158a..5b99bf2dbd08 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1209,8 +1209,7 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, return -EINVAL; } if (fe_substream->pcm->nonatomic && !be_substream->pcm->nonatomic) { - dev_warn(be->dev, "%s: FE is nonatomic but BE is not, forcing BE as nonatomic\n", - __func__); + dev_dbg(be->dev, "FE is nonatomic but BE is not, forcing BE as nonatomic\n"); be_substream->pcm->nonatomic = 1; } -- cgit v1.2.3 From 98418a08a20d3a72e14d88ccb3a48d0bf961ab6a Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Tue, 12 Jul 2022 15:39:02 +0300 Subject: ASoC: SOF: topology: remove unused variable 'ret' is never used. Remove it and return 0 instead. Signed-off-by: Ranjani Sridharan Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20220712123902.14696-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/topology.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 7e54eb1bf77b..9273a70fec25 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -1419,7 +1419,6 @@ static int sof_widget_unload(struct snd_soc_component *scomp, struct soc_bytes_ext *sbe; struct snd_sof_dai *dai; struct soc_enum *se; - int ret = 0; int i; swidget = dobj->private; @@ -1480,7 +1479,7 @@ out: list_del(&swidget->list); kfree(swidget); - return ret; + return 0; } /* -- cgit v1.2.3 From 15d8370cf6d5b3316ad58954086433301363be67 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 12 Jul 2022 15:57:32 +0300 Subject: ASoC: SOF: Intel: hda: Correct the ROM/FW state reporting code The FSR (Firmware State Register) can be found at offset 0 in the SRAM and it is holding information about the state of the ROM/FW. In case of a boot failure it can be used to get the state where the boot process got stuck, it does not itself contains error codes as such. The error code (or the firmware state information) is stored in the next soft register at offset 0x4. Signed-off-by: Peter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Rander Wang Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/20220712125734.30512-2-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda.c | 147 +++++++++++++++++++++++++++++++++++++++------- sound/soc/sof/intel/hda.h | 63 ++++++++++++++++++++ 2 files changed, 190 insertions(+), 20 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index bc07df1fc39f..d519b9802b3b 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -353,7 +353,7 @@ static inline bool hda_sdw_check_wakeen_irq(struct snd_sof_dev *sdev) struct hda_dsp_msg_code { u32 code; - const char *msg; + const char *text; }; #if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG) @@ -382,10 +382,7 @@ module_param_named(use_common_hdmi, hda_codec_use_common_hdmi, bool, 0444); MODULE_PARM_DESC(use_common_hdmi, "SOF HDA use common HDMI codec driver"); #endif -static const struct hda_dsp_msg_code hda_dsp_rom_msg[] = { - {HDA_DSP_ROM_FW_MANIFEST_LOADED, "status: manifest loaded"}, - {HDA_DSP_ROM_FW_FW_LOADED, "status: fw loaded"}, - {HDA_DSP_ROM_FW_ENTERED, "status: fw entered"}, +static const struct hda_dsp_msg_code hda_dsp_rom_fw_error_texts[] = { {HDA_DSP_ROM_CSE_ERROR, "error: cse error"}, {HDA_DSP_ROM_CSE_WRONG_RESPONSE, "error: cse wrong response"}, {HDA_DSP_ROM_IMR_TO_SMALL, "error: IMR too small"}, @@ -404,26 +401,136 @@ static const struct hda_dsp_msg_code hda_dsp_rom_msg[] = { {HDA_DSP_ROM_NULL_FW_ENTRY, "error: null FW entry point"}, }; -static void hda_dsp_get_status(struct snd_sof_dev *sdev, const char *level) +#define FSR_ROM_STATE_ENTRY(state) {FSR_STATE_ROM_##state, #state} +static const struct hda_dsp_msg_code fsr_rom_state_names[] = { + FSR_ROM_STATE_ENTRY(INIT), + FSR_ROM_STATE_ENTRY(INIT_DONE), + FSR_ROM_STATE_ENTRY(CSE_MANIFEST_LOADED), + FSR_ROM_STATE_ENTRY(FW_MANIFEST_LOADED), + FSR_ROM_STATE_ENTRY(FW_FW_LOADED), + FSR_ROM_STATE_ENTRY(FW_ENTERED), + FSR_ROM_STATE_ENTRY(VERIFY_FEATURE_MASK), + FSR_ROM_STATE_ENTRY(GET_LOAD_OFFSET), + FSR_ROM_STATE_ENTRY(FETCH_ROM_EXT), + FSR_ROM_STATE_ENTRY(FETCH_ROM_EXT_DONE), + /* CSE states */ + FSR_ROM_STATE_ENTRY(CSE_IMR_REQUEST), + FSR_ROM_STATE_ENTRY(CSE_IMR_GRANTED), + FSR_ROM_STATE_ENTRY(CSE_VALIDATE_IMAGE_REQUEST), + FSR_ROM_STATE_ENTRY(CSE_IMAGE_VALIDATED), + FSR_ROM_STATE_ENTRY(CSE_IPC_IFACE_INIT), + FSR_ROM_STATE_ENTRY(CSE_IPC_RESET_PHASE_1), + FSR_ROM_STATE_ENTRY(CSE_IPC_OPERATIONAL_ENTRY), + FSR_ROM_STATE_ENTRY(CSE_IPC_OPERATIONAL), + FSR_ROM_STATE_ENTRY(CSE_IPC_DOWN), +}; + +#define FSR_BRINGUP_STATE_ENTRY(state) {FSR_STATE_BRINGUP_##state, #state} +static const struct hda_dsp_msg_code fsr_bringup_state_names[] = { + FSR_BRINGUP_STATE_ENTRY(INIT), + FSR_BRINGUP_STATE_ENTRY(INIT_DONE), + FSR_BRINGUP_STATE_ENTRY(HPSRAM_LOAD), + FSR_BRINGUP_STATE_ENTRY(UNPACK_START), + FSR_BRINGUP_STATE_ENTRY(IMR_RESTORE), + FSR_BRINGUP_STATE_ENTRY(FW_ENTERED), +}; + +#define FSR_WAIT_STATE_ENTRY(state) {FSR_WAIT_FOR_##state, #state} +static const struct hda_dsp_msg_code fsr_wait_state_names[] = { + FSR_WAIT_STATE_ENTRY(IPC_BUSY), + FSR_WAIT_STATE_ENTRY(IPC_DONE), + FSR_WAIT_STATE_ENTRY(CACHE_INVALIDATION), + FSR_WAIT_STATE_ENTRY(LP_SRAM_OFF), + FSR_WAIT_STATE_ENTRY(DMA_BUFFER_FULL), + FSR_WAIT_STATE_ENTRY(CSE_CSR), +}; + +#define FSR_MODULE_NAME_ENTRY(mod) [FSR_MOD_##mod] = #mod +static const char * const fsr_module_names[] = { + FSR_MODULE_NAME_ENTRY(ROM), + FSR_MODULE_NAME_ENTRY(ROM_BYP), + FSR_MODULE_NAME_ENTRY(BASE_FW), + FSR_MODULE_NAME_ENTRY(LP_BOOT), + FSR_MODULE_NAME_ENTRY(BRNGUP), + FSR_MODULE_NAME_ENTRY(ROM_EXT), +}; + +static const char * +hda_dsp_get_state_text(u32 code, const struct hda_dsp_msg_code *msg_code, + size_t array_size) { - const struct sof_intel_dsp_desc *chip; - u32 status; int i; - chip = get_chip_info(sdev->pdata); - status = snd_sof_dsp_read(sdev, HDA_DSP_BAR, - chip->rom_status_reg); - - for (i = 0; i < ARRAY_SIZE(hda_dsp_rom_msg); i++) { - if (status == hda_dsp_rom_msg[i].code) { - dev_printk(level, sdev->dev, "%s - code %8.8x\n", - hda_dsp_rom_msg[i].msg, status); - return; - } + for (i = 0; i < array_size; i++) { + if (code == msg_code[i].code) + return msg_code[i].text; } + return NULL; +} + +static void hda_dsp_get_state(struct snd_sof_dev *sdev, const char *level) +{ + const struct sof_intel_dsp_desc *chip = get_chip_info(sdev->pdata); + const char *state_text, *error_text, *module_text; + u32 fsr, state, wait_state, module, error_code; + + fsr = snd_sof_dsp_read(sdev, HDA_DSP_BAR, chip->rom_status_reg); + state = FSR_TO_STATE_CODE(fsr); + wait_state = FSR_TO_WAIT_STATE_CODE(fsr); + module = FSR_TO_MODULE_CODE(fsr); + + if (module > FSR_MOD_ROM_EXT) + module_text = "unknown"; + else + module_text = fsr_module_names[module]; + + if (module == FSR_MOD_BRNGUP) + state_text = hda_dsp_get_state_text(state, fsr_bringup_state_names, + ARRAY_SIZE(fsr_bringup_state_names)); + else + state_text = hda_dsp_get_state_text(state, fsr_rom_state_names, + ARRAY_SIZE(fsr_rom_state_names)); + /* not for us, must be generic sof message */ - dev_dbg(sdev->dev, "unknown ROM status value %8.8x\n", status); + if (!state_text) { + dev_printk(level, sdev->dev, "%#010x: unknown ROM status value\n", fsr); + return; + } + + if (wait_state) { + const char *wait_state_text; + + wait_state_text = hda_dsp_get_state_text(wait_state, fsr_wait_state_names, + ARRAY_SIZE(fsr_wait_state_names)); + if (!wait_state_text) + wait_state_text = "unknown"; + + dev_printk(level, sdev->dev, + "%#010x: module: %s, state: %s, waiting for: %s, %s\n", + fsr, module_text, state_text, wait_state_text, + fsr & FSR_HALTED ? "not running" : "running"); + } else { + dev_printk(level, sdev->dev, "%#010x: module: %s, state: %s, %s\n", + fsr, module_text, state_text, + fsr & FSR_HALTED ? "not running" : "running"); + } + + error_code = snd_sof_dsp_read(sdev, HDA_DSP_BAR, chip->rom_status_reg + 4); + if (!error_code) + return; + + error_text = hda_dsp_get_state_text(error_code, hda_dsp_rom_fw_error_texts, + ARRAY_SIZE(hda_dsp_rom_fw_error_texts)); + if (!error_text) + error_text = "unknown"; + + if (state == FSR_STATE_FW_ENTERED) + dev_printk(level, sdev->dev, "status code: %#x (%s)\n", error_code, + error_text); + else + dev_printk(level, sdev->dev, "error code: %#x (%s)\n", error_code, + error_text); } static void hda_dsp_get_registers(struct snd_sof_dev *sdev, @@ -482,7 +589,7 @@ void hda_dsp_dump(struct snd_sof_dev *sdev, u32 flags) u32 stack[HDA_DSP_STACK_DUMP_SIZE]; /* print ROM/FW status */ - hda_dsp_get_status(sdev, level); + hda_dsp_get_state(sdev, level); if (flags & SOF_DBG_DUMP_REGS) { u32 status = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_SRAM_REG_FW_STATUS); diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index dc713c20ba1d..77c74e3c09b1 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -187,6 +187,69 @@ #define HDA_DSP_STACK_DUMP_SIZE 32 +/* ROM/FW status register */ +#define FSR_STATE_MASK GENMASK(23, 0) +#define FSR_WAIT_STATE_MASK GENMASK(27, 24) +#define FSR_MODULE_MASK GENMASK(30, 28) +#define FSR_HALTED BIT(31) +#define FSR_TO_STATE_CODE(x) ((x) & FSR_STATE_MASK) +#define FSR_TO_WAIT_STATE_CODE(x) (((x) & FSR_WAIT_STATE_MASK) >> 24) +#define FSR_TO_MODULE_CODE(x) (((x) & FSR_MODULE_MASK) >> 28) + +/* Wait states */ +#define FSR_WAIT_FOR_IPC_BUSY 0x1 +#define FSR_WAIT_FOR_IPC_DONE 0x2 +#define FSR_WAIT_FOR_CACHE_INVALIDATION 0x3 +#define FSR_WAIT_FOR_LP_SRAM_OFF 0x4 +#define FSR_WAIT_FOR_DMA_BUFFER_FULL 0x5 +#define FSR_WAIT_FOR_CSE_CSR 0x6 + +/* Module codes */ +#define FSR_MOD_ROM 0x0 +#define FSR_MOD_ROM_BYP 0x1 +#define FSR_MOD_BASE_FW 0x2 +#define FSR_MOD_LP_BOOT 0x3 +#define FSR_MOD_BRNGUP 0x4 +#define FSR_MOD_ROM_EXT 0x5 + +/* State codes (module dependent) */ +/* Module independent states */ +#define FSR_STATE_INIT 0x0 +#define FSR_STATE_INIT_DONE 0x1 +#define FSR_STATE_FW_ENTERED 0x5 + +/* ROM states */ +#define FSR_STATE_ROM_INIT FSR_STATE_INIT +#define FSR_STATE_ROM_INIT_DONE FSR_STATE_INIT_DONE +#define FSR_STATE_ROM_CSE_MANIFEST_LOADED 0x2 +#define FSR_STATE_ROM_FW_MANIFEST_LOADED 0x3 +#define FSR_STATE_ROM_FW_FW_LOADED 0x4 +#define FSR_STATE_ROM_FW_ENTERED FSR_STATE_FW_ENTERED +#define FSR_STATE_ROM_VERIFY_FEATURE_MASK 0x6 +#define FSR_STATE_ROM_GET_LOAD_OFFSET 0x7 +#define FSR_STATE_ROM_FETCH_ROM_EXT 0x8 +#define FSR_STATE_ROM_FETCH_ROM_EXT_DONE 0x9 + +/* (ROM) CSE states */ +#define FSR_STATE_ROM_CSE_IMR_REQUEST 0x10 +#define FSR_STATE_ROM_CSE_IMR_GRANTED 0x11 +#define FSR_STATE_ROM_CSE_VALIDATE_IMAGE_REQUEST 0x12 +#define FSR_STATE_ROM_CSE_IMAGE_VALIDATED 0x13 + +#define FSR_STATE_ROM_CSE_IPC_IFACE_INIT 0x20 +#define FSR_STATE_ROM_CSE_IPC_RESET_PHASE_1 0x21 +#define FSR_STATE_ROM_CSE_IPC_OPERATIONAL_ENTRY 0x22 +#define FSR_STATE_ROM_CSE_IPC_OPERATIONAL 0x23 +#define FSR_STATE_ROM_CSE_IPC_DOWN 0x24 + +/* BRINGUP (or BRNGUP) states */ +#define FSR_STATE_BRINGUP_INIT FSR_STATE_INIT +#define FSR_STATE_BRINGUP_INIT_DONE FSR_STATE_INIT_DONE +#define FSR_STATE_BRINGUP_HPSRAM_LOAD 0x2 +#define FSR_STATE_BRINGUP_UNPACK_START 0X3 +#define FSR_STATE_BRINGUP_IMR_RESTORE 0x4 +#define FSR_STATE_BRINGUP_FW_ENTERED FSR_STATE_FW_ENTERED + /* ROM status/error values */ #define HDA_DSP_ROM_STS_MASK GENMASK(23, 0) #define HDA_DSP_ROM_INIT 0x1 -- cgit v1.2.3 From 402355e6cdbebf15f2c40cd9469b924c97b94b32 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 12 Jul 2022 16:16:19 +0300 Subject: ASoC: SOF: Intel: hda-dai: Drop misleading comment regarding dma_data The comment in hda_link_dma_hw_params() is no longer valid as the dma_data is set to NULL at system suspend as well. Instead of rewording the comment to state the obvious: try to take the hext_stream from the dma_data and if it is not set then assign a new one and store it as dma_data. Signed-off-by: Peter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Rander Wang Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/20220712131620.13365-2-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-dai.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index ed74a1f264e8..9015ca2024bc 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -221,7 +221,6 @@ static int hda_link_dma_hw_params(struct snd_pcm_substream *substream, struct hdac_bus *bus = hstream->bus; struct hdac_ext_link *link; - /* get stored dma data if resuming from system suspend */ hext_stream = snd_soc_dai_get_dma_data(cpu_dai, substream); if (!hext_stream) { hext_stream = hda_link_stream_assign(bus, substream); -- cgit v1.2.3 From fbabebfb26a8130c10fd91cca687bac87944580d Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 12 Jul 2022 16:16:20 +0300 Subject: ASoC: SOF: Intel: hda-dai: Do snd_hdac_ext_stream_decouple() only once Call snd_hdac_ext_stream_decouple_locked() unconditionally in hda_link_stream_assign(), the snd_hdac_ext_stream_decouple_locked() have internal checks to avoid re-configuring. There is no need to call snd_hdac_ext_stream_decouple() via hda_link_dma_params() as the stream must have been set to decoupled when it got assigned (even if it used local condition to call snd_hdac_ext_stream_decouple_locked()). Signed-off-by: Peter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Rander Wang Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/20220712131620.13365-3-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-dai.c | 9 ++------- 1 file changed, 2 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index 9015ca2024bc..c5b65e4a06be 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -126,12 +126,8 @@ hda_link_stream_assign(struct hdac_bus *bus, } if (res) { - /* - * Decouple host and link DMA. The decoupled flag - * is updated in snd_hdac_ext_stream_decouple(). - */ - if (!res->decoupled) - snd_hdac_ext_stream_decouple_locked(bus, res, true); + /* Make sure that host and link DMA is decoupled. */ + snd_hdac_ext_stream_decouple_locked(bus, res, true); res->link_locked = 1; res->link_substream = substream; @@ -184,7 +180,6 @@ static int hda_link_dma_params(struct hdac_ext_stream *hext_stream, struct hdac_ext_link *link; unsigned int format_val; - snd_hdac_ext_stream_decouple(bus, hext_stream, true); snd_hdac_ext_link_stream_reset(hext_stream); format_val = snd_hdac_calc_stream_format(params->s_freq, params->ch, -- cgit v1.2.3 From d5770daef62d2e4d33015089bab392ef867fd35a Mon Sep 17 00:00:00 2001 From: Daniel Baluta Date: Tue, 12 Jul 2022 17:15:28 +0300 Subject: ASoC: SOF: compress: Dynamically allocate pcm params struct MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit We need to extend sof_ipc_pcm_parmas with additional data in order to send compress_params to SOF FW. The extensions will be done at runtime so we need to dynamically allocate pcm object of type struct sof_ipc_pcm_params. Signed-off-by: Daniel Baluta Reviewed-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Link: https://lore.kernel.org/r/20220712141531.14599-2-daniel.baluta@oss.nxp.com Signed-off-by: Mark Brown --- sound/soc/sof/compress.c | 53 ++++++++++++++++++++++++++---------------------- 1 file changed, 29 insertions(+), 24 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/compress.c b/sound/soc/sof/compress.c index 47639b6344c8..45c2ff61ee4d 100644 --- a/sound/soc/sof/compress.c +++ b/sound/soc/sof/compress.c @@ -168,7 +168,7 @@ static int sof_compr_set_params(struct snd_soc_component *component, struct snd_compr_runtime *crtd = cstream->runtime; struct sof_ipc_pcm_params_reply ipc_params_reply; struct snd_compr_tstamp *tstamp; - struct sof_ipc_pcm_params pcm; + struct sof_ipc_pcm_params *pcm; struct snd_sof_pcm *spcm; int ret; @@ -179,40 +179,42 @@ static int sof_compr_set_params(struct snd_soc_component *component, if (!spcm) return -EINVAL; + pcm = kzalloc(sizeof(*pcm), GFP_KERNEL); + if (!pcm) + return -ENOMEM; + cstream->dma_buffer.dev.type = SNDRV_DMA_TYPE_DEV_SG; cstream->dma_buffer.dev.dev = sdev->dev; ret = snd_compr_malloc_pages(cstream, crtd->buffer_size); if (ret < 0) - return ret; + goto out; ret = create_page_table(component, cstream, crtd->dma_area, crtd->dma_bytes); if (ret < 0) - return ret; - - memset(&pcm, 0, sizeof(pcm)); - - pcm.params.buffer.pages = PFN_UP(crtd->dma_bytes); - pcm.hdr.size = sizeof(pcm); - pcm.hdr.cmd = SOF_IPC_GLB_STREAM_MSG | SOF_IPC_STREAM_PCM_PARAMS; - - pcm.comp_id = spcm->stream[cstream->direction].comp_id; - pcm.params.hdr.size = sizeof(pcm.params); - pcm.params.buffer.phy_addr = spcm->stream[cstream->direction].page_table.addr; - pcm.params.buffer.size = crtd->dma_bytes; - pcm.params.direction = cstream->direction; - pcm.params.channels = params->codec.ch_out; - pcm.params.rate = params->codec.sample_rate; - pcm.params.buffer_fmt = SOF_IPC_BUFFER_INTERLEAVED; - pcm.params.frame_fmt = SOF_IPC_FRAME_S32_LE; - pcm.params.sample_container_bytes = + goto out; + + pcm->params.buffer.pages = PFN_UP(crtd->dma_bytes); + pcm->hdr.size = sizeof(*pcm); + pcm->hdr.cmd = SOF_IPC_GLB_STREAM_MSG | SOF_IPC_STREAM_PCM_PARAMS; + + pcm->comp_id = spcm->stream[cstream->direction].comp_id; + pcm->params.hdr.size = sizeof(pcm->params); + pcm->params.buffer.phy_addr = spcm->stream[cstream->direction].page_table.addr; + pcm->params.buffer.size = crtd->dma_bytes; + pcm->params.direction = cstream->direction; + pcm->params.channels = params->codec.ch_out; + pcm->params.rate = params->codec.sample_rate; + pcm->params.buffer_fmt = SOF_IPC_BUFFER_INTERLEAVED; + pcm->params.frame_fmt = SOF_IPC_FRAME_S32_LE; + pcm->params.sample_container_bytes = snd_pcm_format_physical_width(SNDRV_PCM_FORMAT_S32) >> 3; - pcm.params.host_period_bytes = params->buffer.fragment_size; + pcm->params.host_period_bytes = params->buffer.fragment_size; - ret = sof_ipc_tx_message(sdev->ipc, &pcm, sizeof(pcm), + ret = sof_ipc_tx_message(sdev->ipc, pcm, sizeof(*pcm), &ipc_params_reply, sizeof(ipc_params_reply)); if (ret < 0) { dev_err(component->dev, "error ipc failed\n"); - return ret; + goto out; } tstamp->byte_offset = sdev->stream_box.offset + ipc_params_reply.posn_offset; @@ -220,7 +222,10 @@ static int sof_compr_set_params(struct snd_soc_component *component, spcm->prepared[cstream->direction] = true; - return 0; +out: + kfree(pcm); + + return ret; } static int sof_compr_get_params(struct snd_soc_component *component, -- cgit v1.2.3 From 3f70c360d484466da7420f395d4675ca02436e32 Mon Sep 17 00:00:00 2001 From: Daniel Baluta Date: Tue, 12 Jul 2022 17:15:29 +0300 Subject: ASoC: SOF: Copy compress parameters into extended data MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Allocate memory at the end of sof_ipc_stream_params to store snd_compr_params in order to be sent them to SOF firmware. This will help firmware correctly configure codecs parameters. Notice, that we use 2 bytes from the reserved pool in order to store the extended data length. This is compatible with older FWs where there was no extended data. Signed-off-by: Daniel Baluta Reviewed-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Link: https://lore.kernel.org/r/20220712141531.14599-3-daniel.baluta@oss.nxp.com Signed-off-by: Mark Brown --- sound/soc/sof/compress.c | 17 +++++++++++++---- 1 file changed, 13 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/compress.c b/sound/soc/sof/compress.c index 45c2ff61ee4d..1204dce29ef9 100644 --- a/sound/soc/sof/compress.c +++ b/sound/soc/sof/compress.c @@ -170,6 +170,7 @@ static int sof_compr_set_params(struct snd_soc_component *component, struct snd_compr_tstamp *tstamp; struct sof_ipc_pcm_params *pcm; struct snd_sof_pcm *spcm; + size_t ext_data_size; int ret; tstamp = crtd->private_data; @@ -179,7 +180,12 @@ static int sof_compr_set_params(struct snd_soc_component *component, if (!spcm) return -EINVAL; - pcm = kzalloc(sizeof(*pcm), GFP_KERNEL); + ext_data_size = sizeof(params->codec); + + if (sizeof(*pcm) + ext_data_size > sdev->ipc->max_payload_size) + return -EINVAL; + + pcm = kzalloc(sizeof(*pcm) + ext_data_size, GFP_KERNEL); if (!pcm) return -ENOMEM; @@ -194,11 +200,11 @@ static int sof_compr_set_params(struct snd_soc_component *component, goto out; pcm->params.buffer.pages = PFN_UP(crtd->dma_bytes); - pcm->hdr.size = sizeof(*pcm); + pcm->hdr.size = sizeof(*pcm) + ext_data_size; pcm->hdr.cmd = SOF_IPC_GLB_STREAM_MSG | SOF_IPC_STREAM_PCM_PARAMS; pcm->comp_id = spcm->stream[cstream->direction].comp_id; - pcm->params.hdr.size = sizeof(pcm->params); + pcm->params.hdr.size = sizeof(pcm->params) + ext_data_size; pcm->params.buffer.phy_addr = spcm->stream[cstream->direction].page_table.addr; pcm->params.buffer.size = crtd->dma_bytes; pcm->params.direction = cstream->direction; @@ -209,8 +215,11 @@ static int sof_compr_set_params(struct snd_soc_component *component, pcm->params.sample_container_bytes = snd_pcm_format_physical_width(SNDRV_PCM_FORMAT_S32) >> 3; pcm->params.host_period_bytes = params->buffer.fragment_size; + pcm->params.ext_data_length = ext_data_size; + + memcpy((u8 *)pcm->params.ext_data, ¶ms->codec, ext_data_size); - ret = sof_ipc_tx_message(sdev->ipc, pcm, sizeof(*pcm), + ret = sof_ipc_tx_message(sdev->ipc, pcm, sizeof(*pcm) + ext_data_size, &ipc_params_reply, sizeof(ipc_params_reply)); if (ret < 0) { dev_err(component->dev, "error ipc failed\n"); -- cgit v1.2.3 From 246b135fcdba57a4e77a702580391ae1942c1e3b Mon Sep 17 00:00:00 2001 From: Daniel Baluta Date: Tue, 12 Jul 2022 17:15:30 +0300 Subject: ASoC: SOF: compress: Prevent current kernel running with older FW MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit After introducing extended parameters we need to forbid older firmware versions to run with the current and future kernel versions. Although in theory the communication protocol will still work the semantics at application level are undefined. So, prevent this by disallowing older firmwares to run with newer kernels. Signed-off-by: Daniel Baluta Reviewed-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Link: https://lore.kernel.org/r/20220712141531.14599-4-daniel.baluta@oss.nxp.com Signed-off-by: Mark Brown --- sound/soc/sof/compress.c | 11 +++++++++++ 1 file changed, 11 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/compress.c b/sound/soc/sof/compress.c index 1204dce29ef9..67139e15f862 100644 --- a/sound/soc/sof/compress.c +++ b/sound/soc/sof/compress.c @@ -167,12 +167,23 @@ static int sof_compr_set_params(struct snd_soc_component *component, struct snd_soc_pcm_runtime *rtd = cstream->private_data; struct snd_compr_runtime *crtd = cstream->runtime; struct sof_ipc_pcm_params_reply ipc_params_reply; + struct sof_ipc_fw_ready *ready = &sdev->fw_ready; + struct sof_ipc_fw_version *v = &ready->version; struct snd_compr_tstamp *tstamp; struct sof_ipc_pcm_params *pcm; struct snd_sof_pcm *spcm; size_t ext_data_size; int ret; + if (v->abi_version < SOF_ABI_VER(3, 22, 0)) { + dev_err(component->dev, + "Compress params not supported with FW ABI version %d:%d:%d\n", + SOF_ABI_VERSION_MAJOR(v->abi_version), + SOF_ABI_VERSION_MINOR(v->abi_version), + SOF_ABI_VERSION_PATCH(v->abi_version)); + return -EINVAL; + } + tstamp = crtd->private_data; spcm = snd_sof_find_spcm_dai(component, rtd); -- cgit v1.2.3 From 9b93eda355089b36482f7a2f134bdd24be70f907 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 12 Jul 2022 16:10:22 +0300 Subject: ASoC: SOF: sof-client-probes: Only load the driver if IPC3 is used The current implementation of probes only supports IPC3 and should not be loaded for other IPC implementation. Signed-off-by: Peter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/20220712131022.1124-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/sof-client-probes.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/sof-client-probes.c b/sound/soc/sof/sof-client-probes.c index 1f1ea93a7fbf..e7e78d1a8ec3 100644 --- a/sound/soc/sof/sof-client-probes.c +++ b/sound/soc/sof/sof-client-probes.c @@ -694,6 +694,10 @@ static int sof_probes_client_probe(struct auxiliary_device *auxdev, if (!sof_probes_enabled) return -ENXIO; + /* only ipc3 is supported */ + if (sof_client_get_ipc_type(cdev) != SOF_IPC) + return -ENXIO; + if (!dev->platform_data) { dev_err(dev, "missing platform data\n"); return -ENODEV; -- cgit v1.2.3 From d5bd47f3ca124058a8e87eae4508afeda2132611 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 12 Jul 2022 16:01:03 +0300 Subject: ASoC: SOF: ipc3-topology: Prevent double freeing of ipc_control_data via load_bytes We have sanity checks for byte controls and if any of the fail the locally allocated scontrol->ipc_control_data is freed up, but not set to NULL. On a rollback path of the error the higher level code will also try to free the scontrol->ipc_control_data which will eventually going to lead to memory corruption as double freeing memory is not a good thing. Fixes: b5cee8feb1d4 ("ASoC: SOF: topology: Make control parsing IPC agnostic") Reported-by: Seppo Ingalsuo Signed-off-by: Peter Ujfalusi Reviewed-by: Seppo Ingalsuo Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220712130103.31514-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc3-topology.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/sof/ipc3-topology.c b/sound/soc/sof/ipc3-topology.c index 9448d5338423..b2cc046b9f60 100644 --- a/sound/soc/sof/ipc3-topology.c +++ b/sound/soc/sof/ipc3-topology.c @@ -1644,6 +1644,7 @@ static int sof_ipc3_control_load_bytes(struct snd_sof_dev *sdev, struct snd_sof_ return 0; err: kfree(scontrol->ipc_control_data); + scontrol->ipc_control_data = NULL; return ret; } -- cgit v1.2.3 From 20bcf721068f6418607283cdb0c16cf0b606cfc1 Mon Sep 17 00:00:00 2001 From: Andy Shevchenko Date: Tue, 12 Jul 2022 18:35:16 +0300 Subject: ALSA: hda: cs35l41: Don't dereference fwnode handle Use acpi_fwnode_handle() instead of dereferencing an fwnode handle directly, which is a better coding practice. Signed-off-by: Andy Shevchenko Link: https://lore.kernel.org/r/20220712153519.35692-1-andriy.shevchenko@linux.intel.com Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l41_hda.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index 49b25432a9f5..1a1afa0725e0 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -347,7 +347,7 @@ static int cs35l41_hda_read_acpi(struct cs35l41_hda *cs35l41, const char *hid, i /* To use the same release code for all laptop variants we can't use devm_ version of * gpiod_get here, as CLSA010* don't have a fully functional bios with an _DSD node */ - cs35l41->reset_gpio = fwnode_gpiod_get_index(&adev->fwnode, "reset", cs35l41->index, + cs35l41->reset_gpio = fwnode_gpiod_get_index(acpi_fwnode_handle(adev), "reset", cs35l41->index, GPIOD_OUT_LOW, "cs35l41-reset"); property = "cirrus,speaker-position"; -- cgit v1.2.3 From d60b05b4c7802b45c4f2ac003827384618a17bb4 Mon Sep 17 00:00:00 2001 From: Andy Shevchenko Date: Tue, 12 Jul 2022 18:35:17 +0300 Subject: ALSA: hda: cs35l41: Allow compilation test on non-ACPI configurations ACPI is needed only for functioning of this codec on some platforms, there is no compilation dependency, so make it optional Signed-off-by: Andy Shevchenko Link: https://lore.kernel.org/r/20220712153519.35692-2-andriy.shevchenko@linux.intel.com Signed-off-by: Takashi Iwai --- sound/pci/hda/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 79ade4787d95..e86cf80bdf96 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -97,7 +97,7 @@ config SND_HDA_SCODEC_CS35L41 config SND_HDA_SCODEC_CS35L41_I2C tristate "Build CS35L41 HD-audio side codec support for I2C Bus" depends on I2C - depends on ACPI + depends on ACPI || COMPILE_TEST depends on SND_SOC select SND_HDA_GENERIC select SND_SOC_CS35L41_LIB @@ -113,7 +113,7 @@ comment "Set to Y if you want auto-loading the side codec driver" config SND_HDA_SCODEC_CS35L41_SPI tristate "Build CS35L41 HD-audio codec support for SPI Bus" depends on SPI_MASTER - depends on ACPI + depends on ACPI || COMPILE_TEST depends on SND_SOC select SND_HDA_GENERIC select SND_SOC_CS35L41_LIB -- cgit v1.2.3 From 931c940fc5d96e5ef7a2188abfb14e7c3ab1290e Mon Sep 17 00:00:00 2001 From: Andy Shevchenko Date: Tue, 12 Jul 2022 18:35:18 +0300 Subject: ALSA: hda: cs35l41: Drop wrong use of ACPI_PTR() ACPI_PTR() is more harmful than helpful. For example, in this case if CONFIG_ACPI=n, the ID table left unused which is not what we want. Instead of adding ifdeffery or attribute here and there, drop ACPI_PTR(). Signed-off-by: Andy Shevchenko Link: https://lore.kernel.org/r/20220712153519.35692-3-andriy.shevchenko@linux.intel.com Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l41_hda_i2c.c | 8 +++----- sound/pci/hda/cs35l41_hda_spi.c | 8 +++----- 2 files changed, 6 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/cs35l41_hda_i2c.c b/sound/pci/hda/cs35l41_hda_i2c.c index ec626e0fbedc..df39fc76e6be 100644 --- a/sound/pci/hda/cs35l41_hda_i2c.c +++ b/sound/pci/hda/cs35l41_hda_i2c.c @@ -6,9 +6,9 @@ // // Author: Lucas Tanure +#include #include #include -#include #include "cs35l41_hda.h" @@ -43,19 +43,17 @@ static const struct i2c_device_id cs35l41_hda_i2c_id[] = { {} }; -#ifdef CONFIG_ACPI static const struct acpi_device_id cs35l41_acpi_hda_match[] = { {"CLSA0100", 0 }, {"CSC3551", 0 }, - { }, + {} }; MODULE_DEVICE_TABLE(acpi, cs35l41_acpi_hda_match); -#endif static struct i2c_driver cs35l41_i2c_driver = { .driver = { .name = "cs35l41-hda", - .acpi_match_table = ACPI_PTR(cs35l41_acpi_hda_match), + .acpi_match_table = cs35l41_acpi_hda_match, }, .id_table = cs35l41_hda_i2c_id, .probe = cs35l41_hda_i2c_probe, diff --git a/sound/pci/hda/cs35l41_hda_spi.c b/sound/pci/hda/cs35l41_hda_spi.c index 3a1472e1bc24..2f5afad3719e 100644 --- a/sound/pci/hda/cs35l41_hda_spi.c +++ b/sound/pci/hda/cs35l41_hda_spi.c @@ -6,7 +6,7 @@ // // Author: Lucas Tanure -#include +#include #include #include @@ -39,18 +39,16 @@ static const struct spi_device_id cs35l41_hda_spi_id[] = { {} }; -#ifdef CONFIG_ACPI static const struct acpi_device_id cs35l41_acpi_hda_match[] = { { "CSC3551", 0 }, - {}, + {} }; MODULE_DEVICE_TABLE(acpi, cs35l41_acpi_hda_match); -#endif static struct spi_driver cs35l41_spi_driver = { .driver = { .name = "cs35l41-hda", - .acpi_match_table = ACPI_PTR(cs35l41_acpi_hda_match), + .acpi_match_table = cs35l41_acpi_hda_match, }, .id_table = cs35l41_hda_spi_id, .probe = cs35l41_hda_spi_probe, -- cgit v1.2.3 From 33c1f401939c66858157c0665dc07ad9596cd1bd Mon Sep 17 00:00:00 2001 From: Andy Shevchenko Date: Tue, 12 Jul 2022 18:35:19 +0300 Subject: ALSA: hda: cs35l41: Consolidate selections under SND_HDA_SCODEC_CS35L41 Selections can be propagated via selections, while dependencies are not. Hence, consolidate selections under the SND_HDA_SCODEC_CS35L41 option. Signed-off-by: Andy Shevchenko Link: https://lore.kernel.org/r/20220712153519.35692-4-andriy.shevchenko@linux.intel.com Signed-off-by: Takashi Iwai --- sound/pci/hda/Kconfig | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index e86cf80bdf96..8b73a12d356f 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -93,16 +93,16 @@ config SND_HDA_PATCH_LOADER config SND_HDA_SCODEC_CS35L41 tristate + select SND_HDA_GENERIC + select REGMAP_IRQ config SND_HDA_SCODEC_CS35L41_I2C tristate "Build CS35L41 HD-audio side codec support for I2C Bus" depends on I2C depends on ACPI || COMPILE_TEST depends on SND_SOC - select SND_HDA_GENERIC select SND_SOC_CS35L41_LIB select SND_HDA_SCODEC_CS35L41 - select REGMAP_IRQ help Say Y or M here to include CS35L41 I2C HD-audio side codec support in snd-hda-intel driver, such as ALC287. @@ -115,10 +115,8 @@ config SND_HDA_SCODEC_CS35L41_SPI depends on SPI_MASTER depends on ACPI || COMPILE_TEST depends on SND_SOC - select SND_HDA_GENERIC select SND_SOC_CS35L41_LIB select SND_HDA_SCODEC_CS35L41 - select REGMAP_IRQ help Say Y or M here to include CS35L41 SPI HD-audio side codec support in snd-hda-intel driver, such as ALC287. -- cgit v1.2.3 From 539311aa61a144088779f1354492bbf9ae1ac458 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 13 Jul 2022 12:47:53 +0200 Subject: ALSA: compress: Enable kernel doc markers for some functions The exported functions snd_compress_new() and snd_compr_stop_error() had already kernel-doc-style comments but they were not processed as they weren't marked properly. Let's enable them. This patch also fixes the missing argument id for snd_compress_new comments, too. Reported-by: Mauro Carvalho Chehab Link: https://lore.kernel.org/r/3cd6b93b36b32ad6ae160931aaa00b20688e241a.1656759989.git.mchehab@kernel.org Link: https://lore.kernel.org/r/20220713104759.4365-2-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/compress_offload.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index de514ec8c83d..cf415fe231ed 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -810,7 +810,7 @@ static void error_delayed_work(struct work_struct *work) mutex_unlock(&stream->device->lock); } -/* +/** * snd_compr_stop_error: Report a fatal error on a stream * @stream: pointer to stream * @state: state to transition the stream to @@ -1157,11 +1157,12 @@ static int snd_compress_dev_free(struct snd_device *device) return 0; } -/* +/** * snd_compress_new: create new compress device * @card: sound card pointer * @device: device number * @dirn: device direction, should be of type enum snd_compr_direction + * @id: ID string * @compr: compress device pointer */ int snd_compress_new(struct snd_card *card, int device, -- cgit v1.2.3 From 4e2b70673f2b93ab8e037a2b89c15f146c1ae9b0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 13 Jul 2022 12:47:54 +0200 Subject: ALSA: pcm: Fix missing return value comments for kernel docs Each kernel doc comment expects the definition of the return value in a proper format. This patch adds or fixes the missing entries for PCM API. Link: https://lore.kernel.org/r/20220713104759.4365-3-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/pcm.c | 4 ++++ sound/core/pcm_memory.c | 4 ++++ sound/core/pcm_native.c | 6 ++++++ 3 files changed, 14 insertions(+) (limited to 'sound') diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 977d54320a5c..03fc5fa5813e 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -216,6 +216,8 @@ static const char * const snd_pcm_format_names[] = { /** * snd_pcm_format_name - Return a name string for the given PCM format * @format: PCM format + * + * Return: the format name string */ const char *snd_pcm_format_name(snd_pcm_format_t format) { @@ -1138,6 +1140,8 @@ static int snd_pcm_dev_disconnect(struct snd_device *device) * This adds the given notifier to the global list so that the callback is * called for each registered PCM devices. This exists only for PCM OSS * emulation, so far. + * + * Return: zero if successful, or a negative error code */ int snd_pcm_notify(struct snd_pcm_notify *notify, int nfree) { diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c index b8296b6eb2c1..7bde7fb64011 100644 --- a/sound/core/pcm_memory.c +++ b/sound/core/pcm_memory.c @@ -350,6 +350,8 @@ EXPORT_SYMBOL(snd_pcm_lib_preallocate_pages_for_all); * SNDRV_DMA_TYPE_VMALLOC type. * * Upon successful buffer allocation and setup, the function returns 0. + * + * Return: zero if successful, or a negative error code */ int snd_pcm_set_managed_buffer(struct snd_pcm_substream *substream, int type, struct device *data, size_t size, size_t max) @@ -369,6 +371,8 @@ EXPORT_SYMBOL(snd_pcm_set_managed_buffer); * * Do pre-allocation to all substreams of the given pcm for the specified DMA * type and size, and set the managed_buffer_alloc flag to each substream. + * + * Return: zero if successful, or a negative error code */ int snd_pcm_set_managed_buffer_all(struct snd_pcm *pcm, int type, struct device *data, diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 4adaee62ef33..aa0453e51595 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3412,6 +3412,8 @@ static long snd_pcm_ioctl(struct file *file, unsigned int cmd, * The function is provided primarily for OSS layer and USB gadget drivers, * and it allows only the limited set of ioctls (hw_params, sw_params, * prepare, start, drain, drop, forward). + * + * Return: zero if successful, or a negative error code */ int snd_pcm_kernel_ioctl(struct snd_pcm_substream *substream, unsigned int cmd, void *arg) @@ -3810,6 +3812,8 @@ static const struct vm_operations_struct snd_pcm_vm_ops_data_fault = { * * This is the default mmap handler for PCM data. When mmap pcm_ops is NULL, * this function is invoked implicitly. + * + * Return: zero if successful, or a negative error code */ int snd_pcm_lib_default_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *area) @@ -3836,6 +3840,8 @@ EXPORT_SYMBOL_GPL(snd_pcm_lib_default_mmap); * When your hardware uses the iomapped pages as the hardware buffer and * wants to mmap it, pass this function as mmap pcm_ops. Note that this * is supposed to work only on limited architectures. + * + * Return: zero if successful, or a negative error code */ int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream, struct vm_area_struct *area) -- cgit v1.2.3 From 5c121d6362d60198d9e37429f87e87d5477e3555 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 13 Jul 2022 12:47:55 +0200 Subject: ALSA: dmaengine: Fix missing return value comments for kernel docs Each kernel doc comment expects the definition of the return value in a proper format. This patch adds or fixes the missing entries for PCM dmaengine API. Link: https://lore.kernel.org/r/20220713104759.4365-4-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/pcm_dmaengine.c | 30 ++++++++++++++++++++---------- 1 file changed, 20 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c index af6f717e1e7e..5b2ca028f5aa 100644 --- a/sound/core/pcm_dmaengine.c +++ b/sound/core/pcm_dmaengine.c @@ -48,6 +48,8 @@ EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_get_chan); * * This function can be used to initialize a dma_slave_config from a substream * and hw_params in a dmaengine based PCM driver implementation. + * + * Return: zero if successful, or a negative error code */ int snd_hwparams_to_dma_slave_config(const struct snd_pcm_substream *substream, const struct snd_pcm_hw_params *params, @@ -175,10 +177,10 @@ static int dmaengine_pcm_prepare_and_submit(struct snd_pcm_substream *substream) * @substream: PCM substream * @cmd: Trigger command * - * Returns 0 on success, a negative error code otherwise. - * * This function can be used as the PCM trigger callback for dmaengine based PCM * driver implementations. + * + * Return: 0 on success, a negative error code otherwise */ int snd_dmaengine_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { @@ -223,6 +225,8 @@ EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_trigger); * * This function is deprecated and should not be used by new drivers, as its * results may be unreliable. + * + * Return: PCM position in frames */ snd_pcm_uframes_t snd_dmaengine_pcm_pointer_no_residue(struct snd_pcm_substream *substream) { @@ -237,6 +241,8 @@ EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_pointer_no_residue); * * This function can be used as the PCM pointer callback for dmaengine based PCM * driver implementations. + * + * Return: PCM position in frames */ snd_pcm_uframes_t snd_dmaengine_pcm_pointer(struct snd_pcm_substream *substream) { @@ -266,9 +272,9 @@ EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_pointer); * @filter_fn: Filter function used to request the DMA channel * @filter_data: Data passed to the DMA filter function * - * Returns NULL or the requested DMA channel. - * * This function request a DMA channel for usage with dmaengine PCM. + * + * Return: NULL or the requested DMA channel */ struct dma_chan *snd_dmaengine_pcm_request_channel(dma_filter_fn filter_fn, void *filter_data) @@ -288,11 +294,11 @@ EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_request_channel); * @substream: PCM substream * @chan: DMA channel to use for data transfers * - * Returns 0 on success, a negative error code otherwise. - * * The function should usually be called from the pcm open callback. Note that * this function will use private_data field of the substream's runtime. So it * is not available to your pcm driver implementation. + * + * Return: 0 on success, a negative error code otherwise */ int snd_dmaengine_pcm_open(struct snd_pcm_substream *substream, struct dma_chan *chan) @@ -326,12 +332,12 @@ EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_open); * @filter_fn: Filter function used to request the DMA channel * @filter_data: Data passed to the DMA filter function * - * Returns 0 on success, a negative error code otherwise. - * * This function will request a DMA channel using the passed filter function and * data. The function should usually be called from the pcm open callback. Note * that this function will use private_data field of the substream's runtime. So * it is not available to your pcm driver implementation. + * + * Return: 0 on success, a negative error code otherwise */ int snd_dmaengine_pcm_open_request_chan(struct snd_pcm_substream *substream, dma_filter_fn filter_fn, void *filter_data) @@ -344,6 +350,8 @@ EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_open_request_chan); /** * snd_dmaengine_pcm_close - Close a dmaengine based PCM substream * @substream: PCM substream + * + * Return: 0 on success, a negative error code otherwise */ int snd_dmaengine_pcm_close(struct snd_pcm_substream *substream) { @@ -362,6 +370,8 @@ EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_close); * @substream: PCM substream * * Releases the DMA channel associated with the PCM substream. + * + * Return: zero if successful, or a negative error code */ int snd_dmaengine_pcm_close_release_chan(struct snd_pcm_substream *substream) { @@ -382,10 +392,10 @@ EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_close_release_chan); * @hw: PCM hw params * @chan: DMA channel to use for data transfers * - * Returns 0 on success, a negative error code otherwise. - * * This function will query DMA capability, then refine the pcm hardware * parameters. + * + * Return: 0 on success, a negative error code otherwise */ int snd_dmaengine_pcm_refine_runtime_hwparams( struct snd_pcm_substream *substream, -- cgit v1.2.3 From b05d834ef8f8fbd90b1bacca909c1eeec02e3625 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 13 Jul 2022 12:47:56 +0200 Subject: ALSA: compress: Fix kernel doc warnings Each kernel doc comment expects the definition of the return value and the summary for each struct / enum in a proper format. This patch adds or fixes the missing entries for compress-offload API. Link: https://lore.kernel.org/r/20220713104759.4365-5-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/compress_offload.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index cf415fe231ed..243acad89fd3 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -818,6 +818,8 @@ static void error_delayed_work(struct work_struct *work) * Stop the stream and set its state. * * Should be called with compressed device lock held. + * + * Return: zero if successful, or a negative error code */ int snd_compr_stop_error(struct snd_compr_stream *stream, snd_pcm_state_t state) @@ -1164,6 +1166,8 @@ static int snd_compress_dev_free(struct snd_device *device) * @dirn: device direction, should be of type enum snd_compr_direction * @id: ID string * @compr: compress device pointer + * + * Return: zero if successful, or a negative error code */ int snd_compress_new(struct snd_card *card, int device, int dirn, const char *id, struct snd_compr *compr) -- cgit v1.2.3 From e8406ebc37d2efb7e473e469152f977235a742e1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 13 Jul 2022 12:47:57 +0200 Subject: ALSA: control: Fix missing return value comments for kernel docs Each kernel doc comment expects the definition of the return value in proper format. This patch adds or fixes the missing entries for control API. Link: https://lore.kernel.org/r/20220713104759.4365-6-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/control.c | 16 ++++++++++++++-- 1 file changed, 14 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/core/control.c b/sound/core/control.c index fa04a9233155..4dba3a342458 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -2054,6 +2054,8 @@ static int _snd_ctl_register_ioctl(snd_kctl_ioctl_func_t fcn, struct list_head * * @fcn: ioctl callback function * * called from each device manager like pcm.c, hwdep.c, etc. + * + * Return: zero if successful, or a negative error code */ int snd_ctl_register_ioctl(snd_kctl_ioctl_func_t fcn) { @@ -2066,6 +2068,8 @@ EXPORT_SYMBOL(snd_ctl_register_ioctl); * snd_ctl_register_ioctl_compat - register the device-specific 32bit compat * control-ioctls * @fcn: ioctl callback function + * + * Return: zero if successful, or a negative error code */ int snd_ctl_register_ioctl_compat(snd_kctl_ioctl_func_t fcn) { @@ -2101,6 +2105,8 @@ static int _snd_ctl_unregister_ioctl(snd_kctl_ioctl_func_t fcn, /** * snd_ctl_unregister_ioctl - de-register the device-specific control-ioctls * @fcn: ioctl callback function to unregister + * + * Return: zero if successful, or a negative error code */ int snd_ctl_unregister_ioctl(snd_kctl_ioctl_func_t fcn) { @@ -2113,6 +2119,8 @@ EXPORT_SYMBOL(snd_ctl_unregister_ioctl); * snd_ctl_unregister_ioctl_compat - de-register the device-specific compat * 32bit control-ioctls * @fcn: ioctl callback function to unregister + * + * Return: zero if successful, or a negative error code */ int snd_ctl_unregister_ioctl_compat(snd_kctl_ioctl_func_t fcn) { @@ -2168,7 +2176,7 @@ EXPORT_SYMBOL_GPL(snd_ctl_get_preferred_subdevice); * snd_ctl_request_layer - request to use the layer * @module_name: Name of the kernel module (NULL == build-in) * - * Return an error code when the module cannot be loaded. + * Return: zero if successful, or an error code when the module cannot be loaded */ int snd_ctl_request_layer(const char *module_name) { @@ -2370,6 +2378,8 @@ int snd_ctl_create(struct snd_card *card) * * This is a function that can be used as info callback for a standard * boolean control with a single mono channel. + * + * Return: Zero (always successful) */ int snd_ctl_boolean_mono_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) @@ -2390,6 +2400,8 @@ EXPORT_SYMBOL(snd_ctl_boolean_mono_info); * * This is a function that can be used as info callback for a standard * boolean control with stereo two channels. + * + * Return: Zero (always successful) */ int snd_ctl_boolean_stereo_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) @@ -2413,7 +2425,7 @@ EXPORT_SYMBOL(snd_ctl_boolean_stereo_info); * If the control's accessibility is not the default (readable and writable), * the caller has to fill @info->access. * - * Return: Zero. + * Return: Zero (always successful) */ int snd_ctl_enum_info(struct snd_ctl_elem_info *info, unsigned int channels, unsigned int items, const char *const names[]) -- cgit v1.2.3 From 6eba99d4ce2487f2050b8787029cabfcfb748ee4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 13 Jul 2022 12:47:58 +0200 Subject: ALSA: memalloc: Fix missing return value comments for kernel docs Each kernel doc comment expects the definition of the return value in a proper format. This patch adds or fixes the missing entries for memory allocation helpers. Link: https://lore.kernel.org/r/20220713104759.4365-7-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/memalloc.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 8cfdaee77905..d3885cb02270 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -147,7 +147,7 @@ static void __snd_release_pages(struct device *dev, void *res) * hence it can't work with SNDRV_DMA_TYPE_CONTINUOUS or * SNDRV_DMA_TYPE_VMALLOC type. * - * The function returns the snd_dma_buffer object at success, or NULL if failed. + * Return: the snd_dma_buffer object at success, or NULL if failed */ struct snd_dma_buffer * snd_devm_alloc_dir_pages(struct device *dev, int type, @@ -179,6 +179,8 @@ EXPORT_SYMBOL_GPL(snd_devm_alloc_dir_pages); * snd_dma_buffer_mmap - perform mmap of the given DMA buffer * @dmab: buffer allocation information * @area: VM area information + * + * Return: zero if successful, or a negative error code */ int snd_dma_buffer_mmap(struct snd_dma_buffer *dmab, struct vm_area_struct *area) @@ -219,6 +221,8 @@ EXPORT_SYMBOL_GPL(snd_dma_buffer_sync); * snd_sgbuf_get_addr - return the physical address at the corresponding offset * @dmab: buffer allocation information * @offset: offset in the ring buffer + * + * Return: the physical address */ dma_addr_t snd_sgbuf_get_addr(struct snd_dma_buffer *dmab, size_t offset) { @@ -235,6 +239,8 @@ EXPORT_SYMBOL(snd_sgbuf_get_addr); * snd_sgbuf_get_page - return the physical page at the corresponding offset * @dmab: buffer allocation information * @offset: offset in the ring buffer + * + * Return: the page pointer */ struct page *snd_sgbuf_get_page(struct snd_dma_buffer *dmab, size_t offset) { @@ -253,6 +259,8 @@ EXPORT_SYMBOL(snd_sgbuf_get_page); * @dmab: buffer allocation information * @ofs: offset in the ring buffer * @size: the requested size + * + * Return: the chunk size */ unsigned int snd_sgbuf_get_chunk_size(struct snd_dma_buffer *dmab, unsigned int ofs, unsigned int size) -- cgit v1.2.3 From 281dee6707a869e0f40cac81b6b65c703192aa9d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 13 Jul 2022 12:47:59 +0200 Subject: ALSA: core: Fix missing return value comments for kernel docs Each kernel doc comment expects the definition of the return value in a proper format. This patch adds or fixes the missing entries for the remaining ALSA core API functions. Link: https://lore.kernel.org/r/20220713104759.4365-8-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/device.c | 2 ++ sound/core/info.c | 2 ++ sound/core/init.c | 10 ++++++++++ sound/core/isadma.c | 3 ++- sound/core/vmaster.c | 3 ++- 5 files changed, 18 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/core/device.c b/sound/core/device.c index bf0b04a7ee79..b57d80a17052 100644 --- a/sound/core/device.c +++ b/sound/core/device.c @@ -247,6 +247,8 @@ void snd_device_free_all(struct snd_card *card) * device, either @SNDRV_DEV_BUILD, @SNDRV_DEV_REGISTERED or * @SNDRV_DEV_DISCONNECTED is returned. * Or for a non-existing device, -1 is returned as an error. + * + * Return: the current state, or -1 if not found */ int snd_device_get_state(struct snd_card *card, void *device_data) { diff --git a/sound/core/info.c b/sound/core/info.c index 782fba87cc04..b8058b341178 100644 --- a/sound/core/info.c +++ b/sound/core/info.c @@ -868,6 +868,8 @@ EXPORT_SYMBOL(snd_info_register); * * This proc file entry will be registered via snd_card_register() call, and * it will be removed automatically at the card removal, too. + * + * Return: zero if successful, or a negative error code */ int snd_card_rw_proc_new(struct snd_card *card, const char *name, void *private_data, diff --git a/sound/core/init.c b/sound/core/init.c index 1870aee7b64f..3ac95c66a4b5 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -215,6 +215,8 @@ static void __snd_card_release(struct device *dev, void *data) * via snd_card_free() call in the error; otherwise it may lead to UAF due to * devres call orders. You can use snd_card_free_on_error() helper for * handling it more easily. + * + * Return: zero if successful, or a negative error code */ int snd_devm_card_new(struct device *parent, int idx, const char *xid, struct module *module, size_t extra_size, @@ -249,6 +251,8 @@ EXPORT_SYMBOL_GPL(snd_devm_card_new); * This function handles the explicit snd_card_free() call at the error from * the probe callback. It's just a small helper for simplifying the error * handling with the managed devices. + * + * Return: zero if successful, or a negative error code */ int snd_card_free_on_error(struct device *dev, int ret) { @@ -370,6 +374,8 @@ static int snd_card_init(struct snd_card *card, struct device *parent, * * Returns a card object corresponding to the given index or NULL if not found. * Release the object via snd_card_unref(). + * + * Return: a card object or NULL */ struct snd_card *snd_card_ref(int idx) { @@ -608,6 +614,8 @@ static int snd_card_do_free(struct snd_card *card) * resource immediately, but tries to disconnect at first. When the card * is still in use, the function returns before freeing the resources. * The card resources will be freed when the refcount gets to zero. + * + * Return: zero if successful, or a negative error code */ int snd_card_free_when_closed(struct snd_card *card) { @@ -833,6 +841,8 @@ static const struct attribute_group card_dev_attr_group = { * snd_card_add_dev_attr - Append a new sysfs attribute group to card * @card: card instance * @group: attribute group to append + * + * Return: zero if successful, or a negative error code */ int snd_card_add_dev_attr(struct snd_card *card, const struct attribute_group *group) diff --git a/sound/core/isadma.c b/sound/core/isadma.c index 1f45ede023b4..2602246bd5a0 100644 --- a/sound/core/isadma.c +++ b/sound/core/isadma.c @@ -116,8 +116,9 @@ static void __snd_release_dma(struct device *dev, void *data) * @dma: the dma number * @name: the name string of the requester * - * Returns zero on success, or a negative error code. * The requested DMA will be automatically released at unbinding via devres. + * + * Return: zero on success, or a negative error code */ int snd_devm_request_dma(struct device *dev, int dma, const char *name) { diff --git a/sound/core/vmaster.c b/sound/core/vmaster.c index ab36f9898711..d0f11f37889b 100644 --- a/sound/core/vmaster.c +++ b/sound/core/vmaster.c @@ -494,7 +494,8 @@ EXPORT_SYMBOL_GPL(snd_ctl_sync_vmaster); * @arg: optional function argument * * Apply the function @func to each follower kctl of the given vmaster kctl. - * Returns 0 if successful, or a negative error code. + * + * Return: 0 if successful, or a negative error code */ int snd_ctl_apply_vmaster_followers(struct snd_kcontrol *kctl, int (*func)(struct snd_kcontrol *vfollower, -- cgit v1.2.3 From 73acfba792b06156b7c805162fcd89fdb71646f9 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Tue, 12 Jul 2022 17:42:12 +0300 Subject: ASoC: amd: Fix error pointer dereference The "gpio_pa" pointer is an error pointer, there is no need to try put it. Calling gpiod_put() on it will lead to an error pointer dereference. Fixes: 02527c3f2300 ("ASoC: amd: add Machine driver for Jadeite platform") Signed-off-by: Dan Carpenter Link: https://lore.kernel.org/r/Ys2IRPHWGIwuVs21@kili Signed-off-by: Mark Brown --- sound/soc/amd/acp-es8336.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/amd/acp-es8336.c b/sound/soc/amd/acp-es8336.c index d501618b78f6..2fe8df86053a 100644 --- a/sound/soc/amd/acp-es8336.c +++ b/sound/soc/amd/acp-es8336.c @@ -212,7 +212,6 @@ static int st_es8336_late_probe(struct snd_soc_card *card) if (IS_ERR(gpio_pa)) { ret = dev_err_probe(card->dev, PTR_ERR(gpio_pa), "could not get pa-enable GPIO\n"); - gpiod_put(gpio_pa); put_device(codec_dev); return ret; } -- cgit v1.2.3 From eda26893dabfc6da7a1e1ff5f8628ed9faab3ab9 Mon Sep 17 00:00:00 2001 From: Liang He Date: Wed, 13 Jul 2022 15:12:00 +0800 Subject: ASoc: audio-graph-card2: Fix refcount leak bug in __graph_get_type() We should call of_node_put() for the reference before its replacement as it returned by of_get_parent() which has increased the refcount. Besides, we should also call of_node_put() before return. Fixes: c8c74939f791 ("ASoC: audio-graph-card2: add Multi CPU/Codec support") Signed-off-by: Liang He Link: https://lore.kernel.org/r/20220713071200.366729-1-windhl@126.com Signed-off-by: Mark Brown --- sound/soc/generic/audio-graph-card2.c | 35 +++++++++++++++++++++++++---------- 1 file changed, 25 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/generic/audio-graph-card2.c b/sound/soc/generic/audio-graph-card2.c index 19e31d53422a..18d053a0d05c 100644 --- a/sound/soc/generic/audio-graph-card2.c +++ b/sound/soc/generic/audio-graph-card2.c @@ -229,7 +229,8 @@ enum graph_type { static enum graph_type __graph_get_type(struct device_node *lnk) { - struct device_node *np; + struct device_node *np, *parent_np; + enum graph_type ret; /* * target { @@ -240,19 +241,33 @@ static enum graph_type __graph_get_type(struct device_node *lnk) * }; */ np = of_get_parent(lnk); - if (of_node_name_eq(np, "ports")) - np = of_get_parent(np); + if (of_node_name_eq(np, "ports")) { + parent_np = of_get_parent(np); + of_node_put(np); + np = parent_np; + } - if (of_node_name_eq(np, GRAPH_NODENAME_MULTI)) - return GRAPH_MULTI; + if (of_node_name_eq(np, GRAPH_NODENAME_MULTI)) { + ret = GRAPH_MULTI; + goto out_put; + } - if (of_node_name_eq(np, GRAPH_NODENAME_DPCM)) - return GRAPH_DPCM; + if (of_node_name_eq(np, GRAPH_NODENAME_DPCM)) { + ret = GRAPH_DPCM; + goto out_put; + } - if (of_node_name_eq(np, GRAPH_NODENAME_C2C)) - return GRAPH_C2C; + if (of_node_name_eq(np, GRAPH_NODENAME_C2C)) { + ret = GRAPH_C2C; + goto out_put; + } + + ret = GRAPH_NORMAL; + +out_put: + of_node_put(np); + return ret; - return GRAPH_NORMAL; } static enum graph_type graph_get_type(struct asoc_simple_priv *priv, -- cgit v1.2.3 From 1795c16a436057f95fef5b622d808885dd772d0e Mon Sep 17 00:00:00 2001 From: Randy Dunlap Date: Tue, 12 Jul 2022 11:33:48 -0700 Subject: ASoC: amd: fix Jadeite kconfig warning and build errors MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Since SND_SOC_ES8316 has a hard dependency on I2C and since 'select' does not follow any dependency chains, SND_SOC_AMD_ST_ES8336_MACH also needs to have a hard dependency on I2C. Fixes a kconfig warning and subsequent build errors: WARNING: unmet direct dependencies detected for SND_SOC_ES8316 Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && I2C [=n] Selected by [y]: - SND_SOC_AMD_ST_ES8336_MACH [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_AMD_ACP [=y] && ACPI [=y] && (I2C [=n] || COMPILE_TEST [=y]) sound/soc/codecs/es8316.c:866:1: warning: data definition has no type or storage class 866 | module_i2c_driver(es8316_i2c_driver); sound/soc/codecs/es8316.c:866:1: error: type defaults to ‘int’ in declaration of ‘module_i2c_driver’ [-Werror=implicit-int] sound/soc/codecs/es8316.c:866:1: warning: parameter names (without types) in function declaration sound/soc/codecs/es8316.c:857:26: warning: ‘es8316_i2c_driver’ defined but not used [-Wunused-variable] 857 | static struct i2c_driver es8316_i2c_driver = { Fixes: f94fa8405801 ("ASoC: amd: enable machine driver build for Jadeite platform") Signed-off-by: Randy Dunlap Cc: Vijendar Mukunda Cc: Mark Brown Cc: alsa-devel@alsa-project.org Cc: Jaroslav Kysela Cc: Takashi Iwai Link: https://lore.kernel.org/r/20220712183348.31046-1-rdunlap@infradead.org Signed-off-by: Mark Brown --- sound/soc/amd/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index 9629328c419e..9c2fef2ce89f 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -28,7 +28,7 @@ config SND_SOC_AMD_ST_ES8336_MACH select SND_SOC_ACPI if ACPI select SND_SOC_ES8316 depends on SND_SOC_AMD_ACP && ACPI - depends on I2C || COMPILE_TEST + depends on I2C help This option enables machine driver for Jadeite platform using es8336 codec. -- cgit v1.2.3 From a8d5df69e2ec702d979f7d04ed519caf8691a032 Mon Sep 17 00:00:00 2001 From: Liang He Date: Wed, 13 Jul 2022 18:20:13 +0800 Subject: ASoC: mt6359: Fix refcount leak bug In mt6359_parse_dt() and mt6359_accdet_parse_dt(), we should call of_node_put() for the reference returned by of_get_child_by_name() which has increased the refcount. Fixes: 683530285316 ("ASoC: mt6359: fix failed to parse DT properties") Fixes: eef07b9e0925 ("ASoC: mediatek: mt6359: add MT6359 accdet jack driver") Signed-off-by: Liang He Link: https://lore.kernel.org/r/20220713102013.367336-1-windhl@126.com Signed-off-by: Mark Brown --- sound/soc/codecs/mt6359-accdet.c | 1 + sound/soc/codecs/mt6359.c | 1 + 2 files changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/mt6359-accdet.c b/sound/soc/codecs/mt6359-accdet.c index 6d3d170144a0..c190628e2905 100644 --- a/sound/soc/codecs/mt6359-accdet.c +++ b/sound/soc/codecs/mt6359-accdet.c @@ -675,6 +675,7 @@ static int mt6359_accdet_parse_dt(struct mt6359_accdet *priv) sizeof(struct three_key_threshold)); } + of_node_put(node); dev_warn(priv->dev, "accdet caps=%x\n", priv->caps); return 0; diff --git a/sound/soc/codecs/mt6359.c b/sound/soc/codecs/mt6359.c index 23709b180409..c9a453ce8a2a 100644 --- a/sound/soc/codecs/mt6359.c +++ b/sound/soc/codecs/mt6359.c @@ -2778,6 +2778,7 @@ static int mt6359_parse_dt(struct mt6359_priv *priv) ret = of_property_read_u32(np, "mediatek,mic-type-2", &priv->mux_select[MUX_MIC_TYPE_2]); + of_node_put(np); if (ret) { dev_info(priv->dev, "%s() failed to read mic-type-2, use default (%d)\n", -- cgit v1.2.3 From 2a1be12c4d77d4f7b122568383382e006a60381b Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Fri, 8 Jul 2022 14:13:12 +0800 Subject: ASoC: SOF: Intel: add trigger callback into sdw_callback For IPC4, we need to set pipeline state in BE DAI trigger. Signed-off-by: Bard Liao Reviewed-by: Rander Wang Reviewed-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/20220708061312.25878-3-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-dai.c | 14 +++++++++++--- sound/soc/sof/intel/hda.c | 2 +- sound/soc/sof/intel/hda.h | 1 + 3 files changed, 13 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index c5b65e4a06be..556e883a32ed 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -708,8 +708,7 @@ static const struct snd_soc_dai_ops ipc3_ssp_dai_ops = { .shutdown = ssp_dai_shutdown, }; -static int ipc4_be_dai_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_dai *dai) +static int ipc4_be_dai_common_trigger(struct snd_soc_dai *dai, int cmd, int stream) { struct snd_sof_widget *pipe_widget; struct sof_ipc4_pipeline *pipeline; @@ -718,7 +717,7 @@ static int ipc4_be_dai_trigger(struct snd_pcm_substream *substream, struct snd_sof_dev *sdev; int ret; - w = snd_soc_dai_get_widget(dai, substream->stream); + w = snd_soc_dai_get_widget(dai, stream); swidget = w->dobj.private; pipe_widget = swidget->pipe_widget; pipeline = pipe_widget->private; @@ -753,6 +752,12 @@ static int ipc4_be_dai_trigger(struct snd_pcm_substream *substream, return 0; } +static int ipc4_be_dai_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + return ipc4_be_dai_common_trigger(dai, cmd, substream->stream); +} + static const struct snd_soc_dai_ops ipc4_dmic_dai_ops = { .trigger = ipc4_be_dai_trigger, }; @@ -804,6 +809,9 @@ void hda_set_dai_drv_ops(struct snd_sof_dev *sdev, struct snd_sof_dsp_ops *ops) if (!hda_use_tplg_nhlt) ipc4_data->nhlt = intel_nhlt_init(sdev->dev); + if (IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE)) + sdw_callback.trigger = ipc4_be_dai_common_trigger; + break; } default: diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index d519b9802b3b..b7fa95ea1090 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -147,7 +147,7 @@ static int sdw_free_stream(struct device *dev, return hda_ctrl_dai_widget_free(w, SOF_DAI_CONFIG_FLAGS_NONE, &data); } -static const struct sdw_intel_ops sdw_callback = { +struct sdw_intel_ops sdw_callback = { .params_stream = sdw_params_stream, .free_stream = sdw_free_stream, }; diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index d684908e9a53..5ef3e8775e36 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -842,5 +842,6 @@ irqreturn_t cnl_ipc4_irq_thread(int irq, void *context); int cnl_ipc4_send_msg(struct snd_sof_dev *sdev, struct snd_sof_ipc_msg *msg); irqreturn_t hda_dsp_ipc4_irq_thread(int irq, void *context); int hda_dsp_ipc4_send_msg(struct snd_sof_dev *sdev, struct snd_sof_ipc_msg *msg); +extern struct sdw_intel_ops sdw_callback; #endif -- cgit v1.2.3 From 89422df9548002adfffb73799cebe4909cfc8902 Mon Sep 17 00:00:00 2001 From: Uros Bizjak Date: Wed, 13 Jul 2022 17:19:46 +0200 Subject: ALSA: usb-audio: Use atomic_try_cmpxchg in ep_state_update Use atomic_try_cmpxchg instead of atomic_cmpxchg (*ptr, old, new) == old in ep_state_update. x86 CMPXCHG instruction returns success in ZF flag, so this change saves a compare after cmpxchg (and related move instruction in front of cmpxchg). No functional change intended. Signed-off-by: Uros Bizjak Link: https://lore.kernel.org/r/20220713151946.4743-1-ubizjak@gmail.com Signed-off-by: Takashi Iwai --- sound/usb/endpoint.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index f9c921683948..0d7b73bf7945 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -133,7 +133,7 @@ static inline bool ep_state_running(struct snd_usb_endpoint *ep) static inline bool ep_state_update(struct snd_usb_endpoint *ep, int old, int new) { - return atomic_cmpxchg(&ep->state, old, new) == old; + return atomic_try_cmpxchg(&ep->state, &old, new); } /** -- cgit v1.2.3 From 3233b978af23f11b4ad4f7f11a9a64bd05702b1f Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Thu, 30 Jun 2022 01:23:22 +0100 Subject: ALSA: hda: hda_cs_dsp_ctl: Add Library to support CS_DSP ALSA controls The cs35l41 part contains a DSP which is able to run firmware. The cs_dsp library can be used to control the DSP. These controls can be exposed to userspace using ALSA controls. This library adds apis to be able to interface between cs_dsp and hda drivers and expose the relevant controls as ALSA controls. [ Note: the dependency of CONFIG_SND_HDA_CS_DSP_CONTROLS Kconfig is corrected. Also, this Kconfig isn't enabled now but will be actually enabled in a later patch -- tiwai ] Signed-off-by: Stefan Binding Signed-off-by: Vitaly Rodionov Link: https://lore.kernel.org/r/20220630002335.366545-2-vitalyr@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/Kconfig | 4 + sound/pci/hda/Makefile | 2 + sound/pci/hda/hda_cs_dsp_ctl.c | 193 +++++++++++++++++++++++++++++++++++++++++ sound/pci/hda/hda_cs_dsp_ctl.h | 33 +++++++ 4 files changed, 232 insertions(+) create mode 100644 sound/pci/hda/hda_cs_dsp_ctl.c create mode 100644 sound/pci/hda/hda_cs_dsp_ctl.h (limited to 'sound') diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 8b73a12d356f..a17803953222 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -96,6 +96,10 @@ config SND_HDA_SCODEC_CS35L41 select SND_HDA_GENERIC select REGMAP_IRQ +config SND_HDA_CS_DSP_CONTROLS + tristate + select CS_DSP + config SND_HDA_SCODEC_CS35L41_I2C tristate "Build CS35L41 HD-audio side codec support for I2C Bus" depends on I2C diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile index 3e7bc608d45f..00d306104484 100644 --- a/sound/pci/hda/Makefile +++ b/sound/pci/hda/Makefile @@ -31,6 +31,7 @@ snd-hda-codec-hdmi-objs := patch_hdmi.o hda_eld.o snd-hda-scodec-cs35l41-objs := cs35l41_hda.o snd-hda-scodec-cs35l41-i2c-objs := cs35l41_hda_i2c.o snd-hda-scodec-cs35l41-spi-objs := cs35l41_hda_spi.o +snd-hda-cs-dsp-ctls-objs := hda_cs_dsp_ctl.o # common driver obj-$(CONFIG_SND_HDA) := snd-hda-codec.o @@ -54,6 +55,7 @@ obj-$(CONFIG_SND_HDA_CODEC_HDMI) += snd-hda-codec-hdmi.o obj-$(CONFIG_SND_HDA_SCODEC_CS35L41) += snd-hda-scodec-cs35l41.o obj-$(CONFIG_SND_HDA_SCODEC_CS35L41_I2C) += snd-hda-scodec-cs35l41-i2c.o obj-$(CONFIG_SND_HDA_SCODEC_CS35L41_SPI) += snd-hda-scodec-cs35l41-spi.o +obj-$(CONFIG_SND_HDA_CS_DSP_CONTROLS) += snd-hda-cs-dsp-ctls.o # this must be the last entry after codec drivers; # otherwise the codec patches won't be hooked before the PCI probe diff --git a/sound/pci/hda/hda_cs_dsp_ctl.c b/sound/pci/hda/hda_cs_dsp_ctl.c new file mode 100644 index 000000000000..74e2c5bd1b08 --- /dev/null +++ b/sound/pci/hda/hda_cs_dsp_ctl.c @@ -0,0 +1,193 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// HDA DSP ALSA Control Driver +// +// Copyright 2022 Cirrus Logic, Inc. +// +// Author: Stefan Binding + +#include +#include +#include +#include +#include "hda_cs_dsp_ctl.h" + +#define ADSP_MAX_STD_CTRL_SIZE 512 + +struct hda_cs_dsp_coeff_ctl { + struct cs_dsp_coeff_ctl *cs_ctl; + struct snd_card *card; + struct snd_kcontrol *kctl; +}; + +static const char * const hda_cs_dsp_fw_text[HDA_CS_DSP_NUM_FW] = { + [HDA_CS_DSP_FW_SPK_PROT] = "Prot", + [HDA_CS_DSP_FW_SPK_CALI] = "Cali", + [HDA_CS_DSP_FW_SPK_DIAG] = "Diag", + [HDA_CS_DSP_FW_MISC] = "Misc", +}; + +static int hda_cs_dsp_coeff_info(struct snd_kcontrol *kctl, struct snd_ctl_elem_info *uinfo) +{ + struct hda_cs_dsp_coeff_ctl *ctl = (struct hda_cs_dsp_coeff_ctl *)snd_kcontrol_chip(kctl); + struct cs_dsp_coeff_ctl *cs_ctl = ctl->cs_ctl; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; + uinfo->count = cs_ctl->len; + + return 0; +} + +static int hda_cs_dsp_coeff_put(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) +{ + struct hda_cs_dsp_coeff_ctl *ctl = (struct hda_cs_dsp_coeff_ctl *)snd_kcontrol_chip(kctl); + struct cs_dsp_coeff_ctl *cs_ctl = ctl->cs_ctl; + char *p = ucontrol->value.bytes.data; + int ret = 0; + + mutex_lock(&cs_ctl->dsp->pwr_lock); + ret = cs_dsp_coeff_write_ctrl(cs_ctl, 0, p, cs_ctl->len); + mutex_unlock(&cs_ctl->dsp->pwr_lock); + + return ret; +} + +static int hda_cs_dsp_coeff_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) +{ + struct hda_cs_dsp_coeff_ctl *ctl = (struct hda_cs_dsp_coeff_ctl *)snd_kcontrol_chip(kctl); + struct cs_dsp_coeff_ctl *cs_ctl = ctl->cs_ctl; + char *p = ucontrol->value.bytes.data; + int ret; + + mutex_lock(&cs_ctl->dsp->pwr_lock); + ret = cs_dsp_coeff_read_ctrl(cs_ctl, 0, p, cs_ctl->len); + mutex_unlock(&cs_ctl->dsp->pwr_lock); + + return ret; +} + +static unsigned int wmfw_convert_flags(unsigned int in) +{ + unsigned int out, rd, wr, vol; + + rd = SNDRV_CTL_ELEM_ACCESS_READ; + wr = SNDRV_CTL_ELEM_ACCESS_WRITE; + vol = SNDRV_CTL_ELEM_ACCESS_VOLATILE; + + out = 0; + + if (in) { + out |= rd; + if (in & WMFW_CTL_FLAG_WRITEABLE) + out |= wr; + if (in & WMFW_CTL_FLAG_VOLATILE) + out |= vol; + } else { + out |= rd | wr | vol; + } + + return out; +} + +static int hda_cs_dsp_add_kcontrol(struct hda_cs_dsp_coeff_ctl *ctl, const char *name) +{ + struct cs_dsp_coeff_ctl *cs_ctl = ctl->cs_ctl; + struct snd_kcontrol_new kcontrol = {0}; + struct snd_kcontrol *kctl; + int ret = 0; + + if (cs_ctl->len > ADSP_MAX_STD_CTRL_SIZE) { + dev_err(cs_ctl->dsp->dev, "KControl %s: length %zu exceeds maximum %d\n", name, + cs_ctl->len, ADSP_MAX_STD_CTRL_SIZE); + return -EINVAL; + } + + kcontrol.name = name; + kcontrol.info = hda_cs_dsp_coeff_info; + kcontrol.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + kcontrol.access = wmfw_convert_flags(cs_ctl->flags); + kcontrol.get = hda_cs_dsp_coeff_get; + kcontrol.put = hda_cs_dsp_coeff_put; + + /* Save ctl inside private_data, ctl is owned by cs_dsp, + * and will be freed when cs_dsp removes the control */ + kctl = snd_ctl_new1(&kcontrol, (void *)ctl); + if (!kctl) { + ret = -ENOMEM; + return ret; + } + + ret = snd_ctl_add(ctl->card, kctl); + if (ret) { + dev_err(cs_ctl->dsp->dev, "Failed to add KControl %s = %d\n", kcontrol.name, ret); + return ret; + } + + dev_dbg(cs_ctl->dsp->dev, "Added KControl: %s\n", kcontrol.name); + ctl->kctl = kctl; + + return 0; +} + +int hda_cs_dsp_control_add(struct cs_dsp_coeff_ctl *cs_ctl, struct hda_cs_dsp_ctl_info *info) +{ + struct cs_dsp *cs_dsp = cs_ctl->dsp; + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + struct hda_cs_dsp_coeff_ctl *ctl; + const char *region_name; + int ret; + + if (cs_ctl->flags & WMFW_CTL_FLAG_SYS) + return 0; + + region_name = cs_dsp_mem_region_name(cs_ctl->alg_region.type); + if (!region_name) { + dev_err(cs_dsp->dev, "Unknown region type: %d\n", cs_ctl->alg_region.type); + return -EINVAL; + } + + ret = scnprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "%s %s %.12s %x", info->device_name, + cs_dsp->name, hda_cs_dsp_fw_text[info->fw_type], cs_ctl->alg_region.alg); + + if (cs_ctl->subname) { + int avail = SNDRV_CTL_ELEM_ID_NAME_MAXLEN - ret - 2; + int skip = 0; + + /* Truncate the subname from the start if it is too long */ + if (cs_ctl->subname_len > avail) + skip = cs_ctl->subname_len - avail; + + snprintf(name + ret, SNDRV_CTL_ELEM_ID_NAME_MAXLEN - ret, + " %.*s", cs_ctl->subname_len - skip, cs_ctl->subname + skip); + } + + ctl = kzalloc(sizeof(*ctl), GFP_KERNEL); + if (!ctl) + return -ENOMEM; + + ctl->cs_ctl = cs_ctl; + ctl->card = info->card; + cs_ctl->priv = ctl; + + ret = hda_cs_dsp_add_kcontrol(ctl, name); + if (ret) { + dev_err(cs_dsp->dev, "Error (%d) adding control %s\n", ret, name); + kfree(ctl); + return ret; + } + + return 0; +} +EXPORT_SYMBOL_NS_GPL(hda_cs_dsp_control_add, SND_HDA_CS_DSP_CONTROLS); + +void hda_cs_dsp_control_remove(struct cs_dsp_coeff_ctl *cs_ctl) +{ + struct hda_cs_dsp_coeff_ctl *ctl = cs_ctl->priv; + + kfree(ctl); +} +EXPORT_SYMBOL_NS_GPL(hda_cs_dsp_control_remove, SND_HDA_CS_DSP_CONTROLS); + +MODULE_DESCRIPTION("CS_DSP ALSA Control HDA Library"); +MODULE_AUTHOR("Stefan Binding, "); +MODULE_LICENSE("GPL"); diff --git a/sound/pci/hda/hda_cs_dsp_ctl.h b/sound/pci/hda/hda_cs_dsp_ctl.h new file mode 100644 index 000000000000..1c6d0fc9a2cc --- /dev/null +++ b/sound/pci/hda/hda_cs_dsp_ctl.h @@ -0,0 +1,33 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * HDA DSP ALSA Control Driver + * + * Copyright 2022 Cirrus Logic, Inc. + * + * Author: Stefan Binding + */ + +#ifndef __HDA_CS_DSP_CTL_H__ +#define __HDA_CS_DSP_CTL_H__ + +#include +#include + +enum hda_cs_dsp_fw_id { + HDA_CS_DSP_FW_SPK_PROT, + HDA_CS_DSP_FW_SPK_CALI, + HDA_CS_DSP_FW_SPK_DIAG, + HDA_CS_DSP_FW_MISC, + HDA_CS_DSP_NUM_FW +}; + +struct hda_cs_dsp_ctl_info { + struct snd_card *card; + enum hda_cs_dsp_fw_id fw_type; + const char *device_name; +}; + +int hda_cs_dsp_control_add(struct cs_dsp_coeff_ctl *cs_ctl, struct hda_cs_dsp_ctl_info *info); +void hda_cs_dsp_control_remove(struct cs_dsp_coeff_ctl *cs_ctl); + +#endif /*__HDA_CS_DSP_CTL_H__*/ -- cgit v1.2.3 From e414b05e724f5fbae6e86d074d7668287a603b24 Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Thu, 30 Jun 2022 01:23:23 +0100 Subject: ALSA: hda: hda_cs_dsp_ctl: Add apis to write the controls directly DSP controls are exposed as ALSA controls, however, some of these controls are required to be accessed by the driver. Add apis which allow read/write of these controls. The write api will also notify the ALSA control on value change. Signed-off-by: Stefan Binding Signed-off-by: Vitaly Rodionov Link: https://lore.kernel.org/r/20220630002335.366545-3-vitalyr@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_cs_dsp_ctl.c | 39 +++++++++++++++++++++++++++++++++++++++ sound/pci/hda/hda_cs_dsp_ctl.h | 4 ++++ 2 files changed, 43 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_cs_dsp_ctl.c b/sound/pci/hda/hda_cs_dsp_ctl.c index 74e2c5bd1b08..2351476c9ee6 100644 --- a/sound/pci/hda/hda_cs_dsp_ctl.c +++ b/sound/pci/hda/hda_cs_dsp_ctl.c @@ -188,6 +188,45 @@ void hda_cs_dsp_control_remove(struct cs_dsp_coeff_ctl *cs_ctl) } EXPORT_SYMBOL_NS_GPL(hda_cs_dsp_control_remove, SND_HDA_CS_DSP_CONTROLS); +int hda_cs_dsp_write_ctl(struct cs_dsp *dsp, const char *name, int type, + unsigned int alg, const void *buf, size_t len) +{ + struct cs_dsp_coeff_ctl *cs_ctl; + struct hda_cs_dsp_coeff_ctl *ctl; + int ret; + + cs_ctl = cs_dsp_get_ctl(dsp, name, type, alg); + if (!cs_ctl) + return -EINVAL; + + ctl = cs_ctl->priv; + + ret = cs_dsp_coeff_write_ctrl(cs_ctl, 0, buf, len); + if (ret) + return ret; + + if (cs_ctl->flags & WMFW_CTL_FLAG_SYS) + return 0; + + snd_ctl_notify(ctl->card, SNDRV_CTL_EVENT_MASK_VALUE, &ctl->kctl->id); + + return 0; +} +EXPORT_SYMBOL_NS_GPL(hda_cs_dsp_write_ctl, SND_HDA_CS_DSP_CONTROLS); + +int hda_cs_dsp_read_ctl(struct cs_dsp *dsp, const char *name, int type, + unsigned int alg, void *buf, size_t len) +{ + struct cs_dsp_coeff_ctl *cs_ctl; + + cs_ctl = cs_dsp_get_ctl(dsp, name, type, alg); + if (!cs_ctl) + return -EINVAL; + + return cs_dsp_coeff_read_ctrl(cs_ctl, 0, buf, len); +} +EXPORT_SYMBOL_NS_GPL(hda_cs_dsp_read_ctl, SND_HDA_CS_DSP_CONTROLS); + MODULE_DESCRIPTION("CS_DSP ALSA Control HDA Library"); MODULE_AUTHOR("Stefan Binding, "); MODULE_LICENSE("GPL"); diff --git a/sound/pci/hda/hda_cs_dsp_ctl.h b/sound/pci/hda/hda_cs_dsp_ctl.h index 1c6d0fc9a2cc..c65bfd6878fd 100644 --- a/sound/pci/hda/hda_cs_dsp_ctl.h +++ b/sound/pci/hda/hda_cs_dsp_ctl.h @@ -29,5 +29,9 @@ struct hda_cs_dsp_ctl_info { int hda_cs_dsp_control_add(struct cs_dsp_coeff_ctl *cs_ctl, struct hda_cs_dsp_ctl_info *info); void hda_cs_dsp_control_remove(struct cs_dsp_coeff_ctl *cs_ctl); +int hda_cs_dsp_write_ctl(struct cs_dsp *dsp, const char *name, int type, + unsigned int alg, const void *buf, size_t len); +int hda_cs_dsp_read_ctl(struct cs_dsp *dsp, const char *name, int type, + unsigned int alg, void *buf, size_t len); #endif /*__HDA_CS_DSP_CTL_H__*/ -- cgit v1.2.3 From 22d5cbd273a2ca90ba026ec82f0b9c3e984b0c1c Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Thu, 30 Jun 2022 01:23:24 +0100 Subject: ALSA: hda: cs35l41: Save codec object inside component struct This is required for ALSA control support. Signed-off-by: Stefan Binding Signed-off-by: Vitaly Rodionov Link: https://lore.kernel.org/r/20220630002335.366545-4-vitalyr@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l41_hda.c | 1 + sound/pci/hda/cs35l41_hda.h | 1 + sound/pci/hda/hda_component.h | 1 + sound/pci/hda/patch_realtek.c | 1 + 4 files changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index 1a1afa0725e0..20d3ce8773dd 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -117,6 +117,7 @@ static int cs35l41_hda_bind(struct device *dev, struct device *master, void *mas return -EBUSY; comps->dev = dev; + cs35l41->codec = comps->codec; strscpy(comps->name, dev_name(dev), sizeof(comps->name)); comps->playback_hook = cs35l41_hda_playback_hook; diff --git a/sound/pci/hda/cs35l41_hda.h b/sound/pci/hda/cs35l41_hda.h index a52ffd1f7999..aaf9e16684c2 100644 --- a/sound/pci/hda/cs35l41_hda.h +++ b/sound/pci/hda/cs35l41_hda.h @@ -32,6 +32,7 @@ struct cs35l41_hda { struct regmap *regmap; struct gpio_desc *reset_gpio; struct cs35l41_hw_cfg hw_cfg; + struct hda_codec *codec; int irq; int index; diff --git a/sound/pci/hda/hda_component.h b/sound/pci/hda/hda_component.h index e26c896a13f3..534e845b9cd1 100644 --- a/sound/pci/hda/hda_component.h +++ b/sound/pci/hda/hda_component.h @@ -14,5 +14,6 @@ struct hda_component { struct device *dev; char name[HDA_MAX_NAME_SIZE]; + struct hda_codec *codec; void (*playback_hook)(struct device *dev, int action); }; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 007dd8b5e1f2..44744d568404 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6654,6 +6654,7 @@ static void cs35l41_generic_fixup(struct hda_codec *cdc, int action, const char "%s-%s:00-cs35l41-hda.%d", bus, hid, i); if (!name) return; + spec->comps[i].codec = cdc; component_match_add(dev, &spec->match, component_compare_dev_name, name); } ret = component_master_add_with_match(dev, &comp_master_ops, spec->match); -- cgit v1.2.3 From 2e81e1fffd53ba108481f2f14388b628884efe61 Mon Sep 17 00:00:00 2001 From: Vitaly Rodionov Date: Thu, 30 Jun 2022 01:23:25 +0100 Subject: ALSA: hda: cs35l41: Add initial DSP support and firmware loading This patch adds support for the CS35L41 DSP. The DSP allows for extra features, such as running speaker protection algorithms and hibernations. To utilize these features, the driver must load firmware into the DSP, as well as various tuning files which allow for customization for specific models. [ Slightly simplified Kconfig changes by tiwai ] Signed-off-by: Vitaly Rodionov Signed-off-by: Vitaly Rodionov Link: https://lore.kernel.org/r/20220630002335.366545-5-vitalyr@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/Kconfig | 2 + sound/pci/hda/cs35l41_hda.c | 251 +++++++++++++++++++++++++++++++++++++++++++- sound/pci/hda/cs35l41_hda.h | 13 +++ 3 files changed, 265 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index a17803953222..44c33bc0740e 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -107,6 +107,7 @@ config SND_HDA_SCODEC_CS35L41_I2C depends on SND_SOC select SND_SOC_CS35L41_LIB select SND_HDA_SCODEC_CS35L41 + select SND_HDA_CS_DSP_CONTROLS help Say Y or M here to include CS35L41 I2C HD-audio side codec support in snd-hda-intel driver, such as ALC287. @@ -121,6 +122,7 @@ config SND_HDA_SCODEC_CS35L41_SPI depends on SND_SOC select SND_SOC_CS35L41_LIB select SND_HDA_SCODEC_CS35L41 + select SND_HDA_CS_DSP_CONTROLS help Say Y or M here to include CS35L41 SPI HD-audio side codec support in snd-hda-intel driver, such as ALC287. diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index 20d3ce8773dd..d68d951c434e 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -9,12 +9,22 @@ #include #include #include +#include #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_jack.h" #include "hda_generic.h" #include "hda_component.h" #include "cs35l41_hda.h" +#include "hda_cs_dsp_ctl.h" + +#define CS35L41_FIRMWARE_ROOT "cirrus/" +#define CS35L41_PART "cs35l41" +#define FW_NAME "CSPL" + +#define HALO_STATE_DSP_CTL_NAME "HALO_STATE" +#define HALO_STATE_DSP_CTL_TYPE 5 +#define HALO_STATE_DSP_CTL_ALG 262308 static const struct reg_sequence cs35l41_hda_config[] = { { CS35L41_PLL_CLK_CTRL, 0x00000430 }, // 3072000Hz, BCLK Input, PLL_REFCLK_EN = 1 @@ -27,11 +37,172 @@ static const struct reg_sequence cs35l41_hda_config[] = { { CS35L41_AMP_GAIN_CTRL, 0x00000084 }, // AMP_GAIN_PCM 4.5 dB }; +static const struct reg_sequence cs35l41_hda_config_dsp[] = { + { CS35L41_PLL_CLK_CTRL, 0x00000430 }, // 3072000Hz, BCLK Input, PLL_REFCLK_EN = 1 + { CS35L41_DSP_CLK_CTRL, 0x00000003 }, // DSP CLK EN + { CS35L41_GLOBAL_CLK_CTRL, 0x00000003 }, // GLOBAL_FS = 48 kHz + { CS35L41_SP_ENABLES, 0x00010001 }, // ASP_RX1_EN = 1, ASP_TX1_EN = 1 + { CS35L41_SP_RATE_CTRL, 0x00000021 }, // ASP_BCLK_FREQ = 3.072 MHz + { CS35L41_SP_FORMAT, 0x20200200 }, // 32 bits RX/TX slots, I2S, clk consumer + { CS35L41_SP_HIZ_CTRL, 0x00000003 }, // Hi-Z unused/disabled + { CS35L41_SP_TX_WL, 0x00000018 }, // 24 cycles/slot + { CS35L41_SP_RX_WL, 0x00000018 }, // 24 cycles/slot + { CS35L41_DAC_PCM1_SRC, 0x00000032 }, // DACPCM1_SRC = ERR_VOL + { CS35L41_ASP_TX1_SRC, 0x00000018 }, // ASPTX1 SRC = VMON + { CS35L41_ASP_TX2_SRC, 0x00000019 }, // ASPTX2 SRC = IMON + { CS35L41_ASP_TX3_SRC, 0x00000028 }, // ASPTX3 SRC = VPMON + { CS35L41_ASP_TX4_SRC, 0x00000029 }, // ASPTX4 SRC = VBSTMON + { CS35L41_DSP1_RX1_SRC, 0x00000008 }, // DSP1RX1 SRC = ASPRX1 + { CS35L41_DSP1_RX2_SRC, 0x00000008 }, // DSP1RX2 SRC = ASPRX1 + { CS35L41_DSP1_RX3_SRC, 0x00000018 }, // DSP1RX3 SRC = VMON + { CS35L41_DSP1_RX4_SRC, 0x00000019 }, // DSP1RX4 SRC = IMON + { CS35L41_DSP1_RX5_SRC, 0x00000029 }, // DSP1RX5 SRC = VBSTMON + { CS35L41_AMP_DIG_VOL_CTRL, 0x00000000 }, // AMP_VOL_PCM 0.0 dB + { CS35L41_AMP_GAIN_CTRL, 0x00000233 }, // AMP_GAIN_PCM = 17.5dB AMP_GAIN_PDM = 19.5dB +}; + static const struct reg_sequence cs35l41_hda_mute[] = { { CS35L41_AMP_GAIN_CTRL, 0x00000000 }, // AMP_GAIN_PCM 0.5 dB { CS35L41_AMP_DIG_VOL_CTRL, 0x0000A678 }, // AMP_VOL_PCM Mute }; +static int cs35l41_control_add(struct cs_dsp_coeff_ctl *cs_ctl) +{ + struct cs35l41_hda *cs35l41 = container_of(cs_ctl->dsp, struct cs35l41_hda, cs_dsp); + struct hda_cs_dsp_ctl_info info; + + info.device_name = cs35l41->amp_name; + info.fw_type = HDA_CS_DSP_FW_SPK_PROT; + info.card = cs35l41->codec->card; + + return hda_cs_dsp_control_add(cs_ctl, &info); +} + +static const struct cs_dsp_client_ops client_ops = { + .control_add = cs35l41_control_add, + .control_remove = hda_cs_dsp_control_remove, +}; + +static int cs35l41_request_firmware_file(struct cs35l41_hda *cs35l41, + const struct firmware **firmware, char **filename, + const char *dir, const char *filetype) +{ + const char * const dsp_name = cs35l41->cs_dsp.name; + char *s, c; + int ret = 0; + + *filename = kasprintf(GFP_KERNEL, "%s%s-%s-%s.%s", dir, CS35L41_PART, dsp_name, "spk-prot", + filetype); + + if (*filename == NULL) + return -ENOMEM; + + /* + * Make sure that filename is lower-case and any non alpha-numeric + * characters except full stop and '/' are replaced with hyphens. + */ + s = *filename; + while (*s) { + c = *s; + if (isalnum(c)) + *s = tolower(c); + else if (c != '.' && c != '/') + *s = '-'; + s++; + } + + ret = firmware_request_nowarn(firmware, *filename, cs35l41->dev); + if (ret != 0) { + dev_dbg(cs35l41->dev, "Failed to request '%s'\n", *filename); + kfree(*filename); + *filename = NULL; + } + + return ret; +} + +static int cs35l41_request_firmware_files(struct cs35l41_hda *cs35l41, + const struct firmware **wmfw_firmware, + char **wmfw_filename, + const struct firmware **coeff_firmware, + char **coeff_filename) +{ + int ret; + + /* cirrus/part-dspN-fwtype.wmfw */ + ret = cs35l41_request_firmware_file(cs35l41, wmfw_firmware, wmfw_filename, + CS35L41_FIRMWARE_ROOT, "wmfw"); + if (!ret) { + cs35l41_request_firmware_file(cs35l41, coeff_firmware, coeff_filename, + CS35L41_FIRMWARE_ROOT, "bin"); + return 0; + } + + dev_warn(cs35l41->dev, "Failed to request firmware\n"); + + return ret; +} + +static int cs35l41_init_dsp(struct cs35l41_hda *cs35l41) +{ + const struct firmware *coeff_firmware = NULL; + const struct firmware *wmfw_firmware = NULL; + struct cs_dsp *dsp = &cs35l41->cs_dsp; + char *coeff_filename = NULL; + char *wmfw_filename = NULL; + int ret; + + if (!cs35l41->halo_initialized) { + cs35l41_configure_cs_dsp(cs35l41->dev, cs35l41->regmap, dsp); + dsp->client_ops = &client_ops; + + ret = cs_dsp_halo_init(&cs35l41->cs_dsp); + if (ret) + return ret; + cs35l41->halo_initialized = true; + } + + ret = cs35l41_request_firmware_files(cs35l41, &wmfw_firmware, &wmfw_filename, + &coeff_firmware, &coeff_filename); + if (ret < 0) + return ret; + + dev_dbg(cs35l41->dev, "Loading WMFW Firmware: %s\n", wmfw_filename); + if (coeff_filename) + dev_dbg(cs35l41->dev, "Loading Coefficient File: %s\n", coeff_filename); + else + dev_warn(cs35l41->dev, "No Coefficient File available.\n"); + + ret = cs_dsp_power_up(dsp, wmfw_firmware, wmfw_filename, coeff_firmware, coeff_filename, + FW_NAME); + + release_firmware(wmfw_firmware); + release_firmware(coeff_firmware); + kfree(wmfw_filename); + kfree(coeff_filename); + + return ret; +} + +static void cs35l41_shutdown_dsp(struct cs35l41_hda *cs35l41) +{ + struct cs_dsp *dsp = &cs35l41->cs_dsp; + + cs_dsp_stop(dsp); + cs_dsp_power_down(dsp); + cs35l41->firmware_running = false; + dev_dbg(cs35l41->dev, "Unloaded Firmware\n"); +} + +static void cs35l41_remove_dsp(struct cs35l41_hda *cs35l41) +{ + struct cs_dsp *dsp = &cs35l41->cs_dsp; + + cs35l41_shutdown_dsp(cs35l41); + cs_dsp_remove(dsp); + cs35l41->halo_initialized = false; +} + /* Protection release cycle to get the speaker out of Safe-Mode */ static void cs35l41_error_release(struct device *dev, struct regmap *regmap, unsigned int mask) { @@ -53,9 +224,22 @@ static void cs35l41_hda_playback_hook(struct device *dev, int action) struct regmap *reg = cs35l41->regmap; int ret = 0; + mutex_lock(&cs35l41->fw_mutex); + switch (action) { case HDA_GEN_PCM_ACT_OPEN: - regmap_multi_reg_write(reg, cs35l41_hda_config, ARRAY_SIZE(cs35l41_hda_config)); + if (cs35l41->firmware_running) { + regmap_multi_reg_write(reg, cs35l41_hda_config_dsp, + ARRAY_SIZE(cs35l41_hda_config_dsp)); + regmap_update_bits(cs35l41->regmap, CS35L41_PWR_CTRL2, + CS35L41_VMON_EN_MASK | CS35L41_IMON_EN_MASK, + 1 << CS35L41_VMON_EN_SHIFT | 1 << CS35L41_IMON_EN_SHIFT); + cs35l41_set_cspl_mbox_cmd(cs35l41->dev, cs35l41->regmap, + CSPL_MBOX_CMD_RESUME); + } else { + regmap_multi_reg_write(reg, cs35l41_hda_config, + ARRAY_SIZE(cs35l41_hda_config)); + } ret = regmap_update_bits(reg, CS35L41_PWR_CTRL2, CS35L41_AMP_EN_MASK, 1 << CS35L41_AMP_EN_SHIFT); if (cs35l41->hw_cfg.bst_type == CS35L41_EXT_BOOST) @@ -73,6 +257,13 @@ static void cs35l41_hda_playback_hook(struct device *dev, int action) CS35L41_AMP_EN_MASK, 0 << CS35L41_AMP_EN_SHIFT); if (cs35l41->hw_cfg.bst_type == CS35L41_EXT_BOOST) regmap_write(reg, CS35L41_GPIO1_CTRL1, 0x00000001); + if (cs35l41->firmware_running) { + cs35l41_set_cspl_mbox_cmd(cs35l41->dev, cs35l41->regmap, + CSPL_MBOX_CMD_PAUSE); + regmap_update_bits(cs35l41->regmap, CS35L41_PWR_CTRL2, + CS35L41_VMON_EN_MASK | CS35L41_IMON_EN_MASK, + 0 << CS35L41_VMON_EN_SHIFT | 0 << CS35L41_IMON_EN_SHIFT); + } cs35l41_irq_release(cs35l41); break; default: @@ -80,6 +271,8 @@ static void cs35l41_hda_playback_hook(struct device *dev, int action) break; } + mutex_unlock(&cs35l41->fw_mutex); + if (ret) dev_err(cs35l41->dev, "Regmap access fail: %d\n", ret); } @@ -104,6 +297,51 @@ static int cs35l41_hda_channel_map(struct device *dev, unsigned int tx_num, unsi rx_slot); } +static int cs35l41_smart_amp(struct cs35l41_hda *cs35l41) +{ + int halo_sts; + int ret; + + ret = cs35l41_init_dsp(cs35l41); + if (ret) { + dev_warn(cs35l41->dev, "Cannot Initialize Firmware. Error: %d\n", ret); + goto clean_dsp; + } + + ret = cs35l41_write_fs_errata(cs35l41->dev, cs35l41->regmap); + if (ret) { + dev_err(cs35l41->dev, "Cannot Write FS Errata: %d\n", ret); + goto clean_dsp; + } + + ret = cs_dsp_run(&cs35l41->cs_dsp); + if (ret) { + dev_err(cs35l41->dev, "Fail to start dsp: %d\n", ret); + goto clean_dsp; + } + + ret = read_poll_timeout(hda_cs_dsp_read_ctl, ret, + be32_to_cpu(halo_sts) == HALO_STATE_CODE_RUN, + 1000, 15000, false, &cs35l41->cs_dsp, HALO_STATE_DSP_CTL_NAME, + HALO_STATE_DSP_CTL_TYPE, HALO_STATE_DSP_CTL_ALG, + &halo_sts, sizeof(halo_sts)); + + if (ret) { + dev_err(cs35l41->dev, "Timeout waiting for HALO Core to start. State: %d\n", + halo_sts); + goto clean_dsp; + } + + cs35l41_set_cspl_mbox_cmd(cs35l41->dev, cs35l41->regmap, CSPL_MBOX_CMD_PAUSE); + cs35l41->firmware_running = true; + + return 0; + +clean_dsp: + cs35l41_shutdown_dsp(cs35l41); + return ret; +} + static int cs35l41_hda_bind(struct device *dev, struct device *master, void *master_data) { struct cs35l41_hda *cs35l41 = dev_get_drvdata(dev); @@ -121,6 +359,11 @@ static int cs35l41_hda_bind(struct device *dev, struct device *master, void *mas strscpy(comps->name, dev_name(dev), sizeof(comps->name)); comps->playback_hook = cs35l41_hda_playback_hook; + mutex_lock(&cs35l41->fw_mutex); + if (cs35l41_smart_amp(cs35l41) < 0) + dev_warn(cs35l41->dev, "Cannot Run Firmware, reverting to dsp bypass...\n"); + mutex_unlock(&cs35l41->fw_mutex); + return 0; } @@ -535,6 +778,8 @@ int cs35l41_hda_probe(struct device *dev, const char *device_name, int id, int i if (ret) goto err; + mutex_init(&cs35l41->fw_mutex); + ret = cs35l41_hda_apply_properties(cs35l41); if (ret) goto err; @@ -562,6 +807,9 @@ void cs35l41_hda_remove(struct device *dev) { struct cs35l41_hda *cs35l41 = dev_get_drvdata(dev); + if (cs35l41->halo_initialized) + cs35l41_remove_dsp(cs35l41); + component_del(cs35l41->dev, &cs35l41_hda_comp_ops); if (cs35l41_safe_reset(cs35l41->regmap, cs35l41->hw_cfg.bst_type)) @@ -571,5 +819,6 @@ void cs35l41_hda_remove(struct device *dev) EXPORT_SYMBOL_NS_GPL(cs35l41_hda_remove, SND_HDA_SCODEC_CS35L41); MODULE_DESCRIPTION("CS35L41 HDA Driver"); +MODULE_IMPORT_NS(SND_HDA_CS_DSP_CONTROLS); MODULE_AUTHOR("Lucas Tanure, Cirrus Logic Inc, "); MODULE_LICENSE("GPL"); diff --git a/sound/pci/hda/cs35l41_hda.h b/sound/pci/hda/cs35l41_hda.h index aaf9e16684c2..5814af050944 100644 --- a/sound/pci/hda/cs35l41_hda.h +++ b/sound/pci/hda/cs35l41_hda.h @@ -15,6 +15,9 @@ #include #include +#include +#include + enum cs35l41_hda_spk_pos { CS35l41_LEFT, CS35l41_RIGHT, @@ -39,7 +42,17 @@ struct cs35l41_hda { int channel_index; unsigned volatile long irq_errors; const char *amp_name; + struct mutex fw_mutex; struct regmap_irq_chip_data *irq_data; + bool firmware_running; + bool halo_initialized; + struct cs_dsp cs_dsp; +}; + +enum halo_state { + HALO_STATE_CODE_INIT_DOWNLOAD = 0, + HALO_STATE_CODE_START, + HALO_STATE_CODE_RUN }; int cs35l41_hda_probe(struct device *dev, const char *device_name, int id, int irq, -- cgit v1.2.3 From e99f3c7e3250dd895d2da506d0d910d641136d2c Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Thu, 30 Jun 2022 01:23:26 +0100 Subject: ALSA: hda: cs35l41: Save Subsystem ID inside CS35L41 Driver The Subsystem ID is read from the HDA driver, and will be used by the CS35L41 driver to be able to uniquely identify the laptop, which is required to be able to define firmware to be used by specific models. Signed-off-by: Stefan Binding Signed-off-by: Vitaly Rodionov Link: https://lore.kernel.org/r/20220630002335.366545-6-vitalyr@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l41_hda.c | 3 +++ sound/pci/hda/cs35l41_hda.h | 1 + 2 files changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index d68d951c434e..25cb76437ba5 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -355,6 +355,9 @@ static int cs35l41_hda_bind(struct device *dev, struct device *master, void *mas return -EBUSY; comps->dev = dev; + if (!cs35l41->acpi_subsystem_id) + cs35l41->acpi_subsystem_id = devm_kasprintf(dev, GFP_KERNEL, "%.8x", + comps->codec->core.subsystem_id); cs35l41->codec = comps->codec; strscpy(comps->name, dev_name(dev), sizeof(comps->name)); comps->playback_hook = cs35l41_hda_playback_hook; diff --git a/sound/pci/hda/cs35l41_hda.h b/sound/pci/hda/cs35l41_hda.h index 5814af050944..b57f59a1ba49 100644 --- a/sound/pci/hda/cs35l41_hda.h +++ b/sound/pci/hda/cs35l41_hda.h @@ -42,6 +42,7 @@ struct cs35l41_hda { int channel_index; unsigned volatile long irq_errors; const char *amp_name; + const char *acpi_subsystem_id; struct mutex fw_mutex; struct regmap_irq_chip_data *irq_data; bool firmware_running; -- cgit v1.2.3 From eef375960210fdc1ec2786bddc91ff100444ffb8 Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Thu, 30 Jun 2022 01:23:27 +0100 Subject: ALSA: hda: cs35l41: Support reading subsystem id from ACPI On some laptop models, the ACPI contains the unique Subsystem ID, and this value should be preferred over the value from the HDA driver. Signed-off-by: Stefan Binding Signed-off-by: Vitaly Rodionov Link: https://lore.kernel.org/r/20220630002335.366545-7-vitalyr@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l41_hda.c | 36 ++++++++++++++++++++++++++++++++++++ 1 file changed, 36 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index 25cb76437ba5..effb3ce95c00 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -544,6 +544,36 @@ static int cs35l41_hda_apply_properties(struct cs35l41_hda *cs35l41) return cs35l41_hda_channel_map(cs35l41->dev, 0, NULL, 1, &hw_cfg->spk_pos); } +static int cs35l41_get_acpi_sub_string(struct device *dev, struct acpi_device *adev, + const char **subsysid) +{ + struct acpi_buffer buffer = { ACPI_ALLOCATE_BUFFER, NULL }; + union acpi_object *obj; + acpi_status status; + int ret = 0; + + status = acpi_evaluate_object(adev->handle, "_SUB", NULL, &buffer); + if (ACPI_SUCCESS(status)) { + obj = buffer.pointer; + if (obj->type == ACPI_TYPE_STRING) { + *subsysid = devm_kstrdup(dev, obj->string.pointer, GFP_KERNEL); + if (*subsysid == NULL) { + dev_err(dev, "Cannot allocate Subsystem ID"); + ret = -ENOMEM; + } + } else { + dev_warn(dev, "Warning ACPI _SUB did not return a string\n"); + ret = -ENODEV; + } + acpi_os_free(buffer.pointer); + } else { + dev_dbg(dev, "Warning ACPI _SUB failed: %#x\n", status); + ret = -ENODEV; + } + + return ret; +} + static int cs35l41_hda_read_acpi(struct cs35l41_hda *cs35l41, const char *hid, int id) { struct cs35l41_hw_cfg *hw_cfg = &cs35l41->hw_cfg; @@ -563,6 +593,12 @@ static int cs35l41_hda_read_acpi(struct cs35l41_hda *cs35l41, const char *hid, i physdev = get_device(acpi_get_first_physical_node(adev)); acpi_dev_put(adev); + ret = cs35l41_get_acpi_sub_string(cs35l41->dev, adev, &cs35l41->acpi_subsystem_id); + if (ret) + dev_info(cs35l41->dev, "No Subsystem ID found in ACPI: %d", ret); + else + dev_dbg(cs35l41->dev, "Subsystem ID %s found", cs35l41->acpi_subsystem_id); + property = "cirrus,dev-index"; ret = device_property_count_u32(physdev, property); if (ret <= 0) -- cgit v1.2.3 From bb6eb621f522d1f76ee4593966d2863401892407 Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Thu, 30 Jun 2022 01:23:28 +0100 Subject: ALSA: hda: cs35l41: Support multiple load paths for firmware To be able to support different firmwares and tuning for different models, the driver needs to be able to load a different firmware and coefficient file based on its Subsystem ID. The driver attempts to load the firmware in the following order: /lib/firmware/cirrus/cs35l41-dsp1---dev<#>.wmfw /lib/firmware/cirrus/cs35l41-dsp1--.wmfw /lib/firmware/cirrus/cs35l41-dsp1-.wmfw Signed-off-by: Stefan Binding Signed-off-by: Vitaly Rodionov Link: https://lore.kernel.org/r/20220630002335.366545-8-vitalyr@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l41_hda.c | 52 +++++++++++++++++++++++++++++++++++++++------ 1 file changed, 46 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index effb3ce95c00..d356c1a55ad5 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -85,14 +85,23 @@ static const struct cs_dsp_client_ops client_ops = { static int cs35l41_request_firmware_file(struct cs35l41_hda *cs35l41, const struct firmware **firmware, char **filename, - const char *dir, const char *filetype) + const char *dir, const char *ssid, const char *amp_name, + const char *filetype) { const char * const dsp_name = cs35l41->cs_dsp.name; char *s, c; int ret = 0; - *filename = kasprintf(GFP_KERNEL, "%s%s-%s-%s.%s", dir, CS35L41_PART, dsp_name, "spk-prot", - filetype); + if (ssid && amp_name) + *filename = kasprintf(GFP_KERNEL, "%s%s-%s-%s-%s-%s.%s", dir, CS35L41_PART, + dsp_name, "spk-prot", ssid, amp_name, + filetype); + else if (ssid) + *filename = kasprintf(GFP_KERNEL, "%s%s-%s-%s-%s.%s", dir, CS35L41_PART, + dsp_name, "spk-prot", ssid, filetype); + else + *filename = kasprintf(GFP_KERNEL, "%s%s-%s-%s.%s", dir, CS35L41_PART, + dsp_name, "spk-prot", filetype); if (*filename == NULL) return -ENOMEM; @@ -129,12 +138,43 @@ static int cs35l41_request_firmware_files(struct cs35l41_hda *cs35l41, { int ret; - /* cirrus/part-dspN-fwtype.wmfw */ + /* try cirrus/part-dspN-fwtype-sub<-ampname>.wmfw */ + ret = cs35l41_request_firmware_file(cs35l41, wmfw_firmware, wmfw_filename, + CS35L41_FIRMWARE_ROOT, cs35l41->acpi_subsystem_id, + cs35l41->amp_name, "wmfw"); + if (!ret) { + /* try cirrus/part-dspN-fwtype-sub<-ampname>.bin */ + cs35l41_request_firmware_file(cs35l41, coeff_firmware, coeff_filename, + CS35L41_FIRMWARE_ROOT, cs35l41->acpi_subsystem_id, + cs35l41->amp_name, "bin"); + return 0; + } + + /* try cirrus/part-dspN-fwtype-sub.wmfw */ + ret = cs35l41_request_firmware_file(cs35l41, wmfw_firmware, wmfw_filename, + CS35L41_FIRMWARE_ROOT, cs35l41->acpi_subsystem_id, + NULL, "wmfw"); + if (!ret) { + /* try cirrus/part-dspN-fwtype-sub<-ampname>.bin */ + ret = cs35l41_request_firmware_file(cs35l41, coeff_firmware, coeff_filename, + CS35L41_FIRMWARE_ROOT, + cs35l41->acpi_subsystem_id, + cs35l41->amp_name, "bin"); + if (ret) + /* try cirrus/part-dspN-fwtype-sub.bin */ + cs35l41_request_firmware_file(cs35l41, coeff_firmware, coeff_filename, + CS35L41_FIRMWARE_ROOT, + cs35l41->acpi_subsystem_id, NULL, "bin"); + return 0; + } + + /* fallback try cirrus/part-dspN-fwtype.wmfw */ ret = cs35l41_request_firmware_file(cs35l41, wmfw_firmware, wmfw_filename, - CS35L41_FIRMWARE_ROOT, "wmfw"); + CS35L41_FIRMWARE_ROOT, NULL, NULL, "wmfw"); if (!ret) { + /* fallback try cirrus/part-dspN-fwtype.bin */ cs35l41_request_firmware_file(cs35l41, coeff_firmware, coeff_filename, - CS35L41_FIRMWARE_ROOT, "bin"); + CS35L41_FIRMWARE_ROOT, NULL, NULL, "bin"); return 0; } -- cgit v1.2.3 From 63f4b99f0089a9719aa4441015fe30ff4b6f10e5 Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Thu, 30 Jun 2022 01:23:29 +0100 Subject: ALSA: hda: cs35l41: Support Speaker ID for laptops Some Laptops use a number of gpios to define which vendor is used for a particular laptop. Different coefficient files are used for different vendors. Signed-off-by: Stefan Binding Signed-off-by: Vitaly Rodionov Link: https://lore.kernel.org/r/20220630002335.366545-9-vitalyr@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l41_hda.c | 174 +++++++++++++++++++++++++++++++++++++++++--- sound/pci/hda/cs35l41_hda.h | 1 + 2 files changed, 166 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index d356c1a55ad5..d5d2323cc8c7 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -86,13 +86,19 @@ static const struct cs_dsp_client_ops client_ops = { static int cs35l41_request_firmware_file(struct cs35l41_hda *cs35l41, const struct firmware **firmware, char **filename, const char *dir, const char *ssid, const char *amp_name, - const char *filetype) + int spkid, const char *filetype) { const char * const dsp_name = cs35l41->cs_dsp.name; char *s, c; int ret = 0; - if (ssid && amp_name) + if (spkid > -1 && ssid && amp_name) + *filename = kasprintf(GFP_KERNEL, "%s%s-%s-%s-%s-spkid%d-%s.%s", dir, CS35L41_PART, + dsp_name, "spk-prot", ssid, spkid, amp_name, filetype); + else if (spkid > -1 && ssid) + *filename = kasprintf(GFP_KERNEL, "%s%s-%s-%s-%s-spkid%d.%s", dir, CS35L41_PART, + dsp_name, "spk-prot", ssid, spkid, filetype); + else if (ssid && amp_name) *filename = kasprintf(GFP_KERNEL, "%s%s-%s-%s-%s-%s.%s", dir, CS35L41_PART, dsp_name, "spk-prot", ssid, amp_name, filetype); @@ -130,6 +136,93 @@ static int cs35l41_request_firmware_file(struct cs35l41_hda *cs35l41, return ret; } +static int cs35l41_request_firmware_files_spkid(struct cs35l41_hda *cs35l41, + const struct firmware **wmfw_firmware, + char **wmfw_filename, + const struct firmware **coeff_firmware, + char **coeff_filename) +{ + int ret; + + /* try cirrus/part-dspN-fwtype-sub<-spkidN><-ampname>.wmfw */ + ret = cs35l41_request_firmware_file(cs35l41, wmfw_firmware, wmfw_filename, + CS35L41_FIRMWARE_ROOT, + cs35l41->acpi_subsystem_id, cs35l41->amp_name, + cs35l41->speaker_id, "wmfw"); + if (!ret) { + /* try cirrus/part-dspN-fwtype-sub<-spkidN><-ampname>.bin */ + cs35l41_request_firmware_file(cs35l41, coeff_firmware, coeff_filename, + CS35L41_FIRMWARE_ROOT, + cs35l41->acpi_subsystem_id, cs35l41->amp_name, + cs35l41->speaker_id, "bin"); + return 0; + } + + /* try cirrus/part-dspN-fwtype-sub<-ampname>.wmfw */ + ret = cs35l41_request_firmware_file(cs35l41, wmfw_firmware, wmfw_filename, + CS35L41_FIRMWARE_ROOT, cs35l41->acpi_subsystem_id, + cs35l41->amp_name, -1, "wmfw"); + if (!ret) { + /* try cirrus/part-dspN-fwtype-sub<-spkidN><-ampname>.bin */ + cs35l41_request_firmware_file(cs35l41, coeff_firmware, coeff_filename, + CS35L41_FIRMWARE_ROOT, cs35l41->acpi_subsystem_id, + cs35l41->amp_name, cs35l41->speaker_id, "bin"); + return 0; + } + + /* try cirrus/part-dspN-fwtype-sub<-spkidN>.wmfw */ + ret = cs35l41_request_firmware_file(cs35l41, wmfw_firmware, wmfw_filename, + CS35L41_FIRMWARE_ROOT, cs35l41->acpi_subsystem_id, + NULL, cs35l41->speaker_id, "wmfw"); + if (!ret) { + /* try cirrus/part-dspN-fwtype-sub<-spkidN><-ampname>.bin */ + ret = cs35l41_request_firmware_file(cs35l41, coeff_firmware, coeff_filename, + CS35L41_FIRMWARE_ROOT, + cs35l41->acpi_subsystem_id, + cs35l41->amp_name, cs35l41->speaker_id, "bin"); + if (ret) + /* try cirrus/part-dspN-fwtype-sub<-spkidN>.bin */ + cs35l41_request_firmware_file(cs35l41, coeff_firmware, coeff_filename, + CS35L41_FIRMWARE_ROOT, + cs35l41->acpi_subsystem_id, + NULL, cs35l41->speaker_id, "bin"); + return 0; + } + + /* try cirrus/part-dspN-fwtype-sub.wmfw */ + ret = cs35l41_request_firmware_file(cs35l41, wmfw_firmware, wmfw_filename, + CS35L41_FIRMWARE_ROOT, cs35l41->acpi_subsystem_id, + NULL, -1, "wmfw"); + if (!ret) { + /* try cirrus/part-dspN-fwtype-sub<-spkidN><-ampname>.bin */ + ret = cs35l41_request_firmware_file(cs35l41, coeff_firmware, coeff_filename, + CS35L41_FIRMWARE_ROOT, + cs35l41->acpi_subsystem_id, + cs35l41->amp_name, cs35l41->speaker_id, "bin"); + if (ret) + /* try cirrus/part-dspN-fwtype-sub<-spkidN>.bin */ + cs35l41_request_firmware_file(cs35l41, coeff_firmware, coeff_filename, + CS35L41_FIRMWARE_ROOT, + cs35l41->acpi_subsystem_id, + NULL, cs35l41->speaker_id, "bin"); + return 0; + } + + /* fallback try cirrus/part-dspN-fwtype.wmfw */ + ret = cs35l41_request_firmware_file(cs35l41, wmfw_firmware, wmfw_filename, + CS35L41_FIRMWARE_ROOT, NULL, NULL, -1, "wmfw"); + if (!ret) { + /* fallback try cirrus/part-dspN-fwtype.bin */ + cs35l41_request_firmware_file(cs35l41, coeff_firmware, coeff_filename, + CS35L41_FIRMWARE_ROOT, NULL, NULL, -1, "bin"); + return 0; + } + + dev_warn(cs35l41->dev, "Failed to request firmware\n"); + + return ret; +} + static int cs35l41_request_firmware_files(struct cs35l41_hda *cs35l41, const struct firmware **wmfw_firmware, char **wmfw_filename, @@ -138,43 +231,48 @@ static int cs35l41_request_firmware_files(struct cs35l41_hda *cs35l41, { int ret; + if (cs35l41->speaker_id > -1) + return cs35l41_request_firmware_files_spkid(cs35l41, wmfw_firmware, wmfw_filename, + coeff_firmware, coeff_filename); + /* try cirrus/part-dspN-fwtype-sub<-ampname>.wmfw */ ret = cs35l41_request_firmware_file(cs35l41, wmfw_firmware, wmfw_filename, CS35L41_FIRMWARE_ROOT, cs35l41->acpi_subsystem_id, - cs35l41->amp_name, "wmfw"); + cs35l41->amp_name, -1, "wmfw"); if (!ret) { /* try cirrus/part-dspN-fwtype-sub<-ampname>.bin */ cs35l41_request_firmware_file(cs35l41, coeff_firmware, coeff_filename, CS35L41_FIRMWARE_ROOT, cs35l41->acpi_subsystem_id, - cs35l41->amp_name, "bin"); + cs35l41->amp_name, -1, "bin"); return 0; } /* try cirrus/part-dspN-fwtype-sub.wmfw */ ret = cs35l41_request_firmware_file(cs35l41, wmfw_firmware, wmfw_filename, CS35L41_FIRMWARE_ROOT, cs35l41->acpi_subsystem_id, - NULL, "wmfw"); + NULL, -1, "wmfw"); if (!ret) { /* try cirrus/part-dspN-fwtype-sub<-ampname>.bin */ ret = cs35l41_request_firmware_file(cs35l41, coeff_firmware, coeff_filename, CS35L41_FIRMWARE_ROOT, cs35l41->acpi_subsystem_id, - cs35l41->amp_name, "bin"); + cs35l41->amp_name, -1, "bin"); if (ret) /* try cirrus/part-dspN-fwtype-sub.bin */ cs35l41_request_firmware_file(cs35l41, coeff_firmware, coeff_filename, CS35L41_FIRMWARE_ROOT, - cs35l41->acpi_subsystem_id, NULL, "bin"); + cs35l41->acpi_subsystem_id, + NULL, -1, "bin"); return 0; } /* fallback try cirrus/part-dspN-fwtype.wmfw */ ret = cs35l41_request_firmware_file(cs35l41, wmfw_firmware, wmfw_filename, - CS35L41_FIRMWARE_ROOT, NULL, NULL, "wmfw"); + CS35L41_FIRMWARE_ROOT, NULL, NULL, -1, "wmfw"); if (!ret) { /* fallback try cirrus/part-dspN-fwtype.bin */ cs35l41_request_firmware_file(cs35l41, coeff_firmware, coeff_filename, - CS35L41_FIRMWARE_ROOT, NULL, NULL, "bin"); + CS35L41_FIRMWARE_ROOT, NULL, NULL, -1, "bin"); return 0; } @@ -614,6 +712,61 @@ static int cs35l41_get_acpi_sub_string(struct device *dev, struct acpi_device *a return ret; } +static int cs35l41_get_speaker_id(struct device *dev, int amp_index, + int num_amps, int fixed_gpio_id) +{ + struct gpio_desc *speaker_id_desc; + int speaker_id = -ENODEV; + + if (fixed_gpio_id >= 0) { + dev_dbg(dev, "Found Fixed Speaker ID GPIO (index = %d)\n", fixed_gpio_id); + speaker_id_desc = gpiod_get_index(dev, NULL, fixed_gpio_id, GPIOD_IN); + if (IS_ERR(speaker_id_desc)) { + speaker_id = PTR_ERR(speaker_id_desc); + return speaker_id; + } + speaker_id = gpiod_get_value_cansleep(speaker_id_desc); + gpiod_put(speaker_id_desc); + dev_dbg(dev, "Speaker ID = %d\n", speaker_id); + } else { + int base_index; + int gpios_per_amp; + int count; + int tmp; + int i; + + count = gpiod_count(dev, "spk-id"); + if (count > 0) { + speaker_id = 0; + gpios_per_amp = count / num_amps; + base_index = gpios_per_amp * amp_index; + + if (count % num_amps) + return -EINVAL; + + dev_dbg(dev, "Found %d Speaker ID GPIOs per Amp\n", gpios_per_amp); + + for (i = 0; i < gpios_per_amp; i++) { + speaker_id_desc = gpiod_get_index(dev, "spk-id", i + base_index, + GPIOD_IN); + if (IS_ERR(speaker_id_desc)) { + speaker_id = PTR_ERR(speaker_id_desc); + break; + } + tmp = gpiod_get_value_cansleep(speaker_id_desc); + gpiod_put(speaker_id_desc); + if (tmp < 0) { + speaker_id = tmp; + break; + } + speaker_id |= tmp << i; + } + dev_dbg(dev, "Speaker ID = %d\n", speaker_id); + } + } + return speaker_id; +} + static int cs35l41_hda_read_acpi(struct cs35l41_hda *cs35l41, const char *hid, int id) { struct cs35l41_hw_cfg *hw_cfg = &cs35l41->hw_cfg; @@ -719,6 +872,8 @@ static int cs35l41_hda_read_acpi(struct cs35l41_hda *cs35l41, const char *hid, i else hw_cfg->bst_cap = -1; + cs35l41->speaker_id = cs35l41_get_speaker_id(physdev, cs35l41->index, nval, -1); + if (hw_cfg->bst_ind > 0 || hw_cfg->bst_cap > 0 || hw_cfg->bst_ipk > 0) hw_cfg->bst_type = CS35L41_INT_BOOST; else @@ -752,6 +907,7 @@ no_acpi_dsd: cs35l41->channel_index = 0; cs35l41->reset_gpio = gpiod_get_index(physdev, NULL, 0, GPIOD_OUT_HIGH); cs35l41->hw_cfg.bst_type = CS35L41_EXT_BOOST_NO_VSPK_SWITCH; + cs35l41->speaker_id = cs35l41_get_speaker_id(physdev, 0, 0, 2); hw_cfg->gpio2.func = CS35L41_GPIO2_INT_OPEN_DRAIN; hw_cfg->gpio2.valid = true; cs35l41->hw_cfg.valid = true; diff --git a/sound/pci/hda/cs35l41_hda.h b/sound/pci/hda/cs35l41_hda.h index b57f59a1ba49..a9dbc1c19248 100644 --- a/sound/pci/hda/cs35l41_hda.h +++ b/sound/pci/hda/cs35l41_hda.h @@ -43,6 +43,7 @@ struct cs35l41_hda { unsigned volatile long irq_errors; const char *amp_name; const char *acpi_subsystem_id; + int speaker_id; struct mutex fw_mutex; struct regmap_irq_chip_data *irq_data; bool firmware_running; -- cgit v1.2.3 From fa9b878ff86f4adccddf62492a5894fbdb04f97d Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Fri, 8 Jul 2022 16:48:36 +0300 Subject: ASoC: SOF: ipc-msg-injector: fix copy in sof_msg_inject_ipc4_dfs_write() There are two bugs that have to do with when we copy the payload: size = simple_write_to_buffer(ipc4_msg->data_ptr, priv->max_msg_size, ppos, buffer, count); The value of "*ppos" was supposed to be zero but it is sizeof(ipc4_msg->header_u64) so it will copy the data into the middle of the "ipc4_msg->data_ptr" buffer instead of to the start. The second problem is "buffer" should be "buffer + sizeof(ipc4_msg->header_u64)". This function is used for fuzz testing so the data is normally random and this bug likely does not affect anyone very much. In this context, it's simpler and more appropriate to use copy_from_user() instead of simple_write_to_buffer() so I have re-written the function. Fixes: 066c67624d8c ("ASoC: SOF: ipc-msg-injector: Add support for IPC4 messages") Signed-off-by: Dan Carpenter Link: https://lore.kernel.org/r/Ysg1tB2FKLnRMsel@kili Signed-off-by: Mark Brown --- sound/soc/sof/sof-client-ipc-msg-injector.c | 29 ++++++++++++----------------- 1 file changed, 12 insertions(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/sof-client-ipc-msg-injector.c b/sound/soc/sof/sof-client-ipc-msg-injector.c index 6bdfa527b7f7..752d5320680f 100644 --- a/sound/soc/sof/sof-client-ipc-msg-injector.c +++ b/sound/soc/sof/sof-client-ipc-msg-injector.c @@ -181,7 +181,7 @@ static ssize_t sof_msg_inject_ipc4_dfs_write(struct file *file, struct sof_client_dev *cdev = file->private_data; struct sof_msg_inject_priv *priv = cdev->data; struct sof_ipc4_msg *ipc4_msg = priv->tx_buffer; - ssize_t size; + size_t data_size; int ret; if (*ppos) @@ -191,25 +191,20 @@ static ssize_t sof_msg_inject_ipc4_dfs_write(struct file *file, return -EINVAL; /* copy the header first */ - size = simple_write_to_buffer(&ipc4_msg->header_u64, - sizeof(ipc4_msg->header_u64), - ppos, buffer, count); - if (size < 0) - return size; - if (size != sizeof(ipc4_msg->header_u64)) + if (copy_from_user(&ipc4_msg->header_u64, buffer, + sizeof(ipc4_msg->header_u64))) return -EFAULT; - count -= size; + data_size = count - sizeof(ipc4_msg->header_u64); + if (data_size > priv->max_msg_size) + return -EINVAL; + /* Copy the payload */ - size = simple_write_to_buffer(ipc4_msg->data_ptr, - priv->max_msg_size, ppos, buffer, - count); - if (size < 0) - return size; - if (size != count) + if (copy_from_user(ipc4_msg->data_ptr, + buffer + sizeof(ipc4_msg->header_u64), data_size)) return -EFAULT; - ipc4_msg->data_size = count; + ipc4_msg->data_size = data_size; /* Initialize the reply storage */ ipc4_msg = priv->rx_buffer; @@ -221,9 +216,9 @@ static ssize_t sof_msg_inject_ipc4_dfs_write(struct file *file, /* return the error code if test failed */ if (ret < 0) - size = ret; + return ret; - return size; + return count; }; static int sof_msg_inject_dfs_release(struct inode *inode, struct file *file) -- cgit v1.2.3 From ef30911d3c39fd57884c348c29b9cbff88def155 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 14 Jul 2022 06:28:15 +0000 Subject: ASoC: rsnd: care default case on rsnd_ssiu_busif_err_irq_ctrl() Before, ssiu.c didn't care SSI5-8, thus, commit b1384d4c95088d0 ("ASoC: rsnd: care default case on rsnd_ssiu_busif_err_status_clear()") cares it for status clear. But we should care it for error irq handling, too. This patch cares it. Reported-by: Nguyen Bao Nguyen Reported-by: Nishiyama Kunihiko Signed-off-by: Kuninori Morimoto Link: https://lore.kernel.org/r/871quocio1.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/sh/rcar/ssiu.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index 4b8a63e336c7..d7f4646ee029 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -67,6 +67,8 @@ static void rsnd_ssiu_busif_err_irq_ctrl(struct rsnd_mod *mod, int enable) shift = 1; offset = 1; break; + default: + return; } for (i = 0; i < 4; i++) { -- cgit v1.2.3 From ffb2759df7efbc00187bfd9d1072434a13a54139 Mon Sep 17 00:00:00 2001 From: Zheyu Ma Date: Fri, 15 Jul 2022 09:05:15 +0800 Subject: ALSA: bcd2000: Fix a UAF bug on the error path of probing When the driver fails in snd_card_register() at probe time, it will free the 'bcd2k->midi_out_urb' before killing it, which may cause a UAF bug. The following log can reveal it: [ 50.727020] BUG: KASAN: use-after-free in bcd2000_input_complete+0x1f1/0x2e0 [snd_bcd2000] [ 50.727623] Read of size 8 at addr ffff88810fab0e88 by task swapper/4/0 [ 50.729530] Call Trace: [ 50.732899] bcd2000_input_complete+0x1f1/0x2e0 [snd_bcd2000] Fix this by adding usb_kill_urb() before usb_free_urb(). Fixes: b47a22290d58 ("ALSA: MIDI driver for Behringer BCD2000 USB device") Signed-off-by: Zheyu Ma Cc: Link: https://lore.kernel.org/r/20220715010515.2087925-1-zheyuma97@gmail.com Signed-off-by: Takashi Iwai --- sound/usb/bcd2000/bcd2000.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/bcd2000/bcd2000.c b/sound/usb/bcd2000/bcd2000.c index cd4a0bc6d278..7aec0a95c609 100644 --- a/sound/usb/bcd2000/bcd2000.c +++ b/sound/usb/bcd2000/bcd2000.c @@ -348,7 +348,8 @@ static int bcd2000_init_midi(struct bcd2000 *bcd2k) static void bcd2000_free_usb_related_resources(struct bcd2000 *bcd2k, struct usb_interface *interface) { - /* usb_kill_urb not necessary, urb is aborted automatically */ + usb_kill_urb(bcd2k->midi_out_urb); + usb_kill_urb(bcd2k->midi_in_urb); usb_free_urb(bcd2k->midi_out_urb); usb_free_urb(bcd2k->midi_in_urb); -- cgit v1.2.3 From 1873ebd30cc818eefd151e40a4bd05fd8f83b85a Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Thu, 30 Jun 2022 01:23:30 +0100 Subject: ALSA: hda: cs35l41: Support Hibernation during Suspend CS35L41 supports hibernation during suspend when using DSP firmware. When the driver suspends it will hibernate the part, if firmware is running, and resume will wake from hibernation. CS35L41 driver will suspend/resume when requested by hda driver. Note that suspend/resume and hibernation is only supported when firmware is running. Signed-off-by: Stefan Binding Signed-off-by: Vitaly Rodionov Link: https://lore.kernel.org/r/20220630002335.366545-10-vitalyr@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l41_hda.c | 109 ++++++++++++++++++++++++++++++++++++++-- sound/pci/hda/cs35l41_hda.h | 2 + sound/pci/hda/cs35l41_hda_i2c.c | 1 + sound/pci/hda/cs35l41_hda_spi.c | 1 + sound/pci/hda/hda_component.h | 2 + sound/pci/hda/patch_realtek.c | 25 ++++++++- 6 files changed, 136 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index d5d2323cc8c7..61441bfc9fa9 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -10,6 +10,7 @@ #include #include #include +#include #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_jack.h" @@ -435,6 +436,75 @@ static int cs35l41_hda_channel_map(struct device *dev, unsigned int tx_num, unsi rx_slot); } +static int cs35l41_runtime_suspend(struct device *dev) +{ + struct cs35l41_hda *cs35l41 = dev_get_drvdata(dev); + + dev_dbg(cs35l41->dev, "Suspend\n"); + + if (!cs35l41->firmware_running) + return 0; + + if (cs35l41_enter_hibernate(cs35l41->dev, cs35l41->regmap, cs35l41->hw_cfg.bst_type) < 0) + return 0; + + regcache_cache_only(cs35l41->regmap, true); + regcache_mark_dirty(cs35l41->regmap); + + return 0; +} + +static int cs35l41_runtime_resume(struct device *dev) +{ + struct cs35l41_hda *cs35l41 = dev_get_drvdata(dev); + int ret; + + dev_dbg(cs35l41->dev, "Resume.\n"); + + if (cs35l41->hw_cfg.bst_type == CS35L41_EXT_BOOST_NO_VSPK_SWITCH) { + dev_dbg(cs35l41->dev, "System does not support Resume\n"); + return 0; + } + + if (!cs35l41->firmware_running) + return 0; + + regcache_cache_only(cs35l41->regmap, false); + + ret = cs35l41_exit_hibernate(cs35l41->dev, cs35l41->regmap); + if (ret) { + regcache_cache_only(cs35l41->regmap, true); + return ret; + } + + /* Test key needs to be unlocked to allow the OTP settings to re-apply */ + cs35l41_test_key_unlock(cs35l41->dev, cs35l41->regmap); + ret = regcache_sync(cs35l41->regmap); + cs35l41_test_key_lock(cs35l41->dev, cs35l41->regmap); + if (ret) { + dev_err(cs35l41->dev, "Failed to restore register cache: %d\n", ret); + return ret; + } + + if (cs35l41->hw_cfg.bst_type == CS35L41_EXT_BOOST) + cs35l41_init_boost(cs35l41->dev, cs35l41->regmap, &cs35l41->hw_cfg); + + return 0; +} + +static int cs35l41_hda_suspend_hook(struct device *dev) +{ + dev_dbg(dev, "Request Suspend\n"); + pm_runtime_mark_last_busy(dev); + return pm_runtime_put_autosuspend(dev); +} + +static int cs35l41_hda_resume_hook(struct device *dev) +{ + dev_dbg(dev, "Request Resume\n"); + return pm_runtime_get_sync(dev); +} + static int cs35l41_smart_amp(struct cs35l41_hda *cs35l41) { int halo_sts; @@ -492,19 +562,27 @@ static int cs35l41_hda_bind(struct device *dev, struct device *master, void *mas if (comps->dev) return -EBUSY; + pm_runtime_get_sync(dev); + comps->dev = dev; if (!cs35l41->acpi_subsystem_id) cs35l41->acpi_subsystem_id = devm_kasprintf(dev, GFP_KERNEL, "%.8x", comps->codec->core.subsystem_id); cs35l41->codec = comps->codec; strscpy(comps->name, dev_name(dev), sizeof(comps->name)); - comps->playback_hook = cs35l41_hda_playback_hook; mutex_lock(&cs35l41->fw_mutex); if (cs35l41_smart_amp(cs35l41) < 0) dev_warn(cs35l41->dev, "Cannot Run Firmware, reverting to dsp bypass...\n"); mutex_unlock(&cs35l41->fw_mutex); + comps->playback_hook = cs35l41_hda_playback_hook; + comps->suspend_hook = cs35l41_hda_suspend_hook; + comps->resume_hook = cs35l41_hda_resume_hook; + + pm_runtime_mark_last_busy(dev); + pm_runtime_put_autosuspend(dev); + return 0; } @@ -600,7 +678,7 @@ static const struct regmap_irq cs35l41_reg_irqs[] = { CS35L41_REG_IRQ(IRQ1_STATUS1, AMP_SHORT_ERR), }; -static const struct regmap_irq_chip cs35l41_regmap_irq_chip = { +static struct regmap_irq_chip cs35l41_regmap_irq_chip = { .name = "cs35l41 IRQ1 Controller", .status_base = CS35L41_IRQ1_STATUS1, .mask_base = CS35L41_IRQ1_MASK1, @@ -608,6 +686,7 @@ static const struct regmap_irq_chip cs35l41_regmap_irq_chip = { .num_regs = 4, .irqs = cs35l41_reg_irqs, .num_irqs = ARRAY_SIZE(cs35l41_reg_irqs), + .runtime_pm = true, }; static int cs35l41_hda_apply_properties(struct cs35l41_hda *cs35l41) @@ -1015,13 +1094,23 @@ int cs35l41_hda_probe(struct device *dev, const char *device_name, int id, int i mutex_init(&cs35l41->fw_mutex); + pm_runtime_set_autosuspend_delay(cs35l41->dev, 3000); + pm_runtime_use_autosuspend(cs35l41->dev); + pm_runtime_mark_last_busy(cs35l41->dev); + pm_runtime_set_active(cs35l41->dev); + pm_runtime_get_noresume(cs35l41->dev); + pm_runtime_enable(cs35l41->dev); + ret = cs35l41_hda_apply_properties(cs35l41); if (ret) - goto err; + goto err_pm; + + pm_runtime_put_autosuspend(cs35l41->dev); ret = component_add(cs35l41->dev, &cs35l41_hda_comp_ops); if (ret) { dev_err(cs35l41->dev, "Register component failed: %d\n", ret); + pm_runtime_disable(cs35l41->dev); goto err; } @@ -1029,6 +1118,10 @@ int cs35l41_hda_probe(struct device *dev, const char *device_name, int id, int i return 0; +err_pm: + pm_runtime_disable(cs35l41->dev); + pm_runtime_put_noidle(cs35l41->dev); + err: if (cs35l41_safe_reset(cs35l41->regmap, cs35l41->hw_cfg.bst_type)) gpiod_set_value_cansleep(cs35l41->reset_gpio, 0); @@ -1042,17 +1135,27 @@ void cs35l41_hda_remove(struct device *dev) { struct cs35l41_hda *cs35l41 = dev_get_drvdata(dev); + pm_runtime_get_sync(cs35l41->dev); + pm_runtime_disable(cs35l41->dev); + if (cs35l41->halo_initialized) cs35l41_remove_dsp(cs35l41); component_del(cs35l41->dev, &cs35l41_hda_comp_ops); + pm_runtime_put_noidle(cs35l41->dev); + if (cs35l41_safe_reset(cs35l41->regmap, cs35l41->hw_cfg.bst_type)) gpiod_set_value_cansleep(cs35l41->reset_gpio, 0); gpiod_put(cs35l41->reset_gpio); } EXPORT_SYMBOL_NS_GPL(cs35l41_hda_remove, SND_HDA_SCODEC_CS35L41); +const struct dev_pm_ops cs35l41_hda_pm_ops = { + SET_RUNTIME_PM_OPS(cs35l41_runtime_suspend, cs35l41_runtime_resume, NULL) +}; +EXPORT_SYMBOL_NS_GPL(cs35l41_hda_pm_ops, SND_HDA_SCODEC_CS35L41); + MODULE_DESCRIPTION("CS35L41 HDA Driver"); MODULE_IMPORT_NS(SND_HDA_CS_DSP_CONTROLS); MODULE_AUTHOR("Lucas Tanure, Cirrus Logic Inc, "); diff --git a/sound/pci/hda/cs35l41_hda.h b/sound/pci/hda/cs35l41_hda.h index a9dbc1c19248..439c4b705328 100644 --- a/sound/pci/hda/cs35l41_hda.h +++ b/sound/pci/hda/cs35l41_hda.h @@ -57,6 +57,8 @@ enum halo_state { HALO_STATE_CODE_RUN }; +extern const struct dev_pm_ops cs35l41_hda_pm_ops; + int cs35l41_hda_probe(struct device *dev, const char *device_name, int id, int irq, struct regmap *regmap); void cs35l41_hda_remove(struct device *dev); diff --git a/sound/pci/hda/cs35l41_hda_i2c.c b/sound/pci/hda/cs35l41_hda_i2c.c index df39fc76e6be..9c08fa08c421 100644 --- a/sound/pci/hda/cs35l41_hda_i2c.c +++ b/sound/pci/hda/cs35l41_hda_i2c.c @@ -54,6 +54,7 @@ static struct i2c_driver cs35l41_i2c_driver = { .driver = { .name = "cs35l41-hda", .acpi_match_table = cs35l41_acpi_hda_match, + .pm = &cs35l41_hda_pm_ops, }, .id_table = cs35l41_hda_i2c_id, .probe = cs35l41_hda_i2c_probe, diff --git a/sound/pci/hda/cs35l41_hda_spi.c b/sound/pci/hda/cs35l41_hda_spi.c index 2f5afad3719e..71979cfb4d7e 100644 --- a/sound/pci/hda/cs35l41_hda_spi.c +++ b/sound/pci/hda/cs35l41_hda_spi.c @@ -49,6 +49,7 @@ static struct spi_driver cs35l41_spi_driver = { .driver = { .name = "cs35l41-hda", .acpi_match_table = cs35l41_acpi_hda_match, + .pm = &cs35l41_hda_pm_ops, }, .id_table = cs35l41_hda_spi_id, .probe = cs35l41_hda_spi_probe, diff --git a/sound/pci/hda/hda_component.h b/sound/pci/hda/hda_component.h index 534e845b9cd1..1223621bd62c 100644 --- a/sound/pci/hda/hda_component.h +++ b/sound/pci/hda/hda_component.h @@ -16,4 +16,6 @@ struct hda_component { char name[HDA_MAX_NAME_SIZE]; struct hda_codec *codec; void (*playback_hook)(struct device *dev, int action); + int (*suspend_hook)(struct device *dev); + int (*resume_hook)(struct device *dev); }; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 44744d568404..7c21bc439c46 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4021,15 +4021,22 @@ static void alc5505_dsp_init(struct hda_codec *codec) static int alc269_suspend(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; + int i; if (spec->has_alc5505_dsp) alc5505_dsp_suspend(codec); + + for (i = 0; i < HDA_MAX_COMPONENTS; i++) + if (spec->comps[i].suspend_hook) + spec->comps[i].suspend_hook(spec->comps[i].dev); + return alc_suspend(codec); } static int alc269_resume(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; + int i; if (spec->codec_variant == ALC269_TYPE_ALC269VB) alc269vb_toggle_power_output(codec, 0); @@ -4060,6 +4067,10 @@ static int alc269_resume(struct hda_codec *codec) if (spec->has_alc5505_dsp) alc5505_dsp_resume(codec); + for (i = 0; i < HDA_MAX_COMPONENTS; i++) + if (spec->comps[i].resume_hook) + spec->comps[i].resume_hook(spec->comps[i].dev); + return 0; } #endif /* CONFIG_PM */ @@ -6610,8 +6621,20 @@ static int comp_bind(struct device *dev) { struct hda_codec *cdc = dev_to_hda_codec(dev); struct alc_spec *spec = cdc->spec; + int ret, i; + + ret = component_bind_all(dev, spec->comps); + if (ret) + return ret; - return component_bind_all(dev, spec->comps); + if (snd_hdac_is_power_on(&cdc->core)) { + codec_dbg(cdc, "Resuming after bind.\n"); + for (i = 0; i < HDA_MAX_COMPONENTS; i++) + if (spec->comps[i].resume_hook) + spec->comps[i].resume_hook(spec->comps[i].dev); + } + + return 0; } static void comp_unbind(struct device *dev) -- cgit v1.2.3 From 3e34e2ae29591f0fd84dca905d296da1e127160c Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Thu, 30 Jun 2022 01:23:31 +0100 Subject: ALSA: hda: cs35l41: Read Speaker Calibration data from UEFI variables Speaker Calibration data, specific to an individual speaker is stored inside UEFI variables during calibration, and can be used by the DSP. Signed-off-by: Stefan Binding Signed-off-by: Vitaly Rodionov Link: https://lore.kernel.org/r/20220630002335.366545-11-vitalyr@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l41_hda.c | 101 ++++++++++++++++++++++++++++++++++++++++++++ sound/pci/hda/cs35l41_hda.h | 15 +++++++ 2 files changed, 116 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index 61441bfc9fa9..75edaffb568a 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -26,6 +26,12 @@ #define HALO_STATE_DSP_CTL_NAME "HALO_STATE" #define HALO_STATE_DSP_CTL_TYPE 5 #define HALO_STATE_DSP_CTL_ALG 262308 +#define CAL_R_DSP_CTL_NAME "CAL_R" +#define CAL_STATUS_DSP_CTL_NAME "CAL_STATUS" +#define CAL_CHECKSUM_DSP_CTL_NAME "CAL_CHECKSUM" +#define CAL_AMBIENT_DSP_CTL_NAME "CAL_AMBIENT" +#define CAL_DSP_CTL_TYPE 5 +#define CAL_DSP_CTL_ALG 205 static const struct reg_sequence cs35l41_hda_config[] = { { CS35L41_PLL_CLK_CTRL, 0x00000430 }, // 3072000Hz, BCLK Input, PLL_REFCLK_EN = 1 @@ -282,6 +288,96 @@ static int cs35l41_request_firmware_files(struct cs35l41_hda *cs35l41, return ret; } +#if IS_ENABLED(CONFIG_EFI) +static int cs35l41_apply_calibration(struct cs35l41_hda *cs35l41, unsigned int ambient, + unsigned int r0, unsigned int status, unsigned int checksum) +{ + int ret; + + ret = hda_cs_dsp_write_ctl(&cs35l41->cs_dsp, CAL_AMBIENT_DSP_CTL_NAME, CAL_DSP_CTL_TYPE, + CAL_DSP_CTL_ALG, &ambient, 4); + if (ret) { + dev_err(cs35l41->dev, "Cannot Write Control: %s - %d\n", CAL_AMBIENT_DSP_CTL_NAME, + ret); + return ret; + } + ret = hda_cs_dsp_write_ctl(&cs35l41->cs_dsp, CAL_R_DSP_CTL_NAME, CAL_DSP_CTL_TYPE, + CAL_DSP_CTL_ALG, &r0, 4); + if (ret) { + dev_err(cs35l41->dev, "Cannot Write Control: %s - %d\n", CAL_R_DSP_CTL_NAME, ret); + return ret; + } + ret = hda_cs_dsp_write_ctl(&cs35l41->cs_dsp, CAL_STATUS_DSP_CTL_NAME, CAL_DSP_CTL_TYPE, + CAL_DSP_CTL_ALG, &status, 4); + if (ret) { + dev_err(cs35l41->dev, "Cannot Write Control: %s - %d\n", CAL_STATUS_DSP_CTL_NAME, + ret); + return ret; + } + ret = hda_cs_dsp_write_ctl(&cs35l41->cs_dsp, CAL_CHECKSUM_DSP_CTL_NAME, CAL_DSP_CTL_TYPE, + CAL_DSP_CTL_ALG, &checksum, 4); + if (ret) { + dev_err(cs35l41->dev, "Cannot Write Control: %s - %d\n", CAL_CHECKSUM_DSP_CTL_NAME, + ret); + return ret; + } + + return 0; +} + +static int cs35l41_save_calibration(struct cs35l41_hda *cs35l41) +{ + static efi_guid_t efi_guid = EFI_GUID(0x02f9af02, 0x7734, 0x4233, 0xb4, 0x3d, 0x93, 0xfe, + 0x5a, 0xa3, 0x5d, 0xb3); + static efi_char16_t efi_name[] = L"CirrusSmartAmpCalibrationData"; + const struct cs35l41_amp_efi_data *efi_data; + const struct cs35l41_amp_cal_data *cl; + unsigned long data_size = 0; + efi_status_t status; + int ret = 0; + u8 *data = NULL; + u32 attr; + + /* Get real size of UEFI variable */ + status = efi.get_variable(efi_name, &efi_guid, &attr, &data_size, data); + if (status == EFI_BUFFER_TOO_SMALL) { + ret = -ENODEV; + /* Allocate data buffer of data_size bytes */ + data = vmalloc(data_size); + if (!data) + return -ENOMEM; + /* Get variable contents into buffer */ + status = efi.get_variable(efi_name, &efi_guid, &attr, &data_size, data); + if (status == EFI_SUCCESS) { + efi_data = (struct cs35l41_amp_efi_data *)data; + dev_dbg(cs35l41->dev, "Calibration: Size=%d, Amp Count=%d\n", + efi_data->size, efi_data->count); + if (efi_data->count > cs35l41->index) { + cl = &efi_data->data[cs35l41->index]; + dev_dbg(cs35l41->dev, + "Calibration: Ambient=%02x, Status=%02x, R0=%d\n", + cl->calAmbient, cl->calStatus, cl->calR); + + /* Calibration can only be applied whilst the DSP is not running */ + ret = cs35l41_apply_calibration(cs35l41, + cpu_to_be32(cl->calAmbient), + cpu_to_be32(cl->calR), + cpu_to_be32(cl->calStatus), + cpu_to_be32(cl->calR + 1)); + } + } + vfree(data); + } + return ret; +} +#else +static int cs35l41_save_calibration(struct cs35l41_hda *cs35l41) +{ + dev_warn(cs35l41->dev, "Calibration not supported without EFI support.\n"); + return 0; +} +#endif + static int cs35l41_init_dsp(struct cs35l41_hda *cs35l41) { const struct firmware *coeff_firmware = NULL; @@ -314,7 +410,12 @@ static int cs35l41_init_dsp(struct cs35l41_hda *cs35l41) ret = cs_dsp_power_up(dsp, wmfw_firmware, wmfw_filename, coeff_firmware, coeff_filename, FW_NAME); + if (ret) + goto err_release; + + ret = cs35l41_save_calibration(cs35l41); +err_release: release_firmware(wmfw_firmware); release_firmware(coeff_firmware); kfree(wmfw_filename); diff --git a/sound/pci/hda/cs35l41_hda.h b/sound/pci/hda/cs35l41_hda.h index 439c4b705328..59a9461d0444 100644 --- a/sound/pci/hda/cs35l41_hda.h +++ b/sound/pci/hda/cs35l41_hda.h @@ -10,6 +10,7 @@ #ifndef __CS35L41_HDA_H__ #define __CS35L41_HDA_H__ +#include #include #include #include @@ -18,6 +19,20 @@ #include #include +struct cs35l41_amp_cal_data { + u32 calTarget[2]; + u32 calTime[2]; + s8 calAmbient; + u8 calStatus; + u16 calR; +} __packed; + +struct cs35l41_amp_efi_data { + u32 size; + u32 count; + struct cs35l41_amp_cal_data data[]; +} __packed; + enum cs35l41_hda_spk_pos { CS35l41_LEFT, CS35l41_RIGHT, -- cgit v1.2.3 From 291e7c220b82b28d4c128dfb2abaa51d29969dd5 Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Thu, 30 Jun 2022 01:23:32 +0100 Subject: ALSA: hda: hda_cs_dsp_ctl: Add fw id strings This will be used to define the firmware names. Signed-off-by: Stefan Binding Signed-off-by: Vitaly Rodionov Link: https://lore.kernel.org/r/20220630002335.366545-12-vitalyr@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_cs_dsp_ctl.c | 8 ++++++++ sound/pci/hda/hda_cs_dsp_ctl.h | 2 ++ 2 files changed, 10 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_cs_dsp_ctl.c b/sound/pci/hda/hda_cs_dsp_ctl.c index 2351476c9ee6..89ee549cb7d5 100644 --- a/sound/pci/hda/hda_cs_dsp_ctl.c +++ b/sound/pci/hda/hda_cs_dsp_ctl.c @@ -27,6 +27,14 @@ static const char * const hda_cs_dsp_fw_text[HDA_CS_DSP_NUM_FW] = { [HDA_CS_DSP_FW_MISC] = "Misc", }; +const char * const hda_cs_dsp_fw_ids[HDA_CS_DSP_NUM_FW] = { + [HDA_CS_DSP_FW_SPK_PROT] = "spk-prot", + [HDA_CS_DSP_FW_SPK_CALI] = "spk-cali", + [HDA_CS_DSP_FW_SPK_DIAG] = "spk-diag", + [HDA_CS_DSP_FW_MISC] = "misc", +}; +EXPORT_SYMBOL_NS_GPL(hda_cs_dsp_fw_ids, SND_HDA_CS_DSP_CONTROLS); + static int hda_cs_dsp_coeff_info(struct snd_kcontrol *kctl, struct snd_ctl_elem_info *uinfo) { struct hda_cs_dsp_coeff_ctl *ctl = (struct hda_cs_dsp_coeff_ctl *)snd_kcontrol_chip(kctl); diff --git a/sound/pci/hda/hda_cs_dsp_ctl.h b/sound/pci/hda/hda_cs_dsp_ctl.h index c65bfd6878fd..4babc69cf2f0 100644 --- a/sound/pci/hda/hda_cs_dsp_ctl.h +++ b/sound/pci/hda/hda_cs_dsp_ctl.h @@ -27,6 +27,8 @@ struct hda_cs_dsp_ctl_info { const char *device_name; }; +extern const char * const hda_cs_dsp_fw_ids[HDA_CS_DSP_NUM_FW]; + int hda_cs_dsp_control_add(struct cs_dsp_coeff_ctl *cs_ctl, struct hda_cs_dsp_ctl_info *info); void hda_cs_dsp_control_remove(struct cs_dsp_coeff_ctl *cs_ctl); int hda_cs_dsp_write_ctl(struct cs_dsp *dsp, const char *name, int type, -- cgit v1.2.3 From 4fa58b1d7ec714581bfb1d12370746d29518cd3a Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Thu, 30 Jun 2022 01:23:33 +0100 Subject: ALSA: hda: cs35l41: Add defaulted values into dsp bypass config sequence The config sequences for running with and without firmware and DSP are different. The original behavior assumed that we would only run without DSP only in the case where firmware load failed. This meant the non-firmware sequence was written with the assumtion that various registers would be set to their default value. However, to support the ability to unload the firmware, the non-firmware register sequence must be updated to update all required registers, including values that would be defaulted, in case the firmware sequence, which could have already run, has changed their value. Signed-off-by: Stefan Binding Signed-off-by: Vitaly Rodionov Link: https://lore.kernel.org/r/20220630002335.366545-13-vitalyr@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l41_hda.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index 75edaffb568a..a02a74f68c2d 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -35,11 +35,24 @@ static const struct reg_sequence cs35l41_hda_config[] = { { CS35L41_PLL_CLK_CTRL, 0x00000430 }, // 3072000Hz, BCLK Input, PLL_REFCLK_EN = 1 + { CS35L41_DSP_CLK_CTRL, 0x00000003 }, // DSP CLK EN { CS35L41_GLOBAL_CLK_CTRL, 0x00000003 }, // GLOBAL_FS = 48 kHz { CS35L41_SP_ENABLES, 0x00010000 }, // ASP_RX1_EN = 1 { CS35L41_SP_RATE_CTRL, 0x00000021 }, // ASP_BCLK_FREQ = 3.072 MHz { CS35L41_SP_FORMAT, 0x20200200 }, // 32 bits RX/TX slots, I2S, clk consumer + { CS35L41_SP_HIZ_CTRL, 0x00000002 }, // Hi-Z unused + { CS35L41_SP_TX_WL, 0x00000018 }, // 24 cycles/slot + { CS35L41_SP_RX_WL, 0x00000018 }, // 24 cycles/slot { CS35L41_DAC_PCM1_SRC, 0x00000008 }, // DACPCM1_SRC = ASPRX1 + { CS35L41_ASP_TX1_SRC, 0x00000018 }, // ASPTX1 SRC = VMON + { CS35L41_ASP_TX2_SRC, 0x00000019 }, // ASPTX2 SRC = IMON + { CS35L41_ASP_TX3_SRC, 0x00000032 }, // ASPTX3 SRC = ERRVOL + { CS35L41_ASP_TX4_SRC, 0x00000033 }, // ASPTX4 SRC = CLASSH_TGT + { CS35L41_DSP1_RX1_SRC, 0x00000008 }, // DSP1RX1 SRC = ASPRX1 + { CS35L41_DSP1_RX2_SRC, 0x00000009 }, // DSP1RX2 SRC = ASPRX2 + { CS35L41_DSP1_RX3_SRC, 0x00000018 }, // DSP1RX3 SRC = VMON + { CS35L41_DSP1_RX4_SRC, 0x00000019 }, // DSP1RX4 SRC = IMON + { CS35L41_DSP1_RX5_SRC, 0x00000020 }, // DSP1RX5 SRC = ERRVOL { CS35L41_AMP_DIG_VOL_CTRL, 0x00000000 }, // AMP_VOL_PCM 0.0 dB { CS35L41_AMP_GAIN_CTRL, 0x00000084 }, // AMP_GAIN_PCM 4.5 dB }; -- cgit v1.2.3 From 47ceabd99a28399f8971f4ca0a37ebc0a21dd2a8 Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Thu, 30 Jun 2022 01:23:34 +0100 Subject: ALSA: hda: cs35l41: Support Firmware switching and reloading This is required to support CS35L41 calibration. By default, speaker protection firmware will be loaded, if available. However, different firmware is required to run the calibration sequence, so it is necessary to add support to be able to unload, switch and reload firmware. This patch adds 2 ALSA Controls for each amp: "DSP1 Firmware Load" "DSP1 Firmware Type" "DSP1 Firmware Load" can be used to unload and load the firmware. "DSP1 Firmware Type" can be used to switch the target firmware to be loaded by "DSP1 Firmware Load" Since loading firmware can add new ALSA controls, it is necessary to ensure the firmware loading is run asynchronously from the ALSA control itself to prevent deadlocks. Note: When switching between firmwares, an ALSA control is only added if it has not previously existed. If it had existed previously, it will be re-enabled instead. Signed-off-by: Stefan Binding Signed-off-by: Vitaly Rodionov Link: https://lore.kernel.org/r/20220630002335.366545-14-vitalyr@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l41_hda.c | 182 +++++++++++++++++++++++++++++++++++++++++--- sound/pci/hda/cs35l41_hda.h | 6 ++ 2 files changed, 178 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index a02a74f68c2d..4252c0ac69b7 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -21,7 +21,6 @@ #define CS35L41_FIRMWARE_ROOT "cirrus/" #define CS35L41_PART "cs35l41" -#define FW_NAME "CSPL" #define HALO_STATE_DSP_CTL_NAME "HALO_STATE" #define HALO_STATE_DSP_CTL_TYPE 5 @@ -92,7 +91,7 @@ static int cs35l41_control_add(struct cs_dsp_coeff_ctl *cs_ctl) struct hda_cs_dsp_ctl_info info; info.device_name = cs35l41->amp_name; - info.fw_type = HDA_CS_DSP_FW_SPK_PROT; + info.fw_type = cs35l41->firmware_type; info.card = cs35l41->codec->card; return hda_cs_dsp_control_add(cs_ctl, &info); @@ -114,20 +113,24 @@ static int cs35l41_request_firmware_file(struct cs35l41_hda *cs35l41, if (spkid > -1 && ssid && amp_name) *filename = kasprintf(GFP_KERNEL, "%s%s-%s-%s-%s-spkid%d-%s.%s", dir, CS35L41_PART, - dsp_name, "spk-prot", ssid, spkid, amp_name, filetype); + dsp_name, hda_cs_dsp_fw_ids[cs35l41->firmware_type], + ssid, spkid, amp_name, filetype); else if (spkid > -1 && ssid) *filename = kasprintf(GFP_KERNEL, "%s%s-%s-%s-%s-spkid%d.%s", dir, CS35L41_PART, - dsp_name, "spk-prot", ssid, spkid, filetype); + dsp_name, hda_cs_dsp_fw_ids[cs35l41->firmware_type], + ssid, spkid, filetype); else if (ssid && amp_name) *filename = kasprintf(GFP_KERNEL, "%s%s-%s-%s-%s-%s.%s", dir, CS35L41_PART, - dsp_name, "spk-prot", ssid, amp_name, - filetype); + dsp_name, hda_cs_dsp_fw_ids[cs35l41->firmware_type], + ssid, amp_name, filetype); else if (ssid) *filename = kasprintf(GFP_KERNEL, "%s%s-%s-%s-%s.%s", dir, CS35L41_PART, - dsp_name, "spk-prot", ssid, filetype); + dsp_name, hda_cs_dsp_fw_ids[cs35l41->firmware_type], + ssid, filetype); else *filename = kasprintf(GFP_KERNEL, "%s%s-%s-%s.%s", dir, CS35L41_PART, - dsp_name, "spk-prot", filetype); + dsp_name, hda_cs_dsp_fw_ids[cs35l41->firmware_type], + filetype); if (*filename == NULL) return -ENOMEM; @@ -422,7 +425,7 @@ static int cs35l41_init_dsp(struct cs35l41_hda *cs35l41) dev_warn(cs35l41->dev, "No Coefficient File available.\n"); ret = cs_dsp_power_up(dsp, wmfw_firmware, wmfw_filename, coeff_firmware, coeff_filename, - FW_NAME); + hda_cs_dsp_fw_ids[cs35l41->firmware_type]); if (ret) goto err_release; @@ -451,6 +454,7 @@ static void cs35l41_remove_dsp(struct cs35l41_hda *cs35l41) { struct cs_dsp *dsp = &cs35l41->cs_dsp; + cancel_work_sync(&cs35l41->fw_load_work); cs35l41_shutdown_dsp(cs35l41); cs_dsp_remove(dsp); cs35l41->halo_initialized = false; @@ -481,6 +485,7 @@ static void cs35l41_hda_playback_hook(struct device *dev, int action) switch (action) { case HDA_GEN_PCM_ACT_OPEN: + cs35l41->playback_started = true; if (cs35l41->firmware_running) { regmap_multi_reg_write(reg, cs35l41_hda_config_dsp, ARRAY_SIZE(cs35l41_hda_config_dsp)); @@ -518,6 +523,7 @@ static void cs35l41_hda_playback_hook(struct device *dev, int action) 0 << CS35L41_VMON_EN_SHIFT | 0 << CS35L41_IMON_EN_SHIFT); } cs35l41_irq_release(cs35l41); + cs35l41->playback_started = false; break; default: dev_warn(cs35l41->dev, "Playback action not supported: %d\n", action); @@ -664,10 +670,160 @@ clean_dsp: return ret; } +static void cs35l41_load_firmware(struct cs35l41_hda *cs35l41, bool load) +{ + pm_runtime_get_sync(cs35l41->dev); + + if (cs35l41->firmware_running && !load) { + dev_dbg(cs35l41->dev, "Unloading Firmware\n"); + cs35l41_shutdown_dsp(cs35l41); + } else if (!cs35l41->firmware_running && load) { + dev_dbg(cs35l41->dev, "Loading Firmware\n"); + cs35l41_smart_amp(cs35l41); + } else { + dev_dbg(cs35l41->dev, "Unable to Load firmware.\n"); + } + + pm_runtime_mark_last_busy(cs35l41->dev); + pm_runtime_put_autosuspend(cs35l41->dev); +} + +static int cs35l41_fw_load_ctl_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct cs35l41_hda *cs35l41 = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = cs35l41->request_fw_load; + return 0; +} + +static void cs35l41_fw_load_work(struct work_struct *work) +{ + struct cs35l41_hda *cs35l41 = container_of(work, struct cs35l41_hda, fw_load_work); + + mutex_lock(&cs35l41->fw_mutex); + + /* Recheck if playback is ongoing, mutex will block playback during firmware loading */ + if (cs35l41->playback_started) + dev_err(cs35l41->dev, "Cannot Load/Unload firmware during Playback\n"); + else + cs35l41_load_firmware(cs35l41, cs35l41->request_fw_load); + + cs35l41->fw_request_ongoing = false; + mutex_unlock(&cs35l41->fw_mutex); +} + +static int cs35l41_fw_load_ctl_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct cs35l41_hda *cs35l41 = snd_kcontrol_chip(kcontrol); + unsigned int ret = 0; + + mutex_lock(&cs35l41->fw_mutex); + + if (cs35l41->request_fw_load == ucontrol->value.integer.value[0]) + goto err; + + if (cs35l41->fw_request_ongoing) { + dev_dbg(cs35l41->dev, "Existing request not complete\n"); + ret = -EBUSY; + goto err; + } + + /* Check if playback is ongoing when initial request is made */ + if (cs35l41->playback_started) { + dev_err(cs35l41->dev, "Cannot Load/Unload firmware during Playback\n"); + ret = -EBUSY; + goto err; + } + + cs35l41->fw_request_ongoing = true; + cs35l41->request_fw_load = ucontrol->value.integer.value[0]; + schedule_work(&cs35l41->fw_load_work); + +err: + mutex_unlock(&cs35l41->fw_mutex); + + return ret; +} + +static int cs35l41_fw_type_ctl_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct cs35l41_hda *cs35l41 = snd_kcontrol_chip(kcontrol); + + ucontrol->value.enumerated.item[0] = cs35l41->firmware_type; + + return 0; +} + +static int cs35l41_fw_type_ctl_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct cs35l41_hda *cs35l41 = snd_kcontrol_chip(kcontrol); + + if (ucontrol->value.enumerated.item[0] < HDA_CS_DSP_NUM_FW) { + cs35l41->firmware_type = ucontrol->value.enumerated.item[0]; + return 0; + } + + return -EINVAL; +} + +static int cs35l41_fw_type_ctl_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +{ + return snd_ctl_enum_info(uinfo, 1, ARRAY_SIZE(hda_cs_dsp_fw_ids), hda_cs_dsp_fw_ids); +} + +static int cs35l41_create_controls(struct cs35l41_hda *cs35l41) +{ + char fw_type_ctl_name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + char fw_load_ctl_name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + struct snd_kcontrol_new fw_type_ctl = { + .name = fw_type_ctl_name, + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .info = cs35l41_fw_type_ctl_info, + .get = cs35l41_fw_type_ctl_get, + .put = cs35l41_fw_type_ctl_put, + }; + struct snd_kcontrol_new fw_load_ctl = { + .name = fw_load_ctl_name, + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .info = snd_ctl_boolean_mono_info, + .get = cs35l41_fw_load_ctl_get, + .put = cs35l41_fw_load_ctl_put, + }; + int ret; + + scnprintf(fw_type_ctl_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "%s DSP1 Firmware Type", + cs35l41->amp_name); + scnprintf(fw_load_ctl_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "%s DSP1 Firmware Load", + cs35l41->amp_name); + + ret = snd_ctl_add(cs35l41->codec->card, snd_ctl_new1(&fw_type_ctl, cs35l41)); + if (ret) { + dev_err(cs35l41->dev, "Failed to add KControl %s = %d\n", fw_type_ctl.name, ret); + return ret; + } + + dev_dbg(cs35l41->dev, "Added Control %s\n", fw_type_ctl.name); + + ret = snd_ctl_add(cs35l41->codec->card, snd_ctl_new1(&fw_load_ctl, cs35l41)); + if (ret) { + dev_err(cs35l41->dev, "Failed to add KControl %s = %d\n", fw_load_ctl.name, ret); + return ret; + } + + dev_dbg(cs35l41->dev, "Added Control %s\n", fw_load_ctl.name); + + return 0; +} + static int cs35l41_hda_bind(struct device *dev, struct device *master, void *master_data) { struct cs35l41_hda *cs35l41 = dev_get_drvdata(dev); struct hda_component *comps = master_data; + int ret = 0; if (!comps || cs35l41->index < 0 || cs35l41->index >= HDA_MAX_COMPONENTS) return -EINVAL; @@ -685,11 +841,16 @@ static int cs35l41_hda_bind(struct device *dev, struct device *master, void *mas cs35l41->codec = comps->codec; strscpy(comps->name, dev_name(dev), sizeof(comps->name)); + cs35l41->firmware_type = HDA_CS_DSP_FW_SPK_PROT; + + cs35l41->request_fw_load = true; mutex_lock(&cs35l41->fw_mutex); if (cs35l41_smart_amp(cs35l41) < 0) dev_warn(cs35l41->dev, "Cannot Run Firmware, reverting to dsp bypass...\n"); mutex_unlock(&cs35l41->fw_mutex); + ret = cs35l41_create_controls(cs35l41); + comps->playback_hook = cs35l41_hda_playback_hook; comps->suspend_hook = cs35l41_hda_suspend_hook; comps->resume_hook = cs35l41_hda_resume_hook; @@ -697,7 +858,7 @@ static int cs35l41_hda_bind(struct device *dev, struct device *master, void *mas pm_runtime_mark_last_busy(dev); pm_runtime_put_autosuspend(dev); - return 0; + return ret; } static void cs35l41_hda_unbind(struct device *dev, struct device *master, void *master_data) @@ -1206,6 +1367,7 @@ int cs35l41_hda_probe(struct device *dev, const char *device_name, int id, int i if (ret) goto err; + INIT_WORK(&cs35l41->fw_load_work, cs35l41_fw_load_work); mutex_init(&cs35l41->fw_mutex); pm_runtime_set_autosuspend_delay(cs35l41->dev, 3000); diff --git a/sound/pci/hda/cs35l41_hda.h b/sound/pci/hda/cs35l41_hda.h index 59a9461d0444..bdb35f3be68a 100644 --- a/sound/pci/hda/cs35l41_hda.h +++ b/sound/pci/hda/cs35l41_hda.h @@ -58,11 +58,17 @@ struct cs35l41_hda { unsigned volatile long irq_errors; const char *amp_name; const char *acpi_subsystem_id; + int firmware_type; int speaker_id; struct mutex fw_mutex; + struct work_struct fw_load_work; + struct regmap_irq_chip_data *irq_data; bool firmware_running; + bool request_fw_load; + bool fw_request_ongoing; bool halo_initialized; + bool playback_started; struct cs_dsp cs_dsp; }; -- cgit v1.2.3 From 622f21994506e1dac7b8e4e362c8951426e032c5 Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Thu, 30 Jun 2022 01:23:35 +0100 Subject: ALSA: hda: cs35l41: Add module parameter to control firmware load By default, the driver will automatically load DSP firmware for the amps, if available. Adding this option allows the autoload to be optional, which allows for different configurations. Signed-off-by: Stefan Binding Signed-off-by: Vitaly Rodionov Link: https://lore.kernel.org/r/20220630002335.366545-15-vitalyr@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l41_hda.c | 21 ++++++++++++++++----- 1 file changed, 16 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index 4252c0ac69b7..28798d5c1cf1 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -8,6 +8,7 @@ #include #include +#include #include #include #include @@ -32,6 +33,11 @@ #define CAL_DSP_CTL_TYPE 5 #define CAL_DSP_CTL_ALG 205 +static bool firmware_autostart = 1; +module_param(firmware_autostart, bool, 0444); +MODULE_PARM_DESC(firmware_autostart, "Allow automatic firmware download on boot" + "(0=Disable, 1=Enable) (default=1); "); + static const struct reg_sequence cs35l41_hda_config[] = { { CS35L41_PLL_CLK_CTRL, 0x00000430 }, // 3072000Hz, BCLK Input, PLL_REFCLK_EN = 1 { CS35L41_DSP_CLK_CTRL, 0x00000003 }, // DSP CLK EN @@ -843,11 +849,16 @@ static int cs35l41_hda_bind(struct device *dev, struct device *master, void *mas cs35l41->firmware_type = HDA_CS_DSP_FW_SPK_PROT; - cs35l41->request_fw_load = true; - mutex_lock(&cs35l41->fw_mutex); - if (cs35l41_smart_amp(cs35l41) < 0) - dev_warn(cs35l41->dev, "Cannot Run Firmware, reverting to dsp bypass...\n"); - mutex_unlock(&cs35l41->fw_mutex); + if (firmware_autostart) { + dev_dbg(cs35l41->dev, "Firmware Autostart.\n"); + cs35l41->request_fw_load = true; + mutex_lock(&cs35l41->fw_mutex); + if (cs35l41_smart_amp(cs35l41) < 0) + dev_warn(cs35l41->dev, "Cannot Run Firmware, reverting to dsp bypass...\n"); + mutex_unlock(&cs35l41->fw_mutex); + } else { + dev_dbg(cs35l41->dev, "Firmware Autostart is disabled.\n"); + } ret = cs35l41_create_controls(cs35l41); -- cgit v1.2.3 From e7255c00b10e5e570dd8eb24f59e964eeec38d3b Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Wed, 6 Jul 2022 14:02:26 +0200 Subject: ALSA: hda: Skip event processing for unregistered codecs When codec is unbound but not yet removed, in the eyes of snd_hdac_bus_process_unsol_events() it is still a valid target to delegate work to. Such behaviour may lead to use-after-free errors. Address by verifying if codec is actually registered. Signed-off-by: Cezary Rojewski Link: https://lore.kernel.org/r/20220706120230.427296-6-cezary.rojewski@intel.com Signed-off-by: Takashi Iwai --- sound/hda/hdac_bus.c | 2 +- sound/pci/hda/hda_codec.c | 10 +++++----- sound/soc/codecs/hda.c | 4 ++-- 3 files changed, 8 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/hda/hdac_bus.c b/sound/hda/hdac_bus.c index 71db8592b33d..d497414a5538 100644 --- a/sound/hda/hdac_bus.c +++ b/sound/hda/hdac_bus.c @@ -183,7 +183,7 @@ static void snd_hdac_bus_process_unsol_events(struct work_struct *work) if (!(caddr & (1 << 4))) /* no unsolicited event? */ continue; codec = bus->caddr_tbl[caddr & 0x0f]; - if (!codec || !codec->dev.driver) + if (!codec || !codec->registered) continue; spin_unlock_irq(&bus->reg_lock); drv = drv_to_hdac_driver(codec->dev.driver); diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index b1921f920513..7be74227bf19 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -772,11 +772,11 @@ static void codec_release_pcms(struct hda_codec *codec) */ void snd_hda_codec_cleanup_for_unbind(struct hda_codec *codec) { - if (codec->registered) { + if (codec->core.registered) { /* pm_runtime_put() is called in snd_hdac_device_exit() */ pm_runtime_get_noresume(hda_codec_dev(codec)); pm_runtime_disable(hda_codec_dev(codec)); - codec->registered = 0; + codec->core.registered = 0; } snd_hda_codec_disconnect_pcms(codec); @@ -825,14 +825,14 @@ void snd_hda_codec_display_power(struct hda_codec *codec, bool enable) */ void snd_hda_codec_register(struct hda_codec *codec) { - if (codec->registered) + if (codec->core.registered) return; if (device_is_registered(hda_codec_dev(codec))) { snd_hda_codec_display_power(codec, true); pm_runtime_enable(hda_codec_dev(codec)); /* it was powered up in snd_hda_codec_new(), now all done */ snd_hda_power_down(codec); - codec->registered = 1; + codec->core.registered = 1; } } EXPORT_SYMBOL_GPL(snd_hda_codec_register); @@ -3047,7 +3047,7 @@ void snd_hda_codec_shutdown(struct hda_codec *codec) struct hda_pcm *cpcm; /* Skip the shutdown if codec is not registered */ - if (!codec->registered) + if (!codec->core.registered) return; cancel_delayed_work_sync(&codec->jackpoll_work); diff --git a/sound/soc/codecs/hda.c b/sound/soc/codecs/hda.c index edcb8bc6806b..ad20a3dff9b7 100644 --- a/sound/soc/codecs/hda.c +++ b/sound/soc/codecs/hda.c @@ -274,7 +274,7 @@ static void hda_codec_remove(struct snd_soc_component *component) struct hdac_device *hdev = &codec->core; struct hdac_bus *bus = hdev->bus; struct hdac_ext_link *hlink; - bool was_registered = codec->registered; + bool was_registered = codec->core.registered; /* Don't allow any more runtime suspends */ pm_runtime_forbid(&hdev->dev); @@ -376,7 +376,7 @@ static int hda_hdev_detach(struct hdac_device *hdev) { struct hda_codec *codec = dev_to_hda_codec(&hdev->dev); - if (codec->registered) + if (codec->core.registered) cancel_delayed_work_sync(&codec->jackpoll_work); snd_soc_unregister_component(&hdev->dev); -- cgit v1.2.3 From d59d2277febbad93e42137a50673dd1c16199813 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 15 Jul 2022 20:24:27 +0200 Subject: Revert "ALSA: hda: cs35l41: Allow compilation test on non-ACPI configurations" Since the recent change in CS35L41 codec requires the reference of acpi_dev handle, the current Kconfig may lead to a build breakage. Revert the Kconfig change and re-introduce the hard dependency on CONFIG_ACPI again as a temporary workaround. Fixes: eef375960210 ("ALSA: hda: cs35l41: Support reading subsystem id from ACPI") Link: https://lore.kernel.org/r/20220715182427.18891-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 44c33bc0740e..a8e8cf98befa 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -103,7 +103,7 @@ config SND_HDA_CS_DSP_CONTROLS config SND_HDA_SCODEC_CS35L41_I2C tristate "Build CS35L41 HD-audio side codec support for I2C Bus" depends on I2C - depends on ACPI || COMPILE_TEST + depends on ACPI depends on SND_SOC select SND_SOC_CS35L41_LIB select SND_HDA_SCODEC_CS35L41 @@ -118,7 +118,7 @@ comment "Set to Y if you want auto-loading the side codec driver" config SND_HDA_SCODEC_CS35L41_SPI tristate "Build CS35L41 HD-audio codec support for SPI Bus" depends on SPI_MASTER - depends on ACPI || COMPILE_TEST + depends on ACPI depends on SND_SOC select SND_SOC_CS35L41_LIB select SND_HDA_SCODEC_CS35L41 -- cgit v1.2.3 From 53f07e9b010b966017e32be1ca1bbcbcc4cee73d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 15 Jul 2022 20:29:03 +0200 Subject: Revert "ALSA: hda: Fix page fault in snd_hda_codec_shutdown()" This reverts commit 980b3a8790b402e959a6d773b38b771019682be1. The commit didn't consider the fact that ASoC hdac-hda driver initializes the HD-audio stuff without calling snd_hda_codec_device_init(). Hence this caused a regression leading to Oops. Revert the commit to restore the behavior. Fixes: 980b3a8790b4 ("ALSA: hda: Fix page fault in snd_hda_codec_shutdown()") Link: https://lore.kernel.org/r/3c40df55-3aee-1e08-493b-7b30cd84dc00@linux.intel.com Link: https://lore.kernel.org/r/20220715182903.19594-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 41 +++++++++++++++++++++-------------------- 1 file changed, 21 insertions(+), 20 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 7be74227bf19..7b2e62fa82d5 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -931,28 +931,8 @@ snd_hda_codec_device_init(struct hda_bus *bus, unsigned int codec_addr, } codec->bus = bus; - codec->depop_delay = -1; - codec->fixup_id = HDA_FIXUP_ID_NOT_SET; - codec->core.dev.release = snd_hda_codec_dev_release; - codec->core.exec_verb = codec_exec_verb; codec->core.type = HDA_DEV_LEGACY; - mutex_init(&codec->spdif_mutex); - mutex_init(&codec->control_mutex); - snd_array_init(&codec->mixers, sizeof(struct hda_nid_item), 32); - snd_array_init(&codec->nids, sizeof(struct hda_nid_item), 32); - snd_array_init(&codec->init_pins, sizeof(struct hda_pincfg), 16); - snd_array_init(&codec->driver_pins, sizeof(struct hda_pincfg), 16); - snd_array_init(&codec->cvt_setups, sizeof(struct hda_cvt_setup), 8); - snd_array_init(&codec->spdif_out, sizeof(struct hda_spdif_out), 16); - snd_array_init(&codec->jacktbl, sizeof(struct hda_jack_tbl), 16); - snd_array_init(&codec->verbs, sizeof(struct hda_verb *), 8); - INIT_LIST_HEAD(&codec->conn_list); - INIT_LIST_HEAD(&codec->pcm_list_head); - INIT_DELAYED_WORK(&codec->jackpoll_work, hda_jackpoll_work); - refcount_set(&codec->pcm_ref, 1); - init_waitqueue_head(&codec->remove_sleep); - return codec; } EXPORT_SYMBOL_GPL(snd_hda_codec_device_init); @@ -1005,8 +985,29 @@ int snd_hda_codec_device_new(struct hda_bus *bus, struct snd_card *card, if (snd_BUG_ON(codec_addr > HDA_MAX_CODEC_ADDRESS)) return -EINVAL; + codec->core.dev.release = snd_hda_codec_dev_release; + codec->core.exec_verb = codec_exec_verb; + codec->card = card; codec->addr = codec_addr; + mutex_init(&codec->spdif_mutex); + mutex_init(&codec->control_mutex); + snd_array_init(&codec->mixers, sizeof(struct hda_nid_item), 32); + snd_array_init(&codec->nids, sizeof(struct hda_nid_item), 32); + snd_array_init(&codec->init_pins, sizeof(struct hda_pincfg), 16); + snd_array_init(&codec->driver_pins, sizeof(struct hda_pincfg), 16); + snd_array_init(&codec->cvt_setups, sizeof(struct hda_cvt_setup), 8); + snd_array_init(&codec->spdif_out, sizeof(struct hda_spdif_out), 16); + snd_array_init(&codec->jacktbl, sizeof(struct hda_jack_tbl), 16); + snd_array_init(&codec->verbs, sizeof(struct hda_verb *), 8); + INIT_LIST_HEAD(&codec->conn_list); + INIT_LIST_HEAD(&codec->pcm_list_head); + refcount_set(&codec->pcm_ref, 1); + init_waitqueue_head(&codec->remove_sleep); + + INIT_DELAYED_WORK(&codec->jackpoll_work, hda_jackpoll_work); + codec->depop_delay = -1; + codec->fixup_id = HDA_FIXUP_ID_NOT_SET; #ifdef CONFIG_PM codec->power_jiffies = jiffies; -- cgit v1.2.3 From 48d8bd769fb7e2272beb92a75518d4d6f98b4ccc Mon Sep 17 00:00:00 2001 From: shaomin Deng Date: Thu, 21 Jul 2022 11:05:28 -0400 Subject: ALSA: emu10k1: Fix typo in comments Remove the rebundant word "in" in comments. Signed-off-by: shaomin Deng Link: https://lore.kernel.org/r/20220721150528.22099-1-dengshaomin@cdjrlc.com Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/memory.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c index 9d26535f3fa3..edb3f1763719 100644 --- a/sound/pci/emu10k1/memory.c +++ b/sound/pci/emu10k1/memory.c @@ -324,7 +324,7 @@ snd_emu10k1_alloc_pages(struct snd_emu10k1 *emu, struct snd_pcm_substream *subst return NULL; } /* fill buffer addresses but pointers are not stored so that - * snd_free_pci_page() is not called in in synth_free() + * snd_free_pci_page() is not called in synth_free() */ idx = 0; for (page = blk->first_page; page <= blk->last_page; page++, idx++) { -- cgit v1.2.3 From 84f2a3c182d545261c36f94510134f5e9fb918f5 Mon Sep 17 00:00:00 2001 From: shaomin Deng Date: Thu, 21 Jul 2022 11:55:17 -0400 Subject: ALSA: asihpi: Fix typo in comments Delete the repeated word "in" in comments. Signed-off-by: shaomin Deng Link: https://lore.kernel.org/r/20220721155517.2438-1-dengshaomin@cdjrlc.com Signed-off-by: Takashi Iwai --- sound/pci/asihpi/hpi6205.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/asihpi/hpi6205.c b/sound/pci/asihpi/hpi6205.c index 3d6914c64c4a..27e11b5f70b9 100644 --- a/sound/pci/asihpi/hpi6205.c +++ b/sound/pci/asihpi/hpi6205.c @@ -445,7 +445,7 @@ void HPI_6205(struct hpi_message *phm, struct hpi_response *phr) /* SUBSYSTEM */ /** Create an adapter object and initialise it based on resource information - * passed in in the message + * passed in the message * *** NOTE - you cannot use this function AND the FindAdapters function at the * same time, the application must use only one of them to get the adapters *** */ -- cgit v1.2.3 From e086c37f876fd1f551e2b4f9be97d4a1923cd219 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 22 Jul 2022 16:39:48 +0200 Subject: ALSA: usb-audio: Add quirk for Behringer UMC202HD Just like other Behringer models, UMC202HD (USB ID 1397:0507) requires the quirk for the stable streaming, too. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=215934 Cc: Link: https://lore.kernel.org/r/20220722143948.29804-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 968d90caeefa..168fd802d70b 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1843,6 +1843,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_SHARE_MEDIA_DEVICE | QUIRK_FLAG_ALIGN_TRANSFER), DEVICE_FLG(0x1395, 0x740a, /* Sennheiser DECT */ QUIRK_FLAG_GET_SAMPLE_RATE), + DEVICE_FLG(0x1397, 0x0507, /* Behringer UMC202HD */ + QUIRK_FLAG_PLAYBACK_FIRST | QUIRK_FLAG_GENERIC_IMPLICIT_FB), DEVICE_FLG(0x1397, 0x0508, /* Behringer UMC204HD */ QUIRK_FLAG_PLAYBACK_FIRST | QUIRK_FLAG_GENERIC_IMPLICIT_FB), DEVICE_FLG(0x1397, 0x0509, /* Behringer UMC404HD */ -- cgit v1.2.3 From ccc86a0a02139321df943313a99efa11745bd273 Mon Sep 17 00:00:00 2001 From: wangjianli Date: Sun, 24 Jul 2022 15:14:13 +0800 Subject: ALSA: asihpi: fix repeated words in comments Delete the redundant word 'in'. Signed-off-by: wangjianli Link: https://lore.kernel.org/r/20220724071413.10085-1-wangjianli@cdjrlc.com Signed-off-by: Takashi Iwai --- sound/pci/asihpi/hpi6000.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/asihpi/hpi6000.c b/sound/pci/asihpi/hpi6000.c index aa4d06353126..88d902997b74 100644 --- a/sound/pci/asihpi/hpi6000.c +++ b/sound/pci/asihpi/hpi6000.c @@ -388,7 +388,7 @@ void HPI_6000(struct hpi_message *phm, struct hpi_response *phr) /* SUBSYSTEM */ /* create an adapter object and initialise it based on resource information - * passed in in the message + * passed in the message * NOTE - you cannot use this function AND the FindAdapters function at the * same time, the application must use only one of them to get the adapters */ -- cgit v1.2.3 From 614b9febdc144c56b087dacc7c1268c25ef6d684 Mon Sep 17 00:00:00 2001 From: wangjianli Date: Sun, 24 Jul 2022 15:16:44 +0800 Subject: ALSA: usb/6fire: fix repeated words in comments Delete the redundant word 'in'. Signed-off-by: wangjianli Link: https://lore.kernel.org/r/20220724071644.10630-1-wangjianli@cdjrlc.com Signed-off-by: Takashi Iwai --- sound/usb/6fire/pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/6fire/pcm.c b/sound/usb/6fire/pcm.c index 7168f1c6a37a..32c39d8bd2e5 100644 --- a/sound/usb/6fire/pcm.c +++ b/sound/usb/6fire/pcm.c @@ -175,7 +175,7 @@ static int usb6fire_pcm_stream_start(struct pcm_runtime *rt) } } - /* wait for first out urb to return (sent in in urb handler) */ + /* wait for first out urb to return (sent in urb handler) */ wait_event_timeout(rt->stream_wait_queue, rt->stream_wait_cond, HZ); if (rt->stream_wait_cond) -- cgit v1.2.3 From 4e3b86509f9204f65842561ad8d37cf83c297550 Mon Sep 17 00:00:00 2001 From: wangjianli Date: Sun, 24 Jul 2022 15:18:29 +0800 Subject: ALSA: hiface: fix repeated words in comments Delete the redundant word 'in'. Signed-off-by: wangjianli Link: https://lore.kernel.org/r/20220724071829.11117-1-wangjianli@cdjrlc.com Signed-off-by: Takashi Iwai --- sound/usb/hiface/pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/hiface/pcm.c b/sound/usb/hiface/pcm.c index 71f17f02f341..cf650fab54d7 100644 --- a/sound/usb/hiface/pcm.c +++ b/sound/usb/hiface/pcm.c @@ -225,7 +225,7 @@ static int hiface_pcm_stream_start(struct pcm_runtime *rt) } } - /* wait for first out urb to return (sent in in urb handler) */ + /* wait for first out urb to return (sent in urb handler) */ wait_event_timeout(rt->stream_wait_queue, rt->stream_wait_cond, HZ); if (rt->stream_wait_cond) { -- cgit v1.2.3 From 26ae150bbb6d19767f10800e17ad0fd81f3da67e Mon Sep 17 00:00:00 2001 From: Ren Zhijie Date: Mon, 25 Jul 2022 10:36:11 +0800 Subject: ALSA: hda: cs35l41: Fix build error unused-function MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit If CONFIG_PM_SLEEP is not set, make ARCH=x86_64 CROSS_COMPILE=x86_64-linux-gnu-, will be failed, like this: sound/pci/hda/cs35l41_hda.c:583:12: error: ‘cs35l41_runtime_resume’ defined but not used [-Werror=unused-function] static int cs35l41_runtime_resume(struct device *dev) ^~~~~~~~~~~~~~~~~~~~~~ sound/pci/hda/cs35l41_hda.c:565:12: error: ‘cs35l41_runtime_suspend’ defined but not used [-Werror=unused-function] static int cs35l41_runtime_suspend(struct device *dev) ^~~~~~~~~~~~~~~~~~~~~~~ cc1: all warnings being treated as errors make[3]: *** [sound/pci/hda/cs35l41_hda.o] Error 1 commit 1a3c7bb08826 ("PM: core: Add new *_PM_OPS macros, deprecate old ones"), add new marco RUNTIME_PM_OPS to fix this unused-function problem. Fixes: 1873ebd30cc8 ("ALSA: hda: cs35l41: Support Hibernation during Suspend") Signed-off-by: Ren Zhijie Link: https://lore.kernel.org/r/20220725023611.57055-1-renzhijie2@huawei.com Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l41_hda.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index 28798d5c1cf1..93cf039abb02 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -1439,7 +1439,7 @@ void cs35l41_hda_remove(struct device *dev) EXPORT_SYMBOL_NS_GPL(cs35l41_hda_remove, SND_HDA_SCODEC_CS35L41); const struct dev_pm_ops cs35l41_hda_pm_ops = { - SET_RUNTIME_PM_OPS(cs35l41_runtime_suspend, cs35l41_runtime_resume, NULL) + RUNTIME_PM_OPS(cs35l41_runtime_suspend, cs35l41_runtime_resume, NULL) }; EXPORT_SYMBOL_NS_GPL(cs35l41_hda_pm_ops, SND_HDA_SCODEC_CS35L41); -- cgit v1.2.3 From c578d5da10dc429c6676ab09f3fec0b79b31633a Mon Sep 17 00:00:00 2001 From: Kai-Heng Feng Date: Tue, 19 Jul 2022 22:20:14 +0800 Subject: ALSA: hda/realtek: Enable speaker and mute LEDs for HP laptops Two more HP laptops that use cs35l41 AMP for speaker and GPIO for mute LEDs. So use the existing quirk to enable them accordingly. [ Sort the entries at the SSID order by tiwai ] Signed-off-by: Kai-Heng Feng Reviewed-by: Lucas Tanure Link: https://lore.kernel.org/r/20220719142015.244426-1-kai.heng.feng@canonical.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0e340c0934db..06bb55399564 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9138,6 +9138,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8aa3, "HP ProBook 450 G9 (MB 8AA1)", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8aa8, "HP EliteBook 640 G9 (MB 8AA6)", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8aab, "HP EliteBook 650 G9 (MB 8AA9)", ALC236_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8ad1, "HP EliteBook 840 14 inch G9 Notebook PC", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8ad2, "HP EliteBook 860 16 inch G9 Notebook PC", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), -- cgit v1.2.3 From f81ee579c0898201eb2b5105718bedf34c0401f9 Mon Sep 17 00:00:00 2001 From: Lucas Tanure Date: Wed, 27 Jul 2022 10:59:21 +0100 Subject: ALSA: hda: cs35l41: Use the CS35L41 HDA internal define Follow GPIO1 pattern, use cs35l41 HDA internal define for IRQ and then translate to ASoC cs35l41 define. Signed-off-by: Lucas Tanure Link: https://lore.kernel.org/r/20220727095924.80884-2-tanureal@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l41_hda.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index 93cf039abb02..9af5e2e9be55 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -1014,6 +1014,7 @@ static int cs35l41_hda_apply_properties(struct cs35l41_hda *cs35l41) break; case CS35L41_INTERRUPT: using_irq = true; + hw_cfg->gpio2.func = CS35L41_GPIO2_INT_OPEN_DRAIN; break; default: dev_err(cs35l41->dev, "Invalid GPIO2 function %d\n", hw_cfg->gpio2.func); @@ -1273,7 +1274,7 @@ no_acpi_dsd: cs35l41->reset_gpio = gpiod_get_index(physdev, NULL, 0, GPIOD_OUT_HIGH); cs35l41->hw_cfg.bst_type = CS35L41_EXT_BOOST_NO_VSPK_SWITCH; cs35l41->speaker_id = cs35l41_get_speaker_id(physdev, 0, 0, 2); - hw_cfg->gpio2.func = CS35L41_GPIO2_INT_OPEN_DRAIN; + hw_cfg->gpio2.func = CS35L41_INTERRUPT; hw_cfg->gpio2.valid = true; cs35l41->hw_cfg.valid = true; put_device(physdev); -- cgit v1.2.3 From 1e24881d8b2a7c198a67fe9e5179e9efb2140df7 Mon Sep 17 00:00:00 2001 From: Lucas Tanure Date: Wed, 27 Jul 2022 10:59:22 +0100 Subject: ALSA: hda: cs35l41: Support CLSA0101 Add support for Intel version of Legion 7 laptop. Signed-off-by: Lucas Tanure Link: https://lore.kernel.org/r/20220727095924.80884-3-tanureal@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l41_hda.c | 66 +++++++++++++++++++++++++---------------- sound/pci/hda/cs35l41_hda_i2c.c | 3 ++ sound/pci/hda/patch_realtek.c | 12 ++++++++ 3 files changed, 55 insertions(+), 26 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index 9af5e2e9be55..129bffb431c2 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -1133,6 +1133,45 @@ static int cs35l41_get_speaker_id(struct device *dev, int amp_index, return speaker_id; } +/* + * Device CLSA010(0/1) doesn't have _DSD so a gpiod_get by the label reset won't work. + * And devices created by serial-multi-instantiate don't have their device struct + * pointing to the correct fwnode, so acpi_dev must be used here. + * And devm functions expect that the device requesting the resource has the correct + * fwnode. + */ +static int cs35l41_no_acpi_dsd(struct cs35l41_hda *cs35l41, struct device *physdev, int id, + const char *hid) +{ + struct cs35l41_hw_cfg *hw_cfg = &cs35l41->hw_cfg; + + /* check I2C address to assign the index */ + cs35l41->index = id == 0x40 ? 0 : 1; + cs35l41->channel_index = 0; + cs35l41->reset_gpio = gpiod_get_index(physdev, NULL, 0, GPIOD_OUT_HIGH); + cs35l41->speaker_id = cs35l41_get_speaker_id(physdev, 0, 0, 2); + hw_cfg->spk_pos = cs35l41->index; + hw_cfg->gpio2.func = CS35L41_INTERRUPT; + hw_cfg->gpio2.valid = true; + hw_cfg->valid = true; + put_device(physdev); + + if (strncmp(hid, "CLSA0100", 8) == 0) { + hw_cfg->bst_type = CS35L41_EXT_BOOST_NO_VSPK_SWITCH; + } else if (strncmp(hid, "CLSA0101", 8) == 0) { + hw_cfg->bst_type = CS35L41_EXT_BOOST; + hw_cfg->gpio1.func = CS35l41_VSPK_SWITCH; + hw_cfg->gpio1.valid = true; + } else { + hw_cfg->valid = false; + hw_cfg->gpio1.valid = false; + hw_cfg->gpio2.valid = false; + return -EINVAL; + } + + return 0; +} + static int cs35l41_hda_read_acpi(struct cs35l41_hda *cs35l41, const char *hid, int id) { struct cs35l41_hw_cfg *hw_cfg = &cs35l41->hw_cfg; @@ -1161,7 +1200,7 @@ static int cs35l41_hda_read_acpi(struct cs35l41_hda *cs35l41, const char *hid, i property = "cirrus,dev-index"; ret = device_property_count_u32(physdev, property); if (ret <= 0) - goto no_acpi_dsd; + return cs35l41_no_acpi_dsd(cs35l41, physdev, id, hid); if (ret > ARRAY_SIZE(values)) { ret = -EINVAL; @@ -1255,31 +1294,6 @@ err: dev_err(cs35l41->dev, "Failed property %s: %d\n", property, ret); return ret; - -no_acpi_dsd: - /* - * Device CLSA0100 doesn't have _DSD so a gpiod_get by the label reset won't work. - * And devices created by serial-multi-instantiate don't have their device struct - * pointing to the correct fwnode, so acpi_dev must be used here. - * And devm functions expect that the device requesting the resource has the correct - * fwnode. - */ - if (strncmp(hid, "CLSA0100", 8) != 0) - return -EINVAL; - - /* check I2C address to assign the index */ - cs35l41->index = id == 0x40 ? 0 : 1; - cs35l41->hw_cfg.spk_pos = cs35l41->index; - cs35l41->channel_index = 0; - cs35l41->reset_gpio = gpiod_get_index(physdev, NULL, 0, GPIOD_OUT_HIGH); - cs35l41->hw_cfg.bst_type = CS35L41_EXT_BOOST_NO_VSPK_SWITCH; - cs35l41->speaker_id = cs35l41_get_speaker_id(physdev, 0, 0, 2); - hw_cfg->gpio2.func = CS35L41_INTERRUPT; - hw_cfg->gpio2.valid = true; - cs35l41->hw_cfg.valid = true; - put_device(physdev); - - return 0; } int cs35l41_hda_probe(struct device *dev, const char *device_name, int id, int irq, diff --git a/sound/pci/hda/cs35l41_hda_i2c.c b/sound/pci/hda/cs35l41_hda_i2c.c index 9c08fa08c421..5baacfde4f16 100644 --- a/sound/pci/hda/cs35l41_hda_i2c.c +++ b/sound/pci/hda/cs35l41_hda_i2c.c @@ -22,6 +22,8 @@ static int cs35l41_hda_i2c_probe(struct i2c_client *clt, const struct i2c_device */ if (strstr(dev_name(&clt->dev), "CLSA0100")) device_name = "CLSA0100"; + else if (strstr(dev_name(&clt->dev), "CLSA0101")) + device_name = "CLSA0101"; else if (strstr(dev_name(&clt->dev), "CSC3551")) device_name = "CSC3551"; else @@ -45,6 +47,7 @@ static const struct i2c_device_id cs35l41_hda_i2c_id[] = { static const struct acpi_device_id cs35l41_acpi_hda_match[] = { {"CLSA0100", 0 }, + {"CLSA0101", 0 }, {"CSC3551", 0 }, {} }; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 06bb55399564..e1fbda215975 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6710,6 +6710,12 @@ static void alc287_fixup_legion_16achg6_speakers(struct hda_codec *cdc, const st cs35l41_generic_fixup(cdc, action, "i2c", "CLSA0100", 2); } +static void alc287_fixup_legion_16ithg6_speakers(struct hda_codec *cdc, const struct hda_fixup *fix, + int action) +{ + cs35l41_generic_fixup(cdc, action, "i2c", "CLSA0101", 2); +} + /* for alc295_fixup_hp_top_speakers */ #include "hp_x360_helper.c" @@ -7047,6 +7053,7 @@ enum { ALC245_FIXUP_CS35L41_SPI_4_HP_GPIO_LED, ALC285_FIXUP_HP_SPEAKERS_MICMUTE_LED, ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE, + ALC287_FIXUP_LEGION_16ITHG6, }; /* A special fixup for Lenovo C940 and Yoga Duet 7; @@ -8889,6 +8896,10 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC }, + [ALC287_FIXUP_LEGION_16ITHG6] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc287_fixup_legion_16ithg6_speakers, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -9355,6 +9366,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x384a, "Lenovo Yoga 7 15ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3852, "Lenovo Yoga 7 14ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3853, "Lenovo Yoga 7 15ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), + SND_PCI_QUIRK(0x17aa, 0x3855, "Legion 7 16ITHG6", ALC287_FIXUP_LEGION_16ITHG6), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo B50-70", ALC269_FIXUP_DMIC_THINKPAD_ACPI), -- cgit v1.2.3 From ef34a0ae7a2654bc9e58675e36898217fb2799d8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 28 Jul 2022 14:59:42 +0200 Subject: ALSA: core: Add async signal helpers Currently the call of kill_fasync() from an interrupt handler might lead to potential spin deadlocks, as spotted by syzkaller. Unfortunately, it's not so trivial to fix this lock chain as it's involved with the tasklist_lock that is touched in allover places. As a temporary workaround, this patch provides the way to defer the async signal notification in a work. The new helper functions, snd_fasync_helper() and snd_kill_faync() are replacements for fasync_helper() and kill_fasync(), respectively. In addition, snd_fasync_free() needs to be called at the destructor of the relevant file object. Link: https://lore.kernel.org/r/20220728125945.29533-2-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/misc.c | 94 +++++++++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 94 insertions(+) (limited to 'sound') diff --git a/sound/core/misc.c b/sound/core/misc.c index 50e4aaa6270d..d32a19976a2b 100644 --- a/sound/core/misc.c +++ b/sound/core/misc.c @@ -10,6 +10,7 @@ #include #include #include +#include #include #ifdef CONFIG_SND_DEBUG @@ -145,3 +146,96 @@ snd_pci_quirk_lookup(struct pci_dev *pci, const struct snd_pci_quirk *list) } EXPORT_SYMBOL(snd_pci_quirk_lookup); #endif + +/* + * Deferred async signal helpers + * + * Below are a few helper functions to wrap the async signal handling + * in the deferred work. The main purpose is to avoid the messy deadlock + * around tasklist_lock and co at the kill_fasync() invocation. + * fasync_helper() and kill_fasync() are replaced with snd_fasync_helper() + * and snd_kill_fasync(), respectively. In addition, snd_fasync_free() has + * to be called at releasing the relevant file object. + */ +struct snd_fasync { + struct fasync_struct *fasync; + int signal; + int poll; + int on; + struct list_head list; +}; + +static DEFINE_SPINLOCK(snd_fasync_lock); +static LIST_HEAD(snd_fasync_list); + +static void snd_fasync_work_fn(struct work_struct *work) +{ + struct snd_fasync *fasync; + + spin_lock_irq(&snd_fasync_lock); + while (!list_empty(&snd_fasync_list)) { + fasync = list_first_entry(&snd_fasync_list, struct snd_fasync, list); + list_del_init(&fasync->list); + spin_unlock_irq(&snd_fasync_lock); + if (fasync->on) + kill_fasync(&fasync->fasync, fasync->signal, fasync->poll); + spin_lock_irq(&snd_fasync_lock); + } + spin_unlock_irq(&snd_fasync_lock); +} + +static DECLARE_WORK(snd_fasync_work, snd_fasync_work_fn); + +int snd_fasync_helper(int fd, struct file *file, int on, + struct snd_fasync **fasyncp) +{ + struct snd_fasync *fasync = NULL; + + if (on) { + fasync = kzalloc(sizeof(*fasync), GFP_KERNEL); + if (!fasync) + return -ENOMEM; + INIT_LIST_HEAD(&fasync->list); + } + + spin_lock_irq(&snd_fasync_lock); + if (*fasyncp) { + kfree(fasync); + fasync = *fasyncp; + } else { + if (!fasync) { + spin_unlock_irq(&snd_fasync_lock); + return 0; + } + *fasyncp = fasync; + } + fasync->on = on; + spin_unlock_irq(&snd_fasync_lock); + return fasync_helper(fd, file, on, &fasync->fasync); +} +EXPORT_SYMBOL_GPL(snd_fasync_helper); + +void snd_kill_fasync(struct snd_fasync *fasync, int signal, int poll) +{ + unsigned long flags; + + if (!fasync || !fasync->on) + return; + spin_lock_irqsave(&snd_fasync_lock, flags); + fasync->signal = signal; + fasync->poll = poll; + list_move(&fasync->list, &snd_fasync_list); + schedule_work(&snd_fasync_work); + spin_unlock_irqrestore(&snd_fasync_lock, flags); +} +EXPORT_SYMBOL_GPL(snd_kill_fasync); + +void snd_fasync_free(struct snd_fasync *fasync) +{ + if (!fasync) + return; + fasync->on = 0; + flush_work(&snd_fasync_work); + kfree(fasync); +} +EXPORT_SYMBOL_GPL(snd_fasync_free); -- cgit v1.2.3 From 95cc637c1afd83fb7dd3d7c8a53710488f4caf9c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 28 Jul 2022 14:59:43 +0200 Subject: ALSA: timer: Use deferred fasync helper For avoiding the potential deadlock via kill_fasync() call, use the new fasync helpers to defer the invocation from PCI API. Note that it's merely a workaround. Reported-by: syzbot+1ee0910eca9c94f71f25@syzkaller.appspotmail.com Reported-by: syzbot+49b10793b867871ee26f@syzkaller.appspotmail.com Reported-by: syzbot+8285e973a41b5aa68902@syzkaller.appspotmail.com Link: https://lore.kernel.org/r/20220728125945.29533-3-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/timer.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/core/timer.c b/sound/core/timer.c index b3214baa8919..e08a37c23add 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -83,7 +83,7 @@ struct snd_timer_user { unsigned int filter; struct timespec64 tstamp; /* trigger tstamp */ wait_queue_head_t qchange_sleep; - struct fasync_struct *fasync; + struct snd_fasync *fasync; struct mutex ioctl_lock; }; @@ -1345,7 +1345,7 @@ static void snd_timer_user_interrupt(struct snd_timer_instance *timeri, } __wake: spin_unlock(&tu->qlock); - kill_fasync(&tu->fasync, SIGIO, POLL_IN); + snd_kill_fasync(tu->fasync, SIGIO, POLL_IN); wake_up(&tu->qchange_sleep); } @@ -1383,7 +1383,7 @@ static void snd_timer_user_ccallback(struct snd_timer_instance *timeri, spin_lock_irqsave(&tu->qlock, flags); snd_timer_user_append_to_tqueue(tu, &r1); spin_unlock_irqrestore(&tu->qlock, flags); - kill_fasync(&tu->fasync, SIGIO, POLL_IN); + snd_kill_fasync(tu->fasync, SIGIO, POLL_IN); wake_up(&tu->qchange_sleep); } @@ -1453,7 +1453,7 @@ static void snd_timer_user_tinterrupt(struct snd_timer_instance *timeri, spin_unlock(&tu->qlock); if (append == 0) return; - kill_fasync(&tu->fasync, SIGIO, POLL_IN); + snd_kill_fasync(tu->fasync, SIGIO, POLL_IN); wake_up(&tu->qchange_sleep); } @@ -1521,6 +1521,7 @@ static int snd_timer_user_release(struct inode *inode, struct file *file) snd_timer_instance_free(tu->timeri); } mutex_unlock(&tu->ioctl_lock); + snd_fasync_free(tu->fasync); kfree(tu->queue); kfree(tu->tqueue); kfree(tu); @@ -2135,7 +2136,7 @@ static int snd_timer_user_fasync(int fd, struct file * file, int on) struct snd_timer_user *tu; tu = file->private_data; - return fasync_helper(fd, file, on, &tu->fasync); + return snd_fasync_helper(fd, file, on, &tu->fasync); } static ssize_t snd_timer_user_read(struct file *file, char __user *buffer, -- cgit v1.2.3 From 96b097091c66df4f6fbf5cbff21df6cc02a2f055 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 28 Jul 2022 14:59:44 +0200 Subject: ALSA: pcm: Use deferred fasync helper For avoiding the potential deadlock via kill_fasync() call, use the new fasync helpers to defer the invocation from timer API. Note that it's merely a workaround. Reported-by: syzbot+8285e973a41b5aa68902@syzkaller.appspotmail.com Reported-by: syzbot+669c9abf11a6a011dd09@syzkaller.appspotmail.com Link: https://lore.kernel.org/r/20220728125945.29533-4-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/pcm.c | 1 + sound/core/pcm_lib.c | 2 +- sound/core/pcm_native.c | 2 +- 3 files changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 03fc5fa5813e..2ac742035310 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -1007,6 +1007,7 @@ void snd_pcm_detach_substream(struct snd_pcm_substream *substream) substream->runtime = NULL; } mutex_destroy(&runtime->buffer_mutex); + snd_fasync_free(runtime->fasync); kfree(runtime); put_pid(substream->pid); substream->pid = NULL; diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 1fc7c50ffa62..40751e5aff09 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1822,7 +1822,7 @@ void snd_pcm_period_elapsed_under_stream_lock(struct snd_pcm_substream *substrea snd_timer_interrupt(substream->timer, 1); #endif _end: - kill_fasync(&runtime->fasync, SIGIO, POLL_IN); + snd_kill_fasync(runtime->fasync, SIGIO, POLL_IN); } EXPORT_SYMBOL(snd_pcm_period_elapsed_under_stream_lock); diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index aa0453e51595..ad0541e9e888 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3951,7 +3951,7 @@ static int snd_pcm_fasync(int fd, struct file * file, int on) runtime = substream->runtime; if (runtime->status->state == SNDRV_PCM_STATE_DISCONNECTED) return -EBADFD; - return fasync_helper(fd, file, on, &runtime->fasync); + return snd_fasync_helper(fd, file, on, &runtime->fasync); } /* -- cgit v1.2.3 From 4a971e84a7ae10a38d875cd2d4e487c8d1682ca3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 28 Jul 2022 14:59:45 +0200 Subject: ALSA: control: Use deferred fasync helper For avoiding the potential deadlock via kill_fasync() call, use the new fasync helpers to defer the invocation from the control API. Note that it's merely a workaround. Another note: although we haven't received reports about the deadlock with the control API, the deadlock is still potentially possible, and it's better to align the behavior with other core APIs (PCM and timer); so let's move altogether. Link: https://lore.kernel.org/r/20220728125945.29533-5-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/control.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/core/control.c b/sound/core/control.c index 4dba3a342458..f3e893715369 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -127,6 +127,7 @@ static int snd_ctl_release(struct inode *inode, struct file *file) if (control->vd[idx].owner == ctl) control->vd[idx].owner = NULL; up_write(&card->controls_rwsem); + snd_fasync_free(ctl->fasync); snd_ctl_empty_read_queue(ctl); put_pid(ctl->pid); kfree(ctl); @@ -181,7 +182,7 @@ void snd_ctl_notify(struct snd_card *card, unsigned int mask, _found: wake_up(&ctl->change_sleep); spin_unlock(&ctl->read_lock); - kill_fasync(&ctl->fasync, SIGIO, POLL_IN); + snd_kill_fasync(ctl->fasync, SIGIO, POLL_IN); } read_unlock_irqrestore(&card->ctl_files_rwlock, flags); } @@ -2134,7 +2135,7 @@ static int snd_ctl_fasync(int fd, struct file * file, int on) struct snd_ctl_file *ctl; ctl = file->private_data; - return fasync_helper(fd, file, on, &ctl->fasync); + return snd_fasync_helper(fd, file, on, &ctl->fasync); } /* return the preferred subdevice number if already assigned; @@ -2302,7 +2303,7 @@ static int snd_ctl_dev_disconnect(struct snd_device *device) read_lock_irqsave(&card->ctl_files_rwlock, flags); list_for_each_entry(ctl, &card->ctl_files, list) { wake_up(&ctl->change_sleep); - kill_fasync(&ctl->fasync, SIGIO, POLL_ERR); + snd_kill_fasync(ctl->fasync, SIGIO, POLL_ERR); } read_unlock_irqrestore(&card->ctl_files_rwlock, flags); -- cgit v1.2.3 From 3790a3d6dbbc48e30586e9c3fc752a00e2e11946 Mon Sep 17 00:00:00 2001 From: Philipp Jungkamp Date: Fri, 29 Jul 2022 18:21:03 +0200 Subject: ALSA: hda/realtek: Add quirk for Lenovo Yoga9 14IAP7 The Lenovo Yoga 9 14IAP7 is set up similarly to the Thinkpad X1 7th and 8th Gen. It also has the speakers attached to NID 0x14 and the bass speakers to NID 0x17, but here the codec misreports the NID 0x17 as unconnected. The pincfg and hda verbs connect and activate the bass speaker amplifiers, but the generic driver will connect them to NID 0x06 which has no volume control. Set connection list/preferred connections is required to gain volume control. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=208555 Signed-off-by: Philipp Jungkamp Cc: Link: https://lore.kernel.org/r/20220729162103.6062-1-p.jungkamp@gmx.net Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 109 ++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 109 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e1fbda215975..cca093cb643e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6817,6 +6817,43 @@ static void alc_fixup_dell4_mic_no_presence_quiet(struct hda_codec *codec, } } +static void alc287_fixup_yoga9_14iap7_bass_spk_pin(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + /* + * The Pin Complex 0x17 for the bass speakers is wrongly reported as + * unconnected. + */ + static const struct hda_pintbl pincfgs[] = { + { 0x17, 0x90170121 }, + { } + }; + /* + * Avoid DAC 0x06 and 0x08, as they have no volume controls. + * DAC 0x02 and 0x03 would be fine. + */ + static const hda_nid_t conn[] = { 0x02, 0x03 }; + /* + * Prefer both speakerbar (0x14) and bass speakers (0x17) connected to DAC 0x02. + * Headphones (0x21) are connected to DAC 0x03. + */ + static const hda_nid_t preferred_pairs[] = { + 0x14, 0x02, + 0x17, 0x02, + 0x21, 0x03, + 0 + }; + struct alc_spec *spec = codec->spec; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + snd_hda_apply_pincfgs(codec, pincfgs); + snd_hda_override_conn_list(codec, 0x17, ARRAY_SIZE(conn), conn); + spec->gen.preferred_dacs = preferred_pairs; + break; + } +} + enum { ALC269_FIXUP_GPIO2, ALC269_FIXUP_SONY_VAIO, @@ -7054,6 +7091,8 @@ enum { ALC285_FIXUP_HP_SPEAKERS_MICMUTE_LED, ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE, ALC287_FIXUP_LEGION_16ITHG6, + ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK, + ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN, }; /* A special fixup for Lenovo C940 and Yoga Duet 7; @@ -8900,6 +8939,74 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc287_fixup_legion_16ithg6_speakers, }, + [ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + // enable left speaker + { 0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x41 }, + + { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0xc }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x1a }, + { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + + { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0xf }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x42 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + + { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x10 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x40 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + + { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x2 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + + // enable right speaker + { 0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x46 }, + + { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0xc }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x2a }, + { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + + { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0xf }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x46 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + + { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x10 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x44 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + + { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x2 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + + { }, + }, + }, + [ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc287_fixup_yoga9_14iap7_bass_spk_pin, + .chained = true, + .chain_id = ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -9352,6 +9459,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3176, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x17aa, 0x3178, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x17aa, 0x31af, "ThinkCentre Station", ALC623_FIXUP_LENOVO_THINKSTATION_P340), + SND_PCI_QUIRK(0x17aa, 0x3801, "Lenovo Yoga9 14IAP7", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN), SND_PCI_QUIRK(0x17aa, 0x3802, "Lenovo Yoga DuetITL 2021", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3813, "Legion 7i 15IMHG05", ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3818, "Lenovo C940 / Yoga Duet 7", ALC298_FIXUP_LENOVO_C940_DUET7), @@ -9598,6 +9706,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC285_FIXUP_HP_SPECTRE_X360, .name = "alc285-hp-spectre-x360"}, {.id = ALC285_FIXUP_HP_SPECTRE_X360_EB1, .name = "alc285-hp-spectre-x360-eb1"}, {.id = ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP, .name = "alc287-ideapad-bass-spk-amp"}, + {.id = ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN, .name = "alc287-yoga9-bass-spk-pin"}, {.id = ALC623_FIXUP_LENOVO_THINKSTATION_P340, .name = "alc623-lenovo-thinkstation-p340"}, {.id = ALC255_FIXUP_ACER_HEADPHONE_AND_MIC, .name = "alc255-acer-headphone-and-mic"}, {.id = ALC285_FIXUP_HP_GPIO_AMP_INIT, .name = "alc285-hp-amp-init"}, -- cgit v1.2.3 From be561ffad708f0cee18aee4231f80ffafaf7a419 Mon Sep 17 00:00:00 2001 From: Tim Crawford Date: Sat, 30 Jul 2022 21:22:43 -0600 Subject: ALSA: hda/realtek: Add quirk for Clevo NV45PZ Fixes headset detection on Clevo NV45PZ. Signed-off-by: Tim Crawford Cc: Link: https://lore.kernel.org/r/20220731032243.4300-1-tcrawford@system76.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index cca093cb643e..105468acde90 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9347,6 +9347,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x4018, "Clevo NV40M[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x4019, "Clevo NV40MZ", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x4020, "Clevo NV40MB", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x4041, "Clevo NV4[15]PZ", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x40a1, "Clevo NL40GU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x40c1, "Clevo NL40[CZ]U", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x40d1, "Clevo NL41DU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), -- cgit v1.2.3