diff options
Diffstat (limited to 'sound/soc/meson')
-rw-r--r-- | sound/soc/meson/Kconfig | 8 | ||||
-rw-r--r-- | sound/soc/meson/Makefile | 2 | ||||
-rw-r--r-- | sound/soc/meson/axg-card.c | 87 | ||||
-rw-r--r-- | sound/soc/meson/axg-tdm-formatter.c | 29 | ||||
-rw-r--r-- | sound/soc/meson/axg-tdm-interface.c | 4 | ||||
-rw-r--r-- | sound/soc/meson/g12a-tohdmitx.c | 413 |
6 files changed, 522 insertions, 21 deletions
diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig index 3085bdd318f3..63b38c123103 100644 --- a/sound/soc/meson/Kconfig +++ b/sound/soc/meson/Kconfig @@ -57,6 +57,7 @@ config SND_MESON_AXG_SOUND_CARD imply SND_MESON_AXG_SPDIFOUT imply SND_MESON_AXG_SPDIFIN imply SND_MESON_AXG_PDM + imply SND_MESON_G12A_TOHDMITX if DRM_MESON_DW_HDMI help Select Y or M to add support for the AXG SoC sound card @@ -83,4 +84,11 @@ config SND_MESON_AXG_PDM help Select Y or M to add support for PDM input embedded in the Amlogic AXG SoC family + +config SND_MESON_G12A_TOHDMITX + tristate "Amlogic G12A To HDMI TX Control Support" + imply SND_SOC_HDMI_CODEC + help + Select Y or M to add support for HDMI audio on the g12a SoC + family endmenu diff --git a/sound/soc/meson/Makefile b/sound/soc/meson/Makefile index b45dfb9e2f88..1a8b1470ed84 100644 --- a/sound/soc/meson/Makefile +++ b/sound/soc/meson/Makefile @@ -11,6 +11,7 @@ snd-soc-meson-axg-sound-card-objs := axg-card.o snd-soc-meson-axg-spdifin-objs := axg-spdifin.o snd-soc-meson-axg-spdifout-objs := axg-spdifout.o snd-soc-meson-axg-pdm-objs := axg-pdm.o +snd-soc-meson-g12a-tohdmitx-objs := g12a-tohdmitx.o obj-$(CONFIG_SND_MESON_AXG_FIFO) += snd-soc-meson-axg-fifo.o obj-$(CONFIG_SND_MESON_AXG_FRDDR) += snd-soc-meson-axg-frddr.o @@ -23,3 +24,4 @@ obj-$(CONFIG_SND_MESON_AXG_SOUND_CARD) += snd-soc-meson-axg-sound-card.o obj-$(CONFIG_SND_MESON_AXG_SPDIFIN) += snd-soc-meson-axg-spdifin.o obj-$(CONFIG_SND_MESON_AXG_SPDIFOUT) += snd-soc-meson-axg-spdifout.o obj-$(CONFIG_SND_MESON_AXG_PDM) += snd-soc-meson-axg-pdm.o +obj-$(CONFIG_SND_MESON_G12A_TOHDMITX) += snd-soc-meson-g12a-tohdmitx.o diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c index aa54d2c612c9..14a8321744da 100644 --- a/sound/soc/meson/axg-card.c +++ b/sound/soc/meson/axg-card.c @@ -29,6 +29,18 @@ struct axg_dai_link_tdm_data { struct axg_dai_link_tdm_mask *codec_masks; }; +/* + * Base params for the codec to codec links + * Those will be over-written by the CPU side of the link + */ +static const struct snd_soc_pcm_stream codec_params = { + .formats = SNDRV_PCM_FMTBIT_S24_LE, + .rate_min = 5525, + .rate_max = 192000, + .channels_min = 1, + .channels_max = 8, +}; + #define PREFIX "amlogic," static int axg_card_reallocate_links(struct axg_card *priv, @@ -80,10 +92,11 @@ static int axg_card_parse_dai(struct snd_soc_card *card, static int axg_card_set_link_name(struct snd_soc_card *card, struct snd_soc_dai_link *link, + struct device_node *node, const char *prefix) { char *name = devm_kasprintf(card->dev, GFP_KERNEL, "%s.%s", - prefix, link->cpu_of_node->full_name); + prefix, node->full_name); if (!name) return -ENOMEM; @@ -102,7 +115,8 @@ static void axg_card_clean_references(struct axg_card *priv) if (card->dai_link) { for_each_card_prelinks(card, i, link) { - of_node_put(link->cpu_of_node); + if (link->cpus) + of_node_put(link->cpus->of_node); for_each_link_codecs(link, j, codec) of_node_put(codec->of_node); } @@ -241,6 +255,7 @@ static int axg_card_add_tdm_loopback(struct snd_soc_card *card, struct axg_card *priv = snd_soc_card_get_drvdata(card); struct snd_soc_dai_link *pad = &card->dai_link[*index]; struct snd_soc_dai_link *lb; + struct snd_soc_dai_link_component *dlc; int ret; /* extend links */ @@ -254,11 +269,20 @@ static int axg_card_add_tdm_loopback(struct snd_soc_card *card, if (!lb->name) return -ENOMEM; + dlc = devm_kzalloc(card->dev, 2 * sizeof(*dlc), GFP_KERNEL); + if (!dlc) + return -ENOMEM; + + lb->cpus = &dlc[0]; + lb->codecs = &dlc[1]; + lb->num_cpus = 1; + lb->num_codecs = 1; + lb->stream_name = lb->name; - lb->cpu_of_node = pad->cpu_of_node; - lb->cpu_dai_name = "TDM Loopback"; - lb->codec_name = "snd-soc-dummy"; - lb->codec_dai_name = "snd-soc-dummy-dai"; + lb->cpus->of_node = pad->cpus->of_node; + lb->cpus->dai_name = "TDM Loopback"; + lb->codecs->name = "snd-soc-dummy"; + lb->codecs->dai_name = "snd-soc-dummy-dai"; lb->dpcm_capture = 1; lb->no_pcm = 1; lb->ops = &axg_card_tdm_be_ops; @@ -271,7 +295,7 @@ static int axg_card_add_tdm_loopback(struct snd_soc_card *card, * axg_card_clean_references() will iterate over this link, * make sure the node count is balanced */ - of_node_get(lb->cpu_of_node); + of_node_get(lb->cpus->of_node); /* Let add_links continue where it should */ *index += 1; @@ -413,7 +437,7 @@ static int axg_card_parse_tdm(struct snd_soc_card *card, /* Setup tdm link */ link->ops = &axg_card_tdm_be_ops; link->init = axg_card_tdm_dai_init; - link->dai_fmt = axg_card_parse_daifmt(node, link->cpu_of_node); + link->dai_fmt = axg_card_parse_daifmt(node, link->cpus->of_node); of_property_read_u32(node, "mclk-fs", &be->mclk_fs); @@ -474,7 +498,7 @@ static int axg_card_set_be_link(struct snd_soc_card *card, codec++; } - ret = axg_card_set_link_name(card, link, "be"); + ret = axg_card_set_link_name(card, link, node, "be"); if (ret) dev_err(card->dev, "error setting %pOFn link name\n", np); @@ -483,21 +507,31 @@ static int axg_card_set_be_link(struct snd_soc_card *card, static int axg_card_set_fe_link(struct snd_soc_card *card, struct snd_soc_dai_link *link, + struct device_node *node, bool is_playback) { + struct snd_soc_dai_link_component *codec; + + codec = devm_kzalloc(card->dev, sizeof(*codec), GFP_KERNEL); + if (!codec) + return -ENOMEM; + + link->codecs = codec; + link->num_codecs = 1; + link->dynamic = 1; link->dpcm_merged_format = 1; link->dpcm_merged_chan = 1; link->dpcm_merged_rate = 1; - link->codec_dai_name = "snd-soc-dummy-dai"; - link->codec_name = "snd-soc-dummy"; + link->codecs->dai_name = "snd-soc-dummy-dai"; + link->codecs->name = "snd-soc-dummy"; if (is_playback) link->dpcm_playback = 1; else link->dpcm_capture = 1; - return axg_card_set_link_name(card, link, "fe"); + return axg_card_set_link_name(card, link, node, "fe"); } static int axg_card_cpu_is_capture_fe(struct device_node *np) @@ -515,29 +549,44 @@ static int axg_card_cpu_is_tdm_iface(struct device_node *np) return of_device_is_compatible(np, PREFIX "axg-tdm-iface"); } +static int axg_card_cpu_is_codec(struct device_node *np) +{ + return of_device_is_compatible(np, PREFIX "g12a-tohdmitx"); +} + static int axg_card_add_link(struct snd_soc_card *card, struct device_node *np, int *index) { struct snd_soc_dai_link *dai_link = &card->dai_link[*index]; + struct snd_soc_dai_link_component *cpu; int ret; - ret = axg_card_parse_dai(card, np, &dai_link->cpu_of_node, - &dai_link->cpu_dai_name); + cpu = devm_kzalloc(card->dev, sizeof(*cpu), GFP_KERNEL); + if (!cpu) + return -ENOMEM; + + dai_link->cpus = cpu; + dai_link->num_cpus = 1; + + ret = axg_card_parse_dai(card, np, &dai_link->cpus->of_node, + &dai_link->cpus->dai_name); if (ret) return ret; - if (axg_card_cpu_is_playback_fe(dai_link->cpu_of_node)) - ret = axg_card_set_fe_link(card, dai_link, true); - else if (axg_card_cpu_is_capture_fe(dai_link->cpu_of_node)) - ret = axg_card_set_fe_link(card, dai_link, false); + if (axg_card_cpu_is_playback_fe(dai_link->cpus->of_node)) + ret = axg_card_set_fe_link(card, dai_link, np, true); + else if (axg_card_cpu_is_capture_fe(dai_link->cpus->of_node)) + ret = axg_card_set_fe_link(card, dai_link, np, false); else ret = axg_card_set_be_link(card, dai_link, np); if (ret) return ret; - if (axg_card_cpu_is_tdm_iface(dai_link->cpu_of_node)) + if (axg_card_cpu_is_tdm_iface(dai_link->cpus->of_node)) ret = axg_card_parse_tdm(card, np, index); + else if (axg_card_cpu_is_codec(dai_link->cpus->of_node)) + dai_link->params = &codec_params; return ret; } diff --git a/sound/soc/meson/axg-tdm-formatter.c b/sound/soc/meson/axg-tdm-formatter.c index 0c6cce5c5773..2e498201139f 100644 --- a/sound/soc/meson/axg-tdm-formatter.c +++ b/sound/soc/meson/axg-tdm-formatter.c @@ -7,6 +7,7 @@ #include <linux/module.h> #include <linux/of_platform.h> #include <linux/regmap.h> +#include <linux/reset.h> #include <sound/soc.h> #include "axg-tdm-formatter.h" @@ -20,6 +21,7 @@ struct axg_tdm_formatter { struct clk *lrclk; struct clk *sclk_sel; struct clk *lrclk_sel; + struct reset_control *reset; bool enabled; struct regmap *map; }; @@ -76,6 +78,24 @@ static int axg_tdm_formatter_enable(struct axg_tdm_formatter *formatter) return 0; /* + * On the g12a (and possibly other SoCs), when a stream using + * multiple lanes is restarted, it will sometimes not start + * from the first lane, but randomly from another used one. + * The result is an unexpected and random channel shift. + * + * The hypothesis is that an HW counter is not properly reset + * and the formatter simply starts on the lane it stopped + * before. Unfortunately, there does not seems to be a way to + * reset this through the registers of the block. + * + * However, the g12a has indenpendent reset lines for each audio + * devices. Using this reset before each start solves the issue. + */ + ret = reset_control_reset(formatter->reset); + if (ret) + return ret; + + /* * If sclk is inverted, invert it back and provide the inversion * required by the formatter */ @@ -306,6 +326,15 @@ int axg_tdm_formatter_probe(struct platform_device *pdev) return ret; } + /* Formatter dedicated reset line */ + formatter->reset = reset_control_get_optional_exclusive(dev, NULL); + if (IS_ERR(formatter->reset)) { + ret = PTR_ERR(formatter->reset); + if (ret != -EPROBE_DEFER) + dev_err(dev, "failed to get reset: %d\n", ret); + return ret; + } + return devm_snd_soc_register_component(dev, drv->component_drv, NULL, 0); } diff --git a/sound/soc/meson/axg-tdm-interface.c b/sound/soc/meson/axg-tdm-interface.c index 585ce030b79b..d51f3344be7c 100644 --- a/sound/soc/meson/axg-tdm-interface.c +++ b/sound/soc/meson/axg-tdm-interface.c @@ -306,8 +306,8 @@ static int axg_tdm_iface_hw_params(struct snd_pcm_substream *substream, } break; - case SND_SOC_DAI_FORMAT_DSP_A: - case SND_SOC_DAI_FORMAT_DSP_B: + case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: break; default: diff --git a/sound/soc/meson/g12a-tohdmitx.c b/sound/soc/meson/g12a-tohdmitx.c new file mode 100644 index 000000000000..707ccb192e4c --- /dev/null +++ b/sound/soc/meson/g12a-tohdmitx.c @@ -0,0 +1,413 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Copyright (c) 2019 BayLibre, SAS. +// Author: Jerome Brunet <jbrunet@baylibre.com> + +#include <linux/bitfield.h> +#include <linux/clk.h> +#include <linux/module.h> +#include <sound/pcm_params.h> +#include <linux/regmap.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> + +#include <dt-bindings/sound/meson-g12a-tohdmitx.h> + +#define G12A_TOHDMITX_DRV_NAME "g12a-tohdmitx" + +#define TOHDMITX_CTRL0 0x0 +#define CTRL0_ENABLE_SHIFT 31 +#define CTRL0_I2S_DAT_SEL GENMASK(13, 12) +#define CTRL0_I2S_LRCLK_SEL GENMASK(9, 8) +#define CTRL0_I2S_BLK_CAP_INV BIT(7) +#define CTRL0_I2S_BCLK_O_INV BIT(6) +#define CTRL0_I2S_BCLK_SEL GENMASK(5, 4) +#define CTRL0_SPDIF_CLK_CAP_INV BIT(3) +#define CTRL0_SPDIF_CLK_O_INV BIT(2) +#define CTRL0_SPDIF_SEL BIT(1) +#define CTRL0_SPDIF_CLK_SEL BIT(0) + +struct g12a_tohdmitx_input { + struct snd_pcm_hw_params params; + unsigned int fmt; +}; + +static struct snd_soc_dapm_widget * +g12a_tohdmitx_get_input(struct snd_soc_dapm_widget *w) +{ + struct snd_soc_dapm_path *p = NULL; + struct snd_soc_dapm_widget *in; + + snd_soc_dapm_widget_for_each_source_path(w, p) { + if (!p->connect) + continue; + + /* Check that we still are in the same component */ + if (snd_soc_dapm_to_component(w->dapm) != + snd_soc_dapm_to_component(p->source->dapm)) + continue; + + if (p->source->id == snd_soc_dapm_dai_in) + return p->source; + + in = g12a_tohdmitx_get_input(p->source); + if (in) + return in; + } + + return NULL; +} + +static struct g12a_tohdmitx_input * +g12a_tohdmitx_get_input_data(struct snd_soc_dapm_widget *w) +{ + struct snd_soc_dapm_widget *in = + g12a_tohdmitx_get_input(w); + struct snd_soc_dai *dai; + + if (WARN_ON(!in)) + return NULL; + + dai = in->priv; + + return dai->playback_dma_data; +} + +static const char * const g12a_tohdmitx_i2s_mux_texts[] = { + "I2S A", "I2S B", "I2S C", +}; + +static SOC_ENUM_SINGLE_EXT_DECL(g12a_tohdmitx_i2s_mux_enum, + g12a_tohdmitx_i2s_mux_texts); + +static int g12a_tohdmitx_get_input_val(struct snd_soc_component *component, + unsigned int mask) +{ + unsigned int val; + + snd_soc_component_read(component, TOHDMITX_CTRL0, &val); + return (val & mask) >> __ffs(mask); +} + +static int g12a_tohdmitx_i2s_mux_get_enum(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_dapm_kcontrol_component(kcontrol); + + ucontrol->value.enumerated.item[0] = + g12a_tohdmitx_get_input_val(component, CTRL0_I2S_DAT_SEL); + + return 0; +} + +static int g12a_tohdmitx_i2s_mux_put_enum(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_dapm_kcontrol_component(kcontrol); + struct snd_soc_dapm_context *dapm = + snd_soc_dapm_kcontrol_dapm(kcontrol); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int mux = ucontrol->value.enumerated.item[0]; + unsigned int val = g12a_tohdmitx_get_input_val(component, + CTRL0_I2S_DAT_SEL); + + /* Force disconnect of the mux while updating */ + if (val != mux) + snd_soc_dapm_mux_update_power(dapm, kcontrol, 0, NULL, NULL); + + snd_soc_component_update_bits(component, TOHDMITX_CTRL0, + CTRL0_I2S_DAT_SEL | + CTRL0_I2S_LRCLK_SEL | + CTRL0_I2S_BCLK_SEL, + FIELD_PREP(CTRL0_I2S_DAT_SEL, mux) | + FIELD_PREP(CTRL0_I2S_LRCLK_SEL, mux) | + FIELD_PREP(CTRL0_I2S_BCLK_SEL, mux)); + + snd_soc_dapm_mux_update_power(dapm, kcontrol, mux, e, NULL); + + return 0; +} + +static const struct snd_kcontrol_new g12a_tohdmitx_i2s_mux = + SOC_DAPM_ENUM_EXT("I2S Source", g12a_tohdmitx_i2s_mux_enum, + g12a_tohdmitx_i2s_mux_get_enum, + g12a_tohdmitx_i2s_mux_put_enum); + +static const char * const g12a_tohdmitx_spdif_mux_texts[] = { + "SPDIF A", "SPDIF B", +}; + +static SOC_ENUM_SINGLE_EXT_DECL(g12a_tohdmitx_spdif_mux_enum, + g12a_tohdmitx_spdif_mux_texts); + +static int g12a_tohdmitx_spdif_mux_get_enum(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_dapm_kcontrol_component(kcontrol); + + ucontrol->value.enumerated.item[0] = + g12a_tohdmitx_get_input_val(component, CTRL0_SPDIF_SEL); + + return 0; +} + +static int g12a_tohdmitx_spdif_mux_put_enum(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_dapm_kcontrol_component(kcontrol); + struct snd_soc_dapm_context *dapm = + snd_soc_dapm_kcontrol_dapm(kcontrol); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int mux = ucontrol->value.enumerated.item[0]; + unsigned int val = g12a_tohdmitx_get_input_val(component, + CTRL0_SPDIF_SEL); + + /* Force disconnect of the mux while updating */ + if (val != mux) + snd_soc_dapm_mux_update_power(dapm, kcontrol, 0, NULL, NULL); + + snd_soc_component_update_bits(component, TOHDMITX_CTRL0, + CTRL0_SPDIF_SEL | + CTRL0_SPDIF_CLK_SEL, + FIELD_PREP(CTRL0_SPDIF_SEL, mux) | + FIELD_PREP(CTRL0_SPDIF_CLK_SEL, mux)); + + snd_soc_dapm_mux_update_power(dapm, kcontrol, mux, e, NULL); + + return 0; +} + +static const struct snd_kcontrol_new g12a_tohdmitx_spdif_mux = + SOC_DAPM_ENUM_EXT("SPDIF Source", g12a_tohdmitx_spdif_mux_enum, + g12a_tohdmitx_spdif_mux_get_enum, + g12a_tohdmitx_spdif_mux_put_enum); + +static const struct snd_kcontrol_new g12a_tohdmitx_out_enable = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", TOHDMITX_CTRL0, + CTRL0_ENABLE_SHIFT, 1, 0); + +static const struct snd_soc_dapm_widget g12a_tohdmitx_widgets[] = { + SND_SOC_DAPM_MUX("I2S SRC", SND_SOC_NOPM, 0, 0, + &g12a_tohdmitx_i2s_mux), + SND_SOC_DAPM_SWITCH("I2S OUT EN", SND_SOC_NOPM, 0, 0, + &g12a_tohdmitx_out_enable), + SND_SOC_DAPM_MUX("SPDIF SRC", SND_SOC_NOPM, 0, 0, + &g12a_tohdmitx_spdif_mux), + SND_SOC_DAPM_SWITCH("SPDIF OUT EN", SND_SOC_NOPM, 0, 0, + &g12a_tohdmitx_out_enable), +}; + +static int g12a_tohdmitx_input_probe(struct snd_soc_dai *dai) +{ + struct g12a_tohdmitx_input *data; + + data = kzalloc(sizeof(*data), GFP_KERNEL); + if (!data) + return -ENOMEM; + + dai->playback_dma_data = data; + return 0; +} + +static int g12a_tohdmitx_input_remove(struct snd_soc_dai *dai) +{ + kfree(dai->playback_dma_data); + return 0; +} + +static int g12a_tohdmitx_input_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct g12a_tohdmitx_input *data = dai->playback_dma_data; + + /* Save the stream params for the downstream link */ + memcpy(&data->params, params, sizeof(*params)); + + return 0; +} + +static int g12a_tohdmitx_output_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct g12a_tohdmitx_input *in_data = + g12a_tohdmitx_get_input_data(dai->capture_widget); + + if (!in_data) + return -ENODEV; + + memcpy(params, &in_data->params, sizeof(*params)); + + return 0; +} + +static int g12a_tohdmitx_input_set_fmt(struct snd_soc_dai *dai, + unsigned int fmt) +{ + struct g12a_tohdmitx_input *data = dai->playback_dma_data; + + /* Save the source stream format for the downstream link */ + data->fmt = fmt; + return 0; +} + +static int g12a_tohdmitx_output_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct g12a_tohdmitx_input *in_data = + g12a_tohdmitx_get_input_data(dai->capture_widget); + + if (!in_data) + return -ENODEV; + + if (!in_data->fmt) + return 0; + + return snd_soc_runtime_set_dai_fmt(rtd, in_data->fmt); +} + +static const struct snd_soc_dai_ops g12a_tohdmitx_input_ops = { + .hw_params = g12a_tohdmitx_input_hw_params, + .set_fmt = g12a_tohdmitx_input_set_fmt, +}; + +static const struct snd_soc_dai_ops g12a_tohdmitx_output_ops = { + .hw_params = g12a_tohdmitx_output_hw_params, + .startup = g12a_tohdmitx_output_startup, +}; + +#define TOHDMITX_SPDIF_FORMATS \ + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE) + +#define TOHDMITX_I2S_FORMATS \ + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +#define TOHDMITX_STREAM(xname, xsuffix, xfmt, xchmax) \ +{ \ + .stream_name = xname " " xsuffix, \ + .channels_min = 1, \ + .channels_max = (xchmax), \ + .rate_min = 8000, \ + .rate_max = 192000, \ + .formats = (xfmt), \ +} + +#define TOHDMITX_IN(xname, xid, xfmt, xchmax) { \ + .name = xname, \ + .id = (xid), \ + .playback = TOHDMITX_STREAM(xname, "Playback", xfmt, xchmax), \ + .ops = &g12a_tohdmitx_input_ops, \ + .probe = g12a_tohdmitx_input_probe, \ + .remove = g12a_tohdmitx_input_remove, \ +} + +#define TOHDMITX_OUT(xname, xid, xfmt, xchmax) { \ + .name = xname, \ + .id = (xid), \ + .capture = TOHDMITX_STREAM(xname, "Capture", xfmt, xchmax), \ + .ops = &g12a_tohdmitx_output_ops, \ +} + +static struct snd_soc_dai_driver g12a_tohdmitx_dai_drv[] = { + TOHDMITX_IN("I2S IN A", TOHDMITX_I2S_IN_A, + TOHDMITX_I2S_FORMATS, 8), + TOHDMITX_IN("I2S IN B", TOHDMITX_I2S_IN_B, + TOHDMITX_I2S_FORMATS, 8), + TOHDMITX_IN("I2S IN C", TOHDMITX_I2S_IN_C, + TOHDMITX_I2S_FORMATS, 8), + TOHDMITX_OUT("I2S OUT", TOHDMITX_I2S_OUT, + TOHDMITX_I2S_FORMATS, 8), + TOHDMITX_IN("SPDIF IN A", TOHDMITX_SPDIF_IN_A, + TOHDMITX_SPDIF_FORMATS, 2), + TOHDMITX_IN("SPDIF IN B", TOHDMITX_SPDIF_IN_B, + TOHDMITX_SPDIF_FORMATS, 2), + TOHDMITX_OUT("SPDIF OUT", TOHDMITX_SPDIF_OUT, + TOHDMITX_SPDIF_FORMATS, 2), +}; + +static int g12a_tohdmi_component_probe(struct snd_soc_component *c) +{ + /* Initialize the static clock parameters */ + return snd_soc_component_write(c, TOHDMITX_CTRL0, + CTRL0_I2S_BLK_CAP_INV | CTRL0_SPDIF_CLK_CAP_INV); +} + +static const struct snd_soc_dapm_route g12a_tohdmitx_routes[] = { + { "I2S SRC", "I2S A", "I2S IN A Playback" }, + { "I2S SRC", "I2S B", "I2S IN B Playback" }, + { "I2S SRC", "I2S C", "I2S IN C Playback" }, + { "I2S OUT EN", "Switch", "I2S SRC" }, + { "I2S OUT Capture", NULL, "I2S OUT EN" }, + { "SPDIF SRC", "SPDIF A", "SPDIF IN A Playback" }, + { "SPDIF SRC", "SPDIF B", "SPDIF IN B Playback" }, + { "SPDIF OUT EN", "Switch", "SPDIF SRC" }, + { "SPDIF OUT Capture", NULL, "SPDIF OUT EN" }, +}; + +static const struct snd_soc_component_driver g12a_tohdmitx_component_drv = { + .probe = g12a_tohdmi_component_probe, + .dapm_widgets = g12a_tohdmitx_widgets, + .num_dapm_widgets = ARRAY_SIZE(g12a_tohdmitx_widgets), + .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 = { + .reg_bits = 32, + .val_bits = 32, + .reg_stride = 4, +}; + +static const struct of_device_id g12a_tohdmitx_of_match[] = { + { .compatible = "amlogic,g12a-tohdmitx", }, + {} +}; +MODULE_DEVICE_TABLE(of, g12a_tohdmitx_of_match); + +static int g12a_tohdmitx_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct resource *res; + void __iomem *regs; + struct regmap *map; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + regs = devm_ioremap_resource(dev, res); + if (IS_ERR(regs)) + return PTR_ERR(regs); + + map = devm_regmap_init_mmio(dev, regs, &g12a_tohdmitx_regmap_cfg); + if (IS_ERR(map)) { + dev_err(dev, "failed to init regmap: %ld\n", + PTR_ERR(map)); + return PTR_ERR(map); + } + + return devm_snd_soc_register_component(dev, + &g12a_tohdmitx_component_drv, g12a_tohdmitx_dai_drv, + ARRAY_SIZE(g12a_tohdmitx_dai_drv)); +} + +static struct platform_driver g12a_tohdmitx_pdrv = { + .driver = { + .name = G12A_TOHDMITX_DRV_NAME, + .of_match_table = g12a_tohdmitx_of_match, + }, + .probe = g12a_tohdmitx_probe, +}; +module_platform_driver(g12a_tohdmitx_pdrv); + +MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>"); +MODULE_DESCRIPTION("Amlogic G12a To HDMI Tx Control Codec Driver"); +MODULE_LICENSE("GPL v2"); |