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-rw-r--r--sound/soc/codecs/da7219-aad.c2
-rw-r--r--sound/soc/codecs/hdmi-codec.c11
-rw-r--r--sound/soc/codecs/lpass-tx-macro.c11
-rw-r--r--sound/soc/fsl/Kconfig4
-rw-r--r--sound/soc/intel/avs/boards/da7219.c21
-rw-r--r--sound/soc/intel/avs/boards/max98357a.c22
-rw-r--r--sound/soc/intel/avs/boards/nau8825.c14
-rw-r--r--sound/soc/intel/avs/boards/rt5682.c22
-rw-r--r--sound/soc/intel/avs/boards/ssm4567.c31
-rw-r--r--sound/soc/intel/common/soc-acpi-intel-adl-match.c2
-rw-r--r--sound/soc/qcom/qdsp6/q6prm.c4
-rw-r--r--sound/soc/sof/intel/hda-ctrl.c3
-rw-r--r--sound/soc/sof/intel/hda-dsp.c12
-rw-r--r--sound/soc/sof/intel/pci-apl.c1
-rw-r--r--sound/soc/sof/intel/pci-cnl.c2
-rw-r--r--sound/soc/sof/intel/pci-icl.c1
-rw-r--r--sound/soc/sof/intel/pci-mtl.c1
-rw-r--r--sound/soc/sof/intel/pci-skl.c2
-rw-r--r--sound/soc/sof/intel/pci-tgl.c7
-rw-r--r--sound/soc/sof/intel/pci-tng.c6
-rw-r--r--sound/soc/sof/ipc3-topology.c32
-rw-r--r--sound/soc/sof/ipc3.c5
-rw-r--r--sound/soc/sof/ipc4-control.c3
-rw-r--r--sound/soc/sof/ipc4-topology.c21
-rw-r--r--sound/soc/sof/ipc4-topology.h8
-rw-r--r--sound/soc/sof/sof-audio.c32
-rw-r--r--sound/soc/sof/topology.c34
27 files changed, 228 insertions, 86 deletions
diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c
index 4a4f09f924bc..e3d398b8f54e 100644
--- a/sound/soc/codecs/da7219-aad.c
+++ b/sound/soc/codecs/da7219-aad.c
@@ -968,6 +968,8 @@ int da7219_aad_init(struct snd_soc_component *component)
INIT_WORK(&da7219_aad->hptest_work, da7219_aad_hptest_work);
INIT_WORK(&da7219_aad->jack_det_work, da7219_aad_jack_det_work);
+ mutex_init(&da7219_aad->jack_det_mutex);
+
ret = request_threaded_irq(da7219_aad->irq, da7219_aad_pre_irq_thread,
da7219_aad_irq_thread,
IRQF_TRIGGER_LOW | IRQF_ONESHOT,
diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c
index 01e8ffda2a4b..6d980fbc4207 100644
--- a/sound/soc/codecs/hdmi-codec.c
+++ b/sound/soc/codecs/hdmi-codec.c
@@ -428,8 +428,13 @@ static int hdmi_codec_startup(struct snd_pcm_substream *substream,
{
struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai);
bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+ bool has_capture = !hcp->hcd.no_i2s_capture;
+ bool has_playback = !hcp->hcd.no_i2s_playback;
int ret = 0;
+ if (!((has_playback && tx) || (has_capture && !tx)))
+ return 0;
+
mutex_lock(&hcp->lock);
if (hcp->busy) {
dev_err(dai->dev, "Only one simultaneous stream supported!\n");
@@ -468,6 +473,12 @@ static void hdmi_codec_shutdown(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai);
+ bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+ bool has_capture = !hcp->hcd.no_i2s_capture;
+ bool has_playback = !hcp->hcd.no_i2s_playback;
+
+ if (!((has_playback && tx) || (has_capture && !tx)))
+ return;
hcp->chmap_idx = HDMI_CODEC_CHMAP_IDX_UNKNOWN;
hcp->hcd.ops->audio_shutdown(dai->dev->parent, hcp->hcd.data);
diff --git a/sound/soc/codecs/lpass-tx-macro.c b/sound/soc/codecs/lpass-tx-macro.c
index bf27bdd5be20..473d3cd39554 100644
--- a/sound/soc/codecs/lpass-tx-macro.c
+++ b/sound/soc/codecs/lpass-tx-macro.c
@@ -242,7 +242,7 @@ enum {
struct tx_mute_work {
struct tx_macro *tx;
- u32 decimator;
+ u8 decimator;
struct delayed_work dwork;
};
@@ -635,7 +635,7 @@ exit:
return 0;
}
-static bool is_amic_enabled(struct snd_soc_component *component, int decimator)
+static bool is_amic_enabled(struct snd_soc_component *component, u8 decimator)
{
u16 adc_mux_reg, adc_reg, adc_n;
@@ -849,7 +849,7 @@ static int tx_macro_enable_dec(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm);
- unsigned int decimator;
+ u8 decimator;
u16 tx_vol_ctl_reg, dec_cfg_reg, hpf_gate_reg, tx_gain_ctl_reg;
u8 hpf_cut_off_freq;
int hpf_delay = TX_MACRO_DMIC_HPF_DELAY_MS;
@@ -1064,7 +1064,8 @@ static int tx_macro_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_component *component = dai->component;
- u32 decimator, sample_rate;
+ u32 sample_rate;
+ u8 decimator;
int tx_fs_rate;
struct tx_macro *tx = snd_soc_component_get_drvdata(component);
@@ -1128,7 +1129,7 @@ static int tx_macro_digital_mute(struct snd_soc_dai *dai, int mute, int stream)
{
struct snd_soc_component *component = dai->component;
struct tx_macro *tx = snd_soc_component_get_drvdata(component);
- u16 decimator;
+ u8 decimator;
/* active decimator not set yet */
if (tx->active_decimator[dai->id] == -1)
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index 17db29c25d96..725c530a3636 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -303,6 +303,10 @@ config SND_SOC_IMX_SGTL5000
Say Y if you want to add support for SoC audio on an i.MX board with
a sgtl5000 codec.
+ Note that this is an old driver. Consider enabling
+ SND_SOC_FSL_ASOC_CARD and SND_SOC_SGTL5000 to use the newer
+ driver.
+
config SND_SOC_IMX_SPDIF
tristate "SoC Audio support for i.MX boards with S/PDIF"
select SND_SOC_IMX_PCM_DMA
diff --git a/sound/soc/intel/avs/boards/da7219.c b/sound/soc/intel/avs/boards/da7219.c
index acd43b6108e9..1a1d572cc1d0 100644
--- a/sound/soc/intel/avs/boards/da7219.c
+++ b/sound/soc/intel/avs/boards/da7219.c
@@ -117,6 +117,26 @@ static void avs_da7219_codec_exit(struct snd_soc_pcm_runtime *rtd)
snd_soc_component_set_jack(asoc_rtd_to_codec(rtd, 0)->component, NULL, NULL);
}
+static int
+avs_da7219_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 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_create_dai_link(struct device *dev, const char *platform_name, int ssp_port,
struct snd_soc_dai_link **dai_link)
{
@@ -148,6 +168,7 @@ static int avs_create_dai_link(struct device *dev, const char *platform_name, in
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->be_hw_params_fixup = avs_da7219_be_fixup;
dl->init = avs_da7219_codec_init;
dl->exit = avs_da7219_codec_exit;
dl->nonatomic = 1;
diff --git a/sound/soc/intel/avs/boards/max98357a.c b/sound/soc/intel/avs/boards/max98357a.c
index 921f42caf7e0..183123d08c5a 100644
--- a/sound/soc/intel/avs/boards/max98357a.c
+++ b/sound/soc/intel/avs/boards/max98357a.c
@@ -8,6 +8,7 @@
#include <linux/module.h>
#include <linux/platform_device.h>
+#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/soc-acpi.h>
#include <sound/soc-dapm.h>
@@ -24,6 +25,26 @@ static const struct snd_soc_dapm_route card_base_routes[] = {
{ "Spk", NULL, "Speaker" },
};
+static int
+avs_max98357a_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 convert 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_create_dai_link(struct device *dev, const char *platform_name, int ssp_port,
struct snd_soc_dai_link **dai_link)
{
@@ -55,6 +76,7 @@ static int avs_create_dai_link(struct device *dev, const char *platform_name, in
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->be_hw_params_fixup = avs_max98357a_be_fixup;
dl->nonatomic = 1;
dl->no_pcm = 1;
dl->dpcm_playback = 1;
diff --git a/sound/soc/intel/avs/boards/nau8825.c b/sound/soc/intel/avs/boards/nau8825.c
index b31fa931ba8b..b69fc5567135 100644
--- a/sound/soc/intel/avs/boards/nau8825.c
+++ b/sound/soc/intel/avs/boards/nau8825.c
@@ -33,15 +33,15 @@ avs_nau8825_clock_control(struct snd_soc_dapm_widget *w, struct snd_kcontrol *co
return -EINVAL;
}
- if (!SND_SOC_DAPM_EVENT_ON(event)) {
+ if (SND_SOC_DAPM_EVENT_ON(event))
+ ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_MCLK, 24000000,
+ SND_SOC_CLOCK_IN);
+ 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;
- }
- }
+ if (ret < 0)
+ dev_err(card->dev, "Set sysclk failed: %d\n", ret);
- return 0;
+ return ret;
}
static const struct snd_kcontrol_new card_controls[] = {
diff --git a/sound/soc/intel/avs/boards/rt5682.c b/sound/soc/intel/avs/boards/rt5682.c
index 473e9fe5d0bf..b2c2ba93dcb5 100644
--- a/sound/soc/intel/avs/boards/rt5682.c
+++ b/sound/soc/intel/avs/boards/rt5682.c
@@ -169,6 +169,27 @@ static const struct snd_soc_ops avs_rt5682_ops = {
.hw_params = avs_rt5682_hw_params,
};
+static int
+avs_rt5682_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 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)
{
@@ -201,6 +222,7 @@ static int avs_create_dai_link(struct device *dev, const char *platform_name, in
dl->id = 0;
dl->init = avs_rt5682_codec_init;
dl->exit = avs_rt5682_codec_exit;
+ dl->be_hw_params_fixup = avs_rt5682_be_fixup;
dl->ops = &avs_rt5682_ops;
dl->nonatomic = 1;
dl->no_pcm = 1;
diff --git a/sound/soc/intel/avs/boards/ssm4567.c b/sound/soc/intel/avs/boards/ssm4567.c
index c5db69612762..2b7f5ad92aca 100644
--- a/sound/soc/intel/avs/boards/ssm4567.c
+++ b/sound/soc/intel/avs/boards/ssm4567.c
@@ -15,7 +15,6 @@
#include <sound/soc-acpi.h>
#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[] = {
@@ -34,41 +33,11 @@ static const struct snd_kcontrol_new card_controls[] = {
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[] = {
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 56ee5fef66a8..28dd2046e4ac 100644
--- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c
@@ -559,7 +559,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = {
{
.comp_ids = &essx_83x6,
.drv_name = "sof-essx8336",
- .sof_tplg_filename = "sof-adl-es83x6", /* the tplg suffix is added at run time */
+ .sof_tplg_filename = "sof-adl-es8336", /* the tplg suffix is added at run time */
.tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_SSP_NUMBER |
SND_SOC_ACPI_TPLG_INTEL_SSP_MSB |
SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER,
diff --git a/sound/soc/qcom/qdsp6/q6prm.c b/sound/soc/qcom/qdsp6/q6prm.c
index 3aa63aac4a68..81554d202658 100644
--- a/sound/soc/qcom/qdsp6/q6prm.c
+++ b/sound/soc/qcom/qdsp6/q6prm.c
@@ -184,9 +184,9 @@ int q6prm_set_lpass_clock(struct device *dev, int clk_id, int clk_attr, int clk_
unsigned int freq)
{
if (freq)
- return q6prm_request_lpass_clock(dev, clk_id, clk_attr, clk_attr, freq);
+ return q6prm_request_lpass_clock(dev, clk_id, clk_attr, clk_root, freq);
- return q6prm_release_lpass_clock(dev, clk_id, clk_attr, clk_attr, freq);
+ return q6prm_release_lpass_clock(dev, clk_id, clk_attr, clk_root, freq);
}
EXPORT_SYMBOL_GPL(q6prm_set_lpass_clock);
diff --git a/sound/soc/sof/intel/hda-ctrl.c b/sound/soc/sof/intel/hda-ctrl.c
index a1037512da1f..e1dba6b79065 100644
--- a/sound/soc/sof/intel/hda-ctrl.c
+++ b/sound/soc/sof/intel/hda-ctrl.c
@@ -198,12 +198,15 @@ int hda_dsp_ctrl_init_chip(struct snd_sof_dev *sdev)
goto err;
}
+ usleep_range(500, 1000);
+
/* exit HDA controller reset */
ret = hda_dsp_ctrl_link_reset(sdev, false);
if (ret < 0) {
dev_err(sdev->dev, "error: failed to exit HDA controller reset\n");
goto err;
}
+ usleep_range(1000, 1200);
hda_codec_detect_mask(sdev);
diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c
index e3b69dbc1308..c9231aeacc53 100644
--- a/sound/soc/sof/intel/hda-dsp.c
+++ b/sound/soc/sof/intel/hda-dsp.c
@@ -392,6 +392,12 @@ static int hda_dsp_update_d0i3c_register(struct snd_sof_dev *sdev, u8 value)
snd_sof_dsp_update8(sdev, HDA_DSP_HDA_BAR, chip->d0i3_offset,
SOF_HDA_VS_D0I3C_I3, value);
+ /*
+ * The value written to the D0I3C::I3 bit may not be taken into account immediately.
+ * A delay is recommended before checking if D0I3C::CIP is cleared
+ */
+ usleep_range(30, 40);
+
/* Wait for cmd in progress to be cleared before exiting the function */
ret = hda_dsp_wait_d0i3c_done(sdev);
if (ret < 0) {
@@ -400,6 +406,12 @@ static int hda_dsp_update_d0i3c_register(struct snd_sof_dev *sdev, u8 value)
}
reg = snd_sof_dsp_read8(sdev, HDA_DSP_HDA_BAR, chip->d0i3_offset);
+ /* Confirm d0i3 state changed with paranoia check */
+ if ((reg ^ value) & SOF_HDA_VS_D0I3C_I3) {
+ dev_err(sdev->dev, "failed to update D0I3C!\n");
+ return -EIO;
+ }
+
trace_sof_intel_D0I3C_updated(sdev, reg);
return 0;
diff --git a/sound/soc/sof/intel/pci-apl.c b/sound/soc/sof/intel/pci-apl.c
index 69279dcc92dc..aff6cb573c27 100644
--- a/sound/soc/sof/intel/pci-apl.c
+++ b/sound/soc/sof/intel/pci-apl.c
@@ -78,6 +78,7 @@ static const struct sof_dev_desc glk_desc = {
.nocodec_tplg_filename = "sof-glk-nocodec.tplg",
.ops = &sof_apl_ops,
.ops_init = sof_apl_ops_init,
+ .ops_free = hda_ops_free,
};
/* PCI IDs */
diff --git a/sound/soc/sof/intel/pci-cnl.c b/sound/soc/sof/intel/pci-cnl.c
index 8db3f8d15b55..4c0c1c369dcd 100644
--- a/sound/soc/sof/intel/pci-cnl.c
+++ b/sound/soc/sof/intel/pci-cnl.c
@@ -48,6 +48,7 @@ static const struct sof_dev_desc cnl_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 cfl_desc = {
@@ -111,6 +112,7 @@ static const struct sof_dev_desc cml_desc = {
.nocodec_tplg_filename = "sof-cnl-nocodec.tplg",
.ops = &sof_cnl_ops,
.ops_init = sof_cnl_ops_init,
+ .ops_free = hda_ops_free,
};
/* PCI IDs */
diff --git a/sound/soc/sof/intel/pci-icl.c b/sound/soc/sof/intel/pci-icl.c
index d6cf75e357db..6785669113b3 100644
--- a/sound/soc/sof/intel/pci-icl.c
+++ b/sound/soc/sof/intel/pci-icl.c
@@ -79,6 +79,7 @@ static const struct sof_dev_desc jsl_desc = {
.nocodec_tplg_filename = "sof-jsl-nocodec.tplg",
.ops = &sof_cnl_ops,
.ops_init = sof_cnl_ops_init,
+ .ops_free = hda_ops_free,
};
/* PCI IDs */
diff --git a/sound/soc/sof/intel/pci-mtl.c b/sound/soc/sof/intel/pci-mtl.c
index 6e4e6d4ef5a5..b183dc0014b4 100644
--- a/sound/soc/sof/intel/pci-mtl.c
+++ b/sound/soc/sof/intel/pci-mtl.c
@@ -46,6 +46,7 @@ static const struct sof_dev_desc mtl_desc = {
.nocodec_tplg_filename = "sof-mtl-nocodec.tplg",
.ops = &sof_mtl_ops,
.ops_init = sof_mtl_ops_init,
+ .ops_free = hda_ops_free,
};
/* PCI IDs */
diff --git a/sound/soc/sof/intel/pci-skl.c b/sound/soc/sof/intel/pci-skl.c
index 3a99dc444f92..5b4bccf81965 100644
--- a/sound/soc/sof/intel/pci-skl.c
+++ b/sound/soc/sof/intel/pci-skl.c
@@ -38,6 +38,7 @@ static struct sof_dev_desc skl_desc = {
.nocodec_tplg_filename = "sof-skl-nocodec.tplg",
.ops = &sof_skl_ops,
.ops_init = sof_skl_ops_init,
+ .ops_free = hda_ops_free,
};
static struct sof_dev_desc kbl_desc = {
@@ -61,6 +62,7 @@ static struct sof_dev_desc kbl_desc = {
.nocodec_tplg_filename = "sof-kbl-nocodec.tplg",
.ops = &sof_skl_ops,
.ops_init = sof_skl_ops_init,
+ .ops_free = hda_ops_free,
};
/* PCI IDs */
diff --git a/sound/soc/sof/intel/pci-tgl.c b/sound/soc/sof/intel/pci-tgl.c
index e80c4dfef85a..22e769e0831d 100644
--- a/sound/soc/sof/intel/pci-tgl.c
+++ b/sound/soc/sof/intel/pci-tgl.c
@@ -48,6 +48,7 @@ static const struct sof_dev_desc tgl_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 tglh_desc = {
@@ -110,6 +111,7 @@ static const struct sof_dev_desc ehl_desc = {
.nocodec_tplg_filename = "sof-ehl-nocodec.tplg",
.ops = &sof_tgl_ops,
.ops_init = sof_tgl_ops_init,
+ .ops_free = hda_ops_free,
};
static const struct sof_dev_desc adls_desc = {
@@ -141,6 +143,7 @@ static const struct sof_dev_desc adls_desc = {
.nocodec_tplg_filename = "sof-adl-nocodec.tplg",
.ops = &sof_tgl_ops,
.ops_init = sof_tgl_ops_init,
+ .ops_free = hda_ops_free,
};
static const struct sof_dev_desc adl_desc = {
@@ -172,6 +175,7 @@ static const struct sof_dev_desc adl_desc = {
.nocodec_tplg_filename = "sof-adl-nocodec.tplg",
.ops = &sof_tgl_ops,
.ops_init = sof_tgl_ops_init,
+ .ops_free = hda_ops_free,
};
static const struct sof_dev_desc adl_n_desc = {
@@ -203,6 +207,7 @@ static const struct sof_dev_desc adl_n_desc = {
.nocodec_tplg_filename = "sof-adl-nocodec.tplg",
.ops = &sof_tgl_ops,
.ops_init = sof_tgl_ops_init,
+ .ops_free = hda_ops_free,
};
static const struct sof_dev_desc rpls_desc = {
@@ -234,6 +239,7 @@ static const struct sof_dev_desc rpls_desc = {
.nocodec_tplg_filename = "sof-rpl-nocodec.tplg",
.ops = &sof_tgl_ops,
.ops_init = sof_tgl_ops_init,
+ .ops_free = hda_ops_free,
};
static const struct sof_dev_desc rpl_desc = {
@@ -265,6 +271,7 @@ static const struct sof_dev_desc rpl_desc = {
.nocodec_tplg_filename = "sof-rpl-nocodec.tplg",
.ops = &sof_tgl_ops,
.ops_init = sof_tgl_ops_init,
+ .ops_free = hda_ops_free,
};
/* PCI IDs */
diff --git a/sound/soc/sof/intel/pci-tng.c b/sound/soc/sof/intel/pci-tng.c
index 5b2b409752c5..8c22a00266c0 100644
--- a/sound/soc/sof/intel/pci-tng.c
+++ b/sound/soc/sof/intel/pci-tng.c
@@ -75,11 +75,7 @@ static int tangier_pci_probe(struct snd_sof_dev *sdev)
/* LPE base */
base = pci_resource_start(pci, desc->resindex_lpe_base) - IRAM_OFFSET;
- size = pci_resource_len(pci, desc->resindex_lpe_base);
- if (size < PCI_BAR_SIZE) {
- dev_err(sdev->dev, "error: I/O region is too small.\n");
- return -ENODEV;
- }
+ size = PCI_BAR_SIZE;
dev_dbg(sdev->dev, "LPE PHY base at 0x%x size 0x%x", base, size);
sdev->bar[DSP_BAR] = devm_ioremap(sdev->dev, base, size);
diff --git a/sound/soc/sof/ipc3-topology.c b/sound/soc/sof/ipc3-topology.c
index dceb78bfe17c..b1f425b39db9 100644
--- a/sound/soc/sof/ipc3-topology.c
+++ b/sound/soc/sof/ipc3-topology.c
@@ -2081,7 +2081,9 @@ static int sof_ipc3_dai_config(struct snd_sof_dev *sdev, struct snd_sof_widget *
break;
case SOF_DAI_INTEL_ALH:
if (data) {
- config->dai_index = data->dai_index;
+ /* save the dai_index during hw_params and reuse it for hw_free */
+ if (flags & SOF_DAI_CONFIG_FLAGS_HW_PARAMS)
+ config->dai_index = data->dai_index;
config->alh.stream_id = data->dai_data;
}
break;
@@ -2089,7 +2091,30 @@ static int sof_ipc3_dai_config(struct snd_sof_dev *sdev, struct snd_sof_widget *
break;
}
- config->flags = flags;
+ /*
+ * The dai_config op is invoked several times and the flags argument varies as below:
+ * BE DAI hw_params: When the op is invoked during the BE DAI hw_params, flags contains
+ * SOF_DAI_CONFIG_FLAGS_HW_PARAMS along with quirks
+ * FE DAI hw_params: When invoked during FE DAI hw_params after the DAI widget has
+ * just been set up in the DSP, flags is set to SOF_DAI_CONFIG_FLAGS_HW_PARAMS with no
+ * quirks
+ * BE DAI trigger: When invoked during the BE DAI trigger, flags is set to
+ * SOF_DAI_CONFIG_FLAGS_PAUSE and contains no quirks
+ * BE DAI hw_free: When invoked during the BE DAI hw_free, flags is set to
+ * SOF_DAI_CONFIG_FLAGS_HW_FREE and contains no quirks
+ * FE DAI hw_free: When invoked during the FE DAI hw_free, flags is set to
+ * SOF_DAI_CONFIG_FLAGS_HW_FREE and contains no quirks
+ *
+ * The DAI_CONFIG IPC is sent to the DSP, only after the widget is set up during the FE
+ * DAI hw_params. But since the BE DAI hw_params precedes the FE DAI hw_params, the quirks
+ * need to be preserved when assigning the flags before sending the IPC.
+ * For the case of PAUSE/HW_FREE, since there are no quirks, flags can be used as is.
+ */
+
+ if (flags & SOF_DAI_CONFIG_FLAGS_HW_PARAMS)
+ config->flags |= flags;
+ else
+ config->flags = flags;
/* only send the IPC if the widget is set up in the DSP */
if (swidget->use_count > 0) {
@@ -2097,6 +2122,9 @@ static int sof_ipc3_dai_config(struct snd_sof_dev *sdev, struct snd_sof_widget *
&reply, sizeof(reply));
if (ret < 0)
dev_err(sdev->dev, "Failed to set dai config for %s\n", dai->name);
+
+ /* clear the flags once the IPC has been sent even if it fails */
+ config->flags = SOF_DAI_CONFIG_FLAGS_NONE;
}
return ret;
diff --git a/sound/soc/sof/ipc3.c b/sound/soc/sof/ipc3.c
index 3de64ea2dc9a..4493bbd7faf1 100644
--- a/sound/soc/sof/ipc3.c
+++ b/sound/soc/sof/ipc3.c
@@ -970,8 +970,9 @@ static void sof_ipc3_rx_msg(struct snd_sof_dev *sdev)
return;
}
- if (hdr.size < sizeof(hdr)) {
- dev_err(sdev->dev, "The received message size is invalid\n");
+ if (hdr.size < sizeof(hdr) || hdr.size > SOF_IPC_MSG_MAX_SIZE) {
+ dev_err(sdev->dev, "The received message size is invalid: %u\n",
+ hdr.size);
return;
}
diff --git a/sound/soc/sof/ipc4-control.c b/sound/soc/sof/ipc4-control.c
index 67bd2233fd9a..9a71af1a613a 100644
--- a/sound/soc/sof/ipc4-control.c
+++ b/sound/soc/sof/ipc4-control.c
@@ -97,7 +97,8 @@ sof_ipc4_set_volume_data(struct snd_sof_dev *sdev, struct snd_sof_widget *swidge
}
/* set curve type and duration from topology */
- data.curve_duration = gain->data.curve_duration;
+ data.curve_duration_l = gain->data.curve_duration_l;
+ data.curve_duration_h = gain->data.curve_duration_h;
data.curve_type = gain->data.curve_type;
msg->data_ptr = &data;
diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c
index cb75f685e722..0bb16ed38e48 100644
--- a/sound/soc/sof/ipc4-topology.c
+++ b/sound/soc/sof/ipc4-topology.c
@@ -111,7 +111,7 @@ static const struct sof_topology_token gain_tokens[] = {
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)},
+ offsetof(struct sof_ipc4_gain_data, curve_duration_l)},
{SOF_TKN_GAIN_VAL, SND_SOC_TPLG_TUPLE_TYPE_WORD,
get_token_u32, offsetof(struct sof_ipc4_gain_data, init_val)},
};
@@ -159,7 +159,7 @@ static void sof_ipc4_dbg_audio_format(struct device *dev,
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",
+ " #%d: %uHz, %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);
}
@@ -715,7 +715,7 @@ static int sof_ipc4_widget_setup_comp_pga(struct snd_sof_widget *swidget)
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,
+ swidget->widget->name, gain->data.curve_type, gain->data.curve_duration_l,
gain->data.init_val, gain->base_config.cpc);
ret = sof_ipc4_widget_setup_msg(swidget, &gain->msg);
@@ -1003,6 +1003,7 @@ static void sof_ipc4_unprepare_copier_module(struct snd_sof_widget *swidget)
ipc4_copier = dai->private;
if (ipc4_copier->dai_type == SOF_DAI_INTEL_ALH) {
+ struct sof_ipc4_copier_data *copier_data = &ipc4_copier->data;
struct sof_ipc4_alh_configuration_blob *blob;
unsigned int group_id;
@@ -1012,6 +1013,9 @@ static void sof_ipc4_unprepare_copier_module(struct snd_sof_widget *swidget)
ALH_MULTI_GTW_BASE;
ida_free(&alh_group_ida, group_id);
}
+
+ /* clear the node ID */
+ copier_data->gtw_cfg.node_id &= ~SOF_IPC4_NODE_INDEX_MASK;
}
}
@@ -1963,8 +1967,15 @@ static int sof_ipc4_dai_config(struct snd_sof_dev *sdev, struct snd_sof_widget *
pipeline->skip_during_fe_trigger = true;
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);
+ /*
+ * Do not clear the node ID when this op is invoked with
+ * SOF_DAI_CONFIG_FLAGS_HW_FREE. It is needed to free the group_ida during
+ * unprepare.
+ */
+ if (flags & SOF_DAI_CONFIG_FLAGS_HW_PARAMS) {
+ 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:
diff --git a/sound/soc/sof/ipc4-topology.h b/sound/soc/sof/ipc4-topology.h
index 7877f0638011..22900b572766 100644
--- a/sound/soc/sof/ipc4-topology.h
+++ b/sound/soc/sof/ipc4-topology.h
@@ -46,7 +46,7 @@
#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_NODE_INDEX_INTEL_DMIC(x) ((x) & 0x7)
#define SOF_IPC4_GAIN_ALL_CHANNELS_MASK 0xffffffff
#define SOF_IPC4_VOL_ZERO_DB 0x7fffffff
@@ -279,14 +279,16 @@ struct sof_ipc4_control_data {
* @init_val: Initial value
* @curve_type: Curve type
* @reserved: reserved for future use
- * @curve_duration: Curve duration
+ * @curve_duration_l: Curve duration low part
+ * @curve_duration_h: Curve duration high part
*/
struct sof_ipc4_gain_data {
uint32_t channels;
uint32_t init_val;
uint32_t curve_type;
uint32_t reserved;
- uint32_t curve_duration;
+ uint32_t curve_duration_l;
+ uint32_t curve_duration_h;
} __aligned(8);
/**
diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c
index 760621bfc802..6de388a8d0b8 100644
--- a/sound/soc/sof/sof-audio.c
+++ b/sound/soc/sof/sof-audio.c
@@ -50,9 +50,27 @@ static int sof_widget_free_unlocked(struct snd_sof_dev *sdev,
/* reset route setup status for all routes that contain this widget */
sof_reset_route_setup_status(sdev, swidget);
+ /* free DAI config and continue to free widget even if it fails */
+ if (WIDGET_IS_DAI(swidget->id)) {
+ struct snd_sof_dai_config_data data;
+ unsigned int flags = SOF_DAI_CONFIG_FLAGS_HW_FREE;
+
+ data.dai_data = DMA_CHAN_INVALID;
+
+ if (tplg_ops && tplg_ops->dai_config) {
+ err = tplg_ops->dai_config(sdev, swidget, flags, &data);
+ if (err < 0)
+ dev_err(sdev->dev, "failed to free config for widget %s\n",
+ swidget->widget->name);
+ }
+ }
+
/* continue to disable core even if IPC fails */
- if (tplg_ops && tplg_ops->widget_free)
- err = tplg_ops->widget_free(sdev, swidget);
+ if (tplg_ops && tplg_ops->widget_free) {
+ ret = tplg_ops->widget_free(sdev, swidget);
+ if (ret < 0 && !err)
+ err = ret;
+ }
/*
* disable widget core. continue to route setup status and complete flag
@@ -151,8 +169,12 @@ static int sof_widget_setup_unlocked(struct snd_sof_dev *sdev,
/* send config for DAI components */
if (WIDGET_IS_DAI(swidget->id)) {
- unsigned int flags = SOF_DAI_CONFIG_FLAGS_NONE;
+ unsigned int flags = SOF_DAI_CONFIG_FLAGS_HW_PARAMS;
+ /*
+ * The config flags saved during BE DAI hw_params will be used for IPC3. IPC4 does
+ * not use the flags argument.
+ */
if (tplg_ops && tplg_ops->dai_config) {
ret = tplg_ops->dai_config(sdev, swidget, flags, NULL);
if (ret < 0)
@@ -588,8 +610,8 @@ int sof_widget_list_setup(struct snd_sof_dev *sdev, struct snd_sof_pcm *spcm,
ret = sof_walk_widgets_in_order(sdev, spcm, fe_params, platform_params,
dir, SOF_WIDGET_SETUP);
if (ret < 0) {
- ret = sof_walk_widgets_in_order(sdev, spcm, fe_params, platform_params,
- dir, SOF_WIDGET_UNPREPARE);
+ sof_walk_widgets_in_order(sdev, spcm, fe_params, platform_params,
+ dir, SOF_WIDGET_UNPREPARE);
return ret;
}
diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c
index 4a62ccc71fcb..9f3a038fe21a 100644
--- a/sound/soc/sof/topology.c
+++ b/sound/soc/sof/topology.c
@@ -1388,14 +1388,15 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index,
if (ret < 0) {
dev_err(scomp->dev, "failed to parse component pin tokens for %s\n",
w->name);
- return ret;
+ goto widget_free;
}
if (swidget->num_sink_pins > SOF_WIDGET_MAX_NUM_PINS ||
swidget->num_source_pins > SOF_WIDGET_MAX_NUM_PINS) {
dev_err(scomp->dev, "invalid pins for %s: [sink: %d, src: %d]\n",
swidget->widget->name, swidget->num_sink_pins, swidget->num_source_pins);
- return -EINVAL;
+ ret = -EINVAL;
+ goto widget_free;
}
if (swidget->num_sink_pins > 1) {
@@ -1404,7 +1405,7 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index,
if (ret < 0) {
dev_err(scomp->dev, "failed to parse sink pin binding for %s\n",
w->name);
- return ret;
+ goto widget_free;
}
}
@@ -1414,7 +1415,7 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index,
if (ret < 0) {
dev_err(scomp->dev, "failed to parse source pin binding for %s\n",
w->name);
- return ret;
+ goto widget_free;
}
}
@@ -1436,9 +1437,8 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index,
case snd_soc_dapm_dai_out:
dai = kzalloc(sizeof(*dai), GFP_KERNEL);
if (!dai) {
- kfree(swidget);
- return -ENOMEM;
-
+ ret = -ENOMEM;
+ goto widget_free;
}
ret = sof_widget_parse_tokens(scomp, swidget, tw, token_list, token_list_size);
@@ -1496,8 +1496,7 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index,
tw->shift, swidget->id, tw->name,
strnlen(tw->sname, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) > 0
? tw->sname : "none");
- kfree(swidget);
- return ret;
+ goto widget_free;
}
if (sof_debug_check_flag(SOF_DBG_DISABLE_MULTICORE)) {
@@ -1518,10 +1517,7 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index,
if (ret) {
dev_err(scomp->dev, "widget event binding failed for %s\n",
swidget->widget->name);
- kfree(swidget->private);
- kfree(swidget->tuples);
- kfree(swidget);
- return ret;
+ goto free;
}
}
}
@@ -1532,10 +1528,8 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index,
spipe = kzalloc(sizeof(*spipe), GFP_KERNEL);
if (!spipe) {
- kfree(swidget->private);
- kfree(swidget->tuples);
- kfree(swidget);
- return -ENOMEM;
+ ret = -ENOMEM;
+ goto free;
}
spipe->pipe_widget = swidget;
@@ -1546,6 +1540,12 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index,
w->dobj.private = swidget;
list_add(&swidget->list, &sdev->widget_list);
return ret;
+free:
+ kfree(swidget->private);
+ kfree(swidget->tuples);
+widget_free:
+ kfree(swidget);
+ return ret;
}
static int sof_route_unload(struct snd_soc_component *scomp,