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authorGreg Kroah-Hartman <gregkh@linuxfoundation.org>2025-09-25 11:13:51 +0200
committerGreg Kroah-Hartman <gregkh@linuxfoundation.org>2025-09-25 11:13:51 +0200
commit56fb05093756ed55ba1cdf5d432a68004da67860 (patch)
tree87dc333d4f606f375d6253eb5b8ef6f04674ffa6 /sound/soc
parentb6a153b0829afbc63032e8271d3ca9a19e704e03 (diff)
parentda274362a7bd9ab3a6e46d15945029145ebce672 (diff)
Merge v6.12.49linux-rolling-lts
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Diffstat (limited to 'sound/soc')
-rw-r--r--sound/soc/codecs/wm8940.c9
-rw-r--r--sound/soc/codecs/wm8974.c8
-rw-r--r--sound/soc/intel/catpt/pcm.c23
-rw-r--r--sound/soc/qcom/qdsp6/audioreach.c1
-rw-r--r--sound/soc/qcom/qdsp6/q6apm-lpass-dais.c7
-rw-r--r--sound/soc/sof/intel/hda-stream.c2
6 files changed, 37 insertions, 13 deletions
diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c
index 8a532f7d750c..808a4d4b6f80 100644
--- a/sound/soc/codecs/wm8940.c
+++ b/sound/soc/codecs/wm8940.c
@@ -220,7 +220,7 @@ static const struct snd_kcontrol_new wm8940_snd_controls[] = {
SOC_SINGLE_TLV("Digital Capture Volume", WM8940_ADCVOL,
0, 255, 0, wm8940_adc_tlv),
SOC_ENUM("Mic Bias Level", wm8940_mic_bias_level_enum),
- SOC_SINGLE_TLV("Capture Boost Volue", WM8940_ADCBOOST,
+ SOC_SINGLE_TLV("Capture Boost Volume", WM8940_ADCBOOST,
8, 1, 0, wm8940_capture_boost_vol_tlv),
SOC_SINGLE_TLV("Speaker Playback Volume", WM8940_SPKVOL,
0, 63, 0, wm8940_spk_vol_tlv),
@@ -693,7 +693,12 @@ static int wm8940_update_clocks(struct snd_soc_dai *dai)
f = wm8940_get_mclkdiv(priv->mclk, fs256, &mclkdiv);
if (f != priv->mclk) {
/* The PLL performs best around 90MHz */
- fpll = wm8940_get_mclkdiv(22500000, fs256, &mclkdiv);
+ if (fs256 % 8000)
+ f = 22579200;
+ else
+ f = 24576000;
+
+ fpll = wm8940_get_mclkdiv(f, fs256, &mclkdiv);
}
wm8940_set_dai_pll(dai, 0, 0, priv->mclk, fpll);
diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c
index 0ee3655cad01..c0a8fc867301 100644
--- a/sound/soc/codecs/wm8974.c
+++ b/sound/soc/codecs/wm8974.c
@@ -419,10 +419,14 @@ static int wm8974_update_clocks(struct snd_soc_dai *dai)
fs256 = 256 * priv->fs;
f = wm8974_get_mclkdiv(priv->mclk, fs256, &mclkdiv);
-
if (f != priv->mclk) {
/* The PLL performs best around 90MHz */
- fpll = wm8974_get_mclkdiv(22500000, fs256, &mclkdiv);
+ if (fs256 % 8000)
+ f = 22579200;
+ else
+ f = 24576000;
+
+ fpll = wm8974_get_mclkdiv(f, fs256, &mclkdiv);
}
wm8974_set_dai_pll(dai, 0, 0, priv->mclk, fpll);
diff --git a/sound/soc/intel/catpt/pcm.c b/sound/soc/intel/catpt/pcm.c
index 81a2f0339e05..ff1fa01acb85 100644
--- a/sound/soc/intel/catpt/pcm.c
+++ b/sound/soc/intel/catpt/pcm.c
@@ -568,8 +568,9 @@ static const struct snd_pcm_hardware catpt_pcm_hardware = {
SNDRV_PCM_INFO_RESUME |
SNDRV_PCM_INFO_NO_PERIOD_WAKEUP,
.formats = SNDRV_PCM_FMTBIT_S16_LE |
- SNDRV_PCM_FMTBIT_S24_LE |
SNDRV_PCM_FMTBIT_S32_LE,
+ .subformats = SNDRV_PCM_SUBFMTBIT_MSBITS_24 |
+ SNDRV_PCM_SUBFMTBIT_MSBITS_MAX,
.period_bytes_min = PAGE_SIZE,
.period_bytes_max = CATPT_BUFFER_MAX_SIZE / CATPT_PCM_PERIODS_MIN,
.periods_min = CATPT_PCM_PERIODS_MIN,
@@ -699,14 +700,18 @@ static struct snd_soc_dai_driver dai_drivers[] = {
.channels_min = 2,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_48000,
- .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE,
+ .subformats = SNDRV_PCM_SUBFMTBIT_MSBITS_24 |
+ SNDRV_PCM_SUBFMTBIT_MSBITS_MAX,
},
.capture = {
.stream_name = "Analog Capture",
.channels_min = 2,
.channels_max = 4,
.rates = SNDRV_PCM_RATE_48000,
- .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE,
+ .subformats = SNDRV_PCM_SUBFMTBIT_MSBITS_24 |
+ SNDRV_PCM_SUBFMTBIT_MSBITS_MAX,
},
},
{
@@ -718,7 +723,9 @@ static struct snd_soc_dai_driver dai_drivers[] = {
.channels_min = 2,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_192000,
- .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE,
+ .subformats = SNDRV_PCM_SUBFMTBIT_MSBITS_24 |
+ SNDRV_PCM_SUBFMTBIT_MSBITS_MAX,
},
},
{
@@ -730,7 +737,9 @@ static struct snd_soc_dai_driver dai_drivers[] = {
.channels_min = 2,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_192000,
- .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE,
+ .subformats = SNDRV_PCM_SUBFMTBIT_MSBITS_24 |
+ SNDRV_PCM_SUBFMTBIT_MSBITS_MAX,
},
},
{
@@ -742,7 +751,9 @@ static struct snd_soc_dai_driver dai_drivers[] = {
.channels_min = 2,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_48000,
- .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE,
+ .subformats = SNDRV_PCM_SUBFMTBIT_MSBITS_24 |
+ SNDRV_PCM_SUBFMTBIT_MSBITS_MAX,
},
},
{
diff --git a/sound/soc/qcom/qdsp6/audioreach.c b/sound/soc/qcom/qdsp6/audioreach.c
index 4ebaaf736fb9..3f5eed5afce5 100644
--- a/sound/soc/qcom/qdsp6/audioreach.c
+++ b/sound/soc/qcom/qdsp6/audioreach.c
@@ -971,6 +971,7 @@ static int audioreach_i2s_set_media_format(struct q6apm_graph *graph,
param_data->param_id = PARAM_ID_I2S_INTF_CFG;
param_data->param_size = ic_sz - APM_MODULE_PARAM_DATA_SIZE;
+ intf_cfg->cfg.lpaif_type = module->hw_interface_type;
intf_cfg->cfg.intf_idx = module->hw_interface_idx;
intf_cfg->cfg.sd_line_idx = module->sd_line_idx;
diff --git a/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c b/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c
index 9c98a35ad099..b46aff1110e1 100644
--- a/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c
+++ b/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c
@@ -213,8 +213,10 @@ static int q6apm_lpass_dai_prepare(struct snd_pcm_substream *substream, struct s
return 0;
err:
- q6apm_graph_close(dai_data->graph[dai->id]);
- dai_data->graph[dai->id] = NULL;
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ q6apm_graph_close(dai_data->graph[dai->id]);
+ dai_data->graph[dai->id] = NULL;
+ }
return rc;
}
@@ -260,6 +262,7 @@ 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,
};
static const struct snd_soc_dai_ops q6hdmi_ops = {
diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c
index 3ac63ce67ab1..24f3cc767614 100644
--- a/sound/soc/sof/intel/hda-stream.c
+++ b/sound/soc/sof/intel/hda-stream.c
@@ -864,7 +864,7 @@ int hda_dsp_stream_init(struct snd_sof_dev *sdev)
if (num_capture >= SOF_HDA_CAPTURE_STREAMS) {
dev_err(sdev->dev, "error: too many capture streams %d\n",
- num_playback);
+ num_capture);
return -EINVAL;
}