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-rw-r--r--sound/soc/intel/boards/bytcr_rt5640.c9
-rw-r--r--sound/soc/intel/boards/kbl_da7219_max98927.c55
-rw-r--r--sound/soc/intel/boards/sof_pcm512x.c13
3 files changed, 24 insertions, 53 deletions
diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c
index ca14730232ba..43ee3d095a1b 100644
--- a/sound/soc/intel/boards/bytcr_rt5640.c
+++ b/sound/soc/intel/boards/bytcr_rt5640.c
@@ -286,9 +286,6 @@ static const struct snd_soc_dapm_widget byt_rt5640_widgets[] = {
static const struct snd_soc_dapm_route byt_rt5640_audio_map[] = {
{"Headphone", NULL, "Platform Clock"},
{"Headset Mic", NULL, "Platform Clock"},
- {"Internal Mic", NULL, "Platform Clock"},
- {"Speaker", NULL, "Platform Clock"},
-
{"Headset Mic", NULL, "MICBIAS1"},
{"IN2P", NULL, "Headset Mic"},
{"Headphone", NULL, "HPOL"},
@@ -296,19 +293,23 @@ static const struct snd_soc_dapm_route byt_rt5640_audio_map[] = {
};
static const struct snd_soc_dapm_route byt_rt5640_intmic_dmic1_map[] = {
+ {"Internal Mic", NULL, "Platform Clock"},
{"DMIC1", NULL, "Internal Mic"},
};
static const struct snd_soc_dapm_route byt_rt5640_intmic_dmic2_map[] = {
+ {"Internal Mic", NULL, "Platform Clock"},
{"DMIC2", NULL, "Internal Mic"},
};
static const struct snd_soc_dapm_route byt_rt5640_intmic_in1_map[] = {
+ {"Internal Mic", NULL, "Platform Clock"},
{"Internal Mic", NULL, "MICBIAS1"},
{"IN1P", NULL, "Internal Mic"},
};
static const struct snd_soc_dapm_route byt_rt5640_intmic_in3_map[] = {
+ {"Internal Mic", NULL, "Platform Clock"},
{"Internal Mic", NULL, "MICBIAS1"},
{"IN3P", NULL, "Internal Mic"},
};
@@ -350,6 +351,7 @@ static const struct snd_soc_dapm_route byt_rt5640_ssp0_aif2_map[] = {
};
static const struct snd_soc_dapm_route byt_rt5640_stereo_spk_map[] = {
+ {"Speaker", NULL, "Platform Clock"},
{"Speaker", NULL, "SPOLP"},
{"Speaker", NULL, "SPOLN"},
{"Speaker", NULL, "SPORP"},
@@ -357,6 +359,7 @@ static const struct snd_soc_dapm_route byt_rt5640_stereo_spk_map[] = {
};
static const struct snd_soc_dapm_route byt_rt5640_mono_spk_map[] = {
+ {"Speaker", NULL, "Platform Clock"},
{"Speaker", NULL, "SPOLP"},
{"Speaker", NULL, "SPOLN"},
};
diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c
index e0149cf6127d..884741aa4833 100644
--- a/sound/soc/intel/boards/kbl_da7219_max98927.c
+++ b/sound/soc/intel/boards/kbl_da7219_max98927.c
@@ -197,7 +197,7 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream,
}
if (!strcmp(codec_dai->component->name, MAX98373_DEV0_NAME)) {
ret = snd_soc_dai_set_tdm_slot(codec_dai,
- 0x03, 3, 8, 24);
+ 0x30, 3, 8, 16);
if (ret < 0) {
dev_err(runtime->dev,
"DEV0 TDM slot err:%d\n", ret);
@@ -206,10 +206,10 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream,
}
if (!strcmp(codec_dai->component->name, MAX98373_DEV1_NAME)) {
ret = snd_soc_dai_set_tdm_slot(codec_dai,
- 0x0C, 3, 8, 24);
+ 0xC0, 3, 8, 16);
if (ret < 0) {
dev_err(runtime->dev,
- "DEV0 TDM slot err:%d\n", ret);
+ "DEV1 TDM slot err:%d\n", ret);
return ret;
}
}
@@ -309,24 +309,6 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
* The above 2 loops are mutually exclusive based on the stream direction,
* thus rtd_dpcm variable will never be overwritten
*/
- /*
- * Topology for kblda7219m98373 & kblmax98373 supports only S24_LE,
- * where as kblda7219m98927 & kblmax98927 supports S16_LE by default.
- * Skipping the port wise FE and BE configuration for kblda7219m98373 &
- * kblmax98373 as the topology (FE & BE) supports S24_LE only.
- */
-
- if (!strcmp(rtd->card->name, "kblda7219m98373") ||
- !strcmp(rtd->card->name, "kblmax98373")) {
- /* The ADSP will convert the FE rate to 48k, stereo */
- rate->min = rate->max = 48000;
- chan->min = chan->max = DUAL_CHANNEL;
-
- /* set SSP to 24 bit */
- snd_mask_none(fmt);
- snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE);
- return 0;
- }
/*
* The ADSP will convert the FE rate to 48k, stereo, 24 bit
@@ -477,31 +459,20 @@ static struct snd_pcm_hw_constraint_list constraints_channels_quad = {
static int kbl_fe_startup(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
- struct snd_soc_pcm_runtime *soc_rt = asoc_substream_to_rtd(substream);
/*
* On this platform for PCM device we support,
* 48Khz
* stereo
+ * 16 bit audio
*/
runtime->hw.channels_max = DUAL_CHANNEL;
snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
&constraints_channels);
- /*
- * Setup S24_LE (32 bit container and 24 bit valid data) for
- * kblda7219m98373 & kblmax98373. For kblda7219m98927 &
- * kblmax98927 keeping it as 16/16 due to topology FW dependency.
- */
- if (!strcmp(soc_rt->card->name, "kblda7219m98373") ||
- !strcmp(soc_rt->card->name, "kblmax98373")) {
- runtime->hw.formats = SNDRV_PCM_FMTBIT_S24_LE;
- snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24);
-
- } else {
- runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE;
- snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16);
- }
+
+ runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE;
+ snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16);
snd_pcm_hw_constraint_list(runtime, 0,
SNDRV_PCM_HW_PARAM_RATE, &constraints_rates);
@@ -534,23 +505,11 @@ static int kabylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd,
static int kabylake_dmic_startup(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
- struct snd_soc_pcm_runtime *soc_rt = asoc_substream_to_rtd(substream);
runtime->hw.channels_min = runtime->hw.channels_max = QUAD_CHANNEL;
snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
&constraints_channels_quad);
- /*
- * Topology for kblda7219m98373 & kblmax98373 supports only S24_LE.
- * The DMIC also configured for S24_LE. Forcing the DMIC format to
- * S24_LE due to the topology FW dependency.
- */
- if (!strcmp(soc_rt->card->name, "kblda7219m98373") ||
- !strcmp(soc_rt->card->name, "kblmax98373")) {
- runtime->hw.formats = SNDRV_PCM_FMTBIT_S24_LE;
- snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24);
- }
-
return snd_pcm_hw_constraint_list(substream->runtime, 0,
SNDRV_PCM_HW_PARAM_RATE, &constraints_rates);
}
diff --git a/sound/soc/intel/boards/sof_pcm512x.c b/sound/soc/intel/boards/sof_pcm512x.c
index d2b0456236c7..bdd671f07659 100644
--- a/sound/soc/intel/boards/sof_pcm512x.c
+++ b/sound/soc/intel/boards/sof_pcm512x.c
@@ -26,11 +26,16 @@
#define SOF_PCM512X_SSP_CODEC(quirk) ((quirk) & GENMASK(3, 0))
#define SOF_PCM512X_SSP_CODEC_MASK (GENMASK(3, 0))
+#define SOF_PCM512X_ENABLE_SSP_CAPTURE BIT(4)
+#define SOF_PCM512X_ENABLE_DMIC BIT(5)
#define IDISP_CODEC_MASK 0x4
/* Default: SSP5 */
-static unsigned long sof_pcm512x_quirk = SOF_PCM512X_SSP_CODEC(5);
+static unsigned long sof_pcm512x_quirk =
+ SOF_PCM512X_SSP_CODEC(5) |
+ SOF_PCM512X_ENABLE_SSP_CAPTURE |
+ SOF_PCM512X_ENABLE_DMIC;
static bool is_legacy_cpu;
@@ -245,8 +250,9 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev,
links[id].dpcm_playback = 1;
/*
* capture only supported with specific versions of the Hifiberry DAC+
- * links[id].dpcm_capture = 1;
*/
+ if (sof_pcm512x_quirk & SOF_PCM512X_ENABLE_SSP_CAPTURE)
+ links[id].dpcm_capture = 1;
links[id].no_pcm = 1;
links[id].cpus = &cpus[id];
links[id].num_cpus = 1;
@@ -381,6 +387,9 @@ static int sof_audio_probe(struct platform_device *pdev)
ssp_codec = sof_pcm512x_quirk & SOF_PCM512X_SSP_CODEC_MASK;
+ if (!(sof_pcm512x_quirk & SOF_PCM512X_ENABLE_DMIC))
+ dmic_be_num = 0;
+
/* compute number of dai links */
sof_audio_card_pcm512x.num_links = 1 + dmic_be_num + hdmi_num;
generated by cgit v1.2.3 (git 2.39.0) at 2025-05-02 18:12:41 +0300