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-rw-r--r--drivers/spi/spi-lantiq-ssc.c2
-rw-r--r--sound/soc/codecs/max98357a.c50
-rw-r--r--sound/soc/codecs/max98390.c2
-rw-r--r--sound/soc/fsl/fsl_sai.c5
-rw-r--r--sound/soc/fsl/fsl_sai.h2
-rw-r--r--sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c41
-rw-r--r--sound/soc/intel/boards/skl_hda_dsp_common.h1
-rw-r--r--sound/soc/intel/boards/skl_hda_dsp_generic.c17
-rw-r--r--sound/soc/intel/common/soc-acpi-intel-ehl-match.c2
-rw-r--r--sound/soc/meson/axg-card.c20
-rw-r--r--sound/soc/meson/axg-tdm-formatter.c11
-rw-r--r--sound/soc/meson/axg-tdm-formatter.h1
-rw-r--r--sound/soc/meson/axg-tdm-interface.c26
-rw-r--r--sound/soc/meson/axg-tdmin.c16
-rw-r--r--sound/soc/meson/axg-tdmout.c3
-rw-r--r--sound/soc/meson/gx-card.c18
-rw-r--r--sound/soc/meson/meson-card-utils.c4
-rw-r--r--sound/soc/soc-core.c5
-rw-r--r--sound/soc/soc-dai.c16
-rw-r--r--sound/soc/soc-pcm.c42
20 files changed, 186 insertions, 98 deletions
diff --git a/drivers/spi/spi-lantiq-ssc.c b/drivers/spi/spi-lantiq-ssc.c
index 1fd7ee53d451..a12a5d0cfebf 100644
--- a/drivers/spi/spi-lantiq-ssc.c
+++ b/drivers/spi/spi-lantiq-ssc.c
@@ -899,7 +899,7 @@ static int lantiq_ssc_probe(struct platform_device *pdev)
master->bits_per_word_mask = SPI_BPW_RANGE_MASK(2, 8) |
SPI_BPW_MASK(16) | SPI_BPW_MASK(32);
- spi->wq = alloc_ordered_workqueue(dev_name(dev), 0);
+ spi->wq = alloc_ordered_workqueue(dev_name(dev), WQ_MEM_RECLAIM);
if (!spi->wq) {
err = -ENOMEM;
goto err_clk_put;
diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c
index a8bd793a7867..151f05a68435 100644
--- a/sound/soc/codecs/max98357a.c
+++ b/sound/soc/codecs/max98357a.c
@@ -23,36 +23,61 @@
struct max98357a_priv {
struct gpio_desc *sdmode;
unsigned int sdmode_delay;
+ int sdmode_switch;
};
-static int max98357a_sdmode_event(struct snd_soc_dapm_widget *w,
- struct snd_kcontrol *kcontrol, int event)
+static int max98357a_daiops_trigger(struct snd_pcm_substream *substream,
+ int cmd, struct snd_soc_dai *dai)
{
- struct snd_soc_component *component =
- snd_soc_dapm_to_component(w->dapm);
+ struct snd_soc_component *component = dai->component;
struct max98357a_priv *max98357a =
snd_soc_component_get_drvdata(component);
if (!max98357a->sdmode)
return 0;
- if (event & SND_SOC_DAPM_POST_PMU) {
- msleep(max98357a->sdmode_delay);
- gpiod_set_value(max98357a->sdmode, 1);
- dev_dbg(component->dev, "set sdmode to 1");
- } else if (event & SND_SOC_DAPM_PRE_PMD) {
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ mdelay(max98357a->sdmode_delay);
+ if (max98357a->sdmode_switch) {
+ gpiod_set_value(max98357a->sdmode, 1);
+ dev_dbg(component->dev, "set sdmode to 1");
+ }
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
gpiod_set_value(max98357a->sdmode, 0);
dev_dbg(component->dev, "set sdmode to 0");
+ break;
}
return 0;
}
+static int max98357a_sdmode_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 max98357a_priv *max98357a =
+ snd_soc_component_get_drvdata(component);
+
+ if (event & SND_SOC_DAPM_POST_PMU)
+ max98357a->sdmode_switch = 1;
+ else if (event & SND_SOC_DAPM_POST_PMD)
+ max98357a->sdmode_switch = 0;
+
+ return 0;
+}
+
static const struct snd_soc_dapm_widget max98357a_dapm_widgets[] = {
SND_SOC_DAPM_OUTPUT("Speaker"),
SND_SOC_DAPM_OUT_DRV_E("SD_MODE", SND_SOC_NOPM, 0, 0, NULL, 0,
max98357a_sdmode_event,
- SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
};
static const struct snd_soc_dapm_route max98357a_dapm_routes[] = {
@@ -71,6 +96,10 @@ static const struct snd_soc_component_driver max98357a_component_driver = {
.non_legacy_dai_naming = 1,
};
+static const struct snd_soc_dai_ops max98357a_dai_ops = {
+ .trigger = max98357a_daiops_trigger,
+};
+
static struct snd_soc_dai_driver max98357a_dai_driver = {
.name = "HiFi",
.playback = {
@@ -90,6 +119,7 @@ static struct snd_soc_dai_driver max98357a_dai_driver = {
.channels_min = 1,
.channels_max = 2,
},
+ .ops = &max98357a_dai_ops,
};
static int max98357a_platform_probe(struct platform_device *pdev)
diff --git a/sound/soc/codecs/max98390.c b/sound/soc/codecs/max98390.c
index e6613b52bd78..9859a133b90c 100644
--- a/sound/soc/codecs/max98390.c
+++ b/sound/soc/codecs/max98390.c
@@ -678,7 +678,7 @@ static const struct snd_kcontrol_new max98390_dai_controls =
static const struct snd_soc_dapm_widget max98390_dapm_widgets[] = {
SND_SOC_DAPM_DAC_E("Amp Enable", "HiFi Playback",
- MAX98390_R203A_AMP_EN, 0, 0, max98390_dac_event,
+ SND_SOC_NOPM, 0, 0, max98390_dac_event,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_MUX("DAI Sel Mux", SND_SOC_NOPM, 0, 0,
&max98390_dai_controls),
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index 9d436b0c5718..7031869a023a 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -680,10 +680,11 @@ static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai)
regmap_write(sai->regmap, FSL_SAI_RCSR(ofs), 0);
regmap_update_bits(sai->regmap, FSL_SAI_TCR1(ofs),
- FSL_SAI_CR1_RFW_MASK,
+ FSL_SAI_CR1_RFW_MASK(sai->soc_data->fifo_depth),
sai->soc_data->fifo_depth - FSL_SAI_MAXBURST_TX);
regmap_update_bits(sai->regmap, FSL_SAI_RCR1(ofs),
- FSL_SAI_CR1_RFW_MASK, FSL_SAI_MAXBURST_RX - 1);
+ FSL_SAI_CR1_RFW_MASK(sai->soc_data->fifo_depth),
+ FSL_SAI_MAXBURST_RX - 1);
snd_soc_dai_init_dma_data(cpu_dai, &sai->dma_params_tx,
&sai->dma_params_rx);
diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h
index 76b15deea80c..6aba7d28f5f3 100644
--- a/sound/soc/fsl/fsl_sai.h
+++ b/sound/soc/fsl/fsl_sai.h
@@ -94,7 +94,7 @@
#define FSL_SAI_CSR_FRDE BIT(0)
/* SAI Transmit and Receive Configuration 1 Register */
-#define FSL_SAI_CR1_RFW_MASK 0x1f
+#define FSL_SAI_CR1_RFW_MASK(x) ((x) - 1)
/* SAI Transmit and Receive Configuration 2 Register */
#define FSL_SAI_CR2_SYNC BIT(30)
diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
index b34cf6cf1139..2985f8bf30b2 100644
--- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
+++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
@@ -336,22 +336,45 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
struct snd_interval *chan = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_CHANNELS);
struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
- struct snd_soc_dpcm *dpcm = container_of(
- params, struct snd_soc_dpcm, hw_params);
- struct snd_soc_dai_link *fe_dai_link = dpcm->fe->dai_link;
- struct snd_soc_dai_link *be_dai_link = dpcm->be->dai_link;
+ struct snd_soc_dpcm *dpcm, *rtd_dpcm = NULL;
+
+ /*
+ * The following loop will be called only for playback stream
+ * In this platform, there is only one playback device on every SSP
+ */
+ for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_PLAYBACK, dpcm) {
+ rtd_dpcm = dpcm;
+ break;
+ }
+
+ /*
+ * This following loop will be called only for capture stream
+ * In this platform, there is only one capture device on every SSP
+ */
+ for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_CAPTURE, dpcm) {
+ rtd_dpcm = dpcm;
+ break;
+ }
+
+ if (!rtd_dpcm)
+ return -EINVAL;
+
+ /*
+ * The above 2 loops are mutually exclusive based on the stream direction,
+ * thus rtd_dpcm variable will never be overwritten
+ */
/*
* The ADSP will convert the FE rate to 48k, stereo, 24 bit
*/
- if (!strcmp(fe_dai_link->name, "Kbl Audio Port") ||
- !strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") ||
- !strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) {
+ if (!strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Port") ||
+ !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Headset Playback") ||
+ !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Capture Port")) {
rate->min = rate->max = 48000;
chan->min = chan->max = 2;
snd_mask_none(fmt);
snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE);
- } else if (!strcmp(fe_dai_link->name, "Kbl Audio DMIC cap")) {
+ } else if (!strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio DMIC cap")) {
if (params_channels(params) == 2 ||
DMIC_CH(dmic_constraints) == 2)
chan->min = chan->max = 2;
@@ -362,7 +385,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
* The speaker on the SSP0 supports S16_LE and not S24_LE.
* thus changing the mask here
*/
- if (!strcmp(be_dai_link->name, "SSP0-Codec"))
+ if (!strcmp(rtd_dpcm->be->dai_link->name, "SSP0-Codec"))
snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE);
return 0;
diff --git a/sound/soc/intel/boards/skl_hda_dsp_common.h b/sound/soc/intel/boards/skl_hda_dsp_common.h
index 507750ef67f3..4b0b3959182e 100644
--- a/sound/soc/intel/boards/skl_hda_dsp_common.h
+++ b/sound/soc/intel/boards/skl_hda_dsp_common.h
@@ -33,6 +33,7 @@ struct skl_hda_private {
int dai_index;
const char *platform_name;
bool common_hdmi_codec_drv;
+ bool idisp_codec;
};
extern struct snd_soc_dai_link skl_hda_be_dai_links[HDA_DSP_MAX_BE_DAI_LINKS];
diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c
index 79c8947f840b..ca4900036ead 100644
--- a/sound/soc/intel/boards/skl_hda_dsp_generic.c
+++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c
@@ -79,6 +79,9 @@ skl_hda_add_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link *link)
link->platforms->name = ctx->platform_name;
link->nonatomic = 1;
+ if (!ctx->idisp_codec)
+ return 0;
+
if (strstr(link->name, "HDMI")) {
ret = skl_hda_hdmi_add_pcm(card, ctx->pcm_count);
@@ -118,19 +121,20 @@ static char hda_soc_components[30];
static int skl_hda_fill_card_info(struct snd_soc_acpi_mach_params *mach_params)
{
struct snd_soc_card *card = &hda_soc_card;
+ struct skl_hda_private *ctx = snd_soc_card_get_drvdata(card);
struct snd_soc_dai_link *dai_link;
- u32 codec_count, codec_mask, idisp_mask;
+ u32 codec_count, codec_mask;
int i, num_links, num_route;
codec_mask = mach_params->codec_mask;
codec_count = hweight_long(codec_mask);
- idisp_mask = codec_mask & IDISP_CODEC_MASK;
+ ctx->idisp_codec = !!(codec_mask & IDISP_CODEC_MASK);
if (!codec_count || codec_count > 2 ||
- (codec_count == 2 && !idisp_mask))
+ (codec_count == 2 && !ctx->idisp_codec))
return -EINVAL;
- if (codec_mask == idisp_mask) {
+ if (codec_mask == IDISP_CODEC_MASK) {
/* topology with iDisp as the only HDA codec */
num_links = IDISP_DAI_COUNT + DMIC_DAI_COUNT;
num_route = IDISP_ROUTE_COUNT;
@@ -152,7 +156,7 @@ static int skl_hda_fill_card_info(struct snd_soc_acpi_mach_params *mach_params)
num_route = ARRAY_SIZE(skl_hda_map);
card->dapm_widgets = skl_hda_widgets;
card->num_dapm_widgets = ARRAY_SIZE(skl_hda_widgets);
- if (!idisp_mask) {
+ if (!ctx->idisp_codec) {
for (i = 0; i < IDISP_DAI_COUNT; i++) {
skl_hda_be_dai_links[i].codecs = dummy_codec;
skl_hda_be_dai_links[i].num_codecs =
@@ -211,6 +215,8 @@ static int skl_hda_audio_probe(struct platform_device *pdev)
if (!mach)
return -EINVAL;
+ snd_soc_card_set_drvdata(&hda_soc_card, ctx);
+
ret = skl_hda_fill_card_info(&mach->mach_params);
if (ret < 0) {
dev_err(&pdev->dev, "Unsupported HDAudio/iDisp configuration found\n");
@@ -223,7 +229,6 @@ static int skl_hda_audio_probe(struct platform_device *pdev)
ctx->common_hdmi_codec_drv = mach->mach_params.common_hdmi_codec_drv;
hda_soc_card.dev = &pdev->dev;
- snd_soc_card_set_drvdata(&hda_soc_card, ctx);
if (mach->mach_params.dmic_num > 0) {
snprintf(hda_soc_components, sizeof(hda_soc_components),
diff --git a/sound/soc/intel/common/soc-acpi-intel-ehl-match.c b/sound/soc/intel/common/soc-acpi-intel-ehl-match.c
index 45e07d886013..badafc1d54d2 100644
--- a/sound/soc/intel/common/soc-acpi-intel-ehl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-ehl-match.c
@@ -12,7 +12,7 @@
struct snd_soc_acpi_mach snd_soc_acpi_intel_ehl_machines[] = {
{
- .id = "INTC1027",
+ .id = "10EC5660",
.drv_name = "ehl_rt5660",
.sof_fw_filename = "sof-ehl.ri",
.sof_tplg_filename = "sof-ehl-rt5660.tplg",
diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c
index 89f7f64747cd..33058518c3da 100644
--- a/sound/soc/meson/axg-card.c
+++ b/sound/soc/meson/axg-card.c
@@ -116,7 +116,7 @@ static int axg_card_add_tdm_loopback(struct snd_soc_card *card,
lb = &card->dai_link[*index + 1];
- lb->name = kasprintf(GFP_KERNEL, "%s-lb", pad->name);
+ lb->name = devm_kasprintf(card->dev, GFP_KERNEL, "%s-lb", pad->name);
if (!lb->name)
return -ENOMEM;
@@ -327,20 +327,22 @@ static int axg_card_add_link(struct snd_soc_card *card, struct device_node *np,
return ret;
if (axg_card_cpu_is_playback_fe(dai_link->cpus->of_node))
- ret = meson_card_set_fe_link(card, dai_link, np, true);
+ return meson_card_set_fe_link(card, dai_link, np, true);
else if (axg_card_cpu_is_capture_fe(dai_link->cpus->of_node))
- ret = meson_card_set_fe_link(card, dai_link, np, false);
- else
- ret = meson_card_set_be_link(card, dai_link, np);
+ return meson_card_set_fe_link(card, dai_link, np, false);
+
+ ret = meson_card_set_be_link(card, dai_link, np);
if (ret)
return ret;
- 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)) {
+ if (axg_card_cpu_is_codec(dai_link->cpus->of_node)) {
dai_link->params = &codec_params;
- dai_link->no_pcm = 0; /* link is not a DPCM BE */
+ } else {
+ dai_link->no_pcm = 1;
+ snd_soc_dai_link_set_capabilities(dai_link);
+ if (axg_card_cpu_is_tdm_iface(dai_link->cpus->of_node))
+ ret = axg_card_parse_tdm(card, np, index);
}
return ret;
diff --git a/sound/soc/meson/axg-tdm-formatter.c b/sound/soc/meson/axg-tdm-formatter.c
index 358c8c0d861c..f7e8e9da68a0 100644
--- a/sound/soc/meson/axg-tdm-formatter.c
+++ b/sound/soc/meson/axg-tdm-formatter.c
@@ -70,7 +70,7 @@ EXPORT_SYMBOL_GPL(axg_tdm_formatter_set_channel_masks);
static int axg_tdm_formatter_enable(struct axg_tdm_formatter *formatter)
{
struct axg_tdm_stream *ts = formatter->stream;
- bool invert = formatter->drv->quirks->invert_sclk;
+ bool invert;
int ret;
/* Do nothing if the formatter is already enabled */
@@ -96,11 +96,12 @@ static int axg_tdm_formatter_enable(struct axg_tdm_formatter *formatter)
return ret;
/*
- * If sclk is inverted, invert it back and provide the inversion
- * required by the formatter
+ * If sclk is inverted, it means the bit should latched on the
+ * rising edge which is what our HW expects. If not, we need to
+ * invert it before the formatter.
*/
- invert ^= axg_tdm_sclk_invert(ts->iface->fmt);
- ret = clk_set_phase(formatter->sclk, invert ? 180 : 0);
+ invert = axg_tdm_sclk_invert(ts->iface->fmt);
+ ret = clk_set_phase(formatter->sclk, invert ? 0 : 180);
if (ret)
return ret;
diff --git a/sound/soc/meson/axg-tdm-formatter.h b/sound/soc/meson/axg-tdm-formatter.h
index 9ef98e955cb2..a1f0dcc0ff13 100644
--- a/sound/soc/meson/axg-tdm-formatter.h
+++ b/sound/soc/meson/axg-tdm-formatter.h
@@ -16,7 +16,6 @@ struct snd_kcontrol;
struct axg_tdm_formatter_hw {
unsigned int skew_offset;
- bool invert_sclk;
};
struct axg_tdm_formatter_ops {
diff --git a/sound/soc/meson/axg-tdm-interface.c b/sound/soc/meson/axg-tdm-interface.c
index 6de27238e9df..36df30915378 100644
--- a/sound/soc/meson/axg-tdm-interface.c
+++ b/sound/soc/meson/axg-tdm-interface.c
@@ -119,18 +119,25 @@ 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);
- /* These modes are not supported */
- if (fmt & (SND_SOC_DAIFMT_CBS_CFM | SND_SOC_DAIFMT_CBM_CFS)) {
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ if (!iface->mclk) {
+ dev_err(dai->dev, "cpu clock master: mclk missing\n");
+ return -ENODEV;
+ }
+ break;
+
+ case SND_SOC_DAIFMT_CBM_CFM:
+ break;
+
+ case SND_SOC_DAIFMT_CBS_CFM:
+ case SND_SOC_DAIFMT_CBM_CFS:
dev_err(dai->dev, "only CBS_CFS and CBM_CFM are supported\n");
+ /* Fall-through */
+ default:
return -EINVAL;
}
- /* If the TDM interface is the clock master, it requires mclk */
- if (!iface->mclk && (fmt & SND_SOC_DAIFMT_CBS_CFS)) {
- dev_err(dai->dev, "cpu clock master: mclk missing\n");
- return -ENODEV;
- }
-
iface->fmt = fmt;
return 0;
}
@@ -319,7 +326,8 @@ static int axg_tdm_iface_hw_params(struct snd_pcm_substream *substream,
if (ret)
return ret;
- if (iface->fmt & SND_SOC_DAIFMT_CBS_CFS) {
+ if ((iface->fmt & SND_SOC_DAIFMT_MASTER_MASK) ==
+ SND_SOC_DAIFMT_CBS_CFS) {
ret = axg_tdm_iface_set_sclk(dai, params);
if (ret)
return ret;
diff --git a/sound/soc/meson/axg-tdmin.c b/sound/soc/meson/axg-tdmin.c
index 973d4c02ef8d..88ed95ae886b 100644
--- a/sound/soc/meson/axg-tdmin.c
+++ b/sound/soc/meson/axg-tdmin.c
@@ -228,15 +228,29 @@ static const struct axg_tdm_formatter_driver axg_tdmin_drv = {
.regmap_cfg = &axg_tdmin_regmap_cfg,
.ops = &axg_tdmin_ops,
.quirks = &(const struct axg_tdm_formatter_hw) {
- .invert_sclk = false,
.skew_offset = 2,
},
};
+static const struct axg_tdm_formatter_driver g12a_tdmin_drv = {
+ .component_drv = &axg_tdmin_component_drv,
+ .regmap_cfg = &axg_tdmin_regmap_cfg,
+ .ops = &axg_tdmin_ops,
+ .quirks = &(const struct axg_tdm_formatter_hw) {
+ .skew_offset = 3,
+ },
+};
+
static const struct of_device_id axg_tdmin_of_match[] = {
{
.compatible = "amlogic,axg-tdmin",
.data = &axg_tdmin_drv,
+ }, {
+ .compatible = "amlogic,g12a-tdmin",
+ .data = &g12a_tdmin_drv,
+ }, {
+ .compatible = "amlogic,sm1-tdmin",
+ .data = &g12a_tdmin_drv,
}, {}
};
MODULE_DEVICE_TABLE(of, axg_tdmin_of_match);
diff --git a/sound/soc/meson/axg-tdmout.c b/sound/soc/meson/axg-tdmout.c
index 418ec314b37d..3ceabddae629 100644
--- a/sound/soc/meson/axg-tdmout.c
+++ b/sound/soc/meson/axg-tdmout.c
@@ -238,7 +238,6 @@ static const struct axg_tdm_formatter_driver axg_tdmout_drv = {
.regmap_cfg = &axg_tdmout_regmap_cfg,
.ops = &axg_tdmout_ops,
.quirks = &(const struct axg_tdm_formatter_hw) {
- .invert_sclk = true,
.skew_offset = 1,
},
};
@@ -248,7 +247,6 @@ static const struct axg_tdm_formatter_driver g12a_tdmout_drv = {
.regmap_cfg = &axg_tdmout_regmap_cfg,
.ops = &axg_tdmout_ops,
.quirks = &(const struct axg_tdm_formatter_hw) {
- .invert_sclk = true,
.skew_offset = 2,
},
};
@@ -309,7 +307,6 @@ static const struct axg_tdm_formatter_driver sm1_tdmout_drv = {
.regmap_cfg = &axg_tdmout_regmap_cfg,
.ops = &axg_tdmout_ops,
.quirks = &(const struct axg_tdm_formatter_hw) {
- .invert_sclk = true,
.skew_offset = 2,
},
};
diff --git a/sound/soc/meson/gx-card.c b/sound/soc/meson/gx-card.c
index 4abf7efb7eac..fdd2d5303b2a 100644
--- a/sound/soc/meson/gx-card.c
+++ b/sound/soc/meson/gx-card.c
@@ -96,21 +96,21 @@ static int gx_card_add_link(struct snd_soc_card *card, struct device_node *np,
return ret;
if (gx_card_cpu_identify(dai_link->cpus, "FIFO"))
- ret = meson_card_set_fe_link(card, dai_link, np, true);
- else
- ret = meson_card_set_be_link(card, dai_link, np);
+ return meson_card_set_fe_link(card, dai_link, np, true);
+ ret = meson_card_set_be_link(card, dai_link, np);
if (ret)
return ret;
- /* Check if the cpu is the i2s encoder and parse i2s data */
- if (gx_card_cpu_identify(dai_link->cpus, "I2S Encoder"))
- ret = gx_card_parse_i2s(card, np, index);
-
/* Or apply codec to codec params if necessary */
- else if (gx_card_cpu_identify(dai_link->cpus, "CODEC CTRL")) {
+ if (gx_card_cpu_identify(dai_link->cpus, "CODEC CTRL")) {
dai_link->params = &codec_params;
- dai_link->no_pcm = 0; /* link is not a DPCM BE */
+ } else {
+ dai_link->no_pcm = 1;
+ snd_soc_dai_link_set_capabilities(dai_link);
+ /* Check if the cpu is the i2s encoder and parse i2s data */
+ if (gx_card_cpu_identify(dai_link->cpus, "I2S Encoder"))
+ ret = gx_card_parse_i2s(card, np, index);
}
return ret;
diff --git a/sound/soc/meson/meson-card-utils.c b/sound/soc/meson/meson-card-utils.c
index 5a4a91c88734..c734131ff0d6 100644
--- a/sound/soc/meson/meson-card-utils.c
+++ b/sound/soc/meson/meson-card-utils.c
@@ -147,10 +147,6 @@ int meson_card_set_be_link(struct snd_soc_card *card,
struct device_node *np;
int ret, num_codecs;
- link->no_pcm = 1;
- link->dpcm_playback = 1;
- link->dpcm_capture = 1;
-
num_codecs = of_get_child_count(node);
if (!num_codecs) {
dev_err(card->dev, "be link %s has no codec\n",
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 2b8abf88ec60..f1d641cd48da 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -446,7 +446,6 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime(
dev->parent = card->dev;
dev->release = soc_release_rtd_dev;
- dev->groups = soc_dev_attr_groups;
dev_set_name(dev, "%s", dai_link->name);
@@ -503,6 +502,10 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime(
/* see for_each_card_rtds */
list_add_tail(&rtd->list, &card->rtd_list);
+ ret = device_add_groups(dev, soc_dev_attr_groups);
+ if (ret < 0)
+ goto free_rtd;
+
return rtd;
free_rtd:
diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c
index 457159975b01..cecbbed2de9d 100644
--- a/sound/soc/soc-dai.c
+++ b/sound/soc/soc-dai.c
@@ -400,28 +400,30 @@ void snd_soc_dai_link_set_capabilities(struct snd_soc_dai_link *dai_link)
struct snd_soc_dai_link_component *codec;
struct snd_soc_dai *dai;
bool supported[SNDRV_PCM_STREAM_LAST + 1];
+ bool supported_cpu;
+ bool supported_codec;
int direction;
int i;
for_each_pcm_streams(direction) {
- supported[direction] = true;
+ supported_cpu = false;
+ supported_codec = false;
for_each_link_cpus(dai_link, i, cpu) {
dai = snd_soc_find_dai(cpu);
- if (!dai || !snd_soc_dai_stream_valid(dai, direction)) {
- supported[direction] = false;
+ if (dai && snd_soc_dai_stream_valid(dai, direction)) {
+ supported_cpu = true;
break;
}
}
- if (!supported[direction])
- continue;
for_each_link_codecs(dai_link, i, codec) {
dai = snd_soc_find_dai(codec);
- if (!dai || !snd_soc_dai_stream_valid(dai, direction)) {
- supported[direction] = false;
+ if (dai && snd_soc_dai_stream_valid(dai, direction)) {
+ supported_codec = true;
break;
}
}
+ supported[direction] = supported_cpu && supported_codec;
}
dai_link->dpcm_playback = supported[SNDRV_PCM_STREAM_PLAYBACK];
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index c517064f5391..74baf1fce053 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -2802,30 +2802,36 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
if (rtd->dai_link->dpcm_playback) {
stream = SNDRV_PCM_STREAM_PLAYBACK;
- for_each_rtd_cpu_dais(rtd, i, cpu_dai)
- if (!snd_soc_dai_stream_valid(cpu_dai,
- stream)) {
- dev_err(rtd->card->dev,
- "CPU DAI %s for rtd %s does not support playback\n",
- cpu_dai->name,
- rtd->dai_link->stream_name);
- return -EINVAL;
+ for_each_rtd_cpu_dais(rtd, i, cpu_dai) {
+ if (snd_soc_dai_stream_valid(cpu_dai, stream)) {
+ playback = 1;
+ break;
}
- playback = 1;
+ }
+
+ if (!playback) {
+ dev_err(rtd->card->dev,
+ "No CPU DAIs support playback for stream %s\n",
+ rtd->dai_link->stream_name);
+ return -EINVAL;
+ }
}
if (rtd->dai_link->dpcm_capture) {
stream = SNDRV_PCM_STREAM_CAPTURE;
- for_each_rtd_cpu_dais(rtd, i, cpu_dai)
- if (!snd_soc_dai_stream_valid(cpu_dai,
- stream)) {
- dev_err(rtd->card->dev,
- "CPU DAI %s for rtd %s does not support capture\n",
- cpu_dai->name,
- rtd->dai_link->stream_name);
- return -EINVAL;
+ for_each_rtd_cpu_dais(rtd, i, cpu_dai) {
+ if (snd_soc_dai_stream_valid(cpu_dai, stream)) {
+ capture = 1;
+ break;
}
- capture = 1;
+ }
+
+ if (!capture) {
+ dev_err(rtd->card->dev,
+ "No CPU DAIs support capture for stream %s\n",
+ rtd->dai_link->stream_name);
+ return -EINVAL;
+ }
}
} else {
/* Adapt stream for codec2codec links */
generated by cgit (git 2.34.1) at 2025-10-31 07:35:00 +0000