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-rw-r--r--sound/hda/hdac_bus.c12
-rw-r--r--sound/hda/hdac_controller.c11
-rw-r--r--sound/isa/sscape.c6
-rw-r--r--sound/pci/hda/patch_realtek.c3
-rw-r--r--sound/ppc/snd_ps3.c4
-rw-r--r--sound/soc/amd/acp3x-rt5682-max9836.c2
-rw-r--r--sound/soc/amd/renoir/acp3x-pdm-dma.c29
-rw-r--r--sound/soc/atmel/mchp-i2s-mcc.c2
-rw-r--r--sound/soc/codecs/jz4770.c2
-rw-r--r--sound/soc/codecs/msm8916-wcd-analog.c4
-rw-r--r--sound/soc/codecs/pcm186x.c2
-rw-r--r--sound/soc/codecs/wm8958-dsp2.c4
-rw-r--r--sound/soc/codecs/wm8962.c2
-rw-r--r--sound/soc/codecs/wm8994.c60
-rw-r--r--sound/soc/fsl/fsl-asoc-card.c154
-rw-r--r--sound/soc/fsl/fsl_ssi.c2
-rw-r--r--sound/soc/fsl/mpc5200_dma.c1
-rw-r--r--sound/soc/hisilicon/hi6210-i2s.c4
-rw-r--r--sound/soc/intel/atom/sst-mfld-platform-pcm.c5
-rw-r--r--sound/soc/intel/baytrail/sst-baytrail-pcm.c2
-rw-r--r--sound/soc/intel/boards/bytcht_es8316.c2
-rw-r--r--sound/soc/intel/boards/bytcr_rt5651.c4
-rw-r--r--sound/soc/intel/skylake/skl-pcm.c2
-rw-r--r--sound/soc/meson/axg-tdm-interface.c10
-rw-r--r--sound/soc/pxa/pxa-ssp.c2
-rw-r--r--sound/soc/qcom/qdsp6/q6afe-dai.c210
-rw-r--r--sound/soc/qcom/qdsp6/q6routing.c16
-rw-r--r--sound/soc/rockchip/rockchip_pdm.c6
-rw-r--r--sound/soc/samsung/i2s.c2
-rw-r--r--sound/soc/soc-component.c2
-rw-r--r--sound/soc/soc-core.c2
-rw-r--r--sound/soc/soc-topology.c4
-rw-r--r--sound/soc/sof/intel/hda-dai.c4
-rw-r--r--sound/soc/sof/pcm.c4
-rw-r--r--sound/soc/tegra/tegra186_dspk.c4
-rw-r--r--sound/soc/tegra/tegra210_admaif.c4
-rw-r--r--sound/soc/tegra/tegra210_ahub.c4
-rw-r--r--sound/soc/tegra/tegra210_dmic.c4
-rw-r--r--sound/soc/tegra/tegra210_i2s.c4
-rw-r--r--sound/soc/ti/davinci-i2s.c2
-rw-r--r--sound/soc/ti/n810.c2
-rw-r--r--sound/soc/ti/omap-dmic.c4
-rw-r--r--sound/soc/ti/omap-mcpdm.c8
-rw-r--r--sound/soc/ti/rx51.c2
-rw-r--r--sound/soc/zte/zx-i2s.c4
-rw-r--r--sound/soc/zte/zx-spdif.c2
-rw-r--r--sound/usb/mixer.c8
-rw-r--r--sound/usb/quirks-table.h34
48 files changed, 365 insertions, 302 deletions
diff --git a/sound/hda/hdac_bus.c b/sound/hda/hdac_bus.c
index 09ddab5f5cae..9766f6af8743 100644
--- a/sound/hda/hdac_bus.c
+++ b/sound/hda/hdac_bus.c
@@ -46,6 +46,18 @@ int snd_hdac_bus_init(struct hdac_bus *bus, struct device *dev,
INIT_LIST_HEAD(&bus->hlink_list);
init_waitqueue_head(&bus->rirb_wq);
bus->irq = -1;
+
+ /*
+ * Default value of '8' is as per the HD audio specification (Rev 1.0a).
+ * Following relation is used to derive STRIPE control value.
+ * For sample rate <= 48K:
+ * { ((num_channels * bits_per_sample) / number of SDOs) >= 8 }
+ * For sample rate > 48K:
+ * { ((num_channels * bits_per_sample * rate/48000) /
+ * number of SDOs) >= 8 }
+ */
+ bus->sdo_limit = 8;
+
return 0;
}
EXPORT_SYMBOL_GPL(snd_hdac_bus_init);
diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c
index 011b17cc1efa..b98449fd92f3 100644
--- a/sound/hda/hdac_controller.c
+++ b/sound/hda/hdac_controller.c
@@ -529,17 +529,6 @@ bool snd_hdac_bus_init_chip(struct hdac_bus *bus, bool full_reset)
bus->chip_init = true;
- /*
- * Default value of '8' is as per the HD audio specification (Rev 1.0a).
- * Following relation is used to derive STRIPE control value.
- * For sample rate <= 48K:
- * { ((num_channels * bits_per_sample) / number of SDOs) >= 8 }
- * For sample rate > 48K:
- * { ((num_channels * bits_per_sample * rate/48000) /
- * number of SDOs) >= 8 }
- */
- bus->sdo_limit = 8;
-
return true;
}
EXPORT_SYMBOL_GPL(snd_hdac_bus_init_chip);
diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c
index 5363d88cc4b9..2e5a5c5279e8 100644
--- a/sound/isa/sscape.c
+++ b/sound/isa/sscape.c
@@ -308,7 +308,7 @@ static inline int verify_mpu401(const struct snd_mpu401 *mpu)
}
/*
- * This is apparently the standard way to initailise an MPU-401
+ * This is apparently the standard way to initialise an MPU-401
*/
static inline void initialise_mpu401(const struct snd_mpu401 *mpu)
{
@@ -339,7 +339,7 @@ static void soundscape_free(struct snd_card *c)
}
/*
- * Tell the SoundScape to begin a DMA tranfer using the given channel.
+ * Tell the SoundScape to begin a DMA transfer using the given channel.
* All locking issues are left to the caller.
*/
static void sscape_start_dma_unsafe(unsigned io_base, enum GA_REG reg)
@@ -803,7 +803,7 @@ static int mpu401_open(struct snd_mpu401 *mpu)
}
/*
- * Initialse an MPU-401 subdevice for MIDI support on the SoundScape.
+ * Initialise an MPU-401 subdevice for MIDI support on the SoundScape.
*/
static int create_mpu401(struct snd_card *card, int devnum,
unsigned long port, int irq)
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 7f9d35273734..a1fa983d2a94 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -7694,6 +7694,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x144d, 0xc109, "Samsung Ativ book 9 (NP900X3G)", ALC269_FIXUP_INV_DMIC),
SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
SND_PCI_QUIRK(0x144d, 0xc176, "Samsung Notebook 9 Pro (NP930MBE-K04US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
+ SND_PCI_QUIRK(0x144d, 0xc189, "Samsung Galaxy Flex Book (NT950QCG-X716)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
+ SND_PCI_QUIRK(0x144d, 0xc18a, "Samsung Galaxy Book Ion (NT950XCJ-X716A)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
SND_PCI_QUIRK(0x144d, 0xc740, "Samsung Ativ book 8 (NP870Z5G)", ALC269_FIXUP_ATIV_BOOK_8),
SND_PCI_QUIRK(0x144d, 0xc812, "Samsung Notebook Pen S (NT950SBE-X58)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC),
@@ -7955,6 +7957,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = {
{.id = ALC299_FIXUP_PREDATOR_SPK, .name = "predator-spk"},
{.id = ALC298_FIXUP_HUAWEI_MBX_STEREO, .name = "huawei-mbx-stereo"},
{.id = ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE, .name = "alc256-medion-headset"},
+ {.id = ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, .name = "alc298-samsung-headphone"},
{}
};
#define ALC225_STANDARD_PINS \
diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c
index b8161a08f2ca..58bb49fff184 100644
--- a/sound/ppc/snd_ps3.c
+++ b/sound/ppc/snd_ps3.c
@@ -227,14 +227,14 @@ static int snd_ps3_program_dma(struct snd_ps3_card_info *card,
switch (filltype) {
case SND_PS3_DMA_FILLTYPE_SILENT_FIRSTFILL:
silent = 1;
- /* intentionally fall thru */
+ fallthrough;
case SND_PS3_DMA_FILLTYPE_FIRSTFILL:
ch0_kick_event = PS3_AUDIO_KICK_EVENT_ALWAYS;
break;
case SND_PS3_DMA_FILLTYPE_SILENT_RUNNING:
silent = 1;
- /* intentionally fall thru */
+ fallthrough;
case SND_PS3_DMA_FILLTYPE_RUNNING:
ch0_kick_event = PS3_AUDIO_KICK_EVENT_SERIALOUT0_EMPTY;
break;
diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c
index 55815fdaa1aa..406526e79af3 100644
--- a/sound/soc/amd/acp3x-rt5682-max9836.c
+++ b/sound/soc/amd/acp3x-rt5682-max9836.c
@@ -138,7 +138,7 @@ static int acp3x_1015_hw_params(struct snd_pcm_substream *substream,
srate = params_rate(params);
for_each_rtd_codec_dais(rtd, i, codec_dai) {
- if (strcmp(codec_dai->component->name, "rt1015-aif"))
+ if (strcmp(codec_dai->name, "rt1015-aif"))
continue;
ret = snd_soc_dai_set_bclk_ratio(codec_dai, 64);
if (ret < 0)
diff --git a/sound/soc/amd/renoir/acp3x-pdm-dma.c b/sound/soc/amd/renoir/acp3x-pdm-dma.c
index 623dfd3ea705..7b14d9a81b97 100644
--- a/sound/soc/amd/renoir/acp3x-pdm-dma.c
+++ b/sound/soc/amd/renoir/acp3x-pdm-dma.c
@@ -314,40 +314,30 @@ static int acp_pdm_dma_close(struct snd_soc_component *component,
return 0;
}
-static int acp_pdm_dai_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params,
- struct snd_soc_dai *dai)
+static int acp_pdm_dai_trigger(struct snd_pcm_substream *substream,
+ int cmd, struct snd_soc_dai *dai)
{
struct pdm_stream_instance *rtd;
+ int ret;
+ bool pdm_status;
unsigned int ch_mask;
rtd = substream->runtime->private_data;
- switch (params_channels(params)) {
+ ret = 0;
+ switch (substream->runtime->channels) {
case TWO_CH:
ch_mask = 0x00;
break;
default:
return -EINVAL;
}
- rn_writel(ch_mask, rtd->acp_base + ACP_WOV_PDM_NO_OF_CHANNELS);
- rn_writel(PDM_DECIMATION_FACTOR, rtd->acp_base +
- ACP_WOV_PDM_DECIMATION_FACTOR);
- return 0;
-}
-
-static int acp_pdm_dai_trigger(struct snd_pcm_substream *substream,
- int cmd, struct snd_soc_dai *dai)
-{
- struct pdm_stream_instance *rtd;
- int ret;
- bool pdm_status;
-
- rtd = substream->runtime->private_data;
- ret = 0;
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ rn_writel(ch_mask, rtd->acp_base + ACP_WOV_PDM_NO_OF_CHANNELS);
+ rn_writel(PDM_DECIMATION_FACTOR, rtd->acp_base +
+ ACP_WOV_PDM_DECIMATION_FACTOR);
rtd->bytescount = acp_pdm_get_byte_count(rtd,
substream->stream);
pdm_status = check_pdm_dma_status(rtd->acp_base);
@@ -369,7 +359,6 @@ static int acp_pdm_dai_trigger(struct snd_pcm_substream *substream,
}
static struct snd_soc_dai_ops acp_pdm_dai_ops = {
- .hw_params = acp_pdm_dai_hw_params,
.trigger = acp_pdm_dai_trigger,
};
diff --git a/sound/soc/atmel/mchp-i2s-mcc.c b/sound/soc/atmel/mchp-i2s-mcc.c
index 3cb63886195f..04acc18f2d72 100644
--- a/sound/soc/atmel/mchp-i2s-mcc.c
+++ b/sound/soc/atmel/mchp-i2s-mcc.c
@@ -536,7 +536,7 @@ static int mchp_i2s_mcc_hw_params(struct snd_pcm_substream *substream,
/* cpu is BCLK master */
mrb |= MCHP_I2SMCC_MRB_CLKSEL_INT;
set_divs = 1;
- /* fall through */
+ fallthrough;
case SND_SOC_DAIFMT_CBM_CFM:
/* cpu is slave */
mra |= MCHP_I2SMCC_MRA_MODE_SLAVE;
diff --git a/sound/soc/codecs/jz4770.c b/sound/soc/codecs/jz4770.c
index c0a28f06b09a..298689a07168 100644
--- a/sound/soc/codecs/jz4770.c
+++ b/sound/soc/codecs/jz4770.c
@@ -202,7 +202,7 @@ static int jz4770_codec_set_bias_level(struct snd_soc_component *codec,
REG_CR_VIC_SB_SLEEP, REG_CR_VIC_SB_SLEEP);
regmap_update_bits(regmap, JZ4770_CODEC_REG_CR_VIC,
REG_CR_VIC_SB, REG_CR_VIC_SB);
- /* fall-through */
+ fallthrough;
default:
break;
}
diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c
index 4428c62e25cf..3ddd822240e3 100644
--- a/sound/soc/codecs/msm8916-wcd-analog.c
+++ b/sound/soc/codecs/msm8916-wcd-analog.c
@@ -19,8 +19,8 @@
#define CDC_D_REVISION1 (0xf000)
#define CDC_D_PERPH_SUBTYPE (0xf005)
-#define CDC_D_INT_EN_SET (0x015)
-#define CDC_D_INT_EN_CLR (0x016)
+#define CDC_D_INT_EN_SET (0xf015)
+#define CDC_D_INT_EN_CLR (0xf016)
#define MBHC_SWITCH_INT BIT(7)
#define MBHC_MIC_ELECTRICAL_INS_REM_DET BIT(6)
#define MBHC_BUTTON_PRESS_DET BIT(5)
diff --git a/sound/soc/codecs/pcm186x.c b/sound/soc/codecs/pcm186x.c
index f0da55901dcb..b8845f45549e 100644
--- a/sound/soc/codecs/pcm186x.c
+++ b/sound/soc/codecs/pcm186x.c
@@ -401,7 +401,7 @@ static int pcm186x_set_fmt(struct snd_soc_dai *dai, unsigned int format)
break;
case SND_SOC_DAIFMT_DSP_A:
priv->tdm_offset += 1;
- /* fall through */
+ fallthrough;
/* DSP_A uses the same basic config as DSP_B
* except we need to shift the TDM output by one BCK cycle
*/
diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c
index 68a3b48e6b31..3bce9a14f0f3 100644
--- a/sound/soc/codecs/wm8958-dsp2.c
+++ b/sound/soc/codecs/wm8958-dsp2.c
@@ -412,8 +412,12 @@ int wm8958_aif_ev(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 wm8994 *control = dev_get_drvdata(component->dev->parent);
int i;
+ if (control->type != WM8958)
+ return 0;
+
switch (event) {
case SND_SOC_DAPM_POST_PMU:
case SND_SOC_DAPM_PRE_PMU:
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 317916cb4e27..0623a2251084 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -151,7 +151,6 @@ static const struct reg_default wm8962_reg[] = {
{ 40, 0x0000 }, /* R40 - SPKOUTL volume */
{ 41, 0x0000 }, /* R41 - SPKOUTR volume */
- { 48, 0x0000 }, /* R48 - Additional control(4) */
{ 49, 0x0010 }, /* R49 - Class D Control 1 */
{ 51, 0x0003 }, /* R51 - Class D Control 2 */
@@ -842,6 +841,7 @@ static bool wm8962_readable_register(struct device *dev, unsigned int reg)
case WM8962_SPKOUTL_VOLUME:
case WM8962_SPKOUTR_VOLUME:
case WM8962_THERMAL_SHUTDOWN_STATUS:
+ case WM8962_ADDITIONAL_CONTROL_4:
case WM8962_CLASS_D_CONTROL_1:
case WM8962_CLASS_D_CONTROL_2:
case WM8962_CLOCKING_4:
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index a84ae879d37e..038be667c1a6 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -43,10 +43,12 @@
#define WM8994_NUM_DRC 3
#define WM8994_NUM_EQ 3
-static struct {
+struct wm8994_reg_mask {
unsigned int reg;
unsigned int mask;
-} wm8994_vu_bits[] = {
+};
+
+static struct wm8994_reg_mask wm8994_vu_bits[] = {
{ WM8994_LEFT_LINE_INPUT_1_2_VOLUME, WM8994_IN1_VU },
{ WM8994_RIGHT_LINE_INPUT_1_2_VOLUME, WM8994_IN1_VU },
{ WM8994_LEFT_LINE_INPUT_3_4_VOLUME, WM8994_IN2_VU },
@@ -60,14 +62,10 @@ static struct {
{ WM8994_AIF1_DAC1_LEFT_VOLUME, WM8994_AIF1DAC1_VU },
{ WM8994_AIF1_DAC1_RIGHT_VOLUME, WM8994_AIF1DAC1_VU },
- { WM8994_AIF1_DAC2_LEFT_VOLUME, WM8994_AIF1DAC2_VU },
- { WM8994_AIF1_DAC2_RIGHT_VOLUME, WM8994_AIF1DAC2_VU },
{ WM8994_AIF2_DAC_LEFT_VOLUME, WM8994_AIF2DAC_VU },
{ WM8994_AIF2_DAC_RIGHT_VOLUME, WM8994_AIF2DAC_VU },
{ WM8994_AIF1_ADC1_LEFT_VOLUME, WM8994_AIF1ADC1_VU },
{ WM8994_AIF1_ADC1_RIGHT_VOLUME, WM8994_AIF1ADC1_VU },
- { WM8994_AIF1_ADC2_LEFT_VOLUME, WM8994_AIF1ADC2_VU },
- { WM8994_AIF1_ADC2_RIGHT_VOLUME, WM8994_AIF1ADC2_VU },
{ WM8994_AIF2_ADC_LEFT_VOLUME, WM8994_AIF2ADC_VU },
{ WM8994_AIF2_ADC_RIGHT_VOLUME, WM8994_AIF1ADC2_VU },
{ WM8994_DAC1_LEFT_VOLUME, WM8994_DAC1_VU },
@@ -76,6 +74,14 @@ static struct {
{ WM8994_DAC2_RIGHT_VOLUME, WM8994_DAC2_VU },
};
+/* VU bitfields for ADC2, DAC2 not available on WM1811 */
+static struct wm8994_reg_mask wm8994_adc2_dac2_vu_bits[] = {
+ { WM8994_AIF1_DAC2_LEFT_VOLUME, WM8994_AIF1DAC2_VU },
+ { WM8994_AIF1_DAC2_RIGHT_VOLUME, WM8994_AIF1DAC2_VU },
+ { WM8994_AIF1_ADC2_LEFT_VOLUME, WM8994_AIF1ADC2_VU },
+ { WM8994_AIF1_ADC2_RIGHT_VOLUME, WM8994_AIF1ADC2_VU },
+};
+
static int wm8994_drc_base[] = {
WM8994_AIF1_DRC1_1,
WM8994_AIF1_DRC2_1,
@@ -1030,6 +1036,26 @@ static bool wm8994_check_class_w_digital(struct snd_soc_component *component)
return true;
}
+static void wm8994_update_vu_bits(struct snd_soc_component *component)
+{
+ struct wm8994_priv *wm8994 = snd_soc_component_get_drvdata(component);
+ struct wm8994 *control = wm8994->wm8994;
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(wm8994_vu_bits); i++)
+ snd_soc_component_write(component, wm8994_vu_bits[i].reg,
+ snd_soc_component_read(component,
+ wm8994_vu_bits[i].reg));
+ if (control->type == WM1811)
+ return;
+
+ for (i = 0; i < ARRAY_SIZE(wm8994_adc2_dac2_vu_bits); i++)
+ snd_soc_component_write(component,
+ wm8994_adc2_dac2_vu_bits[i].reg,
+ snd_soc_component_read(component,
+ wm8994_adc2_dac2_vu_bits[i].reg));
+}
+
static int aif_mclk_set(struct snd_soc_component *component, int aif, bool enable)
{
struct wm8994_priv *wm8994 = snd_soc_component_get_drvdata(component);
@@ -1076,7 +1102,7 @@ static int aif1clk_ev(struct snd_soc_dapm_widget *w,
struct wm8994_priv *wm8994 = snd_soc_component_get_drvdata(component);
struct wm8994 *control = wm8994->wm8994;
int mask = WM8994_AIF1DAC1L_ENA | WM8994_AIF1DAC1R_ENA;
- int ret, i;
+ int ret;
int dac;
int adc;
int val;
@@ -1144,10 +1170,7 @@ static int aif1clk_ev(struct snd_soc_dapm_widget *w,
break;
case SND_SOC_DAPM_POST_PMU:
- for (i = 0; i < ARRAY_SIZE(wm8994_vu_bits); i++)
- snd_soc_component_write(component, wm8994_vu_bits[i].reg,
- snd_soc_component_read(component,
- wm8994_vu_bits[i].reg));
+ wm8994_update_vu_bits(component);
break;
case SND_SOC_DAPM_PRE_PMD:
@@ -1181,7 +1204,7 @@ static int aif2clk_ev(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm);
- int ret, i;
+ int ret;
int dac;
int adc;
int val;
@@ -1237,10 +1260,7 @@ static int aif2clk_ev(struct snd_soc_dapm_widget *w,
break;
case SND_SOC_DAPM_POST_PMU:
- for (i = 0; i < ARRAY_SIZE(wm8994_vu_bits); i++)
- snd_soc_component_write(component, wm8994_vu_bits[i].reg,
- snd_soc_component_read(component,
- wm8994_vu_bits[i].reg));
+ wm8994_update_vu_bits(component);
break;
case SND_SOC_DAPM_PRE_PMD:
@@ -4346,6 +4366,14 @@ static int wm8994_component_probe(struct snd_soc_component *component)
wm8994_vu_bits[i].mask,
wm8994_vu_bits[i].mask);
+ if (control->type != WM1811) {
+ for (i = 0; i < ARRAY_SIZE(wm8994_adc2_dac2_vu_bits); i++)
+ snd_soc_component_update_bits(component,
+ wm8994_adc2_dac2_vu_bits[i].reg,
+ wm8994_adc2_dac2_vu_bits[i].mask,
+ wm8994_adc2_dac2_vu_bits[i].mask);
+ }
+
/* Set the low bit of the 3D stereo depth so TLV matches */
snd_soc_component_update_bits(component, WM8994_AIF1_DAC1_FILTERS_2,
1 << WM8994_AIF1DAC1_3D_GAIN_SHIFT,
diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
index de136c0a497d..52adedc03245 100644
--- a/sound/soc/fsl/fsl-asoc-card.c
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -73,6 +73,7 @@ struct cpu_priv {
* @codec_priv: CODEC private data
* @cpu_priv: CPU private data
* @card: ASoC card structure
+ * @streams: Mask of current active streams
* @sample_rate: Current sample rate
* @sample_format: Current sample format
* @asrc_rate: ASRC sample rate used by Back-Ends
@@ -89,6 +90,7 @@ struct fsl_asoc_card_priv {
struct codec_priv codec_priv;
struct cpu_priv cpu_priv;
struct snd_soc_card card;
+ u8 streams;
u32 sample_rate;
snd_pcm_format_t sample_format;
u32 asrc_rate;
@@ -151,21 +153,17 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card);
bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+ struct codec_priv *codec_priv = &priv->codec_priv;
struct cpu_priv *cpu_priv = &priv->cpu_priv;
struct device *dev = rtd->card->dev;
+ unsigned int pll_out;
int ret;
priv->sample_rate = params_rate(params);
priv->sample_format = params_format(params);
+ priv->streams |= BIT(substream->stream);
- /*
- * If codec-dai is DAI Master and all configurations are already in the
- * set_bias_level(), bypass the remaining settings in hw_params().
- * Note: (dai_fmt & CBM_CFM) includes CBM_CFM and CBM_CFS.
- */
- if ((priv->card.set_bias_level &&
- priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM) ||
- fsl_asoc_card_is_ac97(priv))
+ if (fsl_asoc_card_is_ac97(priv))
return 0;
/* Specific configurations of DAIs starts from here */
@@ -174,7 +172,7 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream,
cpu_priv->sysclk_dir[tx]);
if (ret && ret != -ENOTSUPP) {
dev_err(dev, "failed to set sysclk for cpu dai\n");
- return ret;
+ goto fail;
}
if (cpu_priv->slot_width) {
@@ -182,6 +180,68 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream,
cpu_priv->slot_width);
if (ret && ret != -ENOTSUPP) {
dev_err(dev, "failed to set TDM slot for cpu dai\n");
+ goto fail;
+ }
+ }
+
+ /* Specific configuration for PLL */
+ if (codec_priv->pll_id && codec_priv->fll_id) {
+ if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE)
+ pll_out = priv->sample_rate * 384;
+ else
+ pll_out = priv->sample_rate * 256;
+
+ ret = snd_soc_dai_set_pll(asoc_rtd_to_codec(rtd, 0),
+ codec_priv->pll_id,
+ codec_priv->mclk_id,
+ codec_priv->mclk_freq, pll_out);
+ if (ret) {
+ dev_err(dev, "failed to start FLL: %d\n", ret);
+ goto fail;
+ }
+
+ ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, 0),
+ codec_priv->fll_id,
+ pll_out, SND_SOC_CLOCK_IN);
+
+ if (ret && ret != -ENOTSUPP) {
+ dev_err(dev, "failed to set SYSCLK: %d\n", ret);
+ goto fail;
+ }
+ }
+
+ return 0;
+
+fail:
+ priv->streams &= ~BIT(substream->stream);
+ return ret;
+}
+
+static int fsl_asoc_card_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card);
+ struct codec_priv *codec_priv = &priv->codec_priv;
+ struct device *dev = rtd->card->dev;
+ int ret;
+
+ priv->streams &= ~BIT(substream->stream);
+
+ if (!priv->streams && codec_priv->pll_id && codec_priv->fll_id) {
+ /* Force freq to be 0 to avoid error message in codec */
+ ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, 0),
+ codec_priv->mclk_id,
+ 0,
+ SND_SOC_CLOCK_IN);
+ if (ret) {
+ dev_err(dev, "failed to switch away from FLL: %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_dai_set_pll(asoc_rtd_to_codec(rtd, 0),
+ codec_priv->pll_id, 0, 0, 0);
+ if (ret && ret != -ENOTSUPP) {
+ dev_err(dev, "failed to stop FLL: %d\n", ret);
return ret;
}
}
@@ -191,6 +251,7 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream,
static const struct snd_soc_ops fsl_asoc_card_ops = {
.hw_params = fsl_asoc_card_hw_params,
+ .hw_free = fsl_asoc_card_hw_free,
};
static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
@@ -254,75 +315,6 @@ static struct snd_soc_dai_link fsl_asoc_card_dai[] = {
},
};
-static int fsl_asoc_card_set_bias_level(struct snd_soc_card *card,
- struct snd_soc_dapm_context *dapm,
- enum snd_soc_bias_level level)
-{
- struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card);
- struct snd_soc_pcm_runtime *rtd;
- struct snd_soc_dai *codec_dai;
- struct codec_priv *codec_priv = &priv->codec_priv;
- struct device *dev = card->dev;
- unsigned int pll_out;
- int ret;
-
- rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]);
- codec_dai = asoc_rtd_to_codec(rtd, 0);
- if (dapm->dev != codec_dai->dev)
- return 0;
-
- switch (level) {
- case SND_SOC_BIAS_PREPARE:
- if (dapm->bias_level != SND_SOC_BIAS_STANDBY)
- break;
-
- if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE)
- pll_out = priv->sample_rate * 384;
- else
- pll_out = priv->sample_rate * 256;
-
- ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id,
- codec_priv->mclk_id,
- codec_priv->mclk_freq, pll_out);
- if (ret) {
- dev_err(dev, "failed to start FLL: %d\n", ret);
- return ret;
- }
-
- ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->fll_id,
- pll_out, SND_SOC_CLOCK_IN);
- if (ret && ret != -ENOTSUPP) {
- dev_err(dev, "failed to set SYSCLK: %d\n", ret);
- return ret;
- }
- break;
-
- case SND_SOC_BIAS_STANDBY:
- if (dapm->bias_level != SND_SOC_BIAS_PREPARE)
- break;
-
- ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id,
- codec_priv->mclk_freq,
- SND_SOC_CLOCK_IN);
- if (ret && ret != -ENOTSUPP) {
- dev_err(dev, "failed to switch away from FLL: %d\n", ret);
- return ret;
- }
-
- ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id, 0, 0, 0);
- if (ret) {
- dev_err(dev, "failed to stop FLL: %d\n", ret);
- return ret;
- }
- break;
-
- default:
- break;
- }
-
- return 0;
-}
-
static int fsl_asoc_card_audmux_init(struct device_node *np,
struct fsl_asoc_card_priv *priv)
{
@@ -611,7 +603,6 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
/* Diversify the card configurations */
if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) {
codec_dai_name = "cs42888";
- priv->card.set_bias_level = NULL;
priv->cpu_priv.sysclk_freq[TX] = priv->codec_priv.mclk_freq;
priv->cpu_priv.sysclk_freq[RX] = priv->codec_priv.mclk_freq;
priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT;
@@ -628,26 +619,22 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
} else if (of_device_is_compatible(np, "fsl,imx-audio-wm8962")) {
codec_dai_name = "wm8962";
- priv->card.set_bias_level = fsl_asoc_card_set_bias_level;
priv->codec_priv.mclk_id = WM8962_SYSCLK_MCLK;
priv->codec_priv.fll_id = WM8962_SYSCLK_FLL;
priv->codec_priv.pll_id = WM8962_FLL;
priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
} else if (of_device_is_compatible(np, "fsl,imx-audio-wm8960")) {
codec_dai_name = "wm8960-hifi";
- priv->card.set_bias_level = fsl_asoc_card_set_bias_level;
priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO;
priv->codec_priv.pll_id = WM8960_SYSCLK_AUTO;
priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
} else if (of_device_is_compatible(np, "fsl,imx-audio-ac97")) {
codec_dai_name = "ac97-hifi";
- priv->card.set_bias_level = NULL;
priv->dai_fmt = SND_SOC_DAIFMT_AC97;
priv->card.dapm_routes = audio_map_ac97;
priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_ac97);
} else if (of_device_is_compatible(np, "fsl,imx-audio-mqs")) {
codec_dai_name = "fsl-mqs-dai";
- priv->card.set_bias_level = NULL;
priv->dai_fmt = SND_SOC_DAIFMT_LEFT_J |
SND_SOC_DAIFMT_CBS_CFS |
SND_SOC_DAIFMT_NB_NF;
@@ -657,7 +644,6 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_tx);
} else if (of_device_is_compatible(np, "fsl,imx-audio-wm8524")) {
codec_dai_name = "wm8524-hifi";
- priv->card.set_bias_level = NULL;
priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS;
priv->dai_link[1].dpcm_capture = 0;
priv->dai_link[2].dpcm_capture = 0;
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index d8b9c6547142..404be27c15fe 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -898,7 +898,7 @@ static int _fsl_ssi_set_dai_fmt(struct fsl_ssi *ssi, unsigned int fmt)
"missing baudclk for master mode\n");
return -EINVAL;
}
- /* fall through */
+ fallthrough;
case SND_SOC_DAIFMT_CBM_CFS:
ssi->i2s_net |= SSI_SCR_I2S_MODE_MASTER;
break;
diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c
index 9e4f66b6b92b..231984882176 100644
--- a/sound/soc/fsl/mpc5200_dma.c
+++ b/sound/soc/fsl/mpc5200_dma.c
@@ -339,7 +339,6 @@ static int psc_dma_new(struct snd_soc_component *component,
static void psc_dma_free(struct snd_soc_component *component,
struct snd_pcm *pcm)
{
- struct snd_soc_pcm_runtime *rtd = pcm->private_data;
struct snd_pcm_substream *substream;
int stream;
diff --git a/sound/soc/hisilicon/hi6210-i2s.c b/sound/soc/hisilicon/hi6210-i2s.c
index fd5dcd6b9f85..907f5f1f7b44 100644
--- a/sound/soc/hisilicon/hi6210-i2s.c
+++ b/sound/soc/hisilicon/hi6210-i2s.c
@@ -261,13 +261,13 @@ static int hi6210_i2s_hw_params(struct snd_pcm_substream *substream,
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_U16_LE:
signed_data = HII2S_I2S_CFG__S2_CODEC_DATA_FORMAT;
- /* fall through */
+ fallthrough;
case SNDRV_PCM_FORMAT_S16_LE:
bits = HII2S_BITS_16;
break;
case SNDRV_PCM_FORMAT_U24_LE:
signed_data = HII2S_I2S_CFG__S2_CODEC_DATA_FORMAT;
- /* fall through */
+ fallthrough;
case SNDRV_PCM_FORMAT_S24_LE:
bits = HII2S_BITS_24;
break;
diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
index 49b9f18472bc..b1cac7abdc0a 100644
--- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c
+++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
@@ -331,7 +331,7 @@ static int sst_media_open(struct snd_pcm_substream *substream,
ret_val = power_up_sst(stream);
if (ret_val < 0)
- return ret_val;
+ goto out_power_up;
/* Make sure, that the period size is always even */
snd_pcm_hw_constraint_step(substream->runtime, 0,
@@ -340,8 +340,9 @@ static int sst_media_open(struct snd_pcm_substream *substream,
return snd_pcm_hw_constraint_integer(runtime,
SNDRV_PCM_HW_PARAM_PERIODS);
out_ops:
- kfree(stream);
mutex_unlock(&sst_lock);
+out_power_up:
+ kfree(stream);
return ret_val;
}
diff --git a/sound/soc/intel/baytrail/sst-baytrail-pcm.c b/sound/soc/intel/baytrail/sst-baytrail-pcm.c
index 54a66cc6db89..d2cda33b65d5 100644
--- a/sound/soc/intel/baytrail/sst-baytrail-pcm.c
+++ b/sound/soc/intel/baytrail/sst-baytrail-pcm.c
@@ -181,7 +181,7 @@ static int sst_byt_pcm_trigger(struct snd_soc_component *component,
break;
case SNDRV_PCM_TRIGGER_SUSPEND:
pdata->restore_stream = false;
- /* fallthrough */
+ fallthrough;
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
sst_byt_stream_pause(byt, pcm_data->stream);
break;
diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c
index 414ae4bb5224..7ae34b49815c 100644
--- a/sound/soc/intel/boards/bytcht_es8316.c
+++ b/sound/soc/intel/boards/bytcht_es8316.c
@@ -573,7 +573,7 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev)
break;
default:
dev_err(dev, "get speaker GPIO failed: %d\n", ret);
- /* fall through */
+ fallthrough;
case -EPROBE_DEFER:
return ret;
}
diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c
index 4e2897596cea..688b5e0a49e3 100644
--- a/sound/soc/intel/boards/bytcr_rt5651.c
+++ b/sound/soc/intel/boards/bytcr_rt5651.c
@@ -1009,7 +1009,7 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev)
default:
dev_err(&pdev->dev, "Failed to get ext-amp-enable GPIO: %d\n",
ret_val);
- /* fall through */
+ fallthrough;
case -EPROBE_DEFER:
put_device(codec_dev);
return ret_val;
@@ -1029,7 +1029,7 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev)
default:
dev_err(&pdev->dev, "Failed to get hp-detect GPIO: %d\n",
ret_val);
- /* fall through */
+ fallthrough;
case -EPROBE_DEFER:
put_device(codec_dev);
return ret_val;
diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c
index 5dee55e9546b..bbe8d782e0af 100644
--- a/sound/soc/intel/skylake/skl-pcm.c
+++ b/sound/soc/intel/skylake/skl-pcm.c
@@ -488,7 +488,7 @@ static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd,
stream->lpib);
snd_hdac_ext_stream_set_lpib(stream, stream->lpib);
}
- /* fall through */
+ fallthrough;
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
diff --git a/sound/soc/meson/axg-tdm-interface.c b/sound/soc/meson/axg-tdm-interface.c
index 36df30915378..c8664ab80d45 100644
--- a/sound/soc/meson/axg-tdm-interface.c
+++ b/sound/soc/meson/axg-tdm-interface.c
@@ -58,17 +58,17 @@ int axg_tdm_set_tdm_slots(struct snd_soc_dai *dai, u32 *tx_mask,
switch (slot_width) {
case 0:
slot_width = 32;
- /* Fall-through */
+ fallthrough;
case 32:
fmt |= SNDRV_PCM_FMTBIT_S32_LE;
- /* Fall-through */
+ fallthrough;
case 24:
fmt |= SNDRV_PCM_FMTBIT_S24_LE;
fmt |= SNDRV_PCM_FMTBIT_S20_LE;
- /* Fall-through */
+ fallthrough;
case 16:
fmt |= SNDRV_PCM_FMTBIT_S16_LE;
- /* Fall-through */
+ fallthrough;
case 8:
fmt |= SNDRV_PCM_FMTBIT_S8;
break;
@@ -133,7 +133,7 @@ static int axg_tdm_iface_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
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 */
+ fallthrough;
default:
return -EINVAL;
}
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index d1e09ade0190..c4e7307a4437 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -488,7 +488,7 @@ static int pxa_ssp_configure_dai_fmt(struct ssp_priv *priv)
case SND_SOC_DAIFMT_DSP_A:
sspsp |= SSPSP_FSRT;
- /* fall through */
+ fallthrough;
case SND_SOC_DAIFMT_DSP_B:
sscr0 |= SSCR0_MOD | SSCR0_PSP;
sscr1 |= SSCR1_TRAIL | SSCR1_RWOT;
diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c
index 2a5302f1db98..0168af849272 100644
--- a/sound/soc/qcom/qdsp6/q6afe-dai.c
+++ b/sound/soc/qcom/qdsp6/q6afe-dai.c
@@ -1150,206 +1150,206 @@ static int q6afe_of_xlate_dai_name(struct snd_soc_component *component,
}
static const struct snd_soc_dapm_widget q6afe_dai_widgets[] = {
- SND_SOC_DAPM_AIF_IN("HDMI_RX", NULL, 0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("SLIMBUS_0_RX", NULL, 0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("SLIMBUS_1_RX", NULL, 0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("SLIMBUS_2_RX", NULL, 0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("SLIMBUS_3_RX", NULL, 0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("SLIMBUS_4_RX", NULL, 0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("SLIMBUS_5_RX", NULL, 0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("SLIMBUS_6_RX", NULL, 0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("SLIMBUS_0_TX", NULL, 0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("SLIMBUS_1_TX", NULL, 0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("SLIMBUS_2_TX", NULL, 0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("SLIMBUS_3_TX", NULL, 0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("SLIMBUS_4_TX", NULL, 0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("SLIMBUS_5_TX", NULL, 0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("SLIMBUS_6_TX", NULL, 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("HDMI_RX", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("SLIMBUS_0_RX", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("SLIMBUS_1_RX", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("SLIMBUS_2_RX", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("SLIMBUS_3_RX", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("SLIMBUS_4_RX", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("SLIMBUS_5_RX", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("SLIMBUS_6_RX", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("SLIMBUS_0_TX", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("SLIMBUS_1_TX", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("SLIMBUS_2_TX", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("SLIMBUS_3_TX", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("SLIMBUS_4_TX", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("SLIMBUS_5_TX", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("SLIMBUS_6_TX", NULL, 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("QUAT_MI2S_RX", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("QUAT_MI2S_TX", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("TERT_MI2S_RX", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("TERT_MI2S_TX", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("SEC_MI2S_RX", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("SEC_MI2S_TX", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("SEC_MI2S_RX_SD1",
"Secondary MI2S Playback SD1",
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("PRI_MI2S_RX", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("PRI_MI2S_TX", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_0", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_1", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_2", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_3", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_4", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_5", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_6", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_7", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_0", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_1", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_2", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_3", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_4", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_5", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_6", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_7", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_0", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_1", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_2", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_3", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_4", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_5", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_6", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_7", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_0", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_1", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_2", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_3", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_4", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_5", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_6", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_7", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_0", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_1", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_2", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_3", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_4", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_5", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_6", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_7", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_0", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_1", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_2", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_3", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_4", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_5", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_6", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_7", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_0", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_1", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_2", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_3", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_4", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_5", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_6", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_7", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_0", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_1", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_2", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_3", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_4", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_5", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_6", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_7", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_0", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_1", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_2", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_3", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_4", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_5", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_6", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_7", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_0", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_1", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_2", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_3", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_4", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_5", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_6", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_7", NULL,
- 0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("DISPLAY_PORT_RX", "NULL", 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("DISPLAY_PORT_RX", "NULL", 0, SND_SOC_NOPM, 0, 0),
};
static const struct snd_soc_component_driver q6afe_dai_component = {
diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c
index eaa95b5a7b66..25d23e0266c7 100644
--- a/sound/soc/qcom/qdsp6/q6routing.c
+++ b/sound/soc/qcom/qdsp6/q6routing.c
@@ -973,6 +973,20 @@ static int msm_routing_probe(struct snd_soc_component *c)
return 0;
}
+static unsigned int q6routing_reg_read(struct snd_soc_component *component,
+ unsigned int reg)
+{
+ /* default value */
+ return 0;
+}
+
+static int q6routing_reg_write(struct snd_soc_component *component,
+ unsigned int reg, unsigned int val)
+{
+ /* dummy */
+ return 0;
+}
+
static const struct snd_soc_component_driver msm_soc_routing_component = {
.probe = msm_routing_probe,
.name = DRV_NAME,
@@ -981,6 +995,8 @@ static const struct snd_soc_component_driver msm_soc_routing_component = {
.num_dapm_widgets = ARRAY_SIZE(msm_qdsp6_widgets),
.dapm_routes = intercon,
.num_dapm_routes = ARRAY_SIZE(intercon),
+ .read = q6routing_reg_read,
+ .write = q6routing_reg_write,
};
static int q6pcm_routing_probe(struct platform_device *pdev)
diff --git a/sound/soc/rockchip/rockchip_pdm.c b/sound/soc/rockchip/rockchip_pdm.c
index 1707414cfa92..5adb293d0435 100644
--- a/sound/soc/rockchip/rockchip_pdm.c
+++ b/sound/soc/rockchip/rockchip_pdm.c
@@ -229,13 +229,13 @@ static int rockchip_pdm_hw_params(struct snd_pcm_substream *substream,
switch (params_channels(params)) {
case 8:
val |= PDM_PATH3_EN;
- /* fallthrough */
+ fallthrough;
case 6:
val |= PDM_PATH2_EN;
- /* fallthrough */
+ fallthrough;
case 4:
val |= PDM_PATH1_EN;
- /* fallthrough */
+ fallthrough;
case 2:
val |= PDM_PATH0_EN;
break;
diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c
index 80ecb5c7fed0..df53d4ea808f 100644
--- a/sound/soc/samsung/i2s.c
+++ b/sound/soc/samsung/i2s.c
@@ -733,7 +733,7 @@ static int i2s_hw_params(struct snd_pcm_substream *substream,
switch (params_channels(params)) {
case 6:
val |= MOD_DC2_EN;
- /* Fall through */
+ fallthrough;
case 4:
val |= MOD_DC1_EN;
break;
diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c
index f0b4f4bc44a4..5504b92946e3 100644
--- a/sound/soc/soc-component.c
+++ b/sound/soc/soc-component.c
@@ -406,7 +406,7 @@ static unsigned int soc_component_read_no_lock(
ret = -EIO;
if (ret < 0)
- soc_component_ret(component, ret);
+ return soc_component_ret(component, ret);
return val;
}
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 2fe1b2ec7c8f..663e3839f251 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -618,7 +618,7 @@ int snd_soc_suspend(struct device *dev)
"ASoC: idle_bias_off CODEC on over suspend\n");
break;
}
- /* fall through */
+ fallthrough;
case SND_SOC_BIAS_OFF:
snd_soc_component_suspend(component);
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index cee998671318..5b60379237bf 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -1057,7 +1057,7 @@ static int soc_tplg_denum_create(struct soc_tplg *tplg, unsigned int count,
ec->hdr.name);
goto err_denum;
}
- /* fall through */
+ fallthrough;
case SND_SOC_TPLG_CTL_ENUM:
case SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE:
case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT:
@@ -1445,7 +1445,7 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_denum_create(
ec->hdr.name);
goto err_se;
}
- /* fall through */
+ fallthrough;
case SND_SOC_TPLG_CTL_ENUM:
case SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE:
case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT:
diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c
index df1c6997cb4e..c6cb8c212eca 100644
--- a/sound/soc/sof/intel/hda-dai.c
+++ b/sound/soc/sof/intel/hda-dai.c
@@ -310,7 +310,7 @@ static int hda_link_pcm_trigger(struct snd_pcm_substream *substream,
return ret;
}
- /* fallthrough */
+ fallthrough;
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
snd_hdac_ext_link_stream_start(link_dev);
@@ -333,7 +333,7 @@ static int hda_link_pcm_trigger(struct snd_pcm_substream *substream,
link_dev->link_prepared = 0;
- /* fallthrough */
+ fallthrough;
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
snd_hdac_ext_link_stream_clear(link_dev);
break;
diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c
index d730e437e4ba..71c3f29057a7 100644
--- a/sound/soc/sof/pcm.c
+++ b/sound/soc/sof/pcm.c
@@ -361,7 +361,7 @@ static int sof_pcm_trigger(struct snd_soc_component *component,
return ret;
}
- /* fallthrough */
+ fallthrough;
case SNDRV_PCM_TRIGGER_START:
if (spcm->stream[substream->stream].suspend_ignored) {
/*
@@ -386,7 +386,7 @@ static int sof_pcm_trigger(struct snd_soc_component *component,
spcm->stream[substream->stream].suspend_ignored = true;
return 0;
}
- /* fallthrough */
+ fallthrough;
case SNDRV_PCM_TRIGGER_STOP:
stream.hdr.cmd |= SOF_IPC_STREAM_TRIG_STOP;
ipc_first = true;
diff --git a/sound/soc/tegra/tegra186_dspk.c b/sound/soc/tegra/tegra186_dspk.c
index fe7117171a0e..0cbe31e2c7e9 100644
--- a/sound/soc/tegra/tegra186_dspk.c
+++ b/sound/soc/tegra/tegra186_dspk.c
@@ -71,7 +71,7 @@ static int tegra186_dspk_put_control(struct snd_kcontrol *kcontrol,
return 0;
}
-static int tegra186_dspk_runtime_suspend(struct device *dev)
+static int __maybe_unused tegra186_dspk_runtime_suspend(struct device *dev)
{
struct tegra186_dspk *dspk = dev_get_drvdata(dev);
@@ -83,7 +83,7 @@ static int tegra186_dspk_runtime_suspend(struct device *dev)
return 0;
}
-static int tegra186_dspk_runtime_resume(struct device *dev)
+static int __maybe_unused tegra186_dspk_runtime_resume(struct device *dev)
{
struct tegra186_dspk *dspk = dev_get_drvdata(dev);
int err;
diff --git a/sound/soc/tegra/tegra210_admaif.c b/sound/soc/tegra/tegra210_admaif.c
index 4894e8e6ee7f..1268046b345d 100644
--- a/sound/soc/tegra/tegra210_admaif.c
+++ b/sound/soc/tegra/tegra210_admaif.c
@@ -219,7 +219,7 @@ static const struct regmap_config tegra186_admaif_regmap_config = {
.cache_type = REGCACHE_FLAT,
};
-static int tegra_admaif_runtime_suspend(struct device *dev)
+static int __maybe_unused tegra_admaif_runtime_suspend(struct device *dev)
{
struct tegra_admaif *admaif = dev_get_drvdata(dev);
@@ -229,7 +229,7 @@ static int tegra_admaif_runtime_suspend(struct device *dev)
return 0;
}
-static int tegra_admaif_runtime_resume(struct device *dev)
+static int __maybe_unused tegra_admaif_runtime_resume(struct device *dev)
{
struct tegra_admaif *admaif = dev_get_drvdata(dev);
diff --git a/sound/soc/tegra/tegra210_ahub.c b/sound/soc/tegra/tegra210_ahub.c
index 5123a96fdde8..66287a7c9865 100644
--- a/sound/soc/tegra/tegra210_ahub.c
+++ b/sound/soc/tegra/tegra210_ahub.c
@@ -564,7 +564,7 @@ static const struct of_device_id tegra_ahub_of_match[] = {
};
MODULE_DEVICE_TABLE(of, tegra_ahub_of_match);
-static int tegra_ahub_runtime_suspend(struct device *dev)
+static int __maybe_unused tegra_ahub_runtime_suspend(struct device *dev)
{
struct tegra_ahub *ahub = dev_get_drvdata(dev);
@@ -576,7 +576,7 @@ static int tegra_ahub_runtime_suspend(struct device *dev)
return 0;
}
-static int tegra_ahub_runtime_resume(struct device *dev)
+static int __maybe_unused tegra_ahub_runtime_resume(struct device *dev)
{
struct tegra_ahub *ahub = dev_get_drvdata(dev);
int err;
diff --git a/sound/soc/tegra/tegra210_dmic.c b/sound/soc/tegra/tegra210_dmic.c
index d682414ad90d..a661f40bc41c 100644
--- a/sound/soc/tegra/tegra210_dmic.c
+++ b/sound/soc/tegra/tegra210_dmic.c
@@ -40,7 +40,7 @@ static const struct reg_default tegra210_dmic_reg_defaults[] = {
{ TEGRA210_DMIC_LP_BIQUAD_1_COEF_4, 0x0 },
};
-static int tegra210_dmic_runtime_suspend(struct device *dev)
+static int __maybe_unused tegra210_dmic_runtime_suspend(struct device *dev)
{
struct tegra210_dmic *dmic = dev_get_drvdata(dev);
@@ -52,7 +52,7 @@ static int tegra210_dmic_runtime_suspend(struct device *dev)
return 0;
}
-static int tegra210_dmic_runtime_resume(struct device *dev)
+static int __maybe_unused tegra210_dmic_runtime_resume(struct device *dev)
{
struct tegra210_dmic *dmic = dev_get_drvdata(dev);
int err;
diff --git a/sound/soc/tegra/tegra210_i2s.c b/sound/soc/tegra/tegra210_i2s.c
index 722092181583..a383bd5c51cd 100644
--- a/sound/soc/tegra/tegra210_i2s.c
+++ b/sound/soc/tegra/tegra210_i2s.c
@@ -164,7 +164,7 @@ static int tegra210_i2s_init(struct snd_soc_dapm_widget *w,
return tegra210_i2s_sw_reset(compnt, is_playback);
}
-static int tegra210_i2s_runtime_suspend(struct device *dev)
+static int __maybe_unused tegra210_i2s_runtime_suspend(struct device *dev)
{
struct tegra210_i2s *i2s = dev_get_drvdata(dev);
@@ -176,7 +176,7 @@ static int tegra210_i2s_runtime_suspend(struct device *dev)
return 0;
}
-static int tegra210_i2s_runtime_resume(struct device *dev)
+static int __maybe_unused tegra210_i2s_runtime_resume(struct device *dev)
{
struct tegra210_i2s *i2s = dev_get_drvdata(dev);
int err;
diff --git a/sound/soc/ti/davinci-i2s.c b/sound/soc/ti/davinci-i2s.c
index d89b5c928c4d..dd34504c09ba 100644
--- a/sound/soc/ti/davinci-i2s.c
+++ b/sound/soc/ti/davinci-i2s.c
@@ -289,7 +289,7 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
* rate is lowered.
*/
inv_fs = true;
- /* fall through */
+ fallthrough;
case SND_SOC_DAIFMT_DSP_A:
dev->mode = MOD_DSP_A;
break;
diff --git a/sound/soc/ti/n810.c b/sound/soc/ti/n810.c
index 2802a33b9c5f..ed217b34f846 100644
--- a/sound/soc/ti/n810.c
+++ b/sound/soc/ti/n810.c
@@ -46,7 +46,7 @@ static void n810_ext_control(struct snd_soc_dapm_context *dapm)
switch (n810_jack_func) {
case N810_JACK_HS:
line1l = 1;
- /* fall through */
+ fallthrough;
case N810_JACK_HP:
hp = 1;
break;
diff --git a/sound/soc/ti/omap-dmic.c b/sound/soc/ti/omap-dmic.c
index 01abf1be5d78..a26588e9c3bc 100644
--- a/sound/soc/ti/omap-dmic.c
+++ b/sound/soc/ti/omap-dmic.c
@@ -203,10 +203,10 @@ static int omap_dmic_dai_hw_params(struct snd_pcm_substream *substream,
switch (channels) {
case 6:
dmic->ch_enabled |= OMAP_DMIC_UP3_ENABLE;
- /* fall through */
+ fallthrough;
case 4:
dmic->ch_enabled |= OMAP_DMIC_UP2_ENABLE;
- /* fall through */
+ fallthrough;
case 2:
dmic->ch_enabled |= OMAP_DMIC_UP1_ENABLE;
break;
diff --git a/sound/soc/ti/omap-mcpdm.c b/sound/soc/ti/omap-mcpdm.c
index d482b62f314a..fafb2998ad0d 100644
--- a/sound/soc/ti/omap-mcpdm.c
+++ b/sound/soc/ti/omap-mcpdm.c
@@ -309,19 +309,19 @@ static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream,
/* up to 3 channels for capture */
return -EINVAL;
link_mask |= 1 << 4;
- /* fall through */
+ fallthrough;
case 4:
if (stream == SNDRV_PCM_STREAM_CAPTURE)
/* up to 3 channels for capture */
return -EINVAL;
link_mask |= 1 << 3;
- /* fall through */
+ fallthrough;
case 3:
link_mask |= 1 << 2;
- /* fall through */
+ fallthrough;
case 2:
link_mask |= 1 << 1;
- /* fall through */
+ fallthrough;
case 1:
link_mask |= 1 << 0;
break;
diff --git a/sound/soc/ti/rx51.c b/sound/soc/ti/rx51.c
index 2176a95201bf..a2629ccc1dc8 100644
--- a/sound/soc/ti/rx51.c
+++ b/sound/soc/ti/rx51.c
@@ -55,7 +55,7 @@ static void rx51_ext_control(struct snd_soc_dapm_context *dapm)
break;
case RX51_JACK_HS:
hs = 1;
- /* fall through */
+ fallthrough;
case RX51_JACK_HP:
hp = 1;
break;
diff --git a/sound/soc/zte/zx-i2s.c b/sound/soc/zte/zx-i2s.c
index 568cde64ff8b..1c1a44e08a67 100644
--- a/sound/soc/zte/zx-i2s.c
+++ b/sound/soc/zte/zx-i2s.c
@@ -294,7 +294,7 @@ static int zx_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
zx_i2s_rx_dma_en(zx_i2s->reg_base, true);
else
zx_i2s_tx_dma_en(zx_i2s->reg_base, true);
- /* fall thru */
+ fallthrough;
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
if (capture)
@@ -308,7 +308,7 @@ static int zx_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
zx_i2s_rx_dma_en(zx_i2s->reg_base, false);
else
zx_i2s_tx_dma_en(zx_i2s->reg_base, false);
- /* fall thru */
+ fallthrough;
case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
if (capture)
diff --git a/sound/soc/zte/zx-spdif.c b/sound/soc/zte/zx-spdif.c
index a3a07c0730e6..b4168bd532b7 100644
--- a/sound/soc/zte/zx-spdif.c
+++ b/sound/soc/zte/zx-spdif.c
@@ -218,7 +218,7 @@ static int zx_spdif_trigger(struct snd_pcm_substream *substream, int cmd,
val = readl_relaxed(zx_spdif->reg_base + ZX_FIFOCTRL);
val |= ZX_FIFOCTRL_TX_FIFO_RST;
writel_relaxed(val, zx_spdif->reg_base + ZX_FIFOCTRL);
- /* fall thru */
+ fallthrough;
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
zx_spdif_cfg_tx(zx_spdif->reg_base, true);
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index 6b0f3a8469ef..81e987eaf063 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -2371,7 +2371,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid,
int num_ins;
struct usb_mixer_elem_info *cval;
struct snd_kcontrol *kctl;
- int i, err, nameid, type, len;
+ int i, err, nameid, type, len, val;
const struct procunit_info *info;
const struct procunit_value_info *valinfo;
const struct usbmix_name_map *map;
@@ -2474,6 +2474,12 @@ static int build_audio_procunit(struct mixer_build *state, int unitid,
break;
}
+ err = get_cur_ctl_value(cval, cval->control << 8, &val);
+ if (err < 0) {
+ usb_mixer_elem_info_free(cval);
+ return -EINVAL;
+ }
+
kctl = snd_ctl_new1(&mixer_procunit_ctl, cval);
if (!kctl) {
usb_mixer_elem_info_free(cval);
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index d79e3ddc5690..f4fb002e3ef4 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -2680,6 +2680,10 @@ YAMAHA_DEVICE(0x7010, "UB99"),
.data = (const struct snd_usb_audio_quirk[]) {
{
.ifnum = 0,
+ .type = QUIRK_AUDIO_STANDARD_MIXER,
+ },
+ {
+ .ifnum = 0,
.type = QUIRK_AUDIO_FIXED_ENDPOINT,
.data = &(const struct audioformat) {
.formats = SNDRV_PCM_FMTBIT_S24_3LE,
@@ -2690,6 +2694,32 @@ YAMAHA_DEVICE(0x7010, "UB99"),
.attributes = UAC_EP_CS_ATTR_SAMPLE_RATE,
.endpoint = 0x01,
.ep_attr = USB_ENDPOINT_XFER_ISOC,
+ .datainterval = 1,
+ .maxpacksize = 0x024c,
+ .rates = SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000,
+ .rate_min = 44100,
+ .rate_max = 48000,
+ .nr_rates = 2,
+ .rate_table = (unsigned int[]) {
+ 44100, 48000
+ }
+ }
+ },
+ {
+ .ifnum = 0,
+ .type = QUIRK_AUDIO_FIXED_ENDPOINT,
+ .data = &(const struct audioformat) {
+ .formats = SNDRV_PCM_FMTBIT_S24_3LE,
+ .channels = 2,
+ .iface = 0,
+ .altsetting = 1,
+ .altset_idx = 1,
+ .attributes = 0,
+ .endpoint = 0x82,
+ .ep_attr = USB_ENDPOINT_XFER_ISOC,
+ .datainterval = 1,
+ .maxpacksize = 0x0126,
.rates = SNDRV_PCM_RATE_44100 |
SNDRV_PCM_RATE_48000,
.rate_min = 44100,
@@ -3714,8 +3744,8 @@ ALC1220_VB_DESKTOP(0x26ce, 0x0a01), /* Asrock TRX40 Creator */
* they pretend to be 96kHz mono as a workaround for stereo being broken
* by that...
*
- * They also have swapped L-R channels, but that's for userspace to deal
- * with.
+ * They also have an issue with initial stream alignment that causes the
+ * channels to be swapped and out of phase, which is dealt with in quirks.c.
*/
{
.match_flags = USB_DEVICE_ID_MATCH_DEVICE |