[PATCH 06/19] ASoC: codecs: merge .digital_mute() into .mute_stream()
Kuninori Morimoto
kuninori.morimoto.gx at renesas.com
Tue Jun 23 03:20:02 CEST 2020
From: Kuninori Morimoto <kuninori.morimoto.gx at renesas.com>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx at renesas.com>
---
sound/soc/codecs/88pm860x-codec.c | 9 +++++---
sound/soc/codecs/ad193x.c | 7 +++++--
sound/soc/codecs/adau1701.c | 7 +++++--
sound/soc/codecs/cpcap.c | 15 +++++++++----
sound/soc/codecs/cq93vc.c | 7 +++++--
sound/soc/codecs/isabelle.c | 21 +++++++++++++------
sound/soc/codecs/jz4770.c | 7 +++++--
sound/soc/codecs/lm49453.c | 35 ++++++++++++++++++++++---------
sound/soc/codecs/ml26124.c | 7 +++++--
sound/soc/codecs/nau8822.c | 7 +++++--
sound/soc/codecs/rk3328_codec.c | 7 +++++--
sound/soc/codecs/sgtl5000.c | 7 +++++--
sound/soc/codecs/sta529.c | 7 +++++--
sound/soc/codecs/tfa9879.c | 7 +++++--
sound/soc/codecs/twl6040.c | 7 +++++--
sound/soc/codecs/uda134x.c | 7 +++++--
16 files changed, 117 insertions(+), 47 deletions(-)
diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c
index 068914d0ef3d..c668029258b0 100644
--- a/sound/soc/codecs/88pm860x-codec.c
+++ b/sound/soc/codecs/88pm860x-codec.c
@@ -902,11 +902,14 @@ static const struct snd_soc_dapm_route pm860x_dapm_routes[] = {
* Use MUTE_LEFT & MUTE_RIGHT to implement digital mute.
* These bits can also be used to mute.
*/
-static int pm860x_digital_mute(struct snd_soc_dai *codec_dai, int mute)
+static int pm860x_mute(struct snd_soc_dai *codec_dai, int mute, int direction)
{
struct snd_soc_component *component = codec_dai->component;
int data = 0, mask = MUTE_LEFT | MUTE_RIGHT;
+ if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+ return 0;
+
if (mute)
data = mask;
snd_soc_component_update_bits(component, PM860X_DAC_OFFSET, mask, data);
@@ -1136,14 +1139,14 @@ static int pm860x_set_bias_level(struct snd_soc_component *component,
}
static const struct snd_soc_dai_ops pm860x_pcm_dai_ops = {
- .digital_mute = pm860x_digital_mute,
+ .mute_stream = pm860x_mute,
.hw_params = pm860x_pcm_hw_params,
.set_fmt = pm860x_pcm_set_dai_fmt,
.set_sysclk = pm860x_set_dai_sysclk,
};
static const struct snd_soc_dai_ops pm860x_i2s_dai_ops = {
- .digital_mute = pm860x_digital_mute,
+ .mute_stream = pm860x_mute,
.hw_params = pm860x_i2s_hw_params,
.set_fmt = pm860x_i2s_set_dai_fmt,
.set_sysclk = pm860x_set_dai_sysclk,
diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c
index 980e024a5720..5af815f1028d 100644
--- a/sound/soc/codecs/ad193x.c
+++ b/sound/soc/codecs/ad193x.c
@@ -143,10 +143,13 @@ static inline bool ad193x_has_adc(const struct ad193x_priv *ad193x)
* DAI ops entries
*/
-static int ad193x_mute(struct snd_soc_dai *dai, int mute)
+static int ad193x_mute(struct snd_soc_dai *dai, int mute, int direction)
{
struct ad193x_priv *ad193x = snd_soc_component_get_drvdata(dai->component);
+ if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+ return 0;
+
if (mute)
regmap_update_bits(ad193x->regmap, AD193X_DAC_CTRL2,
AD193X_DAC_MASTER_MUTE,
@@ -371,7 +374,7 @@ static int ad193x_startup(struct snd_pcm_substream *substream,
static const struct snd_soc_dai_ops ad193x_dai_ops = {
.startup = ad193x_startup,
.hw_params = ad193x_hw_params,
- .digital_mute = ad193x_mute,
+ .mute_stream = ad193x_mute,
.set_tdm_slot = ad193x_set_tdm_slot,
.set_sysclk = ad193x_set_dai_sysclk,
.set_fmt = ad193x_set_dai_fmt,
diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c
index 115e296b2ad6..2aa41f91386f 100644
--- a/sound/soc/codecs/adau1701.c
+++ b/sound/soc/codecs/adau1701.c
@@ -573,13 +573,16 @@ static int adau1701_set_bias_level(struct snd_soc_component *component,
return 0;
}
-static int adau1701_digital_mute(struct snd_soc_dai *dai, int mute)
+static int adau1701_mute(struct snd_soc_dai *dai, int mute, int direction)
{
struct snd_soc_component *component = dai->component;
unsigned int mask = ADAU1701_DSPCTRL_DAM;
struct adau1701 *adau1701 = snd_soc_component_get_drvdata(component);
unsigned int val;
+ if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+ return 0;
+
if (mute)
val = 0;
else
@@ -631,7 +634,7 @@ static int adau1701_startup(struct snd_pcm_substream *substream,
static const struct snd_soc_dai_ops adau1701_dai_ops = {
.set_fmt = adau1701_set_dai_fmt,
.hw_params = adau1701_hw_params,
- .digital_mute = adau1701_digital_mute,
+ .mute_stream = adau1701_mute,
.startup = adau1701_startup,
};
diff --git a/sound/soc/codecs/cpcap.c b/sound/soc/codecs/cpcap.c
index d7f05b384f1f..24a1d1196d3a 100644
--- a/sound/soc/codecs/cpcap.c
+++ b/sound/soc/codecs/cpcap.c
@@ -1216,7 +1216,7 @@ static int cpcap_hifi_set_dai_fmt(struct snd_soc_dai *codec_dai,
return regmap_update_bits(cpcap->regmap, reg, mask, val);
}
-static int cpcap_hifi_set_mute(struct snd_soc_dai *dai, int mute)
+static int cpcap_hifi_set_mute(struct snd_soc_dai *dai, int mute, int direction)
{
struct snd_soc_component *component = dai->component;
struct cpcap_audio *cpcap = snd_soc_component_get_drvdata(component);
@@ -1224,6 +1224,9 @@ static int cpcap_hifi_set_mute(struct snd_soc_dai *dai, int mute)
static const u16 mask = BIT(CPCAP_BIT_ST_DAC_SW);
u16 val;
+ if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+ return 0;
+
if (mute)
val = 0;
else
@@ -1237,7 +1240,7 @@ static const struct snd_soc_dai_ops cpcap_dai_hifi_ops = {
.hw_params = cpcap_hifi_hw_params,
.set_sysclk = cpcap_hifi_set_dai_sysclk,
.set_fmt = cpcap_hifi_set_dai_fmt,
- .digital_mute = cpcap_hifi_set_mute,
+ .mute_stream = cpcap_hifi_set_mute,
};
static int cpcap_voice_hw_params(struct snd_pcm_substream *substream,
@@ -1370,7 +1373,8 @@ static int cpcap_voice_set_dai_fmt(struct snd_soc_dai *codec_dai,
return 0;
}
-static int cpcap_voice_set_mute(struct snd_soc_dai *dai, int mute)
+static int cpcap_voice_set_mute(struct snd_soc_dai *dai,
+ int mute, int direction)
{
struct snd_soc_component *component = dai->component;
struct cpcap_audio *cpcap = snd_soc_component_get_drvdata(component);
@@ -1378,6 +1382,9 @@ static int cpcap_voice_set_mute(struct snd_soc_dai *dai, int mute)
static const u16 mask = BIT(CPCAP_BIT_CDC_SW);
u16 val;
+ if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+ return 0;
+
if (mute)
val = 0;
else
@@ -1391,7 +1398,7 @@ static const struct snd_soc_dai_ops cpcap_dai_voice_ops = {
.hw_params = cpcap_voice_hw_params,
.set_sysclk = cpcap_voice_set_dai_sysclk,
.set_fmt = cpcap_voice_set_dai_fmt,
- .digital_mute = cpcap_voice_set_mute,
+ .mute_stream = cpcap_voice_set_mute,
};
static struct snd_soc_dai_driver cpcap_dai[] = {
diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c
index b0cc61178a41..4a41cc551bdd 100644
--- a/sound/soc/codecs/cq93vc.c
+++ b/sound/soc/codecs/cq93vc.c
@@ -30,11 +30,14 @@ static const struct snd_kcontrol_new cq93vc_snd_controls[] = {
SOC_SINGLE("Mono DAC Playback Volume", DAVINCI_VC_REG09, 0, 0x3f, 0),
};
-static int cq93vc_mute(struct snd_soc_dai *dai, int mute)
+static int cq93vc_mute(struct snd_soc_dai *dai, int mute, int direction)
{
struct snd_soc_component *component = dai->component;
u8 reg;
+ if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+ return 0;
+
if (mute)
reg = DAVINCI_VC_REG09_MUTE;
else
@@ -87,7 +90,7 @@ static int cq93vc_set_bias_level(struct snd_soc_component *component,
#define CQ93VC_FORMATS (SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE)
static const struct snd_soc_dai_ops cq93vc_dai_ops = {
- .digital_mute = cq93vc_mute,
+ .mute_stream = cq93vc_mute,
.set_sysclk = cq93vc_set_dai_sysclk,
};
diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c
index 3626f70f7768..204b89d6aa4a 100644
--- a/sound/soc/codecs/isabelle.c
+++ b/sound/soc/codecs/isabelle.c
@@ -860,24 +860,33 @@ static const struct snd_soc_dapm_route isabelle_intercon[] = {
{ "LINEOUT2", NULL, "LINEOUT2 Driver" },
};
-static int isabelle_hs_mute(struct snd_soc_dai *dai, int mute)
+static int isabelle_hs_mute(struct snd_soc_dai *dai, int mute, int direction)
{
+ if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+ return 0;
+
snd_soc_component_update_bits(dai->component, ISABELLE_DAC1_SOFTRAMP_REG,
BIT(4), (mute ? BIT(4) : 0));
return 0;
}
-static int isabelle_hf_mute(struct snd_soc_dai *dai, int mute)
+static int isabelle_hf_mute(struct snd_soc_dai *dai, int mute, int direction)
{
+ if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+ return 0;
+
snd_soc_component_update_bits(dai->component, ISABELLE_DAC2_SOFTRAMP_REG,
BIT(4), (mute ? BIT(4) : 0));
return 0;
}
-static int isabelle_line_mute(struct snd_soc_dai *dai, int mute)
+static int isabelle_line_mute(struct snd_soc_dai *dai, int mute, int direction)
{
+ if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+ return 0;
+
snd_soc_component_update_bits(dai->component, ISABELLE_DAC3_SOFTRAMP_REG,
BIT(4), (mute ? BIT(4) : 0));
@@ -1014,19 +1023,19 @@ static int isabelle_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
static const struct snd_soc_dai_ops isabelle_hs_dai_ops = {
.hw_params = isabelle_hw_params,
.set_fmt = isabelle_set_dai_fmt,
- .digital_mute = isabelle_hs_mute,
+ .mute_stream = isabelle_hs_mute,
};
static const struct snd_soc_dai_ops isabelle_hf_dai_ops = {
.hw_params = isabelle_hw_params,
.set_fmt = isabelle_set_dai_fmt,
- .digital_mute = isabelle_hf_mute,
+ .mute_stream = isabelle_hf_mute,
};
static const struct snd_soc_dai_ops isabelle_line_dai_ops = {
.hw_params = isabelle_hw_params,
.set_fmt = isabelle_set_dai_fmt,
- .digital_mute = isabelle_line_mute,
+ .mute_stream = isabelle_line_mute,
};
static const struct snd_soc_dai_ops isabelle_ul_dai_ops = {
diff --git a/sound/soc/codecs/jz4770.c b/sound/soc/codecs/jz4770.c
index 34775aa62402..c9e2ac155421 100644
--- a/sound/soc/codecs/jz4770.c
+++ b/sound/soc/codecs/jz4770.c
@@ -264,7 +264,7 @@ static int jz4770_codec_pcm_trigger(struct snd_pcm_substream *substream,
return ret;
}
-static int jz4770_codec_digital_mute(struct snd_soc_dai *dai, int mute)
+static int jz4770_codec_mute(struct snd_soc_dai *dai, int mute, int direction)
{
struct snd_soc_component *codec = dai->component;
struct jz_codec *jz_codec = snd_soc_component_get_drvdata(codec);
@@ -272,6 +272,9 @@ static int jz4770_codec_digital_mute(struct snd_soc_dai *dai, int mute)
unsigned int val;
int change, err;
+ if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+ return 0;
+
change = snd_soc_component_update_bits(codec, JZ4770_CODEC_REG_CR_DAC,
REG_CR_DAC_MUTE,
mute ? REG_CR_DAC_MUTE : 0);
@@ -753,7 +756,7 @@ static const struct snd_soc_dai_ops jz4770_codec_dai_ops = {
.shutdown = jz4770_codec_shutdown,
.hw_params = jz4770_codec_hw_params,
.trigger = jz4770_codec_pcm_trigger,
- .digital_mute = jz4770_codec_digital_mute,
+ .mute_stream = jz4770_codec_mute,
};
#define JZ_CODEC_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \
diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c
index f864b07cb0b8..32a384d0ec18 100644
--- a/sound/soc/codecs/lm49453.c
+++ b/sound/soc/codecs/lm49453.c
@@ -1218,36 +1218,51 @@ static int lm49453_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id,
return 0;
}
-static int lm49453_hp_mute(struct snd_soc_dai *dai, int mute)
+static int lm49453_hp_mute(struct snd_soc_dai *dai, int mute, int direction)
{
+ if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+ return 0;
+
snd_soc_component_update_bits(dai->component, LM49453_P0_DAC_DSP_REG, BIT(1)|BIT(0),
(mute ? (BIT(1)|BIT(0)) : 0));
return 0;
}
-static int lm49453_lo_mute(struct snd_soc_dai *dai, int mute)
+static int lm49453_lo_mute(struct snd_soc_dai *dai, int mute, int direction)
{
+ if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+ return 0;
+
snd_soc_component_update_bits(dai->component, LM49453_P0_DAC_DSP_REG, BIT(3)|BIT(2),
(mute ? (BIT(3)|BIT(2)) : 0));
return 0;
}
-static int lm49453_ls_mute(struct snd_soc_dai *dai, int mute)
+static int lm49453_ls_mute(struct snd_soc_dai *dai, int mute, int direction)
{
+ if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+ return 0;
+
snd_soc_component_update_bits(dai->component, LM49453_P0_DAC_DSP_REG, BIT(5)|BIT(4),
(mute ? (BIT(5)|BIT(4)) : 0));
return 0;
}
-static int lm49453_ep_mute(struct snd_soc_dai *dai, int mute)
+static int lm49453_ep_mute(struct snd_soc_dai *dai, int mute, int direction)
{
+ if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+ return 0;
+
snd_soc_component_update_bits(dai->component, LM49453_P0_DAC_DSP_REG, BIT(4),
(mute ? BIT(4) : 0));
return 0;
}
-static int lm49453_ha_mute(struct snd_soc_dai *dai, int mute)
+static int lm49453_ha_mute(struct snd_soc_dai *dai, int mute, int direction)
{
+ if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+ return 0;
+
snd_soc_component_update_bits(dai->component, LM49453_P0_DAC_DSP_REG, BIT(7)|BIT(6),
(mute ? (BIT(7)|BIT(6)) : 0));
return 0;
@@ -1288,35 +1303,35 @@ static const struct snd_soc_dai_ops lm49453_headset_dai_ops = {
.hw_params = lm49453_hw_params,
.set_sysclk = lm49453_set_dai_sysclk,
.set_fmt = lm49453_set_dai_fmt,
- .digital_mute = lm49453_hp_mute,
+ .mute_stream = lm49453_hp_mute,
};
static const struct snd_soc_dai_ops lm49453_speaker_dai_ops = {
.hw_params = lm49453_hw_params,
.set_sysclk = lm49453_set_dai_sysclk,
.set_fmt = lm49453_set_dai_fmt,
- .digital_mute = lm49453_ls_mute,
+ .mute_stream = lm49453_ls_mute,
};
static const struct snd_soc_dai_ops lm49453_haptic_dai_ops = {
.hw_params = lm49453_hw_params,
.set_sysclk = lm49453_set_dai_sysclk,
.set_fmt = lm49453_set_dai_fmt,
- .digital_mute = lm49453_ha_mute,
+ .mute_stream = lm49453_ha_mute,
};
static const struct snd_soc_dai_ops lm49453_ep_dai_ops = {
.hw_params = lm49453_hw_params,
.set_sysclk = lm49453_set_dai_sysclk,
.set_fmt = lm49453_set_dai_fmt,
- .digital_mute = lm49453_ep_mute,
+ .mute_stream = lm49453_ep_mute,
};
static const struct snd_soc_dai_ops lm49453_lineout_dai_ops = {
.hw_params = lm49453_hw_params,
.set_sysclk = lm49453_set_dai_sysclk,
.set_fmt = lm49453_set_dai_fmt,
- .digital_mute = lm49453_lo_mute,
+ .mute_stream = lm49453_lo_mute,
};
/* LM49453 dai structure. */
diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c
index 55823bc95d06..cbfd3919e31c 100644
--- a/sound/soc/codecs/ml26124.c
+++ b/sound/soc/codecs/ml26124.c
@@ -372,11 +372,14 @@ static int ml26124_hw_params(struct snd_pcm_substream *substream,
return 0;
}
-static int ml26124_mute(struct snd_soc_dai *dai, int mute)
+static int ml26124_mute(struct snd_soc_dai *dai, int mute, int direction)
{
struct snd_soc_component *component = dai->component;
struct ml26124_priv *priv = snd_soc_component_get_drvdata(component);
+ if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+ return 0;
+
switch (priv->substream->stream) {
case SNDRV_PCM_STREAM_CAPTURE:
snd_soc_component_update_bits(component, ML26124_REC_PLYBAK_RUN, BIT(0), 1);
@@ -492,7 +495,7 @@ static int ml26124_set_bias_level(struct snd_soc_component *component,
static const struct snd_soc_dai_ops ml26124_dai_ops = {
.hw_params = ml26124_hw_params,
- .digital_mute = ml26124_mute,
+ .mute_stream = ml26124_mute,
.set_fmt = ml26124_set_dai_fmt,
.set_sysclk = ml26124_set_dai_sysclk,
};
diff --git a/sound/soc/codecs/nau8822.c b/sound/soc/codecs/nau8822.c
index 79928ddeb7a1..9e7937f889cd 100644
--- a/sound/soc/codecs/nau8822.c
+++ b/sound/soc/codecs/nau8822.c
@@ -900,10 +900,13 @@ static int nau8822_hw_params(struct snd_pcm_substream *substream,
return 0;
}
-static int nau8822_mute(struct snd_soc_dai *dai, int mute)
+static int nau8822_mute(struct snd_soc_dai *dai, int mute, int direction)
{
struct snd_soc_component *component = dai->component;
+ if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+ return 0;
+
dev_dbg(component->dev, "%s: %d\n", __func__, mute);
if (mute)
@@ -967,7 +970,7 @@ static int nau8822_set_bias_level(struct snd_soc_component *component,
static const struct snd_soc_dai_ops nau8822_dai_ops = {
.hw_params = nau8822_hw_params,
- .digital_mute = nau8822_mute,
+ .mute_stream = nau8822_mute,
.set_fmt = nau8822_set_dai_fmt,
.set_sysclk = nau8822_set_dai_sysclk,
.set_pll = nau8822_set_pll,
diff --git a/sound/soc/codecs/rk3328_codec.c b/sound/soc/codecs/rk3328_codec.c
index 115706a55577..7f235d0da3f5 100644
--- a/sound/soc/codecs/rk3328_codec.c
+++ b/sound/soc/codecs/rk3328_codec.c
@@ -107,12 +107,15 @@ static int rk3328_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
return 0;
}
-static int rk3328_digital_mute(struct snd_soc_dai *dai, int mute)
+static int rk3328_mute(struct snd_soc_dai *dai, int mute, int direction)
{
struct rk3328_codec_priv *rk3328 =
snd_soc_component_get_drvdata(dai->component);
unsigned int val;
+ if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+ return 0;
+
if (mute)
val = HPOUTL_MUTE | HPOUTR_MUTE;
else
@@ -316,7 +319,7 @@ static void rk3328_pcm_shutdown(struct snd_pcm_substream *substream,
static const struct snd_soc_dai_ops rk3328_dai_ops = {
.hw_params = rk3328_hw_params,
.set_fmt = rk3328_set_dai_fmt,
- .digital_mute = rk3328_digital_mute,
+ .mute_stream = rk3328_mute,
.startup = rk3328_pcm_startup,
.shutdown = rk3328_pcm_shutdown,
};
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index eb08976a7d06..9a87c08caca3 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -775,11 +775,14 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = {
};
/* mute the codec used by alsa core */
-static int sgtl5000_digital_mute(struct snd_soc_dai *codec_dai, int mute)
+static int sgtl5000_mute(struct snd_soc_dai *codec_dai, int mute, int direction)
{
struct snd_soc_component *component = codec_dai->component;
u16 i2s_pwr = SGTL5000_I2S_IN_POWERUP;
+ if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+ return 0;
+
/*
* During 'digital mute' do not mute DAC
* because LINE_IN would be muted aswell. We want to mute
@@ -1160,7 +1163,7 @@ static int sgtl5000_set_bias_level(struct snd_soc_component *component,
static const struct snd_soc_dai_ops sgtl5000_ops = {
.hw_params = sgtl5000_pcm_hw_params,
- .digital_mute = sgtl5000_digital_mute,
+ .mute_stream = sgtl5000_mute,
.set_fmt = sgtl5000_set_dai_fmt,
.set_sysclk = sgtl5000_set_dai_sysclk,
};
diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c
index 2881a0f7bb39..3bcb55b240df 100644
--- a/sound/soc/codecs/sta529.c
+++ b/sound/soc/codecs/sta529.c
@@ -251,10 +251,13 @@ static int sta529_hw_params(struct snd_pcm_substream *substream,
return 0;
}
-static int sta529_mute(struct snd_soc_dai *dai, int mute)
+static int sta529_mute(struct snd_soc_dai *dai, int mute, int direction)
{
u8 val = 0;
+ if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+ return 0;
+
if (mute)
val |= CODEC_MUTE_VAL;
@@ -291,7 +294,7 @@ static int sta529_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt)
static const struct snd_soc_dai_ops sta529_dai_ops = {
.hw_params = sta529_hw_params,
.set_fmt = sta529_set_dai_fmt,
- .digital_mute = sta529_mute,
+ .mute_stream = sta529_mute,
};
static struct snd_soc_dai_driver sta529_dai = {
diff --git a/sound/soc/codecs/tfa9879.c b/sound/soc/codecs/tfa9879.c
index abc114a3ae2b..d59a2b1e380c 100644
--- a/sound/soc/codecs/tfa9879.c
+++ b/sound/soc/codecs/tfa9879.c
@@ -93,10 +93,13 @@ static int tfa9879_hw_params(struct snd_pcm_substream *substream,
return 0;
}
-static int tfa9879_digital_mute(struct snd_soc_dai *dai, int mute)
+static int tfa9879_mute(struct snd_soc_dai *dai, int mute, int direction)
{
struct snd_soc_component *component = dai->component;
+ if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+ return 0;
+
snd_soc_component_update_bits(component, TFA9879_MISC_CONTROL,
TFA9879_S_MUTE_MASK,
!!mute << TFA9879_S_MUTE_SHIFT);
@@ -251,7 +254,7 @@ static const struct regmap_config tfa9879_regmap = {
static const struct snd_soc_dai_ops tfa9879_dai_ops = {
.hw_params = tfa9879_hw_params,
- .digital_mute = tfa9879_digital_mute,
+ .mute_stream = tfa9879_mute,
.set_fmt = tfa9879_set_fmt,
};
diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
index f34637afee51..9fead4faa739 100644
--- a/sound/soc/codecs/twl6040.c
+++ b/sound/soc/codecs/twl6040.c
@@ -997,8 +997,11 @@ static void twl6040_mute_path(struct snd_soc_component *component, enum twl6040_
}
}
-static int twl6040_digital_mute(struct snd_soc_dai *dai, int mute)
+static int twl6040_mute(struct snd_soc_dai *dai, int mute, int direction)
{
+ if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+ return 0;
+
switch (dai->id) {
case TWL6040_DAI_LEGACY:
twl6040_mute_path(dai->component, TWL6040_DAI_DL1, mute);
@@ -1020,7 +1023,7 @@ static const struct snd_soc_dai_ops twl6040_dai_ops = {
.hw_params = twl6040_hw_params,
.prepare = twl6040_prepare,
.set_sysclk = twl6040_set_dai_sysclk,
- .digital_mute = twl6040_digital_mute,
+ .mute_stream = twl6040_mute,
};
static struct snd_soc_dai_driver twl6040_dai[] = {
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index 1cc7f56912dc..a58291581813 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -117,12 +117,15 @@ static inline void uda134x_reset(struct snd_soc_component *component)
regmap_update_bits(uda134x->regmap, UDA134X_STATUS0, mask, 0);
}
-static int uda134x_mute(struct snd_soc_dai *dai, int mute)
+static int uda134x_mute(struct snd_soc_dai *dai, int mute, int direction)
{
struct uda134x_priv *uda134x = snd_soc_component_get_drvdata(dai->component);
unsigned int mask = 1<<2;
unsigned int val;
+ if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+ return 0;
+
pr_debug("%s mute: %d\n", __func__, mute);
if (mute)
@@ -416,7 +419,7 @@ static const struct snd_soc_dai_ops uda134x_dai_ops = {
.startup = uda134x_startup,
.shutdown = uda134x_shutdown,
.hw_params = uda134x_hw_params,
- .digital_mute = uda134x_mute,
+ .mute_stream = uda134x_mute,
.set_sysclk = uda134x_set_dai_sysclk,
.set_fmt = uda134x_set_dai_fmt,
};
--
2.25.1
More information about the Alsa-devel
mailing list