[PATCH 15/19] ASoC: codecs: da*: merge .digital_mute() into .mute_stream()
Kuninori Morimoto
kuninori.morimoto.gx at renesas.com
Tue Jun 23 03:20:54 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/da7210.c | 7 +++++--
sound/soc/codecs/da7213.c | 7 +++++--
sound/soc/codecs/da9055.c | 7 +++++--
3 files changed, 15 insertions(+), 6 deletions(-)
diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c
index 0c99dcf242e4..8e5e5cd4bcbe 100644
--- a/sound/soc/codecs/da7210.c
+++ b/sound/soc/codecs/da7210.c
@@ -924,11 +924,14 @@ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt)
return 0;
}
-static int da7210_mute(struct snd_soc_dai *dai, int mute)
+static int da7210_mute(struct snd_soc_dai *dai, int mute, int direction)
{
struct snd_soc_component *component = dai->component;
u8 mute_reg = snd_soc_component_read(component, DA7210_DAC_HPF) & 0xFB;
+ if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+ return 0;
+
if (mute)
snd_soc_component_write(component, DA7210_DAC_HPF, mute_reg | 0x4);
else
@@ -1034,7 +1037,7 @@ static const struct snd_soc_dai_ops da7210_dai_ops = {
.set_fmt = da7210_set_dai_fmt,
.set_sysclk = da7210_set_dai_sysclk,
.set_pll = da7210_set_dai_pll,
- .digital_mute = da7210_mute,
+ .mute_stream = da7210_mute,
};
static struct snd_soc_dai_driver da7210_dai = {
diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c
index cc4ae7b311b4..55f8097112e1 100644
--- a/sound/soc/codecs/da7213.c
+++ b/sound/soc/codecs/da7213.c
@@ -1321,10 +1321,13 @@ static int da7213_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
return 0;
}
-static int da7213_mute(struct snd_soc_dai *dai, int mute)
+static int da7213_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;
+
if (mute) {
snd_soc_component_update_bits(component, DA7213_DAC_L_CTRL,
DA7213_MUTE_EN, DA7213_MUTE_EN);
@@ -1507,7 +1510,7 @@ static int da7213_set_component_pll(struct snd_soc_component *component,
static const struct snd_soc_dai_ops da7213_dai_ops = {
.hw_params = da7213_hw_params,
.set_fmt = da7213_set_dai_fmt,
- .digital_mute = da7213_mute,
+ .mute_stream = da7213_mute,
};
static struct snd_soc_dai_driver da7213_dai = {
diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c
index e93436ccb674..e388b1c0ba19 100644
--- a/sound/soc/codecs/da9055.c
+++ b/sound/soc/codecs/da9055.c
@@ -1211,10 +1211,13 @@ static int da9055_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
return 0;
}
-static int da9055_mute(struct snd_soc_dai *dai, int mute)
+static int da9055_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;
+
if (mute) {
snd_soc_component_update_bits(component, DA9055_DAC_L_CTRL,
DA9055_DAC_L_MUTE_EN, DA9055_DAC_L_MUTE_EN);
@@ -1324,7 +1327,7 @@ static const struct snd_soc_dai_ops da9055_dai_ops = {
.set_fmt = da9055_set_dai_fmt,
.set_sysclk = da9055_set_dai_sysclk,
.set_pll = da9055_set_dai_pll,
- .digital_mute = da9055_mute,
+ .mute_stream = da9055_mute,
};
static struct snd_soc_dai_driver da9055_dai = {
--
2.25.1
More information about the Alsa-devel
mailing list