From: Daniel Mack <zonque@gmail.com>
To: alsa-devel@alsa-project.org
Cc: broonie@opensource.wolfsonmicro.com, subaparts@yandex.ru,
lgirdwood@gmail.com, Daniel Mack <zonque@gmail.com>
Subject: [PATCH 1/2] ASoC: cs4271: switch to mute_stream
Date: Fri, 15 Mar 2013 06:35:47 +0100 [thread overview]
Message-ID: <1363325748-6429-1-git-send-email-zonque@gmail.com> (raw)
Use the newly introduced mute_stream DAI operation, and don't mute the
codec if it's called for the _CAPTURE stream.
Signed-off-by: Daniel Mack <zonque@gmail.com>
---
sound/soc/codecs/cs4271.c | 7 +++++--
1 file changed, 5 insertions(+), 2 deletions(-)
diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c
index ac0d3b4..03036b3 100644
--- a/sound/soc/codecs/cs4271.c
+++ b/sound/soc/codecs/cs4271.c
@@ -388,7 +388,7 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream,
return cs4271_set_deemph(codec);
}
-static int cs4271_digital_mute(struct snd_soc_dai *dai, int mute)
+static int cs4271_mute_stream(struct snd_soc_dai *dai, int mute, int stream)
{
struct snd_soc_codec *codec = dai->codec;
struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec);
@@ -396,6 +396,9 @@ static int cs4271_digital_mute(struct snd_soc_dai *dai, int mute)
int val_a = 0;
int val_b = 0;
+ if (stream != SNDRV_PCM_STREAM_PLAYBACK)
+ return 0;
+
if (mute) {
val_a = CS4271_VOLA_MUTE;
val_b = CS4271_VOLB_MUTE;
@@ -442,7 +445,7 @@ static const struct snd_soc_dai_ops cs4271_dai_ops = {
.hw_params = cs4271_hw_params,
.set_sysclk = cs4271_set_dai_sysclk,
.set_fmt = cs4271_set_dai_fmt,
- .digital_mute = cs4271_digital_mute,
+ .mute_stream = cs4271_mute_stream,
};
static struct snd_soc_dai_driver cs4271_dai = {
--
1.8.1.4
next reply other threads:[~2013-03-15 5:36 UTC|newest]
Thread overview: 2+ messages / expand[flat|nested] mbox.gz Atom feed top
2013-03-15 5:35 Daniel Mack [this message]
2013-03-15 5:35 ` [PATCH 2/2] ASoC: cs4271: preserve "Master Playback Switch" setting Daniel Mack
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