[alsa-devel] [PATCH 1/2 resend] ASoC: cs4271: switch to mute_stream

[Daniel Mack]

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 at gmail.com>
Acked-by: Alexander Sverdlin <subaparts at yandex.ru>
---
 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