[alsa-devel] [PATCH v2 1/2] ASoC: soc-pcm: Use valid condition for snd_soc_dai_digital_mute() in hw_free()

[Nicolin Chen]

The snd_soc_dai_digital_mute() here will be never executed because we only
decrease codec->active in snd_soc_close(). Thus correct it.

Signed-off-by: Nicolin Chen <b42378 at freescale.com>
---
 sound/soc/soc-pcm.c | 5 +++--
 1 file changed, 3 insertions(+), 2 deletions(-)

diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index eb340a8..3774471 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -689,7 +689,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
 	struct snd_soc_platform *platform = rtd->platform;
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_codec *codec = rtd->codec;
+	bool playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
 
 	mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
 
@@ -707,7 +707,8 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
 	}
 
 	/* apply codec digital mute */
-	if (!codec->active)
+	if ((playback && codec_dai->playback_active == 1) ||
+	    (!playback && codec_dai->capture_active == 1))
 		snd_soc_dai_digital_mute(codec_dai, 1, substream->stream);
 
 	/* free any machine hw params */
-- 
1.8.4