[alsa-devel] [PATCH] 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 | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 22af038..7d19117 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)
 	}
 
 	/* apply codec digital mute */
-	if (!codec->active)
+	if (codec->active == 1)
 		snd_soc_dai_digital_mute(codec_dai, 1, substream->stream);
 
 	/* free any machine hw params */
-- 
1.8.4