ASoC: core: Allow digital mute for capture
Help avoid noise from the power up of the capture path propagating through
into the start of the recording (especially noise caused by the ramp of
microphone biases) by keeping the capture muted until after we've finished
powering things up with DAPM in the same manner we do for playback. This
allows us to take advantage of soft mute support in the hardware more
effectively and is more consistent.
The core code using the existing digital mute operation is updated to take
advantage of this. Some additional cases in the soc-pcm code and suspend
will need separate handling but these are less practically relevant than
the main runtime stream start/stop case.
Rather than refactor the digital mute function in every single driver a
new operation is added for drivers taking advantage of this functionality,
the old operation should be phased out over time.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by Vinod Koul <vinod.koul@intel.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 3953cea..a680f23 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -126,7 +126,8 @@
int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate);
/* Digital Audio Interface mute */
-int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute);
+int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute,
+ int direction);
struct snd_soc_dai_ops {
/*
@@ -157,6 +158,7 @@
* Called by soc-core to minimise any pops.
*/
int (*digital_mute)(struct snd_soc_dai *dai, int mute);
+ int (*mute_stream)(struct snd_soc_dai *dai, int mute, int stream);
/*
* ALSA PCM audio operations - all optional.