[alsa-devel] [PATCH] ASoC: soc-pcm: Use valid condition for snd_soc_dai_digital_mute() in hw_free()
b42378 at freescale.com
Mon Dec 2 16:25:18 CET 2013
On Mon, Dec 02, 2013 at 12:19:16PM +0000, Mark Brown wrote:
> On Mon, Dec 02, 2013 at 11:05:31AM +0100, Takashi Iwai wrote:
> > Nicolin Chen wrote:
> > > The snd_soc_dai_digital_mute() here will be never executed because we only
> > > decrease codec->active in snd_soc_close(). Thus correct it.
> > A couple of minor problems by this approach:
> > - snd_soc_dai_digital_mute() will be called twice in the normal PCM
> > close path.
> Right, it's called in close() so it's redundant to call it here most of
> the time. On the other hand it will at least get called slightly sooner
> which is good and in the case of reconfiguration we'll mute while doing
> that - doing the mute earlier is better.
> > - In full duplex mode where both playback/capture streams share the
> > same codec, one of the directions won't be handled correctly because
> > codec->active is incremented/decremented twice (once for each
> > direction).
> Indeed, the check needs to be on playback_active or capture_active
> rather than on the overall DAI.
Sir, thank you for the hint. I will reconsider about the modification.
And also thank Iwai-san for pointing out the flaws in my patch.
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