[alsa-devel] [PATCH v2] ASoC: Handle multiple codecs with split playback / capture

[Ricard Wanderlof]

On Thu, 20 Aug 2015, Pierre-Louis Bossart wrote:

> >   /**
> >    * snd_soc_runtime_activate() - Increment active count for PCM runtime components
> >    * @rtd: ASoC PCM runtime that is activated
> > @@ -371,6 +389,20 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream)
> >
> >   	/* first calculate min/max only for CODECs in the DAI link */
> >   	for (i = 0; i < rtd->num_codecs; i++) {
> > +
> > +		/*
> > +		 * Skip CODECs which don't support the current stream type.
> > +		 * Otherwise, since the rate, channel, and format values will
> > +		 * zero in that case, we would have no usable settings left,
> > +		 * causing the resulting setup to fail.
> > +		 * At least one CODEC should match, otherwise we should have
> > +		 * bailed out on a higher level, since there would be no
> > +		 * CODEC to support the transfer direction in that case.
> > +		 */
> > +		if (!snd_soc_dai_stream_valid(rtd->codec_dais[i],
> > +					      substream->stream))
> 
> Maybe I misunderstood but shouldn't there be some sort of verification 
> that the codecs can use the same number of slots between playback and 
> capture if they share the same LRCLK/FS? e.g. it's not uncommon to have 
> 4 mic capture and 2 ch playback. If the capture and playback is handled 
> by different chips you'd still need to maintain some level of consistency.

You're probably right, although it may be that that sort of thing is 
handled at a higher level, e.g. the machine driver in such a setup would 
configure both codecs to use TDM before we even get to this function, I 
don't know.

I think we'd have to take care of that situation if and when it arises. 
The setup I have here doesn't allow me to test it.

/Ricard
-- 
Ricard Wolf Wanderlöf                           ricardw(at)axis.com
Axis Communications AB, Lund, Sweden            www.axis.com
Phone +46 46 272 2016                           Fax +46 46 13 61 30