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

[Pierre-Louis Bossart]

On 8/19/15 10:32 AM, Ricard Wanderlof wrote:
>
> Add the capability to use multiple codecs on the same DAI linke where
> one codec is used for playback and another one is used for capture.
>
> Tested on a setup using an SSM2518 for playback and an ICS43432 I2S MEMS
> microphone for capture.
>
> Signed-off-by: Ricard Wanderlof <ricardw at axis.com>
> ---
>
> The patch was created after a discussion on the alsa-devel mailing list
> as to how best to implement this functionality.
> (http://mailman.alsa-project.org/pipermail/alsa-devel/2015-June/093214.html).
>
> V2: Minor code change, otherwise the differences compared to V1 are solely
> related to comments, in particular considerations given to the
> potential consequences of the patch. After consideration, it seems to me
> that the patch as it stands augments the current framework with the
> functionality indicated in the commit message above, without restricting
> other current uses. Further functionality could be added, but IMHO that
> should be done as the need arises, when it can be properly tested in a
> real-world setup.
>
>   sound/soc/soc-pcm.c |   49 +++++++++++++++++++++++++++++++++++++++++++++++++
>   1 file changed, 49 insertions(+)
>
> diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
> index 35fe58f4..08407ba 100644
> --- a/sound/soc/soc-pcm.c
> +++ b/sound/soc/soc-pcm.c
> @@ -34,6 +34,24 @@
>
>   #define DPCM_MAX_BE_USERS	8
>
> +/*
> + * snd_soc_dai_stream_valid() - check if a DAI supports the given stream
> + *
> + * Returns true if the DAI supports the indicated stream type.
> + */
> +static bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int stream)
> +{
> +	struct snd_soc_pcm_stream *codec_stream;
> +
> +	if (stream == SNDRV_PCM_STREAM_PLAYBACK)
> +		codec_stream = &dai->driver->playback;
> +	else
> +		codec_stream = &dai->driver->capture;
> +
> +	/* If the codec specifies any rate at all, it supports the stream. */
> +	return codec_stream->rates;
> +}
> +
>   /**
>    * 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.

> +			continue;
> +
>   		codec_dai_drv = rtd->codec_dais[i]->driver;
>   		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
>   			codec_stream = &codec_dai_drv->playback;
> @@ -827,6 +859,23 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
>   		struct snd_soc_dai *codec_dai = rtd->codec_dais[i];
>   		struct snd_pcm_hw_params codec_params;
>
> +		/*
> +		 * Skip CODECs which don't support the current stream type,
> +		 * the idea being that if a CODEC is not used for the currently
> +		 * set up transfer direction, it should not need to be
> +		 * configured, especially since the configuration used might
> +		 * not even be supported by that CODEC. There may be cases
> +		 * however where a CODEC needs to be set up although it is
> +		 * actually not being used for the transfer, e.g. if a
> +		 * capture-only CODEC is acting as an LRCLK and/or BCLK master
> +		 * for the DAI link including a playback-only CODEC.
> +		 * If this becomes necessary, we will have to augment the
> +		 * machine driver setup with information on how to act, so
> +		 * we can do the right thing here.
> +		 */
> +		if (!snd_soc_dai_stream_valid(codec_dai, substream->stream))
> +			continue;
> +
>   		/* copy params for each codec */
>   		codec_params = *params;
>
>