[alsa-devel] ASoC: multicodec: No matching channels with system with DACs and ADCs on same bus

[Caleb Crome]

Hi again,
  I have a system that has separate DACs and ADCs, as opposed to
bidirectional CODECS.

The ADCs have a snd_soc_dai_driver structure that have a .capture
field defined with a .rates parameter, and the DACs have a
snd_soc_dai_driver structure that have a .playback field defined with
a .rates parameter.

However, the ADCs have no .playback.rates, and the DACs have no
.capture.rates. Even though the ADC can record at a given rate and the
DAC can play at that same rate, the soc_pcm_init_runtime_hw function
calculates the rates in this loop:

------------------------------
    /* first calculate min/max only for CODECs in the DAI link */
    for (i = 0; i < rtd->num_codecs; i++) {
        codec_dai_drv = rtd->codec_dais[i]->driver;
        if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
            codec_stream = &codec_dai_drv->playback;
        else
            codec_stream = &codec_dai_drv->capture;
        chan_min = max(chan_min, codec_stream->channels_min);
        chan_max = min(chan_max, codec_stream->channels_max);
        rate_min = max(rate_min, codec_stream->rate_min);
        rate_max = min_not_zero(rate_max, codec_stream->rate_max);
        formats &= codec_stream->formats;
        rates = snd_pcm_rate_mask_intersect(codec_stream->rates, rates);
    }
------------------------------

which can't work out that one side is play only and the other is
record only, yet they do in fact have matching rates.  So, I get a "no
matching channels" error.


Here are my example drivers declarations:
for the ADC:
----------------------
static struct snd_soc_dai_driver cs53l30_dai = {
    .name = "cs53l30-hifi",
    .capture = {
        .stream_name = "Capture",
        .channels_min = 4,
        .channels_max = 16,
        .rates = SNDRV_PCM_RATE_16000,
        .formats =SNDRV_PCM_FMTBIT_S16_LE,
    },
};
--------------------

and for the DAC

----------------------
static struct snd_soc_dai_driver tlv320dac3100_dai = {
    .name = "tlv320dac3100-hifi",
    .playback = {
        .stream_name = "Playback",
        .channels_min = 2,
        .channels_max = 16,
        .rates = SNDRV_PCM_RATE_16000,
        .formats =SNDRV_PCM_FMTBIT_S16_LE,
    },
    .ops = &tlv320dac3100_dai_ops,
};
------------------------------

I was able to fake it out by adding a .playback field to my ADC with 0
channels, and a .capture field to my DAC with 0 channels.

My question is: what's the right way to manage this?  I assume we
should probably tweak soc_pcm_init_runtime_hw, right?

Thanks,
  -Caleb