[alsa-devel] ASoC : playback and capture counts in soc_new_pcm

[JongHo Kim]

Hi I am developing the ALSA ASoC part. I has some problem.
Now my situation must only use the cpu_dai(AP I2S IP) to half duplex.
So, I want to allocate either a capture substream or a playback substream.
But ASoC(soc-pcm.c) referred to the playback(capture)_channel_min of
codec_dai
at the calling snd_pcm_new in soc_new_pcm.
I do not want to modify the codec_dai codes.
As a result, hope to see below

/dev/snd/pcmC0D0p
/dev/snd/pcmC0D1c

attach the ASCII schematic diagram.

Slaves                          Master
+--------------+               +-----------------+
|I2S.0 data_out|---------------|data_in(AIF1)    |
|I2S.0 bclk    |-------+-------|bclk             |
|I2S.0 lrclk   |-------|--+----|lrclk            |
+--------------+       |  |    |                 |
+--------------+       |  |    |   AUDIO CODEC   |
|I2S.1 bclk    |-------+  |    |                 |
|I2S.1 lrclk   |----------+    |                 |
|I2S.1 data_in |---------------|data_out(AIF1)   |
+--------------+               +-----------------+

attach the i2s and dai_link codes.
-----------------------------------------------------------------------------
static struct snd_soc_dai_driver board_i2s_dai_hf_playback_template = {
.probe = board_i2s_probe,
.playback = {
.stream_name = "Playback",
.channels_min = 1,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_96000,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
},
.ops = &board_i2s_dai_ops,
};

static struct snd_soc_dai_driver board_i2s_dai_hf_capture_template = {
.probe = board_i2s_probe,
.capture = {
.stream_name = "Capture",
.channels_min = 1,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_96000,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
},
.ops = &board_i2s_dai_ops,
};

static struct snd_soc_dai_link board_dai[] = {
{
.name = "audcodec_l0",
.stream_name = "audcodec HiFi",
.cpu_dai_name = "board-i2s.0",
.codec_dai_name = "audcodec-aif1",
.platform_name = "board-pcm-audio",
.codec_name = "audcodec-codec",
.ops = &board_ops,
},
{
.name = "audcodec_l1",
.stream_name = "audcodec HiFi",
.cpu_dai_name = "board-i2s.1",
.codec_dai_name = "audcodec-aif1",
.platform_name = "board-pcm-audio",
.codec_name = "audcodec-codec",
.ops = &board_ops,
},
};
-----------------------------------------------------------------------------

So, I suggest the next diff.

--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -2014,9 +2014,11 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int
num)
                if (cpu_dai->driver->capture.channels_min)
                        capture = 1;
        } else {
-               if (codec_dai->driver->playback.channels_min)
+               if (codec_dai->driver->playback.channels_min &&
+                   cpu_dai->driver->playback.channels_min)
                        playback = 1;
-               if (codec_dai->driver->capture.channels_min)
+               if (codec_dai->driver->playback.channels_min &&
+                   cpu_dai->driver->playback.channels_min)
                        capture = 1;
        }

Thanks.