@@ -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))
+ 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;
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. No verification is made that the playback and capture codec setups are compatible in terms of number of TDM slots (where applicable). Signed-off-by: Ricard Wanderlof <ricardw@axis.com> --- V3: Updated commit message regarding the fact that the patch does not consider the slot setup in a TDM system with separate capture and playback codecs, after a comment from Pierre-Louis Bossart. Such a setup was not supported anyway before this patch, so let's handle that situation when and if it arises. 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. 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). sound/soc/soc-pcm.c | 49 +++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 49 insertions(+)