[v4,01/12] ASoC: Intel: mfld-pcm: add FE and BE ops

Commit Message

Subhransu S. Prusty Aug. 4, 2014, 9:45 a.m. UTC

From: Vinod Koul <[e-mail]>

Now that we have added code for managing DSP pipelines we need to add the code
for DSPs FrontEnd and Backend dai.

Signed-off-by: Vinod Koul <[e-mail]>
Signed-off-by: Subhransu S. Prusty <[e-mail]>
---
 sound/soc/intel/sst-mfld-platform-pcm.c | 113 ++++++++++++++++++++++++--------
 1 file changed, 84 insertions(+), 29 deletions(-)

Comments

Mark Brown Aug. 13, 2014, 7:35 p.m. UTC | #1

On Mon, Aug 04, 2014 at 03:15:52PM +0530, Subhransu S. Prusty wrote:
> From: Vinod Koul <[e-mail]>
> 
> Now that we have added code for managing DSP pipelines we need to add the code
> for DSPs FrontEnd and Backend dai.

This would be a lot easier to apply if the changelog were clearer, I
can't tell how we get from having code to manage the pipelines to
needing or wanting this code.  Is the current code broken and this a bug
fix or is this preparing for future changes?  The changelog says that
this is adding ops but the diff is adding DAIs none of which have any
ops defined.

I'm also a bit surprised to see this without anything to add routes
between the front and back ends or updates to the machine drivers for
new DAI names - won't this cause problems for the machine drivers?

Patch

diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c
index a89ff7e18e1a..364034024cee 100644
--- a/sound/soc/intel/sst-mfld-platform-pcm.c
+++ b/sound/soc/intel/sst-mfld-platform-pcm.c
@@ -101,35 +101,6 @@  static struct sst_dev_stream_map dpcm_strm_map[] = {
 	{MERR_DPCM_AUDIO, 0, SNDRV_PCM_STREAM_CAPTURE, PIPE_PCM1_OUT, SST_TASK_ID_MEDIA, 0},
 };
 
-/* MFLD - MSIC */
-static struct snd_soc_dai_driver sst_platform_dai[] = {
-{
-	.name = "Headset-cpu-dai",
-	.id = 0,
-	.playback = {
-		.channels_min = SST_STEREO,
-		.channels_max = SST_STEREO,
-		.rates = SNDRV_PCM_RATE_48000,
-		.formats = SNDRV_PCM_FMTBIT_S24_LE,
-	},
-	.capture = {
-		.channels_min = 1,
-		.channels_max = 5,
-		.rates = SNDRV_PCM_RATE_48000,
-		.formats = SNDRV_PCM_FMTBIT_S24_LE,
-	},
-},
-{
-	.name = "Compress-cpu-dai",
-	.compress_dai = 1,
-	.playback = {
-		.channels_min = SST_STEREO,
-		.channels_max = SST_STEREO,
-		.rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE,
-	},
-},
-};
 
 /* helper functions */
 void sst_set_stream_status(struct sst_runtime_stream *stream,
@@ -441,6 +412,90 @@  static struct snd_soc_dai_ops sst_media_dai_ops = {
 	.hw_free = sst_media_hw_free,
 };
 
+static struct snd_soc_dai_driver sst_platform_dai[] = {
+{
+	.name = "media-cpu-dai",
+	.ops = &sst_media_dai_ops,
+	.playback = {
+		.stream_name = "Headset Playback",
+		.channels_min = SST_STEREO,
+		.channels_max = SST_STEREO,
+		.rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+	.capture = {
+		.stream_name = "Headset Capture",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+},
+{
+	.name = "compress-cpu-dai",
+	.compress_dai = 1,
+	.playback = {
+		.stream_name = "Compress Playback",
+		.channels_min = SST_STEREO,
+		.channels_max = SST_STEREO,
+		.rates = SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+},
+/*BE CPU  Dais */
+{
+	.name = "ssp0-port",
+	.playback = {
+		.stream_name = "ssp0 Tx",
+		.channels_min = SST_STEREO,
+		.channels_max = SST_STEREO,
+		.rates = SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+	.capture = {
+		.stream_name = "ssp0 Rx",
+		.channels_min = SST_STEREO,
+		.channels_max = SST_STEREO,
+		.rates = SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+},
+{
+	.name = "ssp1-port",
+	.playback = {
+		.stream_name = "ssp1 Tx",
+		.channels_min = SST_STEREO,
+		.channels_max = SST_STEREO,
+		.rates = SNDRV_PCM_RATE_8000|SNDRV_PCM_RATE_16000|SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+	.capture = {
+		.stream_name = "ssp1 Rx",
+		.channels_min = SST_STEREO,
+		.channels_max = SST_STEREO,
+		.rates = SNDRV_PCM_RATE_8000|SNDRV_PCM_RATE_16000|SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+},
+{
+	.name = "ssp2-port",
+	.playback = {
+		.stream_name = "ssp2 Tx",
+		.channels_min = SST_STEREO,
+		.channels_max = SST_STEREO,
+		.rates = SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+	.capture = {
+		.stream_name = "ssp2 Rx",
+		.channels_min = SST_STEREO,
+		.channels_max = SST_STEREO,
+		.rates = SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+},
+};
+
 static int sst_platform_open(struct snd_pcm_substream *substream)
 {
 	struct snd_pcm_runtime *runtime;