Commit c82351da authored by Vinod Koul's avatar Vinod Koul Committed by Mark Brown

ASoC: Intel: mfld-pcm: add FE and BE ops

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: default avatarSubhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: default avatarVinod Koul <vinod.koul@intel.com>
Signed-off-by: default avatarMark Brown <broonie@kernel.org>
parent e4f5ccd0
...@@ -101,35 +101,11 @@ static struct sst_dev_stream_map dpcm_strm_map[] = { ...@@ -101,35 +101,11 @@ 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}, {MERR_DPCM_AUDIO, 0, SNDRV_PCM_STREAM_CAPTURE, PIPE_PCM1_OUT, SST_TASK_ID_MEDIA, 0},
}; };
/* MFLD - MSIC */ static int sst_media_digital_mute(struct snd_soc_dai *dai, int mute, int stream)
static struct snd_soc_dai_driver sst_platform_dai[] = {
{ {
.name = "Headset-cpu-dai",
.id = 0, return sst_send_pipe_gains(dai, stream, mute);
.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 */ /* helper functions */
void sst_set_stream_status(struct sst_runtime_stream *stream, void sst_set_stream_status(struct sst_runtime_stream *stream,
...@@ -451,12 +427,133 @@ static int sst_media_hw_free(struct snd_pcm_substream *substream, ...@@ -451,12 +427,133 @@ static int sst_media_hw_free(struct snd_pcm_substream *substream,
return snd_pcm_lib_free_pages(substream); return snd_pcm_lib_free_pages(substream);
} }
static int sst_enable_ssp(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
int ret = 0;
if (!dai->active) {
ret = sst_handle_vb_timer(dai, true);
if (ret)
return ret;
ret = send_ssp_cmd(dai, dai->name, 1);
}
return ret;
}
static void sst_disable_ssp(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
if (!dai->active) {
send_ssp_cmd(dai, dai->name, 0);
sst_handle_vb_timer(dai, false);
}
}
static struct snd_soc_dai_ops sst_media_dai_ops = { static struct snd_soc_dai_ops sst_media_dai_ops = {
.startup = sst_media_open, .startup = sst_media_open,
.shutdown = sst_media_close, .shutdown = sst_media_close,
.prepare = sst_media_prepare, .prepare = sst_media_prepare,
.hw_params = sst_media_hw_params, .hw_params = sst_media_hw_params,
.hw_free = sst_media_hw_free, .hw_free = sst_media_hw_free,
.mute_stream = sst_media_digital_mute,
};
static struct snd_soc_dai_ops sst_compr_dai_ops = {
.mute_stream = sst_media_digital_mute,
};
static struct snd_soc_dai_ops sst_be_dai_ops = {
.startup = sst_enable_ssp,
.shutdown = sst_disable_ssp,
};
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,
.ops = &sst_compr_dai_ops,
.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",
.ops = &sst_be_dai_ops,
.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",
.ops = &sst_be_dai_ops,
.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",
.ops = &sst_be_dai_ops,
.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) static int sst_platform_open(struct snd_pcm_substream *substream)
......
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