From c82351da2e9f2b14d5664e41b021ec1fd948b932 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Wed, 15 Oct 2014 20:12:59 +0530 Subject: [PATCH] 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: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst-mfld-platform-pcm.c | 153 +++++++++++++++++++----- 1 file changed, 125 insertions(+), 28 deletions(-) diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c index aa9b600dfc9b..e7cf18d1d421 100644 --- a/sound/soc/intel/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/sst-mfld-platform-pcm.c @@ -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}, }; -/* MFLD - MSIC */ -static struct snd_soc_dai_driver sst_platform_dai[] = { +static int sst_media_digital_mute(struct snd_soc_dai *dai, int mute, int stream) { - .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, - }, -}, -}; + + return sst_send_pipe_gains(dai, stream, mute); +} /* helper functions */ 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, 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 = { .startup = sst_media_open, .shutdown = sst_media_close, .prepare = sst_media_prepare, .hw_params = sst_media_hw_params, .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) -- 2.20.1