| ============================================== |
| Creating codec to codec dai link for ALSA dapm |
| ============================================== |
| |
| Mostly the flow of audio is always from CPU to codec so your system |
| will look as below: |
| :: |
| |
| --------- --------- |
| | | dai | | |
| CPU -------> codec |
| | | | | |
| --------- --------- |
| |
| In case your system looks as below: |
| :: |
| |
| --------- |
| | | |
| codec-2 |
| | | |
| --------- |
| | |
| dai-2 |
| | |
| ---------- --------- |
| | | dai-1 | | |
| CPU -------> codec-1 |
| | | | | |
| ---------- --------- |
| | |
| dai-3 |
| | |
| --------- |
| | | |
| codec-3 |
| | | |
| --------- |
| |
| Suppose codec-2 is a bluetooth chip and codec-3 is connected to |
| a speaker and you have a below scenario: |
| codec-2 will receive the audio data and the user wants to play that |
| audio through codec-3 without involving the CPU.This |
| aforementioned case is the ideal case when codec to codec |
| connection should be used. |
| |
| Your dai_link should appear as below in your machine |
| file: |
| :: |
| |
| /* |
| * this pcm stream only supports 24 bit, 2 channel and |
| * 48k sampling rate. |
| */ |
| static const struct snd_soc_pcm_stream dsp_codec_params = { |
| .formats = SNDRV_PCM_FMTBIT_S24_LE, |
| .rate_min = 48000, |
| .rate_max = 48000, |
| .channels_min = 2, |
| .channels_max = 2, |
| }; |
| |
| { |
| .name = "CPU-DSP", |
| .stream_name = "CPU-DSP", |
| .cpu_dai_name = "samsung-i2s.0", |
| .codec_name = "codec-2, |
| .codec_dai_name = "codec-2-dai_name", |
| .platform_name = "samsung-i2s.0", |
| .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
| | SND_SOC_DAIFMT_CBM_CFM, |
| .ignore_suspend = 1, |
| .params = &dsp_codec_params, |
| }, |
| { |
| .name = "DSP-CODEC", |
| .stream_name = "DSP-CODEC", |
| .cpu_dai_name = "wm0010-sdi2", |
| .codec_name = "codec-3, |
| .codec_dai_name = "codec-3-dai_name", |
| .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
| | SND_SOC_DAIFMT_CBM_CFM, |
| .ignore_suspend = 1, |
| .params = &dsp_codec_params, |
| }, |
| |
| Above code snippet is motivated from sound/soc/samsung/speyside.c. |
| |
| Note the "params" callback which lets the dapm know that this |
| dai_link is a codec to codec connection. |
| |
| In dapm core a route is created between cpu_dai playback widget |
| and codec_dai capture widget for playback path and vice-versa is |
| true for capture path. In order for this aforementioned route to get |
| triggered, DAPM needs to find a valid endpoint which could be either |
| a sink or source widget corresponding to playback and capture path |
| respectively. |
| |
| In order to trigger this dai_link widget, a thin codec driver for |
| the speaker amp can be created as demonstrated in wm8727.c file, it |
| sets appropriate constraints for the device even if it needs no control. |
| |
| Make sure to name your corresponding cpu and codec playback and capture |
| dai names ending with "Playback" and "Capture" respectively as dapm core |
| will link and power those dais based on the name. |
| |
| Note that in current device tree there is no way to mark a dai_link |
| as codec to codec. However, it may change in future. |