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Diffstat (limited to 'Documentation/sound/alsa/soc/codec_to_codec.txt')
| -rw-r--r-- | Documentation/sound/alsa/soc/codec_to_codec.txt | 103 |
1 files changed, 103 insertions, 0 deletions
diff --git a/Documentation/sound/alsa/soc/codec_to_codec.txt b/Documentation/sound/alsa/soc/codec_to_codec.txt new file mode 100644 index 000000000000..704a6483652c --- /dev/null +++ b/Documentation/sound/alsa/soc/codec_to_codec.txt @@ -0,0 +1,103 @@ +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. |