Documentation/sound/soc/codec-to-codec.rst

==============================================
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.

A dai_link in a "simple-audio-card" will automatically be detected as
codec to codec when all DAIs on the link belong to codec components.
The dai_link will be initialized with the subset of stream parameters
(channels, format, sample rate) supported by all DAIs on the link. Since
there is no way to provide these parameters in the device tree, this is
mostly useful for communication with simple fixed-function codecs, such
as a Bluetooth controller or cellular modem.