[Feature]add MT2731_MP2_MR2_SVN388 baseline version
Change-Id: Ief04314834b31e27effab435d3ca8ba33b499059
diff --git a/src/kernel/linux/v4.14/Documentation/sound/soc/codec-to-codec.rst b/src/kernel/linux/v4.14/Documentation/sound/soc/codec-to-codec.rst
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index 0000000..810109d
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+==============================================
+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.