ASoC: doc: ReSTize codec_to_codec.txt

Yet another simple conversion from a plain text file.
Renamed to codec-to-codec.rst to align with others.

Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

1 files changed, 0 insertions, 103 deletions

diff --git a/Documentation/sound/alsa/soc/codec_to_codec.txt b/Documentation/sound/alsa/soc/codec_to_codec.txt
deleted file mode 100644
index 704a6483652c..000000000000
--- a/Documentation/sound/alsa/soc/codec_to_codec.txt
+++ /dev/null
@@ -1,103 +0,0 @@
-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.