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-rw-r--r--Documentation/sound/soc/codec-to-codec.rst (renamed from Documentation/sound/alsa/soc/codec_to_codec.txt)79
-rw-r--r--Documentation/sound/soc/index.rst1
2 files changed, 43 insertions, 37 deletions
diff --git a/Documentation/sound/alsa/soc/codec_to_codec.txt b/Documentation/sound/soc/codec-to-codec.rst
index 704a6483652c..810109d7500d 100644
--- a/Documentation/sound/alsa/soc/codec_to_codec.txt
+++ b/Documentation/sound/soc/codec-to-codec.rst
@@ -1,37 +1,41 @@
+==============================================
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
-| | | |
- --------- ---------
+ --------- ---------
+ | | dai | |
+ CPU -------> codec
+ | | | |
+ --------- ---------
In case your system looks as below:
- ---------
- | |
- codec-2
- | |
- ---------
- |
- dai-2
- |
- ---------- ---------
-| | dai-1 | |
- CPU -------> codec-1
-| | | |
- ---------- ---------
- |
- dai-3
- |
- ---------
- | |
- codec-3
- | |
- ---------
+::
+
+ ---------
+ | |
+ 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:
@@ -42,20 +46,21 @@ 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 = {
+ /*
+ * 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",
@@ -66,8 +71,8 @@ static const struct snd_soc_pcm_stream dsp_codec_params = {
| SND_SOC_DAIFMT_CBM_CFM,
.ignore_suspend = 1,
.params = &dsp_codec_params,
-},
-{
+ },
+ {
.name = "DSP-CODEC",
.stream_name = "DSP-CODEC",
.cpu_dai_name = "wm0010-sdi2",
@@ -77,7 +82,7 @@ static const struct snd_soc_pcm_stream dsp_codec_params = {
| SND_SOC_DAIFMT_CBM_CFM,
.ignore_suspend = 1,
.params = &dsp_codec_params,
-},
+ },
Above code snippet is motivated from sound/soc/samsung/speyside.c.
diff --git a/Documentation/sound/soc/index.rst b/Documentation/sound/soc/index.rst
index e142a0f25c5b..e57df2dab2fd 100644
--- a/Documentation/sound/soc/index.rst
+++ b/Documentation/sound/soc/index.rst
@@ -17,3 +17,4 @@ The documentation is spilt into the following sections:-
clocking
jack
dpcm
+ codec-to-codec