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ASoC: doc: ReSTize codec_to_codec.txt

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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>
This commit is contained in:
Takashi Iwai 2016-11-11 16:55:29 +01:00
parent 76228a2b54
commit c6ab9e57e8
2 changed files with 43 additions and 37 deletions
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79
Documentation/sound/alsa/soc/codec_to_codec.txt → Documentation/sound/soc/codec-to-codec.rst
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@ -1,37 +1,41 @@
==============================================
Creating codec to codec dai link for ALSA dapm Creating codec to codec dai link for ALSA dapm
=================================================== ==============================================
Mostly the flow of audio is always from CPU to codec so your system Mostly the flow of audio is always from CPU to codec so your system
will look as below: will look as below:
::
--------- --------- --------- ---------
| | dai | | | | dai | |
CPU -------> codec CPU -------> codec
| | | | | | | |
--------- --------- --------- ---------
In case your system looks as below: In case your system looks as below:
--------- ::
| |
codec-2 ---------
| | | |
--------- codec-2
| | |
dai-2 ---------
| |
---------- --------- dai-2
| | dai-1 | | |
CPU -------> codec-1 ---------- ---------
| | | | | | dai-1 | |
---------- --------- CPU -------> codec-1
| | | | |
dai-3 ---------- ---------
| |
--------- dai-3
| | |
codec-3 ---------
| | | |
--------- codec-3
| |
---------
Suppose codec-2 is a bluetooth chip and codec-3 is connected to Suppose codec-2 is a bluetooth chip and codec-3 is connected to
a speaker and you have a below scenario: 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 Your dai_link should appear as below in your machine
file: file:
::
/* /*
* this pcm stream only supports 24 bit, 2 channel and * this pcm stream only supports 24 bit, 2 channel and
* 48k sampling rate. * 48k sampling rate.
*/ */
static const struct snd_soc_pcm_stream dsp_codec_params = { static const struct snd_soc_pcm_stream dsp_codec_params = {
.formats = SNDRV_PCM_FMTBIT_S24_LE, .formats = SNDRV_PCM_FMTBIT_S24_LE,
.rate_min = 48000, .rate_min = 48000,
.rate_max = 48000, .rate_max = 48000,
.channels_min = 2, .channels_min = 2,
.channels_max = 2, .channels_max = 2,
}; };
{ {
.name = "CPU-DSP", .name = "CPU-DSP",
.stream_name = "CPU-DSP", .stream_name = "CPU-DSP",
.cpu_dai_name = "samsung-i2s.0", .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, | SND_SOC_DAIFMT_CBM_CFM,
.ignore_suspend = 1, .ignore_suspend = 1,
.params = &dsp_codec_params, .params = &dsp_codec_params,
}, },
{ {
.name = "DSP-CODEC", .name = "DSP-CODEC",
.stream_name = "DSP-CODEC", .stream_name = "DSP-CODEC",
.cpu_dai_name = "wm0010-sdi2", .cpu_dai_name = "wm0010-sdi2",
@ -77,7 +82,7 @@ static const struct snd_soc_pcm_stream dsp_codec_params = {
| SND_SOC_DAIFMT_CBM_CFM, | SND_SOC_DAIFMT_CBM_CFM,
.ignore_suspend = 1, .ignore_suspend = 1,
.params = &dsp_codec_params, .params = &dsp_codec_params,
}, },
Above code snippet is motivated from sound/soc/samsung/speyside.c. Above code snippet is motivated from sound/soc/samsung/speyside.c.

1
Documentation/sound/soc/index.rst
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@ -17,3 +17,4 @@ The documentation is spilt into the following sections:-
clocking clocking
jack jack
dpcm dpcm
codec-to-codec