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| author | Takashi Iwai <tiwai@suse.de> | 2016-11-11 16:55:29 +0100 |
|---|---|---|
| committer | Takashi Iwai <tiwai@suse.de> | 2016-11-11 17:35:10 +0100 |
| commit | c6ab9e57e84ee015bb9c5de213d9f85e5fd4e085 (patch) | |
| tree | adcc13f5864783da1342be5be836447a75d03b54 /Documentation/sound | |
| parent | Merge branch 'topic/doc' of git://git.kernel.org/pub/scm/linux/kernel/git/bro... (diff) | |
| download | linux-c6ab9e57e84ee015bb9c5de213d9f85e5fd4e085.tar.xz linux-c6ab9e57e84ee015bb9c5de213d9f85e5fd4e085.zip | |
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>
Diffstat (limited to 'Documentation/sound')
| -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.rst | 1 |
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 |