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

Signed-off-by: Takashi Iwai tiwai@suse.de ---

Mark, this is the rest one in your branch, which I pulled now. It's already in topic/restize-docs branch, so I can merge to for-next as is if it's OK.

.../codec_to_codec.txt => soc/codec-to-codec.rst} | 79 ++++++++++++---------- Documentation/sound/soc/index.rst | 1 + 2 files changed, 43 insertions(+), 37 deletions(-) rename Documentation/sound/{alsa/soc/codec_to_codec.txt => soc/codec-to-codec.rst} (68%)

diff --git a/Documentation/sound/alsa/soc/codec_to_codec.txt b/Documentation/sound/soc/codec-to-codec.rst similarity index 68% rename from Documentation/sound/alsa/soc/codec_to_codec.txt rename to 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