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author | Linus Torvalds <torvalds@linux-foundation.org> | 2016-12-12 21:58:13 -0800 |
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committer | Linus Torvalds <torvalds@linux-foundation.org> | 2016-12-12 21:58:13 -0800 |
commit | e7aa8c2eb11ba69b1b69099c3c7bd6be3087b0ba (patch) | |
tree | f63906f41699c8e38af9d12b063e2ceab0286ef2 /Documentation/sound/soc/codec.rst | |
parent | e34bac726d27056081d0250c0e173e4b155aa340 (diff) | |
parent | 868c97a846a73e937d835b09b8c885a69df50ec8 (diff) | |
download | linux-e7aa8c2eb11ba69b1b69099c3c7bd6be3087b0ba.tar.gz linux-e7aa8c2eb11ba69b1b69099c3c7bd6be3087b0ba.tar.bz2 linux-e7aa8c2eb11ba69b1b69099c3c7bd6be3087b0ba.zip |
Merge tag 'docs-4.10' of git://git.lwn.net/linux
Pull documentation update from Jonathan Corbet:
"These are the documentation changes for 4.10.
It's another busy cycle for the docs tree, as the sphinx conversion
continues. Highlights include:
- Further work on PDF output, which remains a bit of a pain but
should be more solid now.
- Five more DocBook template files converted to Sphinx. Only 27 to
go... Lots of plain-text files have also been converted and
integrated.
- Images in binary formats have been replaced with more
source-friendly versions.
- Various bits of organizational work, including the renaming of
various files discussed at the kernel summit.
- New documentation for the device_link mechanism.
... and, of course, lots of typo fixes and small updates"
* tag 'docs-4.10' of git://git.lwn.net/linux: (193 commits)
dma-buf: Extract dma-buf.rst
Update Documentation/00-INDEX
docs: 00-INDEX: document directories/files with no docs
docs: 00-INDEX: remove non-existing entries
docs: 00-INDEX: add missing entries for documentation files/dirs
docs: 00-INDEX: consolidate process/ and admin-guide/ description
scripts: add a script to check if Documentation/00-INDEX is sane
Docs: change sh -> awk in REPORTING-BUGS
Documentation/core-api/device_link: Add initial documentation
core-api: remove an unexpected unident
ppc/idle: Add documentation for powersave=off
Doc: Correct typo, "Introdution" => "Introduction"
Documentation/atomic_ops.txt: convert to ReST markup
Documentation/local_ops.txt: convert to ReST markup
Documentation/assoc_array.txt: convert to ReST markup
docs-rst: parse-headers.pl: cleanup the documentation
docs-rst: fix media cleandocs target
docs-rst: media/Makefile: reorganize the rules
docs-rst: media: build SVG from graphviz files
docs-rst: replace bayer.png by a SVG image
...
Diffstat (limited to 'Documentation/sound/soc/codec.rst')
-rw-r--r-- | Documentation/sound/soc/codec.rst | 190 |
1 files changed, 190 insertions, 0 deletions
diff --git a/Documentation/sound/soc/codec.rst b/Documentation/sound/soc/codec.rst new file mode 100644 index 000000000000..f87612b94812 --- /dev/null +++ b/Documentation/sound/soc/codec.rst @@ -0,0 +1,190 @@ +======================= +ASoC Codec Class Driver +======================= + +The codec class driver is generic and hardware independent code that configures +the codec, FM, MODEM, BT or external DSP to provide audio capture and playback. +It should contain no code that is specific to the target platform or machine. +All platform and machine specific code should be added to the platform and +machine drivers respectively. + +Each codec class driver *must* provide the following features:- + +1. Codec DAI and PCM configuration +2. Codec control IO - using RegMap API +3. Mixers and audio controls +4. Codec audio operations +5. DAPM description. +6. DAPM event handler. + +Optionally, codec drivers can also provide:- + +7. DAC Digital mute control. + +Its probably best to use this guide in conjunction with the existing codec +driver code in sound/soc/codecs/ + +ASoC Codec driver breakdown +=========================== + +Codec DAI and PCM configuration +------------------------------- +Each codec driver must have a struct snd_soc_dai_driver to define its DAI and +PCM capabilities and operations. This struct is exported so that it can be +registered with the core by your machine driver. + +e.g. +:: + + static struct snd_soc_dai_ops wm8731_dai_ops = { + .prepare = wm8731_pcm_prepare, + .hw_params = wm8731_hw_params, + .shutdown = wm8731_shutdown, + .digital_mute = wm8731_mute, + .set_sysclk = wm8731_set_dai_sysclk, + .set_fmt = wm8731_set_dai_fmt, + }; + + struct snd_soc_dai_driver wm8731_dai = { + .name = "wm8731-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM8731_RATES, + .formats = WM8731_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM8731_RATES, + .formats = WM8731_FORMATS,}, + .ops = &wm8731_dai_ops, + .symmetric_rates = 1, + }; + + +Codec control IO +---------------- +The codec can usually be controlled via an I2C or SPI style interface +(AC97 combines control with data in the DAI). The codec driver should use the +Regmap API for all codec IO. Please see include/linux/regmap.h and existing +codec drivers for example regmap usage. + + +Mixers and audio controls +------------------------- +All the codec mixers and audio controls can be defined using the convenience +macros defined in soc.h. +:: + + #define SOC_SINGLE(xname, reg, shift, mask, invert) + +Defines a single control as follows:- +:: + + xname = Control name e.g. "Playback Volume" + reg = codec register + shift = control bit(s) offset in register + mask = control bit size(s) e.g. mask of 7 = 3 bits + invert = the control is inverted + +Other macros include:- +:: + + #define SOC_DOUBLE(xname, reg, shift_left, shift_right, mask, invert) + +A stereo control +:: + + #define SOC_DOUBLE_R(xname, reg_left, reg_right, shift, mask, invert) + +A stereo control spanning 2 registers +:: + + #define SOC_ENUM_SINGLE(xreg, xshift, xmask, xtexts) + +Defines an single enumerated control as follows:- +:: + + xreg = register + xshift = control bit(s) offset in register + xmask = control bit(s) size + xtexts = pointer to array of strings that describe each setting + + #define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xtexts) + +Defines a stereo enumerated control + + +Codec Audio Operations +---------------------- +The codec driver also supports the following ALSA PCM operations:- +:: + + /* SoC audio ops */ + struct snd_soc_ops { + int (*startup)(struct snd_pcm_substream *); + void (*shutdown)(struct snd_pcm_substream *); + int (*hw_params)(struct snd_pcm_substream *, struct snd_pcm_hw_params *); + int (*hw_free)(struct snd_pcm_substream *); + int (*prepare)(struct snd_pcm_substream *); + }; + +Please refer to the ALSA driver PCM documentation for details. +http://www.alsa-project.org/~iwai/writing-an-alsa-driver/ + + +DAPM description +---------------- +The Dynamic Audio Power Management description describes the codec power +components and their relationships and registers to the ASoC core. +Please read dapm.txt for details of building the description. + +Please also see the examples in other codec drivers. + + +DAPM event handler +------------------ +This function is a callback that handles codec domain PM calls and system +domain PM calls (e.g. suspend and resume). It is used to put the codec +to sleep when not in use. + +Power states:- +:: + + SNDRV_CTL_POWER_D0: /* full On */ + /* vref/mid, clk and osc on, active */ + + SNDRV_CTL_POWER_D1: /* partial On */ + SNDRV_CTL_POWER_D2: /* partial On */ + + SNDRV_CTL_POWER_D3hot: /* Off, with power */ + /* everything off except vref/vmid, inactive */ + + SNDRV_CTL_POWER_D3cold: /* Everything Off, without power */ + + +Codec DAC digital mute control +------------------------------ +Most codecs have a digital mute before the DACs that can be used to +minimise any system noise. The mute stops any digital data from +entering the DAC. + +A callback can be created that is called by the core for each codec DAI +when the mute is applied or freed. + +i.e. +:: + + static int wm8974_mute(struct snd_soc_dai *dai, int mute) + { + struct snd_soc_codec *codec = dai->codec; + u16 mute_reg = snd_soc_read(codec, WM8974_DAC) & 0xffbf; + + if (mute) + snd_soc_write(codec, WM8974_DAC, mute_reg | 0x40); + else + snd_soc_write(codec, WM8974_DAC, mute_reg); + return 0; + } |