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authorTakashi Iwai <tiwai@suse.de>2016-11-11 00:16:04 +0300
committerTakashi Iwai <tiwai@suse.de>2016-11-11 19:33:07 +0300
commit693ba474a39a2c22e1576995139c9bfdd8b554c8 (patch)
tree306e5f9a3a65bed5907f6476033ab6da00be5173 /Documentation/sound/alsa
parent8e5336a14476e7350b0cf78a99541de6ed51655c (diff)
downloadlinux-693ba474a39a2c22e1576995139c9bfdd8b554c8.tar.xz
ASoC: doc: ReSTize codec.txt
A simple conversion from a plain text file. The section numbers are dropped to align with other documents. Acked-by: Mark Brown <broonie@kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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diff --git a/Documentation/sound/alsa/soc/codec.txt b/Documentation/sound/alsa/soc/codec.txt
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-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
-===========================
-
-1 - 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,
-};
-
-
-2 - 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.
-
-
-3 - 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
-
-
-4 - 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/
-
-
-5 - 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.
-
-
-6 - 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 */
-
-
-7 - 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;
-}