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authorDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-27 10:05:51 +0000
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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
+===========================
+
+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.rst 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_component *component = dai->component;
+ u16 mute_reg = snd_soc_component_read32(component, WM8974_DAC) & 0xffbf;
+
+ if (mute)
+ snd_soc_component_write(component, WM8974_DAC, mute_reg | 0x40);
+ else
+ snd_soc_component_write(component, WM8974_DAC, mute_reg);
+ return 0;
+ }