191 lines
5.2 KiB
ReStructuredText
191 lines
5.2 KiB
ReStructuredText
=======================
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ASoC Codec Class Driver
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=======================
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The codec class driver is generic and hardware independent code that configures
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the codec, FM, MODEM, BT or external DSP to provide audio capture and playback.
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It should contain no code that is specific to the target platform or machine.
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All platform and machine specific code should be added to the platform and
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machine drivers respectively.
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Each codec class driver *must* provide the following features:-
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1. Codec DAI and PCM configuration
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2. Codec control IO - using RegMap API
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3. Mixers and audio controls
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4. Codec audio operations
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5. DAPM description.
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6. DAPM event handler.
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Optionally, codec drivers can also provide:-
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7. DAC Digital mute control.
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Its probably best to use this guide in conjunction with the existing codec
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driver code in sound/soc/codecs/
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ASoC Codec driver breakdown
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===========================
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Codec DAI and PCM configuration
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-------------------------------
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Each codec driver must have a struct snd_soc_dai_driver to define its DAI and
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PCM capabilities and operations. This struct is exported so that it can be
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registered with the core by your machine driver.
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e.g.
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::
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static struct snd_soc_dai_ops wm8731_dai_ops = {
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.prepare = wm8731_pcm_prepare,
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.hw_params = wm8731_hw_params,
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.shutdown = wm8731_shutdown,
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.digital_mute = wm8731_mute,
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.set_sysclk = wm8731_set_dai_sysclk,
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.set_fmt = wm8731_set_dai_fmt,
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};
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struct snd_soc_dai_driver wm8731_dai = {
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.name = "wm8731-hifi",
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.playback = {
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.stream_name = "Playback",
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.channels_min = 1,
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.channels_max = 2,
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.rates = WM8731_RATES,
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.formats = WM8731_FORMATS,},
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.capture = {
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.stream_name = "Capture",
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.channels_min = 1,
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.channels_max = 2,
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.rates = WM8731_RATES,
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.formats = WM8731_FORMATS,},
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.ops = &wm8731_dai_ops,
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.symmetric_rates = 1,
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};
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Codec control IO
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----------------
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The codec can usually be controlled via an I2C or SPI style interface
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(AC97 combines control with data in the DAI). The codec driver should use the
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Regmap API for all codec IO. Please see include/linux/regmap.h and existing
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codec drivers for example regmap usage.
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Mixers and audio controls
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-------------------------
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All the codec mixers and audio controls can be defined using the convenience
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macros defined in soc.h.
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::
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#define SOC_SINGLE(xname, reg, shift, mask, invert)
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Defines a single control as follows:-
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::
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xname = Control name e.g. "Playback Volume"
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reg = codec register
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shift = control bit(s) offset in register
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mask = control bit size(s) e.g. mask of 7 = 3 bits
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invert = the control is inverted
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Other macros include:-
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::
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#define SOC_DOUBLE(xname, reg, shift_left, shift_right, mask, invert)
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A stereo control
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::
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#define SOC_DOUBLE_R(xname, reg_left, reg_right, shift, mask, invert)
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A stereo control spanning 2 registers
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::
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#define SOC_ENUM_SINGLE(xreg, xshift, xmask, xtexts)
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Defines an single enumerated control as follows:-
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::
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xreg = register
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xshift = control bit(s) offset in register
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xmask = control bit(s) size
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xtexts = pointer to array of strings that describe each setting
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#define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xtexts)
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Defines a stereo enumerated control
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Codec Audio Operations
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----------------------
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The codec driver also supports the following ALSA PCM operations:-
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::
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/* SoC audio ops */
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struct snd_soc_ops {
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int (*startup)(struct snd_pcm_substream *);
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void (*shutdown)(struct snd_pcm_substream *);
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int (*hw_params)(struct snd_pcm_substream *, struct snd_pcm_hw_params *);
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int (*hw_free)(struct snd_pcm_substream *);
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int (*prepare)(struct snd_pcm_substream *);
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};
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Please refer to the ALSA driver PCM documentation for details.
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http://www.alsa-project.org/~iwai/writing-an-alsa-driver/
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DAPM description
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----------------
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The Dynamic Audio Power Management description describes the codec power
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components and their relationships and registers to the ASoC core.
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Please read dapm.rst for details of building the description.
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Please also see the examples in other codec drivers.
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DAPM event handler
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------------------
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This function is a callback that handles codec domain PM calls and system
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domain PM calls (e.g. suspend and resume). It is used to put the codec
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to sleep when not in use.
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Power states:-
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::
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SNDRV_CTL_POWER_D0: /* full On */
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/* vref/mid, clk and osc on, active */
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SNDRV_CTL_POWER_D1: /* partial On */
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SNDRV_CTL_POWER_D2: /* partial On */
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SNDRV_CTL_POWER_D3hot: /* Off, with power */
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/* everything off except vref/vmid, inactive */
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SNDRV_CTL_POWER_D3cold: /* Everything Off, without power */
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Codec DAC digital mute control
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------------------------------
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Most codecs have a digital mute before the DACs that can be used to
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minimise any system noise. The mute stops any digital data from
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entering the DAC.
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A callback can be created that is called by the core for each codec DAI
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when the mute is applied or freed.
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i.e.
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::
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static int wm8974_mute(struct snd_soc_dai *dai, int mute)
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{
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struct snd_soc_component *component = dai->component;
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u16 mute_reg = snd_soc_component_read32(component, WM8974_DAC) & 0xffbf;
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if (mute)
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snd_soc_component_write(component, WM8974_DAC, mute_reg | 0x40);
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else
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snd_soc_component_write(component, WM8974_DAC, mute_reg);
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return 0;
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}
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