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Diffstat (limited to 'Documentation/sound/soc/codec.rst')
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diff --git a/Documentation/sound/soc/codec.rst b/Documentation/sound/soc/codec.rst new file mode 100644 index 000000000..8a9737eb7 --- /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.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; + } |