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;
}