ASoC Codec Driver
=================

The codec driver is generic and hardware independent code that configures the
codec 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 driver must provide the following features:-

 1) Digital audio interface (DAI) description
 2) Digital audio interface configuration
 3) PCM's description
 4) Codec control IO - using I2C, 3 Wire(SPI) or both API's
 5) Mixers and audio controls
 6) Sysclk configuration
 7) Codec audio operations

Optionally, codec drivers can also provide:-

 8) DAPM description.
 9) DAPM event handler.
10) DAC Digital mute control.

It's probably best to use this guide in conjuction with the existing codec
driver code in sound/soc/codecs/

ASoC Codec driver breakdown
===========================

1 - Digital Audio Interface (DAI) description
---------------------------------------------
The DAI is a digital audio data transfer link between the codec and host SoC
CPU. It typically has data transfer capabilities in both directions
(playback and capture) and can run at a variety of different speeds.
Supported interfaces currently include AC97, I2S and generic PCM style links.
Please read DAI.txt for implementation information.


2 - Digital Audio Interface (DAI) configuration
-----------------------------------------------
DAI configuration is handled by the codec_pcm_prepare function and is
responsible for configuring and starting the DAI on the codec. This can be
called multiple times and is atomic. It can access the runtime parameters.

This usually consists of a large function with numerous switch statements to
set up each configuration option. These options are set by the core at runtime.


3 - Codec PCM's
---------------
Each codec must have it's PCM's defined. This defines the number of channels,
stream names, callbacks and codec name. It is also used to register the DAI
with the ASoC core. The PCM structure also associates the DAI capabilities with
the ALSA PCM.

e.g.

static struct snd_soc_pcm_codec wm8731_pcm_client = {
	.name = "WM8731",
	.playback = {
		.stream_name = "Playback",
		.channels_min = 1,
		.channels_max = 2,
	},
	.capture = {
		.stream_name = "Capture",
		.channels_min = 1,
		.channels_max = 2,
	},
	.config_sysclk = wm8731_config_sysclk,
	.ops = {
		.prepare = wm8731_pcm_prepare,
	},
	.caps = {
		.num_modes = ARRAY_SIZE(wm8731_hwfmt),
		.modes = &wm8731_hwfmt[0],
	},
};


4 - Codec control IO
--------------------
The codec can ususally be controlled via an I2C or SPI style interface (AC97
combines control with data in the DAI). The codec drivers will have to provide
functions to read and write the codec registers along with supplying a register
cache:-

	/* IO control data and register cache */
    void *control_data; /* codec control (i2c/3wire) data */
    void *reg_cache;

Codec read/write should do any data formatting and call the hardware read write
below to perform the IO. These functions are called by the core and alsa when
performing DAPM or changing the mixer:-

    unsigned int (*read)(struct snd_soc_codec *, unsigned int);
    int (*write)(struct snd_soc_codec *, unsigned int, unsigned int);

Codec hardware IO functions - usually points to either the I2C, SPI or AC97
read/write:-

	hw_write_t hw_write;
	hw_read_t hw_read;


5 - 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


6 - System clock configuration.
-------------------------------
The system clock that drives the audio subsystem can change depending on sample
rate and the system power state. i.e.

o Higher sample rates sometimes need a higher system clock.
o Low system power states can sometimes limit the available clocks.

This function is a callback that the machine driver can call to set and
determine if the clock and sample rate combination is supported by the codec at
the present time (and system state).

NOTE: If the codec has a PLL then it has a lot more flexability wrt clock and
sample rate combinations.

Your config_sysclock function should return the MCLK if it's a valid
combination for your codec else 0;

Please read clocking.txt now.


7 - Codec Audio Operations
--------------------------
The codec driver also supports the following alsa 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/c436.htm


8 - DAPM description.
---------------------
The Dynamic Audio Power Management description describes the codec's power
components, 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.


9 - 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's 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 */


10 - Codec DAC digital mute control.
------------------------------------
Most codecs have a digital mute before the DAC's 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_codec *codec,
	struct snd_soc_codec_dai *dai, int mute)
{
	u16 mute_reg = wm8974_read_reg_cache(codec, WM8974_DAC) & 0xffbf;
	if(mute)
		wm8974_write(codec, WM8974_DAC, mute_reg | 0x40);
	else
		wm8974_write(codec, WM8974_DAC, mute_reg);
	return 0;
}
