aboutsummaryrefslogtreecommitdiffstats
path: root/Documentation/sound/alsa/soc/codec.txt
blob: 48983c75aad9cacd7052a484408425602ff6f98b (plain) (blame)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
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) Codec DAI and PCM configuration
 2) Codec control IO - using I2C, 3 Wire(SPI) or both API's
 3) Mixers and audio controls
 4) Codec audio operations

Optionally, codec drivers can also provide:-

 5) DAPM description.
 6) DAPM event handler.
 7) 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 - Codec DAI and PCM configuration
-----------------------------------
Each codec driver must have a struct snd_soc_codec_dai to define it's DAI and
PCM's capablities and operations. This struct is exported so that it can be
registered with the core by your machine driver.

e.g.

struct snd_soc_codec_dai wm8731_dai = {
	.name = "WM8731",
	/* playback capabilities */
	.playback = {
		.stream_name = "Playback",
		.channels_min = 1,
		.channels_max = 2,
		.rates = WM8731_RATES,
		.formats = WM8731_FORMATS,},
	/* capture capabilities */
	.capture = {
		.stream_name = "Capture",
		.channels_min = 1,
		.channels_max = 2,
		.rates = WM8731_RATES,
		.formats = WM8731_FORMATS,},
	/* pcm operations - see section 4 below */
	.ops = {
		.prepare = wm8731_pcm_prepare,
		.hw_params = wm8731_hw_params,
		.shutdown = wm8731_shutdown,
	},
	/* DAI operations - see DAI.txt */
	.dai_ops = {
		.digital_mute = wm8731_mute,
		.set_sysclk = wm8731_set_dai_sysclk,
		.set_fmt = wm8731_set_dai_fmt,
	}
};
EXPORT_SYMBOL_GPL(wm8731_dai);


2 - 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;


3 - 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


4 - 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


5 - 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.


6 - 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 */


7 - 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;
}