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
|
=======================
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;
}
|
