1=======================
2ASoC Codec Class Driver
3=======================
4
5The codec class driver is generic and hardware independent code that configures
6the codec, FM, MODEM, BT or external DSP to provide audio capture and playback.
7It should contain no code that is specific to the target platform or machine.
8All platform and machine specific code should be added to the platform and
9machine drivers respectively.
10
11Each codec class driver *must* provide the following features:-
12
131. Codec DAI and PCM configuration
142. Codec control IO - using RegMap API
153. Mixers and audio controls
164. Codec audio operations
175. DAPM description.
186. DAPM event handler.
19
20Optionally, codec drivers can also provide:-
21
227. DAC Digital mute control.
23
24Its probably best to use this guide in conjunction with the existing codec
25driver code in sound/soc/codecs/
26
27ASoC Codec driver breakdown
28===========================
29
30Codec DAI and PCM configuration
31-------------------------------
32Each codec driver must have a struct snd_soc_dai_driver to define its DAI and
33PCM capabilities and operations. This struct is exported so that it can be
34registered with the core by your machine driver.
35
36e.g.
37::
38
39  static struct snd_soc_dai_ops wm8731_dai_ops = {
40	.prepare	= wm8731_pcm_prepare,
41	.hw_params	= wm8731_hw_params,
42	.shutdown	= wm8731_shutdown,
43	.digital_mute	= wm8731_mute,
44	.set_sysclk	= wm8731_set_dai_sysclk,
45	.set_fmt	= wm8731_set_dai_fmt,
46  };
47
48  struct snd_soc_dai_driver wm8731_dai = {
49	.name = "wm8731-hifi",
50	.playback = {
51		.stream_name = "Playback",
52		.channels_min = 1,
53		.channels_max = 2,
54		.rates = WM8731_RATES,
55		.formats = WM8731_FORMATS,},
56	.capture = {
57		.stream_name = "Capture",
58		.channels_min = 1,
59		.channels_max = 2,
60		.rates = WM8731_RATES,
61		.formats = WM8731_FORMATS,},
62	.ops = &wm8731_dai_ops,
63	.symmetric_rates = 1,
64  };
65
66
67Codec control IO
68----------------
69The codec can usually be controlled via an I2C or SPI style interface
70(AC97 combines control with data in the DAI). The codec driver should use the
71Regmap API for all codec IO. Please see include/linux/regmap.h and existing
72codec drivers for example regmap usage.
73
74
75Mixers and audio controls
76-------------------------
77All the codec mixers and audio controls can be defined using the convenience
78macros defined in soc.h.
79::
80
81    #define SOC_SINGLE(xname, reg, shift, mask, invert)
82
83Defines a single control as follows:-
84::
85
86  xname = Control name e.g. "Playback Volume"
87  reg = codec register
88  shift = control bit(s) offset in register
89  mask = control bit size(s) e.g. mask of 7 = 3 bits
90  invert = the control is inverted
91
92Other macros include:-
93::
94
95    #define SOC_DOUBLE(xname, reg, shift_left, shift_right, mask, invert)
96
97A stereo control
98::
99
100    #define SOC_DOUBLE_R(xname, reg_left, reg_right, shift, mask, invert)
101
102A stereo control spanning 2 registers
103::
104
105    #define SOC_ENUM_SINGLE(xreg, xshift, xmask, xtexts)
106
107Defines an single enumerated control as follows:-
108::
109
110   xreg = register
111   xshift = control bit(s) offset in register
112   xmask = control bit(s) size
113   xtexts = pointer to array of strings that describe each setting
114
115   #define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xtexts)
116
117Defines a stereo enumerated control
118
119
120Codec Audio Operations
121----------------------
122The codec driver also supports the following ALSA PCM operations:-
123::
124
125  /* SoC audio ops */
126  struct snd_soc_ops {
127	int (*startup)(struct snd_pcm_substream *);
128	void (*shutdown)(struct snd_pcm_substream *);
129	int (*hw_params)(struct snd_pcm_substream *, struct snd_pcm_hw_params *);
130	int (*hw_free)(struct snd_pcm_substream *);
131	int (*prepare)(struct snd_pcm_substream *);
132  };
133
134Please refer to the ALSA driver PCM documentation for details.
135http://www.alsa-project.org/~iwai/writing-an-alsa-driver/
136
137
138DAPM description
139----------------
140The Dynamic Audio Power Management description describes the codec power
141components and their relationships and registers to the ASoC core.
142Please read dapm.rst for details of building the description.
143
144Please also see the examples in other codec drivers.
145
146
147DAPM event handler
148------------------
149This function is a callback that handles codec domain PM calls and system
150domain PM calls (e.g. suspend and resume). It is used to put the codec
151to sleep when not in use.
152
153Power states:-
154::
155
156	SNDRV_CTL_POWER_D0: /* full On */
157	/* vref/mid, clk and osc on, active */
158
159	SNDRV_CTL_POWER_D1: /* partial On */
160	SNDRV_CTL_POWER_D2: /* partial On */
161
162	SNDRV_CTL_POWER_D3hot: /* Off, with power */
163	/* everything off except vref/vmid, inactive */
164
165	SNDRV_CTL_POWER_D3cold: /* Everything Off, without power */
166
167
168Codec DAC digital mute control
169------------------------------
170Most codecs have a digital mute before the DACs that can be used to
171minimise any system noise.  The mute stops any digital data from
172entering the DAC.
173
174A callback can be created that is called by the core for each codec DAI
175when the mute is applied or freed.
176
177i.e.
178::
179
180  static int wm8974_mute(struct snd_soc_dai *dai, int mute)
181  {
182	struct snd_soc_component *component = dai->component;
183	u16 mute_reg = snd_soc_component_read32(component, WM8974_DAC) & 0xffbf;
184
185	if (mute)
186		snd_soc_component_write(component, WM8974_DAC, mute_reg | 0x40);
187	else
188		snd_soc_component_write(component, WM8974_DAC, mute_reg);
189	return 0;
190  }
191