summaryrefslogtreecommitdiff
path: root/include/sound/soc-dai.h
blob: ea07b4bd516189964b9b1abf9f0ff9624abab122 (plain)
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
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
/*
 * linux/sound/soc-dai.h -- ALSA SoC Layer
 *
 * Copyright:	2005-2008 Wolfson Microelectronics. PLC.
 *
 * This program is free software; you can redistribute it and/or modify
 * it under the terms of the GNU General Public License version 2 as
 * published by the Free Software Foundation.
 *
 * Digital Audio Interface (DAI) API.
 */

#ifndef __LINUX_SND_SOC_DAI_H
#define __LINUX_SND_SOC_DAI_H


#include <linux/list.h>

struct snd_pcm_substream;

/*
 * DAI hardware audio formats.
 *
 * Describes the physical PCM data formating and clocking. Add new formats
 * to the end.
 */
#define SND_SOC_DAIFMT_I2S		0 /* I2S mode */
#define SND_SOC_DAIFMT_RIGHT_J		1 /* Right Justified mode */
#define SND_SOC_DAIFMT_LEFT_J		2 /* Left Justified mode */
#define SND_SOC_DAIFMT_DSP_A		3 /* L data msb after FRM LRC */
#define SND_SOC_DAIFMT_DSP_B		4 /* L data msb during FRM LRC */
#define SND_SOC_DAIFMT_AC97		5 /* AC97 */

/* left and right justified also known as MSB and LSB respectively */
#define SND_SOC_DAIFMT_MSB		SND_SOC_DAIFMT_LEFT_J
#define SND_SOC_DAIFMT_LSB		SND_SOC_DAIFMT_RIGHT_J

/*
 * DAI Clock gating.
 *
 * DAI bit clocks can be be gated (disabled) when not the DAI is not
 * sending or receiving PCM data in a frame. This can be used to save power.
 */
#define SND_SOC_DAIFMT_CONT		(0 << 4) /* continuous clock */
#define SND_SOC_DAIFMT_GATED		(1 << 4) /* clock is gated */

/*
 * DAI Left/Right Clocks.
 *
 * Specifies whether the DAI can support different samples for similtanious
 * playback and capture. This usually requires a seperate physical frame
 * clock for playback and capture.
 */
#define SND_SOC_DAIFMT_SYNC		(0 << 5) /* Tx FRM = Rx FRM */
#define SND_SOC_DAIFMT_ASYNC		(1 << 5) /* Tx FRM ~ Rx FRM */

/*
 * TDM
 *
 * Time Division Multiplexing. Allows PCM data to be multplexed with other
 * data on the DAI.
 */
#define SND_SOC_DAIFMT_TDM		(1 << 6)

/*
 * DAI hardware signal inversions.
 *
 * Specifies whether the DAI can also support inverted clocks for the specified
 * format.
 */
#define SND_SOC_DAIFMT_NB_NF		(0 << 8) /* normal bit clock + frame */
#define SND_SOC_DAIFMT_NB_IF		(1 << 8) /* normal bclk + inv frm */
#define SND_SOC_DAIFMT_IB_NF		(2 << 8) /* invert bclk + nor frm */
#define SND_SOC_DAIFMT_IB_IF		(3 << 8) /* invert bclk + frm */

/*
 * DAI hardware clock masters.
 *
 * This is wrt the codec, the inverse is true for the interface
 * i.e. if the codec is clk and frm master then the interface is
 * clk and frame slave.
 */
#define SND_SOC_DAIFMT_CBM_CFM		(0 << 12) /* codec clk & frm master */
#define SND_SOC_DAIFMT_CBS_CFM		(1 << 12) /* codec clk slave & frm master */
#define SND_SOC_DAIFMT_CBM_CFS		(2 << 12) /* codec clk master & frame slave */
#define SND_SOC_DAIFMT_CBS_CFS		(3 << 12) /* codec clk & frm slave */

#define SND_SOC_DAIFMT_FORMAT_MASK	0x000f
#define SND_SOC_DAIFMT_CLOCK_MASK	0x00f0
#define SND_SOC_DAIFMT_INV_MASK		0x0f00
#define SND_SOC_DAIFMT_MASTER_MASK	0xf000

/*
 * Master Clock Directions
 */
#define SND_SOC_CLOCK_IN		0
#define SND_SOC_CLOCK_OUT		1

#define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S16_LE |\
                               SNDRV_PCM_FMTBIT_S32_LE)

struct snd_soc_dai_ops;
struct snd_soc_dai;
struct snd_ac97_bus_ops;

/* Digital Audio Interface registration */
int snd_soc_register_dai(struct snd_soc_dai *dai);
void snd_soc_unregister_dai(struct snd_soc_dai *dai);
int snd_soc_register_dais(struct snd_soc_dai *dai, size_t count);
void snd_soc_unregister_dais(struct snd_soc_dai *dai, size_t count);

/* Digital Audio Interface clocking API.*/
int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
	unsigned int freq, int dir);

int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai,
	int div_id, int div);

int snd_soc_dai_set_pll(struct snd_soc_dai *dai,
	int pll_id, unsigned int freq_in, unsigned int freq_out);

/* Digital Audio interface formatting */
int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt);

int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
	unsigned int mask, int slots);

int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate);

/* Digital Audio Interface mute */
int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute);

/*
 * Digital Audio Interface.
 *
 * Describes the Digital Audio Interface in terms of it's ALSA, DAI and AC97
 * operations an capabilities. Codec and platfom drivers will register a this
 * structure for every DAI they have.
 *
 * This structure covers the clocking, formating and ALSA operations for each
 * interface a
 */
struct snd_soc_dai_ops {
	/*
	 * DAI clocking configuration, all optional.
	 * Called by soc_card drivers, normally in their hw_params.
	 */
	int (*set_sysclk)(struct snd_soc_dai *dai,
		int clk_id, unsigned int freq, int dir);
	int (*set_pll)(struct snd_soc_dai *dai,
		int pll_id, unsigned int freq_in, unsigned int freq_out);
	int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div);

	/*
	 * DAI format configuration
	 * Called by soc_card drivers, normally in their hw_params.
	 */
	int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt);
	int (*set_tdm_slot)(struct snd_soc_dai *dai,
		unsigned int mask, int slots);
	int (*set_tristate)(struct snd_soc_dai *dai, int tristate);

	/*
	 * DAI digital mute - optional.
	 * Called by soc-core to minimise any pops.
	 */
	int (*digital_mute)(struct snd_soc_dai *dai, int mute);

	/*
	 * ALSA PCM audio operations - all optional.
	 * Called by soc-core during audio PCM operations.
	 */
	int (*startup)(struct snd_pcm_substream *,
		struct snd_soc_dai *);
	void (*shutdown)(struct snd_pcm_substream *,
		struct snd_soc_dai *);
	int (*hw_params)(struct snd_pcm_substream *,
		struct snd_pcm_hw_params *, struct snd_soc_dai *);
	int (*hw_free)(struct snd_pcm_substream *,
		struct snd_soc_dai *);
	int (*prepare)(struct snd_pcm_substream *,
		struct snd_soc_dai *);
	int (*trigger)(struct snd_pcm_substream *, int,
		struct snd_soc_dai *);
};

/*
 * Digital Audio Interface runtime data.
 *
 * Holds runtime data for a DAI.
 */
struct snd_soc_dai {
	/* DAI description */
	char *name;
	unsigned int id;
	int ac97_control;

	struct device *dev;

	/* DAI callbacks */
	int (*probe)(struct platform_device *pdev,
		     struct snd_soc_dai *dai);
	void (*remove)(struct platform_device *pdev,
		       struct snd_soc_dai *dai);
	int (*suspend)(struct snd_soc_dai *dai);
	int (*resume)(struct snd_soc_dai *dai);

	/* ops */
	struct snd_soc_dai_ops *ops;

	/* DAI capabilities */
	struct snd_soc_pcm_stream capture;
	struct snd_soc_pcm_stream playback;
	unsigned int symmetric_rates:1;

	/* DAI runtime info */
	struct snd_pcm_runtime *runtime;
	struct snd_soc_codec *codec;
	unsigned int active;
	unsigned char pop_wait:1;
	void *dma_data;

	/* DAI private data */
	void *private_data;

	/* parent codec/platform */
	union {
		struct snd_soc_codec *codec;
		struct snd_soc_platform *platform;
	};

	struct list_head list;
};

#endif