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1/*
2 * linux/sound/soc-dai.h -- ALSA SoC Layer
3 *
4 * Copyright: 2005-2008 Wolfson Microelectronics. PLC.
5 *
6 * This program is free software; you can redistribute it and/or modify
7 * it under the terms of the GNU General Public License version 2 as
8 * published by the Free Software Foundation.
9 *
10 * Digital Audio Interface (DAI) API.
11 */
12
13#ifndef __LINUX_SND_SOC_DAI_H
14#define __LINUX_SND_SOC_DAI_H
15
16
17#include <linux/list.h>
18
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19#include <sound/soc.h>
20
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21struct snd_pcm_substream;
22
23/*
24 * DAI hardware audio formats.
25 *
26 * Describes the physical PCM data formating and clocking. Add new formats
27 * to the end.
28 */
29#define SND_SOC_DAIFMT_I2S 0 /* I2S mode */
30#define SND_SOC_DAIFMT_RIGHT_J 1 /* Right Justified mode */
31#define SND_SOC_DAIFMT_LEFT_J 2 /* Left Justified mode */
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32#define SND_SOC_DAIFMT_DSP_A 3 /* L data MSB after FRM LRC */
33#define SND_SOC_DAIFMT_DSP_B 4 /* L data MSB during FRM LRC */
a47cbe72 34#define SND_SOC_DAIFMT_AC97 5 /* AC97 */
be2500b8 35#define SND_SOC_DAIFMT_PDM 6 /* Pulse density modulation */
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36
37/* left and right justified also known as MSB and LSB respectively */
38#define SND_SOC_DAIFMT_MSB SND_SOC_DAIFMT_LEFT_J
39#define SND_SOC_DAIFMT_LSB SND_SOC_DAIFMT_RIGHT_J
40
41/*
42 * DAI Clock gating.
43 *
47db8e89 44 * DAI bit clocks can be be gated (disabled) when the DAI is not
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45 * sending or receiving PCM data in a frame. This can be used to save power.
46 */
47#define SND_SOC_DAIFMT_CONT (0 << 4) /* continuous clock */
48#define SND_SOC_DAIFMT_GATED (1 << 4) /* clock is gated */
49
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50/*
51 * DAI hardware signal inversions.
52 *
53 * Specifies whether the DAI can also support inverted clocks for the specified
54 * format.
55 */
56#define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bit clock + frame */
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57#define SND_SOC_DAIFMT_NB_IF (1 << 8) /* normal BCLK + inv FRM */
58#define SND_SOC_DAIFMT_IB_NF (2 << 8) /* invert BCLK + nor FRM */
59#define SND_SOC_DAIFMT_IB_IF (3 << 8) /* invert BCLK + FRM */
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60
61/*
62 * DAI hardware clock masters.
63 *
64 * This is wrt the codec, the inverse is true for the interface
47db8e89 65 * i.e. if the codec is clk and FRM master then the interface is
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66 * clk and frame slave.
67 */
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68#define SND_SOC_DAIFMT_CBM_CFM (0 << 12) /* codec clk & FRM master */
69#define SND_SOC_DAIFMT_CBS_CFM (1 << 12) /* codec clk slave & FRM master */
a47cbe72 70#define SND_SOC_DAIFMT_CBM_CFS (2 << 12) /* codec clk master & frame slave */
47db8e89 71#define SND_SOC_DAIFMT_CBS_CFS (3 << 12) /* codec clk & FRM slave */
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72
73#define SND_SOC_DAIFMT_FORMAT_MASK 0x000f
74#define SND_SOC_DAIFMT_CLOCK_MASK 0x00f0
75#define SND_SOC_DAIFMT_INV_MASK 0x0f00
76#define SND_SOC_DAIFMT_MASTER_MASK 0xf000
77
78/*
79 * Master Clock Directions
80 */
81#define SND_SOC_CLOCK_IN 0
82#define SND_SOC_CLOCK_OUT 1
83
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84#define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S8 |\
85 SNDRV_PCM_FMTBIT_S16_LE |\
86 SNDRV_PCM_FMTBIT_S16_BE |\
87 SNDRV_PCM_FMTBIT_S20_3LE |\
88 SNDRV_PCM_FMTBIT_S20_3BE |\
89 SNDRV_PCM_FMTBIT_S24_3LE |\
90 SNDRV_PCM_FMTBIT_S24_3BE |\
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91 SNDRV_PCM_FMTBIT_S32_LE |\
92 SNDRV_PCM_FMTBIT_S32_BE)
33f503c9 93
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94struct snd_soc_dai_ops;
95struct snd_soc_dai;
96struct snd_ac97_bus_ops;
97
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98/* Digital Audio Interface registration */
99int snd_soc_register_dai(struct snd_soc_dai *dai);
100void snd_soc_unregister_dai(struct snd_soc_dai *dai);
101int snd_soc_register_dais(struct snd_soc_dai *dai, size_t count);
102void snd_soc_unregister_dais(struct snd_soc_dai *dai, size_t count);
103
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104/* Digital Audio Interface clocking API.*/
105int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
106 unsigned int freq, int dir);
107
108int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai,
109 int div_id, int div);
110
111int snd_soc_dai_set_pll(struct snd_soc_dai *dai,
85488037 112 int pll_id, int source, unsigned int freq_in, unsigned int freq_out);
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113
114/* Digital Audio interface formatting */
115int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt);
116
117int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
a5479e38 118 unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width);
a47cbe72 119
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120int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai,
121 unsigned int tx_num, unsigned int *tx_slot,
122 unsigned int rx_num, unsigned int *rx_slot);
123
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124int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate);
125
126/* Digital Audio Interface mute */
127int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute);
128
129/*
130 * Digital Audio Interface.
131 *
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132 * Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97
133 * operations and capabilities. Codec and platform drivers will register this
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134 * structure for every DAI they have.
135 *
136 * This structure covers the clocking, formating and ALSA operations for each
47db8e89 137 * interface.
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138 */
139struct snd_soc_dai_ops {
140 /*
141 * DAI clocking configuration, all optional.
142 * Called by soc_card drivers, normally in their hw_params.
143 */
144 int (*set_sysclk)(struct snd_soc_dai *dai,
145 int clk_id, unsigned int freq, int dir);
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146 int (*set_pll)(struct snd_soc_dai *dai, int pll_id, int source,
147 unsigned int freq_in, unsigned int freq_out);
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148 int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div);
149
150 /*
151 * DAI format configuration
152 * Called by soc_card drivers, normally in their hw_params.
153 */
154 int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt);
155 int (*set_tdm_slot)(struct snd_soc_dai *dai,
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156 unsigned int tx_mask, unsigned int rx_mask,
157 int slots, int slot_width);
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158 int (*set_channel_map)(struct snd_soc_dai *dai,
159 unsigned int tx_num, unsigned int *tx_slot,
160 unsigned int rx_num, unsigned int *rx_slot);
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161 int (*set_tristate)(struct snd_soc_dai *dai, int tristate);
162
163 /*
164 * DAI digital mute - optional.
165 * Called by soc-core to minimise any pops.
166 */
167 int (*digital_mute)(struct snd_soc_dai *dai, int mute);
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168
169 /*
170 * ALSA PCM audio operations - all optional.
171 * Called by soc-core during audio PCM operations.
172 */
173 int (*startup)(struct snd_pcm_substream *,
174 struct snd_soc_dai *);
175 void (*shutdown)(struct snd_pcm_substream *,
176 struct snd_soc_dai *);
177 int (*hw_params)(struct snd_pcm_substream *,
178 struct snd_pcm_hw_params *, struct snd_soc_dai *);
179 int (*hw_free)(struct snd_pcm_substream *,
180 struct snd_soc_dai *);
181 int (*prepare)(struct snd_pcm_substream *,
182 struct snd_soc_dai *);
183 int (*trigger)(struct snd_pcm_substream *, int,
184 struct snd_soc_dai *);
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185 /*
186 * For hardware based FIFO caused delay reporting.
187 * Optional.
188 */
189 snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *,
190 struct snd_soc_dai *);
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191};
192
193/*
194 * Digital Audio Interface runtime data.
195 *
196 * Holds runtime data for a DAI.
197 */
198struct snd_soc_dai {
199 /* DAI description */
200 char *name;
201 unsigned int id;
3ba9e10a 202 int ac97_control;
a47cbe72 203
9115171a 204 struct device *dev;
474828a4 205 void *ac97_pdata; /* platform_data for the ac97 codec */
9115171a 206
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207 /* DAI callbacks */
208 int (*probe)(struct platform_device *pdev,
209 struct snd_soc_dai *dai);
210 void (*remove)(struct platform_device *pdev,
211 struct snd_soc_dai *dai);
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212 int (*suspend)(struct snd_soc_dai *dai);
213 int (*resume)(struct snd_soc_dai *dai);
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214
215 /* ops */
6335d055 216 struct snd_soc_dai_ops *ops;
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217
218 /* DAI capabilities */
219 struct snd_soc_pcm_stream capture;
220 struct snd_soc_pcm_stream playback;
06f409d7 221 unsigned int symmetric_rates:1;
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222
223 /* DAI runtime info */
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224 struct snd_soc_codec *codec;
225 unsigned int active;
226 unsigned char pop_wait:1;
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227
228 /* DAI private data */
229 void *private_data;
230
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231 /* parent platform */
232 struct snd_soc_platform *platform;
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233
234 struct list_head list;
235};
236
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237static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai *dai,
238 const struct snd_pcm_substream *ss)
239{
240 return (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
241 dai->playback.dma_data : dai->capture.dma_data;
242}
243
244static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai,
245 const struct snd_pcm_substream *ss,
246 void *data)
247{
248 if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK)
249 dai->playback.dma_data = data;
250 else
251 dai->capture.dma_data = data;
252}
253
a47cbe72 254#endif