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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
19struct snd_pcm_substream;
20
21/*
22 * DAI hardware audio formats.
23 *
24 * Describes the physical PCM data formating and clocking. Add new formats
25 * to the end.
26 */
27#define SND_SOC_DAIFMT_I2S 0 /* I2S mode */
28#define SND_SOC_DAIFMT_RIGHT_J 1 /* Right Justified mode */
29#define SND_SOC_DAIFMT_LEFT_J 2 /* Left Justified mode */
30#define SND_SOC_DAIFMT_DSP_A 3 /* L data MSB after FRM LRC */
31#define SND_SOC_DAIFMT_DSP_B 4 /* L data MSB during FRM LRC */
32#define SND_SOC_DAIFMT_AC97 5 /* AC97 */
33#define SND_SOC_DAIFMT_PDM 6 /* Pulse density modulation */
34
35/* left and right justified also known as MSB and LSB respectively */
36#define SND_SOC_DAIFMT_MSB SND_SOC_DAIFMT_LEFT_J
37#define SND_SOC_DAIFMT_LSB SND_SOC_DAIFMT_RIGHT_J
38
39/*
40 * DAI Clock gating.
41 *
42 * DAI bit clocks can be be gated (disabled) when the DAI is not
43 * sending or receiving PCM data in a frame. This can be used to save power.
44 */
45#define SND_SOC_DAIFMT_CONT (0 << 4) /* continuous clock */
46#define SND_SOC_DAIFMT_GATED (1 << 4) /* clock is gated */
47
48/*
49 * DAI hardware signal inversions.
50 *
51 * Specifies whether the DAI can also support inverted clocks for the specified
52 * format.
53 */
54#define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bit clock + frame */
55#define SND_SOC_DAIFMT_NB_IF (1 << 8) /* normal BCLK + inv FRM */
56#define SND_SOC_DAIFMT_IB_NF (2 << 8) /* invert BCLK + nor FRM */
57#define SND_SOC_DAIFMT_IB_IF (3 << 8) /* invert BCLK + FRM */
58
59/*
60 * DAI hardware clock masters.
61 *
62 * This is wrt the codec, the inverse is true for the interface
63 * i.e. if the codec is clk and FRM master then the interface is
64 * clk and frame slave.
65 */
66#define SND_SOC_DAIFMT_CBM_CFM (0 << 12) /* codec clk & FRM master */
67#define SND_SOC_DAIFMT_CBS_CFM (1 << 12) /* codec clk slave & FRM master */
68#define SND_SOC_DAIFMT_CBM_CFS (2 << 12) /* codec clk master & frame slave */
69#define SND_SOC_DAIFMT_CBS_CFS (3 << 12) /* codec clk & FRM slave */
70
71#define SND_SOC_DAIFMT_FORMAT_MASK 0x000f
72#define SND_SOC_DAIFMT_CLOCK_MASK 0x00f0
73#define SND_SOC_DAIFMT_INV_MASK 0x0f00
74#define SND_SOC_DAIFMT_MASTER_MASK 0xf000
75
76/*
77 * Master Clock Directions
78 */
79#define SND_SOC_CLOCK_IN 0
80#define SND_SOC_CLOCK_OUT 1
81
82#define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S8 |\
83 SNDRV_PCM_FMTBIT_S16_LE |\
84 SNDRV_PCM_FMTBIT_S16_BE |\
85 SNDRV_PCM_FMTBIT_S20_3LE |\
86 SNDRV_PCM_FMTBIT_S20_3BE |\
87 SNDRV_PCM_FMTBIT_S24_3LE |\
88 SNDRV_PCM_FMTBIT_S24_3BE |\
89 SNDRV_PCM_FMTBIT_S32_LE |\
90 SNDRV_PCM_FMTBIT_S32_BE)
91
92struct snd_soc_dai_driver;
93struct snd_soc_dai;
94struct snd_ac97_bus_ops;
95
96/* Digital Audio Interface registration */
97int snd_soc_register_dai(struct device *dev,
98 struct snd_soc_dai_driver *dai_drv);
99void snd_soc_unregister_dai(struct device *dev);
100int snd_soc_register_dais(struct device *dev,
101 struct snd_soc_dai_driver *dai_drv, size_t count);
102void snd_soc_unregister_dais(struct device *dev, size_t count);
103
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,
112 int pll_id, int source, unsigned int freq_in, unsigned int freq_out);
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,
118 unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width);
119
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
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
129struct snd_soc_dai_ops {
130 /*
131 * DAI clocking configuration, all optional.
132 * Called by soc_card drivers, normally in their hw_params.
133 */
134 int (*set_sysclk)(struct snd_soc_dai *dai,
135 int clk_id, unsigned int freq, int dir);
136 int (*set_pll)(struct snd_soc_dai *dai, int pll_id, int source,
137 unsigned int freq_in, unsigned int freq_out);
138 int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div);
139
140 /*
141 * DAI format configuration
142 * Called by soc_card drivers, normally in their hw_params.
143 */
144 int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt);
145 int (*set_tdm_slot)(struct snd_soc_dai *dai,
146 unsigned int tx_mask, unsigned int rx_mask,
147 int slots, int slot_width);
148 int (*set_channel_map)(struct snd_soc_dai *dai,
149 unsigned int tx_num, unsigned int *tx_slot,
150 unsigned int rx_num, unsigned int *rx_slot);
151 int (*set_tristate)(struct snd_soc_dai *dai, int tristate);
152
153 /*
154 * DAI digital mute - optional.
155 * Called by soc-core to minimise any pops.
156 */
157 int (*digital_mute)(struct snd_soc_dai *dai, int mute);
158
159 /*
160 * ALSA PCM audio operations - all optional.
161 * Called by soc-core during audio PCM operations.
162 */
163 int (*startup)(struct snd_pcm_substream *,
164 struct snd_soc_dai *);
165 void (*shutdown)(struct snd_pcm_substream *,
166 struct snd_soc_dai *);
167 int (*hw_params)(struct snd_pcm_substream *,
168 struct snd_pcm_hw_params *, struct snd_soc_dai *);
169 int (*hw_free)(struct snd_pcm_substream *,
170 struct snd_soc_dai *);
171 int (*prepare)(struct snd_pcm_substream *,
172 struct snd_soc_dai *);
173 int (*trigger)(struct snd_pcm_substream *, int,
174 struct snd_soc_dai *);
175 /*
176 * For hardware based FIFO caused delay reporting.
177 * Optional.
178 */
179 snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *,
180 struct snd_soc_dai *);
181};
182
183/*
184 * Digital Audio Interface Driver.
185 *
186 * Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97
187 * operations and capabilities. Codec and platform drivers will register this
188 * structure for every DAI they have.
189 *
190 * This structure covers the clocking, formating and ALSA operations for each
191 * interface.
192 */
193struct snd_soc_dai_driver {
194 /* DAI description */
195 const char *name;
196 unsigned int id;
197 int ac97_control;
198
199 /* DAI driver callbacks */
200 int (*probe)(struct snd_soc_dai *dai);
201 int (*remove)(struct snd_soc_dai *dai);
202 int (*suspend)(struct snd_soc_dai *dai);
203 int (*resume)(struct snd_soc_dai *dai);
204
205 /* ops */
206 const struct snd_soc_dai_ops *ops;
207
208 /* DAI capabilities */
209 struct snd_soc_pcm_stream capture;
210 struct snd_soc_pcm_stream playback;
211 unsigned int symmetric_rates:1;
212
213 /* probe ordering - for components with runtime dependencies */
214 int probe_order;
215 int remove_order;
216};
217
218/*
219 * Digital Audio Interface runtime data.
220 *
221 * Holds runtime data for a DAI.
222 */
223struct snd_soc_dai {
224 const char *name;
225 int id;
226 struct device *dev;
227 void *ac97_pdata; /* platform_data for the ac97 codec */
228
229 /* driver ops */
230 struct snd_soc_dai_driver *driver;
231
232 /* DAI runtime info */
233 unsigned int capture_active:1; /* stream is in use */
234 unsigned int playback_active:1; /* stream is in use */
235 unsigned int symmetric_rates:1;
236 struct snd_pcm_runtime *runtime;
237 unsigned int active;
238 unsigned char pop_wait:1;
239 unsigned char probed:1;
240
241 /* DAI DMA data */
242 void *playback_dma_data;
243 void *capture_dma_data;
244
245 /* parent platform/codec */
246 union {
247 struct snd_soc_platform *platform;
248 struct snd_soc_codec *codec;
249 };
250 struct snd_soc_card *card;
251
252 struct list_head list;
253 struct list_head card_list;
254};
255
256static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai *dai,
257 const struct snd_pcm_substream *ss)
258{
259 return (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
260 dai->playback_dma_data : dai->capture_dma_data;
261}
262
263static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai,
264 const struct snd_pcm_substream *ss,
265 void *data)
266{
267 if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK)
268 dai->playback_dma_data = data;
269 else
270 dai->capture_dma_data = data;
271}
272
273static inline void snd_soc_dai_set_drvdata(struct snd_soc_dai *dai,
274 void *data)
275{
276 dev_set_drvdata(dai->dev, data);
277}
278
279static inline void *snd_soc_dai_get_drvdata(struct snd_soc_dai *dai)
280{
281 return dev_get_drvdata(dai->dev);
282}
283
284#endif
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#include <sound/asoc.h>
19
20struct snd_pcm_substream;
21struct snd_soc_dapm_widget;
22struct snd_compr_stream;
23
24/*
25 * DAI hardware audio formats.
26 *
27 * Describes the physical PCM data formating and clocking. Add new formats
28 * to the end.
29 */
30#define SND_SOC_DAIFMT_I2S SND_SOC_DAI_FORMAT_I2S
31#define SND_SOC_DAIFMT_RIGHT_J SND_SOC_DAI_FORMAT_RIGHT_J
32#define SND_SOC_DAIFMT_LEFT_J SND_SOC_DAI_FORMAT_LEFT_J
33#define SND_SOC_DAIFMT_DSP_A SND_SOC_DAI_FORMAT_DSP_A
34#define SND_SOC_DAIFMT_DSP_B SND_SOC_DAI_FORMAT_DSP_B
35#define SND_SOC_DAIFMT_AC97 SND_SOC_DAI_FORMAT_AC97
36#define SND_SOC_DAIFMT_PDM SND_SOC_DAI_FORMAT_PDM
37
38/* left and right justified also known as MSB and LSB respectively */
39#define SND_SOC_DAIFMT_MSB SND_SOC_DAIFMT_LEFT_J
40#define SND_SOC_DAIFMT_LSB SND_SOC_DAIFMT_RIGHT_J
41
42/*
43 * DAI Clock gating.
44 *
45 * DAI bit clocks can be be gated (disabled) when the DAI is not
46 * sending or receiving PCM data in a frame. This can be used to save power.
47 */
48#define SND_SOC_DAIFMT_CONT (1 << 4) /* continuous clock */
49#define SND_SOC_DAIFMT_GATED (0 << 4) /* clock is gated */
50
51/*
52 * DAI hardware signal polarity.
53 *
54 * Specifies whether the DAI can also support inverted clocks for the specified
55 * format.
56 *
57 * BCLK:
58 * - "normal" polarity means signal is available at rising edge of BCLK
59 * - "inverted" polarity means signal is available at falling edge of BCLK
60 *
61 * FSYNC "normal" polarity depends on the frame format:
62 * - I2S: frame consists of left then right channel data. Left channel starts
63 * with falling FSYNC edge, right channel starts with rising FSYNC edge.
64 * - Left/Right Justified: frame consists of left then right channel data.
65 * Left channel starts with rising FSYNC edge, right channel starts with
66 * falling FSYNC edge.
67 * - DSP A/B: Frame starts with rising FSYNC edge.
68 * - AC97: Frame starts with rising FSYNC edge.
69 *
70 * "Negative" FSYNC polarity is the one opposite of "normal" polarity.
71 */
72#define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bit clock + frame */
73#define SND_SOC_DAIFMT_NB_IF (2 << 8) /* normal BCLK + inv FRM */
74#define SND_SOC_DAIFMT_IB_NF (3 << 8) /* invert BCLK + nor FRM */
75#define SND_SOC_DAIFMT_IB_IF (4 << 8) /* invert BCLK + FRM */
76
77/*
78 * DAI hardware clock masters.
79 *
80 * This is wrt the codec, the inverse is true for the interface
81 * i.e. if the codec is clk and FRM master then the interface is
82 * clk and frame slave.
83 */
84#define SND_SOC_DAIFMT_CBM_CFM (1 << 12) /* codec clk & FRM master */
85#define SND_SOC_DAIFMT_CBS_CFM (2 << 12) /* codec clk slave & FRM master */
86#define SND_SOC_DAIFMT_CBM_CFS (3 << 12) /* codec clk master & frame slave */
87#define SND_SOC_DAIFMT_CBS_CFS (4 << 12) /* codec clk & FRM slave */
88
89#define SND_SOC_DAIFMT_FORMAT_MASK 0x000f
90#define SND_SOC_DAIFMT_CLOCK_MASK 0x00f0
91#define SND_SOC_DAIFMT_INV_MASK 0x0f00
92#define SND_SOC_DAIFMT_MASTER_MASK 0xf000
93
94/*
95 * Master Clock Directions
96 */
97#define SND_SOC_CLOCK_IN 0
98#define SND_SOC_CLOCK_OUT 1
99
100#define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S8 |\
101 SNDRV_PCM_FMTBIT_S16_LE |\
102 SNDRV_PCM_FMTBIT_S16_BE |\
103 SNDRV_PCM_FMTBIT_S20_3LE |\
104 SNDRV_PCM_FMTBIT_S20_3BE |\
105 SNDRV_PCM_FMTBIT_S20_LE |\
106 SNDRV_PCM_FMTBIT_S20_BE |\
107 SNDRV_PCM_FMTBIT_S24_3LE |\
108 SNDRV_PCM_FMTBIT_S24_3BE |\
109 SNDRV_PCM_FMTBIT_S32_LE |\
110 SNDRV_PCM_FMTBIT_S32_BE)
111
112struct snd_soc_dai_driver;
113struct snd_soc_dai;
114struct snd_ac97_bus_ops;
115
116/* Digital Audio Interface clocking API.*/
117int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
118 unsigned int freq, int dir);
119
120int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai,
121 int div_id, int div);
122
123int snd_soc_dai_set_pll(struct snd_soc_dai *dai,
124 int pll_id, int source, unsigned int freq_in, unsigned int freq_out);
125
126int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio);
127
128/* Digital Audio interface formatting */
129int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt);
130
131int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
132 unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width);
133
134int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai,
135 unsigned int tx_num, unsigned int *tx_slot,
136 unsigned int rx_num, unsigned int *rx_slot);
137
138int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate);
139
140/* Digital Audio Interface mute */
141int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute,
142 int direction);
143
144int snd_soc_dai_is_dummy(struct snd_soc_dai *dai);
145
146struct snd_soc_dai_ops {
147 /*
148 * DAI clocking configuration, all optional.
149 * Called by soc_card drivers, normally in their hw_params.
150 */
151 int (*set_sysclk)(struct snd_soc_dai *dai,
152 int clk_id, unsigned int freq, int dir);
153 int (*set_pll)(struct snd_soc_dai *dai, int pll_id, int source,
154 unsigned int freq_in, unsigned int freq_out);
155 int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div);
156 int (*set_bclk_ratio)(struct snd_soc_dai *dai, unsigned int ratio);
157
158 /*
159 * DAI format configuration
160 * Called by soc_card drivers, normally in their hw_params.
161 */
162 int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt);
163 int (*xlate_tdm_slot_mask)(unsigned int slots,
164 unsigned int *tx_mask, unsigned int *rx_mask);
165 int (*set_tdm_slot)(struct snd_soc_dai *dai,
166 unsigned int tx_mask, unsigned int rx_mask,
167 int slots, int slot_width);
168 int (*set_channel_map)(struct snd_soc_dai *dai,
169 unsigned int tx_num, unsigned int *tx_slot,
170 unsigned int rx_num, unsigned int *rx_slot);
171 int (*set_tristate)(struct snd_soc_dai *dai, int tristate);
172
173 /*
174 * DAI digital mute - optional.
175 * Called by soc-core to minimise any pops.
176 */
177 int (*digital_mute)(struct snd_soc_dai *dai, int mute);
178 int (*mute_stream)(struct snd_soc_dai *dai, int mute, int stream);
179
180 /*
181 * ALSA PCM audio operations - all optional.
182 * Called by soc-core during audio PCM operations.
183 */
184 int (*startup)(struct snd_pcm_substream *,
185 struct snd_soc_dai *);
186 void (*shutdown)(struct snd_pcm_substream *,
187 struct snd_soc_dai *);
188 int (*hw_params)(struct snd_pcm_substream *,
189 struct snd_pcm_hw_params *, struct snd_soc_dai *);
190 int (*hw_free)(struct snd_pcm_substream *,
191 struct snd_soc_dai *);
192 int (*prepare)(struct snd_pcm_substream *,
193 struct snd_soc_dai *);
194 /*
195 * NOTE: Commands passed to the trigger function are not necessarily
196 * compatible with the current state of the dai. For example this
197 * sequence of commands is possible: START STOP STOP.
198 * So do not unconditionally use refcounting functions in the trigger
199 * function, e.g. clk_enable/disable.
200 */
201 int (*trigger)(struct snd_pcm_substream *, int,
202 struct snd_soc_dai *);
203 int (*bespoke_trigger)(struct snd_pcm_substream *, int,
204 struct snd_soc_dai *);
205 /*
206 * For hardware based FIFO caused delay reporting.
207 * Optional.
208 */
209 snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *,
210 struct snd_soc_dai *);
211};
212
213struct snd_soc_cdai_ops {
214 /*
215 * for compress ops
216 */
217 int (*startup)(struct snd_compr_stream *,
218 struct snd_soc_dai *);
219 int (*shutdown)(struct snd_compr_stream *,
220 struct snd_soc_dai *);
221 int (*set_params)(struct snd_compr_stream *,
222 struct snd_compr_params *, struct snd_soc_dai *);
223 int (*get_params)(struct snd_compr_stream *,
224 struct snd_codec *, struct snd_soc_dai *);
225 int (*set_metadata)(struct snd_compr_stream *,
226 struct snd_compr_metadata *, struct snd_soc_dai *);
227 int (*get_metadata)(struct snd_compr_stream *,
228 struct snd_compr_metadata *, struct snd_soc_dai *);
229 int (*trigger)(struct snd_compr_stream *, int,
230 struct snd_soc_dai *);
231 int (*pointer)(struct snd_compr_stream *,
232 struct snd_compr_tstamp *, struct snd_soc_dai *);
233 int (*ack)(struct snd_compr_stream *, size_t,
234 struct snd_soc_dai *);
235};
236
237/*
238 * Digital Audio Interface Driver.
239 *
240 * Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97
241 * operations and capabilities. Codec and platform drivers will register this
242 * structure for every DAI they have.
243 *
244 * This structure covers the clocking, formating and ALSA operations for each
245 * interface.
246 */
247struct snd_soc_dai_driver {
248 /* DAI description */
249 const char *name;
250 unsigned int id;
251 unsigned int base;
252 struct snd_soc_dobj dobj;
253
254 /* DAI driver callbacks */
255 int (*probe)(struct snd_soc_dai *dai);
256 int (*remove)(struct snd_soc_dai *dai);
257 int (*suspend)(struct snd_soc_dai *dai);
258 int (*resume)(struct snd_soc_dai *dai);
259 /* compress dai */
260 int (*compress_new)(struct snd_soc_pcm_runtime *rtd, int num);
261 /* Optional Callback used at pcm creation*/
262 int (*pcm_new)(struct snd_soc_pcm_runtime *rtd,
263 struct snd_soc_dai *dai);
264 /* DAI is also used for the control bus */
265 bool bus_control;
266
267 /* ops */
268 const struct snd_soc_dai_ops *ops;
269 const struct snd_soc_cdai_ops *cops;
270
271 /* DAI capabilities */
272 struct snd_soc_pcm_stream capture;
273 struct snd_soc_pcm_stream playback;
274 unsigned int symmetric_rates:1;
275 unsigned int symmetric_channels:1;
276 unsigned int symmetric_samplebits:1;
277
278 /* probe ordering - for components with runtime dependencies */
279 int probe_order;
280 int remove_order;
281};
282
283/*
284 * Digital Audio Interface runtime data.
285 *
286 * Holds runtime data for a DAI.
287 */
288struct snd_soc_dai {
289 const char *name;
290 int id;
291 struct device *dev;
292
293 /* driver ops */
294 struct snd_soc_dai_driver *driver;
295
296 /* DAI runtime info */
297 unsigned int capture_active:1; /* stream is in use */
298 unsigned int playback_active:1; /* stream is in use */
299 unsigned int probed:1;
300
301 unsigned int active;
302
303 struct snd_soc_dapm_widget *playback_widget;
304 struct snd_soc_dapm_widget *capture_widget;
305
306 /* DAI DMA data */
307 void *playback_dma_data;
308 void *capture_dma_data;
309
310 /* Symmetry data - only valid if symmetry is being enforced */
311 unsigned int rate;
312 unsigned int channels;
313 unsigned int sample_bits;
314
315 /* parent platform/codec */
316 struct snd_soc_codec *codec;
317 struct snd_soc_component *component;
318
319 /* CODEC TDM slot masks and params (for fixup) */
320 unsigned int tx_mask;
321 unsigned int rx_mask;
322
323 struct list_head list;
324};
325
326static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai *dai,
327 const struct snd_pcm_substream *ss)
328{
329 return (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
330 dai->playback_dma_data : dai->capture_dma_data;
331}
332
333static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai,
334 const struct snd_pcm_substream *ss,
335 void *data)
336{
337 if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK)
338 dai->playback_dma_data = data;
339 else
340 dai->capture_dma_data = data;
341}
342
343static inline void snd_soc_dai_init_dma_data(struct snd_soc_dai *dai,
344 void *playback, void *capture)
345{
346 dai->playback_dma_data = playback;
347 dai->capture_dma_data = capture;
348}
349
350static inline void snd_soc_dai_set_drvdata(struct snd_soc_dai *dai,
351 void *data)
352{
353 dev_set_drvdata(dai->dev, data);
354}
355
356static inline void *snd_soc_dai_get_drvdata(struct snd_soc_dai *dai)
357{
358 return dev_get_drvdata(dai->dev);
359}
360
361#endif