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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
19struct snd_pcm_substream;
20struct snd_soc_dapm_widget;
21struct snd_compr_stream;
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 1 /* I2S mode */
30#define SND_SOC_DAIFMT_RIGHT_J 2 /* Right Justified mode */
31#define SND_SOC_DAIFMT_LEFT_J 3 /* Left Justified mode */
32#define SND_SOC_DAIFMT_DSP_A 4 /* L data MSB after FRM LRC */
33#define SND_SOC_DAIFMT_DSP_B 5 /* L data MSB during FRM LRC */
34#define SND_SOC_DAIFMT_AC97 6 /* AC97 */
35#define SND_SOC_DAIFMT_PDM 7 /* Pulse density modulation */
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 *
44 * DAI bit clocks can be be gated (disabled) when the DAI is not
45 * sending or receiving PCM data in a frame. This can be used to save power.
46 */
47#define SND_SOC_DAIFMT_CONT (1 << 4) /* continuous clock */
48#define SND_SOC_DAIFMT_GATED (0 << 4) /* clock is gated */
49
50/*
51 * DAI hardware signal polarity.
52 *
53 * Specifies whether the DAI can also support inverted clocks for the specified
54 * format.
55 *
56 * BCLK:
57 * - "normal" polarity means signal is available at rising edge of BCLK
58 * - "inverted" polarity means signal is available at falling edge of BCLK
59 *
60 * FSYNC "normal" polarity depends on the frame format:
61 * - I2S: frame consists of left then right channel data. Left channel starts
62 * with falling FSYNC edge, right channel starts with rising FSYNC edge.
63 * - Left/Right Justified: frame consists of left then right channel data.
64 * Left channel starts with rising FSYNC edge, right channel starts with
65 * falling FSYNC edge.
66 * - DSP A/B: Frame starts with rising FSYNC edge.
67 * - AC97: Frame starts with rising FSYNC edge.
68 *
69 * "Negative" FSYNC polarity is the one opposite of "normal" polarity.
70 */
71#define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bit clock + frame */
72#define SND_SOC_DAIFMT_NB_IF (2 << 8) /* normal BCLK + inv FRM */
73#define SND_SOC_DAIFMT_IB_NF (3 << 8) /* invert BCLK + nor FRM */
74#define SND_SOC_DAIFMT_IB_IF (4 << 8) /* invert BCLK + FRM */
75
76/*
77 * DAI hardware clock masters.
78 *
79 * This is wrt the codec, the inverse is true for the interface
80 * i.e. if the codec is clk and FRM master then the interface is
81 * clk and frame slave.
82 */
83#define SND_SOC_DAIFMT_CBM_CFM (1 << 12) /* codec clk & FRM master */
84#define SND_SOC_DAIFMT_CBS_CFM (2 << 12) /* codec clk slave & FRM master */
85#define SND_SOC_DAIFMT_CBM_CFS (3 << 12) /* codec clk master & frame slave */
86#define SND_SOC_DAIFMT_CBS_CFS (4 << 12) /* codec clk & FRM slave */
87
88#define SND_SOC_DAIFMT_FORMAT_MASK 0x000f
89#define SND_SOC_DAIFMT_CLOCK_MASK 0x00f0
90#define SND_SOC_DAIFMT_INV_MASK 0x0f00
91#define SND_SOC_DAIFMT_MASTER_MASK 0xf000
92
93/*
94 * Master Clock Directions
95 */
96#define SND_SOC_CLOCK_IN 0
97#define SND_SOC_CLOCK_OUT 1
98
99#define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S8 |\
100 SNDRV_PCM_FMTBIT_S16_LE |\
101 SNDRV_PCM_FMTBIT_S16_BE |\
102 SNDRV_PCM_FMTBIT_S20_3LE |\
103 SNDRV_PCM_FMTBIT_S20_3BE |\
104 SNDRV_PCM_FMTBIT_S24_3LE |\
105 SNDRV_PCM_FMTBIT_S24_3BE |\
106 SNDRV_PCM_FMTBIT_S32_LE |\
107 SNDRV_PCM_FMTBIT_S32_BE)
108
109struct snd_soc_dai_driver;
110struct snd_soc_dai;
111struct snd_ac97_bus_ops;
112
113/* Digital Audio Interface clocking API.*/
114int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
115 unsigned int freq, int dir);
116
117int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai,
118 int div_id, int div);
119
120int snd_soc_dai_set_pll(struct snd_soc_dai *dai,
121 int pll_id, int source, unsigned int freq_in, unsigned int freq_out);
122
123int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio);
124
125/* Digital Audio interface formatting */
126int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt);
127
128int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
129 unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width);
130
131int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai,
132 unsigned int tx_num, unsigned int *tx_slot,
133 unsigned int rx_num, unsigned int *rx_slot);
134
135int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate);
136
137/* Digital Audio Interface mute */
138int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute,
139 int direction);
140
141int snd_soc_dai_is_dummy(struct snd_soc_dai *dai);
142
143struct snd_soc_dai_ops {
144 /*
145 * DAI clocking configuration, all optional.
146 * Called by soc_card drivers, normally in their hw_params.
147 */
148 int (*set_sysclk)(struct snd_soc_dai *dai,
149 int clk_id, unsigned int freq, int dir);
150 int (*set_pll)(struct snd_soc_dai *dai, int pll_id, int source,
151 unsigned int freq_in, unsigned int freq_out);
152 int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div);
153 int (*set_bclk_ratio)(struct snd_soc_dai *dai, unsigned int ratio);
154
155 /*
156 * DAI format configuration
157 * Called by soc_card drivers, normally in their hw_params.
158 */
159 int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt);
160 int (*xlate_tdm_slot_mask)(unsigned int slots,
161 unsigned int *tx_mask, unsigned int *rx_mask);
162 int (*set_tdm_slot)(struct snd_soc_dai *dai,
163 unsigned int tx_mask, unsigned int rx_mask,
164 int slots, int slot_width);
165 int (*set_channel_map)(struct snd_soc_dai *dai,
166 unsigned int tx_num, unsigned int *tx_slot,
167 unsigned int rx_num, unsigned int *rx_slot);
168 int (*set_tristate)(struct snd_soc_dai *dai, int tristate);
169
170 /*
171 * DAI digital mute - optional.
172 * Called by soc-core to minimise any pops.
173 */
174 int (*digital_mute)(struct snd_soc_dai *dai, int mute);
175 int (*mute_stream)(struct snd_soc_dai *dai, int mute, int stream);
176
177 /*
178 * ALSA PCM audio operations - all optional.
179 * Called by soc-core during audio PCM operations.
180 */
181 int (*startup)(struct snd_pcm_substream *,
182 struct snd_soc_dai *);
183 void (*shutdown)(struct snd_pcm_substream *,
184 struct snd_soc_dai *);
185 int (*hw_params)(struct snd_pcm_substream *,
186 struct snd_pcm_hw_params *, struct snd_soc_dai *);
187 int (*hw_free)(struct snd_pcm_substream *,
188 struct snd_soc_dai *);
189 int (*prepare)(struct snd_pcm_substream *,
190 struct snd_soc_dai *);
191 /*
192 * NOTE: Commands passed to the trigger function are not necessarily
193 * compatible with the current state of the dai. For example this
194 * sequence of commands is possible: START STOP STOP.
195 * So do not unconditionally use refcounting functions in the trigger
196 * function, e.g. clk_enable/disable.
197 */
198 int (*trigger)(struct snd_pcm_substream *, int,
199 struct snd_soc_dai *);
200 int (*bespoke_trigger)(struct snd_pcm_substream *, int,
201 struct snd_soc_dai *);
202 /*
203 * For hardware based FIFO caused delay reporting.
204 * Optional.
205 */
206 snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *,
207 struct snd_soc_dai *);
208};
209
210/*
211 * Digital Audio Interface Driver.
212 *
213 * Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97
214 * operations and capabilities. Codec and platform drivers will register this
215 * structure for every DAI they have.
216 *
217 * This structure covers the clocking, formating and ALSA operations for each
218 * interface.
219 */
220struct snd_soc_dai_driver {
221 /* DAI description */
222 const char *name;
223 unsigned int id;
224 unsigned int base;
225 struct snd_soc_dobj dobj;
226
227 /* DAI driver callbacks */
228 int (*probe)(struct snd_soc_dai *dai);
229 int (*remove)(struct snd_soc_dai *dai);
230 int (*suspend)(struct snd_soc_dai *dai);
231 int (*resume)(struct snd_soc_dai *dai);
232 /* compress dai */
233 int (*compress_new)(struct snd_soc_pcm_runtime *rtd, int num);
234 /* DAI is also used for the control bus */
235 bool bus_control;
236
237 /* ops */
238 const struct snd_soc_dai_ops *ops;
239
240 /* DAI capabilities */
241 struct snd_soc_pcm_stream capture;
242 struct snd_soc_pcm_stream playback;
243 unsigned int symmetric_rates:1;
244 unsigned int symmetric_channels:1;
245 unsigned int symmetric_samplebits:1;
246
247 /* probe ordering - for components with runtime dependencies */
248 int probe_order;
249 int remove_order;
250};
251
252/*
253 * Digital Audio Interface runtime data.
254 *
255 * Holds runtime data for a DAI.
256 */
257struct snd_soc_dai {
258 const char *name;
259 int id;
260 struct device *dev;
261
262 /* driver ops */
263 struct snd_soc_dai_driver *driver;
264
265 /* DAI runtime info */
266 unsigned int capture_active:1; /* stream is in use */
267 unsigned int playback_active:1; /* stream is in use */
268 unsigned int symmetric_rates:1;
269 unsigned int symmetric_channels:1;
270 unsigned int symmetric_samplebits:1;
271 unsigned int active;
272 unsigned char probed:1;
273
274 struct snd_soc_dapm_widget *playback_widget;
275 struct snd_soc_dapm_widget *capture_widget;
276
277 /* DAI DMA data */
278 void *playback_dma_data;
279 void *capture_dma_data;
280
281 /* Symmetry data - only valid if symmetry is being enforced */
282 unsigned int rate;
283 unsigned int channels;
284 unsigned int sample_bits;
285
286 /* parent platform/codec */
287 struct snd_soc_codec *codec;
288 struct snd_soc_component *component;
289
290 /* CODEC TDM slot masks and params (for fixup) */
291 unsigned int tx_mask;
292 unsigned int rx_mask;
293
294 struct list_head list;
295};
296
297static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai *dai,
298 const struct snd_pcm_substream *ss)
299{
300 return (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
301 dai->playback_dma_data : dai->capture_dma_data;
302}
303
304static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai,
305 const struct snd_pcm_substream *ss,
306 void *data)
307{
308 if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK)
309 dai->playback_dma_data = data;
310 else
311 dai->capture_dma_data = data;
312}
313
314static inline void snd_soc_dai_init_dma_data(struct snd_soc_dai *dai,
315 void *playback, void *capture)
316{
317 dai->playback_dma_data = playback;
318 dai->capture_dma_data = capture;
319}
320
321static inline void snd_soc_dai_set_drvdata(struct snd_soc_dai *dai,
322 void *data)
323{
324 dev_set_drvdata(dai->dev, data);
325}
326
327static inline void *snd_soc_dai_get_drvdata(struct snd_soc_dai *dai)
328{
329 return dev_get_drvdata(dai->dev);
330}
331
332#endif