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a47cbe72 MB |
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> | |
d2e3cb85 | 18 | #include <sound/asoc.h> |
a47cbe72 MB |
19 | |
20 | struct snd_pcm_substream; | |
888df395 | 21 | struct snd_soc_dapm_widget; |
49681077 | 22 | struct snd_compr_stream; |
a47cbe72 MB |
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 | */ | |
d2e3cb85 ML |
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 | |
a47cbe72 MB |
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 | * | |
47db8e89 | 45 | * DAI bit clocks can be be gated (disabled) when the DAI is not |
a47cbe72 MB |
46 | * sending or receiving PCM data in a frame. This can be used to save power. |
47 | */ | |
75d9ac46 | 48 | #define SND_SOC_DAIFMT_CONT (1 << 4) /* continuous clock */ |
eef28e10 | 49 | #define SND_SOC_DAIFMT_GATED (0 << 4) /* clock is gated */ |
a47cbe72 | 50 | |
a47cbe72 | 51 | /* |
1d387a3f | 52 | * DAI hardware signal polarity. |
a47cbe72 MB |
53 | * |
54 | * Specifies whether the DAI can also support inverted clocks for the specified | |
55 | * format. | |
1d387a3f AP |
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. | |
a47cbe72 | 71 | */ |
5d163336 | 72 | #define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bit clock + frame */ |
75d9ac46 MB |
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 */ | |
a47cbe72 MB |
76 | |
77 | /* | |
78 | * DAI hardware clock masters. | |
79 | * | |
80 | * This is wrt the codec, the inverse is true for the interface | |
47db8e89 | 81 | * i.e. if the codec is clk and FRM master then the interface is |
a47cbe72 MB |
82 | * clk and frame slave. |
83 | */ | |
75d9ac46 MB |
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 */ | |
a47cbe72 MB |
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 | ||
8f738d58 MB |
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 |\ | |
823dbb6e MS |
105 | SNDRV_PCM_FMTBIT_S20_LE |\ |
106 | SNDRV_PCM_FMTBIT_S20_BE |\ | |
8f738d58 MB |
107 | SNDRV_PCM_FMTBIT_S24_3LE |\ |
108 | SNDRV_PCM_FMTBIT_S24_3BE |\ | |
d34c4307 JS |
109 | SNDRV_PCM_FMTBIT_S32_LE |\ |
110 | SNDRV_PCM_FMTBIT_S32_BE) | |
33f503c9 | 111 | |
f0fba2ad | 112 | struct snd_soc_dai_driver; |
a47cbe72 MB |
113 | struct snd_soc_dai; |
114 | struct snd_ac97_bus_ops; | |
115 | ||
116 | /* Digital Audio Interface clocking API.*/ | |
117 | int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, | |
118 | unsigned int freq, int dir); | |
119 | ||
120 | int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai, | |
121 | int div_id, int div); | |
122 | ||
123 | int snd_soc_dai_set_pll(struct snd_soc_dai *dai, | |
85488037 | 124 | int pll_id, int source, unsigned int freq_in, unsigned int freq_out); |
a47cbe72 | 125 | |
e54cf76b LG |
126 | int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio); |
127 | ||
a47cbe72 MB |
128 | /* Digital Audio interface formatting */ |
129 | int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); | |
130 | ||
131 | int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, | |
a5479e38 | 132 | unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width); |
a47cbe72 | 133 | |
472df3cb BS |
134 | int 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 | ||
a47cbe72 MB |
138 | int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); |
139 | ||
140 | /* Digital Audio Interface mute */ | |
da18396f MB |
141 | int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute, |
142 | int direction); | |
a47cbe72 | 143 | |
bece9e95 LG |
144 | int snd_soc_dai_is_dummy(struct snd_soc_dai *dai); |
145 | ||
a47cbe72 MB |
146 | struct 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); | |
85488037 MB |
153 | int (*set_pll)(struct snd_soc_dai *dai, int pll_id, int source, |
154 | unsigned int freq_in, unsigned int freq_out); | |
a47cbe72 | 155 | int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div); |
e54cf76b | 156 | int (*set_bclk_ratio)(struct snd_soc_dai *dai, unsigned int ratio); |
a47cbe72 MB |
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); | |
e5c21514 | 163 | int (*xlate_tdm_slot_mask)(unsigned int slots, |
89c67857 | 164 | unsigned int *tx_mask, unsigned int *rx_mask); |
a47cbe72 | 165 | int (*set_tdm_slot)(struct snd_soc_dai *dai, |
a5479e38 DR |
166 | unsigned int tx_mask, unsigned int rx_mask, |
167 | int slots, int slot_width); | |
472df3cb BS |
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); | |
a47cbe72 MB |
171 | int (*set_tristate)(struct snd_soc_dai *dai, int tristate); |
172 | ||
97349b60 SN |
173 | int (*set_sdw_stream)(struct snd_soc_dai *dai, |
174 | void *stream, int direction); | |
a47cbe72 MB |
175 | /* |
176 | * DAI digital mute - optional. | |
177 | * Called by soc-core to minimise any pops. | |
178 | */ | |
179 | int (*digital_mute)(struct snd_soc_dai *dai, int mute); | |
da18396f | 180 | int (*mute_stream)(struct snd_soc_dai *dai, int mute, int stream); |
dee89c4d MB |
181 | |
182 | /* | |
183 | * ALSA PCM audio operations - all optional. | |
184 | * Called by soc-core during audio PCM operations. | |
185 | */ | |
186 | int (*startup)(struct snd_pcm_substream *, | |
187 | struct snd_soc_dai *); | |
188 | void (*shutdown)(struct snd_pcm_substream *, | |
189 | struct snd_soc_dai *); | |
190 | int (*hw_params)(struct snd_pcm_substream *, | |
191 | struct snd_pcm_hw_params *, struct snd_soc_dai *); | |
192 | int (*hw_free)(struct snd_pcm_substream *, | |
193 | struct snd_soc_dai *); | |
194 | int (*prepare)(struct snd_pcm_substream *, | |
195 | struct snd_soc_dai *); | |
9f1614aa MP |
196 | /* |
197 | * NOTE: Commands passed to the trigger function are not necessarily | |
198 | * compatible with the current state of the dai. For example this | |
199 | * sequence of commands is possible: START STOP STOP. | |
200 | * So do not unconditionally use refcounting functions in the trigger | |
201 | * function, e.g. clk_enable/disable. | |
202 | */ | |
dee89c4d MB |
203 | int (*trigger)(struct snd_pcm_substream *, int, |
204 | struct snd_soc_dai *); | |
07bf84aa LG |
205 | int (*bespoke_trigger)(struct snd_pcm_substream *, int, |
206 | struct snd_soc_dai *); | |
258020d0 PU |
207 | /* |
208 | * For hardware based FIFO caused delay reporting. | |
209 | * Optional. | |
210 | */ | |
211 | snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *, | |
212 | struct snd_soc_dai *); | |
a47cbe72 MB |
213 | }; |
214 | ||
2e622ae4 VK |
215 | struct snd_soc_cdai_ops { |
216 | /* | |
217 | * for compress ops | |
218 | */ | |
219 | int (*startup)(struct snd_compr_stream *, | |
220 | struct snd_soc_dai *); | |
221 | int (*shutdown)(struct snd_compr_stream *, | |
222 | struct snd_soc_dai *); | |
223 | int (*set_params)(struct snd_compr_stream *, | |
224 | struct snd_compr_params *, struct snd_soc_dai *); | |
225 | int (*get_params)(struct snd_compr_stream *, | |
226 | struct snd_codec *, struct snd_soc_dai *); | |
227 | int (*set_metadata)(struct snd_compr_stream *, | |
228 | struct snd_compr_metadata *, struct snd_soc_dai *); | |
229 | int (*get_metadata)(struct snd_compr_stream *, | |
230 | struct snd_compr_metadata *, struct snd_soc_dai *); | |
231 | int (*trigger)(struct snd_compr_stream *, int, | |
232 | struct snd_soc_dai *); | |
233 | int (*pointer)(struct snd_compr_stream *, | |
234 | struct snd_compr_tstamp *, struct snd_soc_dai *); | |
235 | int (*ack)(struct snd_compr_stream *, size_t, | |
236 | struct snd_soc_dai *); | |
237 | }; | |
238 | ||
a47cbe72 | 239 | /* |
f0fba2ad | 240 | * Digital Audio Interface Driver. |
a47cbe72 | 241 | * |
f0fba2ad LG |
242 | * Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97 |
243 | * operations and capabilities. Codec and platform drivers will register this | |
244 | * structure for every DAI they have. | |
245 | * | |
246 | * This structure covers the clocking, formating and ALSA operations for each | |
247 | * interface. | |
a47cbe72 | 248 | */ |
f0fba2ad | 249 | struct snd_soc_dai_driver { |
a47cbe72 | 250 | /* DAI description */ |
f0fba2ad | 251 | const char *name; |
a47cbe72 | 252 | unsigned int id; |
62368294 | 253 | unsigned int base; |
68003e6c | 254 | struct snd_soc_dobj dobj; |
a47cbe72 | 255 | |
f0fba2ad LG |
256 | /* DAI driver callbacks */ |
257 | int (*probe)(struct snd_soc_dai *dai); | |
258 | int (*remove)(struct snd_soc_dai *dai); | |
dc7d7b83 MB |
259 | int (*suspend)(struct snd_soc_dai *dai); |
260 | int (*resume)(struct snd_soc_dai *dai); | |
49681077 | 261 | /* compress dai */ |
6f0c4226 | 262 | int (*compress_new)(struct snd_soc_pcm_runtime *rtd, int num); |
25f7b701 AP |
263 | /* Optional Callback used at pcm creation*/ |
264 | int (*pcm_new)(struct snd_soc_pcm_runtime *rtd, | |
265 | struct snd_soc_dai *dai); | |
bc263214 LPC |
266 | /* DAI is also used for the control bus */ |
267 | bool bus_control; | |
a47cbe72 MB |
268 | |
269 | /* ops */ | |
1ee46ebd | 270 | const struct snd_soc_dai_ops *ops; |
2e622ae4 | 271 | const struct snd_soc_cdai_ops *cops; |
a47cbe72 MB |
272 | |
273 | /* DAI capabilities */ | |
274 | struct snd_soc_pcm_stream capture; | |
275 | struct snd_soc_pcm_stream playback; | |
06f409d7 | 276 | unsigned int symmetric_rates:1; |
3635bf09 NC |
277 | unsigned int symmetric_channels:1; |
278 | unsigned int symmetric_samplebits:1; | |
0168bf0d LG |
279 | |
280 | /* probe ordering - for components with runtime dependencies */ | |
281 | int probe_order; | |
282 | int remove_order; | |
f0fba2ad LG |
283 | }; |
284 | ||
285 | /* | |
286 | * Digital Audio Interface runtime data. | |
287 | * | |
288 | * Holds runtime data for a DAI. | |
289 | */ | |
290 | struct snd_soc_dai { | |
291 | const char *name; | |
292 | int id; | |
293 | struct device *dev; | |
f0fba2ad LG |
294 | |
295 | /* driver ops */ | |
296 | struct snd_soc_dai_driver *driver; | |
a47cbe72 MB |
297 | |
298 | /* DAI runtime info */ | |
40c57963 JB |
299 | unsigned int capture_active; /* stream usage count */ |
300 | unsigned int playback_active; /* stream usage count */ | |
39b2238e KM |
301 | unsigned int probed:1; |
302 | ||
a47cbe72 | 303 | unsigned int active; |
a47cbe72 | 304 | |
888df395 MB |
305 | struct snd_soc_dapm_widget *playback_widget; |
306 | struct snd_soc_dapm_widget *capture_widget; | |
307 | ||
f0fba2ad LG |
308 | /* DAI DMA data */ |
309 | void *playback_dma_data; | |
310 | void *capture_dma_data; | |
a47cbe72 | 311 | |
17841020 DA |
312 | /* Symmetry data - only valid if symmetry is being enforced */ |
313 | unsigned int rate; | |
3635bf09 NC |
314 | unsigned int channels; |
315 | unsigned int sample_bits; | |
17841020 | 316 | |
f0fba2ad | 317 | /* parent platform/codec */ |
6106d129 | 318 | struct snd_soc_component *component; |
2466ab97 | 319 | |
88bd870f BC |
320 | /* CODEC TDM slot masks and params (for fixup) */ |
321 | unsigned int tx_mask; | |
322 | unsigned int rx_mask; | |
323 | ||
a47cbe72 MB |
324 | struct list_head list; |
325 | }; | |
326 | ||
fd23b7de DM |
327 | static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai *dai, |
328 | const struct snd_pcm_substream *ss) | |
329 | { | |
330 | return (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) ? | |
f0fba2ad | 331 | dai->playback_dma_data : dai->capture_dma_data; |
fd23b7de DM |
332 | } |
333 | ||
334 | static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai, | |
335 | const struct snd_pcm_substream *ss, | |
336 | void *data) | |
337 | { | |
338 | if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) | |
f0fba2ad | 339 | dai->playback_dma_data = data; |
fd23b7de | 340 | else |
f0fba2ad LG |
341 | dai->capture_dma_data = data; |
342 | } | |
343 | ||
ecfc0c04 MB |
344 | static inline void snd_soc_dai_init_dma_data(struct snd_soc_dai *dai, |
345 | void *playback, void *capture) | |
346 | { | |
347 | dai->playback_dma_data = playback; | |
348 | dai->capture_dma_data = capture; | |
349 | } | |
350 | ||
f0fba2ad LG |
351 | static inline void snd_soc_dai_set_drvdata(struct snd_soc_dai *dai, |
352 | void *data) | |
353 | { | |
354 | dev_set_drvdata(dai->dev, data); | |
355 | } | |
356 | ||
357 | static inline void *snd_soc_dai_get_drvdata(struct snd_soc_dai *dai) | |
358 | { | |
359 | return dev_get_drvdata(dai->dev); | |
fd23b7de DM |
360 | } |
361 | ||
97349b60 SN |
362 | /** |
363 | * snd_soc_dai_set_sdw_stream() - Configures a DAI for SDW stream operation | |
364 | * @dai: DAI | |
365 | * @stream: STREAM | |
366 | * @direction: Stream direction(Playback/Capture) | |
367 | * SoundWire subsystem doesn't have a notion of direction and we reuse | |
368 | * the ASoC stream direction to configure sink/source ports. | |
369 | * Playback maps to source ports and Capture for sink ports. | |
370 | * | |
371 | * This should be invoked with NULL to clear the stream set previously. | |
372 | * Returns 0 on success, a negative error code otherwise. | |
373 | */ | |
374 | static inline int snd_soc_dai_set_sdw_stream(struct snd_soc_dai *dai, | |
375 | void *stream, int direction) | |
376 | { | |
377 | if (dai->driver->ops->set_sdw_stream) | |
378 | return dai->driver->ops->set_sdw_stream(dai, stream, direction); | |
379 | else | |
380 | return -ENOTSUPP; | |
381 | } | |
382 | ||
a47cbe72 | 383 | #endif |