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1/* SPDX-License-Identifier: GPL-2.0
2 *
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3 * linux/sound/soc-dai.h -- ALSA SoC Layer
4 *
5 * Copyright: 2005-2008 Wolfson Microelectronics. PLC.
6 *
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7 * Digital Audio Interface (DAI) API.
8 */
9
10#ifndef __LINUX_SND_SOC_DAI_H
11#define __LINUX_SND_SOC_DAI_H
12
13
14#include <linux/list.h>
d2e3cb85 15#include <sound/asoc.h>
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16
17struct snd_pcm_substream;
888df395 18struct snd_soc_dapm_widget;
49681077 19struct snd_compr_stream;
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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 */
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27#define SND_SOC_DAIFMT_I2S SND_SOC_DAI_FORMAT_I2S
28#define SND_SOC_DAIFMT_RIGHT_J SND_SOC_DAI_FORMAT_RIGHT_J
29#define SND_SOC_DAIFMT_LEFT_J SND_SOC_DAI_FORMAT_LEFT_J
30#define SND_SOC_DAIFMT_DSP_A SND_SOC_DAI_FORMAT_DSP_A
31#define SND_SOC_DAIFMT_DSP_B SND_SOC_DAI_FORMAT_DSP_B
32#define SND_SOC_DAIFMT_AC97 SND_SOC_DAI_FORMAT_AC97
33#define SND_SOC_DAIFMT_PDM SND_SOC_DAI_FORMAT_PDM
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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 *
47db8e89 42 * DAI bit clocks can be be gated (disabled) when the DAI is not
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43 * sending or receiving PCM data in a frame. This can be used to save power.
44 */
75d9ac46 45#define SND_SOC_DAIFMT_CONT (1 << 4) /* continuous clock */
eef28e10 46#define SND_SOC_DAIFMT_GATED (0 << 4) /* clock is gated */
a47cbe72 47
a47cbe72 48/*
1d387a3f 49 * DAI hardware signal polarity.
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50 *
51 * Specifies whether the DAI can also support inverted clocks for the specified
52 * format.
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53 *
54 * BCLK:
55 * - "normal" polarity means signal is available at rising edge of BCLK
56 * - "inverted" polarity means signal is available at falling edge of BCLK
57 *
58 * FSYNC "normal" polarity depends on the frame format:
59 * - I2S: frame consists of left then right channel data. Left channel starts
60 * with falling FSYNC edge, right channel starts with rising FSYNC edge.
61 * - Left/Right Justified: frame consists of left then right channel data.
62 * Left channel starts with rising FSYNC edge, right channel starts with
63 * falling FSYNC edge.
64 * - DSP A/B: Frame starts with rising FSYNC edge.
65 * - AC97: Frame starts with rising FSYNC edge.
66 *
67 * "Negative" FSYNC polarity is the one opposite of "normal" polarity.
a47cbe72 68 */
5d163336 69#define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bit clock + frame */
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70#define SND_SOC_DAIFMT_NB_IF (2 << 8) /* normal BCLK + inv FRM */
71#define SND_SOC_DAIFMT_IB_NF (3 << 8) /* invert BCLK + nor FRM */
72#define SND_SOC_DAIFMT_IB_IF (4 << 8) /* invert BCLK + FRM */
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73
74/*
75 * DAI hardware clock masters.
76 *
77 * This is wrt the codec, the inverse is true for the interface
47db8e89 78 * i.e. if the codec is clk and FRM master then the interface is
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79 * clk and frame slave.
80 */
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81#define SND_SOC_DAIFMT_CBM_CFM (1 << 12) /* codec clk & FRM master */
82#define SND_SOC_DAIFMT_CBS_CFM (2 << 12) /* codec clk slave & FRM master */
83#define SND_SOC_DAIFMT_CBM_CFS (3 << 12) /* codec clk master & frame slave */
84#define SND_SOC_DAIFMT_CBS_CFS (4 << 12) /* codec clk & FRM slave */
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85
86#define SND_SOC_DAIFMT_FORMAT_MASK 0x000f
87#define SND_SOC_DAIFMT_CLOCK_MASK 0x00f0
88#define SND_SOC_DAIFMT_INV_MASK 0x0f00
89#define SND_SOC_DAIFMT_MASTER_MASK 0xf000
90
91/*
92 * Master Clock Directions
93 */
94#define SND_SOC_CLOCK_IN 0
95#define SND_SOC_CLOCK_OUT 1
96
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97#define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S8 |\
98 SNDRV_PCM_FMTBIT_S16_LE |\
99 SNDRV_PCM_FMTBIT_S16_BE |\
100 SNDRV_PCM_FMTBIT_S20_3LE |\
101 SNDRV_PCM_FMTBIT_S20_3BE |\
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102 SNDRV_PCM_FMTBIT_S20_LE |\
103 SNDRV_PCM_FMTBIT_S20_BE |\
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104 SNDRV_PCM_FMTBIT_S24_3LE |\
105 SNDRV_PCM_FMTBIT_S24_3BE |\
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106 SNDRV_PCM_FMTBIT_S32_LE |\
107 SNDRV_PCM_FMTBIT_S32_BE)
33f503c9 108
f0fba2ad 109struct snd_soc_dai_driver;
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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,
85488037 121 int pll_id, int source, unsigned int freq_in, unsigned int freq_out);
a47cbe72 122
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123int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio);
124
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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,
a5479e38 129 unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width);
a47cbe72 130
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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
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135int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate);
136
137/* Digital Audio Interface mute */
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138int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute,
139 int direction);
a47cbe72 140
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141
142int snd_soc_dai_get_channel_map(struct snd_soc_dai *dai,
143 unsigned int *tx_num, unsigned int *tx_slot,
144 unsigned int *rx_num, unsigned int *rx_slot);
145
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146int snd_soc_dai_is_dummy(struct snd_soc_dai *dai);
147
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148int snd_soc_dai_hw_params(struct snd_soc_dai *dai,
149 struct snd_pcm_substream *substream,
150 struct snd_pcm_hw_params *params);
151
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152struct snd_soc_dai_ops {
153 /*
154 * DAI clocking configuration, all optional.
155 * Called by soc_card drivers, normally in their hw_params.
156 */
157 int (*set_sysclk)(struct snd_soc_dai *dai,
158 int clk_id, unsigned int freq, int dir);
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159 int (*set_pll)(struct snd_soc_dai *dai, int pll_id, int source,
160 unsigned int freq_in, unsigned int freq_out);
a47cbe72 161 int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div);
e54cf76b 162 int (*set_bclk_ratio)(struct snd_soc_dai *dai, unsigned int ratio);
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163
164 /*
165 * DAI format configuration
166 * Called by soc_card drivers, normally in their hw_params.
167 */
168 int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt);
e5c21514 169 int (*xlate_tdm_slot_mask)(unsigned int slots,
89c67857 170 unsigned int *tx_mask, unsigned int *rx_mask);
a47cbe72 171 int (*set_tdm_slot)(struct snd_soc_dai *dai,
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172 unsigned int tx_mask, unsigned int rx_mask,
173 int slots, int slot_width);
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174 int (*set_channel_map)(struct snd_soc_dai *dai,
175 unsigned int tx_num, unsigned int *tx_slot,
176 unsigned int rx_num, unsigned int *rx_slot);
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177 int (*get_channel_map)(struct snd_soc_dai *dai,
178 unsigned int *tx_num, unsigned int *tx_slot,
179 unsigned int *rx_num, unsigned int *rx_slot);
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180 int (*set_tristate)(struct snd_soc_dai *dai, int tristate);
181
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182 int (*set_sdw_stream)(struct snd_soc_dai *dai,
183 void *stream, int direction);
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184 /*
185 * DAI digital mute - optional.
186 * Called by soc-core to minimise any pops.
187 */
188 int (*digital_mute)(struct snd_soc_dai *dai, int mute);
da18396f 189 int (*mute_stream)(struct snd_soc_dai *dai, int mute, int stream);
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190
191 /*
192 * ALSA PCM audio operations - all optional.
193 * Called by soc-core during audio PCM operations.
194 */
195 int (*startup)(struct snd_pcm_substream *,
196 struct snd_soc_dai *);
197 void (*shutdown)(struct snd_pcm_substream *,
198 struct snd_soc_dai *);
199 int (*hw_params)(struct snd_pcm_substream *,
200 struct snd_pcm_hw_params *, struct snd_soc_dai *);
201 int (*hw_free)(struct snd_pcm_substream *,
202 struct snd_soc_dai *);
203 int (*prepare)(struct snd_pcm_substream *,
204 struct snd_soc_dai *);
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205 /*
206 * NOTE: Commands passed to the trigger function are not necessarily
207 * compatible with the current state of the dai. For example this
208 * sequence of commands is possible: START STOP STOP.
209 * So do not unconditionally use refcounting functions in the trigger
210 * function, e.g. clk_enable/disable.
211 */
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212 int (*trigger)(struct snd_pcm_substream *, int,
213 struct snd_soc_dai *);
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214 int (*bespoke_trigger)(struct snd_pcm_substream *, int,
215 struct snd_soc_dai *);
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216 /*
217 * For hardware based FIFO caused delay reporting.
218 * Optional.
219 */
220 snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *,
221 struct snd_soc_dai *);
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222};
223
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224struct snd_soc_cdai_ops {
225 /*
226 * for compress ops
227 */
228 int (*startup)(struct snd_compr_stream *,
229 struct snd_soc_dai *);
230 int (*shutdown)(struct snd_compr_stream *,
231 struct snd_soc_dai *);
232 int (*set_params)(struct snd_compr_stream *,
233 struct snd_compr_params *, struct snd_soc_dai *);
234 int (*get_params)(struct snd_compr_stream *,
235 struct snd_codec *, struct snd_soc_dai *);
236 int (*set_metadata)(struct snd_compr_stream *,
237 struct snd_compr_metadata *, struct snd_soc_dai *);
238 int (*get_metadata)(struct snd_compr_stream *,
239 struct snd_compr_metadata *, struct snd_soc_dai *);
240 int (*trigger)(struct snd_compr_stream *, int,
241 struct snd_soc_dai *);
242 int (*pointer)(struct snd_compr_stream *,
243 struct snd_compr_tstamp *, struct snd_soc_dai *);
244 int (*ack)(struct snd_compr_stream *, size_t,
245 struct snd_soc_dai *);
246};
247
a47cbe72 248/*
f0fba2ad 249 * Digital Audio Interface Driver.
a47cbe72 250 *
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251 * Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97
252 * operations and capabilities. Codec and platform drivers will register this
253 * structure for every DAI they have.
254 *
255 * This structure covers the clocking, formating and ALSA operations for each
256 * interface.
a47cbe72 257 */
f0fba2ad 258struct snd_soc_dai_driver {
a47cbe72 259 /* DAI description */
f0fba2ad 260 const char *name;
a47cbe72 261 unsigned int id;
62368294 262 unsigned int base;
68003e6c 263 struct snd_soc_dobj dobj;
a47cbe72 264
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265 /* DAI driver callbacks */
266 int (*probe)(struct snd_soc_dai *dai);
267 int (*remove)(struct snd_soc_dai *dai);
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268 int (*suspend)(struct snd_soc_dai *dai);
269 int (*resume)(struct snd_soc_dai *dai);
49681077 270 /* compress dai */
6f0c4226 271 int (*compress_new)(struct snd_soc_pcm_runtime *rtd, int num);
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272 /* Optional Callback used at pcm creation*/
273 int (*pcm_new)(struct snd_soc_pcm_runtime *rtd,
274 struct snd_soc_dai *dai);
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275 /* DAI is also used for the control bus */
276 bool bus_control;
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277
278 /* ops */
1ee46ebd 279 const struct snd_soc_dai_ops *ops;
2e622ae4 280 const struct snd_soc_cdai_ops *cops;
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281
282 /* DAI capabilities */
283 struct snd_soc_pcm_stream capture;
284 struct snd_soc_pcm_stream playback;
06f409d7 285 unsigned int symmetric_rates:1;
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286 unsigned int symmetric_channels:1;
287 unsigned int symmetric_samplebits:1;
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288
289 /* probe ordering - for components with runtime dependencies */
290 int probe_order;
291 int remove_order;
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292};
293
294/*
295 * Digital Audio Interface runtime data.
296 *
297 * Holds runtime data for a DAI.
298 */
299struct snd_soc_dai {
300 const char *name;
301 int id;
302 struct device *dev;
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303
304 /* driver ops */
305 struct snd_soc_dai_driver *driver;
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306
307 /* DAI runtime info */
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308 unsigned int capture_active; /* stream usage count */
309 unsigned int playback_active; /* stream usage count */
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310 unsigned int probed:1;
311
a47cbe72 312 unsigned int active;
a47cbe72 313
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314 struct snd_soc_dapm_widget *playback_widget;
315 struct snd_soc_dapm_widget *capture_widget;
316
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317 /* DAI DMA data */
318 void *playback_dma_data;
319 void *capture_dma_data;
a47cbe72 320
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321 /* Symmetry data - only valid if symmetry is being enforced */
322 unsigned int rate;
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323 unsigned int channels;
324 unsigned int sample_bits;
17841020 325
f0fba2ad 326 /* parent platform/codec */
6106d129 327 struct snd_soc_component *component;
2466ab97 328
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329 /* CODEC TDM slot masks and params (for fixup) */
330 unsigned int tx_mask;
331 unsigned int rx_mask;
332
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333 struct list_head list;
334};
335
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336static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai *dai,
337 const struct snd_pcm_substream *ss)
338{
339 return (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
f0fba2ad 340 dai->playback_dma_data : dai->capture_dma_data;
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341}
342
343static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai,
344 const struct snd_pcm_substream *ss,
345 void *data)
346{
347 if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK)
f0fba2ad 348 dai->playback_dma_data = data;
fd23b7de 349 else
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350 dai->capture_dma_data = data;
351}
352
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353static inline void snd_soc_dai_init_dma_data(struct snd_soc_dai *dai,
354 void *playback, void *capture)
355{
356 dai->playback_dma_data = playback;
357 dai->capture_dma_data = capture;
358}
359
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360static inline void snd_soc_dai_set_drvdata(struct snd_soc_dai *dai,
361 void *data)
362{
363 dev_set_drvdata(dai->dev, data);
364}
365
366static inline void *snd_soc_dai_get_drvdata(struct snd_soc_dai *dai)
367{
368 return dev_get_drvdata(dai->dev);
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369}
370
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371/**
372 * snd_soc_dai_set_sdw_stream() - Configures a DAI for SDW stream operation
373 * @dai: DAI
374 * @stream: STREAM
375 * @direction: Stream direction(Playback/Capture)
376 * SoundWire subsystem doesn't have a notion of direction and we reuse
377 * the ASoC stream direction to configure sink/source ports.
378 * Playback maps to source ports and Capture for sink ports.
379 *
380 * This should be invoked with NULL to clear the stream set previously.
381 * Returns 0 on success, a negative error code otherwise.
382 */
383static inline int snd_soc_dai_set_sdw_stream(struct snd_soc_dai *dai,
384 void *stream, int direction)
385{
386 if (dai->driver->ops->set_sdw_stream)
387 return dai->driver->ops->set_sdw_stream(dai, stream, direction);
388 else
389 return -ENOTSUPP;
390}
391
a47cbe72 392#endif