1 /* SPDX-License-Identifier: GPL-2.0
2 *
3 * linux/sound/soc-dai.h -- ALSA SoC Layer
4 *
5 * Copyright: 2005-2008 Wolfson Microelectronics. PLC.
6 *
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>
15 #include <sound/asoc.h>
16
17 struct snd_pcm_substream;
18 struct snd_soc_dapm_widget;
19 struct snd_compr_stream;
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 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
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 (1 << 4) /* continuous clock */
46 #define SND_SOC_DAIFMT_GATED (0 << 4) /* clock is gated */
47
48 /*
49 * DAI hardware signal polarity.
50 *
51 * Specifies whether the DAI can also support inverted clocks for the specified
52 * format.
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.
68 */
69 #define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bit clock + frame */
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 */
73
74 /*
75 * DAI hardware clock masters.
76 *
77 * This is wrt the codec, the inverse is true for the interface
78 * i.e. if the codec is clk and FRM master then the interface is
79 * clk and frame slave.
80 */
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 */
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
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 |\
102 SNDRV_PCM_FMTBIT_S20_LE |\
103 SNDRV_PCM_FMTBIT_S20_BE |\
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
109 struct snd_soc_dai_driver;
110 struct snd_soc_dai;
111 struct snd_ac97_bus_ops;
112
113 /* Digital Audio Interface clocking API.*/
114 int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
115 unsigned int freq, int dir);
116
117 int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai,
118 int div_id, int div);
119
120 int 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
123 int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio);
124
125 /* Digital Audio interface formatting */
126 int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt);
127
128 int 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
131 int 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
135 int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate);
136
137 /* Digital Audio Interface mute */
138 int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute,
139 int direction);
140
141
142 int 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
146 int snd_soc_dai_is_dummy(struct snd_soc_dai *dai);
147
148 int snd_soc_dai_hw_params(struct snd_soc_dai *dai,
149 struct snd_pcm_substream *substream,
150 struct snd_pcm_hw_params *params);
151 void snd_soc_dai_hw_free(struct snd_soc_dai *dai,
152 struct snd_pcm_substream *substream);
153 int snd_soc_dai_startup(struct snd_soc_dai *dai,
154 struct snd_pcm_substream *substream);
155 void snd_soc_dai_shutdown(struct snd_soc_dai *dai,
156 struct snd_pcm_substream *substream);
157 int snd_soc_dai_prepare(struct snd_soc_dai *dai,
158 struct snd_pcm_substream *substream);
159 int snd_soc_dai_trigger(struct snd_soc_dai *dai,
160 struct snd_pcm_substream *substream, int cmd);
161 int snd_soc_dai_bespoke_trigger(struct snd_soc_dai *dai,
162 struct snd_pcm_substream *substream, int cmd);
163 snd_pcm_sframes_t snd_soc_dai_delay(struct snd_soc_dai *dai,
164 struct snd_pcm_substream *substream);
165 void snd_soc_dai_suspend(struct snd_soc_dai *dai);
166 void snd_soc_dai_resume(struct snd_soc_dai *dai);
167 int snd_soc_dai_probe(struct snd_soc_dai *dai);
168 int snd_soc_dai_remove(struct snd_soc_dai *dai);
169 int snd_soc_dai_compress_new(struct snd_soc_dai *dai,
170 struct snd_soc_pcm_runtime *rtd, int num);
171 bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int stream);
172
173 struct snd_soc_dai_ops {
174 /*
175 * DAI clocking configuration, all optional.
176 * Called by soc_card drivers, normally in their hw_params.
177 */
178 int (*set_sysclk)(struct snd_soc_dai *dai,
179 int clk_id, unsigned int freq, int dir);
180 int (*set_pll)(struct snd_soc_dai *dai, int pll_id, int source,
181 unsigned int freq_in, unsigned int freq_out);
182 int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div);
183 int (*set_bclk_ratio)(struct snd_soc_dai *dai, unsigned int ratio);
184
185 /*
186 * DAI format configuration
187 * Called by soc_card drivers, normally in their hw_params.
188 */
189 int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt);
190 int (*xlate_tdm_slot_mask)(unsigned int slots,
191 unsigned int *tx_mask, unsigned int *rx_mask);
192 int (*set_tdm_slot)(struct snd_soc_dai *dai,
193 unsigned int tx_mask, unsigned int rx_mask,
194 int slots, int slot_width);
195 int (*set_channel_map)(struct snd_soc_dai *dai,
196 unsigned int tx_num, unsigned int *tx_slot,
197 unsigned int rx_num, unsigned int *rx_slot);
198 int (*get_channel_map)(struct snd_soc_dai *dai,
199 unsigned int *tx_num, unsigned int *tx_slot,
200 unsigned int *rx_num, unsigned int *rx_slot);
201 int (*set_tristate)(struct snd_soc_dai *dai, int tristate);
202
203 int (*set_sdw_stream)(struct snd_soc_dai *dai,
204 void *stream, int direction);
205 /*
206 * DAI digital mute - optional.
207 * Called by soc-core to minimise any pops.
208 */
209 int (*digital_mute)(struct snd_soc_dai *dai, int mute);
210 int (*mute_stream)(struct snd_soc_dai *dai, int mute, int stream);
211
212 /*
213 * ALSA PCM audio operations - all optional.
214 * Called by soc-core during audio PCM operations.
215 */
216 int (*startup)(struct snd_pcm_substream *,
217 struct snd_soc_dai *);
218 void (*shutdown)(struct snd_pcm_substream *,
219 struct snd_soc_dai *);
220 int (*hw_params)(struct snd_pcm_substream *,
221 struct snd_pcm_hw_params *, struct snd_soc_dai *);
222 int (*hw_free)(struct snd_pcm_substream *,
223 struct snd_soc_dai *);
224 int (*prepare)(struct snd_pcm_substream *,
225 struct snd_soc_dai *);
226 /*
227 * NOTE: Commands passed to the trigger function are not necessarily
228 * compatible with the current state of the dai. For example this
229 * sequence of commands is possible: START STOP STOP.
230 * So do not unconditionally use refcounting functions in the trigger
231 * function, e.g. clk_enable/disable.
232 */
233 int (*trigger)(struct snd_pcm_substream *, int,
234 struct snd_soc_dai *);
235 int (*bespoke_trigger)(struct snd_pcm_substream *, int,
236 struct snd_soc_dai *);
237 /*
238 * For hardware based FIFO caused delay reporting.
239 * Optional.
240 */
241 snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *,
242 struct snd_soc_dai *);
243 };
244
245 struct snd_soc_cdai_ops {
246 /*
247 * for compress ops
248 */
249 int (*startup)(struct snd_compr_stream *,
250 struct snd_soc_dai *);
251 int (*shutdown)(struct snd_compr_stream *,
252 struct snd_soc_dai *);
253 int (*set_params)(struct snd_compr_stream *,
254 struct snd_compr_params *, struct snd_soc_dai *);
255 int (*get_params)(struct snd_compr_stream *,
256 struct snd_codec *, struct snd_soc_dai *);
257 int (*set_metadata)(struct snd_compr_stream *,
258 struct snd_compr_metadata *, struct snd_soc_dai *);
259 int (*get_metadata)(struct snd_compr_stream *,
260 struct snd_compr_metadata *, struct snd_soc_dai *);
261 int (*trigger)(struct snd_compr_stream *, int,
262 struct snd_soc_dai *);
263 int (*pointer)(struct snd_compr_stream *,
264 struct snd_compr_tstamp *, struct snd_soc_dai *);
265 int (*ack)(struct snd_compr_stream *, size_t,
266 struct snd_soc_dai *);
267 };
268
269 /*
270 * Digital Audio Interface Driver.
271 *
272 * Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97
273 * operations and capabilities. Codec and platform drivers will register this
274 * structure for every DAI they have.
275 *
276 * This structure covers the clocking, formating and ALSA operations for each
277 * interface.
278 */
279 struct snd_soc_dai_driver {
280 /* DAI description */
281 const char *name;
282 unsigned int id;
283 unsigned int base;
284 struct snd_soc_dobj dobj;
285
286 /* DAI driver callbacks */
287 int (*probe)(struct snd_soc_dai *dai);
288 int (*remove)(struct snd_soc_dai *dai);
289 int (*suspend)(struct snd_soc_dai *dai);
290 int (*resume)(struct snd_soc_dai *dai);
291 /* compress dai */
292 int (*compress_new)(struct snd_soc_pcm_runtime *rtd, int num);
293 /* Optional Callback used at pcm creation*/
294 int (*pcm_new)(struct snd_soc_pcm_runtime *rtd,
295 struct snd_soc_dai *dai);
296
297 /* ops */
298 const struct snd_soc_dai_ops *ops;
299 const struct snd_soc_cdai_ops *cops;
300
301 /* DAI capabilities */
302 struct snd_soc_pcm_stream capture;
303 struct snd_soc_pcm_stream playback;
304 unsigned int symmetric_rates:1;
305 unsigned int symmetric_channels:1;
306 unsigned int symmetric_samplebits:1;
307 unsigned int bus_control:1; /* DAI is also used for the control bus */
308
309 /* probe ordering - for components with runtime dependencies */
310 int probe_order;
311 int remove_order;
312 };
313
314 /*
315 * Digital Audio Interface runtime data.
316 *
317 * Holds runtime data for a DAI.
318 */
319 struct snd_soc_dai {
320 const char *name;
321 int id;
322 struct device *dev;
323
324 /* driver ops */
325 struct snd_soc_dai_driver *driver;
326
327 /* DAI runtime info */
328 unsigned int capture_active; /* stream usage count */
329 unsigned int playback_active; /* stream usage count */
330 unsigned int probed:1;
331
332 unsigned int active;
333
334 struct snd_soc_dapm_widget *playback_widget;
335 struct snd_soc_dapm_widget *capture_widget;
336
337 /* DAI DMA data */
338 void *playback_dma_data;
339 void *capture_dma_data;
340
341 /* Symmetry data - only valid if symmetry is being enforced */
342 unsigned int rate;
343 unsigned int channels;
344 unsigned int sample_bits;
345
346 /* parent platform/codec */
347 struct snd_soc_component *component;
348
349 /* CODEC TDM slot masks and params (for fixup) */
350 unsigned int tx_mask;
351 unsigned int rx_mask;
352
353 struct list_head list;
354 };
355
snd_soc_dai_get_dma_data(const struct snd_soc_dai * dai,const struct snd_pcm_substream * ss)356 static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai *dai,
357 const struct snd_pcm_substream *ss)
358 {
359 return (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
360 dai->playback_dma_data : dai->capture_dma_data;
361 }
362
snd_soc_dai_set_dma_data(struct snd_soc_dai * dai,const struct snd_pcm_substream * ss,void * data)363 static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai,
364 const struct snd_pcm_substream *ss,
365 void *data)
366 {
367 if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK)
368 dai->playback_dma_data = data;
369 else
370 dai->capture_dma_data = data;
371 }
372
snd_soc_dai_init_dma_data(struct snd_soc_dai * dai,void * playback,void * capture)373 static inline void snd_soc_dai_init_dma_data(struct snd_soc_dai *dai,
374 void *playback, void *capture)
375 {
376 dai->playback_dma_data = playback;
377 dai->capture_dma_data = capture;
378 }
379
snd_soc_dai_set_drvdata(struct snd_soc_dai * dai,void * data)380 static inline void snd_soc_dai_set_drvdata(struct snd_soc_dai *dai,
381 void *data)
382 {
383 dev_set_drvdata(dai->dev, data);
384 }
385
snd_soc_dai_get_drvdata(struct snd_soc_dai * dai)386 static inline void *snd_soc_dai_get_drvdata(struct snd_soc_dai *dai)
387 {
388 return dev_get_drvdata(dai->dev);
389 }
390
391 /**
392 * snd_soc_dai_set_sdw_stream() - Configures a DAI for SDW stream operation
393 * @dai: DAI
394 * @stream: STREAM
395 * @direction: Stream direction(Playback/Capture)
396 * SoundWire subsystem doesn't have a notion of direction and we reuse
397 * the ASoC stream direction to configure sink/source ports.
398 * Playback maps to source ports and Capture for sink ports.
399 *
400 * This should be invoked with NULL to clear the stream set previously.
401 * Returns 0 on success, a negative error code otherwise.
402 */
snd_soc_dai_set_sdw_stream(struct snd_soc_dai * dai,void * stream,int direction)403 static inline int snd_soc_dai_set_sdw_stream(struct snd_soc_dai *dai,
404 void *stream, int direction)
405 {
406 if (dai->driver->ops->set_sdw_stream)
407 return dai->driver->ops->set_sdw_stream(dai, stream, direction);
408 else
409 return -ENOTSUPP;
410 }
411
412 #endif
413
