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 struct snd_soc_dai_ops {
149 /*
150 * DAI clocking configuration, all optional.
151 * Called by soc_card drivers, normally in their hw_params.
152 */
153 int (*set_sysclk)(struct snd_soc_dai *dai,
154 int clk_id, unsigned int freq, int dir);
155 int (*set_pll)(struct snd_soc_dai *dai, int pll_id, int source,
156 unsigned int freq_in, unsigned int freq_out);
157 int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div);
158 int (*set_bclk_ratio)(struct snd_soc_dai *dai, unsigned int ratio);
159
160 /*
161 * DAI format configuration
162 * Called by soc_card drivers, normally in their hw_params.
163 */
164 int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt);
165 int (*xlate_tdm_slot_mask)(unsigned int slots,
166 unsigned int *tx_mask, unsigned int *rx_mask);
167 int (*set_tdm_slot)(struct snd_soc_dai *dai,
168 unsigned int tx_mask, unsigned int rx_mask,
169 int slots, int slot_width);
170 int (*set_channel_map)(struct snd_soc_dai *dai,
171 unsigned int tx_num, unsigned int *tx_slot,
172 unsigned int rx_num, unsigned int *rx_slot);
173 int (*get_channel_map)(struct snd_soc_dai *dai,
174 unsigned int *tx_num, unsigned int *tx_slot,
175 unsigned int *rx_num, unsigned int *rx_slot);
176 int (*set_tristate)(struct snd_soc_dai *dai, int tristate);
177
178 int (*set_sdw_stream)(struct snd_soc_dai *dai,
179 void *stream, int direction);
180 /*
181 * DAI digital mute - optional.
182 * Called by soc-core to minimise any pops.
183 */
184 int (*digital_mute)(struct snd_soc_dai *dai, int mute);
185 int (*mute_stream)(struct snd_soc_dai *dai, int mute, int stream);
186
187 /*
188 * ALSA PCM audio operations - all optional.
189 * Called by soc-core during audio PCM operations.
190 */
191 int (*startup)(struct snd_pcm_substream *,
192 struct snd_soc_dai *);
193 void (*shutdown)(struct snd_pcm_substream *,
194 struct snd_soc_dai *);
195 int (*hw_params)(struct snd_pcm_substream *,
196 struct snd_pcm_hw_params *, struct snd_soc_dai *);
197 int (*hw_free)(struct snd_pcm_substream *,
198 struct snd_soc_dai *);
199 int (*prepare)(struct snd_pcm_substream *,
200 struct snd_soc_dai *);
201 /*
202 * NOTE: Commands passed to the trigger function are not necessarily
203 * compatible with the current state of the dai. For example this
204 * sequence of commands is possible: START STOP STOP.
205 * So do not unconditionally use refcounting functions in the trigger
206 * function, e.g. clk_enable/disable.
207 */
208 int (*trigger)(struct snd_pcm_substream *, int,
209 struct snd_soc_dai *);
210 int (*bespoke_trigger)(struct snd_pcm_substream *, int,
211 struct snd_soc_dai *);
212 /*
213 * For hardware based FIFO caused delay reporting.
214 * Optional.
215 */
216 snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *,
217 struct snd_soc_dai *);
218 };
219
220 struct snd_soc_cdai_ops {
221 /*
222 * for compress ops
223 */
224 int (*startup)(struct snd_compr_stream *,
225 struct snd_soc_dai *);
226 int (*shutdown)(struct snd_compr_stream *,
227 struct snd_soc_dai *);
228 int (*set_params)(struct snd_compr_stream *,
229 struct snd_compr_params *, struct snd_soc_dai *);
230 int (*get_params)(struct snd_compr_stream *,
231 struct snd_codec *, struct snd_soc_dai *);
232 int (*set_metadata)(struct snd_compr_stream *,
233 struct snd_compr_metadata *, struct snd_soc_dai *);
234 int (*get_metadata)(struct snd_compr_stream *,
235 struct snd_compr_metadata *, struct snd_soc_dai *);
236 int (*trigger)(struct snd_compr_stream *, int,
237 struct snd_soc_dai *);
238 int (*pointer)(struct snd_compr_stream *,
239 struct snd_compr_tstamp *, struct snd_soc_dai *);
240 int (*ack)(struct snd_compr_stream *, size_t,
241 struct snd_soc_dai *);
242 };
243
244 /*
245 * Digital Audio Interface Driver.
246 *
247 * Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97
248 * operations and capabilities. Codec and platform drivers will register this
249 * structure for every DAI they have.
250 *
251 * This structure covers the clocking, formating and ALSA operations for each
252 * interface.
253 */
254 struct snd_soc_dai_driver {
255 /* DAI description */
256 const char *name;
257 unsigned int id;
258 unsigned int base;
259 struct snd_soc_dobj dobj;
260
261 /* DAI driver callbacks */
262 int (*probe)(struct snd_soc_dai *dai);
263 int (*remove)(struct snd_soc_dai *dai);
264 int (*suspend)(struct snd_soc_dai *dai);
265 int (*resume)(struct snd_soc_dai *dai);
266 /* compress dai */
267 int (*compress_new)(struct snd_soc_pcm_runtime *rtd, int num);
268 /* Optional Callback used at pcm creation*/
269 int (*pcm_new)(struct snd_soc_pcm_runtime *rtd,
270 struct snd_soc_dai *dai);
271 /* DAI is also used for the control bus */
272 bool bus_control;
273
274 /* ops */
275 const struct snd_soc_dai_ops *ops;
276 const struct snd_soc_cdai_ops *cops;
277
278 /* DAI capabilities */
279 struct snd_soc_pcm_stream capture;
280 struct snd_soc_pcm_stream playback;
281 unsigned int symmetric_rates:1;
282 unsigned int symmetric_channels:1;
283 unsigned int symmetric_samplebits:1;
284
285 /* probe ordering - for components with runtime dependencies */
286 int probe_order;
287 int remove_order;
288 };
289
290 /*
291 * Digital Audio Interface runtime data.
292 *
293 * Holds runtime data for a DAI.
294 */
295 struct snd_soc_dai {
296 const char *name;
297 int id;
298 struct device *dev;
299
300 /* driver ops */
301 struct snd_soc_dai_driver *driver;
302
303 /* DAI runtime info */
304 unsigned int capture_active; /* stream usage count */
305 unsigned int playback_active; /* stream usage count */
306 unsigned int probed:1;
307
308 unsigned int active;
309
310 struct snd_soc_dapm_widget *playback_widget;
311 struct snd_soc_dapm_widget *capture_widget;
312
313 /* DAI DMA data */
314 void *playback_dma_data;
315 void *capture_dma_data;
316
317 /* Symmetry data - only valid if symmetry is being enforced */
318 unsigned int rate;
319 unsigned int channels;
320 unsigned int sample_bits;
321
322 /* parent platform/codec */
323 struct snd_soc_component *component;
324
325 /* CODEC TDM slot masks and params (for fixup) */
326 unsigned int tx_mask;
327 unsigned int rx_mask;
328
329 struct list_head list;
330 };
331
snd_soc_dai_get_dma_data(const struct snd_soc_dai * dai,const struct snd_pcm_substream * ss)332 static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai *dai,
333 const struct snd_pcm_substream *ss)
334 {
335 return (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
336 dai->playback_dma_data : dai->capture_dma_data;
337 }
338
snd_soc_dai_set_dma_data(struct snd_soc_dai * dai,const struct snd_pcm_substream * ss,void * data)339 static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai,
340 const struct snd_pcm_substream *ss,
341 void *data)
342 {
343 if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK)
344 dai->playback_dma_data = data;
345 else
346 dai->capture_dma_data = data;
347 }
348
snd_soc_dai_init_dma_data(struct snd_soc_dai * dai,void * playback,void * capture)349 static inline void snd_soc_dai_init_dma_data(struct snd_soc_dai *dai,
350 void *playback, void *capture)
351 {
352 dai->playback_dma_data = playback;
353 dai->capture_dma_data = capture;
354 }
355
snd_soc_dai_set_drvdata(struct snd_soc_dai * dai,void * data)356 static inline void snd_soc_dai_set_drvdata(struct snd_soc_dai *dai,
357 void *data)
358 {
359 dev_set_drvdata(dai->dev, data);
360 }
361
snd_soc_dai_get_drvdata(struct snd_soc_dai * dai)362 static inline void *snd_soc_dai_get_drvdata(struct snd_soc_dai *dai)
363 {
364 return dev_get_drvdata(dai->dev);
365 }
366
367 /**
368 * snd_soc_dai_set_sdw_stream() - Configures a DAI for SDW stream operation
369 * @dai: DAI
370 * @stream: STREAM
371 * @direction: Stream direction(Playback/Capture)
372 * SoundWire subsystem doesn't have a notion of direction and we reuse
373 * the ASoC stream direction to configure sink/source ports.
374 * Playback maps to source ports and Capture for sink ports.
375 *
376 * This should be invoked with NULL to clear the stream set previously.
377 * Returns 0 on success, a negative error code otherwise.
378 */
snd_soc_dai_set_sdw_stream(struct snd_soc_dai * dai,void * stream,int direction)379 static inline int snd_soc_dai_set_sdw_stream(struct snd_soc_dai *dai,
380 void *stream, int direction)
381 {
382 if (dai->driver->ops->set_sdw_stream)
383 return dai->driver->ops->set_sdw_stream(dai, stream, direction);
384 else
385 return -ENOTSUPP;
386 }
387
388 #endif
389