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TOMOYO Linux Cross Reference
Linux/include/sound/soc-dai.h

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  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 
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 
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 
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 
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 
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  */
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 

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