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 void *(*get_sdw_stream)(struct snd_soc_dai *dai, int direction); 206 207 /* 208 * DAI digital mute - optional. 209 * Called by soc-core to minimise any pops. 210 */ 211 int (*digital_mute)(struct snd_soc_dai *dai, int mute); 212 int (*mute_stream)(struct snd_soc_dai *dai, int mute, int stream); 213 214 /* 215 * ALSA PCM audio operations - all optional. 216 * Called by soc-core during audio PCM operations. 217 */ 218 int (*startup)(struct snd_pcm_substream *, 219 struct snd_soc_dai *); 220 void (*shutdown)(struct snd_pcm_substream *, 221 struct snd_soc_dai *); 222 int (*hw_params)(struct snd_pcm_substream *, 223 struct snd_pcm_hw_params *, struct snd_soc_dai *); 224 int (*hw_free)(struct snd_pcm_substream *, 225 struct snd_soc_dai *); 226 int (*prepare)(struct snd_pcm_substream *, 227 struct snd_soc_dai *); 228 /* 229 * NOTE: Commands passed to the trigger function are not necessarily 230 * compatible with the current state of the dai. For example this 231 * sequence of commands is possible: START STOP STOP. 232 * So do not unconditionally use refcounting functions in the trigger 233 * function, e.g. clk_enable/disable. 234 */ 235 int (*trigger)(struct snd_pcm_substream *, int, 236 struct snd_soc_dai *); 237 int (*bespoke_trigger)(struct snd_pcm_substream *, int, 238 struct snd_soc_dai *); 239 /* 240 * For hardware based FIFO caused delay reporting. 241 * Optional. 242 */ 243 snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *, 244 struct snd_soc_dai *); 245 }; 246 247 struct snd_soc_cdai_ops { 248 /* 249 * for compress ops 250 */ 251 int (*startup)(struct snd_compr_stream *, 252 struct snd_soc_dai *); 253 int (*shutdown)(struct snd_compr_stream *, 254 struct snd_soc_dai *); 255 int (*set_params)(struct snd_compr_stream *, 256 struct snd_compr_params *, struct snd_soc_dai *); 257 int (*get_params)(struct snd_compr_stream *, 258 struct snd_codec *, struct snd_soc_dai *); 259 int (*set_metadata)(struct snd_compr_stream *, 260 struct snd_compr_metadata *, struct snd_soc_dai *); 261 int (*get_metadata)(struct snd_compr_stream *, 262 struct snd_compr_metadata *, struct snd_soc_dai *); 263 int (*trigger)(struct snd_compr_stream *, int, 264 struct snd_soc_dai *); 265 int (*pointer)(struct snd_compr_stream *, 266 struct snd_compr_tstamp *, struct snd_soc_dai *); 267 int (*ack)(struct snd_compr_stream *, size_t, 268 struct snd_soc_dai *); 269 }; 270 271 /* 272 * Digital Audio Interface Driver. 273 * 274 * Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97 275 * operations and capabilities. Codec and platform drivers will register this 276 * structure for every DAI they have. 277 * 278 * This structure covers the clocking, formating and ALSA operations for each 279 * interface. 280 */ 281 struct snd_soc_dai_driver { 282 /* DAI description */ 283 const char *name; 284 unsigned int id; 285 unsigned int base; 286 struct snd_soc_dobj dobj; 287 288 /* DAI driver callbacks */ 289 int (*probe)(struct snd_soc_dai *dai); 290 int (*remove)(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; 307 308 /* probe ordering - for components with runtime dependencies */ 309 int probe_order; 310 int remove_order; 311 }; 312 313 /* 314 * Digital Audio Interface runtime data. 315 * 316 * Holds runtime data for a DAI. 317 */ 318 struct snd_soc_dai { 319 const char *name; 320 int id; 321 struct device *dev; 322 323 /* driver ops */ 324 struct snd_soc_dai_driver *driver; 325 326 /* DAI runtime info */ 327 unsigned int stream_active[SNDRV_PCM_STREAM_LAST + 1]; /* usage count */ 328 329 unsigned int active; 330 331 struct snd_soc_dapm_widget *playback_widget; 332 struct snd_soc_dapm_widget *capture_widget; 333 334 /* DAI DMA data */ 335 void *playback_dma_data; 336 void *capture_dma_data; 337 338 /* Symmetry data - only valid if symmetry is being enforced */ 339 unsigned int rate; 340 unsigned int channels; 341 unsigned int sample_bits; 342 343 /* parent platform/codec */ 344 struct snd_soc_component *component; 345 346 /* CODEC TDM slot masks and params (for fixup) */ 347 unsigned int tx_mask; 348 unsigned int rx_mask; 349 350 struct list_head list; 351 352 /* bit field */ 353 unsigned int probed:1; 354 }; 355 356 static inline struct snd_soc_pcm_stream * 357 snd_soc_dai_get_pcm_stream(const struct snd_soc_dai *dai, int stream) 358 { 359 return (stream == SNDRV_PCM_STREAM_PLAYBACK) ? 360 &dai->driver->playback : &dai->driver->capture; 361 } 362 363 static inline 364 struct snd_soc_dapm_widget *snd_soc_dai_get_widget( 365 struct snd_soc_dai *dai, int stream) 366 { 367 return (stream == SNDRV_PCM_STREAM_PLAYBACK) ? 368 dai->playback_widget : dai->capture_widget; 369 } 370 371 static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai *dai, 372 const struct snd_pcm_substream *ss) 373 { 374 return (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) ? 375 dai->playback_dma_data : dai->capture_dma_data; 376 } 377 378 static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai, 379 const struct snd_pcm_substream *ss, 380 void *data) 381 { 382 if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) 383 dai->playback_dma_data = data; 384 else 385 dai->capture_dma_data = data; 386 } 387 388 static inline void snd_soc_dai_init_dma_data(struct snd_soc_dai *dai, 389 void *playback, void *capture) 390 { 391 dai->playback_dma_data = playback; 392 dai->capture_dma_data = capture; 393 } 394 395 static inline void snd_soc_dai_set_drvdata(struct snd_soc_dai *dai, 396 void *data) 397 { 398 dev_set_drvdata(dai->dev, data); 399 } 400 401 static inline void *snd_soc_dai_get_drvdata(struct snd_soc_dai *dai) 402 { 403 return dev_get_drvdata(dai->dev); 404 } 405 406 /** 407 * snd_soc_dai_set_sdw_stream() - Configures a DAI for SDW stream operation 408 * @dai: DAI 409 * @stream: STREAM 410 * @direction: Stream direction(Playback/Capture) 411 * SoundWire subsystem doesn't have a notion of direction and we reuse 412 * the ASoC stream direction to configure sink/source ports. 413 * Playback maps to source ports and Capture for sink ports. 414 * 415 * This should be invoked with NULL to clear the stream set previously. 416 * Returns 0 on success, a negative error code otherwise. 417 */ 418 static inline int snd_soc_dai_set_sdw_stream(struct snd_soc_dai *dai, 419 void *stream, int direction) 420 { 421 if (dai->driver->ops->set_sdw_stream) 422 return dai->driver->ops->set_sdw_stream(dai, stream, direction); 423 else 424 return -ENOTSUPP; 425 } 426 427 /** 428 * snd_soc_dai_get_sdw_stream() - Retrieves SDW stream from DAI 429 * @dai: DAI 430 * @direction: Stream direction(Playback/Capture) 431 * 432 * This routine only retrieves that was previously configured 433 * with snd_soc_dai_get_sdw_stream() 434 * 435 * Returns pointer to stream or -ENOTSUPP if callback is not supported; 436 */ 437 static inline void *snd_soc_dai_get_sdw_stream(struct snd_soc_dai *dai, 438 int direction) 439 { 440 if (dai->driver->ops->get_sdw_stream) 441 return dai->driver->ops->get_sdw_stream(dai, direction); 442 else 443 return ERR_PTR(-ENOTSUPP); 444 } 445 446 #endif 447