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