1 /* 2 * linux/sound/soc-dai.h -- ALSA SoC Layer 3 * 4 * Copyright: 2005-2008 Wolfson Microelectronics. PLC. 5 * 6 * This program is free software; you can redistribute it and/or modify 7 * it under the terms of the GNU General Public License version 2 as 8 * published by the Free Software Foundation. 9 * 10 * Digital Audio Interface (DAI) API. 11 */ 12 13 #ifndef __LINUX_SND_SOC_DAI_H 14 #define __LINUX_SND_SOC_DAI_H 15 16 17 #include <linux/list.h> 18 #include <sound/asoc.h> 19 20 struct snd_pcm_substream; 21 struct snd_soc_dapm_widget; 22 struct snd_compr_stream; 23 24 /* 25 * DAI hardware audio formats. 26 * 27 * Describes the physical PCM data formating and clocking. Add new formats 28 * to the end. 29 */ 30 #define SND_SOC_DAIFMT_I2S SND_SOC_DAI_FORMAT_I2S 31 #define SND_SOC_DAIFMT_RIGHT_J SND_SOC_DAI_FORMAT_RIGHT_J 32 #define SND_SOC_DAIFMT_LEFT_J SND_SOC_DAI_FORMAT_LEFT_J 33 #define SND_SOC_DAIFMT_DSP_A SND_SOC_DAI_FORMAT_DSP_A 34 #define SND_SOC_DAIFMT_DSP_B SND_SOC_DAI_FORMAT_DSP_B 35 #define SND_SOC_DAIFMT_AC97 SND_SOC_DAI_FORMAT_AC97 36 #define SND_SOC_DAIFMT_PDM SND_SOC_DAI_FORMAT_PDM 37 38 /* left and right justified also known as MSB and LSB respectively */ 39 #define SND_SOC_DAIFMT_MSB SND_SOC_DAIFMT_LEFT_J 40 #define SND_SOC_DAIFMT_LSB SND_SOC_DAIFMT_RIGHT_J 41 42 /* 43 * DAI Clock gating. 44 * 45 * DAI bit clocks can be be gated (disabled) when the DAI is not 46 * sending or receiving PCM data in a frame. This can be used to save power. 47 */ 48 #define SND_SOC_DAIFMT_CONT (1 << 4) /* continuous clock */ 49 #define SND_SOC_DAIFMT_GATED (0 << 4) /* clock is gated */ 50 51 /* 52 * DAI hardware signal polarity. 53 * 54 * Specifies whether the DAI can also support inverted clocks for the specified 55 * format. 56 * 57 * BCLK: 58 * - "normal" polarity means signal is available at rising edge of BCLK 59 * - "inverted" polarity means signal is available at falling edge of BCLK 60 * 61 * FSYNC "normal" polarity depends on the frame format: 62 * - I2S: frame consists of left then right channel data. Left channel starts 63 * with falling FSYNC edge, right channel starts with rising FSYNC edge. 64 * - Left/Right Justified: frame consists of left then right channel data. 65 * Left channel starts with rising FSYNC edge, right channel starts with 66 * falling FSYNC edge. 67 * - DSP A/B: Frame starts with rising FSYNC edge. 68 * - AC97: Frame starts with rising FSYNC edge. 69 * 70 * "Negative" FSYNC polarity is the one opposite of "normal" polarity. 71 */ 72 #define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bit clock + frame */ 73 #define SND_SOC_DAIFMT_NB_IF (2 << 8) /* normal BCLK + inv FRM */ 74 #define SND_SOC_DAIFMT_IB_NF (3 << 8) /* invert BCLK + nor FRM */ 75 #define SND_SOC_DAIFMT_IB_IF (4 << 8) /* invert BCLK + FRM */ 76 77 /* 78 * DAI hardware clock masters. 79 * 80 * This is wrt the codec, the inverse is true for the interface 81 * i.e. if the codec is clk and FRM master then the interface is 82 * clk and frame slave. 83 */ 84 #define SND_SOC_DAIFMT_CBM_CFM (1 << 12) /* codec clk & FRM master */ 85 #define SND_SOC_DAIFMT_CBS_CFM (2 << 12) /* codec clk slave & FRM master */ 86 #define SND_SOC_DAIFMT_CBM_CFS (3 << 12) /* codec clk master & frame slave */ 87 #define SND_SOC_DAIFMT_CBS_CFS (4 << 12) /* codec clk & FRM slave */ 88 89 #define SND_SOC_DAIFMT_FORMAT_MASK 0x000f 90 #define SND_SOC_DAIFMT_CLOCK_MASK 0x00f0 91 #define SND_SOC_DAIFMT_INV_MASK 0x0f00 92 #define SND_SOC_DAIFMT_MASTER_MASK 0xf000 93 94 /* 95 * Master Clock Directions 96 */ 97 #define SND_SOC_CLOCK_IN 0 98 #define SND_SOC_CLOCK_OUT 1 99 100 #define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S8 |\ 101 SNDRV_PCM_FMTBIT_S16_LE |\ 102 SNDRV_PCM_FMTBIT_S16_BE |\ 103 SNDRV_PCM_FMTBIT_S20_3LE |\ 104 SNDRV_PCM_FMTBIT_S20_3BE |\ 105 SNDRV_PCM_FMTBIT_S20_LE |\ 106 SNDRV_PCM_FMTBIT_S20_BE |\ 107 SNDRV_PCM_FMTBIT_S24_3LE |\ 108 SNDRV_PCM_FMTBIT_S24_3BE |\ 109 SNDRV_PCM_FMTBIT_S32_LE |\ 110 SNDRV_PCM_FMTBIT_S32_BE) 111 112 struct snd_soc_dai_driver; 113 struct snd_soc_dai; 114 struct snd_ac97_bus_ops; 115 116 /* Digital Audio Interface clocking API.*/ 117 int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, 118 unsigned int freq, int dir); 119 120 int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai, 121 int div_id, int div); 122 123 int snd_soc_dai_set_pll(struct snd_soc_dai *dai, 124 int pll_id, int source, unsigned int freq_in, unsigned int freq_out); 125 126 int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio); 127 128 /* Digital Audio interface formatting */ 129 int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); 130 131 int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, 132 unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width); 133 134 int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai, 135 unsigned int tx_num, unsigned int *tx_slot, 136 unsigned int rx_num, unsigned int *rx_slot); 137 138 int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); 139 140 /* Digital Audio Interface mute */ 141 int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute, 142 int direction); 143 144 int snd_soc_dai_is_dummy(struct snd_soc_dai *dai); 145 146 struct snd_soc_dai_ops { 147 /* 148 * DAI clocking configuration, all optional. 149 * Called by soc_card drivers, normally in their hw_params. 150 */ 151 int (*set_sysclk)(struct snd_soc_dai *dai, 152 int clk_id, unsigned int freq, int dir); 153 int (*set_pll)(struct snd_soc_dai *dai, int pll_id, int source, 154 unsigned int freq_in, unsigned int freq_out); 155 int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div); 156 int (*set_bclk_ratio)(struct snd_soc_dai *dai, unsigned int ratio); 157 158 /* 159 * DAI format configuration 160 * Called by soc_card drivers, normally in their hw_params. 161 */ 162 int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt); 163 int (*xlate_tdm_slot_mask)(unsigned int slots, 164 unsigned int *tx_mask, unsigned int *rx_mask); 165 int (*set_tdm_slot)(struct snd_soc_dai *dai, 166 unsigned int tx_mask, unsigned int rx_mask, 167 int slots, int slot_width); 168 int (*set_channel_map)(struct snd_soc_dai *dai, 169 unsigned int tx_num, unsigned int *tx_slot, 170 unsigned int rx_num, unsigned int *rx_slot); 171 int (*set_tristate)(struct snd_soc_dai *dai, int tristate); 172 173 int (*set_sdw_stream)(struct snd_soc_dai *dai, 174 void *stream, int direction); 175 /* 176 * DAI digital mute - optional. 177 * Called by soc-core to minimise any pops. 178 */ 179 int (*digital_mute)(struct snd_soc_dai *dai, int mute); 180 int (*mute_stream)(struct snd_soc_dai *dai, int mute, int stream); 181 182 /* 183 * ALSA PCM audio operations - all optional. 184 * Called by soc-core during audio PCM operations. 185 */ 186 int (*startup)(struct snd_pcm_substream *, 187 struct snd_soc_dai *); 188 void (*shutdown)(struct snd_pcm_substream *, 189 struct snd_soc_dai *); 190 int (*hw_params)(struct snd_pcm_substream *, 191 struct snd_pcm_hw_params *, struct snd_soc_dai *); 192 int (*hw_free)(struct snd_pcm_substream *, 193 struct snd_soc_dai *); 194 int (*prepare)(struct snd_pcm_substream *, 195 struct snd_soc_dai *); 196 /* 197 * NOTE: Commands passed to the trigger function are not necessarily 198 * compatible with the current state of the dai. For example this 199 * sequence of commands is possible: START STOP STOP. 200 * So do not unconditionally use refcounting functions in the trigger 201 * function, e.g. clk_enable/disable. 202 */ 203 int (*trigger)(struct snd_pcm_substream *, int, 204 struct snd_soc_dai *); 205 int (*bespoke_trigger)(struct snd_pcm_substream *, int, 206 struct snd_soc_dai *); 207 /* 208 * For hardware based FIFO caused delay reporting. 209 * Optional. 210 */ 211 snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *, 212 struct snd_soc_dai *); 213 }; 214 215 struct snd_soc_cdai_ops { 216 /* 217 * for compress ops 218 */ 219 int (*startup)(struct snd_compr_stream *, 220 struct snd_soc_dai *); 221 int (*shutdown)(struct snd_compr_stream *, 222 struct snd_soc_dai *); 223 int (*set_params)(struct snd_compr_stream *, 224 struct snd_compr_params *, struct snd_soc_dai *); 225 int (*get_params)(struct snd_compr_stream *, 226 struct snd_codec *, struct snd_soc_dai *); 227 int (*set_metadata)(struct snd_compr_stream *, 228 struct snd_compr_metadata *, struct snd_soc_dai *); 229 int (*get_metadata)(struct snd_compr_stream *, 230 struct snd_compr_metadata *, struct snd_soc_dai *); 231 int (*trigger)(struct snd_compr_stream *, int, 232 struct snd_soc_dai *); 233 int (*pointer)(struct snd_compr_stream *, 234 struct snd_compr_tstamp *, struct snd_soc_dai *); 235 int (*ack)(struct snd_compr_stream *, size_t, 236 struct snd_soc_dai *); 237 }; 238 239 /* 240 * Digital Audio Interface Driver. 241 * 242 * Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97 243 * operations and capabilities. Codec and platform drivers will register this 244 * structure for every DAI they have. 245 * 246 * This structure covers the clocking, formating and ALSA operations for each 247 * interface. 248 */ 249 struct snd_soc_dai_driver { 250 /* DAI description */ 251 const char *name; 252 unsigned int id; 253 unsigned int base; 254 struct snd_soc_dobj dobj; 255 256 /* DAI driver callbacks */ 257 int (*probe)(struct snd_soc_dai *dai); 258 int (*remove)(struct snd_soc_dai *dai); 259 int (*suspend)(struct snd_soc_dai *dai); 260 int (*resume)(struct snd_soc_dai *dai); 261 /* compress dai */ 262 int (*compress_new)(struct snd_soc_pcm_runtime *rtd, int num); 263 /* Optional Callback used at pcm creation*/ 264 int (*pcm_new)(struct snd_soc_pcm_runtime *rtd, 265 struct snd_soc_dai *dai); 266 /* DAI is also used for the control bus */ 267 bool bus_control; 268 269 /* ops */ 270 const struct snd_soc_dai_ops *ops; 271 const struct snd_soc_cdai_ops *cops; 272 273 /* DAI capabilities */ 274 struct snd_soc_pcm_stream capture; 275 struct snd_soc_pcm_stream playback; 276 unsigned int symmetric_rates:1; 277 unsigned int symmetric_channels:1; 278 unsigned int symmetric_samplebits:1; 279 280 /* probe ordering - for components with runtime dependencies */ 281 int probe_order; 282 int remove_order; 283 }; 284 285 /* 286 * Digital Audio Interface runtime data. 287 * 288 * Holds runtime data for a DAI. 289 */ 290 struct snd_soc_dai { 291 const char *name; 292 int id; 293 struct device *dev; 294 295 /* driver ops */ 296 struct snd_soc_dai_driver *driver; 297 298 /* DAI runtime info */ 299 unsigned int capture_active; /* stream usage count */ 300 unsigned int playback_active; /* stream usage count */ 301 unsigned int probed:1; 302 303 unsigned int active; 304 305 struct snd_soc_dapm_widget *playback_widget; 306 struct snd_soc_dapm_widget *capture_widget; 307 308 /* DAI DMA data */ 309 void *playback_dma_data; 310 void *capture_dma_data; 311 312 /* Symmetry data - only valid if symmetry is being enforced */ 313 unsigned int rate; 314 unsigned int channels; 315 unsigned int sample_bits; 316 317 /* parent platform/codec */ 318 struct snd_soc_component *component; 319 320 /* CODEC TDM slot masks and params (for fixup) */ 321 unsigned int tx_mask; 322 unsigned int rx_mask; 323 324 struct list_head list; 325 }; 326 327 static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai *dai, 328 const struct snd_pcm_substream *ss) 329 { 330 return (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) ? 331 dai->playback_dma_data : dai->capture_dma_data; 332 } 333 334 static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai, 335 const struct snd_pcm_substream *ss, 336 void *data) 337 { 338 if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) 339 dai->playback_dma_data = data; 340 else 341 dai->capture_dma_data = data; 342 } 343 344 static inline void snd_soc_dai_init_dma_data(struct snd_soc_dai *dai, 345 void *playback, void *capture) 346 { 347 dai->playback_dma_data = playback; 348 dai->capture_dma_data = capture; 349 } 350 351 static inline void snd_soc_dai_set_drvdata(struct snd_soc_dai *dai, 352 void *data) 353 { 354 dev_set_drvdata(dai->dev, data); 355 } 356 357 static inline void *snd_soc_dai_get_drvdata(struct snd_soc_dai *dai) 358 { 359 return dev_get_drvdata(dai->dev); 360 } 361 362 /** 363 * snd_soc_dai_set_sdw_stream() - Configures a DAI for SDW stream operation 364 * @dai: DAI 365 * @stream: STREAM 366 * @direction: Stream direction(Playback/Capture) 367 * SoundWire subsystem doesn't have a notion of direction and we reuse 368 * the ASoC stream direction to configure sink/source ports. 369 * Playback maps to source ports and Capture for sink ports. 370 * 371 * This should be invoked with NULL to clear the stream set previously. 372 * Returns 0 on success, a negative error code otherwise. 373 */ 374 static inline int snd_soc_dai_set_sdw_stream(struct snd_soc_dai *dai, 375 void *stream, int direction) 376 { 377 if (dai->driver->ops->set_sdw_stream) 378 return dai->driver->ops->set_sdw_stream(dai, stream, direction); 379 else 380 return -ENOTSUPP; 381 } 382 383 #endif 384