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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 20struct snd_pcm_substream; 21struct snd_soc_dapm_widget; 22struct 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_S24_3LE |\ 106 SNDRV_PCM_FMTBIT_S24_3BE |\ 107 SNDRV_PCM_FMTBIT_S32_LE |\ 108 SNDRV_PCM_FMTBIT_S32_BE) 109 110struct snd_soc_dai_driver; 111struct snd_soc_dai; 112struct snd_ac97_bus_ops; 113 114/* Digital Audio Interface clocking API.*/ 115int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, 116 unsigned int freq, int dir); 117 118int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai, 119 int div_id, int div); 120 121int snd_soc_dai_set_pll(struct snd_soc_dai *dai, 122 int pll_id, int source, unsigned int freq_in, unsigned int freq_out); 123 124int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio); 125 126/* Digital Audio interface formatting */ 127int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); 128 129int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, 130 unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width); 131 132int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai, 133 unsigned int tx_num, unsigned int *tx_slot, 134 unsigned int rx_num, unsigned int *rx_slot); 135 136int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); 137 138/* Digital Audio Interface mute */ 139int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute, 140 int direction); 141 142int snd_soc_dai_is_dummy(struct snd_soc_dai *dai); 143 144struct snd_soc_dai_ops { 145 /* 146 * DAI clocking configuration, all optional. 147 * Called by soc_card drivers, normally in their hw_params. 148 */ 149 int (*set_sysclk)(struct snd_soc_dai *dai, 150 int clk_id, unsigned int freq, int dir); 151 int (*set_pll)(struct snd_soc_dai *dai, int pll_id, int source, 152 unsigned int freq_in, unsigned int freq_out); 153 int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div); 154 int (*set_bclk_ratio)(struct snd_soc_dai *dai, unsigned int ratio); 155 156 /* 157 * DAI format configuration 158 * Called by soc_card drivers, normally in their hw_params. 159 */ 160 int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt); 161 int (*xlate_tdm_slot_mask)(unsigned int slots, 162 unsigned int *tx_mask, unsigned int *rx_mask); 163 int (*set_tdm_slot)(struct snd_soc_dai *dai, 164 unsigned int tx_mask, unsigned int rx_mask, 165 int slots, int slot_width); 166 int (*set_channel_map)(struct snd_soc_dai *dai, 167 unsigned int tx_num, unsigned int *tx_slot, 168 unsigned int rx_num, unsigned int *rx_slot); 169 int (*set_tristate)(struct snd_soc_dai *dai, int tristate); 170 171 /* 172 * DAI digital mute - optional. 173 * Called by soc-core to minimise any pops. 174 */ 175 int (*digital_mute)(struct snd_soc_dai *dai, int mute); 176 int (*mute_stream)(struct snd_soc_dai *dai, int mute, int stream); 177 178 /* 179 * ALSA PCM audio operations - all optional. 180 * Called by soc-core during audio PCM operations. 181 */ 182 int (*startup)(struct snd_pcm_substream *, 183 struct snd_soc_dai *); 184 void (*shutdown)(struct snd_pcm_substream *, 185 struct snd_soc_dai *); 186 int (*hw_params)(struct snd_pcm_substream *, 187 struct snd_pcm_hw_params *, struct snd_soc_dai *); 188 int (*hw_free)(struct snd_pcm_substream *, 189 struct snd_soc_dai *); 190 int (*prepare)(struct snd_pcm_substream *, 191 struct snd_soc_dai *); 192 /* 193 * NOTE: Commands passed to the trigger function are not necessarily 194 * compatible with the current state of the dai. For example this 195 * sequence of commands is possible: START STOP STOP. 196 * So do not unconditionally use refcounting functions in the trigger 197 * function, e.g. clk_enable/disable. 198 */ 199 int (*trigger)(struct snd_pcm_substream *, int, 200 struct snd_soc_dai *); 201 int (*bespoke_trigger)(struct snd_pcm_substream *, int, 202 struct snd_soc_dai *); 203 /* 204 * For hardware based FIFO caused delay reporting. 205 * Optional. 206 */ 207 snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *, 208 struct snd_soc_dai *); 209}; 210 211struct snd_soc_cdai_ops { 212 /* 213 * for compress ops 214 */ 215 int (*startup)(struct snd_compr_stream *, 216 struct snd_soc_dai *); 217 int (*shutdown)(struct snd_compr_stream *, 218 struct snd_soc_dai *); 219 int (*set_params)(struct snd_compr_stream *, 220 struct snd_compr_params *, struct snd_soc_dai *); 221 int (*get_params)(struct snd_compr_stream *, 222 struct snd_codec *, struct snd_soc_dai *); 223 int (*set_metadata)(struct snd_compr_stream *, 224 struct snd_compr_metadata *, struct snd_soc_dai *); 225 int (*get_metadata)(struct snd_compr_stream *, 226 struct snd_compr_metadata *, struct snd_soc_dai *); 227 int (*trigger)(struct snd_compr_stream *, int, 228 struct snd_soc_dai *); 229 int (*pointer)(struct snd_compr_stream *, 230 struct snd_compr_tstamp *, struct snd_soc_dai *); 231 int (*ack)(struct snd_compr_stream *, size_t, 232 struct snd_soc_dai *); 233}; 234 235/* 236 * Digital Audio Interface Driver. 237 * 238 * Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97 239 * operations and capabilities. Codec and platform drivers will register this 240 * structure for every DAI they have. 241 * 242 * This structure covers the clocking, formating and ALSA operations for each 243 * interface. 244 */ 245struct snd_soc_dai_driver { 246 /* DAI description */ 247 const char *name; 248 unsigned int id; 249 unsigned int base; 250 struct snd_soc_dobj dobj; 251 252 /* DAI driver callbacks */ 253 int (*probe)(struct snd_soc_dai *dai); 254 int (*remove)(struct snd_soc_dai *dai); 255 int (*suspend)(struct snd_soc_dai *dai); 256 int (*resume)(struct snd_soc_dai *dai); 257 /* compress dai */ 258 int (*compress_new)(struct snd_soc_pcm_runtime *rtd, int num); 259 /* Optional Callback used at pcm creation*/ 260 int (*pcm_new)(struct snd_soc_pcm_runtime *rtd, 261 struct snd_soc_dai *dai); 262 /* DAI is also used for the control bus */ 263 bool bus_control; 264 265 /* ops */ 266 const struct snd_soc_dai_ops *ops; 267 const struct snd_soc_cdai_ops *cops; 268 269 /* DAI capabilities */ 270 struct snd_soc_pcm_stream capture; 271 struct snd_soc_pcm_stream playback; 272 unsigned int symmetric_rates:1; 273 unsigned int symmetric_channels:1; 274 unsigned int symmetric_samplebits:1; 275 276 /* probe ordering - for components with runtime dependencies */ 277 int probe_order; 278 int remove_order; 279}; 280 281/* 282 * Digital Audio Interface runtime data. 283 * 284 * Holds runtime data for a DAI. 285 */ 286struct snd_soc_dai { 287 const char *name; 288 int id; 289 struct device *dev; 290 291 /* driver ops */ 292 struct snd_soc_dai_driver *driver; 293 294 /* DAI runtime info */ 295 unsigned int capture_active:1; /* stream is in use */ 296 unsigned int playback_active:1; /* stream is in use */ 297 unsigned int symmetric_rates:1; 298 unsigned int symmetric_channels:1; 299 unsigned int symmetric_samplebits:1; 300 unsigned int probed:1; 301 302 unsigned int active; 303 304 struct snd_soc_dapm_widget *playback_widget; 305 struct snd_soc_dapm_widget *capture_widget; 306 307 /* DAI DMA data */ 308 void *playback_dma_data; 309 void *capture_dma_data; 310 311 /* Symmetry data - only valid if symmetry is being enforced */ 312 unsigned int rate; 313 unsigned int channels; 314 unsigned int sample_bits; 315 316 /* parent platform/codec */ 317 struct snd_soc_codec *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 327static 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 334static 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 344static 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 351static 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 357static inline void *snd_soc_dai_get_drvdata(struct snd_soc_dai *dai) 358{ 359 return dev_get_drvdata(dai->dev); 360} 361 362#endif