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 /* 174 * DAI digital mute - optional. 175 * Called by soc-core to minimise any pops. 176 */ 177 int (*digital_mute)(struct snd_soc_dai *dai, int mute); 178 int (*mute_stream)(struct snd_soc_dai *dai, int mute, int stream); 179 180 /* 181 * ALSA PCM audio operations - all optional. 182 * Called by soc-core during audio PCM operations. 183 */ 184 int (*startup)(struct snd_pcm_substream *, 185 struct snd_soc_dai *); 186 void (*shutdown)(struct snd_pcm_substream *, 187 struct snd_soc_dai *); 188 int (*hw_params)(struct snd_pcm_substream *, 189 struct snd_pcm_hw_params *, struct snd_soc_dai *); 190 int (*hw_free)(struct snd_pcm_substream *, 191 struct snd_soc_dai *); 192 int (*prepare)(struct snd_pcm_substream *, 193 struct snd_soc_dai *); 194 /* 195 * NOTE: Commands passed to the trigger function are not necessarily 196 * compatible with the current state of the dai. For example this 197 * sequence of commands is possible: START STOP STOP. 198 * So do not unconditionally use refcounting functions in the trigger 199 * function, e.g. clk_enable/disable. 200 */ 201 int (*trigger)(struct snd_pcm_substream *, int, 202 struct snd_soc_dai *); 203 int (*bespoke_trigger)(struct snd_pcm_substream *, int, 204 struct snd_soc_dai *); 205 /* 206 * For hardware based FIFO caused delay reporting. 207 * Optional. 208 */ 209 snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *, 210 struct snd_soc_dai *); 211 }; 212 213 struct snd_soc_cdai_ops { 214 /* 215 * for compress ops 216 */ 217 int (*startup)(struct snd_compr_stream *, 218 struct snd_soc_dai *); 219 int (*shutdown)(struct snd_compr_stream *, 220 struct snd_soc_dai *); 221 int (*set_params)(struct snd_compr_stream *, 222 struct snd_compr_params *, struct snd_soc_dai *); 223 int (*get_params)(struct snd_compr_stream *, 224 struct snd_codec *, struct snd_soc_dai *); 225 int (*set_metadata)(struct snd_compr_stream *, 226 struct snd_compr_metadata *, struct snd_soc_dai *); 227 int (*get_metadata)(struct snd_compr_stream *, 228 struct snd_compr_metadata *, struct snd_soc_dai *); 229 int (*trigger)(struct snd_compr_stream *, int, 230 struct snd_soc_dai *); 231 int (*pointer)(struct snd_compr_stream *, 232 struct snd_compr_tstamp *, struct snd_soc_dai *); 233 int (*ack)(struct snd_compr_stream *, size_t, 234 struct snd_soc_dai *); 235 }; 236 237 /* 238 * Digital Audio Interface Driver. 239 * 240 * Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97 241 * operations and capabilities. Codec and platform drivers will register this 242 * structure for every DAI they have. 243 * 244 * This structure covers the clocking, formating and ALSA operations for each 245 * interface. 246 */ 247 struct snd_soc_dai_driver { 248 /* DAI description */ 249 const char *name; 250 unsigned int id; 251 unsigned int base; 252 struct snd_soc_dobj dobj; 253 254 /* DAI driver callbacks */ 255 int (*probe)(struct snd_soc_dai *dai); 256 int (*remove)(struct snd_soc_dai *dai); 257 int (*suspend)(struct snd_soc_dai *dai); 258 int (*resume)(struct snd_soc_dai *dai); 259 /* compress dai */ 260 int (*compress_new)(struct snd_soc_pcm_runtime *rtd, int num); 261 /* Optional Callback used at pcm creation*/ 262 int (*pcm_new)(struct snd_soc_pcm_runtime *rtd, 263 struct snd_soc_dai *dai); 264 /* DAI is also used for the control bus */ 265 bool bus_control; 266 267 /* ops */ 268 const struct snd_soc_dai_ops *ops; 269 const struct snd_soc_cdai_ops *cops; 270 271 /* DAI capabilities */ 272 struct snd_soc_pcm_stream capture; 273 struct snd_soc_pcm_stream playback; 274 unsigned int symmetric_rates:1; 275 unsigned int symmetric_channels:1; 276 unsigned int symmetric_samplebits:1; 277 278 /* probe ordering - for components with runtime dependencies */ 279 int probe_order; 280 int remove_order; 281 }; 282 283 /* 284 * Digital Audio Interface runtime data. 285 * 286 * Holds runtime data for a DAI. 287 */ 288 struct snd_soc_dai { 289 const char *name; 290 int id; 291 struct device *dev; 292 293 /* driver ops */ 294 struct snd_soc_dai_driver *driver; 295 296 /* DAI runtime info */ 297 unsigned int capture_active:1; /* stream is in use */ 298 unsigned int playback_active:1; /* stream is in use */ 299 unsigned int probed:1; 300 301 unsigned int active; 302 303 struct snd_soc_dapm_widget *playback_widget; 304 struct snd_soc_dapm_widget *capture_widget; 305 306 /* DAI DMA data */ 307 void *playback_dma_data; 308 void *capture_dma_data; 309 310 /* Symmetry data - only valid if symmetry is being enforced */ 311 unsigned int rate; 312 unsigned int channels; 313 unsigned int sample_bits; 314 315 /* parent platform/codec */ 316 struct snd_soc_codec *codec; 317 struct snd_soc_component *component; 318 319 /* CODEC TDM slot masks and params (for fixup) */ 320 unsigned int tx_mask; 321 unsigned int rx_mask; 322 323 struct list_head list; 324 }; 325 326 static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai *dai, 327 const struct snd_pcm_substream *ss) 328 { 329 return (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) ? 330 dai->playback_dma_data : dai->capture_dma_data; 331 } 332 333 static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai, 334 const struct snd_pcm_substream *ss, 335 void *data) 336 { 337 if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) 338 dai->playback_dma_data = data; 339 else 340 dai->capture_dma_data = data; 341 } 342 343 static inline void snd_soc_dai_init_dma_data(struct snd_soc_dai *dai, 344 void *playback, void *capture) 345 { 346 dai->playback_dma_data = playback; 347 dai->capture_dma_data = capture; 348 } 349 350 static inline void snd_soc_dai_set_drvdata(struct snd_soc_dai *dai, 351 void *data) 352 { 353 dev_set_drvdata(dai->dev, data); 354 } 355 356 static inline void *snd_soc_dai_get_drvdata(struct snd_soc_dai *dai) 357 { 358 return dev_get_drvdata(dai->dev); 359 } 360 361 #endif 362