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_S24_3LE |\ 106 SNDRV_PCM_FMTBIT_S24_3BE |\ 107 SNDRV_PCM_FMTBIT_S32_LE |\ 108 SNDRV_PCM_FMTBIT_S32_BE) 109 110 struct snd_soc_dai_driver; 111 struct snd_soc_dai; 112 struct snd_ac97_bus_ops; 113 114 /* Digital Audio Interface clocking API.*/ 115 int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, 116 unsigned int freq, int dir); 117 118 int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai, 119 int div_id, int div); 120 121 int snd_soc_dai_set_pll(struct snd_soc_dai *dai, 122 int pll_id, int source, unsigned int freq_in, unsigned int freq_out); 123 124 int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio); 125 126 /* Digital Audio interface formatting */ 127 int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); 128 129 int 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); 131 132 int 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); 135 136 int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); 137 138 /* Digital Audio Interface mute */ 139 int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute, 140 int direction); 141 142 int snd_soc_dai_is_dummy(struct snd_soc_dai *dai); 143 144 struct 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 }; 210 211 struct 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 */ 245 struct 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 /* DAI is also used for the control bus */ 260 bool bus_control; 261 262 /* ops */ 263 const struct snd_soc_dai_ops *ops; 264 const struct snd_soc_cdai_ops *cops; 265 266 /* DAI capabilities */ 267 struct snd_soc_pcm_stream capture; 268 struct snd_soc_pcm_stream playback; 269 unsigned int symmetric_rates:1; 270 unsigned int symmetric_channels:1; 271 unsigned int symmetric_samplebits:1; 272 273 /* probe ordering - for components with runtime dependencies */ 274 int probe_order; 275 int remove_order; 276 }; 277 278 /* 279 * Digital Audio Interface runtime data. 280 * 281 * Holds runtime data for a DAI. 282 */ 283 struct snd_soc_dai { 284 const char *name; 285 int id; 286 struct device *dev; 287 288 /* driver ops */ 289 struct snd_soc_dai_driver *driver; 290 291 /* DAI runtime info */ 292 unsigned int capture_active:1; /* stream is in use */ 293 unsigned int playback_active:1; /* stream is in use */ 294 unsigned int symmetric_rates:1; 295 unsigned int symmetric_channels:1; 296 unsigned int symmetric_samplebits:1; 297 unsigned int probed:1; 298 299 unsigned int active; 300 301 struct snd_soc_dapm_widget *playback_widget; 302 struct snd_soc_dapm_widget *capture_widget; 303 304 /* DAI DMA data */ 305 void *playback_dma_data; 306 void *capture_dma_data; 307 308 /* Symmetry data - only valid if symmetry is being enforced */ 309 unsigned int rate; 310 unsigned int channels; 311 unsigned int sample_bits; 312 313 /* parent platform/codec */ 314 struct snd_soc_codec *codec; 315 struct snd_soc_component *component; 316 317 /* CODEC TDM slot masks and params (for fixup) */ 318 unsigned int tx_mask; 319 unsigned int rx_mask; 320 321 struct list_head list; 322 }; 323 324 static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai *dai, 325 const struct snd_pcm_substream *ss) 326 { 327 return (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) ? 328 dai->playback_dma_data : dai->capture_dma_data; 329 } 330 331 static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai, 332 const struct snd_pcm_substream *ss, 333 void *data) 334 { 335 if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) 336 dai->playback_dma_data = data; 337 else 338 dai->capture_dma_data = data; 339 } 340 341 static inline void snd_soc_dai_init_dma_data(struct snd_soc_dai *dai, 342 void *playback, void *capture) 343 { 344 dai->playback_dma_data = playback; 345 dai->capture_dma_data = capture; 346 } 347 348 static inline void snd_soc_dai_set_drvdata(struct snd_soc_dai *dai, 349 void *data) 350 { 351 dev_set_drvdata(dai->dev, data); 352 } 353 354 static inline void *snd_soc_dai_get_drvdata(struct snd_soc_dai *dai) 355 { 356 return dev_get_drvdata(dai->dev); 357 } 358 359 #endif 360