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 19 struct snd_pcm_substream; 20 struct snd_soc_dapm_widget; 21 struct snd_compr_stream; 22 23 /* 24 * DAI hardware audio formats. 25 * 26 * Describes the physical PCM data formating and clocking. Add new formats 27 * to the end. 28 */ 29 #define SND_SOC_DAIFMT_I2S 1 /* I2S mode */ 30 #define SND_SOC_DAIFMT_RIGHT_J 2 /* Right Justified mode */ 31 #define SND_SOC_DAIFMT_LEFT_J 3 /* Left Justified mode */ 32 #define SND_SOC_DAIFMT_DSP_A 4 /* L data MSB after FRM LRC */ 33 #define SND_SOC_DAIFMT_DSP_B 5 /* L data MSB during FRM LRC */ 34 #define SND_SOC_DAIFMT_AC97 6 /* AC97 */ 35 #define SND_SOC_DAIFMT_PDM 7 /* Pulse density modulation */ 36 37 /* left and right justified also known as MSB and LSB respectively */ 38 #define SND_SOC_DAIFMT_MSB SND_SOC_DAIFMT_LEFT_J 39 #define SND_SOC_DAIFMT_LSB SND_SOC_DAIFMT_RIGHT_J 40 41 /* 42 * DAI Clock gating. 43 * 44 * DAI bit clocks can be be gated (disabled) when the DAI is not 45 * sending or receiving PCM data in a frame. This can be used to save power. 46 */ 47 #define SND_SOC_DAIFMT_CONT (1 << 4) /* continuous clock */ 48 #define SND_SOC_DAIFMT_GATED (0 << 4) /* clock is gated */ 49 50 /* 51 * DAI hardware signal polarity. 52 * 53 * Specifies whether the DAI can also support inverted clocks for the specified 54 * format. 55 * 56 * BCLK: 57 * - "normal" polarity means signal is available at rising edge of BCLK 58 * - "inverted" polarity means signal is available at falling edge of BCLK 59 * 60 * FSYNC "normal" polarity depends on the frame format: 61 * - I2S: frame consists of left then right channel data. Left channel starts 62 * with falling FSYNC edge, right channel starts with rising FSYNC edge. 63 * - Left/Right Justified: frame consists of left then right channel data. 64 * Left channel starts with rising FSYNC edge, right channel starts with 65 * falling FSYNC edge. 66 * - DSP A/B: Frame starts with rising FSYNC edge. 67 * - AC97: Frame starts with rising FSYNC edge. 68 * 69 * "Negative" FSYNC polarity is the one opposite of "normal" polarity. 70 */ 71 #define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bit clock + frame */ 72 #define SND_SOC_DAIFMT_NB_IF (2 << 8) /* normal BCLK + inv FRM */ 73 #define SND_SOC_DAIFMT_IB_NF (3 << 8) /* invert BCLK + nor FRM */ 74 #define SND_SOC_DAIFMT_IB_IF (4 << 8) /* invert BCLK + FRM */ 75 76 /* 77 * DAI hardware clock masters. 78 * 79 * This is wrt the codec, the inverse is true for the interface 80 * i.e. if the codec is clk and FRM master then the interface is 81 * clk and frame slave. 82 */ 83 #define SND_SOC_DAIFMT_CBM_CFM (1 << 12) /* codec clk & FRM master */ 84 #define SND_SOC_DAIFMT_CBS_CFM (2 << 12) /* codec clk slave & FRM master */ 85 #define SND_SOC_DAIFMT_CBM_CFS (3 << 12) /* codec clk master & frame slave */ 86 #define SND_SOC_DAIFMT_CBS_CFS (4 << 12) /* codec clk & FRM slave */ 87 88 #define SND_SOC_DAIFMT_FORMAT_MASK 0x000f 89 #define SND_SOC_DAIFMT_CLOCK_MASK 0x00f0 90 #define SND_SOC_DAIFMT_INV_MASK 0x0f00 91 #define SND_SOC_DAIFMT_MASTER_MASK 0xf000 92 93 /* 94 * Master Clock Directions 95 */ 96 #define SND_SOC_CLOCK_IN 0 97 #define SND_SOC_CLOCK_OUT 1 98 99 #define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S8 |\ 100 SNDRV_PCM_FMTBIT_S16_LE |\ 101 SNDRV_PCM_FMTBIT_S16_BE |\ 102 SNDRV_PCM_FMTBIT_S20_3LE |\ 103 SNDRV_PCM_FMTBIT_S20_3BE |\ 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 int snd_soc_dai_is_dummy(struct snd_soc_dai *dai); 142 143 struct snd_soc_dai_ops { 144 /* 145 * DAI clocking configuration, all optional. 146 * Called by soc_card drivers, normally in their hw_params. 147 */ 148 int (*set_sysclk)(struct snd_soc_dai *dai, 149 int clk_id, unsigned int freq, int dir); 150 int (*set_pll)(struct snd_soc_dai *dai, int pll_id, int source, 151 unsigned int freq_in, unsigned int freq_out); 152 int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div); 153 int (*set_bclk_ratio)(struct snd_soc_dai *dai, unsigned int ratio); 154 155 /* 156 * DAI format configuration 157 * Called by soc_card drivers, normally in their hw_params. 158 */ 159 int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt); 160 int (*xlate_tdm_slot_mask)(unsigned int slots, 161 unsigned int *tx_mask, unsigned int *rx_mask); 162 int (*set_tdm_slot)(struct snd_soc_dai *dai, 163 unsigned int tx_mask, unsigned int rx_mask, 164 int slots, int slot_width); 165 int (*set_channel_map)(struct snd_soc_dai *dai, 166 unsigned int tx_num, unsigned int *tx_slot, 167 unsigned int rx_num, unsigned int *rx_slot); 168 int (*set_tristate)(struct snd_soc_dai *dai, int tristate); 169 170 /* 171 * DAI digital mute - optional. 172 * Called by soc-core to minimise any pops. 173 */ 174 int (*digital_mute)(struct snd_soc_dai *dai, int mute); 175 int (*mute_stream)(struct snd_soc_dai *dai, int mute, int stream); 176 177 /* 178 * ALSA PCM audio operations - all optional. 179 * Called by soc-core during audio PCM operations. 180 */ 181 int (*startup)(struct snd_pcm_substream *, 182 struct snd_soc_dai *); 183 void (*shutdown)(struct snd_pcm_substream *, 184 struct snd_soc_dai *); 185 int (*hw_params)(struct snd_pcm_substream *, 186 struct snd_pcm_hw_params *, struct snd_soc_dai *); 187 int (*hw_free)(struct snd_pcm_substream *, 188 struct snd_soc_dai *); 189 int (*prepare)(struct snd_pcm_substream *, 190 struct snd_soc_dai *); 191 /* 192 * NOTE: Commands passed to the trigger function are not necessarily 193 * compatible with the current state of the dai. For example this 194 * sequence of commands is possible: START STOP STOP. 195 * So do not unconditionally use refcounting functions in the trigger 196 * function, e.g. clk_enable/disable. 197 */ 198 int (*trigger)(struct snd_pcm_substream *, int, 199 struct snd_soc_dai *); 200 int (*bespoke_trigger)(struct snd_pcm_substream *, int, 201 struct snd_soc_dai *); 202 /* 203 * For hardware based FIFO caused delay reporting. 204 * Optional. 205 */ 206 snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *, 207 struct snd_soc_dai *); 208 }; 209 210 /* 211 * Digital Audio Interface Driver. 212 * 213 * Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97 214 * operations and capabilities. Codec and platform drivers will register this 215 * structure for every DAI they have. 216 * 217 * This structure covers the clocking, formating and ALSA operations for each 218 * interface. 219 */ 220 struct snd_soc_dai_driver { 221 /* DAI description */ 222 const char *name; 223 unsigned int id; 224 unsigned int base; 225 226 /* DAI driver callbacks */ 227 int (*probe)(struct snd_soc_dai *dai); 228 int (*remove)(struct snd_soc_dai *dai); 229 int (*suspend)(struct snd_soc_dai *dai); 230 int (*resume)(struct snd_soc_dai *dai); 231 /* compress dai */ 232 int (*compress_new)(struct snd_soc_pcm_runtime *rtd, int num); 233 /* DAI is also used for the control bus */ 234 bool bus_control; 235 236 /* ops */ 237 const struct snd_soc_dai_ops *ops; 238 239 /* DAI capabilities */ 240 struct snd_soc_pcm_stream capture; 241 struct snd_soc_pcm_stream playback; 242 unsigned int symmetric_rates:1; 243 unsigned int symmetric_channels:1; 244 unsigned int symmetric_samplebits:1; 245 246 /* probe ordering - for components with runtime dependencies */ 247 int probe_order; 248 int remove_order; 249 }; 250 251 /* 252 * Digital Audio Interface runtime data. 253 * 254 * Holds runtime data for a DAI. 255 */ 256 struct snd_soc_dai { 257 const char *name; 258 int id; 259 struct device *dev; 260 261 /* driver ops */ 262 struct snd_soc_dai_driver *driver; 263 264 /* DAI runtime info */ 265 unsigned int capture_active:1; /* stream is in use */ 266 unsigned int playback_active:1; /* stream is in use */ 267 unsigned int symmetric_rates:1; 268 unsigned int symmetric_channels:1; 269 unsigned int symmetric_samplebits:1; 270 unsigned int active; 271 unsigned char probed:1; 272 273 struct snd_soc_dapm_widget *playback_widget; 274 struct snd_soc_dapm_widget *capture_widget; 275 276 /* DAI DMA data */ 277 void *playback_dma_data; 278 void *capture_dma_data; 279 280 /* Symmetry data - only valid if symmetry is being enforced */ 281 unsigned int rate; 282 unsigned int channels; 283 unsigned int sample_bits; 284 285 /* parent platform/codec */ 286 struct snd_soc_codec *codec; 287 struct snd_soc_component *component; 288 289 /* CODEC TDM slot masks and params (for fixup) */ 290 unsigned int tx_mask; 291 unsigned int rx_mask; 292 293 struct list_head list; 294 }; 295 296 static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai *dai, 297 const struct snd_pcm_substream *ss) 298 { 299 return (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) ? 300 dai->playback_dma_data : dai->capture_dma_data; 301 } 302 303 static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai, 304 const struct snd_pcm_substream *ss, 305 void *data) 306 { 307 if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) 308 dai->playback_dma_data = data; 309 else 310 dai->capture_dma_data = data; 311 } 312 313 static inline void snd_soc_dai_init_dma_data(struct snd_soc_dai *dai, 314 void *playback, void *capture) 315 { 316 dai->playback_dma_data = playback; 317 dai->capture_dma_data = capture; 318 } 319 320 static inline void snd_soc_dai_set_drvdata(struct snd_soc_dai *dai, 321 void *data) 322 { 323 dev_set_drvdata(dai->dev, data); 324 } 325 326 static inline void *snd_soc_dai_get_drvdata(struct snd_soc_dai *dai) 327 { 328 return dev_get_drvdata(dai->dev); 329 } 330 331 #endif 332