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 inversions. 52 * 53 * Specifies whether the DAI can also support inverted clocks for the specified 54 * format. 55 */ 56 #define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bit clock + frame */ 57 #define SND_SOC_DAIFMT_NB_IF (2 << 8) /* normal BCLK + inv FRM */ 58 #define SND_SOC_DAIFMT_IB_NF (3 << 8) /* invert BCLK + nor FRM */ 59 #define SND_SOC_DAIFMT_IB_IF (4 << 8) /* invert BCLK + FRM */ 60 61 /* 62 * DAI hardware clock masters. 63 * 64 * This is wrt the codec, the inverse is true for the interface 65 * i.e. if the codec is clk and FRM master then the interface is 66 * clk and frame slave. 67 */ 68 #define SND_SOC_DAIFMT_CBM_CFM (1 << 12) /* codec clk & FRM master */ 69 #define SND_SOC_DAIFMT_CBS_CFM (2 << 12) /* codec clk slave & FRM master */ 70 #define SND_SOC_DAIFMT_CBM_CFS (3 << 12) /* codec clk master & frame slave */ 71 #define SND_SOC_DAIFMT_CBS_CFS (4 << 12) /* codec clk & FRM slave */ 72 73 #define SND_SOC_DAIFMT_FORMAT_MASK 0x000f 74 #define SND_SOC_DAIFMT_CLOCK_MASK 0x00f0 75 #define SND_SOC_DAIFMT_INV_MASK 0x0f00 76 #define SND_SOC_DAIFMT_MASTER_MASK 0xf000 77 78 /* 79 * Master Clock Directions 80 */ 81 #define SND_SOC_CLOCK_IN 0 82 #define SND_SOC_CLOCK_OUT 1 83 84 #define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S8 |\ 85 SNDRV_PCM_FMTBIT_S16_LE |\ 86 SNDRV_PCM_FMTBIT_S16_BE |\ 87 SNDRV_PCM_FMTBIT_S20_3LE |\ 88 SNDRV_PCM_FMTBIT_S20_3BE |\ 89 SNDRV_PCM_FMTBIT_S24_3LE |\ 90 SNDRV_PCM_FMTBIT_S24_3BE |\ 91 SNDRV_PCM_FMTBIT_S32_LE |\ 92 SNDRV_PCM_FMTBIT_S32_BE) 93 94 struct snd_soc_dai_driver; 95 struct snd_soc_dai; 96 struct snd_ac97_bus_ops; 97 98 /* Digital Audio Interface clocking API.*/ 99 int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, 100 unsigned int freq, int dir); 101 102 int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai, 103 int div_id, int div); 104 105 int snd_soc_dai_set_pll(struct snd_soc_dai *dai, 106 int pll_id, int source, unsigned int freq_in, unsigned int freq_out); 107 108 int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio); 109 110 /* Digital Audio interface formatting */ 111 int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); 112 113 int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, 114 unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width); 115 116 int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai, 117 unsigned int tx_num, unsigned int *tx_slot, 118 unsigned int rx_num, unsigned int *rx_slot); 119 120 int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); 121 122 /* Digital Audio Interface mute */ 123 int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute, 124 int direction); 125 126 int snd_soc_dai_is_dummy(struct snd_soc_dai *dai); 127 128 struct snd_soc_dai_ops { 129 /* 130 * DAI clocking configuration, all optional. 131 * Called by soc_card drivers, normally in their hw_params. 132 */ 133 int (*set_sysclk)(struct snd_soc_dai *dai, 134 int clk_id, unsigned int freq, int dir); 135 int (*set_pll)(struct snd_soc_dai *dai, int pll_id, int source, 136 unsigned int freq_in, unsigned int freq_out); 137 int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div); 138 int (*set_bclk_ratio)(struct snd_soc_dai *dai, unsigned int ratio); 139 140 /* 141 * DAI format configuration 142 * Called by soc_card drivers, normally in their hw_params. 143 */ 144 int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt); 145 int (*set_tdm_slot)(struct snd_soc_dai *dai, 146 unsigned int tx_mask, unsigned int rx_mask, 147 int slots, int slot_width); 148 int (*set_channel_map)(struct snd_soc_dai *dai, 149 unsigned int tx_num, unsigned int *tx_slot, 150 unsigned int rx_num, unsigned int *rx_slot); 151 int (*set_tristate)(struct snd_soc_dai *dai, int tristate); 152 153 /* 154 * DAI digital mute - optional. 155 * Called by soc-core to minimise any pops. 156 */ 157 int (*digital_mute)(struct snd_soc_dai *dai, int mute); 158 int (*mute_stream)(struct snd_soc_dai *dai, int mute, int stream); 159 160 /* 161 * ALSA PCM audio operations - all optional. 162 * Called by soc-core during audio PCM operations. 163 */ 164 int (*startup)(struct snd_pcm_substream *, 165 struct snd_soc_dai *); 166 void (*shutdown)(struct snd_pcm_substream *, 167 struct snd_soc_dai *); 168 int (*hw_params)(struct snd_pcm_substream *, 169 struct snd_pcm_hw_params *, struct snd_soc_dai *); 170 int (*hw_free)(struct snd_pcm_substream *, 171 struct snd_soc_dai *); 172 int (*prepare)(struct snd_pcm_substream *, 173 struct snd_soc_dai *); 174 /* 175 * NOTE: Commands passed to the trigger function are not necessarily 176 * compatible with the current state of the dai. For example this 177 * sequence of commands is possible: START STOP STOP. 178 * So do not unconditionally use refcounting functions in the trigger 179 * function, e.g. clk_enable/disable. 180 */ 181 int (*trigger)(struct snd_pcm_substream *, int, 182 struct snd_soc_dai *); 183 int (*bespoke_trigger)(struct snd_pcm_substream *, int, 184 struct snd_soc_dai *); 185 /* 186 * For hardware based FIFO caused delay reporting. 187 * Optional. 188 */ 189 snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *, 190 struct snd_soc_dai *); 191 }; 192 193 /* 194 * Digital Audio Interface Driver. 195 * 196 * Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97 197 * operations and capabilities. Codec and platform drivers will register this 198 * structure for every DAI they have. 199 * 200 * This structure covers the clocking, formating and ALSA operations for each 201 * interface. 202 */ 203 struct snd_soc_dai_driver { 204 /* DAI description */ 205 const char *name; 206 unsigned int id; 207 int ac97_control; 208 unsigned int base; 209 210 /* DAI driver callbacks */ 211 int (*probe)(struct snd_soc_dai *dai); 212 int (*remove)(struct snd_soc_dai *dai); 213 int (*suspend)(struct snd_soc_dai *dai); 214 int (*resume)(struct snd_soc_dai *dai); 215 /* compress dai */ 216 bool compress_dai; 217 218 /* ops */ 219 const struct snd_soc_dai_ops *ops; 220 221 /* DAI capabilities */ 222 struct snd_soc_pcm_stream capture; 223 struct snd_soc_pcm_stream playback; 224 unsigned int symmetric_rates:1; 225 unsigned int symmetric_channels:1; 226 unsigned int symmetric_samplebits:1; 227 228 /* probe ordering - for components with runtime dependencies */ 229 int probe_order; 230 int remove_order; 231 }; 232 233 /* 234 * Digital Audio Interface runtime data. 235 * 236 * Holds runtime data for a DAI. 237 */ 238 struct snd_soc_dai { 239 const char *name; 240 int id; 241 struct device *dev; 242 void *ac97_pdata; /* platform_data for the ac97 codec */ 243 244 /* driver ops */ 245 struct snd_soc_dai_driver *driver; 246 247 /* DAI runtime info */ 248 unsigned int capture_active:1; /* stream is in use */ 249 unsigned int playback_active:1; /* stream is in use */ 250 unsigned int symmetric_rates:1; 251 unsigned int symmetric_channels:1; 252 unsigned int symmetric_samplebits:1; 253 struct snd_pcm_runtime *runtime; 254 unsigned int active; 255 unsigned char probed:1; 256 257 struct snd_soc_dapm_widget *playback_widget; 258 struct snd_soc_dapm_widget *capture_widget; 259 struct snd_soc_dapm_context dapm; 260 261 /* DAI DMA data */ 262 void *playback_dma_data; 263 void *capture_dma_data; 264 265 /* Symmetry data - only valid if symmetry is being enforced */ 266 unsigned int rate; 267 unsigned int channels; 268 unsigned int sample_bits; 269 270 /* parent platform/codec */ 271 struct snd_soc_platform *platform; 272 struct snd_soc_codec *codec; 273 274 struct snd_soc_card *card; 275 276 struct list_head list; 277 struct list_head card_list; 278 }; 279 280 static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai *dai, 281 const struct snd_pcm_substream *ss) 282 { 283 return (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) ? 284 dai->playback_dma_data : dai->capture_dma_data; 285 } 286 287 static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai, 288 const struct snd_pcm_substream *ss, 289 void *data) 290 { 291 if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) 292 dai->playback_dma_data = data; 293 else 294 dai->capture_dma_data = data; 295 } 296 297 static inline void snd_soc_dai_init_dma_data(struct snd_soc_dai *dai, 298 void *playback, void *capture) 299 { 300 dai->playback_dma_data = playback; 301 dai->capture_dma_data = capture; 302 } 303 304 static inline void snd_soc_dai_set_drvdata(struct snd_soc_dai *dai, 305 void *data) 306 { 307 dev_set_drvdata(dai->dev, data); 308 } 309 310 static inline void *snd_soc_dai_get_drvdata(struct snd_soc_dai *dai) 311 { 312 return dev_get_drvdata(dai->dev); 313 } 314 315 #endif 316