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1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48 49 50 51 52 53 54 55 56 57 58 59 60 61 62 63 64 65 66 67 68 69 70 71 72 73 74 75 76 77 78 79 80 81 82 83 84 85 86 87 88 89 90 91 92 93 94 95 96 97 98 99 100 101 102 103 104 105 106 107 108 109 110 111 112 113 114 115 116 117 118 119 120 121 122 123 124 125 126 127 128 129 130 131 132 133 134 135 136 137 138 139 140 141 142 143 144 145 146 147 148 149 150 151 152 153 154 155 156 157 158 159 160 161 162 163 164 165 166 167 168 169 170 171 172 173 174 175 176 177 178 179 180 181 182 183 184 185 186 187 188 189 190 191 192 193 194 195 196 197 198 199 200 201 202 203 204 205 206 207 208 209 210 211 212 213 214 215 216 217 218 219 220 221 222 223 224 225 226 227 228 229 230 231 232 233 234 235 236 237 238 239 240 241 242 243 244 245 246 247 248 249 250 251 252 253 254 255 256 257 258 259 260 261 262 263 264 265 266 267 268 269 270 271 272 273 274 275 276 277 278 279 280 281 282 283 284 285 286 287 288 289 290 291 292 293 294 295 296 297 298 299 300 301 302 303 304 305 306 307 308 309 310 311 312 313 314 315 316 317 318 319 320 321 322 323 324 325 326 327 328 329 330 331 | /* * linux/sound/soc-dai.h -- ALSA SoC Layer * * Copyright: 2005-2008 Wolfson Microelectronics. PLC. * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as * published by the Free Software Foundation. * * Digital Audio Interface (DAI) API. */ #ifndef __LINUX_SND_SOC_DAI_H #define __LINUX_SND_SOC_DAI_H #include <linux/list.h> struct snd_pcm_substream; struct snd_soc_dapm_widget; struct snd_compr_stream; /* * DAI hardware audio formats. * * Describes the physical PCM data formating and clocking. Add new formats * to the end. */ #define SND_SOC_DAIFMT_I2S 1 /* I2S mode */ #define SND_SOC_DAIFMT_RIGHT_J 2 /* Right Justified mode */ #define SND_SOC_DAIFMT_LEFT_J 3 /* Left Justified mode */ #define SND_SOC_DAIFMT_DSP_A 4 /* L data MSB after FRM LRC */ #define SND_SOC_DAIFMT_DSP_B 5 /* L data MSB during FRM LRC */ #define SND_SOC_DAIFMT_AC97 6 /* AC97 */ #define SND_SOC_DAIFMT_PDM 7 /* Pulse density modulation */ /* left and right justified also known as MSB and LSB respectively */ #define SND_SOC_DAIFMT_MSB SND_SOC_DAIFMT_LEFT_J #define SND_SOC_DAIFMT_LSB SND_SOC_DAIFMT_RIGHT_J /* * DAI Clock gating. * * DAI bit clocks can be be gated (disabled) when the DAI is not * sending or receiving PCM data in a frame. This can be used to save power. */ #define SND_SOC_DAIFMT_CONT (1 << 4) /* continuous clock */ #define SND_SOC_DAIFMT_GATED (0 << 4) /* clock is gated */ /* * DAI hardware signal polarity. * * Specifies whether the DAI can also support inverted clocks for the specified * format. * * BCLK: * - "normal" polarity means signal is available at rising edge of BCLK * - "inverted" polarity means signal is available at falling edge of BCLK * * FSYNC "normal" polarity depends on the frame format: * - I2S: frame consists of left then right channel data. Left channel starts * with falling FSYNC edge, right channel starts with rising FSYNC edge. * - Left/Right Justified: frame consists of left then right channel data. * Left channel starts with rising FSYNC edge, right channel starts with * falling FSYNC edge. * - DSP A/B: Frame starts with rising FSYNC edge. * - AC97: Frame starts with rising FSYNC edge. * * "Negative" FSYNC polarity is the one opposite of "normal" polarity. */ #define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bit clock + frame */ #define SND_SOC_DAIFMT_NB_IF (2 << 8) /* normal BCLK + inv FRM */ #define SND_SOC_DAIFMT_IB_NF (3 << 8) /* invert BCLK + nor FRM */ #define SND_SOC_DAIFMT_IB_IF (4 << 8) /* invert BCLK + FRM */ /* * DAI hardware clock masters. * * This is wrt the codec, the inverse is true for the interface * i.e. if the codec is clk and FRM master then the interface is * clk and frame slave. */ #define SND_SOC_DAIFMT_CBM_CFM (1 << 12) /* codec clk & FRM master */ #define SND_SOC_DAIFMT_CBS_CFM (2 << 12) /* codec clk slave & FRM master */ #define SND_SOC_DAIFMT_CBM_CFS (3 << 12) /* codec clk master & frame slave */ #define SND_SOC_DAIFMT_CBS_CFS (4 << 12) /* codec clk & FRM slave */ #define SND_SOC_DAIFMT_FORMAT_MASK 0x000f #define SND_SOC_DAIFMT_CLOCK_MASK 0x00f0 #define SND_SOC_DAIFMT_INV_MASK 0x0f00 #define SND_SOC_DAIFMT_MASTER_MASK 0xf000 /* * Master Clock Directions */ #define SND_SOC_CLOCK_IN 0 #define SND_SOC_CLOCK_OUT 1 #define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S8 |\ SNDRV_PCM_FMTBIT_S16_LE |\ SNDRV_PCM_FMTBIT_S16_BE |\ SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S20_3BE |\ SNDRV_PCM_FMTBIT_S24_3LE |\ SNDRV_PCM_FMTBIT_S24_3BE |\ SNDRV_PCM_FMTBIT_S32_LE |\ SNDRV_PCM_FMTBIT_S32_BE) struct snd_soc_dai_driver; struct snd_soc_dai; struct snd_ac97_bus_ops; /* Digital Audio Interface clocking API.*/ int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, unsigned int freq, int dir); int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div); int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int pll_id, int source, unsigned int freq_in, unsigned int freq_out); int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio); /* Digital Audio interface formatting */ int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width); int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai, unsigned int tx_num, unsigned int *tx_slot, unsigned int rx_num, unsigned int *rx_slot); int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); /* Digital Audio Interface mute */ int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute, int direction); int snd_soc_dai_is_dummy(struct snd_soc_dai *dai); struct snd_soc_dai_ops { /* * DAI clocking configuration, all optional. * Called by soc_card drivers, normally in their hw_params. */ int (*set_sysclk)(struct snd_soc_dai *dai, int clk_id, unsigned int freq, int dir); int (*set_pll)(struct snd_soc_dai *dai, int pll_id, int source, unsigned int freq_in, unsigned int freq_out); int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div); int (*set_bclk_ratio)(struct snd_soc_dai *dai, unsigned int ratio); /* * DAI format configuration * Called by soc_card drivers, normally in their hw_params. */ int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt); int (*xlate_tdm_slot_mask)(unsigned int slots, unsigned int *tx_mask, unsigned int *rx_mask); int (*set_tdm_slot)(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width); int (*set_channel_map)(struct snd_soc_dai *dai, unsigned int tx_num, unsigned int *tx_slot, unsigned int rx_num, unsigned int *rx_slot); int (*set_tristate)(struct snd_soc_dai *dai, int tristate); /* * DAI digital mute - optional. * Called by soc-core to minimise any pops. */ int (*digital_mute)(struct snd_soc_dai *dai, int mute); int (*mute_stream)(struct snd_soc_dai *dai, int mute, int stream); /* * ALSA PCM audio operations - all optional. * Called by soc-core during audio PCM operations. */ int (*startup)(struct snd_pcm_substream *, struct snd_soc_dai *); void (*shutdown)(struct snd_pcm_substream *, struct snd_soc_dai *); int (*hw_params)(struct snd_pcm_substream *, struct snd_pcm_hw_params *, struct snd_soc_dai *); int (*hw_free)(struct snd_pcm_substream *, struct snd_soc_dai *); int (*prepare)(struct snd_pcm_substream *, struct snd_soc_dai *); /* * NOTE: Commands passed to the trigger function are not necessarily * compatible with the current state of the dai. For example this * sequence of commands is possible: START STOP STOP. * So do not unconditionally use refcounting functions in the trigger * function, e.g. clk_enable/disable. */ int (*trigger)(struct snd_pcm_substream *, int, struct snd_soc_dai *); int (*bespoke_trigger)(struct snd_pcm_substream *, int, struct snd_soc_dai *); /* * For hardware based FIFO caused delay reporting. * Optional. */ snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *, struct snd_soc_dai *); }; /* * Digital Audio Interface Driver. * * Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97 * operations and capabilities. Codec and platform drivers will register this * structure for every DAI they have. * * This structure covers the clocking, formating and ALSA operations for each * interface. */ struct snd_soc_dai_driver { /* DAI description */ const char *name; unsigned int id; unsigned int base; /* DAI driver callbacks */ int (*probe)(struct snd_soc_dai *dai); int (*remove)(struct snd_soc_dai *dai); int (*suspend)(struct snd_soc_dai *dai); int (*resume)(struct snd_soc_dai *dai); /* compress dai */ int (*compress_new)(struct snd_soc_pcm_runtime *rtd, int num); /* DAI is also used for the control bus */ bool bus_control; /* ops */ const struct snd_soc_dai_ops *ops; /* DAI capabilities */ struct snd_soc_pcm_stream capture; struct snd_soc_pcm_stream playback; unsigned int symmetric_rates:1; unsigned int symmetric_channels:1; unsigned int symmetric_samplebits:1; /* probe ordering - for components with runtime dependencies */ int probe_order; int remove_order; }; /* * Digital Audio Interface runtime data. * * Holds runtime data for a DAI. */ struct snd_soc_dai { const char *name; int id; struct device *dev; /* driver ops */ struct snd_soc_dai_driver *driver; /* DAI runtime info */ unsigned int capture_active:1; /* stream is in use */ unsigned int playback_active:1; /* stream is in use */ unsigned int symmetric_rates:1; unsigned int symmetric_channels:1; unsigned int symmetric_samplebits:1; unsigned int active; unsigned char probed:1; struct snd_soc_dapm_widget *playback_widget; struct snd_soc_dapm_widget *capture_widget; /* DAI DMA data */ void *playback_dma_data; void *capture_dma_data; /* Symmetry data - only valid if symmetry is being enforced */ unsigned int rate; unsigned int channels; unsigned int sample_bits; /* parent platform/codec */ struct snd_soc_codec *codec; struct snd_soc_component *component; /* CODEC TDM slot masks and params (for fixup) */ unsigned int tx_mask; unsigned int rx_mask; struct list_head list; }; static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai *dai, const struct snd_pcm_substream *ss) { return (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) ? dai->playback_dma_data : dai->capture_dma_data; } static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai, const struct snd_pcm_substream *ss, void *data) { if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) dai->playback_dma_data = data; else dai->capture_dma_data = data; } static inline void snd_soc_dai_init_dma_data(struct snd_soc_dai *dai, void *playback, void *capture) { dai->playback_dma_data = playback; dai->capture_dma_data = capture; } static inline void snd_soc_dai_set_drvdata(struct snd_soc_dai *dai, void *data) { dev_set_drvdata(dai->dev, data); } static inline void *snd_soc_dai_get_drvdata(struct snd_soc_dai *dai) { return dev_get_drvdata(dai->dev); } #endif |