| 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 (*xlate_tdm_slot_mask)(unsigned int slots, |
| 146 | unsigned int *tx_mask, unsigned int *rx_mask); |
| 147 | int (*set_tdm_slot)(struct snd_soc_dai *dai, |
| 148 | unsigned int tx_mask, unsigned int rx_mask, |
| 149 | int slots, int slot_width); |
| 150 | int (*set_channel_map)(struct snd_soc_dai *dai, |
| 151 | unsigned int tx_num, unsigned int *tx_slot, |
| 152 | unsigned int rx_num, unsigned int *rx_slot); |
| 153 | int (*set_tristate)(struct snd_soc_dai *dai, int tristate); |
| 154 | |
| 155 | /* |
| 156 | * DAI digital mute - optional. |
| 157 | * Called by soc-core to minimise any pops. |
| 158 | */ |
| 159 | int (*digital_mute)(struct snd_soc_dai *dai, int mute); |
| 160 | int (*mute_stream)(struct snd_soc_dai *dai, int mute, int stream); |
| 161 | |
| 162 | /* |
| 163 | * ALSA PCM audio operations - all optional. |
| 164 | * Called by soc-core during audio PCM operations. |
| 165 | */ |
| 166 | int (*startup)(struct snd_pcm_substream *, |
| 167 | struct snd_soc_dai *); |
| 168 | void (*shutdown)(struct snd_pcm_substream *, |
| 169 | struct snd_soc_dai *); |
| 170 | int (*hw_params)(struct snd_pcm_substream *, |
| 171 | struct snd_pcm_hw_params *, struct snd_soc_dai *); |
| 172 | int (*hw_free)(struct snd_pcm_substream *, |
| 173 | struct snd_soc_dai *); |
| 174 | int (*prepare)(struct snd_pcm_substream *, |
| 175 | struct snd_soc_dai *); |
| 176 | /* |
| 177 | * NOTE: Commands passed to the trigger function are not necessarily |
| 178 | * compatible with the current state of the dai. For example this |
| 179 | * sequence of commands is possible: START STOP STOP. |
| 180 | * So do not unconditionally use refcounting functions in the trigger |
| 181 | * function, e.g. clk_enable/disable. |
| 182 | */ |
| 183 | int (*trigger)(struct snd_pcm_substream *, int, |
| 184 | struct snd_soc_dai *); |
| 185 | int (*bespoke_trigger)(struct snd_pcm_substream *, int, |
| 186 | struct snd_soc_dai *); |
| 187 | /* |
| 188 | * For hardware based FIFO caused delay reporting. |
| 189 | * Optional. |
| 190 | */ |
| 191 | snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *, |
| 192 | struct snd_soc_dai *); |
| 193 | }; |
| 194 | |
| 195 | /* |
| 196 | * Digital Audio Interface Driver. |
| 197 | * |
| 198 | * Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97 |
| 199 | * operations and capabilities. Codec and platform drivers will register this |
| 200 | * structure for every DAI they have. |
| 201 | * |
| 202 | * This structure covers the clocking, formating and ALSA operations for each |
| 203 | * interface. |
| 204 | */ |
| 205 | struct snd_soc_dai_driver { |
| 206 | /* DAI description */ |
| 207 | const char *name; |
| 208 | unsigned int id; |
| 209 | unsigned int base; |
| 210 | |
| 211 | /* DAI driver callbacks */ |
| 212 | int (*probe)(struct snd_soc_dai *dai); |
| 213 | int (*remove)(struct snd_soc_dai *dai); |
| 214 | int (*suspend)(struct snd_soc_dai *dai); |
| 215 | int (*resume)(struct snd_soc_dai *dai); |
| 216 | /* compress dai */ |
| 217 | bool compress_dai; |
| 218 | /* DAI is also used for the control bus */ |
| 219 | bool bus_control; |
| 220 | |
| 221 | /* ops */ |
| 222 | const struct snd_soc_dai_ops *ops; |
| 223 | |
| 224 | /* DAI capabilities */ |
| 225 | struct snd_soc_pcm_stream capture; |
| 226 | struct snd_soc_pcm_stream playback; |
| 227 | unsigned int symmetric_rates:1; |
| 228 | unsigned int symmetric_channels:1; |
| 229 | unsigned int symmetric_samplebits:1; |
| 230 | |
| 231 | /* probe ordering - for components with runtime dependencies */ |
| 232 | int probe_order; |
| 233 | int remove_order; |
| 234 | }; |
| 235 | |
| 236 | /* |
| 237 | * Digital Audio Interface runtime data. |
| 238 | * |
| 239 | * Holds runtime data for a DAI. |
| 240 | */ |
| 241 | struct snd_soc_dai { |
| 242 | const char *name; |
| 243 | int id; |
| 244 | struct device *dev; |
| 245 | |
| 246 | /* driver ops */ |
| 247 | struct snd_soc_dai_driver *driver; |
| 248 | |
| 249 | /* DAI runtime info */ |
| 250 | unsigned int capture_active:1; /* stream is in use */ |
| 251 | unsigned int playback_active:1; /* stream is in use */ |
| 252 | unsigned int symmetric_rates:1; |
| 253 | unsigned int symmetric_channels:1; |
| 254 | unsigned int symmetric_samplebits:1; |
| 255 | unsigned int active; |
| 256 | unsigned char probed:1; |
| 257 | |
| 258 | struct snd_soc_dapm_widget *playback_widget; |
| 259 | struct snd_soc_dapm_widget *capture_widget; |
| 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_codec *codec; |
| 272 | struct snd_soc_component *component; |
| 273 | |
| 274 | /* CODEC TDM slot masks and params (for fixup) */ |
| 275 | unsigned int tx_mask; |
| 276 | unsigned int rx_mask; |
| 277 | |
| 278 | struct list_head list; |
| 279 | }; |
| 280 | |
| 281 | static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai *dai, |
| 282 | const struct snd_pcm_substream *ss) |
| 283 | { |
| 284 | return (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) ? |
| 285 | dai->playback_dma_data : dai->capture_dma_data; |
| 286 | } |
| 287 | |
| 288 | static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai, |
| 289 | const struct snd_pcm_substream *ss, |
| 290 | void *data) |
| 291 | { |
| 292 | if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) |
| 293 | dai->playback_dma_data = data; |
| 294 | else |
| 295 | dai->capture_dma_data = data; |
| 296 | } |
| 297 | |
| 298 | static inline void snd_soc_dai_init_dma_data(struct snd_soc_dai *dai, |
| 299 | void *playback, void *capture) |
| 300 | { |
| 301 | dai->playback_dma_data = playback; |
| 302 | dai->capture_dma_data = capture; |
| 303 | } |
| 304 | |
| 305 | static inline void snd_soc_dai_set_drvdata(struct snd_soc_dai *dai, |
| 306 | void *data) |
| 307 | { |
| 308 | dev_set_drvdata(dai->dev, data); |
| 309 | } |
| 310 | |
| 311 | static inline void *snd_soc_dai_get_drvdata(struct snd_soc_dai *dai) |
| 312 | { |
| 313 | return dev_get_drvdata(dai->dev); |
| 314 | } |
| 315 | |
| 316 | #endif |