Mark Brown | a47cbe7 | 2008-07-23 14:03:07 +0100 | [diff] [blame] | 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 | |
| 21 | /* |
| 22 | * DAI hardware audio formats. |
| 23 | * |
| 24 | * Describes the physical PCM data formating and clocking. Add new formats |
| 25 | * to the end. |
| 26 | */ |
| 27 | #define SND_SOC_DAIFMT_I2S 0 /* I2S mode */ |
| 28 | #define SND_SOC_DAIFMT_RIGHT_J 1 /* Right Justified mode */ |
| 29 | #define SND_SOC_DAIFMT_LEFT_J 2 /* Left Justified mode */ |
Peter Meerwald | 47db8e89 | 2009-07-13 23:05:11 +0100 | [diff] [blame] | 30 | #define SND_SOC_DAIFMT_DSP_A 3 /* L data MSB after FRM LRC */ |
| 31 | #define SND_SOC_DAIFMT_DSP_B 4 /* L data MSB during FRM LRC */ |
Mark Brown | a47cbe7 | 2008-07-23 14:03:07 +0100 | [diff] [blame] | 32 | #define SND_SOC_DAIFMT_AC97 5 /* AC97 */ |
Lopez Cruz, Misael | be2500b | 2009-09-25 21:02:49 -0500 | [diff] [blame] | 33 | #define SND_SOC_DAIFMT_PDM 6 /* Pulse density modulation */ |
Mark Brown | a47cbe7 | 2008-07-23 14:03:07 +0100 | [diff] [blame] | 34 | |
| 35 | /* left and right justified also known as MSB and LSB respectively */ |
| 36 | #define SND_SOC_DAIFMT_MSB SND_SOC_DAIFMT_LEFT_J |
| 37 | #define SND_SOC_DAIFMT_LSB SND_SOC_DAIFMT_RIGHT_J |
| 38 | |
| 39 | /* |
| 40 | * DAI Clock gating. |
| 41 | * |
Peter Meerwald | 47db8e89 | 2009-07-13 23:05:11 +0100 | [diff] [blame] | 42 | * DAI bit clocks can be be gated (disabled) when the DAI is not |
Mark Brown | a47cbe7 | 2008-07-23 14:03:07 +0100 | [diff] [blame] | 43 | * sending or receiving PCM data in a frame. This can be used to save power. |
| 44 | */ |
| 45 | #define SND_SOC_DAIFMT_CONT (0 << 4) /* continuous clock */ |
| 46 | #define SND_SOC_DAIFMT_GATED (1 << 4) /* clock is gated */ |
| 47 | |
| 48 | /* |
Mark Brown | a47cbe7 | 2008-07-23 14:03:07 +0100 | [diff] [blame] | 49 | * DAI hardware signal inversions. |
| 50 | * |
| 51 | * Specifies whether the DAI can also support inverted clocks for the specified |
| 52 | * format. |
| 53 | */ |
| 54 | #define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bit clock + frame */ |
Peter Meerwald | 47db8e89 | 2009-07-13 23:05:11 +0100 | [diff] [blame] | 55 | #define SND_SOC_DAIFMT_NB_IF (1 << 8) /* normal BCLK + inv FRM */ |
| 56 | #define SND_SOC_DAIFMT_IB_NF (2 << 8) /* invert BCLK + nor FRM */ |
| 57 | #define SND_SOC_DAIFMT_IB_IF (3 << 8) /* invert BCLK + FRM */ |
Mark Brown | a47cbe7 | 2008-07-23 14:03:07 +0100 | [diff] [blame] | 58 | |
| 59 | /* |
| 60 | * DAI hardware clock masters. |
| 61 | * |
| 62 | * This is wrt the codec, the inverse is true for the interface |
Peter Meerwald | 47db8e89 | 2009-07-13 23:05:11 +0100 | [diff] [blame] | 63 | * i.e. if the codec is clk and FRM master then the interface is |
Mark Brown | a47cbe7 | 2008-07-23 14:03:07 +0100 | [diff] [blame] | 64 | * clk and frame slave. |
| 65 | */ |
Peter Meerwald | 47db8e89 | 2009-07-13 23:05:11 +0100 | [diff] [blame] | 66 | #define SND_SOC_DAIFMT_CBM_CFM (0 << 12) /* codec clk & FRM master */ |
| 67 | #define SND_SOC_DAIFMT_CBS_CFM (1 << 12) /* codec clk slave & FRM master */ |
Mark Brown | a47cbe7 | 2008-07-23 14:03:07 +0100 | [diff] [blame] | 68 | #define SND_SOC_DAIFMT_CBM_CFS (2 << 12) /* codec clk master & frame slave */ |
Peter Meerwald | 47db8e89 | 2009-07-13 23:05:11 +0100 | [diff] [blame] | 69 | #define SND_SOC_DAIFMT_CBS_CFS (3 << 12) /* codec clk & FRM slave */ |
Mark Brown | a47cbe7 | 2008-07-23 14:03:07 +0100 | [diff] [blame] | 70 | |
| 71 | #define SND_SOC_DAIFMT_FORMAT_MASK 0x000f |
| 72 | #define SND_SOC_DAIFMT_CLOCK_MASK 0x00f0 |
| 73 | #define SND_SOC_DAIFMT_INV_MASK 0x0f00 |
| 74 | #define SND_SOC_DAIFMT_MASTER_MASK 0xf000 |
| 75 | |
| 76 | /* |
| 77 | * Master Clock Directions |
| 78 | */ |
| 79 | #define SND_SOC_CLOCK_IN 0 |
| 80 | #define SND_SOC_CLOCK_OUT 1 |
| 81 | |
Mark Brown | 8f738d5 | 2009-08-09 20:08:31 +0100 | [diff] [blame] | 82 | #define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S8 |\ |
| 83 | SNDRV_PCM_FMTBIT_S16_LE |\ |
| 84 | SNDRV_PCM_FMTBIT_S16_BE |\ |
| 85 | SNDRV_PCM_FMTBIT_S20_3LE |\ |
| 86 | SNDRV_PCM_FMTBIT_S20_3BE |\ |
| 87 | SNDRV_PCM_FMTBIT_S24_3LE |\ |
| 88 | SNDRV_PCM_FMTBIT_S24_3BE |\ |
Jon Smirl | d34c430 | 2009-05-13 21:59:14 -0400 | [diff] [blame] | 89 | SNDRV_PCM_FMTBIT_S32_LE |\ |
| 90 | SNDRV_PCM_FMTBIT_S32_BE) |
Mark Brown | 33f503c | 2009-05-02 12:24:55 +0100 | [diff] [blame] | 91 | |
Liam Girdwood | f0fba2a | 2010-03-17 20:15:21 +0000 | [diff] [blame] | 92 | struct snd_soc_dai_driver; |
Mark Brown | a47cbe7 | 2008-07-23 14:03:07 +0100 | [diff] [blame] | 93 | struct snd_soc_dai; |
| 94 | struct snd_ac97_bus_ops; |
| 95 | |
Mark Brown | 9115171 | 2008-11-30 23:31:24 +0000 | [diff] [blame] | 96 | /* Digital Audio Interface registration */ |
Liam Girdwood | f0fba2a | 2010-03-17 20:15:21 +0000 | [diff] [blame] | 97 | int snd_soc_register_dai(struct device *dev, |
| 98 | struct snd_soc_dai_driver *dai_drv); |
| 99 | void snd_soc_unregister_dai(struct device *dev); |
| 100 | int snd_soc_register_dais(struct device *dev, |
| 101 | struct snd_soc_dai_driver *dai_drv, size_t count); |
| 102 | void snd_soc_unregister_dais(struct device *dev, size_t count); |
Mark Brown | 9115171 | 2008-11-30 23:31:24 +0000 | [diff] [blame] | 103 | |
Mark Brown | a47cbe7 | 2008-07-23 14:03:07 +0100 | [diff] [blame] | 104 | /* Digital Audio Interface clocking API.*/ |
| 105 | int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, |
| 106 | unsigned int freq, int dir); |
| 107 | |
| 108 | int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai, |
| 109 | int div_id, int div); |
| 110 | |
| 111 | int snd_soc_dai_set_pll(struct snd_soc_dai *dai, |
Mark Brown | 8548803 | 2009-09-05 18:52:16 +0100 | [diff] [blame] | 112 | int pll_id, int source, unsigned int freq_in, unsigned int freq_out); |
Mark Brown | a47cbe7 | 2008-07-23 14:03:07 +0100 | [diff] [blame] | 113 | |
| 114 | /* Digital Audio interface formatting */ |
| 115 | int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); |
| 116 | |
| 117 | int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, |
Daniel Ribeiro | a5479e3 | 2009-06-15 21:44:31 -0300 | [diff] [blame] | 118 | unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width); |
Mark Brown | a47cbe7 | 2008-07-23 14:03:07 +0100 | [diff] [blame] | 119 | |
Barry Song | 472df3c | 2009-09-12 01:16:29 +0800 | [diff] [blame] | 120 | int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai, |
| 121 | unsigned int tx_num, unsigned int *tx_slot, |
| 122 | unsigned int rx_num, unsigned int *rx_slot); |
| 123 | |
Mark Brown | a47cbe7 | 2008-07-23 14:03:07 +0100 | [diff] [blame] | 124 | int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); |
| 125 | |
| 126 | /* Digital Audio Interface mute */ |
| 127 | int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute); |
| 128 | |
Mark Brown | a47cbe7 | 2008-07-23 14:03:07 +0100 | [diff] [blame] | 129 | struct snd_soc_dai_ops { |
| 130 | /* |
| 131 | * DAI clocking configuration, all optional. |
| 132 | * Called by soc_card drivers, normally in their hw_params. |
| 133 | */ |
| 134 | int (*set_sysclk)(struct snd_soc_dai *dai, |
| 135 | int clk_id, unsigned int freq, int dir); |
Mark Brown | 8548803 | 2009-09-05 18:52:16 +0100 | [diff] [blame] | 136 | int (*set_pll)(struct snd_soc_dai *dai, int pll_id, int source, |
| 137 | unsigned int freq_in, unsigned int freq_out); |
Mark Brown | a47cbe7 | 2008-07-23 14:03:07 +0100 | [diff] [blame] | 138 | int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div); |
| 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, |
Daniel Ribeiro | a5479e3 | 2009-06-15 21:44:31 -0300 | [diff] [blame] | 146 | unsigned int tx_mask, unsigned int rx_mask, |
| 147 | int slots, int slot_width); |
Barry Song | 472df3c | 2009-09-12 01:16:29 +0800 | [diff] [blame] | 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); |
Mark Brown | a47cbe7 | 2008-07-23 14:03:07 +0100 | [diff] [blame] | 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); |
Mark Brown | dee89c4 | 2008-11-18 22:11:38 +0000 | [diff] [blame] | 158 | |
| 159 | /* |
| 160 | * ALSA PCM audio operations - all optional. |
| 161 | * Called by soc-core during audio PCM operations. |
| 162 | */ |
| 163 | int (*startup)(struct snd_pcm_substream *, |
| 164 | struct snd_soc_dai *); |
| 165 | void (*shutdown)(struct snd_pcm_substream *, |
| 166 | struct snd_soc_dai *); |
| 167 | int (*hw_params)(struct snd_pcm_substream *, |
| 168 | struct snd_pcm_hw_params *, struct snd_soc_dai *); |
| 169 | int (*hw_free)(struct snd_pcm_substream *, |
| 170 | struct snd_soc_dai *); |
| 171 | int (*prepare)(struct snd_pcm_substream *, |
| 172 | struct snd_soc_dai *); |
| 173 | int (*trigger)(struct snd_pcm_substream *, int, |
| 174 | struct snd_soc_dai *); |
Peter Ujfalusi | 258020d | 2010-03-03 15:08:07 +0200 | [diff] [blame] | 175 | /* |
| 176 | * For hardware based FIFO caused delay reporting. |
| 177 | * Optional. |
| 178 | */ |
| 179 | snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *, |
| 180 | struct snd_soc_dai *); |
Mark Brown | a47cbe7 | 2008-07-23 14:03:07 +0100 | [diff] [blame] | 181 | }; |
| 182 | |
| 183 | /* |
Liam Girdwood | f0fba2a | 2010-03-17 20:15:21 +0000 | [diff] [blame] | 184 | * Digital Audio Interface Driver. |
Mark Brown | a47cbe7 | 2008-07-23 14:03:07 +0100 | [diff] [blame] | 185 | * |
Liam Girdwood | f0fba2a | 2010-03-17 20:15:21 +0000 | [diff] [blame] | 186 | * Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97 |
| 187 | * operations and capabilities. Codec and platform drivers will register this |
| 188 | * structure for every DAI they have. |
| 189 | * |
| 190 | * This structure covers the clocking, formating and ALSA operations for each |
| 191 | * interface. |
Mark Brown | a47cbe7 | 2008-07-23 14:03:07 +0100 | [diff] [blame] | 192 | */ |
Liam Girdwood | f0fba2a | 2010-03-17 20:15:21 +0000 | [diff] [blame] | 193 | struct snd_soc_dai_driver { |
Mark Brown | a47cbe7 | 2008-07-23 14:03:07 +0100 | [diff] [blame] | 194 | /* DAI description */ |
Liam Girdwood | f0fba2a | 2010-03-17 20:15:21 +0000 | [diff] [blame] | 195 | const char *name; |
Mark Brown | a47cbe7 | 2008-07-23 14:03:07 +0100 | [diff] [blame] | 196 | unsigned int id; |
Mark Brown | 3ba9e10a | 2008-11-24 18:01:05 +0000 | [diff] [blame] | 197 | int ac97_control; |
Mark Brown | a47cbe7 | 2008-07-23 14:03:07 +0100 | [diff] [blame] | 198 | |
Liam Girdwood | f0fba2a | 2010-03-17 20:15:21 +0000 | [diff] [blame] | 199 | /* DAI driver callbacks */ |
| 200 | int (*probe)(struct snd_soc_dai *dai); |
| 201 | int (*remove)(struct snd_soc_dai *dai); |
Mark Brown | dc7d7b8 | 2008-12-03 18:21:52 +0000 | [diff] [blame] | 202 | int (*suspend)(struct snd_soc_dai *dai); |
| 203 | int (*resume)(struct snd_soc_dai *dai); |
Mark Brown | a47cbe7 | 2008-07-23 14:03:07 +0100 | [diff] [blame] | 204 | |
| 205 | /* ops */ |
Mark Brown | 1ee46eb | 2010-12-02 16:10:09 +0000 | [diff] [blame] | 206 | const struct snd_soc_dai_ops *ops; |
Mark Brown | a47cbe7 | 2008-07-23 14:03:07 +0100 | [diff] [blame] | 207 | |
| 208 | /* DAI capabilities */ |
| 209 | struct snd_soc_pcm_stream capture; |
| 210 | struct snd_soc_pcm_stream playback; |
Mark Brown | 06f409d | 2009-04-07 18:10:13 +0100 | [diff] [blame] | 211 | unsigned int symmetric_rates:1; |
Liam Girdwood | 0168bf0 | 2011-06-07 16:08:05 +0100 | [diff] [blame] | 212 | |
| 213 | /* probe ordering - for components with runtime dependencies */ |
| 214 | int probe_order; |
| 215 | int remove_order; |
Liam Girdwood | f0fba2a | 2010-03-17 20:15:21 +0000 | [diff] [blame] | 216 | }; |
| 217 | |
| 218 | /* |
| 219 | * Digital Audio Interface runtime data. |
| 220 | * |
| 221 | * Holds runtime data for a DAI. |
| 222 | */ |
| 223 | struct snd_soc_dai { |
| 224 | const char *name; |
| 225 | int id; |
| 226 | struct device *dev; |
| 227 | void *ac97_pdata; /* platform_data for the ac97 codec */ |
| 228 | |
| 229 | /* driver ops */ |
| 230 | struct snd_soc_dai_driver *driver; |
Mark Brown | a47cbe7 | 2008-07-23 14:03:07 +0100 | [diff] [blame] | 231 | |
| 232 | /* DAI runtime info */ |
Liam Girdwood | f0fba2a | 2010-03-17 20:15:21 +0000 | [diff] [blame] | 233 | unsigned int capture_active:1; /* stream is in use */ |
| 234 | unsigned int playback_active:1; /* stream is in use */ |
| 235 | unsigned int symmetric_rates:1; |
| 236 | struct snd_pcm_runtime *runtime; |
Mark Brown | a47cbe7 | 2008-07-23 14:03:07 +0100 | [diff] [blame] | 237 | unsigned int active; |
| 238 | unsigned char pop_wait:1; |
Liam Girdwood | f0fba2a | 2010-03-17 20:15:21 +0000 | [diff] [blame] | 239 | unsigned char probed:1; |
Mark Brown | a47cbe7 | 2008-07-23 14:03:07 +0100 | [diff] [blame] | 240 | |
Liam Girdwood | f0fba2a | 2010-03-17 20:15:21 +0000 | [diff] [blame] | 241 | /* DAI DMA data */ |
| 242 | void *playback_dma_data; |
| 243 | void *capture_dma_data; |
Mark Brown | a47cbe7 | 2008-07-23 14:03:07 +0100 | [diff] [blame] | 244 | |
Dong Aisheng | 1784102 | 2011-08-29 17:15:14 +0800 | [diff] [blame^] | 245 | /* Symmetry data - only valid if symmetry is being enforced */ |
| 246 | unsigned int rate; |
| 247 | |
Liam Girdwood | f0fba2a | 2010-03-17 20:15:21 +0000 | [diff] [blame] | 248 | /* parent platform/codec */ |
| 249 | union { |
| 250 | struct snd_soc_platform *platform; |
| 251 | struct snd_soc_codec *codec; |
| 252 | }; |
| 253 | struct snd_soc_card *card; |
Mark Brown | a47cbe7 | 2008-07-23 14:03:07 +0100 | [diff] [blame] | 254 | |
| 255 | struct list_head list; |
Liam Girdwood | f0fba2a | 2010-03-17 20:15:21 +0000 | [diff] [blame] | 256 | struct list_head card_list; |
Mark Brown | a47cbe7 | 2008-07-23 14:03:07 +0100 | [diff] [blame] | 257 | }; |
| 258 | |
Daniel Mack | fd23b7d | 2010-03-19 14:52:55 +0000 | [diff] [blame] | 259 | static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai *dai, |
| 260 | const struct snd_pcm_substream *ss) |
| 261 | { |
| 262 | return (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) ? |
Liam Girdwood | f0fba2a | 2010-03-17 20:15:21 +0000 | [diff] [blame] | 263 | dai->playback_dma_data : dai->capture_dma_data; |
Daniel Mack | fd23b7d | 2010-03-19 14:52:55 +0000 | [diff] [blame] | 264 | } |
| 265 | |
| 266 | static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai, |
| 267 | const struct snd_pcm_substream *ss, |
| 268 | void *data) |
| 269 | { |
| 270 | if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) |
Liam Girdwood | f0fba2a | 2010-03-17 20:15:21 +0000 | [diff] [blame] | 271 | dai->playback_dma_data = data; |
Daniel Mack | fd23b7d | 2010-03-19 14:52:55 +0000 | [diff] [blame] | 272 | else |
Liam Girdwood | f0fba2a | 2010-03-17 20:15:21 +0000 | [diff] [blame] | 273 | dai->capture_dma_data = data; |
| 274 | } |
| 275 | |
| 276 | static inline void snd_soc_dai_set_drvdata(struct snd_soc_dai *dai, |
| 277 | void *data) |
| 278 | { |
| 279 | dev_set_drvdata(dai->dev, data); |
| 280 | } |
| 281 | |
| 282 | static inline void *snd_soc_dai_get_drvdata(struct snd_soc_dai *dai) |
| 283 | { |
| 284 | return dev_get_drvdata(dai->dev); |
Daniel Mack | fd23b7d | 2010-03-19 14:52:55 +0000 | [diff] [blame] | 285 | } |
| 286 | |
Mark Brown | a47cbe7 | 2008-07-23 14:03:07 +0100 | [diff] [blame] | 287 | #endif |