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 #include <sound/soc.h> 20 21 struct snd_pcm_substream; 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 0 /* I2S mode */ 30 #define SND_SOC_DAIFMT_RIGHT_J 1 /* Right Justified mode */ 31 #define SND_SOC_DAIFMT_LEFT_J 2 /* Left Justified mode */ 32 #define SND_SOC_DAIFMT_DSP_A 3 /* L data MSB after FRM LRC */ 33 #define SND_SOC_DAIFMT_DSP_B 4 /* L data MSB during FRM LRC */ 34 #define SND_SOC_DAIFMT_AC97 5 /* AC97 */ 35 #define SND_SOC_DAIFMT_PDM 6 /* 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 (0 << 4) /* continuous clock */ 48 #define SND_SOC_DAIFMT_GATED (1 << 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 (1 << 8) /* normal BCLK + inv FRM */ 58 #define SND_SOC_DAIFMT_IB_NF (2 << 8) /* invert BCLK + nor FRM */ 59 #define SND_SOC_DAIFMT_IB_IF (3 << 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 (0 << 12) /* codec clk & FRM master */ 69 #define SND_SOC_DAIFMT_CBS_CFM (1 << 12) /* codec clk slave & FRM master */ 70 #define SND_SOC_DAIFMT_CBM_CFS (2 << 12) /* codec clk master & frame slave */ 71 #define SND_SOC_DAIFMT_CBS_CFS (3 << 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_ops; 95 struct snd_soc_dai; 96 struct snd_ac97_bus_ops; 97 98 /* Digital Audio Interface registration */ 99 int snd_soc_register_dai(struct snd_soc_dai *dai); 100 void snd_soc_unregister_dai(struct snd_soc_dai *dai); 101 int snd_soc_register_dais(struct snd_soc_dai *dai, size_t count); 102 void snd_soc_unregister_dais(struct snd_soc_dai *dai, size_t count); 103 104 /* 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, 112 int pll_id, int source, unsigned int freq_in, unsigned int freq_out); 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, 118 unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width); 119 120 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); 123 124 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); 128 129 /* 130 * Digital Audio Interface. 131 * 132 * Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97 133 * operations and capabilities. Codec and platform drivers will register this 134 * structure for every DAI they have. 135 * 136 * This structure covers the clocking, formating and ALSA operations for each 137 * interface. 138 */ 139 struct snd_soc_dai_ops { 140 /* 141 * DAI clocking configuration, all optional. 142 * Called by soc_card drivers, normally in their hw_params. 143 */ 144 int (*set_sysclk)(struct snd_soc_dai *dai, 145 int clk_id, unsigned int freq, int dir); 146 int (*set_pll)(struct snd_soc_dai *dai, int pll_id, int source, 147 unsigned int freq_in, unsigned int freq_out); 148 int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div); 149 150 /* 151 * DAI format configuration 152 * Called by soc_card drivers, normally in their hw_params. 153 */ 154 int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt); 155 int (*set_tdm_slot)(struct snd_soc_dai *dai, 156 unsigned int tx_mask, unsigned int rx_mask, 157 int slots, int slot_width); 158 int (*set_channel_map)(struct snd_soc_dai *dai, 159 unsigned int tx_num, unsigned int *tx_slot, 160 unsigned int rx_num, unsigned int *rx_slot); 161 int (*set_tristate)(struct snd_soc_dai *dai, int tristate); 162 163 /* 164 * DAI digital mute - optional. 165 * Called by soc-core to minimise any pops. 166 */ 167 int (*digital_mute)(struct snd_soc_dai *dai, int mute); 168 169 /* 170 * ALSA PCM audio operations - all optional. 171 * Called by soc-core during audio PCM operations. 172 */ 173 int (*startup)(struct snd_pcm_substream *, 174 struct snd_soc_dai *); 175 void (*shutdown)(struct snd_pcm_substream *, 176 struct snd_soc_dai *); 177 int (*hw_params)(struct snd_pcm_substream *, 178 struct snd_pcm_hw_params *, struct snd_soc_dai *); 179 int (*hw_free)(struct snd_pcm_substream *, 180 struct snd_soc_dai *); 181 int (*prepare)(struct snd_pcm_substream *, 182 struct snd_soc_dai *); 183 int (*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 runtime data. 195 * 196 * Holds runtime data for a DAI. 197 */ 198 struct snd_soc_dai { 199 /* DAI description */ 200 char *name; 201 unsigned int id; 202 int ac97_control; 203 204 struct device *dev; 205 void *ac97_pdata; /* platform_data for the ac97 codec */ 206 207 /* DAI callbacks */ 208 int (*probe)(struct platform_device *pdev, 209 struct snd_soc_dai *dai); 210 void (*remove)(struct platform_device *pdev, 211 struct snd_soc_dai *dai); 212 int (*suspend)(struct snd_soc_dai *dai); 213 int (*resume)(struct snd_soc_dai *dai); 214 215 /* ops */ 216 struct snd_soc_dai_ops *ops; 217 218 /* DAI capabilities */ 219 struct snd_soc_pcm_stream capture; 220 struct snd_soc_pcm_stream playback; 221 unsigned int symmetric_rates:1; 222 223 /* DAI runtime info */ 224 struct snd_soc_codec *codec; 225 unsigned int active; 226 unsigned char pop_wait:1; 227 228 /* DAI private data */ 229 void *private_data; 230 231 /* parent platform */ 232 struct snd_soc_platform *platform; 233 234 struct list_head list; 235 }; 236 237 static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai *dai, 238 const struct snd_pcm_substream *ss) 239 { 240 return (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) ? 241 dai->playback.dma_data : dai->capture.dma_data; 242 } 243 244 static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai, 245 const struct snd_pcm_substream *ss, 246 void *data) 247 { 248 if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) 249 dai->playback.dma_data = data; 250 else 251 dai->capture.dma_data = data; 252 } 253 254 #endif 255