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 */ |
| 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 */ |
| 32 | #define SND_SOC_DAIFMT_AC97 5 /* AC97 */ |
| 33 | |
| 34 | /* left and right justified also known as MSB and LSB respectively */ |
| 35 | #define SND_SOC_DAIFMT_MSB SND_SOC_DAIFMT_LEFT_J |
| 36 | #define SND_SOC_DAIFMT_LSB SND_SOC_DAIFMT_RIGHT_J |
| 37 | |
| 38 | /* |
| 39 | * DAI Clock gating. |
| 40 | * |
| 41 | * DAI bit clocks can be be gated (disabled) when not the DAI is not |
| 42 | * sending or receiving PCM data in a frame. This can be used to save power. |
| 43 | */ |
| 44 | #define SND_SOC_DAIFMT_CONT (0 << 4) /* continuous clock */ |
| 45 | #define SND_SOC_DAIFMT_GATED (1 << 4) /* clock is gated */ |
| 46 | |
| 47 | /* |
| 48 | * DAI Left/Right Clocks. |
| 49 | * |
| 50 | * Specifies whether the DAI can support different samples for similtanious |
| 51 | * playback and capture. This usually requires a seperate physical frame |
| 52 | * clock for playback and capture. |
| 53 | */ |
| 54 | #define SND_SOC_DAIFMT_SYNC (0 << 5) /* Tx FRM = Rx FRM */ |
| 55 | #define SND_SOC_DAIFMT_ASYNC (1 << 5) /* Tx FRM ~ Rx FRM */ |
| 56 | |
| 57 | /* |
| 58 | * TDM |
| 59 | * |
| 60 | * Time Division Multiplexing. Allows PCM data to be multplexed with other |
| 61 | * data on the DAI. |
| 62 | */ |
| 63 | #define SND_SOC_DAIFMT_TDM (1 << 6) |
| 64 | |
| 65 | /* |
| 66 | * DAI hardware signal inversions. |
| 67 | * |
| 68 | * Specifies whether the DAI can also support inverted clocks for the specified |
| 69 | * format. |
| 70 | */ |
| 71 | #define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bit clock + frame */ |
| 72 | #define SND_SOC_DAIFMT_NB_IF (1 << 8) /* normal bclk + inv frm */ |
| 73 | #define SND_SOC_DAIFMT_IB_NF (2 << 8) /* invert bclk + nor frm */ |
| 74 | #define SND_SOC_DAIFMT_IB_IF (3 << 8) /* invert bclk + frm */ |
| 75 | |
| 76 | /* |
| 77 | * DAI hardware clock masters. |
| 78 | * |
| 79 | * This is wrt the codec, the inverse is true for the interface |
| 80 | * i.e. if the codec is clk and frm master then the interface is |
| 81 | * clk and frame slave. |
| 82 | */ |
| 83 | #define SND_SOC_DAIFMT_CBM_CFM (0 << 12) /* codec clk & frm master */ |
| 84 | #define SND_SOC_DAIFMT_CBS_CFM (1 << 12) /* codec clk slave & frm master */ |
| 85 | #define SND_SOC_DAIFMT_CBM_CFS (2 << 12) /* codec clk master & frame slave */ |
| 86 | #define SND_SOC_DAIFMT_CBS_CFS (3 << 12) /* codec clk & frm slave */ |
| 87 | |
| 88 | #define SND_SOC_DAIFMT_FORMAT_MASK 0x000f |
| 89 | #define SND_SOC_DAIFMT_CLOCK_MASK 0x00f0 |
| 90 | #define SND_SOC_DAIFMT_INV_MASK 0x0f00 |
| 91 | #define SND_SOC_DAIFMT_MASTER_MASK 0xf000 |
| 92 | |
| 93 | /* |
| 94 | * Master Clock Directions |
| 95 | */ |
| 96 | #define SND_SOC_CLOCK_IN 0 |
| 97 | #define SND_SOC_CLOCK_OUT 1 |
| 98 | |
| 99 | struct snd_soc_dai_ops; |
| 100 | struct snd_soc_dai; |
| 101 | struct snd_ac97_bus_ops; |
| 102 | |
Mark Brown | 9115171 | 2008-11-30 23:31:24 +0000 | [diff] [blame] | 103 | /* Digital Audio Interface registration */ |
| 104 | int snd_soc_register_dai(struct snd_soc_dai *dai); |
| 105 | void snd_soc_unregister_dai(struct snd_soc_dai *dai); |
| 106 | int snd_soc_register_dais(struct snd_soc_dai *dai, size_t count); |
| 107 | void snd_soc_unregister_dais(struct snd_soc_dai *dai, size_t count); |
| 108 | |
Mark Brown | a47cbe7 | 2008-07-23 14:03:07 +0100 | [diff] [blame] | 109 | /* Digital Audio Interface clocking API.*/ |
| 110 | int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, |
| 111 | unsigned int freq, int dir); |
| 112 | |
| 113 | int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai, |
| 114 | int div_id, int div); |
| 115 | |
| 116 | int snd_soc_dai_set_pll(struct snd_soc_dai *dai, |
| 117 | int pll_id, unsigned int freq_in, unsigned int freq_out); |
| 118 | |
| 119 | /* Digital Audio interface formatting */ |
| 120 | int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); |
| 121 | |
| 122 | int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, |
| 123 | unsigned int mask, int slots); |
| 124 | |
| 125 | int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); |
| 126 | |
| 127 | /* Digital Audio Interface mute */ |
| 128 | int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute); |
| 129 | |
| 130 | /* |
| 131 | * Digital Audio Interface. |
| 132 | * |
| 133 | * Describes the Digital Audio Interface in terms of it's ALSA, DAI and AC97 |
| 134 | * operations an capabilities. Codec and platfom drivers will register a this |
| 135 | * structure for every DAI they have. |
| 136 | * |
| 137 | * This structure covers the clocking, formating and ALSA operations for each |
| 138 | * interface a |
| 139 | */ |
| 140 | struct snd_soc_dai_ops { |
| 141 | /* |
| 142 | * DAI clocking configuration, all optional. |
| 143 | * Called by soc_card drivers, normally in their hw_params. |
| 144 | */ |
| 145 | int (*set_sysclk)(struct snd_soc_dai *dai, |
| 146 | int clk_id, unsigned int freq, int dir); |
| 147 | int (*set_pll)(struct snd_soc_dai *dai, |
| 148 | int pll_id, unsigned int freq_in, unsigned int freq_out); |
| 149 | int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div); |
| 150 | |
| 151 | /* |
| 152 | * DAI format configuration |
| 153 | * Called by soc_card drivers, normally in their hw_params. |
| 154 | */ |
| 155 | int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt); |
| 156 | int (*set_tdm_slot)(struct snd_soc_dai *dai, |
| 157 | unsigned int mask, int slots); |
| 158 | int (*set_tristate)(struct snd_soc_dai *dai, int tristate); |
| 159 | |
| 160 | /* |
| 161 | * DAI digital mute - optional. |
| 162 | * Called by soc-core to minimise any pops. |
| 163 | */ |
| 164 | int (*digital_mute)(struct snd_soc_dai *dai, int mute); |
Mark Brown | dee89c4 | 2008-11-18 22:11:38 +0000 | [diff] [blame] | 165 | |
| 166 | /* |
| 167 | * ALSA PCM audio operations - all optional. |
| 168 | * Called by soc-core during audio PCM operations. |
| 169 | */ |
| 170 | int (*startup)(struct snd_pcm_substream *, |
| 171 | struct snd_soc_dai *); |
| 172 | void (*shutdown)(struct snd_pcm_substream *, |
| 173 | struct snd_soc_dai *); |
| 174 | int (*hw_params)(struct snd_pcm_substream *, |
| 175 | struct snd_pcm_hw_params *, struct snd_soc_dai *); |
| 176 | int (*hw_free)(struct snd_pcm_substream *, |
| 177 | struct snd_soc_dai *); |
| 178 | int (*prepare)(struct snd_pcm_substream *, |
| 179 | struct snd_soc_dai *); |
| 180 | int (*trigger)(struct snd_pcm_substream *, int, |
| 181 | struct snd_soc_dai *); |
Mark Brown | a47cbe7 | 2008-07-23 14:03:07 +0100 | [diff] [blame] | 182 | }; |
| 183 | |
| 184 | /* |
| 185 | * Digital Audio Interface runtime data. |
| 186 | * |
| 187 | * Holds runtime data for a DAI. |
| 188 | */ |
| 189 | struct snd_soc_dai { |
| 190 | /* DAI description */ |
| 191 | char *name; |
| 192 | unsigned int id; |
Mark Brown | 3ba9e10 | 2008-11-24 18:01:05 +0000 | [diff] [blame] | 193 | int ac97_control; |
Mark Brown | a47cbe7 | 2008-07-23 14:03:07 +0100 | [diff] [blame] | 194 | |
Mark Brown | 9115171 | 2008-11-30 23:31:24 +0000 | [diff] [blame] | 195 | struct device *dev; |
| 196 | |
Mark Brown | a47cbe7 | 2008-07-23 14:03:07 +0100 | [diff] [blame] | 197 | /* DAI callbacks */ |
| 198 | int (*probe)(struct platform_device *pdev, |
| 199 | struct snd_soc_dai *dai); |
| 200 | void (*remove)(struct platform_device *pdev, |
| 201 | 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 */ |
Eric Miao | 6335d05 | 2009-03-03 09:41:00 +0800 | [diff] [blame] | 206 | 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; |
| 211 | |
| 212 | /* DAI runtime info */ |
| 213 | struct snd_pcm_runtime *runtime; |
| 214 | struct snd_soc_codec *codec; |
| 215 | unsigned int active; |
| 216 | unsigned char pop_wait:1; |
| 217 | void *dma_data; |
| 218 | |
| 219 | /* DAI private data */ |
| 220 | void *private_data; |
| 221 | |
| 222 | /* parent codec/platform */ |
| 223 | union { |
| 224 | struct snd_soc_codec *codec; |
| 225 | struct snd_soc_platform *platform; |
| 226 | }; |
| 227 | |
| 228 | struct list_head list; |
| 229 | }; |
| 230 | |
| 231 | #endif |