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 */ |
| 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 | * |
Peter Meerwald | 47db8e89 | 2009-07-13 23:05:11 +0100 | [diff] [blame] | 41 | * 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] | 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 | /* |
Mark Brown | a47cbe7 | 2008-07-23 14:03:07 +0100 | [diff] [blame] | 48 | * DAI hardware signal inversions. |
| 49 | * |
| 50 | * Specifies whether the DAI can also support inverted clocks for the specified |
| 51 | * format. |
| 52 | */ |
| 53 | #define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bit clock + frame */ |
Peter Meerwald | 47db8e89 | 2009-07-13 23:05:11 +0100 | [diff] [blame] | 54 | #define SND_SOC_DAIFMT_NB_IF (1 << 8) /* normal BCLK + inv FRM */ |
| 55 | #define SND_SOC_DAIFMT_IB_NF (2 << 8) /* invert BCLK + nor FRM */ |
| 56 | #define SND_SOC_DAIFMT_IB_IF (3 << 8) /* invert BCLK + FRM */ |
Mark Brown | a47cbe7 | 2008-07-23 14:03:07 +0100 | [diff] [blame] | 57 | |
| 58 | /* |
| 59 | * DAI hardware clock masters. |
| 60 | * |
| 61 | * This is wrt the codec, the inverse is true for the interface |
Peter Meerwald | 47db8e89 | 2009-07-13 23:05:11 +0100 | [diff] [blame] | 62 | * 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] | 63 | * clk and frame slave. |
| 64 | */ |
Peter Meerwald | 47db8e89 | 2009-07-13 23:05:11 +0100 | [diff] [blame] | 65 | #define SND_SOC_DAIFMT_CBM_CFM (0 << 12) /* codec clk & FRM master */ |
| 66 | #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] | 67 | #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] | 68 | #define SND_SOC_DAIFMT_CBS_CFS (3 << 12) /* codec clk & FRM slave */ |
Mark Brown | a47cbe7 | 2008-07-23 14:03:07 +0100 | [diff] [blame] | 69 | |
| 70 | #define SND_SOC_DAIFMT_FORMAT_MASK 0x000f |
| 71 | #define SND_SOC_DAIFMT_CLOCK_MASK 0x00f0 |
| 72 | #define SND_SOC_DAIFMT_INV_MASK 0x0f00 |
| 73 | #define SND_SOC_DAIFMT_MASTER_MASK 0xf000 |
| 74 | |
| 75 | /* |
| 76 | * Master Clock Directions |
| 77 | */ |
| 78 | #define SND_SOC_CLOCK_IN 0 |
| 79 | #define SND_SOC_CLOCK_OUT 1 |
| 80 | |
Mark Brown | 33f503c | 2009-05-02 12:24:55 +0100 | [diff] [blame] | 81 | #define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S16_LE |\ |
Jon Smirl | d34c430 | 2009-05-13 21:59:14 -0400 | [diff] [blame] | 82 | SNDRV_PCM_FMTBIT_S32_LE |\ |
| 83 | SNDRV_PCM_FMTBIT_S32_BE) |
Mark Brown | 33f503c | 2009-05-02 12:24:55 +0100 | [diff] [blame] | 84 | |
Mark Brown | a47cbe7 | 2008-07-23 14:03:07 +0100 | [diff] [blame] | 85 | struct snd_soc_dai_ops; |
| 86 | struct snd_soc_dai; |
| 87 | struct snd_ac97_bus_ops; |
| 88 | |
Mark Brown | 9115171 | 2008-11-30 23:31:24 +0000 | [diff] [blame] | 89 | /* Digital Audio Interface registration */ |
| 90 | int snd_soc_register_dai(struct snd_soc_dai *dai); |
| 91 | void snd_soc_unregister_dai(struct snd_soc_dai *dai); |
| 92 | int snd_soc_register_dais(struct snd_soc_dai *dai, size_t count); |
| 93 | void snd_soc_unregister_dais(struct snd_soc_dai *dai, size_t count); |
| 94 | |
Mark Brown | a47cbe7 | 2008-07-23 14:03:07 +0100 | [diff] [blame] | 95 | /* Digital Audio Interface clocking API.*/ |
| 96 | int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, |
| 97 | unsigned int freq, int dir); |
| 98 | |
| 99 | int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai, |
| 100 | int div_id, int div); |
| 101 | |
| 102 | int snd_soc_dai_set_pll(struct snd_soc_dai *dai, |
| 103 | int pll_id, unsigned int freq_in, unsigned int freq_out); |
| 104 | |
| 105 | /* Digital Audio interface formatting */ |
| 106 | int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); |
| 107 | |
| 108 | int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, |
Daniel Ribeiro | a5479e3 | 2009-06-15 21:44:31 -0300 | [diff] [blame^] | 109 | 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] | 110 | |
| 111 | int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); |
| 112 | |
| 113 | /* Digital Audio Interface mute */ |
| 114 | int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute); |
| 115 | |
| 116 | /* |
| 117 | * Digital Audio Interface. |
| 118 | * |
Peter Meerwald | 47db8e89 | 2009-07-13 23:05:11 +0100 | [diff] [blame] | 119 | * Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97 |
| 120 | * operations and capabilities. Codec and platform drivers will register this |
Mark Brown | a47cbe7 | 2008-07-23 14:03:07 +0100 | [diff] [blame] | 121 | * structure for every DAI they have. |
| 122 | * |
| 123 | * This structure covers the clocking, formating and ALSA operations for each |
Peter Meerwald | 47db8e89 | 2009-07-13 23:05:11 +0100 | [diff] [blame] | 124 | * interface. |
Mark Brown | a47cbe7 | 2008-07-23 14:03:07 +0100 | [diff] [blame] | 125 | */ |
| 126 | struct snd_soc_dai_ops { |
| 127 | /* |
| 128 | * DAI clocking configuration, all optional. |
| 129 | * Called by soc_card drivers, normally in their hw_params. |
| 130 | */ |
| 131 | int (*set_sysclk)(struct snd_soc_dai *dai, |
| 132 | int clk_id, unsigned int freq, int dir); |
| 133 | int (*set_pll)(struct snd_soc_dai *dai, |
| 134 | int pll_id, unsigned int freq_in, unsigned int freq_out); |
| 135 | int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div); |
| 136 | |
| 137 | /* |
| 138 | * DAI format configuration |
| 139 | * Called by soc_card drivers, normally in their hw_params. |
| 140 | */ |
| 141 | int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt); |
| 142 | int (*set_tdm_slot)(struct snd_soc_dai *dai, |
Daniel Ribeiro | a5479e3 | 2009-06-15 21:44:31 -0300 | [diff] [blame^] | 143 | unsigned int tx_mask, unsigned int rx_mask, |
| 144 | int slots, int slot_width); |
Mark Brown | a47cbe7 | 2008-07-23 14:03:07 +0100 | [diff] [blame] | 145 | int (*set_tristate)(struct snd_soc_dai *dai, int tristate); |
| 146 | |
| 147 | /* |
| 148 | * DAI digital mute - optional. |
| 149 | * Called by soc-core to minimise any pops. |
| 150 | */ |
| 151 | int (*digital_mute)(struct snd_soc_dai *dai, int mute); |
Mark Brown | dee89c4 | 2008-11-18 22:11:38 +0000 | [diff] [blame] | 152 | |
| 153 | /* |
| 154 | * ALSA PCM audio operations - all optional. |
| 155 | * Called by soc-core during audio PCM operations. |
| 156 | */ |
| 157 | int (*startup)(struct snd_pcm_substream *, |
| 158 | struct snd_soc_dai *); |
| 159 | void (*shutdown)(struct snd_pcm_substream *, |
| 160 | struct snd_soc_dai *); |
| 161 | int (*hw_params)(struct snd_pcm_substream *, |
| 162 | struct snd_pcm_hw_params *, struct snd_soc_dai *); |
| 163 | int (*hw_free)(struct snd_pcm_substream *, |
| 164 | struct snd_soc_dai *); |
| 165 | int (*prepare)(struct snd_pcm_substream *, |
| 166 | struct snd_soc_dai *); |
| 167 | int (*trigger)(struct snd_pcm_substream *, int, |
| 168 | struct snd_soc_dai *); |
Mark Brown | a47cbe7 | 2008-07-23 14:03:07 +0100 | [diff] [blame] | 169 | }; |
| 170 | |
| 171 | /* |
| 172 | * Digital Audio Interface runtime data. |
| 173 | * |
| 174 | * Holds runtime data for a DAI. |
| 175 | */ |
| 176 | struct snd_soc_dai { |
| 177 | /* DAI description */ |
| 178 | char *name; |
| 179 | unsigned int id; |
Mark Brown | 3ba9e10a | 2008-11-24 18:01:05 +0000 | [diff] [blame] | 180 | int ac97_control; |
Mark Brown | a47cbe7 | 2008-07-23 14:03:07 +0100 | [diff] [blame] | 181 | |
Mark Brown | 9115171 | 2008-11-30 23:31:24 +0000 | [diff] [blame] | 182 | struct device *dev; |
Marek Vasut | 474828a | 2009-07-22 13:01:03 +0200 | [diff] [blame] | 183 | void *ac97_pdata; /* platform_data for the ac97 codec */ |
Mark Brown | 9115171 | 2008-11-30 23:31:24 +0000 | [diff] [blame] | 184 | |
Mark Brown | a47cbe7 | 2008-07-23 14:03:07 +0100 | [diff] [blame] | 185 | /* DAI callbacks */ |
| 186 | int (*probe)(struct platform_device *pdev, |
| 187 | struct snd_soc_dai *dai); |
| 188 | void (*remove)(struct platform_device *pdev, |
| 189 | struct snd_soc_dai *dai); |
Mark Brown | dc7d7b8 | 2008-12-03 18:21:52 +0000 | [diff] [blame] | 190 | int (*suspend)(struct snd_soc_dai *dai); |
| 191 | int (*resume)(struct snd_soc_dai *dai); |
Mark Brown | a47cbe7 | 2008-07-23 14:03:07 +0100 | [diff] [blame] | 192 | |
| 193 | /* ops */ |
Eric Miao | 6335d05 | 2009-03-03 09:41:00 +0800 | [diff] [blame] | 194 | struct snd_soc_dai_ops *ops; |
Mark Brown | a47cbe7 | 2008-07-23 14:03:07 +0100 | [diff] [blame] | 195 | |
| 196 | /* DAI capabilities */ |
| 197 | struct snd_soc_pcm_stream capture; |
| 198 | struct snd_soc_pcm_stream playback; |
Mark Brown | 06f409d | 2009-04-07 18:10:13 +0100 | [diff] [blame] | 199 | unsigned int symmetric_rates:1; |
Mark Brown | a47cbe7 | 2008-07-23 14:03:07 +0100 | [diff] [blame] | 200 | |
| 201 | /* DAI runtime info */ |
| 202 | struct snd_pcm_runtime *runtime; |
| 203 | struct snd_soc_codec *codec; |
| 204 | unsigned int active; |
| 205 | unsigned char pop_wait:1; |
| 206 | void *dma_data; |
| 207 | |
| 208 | /* DAI private data */ |
| 209 | void *private_data; |
| 210 | |
Mark Brown | bbd9930 | 2009-05-05 10:27:38 +0100 | [diff] [blame] | 211 | /* parent platform */ |
| 212 | struct snd_soc_platform *platform; |
Mark Brown | a47cbe7 | 2008-07-23 14:03:07 +0100 | [diff] [blame] | 213 | |
| 214 | struct list_head list; |
| 215 | }; |
| 216 | |
| 217 | #endif |