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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 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 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 */ |
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 */ |
57 | |
58 | /* |
59 | * DAI hardware clock masters. |
60 | * |
61 | * This is wrt the codec, the inverse is true for the interface |
62 | * i.e. if the codec is clk and FRM master then the interface is |
63 | * clk and frame slave. |
64 | */ |
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 */ |
67 | #define SND_SOC_DAIFMT_CBM_CFS (2 << 12) /* codec clk master & frame slave */ |
68 | #define SND_SOC_DAIFMT_CBS_CFS (3 << 12) /* codec clk & FRM slave */ |
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 | |
81 | #define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S8 |\ |
82 | SNDRV_PCM_FMTBIT_S16_LE |\ |
83 | SNDRV_PCM_FMTBIT_S16_BE |\ |
84 | SNDRV_PCM_FMTBIT_S20_3LE |\ |
85 | SNDRV_PCM_FMTBIT_S20_3BE |\ |
86 | SNDRV_PCM_FMTBIT_S24_3LE |\ |
87 | SNDRV_PCM_FMTBIT_S24_3BE |\ |
88 | SNDRV_PCM_FMTBIT_S32_LE |\ |
89 | SNDRV_PCM_FMTBIT_S32_BE) |
90 | |
91 | struct snd_soc_dai_ops; |
92 | struct snd_soc_dai; |
93 | struct snd_ac97_bus_ops; |
94 | |
95 | /* Digital Audio Interface registration */ |
96 | int snd_soc_register_dai(struct snd_soc_dai *dai); |
97 | void snd_soc_unregister_dai(struct snd_soc_dai *dai); |
98 | int snd_soc_register_dais(struct snd_soc_dai *dai, size_t count); |
99 | void snd_soc_unregister_dais(struct snd_soc_dai *dai, size_t count); |
100 | |
101 | /* Digital Audio Interface clocking API.*/ |
102 | int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, |
103 | unsigned int freq, int dir); |
104 | |
105 | int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai, |
106 | int div_id, int div); |
107 | |
108 | int snd_soc_dai_set_pll(struct snd_soc_dai *dai, |
109 | int pll_id, unsigned int freq_in, unsigned int freq_out); |
110 | |
111 | /* Digital Audio interface formatting */ |
112 | int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); |
113 | |
114 | int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, |
115 | unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width); |
116 | |
117 | int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); |
118 | |
119 | /* Digital Audio Interface mute */ |
120 | int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute); |
121 | |
122 | /* |
123 | * Digital Audio Interface. |
124 | * |
125 | * Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97 |
126 | * operations and capabilities. Codec and platform drivers will register this |
127 | * structure for every DAI they have. |
128 | * |
129 | * This structure covers the clocking, formating and ALSA operations for each |
130 | * interface. |
131 | */ |
132 | struct snd_soc_dai_ops { |
133 | /* |
134 | * DAI clocking configuration, all optional. |
135 | * Called by soc_card drivers, normally in their hw_params. |
136 | */ |
137 | int (*set_sysclk)(struct snd_soc_dai *dai, |
138 | int clk_id, unsigned int freq, int dir); |
139 | int (*set_pll)(struct snd_soc_dai *dai, |
140 | int pll_id, unsigned int freq_in, unsigned int freq_out); |
141 | int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div); |
142 | |
143 | /* |
144 | * DAI format configuration |
145 | * Called by soc_card drivers, normally in their hw_params. |
146 | */ |
147 | int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt); |
148 | int (*set_tdm_slot)(struct snd_soc_dai *dai, |
149 | unsigned int tx_mask, unsigned int rx_mask, |
150 | int slots, int slot_width); |
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); |
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 *); |
175 | }; |
176 | |
177 | /* |
178 | * Digital Audio Interface runtime data. |
179 | * |
180 | * Holds runtime data for a DAI. |
181 | */ |
182 | struct snd_soc_dai { |
183 | /* DAI description */ |
184 | char *name; |
185 | unsigned int id; |
186 | int ac97_control; |
187 | |
188 | struct device *dev; |
189 | void *ac97_pdata; /* platform_data for the ac97 codec */ |
190 | |
191 | /* DAI callbacks */ |
192 | int (*probe)(struct platform_device *pdev, |
193 | struct snd_soc_dai *dai); |
194 | void (*remove)(struct platform_device *pdev, |
195 | struct snd_soc_dai *dai); |
196 | int (*suspend)(struct snd_soc_dai *dai); |
197 | int (*resume)(struct snd_soc_dai *dai); |
198 | |
199 | /* ops */ |
200 | struct snd_soc_dai_ops *ops; |
201 | |
202 | /* DAI capabilities */ |
203 | struct snd_soc_pcm_stream capture; |
204 | struct snd_soc_pcm_stream playback; |
205 | unsigned int symmetric_rates:1; |
206 | |
207 | /* DAI runtime info */ |
208 | struct snd_pcm_runtime *runtime; |
209 | struct snd_soc_codec *codec; |
210 | unsigned int active; |
211 | unsigned char pop_wait:1; |
212 | void *dma_data; |
213 | |
214 | /* DAI private data */ |
215 | void *private_data; |
216 | |
217 | /* parent platform */ |
218 | struct snd_soc_platform *platform; |
219 | |
220 | struct list_head list; |
221 | }; |
222 | |
223 | #endif |
224 |
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