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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 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 | ||
103 | /* Digital Audio Interface clocking API.*/ | |
104 | int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, | |
105 | unsigned int freq, int dir); | |
106 | ||
107 | int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai, | |
108 | int div_id, int div); | |
109 | ||
110 | int snd_soc_dai_set_pll(struct snd_soc_dai *dai, | |
111 | int pll_id, unsigned int freq_in, unsigned int freq_out); | |
112 | ||
113 | /* Digital Audio interface formatting */ | |
114 | int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); | |
115 | ||
116 | int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, | |
117 | unsigned int mask, int slots); | |
118 | ||
119 | int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); | |
120 | ||
121 | /* Digital Audio Interface mute */ | |
122 | int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute); | |
123 | ||
124 | /* | |
125 | * Digital Audio Interface. | |
126 | * | |
127 | * Describes the Digital Audio Interface in terms of it's ALSA, DAI and AC97 | |
128 | * operations an capabilities. Codec and platfom drivers will register a this | |
129 | * structure for every DAI they have. | |
130 | * | |
131 | * This structure covers the clocking, formating and ALSA operations for each | |
132 | * interface a | |
133 | */ | |
134 | struct snd_soc_dai_ops { | |
135 | /* | |
136 | * DAI clocking configuration, all optional. | |
137 | * Called by soc_card drivers, normally in their hw_params. | |
138 | */ | |
139 | int (*set_sysclk)(struct snd_soc_dai *dai, | |
140 | int clk_id, unsigned int freq, int dir); | |
141 | int (*set_pll)(struct snd_soc_dai *dai, | |
142 | int pll_id, unsigned int freq_in, unsigned int freq_out); | |
143 | int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div); | |
144 | ||
145 | /* | |
146 | * DAI format configuration | |
147 | * Called by soc_card drivers, normally in their hw_params. | |
148 | */ | |
149 | int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt); | |
150 | int (*set_tdm_slot)(struct snd_soc_dai *dai, | |
151 | unsigned int mask, int slots); | |
152 | int (*set_tristate)(struct snd_soc_dai *dai, int tristate); | |
153 | ||
154 | /* | |
155 | * DAI digital mute - optional. | |
156 | * Called by soc-core to minimise any pops. | |
157 | */ | |
158 | int (*digital_mute)(struct snd_soc_dai *dai, int mute); | |
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159 | |
160 | /* | |
161 | * ALSA PCM audio operations - all optional. | |
162 | * Called by soc-core during audio PCM operations. | |
163 | */ | |
164 | int (*startup)(struct snd_pcm_substream *, | |
165 | struct snd_soc_dai *); | |
166 | void (*shutdown)(struct snd_pcm_substream *, | |
167 | struct snd_soc_dai *); | |
168 | int (*hw_params)(struct snd_pcm_substream *, | |
169 | struct snd_pcm_hw_params *, struct snd_soc_dai *); | |
170 | int (*hw_free)(struct snd_pcm_substream *, | |
171 | struct snd_soc_dai *); | |
172 | int (*prepare)(struct snd_pcm_substream *, | |
173 | struct snd_soc_dai *); | |
174 | int (*trigger)(struct snd_pcm_substream *, int, | |
175 | struct snd_soc_dai *); | |
a47cbe72 MB |
176 | }; |
177 | ||
178 | /* | |
179 | * Digital Audio Interface runtime data. | |
180 | * | |
181 | * Holds runtime data for a DAI. | |
182 | */ | |
183 | struct snd_soc_dai { | |
184 | /* DAI description */ | |
185 | char *name; | |
186 | unsigned int id; | |
187 | unsigned char type; | |
188 | ||
189 | /* DAI callbacks */ | |
190 | int (*probe)(struct platform_device *pdev, | |
191 | struct snd_soc_dai *dai); | |
192 | void (*remove)(struct platform_device *pdev, | |
193 | struct snd_soc_dai *dai); | |
194 | int (*suspend)(struct platform_device *pdev, | |
195 | struct snd_soc_dai *dai); | |
196 | int (*resume)(struct platform_device *pdev, | |
197 | struct snd_soc_dai *dai); | |
198 | ||
199 | /* ops */ | |
dee89c4d | 200 | struct snd_soc_dai_ops ops; |
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201 | |
202 | /* DAI capabilities */ | |
203 | struct snd_soc_pcm_stream capture; | |
204 | struct snd_soc_pcm_stream playback; | |
205 | ||
206 | /* DAI runtime info */ | |
207 | struct snd_pcm_runtime *runtime; | |
208 | struct snd_soc_codec *codec; | |
209 | unsigned int active; | |
210 | unsigned char pop_wait:1; | |
211 | void *dma_data; | |
212 | ||
213 | /* DAI private data */ | |
214 | void *private_data; | |
215 | ||
216 | /* parent codec/platform */ | |
217 | union { | |
218 | struct snd_soc_codec *codec; | |
219 | struct snd_soc_platform *platform; | |
220 | }; | |
221 | ||
222 | struct list_head list; | |
223 | }; | |
224 | ||
225 | #endif |