af_unix: limit recursion level
[linux-2.6/linux-acpi-2.6/ibm-acpi-2.6.git] / include / sound / soc-dai.h
blob377693a143855fe24a827aaaa035f1c55cccfe61
1 /*
2 * linux/sound/soc-dai.h -- ALSA SoC Layer
4 * Copyright: 2005-2008 Wolfson Microelectronics. PLC.
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.
10 * Digital Audio Interface (DAI) API.
13 #ifndef __LINUX_SND_SOC_DAI_H
14 #define __LINUX_SND_SOC_DAI_H
17 #include <linux/list.h>
19 #include <sound/soc.h>
21 struct snd_pcm_substream;
24 * DAI hardware audio formats.
26 * Describes the physical PCM data formating and clocking. Add new formats
27 * to the end.
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 */
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
42 * DAI Clock gating.
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.
47 #define SND_SOC_DAIFMT_CONT (0 << 4) /* continuous clock */
48 #define SND_SOC_DAIFMT_GATED (1 << 4) /* clock is gated */
51 * DAI hardware signal inversions.
53 * Specifies whether the DAI can also support inverted clocks for the specified
54 * format.
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 */
62 * DAI hardware clock masters.
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.
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 */
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
79 * Master Clock Directions
81 #define SND_SOC_CLOCK_IN 0
82 #define SND_SOC_CLOCK_OUT 1
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)
94 struct snd_soc_dai_ops;
95 struct snd_soc_dai;
96 struct snd_ac97_bus_ops;
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);
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);
108 int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai,
109 int div_id, int div);
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);
114 /* Digital Audio interface formatting */
115 int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt);
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);
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);
124 int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate);
126 /* Digital Audio Interface mute */
127 int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute);
130 * Digital Audio Interface.
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.
136 * This structure covers the clocking, formating and ALSA operations for each
137 * interface.
139 struct snd_soc_dai_ops {
141 * DAI clocking configuration, all optional.
142 * Called by soc_card drivers, normally in their hw_params.
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);
151 * DAI format configuration
152 * Called by soc_card drivers, normally in their hw_params.
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);
164 * DAI digital mute - optional.
165 * Called by soc-core to minimise any pops.
167 int (*digital_mute)(struct snd_soc_dai *dai, int mute);
170 * ALSA PCM audio operations - all optional.
171 * Called by soc-core during audio PCM operations.
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 *);
186 * For hardware based FIFO caused delay reporting.
187 * Optional.
189 snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *,
190 struct snd_soc_dai *);
194 * Digital Audio Interface runtime data.
196 * Holds runtime data for a DAI.
198 struct snd_soc_dai {
199 /* DAI description */
200 char *name;
201 unsigned int id;
202 int ac97_control;
204 struct device *dev;
205 void *ac97_pdata; /* platform_data for the ac97 codec */
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);
215 /* ops */
216 struct snd_soc_dai_ops *ops;
218 /* DAI capabilities */
219 struct snd_soc_pcm_stream capture;
220 struct snd_soc_pcm_stream playback;
221 unsigned int symmetric_rates:1;
223 /* DAI runtime info */
224 struct snd_soc_codec *codec;
225 unsigned int active;
226 unsigned char pop_wait:1;
228 /* DAI private data */
229 void *private_data;
231 /* parent platform */
232 struct snd_soc_platform *platform;
234 struct list_head list;
237 static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai *dai,
238 const struct snd_pcm_substream *ss)
240 return (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
241 dai->playback.dma_data : dai->capture.dma_data;
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)
248 if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK)
249 dai->playback.dma_data = data;
250 else
251 dai->capture.dma_data = data;
254 #endif