ALSA: hda - Fix CORB reset to follow specification
[linux/fpc-iii.git] / include / sound / soc-dai.h
blob71f27c403194f19ed26a189bfed3252f7177747b
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 struct snd_pcm_substream;
20 struct snd_soc_dapm_widget;
21 struct snd_compr_stream;
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 1 /* I2S mode */
30 #define SND_SOC_DAIFMT_RIGHT_J 2 /* Right Justified mode */
31 #define SND_SOC_DAIFMT_LEFT_J 3 /* Left Justified mode */
32 #define SND_SOC_DAIFMT_DSP_A 4 /* L data MSB after FRM LRC */
33 #define SND_SOC_DAIFMT_DSP_B 5 /* L data MSB during FRM LRC */
34 #define SND_SOC_DAIFMT_AC97 6 /* AC97 */
35 #define SND_SOC_DAIFMT_PDM 7 /* 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 (1 << 4) /* continuous clock */
48 #define SND_SOC_DAIFMT_GATED (0 << 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 (2 << 8) /* normal BCLK + inv FRM */
58 #define SND_SOC_DAIFMT_IB_NF (3 << 8) /* invert BCLK + nor FRM */
59 #define SND_SOC_DAIFMT_IB_IF (4 << 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 (1 << 12) /* codec clk & FRM master */
69 #define SND_SOC_DAIFMT_CBS_CFM (2 << 12) /* codec clk slave & FRM master */
70 #define SND_SOC_DAIFMT_CBM_CFS (3 << 12) /* codec clk master & frame slave */
71 #define SND_SOC_DAIFMT_CBS_CFS (4 << 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_driver;
95 struct snd_soc_dai;
96 struct snd_ac97_bus_ops;
98 /* Digital Audio Interface clocking API.*/
99 int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
100 unsigned int freq, int dir);
102 int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai,
103 int div_id, int div);
105 int snd_soc_dai_set_pll(struct snd_soc_dai *dai,
106 int pll_id, int source, unsigned int freq_in, unsigned int freq_out);
108 int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio);
110 /* Digital Audio interface formatting */
111 int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt);
113 int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
114 unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width);
116 int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai,
117 unsigned int tx_num, unsigned int *tx_slot,
118 unsigned int rx_num, unsigned int *rx_slot);
120 int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate);
122 /* Digital Audio Interface mute */
123 int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute,
124 int direction);
126 int snd_soc_dai_is_dummy(struct snd_soc_dai *dai);
128 struct snd_soc_dai_ops {
130 * DAI clocking configuration, all optional.
131 * Called by soc_card drivers, normally in their hw_params.
133 int (*set_sysclk)(struct snd_soc_dai *dai,
134 int clk_id, unsigned int freq, int dir);
135 int (*set_pll)(struct snd_soc_dai *dai, int pll_id, int source,
136 unsigned int freq_in, unsigned int freq_out);
137 int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div);
138 int (*set_bclk_ratio)(struct snd_soc_dai *dai, unsigned int ratio);
141 * DAI format configuration
142 * Called by soc_card drivers, normally in their hw_params.
144 int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt);
145 int (*set_tdm_slot)(struct snd_soc_dai *dai,
146 unsigned int tx_mask, unsigned int rx_mask,
147 int slots, int slot_width);
148 int (*set_channel_map)(struct snd_soc_dai *dai,
149 unsigned int tx_num, unsigned int *tx_slot,
150 unsigned int rx_num, unsigned int *rx_slot);
151 int (*set_tristate)(struct snd_soc_dai *dai, int tristate);
154 * DAI digital mute - optional.
155 * Called by soc-core to minimise any pops.
157 int (*digital_mute)(struct snd_soc_dai *dai, int mute);
158 int (*mute_stream)(struct snd_soc_dai *dai, int mute, int stream);
161 * ALSA PCM audio operations - all optional.
162 * Called by soc-core during audio PCM operations.
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 *);
175 * NOTE: Commands passed to the trigger function are not necessarily
176 * compatible with the current state of the dai. For example this
177 * sequence of commands is possible: START STOP STOP.
178 * So do not unconditionally use refcounting functions in the trigger
179 * function, e.g. clk_enable/disable.
181 int (*trigger)(struct snd_pcm_substream *, int,
182 struct snd_soc_dai *);
183 int (*bespoke_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 Driver.
196 * Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97
197 * operations and capabilities. Codec and platform drivers will register this
198 * structure for every DAI they have.
200 * This structure covers the clocking, formating and ALSA operations for each
201 * interface.
203 struct snd_soc_dai_driver {
204 /* DAI description */
205 const char *name;
206 unsigned int id;
207 int ac97_control;
208 unsigned int base;
210 /* DAI driver callbacks */
211 int (*probe)(struct snd_soc_dai *dai);
212 int (*remove)(struct snd_soc_dai *dai);
213 int (*suspend)(struct snd_soc_dai *dai);
214 int (*resume)(struct snd_soc_dai *dai);
215 /* compress dai */
216 bool compress_dai;
218 /* ops */
219 const struct snd_soc_dai_ops *ops;
221 /* DAI capabilities */
222 struct snd_soc_pcm_stream capture;
223 struct snd_soc_pcm_stream playback;
224 unsigned int symmetric_rates:1;
225 unsigned int symmetric_channels:1;
226 unsigned int symmetric_samplebits:1;
228 /* probe ordering - for components with runtime dependencies */
229 int probe_order;
230 int remove_order;
234 * Digital Audio Interface runtime data.
236 * Holds runtime data for a DAI.
238 struct snd_soc_dai {
239 const char *name;
240 int id;
241 struct device *dev;
242 void *ac97_pdata; /* platform_data for the ac97 codec */
244 /* driver ops */
245 struct snd_soc_dai_driver *driver;
247 /* DAI runtime info */
248 unsigned int capture_active:1; /* stream is in use */
249 unsigned int playback_active:1; /* stream is in use */
250 unsigned int symmetric_rates:1;
251 unsigned int symmetric_channels:1;
252 unsigned int symmetric_samplebits:1;
253 struct snd_pcm_runtime *runtime;
254 unsigned int active;
255 unsigned char probed:1;
257 struct snd_soc_dapm_widget *playback_widget;
258 struct snd_soc_dapm_widget *capture_widget;
259 struct snd_soc_dapm_context dapm;
261 /* DAI DMA data */
262 void *playback_dma_data;
263 void *capture_dma_data;
265 /* Symmetry data - only valid if symmetry is being enforced */
266 unsigned int rate;
267 unsigned int channels;
268 unsigned int sample_bits;
270 /* parent platform/codec */
271 struct snd_soc_platform *platform;
272 struct snd_soc_codec *codec;
274 struct snd_soc_card *card;
276 struct list_head list;
277 struct list_head card_list;
280 static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai *dai,
281 const struct snd_pcm_substream *ss)
283 return (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
284 dai->playback_dma_data : dai->capture_dma_data;
287 static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai,
288 const struct snd_pcm_substream *ss,
289 void *data)
291 if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK)
292 dai->playback_dma_data = data;
293 else
294 dai->capture_dma_data = data;
297 static inline void snd_soc_dai_init_dma_data(struct snd_soc_dai *dai,
298 void *playback, void *capture)
300 dai->playback_dma_data = playback;
301 dai->capture_dma_data = capture;
304 static inline void snd_soc_dai_set_drvdata(struct snd_soc_dai *dai,
305 void *data)
307 dev_set_drvdata(dai->dev, data);
310 static inline void *snd_soc_dai_get_drvdata(struct snd_soc_dai *dai)
312 return dev_get_drvdata(dai->dev);
315 #endif