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Mark Browna47cbe72008-07-23 14:03:07 +01001/*
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
19struct snd_pcm_substream;
Mark Brown888df392012-02-16 19:37:51 -080020struct snd_soc_dapm_widget;
Vinod Koul49681072012-08-16 17:10:40 +053021struct snd_compr_stream;
Mark Browna47cbe72008-07-23 14:03:07 +010022
23/*
24 * DAI hardware audio formats.
25 *
26 * Describes the physical PCM data formating and clocking. Add new formats
27 * to the end.
28 */
Mark Brown75d9ac42011-09-27 16:41:01 +010029#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 */
Mark Browna47cbe72008-07-23 14:03:07 +010036
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
40
41/*
42 * DAI Clock gating.
43 *
Peter Meerwald47db8e82009-07-13 23:05:11 +010044 * DAI bit clocks can be be gated (disabled) when the DAI is not
Mark Browna47cbe72008-07-23 14:03:07 +010045 * sending or receiving PCM data in a frame. This can be used to save power.
46 */
Mark Brown75d9ac42011-09-27 16:41:01 +010047#define SND_SOC_DAIFMT_CONT (1 << 4) /* continuous clock */
Kuninori Morimotoeef28e102013-01-29 21:03:13 -080048#define SND_SOC_DAIFMT_GATED (0 << 4) /* clock is gated */
Mark Browna47cbe72008-07-23 14:03:07 +010049
50/*
Anatol Pomozov1d387a3fd2015-10-08 09:37:51 -070051 * DAI hardware signal polarity.
Mark Browna47cbe72008-07-23 14:03:07 +010052 *
53 * Specifies whether the DAI can also support inverted clocks for the specified
54 * format.
Anatol Pomozov1d387a3fd2015-10-08 09:37:51 -070055 *
56 * BCLK:
57 * - "normal" polarity means signal is available at rising edge of BCLK
58 * - "inverted" polarity means signal is available at falling edge of BCLK
59 *
60 * FSYNC "normal" polarity depends on the frame format:
61 * - I2S: frame consists of left then right channel data. Left channel starts
62 * with falling FSYNC edge, right channel starts with rising FSYNC edge.
63 * - Left/Right Justified: frame consists of left then right channel data.
64 * Left channel starts with rising FSYNC edge, right channel starts with
65 * falling FSYNC edge.
66 * - DSP A/B: Frame starts with rising FSYNC edge.
67 * - AC97: Frame starts with rising FSYNC edge.
68 *
69 * "Negative" FSYNC polarity is the one opposite of "normal" polarity.
Mark Browna47cbe72008-07-23 14:03:07 +010070 */
Kuninori Morimoto5d163332013-01-15 20:18:23 -080071#define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bit clock + frame */
Mark Brown75d9ac42011-09-27 16:41:01 +010072#define SND_SOC_DAIFMT_NB_IF (2 << 8) /* normal BCLK + inv FRM */
73#define SND_SOC_DAIFMT_IB_NF (3 << 8) /* invert BCLK + nor FRM */
74#define SND_SOC_DAIFMT_IB_IF (4 << 8) /* invert BCLK + FRM */
Mark Browna47cbe72008-07-23 14:03:07 +010075
76/*
77 * DAI hardware clock masters.
78 *
79 * This is wrt the codec, the inverse is true for the interface
Peter Meerwald47db8e82009-07-13 23:05:11 +010080 * i.e. if the codec is clk and FRM master then the interface is
Mark Browna47cbe72008-07-23 14:03:07 +010081 * clk and frame slave.
82 */
Mark Brown75d9ac42011-09-27 16:41:01 +010083#define SND_SOC_DAIFMT_CBM_CFM (1 << 12) /* codec clk & FRM master */
84#define SND_SOC_DAIFMT_CBS_CFM (2 << 12) /* codec clk slave & FRM master */
85#define SND_SOC_DAIFMT_CBM_CFS (3 << 12) /* codec clk master & frame slave */
86#define SND_SOC_DAIFMT_CBS_CFS (4 << 12) /* codec clk & FRM slave */
Mark Browna47cbe72008-07-23 14:03:07 +010087
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
Mark Brown8f738d52009-08-09 20:08:31 +010099#define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S8 |\
100 SNDRV_PCM_FMTBIT_S16_LE |\
101 SNDRV_PCM_FMTBIT_S16_BE |\
102 SNDRV_PCM_FMTBIT_S20_3LE |\
103 SNDRV_PCM_FMTBIT_S20_3BE |\
104 SNDRV_PCM_FMTBIT_S24_3LE |\
105 SNDRV_PCM_FMTBIT_S24_3BE |\
Jon Smirld34c4302009-05-13 21:59:14 -0400106 SNDRV_PCM_FMTBIT_S32_LE |\
107 SNDRV_PCM_FMTBIT_S32_BE)
Mark Brown33f503c2009-05-02 12:24:55 +0100108
Liam Girdwoodf0fba2a2010-03-17 20:15:21 +0000109struct snd_soc_dai_driver;
Mark Browna47cbe72008-07-23 14:03:07 +0100110struct snd_soc_dai;
111struct snd_ac97_bus_ops;
112
113/* Digital Audio Interface clocking API.*/
114int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
115 unsigned int freq, int dir);
116
117int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai,
118 int div_id, int div);
119
120int snd_soc_dai_set_pll(struct snd_soc_dai *dai,
Mark Brown85488032009-09-05 18:52:16 +0100121 int pll_id, int source, unsigned int freq_in, unsigned int freq_out);
Mark Browna47cbe72008-07-23 14:03:07 +0100122
Liam Girdwoode54cf762013-09-16 13:01:46 +0100123int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio);
124
Mark Browna47cbe72008-07-23 14:03:07 +0100125/* Digital Audio interface formatting */
126int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt);
127
128int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
Daniel Ribeiroa5479e32009-06-15 21:44:31 -0300129 unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width);
Mark Browna47cbe72008-07-23 14:03:07 +0100130
Barry Song472df3c2009-09-12 01:16:29 +0800131int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai,
132 unsigned int tx_num, unsigned int *tx_slot,
133 unsigned int rx_num, unsigned int *rx_slot);
134
Mark Browna47cbe72008-07-23 14:03:07 +0100135int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate);
136
137/* Digital Audio Interface mute */
Mark Brownda183962013-02-06 15:44:07 +0000138int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute,
139 int direction);
Mark Browna47cbe72008-07-23 14:03:07 +0100140
Banajit Goswami5ae53a82015-04-10 16:40:29 -0700141int snd_soc_dai_get_channel_map(struct snd_soc_dai *dai,
142 unsigned int *tx_num, unsigned int *tx_slot,
143 unsigned int *rx_num, unsigned int *rx_slot);
144
Liam Girdwoodbece9e92014-01-08 10:40:18 +0000145int snd_soc_dai_is_dummy(struct snd_soc_dai *dai);
146
Mark Browna47cbe72008-07-23 14:03:07 +0100147struct snd_soc_dai_ops {
148 /*
149 * DAI clocking configuration, all optional.
150 * Called by soc_card drivers, normally in their hw_params.
151 */
152 int (*set_sysclk)(struct snd_soc_dai *dai,
153 int clk_id, unsigned int freq, int dir);
Mark Brown85488032009-09-05 18:52:16 +0100154 int (*set_pll)(struct snd_soc_dai *dai, int pll_id, int source,
155 unsigned int freq_in, unsigned int freq_out);
Mark Browna47cbe72008-07-23 14:03:07 +0100156 int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div);
Liam Girdwoode54cf762013-09-16 13:01:46 +0100157 int (*set_bclk_ratio)(struct snd_soc_dai *dai, unsigned int ratio);
Mark Browna47cbe72008-07-23 14:03:07 +0100158
159 /*
160 * DAI format configuration
161 * Called by soc_card drivers, normally in their hw_params.
162 */
163 int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt);
Xiubo Lie5c21512014-03-21 14:17:12 +0800164 int (*xlate_tdm_slot_mask)(unsigned int slots,
Xiubo Li89c67852014-02-14 09:34:35 +0800165 unsigned int *tx_mask, unsigned int *rx_mask);
Mark Browna47cbe72008-07-23 14:03:07 +0100166 int (*set_tdm_slot)(struct snd_soc_dai *dai,
Daniel Ribeiroa5479e32009-06-15 21:44:31 -0300167 unsigned int tx_mask, unsigned int rx_mask,
168 int slots, int slot_width);
Barry Song472df3c2009-09-12 01:16:29 +0800169 int (*set_channel_map)(struct snd_soc_dai *dai,
170 unsigned int tx_num, unsigned int *tx_slot,
171 unsigned int rx_num, unsigned int *rx_slot);
Mark Browna47cbe72008-07-23 14:03:07 +0100172 int (*set_tristate)(struct snd_soc_dai *dai, int tristate);
Banajit Goswami5ae53a82015-04-10 16:40:29 -0700173 int (*get_channel_map)(struct snd_soc_dai *dai,
174 unsigned int *tx_num, unsigned int *tx_slot,
175 unsigned int *rx_num, unsigned int *rx_slot);
Mark Browna47cbe72008-07-23 14:03:07 +0100176
177 /*
178 * DAI digital mute - optional.
179 * Called by soc-core to minimise any pops.
180 */
181 int (*digital_mute)(struct snd_soc_dai *dai, int mute);
Mark Brownda183962013-02-06 15:44:07 +0000182 int (*mute_stream)(struct snd_soc_dai *dai, int mute, int stream);
Mark Browndee89c42008-11-18 22:11:38 +0000183
184 /*
185 * ALSA PCM audio operations - all optional.
186 * Called by soc-core during audio PCM operations.
187 */
188 int (*startup)(struct snd_pcm_substream *,
189 struct snd_soc_dai *);
190 void (*shutdown)(struct snd_pcm_substream *,
191 struct snd_soc_dai *);
192 int (*hw_params)(struct snd_pcm_substream *,
193 struct snd_pcm_hw_params *, struct snd_soc_dai *);
194 int (*hw_free)(struct snd_pcm_substream *,
195 struct snd_soc_dai *);
196 int (*prepare)(struct snd_pcm_substream *,
197 struct snd_soc_dai *);
Markus Pargmann9f1614a2013-10-11 12:11:02 +0200198 /*
199 * NOTE: Commands passed to the trigger function are not necessarily
200 * compatible with the current state of the dai. For example this
201 * sequence of commands is possible: START STOP STOP.
202 * So do not unconditionally use refcounting functions in the trigger
203 * function, e.g. clk_enable/disable.
204 */
Mark Browndee89c42008-11-18 22:11:38 +0000205 int (*trigger)(struct snd_pcm_substream *, int,
206 struct snd_soc_dai *);
Liam Girdwood07bf84a2012-04-25 12:12:52 +0100207 int (*bespoke_trigger)(struct snd_pcm_substream *, int,
208 struct snd_soc_dai *);
Peter Ujfalusi258020d2010-03-03 15:08:07 +0200209 /*
210 * For hardware based FIFO caused delay reporting.
211 * Optional.
212 */
213 snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *,
214 struct snd_soc_dai *);
Mark Browna47cbe72008-07-23 14:03:07 +0100215};
216
217/*
Liam Girdwoodf0fba2a2010-03-17 20:15:21 +0000218 * Digital Audio Interface Driver.
Mark Browna47cbe72008-07-23 14:03:07 +0100219 *
Liam Girdwoodf0fba2a2010-03-17 20:15:21 +0000220 * Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97
221 * operations and capabilities. Codec and platform drivers will register this
222 * structure for every DAI they have.
223 *
224 * This structure covers the clocking, formating and ALSA operations for each
225 * interface.
Mark Browna47cbe72008-07-23 14:03:07 +0100226 */
Liam Girdwoodf0fba2a2010-03-17 20:15:21 +0000227struct snd_soc_dai_driver {
Mark Browna47cbe72008-07-23 14:03:07 +0100228 /* DAI description */
Liam Girdwoodf0fba2a2010-03-17 20:15:21 +0000229 const char *name;
Mark Browna47cbe72008-07-23 14:03:07 +0100230 unsigned int id;
Mark Brown62368292012-05-01 20:03:32 +0100231 unsigned int base;
Mengdong Lin68003e62015-12-31 16:40:43 +0800232 struct snd_soc_dobj dobj;
Mark Browna47cbe72008-07-23 14:03:07 +0100233
Liam Girdwoodf0fba2a2010-03-17 20:15:21 +0000234 /* DAI driver callbacks */
235 int (*probe)(struct snd_soc_dai *dai);
236 int (*remove)(struct snd_soc_dai *dai);
Mark Browndc7d7b82008-12-03 18:21:52 +0000237 int (*suspend)(struct snd_soc_dai *dai);
238 int (*resume)(struct snd_soc_dai *dai);
Vinod Koul49681072012-08-16 17:10:40 +0530239 /* compress dai */
Jie Yang6f0c4222015-10-13 23:41:00 +0800240 int (*compress_new)(struct snd_soc_pcm_runtime *rtd, int num);
Lars-Peter Clausenbc263212014-11-10 22:41:52 +0100241 /* DAI is also used for the control bus */
242 bool bus_control;
Mark Browna47cbe72008-07-23 14:03:07 +0100243
244 /* ops */
Mark Brown1ee46eb2010-12-02 16:10:09 +0000245 const struct snd_soc_dai_ops *ops;
Mark Browna47cbe72008-07-23 14:03:07 +0100246
247 /* DAI capabilities */
248 struct snd_soc_pcm_stream capture;
249 struct snd_soc_pcm_stream playback;
Mark Brown06f409d2009-04-07 18:10:13 +0100250 unsigned int symmetric_rates:1;
Nicolin Chen3635bf02013-11-13 18:56:24 +0800251 unsigned int symmetric_channels:1;
252 unsigned int symmetric_samplebits:1;
Liam Girdwood0168bf02011-06-07 16:08:05 +0100253
254 /* probe ordering - for components with runtime dependencies */
255 int probe_order;
256 int remove_order;
Liam Girdwoodf0fba2a2010-03-17 20:15:21 +0000257};
258
259/*
260 * Digital Audio Interface runtime data.
261 *
262 * Holds runtime data for a DAI.
263 */
264struct snd_soc_dai {
265 const char *name;
266 int id;
267 struct device *dev;
Liam Girdwoodf0fba2a2010-03-17 20:15:21 +0000268
269 /* driver ops */
270 struct snd_soc_dai_driver *driver;
Mark Browna47cbe72008-07-23 14:03:07 +0100271
272 /* DAI runtime info */
Walter Yang24430542016-11-04 21:48:03 +0800273 unsigned int capture_active; /* stream is in use */
Vidyakumar Athota25f34112016-08-15 14:02:11 -0700274 unsigned int playback_active; /* stream is in use */
Liam Girdwoodf0fba2a2010-03-17 20:15:21 +0000275 unsigned int symmetric_rates:1;
Nicolin Chen3635bf02013-11-13 18:56:24 +0800276 unsigned int symmetric_channels:1;
277 unsigned int symmetric_samplebits:1;
Mark Browna47cbe72008-07-23 14:03:07 +0100278 unsigned int active;
Liam Girdwoodf0fba2a2010-03-17 20:15:21 +0000279 unsigned char probed:1;
Mark Browna47cbe72008-07-23 14:03:07 +0100280
Mark Brown888df392012-02-16 19:37:51 -0800281 struct snd_soc_dapm_widget *playback_widget;
282 struct snd_soc_dapm_widget *capture_widget;
283
Liam Girdwoodf0fba2a2010-03-17 20:15:21 +0000284 /* DAI DMA data */
285 void *playback_dma_data;
286 void *capture_dma_data;
Mark Browna47cbe72008-07-23 14:03:07 +0100287
Dong Aisheng17841022011-08-29 17:15:14 +0800288 /* Symmetry data - only valid if symmetry is being enforced */
289 unsigned int rate;
Nicolin Chen3635bf02013-11-13 18:56:24 +0800290 unsigned int channels;
291 unsigned int sample_bits;
Dong Aisheng17841022011-08-29 17:15:14 +0800292
Liam Girdwoodf0fba2a2010-03-17 20:15:21 +0000293 /* parent platform/codec */
Mark Brown2466ab92012-03-02 16:18:30 +0000294 struct snd_soc_codec *codec;
Lars-Peter Clausen6106d122014-03-05 13:17:46 +0100295 struct snd_soc_component *component;
Mark Brown2466ab92012-03-02 16:18:30 +0000296
Benoit Cousson88bd8702014-07-08 23:19:34 +0200297 /* CODEC TDM slot masks and params (for fixup) */
298 unsigned int tx_mask;
299 unsigned int rx_mask;
300
Mark Browna47cbe72008-07-23 14:03:07 +0100301 struct list_head list;
302};
303
Daniel Mackfd23b7d2010-03-19 14:52:55 +0000304static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai *dai,
305 const struct snd_pcm_substream *ss)
306{
307 return (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
Liam Girdwoodf0fba2a2010-03-17 20:15:21 +0000308 dai->playback_dma_data : dai->capture_dma_data;
Daniel Mackfd23b7d2010-03-19 14:52:55 +0000309}
310
311static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai,
312 const struct snd_pcm_substream *ss,
313 void *data)
314{
315 if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK)
Liam Girdwoodf0fba2a2010-03-17 20:15:21 +0000316 dai->playback_dma_data = data;
Daniel Mackfd23b7d2010-03-19 14:52:55 +0000317 else
Liam Girdwoodf0fba2a2010-03-17 20:15:21 +0000318 dai->capture_dma_data = data;
319}
320
Mark Brownecfc0c02013-10-17 21:13:19 +0100321static inline void snd_soc_dai_init_dma_data(struct snd_soc_dai *dai,
322 void *playback, void *capture)
323{
324 dai->playback_dma_data = playback;
325 dai->capture_dma_data = capture;
326}
327
Liam Girdwoodf0fba2a2010-03-17 20:15:21 +0000328static inline void snd_soc_dai_set_drvdata(struct snd_soc_dai *dai,
329 void *data)
330{
331 dev_set_drvdata(dai->dev, data);
332}
333
334static inline void *snd_soc_dai_get_drvdata(struct snd_soc_dai *dai)
335{
336 return dev_get_drvdata(dai->dev);
Daniel Mackfd23b7d2010-03-19 14:52:55 +0000337}
338
Mark Browna47cbe72008-07-23 14:03:07 +0100339#endif