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ASoC: soc-dai: add snd_soc_dai_compr_set_params()
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1/* SPDX-License-Identifier: GPL-2.0
2 *
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3 * linux/sound/soc-dai.h -- ALSA SoC Layer
4 *
5 * Copyright: 2005-2008 Wolfson Microelectronics. PLC.
6 *
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7 * Digital Audio Interface (DAI) API.
8 */
9
10#ifndef __LINUX_SND_SOC_DAI_H
11#define __LINUX_SND_SOC_DAI_H
12
13
14#include <linux/list.h>
d2e3cb85 15#include <sound/asoc.h>
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16
17struct snd_pcm_substream;
888df395 18struct snd_soc_dapm_widget;
49681077 19struct snd_compr_stream;
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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 */
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27#define SND_SOC_DAIFMT_I2S SND_SOC_DAI_FORMAT_I2S
28#define SND_SOC_DAIFMT_RIGHT_J SND_SOC_DAI_FORMAT_RIGHT_J
29#define SND_SOC_DAIFMT_LEFT_J SND_SOC_DAI_FORMAT_LEFT_J
30#define SND_SOC_DAIFMT_DSP_A SND_SOC_DAI_FORMAT_DSP_A
31#define SND_SOC_DAIFMT_DSP_B SND_SOC_DAI_FORMAT_DSP_B
32#define SND_SOC_DAIFMT_AC97 SND_SOC_DAI_FORMAT_AC97
33#define SND_SOC_DAIFMT_PDM SND_SOC_DAI_FORMAT_PDM
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34
35/* left and right justified also known as MSB and LSB respectively */
36#define SND_SOC_DAIFMT_MSB SND_SOC_DAIFMT_LEFT_J
37#define SND_SOC_DAIFMT_LSB SND_SOC_DAIFMT_RIGHT_J
38
39/*
40 * DAI Clock gating.
41 *
47db8e89 42 * DAI bit clocks can be be gated (disabled) when the DAI is not
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43 * sending or receiving PCM data in a frame. This can be used to save power.
44 */
75d9ac46 45#define SND_SOC_DAIFMT_CONT (1 << 4) /* continuous clock */
eef28e10 46#define SND_SOC_DAIFMT_GATED (0 << 4) /* clock is gated */
a47cbe72 47
a47cbe72 48/*
1d387a3f 49 * DAI hardware signal polarity.
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50 *
51 * Specifies whether the DAI can also support inverted clocks for the specified
52 * format.
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53 *
54 * BCLK:
55 * - "normal" polarity means signal is available at rising edge of BCLK
56 * - "inverted" polarity means signal is available at falling edge of BCLK
57 *
58 * FSYNC "normal" polarity depends on the frame format:
59 * - I2S: frame consists of left then right channel data. Left channel starts
60 * with falling FSYNC edge, right channel starts with rising FSYNC edge.
61 * - Left/Right Justified: frame consists of left then right channel data.
62 * Left channel starts with rising FSYNC edge, right channel starts with
63 * falling FSYNC edge.
64 * - DSP A/B: Frame starts with rising FSYNC edge.
65 * - AC97: Frame starts with rising FSYNC edge.
66 *
67 * "Negative" FSYNC polarity is the one opposite of "normal" polarity.
a47cbe72 68 */
5d163336 69#define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bit clock + frame */
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70#define SND_SOC_DAIFMT_NB_IF (2 << 8) /* normal BCLK + inv FRM */
71#define SND_SOC_DAIFMT_IB_NF (3 << 8) /* invert BCLK + nor FRM */
72#define SND_SOC_DAIFMT_IB_IF (4 << 8) /* invert BCLK + FRM */
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73
74/*
75 * DAI hardware clock masters.
76 *
77 * This is wrt the codec, the inverse is true for the interface
47db8e89 78 * i.e. if the codec is clk and FRM master then the interface is
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79 * clk and frame slave.
80 */
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81#define SND_SOC_DAIFMT_CBM_CFM (1 << 12) /* codec clk & FRM master */
82#define SND_SOC_DAIFMT_CBS_CFM (2 << 12) /* codec clk slave & FRM master */
83#define SND_SOC_DAIFMT_CBM_CFS (3 << 12) /* codec clk master & frame slave */
84#define SND_SOC_DAIFMT_CBS_CFS (4 << 12) /* codec clk & FRM slave */
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85
86#define SND_SOC_DAIFMT_FORMAT_MASK 0x000f
87#define SND_SOC_DAIFMT_CLOCK_MASK 0x00f0
88#define SND_SOC_DAIFMT_INV_MASK 0x0f00
89#define SND_SOC_DAIFMT_MASTER_MASK 0xf000
90
91/*
92 * Master Clock Directions
93 */
94#define SND_SOC_CLOCK_IN 0
95#define SND_SOC_CLOCK_OUT 1
96
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97#define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S8 |\
98 SNDRV_PCM_FMTBIT_S16_LE |\
99 SNDRV_PCM_FMTBIT_S16_BE |\
100 SNDRV_PCM_FMTBIT_S20_3LE |\
101 SNDRV_PCM_FMTBIT_S20_3BE |\
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102 SNDRV_PCM_FMTBIT_S20_LE |\
103 SNDRV_PCM_FMTBIT_S20_BE |\
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104 SNDRV_PCM_FMTBIT_S24_3LE |\
105 SNDRV_PCM_FMTBIT_S24_3BE |\
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106 SNDRV_PCM_FMTBIT_S32_LE |\
107 SNDRV_PCM_FMTBIT_S32_BE)
33f503c9 108
f0fba2ad 109struct snd_soc_dai_driver;
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110struct 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,
85488037 121 int pll_id, int source, unsigned int freq_in, unsigned int freq_out);
a47cbe72 122
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123int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio);
124
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125/* 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,
a5479e38 129 unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width);
a47cbe72 130
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131int 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
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135int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate);
136
137/* Digital Audio Interface mute */
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138int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute,
139 int direction);
a47cbe72 140
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141
142int snd_soc_dai_get_channel_map(struct snd_soc_dai *dai,
143 unsigned int *tx_num, unsigned int *tx_slot,
144 unsigned int *rx_num, unsigned int *rx_slot);
145
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146int snd_soc_dai_is_dummy(struct snd_soc_dai *dai);
147
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148int snd_soc_dai_hw_params(struct snd_soc_dai *dai,
149 struct snd_pcm_substream *substream,
150 struct snd_pcm_hw_params *params);
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151void snd_soc_dai_hw_free(struct snd_soc_dai *dai,
152 struct snd_pcm_substream *substream);
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153int snd_soc_dai_startup(struct snd_soc_dai *dai,
154 struct snd_pcm_substream *substream);
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155void snd_soc_dai_shutdown(struct snd_soc_dai *dai,
156 struct snd_pcm_substream *substream);
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157snd_pcm_sframes_t snd_soc_dai_delay(struct snd_soc_dai *dai,
158 struct snd_pcm_substream *substream);
e0f22622 159void snd_soc_dai_suspend(struct snd_soc_dai *dai);
24b09d05 160void snd_soc_dai_resume(struct snd_soc_dai *dai);
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161int snd_soc_dai_compress_new(struct snd_soc_dai *dai,
162 struct snd_soc_pcm_runtime *rtd, int num);
467fece8 163bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int stream);
aa6166c2 164
51801aea 165int snd_soc_pcm_dai_probe(struct snd_soc_pcm_runtime *rtd, int order);
7eaa313b 166int snd_soc_pcm_dai_remove(struct snd_soc_pcm_runtime *rtd, int order);
0b73ba55 167int snd_soc_pcm_dai_new(struct snd_soc_pcm_runtime *rtd);
d108c7fd 168int snd_soc_pcm_dai_prepare(struct snd_pcm_substream *substream);
42f2472d 169int snd_soc_pcm_dai_trigger(struct snd_pcm_substream *substream, int cmd);
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170int snd_soc_pcm_dai_bespoke_trigger(struct snd_pcm_substream *substream,
171 int cmd);
0b73ba55 172
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173int snd_soc_dai_compr_startup(struct snd_soc_dai *dai,
174 struct snd_compr_stream *cstream);
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175void snd_soc_dai_compr_shutdown(struct snd_soc_dai *dai,
176 struct snd_compr_stream *cstream);
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177int snd_soc_dai_compr_trigger(struct snd_soc_dai *dai,
178 struct snd_compr_stream *cstream, int cmd);
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179int snd_soc_dai_compr_set_params(struct snd_soc_dai *dai,
180 struct snd_compr_stream *cstream,
181 struct snd_compr_params *params);
b5ae4cce 182
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183struct snd_soc_dai_ops {
184 /*
185 * DAI clocking configuration, all optional.
186 * Called by soc_card drivers, normally in their hw_params.
187 */
188 int (*set_sysclk)(struct snd_soc_dai *dai,
189 int clk_id, unsigned int freq, int dir);
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190 int (*set_pll)(struct snd_soc_dai *dai, int pll_id, int source,
191 unsigned int freq_in, unsigned int freq_out);
a47cbe72 192 int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div);
e54cf76b 193 int (*set_bclk_ratio)(struct snd_soc_dai *dai, unsigned int ratio);
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194
195 /*
196 * DAI format configuration
197 * Called by soc_card drivers, normally in their hw_params.
198 */
199 int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt);
e5c21514 200 int (*xlate_tdm_slot_mask)(unsigned int slots,
89c67857 201 unsigned int *tx_mask, unsigned int *rx_mask);
a47cbe72 202 int (*set_tdm_slot)(struct snd_soc_dai *dai,
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203 unsigned int tx_mask, unsigned int rx_mask,
204 int slots, int slot_width);
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205 int (*set_channel_map)(struct snd_soc_dai *dai,
206 unsigned int tx_num, unsigned int *tx_slot,
207 unsigned int rx_num, unsigned int *rx_slot);
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208 int (*get_channel_map)(struct snd_soc_dai *dai,
209 unsigned int *tx_num, unsigned int *tx_slot,
210 unsigned int *rx_num, unsigned int *rx_slot);
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211 int (*set_tristate)(struct snd_soc_dai *dai, int tristate);
212
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213 int (*set_sdw_stream)(struct snd_soc_dai *dai,
214 void *stream, int direction);
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215 void *(*get_sdw_stream)(struct snd_soc_dai *dai, int direction);
216
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217 /*
218 * DAI digital mute - optional.
219 * Called by soc-core to minimise any pops.
220 */
221 int (*digital_mute)(struct snd_soc_dai *dai, int mute);
da18396f 222 int (*mute_stream)(struct snd_soc_dai *dai, int mute, int stream);
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223
224 /*
225 * ALSA PCM audio operations - all optional.
226 * Called by soc-core during audio PCM operations.
227 */
228 int (*startup)(struct snd_pcm_substream *,
229 struct snd_soc_dai *);
230 void (*shutdown)(struct snd_pcm_substream *,
231 struct snd_soc_dai *);
232 int (*hw_params)(struct snd_pcm_substream *,
233 struct snd_pcm_hw_params *, struct snd_soc_dai *);
234 int (*hw_free)(struct snd_pcm_substream *,
235 struct snd_soc_dai *);
236 int (*prepare)(struct snd_pcm_substream *,
237 struct snd_soc_dai *);
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238 /*
239 * NOTE: Commands passed to the trigger function are not necessarily
240 * compatible with the current state of the dai. For example this
241 * sequence of commands is possible: START STOP STOP.
242 * So do not unconditionally use refcounting functions in the trigger
243 * function, e.g. clk_enable/disable.
244 */
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245 int (*trigger)(struct snd_pcm_substream *, int,
246 struct snd_soc_dai *);
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247 int (*bespoke_trigger)(struct snd_pcm_substream *, int,
248 struct snd_soc_dai *);
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249 /*
250 * For hardware based FIFO caused delay reporting.
251 * Optional.
252 */
253 snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *,
254 struct snd_soc_dai *);
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255};
256
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257struct snd_soc_cdai_ops {
258 /*
259 * for compress ops
260 */
261 int (*startup)(struct snd_compr_stream *,
262 struct snd_soc_dai *);
263 int (*shutdown)(struct snd_compr_stream *,
264 struct snd_soc_dai *);
265 int (*set_params)(struct snd_compr_stream *,
266 struct snd_compr_params *, struct snd_soc_dai *);
267 int (*get_params)(struct snd_compr_stream *,
268 struct snd_codec *, struct snd_soc_dai *);
269 int (*set_metadata)(struct snd_compr_stream *,
270 struct snd_compr_metadata *, struct snd_soc_dai *);
271 int (*get_metadata)(struct snd_compr_stream *,
272 struct snd_compr_metadata *, struct snd_soc_dai *);
273 int (*trigger)(struct snd_compr_stream *, int,
274 struct snd_soc_dai *);
275 int (*pointer)(struct snd_compr_stream *,
276 struct snd_compr_tstamp *, struct snd_soc_dai *);
277 int (*ack)(struct snd_compr_stream *, size_t,
278 struct snd_soc_dai *);
279};
280
a47cbe72 281/*
f0fba2ad 282 * Digital Audio Interface Driver.
a47cbe72 283 *
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284 * Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97
285 * operations and capabilities. Codec and platform drivers will register this
286 * structure for every DAI they have.
287 *
288 * This structure covers the clocking, formating and ALSA operations for each
289 * interface.
a47cbe72 290 */
f0fba2ad 291struct snd_soc_dai_driver {
a47cbe72 292 /* DAI description */
f0fba2ad 293 const char *name;
a47cbe72 294 unsigned int id;
62368294 295 unsigned int base;
68003e6c 296 struct snd_soc_dobj dobj;
a47cbe72 297
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298 /* DAI driver callbacks */
299 int (*probe)(struct snd_soc_dai *dai);
300 int (*remove)(struct snd_soc_dai *dai);
49681077 301 /* compress dai */
6f0c4226 302 int (*compress_new)(struct snd_soc_pcm_runtime *rtd, int num);
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303 /* Optional Callback used at pcm creation*/
304 int (*pcm_new)(struct snd_soc_pcm_runtime *rtd,
305 struct snd_soc_dai *dai);
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306
307 /* ops */
1ee46ebd 308 const struct snd_soc_dai_ops *ops;
2e622ae4 309 const struct snd_soc_cdai_ops *cops;
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310
311 /* DAI capabilities */
312 struct snd_soc_pcm_stream capture;
313 struct snd_soc_pcm_stream playback;
06f409d7 314 unsigned int symmetric_rates:1;
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315 unsigned int symmetric_channels:1;
316 unsigned int symmetric_samplebits:1;
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317
318 /* probe ordering - for components with runtime dependencies */
319 int probe_order;
320 int remove_order;
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321};
322
323/*
324 * Digital Audio Interface runtime data.
325 *
326 * Holds runtime data for a DAI.
327 */
328struct snd_soc_dai {
329 const char *name;
330 int id;
331 struct device *dev;
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332
333 /* driver ops */
334 struct snd_soc_dai_driver *driver;
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335
336 /* DAI runtime info */
0f6011fd 337 unsigned int stream_active[SNDRV_PCM_STREAM_LAST + 1]; /* usage count */
39b2238e 338
a47cbe72 339 unsigned int active;
a47cbe72 340
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341 struct snd_soc_dapm_widget *playback_widget;
342 struct snd_soc_dapm_widget *capture_widget;
343
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344 /* DAI DMA data */
345 void *playback_dma_data;
346 void *capture_dma_data;
a47cbe72 347
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348 /* Symmetry data - only valid if symmetry is being enforced */
349 unsigned int rate;
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350 unsigned int channels;
351 unsigned int sample_bits;
17841020 352
f0fba2ad 353 /* parent platform/codec */
6106d129 354 struct snd_soc_component *component;
2466ab97 355
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356 /* CODEC TDM slot masks and params (for fixup) */
357 unsigned int tx_mask;
358 unsigned int rx_mask;
359
a47cbe72 360 struct list_head list;
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361
362 /* bit field */
363 unsigned int probed:1;
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364};
365
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366static inline struct snd_soc_pcm_stream *
367snd_soc_dai_get_pcm_stream(const struct snd_soc_dai *dai, int stream)
368{
369 return (stream == SNDRV_PCM_STREAM_PLAYBACK) ?
370 &dai->driver->playback : &dai->driver->capture;
371}
372
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373static inline
374struct snd_soc_dapm_widget *snd_soc_dai_get_widget(
375 struct snd_soc_dai *dai, int stream)
376{
377 return (stream == SNDRV_PCM_STREAM_PLAYBACK) ?
378 dai->playback_widget : dai->capture_widget;
379}
380
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381static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai *dai,
382 const struct snd_pcm_substream *ss)
383{
384 return (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
f0fba2ad 385 dai->playback_dma_data : dai->capture_dma_data;
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386}
387
388static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai,
389 const struct snd_pcm_substream *ss,
390 void *data)
391{
392 if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK)
f0fba2ad 393 dai->playback_dma_data = data;
fd23b7de 394 else
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395 dai->capture_dma_data = data;
396}
397
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398static inline void snd_soc_dai_init_dma_data(struct snd_soc_dai *dai,
399 void *playback, void *capture)
400{
401 dai->playback_dma_data = playback;
402 dai->capture_dma_data = capture;
403}
404
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405static inline void snd_soc_dai_set_drvdata(struct snd_soc_dai *dai,
406 void *data)
407{
408 dev_set_drvdata(dai->dev, data);
409}
410
411static inline void *snd_soc_dai_get_drvdata(struct snd_soc_dai *dai)
412{
413 return dev_get_drvdata(dai->dev);
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414}
415
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416/**
417 * snd_soc_dai_set_sdw_stream() - Configures a DAI for SDW stream operation
418 * @dai: DAI
419 * @stream: STREAM
420 * @direction: Stream direction(Playback/Capture)
421 * SoundWire subsystem doesn't have a notion of direction and we reuse
422 * the ASoC stream direction to configure sink/source ports.
423 * Playback maps to source ports and Capture for sink ports.
424 *
425 * This should be invoked with NULL to clear the stream set previously.
426 * Returns 0 on success, a negative error code otherwise.
427 */
428static inline int snd_soc_dai_set_sdw_stream(struct snd_soc_dai *dai,
429 void *stream, int direction)
430{
431 if (dai->driver->ops->set_sdw_stream)
432 return dai->driver->ops->set_sdw_stream(dai, stream, direction);
433 else
434 return -ENOTSUPP;
435}
436
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437/**
438 * snd_soc_dai_get_sdw_stream() - Retrieves SDW stream from DAI
439 * @dai: DAI
440 * @direction: Stream direction(Playback/Capture)
441 *
442 * This routine only retrieves that was previously configured
443 * with snd_soc_dai_get_sdw_stream()
444 *
308811a3 445 * Returns pointer to stream or -ENOTSUPP if callback is not supported;
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446 */
447static inline void *snd_soc_dai_get_sdw_stream(struct snd_soc_dai *dai,
448 int direction)
449{
450 if (dai->driver->ops->get_sdw_stream)
451 return dai->driver->ops->get_sdw_stream(dai, direction);
452 else
308811a3 453 return ERR_PTR(-ENOTSUPP);
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454}
455
a47cbe72 456#endif