1 /* SPDX-License-Identifier: GPL-2.0
3 * linux/sound/soc-dai.h -- ALSA SoC Layer
5 * Copyright: 2005-2008 Wolfson Microelectronics. PLC.
7 * Digital Audio Interface (DAI) API.
10 #ifndef __LINUX_SND_SOC_DAI_H
11 #define __LINUX_SND_SOC_DAI_H
14 #include <linux/list.h>
15 #include <sound/asoc.h>
17 struct snd_pcm_substream
;
18 struct snd_soc_dapm_widget
;
19 struct snd_compr_stream
;
22 * DAI hardware audio formats.
24 * Describes the physical PCM data formating and clocking. Add new formats
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
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
42 * DAI bit clocks can be be gated (disabled) when the DAI is not
43 * sending or receiving PCM data in a frame. This can be used to save power.
45 #define SND_SOC_DAIFMT_CONT (1 << 4) /* continuous clock */
46 #define SND_SOC_DAIFMT_GATED (0 << 4) /* clock is gated */
49 * DAI hardware signal polarity.
51 * Specifies whether the DAI can also support inverted clocks for the specified
55 * - "normal" polarity means signal is available at rising edge of BCLK
56 * - "inverted" polarity means signal is available at falling edge of BCLK
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
64 * - DSP A/B: Frame starts with rising FSYNC edge.
65 * - AC97: Frame starts with rising FSYNC edge.
67 * "Negative" FSYNC polarity is the one opposite of "normal" polarity.
69 #define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bit clock + frame */
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 */
75 * DAI hardware clock masters.
77 * This is wrt the codec, the inverse is true for the interface
78 * i.e. if the codec is clk and FRM master then the interface is
79 * clk and frame slave.
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 */
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
92 * Master Clock Directions
94 #define SND_SOC_CLOCK_IN 0
95 #define SND_SOC_CLOCK_OUT 1
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 |\
102 SNDRV_PCM_FMTBIT_S20_LE |\
103 SNDRV_PCM_FMTBIT_S20_BE |\
104 SNDRV_PCM_FMTBIT_S24_3LE |\
105 SNDRV_PCM_FMTBIT_S24_3BE |\
106 SNDRV_PCM_FMTBIT_S32_LE |\
107 SNDRV_PCM_FMTBIT_S32_BE)
109 struct snd_soc_dai_driver
;
111 struct snd_ac97_bus_ops
;
113 /* Digital Audio Interface clocking API.*/
114 int snd_soc_dai_set_sysclk(struct snd_soc_dai
*dai
, int clk_id
,
115 unsigned int freq
, int dir
);
117 int snd_soc_dai_set_clkdiv(struct snd_soc_dai
*dai
,
118 int div_id
, int div
);
120 int snd_soc_dai_set_pll(struct snd_soc_dai
*dai
,
121 int pll_id
, int source
, unsigned int freq_in
, unsigned int freq_out
);
123 int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai
*dai
, unsigned int ratio
);
125 /* Digital Audio interface formatting */
126 int snd_soc_dai_set_fmt(struct snd_soc_dai
*dai
, unsigned int fmt
);
128 int snd_soc_dai_set_tdm_slot(struct snd_soc_dai
*dai
,
129 unsigned int tx_mask
, unsigned int rx_mask
, int slots
, int slot_width
);
131 int 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
);
135 int snd_soc_dai_set_tristate(struct snd_soc_dai
*dai
, int tristate
);
137 /* Digital Audio Interface mute */
138 int snd_soc_dai_digital_mute(struct snd_soc_dai
*dai
, int mute
,
142 int 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
);
146 int snd_soc_dai_is_dummy(struct snd_soc_dai
*dai
);
148 int snd_soc_dai_hw_params(struct snd_soc_dai
*dai
,
149 struct snd_pcm_substream
*substream
,
150 struct snd_pcm_hw_params
*params
);
151 void snd_soc_dai_hw_free(struct snd_soc_dai
*dai
,
152 struct snd_pcm_substream
*substream
);
153 int snd_soc_dai_startup(struct snd_soc_dai
*dai
,
154 struct snd_pcm_substream
*substream
);
155 void snd_soc_dai_shutdown(struct snd_soc_dai
*dai
,
156 struct snd_pcm_substream
*substream
);
157 int snd_soc_dai_prepare(struct snd_soc_dai
*dai
,
158 struct snd_pcm_substream
*substream
);
159 int snd_soc_dai_trigger(struct snd_soc_dai
*dai
,
160 struct snd_pcm_substream
*substream
, int cmd
);
161 int snd_soc_dai_bespoke_trigger(struct snd_soc_dai
*dai
,
162 struct snd_pcm_substream
*substream
, int cmd
);
163 snd_pcm_sframes_t
snd_soc_dai_delay(struct snd_soc_dai
*dai
,
164 struct snd_pcm_substream
*substream
);
165 void snd_soc_dai_suspend(struct snd_soc_dai
*dai
);
166 void snd_soc_dai_resume(struct snd_soc_dai
*dai
);
167 int snd_soc_dai_probe(struct snd_soc_dai
*dai
);
168 int snd_soc_dai_remove(struct snd_soc_dai
*dai
);
169 int snd_soc_dai_compress_new(struct snd_soc_dai
*dai
,
170 struct snd_soc_pcm_runtime
*rtd
, int num
);
171 bool snd_soc_dai_stream_valid(struct snd_soc_dai
*dai
, int stream
);
173 struct snd_soc_dai_ops
{
175 * DAI clocking configuration, all optional.
176 * Called by soc_card drivers, normally in their hw_params.
178 int (*set_sysclk
)(struct snd_soc_dai
*dai
,
179 int clk_id
, unsigned int freq
, int dir
);
180 int (*set_pll
)(struct snd_soc_dai
*dai
, int pll_id
, int source
,
181 unsigned int freq_in
, unsigned int freq_out
);
182 int (*set_clkdiv
)(struct snd_soc_dai
*dai
, int div_id
, int div
);
183 int (*set_bclk_ratio
)(struct snd_soc_dai
*dai
, unsigned int ratio
);
186 * DAI format configuration
187 * Called by soc_card drivers, normally in their hw_params.
189 int (*set_fmt
)(struct snd_soc_dai
*dai
, unsigned int fmt
);
190 int (*xlate_tdm_slot_mask
)(unsigned int slots
,
191 unsigned int *tx_mask
, unsigned int *rx_mask
);
192 int (*set_tdm_slot
)(struct snd_soc_dai
*dai
,
193 unsigned int tx_mask
, unsigned int rx_mask
,
194 int slots
, int slot_width
);
195 int (*set_channel_map
)(struct snd_soc_dai
*dai
,
196 unsigned int tx_num
, unsigned int *tx_slot
,
197 unsigned int rx_num
, unsigned int *rx_slot
);
198 int (*get_channel_map
)(struct snd_soc_dai
*dai
,
199 unsigned int *tx_num
, unsigned int *tx_slot
,
200 unsigned int *rx_num
, unsigned int *rx_slot
);
201 int (*set_tristate
)(struct snd_soc_dai
*dai
, int tristate
);
203 int (*set_sdw_stream
)(struct snd_soc_dai
*dai
,
204 void *stream
, int direction
);
205 void *(*get_sdw_stream
)(struct snd_soc_dai
*dai
, int direction
);
208 * DAI digital mute - optional.
209 * Called by soc-core to minimise any pops.
211 int (*digital_mute
)(struct snd_soc_dai
*dai
, int mute
);
212 int (*mute_stream
)(struct snd_soc_dai
*dai
, int mute
, int stream
);
215 * ALSA PCM audio operations - all optional.
216 * Called by soc-core during audio PCM operations.
218 int (*startup
)(struct snd_pcm_substream
*,
219 struct snd_soc_dai
*);
220 void (*shutdown
)(struct snd_pcm_substream
*,
221 struct snd_soc_dai
*);
222 int (*hw_params
)(struct snd_pcm_substream
*,
223 struct snd_pcm_hw_params
*, struct snd_soc_dai
*);
224 int (*hw_free
)(struct snd_pcm_substream
*,
225 struct snd_soc_dai
*);
226 int (*prepare
)(struct snd_pcm_substream
*,
227 struct snd_soc_dai
*);
229 * NOTE: Commands passed to the trigger function are not necessarily
230 * compatible with the current state of the dai. For example this
231 * sequence of commands is possible: START STOP STOP.
232 * So do not unconditionally use refcounting functions in the trigger
233 * function, e.g. clk_enable/disable.
235 int (*trigger
)(struct snd_pcm_substream
*, int,
236 struct snd_soc_dai
*);
237 int (*bespoke_trigger
)(struct snd_pcm_substream
*, int,
238 struct snd_soc_dai
*);
240 * For hardware based FIFO caused delay reporting.
243 snd_pcm_sframes_t (*delay
)(struct snd_pcm_substream
*,
244 struct snd_soc_dai
*);
247 struct snd_soc_cdai_ops
{
251 int (*startup
)(struct snd_compr_stream
*,
252 struct snd_soc_dai
*);
253 int (*shutdown
)(struct snd_compr_stream
*,
254 struct snd_soc_dai
*);
255 int (*set_params
)(struct snd_compr_stream
*,
256 struct snd_compr_params
*, struct snd_soc_dai
*);
257 int (*get_params
)(struct snd_compr_stream
*,
258 struct snd_codec
*, struct snd_soc_dai
*);
259 int (*set_metadata
)(struct snd_compr_stream
*,
260 struct snd_compr_metadata
*, struct snd_soc_dai
*);
261 int (*get_metadata
)(struct snd_compr_stream
*,
262 struct snd_compr_metadata
*, struct snd_soc_dai
*);
263 int (*trigger
)(struct snd_compr_stream
*, int,
264 struct snd_soc_dai
*);
265 int (*pointer
)(struct snd_compr_stream
*,
266 struct snd_compr_tstamp
*, struct snd_soc_dai
*);
267 int (*ack
)(struct snd_compr_stream
*, size_t,
268 struct snd_soc_dai
*);
272 * Digital Audio Interface Driver.
274 * Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97
275 * operations and capabilities. Codec and platform drivers will register this
276 * structure for every DAI they have.
278 * This structure covers the clocking, formating and ALSA operations for each
281 struct snd_soc_dai_driver
{
282 /* DAI description */
286 struct snd_soc_dobj dobj
;
288 /* DAI driver callbacks */
289 int (*probe
)(struct snd_soc_dai
*dai
);
290 int (*remove
)(struct snd_soc_dai
*dai
);
292 int (*compress_new
)(struct snd_soc_pcm_runtime
*rtd
, int num
);
293 /* Optional Callback used at pcm creation*/
294 int (*pcm_new
)(struct snd_soc_pcm_runtime
*rtd
,
295 struct snd_soc_dai
*dai
);
298 const struct snd_soc_dai_ops
*ops
;
299 const struct snd_soc_cdai_ops
*cops
;
301 /* DAI capabilities */
302 struct snd_soc_pcm_stream capture
;
303 struct snd_soc_pcm_stream playback
;
304 unsigned int symmetric_rates
:1;
305 unsigned int symmetric_channels
:1;
306 unsigned int symmetric_samplebits
:1;
308 /* probe ordering - for components with runtime dependencies */
314 * Digital Audio Interface runtime data.
316 * Holds runtime data for a DAI.
324 struct snd_soc_dai_driver
*driver
;
326 /* DAI runtime info */
327 unsigned int stream_active
[SNDRV_PCM_STREAM_LAST
+ 1]; /* usage count */
331 struct snd_soc_dapm_widget
*playback_widget
;
332 struct snd_soc_dapm_widget
*capture_widget
;
335 void *playback_dma_data
;
336 void *capture_dma_data
;
338 /* Symmetry data - only valid if symmetry is being enforced */
340 unsigned int channels
;
341 unsigned int sample_bits
;
343 /* parent platform/codec */
344 struct snd_soc_component
*component
;
346 /* CODEC TDM slot masks and params (for fixup) */
347 unsigned int tx_mask
;
348 unsigned int rx_mask
;
350 struct list_head list
;
353 unsigned int probed
:1;
354 unsigned int started
[SNDRV_PCM_STREAM_LAST
+ 1];
357 static inline struct snd_soc_pcm_stream
*
358 snd_soc_dai_get_pcm_stream(const struct snd_soc_dai
*dai
, int stream
)
360 return (stream
== SNDRV_PCM_STREAM_PLAYBACK
) ?
361 &dai
->driver
->playback
: &dai
->driver
->capture
;
365 struct snd_soc_dapm_widget
*snd_soc_dai_get_widget(
366 struct snd_soc_dai
*dai
, int stream
)
368 return (stream
== SNDRV_PCM_STREAM_PLAYBACK
) ?
369 dai
->playback_widget
: dai
->capture_widget
;
372 static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai
*dai
,
373 const struct snd_pcm_substream
*ss
)
375 return (ss
->stream
== SNDRV_PCM_STREAM_PLAYBACK
) ?
376 dai
->playback_dma_data
: dai
->capture_dma_data
;
379 static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai
*dai
,
380 const struct snd_pcm_substream
*ss
,
383 if (ss
->stream
== SNDRV_PCM_STREAM_PLAYBACK
)
384 dai
->playback_dma_data
= data
;
386 dai
->capture_dma_data
= data
;
389 static inline void snd_soc_dai_init_dma_data(struct snd_soc_dai
*dai
,
390 void *playback
, void *capture
)
392 dai
->playback_dma_data
= playback
;
393 dai
->capture_dma_data
= capture
;
396 static inline void snd_soc_dai_set_drvdata(struct snd_soc_dai
*dai
,
399 dev_set_drvdata(dai
->dev
, data
);
402 static inline void *snd_soc_dai_get_drvdata(struct snd_soc_dai
*dai
)
404 return dev_get_drvdata(dai
->dev
);
408 * snd_soc_dai_set_sdw_stream() - Configures a DAI for SDW stream operation
411 * @direction: Stream direction(Playback/Capture)
412 * SoundWire subsystem doesn't have a notion of direction and we reuse
413 * the ASoC stream direction to configure sink/source ports.
414 * Playback maps to source ports and Capture for sink ports.
416 * This should be invoked with NULL to clear the stream set previously.
417 * Returns 0 on success, a negative error code otherwise.
419 static inline int snd_soc_dai_set_sdw_stream(struct snd_soc_dai
*dai
,
420 void *stream
, int direction
)
422 if (dai
->driver
->ops
->set_sdw_stream
)
423 return dai
->driver
->ops
->set_sdw_stream(dai
, stream
, direction
);
429 * snd_soc_dai_get_sdw_stream() - Retrieves SDW stream from DAI
431 * @direction: Stream direction(Playback/Capture)
433 * This routine only retrieves that was previously configured
434 * with snd_soc_dai_get_sdw_stream()
436 * Returns pointer to stream or -ENOTSUPP if callback is not supported;
438 static inline void *snd_soc_dai_get_sdw_stream(struct snd_soc_dai
*dai
,
441 if (dai
->driver
->ops
->get_sdw_stream
)
442 return dai
->driver
->ops
->get_sdw_stream(dai
, direction
);
444 return ERR_PTR(-ENOTSUPP
);