1 /* SPDX-License-Identifier: GPL-2.0
2  *
3  * linux/sound/soc-dai.h -- ALSA SoC Layer
4  *
5  * Copyright:	2005-2008 Wolfson Microelectronics. PLC.
6  *
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>
15 #include <sound/asoc.h>
16 
17 struct snd_pcm_substream;
18 struct snd_soc_dapm_widget;
19 struct snd_compr_stream;
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  */
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
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  *
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.
44  */
45 #define SND_SOC_DAIFMT_CONT		(1 << 4) /* continuous clock */
46 #define SND_SOC_DAIFMT_GATED		(0 << 4) /* clock is gated */
47 
48 /*
49  * DAI hardware signal polarity.
50  *
51  * Specifies whether the DAI can also support inverted clocks for the specified
52  * format.
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.
68  */
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 */
73 
74 /*
75  * DAI hardware clock masters.
76  *
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.
80  */
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 */
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 
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)
108 
109 struct snd_soc_dai_driver;
110 struct snd_soc_dai;
111 struct snd_ac97_bus_ops;
112 
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);
116 
117 int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai,
118 	int div_id, int div);
119 
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);
122 
123 int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio);
124 
125 /* Digital Audio interface formatting */
126 int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt);
127 
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);
130 
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);
134 
135 int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate);
136 
137 /* Digital Audio Interface mute */
138 int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute,
139 			     int direction);
140 
141 
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);
145 
146 int snd_soc_dai_is_dummy(struct snd_soc_dai *dai);
147 
148 struct snd_soc_dai_ops {
149 	/*
150 	 * DAI clocking configuration, all optional.
151 	 * Called by soc_card drivers, normally in their hw_params.
152 	 */
153 	int (*set_sysclk)(struct snd_soc_dai *dai,
154 		int clk_id, unsigned int freq, int dir);
155 	int (*set_pll)(struct snd_soc_dai *dai, int pll_id, int source,
156 		unsigned int freq_in, unsigned int freq_out);
157 	int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div);
158 	int (*set_bclk_ratio)(struct snd_soc_dai *dai, unsigned int ratio);
159 
160 	/*
161 	 * DAI format configuration
162 	 * Called by soc_card drivers, normally in their hw_params.
163 	 */
164 	int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt);
165 	int (*xlate_tdm_slot_mask)(unsigned int slots,
166 		unsigned int *tx_mask, unsigned int *rx_mask);
167 	int (*set_tdm_slot)(struct snd_soc_dai *dai,
168 		unsigned int tx_mask, unsigned int rx_mask,
169 		int slots, int slot_width);
170 	int (*set_channel_map)(struct snd_soc_dai *dai,
171 		unsigned int tx_num, unsigned int *tx_slot,
172 		unsigned int rx_num, unsigned int *rx_slot);
173 	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);
176 	int (*set_tristate)(struct snd_soc_dai *dai, int tristate);
177 
178 	int (*set_sdw_stream)(struct snd_soc_dai *dai,
179 			void *stream, int direction);
180 	/*
181 	 * DAI digital mute - optional.
182 	 * Called by soc-core to minimise any pops.
183 	 */
184 	int (*digital_mute)(struct snd_soc_dai *dai, int mute);
185 	int (*mute_stream)(struct snd_soc_dai *dai, int mute, int stream);
186 
187 	/*
188 	 * ALSA PCM audio operations - all optional.
189 	 * Called by soc-core during audio PCM operations.
190 	 */
191 	int (*startup)(struct snd_pcm_substream *,
192 		struct snd_soc_dai *);
193 	void (*shutdown)(struct snd_pcm_substream *,
194 		struct snd_soc_dai *);
195 	int (*hw_params)(struct snd_pcm_substream *,
196 		struct snd_pcm_hw_params *, struct snd_soc_dai *);
197 	int (*hw_free)(struct snd_pcm_substream *,
198 		struct snd_soc_dai *);
199 	int (*prepare)(struct snd_pcm_substream *,
200 		struct snd_soc_dai *);
201 	/*
202 	 * NOTE: Commands passed to the trigger function are not necessarily
203 	 * compatible with the current state of the dai. For example this
204 	 * sequence of commands is possible: START STOP STOP.
205 	 * So do not unconditionally use refcounting functions in the trigger
206 	 * function, e.g. clk_enable/disable.
207 	 */
208 	int (*trigger)(struct snd_pcm_substream *, int,
209 		struct snd_soc_dai *);
210 	int (*bespoke_trigger)(struct snd_pcm_substream *, int,
211 		struct snd_soc_dai *);
212 	/*
213 	 * For hardware based FIFO caused delay reporting.
214 	 * Optional.
215 	 */
216 	snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *,
217 		struct snd_soc_dai *);
218 };
219 
220 struct snd_soc_cdai_ops {
221 	/*
222 	 * for compress ops
223 	 */
224 	int (*startup)(struct snd_compr_stream *,
225 			struct snd_soc_dai *);
226 	int (*shutdown)(struct snd_compr_stream *,
227 			struct snd_soc_dai *);
228 	int (*set_params)(struct snd_compr_stream *,
229 			struct snd_compr_params *, struct snd_soc_dai *);
230 	int (*get_params)(struct snd_compr_stream *,
231 			struct snd_codec *, struct snd_soc_dai *);
232 	int (*set_metadata)(struct snd_compr_stream *,
233 			struct snd_compr_metadata *, struct snd_soc_dai *);
234 	int (*get_metadata)(struct snd_compr_stream *,
235 			struct snd_compr_metadata *, struct snd_soc_dai *);
236 	int (*trigger)(struct snd_compr_stream *, int,
237 			struct snd_soc_dai *);
238 	int (*pointer)(struct snd_compr_stream *,
239 			struct snd_compr_tstamp *, struct snd_soc_dai *);
240 	int (*ack)(struct snd_compr_stream *, size_t,
241 			struct snd_soc_dai *);
242 };
243 
244 /*
245  * Digital Audio Interface Driver.
246  *
247  * Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97
248  * operations and capabilities. Codec and platform drivers will register this
249  * structure for every DAI they have.
250  *
251  * This structure covers the clocking, formating and ALSA operations for each
252  * interface.
253  */
254 struct snd_soc_dai_driver {
255 	/* DAI description */
256 	const char *name;
257 	unsigned int id;
258 	unsigned int base;
259 	struct snd_soc_dobj dobj;
260 
261 	/* DAI driver callbacks */
262 	int (*probe)(struct snd_soc_dai *dai);
263 	int (*remove)(struct snd_soc_dai *dai);
264 	int (*suspend)(struct snd_soc_dai *dai);
265 	int (*resume)(struct snd_soc_dai *dai);
266 	/* compress dai */
267 	int (*compress_new)(struct snd_soc_pcm_runtime *rtd, int num);
268 	/* Optional Callback used at pcm creation*/
269 	int (*pcm_new)(struct snd_soc_pcm_runtime *rtd,
270 		       struct snd_soc_dai *dai);
271 	/* DAI is also used for the control bus */
272 	bool bus_control;
273 
274 	/* ops */
275 	const struct snd_soc_dai_ops *ops;
276 	const struct snd_soc_cdai_ops *cops;
277 
278 	/* DAI capabilities */
279 	struct snd_soc_pcm_stream capture;
280 	struct snd_soc_pcm_stream playback;
281 	unsigned int symmetric_rates:1;
282 	unsigned int symmetric_channels:1;
283 	unsigned int symmetric_samplebits:1;
284 
285 	/* probe ordering - for components with runtime dependencies */
286 	int probe_order;
287 	int remove_order;
288 };
289 
290 /*
291  * Digital Audio Interface runtime data.
292  *
293  * Holds runtime data for a DAI.
294  */
295 struct snd_soc_dai {
296 	const char *name;
297 	int id;
298 	struct device *dev;
299 
300 	/* driver ops */
301 	struct snd_soc_dai_driver *driver;
302 
303 	/* DAI runtime info */
304 	unsigned int capture_active;		/* stream usage count */
305 	unsigned int playback_active;		/* stream usage count */
306 	unsigned int probed:1;
307 
308 	unsigned int active;
309 
310 	struct snd_soc_dapm_widget *playback_widget;
311 	struct snd_soc_dapm_widget *capture_widget;
312 
313 	/* DAI DMA data */
314 	void *playback_dma_data;
315 	void *capture_dma_data;
316 
317 	/* Symmetry data - only valid if symmetry is being enforced */
318 	unsigned int rate;
319 	unsigned int channels;
320 	unsigned int sample_bits;
321 
322 	/* parent platform/codec */
323 	struct snd_soc_component *component;
324 
325 	/* CODEC TDM slot masks and params (for fixup) */
326 	unsigned int tx_mask;
327 	unsigned int rx_mask;
328 
329 	struct list_head list;
330 };
331 
snd_soc_dai_get_dma_data(const struct snd_soc_dai * dai,const struct snd_pcm_substream * ss)332 static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai *dai,
333 					     const struct snd_pcm_substream *ss)
334 {
335 	return (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
336 		dai->playback_dma_data : dai->capture_dma_data;
337 }
338 
snd_soc_dai_set_dma_data(struct snd_soc_dai * dai,const struct snd_pcm_substream * ss,void * data)339 static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai,
340 					    const struct snd_pcm_substream *ss,
341 					    void *data)
342 {
343 	if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK)
344 		dai->playback_dma_data = data;
345 	else
346 		dai->capture_dma_data = data;
347 }
348 
snd_soc_dai_init_dma_data(struct snd_soc_dai * dai,void * playback,void * capture)349 static inline void snd_soc_dai_init_dma_data(struct snd_soc_dai *dai,
350 					     void *playback, void *capture)
351 {
352 	dai->playback_dma_data = playback;
353 	dai->capture_dma_data = capture;
354 }
355 
snd_soc_dai_set_drvdata(struct snd_soc_dai * dai,void * data)356 static inline void snd_soc_dai_set_drvdata(struct snd_soc_dai *dai,
357 		void *data)
358 {
359 	dev_set_drvdata(dai->dev, data);
360 }
361 
snd_soc_dai_get_drvdata(struct snd_soc_dai * dai)362 static inline void *snd_soc_dai_get_drvdata(struct snd_soc_dai *dai)
363 {
364 	return dev_get_drvdata(dai->dev);
365 }
366 
367 /**
368  * snd_soc_dai_set_sdw_stream() - Configures a DAI for SDW stream operation
369  * @dai: DAI
370  * @stream: STREAM
371  * @direction: Stream direction(Playback/Capture)
372  * SoundWire subsystem doesn't have a notion of direction and we reuse
373  * the ASoC stream direction to configure sink/source ports.
374  * Playback maps to source ports and Capture for sink ports.
375  *
376  * This should be invoked with NULL to clear the stream set previously.
377  * Returns 0 on success, a negative error code otherwise.
378  */
snd_soc_dai_set_sdw_stream(struct snd_soc_dai * dai,void * stream,int direction)379 static inline int snd_soc_dai_set_sdw_stream(struct snd_soc_dai *dai,
380 				void *stream, int direction)
381 {
382 	if (dai->driver->ops->set_sdw_stream)
383 		return dai->driver->ops->set_sdw_stream(dai, stream, direction);
384 	else
385 		return -ENOTSUPP;
386 }
387 
388 #endif
389