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1 /*
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 
19 struct snd_pcm_substream;
20 struct snd_soc_dapm_widget;
21 struct snd_compr_stream;
22 
23 /*
24  * DAI hardware audio formats.
25  *
26  * Describes the physical PCM data formating and clocking. Add new formats
27  * to the end.
28  */
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 */
36 
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  *
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.
46  */
47 #define SND_SOC_DAIFMT_CONT		(1 << 4) /* continuous clock */
48 #define SND_SOC_DAIFMT_GATED		(0 << 4) /* clock is gated */
49 
50 /*
51  * DAI hardware signal polarity.
52  *
53  * Specifies whether the DAI can also support inverted clocks for the specified
54  * format.
55  *
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.
70  */
71 #define SND_SOC_DAIFMT_NB_NF		(0 << 8) /* normal bit clock + frame */
72 #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 */
75 
76 /*
77  * DAI hardware clock masters.
78  *
79  * This is wrt the codec, the inverse is true for the interface
80  * i.e. if the codec is clk and FRM master then the interface is
81  * clk and frame slave.
82  */
83 #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 */
87 
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 
99 #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 |\
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 int snd_soc_dai_is_dummy(struct snd_soc_dai *dai);
142 
143 struct snd_soc_dai_ops {
144 	/*
145 	 * DAI clocking configuration, all optional.
146 	 * Called by soc_card drivers, normally in their hw_params.
147 	 */
148 	int (*set_sysclk)(struct snd_soc_dai *dai,
149 		int clk_id, unsigned int freq, int dir);
150 	int (*set_pll)(struct snd_soc_dai *dai, int pll_id, int source,
151 		unsigned int freq_in, unsigned int freq_out);
152 	int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div);
153 	int (*set_bclk_ratio)(struct snd_soc_dai *dai, unsigned int ratio);
154 
155 	/*
156 	 * DAI format configuration
157 	 * Called by soc_card drivers, normally in their hw_params.
158 	 */
159 	int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt);
160 	int (*xlate_tdm_slot_mask)(unsigned int slots,
161 		unsigned int *tx_mask, unsigned int *rx_mask);
162 	int (*set_tdm_slot)(struct snd_soc_dai *dai,
163 		unsigned int tx_mask, unsigned int rx_mask,
164 		int slots, int slot_width);
165 	int (*set_channel_map)(struct snd_soc_dai *dai,
166 		unsigned int tx_num, unsigned int *tx_slot,
167 		unsigned int rx_num, unsigned int *rx_slot);
168 	int (*set_tristate)(struct snd_soc_dai *dai, int tristate);
169 
170 	/*
171 	 * DAI digital mute - optional.
172 	 * Called by soc-core to minimise any pops.
173 	 */
174 	int (*digital_mute)(struct snd_soc_dai *dai, int mute);
175 	int (*mute_stream)(struct snd_soc_dai *dai, int mute, int stream);
176 
177 	/*
178 	 * ALSA PCM audio operations - all optional.
179 	 * Called by soc-core during audio PCM operations.
180 	 */
181 	int (*startup)(struct snd_pcm_substream *,
182 		struct snd_soc_dai *);
183 	void (*shutdown)(struct snd_pcm_substream *,
184 		struct snd_soc_dai *);
185 	int (*hw_params)(struct snd_pcm_substream *,
186 		struct snd_pcm_hw_params *, struct snd_soc_dai *);
187 	int (*hw_free)(struct snd_pcm_substream *,
188 		struct snd_soc_dai *);
189 	int (*prepare)(struct snd_pcm_substream *,
190 		struct snd_soc_dai *);
191 	/*
192 	 * NOTE: Commands passed to the trigger function are not necessarily
193 	 * compatible with the current state of the dai. For example this
194 	 * sequence of commands is possible: START STOP STOP.
195 	 * So do not unconditionally use refcounting functions in the trigger
196 	 * function, e.g. clk_enable/disable.
197 	 */
198 	int (*trigger)(struct snd_pcm_substream *, int,
199 		struct snd_soc_dai *);
200 	int (*bespoke_trigger)(struct snd_pcm_substream *, int,
201 		struct snd_soc_dai *);
202 	/*
203 	 * For hardware based FIFO caused delay reporting.
204 	 * Optional.
205 	 */
206 	snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *,
207 		struct snd_soc_dai *);
208 };
209 
210 /*
211  * Digital Audio Interface Driver.
212  *
213  * Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97
214  * operations and capabilities. Codec and platform drivers will register this
215  * structure for every DAI they have.
216  *
217  * This structure covers the clocking, formating and ALSA operations for each
218  * interface.
219  */
220 struct snd_soc_dai_driver {
221 	/* DAI description */
222 	const char *name;
223 	unsigned int id;
224 	unsigned int base;
225 	struct snd_soc_dobj dobj;
226 
227 	/* DAI driver callbacks */
228 	int (*probe)(struct snd_soc_dai *dai);
229 	int (*remove)(struct snd_soc_dai *dai);
230 	int (*suspend)(struct snd_soc_dai *dai);
231 	int (*resume)(struct snd_soc_dai *dai);
232 	/* compress dai */
233 	int (*compress_new)(struct snd_soc_pcm_runtime *rtd, int num);
234 	/* DAI is also used for the control bus */
235 	bool bus_control;
236 
237 	/* ops */
238 	const struct snd_soc_dai_ops *ops;
239 
240 	/* DAI capabilities */
241 	struct snd_soc_pcm_stream capture;
242 	struct snd_soc_pcm_stream playback;
243 	unsigned int symmetric_rates:1;
244 	unsigned int symmetric_channels:1;
245 	unsigned int symmetric_samplebits:1;
246 
247 	/* probe ordering - for components with runtime dependencies */
248 	int probe_order;
249 	int remove_order;
250 };
251 
252 /*
253  * Digital Audio Interface runtime data.
254  *
255  * Holds runtime data for a DAI.
256  */
257 struct snd_soc_dai {
258 	const char *name;
259 	int id;
260 	struct device *dev;
261 
262 	/* driver ops */
263 	struct snd_soc_dai_driver *driver;
264 
265 	/* DAI runtime info */
266 	unsigned int capture_active:1;		/* stream is in use */
267 	unsigned int playback_active:1;		/* stream is in use */
268 	unsigned int symmetric_rates:1;
269 	unsigned int symmetric_channels:1;
270 	unsigned int symmetric_samplebits:1;
271 	unsigned int active;
272 	unsigned char probed:1;
273 
274 	struct snd_soc_dapm_widget *playback_widget;
275 	struct snd_soc_dapm_widget *capture_widget;
276 
277 	/* DAI DMA data */
278 	void *playback_dma_data;
279 	void *capture_dma_data;
280 
281 	/* Symmetry data - only valid if symmetry is being enforced */
282 	unsigned int rate;
283 	unsigned int channels;
284 	unsigned int sample_bits;
285 
286 	/* parent platform/codec */
287 	struct snd_soc_codec *codec;
288 	struct snd_soc_component *component;
289 
290 	/* CODEC TDM slot masks and params (for fixup) */
291 	unsigned int tx_mask;
292 	unsigned int rx_mask;
293 
294 	struct list_head list;
295 };
296 
snd_soc_dai_get_dma_data(const struct snd_soc_dai * dai,const struct snd_pcm_substream * ss)297 static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai *dai,
298 					     const struct snd_pcm_substream *ss)
299 {
300 	return (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
301 		dai->playback_dma_data : dai->capture_dma_data;
302 }
303 
snd_soc_dai_set_dma_data(struct snd_soc_dai * dai,const struct snd_pcm_substream * ss,void * data)304 static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai,
305 					    const struct snd_pcm_substream *ss,
306 					    void *data)
307 {
308 	if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK)
309 		dai->playback_dma_data = data;
310 	else
311 		dai->capture_dma_data = data;
312 }
313 
snd_soc_dai_init_dma_data(struct snd_soc_dai * dai,void * playback,void * capture)314 static inline void snd_soc_dai_init_dma_data(struct snd_soc_dai *dai,
315 					     void *playback, void *capture)
316 {
317 	dai->playback_dma_data = playback;
318 	dai->capture_dma_data = capture;
319 }
320 
snd_soc_dai_set_drvdata(struct snd_soc_dai * dai,void * data)321 static inline void snd_soc_dai_set_drvdata(struct snd_soc_dai *dai,
322 		void *data)
323 {
324 	dev_set_drvdata(dai->dev, data);
325 }
326 
snd_soc_dai_get_drvdata(struct snd_soc_dai * dai)327 static inline void *snd_soc_dai_get_drvdata(struct snd_soc_dai *dai)
328 {
329 	return dev_get_drvdata(dai->dev);
330 }
331 
332 #endif
333