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v3.1
  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
 19struct snd_pcm_substream;
 
 
 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		0 /* I2S mode */
 28#define SND_SOC_DAIFMT_RIGHT_J		1 /* Right Justified mode */
 29#define SND_SOC_DAIFMT_LEFT_J		2 /* Left Justified mode */
 30#define SND_SOC_DAIFMT_DSP_A		3 /* L data MSB after FRM LRC */
 31#define SND_SOC_DAIFMT_DSP_B		4 /* L data MSB during FRM LRC */
 32#define SND_SOC_DAIFMT_AC97		5 /* AC97 */
 33#define SND_SOC_DAIFMT_PDM		6 /* Pulse density modulation */
 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		(0 << 4) /* continuous clock */
 46#define SND_SOC_DAIFMT_GATED		(1 << 4) /* clock is gated */
 47
 48/*
 49 * DAI hardware signal inversions.
 50 *
 51 * Specifies whether the DAI can also support inverted clocks for the specified
 52 * format.
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 53 */
 54#define SND_SOC_DAIFMT_NB_NF		(0 << 8) /* normal bit clock + frame */
 55#define SND_SOC_DAIFMT_NB_IF		(1 << 8) /* normal BCLK + inv FRM */
 56#define SND_SOC_DAIFMT_IB_NF		(2 << 8) /* invert BCLK + nor FRM */
 57#define SND_SOC_DAIFMT_IB_IF		(3 << 8) /* invert BCLK + FRM */
 58
 59/*
 60 * DAI hardware clock masters.
 61 *
 62 * This is wrt the codec, the inverse is true for the interface
 63 * i.e. if the codec is clk and FRM master then the interface is
 64 * clk and frame slave.
 65 */
 66#define SND_SOC_DAIFMT_CBM_CFM		(0 << 12) /* codec clk & FRM master */
 67#define SND_SOC_DAIFMT_CBS_CFM		(1 << 12) /* codec clk slave & FRM master */
 68#define SND_SOC_DAIFMT_CBM_CFS		(2 << 12) /* codec clk master & frame slave */
 69#define SND_SOC_DAIFMT_CBS_CFS		(3 << 12) /* codec clk & FRM slave */
 70
 71#define SND_SOC_DAIFMT_FORMAT_MASK	0x000f
 72#define SND_SOC_DAIFMT_CLOCK_MASK	0x00f0
 73#define SND_SOC_DAIFMT_INV_MASK		0x0f00
 74#define SND_SOC_DAIFMT_MASTER_MASK	0xf000
 75
 76/*
 77 * Master Clock Directions
 78 */
 79#define SND_SOC_CLOCK_IN		0
 80#define SND_SOC_CLOCK_OUT		1
 81
 82#define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S8 |\
 83			       SNDRV_PCM_FMTBIT_S16_LE |\
 84			       SNDRV_PCM_FMTBIT_S16_BE |\
 85			       SNDRV_PCM_FMTBIT_S20_3LE |\
 86			       SNDRV_PCM_FMTBIT_S20_3BE |\
 
 
 87			       SNDRV_PCM_FMTBIT_S24_3LE |\
 88			       SNDRV_PCM_FMTBIT_S24_3BE |\
 89                               SNDRV_PCM_FMTBIT_S32_LE |\
 90                               SNDRV_PCM_FMTBIT_S32_BE)
 91
 92struct snd_soc_dai_driver;
 93struct snd_soc_dai;
 94struct snd_ac97_bus_ops;
 95
 96/* Digital Audio Interface registration */
 97int snd_soc_register_dai(struct device *dev,
 98		struct snd_soc_dai_driver *dai_drv);
 99void snd_soc_unregister_dai(struct device *dev);
100int snd_soc_register_dais(struct device *dev,
101		struct snd_soc_dai_driver *dai_drv, size_t count);
102void snd_soc_unregister_dais(struct device *dev, size_t count);
103
104/* Digital Audio Interface clocking API.*/
105int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
106	unsigned int freq, int dir);
107
108int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai,
109	int div_id, int div);
110
111int snd_soc_dai_set_pll(struct snd_soc_dai *dai,
112	int pll_id, int source, unsigned int freq_in, unsigned int freq_out);
113
 
 
114/* Digital Audio interface formatting */
115int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt);
116
117int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
118	unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width);
119
120int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai,
121	unsigned int tx_num, unsigned int *tx_slot,
122	unsigned int rx_num, unsigned int *rx_slot);
123
124int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate);
125
126/* Digital Audio Interface mute */
127int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute);
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
128
129struct snd_soc_dai_ops {
130	/*
131	 * DAI clocking configuration, all optional.
132	 * Called by soc_card drivers, normally in their hw_params.
133	 */
134	int (*set_sysclk)(struct snd_soc_dai *dai,
135		int clk_id, unsigned int freq, int dir);
136	int (*set_pll)(struct snd_soc_dai *dai, int pll_id, int source,
137		unsigned int freq_in, unsigned int freq_out);
138	int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div);
 
139
140	/*
141	 * DAI format configuration
142	 * Called by soc_card drivers, normally in their hw_params.
143	 */
144	int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt);
 
 
145	int (*set_tdm_slot)(struct snd_soc_dai *dai,
146		unsigned int tx_mask, unsigned int rx_mask,
147		int slots, int slot_width);
148	int (*set_channel_map)(struct snd_soc_dai *dai,
149		unsigned int tx_num, unsigned int *tx_slot,
150		unsigned int rx_num, unsigned int *rx_slot);
 
 
 
151	int (*set_tristate)(struct snd_soc_dai *dai, int tristate);
152
 
 
153	/*
154	 * DAI digital mute - optional.
155	 * Called by soc-core to minimise any pops.
156	 */
157	int (*digital_mute)(struct snd_soc_dai *dai, int mute);
 
158
159	/*
160	 * ALSA PCM audio operations - all optional.
161	 * Called by soc-core during audio PCM operations.
162	 */
163	int (*startup)(struct snd_pcm_substream *,
164		struct snd_soc_dai *);
165	void (*shutdown)(struct snd_pcm_substream *,
166		struct snd_soc_dai *);
167	int (*hw_params)(struct snd_pcm_substream *,
168		struct snd_pcm_hw_params *, struct snd_soc_dai *);
169	int (*hw_free)(struct snd_pcm_substream *,
170		struct snd_soc_dai *);
171	int (*prepare)(struct snd_pcm_substream *,
172		struct snd_soc_dai *);
 
 
 
 
 
 
 
173	int (*trigger)(struct snd_pcm_substream *, int,
174		struct snd_soc_dai *);
 
 
175	/*
176	 * For hardware based FIFO caused delay reporting.
177	 * Optional.
178	 */
179	snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *,
180		struct snd_soc_dai *);
181};
182
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
183/*
184 * Digital Audio Interface Driver.
185 *
186 * Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97
187 * operations and capabilities. Codec and platform drivers will register this
188 * structure for every DAI they have.
189 *
190 * This structure covers the clocking, formating and ALSA operations for each
191 * interface.
192 */
193struct snd_soc_dai_driver {
194	/* DAI description */
195	const char *name;
196	unsigned int id;
197	int ac97_control;
 
198
199	/* DAI driver callbacks */
200	int (*probe)(struct snd_soc_dai *dai);
201	int (*remove)(struct snd_soc_dai *dai);
202	int (*suspend)(struct snd_soc_dai *dai);
203	int (*resume)(struct snd_soc_dai *dai);
 
 
 
 
 
204
205	/* ops */
206	const struct snd_soc_dai_ops *ops;
 
207
208	/* DAI capabilities */
209	struct snd_soc_pcm_stream capture;
210	struct snd_soc_pcm_stream playback;
211	unsigned int symmetric_rates:1;
 
 
 
212
213	/* probe ordering - for components with runtime dependencies */
214	int probe_order;
215	int remove_order;
216};
217
218/*
219 * Digital Audio Interface runtime data.
220 *
221 * Holds runtime data for a DAI.
222 */
223struct snd_soc_dai {
224	const char *name;
225	int id;
226	struct device *dev;
227	void *ac97_pdata;	/* platform_data for the ac97 codec */
228
229	/* driver ops */
230	struct snd_soc_dai_driver *driver;
231
232	/* DAI runtime info */
233	unsigned int capture_active:1;		/* stream is in use */
234	unsigned int playback_active:1;		/* stream is in use */
235	unsigned int symmetric_rates:1;
236	struct snd_pcm_runtime *runtime;
237	unsigned int active;
238	unsigned char pop_wait:1;
239	unsigned char probed:1;
 
240
241	/* DAI DMA data */
242	void *playback_dma_data;
243	void *capture_dma_data;
244
 
 
 
 
 
245	/* parent platform/codec */
246	union {
247		struct snd_soc_platform *platform;
248		struct snd_soc_codec *codec;
249	};
250	struct snd_soc_card *card;
251
252	struct list_head list;
253	struct list_head card_list;
254};
255
256static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai *dai,
257					     const struct snd_pcm_substream *ss)
258{
259	return (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
260		dai->playback_dma_data : dai->capture_dma_data;
261}
262
263static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai,
264					    const struct snd_pcm_substream *ss,
265					    void *data)
266{
267	if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK)
268		dai->playback_dma_data = data;
269	else
270		dai->capture_dma_data = data;
271}
272
 
 
 
 
 
 
 
273static inline void snd_soc_dai_set_drvdata(struct snd_soc_dai *dai,
274		void *data)
275{
276	dev_set_drvdata(dai->dev, data);
277}
278
279static inline void *snd_soc_dai_get_drvdata(struct snd_soc_dai *dai)
280{
281	return dev_get_drvdata(dai->dev);
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
282}
283
284#endif
v5.4
  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
 17struct snd_pcm_substream;
 18struct snd_soc_dapm_widget;
 19struct 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
109struct snd_soc_dai_driver;
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,
121	int pll_id, int source, unsigned int freq_in, unsigned int freq_out);
122
123int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio);
124
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,
129	unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width);
130
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
135int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate);
136
137/* Digital Audio Interface mute */
138int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute,
139			     int direction);
140
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
146int snd_soc_dai_is_dummy(struct snd_soc_dai *dai);
147
148int snd_soc_dai_hw_params(struct snd_soc_dai *dai,
149			  struct snd_pcm_substream *substream,
150			  struct snd_pcm_hw_params *params);
151void snd_soc_dai_hw_free(struct snd_soc_dai *dai,
152			 struct snd_pcm_substream *substream);
153int snd_soc_dai_startup(struct snd_soc_dai *dai,
154			struct snd_pcm_substream *substream);
155void snd_soc_dai_shutdown(struct snd_soc_dai *dai,
156			  struct snd_pcm_substream *substream);
157int snd_soc_dai_prepare(struct snd_soc_dai *dai,
158			struct snd_pcm_substream *substream);
159int snd_soc_dai_trigger(struct snd_soc_dai *dai,
160			struct snd_pcm_substream *substream, int cmd);
161int snd_soc_dai_bespoke_trigger(struct snd_soc_dai *dai,
162			struct snd_pcm_substream *substream, int cmd);
163snd_pcm_sframes_t snd_soc_dai_delay(struct snd_soc_dai *dai,
164				    struct snd_pcm_substream *substream);
165void snd_soc_dai_suspend(struct snd_soc_dai *dai);
166void snd_soc_dai_resume(struct snd_soc_dai *dai);
167int snd_soc_dai_probe(struct snd_soc_dai *dai);
168int snd_soc_dai_remove(struct snd_soc_dai *dai);
169int snd_soc_dai_compress_new(struct snd_soc_dai *dai,
170			     struct snd_soc_pcm_runtime *rtd, int num);
171bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int stream);
172
173struct snd_soc_dai_ops {
174	/*
175	 * DAI clocking configuration, all optional.
176	 * Called by soc_card drivers, normally in their hw_params.
177	 */
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);
184
185	/*
186	 * DAI format configuration
187	 * Called by soc_card drivers, normally in their hw_params.
188	 */
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);
202
203	int (*set_sdw_stream)(struct snd_soc_dai *dai,
204			void *stream, int direction);
205	/*
206	 * DAI digital mute - optional.
207	 * Called by soc-core to minimise any pops.
208	 */
209	int (*digital_mute)(struct snd_soc_dai *dai, int mute);
210	int (*mute_stream)(struct snd_soc_dai *dai, int mute, int stream);
211
212	/*
213	 * ALSA PCM audio operations - all optional.
214	 * Called by soc-core during audio PCM operations.
215	 */
216	int (*startup)(struct snd_pcm_substream *,
217		struct snd_soc_dai *);
218	void (*shutdown)(struct snd_pcm_substream *,
219		struct snd_soc_dai *);
220	int (*hw_params)(struct snd_pcm_substream *,
221		struct snd_pcm_hw_params *, struct snd_soc_dai *);
222	int (*hw_free)(struct snd_pcm_substream *,
223		struct snd_soc_dai *);
224	int (*prepare)(struct snd_pcm_substream *,
225		struct snd_soc_dai *);
226	/*
227	 * NOTE: Commands passed to the trigger function are not necessarily
228	 * compatible with the current state of the dai. For example this
229	 * sequence of commands is possible: START STOP STOP.
230	 * So do not unconditionally use refcounting functions in the trigger
231	 * function, e.g. clk_enable/disable.
232	 */
233	int (*trigger)(struct snd_pcm_substream *, int,
234		struct snd_soc_dai *);
235	int (*bespoke_trigger)(struct snd_pcm_substream *, int,
236		struct snd_soc_dai *);
237	/*
238	 * For hardware based FIFO caused delay reporting.
239	 * Optional.
240	 */
241	snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *,
242		struct snd_soc_dai *);
243};
244
245struct snd_soc_cdai_ops {
246	/*
247	 * for compress ops
248	 */
249	int (*startup)(struct snd_compr_stream *,
250			struct snd_soc_dai *);
251	int (*shutdown)(struct snd_compr_stream *,
252			struct snd_soc_dai *);
253	int (*set_params)(struct snd_compr_stream *,
254			struct snd_compr_params *, struct snd_soc_dai *);
255	int (*get_params)(struct snd_compr_stream *,
256			struct snd_codec *, struct snd_soc_dai *);
257	int (*set_metadata)(struct snd_compr_stream *,
258			struct snd_compr_metadata *, struct snd_soc_dai *);
259	int (*get_metadata)(struct snd_compr_stream *,
260			struct snd_compr_metadata *, struct snd_soc_dai *);
261	int (*trigger)(struct snd_compr_stream *, int,
262			struct snd_soc_dai *);
263	int (*pointer)(struct snd_compr_stream *,
264			struct snd_compr_tstamp *, struct snd_soc_dai *);
265	int (*ack)(struct snd_compr_stream *, size_t,
266			struct snd_soc_dai *);
267};
268
269/*
270 * Digital Audio Interface Driver.
271 *
272 * Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97
273 * operations and capabilities. Codec and platform drivers will register this
274 * structure for every DAI they have.
275 *
276 * This structure covers the clocking, formating and ALSA operations for each
277 * interface.
278 */
279struct snd_soc_dai_driver {
280	/* DAI description */
281	const char *name;
282	unsigned int id;
283	unsigned int base;
284	struct snd_soc_dobj dobj;
285
286	/* DAI driver callbacks */
287	int (*probe)(struct snd_soc_dai *dai);
288	int (*remove)(struct snd_soc_dai *dai);
289	int (*suspend)(struct snd_soc_dai *dai);
290	int (*resume)(struct snd_soc_dai *dai);
291	/* compress 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);
296
297	/* ops */
298	const struct snd_soc_dai_ops *ops;
299	const struct snd_soc_cdai_ops *cops;
300
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;
307	unsigned int bus_control:1; /* DAI is also used for the control bus */
308
309	/* probe ordering - for components with runtime dependencies */
310	int probe_order;
311	int remove_order;
312};
313
314/*
315 * Digital Audio Interface runtime data.
316 *
317 * Holds runtime data for a DAI.
318 */
319struct snd_soc_dai {
320	const char *name;
321	int id;
322	struct device *dev;
 
323
324	/* driver ops */
325	struct snd_soc_dai_driver *driver;
326
327	/* DAI runtime info */
328	unsigned int capture_active;		/* stream usage count */
329	unsigned int playback_active;		/* stream usage count */
330	unsigned int probed:1;
331
332	unsigned int active;
333
334	struct snd_soc_dapm_widget *playback_widget;
335	struct snd_soc_dapm_widget *capture_widget;
336
337	/* DAI DMA data */
338	void *playback_dma_data;
339	void *capture_dma_data;
340
341	/* Symmetry data - only valid if symmetry is being enforced */
342	unsigned int rate;
343	unsigned int channels;
344	unsigned int sample_bits;
345
346	/* parent platform/codec */
347	struct snd_soc_component *component;
348
349	/* CODEC TDM slot masks and params (for fixup) */
350	unsigned int tx_mask;
351	unsigned int rx_mask;
352
353	struct list_head list;
 
354};
355
356static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai *dai,
357					     const struct snd_pcm_substream *ss)
358{
359	return (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
360		dai->playback_dma_data : dai->capture_dma_data;
361}
362
363static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai,
364					    const struct snd_pcm_substream *ss,
365					    void *data)
366{
367	if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK)
368		dai->playback_dma_data = data;
369	else
370		dai->capture_dma_data = data;
371}
372
373static inline void snd_soc_dai_init_dma_data(struct snd_soc_dai *dai,
374					     void *playback, void *capture)
375{
376	dai->playback_dma_data = playback;
377	dai->capture_dma_data = capture;
378}
379
380static inline void snd_soc_dai_set_drvdata(struct snd_soc_dai *dai,
381		void *data)
382{
383	dev_set_drvdata(dai->dev, data);
384}
385
386static inline void *snd_soc_dai_get_drvdata(struct snd_soc_dai *dai)
387{
388	return dev_get_drvdata(dai->dev);
389}
390
391/**
392 * snd_soc_dai_set_sdw_stream() - Configures a DAI for SDW stream operation
393 * @dai: DAI
394 * @stream: STREAM
395 * @direction: Stream direction(Playback/Capture)
396 * SoundWire subsystem doesn't have a notion of direction and we reuse
397 * the ASoC stream direction to configure sink/source ports.
398 * Playback maps to source ports and Capture for sink ports.
399 *
400 * This should be invoked with NULL to clear the stream set previously.
401 * Returns 0 on success, a negative error code otherwise.
402 */
403static inline int snd_soc_dai_set_sdw_stream(struct snd_soc_dai *dai,
404				void *stream, int direction)
405{
406	if (dai->driver->ops->set_sdw_stream)
407		return dai->driver->ops->set_sdw_stream(dai, stream, direction);
408	else
409		return -ENOTSUPP;
410}
411
412#endif