[linux.git] / include / sound / soc-dai.h

/*
 * linux/sound/soc-dai.h -- ALSA SoC Layer
 *
 * Copyright:	2005-2008 Wolfson Microelectronics. PLC.
 *
 * This program is free software; you can redistribute it and/or modify
 * it under the terms of the GNU General Public License version 2 as
 * published by the Free Software Foundation.
 *
 * Digital Audio Interface (DAI) API.
 */

#ifndef __LINUX_SND_SOC_DAI_H
#define __LINUX_SND_SOC_DAI_H


#include <linux/list.h>
#include <sound/asoc.h>

struct snd_pcm_substream;
struct snd_soc_dapm_widget;
struct snd_compr_stream;

/*
 * DAI hardware audio formats.
 *
 * Describes the physical PCM data formating and clocking. Add new formats
 * to the end.
 */
#define SND_SOC_DAIFMT_I2S		SND_SOC_DAI_FORMAT_I2S
#define SND_SOC_DAIFMT_RIGHT_J		SND_SOC_DAI_FORMAT_RIGHT_J
#define SND_SOC_DAIFMT_LEFT_J		SND_SOC_DAI_FORMAT_LEFT_J
#define SND_SOC_DAIFMT_DSP_A		SND_SOC_DAI_FORMAT_DSP_A
#define SND_SOC_DAIFMT_DSP_B		SND_SOC_DAI_FORMAT_DSP_B
#define SND_SOC_DAIFMT_AC97		SND_SOC_DAI_FORMAT_AC97
#define SND_SOC_DAIFMT_PDM		SND_SOC_DAI_FORMAT_PDM

/* left and right justified also known as MSB and LSB respectively */
#define SND_SOC_DAIFMT_MSB		SND_SOC_DAIFMT_LEFT_J
#define SND_SOC_DAIFMT_LSB		SND_SOC_DAIFMT_RIGHT_J

/*
 * DAI Clock gating.
 *
 * DAI bit clocks can be be gated (disabled) when the DAI is not
 * sending or receiving PCM data in a frame. This can be used to save power.
 */
#define SND_SOC_DAIFMT_CONT		(1 << 4) /* continuous clock */
#define SND_SOC_DAIFMT_GATED		(0 << 4) /* clock is gated */

/*
 * DAI hardware signal polarity.
 *
 * Specifies whether the DAI can also support inverted clocks for the specified
 * format.
 *
 * BCLK:
 * - "normal" polarity means signal is available at rising edge of BCLK
 * - "inverted" polarity means signal is available at falling edge of BCLK
 *
 * FSYNC "normal" polarity depends on the frame format:
 * - I2S: frame consists of left then right channel data. Left channel starts
 *      with falling FSYNC edge, right channel starts with rising FSYNC edge.
 * - Left/Right Justified: frame consists of left then right channel data.
 *      Left channel starts with rising FSYNC edge, right channel starts with
 *      falling FSYNC edge.
 * - DSP A/B: Frame starts with rising FSYNC edge.
 * - AC97: Frame starts with rising FSYNC edge.
 *
 * "Negative" FSYNC polarity is the one opposite of "normal" polarity.
 */
#define SND_SOC_DAIFMT_NB_NF		(0 << 8) /* normal bit clock + frame */
#define SND_SOC_DAIFMT_NB_IF		(2 << 8) /* normal BCLK + inv FRM */
#define SND_SOC_DAIFMT_IB_NF		(3 << 8) /* invert BCLK + nor FRM */
#define SND_SOC_DAIFMT_IB_IF		(4 << 8) /* invert BCLK + FRM */

/*
 * DAI hardware clock masters.
 *
 * This is wrt the codec, the inverse is true for the interface
 * i.e. if the codec is clk and FRM master then the interface is
 * clk and frame slave.
 */
#define SND_SOC_DAIFMT_CBM_CFM		(1 << 12) /* codec clk & FRM master */
#define SND_SOC_DAIFMT_CBS_CFM		(2 << 12) /* codec clk slave & FRM master */
#define SND_SOC_DAIFMT_CBM_CFS		(3 << 12) /* codec clk master & frame slave */
#define SND_SOC_DAIFMT_CBS_CFS		(4 << 12) /* codec clk & FRM slave */

#define SND_SOC_DAIFMT_FORMAT_MASK	0x000f
#define SND_SOC_DAIFMT_CLOCK_MASK	0x00f0
#define SND_SOC_DAIFMT_INV_MASK		0x0f00
#define SND_SOC_DAIFMT_MASTER_MASK	0xf000

/*
 * Master Clock Directions
 */
#define SND_SOC_CLOCK_IN		0
#define SND_SOC_CLOCK_OUT		1

#define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S8 |\
			       SNDRV_PCM_FMTBIT_S16_LE |\
			       SNDRV_PCM_FMTBIT_S16_BE |\
			       SNDRV_PCM_FMTBIT_S20_3LE |\
			       SNDRV_PCM_FMTBIT_S20_3BE |\
			       SNDRV_PCM_FMTBIT_S24_3LE |\
			       SNDRV_PCM_FMTBIT_S24_3BE |\
                               SNDRV_PCM_FMTBIT_S32_LE |\
                               SNDRV_PCM_FMTBIT_S32_BE)

struct snd_soc_dai_driver;
struct snd_soc_dai;
struct snd_ac97_bus_ops;

/* Digital Audio Interface clocking API.*/
int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
	unsigned int freq, int dir);

int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai,
	int div_id, int div);

int snd_soc_dai_set_pll(struct snd_soc_dai *dai,
	int pll_id, int source, unsigned int freq_in, unsigned int freq_out);

int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio);

/* Digital Audio interface formatting */
int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt);

int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
	unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width);

int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai,
	unsigned int tx_num, unsigned int *tx_slot,
	unsigned int rx_num, unsigned int *rx_slot);

int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate);

/* Digital Audio Interface mute */
int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute,
			     int direction);

int snd_soc_dai_is_dummy(struct snd_soc_dai *dai);

struct snd_soc_dai_ops {
	/*
	 * DAI clocking configuration, all optional.
	 * Called by soc_card drivers, normally in their hw_params.
	 */
	int (*set_sysclk)(struct snd_soc_dai *dai,
		int clk_id, unsigned int freq, int dir);
	int (*set_pll)(struct snd_soc_dai *dai, int pll_id, int source,
		unsigned int freq_in, unsigned int freq_out);
	int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div);
	int (*set_bclk_ratio)(struct snd_soc_dai *dai, unsigned int ratio);

	/*
	 * DAI format configuration
	 * Called by soc_card drivers, normally in their hw_params.
	 */
	int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt);
	int (*xlate_tdm_slot_mask)(unsigned int slots,
		unsigned int *tx_mask, unsigned int *rx_mask);
	int (*set_tdm_slot)(struct snd_soc_dai *dai,
		unsigned int tx_mask, unsigned int rx_mask,
		int slots, int slot_width);
	int (*set_channel_map)(struct snd_soc_dai *dai,
		unsigned int tx_num, unsigned int *tx_slot,
		unsigned int rx_num, unsigned int *rx_slot);
	int (*set_tristate)(struct snd_soc_dai *dai, int tristate);

	/*
	 * DAI digital mute - optional.
	 * Called by soc-core to minimise any pops.
	 */
	int (*digital_mute)(struct snd_soc_dai *dai, int mute);
	int (*mute_stream)(struct snd_soc_dai *dai, int mute, int stream);

	/*
	 * ALSA PCM audio operations - all optional.
	 * Called by soc-core during audio PCM operations.
	 */
	int (*startup)(struct snd_pcm_substream *,
		struct snd_soc_dai *);
	void (*shutdown)(struct snd_pcm_substream *,
		struct snd_soc_dai *);
	int (*hw_params)(struct snd_pcm_substream *,
		struct snd_pcm_hw_params *, struct snd_soc_dai *);
	int (*hw_free)(struct snd_pcm_substream *,
		struct snd_soc_dai *);
	int (*prepare)(struct snd_pcm_substream *,
		struct snd_soc_dai *);
	/*
	 * NOTE: Commands passed to the trigger function are not necessarily
	 * compatible with the current state of the dai. For example this
	 * sequence of commands is possible: START STOP STOP.
	 * So do not unconditionally use refcounting functions in the trigger
	 * function, e.g. clk_enable/disable.
	 */
	int (*trigger)(struct snd_pcm_substream *, int,
		struct snd_soc_dai *);
	int (*bespoke_trigger)(struct snd_pcm_substream *, int,
		struct snd_soc_dai *);
	/*
	 * For hardware based FIFO caused delay reporting.
	 * Optional.
	 */
	snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *,
		struct snd_soc_dai *);
};

struct snd_soc_cdai_ops {
	/*
	 * for compress ops
	 */
	int (*startup)(struct snd_compr_stream *,
			struct snd_soc_dai *);
	int (*shutdown)(struct snd_compr_stream *,
			struct snd_soc_dai *);
	int (*set_params)(struct snd_compr_stream *,
			struct snd_compr_params *, struct snd_soc_dai *);
	int (*get_params)(struct snd_compr_stream *,
			struct snd_codec *, struct snd_soc_dai *);
	int (*set_metadata)(struct snd_compr_stream *,
			struct snd_compr_metadata *, struct snd_soc_dai *);
	int (*get_metadata)(struct snd_compr_stream *,
			struct snd_compr_metadata *, struct snd_soc_dai *);
	int (*trigger)(struct snd_compr_stream *, int,
			struct snd_soc_dai *);
	int (*pointer)(struct snd_compr_stream *,
			struct snd_compr_tstamp *, struct snd_soc_dai *);
	int (*ack)(struct snd_compr_stream *, size_t,
			struct snd_soc_dai *);
};

/*
 * Digital Audio Interface Driver.
 *
 * Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97
 * operations and capabilities. Codec and platform drivers will register this
 * structure for every DAI they have.
 *
 * This structure covers the clocking, formating and ALSA operations for each
 * interface.
 */
struct snd_soc_dai_driver {
	/* DAI description */
	const char *name;
	unsigned int id;
	unsigned int base;
	struct snd_soc_dobj dobj;

	/* DAI driver callbacks */
	int (*probe)(struct snd_soc_dai *dai);
	int (*remove)(struct snd_soc_dai *dai);
	int (*suspend)(struct snd_soc_dai *dai);
	int (*resume)(struct snd_soc_dai *dai);
	/* compress dai */
	int (*compress_new)(struct snd_soc_pcm_runtime *rtd, int num);
	/* Optional Callback used at pcm creation*/
	int (*pcm_new)(struct snd_soc_pcm_runtime *rtd,
		       struct snd_soc_dai *dai);
	/* DAI is also used for the control bus */
	bool bus_control;

	/* ops */
	const struct snd_soc_dai_ops *ops;
	const struct snd_soc_cdai_ops *cops;

	/* DAI capabilities */
	struct snd_soc_pcm_stream capture;
	struct snd_soc_pcm_stream playback;
	unsigned int symmetric_rates:1;
	unsigned int symmetric_channels:1;
	unsigned int symmetric_samplebits:1;

	/* probe ordering - for components with runtime dependencies */
	int probe_order;
	int remove_order;
};

/*
 * Digital Audio Interface runtime data.
 *
 * Holds runtime data for a DAI.
 */
struct snd_soc_dai {
	const char *name;
	int id;
	struct device *dev;

	/* driver ops */
	struct snd_soc_dai_driver *driver;

	/* DAI runtime info */
	unsigned int capture_active:1;		/* stream is in use */
	unsigned int playback_active:1;		/* stream is in use */
	unsigned int symmetric_rates:1;
	unsigned int symmetric_channels:1;
	unsigned int symmetric_samplebits:1;
	unsigned int probed:1;

	unsigned int active;

	struct snd_soc_dapm_widget *playback_widget;
	struct snd_soc_dapm_widget *capture_widget;

	/* DAI DMA data */
	void *playback_dma_data;
	void *capture_dma_data;

	/* Symmetry data - only valid if symmetry is being enforced */
	unsigned int rate;
	unsigned int channels;
	unsigned int sample_bits;

	/* parent platform/codec */
	struct snd_soc_codec *codec;
	struct snd_soc_component *component;

	/* CODEC TDM slot masks and params (for fixup) */
	unsigned int tx_mask;
	unsigned int rx_mask;

	struct list_head list;
};

static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai *dai,
					     const struct snd_pcm_substream *ss)
{
	return (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
		dai->playback_dma_data : dai->capture_dma_data;
}

static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai,
					    const struct snd_pcm_substream *ss,
					    void *data)
{
	if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK)
		dai->playback_dma_data = data;
	else
		dai->capture_dma_data = data;
}

static inline void snd_soc_dai_init_dma_data(struct snd_soc_dai *dai,
					     void *playback, void *capture)
{
	dai->playback_dma_data = playback;
	dai->capture_dma_data = capture;
}

static inline void snd_soc_dai_set_drvdata(struct snd_soc_dai *dai,
		void *data)
{
	dev_set_drvdata(dai->dev, data);
}

static inline void *snd_soc_dai_get_drvdata(struct snd_soc_dai *dai)
{
	return dev_get_drvdata(dai->dev);
}

#endif
Commit	Line	Data
a47cbe72 MB	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>
d2e3cb85	18	#include <sound/asoc.h>
a47cbe72 MB	19
a47cbe72 MB	20	struct snd_pcm_substream;
888df395	21	struct snd_soc_dapm_widget;
49681077	22	struct snd_compr_stream;
a47cbe72 MB	23
	24	/*
	25	* DAI hardware audio formats.
	26	*
	27	* Describes the physical PCM data formating and clocking. Add new formats
	28	* to the end.
	29	*/
d2e3cb85 ML	30	#define SND_SOC_DAIFMT_I2S SND_SOC_DAI_FORMAT_I2S
	31	#define SND_SOC_DAIFMT_RIGHT_J SND_SOC_DAI_FORMAT_RIGHT_J
	32	#define SND_SOC_DAIFMT_LEFT_J SND_SOC_DAI_FORMAT_LEFT_J
	33	#define SND_SOC_DAIFMT_DSP_A SND_SOC_DAI_FORMAT_DSP_A
	34	#define SND_SOC_DAIFMT_DSP_B SND_SOC_DAI_FORMAT_DSP_B
	35	#define SND_SOC_DAIFMT_AC97 SND_SOC_DAI_FORMAT_AC97
	36	#define SND_SOC_DAIFMT_PDM SND_SOC_DAI_FORMAT_PDM
a47cbe72 MB	37
	38	/* left and right justified also known as MSB and LSB respectively */
	39	#define SND_SOC_DAIFMT_MSB SND_SOC_DAIFMT_LEFT_J
	40	#define SND_SOC_DAIFMT_LSB SND_SOC_DAIFMT_RIGHT_J
	41
	42	/*
	43	* DAI Clock gating.
	44	*
47db8e89	45	* DAI bit clocks can be be gated (disabled) when the DAI is not
a47cbe72 MB	46	* sending or receiving PCM data in a frame. This can be used to save power.
a47cbe72 MB	47	*/
75d9ac46	48	#define SND_SOC_DAIFMT_CONT (1 << 4) /* continuous clock */
eef28e10	49	#define SND_SOC_DAIFMT_GATED (0 << 4) /* clock is gated */
a47cbe72	50
a47cbe72	51	/*
1d387a3f	52	* DAI hardware signal polarity.
a47cbe72 MB	53	*
	54	* Specifies whether the DAI can also support inverted clocks for the specified
	55	* format.
1d387a3f AP	56	*
	57	* BCLK:
	58	* - "normal" polarity means signal is available at rising edge of BCLK
	59	* - "inverted" polarity means signal is available at falling edge of BCLK
	60	*
	61	* FSYNC "normal" polarity depends on the frame format:
	62	* - I2S: frame consists of left then right channel data. Left channel starts
	63	* with falling FSYNC edge, right channel starts with rising FSYNC edge.
	64	* - Left/Right Justified: frame consists of left then right channel data.
	65	* Left channel starts with rising FSYNC edge, right channel starts with
	66	* falling FSYNC edge.
	67	* - DSP A/B: Frame starts with rising FSYNC edge.
	68	* - AC97: Frame starts with rising FSYNC edge.
	69	*
	70	* "Negative" FSYNC polarity is the one opposite of "normal" polarity.
a47cbe72	71	*/
5d163336	72	#define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bit clock + frame */
75d9ac46 MB	73	#define SND_SOC_DAIFMT_NB_IF (2 << 8) /* normal BCLK + inv FRM */
	74	#define SND_SOC_DAIFMT_IB_NF (3 << 8) /* invert BCLK + nor FRM */
	75	#define SND_SOC_DAIFMT_IB_IF (4 << 8) /* invert BCLK + FRM */
a47cbe72 MB	76
	77	/*
	78	* DAI hardware clock masters.
	79	*
	80	* This is wrt the codec, the inverse is true for the interface
47db8e89	81	* i.e. if the codec is clk and FRM master then the interface is
a47cbe72 MB	82	* clk and frame slave.
a47cbe72 MB	83	*/
75d9ac46 MB	84	#define SND_SOC_DAIFMT_CBM_CFM (1 << 12) /* codec clk & FRM master */
	85	#define SND_SOC_DAIFMT_CBS_CFM (2 << 12) /* codec clk slave & FRM master */
	86	#define SND_SOC_DAIFMT_CBM_CFS (3 << 12) /* codec clk master & frame slave */
	87	#define SND_SOC_DAIFMT_CBS_CFS (4 << 12) /* codec clk & FRM slave */
a47cbe72 MB	88
	89	#define SND_SOC_DAIFMT_FORMAT_MASK 0x000f
	90	#define SND_SOC_DAIFMT_CLOCK_MASK 0x00f0
	91	#define SND_SOC_DAIFMT_INV_MASK 0x0f00
	92	#define SND_SOC_DAIFMT_MASTER_MASK 0xf000
	93
	94	/*
	95	* Master Clock Directions
	96	*/
	97	#define SND_SOC_CLOCK_IN 0
	98	#define SND_SOC_CLOCK_OUT 1
	99
8f738d58 MB	100	#define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S8 \|\
	101	SNDRV_PCM_FMTBIT_S16_LE \|\
	102	SNDRV_PCM_FMTBIT_S16_BE \|\
	103	SNDRV_PCM_FMTBIT_S20_3LE \|\
	104	SNDRV_PCM_FMTBIT_S20_3BE \|\
	105	SNDRV_PCM_FMTBIT_S24_3LE \|\
	106	SNDRV_PCM_FMTBIT_S24_3BE \|\
d34c4307 JS	107	SNDRV_PCM_FMTBIT_S32_LE \|\
d34c4307 JS	108	SNDRV_PCM_FMTBIT_S32_BE)
33f503c9	109
f0fba2ad	110	struct snd_soc_dai_driver;
a47cbe72 MB	111	struct snd_soc_dai;
	112	struct snd_ac97_bus_ops;
	113
	114	/* Digital Audio Interface clocking API.*/
	115	int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
	116	unsigned int freq, int dir);
	117
	118	int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai,
	119	int div_id, int div);
	120
	121	int snd_soc_dai_set_pll(struct snd_soc_dai *dai,
85488037	122	int pll_id, int source, unsigned int freq_in, unsigned int freq_out);
a47cbe72	123
e54cf76b LG	124	int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio);
e54cf76b LG	125
a47cbe72 MB	126	/* Digital Audio interface formatting */
	127	int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt);
	128
	129	int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
a5479e38	130	unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width);
a47cbe72	131
472df3cb BS	132	int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai,
	133	unsigned int tx_num, unsigned int *tx_slot,
	134	unsigned int rx_num, unsigned int *rx_slot);
	135
a47cbe72 MB	136	int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate);
	137
	138	/* Digital Audio Interface mute */
da18396f MB	139	int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute,
da18396f MB	140	int direction);
a47cbe72	141
bece9e95 LG	142	int snd_soc_dai_is_dummy(struct snd_soc_dai *dai);
bece9e95 LG	143
a47cbe72 MB	144	struct snd_soc_dai_ops {
	145	/*
	146	* DAI clocking configuration, all optional.
	147	* Called by soc_card drivers, normally in their hw_params.
	148	*/
	149	int (set_sysclk)(struct snd_soc_dai dai,
	150	int clk_id, unsigned int freq, int dir);
85488037 MB	151	int (set_pll)(struct snd_soc_dai dai, int pll_id, int source,
85488037 MB	152	unsigned int freq_in, unsigned int freq_out);
a47cbe72	153	int (set_clkdiv)(struct snd_soc_dai dai, int div_id, int div);
e54cf76b	154	int (set_bclk_ratio)(struct snd_soc_dai dai, unsigned int ratio);
a47cbe72 MB	155
	156	/*
	157	* DAI format configuration
	158	* Called by soc_card drivers, normally in their hw_params.
	159	*/
	160	int (set_fmt)(struct snd_soc_dai dai, unsigned int fmt);
e5c21514	161	int (*xlate_tdm_slot_mask)(unsigned int slots,
89c67857	162	unsigned int tx_mask, unsigned int rx_mask);
a47cbe72	163	int (set_tdm_slot)(struct snd_soc_dai dai,
a5479e38 DR	164	unsigned int tx_mask, unsigned int rx_mask,
a5479e38 DR	165	int slots, int slot_width);
472df3cb BS	166	int (set_channel_map)(struct snd_soc_dai dai,
	167	unsigned int tx_num, unsigned int *tx_slot,
	168	unsigned int rx_num, unsigned int *rx_slot);
a47cbe72 MB	169	int (set_tristate)(struct snd_soc_dai dai, int tristate);
	170
	171	/*
	172	* DAI digital mute - optional.
	173	* Called by soc-core to minimise any pops.
	174	*/
	175	int (digital_mute)(struct snd_soc_dai dai, int mute);
da18396f	176	int (mute_stream)(struct snd_soc_dai dai, int mute, int stream);
dee89c4d MB	177
	178	/*
	179	* ALSA PCM audio operations - all optional.
	180	* Called by soc-core during audio PCM operations.
	181	*/
	182	int (startup)(struct snd_pcm_substream ,
	183	struct snd_soc_dai *);
	184	void (shutdown)(struct snd_pcm_substream ,
	185	struct snd_soc_dai *);
	186	int (hw_params)(struct snd_pcm_substream ,
	187	struct snd_pcm_hw_params , struct snd_soc_dai );
	188	int (hw_free)(struct snd_pcm_substream ,
	189	struct snd_soc_dai *);
	190	int (prepare)(struct snd_pcm_substream ,
	191	struct snd_soc_dai *);
9f1614aa MSP	192	/*
	193	* NOTE: Commands passed to the trigger function are not necessarily
	194	* compatible with the current state of the dai. For example this
	195	* sequence of commands is possible: START STOP STOP.
	196	* So do not unconditionally use refcounting functions in the trigger
	197	* function, e.g. clk_enable/disable.
	198	*/
dee89c4d MB	199	int (trigger)(struct snd_pcm_substream , int,
dee89c4d MB	200	struct snd_soc_dai *);
07bf84aa LG	201	int (bespoke_trigger)(struct snd_pcm_substream , int,
07bf84aa LG	202	struct snd_soc_dai *);
258020d0 PU	203	/*
	204	* For hardware based FIFO caused delay reporting.
	205	* Optional.
	206	*/
	207	snd_pcm_sframes_t (delay)(struct snd_pcm_substream ,
	208	struct snd_soc_dai *);
a47cbe72 MB	209	};
a47cbe72 MB	210
2e622ae4 VK	211	struct snd_soc_cdai_ops {
	212	/*
	213	* for compress ops
	214	*/
	215	int (startup)(struct snd_compr_stream ,
	216	struct snd_soc_dai *);
	217	int (shutdown)(struct snd_compr_stream ,
	218	struct snd_soc_dai *);
	219	int (set_params)(struct snd_compr_stream ,
	220	struct snd_compr_params , struct snd_soc_dai );
	221	int (get_params)(struct snd_compr_stream ,
	222	struct snd_codec , struct snd_soc_dai );
	223	int (set_metadata)(struct snd_compr_stream ,
	224	struct snd_compr_metadata , struct snd_soc_dai );
	225	int (get_metadata)(struct snd_compr_stream ,
	226	struct snd_compr_metadata , struct snd_soc_dai );
	227	int (trigger)(struct snd_compr_stream , int,
	228	struct snd_soc_dai *);
	229	int (pointer)(struct snd_compr_stream ,
	230	struct snd_compr_tstamp , struct snd_soc_dai );
	231	int (ack)(struct snd_compr_stream , size_t,
	232	struct snd_soc_dai *);
	233	};
	234
a47cbe72	235	/*
f0fba2ad	236	* Digital Audio Interface Driver.
a47cbe72	237	*
f0fba2ad LG	238	* Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97
	239	* operations and capabilities. Codec and platform drivers will register this
	240	* structure for every DAI they have.
	241	*
	242	* This structure covers the clocking, formating and ALSA operations for each
	243	* interface.
a47cbe72	244	*/
f0fba2ad	245	struct snd_soc_dai_driver {
a47cbe72	246	/* DAI description */
f0fba2ad	247	const char *name;
a47cbe72	248	unsigned int id;
62368294	249	unsigned int base;
68003e6c	250	struct snd_soc_dobj dobj;
a47cbe72	251
f0fba2ad LG	252	/* DAI driver callbacks */
	253	int (probe)(struct snd_soc_dai dai);
	254	int (remove)(struct snd_soc_dai dai);
dc7d7b83 MB	255	int (suspend)(struct snd_soc_dai dai);
dc7d7b83 MB	256	int (resume)(struct snd_soc_dai dai);
49681077	257	/* compress dai */
6f0c4226	258	int (compress_new)(struct snd_soc_pcm_runtime rtd, int num);
25f7b701 AP	259	/* Optional Callback used at pcm creation*/
	260	int (pcm_new)(struct snd_soc_pcm_runtime rtd,
	261	struct snd_soc_dai *dai);
bc263214 LPC	262	/* DAI is also used for the control bus */
bc263214 LPC	263	bool bus_control;
a47cbe72 MB	264
a47cbe72 MB	265	/* ops */
1ee46ebd	266	const struct snd_soc_dai_ops *ops;
2e622ae4	267	const struct snd_soc_cdai_ops *cops;
a47cbe72 MB	268
	269	/* DAI capabilities */
	270	struct snd_soc_pcm_stream capture;
	271	struct snd_soc_pcm_stream playback;
06f409d7	272	unsigned int symmetric_rates:1;
3635bf09 NC	273	unsigned int symmetric_channels:1;
3635bf09 NC	274	unsigned int symmetric_samplebits:1;
0168bf0d LG	275
	276	/* probe ordering - for components with runtime dependencies */
	277	int probe_order;
	278	int remove_order;
f0fba2ad LG	279	};
	280
	281	/*
	282	* Digital Audio Interface runtime data.
	283	*
	284	* Holds runtime data for a DAI.
	285	*/
	286	struct snd_soc_dai {
	287	const char *name;
	288	int id;
	289	struct device *dev;
f0fba2ad LG	290
	291	/* driver ops */
	292	struct snd_soc_dai_driver *driver;
a47cbe72 MB	293
a47cbe72 MB	294	/* DAI runtime info */
f0fba2ad LG	295	unsigned int capture_active:1; /* stream is in use */
	296	unsigned int playback_active:1; /* stream is in use */
	297	unsigned int symmetric_rates:1;
3635bf09 NC	298	unsigned int symmetric_channels:1;
3635bf09 NC	299	unsigned int symmetric_samplebits:1;
39b2238e KM	300	unsigned int probed:1;
39b2238e KM	301
a47cbe72	302	unsigned int active;
a47cbe72	303
888df395 MB	304	struct snd_soc_dapm_widget *playback_widget;
	305	struct snd_soc_dapm_widget *capture_widget;
	306
f0fba2ad LG	307	/* DAI DMA data */
	308	void *playback_dma_data;
	309	void *capture_dma_data;
a47cbe72	310
17841020 DA	311	/* Symmetry data - only valid if symmetry is being enforced */
17841020 DA	312	unsigned int rate;
3635bf09 NC	313	unsigned int channels;
3635bf09 NC	314	unsigned int sample_bits;
17841020	315
f0fba2ad	316	/* parent platform/codec */
2466ab97	317	struct snd_soc_codec *codec;
6106d129	318	struct snd_soc_component *component;
2466ab97	319
88bd870f BC	320	/* CODEC TDM slot masks and params (for fixup) */
	321	unsigned int tx_mask;
	322	unsigned int rx_mask;
	323
a47cbe72 MB	324	struct list_head list;
	325	};
	326
fd23b7de DM	327	static inline void snd_soc_dai_get_dma_data(const struct snd_soc_dai dai,
	328	const struct snd_pcm_substream *ss)
	329	{
	330	return (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
f0fba2ad	331	dai->playback_dma_data : dai->capture_dma_data;
fd23b7de DM	332	}
	333
	334	static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai,
	335	const struct snd_pcm_substream *ss,
	336	void *data)
	337	{
	338	if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK)
f0fba2ad	339	dai->playback_dma_data = data;
fd23b7de	340	else
f0fba2ad LG	341	dai->capture_dma_data = data;
	342	}
	343
ecfc0c04 MB	344	static inline void snd_soc_dai_init_dma_data(struct snd_soc_dai *dai,
	345	void playback, void capture)
	346	{
	347	dai->playback_dma_data = playback;
	348	dai->capture_dma_data = capture;
	349	}
	350
f0fba2ad LG	351	static inline void snd_soc_dai_set_drvdata(struct snd_soc_dai *dai,
	352	void *data)
	353	{
	354	dev_set_drvdata(dai->dev, data);
	355	}
	356
	357	static inline void snd_soc_dai_get_drvdata(struct snd_soc_dai dai)
	358	{
	359	return dev_get_drvdata(dai->dev);
fd23b7de DM	360	}
fd23b7de DM	361
a47cbe72	362	#endif