[linux-2.6-block.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>

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		1 /* I2S mode */
#define SND_SOC_DAIFMT_RIGHT_J		2 /* Right Justified mode */
#define SND_SOC_DAIFMT_LEFT_J		3 /* Left Justified mode */
#define SND_SOC_DAIFMT_DSP_A		4 /* L data MSB after FRM LRC */
#define SND_SOC_DAIFMT_DSP_B		5 /* L data MSB during FRM LRC */
#define SND_SOC_DAIFMT_AC97		6 /* AC97 */
#define SND_SOC_DAIFMT_PDM		7 /* Pulse density modulation */

/* 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 *);
};

/*
 * 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);
	/* DAI is also used for the control bus */
	bool bus_control;

	/* ops */
	const struct snd_soc_dai_ops *ops;

	/* 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 active;
	unsigned char probed:1;

	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>
	18
	19	struct snd_pcm_substream;
888df395	20	struct snd_soc_dapm_widget;
49681077	21	struct snd_compr_stream;
a47cbe72 MB	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	*/
75d9ac46 MB	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 */
a47cbe72 MB	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	*
47db8e89	44	* DAI bit clocks can be be gated (disabled) when the DAI is not
a47cbe72 MB	45	* sending or receiving PCM data in a frame. This can be used to save power.
a47cbe72 MB	46	*/
75d9ac46	47	#define SND_SOC_DAIFMT_CONT (1 << 4) /* continuous clock */
eef28e10	48	#define SND_SOC_DAIFMT_GATED (0 << 4) /* clock is gated */
a47cbe72	49
a47cbe72	50	/*
1d387a3f	51	* DAI hardware signal polarity.
a47cbe72 MB	52	*
	53	* Specifies whether the DAI can also support inverted clocks for the specified
	54	* format.
1d387a3f AP	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.
a47cbe72	70	*/
5d163336	71	#define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bit clock + frame */
75d9ac46 MB	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 */
a47cbe72 MB	75
	76	/*
	77	* DAI hardware clock masters.
	78	*
	79	* This is wrt the codec, the inverse is true for the interface
47db8e89	80	* i.e. if the codec is clk and FRM master then the interface is
a47cbe72 MB	81	* clk and frame slave.
a47cbe72 MB	82	*/
75d9ac46 MB	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 */
a47cbe72 MB	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
8f738d58 MB	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 \|\
d34c4307 JS	106	SNDRV_PCM_FMTBIT_S32_LE \|\
d34c4307 JS	107	SNDRV_PCM_FMTBIT_S32_BE)
33f503c9	108
f0fba2ad	109	struct snd_soc_dai_driver;
a47cbe72 MB	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,
85488037	121	int pll_id, int source, unsigned int freq_in, unsigned int freq_out);
a47cbe72	122
e54cf76b LG	123	int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio);
e54cf76b LG	124
a47cbe72 MB	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,
a5479e38	129	unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width);
a47cbe72	130
472df3cb BS	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
a47cbe72 MB	135	int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate);
	136
	137	/* Digital Audio Interface mute */
da18396f MB	138	int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute,
da18396f MB	139	int direction);
a47cbe72	140
bece9e95 LG	141	int snd_soc_dai_is_dummy(struct snd_soc_dai *dai);
bece9e95 LG	142
a47cbe72 MB	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);
85488037 MB	150	int (set_pll)(struct snd_soc_dai dai, int pll_id, int source,
85488037 MB	151	unsigned int freq_in, unsigned int freq_out);
a47cbe72	152	int (set_clkdiv)(struct snd_soc_dai dai, int div_id, int div);
e54cf76b	153	int (set_bclk_ratio)(struct snd_soc_dai dai, unsigned int ratio);
a47cbe72 MB	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);
e5c21514	160	int (*xlate_tdm_slot_mask)(unsigned int slots,
89c67857	161	unsigned int tx_mask, unsigned int rx_mask);
a47cbe72	162	int (set_tdm_slot)(struct snd_soc_dai dai,
a5479e38 DR	163	unsigned int tx_mask, unsigned int rx_mask,
a5479e38 DR	164	int slots, int slot_width);
472df3cb BS	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);
a47cbe72 MB	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);
da18396f	175	int (mute_stream)(struct snd_soc_dai dai, int mute, int stream);
dee89c4d MB	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 *);
9f1614aa MP	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	*/
dee89c4d MB	198	int (trigger)(struct snd_pcm_substream , int,
dee89c4d MB	199	struct snd_soc_dai *);
07bf84aa LG	200	int (bespoke_trigger)(struct snd_pcm_substream , int,
07bf84aa LG	201	struct snd_soc_dai *);
258020d0 PU	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 *);
a47cbe72 MB	208	};
	209
	210	/*
f0fba2ad	211	* Digital Audio Interface Driver.
a47cbe72	212	*
f0fba2ad LG	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.
a47cbe72	219	*/
f0fba2ad	220	struct snd_soc_dai_driver {
a47cbe72	221	/* DAI description */
f0fba2ad	222	const char *name;
a47cbe72	223	unsigned int id;
62368294	224	unsigned int base;
68003e6c	225	struct snd_soc_dobj dobj;
a47cbe72	226
f0fba2ad LG	227	/* DAI driver callbacks */
	228	int (probe)(struct snd_soc_dai dai);
	229	int (remove)(struct snd_soc_dai dai);
dc7d7b83 MB	230	int (suspend)(struct snd_soc_dai dai);
dc7d7b83 MB	231	int (resume)(struct snd_soc_dai dai);
49681077	232	/* compress dai */
6f0c4226	233	int (compress_new)(struct snd_soc_pcm_runtime rtd, int num);
bc263214 LPC	234	/* DAI is also used for the control bus */
bc263214 LPC	235	bool bus_control;
a47cbe72 MB	236
a47cbe72 MB	237	/* ops */
1ee46ebd	238	const struct snd_soc_dai_ops *ops;
a47cbe72 MB	239
	240	/* DAI capabilities */
	241	struct snd_soc_pcm_stream capture;
	242	struct snd_soc_pcm_stream playback;
06f409d7	243	unsigned int symmetric_rates:1;
3635bf09 NC	244	unsigned int symmetric_channels:1;
3635bf09 NC	245	unsigned int symmetric_samplebits:1;
0168bf0d LG	246
	247	/* probe ordering - for components with runtime dependencies */
	248	int probe_order;
	249	int remove_order;
f0fba2ad LG	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;
f0fba2ad LG	261
	262	/* driver ops */
	263	struct snd_soc_dai_driver *driver;
a47cbe72 MB	264
a47cbe72 MB	265	/* DAI runtime info */
f0fba2ad LG	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;
3635bf09 NC	269	unsigned int symmetric_channels:1;
3635bf09 NC	270	unsigned int symmetric_samplebits:1;
a47cbe72	271	unsigned int active;
f0fba2ad	272	unsigned char probed:1;
a47cbe72	273
888df395 MB	274	struct snd_soc_dapm_widget *playback_widget;
	275	struct snd_soc_dapm_widget *capture_widget;
	276
f0fba2ad LG	277	/* DAI DMA data */
	278	void *playback_dma_data;
	279	void *capture_dma_data;
a47cbe72	280
17841020 DA	281	/* Symmetry data - only valid if symmetry is being enforced */
17841020 DA	282	unsigned int rate;
3635bf09 NC	283	unsigned int channels;
3635bf09 NC	284	unsigned int sample_bits;
17841020	285
f0fba2ad	286	/* parent platform/codec */
2466ab97	287	struct snd_soc_codec *codec;
6106d129	288	struct snd_soc_component *component;
2466ab97	289
88bd870f BC	290	/* CODEC TDM slot masks and params (for fixup) */
	291	unsigned int tx_mask;
	292	unsigned int rx_mask;
	293
a47cbe72 MB	294	struct list_head list;
	295	};
	296
fd23b7de DM	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) ?
f0fba2ad	301	dai->playback_dma_data : dai->capture_dma_data;
fd23b7de DM	302	}
	303
	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)
f0fba2ad	309	dai->playback_dma_data = data;
fd23b7de	310	else
f0fba2ad LG	311	dai->capture_dma_data = data;
	312	}
	313
ecfc0c04 MB	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
f0fba2ad LG	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
	327	static inline void snd_soc_dai_get_drvdata(struct snd_soc_dai dai)
	328	{
	329	return dev_get_drvdata(dai->dev);
fd23b7de DM	330	}
fd23b7de DM	331
a47cbe72	332	#endif