[mirror_ubuntu-zesty-kernel.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 inversions.
 *
 * Specifies whether the DAI can also support inverted clocks for the specified
 * format.
 */
#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 (*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;
	int ac97_control;
	unsigned int base;

	/* 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 */
	bool compress_dai;

	/* 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;
	void *ac97_pdata;	/* platform_data for the ac97 codec */

	/* 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;
	struct snd_pcm_runtime *runtime;
	unsigned int active;
	unsigned char probed:1;

	struct snd_soc_dapm_widget *playback_widget;
	struct snd_soc_dapm_widget *capture_widget;
	struct snd_soc_dapm_context dapm;

	/* 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_platform *platform;
	struct snd_soc_codec *codec;

	struct snd_soc_card *card;

	struct list_head list;
	struct list_head card_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 MB	50	/*
	51	* DAI hardware signal inversions.
	52	*
	53	* Specifies whether the DAI can also support inverted clocks for the specified
	54	* format.
	55	*/
5d163336	56	#define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bit clock + frame */
75d9ac46 MB	57	#define SND_SOC_DAIFMT_NB_IF (2 << 8) /* normal BCLK + inv FRM */
	58	#define SND_SOC_DAIFMT_IB_NF (3 << 8) /* invert BCLK + nor FRM */
	59	#define SND_SOC_DAIFMT_IB_IF (4 << 8) /* invert BCLK + FRM */
a47cbe72 MB	60
	61	/*
	62	* DAI hardware clock masters.
	63	*
	64	* This is wrt the codec, the inverse is true for the interface
47db8e89	65	* i.e. if the codec is clk and FRM master then the interface is
a47cbe72 MB	66	* clk and frame slave.
a47cbe72 MB	67	*/
75d9ac46 MB	68	#define SND_SOC_DAIFMT_CBM_CFM (1 << 12) /* codec clk & FRM master */
	69	#define SND_SOC_DAIFMT_CBS_CFM (2 << 12) /* codec clk slave & FRM master */
	70	#define SND_SOC_DAIFMT_CBM_CFS (3 << 12) /* codec clk master & frame slave */
	71	#define SND_SOC_DAIFMT_CBS_CFS (4 << 12) /* codec clk & FRM slave */
a47cbe72 MB	72
	73	#define SND_SOC_DAIFMT_FORMAT_MASK 0x000f
	74	#define SND_SOC_DAIFMT_CLOCK_MASK 0x00f0
	75	#define SND_SOC_DAIFMT_INV_MASK 0x0f00
	76	#define SND_SOC_DAIFMT_MASTER_MASK 0xf000
	77
	78	/*
	79	* Master Clock Directions
	80	*/
	81	#define SND_SOC_CLOCK_IN 0
	82	#define SND_SOC_CLOCK_OUT 1
	83
8f738d58 MB	84	#define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S8 \|\
	85	SNDRV_PCM_FMTBIT_S16_LE \|\
	86	SNDRV_PCM_FMTBIT_S16_BE \|\
	87	SNDRV_PCM_FMTBIT_S20_3LE \|\
	88	SNDRV_PCM_FMTBIT_S20_3BE \|\
	89	SNDRV_PCM_FMTBIT_S24_3LE \|\
	90	SNDRV_PCM_FMTBIT_S24_3BE \|\
d34c4307 JS	91	SNDRV_PCM_FMTBIT_S32_LE \|\
d34c4307 JS	92	SNDRV_PCM_FMTBIT_S32_BE)
33f503c9	93
f0fba2ad	94	struct snd_soc_dai_driver;
a47cbe72 MB	95	struct snd_soc_dai;
	96	struct snd_ac97_bus_ops;
	97
	98	/* Digital Audio Interface clocking API.*/
	99	int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
	100	unsigned int freq, int dir);
	101
	102	int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai,
	103	int div_id, int div);
	104
	105	int snd_soc_dai_set_pll(struct snd_soc_dai *dai,
85488037	106	int pll_id, int source, unsigned int freq_in, unsigned int freq_out);
a47cbe72	107
e54cf76b LG	108	int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio);
e54cf76b LG	109
a47cbe72 MB	110	/* Digital Audio interface formatting */
	111	int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt);
	112
	113	int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
a5479e38	114	unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width);
a47cbe72	115
472df3cb BS	116	int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai,
	117	unsigned int tx_num, unsigned int *tx_slot,
	118	unsigned int rx_num, unsigned int *rx_slot);
	119
a47cbe72 MB	120	int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate);
	121
	122	/* Digital Audio Interface mute */
da18396f MB	123	int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute,
da18396f MB	124	int direction);
a47cbe72	125
bece9e95 LG	126	int snd_soc_dai_is_dummy(struct snd_soc_dai *dai);
bece9e95 LG	127
a47cbe72 MB	128	struct snd_soc_dai_ops {
	129	/*
	130	* DAI clocking configuration, all optional.
	131	* Called by soc_card drivers, normally in their hw_params.
	132	*/
	133	int (set_sysclk)(struct snd_soc_dai dai,
	134	int clk_id, unsigned int freq, int dir);
85488037 MB	135	int (set_pll)(struct snd_soc_dai dai, int pll_id, int source,
85488037 MB	136	unsigned int freq_in, unsigned int freq_out);
a47cbe72	137	int (set_clkdiv)(struct snd_soc_dai dai, int div_id, int div);
e54cf76b	138	int (set_bclk_ratio)(struct snd_soc_dai dai, unsigned int ratio);
a47cbe72 MB	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,
a5479e38 DR	146	unsigned int tx_mask, unsigned int rx_mask,
a5479e38 DR	147	int slots, int slot_width);
472df3cb BS	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);
a47cbe72 MB	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);
da18396f	158	int (mute_stream)(struct snd_soc_dai dai, int mute, int stream);
dee89c4d MB	159
	160	/*
	161	* ALSA PCM audio operations - all optional.
	162	* Called by soc-core during audio PCM operations.
	163	*/
	164	int (startup)(struct snd_pcm_substream ,
	165	struct snd_soc_dai *);
	166	void (shutdown)(struct snd_pcm_substream ,
	167	struct snd_soc_dai *);
	168	int (hw_params)(struct snd_pcm_substream ,
	169	struct snd_pcm_hw_params , struct snd_soc_dai );
	170	int (hw_free)(struct snd_pcm_substream ,
	171	struct snd_soc_dai *);
	172	int (prepare)(struct snd_pcm_substream ,
	173	struct snd_soc_dai *);
9f1614aa MP	174	/*
	175	* NOTE: Commands passed to the trigger function are not necessarily
	176	* compatible with the current state of the dai. For example this
	177	* sequence of commands is possible: START STOP STOP.
	178	* So do not unconditionally use refcounting functions in the trigger
	179	* function, e.g. clk_enable/disable.
	180	*/
dee89c4d MB	181	int (trigger)(struct snd_pcm_substream , int,
dee89c4d MB	182	struct snd_soc_dai *);
07bf84aa LG	183	int (bespoke_trigger)(struct snd_pcm_substream , int,
07bf84aa LG	184	struct snd_soc_dai *);
258020d0 PU	185	/*
	186	* For hardware based FIFO caused delay reporting.
	187	* Optional.
	188	*/
	189	snd_pcm_sframes_t (delay)(struct snd_pcm_substream ,
	190	struct snd_soc_dai *);
a47cbe72 MB	191	};
	192
	193	/*
f0fba2ad	194	* Digital Audio Interface Driver.
a47cbe72	195	*
f0fba2ad LG	196	* Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97
	197	* operations and capabilities. Codec and platform drivers will register this
	198	* structure for every DAI they have.
	199	*
	200	* This structure covers the clocking, formating and ALSA operations for each
	201	* interface.
a47cbe72	202	*/
f0fba2ad	203	struct snd_soc_dai_driver {
a47cbe72	204	/* DAI description */
f0fba2ad	205	const char *name;
a47cbe72	206	unsigned int id;
3ba9e10a	207	int ac97_control;
62368294	208	unsigned int base;
a47cbe72	209
f0fba2ad LG	210	/* DAI driver callbacks */
	211	int (probe)(struct snd_soc_dai dai);
	212	int (remove)(struct snd_soc_dai dai);
dc7d7b83 MB	213	int (suspend)(struct snd_soc_dai dai);
dc7d7b83 MB	214	int (resume)(struct snd_soc_dai dai);
49681077 VK	215	/* compress dai */
49681077 VK	216	bool compress_dai;
a47cbe72 MB	217
a47cbe72 MB	218	/* ops */
1ee46ebd	219	const struct snd_soc_dai_ops *ops;
a47cbe72 MB	220
	221	/* DAI capabilities */
	222	struct snd_soc_pcm_stream capture;
	223	struct snd_soc_pcm_stream playback;
06f409d7	224	unsigned int symmetric_rates:1;
3635bf09 NC	225	unsigned int symmetric_channels:1;
3635bf09 NC	226	unsigned int symmetric_samplebits:1;
0168bf0d LG	227
	228	/* probe ordering - for components with runtime dependencies */
	229	int probe_order;
	230	int remove_order;
f0fba2ad LG	231	};
	232
	233	/*
	234	* Digital Audio Interface runtime data.
	235	*
	236	* Holds runtime data for a DAI.
	237	*/
	238	struct snd_soc_dai {
	239	const char *name;
	240	int id;
	241	struct device *dev;
	242	void ac97_pdata; / platform_data for the ac97 codec */
	243
	244	/* driver ops */
	245	struct snd_soc_dai_driver *driver;
a47cbe72 MB	246
a47cbe72 MB	247	/* DAI runtime info */
f0fba2ad LG	248	unsigned int capture_active:1; /* stream is in use */
	249	unsigned int playback_active:1; /* stream is in use */
	250	unsigned int symmetric_rates:1;
3635bf09 NC	251	unsigned int symmetric_channels:1;
3635bf09 NC	252	unsigned int symmetric_samplebits:1;
f0fba2ad	253	struct snd_pcm_runtime *runtime;
a47cbe72	254	unsigned int active;
f0fba2ad	255	unsigned char probed:1;
a47cbe72	256
888df395 MB	257	struct snd_soc_dapm_widget *playback_widget;
888df395 MB	258	struct snd_soc_dapm_widget *capture_widget;
be09ad90	259	struct snd_soc_dapm_context dapm;
888df395	260
f0fba2ad LG	261	/* DAI DMA data */
	262	void *playback_dma_data;
	263	void *capture_dma_data;
a47cbe72	264
17841020 DA	265	/* Symmetry data - only valid if symmetry is being enforced */
17841020 DA	266	unsigned int rate;
3635bf09 NC	267	unsigned int channels;
3635bf09 NC	268	unsigned int sample_bits;
17841020	269
f0fba2ad	270	/* parent platform/codec */
2466ab97 MB	271	struct snd_soc_platform *platform;
	272	struct snd_soc_codec *codec;
	273
f0fba2ad	274	struct snd_soc_card *card;
a47cbe72 MB	275
a47cbe72 MB	276	struct list_head list;
f0fba2ad	277	struct list_head card_list;
a47cbe72 MB	278	};
a47cbe72 MB	279
fd23b7de DM	280	static inline void snd_soc_dai_get_dma_data(const struct snd_soc_dai dai,
	281	const struct snd_pcm_substream *ss)
	282	{
	283	return (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
f0fba2ad	284	dai->playback_dma_data : dai->capture_dma_data;
fd23b7de DM	285	}
	286
	287	static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai,
	288	const struct snd_pcm_substream *ss,
	289	void *data)
	290	{
	291	if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK)
f0fba2ad	292	dai->playback_dma_data = data;
fd23b7de	293	else
f0fba2ad LG	294	dai->capture_dma_data = data;
	295	}
	296
ecfc0c04 MB	297	static inline void snd_soc_dai_init_dma_data(struct snd_soc_dai *dai,
	298	void playback, void capture)
	299	{
	300	dai->playback_dma_data = playback;
	301	dai->capture_dma_data = capture;
	302	}
	303
f0fba2ad LG	304	static inline void snd_soc_dai_set_drvdata(struct snd_soc_dai *dai,
	305	void *data)
	306	{
	307	dev_set_drvdata(dai->dev, data);
	308	}
	309
	310	static inline void snd_soc_dai_get_drvdata(struct snd_soc_dai dai)
	311	{
	312	return dev_get_drvdata(dai->dev);
fd23b7de DM	313	}
fd23b7de DM	314
a47cbe72	315	#endif