[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;

/*
 * DAI hardware audio formats.
 *
 * Describes the physical PCM data formating and clocking. Add new formats
 * to the end.
 */
#define SND_SOC_DAIFMT_I2S		0 /* I2S mode */
#define SND_SOC_DAIFMT_RIGHT_J		1 /* Right Justified mode */
#define SND_SOC_DAIFMT_LEFT_J		2 /* Left Justified mode */
#define SND_SOC_DAIFMT_DSP_A		3 /* L data msb after FRM LRC */
#define SND_SOC_DAIFMT_DSP_B		4 /* L data msb during FRM LRC */
#define SND_SOC_DAIFMT_AC97		5 /* AC97 */

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

/*
 * DAI Left/Right Clocks.
 *
 * Specifies whether the DAI can support different samples for similtanious
 * playback and capture. This usually requires a seperate physical frame
 * clock for playback and capture.
 */
#define SND_SOC_DAIFMT_SYNC		(0 << 5) /* Tx FRM = Rx FRM */
#define SND_SOC_DAIFMT_ASYNC		(1 << 5) /* Tx FRM ~ Rx FRM */

/*
 * TDM
 *
 * Time Division Multiplexing. Allows PCM data to be multplexed with other
 * data on the DAI.
 */
#define SND_SOC_DAIFMT_TDM		(1 << 6)

/*
 * 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		(1 << 8) /* normal bclk + inv frm */
#define SND_SOC_DAIFMT_IB_NF		(2 << 8) /* invert bclk + nor frm */
#define SND_SOC_DAIFMT_IB_IF		(3 << 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		(0 << 12) /* codec clk & frm master */
#define SND_SOC_DAIFMT_CBS_CFM		(1 << 12) /* codec clk slave & frm master */
#define SND_SOC_DAIFMT_CBM_CFS		(2 << 12) /* codec clk master & frame slave */
#define SND_SOC_DAIFMT_CBS_CFS		(3 << 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

struct snd_soc_dai_ops;
struct snd_soc_dai;
struct snd_ac97_bus_ops;

/* Digital Audio Interface registration */
int snd_soc_register_dai(struct snd_soc_dai *dai);
void snd_soc_unregister_dai(struct snd_soc_dai *dai);
int snd_soc_register_dais(struct snd_soc_dai *dai, size_t count);
void snd_soc_unregister_dais(struct snd_soc_dai *dai, size_t count);

/* 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, unsigned int freq_in, unsigned int freq_out);

/* 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 mask, int slots);

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);

/*
 * Digital Audio Interface.
 *
 * Describes the Digital Audio Interface in terms of it's ALSA, DAI and AC97
 * operations an capabilities. Codec and platfom drivers will register a this
 * structure for every DAI they have.
 *
 * This structure covers the clocking, formating and ALSA operations for each
 * interface a
 */
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, unsigned int freq_in, unsigned int freq_out);
	int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div);

	/*
	 * 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 mask, int slots);
	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);

	/*
	 * 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 *);
	int (*trigger)(struct snd_pcm_substream *, int,
		struct snd_soc_dai *);
};

/*
 * Digital Audio Interface runtime data.
 *
 * Holds runtime data for a DAI.
 */
struct snd_soc_dai {
	/* DAI description */
	char *name;
	unsigned int id;
	int ac97_control;

	struct device *dev;

	/* DAI callbacks */
	int (*probe)(struct platform_device *pdev,
		     struct snd_soc_dai *dai);
	void (*remove)(struct platform_device *pdev,
		       struct snd_soc_dai *dai);
	int (*suspend)(struct snd_soc_dai *dai);
	int (*resume)(struct snd_soc_dai *dai);

	/* ops */
	struct snd_soc_dai_ops *ops;

	/* DAI capabilities */
	struct snd_soc_pcm_stream capture;
	struct snd_soc_pcm_stream playback;

	/* DAI runtime info */
	struct snd_pcm_runtime *runtime;
	struct snd_soc_codec *codec;
	unsigned int active;
	unsigned char pop_wait:1;
	void *dma_data;

	/* DAI private data */
	void *private_data;

	/* parent codec/platform */
	union {
		struct snd_soc_codec *codec;
		struct snd_soc_platform *platform;
	};

	struct list_head list;
};

#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;
	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
	34	/* left and right justified also known as MSB and LSB respectively */
	35	#define SND_SOC_DAIFMT_MSB SND_SOC_DAIFMT_LEFT_J
	36	#define SND_SOC_DAIFMT_LSB SND_SOC_DAIFMT_RIGHT_J
	37
	38	/*
	39	* DAI Clock gating.
	40	*
	41	* DAI bit clocks can be be gated (disabled) when not the DAI is not
	42	* sending or receiving PCM data in a frame. This can be used to save power.
	43	*/
	44	#define SND_SOC_DAIFMT_CONT (0 << 4) /* continuous clock */
	45	#define SND_SOC_DAIFMT_GATED (1 << 4) /* clock is gated */
	46
	47	/*
	48	* DAI Left/Right Clocks.
	49	*
	50	* Specifies whether the DAI can support different samples for similtanious
	51	* playback and capture. This usually requires a seperate physical frame
	52	* clock for playback and capture.
	53	*/
	54	#define SND_SOC_DAIFMT_SYNC (0 << 5) /* Tx FRM = Rx FRM */
	55	#define SND_SOC_DAIFMT_ASYNC (1 << 5) /* Tx FRM ~ Rx FRM */
	56
	57	/*
	58	* TDM
	59	*
	60	* Time Division Multiplexing. Allows PCM data to be multplexed with other
	61	* data on the DAI.
	62	*/
	63	#define SND_SOC_DAIFMT_TDM (1 << 6)
	64
65	/*
66	* DAI hardware signal inversions.
67	*
68	* Specifies whether the DAI can also support inverted clocks for the specified
69	* format.
70	*/
71	#define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bit clock + frame */
72	#define SND_SOC_DAIFMT_NB_IF (1 << 8) /* normal bclk + inv frm */
73	#define SND_SOC_DAIFMT_IB_NF (2 << 8) /* invert bclk + nor frm */
74	#define SND_SOC_DAIFMT_IB_IF (3 << 8) /* invert bclk + frm */
75
76	/*
77	* DAI hardware clock masters.
78	*
79	* This is wrt the codec, the inverse is true for the interface
80	* i.e. if the codec is clk and frm master then the interface is
81	* clk and frame slave.
82	*/
83	#define SND_SOC_DAIFMT_CBM_CFM (0 << 12) /* codec clk & frm master */
84	#define SND_SOC_DAIFMT_CBS_CFM (1 << 12) /* codec clk slave & frm master */
85	#define SND_SOC_DAIFMT_CBM_CFS (2 << 12) /* codec clk master & frame slave */
86	#define SND_SOC_DAIFMT_CBS_CFS (3 << 12) /* codec clk & frm slave */
87
88	#define SND_SOC_DAIFMT_FORMAT_MASK 0x000f
89	#define SND_SOC_DAIFMT_CLOCK_MASK 0x00f0
90	#define SND_SOC_DAIFMT_INV_MASK 0x0f00
91	#define SND_SOC_DAIFMT_MASTER_MASK 0xf000
92
93	/*
94	* Master Clock Directions
95	*/
96	#define SND_SOC_CLOCK_IN 0
97	#define SND_SOC_CLOCK_OUT 1
98
99	struct snd_soc_dai_ops;
100	struct snd_soc_dai;
101	struct snd_ac97_bus_ops;
102
9115171a MB	103	/* Digital Audio Interface registration */
	104	int snd_soc_register_dai(struct snd_soc_dai *dai);
	105	void snd_soc_unregister_dai(struct snd_soc_dai *dai);
	106	int snd_soc_register_dais(struct snd_soc_dai *dai, size_t count);
	107	void snd_soc_unregister_dais(struct snd_soc_dai *dai, size_t count);
	108
a47cbe72 MB	109	/* Digital Audio Interface clocking API.*/
	110	int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
	111	unsigned int freq, int dir);
	112
	113	int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai,
	114	int div_id, int div);
	115
	116	int snd_soc_dai_set_pll(struct snd_soc_dai *dai,
	117	int pll_id, unsigned int freq_in, unsigned int freq_out);
	118
	119	/* Digital Audio interface formatting */
	120	int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt);
	121
	122	int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
	123	unsigned int mask, int slots);
	124
	125	int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate);
	126
	127	/* Digital Audio Interface mute */
	128	int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute);
	129
	130	/*
	131	* Digital Audio Interface.
	132	*
	133	* Describes the Digital Audio Interface in terms of it's ALSA, DAI and AC97
	134	* operations an capabilities. Codec and platfom drivers will register a this
	135	* structure for every DAI they have.
	136	*
	137	* This structure covers the clocking, formating and ALSA operations for each
	138	* interface a
	139	*/
	140	struct snd_soc_dai_ops {
	141	/*
	142	* DAI clocking configuration, all optional.
	143	* Called by soc_card drivers, normally in their hw_params.
	144	*/
	145	int (set_sysclk)(struct snd_soc_dai dai,
	146	int clk_id, unsigned int freq, int dir);
	147	int (set_pll)(struct snd_soc_dai dai,
	148	int pll_id, unsigned int freq_in, unsigned int freq_out);
	149	int (set_clkdiv)(struct snd_soc_dai dai, int div_id, int div);
	150
	151	/*
	152	* DAI format configuration
	153	* Called by soc_card drivers, normally in their hw_params.
	154	*/
	155	int (set_fmt)(struct snd_soc_dai dai, unsigned int fmt);
	156	int (set_tdm_slot)(struct snd_soc_dai dai,
	157	unsigned int mask, int slots);
	158	int (set_tristate)(struct snd_soc_dai dai, int tristate);
	159
	160	/*
	161	* DAI digital mute - optional.
	162	* Called by soc-core to minimise any pops.
	163	*/
	164	int (digital_mute)(struct snd_soc_dai dai, int mute);
dee89c4d MB	165
	166	/*
	167	* ALSA PCM audio operations - all optional.
	168	* Called by soc-core during audio PCM operations.
	169	*/
	170	int (startup)(struct snd_pcm_substream ,
	171	struct snd_soc_dai *);
	172	void (shutdown)(struct snd_pcm_substream ,
	173	struct snd_soc_dai *);
	174	int (hw_params)(struct snd_pcm_substream ,
	175	struct snd_pcm_hw_params , struct snd_soc_dai );
	176	int (hw_free)(struct snd_pcm_substream ,
	177	struct snd_soc_dai *);
	178	int (prepare)(struct snd_pcm_substream ,
	179	struct snd_soc_dai *);
	180	int (trigger)(struct snd_pcm_substream , int,
	181	struct snd_soc_dai *);
a47cbe72 MB	182	};
	183
	184	/*
	185	* Digital Audio Interface runtime data.
	186	*
	187	* Holds runtime data for a DAI.
	188	*/
	189	struct snd_soc_dai {
	190	/* DAI description */
	191	char *name;
	192	unsigned int id;
3ba9e10a	193	int ac97_control;
a47cbe72	194
9115171a MB	195	struct device *dev;
9115171a MB	196
a47cbe72 MB	197	/* DAI callbacks */
	198	int (probe)(struct platform_device pdev,
	199	struct snd_soc_dai *dai);
	200	void (remove)(struct platform_device pdev,
	201	struct snd_soc_dai *dai);
dc7d7b83 MB	202	int (suspend)(struct snd_soc_dai dai);
dc7d7b83 MB	203	int (resume)(struct snd_soc_dai dai);
a47cbe72 MB	204
a47cbe72 MB	205	/* ops */
6335d055	206	struct snd_soc_dai_ops *ops;
a47cbe72 MB	207
	208	/* DAI capabilities */
	209	struct snd_soc_pcm_stream capture;
	210	struct snd_soc_pcm_stream playback;
	211
	212	/* DAI runtime info */
	213	struct snd_pcm_runtime *runtime;
	214	struct snd_soc_codec *codec;
	215	unsigned int active;
	216	unsigned char pop_wait:1;
	217	void *dma_data;
	218
	219	/* DAI private data */
	220	void *private_data;
	221
	222	/* parent codec/platform */
	223	union {
	224	struct snd_soc_codec *codec;
	225	struct snd_soc_platform *platform;
	226	};
	227
	228	struct list_head list;
	229	};
	230
	231	#endif