[mirror_ubuntu-jammy-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>

#include <sound/soc.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 */
#define SND_SOC_DAIFMT_PDM		6 /* 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		(0 << 4) /* continuous clock */
#define SND_SOC_DAIFMT_GATED		(1 << 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		(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

#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_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, int source, 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 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);

/*
 * Digital Audio Interface.
 *
 * 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_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);

	/*
	 * 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);

	/*
	 * 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 *);
	/*
	 * For hardware based FIFO caused delay reporting.
	 * Optional.
	 */
	snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *,
		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;
	void *ac97_pdata;	/* platform_data for the ac97 codec */

	/* 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;
	unsigned int symmetric_rates:1;

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

	/* DAI private data */
	void *private_data;

	/* parent platform */
	struct snd_soc_platform *platform;

	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;
}

#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
84740ac1 GL	19	#include <sound/soc.h>
84740ac1 GL	20
a47cbe72 MB	21	struct snd_pcm_substream;
	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	*/
	29	#define SND_SOC_DAIFMT_I2S 0 /* I2S mode */
	30	#define SND_SOC_DAIFMT_RIGHT_J 1 /* Right Justified mode */
	31	#define SND_SOC_DAIFMT_LEFT_J 2 /* Left Justified mode */
47db8e89 PM	32	#define SND_SOC_DAIFMT_DSP_A 3 /* L data MSB after FRM LRC */
47db8e89 PM	33	#define SND_SOC_DAIFMT_DSP_B 4 /* L data MSB during FRM LRC */
a47cbe72	34	#define SND_SOC_DAIFMT_AC97 5 /* AC97 */
be2500b8	35	#define SND_SOC_DAIFMT_PDM 6 /* 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.
	46	*/
	47	#define SND_SOC_DAIFMT_CONT (0 << 4) /* continuous clock */
	48	#define SND_SOC_DAIFMT_GATED (1 << 4) /* clock is gated */
	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	*/
	56	#define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bit clock + frame */
47db8e89 PM	57	#define SND_SOC_DAIFMT_NB_IF (1 << 8) /* normal BCLK + inv FRM */
	58	#define SND_SOC_DAIFMT_IB_NF (2 << 8) /* invert BCLK + nor FRM */
	59	#define SND_SOC_DAIFMT_IB_IF (3 << 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	*/
47db8e89 PM	68	#define SND_SOC_DAIFMT_CBM_CFM (0 << 12) /* codec clk & FRM master */
47db8e89 PM	69	#define SND_SOC_DAIFMT_CBS_CFM (1 << 12) /* codec clk slave & FRM master */
a47cbe72	70	#define SND_SOC_DAIFMT_CBM_CFS (2 << 12) /* codec clk master & frame slave */
47db8e89	71	#define SND_SOC_DAIFMT_CBS_CFS (3 << 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
a47cbe72 MB	94	struct snd_soc_dai_ops;
	95	struct snd_soc_dai;
	96	struct snd_ac97_bus_ops;
	97
9115171a MB	98	/* Digital Audio Interface registration */
	99	int snd_soc_register_dai(struct snd_soc_dai *dai);
	100	void snd_soc_unregister_dai(struct snd_soc_dai *dai);
	101	int snd_soc_register_dais(struct snd_soc_dai *dai, size_t count);
	102	void snd_soc_unregister_dais(struct snd_soc_dai *dai, size_t count);
	103
a47cbe72 MB	104	/* Digital Audio Interface clocking API.*/
	105	int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
	106	unsigned int freq, int dir);
	107
	108	int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai,
	109	int div_id, int div);
	110
	111	int snd_soc_dai_set_pll(struct snd_soc_dai *dai,
85488037	112	int pll_id, int source, unsigned int freq_in, unsigned int freq_out);
a47cbe72 MB	113
	114	/* Digital Audio interface formatting */
	115	int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt);
	116
	117	int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
a5479e38	118	unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width);
a47cbe72	119
472df3cb BS	120	int 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
a47cbe72 MB	124	int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate);
	125
	126	/* Digital Audio Interface mute */
	127	int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute);
	128
	129	/*
	130	* Digital Audio Interface.
	131	*
47db8e89 PM	132	* Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97
47db8e89 PM	133	* operations and capabilities. Codec and platform drivers will register this
a47cbe72 MB	134	* structure for every DAI they have.
	135	*
	136	* This structure covers the clocking, formating and ALSA operations for each
47db8e89	137	* interface.
a47cbe72 MB	138	*/
	139	struct snd_soc_dai_ops {
	140	/*
	141	* DAI clocking configuration, all optional.
	142	* Called by soc_card drivers, normally in their hw_params.
	143	*/
	144	int (set_sysclk)(struct snd_soc_dai dai,
	145	int clk_id, unsigned int freq, int dir);
85488037 MB	146	int (set_pll)(struct snd_soc_dai dai, int pll_id, int source,
85488037 MB	147	unsigned int freq_in, unsigned int freq_out);
a47cbe72 MB	148	int (set_clkdiv)(struct snd_soc_dai dai, int div_id, int div);
	149
	150	/*
	151	* DAI format configuration
	152	* Called by soc_card drivers, normally in their hw_params.
	153	*/
	154	int (set_fmt)(struct snd_soc_dai dai, unsigned int fmt);
	155	int (set_tdm_slot)(struct snd_soc_dai dai,
a5479e38 DR	156	unsigned int tx_mask, unsigned int rx_mask,
a5479e38 DR	157	int slots, int slot_width);
472df3cb BS	158	int (set_channel_map)(struct snd_soc_dai dai,
	159	unsigned int tx_num, unsigned int *tx_slot,
	160	unsigned int rx_num, unsigned int *rx_slot);
a47cbe72 MB	161	int (set_tristate)(struct snd_soc_dai dai, int tristate);
	162
	163	/*
	164	* DAI digital mute - optional.
	165	* Called by soc-core to minimise any pops.
	166	*/
	167	int (digital_mute)(struct snd_soc_dai dai, int mute);
dee89c4d MB	168
	169	/*
	170	* ALSA PCM audio operations - all optional.
	171	* Called by soc-core during audio PCM operations.
	172	*/
	173	int (startup)(struct snd_pcm_substream ,
	174	struct snd_soc_dai *);
	175	void (shutdown)(struct snd_pcm_substream ,
	176	struct snd_soc_dai *);
	177	int (hw_params)(struct snd_pcm_substream ,
	178	struct snd_pcm_hw_params , struct snd_soc_dai );
	179	int (hw_free)(struct snd_pcm_substream ,
	180	struct snd_soc_dai *);
	181	int (prepare)(struct snd_pcm_substream ,
	182	struct snd_soc_dai *);
	183	int (trigger)(struct snd_pcm_substream , int,
	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	/*
	194	* Digital Audio Interface runtime data.
	195	*
	196	* Holds runtime data for a DAI.
	197	*/
	198	struct snd_soc_dai {
	199	/* DAI description */
	200	char *name;
	201	unsigned int id;
3ba9e10a	202	int ac97_control;
a47cbe72	203
9115171a	204	struct device *dev;
474828a4	205	void ac97_pdata; / platform_data for the ac97 codec */
9115171a	206
a47cbe72 MB	207	/* DAI callbacks */
	208	int (probe)(struct platform_device pdev,
	209	struct snd_soc_dai *dai);
	210	void (remove)(struct platform_device pdev,
	211	struct snd_soc_dai *dai);
dc7d7b83 MB	212	int (suspend)(struct snd_soc_dai dai);
dc7d7b83 MB	213	int (resume)(struct snd_soc_dai dai);
a47cbe72 MB	214
a47cbe72 MB	215	/* ops */
6335d055	216	struct snd_soc_dai_ops *ops;
a47cbe72 MB	217
	218	/* DAI capabilities */
	219	struct snd_soc_pcm_stream capture;
	220	struct snd_soc_pcm_stream playback;
06f409d7	221	unsigned int symmetric_rates:1;
a47cbe72 MB	222
a47cbe72 MB	223	/* DAI runtime info */
a47cbe72 MB	224	struct snd_soc_codec *codec;
	225	unsigned int active;
	226	unsigned char pop_wait:1;
a47cbe72 MB	227
	228	/* DAI private data */
	229	void *private_data;
	230
bbd99307 MB	231	/* parent platform */
bbd99307 MB	232	struct snd_soc_platform *platform;
a47cbe72 MB	233
	234	struct list_head list;
	235	};
	236
fd23b7de DM	237	static inline void snd_soc_dai_get_dma_data(const struct snd_soc_dai dai,
	238	const struct snd_pcm_substream *ss)
	239	{
	240	return (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
	241	dai->playback.dma_data : dai->capture.dma_data;
	242	}
	243
	244	static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai,
	245	const struct snd_pcm_substream *ss,
	246	void *data)
	247	{
	248	if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK)
	249	dai->playback.dma_data = data;
	250	else
	251	dai->capture.dma_data = data;
	252	}
	253
a47cbe72	254	#endif