include/sound/soc-dai.h - linux - Git at Google

 /*
  * 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
	/*
	* 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