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			224 lines
		
	
	
		
			6.6 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
			
		
		
	
	
			224 lines
		
	
	
		
			6.6 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
| /*
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|  * linux/sound/soc-dai.h -- ALSA SoC Layer
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|  *
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|  * Copyright:	2005-2008 Wolfson Microelectronics. PLC.
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|  *
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|  * This program is free software; you can redistribute it and/or modify
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|  * it under the terms of the GNU General Public License version 2 as
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|  * published by the Free Software Foundation.
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|  *
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|  * Digital Audio Interface (DAI) API.
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|  */
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| 
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| #ifndef __LINUX_SND_SOC_DAI_H
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| #define __LINUX_SND_SOC_DAI_H
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| 
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| 
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| #include <linux/list.h>
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| 
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| struct snd_pcm_substream;
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| 
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| /*
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|  * DAI hardware audio formats.
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|  *
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|  * Describes the physical PCM data formating and clocking. Add new formats
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|  * to the end.
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|  */
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| #define SND_SOC_DAIFMT_I2S		0 /* I2S mode */
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| #define SND_SOC_DAIFMT_RIGHT_J		1 /* Right Justified mode */
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| #define SND_SOC_DAIFMT_LEFT_J		2 /* Left Justified mode */
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| #define SND_SOC_DAIFMT_DSP_A		3 /* L data MSB after FRM LRC */
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| #define SND_SOC_DAIFMT_DSP_B		4 /* L data MSB during FRM LRC */
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| #define SND_SOC_DAIFMT_AC97		5 /* AC97 */
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| 
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| /* left and right justified also known as MSB and LSB respectively */
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| #define SND_SOC_DAIFMT_MSB		SND_SOC_DAIFMT_LEFT_J
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| #define SND_SOC_DAIFMT_LSB		SND_SOC_DAIFMT_RIGHT_J
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| 
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| /*
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|  * DAI Clock gating.
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|  *
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|  * DAI bit clocks can be be gated (disabled) when the DAI is not
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|  * sending or receiving PCM data in a frame. This can be used to save power.
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|  */
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| #define SND_SOC_DAIFMT_CONT		(0 << 4) /* continuous clock */
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| #define SND_SOC_DAIFMT_GATED		(1 << 4) /* clock is gated */
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| 
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| /*
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|  * DAI hardware signal inversions.
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|  *
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|  * Specifies whether the DAI can also support inverted clocks for the specified
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|  * format.
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|  */
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| #define SND_SOC_DAIFMT_NB_NF		(0 << 8) /* normal bit clock + frame */
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| #define SND_SOC_DAIFMT_NB_IF		(1 << 8) /* normal BCLK + inv FRM */
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| #define SND_SOC_DAIFMT_IB_NF		(2 << 8) /* invert BCLK + nor FRM */
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| #define SND_SOC_DAIFMT_IB_IF		(3 << 8) /* invert BCLK + FRM */
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| 
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| /*
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|  * DAI hardware clock masters.
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|  *
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|  * This is wrt the codec, the inverse is true for the interface
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|  * i.e. if the codec is clk and FRM master then the interface is
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|  * clk and frame slave.
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|  */
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| #define SND_SOC_DAIFMT_CBM_CFM		(0 << 12) /* codec clk & FRM master */
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| #define SND_SOC_DAIFMT_CBS_CFM		(1 << 12) /* codec clk slave & FRM master */
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| #define SND_SOC_DAIFMT_CBM_CFS		(2 << 12) /* codec clk master & frame slave */
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| #define SND_SOC_DAIFMT_CBS_CFS		(3 << 12) /* codec clk & FRM slave */
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| 
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| #define SND_SOC_DAIFMT_FORMAT_MASK	0x000f
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| #define SND_SOC_DAIFMT_CLOCK_MASK	0x00f0
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| #define SND_SOC_DAIFMT_INV_MASK		0x0f00
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| #define SND_SOC_DAIFMT_MASTER_MASK	0xf000
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| 
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| /*
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|  * Master Clock Directions
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|  */
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| #define SND_SOC_CLOCK_IN		0
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| #define SND_SOC_CLOCK_OUT		1
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| 
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| #define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S8 |\
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| 			       SNDRV_PCM_FMTBIT_S16_LE |\
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| 			       SNDRV_PCM_FMTBIT_S16_BE |\
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| 			       SNDRV_PCM_FMTBIT_S20_3LE |\
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| 			       SNDRV_PCM_FMTBIT_S20_3BE |\
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| 			       SNDRV_PCM_FMTBIT_S24_3LE |\
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| 			       SNDRV_PCM_FMTBIT_S24_3BE |\
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|                                SNDRV_PCM_FMTBIT_S32_LE |\
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|                                SNDRV_PCM_FMTBIT_S32_BE)
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| 
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| struct snd_soc_dai_ops;
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| struct snd_soc_dai;
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| struct snd_ac97_bus_ops;
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| 
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| /* Digital Audio Interface registration */
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| int snd_soc_register_dai(struct snd_soc_dai *dai);
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| void snd_soc_unregister_dai(struct snd_soc_dai *dai);
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| int snd_soc_register_dais(struct snd_soc_dai *dai, size_t count);
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| void snd_soc_unregister_dais(struct snd_soc_dai *dai, size_t count);
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| 
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| /* Digital Audio Interface clocking API.*/
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| int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
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| 	unsigned int freq, int dir);
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| 
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| int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai,
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| 	int div_id, int div);
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| 
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| int snd_soc_dai_set_pll(struct snd_soc_dai *dai,
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| 	int pll_id, unsigned int freq_in, unsigned int freq_out);
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| 
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| /* Digital Audio interface formatting */
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| int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt);
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| 
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| int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
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| 	unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width);
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| 
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| int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate);
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| 
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| /* Digital Audio Interface mute */
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| int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute);
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| 
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| /*
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|  * Digital Audio Interface.
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|  *
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|  * Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97
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|  * operations and capabilities. Codec and platform drivers will register this
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|  * structure for every DAI they have.
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|  *
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|  * This structure covers the clocking, formating and ALSA operations for each
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|  * interface.
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|  */
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| struct snd_soc_dai_ops {
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| 	/*
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| 	 * DAI clocking configuration, all optional.
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| 	 * Called by soc_card drivers, normally in their hw_params.
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| 	 */
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| 	int (*set_sysclk)(struct snd_soc_dai *dai,
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| 		int clk_id, unsigned int freq, int dir);
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| 	int (*set_pll)(struct snd_soc_dai *dai,
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| 		int pll_id, unsigned int freq_in, unsigned int freq_out);
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| 	int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div);
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| 
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| 	/*
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| 	 * DAI format configuration
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| 	 * Called by soc_card drivers, normally in their hw_params.
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| 	 */
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| 	int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt);
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| 	int (*set_tdm_slot)(struct snd_soc_dai *dai,
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| 		unsigned int tx_mask, unsigned int rx_mask,
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| 		int slots, int slot_width);
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| 	int (*set_tristate)(struct snd_soc_dai *dai, int tristate);
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| 
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| 	/*
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| 	 * DAI digital mute - optional.
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| 	 * Called by soc-core to minimise any pops.
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| 	 */
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| 	int (*digital_mute)(struct snd_soc_dai *dai, int mute);
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| 
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| 	/*
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| 	 * ALSA PCM audio operations - all optional.
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| 	 * Called by soc-core during audio PCM operations.
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| 	 */
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| 	int (*startup)(struct snd_pcm_substream *,
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| 		struct snd_soc_dai *);
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| 	void (*shutdown)(struct snd_pcm_substream *,
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| 		struct snd_soc_dai *);
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| 	int (*hw_params)(struct snd_pcm_substream *,
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| 		struct snd_pcm_hw_params *, struct snd_soc_dai *);
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| 	int (*hw_free)(struct snd_pcm_substream *,
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| 		struct snd_soc_dai *);
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| 	int (*prepare)(struct snd_pcm_substream *,
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| 		struct snd_soc_dai *);
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| 	int (*trigger)(struct snd_pcm_substream *, int,
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| 		struct snd_soc_dai *);
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| };
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| 
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| /*
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|  * Digital Audio Interface runtime data.
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|  *
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|  * Holds runtime data for a DAI.
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|  */
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| struct snd_soc_dai {
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| 	/* DAI description */
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| 	char *name;
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| 	unsigned int id;
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| 	int ac97_control;
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| 
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| 	struct device *dev;
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| 	void *ac97_pdata;	/* platform_data for the ac97 codec */
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| 
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| 	/* DAI callbacks */
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| 	int (*probe)(struct platform_device *pdev,
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| 		     struct snd_soc_dai *dai);
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| 	void (*remove)(struct platform_device *pdev,
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| 		       struct snd_soc_dai *dai);
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| 	int (*suspend)(struct snd_soc_dai *dai);
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| 	int (*resume)(struct snd_soc_dai *dai);
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| 
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| 	/* ops */
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| 	struct snd_soc_dai_ops *ops;
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| 
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| 	/* DAI capabilities */
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| 	struct snd_soc_pcm_stream capture;
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| 	struct snd_soc_pcm_stream playback;
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| 	unsigned int symmetric_rates:1;
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| 
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| 	/* DAI runtime info */
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| 	struct snd_pcm_runtime *runtime;
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| 	struct snd_soc_codec *codec;
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| 	unsigned int active;
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| 	unsigned char pop_wait:1;
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| 	void *dma_data;
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| 
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| 	/* DAI private data */
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| 	void *private_data;
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| 
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| 	/* parent platform */
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| 	struct snd_soc_platform *platform;
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| 
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| 	struct list_head list;
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| };
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| 
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| #endif
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