include/sound/soc-dai.h - third_party/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>
 #include <sound/asoc.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		SND_SOC_DAI_FORMAT_I2S
 #define SND_SOC_DAIFMT_RIGHT_J		SND_SOC_DAI_FORMAT_RIGHT_J
 #define SND_SOC_DAIFMT_LEFT_J		SND_SOC_DAI_FORMAT_LEFT_J
 #define SND_SOC_DAIFMT_DSP_A		SND_SOC_DAI_FORMAT_DSP_A
 #define SND_SOC_DAIFMT_DSP_B		SND_SOC_DAI_FORMAT_DSP_B
 #define SND_SOC_DAIFMT_AC97		SND_SOC_DAI_FORMAT_AC97
 #define SND_SOC_DAIFMT_PDM		SND_SOC_DAI_FORMAT_PDM

 /* 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 polarity.
  *
  * Specifies whether the DAI can also support inverted clocks for the specified
  * format.
  *
  * BCLK:
  * - "normal" polarity means signal is available at rising edge of BCLK
  * - "inverted" polarity means signal is available at falling edge of BCLK
  *
  * FSYNC "normal" polarity depends on the frame format:
  * - I2S: frame consists of left then right channel data. Left channel starts
  *      with falling FSYNC edge, right channel starts with rising FSYNC edge.
  * - Left/Right Justified: frame consists of left then right channel data.
  *      Left channel starts with rising FSYNC edge, right channel starts with
  *      falling FSYNC edge.
  * - DSP A/B: Frame starts with rising FSYNC edge.
  * - AC97: Frame starts with rising FSYNC edge.
  *
  * "Negative" FSYNC polarity is the one opposite of "normal" polarity.
  */
 #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_S20_LE |\
 			       SNDRV_PCM_FMTBIT_S20_BE |\
 			       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 (*xlate_tdm_slot_mask)(unsigned int slots,
 		unsigned int *tx_mask, unsigned int *rx_mask);
 	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);

 	int (*set_sdw_stream)(struct snd_soc_dai *dai,
 			void *stream, int direction);
 	/*
 	 * 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 *);
 };

 struct snd_soc_cdai_ops {
 	/*
 	 * for compress ops
 	 */
 	int (*startup)(struct snd_compr_stream *,
 			struct snd_soc_dai *);
 	int (*shutdown)(struct snd_compr_stream *,
 			struct snd_soc_dai *);
 	int (*set_params)(struct snd_compr_stream *,
 			struct snd_compr_params *, struct snd_soc_dai *);
 	int (*get_params)(struct snd_compr_stream *,
 			struct snd_codec *, struct snd_soc_dai *);
 	int (*set_metadata)(struct snd_compr_stream *,
 			struct snd_compr_metadata *, struct snd_soc_dai *);
 	int (*get_metadata)(struct snd_compr_stream *,
 			struct snd_compr_metadata *, struct snd_soc_dai *);
 	int (*trigger)(struct snd_compr_stream *, int,
 			struct snd_soc_dai *);
 	int (*pointer)(struct snd_compr_stream *,
 			struct snd_compr_tstamp *, struct snd_soc_dai *);
 	int (*ack)(struct snd_compr_stream *, size_t,
 			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;
 	unsigned int base;
 	struct snd_soc_dobj dobj;

 	/* 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 */
 	int (*compress_new)(struct snd_soc_pcm_runtime *rtd, int num);
 	/* Optional Callback used at pcm creation*/
 	int (*pcm_new)(struct snd_soc_pcm_runtime *rtd,
 		       struct snd_soc_dai *dai);
 	/* DAI is also used for the control bus */
 	bool bus_control;

 	/* ops */
 	const struct snd_soc_dai_ops *ops;
 	const struct snd_soc_cdai_ops *cops;

 	/* 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;

 	/* driver ops */
 	struct snd_soc_dai_driver *driver;

 	/* DAI runtime info */
 	unsigned int capture_active;		/* stream usage count */
 	unsigned int playback_active;		/* stream usage count */
 	unsigned int probed:1;

 	unsigned int active;

 	struct snd_soc_dapm_widget *playback_widget;
 	struct snd_soc_dapm_widget *capture_widget;

 	/* 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_component *component;

 	/* CODEC TDM slot masks and params (for fixup) */
 	unsigned int tx_mask;
 	unsigned int rx_mask;

 	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;
 }

 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);
 }

 /**
  * snd_soc_dai_set_sdw_stream() - Configures a DAI for SDW stream operation
  * @dai: DAI
  * @stream: STREAM
  * @direction: Stream direction(Playback/Capture)
  * SoundWire subsystem doesn't have a notion of direction and we reuse
  * the ASoC stream direction to configure sink/source ports.
  * Playback maps to source ports and Capture for sink ports.
  *
  * This should be invoked with NULL to clear the stream set previously.
  * Returns 0 on success, a negative error code otherwise.
  */
 static inline int snd_soc_dai_set_sdw_stream(struct snd_soc_dai *dai,
 				void *stream, int direction)
 {
 	if (dai->driver->ops->set_sdw_stream)
 		return dai->driver->ops->set_sdw_stream(dai, stream, direction);
 	else
 		return -ENOTSUPP;
 }

 #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>
	#include <sound/asoc.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 SND_SOC_DAI_FORMAT_I2S
	#define SND_SOC_DAIFMT_RIGHT_J SND_SOC_DAI_FORMAT_RIGHT_J
	#define SND_SOC_DAIFMT_LEFT_J SND_SOC_DAI_FORMAT_LEFT_J
	#define SND_SOC_DAIFMT_DSP_A SND_SOC_DAI_FORMAT_DSP_A
	#define SND_SOC_DAIFMT_DSP_B SND_SOC_DAI_FORMAT_DSP_B
	#define SND_SOC_DAIFMT_AC97 SND_SOC_DAI_FORMAT_AC97
	#define SND_SOC_DAIFMT_PDM SND_SOC_DAI_FORMAT_PDM

	/* 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 polarity.
	*
	* Specifies whether the DAI can also support inverted clocks for the specified
	* format.
	*
	* BCLK:
	* - "normal" polarity means signal is available at rising edge of BCLK
	* - "inverted" polarity means signal is available at falling edge of BCLK
	*
	* FSYNC "normal" polarity depends on the frame format:
	* - I2S: frame consists of left then right channel data. Left channel starts
	* with falling FSYNC edge, right channel starts with rising FSYNC edge.
	* - Left/Right Justified: frame consists of left then right channel data.
	* Left channel starts with rising FSYNC edge, right channel starts with
	* falling FSYNC edge.
	* - DSP A/B: Frame starts with rising FSYNC edge.
	* - AC97: Frame starts with rising FSYNC edge.
	*
	* "Negative" FSYNC polarity is the one opposite of "normal" polarity.
	*/
	#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_S20_LE \|\
	SNDRV_PCM_FMTBIT_S20_BE \|\
	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 (*xlate_tdm_slot_mask)(unsigned int slots,
	unsigned int tx_mask, unsigned int rx_mask);
	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);

	int (set_sdw_stream)(struct snd_soc_dai dai,
	void *stream, int direction);
	/*
	* 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 *);
	};

	struct snd_soc_cdai_ops {
	/*
	* for compress ops
	*/
	int (startup)(struct snd_compr_stream ,
	struct snd_soc_dai *);
	int (shutdown)(struct snd_compr_stream ,
	struct snd_soc_dai *);
	int (set_params)(struct snd_compr_stream ,
	struct snd_compr_params , struct snd_soc_dai );
	int (get_params)(struct snd_compr_stream ,
	struct snd_codec , struct snd_soc_dai );
	int (set_metadata)(struct snd_compr_stream ,
	struct snd_compr_metadata , struct snd_soc_dai );
	int (get_metadata)(struct snd_compr_stream ,
	struct snd_compr_metadata , struct snd_soc_dai );
	int (trigger)(struct snd_compr_stream , int,
	struct snd_soc_dai *);
	int (pointer)(struct snd_compr_stream ,
	struct snd_compr_tstamp , struct snd_soc_dai );
	int (ack)(struct snd_compr_stream , size_t,
	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;
	unsigned int base;
	struct snd_soc_dobj dobj;

	/* 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 */
	int (compress_new)(struct snd_soc_pcm_runtime rtd, int num);
	/* Optional Callback used at pcm creation*/
	int (pcm_new)(struct snd_soc_pcm_runtime rtd,
	struct snd_soc_dai *dai);
	/* DAI is also used for the control bus */
	bool bus_control;

	/* ops */
	const struct snd_soc_dai_ops *ops;
	const struct snd_soc_cdai_ops *cops;

	/* 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;

	/* driver ops */
	struct snd_soc_dai_driver *driver;

	/* DAI runtime info */
	unsigned int capture_active; /* stream usage count */
	unsigned int playback_active; /* stream usage count */
	unsigned int probed:1;

	unsigned int active;

	struct snd_soc_dapm_widget *playback_widget;
	struct snd_soc_dapm_widget *capture_widget;

	/* 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_component *component;

	/* CODEC TDM slot masks and params (for fixup) */
	unsigned int tx_mask;
	unsigned int rx_mask;

	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;
	}

	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);
	}

	/**
	* snd_soc_dai_set_sdw_stream() - Configures a DAI for SDW stream operation
	* @dai: DAI
	* @stream: STREAM
	* @direction: Stream direction(Playback/Capture)
	* SoundWire subsystem doesn't have a notion of direction and we reuse
	* the ASoC stream direction to configure sink/source ports.
	* Playback maps to source ports and Capture for sink ports.
	*
	* This should be invoked with NULL to clear the stream set previously.
	* Returns 0 on success, a negative error code otherwise.
	*/
	static inline int snd_soc_dai_set_sdw_stream(struct snd_soc_dai *dai,
	void *stream, int direction)
	{
	if (dai->driver->ops->set_sdw_stream)
	return dai->driver->ops->set_sdw_stream(dai, stream, direction);
	else
	return -ENOTSUPP;
	}

	#endif