/*
   * 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;
  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		1 /* I2S mode */
  #define SND_SOC_DAIFMT_RIGHT_J		2 /* Right Justified mode */
  #define SND_SOC_DAIFMT_LEFT_J		3 /* Left Justified mode */
  #define SND_SOC_DAIFMT_DSP_A		4 /* L data MSB after FRM LRC */
  #define SND_SOC_DAIFMT_DSP_B		5 /* L data MSB during FRM LRC */
  #define SND_SOC_DAIFMT_AC97		6 /* AC97 */
  #define SND_SOC_DAIFMT_PDM		7 /* 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		(1 << 4) /* continuous clock */
  #define SND_SOC_DAIFMT_GATED		(0 << 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		(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_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 (*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);
  	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 *);
  };
  
  /*
   * 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;
  	int ac97_control;
  	unsigned int base;
  
  	/* 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 */
  	bool compress_dai;
  
  	/* ops */
  	const 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;
  	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;
  	void *ac97_pdata;	/* platform_data for the ac97 codec */
  
  	/* driver ops */
  	struct snd_soc_dai_driver *driver;
  
  	/* DAI runtime info */
  	unsigned int capture_active:1;		/* stream is in use */
  	unsigned int playback_active:1;		/* stream is in use */
  	unsigned int symmetric_rates:1;
  	unsigned int symmetric_channels:1;
  	unsigned int symmetric_samplebits:1;
  	struct snd_pcm_runtime *runtime;
  	unsigned int active;
  	unsigned char probed:1;
  
  	struct snd_soc_dapm_widget *playback_widget;
  	struct snd_soc_dapm_widget *capture_widget;
  	struct snd_soc_dapm_context dapm;
  
  	/* 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_platform *platform;
  	struct snd_soc_codec *codec;
  
  	struct snd_soc_card *card;
  
  	struct list_head list;
  	struct list_head card_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);
  }
  
  #endif