/* SPDX-License-Identifier: GPL-2.0
   *

   * linux/sound/soc-dai.h -- ALSA SoC Layer
   *
   * Copyright:	2005-2008 Wolfson Microelectronics. PLC.
   *

   * 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_get_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_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 (*get_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