[linux-2.6-block.git] / include / sound / soc-dai.h

/* 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);

int snd_soc_dai_hw_params(struct snd_soc_dai *dai,
			  struct snd_pcm_substream *substream,
			  struct snd_pcm_hw_params *params);
void snd_soc_dai_hw_free(struct snd_soc_dai *dai,
			 struct snd_pcm_substream *substream);
int snd_soc_dai_startup(struct snd_soc_dai *dai,
			struct snd_pcm_substream *substream);
void snd_soc_dai_shutdown(struct snd_soc_dai *dai,
			  struct snd_pcm_substream *substream);
int snd_soc_dai_prepare(struct snd_soc_dai *dai,
			struct snd_pcm_substream *substream);
int snd_soc_dai_trigger(struct snd_soc_dai *dai,
			struct snd_pcm_substream *substream, int cmd);
int snd_soc_dai_bespoke_trigger(struct snd_soc_dai *dai,
			struct snd_pcm_substream *substream, int cmd);
snd_pcm_sframes_t snd_soc_dai_delay(struct snd_soc_dai *dai,
				    struct snd_pcm_substream *substream);
void snd_soc_dai_suspend(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
Commit	Line	Data
e2cfd2c9 KM	1	/* SPDX-License-Identifier: GPL-2.0
e2cfd2c9 KM	2	*
a47cbe72 MB	3	* linux/sound/soc-dai.h -- ALSA SoC Layer
	4	*
	5	* Copyright: 2005-2008 Wolfson Microelectronics. PLC.
	6	*
a47cbe72 MB	7	* Digital Audio Interface (DAI) API.
	8	*/
	9
	10	#ifndef __LINUX_SND_SOC_DAI_H
	11	#define __LINUX_SND_SOC_DAI_H
	12
	13
	14	#include <linux/list.h>
d2e3cb85	15	#include <sound/asoc.h>
a47cbe72 MB	16
a47cbe72 MB	17	struct snd_pcm_substream;
888df395	18	struct snd_soc_dapm_widget;
49681077	19	struct snd_compr_stream;
a47cbe72 MB	20
	21	/*
	22	* DAI hardware audio formats.
	23	*
	24	* Describes the physical PCM data formating and clocking. Add new formats
	25	* to the end.
	26	*/
d2e3cb85 ML	27	#define SND_SOC_DAIFMT_I2S SND_SOC_DAI_FORMAT_I2S
	28	#define SND_SOC_DAIFMT_RIGHT_J SND_SOC_DAI_FORMAT_RIGHT_J
	29	#define SND_SOC_DAIFMT_LEFT_J SND_SOC_DAI_FORMAT_LEFT_J
	30	#define SND_SOC_DAIFMT_DSP_A SND_SOC_DAI_FORMAT_DSP_A
	31	#define SND_SOC_DAIFMT_DSP_B SND_SOC_DAI_FORMAT_DSP_B
	32	#define SND_SOC_DAIFMT_AC97 SND_SOC_DAI_FORMAT_AC97
	33	#define SND_SOC_DAIFMT_PDM SND_SOC_DAI_FORMAT_PDM
a47cbe72 MB	34
	35	/* left and right justified also known as MSB and LSB respectively */
	36	#define SND_SOC_DAIFMT_MSB SND_SOC_DAIFMT_LEFT_J
	37	#define SND_SOC_DAIFMT_LSB SND_SOC_DAIFMT_RIGHT_J
	38
	39	/*
	40	* DAI Clock gating.
	41	*
47db8e89	42	* DAI bit clocks can be be gated (disabled) when the DAI is not
a47cbe72 MB	43	* sending or receiving PCM data in a frame. This can be used to save power.
a47cbe72 MB	44	*/
75d9ac46	45	#define SND_SOC_DAIFMT_CONT (1 << 4) /* continuous clock */
eef28e10	46	#define SND_SOC_DAIFMT_GATED (0 << 4) /* clock is gated */
a47cbe72	47
a47cbe72	48	/*
1d387a3f	49	* DAI hardware signal polarity.
a47cbe72 MB	50	*
	51	* Specifies whether the DAI can also support inverted clocks for the specified
	52	* format.
1d387a3f AP	53	*
	54	* BCLK:
	55	* - "normal" polarity means signal is available at rising edge of BCLK
	56	* - "inverted" polarity means signal is available at falling edge of BCLK
	57	*
	58	* FSYNC "normal" polarity depends on the frame format:
	59	* - I2S: frame consists of left then right channel data. Left channel starts
	60	* with falling FSYNC edge, right channel starts with rising FSYNC edge.
	61	* - Left/Right Justified: frame consists of left then right channel data.
	62	* Left channel starts with rising FSYNC edge, right channel starts with
	63	* falling FSYNC edge.
	64	* - DSP A/B: Frame starts with rising FSYNC edge.
	65	* - AC97: Frame starts with rising FSYNC edge.
	66	*
	67	* "Negative" FSYNC polarity is the one opposite of "normal" polarity.
a47cbe72	68	*/
5d163336	69	#define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bit clock + frame */
75d9ac46 MB	70	#define SND_SOC_DAIFMT_NB_IF (2 << 8) /* normal BCLK + inv FRM */
	71	#define SND_SOC_DAIFMT_IB_NF (3 << 8) /* invert BCLK + nor FRM */
	72	#define SND_SOC_DAIFMT_IB_IF (4 << 8) /* invert BCLK + FRM */
a47cbe72 MB	73
	74	/*
	75	* DAI hardware clock masters.
	76	*
	77	* This is wrt the codec, the inverse is true for the interface
47db8e89	78	* i.e. if the codec is clk and FRM master then the interface is
a47cbe72 MB	79	* clk and frame slave.
a47cbe72 MB	80	*/
75d9ac46 MB	81	#define SND_SOC_DAIFMT_CBM_CFM (1 << 12) /* codec clk & FRM master */
	82	#define SND_SOC_DAIFMT_CBS_CFM (2 << 12) /* codec clk slave & FRM master */
	83	#define SND_SOC_DAIFMT_CBM_CFS (3 << 12) /* codec clk master & frame slave */
	84	#define SND_SOC_DAIFMT_CBS_CFS (4 << 12) /* codec clk & FRM slave */
a47cbe72 MB	85
	86	#define SND_SOC_DAIFMT_FORMAT_MASK 0x000f
	87	#define SND_SOC_DAIFMT_CLOCK_MASK 0x00f0
	88	#define SND_SOC_DAIFMT_INV_MASK 0x0f00
	89	#define SND_SOC_DAIFMT_MASTER_MASK 0xf000
	90
	91	/*
	92	* Master Clock Directions
	93	*/
	94	#define SND_SOC_CLOCK_IN 0
	95	#define SND_SOC_CLOCK_OUT 1
	96
8f738d58 MB	97	#define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S8 \|\
	98	SNDRV_PCM_FMTBIT_S16_LE \|\
	99	SNDRV_PCM_FMTBIT_S16_BE \|\
	100	SNDRV_PCM_FMTBIT_S20_3LE \|\
	101	SNDRV_PCM_FMTBIT_S20_3BE \|\
823dbb6e MS	102	SNDRV_PCM_FMTBIT_S20_LE \|\
823dbb6e MS	103	SNDRV_PCM_FMTBIT_S20_BE \|\
8f738d58 MB	104	SNDRV_PCM_FMTBIT_S24_3LE \|\
8f738d58 MB	105	SNDRV_PCM_FMTBIT_S24_3BE \|\
d34c4307 JS	106	SNDRV_PCM_FMTBIT_S32_LE \|\
d34c4307 JS	107	SNDRV_PCM_FMTBIT_S32_BE)
33f503c9	108
f0fba2ad	109	struct snd_soc_dai_driver;
a47cbe72 MB	110	struct snd_soc_dai;
	111	struct snd_ac97_bus_ops;
	112
	113	/* Digital Audio Interface clocking API.*/
	114	int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
	115	unsigned int freq, int dir);
	116
	117	int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai,
	118	int div_id, int div);
	119
	120	int snd_soc_dai_set_pll(struct snd_soc_dai *dai,
85488037	121	int pll_id, int source, unsigned int freq_in, unsigned int freq_out);
a47cbe72	122
e54cf76b LG	123	int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio);
e54cf76b LG	124
a47cbe72 MB	125	/* Digital Audio interface formatting */
	126	int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt);
	127
	128	int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
a5479e38	129	unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width);
a47cbe72	130
472df3cb BS	131	int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai,
	132	unsigned int tx_num, unsigned int *tx_slot,
	133	unsigned int rx_num, unsigned int *rx_slot);
	134
a47cbe72 MB	135	int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate);
	136
	137	/* Digital Audio Interface mute */
da18396f MB	138	int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute,
da18396f MB	139	int direction);
a47cbe72	140
467b061f SK	141
	142	int snd_soc_dai_get_channel_map(struct snd_soc_dai *dai,
	143	unsigned int tx_num, unsigned int tx_slot,
	144	unsigned int rx_num, unsigned int rx_slot);
	145
bece9e95 LG	146	int snd_soc_dai_is_dummy(struct snd_soc_dai *dai);
bece9e95 LG	147
aa6166c2 KM	148	int snd_soc_dai_hw_params(struct snd_soc_dai *dai,
	149	struct snd_pcm_substream *substream,
	150	struct snd_pcm_hw_params *params);
846faaed KM	151	void snd_soc_dai_hw_free(struct snd_soc_dai *dai,
846faaed KM	152	struct snd_pcm_substream *substream);
5a52a045 KM	153	int snd_soc_dai_startup(struct snd_soc_dai *dai,
5a52a045 KM	154	struct snd_pcm_substream *substream);
330fcb51 KM	155	void snd_soc_dai_shutdown(struct snd_soc_dai *dai,
330fcb51 KM	156	struct snd_pcm_substream *substream);
4beb8e10 KM	157	int snd_soc_dai_prepare(struct snd_soc_dai *dai,
4beb8e10 KM	158	struct snd_pcm_substream *substream);
95aef355 KM	159	int snd_soc_dai_trigger(struct snd_soc_dai *dai,
95aef355 KM	160	struct snd_pcm_substream *substream, int cmd);
5c0769af KM	161	int snd_soc_dai_bespoke_trigger(struct snd_soc_dai *dai,
5c0769af KM	162	struct snd_pcm_substream *substream, int cmd);
1dea80d4 KM	163	snd_pcm_sframes_t snd_soc_dai_delay(struct snd_soc_dai *dai,
1dea80d4 KM	164	struct snd_pcm_substream *substream);
e0f22622	165	void snd_soc_dai_suspend(struct snd_soc_dai *dai);
aa6166c2	166
a47cbe72 MB	167	struct snd_soc_dai_ops {
	168	/*
	169	* DAI clocking configuration, all optional.
	170	* Called by soc_card drivers, normally in their hw_params.
	171	*/
	172	int (set_sysclk)(struct snd_soc_dai dai,
	173	int clk_id, unsigned int freq, int dir);
85488037 MB	174	int (set_pll)(struct snd_soc_dai dai, int pll_id, int source,
85488037 MB	175	unsigned int freq_in, unsigned int freq_out);
a47cbe72	176	int (set_clkdiv)(struct snd_soc_dai dai, int div_id, int div);
e54cf76b	177	int (set_bclk_ratio)(struct snd_soc_dai dai, unsigned int ratio);
a47cbe72 MB	178
	179	/*
	180	* DAI format configuration
	181	* Called by soc_card drivers, normally in their hw_params.
	182	*/
	183	int (set_fmt)(struct snd_soc_dai dai, unsigned int fmt);
e5c21514	184	int (*xlate_tdm_slot_mask)(unsigned int slots,
89c67857	185	unsigned int tx_mask, unsigned int rx_mask);
a47cbe72	186	int (set_tdm_slot)(struct snd_soc_dai dai,
a5479e38 DR	187	unsigned int tx_mask, unsigned int rx_mask,
a5479e38 DR	188	int slots, int slot_width);
472df3cb BS	189	int (set_channel_map)(struct snd_soc_dai dai,
	190	unsigned int tx_num, unsigned int *tx_slot,
	191	unsigned int rx_num, unsigned int *rx_slot);
467b061f SK	192	int (get_channel_map)(struct snd_soc_dai dai,
	193	unsigned int tx_num, unsigned int tx_slot,
	194	unsigned int rx_num, unsigned int rx_slot);
a47cbe72 MB	195	int (set_tristate)(struct snd_soc_dai dai, int tristate);
a47cbe72 MB	196
97349b60 SN	197	int (set_sdw_stream)(struct snd_soc_dai dai,
97349b60 SN	198	void *stream, int direction);
a47cbe72 MB	199	/*
	200	* DAI digital mute - optional.
	201	* Called by soc-core to minimise any pops.
	202	*/
	203	int (digital_mute)(struct snd_soc_dai dai, int mute);
da18396f	204	int (mute_stream)(struct snd_soc_dai dai, int mute, int stream);
dee89c4d MB	205
	206	/*
	207	* ALSA PCM audio operations - all optional.
	208	* Called by soc-core during audio PCM operations.
	209	*/
	210	int (startup)(struct snd_pcm_substream ,
	211	struct snd_soc_dai *);
	212	void (shutdown)(struct snd_pcm_substream ,
	213	struct snd_soc_dai *);
	214	int (hw_params)(struct snd_pcm_substream ,
	215	struct snd_pcm_hw_params , struct snd_soc_dai );
	216	int (hw_free)(struct snd_pcm_substream ,
	217	struct snd_soc_dai *);
	218	int (prepare)(struct snd_pcm_substream ,
	219	struct snd_soc_dai *);
9f1614aa MP	220	/*
	221	* NOTE: Commands passed to the trigger function are not necessarily
	222	* compatible with the current state of the dai. For example this
	223	* sequence of commands is possible: START STOP STOP.
	224	* So do not unconditionally use refcounting functions in the trigger
	225	* function, e.g. clk_enable/disable.
	226	*/
dee89c4d MB	227	int (trigger)(struct snd_pcm_substream , int,
dee89c4d MB	228	struct snd_soc_dai *);
07bf84aa LG	229	int (bespoke_trigger)(struct snd_pcm_substream , int,
07bf84aa LG	230	struct snd_soc_dai *);
258020d0 PU	231	/*
	232	* For hardware based FIFO caused delay reporting.
	233	* Optional.
	234	*/
	235	snd_pcm_sframes_t (delay)(struct snd_pcm_substream ,
	236	struct snd_soc_dai *);
a47cbe72 MB	237	};
a47cbe72 MB	238
2e622ae4 VK	239	struct snd_soc_cdai_ops {
	240	/*
	241	* for compress ops
	242	*/
	243	int (startup)(struct snd_compr_stream ,
	244	struct snd_soc_dai *);
	245	int (shutdown)(struct snd_compr_stream ,
	246	struct snd_soc_dai *);
	247	int (set_params)(struct snd_compr_stream ,
	248	struct snd_compr_params , struct snd_soc_dai );
	249	int (get_params)(struct snd_compr_stream ,
	250	struct snd_codec , struct snd_soc_dai );
	251	int (set_metadata)(struct snd_compr_stream ,
	252	struct snd_compr_metadata , struct snd_soc_dai );
	253	int (get_metadata)(struct snd_compr_stream ,
	254	struct snd_compr_metadata , struct snd_soc_dai );
	255	int (trigger)(struct snd_compr_stream , int,
	256	struct snd_soc_dai *);
	257	int (pointer)(struct snd_compr_stream ,
	258	struct snd_compr_tstamp , struct snd_soc_dai );
	259	int (ack)(struct snd_compr_stream , size_t,
	260	struct snd_soc_dai *);
	261	};
	262
a47cbe72	263	/*
f0fba2ad	264	* Digital Audio Interface Driver.
a47cbe72	265	*
f0fba2ad LG	266	* Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97
	267	* operations and capabilities. Codec and platform drivers will register this
	268	* structure for every DAI they have.
	269	*
	270	* This structure covers the clocking, formating and ALSA operations for each
	271	* interface.
a47cbe72	272	*/
f0fba2ad	273	struct snd_soc_dai_driver {
a47cbe72	274	/* DAI description */
f0fba2ad	275	const char *name;
a47cbe72	276	unsigned int id;
62368294	277	unsigned int base;
68003e6c	278	struct snd_soc_dobj dobj;
a47cbe72	279
f0fba2ad LG	280	/* DAI driver callbacks */
	281	int (probe)(struct snd_soc_dai dai);
	282	int (remove)(struct snd_soc_dai dai);
dc7d7b83 MB	283	int (suspend)(struct snd_soc_dai dai);
dc7d7b83 MB	284	int (resume)(struct snd_soc_dai dai);
49681077	285	/* compress dai */
6f0c4226	286	int (compress_new)(struct snd_soc_pcm_runtime rtd, int num);
25f7b701 AP	287	/* Optional Callback used at pcm creation*/
	288	int (pcm_new)(struct snd_soc_pcm_runtime rtd,
	289	struct snd_soc_dai *dai);
bc263214 LPC	290	/* DAI is also used for the control bus */
bc263214 LPC	291	bool bus_control;
a47cbe72 MB	292
a47cbe72 MB	293	/* ops */
1ee46ebd	294	const struct snd_soc_dai_ops *ops;
2e622ae4	295	const struct snd_soc_cdai_ops *cops;
a47cbe72 MB	296
	297	/* DAI capabilities */
	298	struct snd_soc_pcm_stream capture;
	299	struct snd_soc_pcm_stream playback;
06f409d7	300	unsigned int symmetric_rates:1;
3635bf09 NC	301	unsigned int symmetric_channels:1;
3635bf09 NC	302	unsigned int symmetric_samplebits:1;
0168bf0d LG	303
	304	/* probe ordering - for components with runtime dependencies */
	305	int probe_order;
	306	int remove_order;
f0fba2ad LG	307	};
	308
	309	/*
	310	* Digital Audio Interface runtime data.
	311	*
	312	* Holds runtime data for a DAI.
	313	*/
	314	struct snd_soc_dai {
	315	const char *name;
	316	int id;
	317	struct device *dev;
f0fba2ad LG	318
	319	/* driver ops */
	320	struct snd_soc_dai_driver *driver;
a47cbe72 MB	321
a47cbe72 MB	322	/* DAI runtime info */
40c57963 JB	323	unsigned int capture_active; /* stream usage count */
40c57963 JB	324	unsigned int playback_active; /* stream usage count */
39b2238e KM	325	unsigned int probed:1;
39b2238e KM	326
a47cbe72	327	unsigned int active;
a47cbe72	328
888df395 MB	329	struct snd_soc_dapm_widget *playback_widget;
	330	struct snd_soc_dapm_widget *capture_widget;
	331
f0fba2ad LG	332	/* DAI DMA data */
	333	void *playback_dma_data;
	334	void *capture_dma_data;
a47cbe72	335
17841020 DA	336	/* Symmetry data - only valid if symmetry is being enforced */
17841020 DA	337	unsigned int rate;
3635bf09 NC	338	unsigned int channels;
3635bf09 NC	339	unsigned int sample_bits;
17841020	340
f0fba2ad	341	/* parent platform/codec */
6106d129	342	struct snd_soc_component *component;
2466ab97	343
88bd870f BC	344	/* CODEC TDM slot masks and params (for fixup) */
	345	unsigned int tx_mask;
	346	unsigned int rx_mask;
	347
a47cbe72 MB	348	struct list_head list;
	349	};
	350
fd23b7de DM	351	static inline void snd_soc_dai_get_dma_data(const struct snd_soc_dai dai,
	352	const struct snd_pcm_substream *ss)
	353	{
	354	return (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
f0fba2ad	355	dai->playback_dma_data : dai->capture_dma_data;
fd23b7de DM	356	}
	357
	358	static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai,
	359	const struct snd_pcm_substream *ss,
	360	void *data)
	361	{
	362	if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK)
f0fba2ad	363	dai->playback_dma_data = data;
fd23b7de	364	else
f0fba2ad LG	365	dai->capture_dma_data = data;
	366	}
	367
ecfc0c04 MB	368	static inline void snd_soc_dai_init_dma_data(struct snd_soc_dai *dai,
	369	void playback, void capture)
	370	{
	371	dai->playback_dma_data = playback;
	372	dai->capture_dma_data = capture;
	373	}
	374
f0fba2ad LG	375	static inline void snd_soc_dai_set_drvdata(struct snd_soc_dai *dai,
	376	void *data)
	377	{
	378	dev_set_drvdata(dai->dev, data);
	379	}
	380
	381	static inline void snd_soc_dai_get_drvdata(struct snd_soc_dai dai)
	382	{
	383	return dev_get_drvdata(dai->dev);
fd23b7de DM	384	}
fd23b7de DM	385
97349b60 SN	386	/**
	387	* snd_soc_dai_set_sdw_stream() - Configures a DAI for SDW stream operation
	388	* @dai: DAI
	389	* @stream: STREAM
	390	* @direction: Stream direction(Playback/Capture)
	391	* SoundWire subsystem doesn't have a notion of direction and we reuse
	392	* the ASoC stream direction to configure sink/source ports.
	393	* Playback maps to source ports and Capture for sink ports.
	394	*
	395	* This should be invoked with NULL to clear the stream set previously.
	396	* Returns 0 on success, a negative error code otherwise.
	397	*/
	398	static inline int snd_soc_dai_set_sdw_stream(struct snd_soc_dai *dai,
	399	void *stream, int direction)
	400	{
	401	if (dai->driver->ops->set_sdw_stream)
	402	return dai->driver->ops->set_sdw_stream(dai, stream, direction);
	403	else
	404	return -ENOTSUPP;
	405	}
	406
a47cbe72	407	#endif