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Name	Size	Permission
ac97	[ DIR ]	drwxr-xr-x
sof	[ DIR ]	drwxr-xr-x
ac97_codec.h	15.65 KB	-rw-r--r--
aci.h	2.4 KB	-rw-r--r--
ad1816a.h	4.73 KB	-rw-r--r--
ad1843.h	1.48 KB	-rw-r--r--
adau1373.h	706 B	-rw-r--r--
aess.h	1.63 KB	-rw-r--r--
ak4113.h	10.22 KB	-rw-r--r--
ak4114.h	9.54 KB	-rw-r--r--
ak4117.h	8.34 KB	-rw-r--r--
ak4531_codec.h	2.42 KB	-rw-r--r--
ak4641.h	476 B	-rw-r--r--
ak4xxx-adda.h	2.66 KB	-rw-r--r--
alc5623.h	536 B	-rw-r--r--
asequencer.h	2.91 KB	-rw-r--r--
asound.h	590 B	-rw-r--r--
asoundef.h	16.02 KB	-rw-r--r--
compress_driver.h	6.72 KB	-rw-r--r--
control.h	8.17 KB	-rw-r--r--
core.h	13.51 KB	-rw-r--r--
cs35l33.h	888 B	-rw-r--r--
cs35l34.h	741 B	-rw-r--r--
cs35l35.h	2.21 KB	-rw-r--r--
cs35l36.h	772 B	-rw-r--r--
cs4231-regs.h	7.6 KB	-rw-r--r--
cs4271.h	961 B	-rw-r--r--
cs42l52.h	592 B	-rw-r--r--
cs42l56.h	1.02 KB	-rw-r--r--
cs42l73.h	361 B	-rw-r--r--
cs8403.h	7.95 KB	-rw-r--r--
cs8427.h	9.72 KB	-rw-r--r--
da7213.h	1.01 KB	-rw-r--r--
da7218.h	2.41 KB	-rw-r--r--
da7219-aad.h	2.21 KB	-rw-r--r--
da7219.h	998 B	-rw-r--r--
da9055.h	707 B	-rw-r--r--
designware_i2s.h	1.59 KB	-rw-r--r--
dmaengine_pcm.h	5.97 KB	-rw-r--r--
emu10k1.h	88.67 KB	-rw-r--r--
emu10k1_synth.h	693 B	-rw-r--r--
emu8000.h	3.34 KB	-rw-r--r--
emu8000_reg.h	9.54 KB	-rw-r--r--
emux_legacy.h	4.7 KB	-rw-r--r--
emux_synth.h	6.79 KB	-rw-r--r--
es1688.h	2.85 KB	-rw-r--r--
gus.h	19.53 KB	-rw-r--r--
hda_chmap.h	2.6 KB	-rw-r--r--
hda_codec.h	17.14 KB	-rw-r--r--
hda_component.h	2.04 KB	-rw-r--r--
hda_hwdep.h	736 B	-rw-r--r--
hda_i915.h	589 B	-rw-r--r--
hda_register.h	10.21 KB	-rw-r--r--
hda_regmap.h	6.77 KB	-rw-r--r--
hda_verbs.h	16.89 KB	-rw-r--r--
hdaudio.h	20.93 KB	-rw-r--r--
hdaudio_ext.h	5.46 KB	-rw-r--r--
hdmi-codec.h	2.47 KB	-rw-r--r--
hwdep.h	1.88 KB	-rw-r--r--
i2c.h	2.79 KB	-rw-r--r--
info.h	7.86 KB	-rw-r--r--
initval.h	2.41 KB	-rw-r--r--
intel-dsp-config.h	639 B	-rw-r--r--
intel-nhlt.h	2.79 KB	-rw-r--r--
jack.h	3.08 KB	-rw-r--r--
l3.h	524 B	-rw-r--r--
madera-pdata.h	1.95 KB	-rw-r--r--
max9768.h	729 B	-rw-r--r--
max98088.h	1.05 KB	-rw-r--r--
max98090.h	534 B	-rw-r--r--
max98095.h	1.33 KB	-rw-r--r--
memalloc.h	3.72 KB	-rw-r--r--
minors.h	3.65 KB	-rw-r--r--
mixer_oss.h	1.73 KB	-rw-r--r--
mpu401.h	3.81 KB	-rw-r--r--
omap-hdmi-audio.h	970 B	-rw-r--r--
opl3.h	11.63 KB	-rw-r--r--
opl4.h	459 B	-rw-r--r--
pcm-indirect.h	5.04 KB	-rw-r--r--
pcm.h	47.05 KB	-rw-r--r--
pcm_drm_eld.h	183 B	-rw-r--r--
pcm_iec958.h	332 B	-rw-r--r--
pcm_oss.h	2.01 KB	-rw-r--r--
pcm_params.h	8.53 KB	-rw-r--r--
pt2258.h	513 B	-rw-r--r--
pxa2xx-lib.h	1.6 KB	-rw-r--r--
rawmidi.h	5.56 KB	-rw-r--r--
rt286.h	314 B	-rw-r--r--
rt298.h	373 B	-rw-r--r--
rt5514.h	399 B	-rw-r--r--
rt5645.h	665 B	-rw-r--r--
rt5659.h	880 B	-rw-r--r--
rt5660.h	578 B	-rw-r--r--
rt5663.h	476 B	-rw-r--r--
rt5665.h	761 B	-rw-r--r--
rt5668.h	646 B	-rw-r--r--
rt5670.h	533 B	-rw-r--r--
rt5682.h	646 B	-rw-r--r--
s3c24xx_uda134x.h	229 B	-rw-r--r--
sb.h	10.3 KB	-rw-r--r--
sb16_csp.h	2.05 KB	-rw-r--r--
seq_device.h	2.11 KB	-rw-r--r--
seq_kernel.h	3.3 KB	-rw-r--r--
seq_midi_emul.h	6.59 KB	-rw-r--r--
seq_midi_event.h	1.32 KB	-rw-r--r--
seq_oss.h	2.21 KB	-rw-r--r--
seq_oss_legacy.h	360 B	-rw-r--r--
seq_virmidi.h	2.07 KB	-rw-r--r--
sh_dac_audio.h	441 B	-rw-r--r--
sh_fsi.h	693 B	-rw-r--r--
simple_card.h	529 B	-rw-r--r--
simple_card_utils.h	6.56 KB	-rw-r--r--
snd_wavefront.h	5.57 KB	-rw-r--r--
soc-acpi-intel-match.h	1.55 KB	-rw-r--r--
soc-acpi.h	3.55 KB	-rw-r--r--
soc-component.h	13.34 KB	-rw-r--r--
soc-dai.h	13.17 KB	-rw-r--r--
soc-dapm.h	31.27 KB	-rw-r--r--
soc-dpcm.h	5.19 KB	-rw-r--r--
soc-topology.h	6.08 KB	-rw-r--r--
soc.h	47.42 KB	-rw-r--r--
sof.h	2.43 KB	-rw-r--r--
soundfont.h	3.81 KB	-rw-r--r--
spear_dma.h	350 B	-rw-r--r--
spear_spdif.h	345 B	-rw-r--r--
sta32x.h	1015 B	-rw-r--r--
sta350.h	1.44 KB	-rw-r--r--
tas2552-plat.h	283 B	-rw-r--r--
tas5086.h	210 B	-rw-r--r--
tea6330t.h	468 B	-rw-r--r--
timer.h	4.88 KB	-rw-r--r--
tlv.h	1.59 KB	-rw-r--r--
tlv320aic32x4.h	1.29 KB	-rw-r--r--
tlv320aic3x.h	1.74 KB	-rw-r--r--
tlv320dac33-plat.h	574 B	-rw-r--r--
tpa6130a2-plat.h	291 B	-rw-r--r--
uda134x.h	451 B	-rw-r--r--
uda1380.h	335 B	-rw-r--r--
util_mem.h	1.58 KB	-rw-r--r--
vx_core.h	14.67 KB	-rw-r--r--
wavefront.h	18.08 KB	-rw-r--r--
wm0010.h	466 B	-rw-r--r--
wm1250-ev1.h	510 B	-rw-r--r--
wm2000.h	479 B	-rw-r--r--
wm2200.h	1.32 KB	-rw-r--r--
wm5100.h	1.1 KB	-rw-r--r--
wm8903.h	15.02 KB	-rw-r--r--
wm8904.h	7.21 KB	-rw-r--r--
wm8955.h	442 B	-rw-r--r--
wm8960.h	457 B	-rw-r--r--
wm8962.h	1.65 KB	-rw-r--r--
wm8993.h	1.05 KB	-rw-r--r--
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wm9081.h	515 B	-rw-r--r--
wm9090.h	634 B	-rw-r--r--
wss.h	7.76 KB	-rw-r--r--

Code Editor : 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

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);
void snd_soc_dai_resume(struct snd_soc_dai *dai);
int snd_soc_dai_probe(struct snd_soc_dai *dai);
int snd_soc_dai_remove(struct snd_soc_dai *dai);
int snd_soc_dai_compress_new(struct snd_soc_dai *dai,
			     struct snd_soc_pcm_runtime *rtd, int num);
bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int stream);

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);

/* 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;
	unsigned int bus_control:1; /* DAI is also used for the control bus */

/* 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

${this.title}

Code Editor : soc-dai.h