linux/include/sound/soc-dai.h
Kuninori Morimoto 3653480c68
ASoC: soc-dai.h: cleanup Playback/Capture data for snd_soc_dai
Current snd_soc_dai has data for Playback/Capture, but it is very
random. Someone is array (A), someone is playback/capture (B),
and someone is tx/rx (C);

	struct snd_soc_dai {
		...
(A)		unsigned int stream_active[SNDRV_PCM_STREAM_LAST + 1];

(B)		struct snd_soc_dapm_widget *playback_widget;
(B)		struct snd_soc_dapm_widget *capture_widget;

(B)		void *playback_dma_data;
(B)		void *capture_dma_data;

		...

(C)		unsigned int tx_mask;
(C)		unsigned int rx_mask;
	};

Because of it, the code was very complicated.
This patch creates new data structure to merge these into one,
and tidyup the code.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/87cz6vea1v.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
2023-01-31 11:05:12 +00:00

587 lines
21 KiB
C

/* 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
/* Describes the possible PCM format */
/*
* use SND_SOC_DAI_FORMAT_xx as eash shift.
* see
* snd_soc_runtime_get_dai_fmt()
*/
#define SND_SOC_POSSIBLE_DAIFMT_FORMAT_SHIFT 0
#define SND_SOC_POSSIBLE_DAIFMT_FORMAT_MASK (0xFFFF << SND_SOC_POSSIBLE_DAIFMT_FORMAT_SHIFT)
#define SND_SOC_POSSIBLE_DAIFMT_I2S (1 << SND_SOC_DAI_FORMAT_I2S)
#define SND_SOC_POSSIBLE_DAIFMT_RIGHT_J (1 << SND_SOC_DAI_FORMAT_RIGHT_J)
#define SND_SOC_POSSIBLE_DAIFMT_LEFT_J (1 << SND_SOC_DAI_FORMAT_LEFT_J)
#define SND_SOC_POSSIBLE_DAIFMT_DSP_A (1 << SND_SOC_DAI_FORMAT_DSP_A)
#define SND_SOC_POSSIBLE_DAIFMT_DSP_B (1 << SND_SOC_DAI_FORMAT_DSP_B)
#define SND_SOC_POSSIBLE_DAIFMT_AC97 (1 << SND_SOC_DAI_FORMAT_AC97)
#define SND_SOC_POSSIBLE_DAIFMT_PDM (1 << SND_SOC_DAI_FORMAT_PDM)
/*
* DAI Clock gating.
*
* DAI bit clocks can 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 */
/* Describes the possible PCM format */
/*
* define GATED -> CONT. GATED will be selected if both are selected.
* see
* snd_soc_runtime_get_dai_fmt()
*/
#define SND_SOC_POSSIBLE_DAIFMT_CLOCK_SHIFT 16
#define SND_SOC_POSSIBLE_DAIFMT_CLOCK_MASK (0xFFFF << SND_SOC_POSSIBLE_DAIFMT_CLOCK_SHIFT)
#define SND_SOC_POSSIBLE_DAIFMT_GATED (0x1ULL << SND_SOC_POSSIBLE_DAIFMT_CLOCK_SHIFT)
#define SND_SOC_POSSIBLE_DAIFMT_CONT (0x2ULL << SND_SOC_POSSIBLE_DAIFMT_CLOCK_SHIFT)
/*
* 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 */
/* Describes the possible PCM format */
#define SND_SOC_POSSIBLE_DAIFMT_INV_SHIFT 32
#define SND_SOC_POSSIBLE_DAIFMT_INV_MASK (0xFFFFULL << SND_SOC_POSSIBLE_DAIFMT_INV_SHIFT)
#define SND_SOC_POSSIBLE_DAIFMT_NB_NF (0x1ULL << SND_SOC_POSSIBLE_DAIFMT_INV_SHIFT)
#define SND_SOC_POSSIBLE_DAIFMT_NB_IF (0x2ULL << SND_SOC_POSSIBLE_DAIFMT_INV_SHIFT)
#define SND_SOC_POSSIBLE_DAIFMT_IB_NF (0x4ULL << SND_SOC_POSSIBLE_DAIFMT_INV_SHIFT)
#define SND_SOC_POSSIBLE_DAIFMT_IB_IF (0x8ULL << SND_SOC_POSSIBLE_DAIFMT_INV_SHIFT)
/*
* DAI hardware clock providers/consumers
*
* This is wrt the codec, the inverse is true for the interface
* i.e. if the codec is clk and FRM provider then the interface is
* clk and frame consumer.
*/
#define SND_SOC_DAIFMT_CBP_CFP (1 << 12) /* codec clk provider & frame provider */
#define SND_SOC_DAIFMT_CBC_CFP (2 << 12) /* codec clk consumer & frame provider */
#define SND_SOC_DAIFMT_CBP_CFC (3 << 12) /* codec clk provider & frame consumer */
#define SND_SOC_DAIFMT_CBC_CFC (4 << 12) /* codec clk consumer & frame consumer */
/* previous definitions kept for backwards-compatibility, do not use in new contributions */
#define SND_SOC_DAIFMT_CBM_CFM SND_SOC_DAIFMT_CBP_CFP
#define SND_SOC_DAIFMT_CBS_CFM SND_SOC_DAIFMT_CBC_CFP
#define SND_SOC_DAIFMT_CBM_CFS SND_SOC_DAIFMT_CBP_CFC
#define SND_SOC_DAIFMT_CBS_CFS SND_SOC_DAIFMT_CBC_CFC
/* when passed to set_fmt directly indicate if the device is provider or consumer */
#define SND_SOC_DAIFMT_BP_FP SND_SOC_DAIFMT_CBP_CFP
#define SND_SOC_DAIFMT_BC_FP SND_SOC_DAIFMT_CBC_CFP
#define SND_SOC_DAIFMT_BP_FC SND_SOC_DAIFMT_CBP_CFC
#define SND_SOC_DAIFMT_BC_FC SND_SOC_DAIFMT_CBC_CFC
/* Describes the possible PCM format */
#define SND_SOC_POSSIBLE_DAIFMT_CLOCK_PROVIDER_SHIFT 48
#define SND_SOC_POSSIBLE_DAIFMT_CLOCK_PROVIDER_MASK (0xFFFFULL << SND_SOC_POSSIBLE_DAIFMT_CLOCK_PROVIDER_SHIFT)
#define SND_SOC_POSSIBLE_DAIFMT_CBP_CFP (0x1ULL << SND_SOC_POSSIBLE_DAIFMT_CLOCK_PROVIDER_SHIFT)
#define SND_SOC_POSSIBLE_DAIFMT_CBC_CFP (0x2ULL << SND_SOC_POSSIBLE_DAIFMT_CLOCK_PROVIDER_SHIFT)
#define SND_SOC_POSSIBLE_DAIFMT_CBP_CFC (0x4ULL << SND_SOC_POSSIBLE_DAIFMT_CLOCK_PROVIDER_SHIFT)
#define SND_SOC_POSSIBLE_DAIFMT_CBC_CFC (0x8ULL << SND_SOC_POSSIBLE_DAIFMT_CLOCK_PROVIDER_SHIFT)
#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_CLOCK_PROVIDER_MASK 0xf000
#define SND_SOC_DAIFMT_MASTER_MASK SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK
/*
* 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_get_fmt_max_priority(struct snd_soc_pcm_runtime *rtd);
u64 snd_soc_dai_get_fmt(struct snd_soc_dai *dai, int priority);
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 rollback);
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 rollback);
void snd_soc_dai_suspend(struct snd_soc_dai *dai);
void snd_soc_dai_resume(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);
void snd_soc_dai_link_set_capabilities(struct snd_soc_dai_link *dai_link);
void snd_soc_dai_action(struct snd_soc_dai *dai,
int stream, int action);
static inline void snd_soc_dai_activate(struct snd_soc_dai *dai,
int stream)
{
snd_soc_dai_action(dai, stream, 1);
}
static inline void snd_soc_dai_deactivate(struct snd_soc_dai *dai,
int stream)
{
snd_soc_dai_action(dai, stream, -1);
}
int snd_soc_dai_active(struct snd_soc_dai *dai);
int snd_soc_pcm_dai_probe(struct snd_soc_pcm_runtime *rtd, int order);
int snd_soc_pcm_dai_remove(struct snd_soc_pcm_runtime *rtd, int order);
int snd_soc_pcm_dai_new(struct snd_soc_pcm_runtime *rtd);
int snd_soc_pcm_dai_prepare(struct snd_pcm_substream *substream);
int snd_soc_pcm_dai_trigger(struct snd_pcm_substream *substream, int cmd,
int rollback);
int snd_soc_pcm_dai_bespoke_trigger(struct snd_pcm_substream *substream,
int cmd);
void snd_soc_pcm_dai_delay(struct snd_pcm_substream *substream,
snd_pcm_sframes_t *cpu_delay, snd_pcm_sframes_t *codec_delay);
int snd_soc_dai_compr_startup(struct snd_soc_dai *dai,
struct snd_compr_stream *cstream);
void snd_soc_dai_compr_shutdown(struct snd_soc_dai *dai,
struct snd_compr_stream *cstream,
int rollback);
int snd_soc_dai_compr_trigger(struct snd_soc_dai *dai,
struct snd_compr_stream *cstream, int cmd);
int snd_soc_dai_compr_set_params(struct snd_soc_dai *dai,
struct snd_compr_stream *cstream,
struct snd_compr_params *params);
int snd_soc_dai_compr_get_params(struct snd_soc_dai *dai,
struct snd_compr_stream *cstream,
struct snd_codec *params);
int snd_soc_dai_compr_ack(struct snd_soc_dai *dai,
struct snd_compr_stream *cstream,
size_t bytes);
int snd_soc_dai_compr_pointer(struct snd_soc_dai *dai,
struct snd_compr_stream *cstream,
struct snd_compr_tstamp *tstamp);
int snd_soc_dai_compr_set_metadata(struct snd_soc_dai *dai,
struct snd_compr_stream *cstream,
struct snd_compr_metadata *metadata);
int snd_soc_dai_compr_get_metadata(struct snd_soc_dai *dai,
struct snd_compr_stream *cstream,
struct snd_compr_metadata *metadata);
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_stream)(struct snd_soc_dai *dai,
void *stream, int direction);
void *(*get_stream)(struct snd_soc_dai *dai, int direction);
/*
* DAI digital mute - optional.
* Called by soc-core to minimise any pops.
*/
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 *);
/*
* Format list for auto selection.
* Format will be increased if priority format was
* not selected.
* see
* snd_soc_dai_get_fmt()
*/
u64 *auto_selectable_formats;
int num_auto_selectable_formats;
/* bit field */
unsigned int no_capture_mute:1;
};
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);
/* 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_rate:1;
unsigned int symmetric_channels:1;
unsigned int symmetric_sample_bits:1;
/* probe ordering - for components with runtime dependencies */
int probe_order;
int remove_order;
};
/* for Playback/Capture */
struct snd_soc_dai_stream {
struct snd_soc_dapm_widget *widget;
unsigned int active; /* usage count */
unsigned int tdm_mask; /* CODEC TDM slot masks and params (for fixup) */
void *dma_data; /* DAI DMA data */
};
/*
* 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 */
struct snd_soc_dai_stream stream[SNDRV_PCM_STREAM_LAST + 1];
/* 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;
struct list_head list;
/* function mark */
struct snd_pcm_substream *mark_startup;
struct snd_pcm_substream *mark_hw_params;
struct snd_pcm_substream *mark_trigger;
struct snd_compr_stream *mark_compr_startup;
/* bit field */
unsigned int probed:1;
};
static inline struct snd_soc_pcm_stream *
snd_soc_dai_get_pcm_stream(const struct snd_soc_dai *dai, int stream)
{
return (stream == SNDRV_PCM_STREAM_PLAYBACK) ?
&dai->driver->playback : &dai->driver->capture;
}
#define snd_soc_dai_get_widget_playback(dai) snd_soc_dai_get_widget(dai, SNDRV_PCM_STREAM_PLAYBACK)
#define snd_soc_dai_get_widget_capture(dai) snd_soc_dai_get_widget(dai, SNDRV_PCM_STREAM_CAPTURE)
static inline
struct snd_soc_dapm_widget *snd_soc_dai_get_widget(struct snd_soc_dai *dai, int stream)
{
return dai->stream[stream].widget;
}
#define snd_soc_dai_set_widget_playback(dai, widget) snd_soc_dai_set_widget(dai, SNDRV_PCM_STREAM_PLAYBACK, widget)
#define snd_soc_dai_set_widget_capture(dai, widget) snd_soc_dai_set_widget(dai, SNDRV_PCM_STREAM_CAPTURE, widget)
static inline
void snd_soc_dai_set_widget(struct snd_soc_dai *dai, int stream, struct snd_soc_dapm_widget *widget)
{
dai->stream[stream].widget = widget;
}
#define snd_soc_dai_dma_data_get_playback(dai) snd_soc_dai_dma_data_get(dai, SNDRV_PCM_STREAM_PLAYBACK)
#define snd_soc_dai_dma_data_get_capture(dai) snd_soc_dai_dma_data_get(dai, SNDRV_PCM_STREAM_CAPTURE)
#define snd_soc_dai_get_dma_data(dai, ss) snd_soc_dai_dma_data_get(dai, ss->stream)
static inline void *snd_soc_dai_dma_data_get(const struct snd_soc_dai *dai, int stream)
{
return dai->stream[stream].dma_data;
}
#define snd_soc_dai_dma_data_set_playback(dai, data) snd_soc_dai_dma_data_set(dai, SNDRV_PCM_STREAM_PLAYBACK, data)
#define snd_soc_dai_dma_data_set_capture(dai, data) snd_soc_dai_dma_data_set(dai, SNDRV_PCM_STREAM_CAPTURE, data)
#define snd_soc_dai_set_dma_data(dai, ss, data) snd_soc_dai_dma_data_set(dai, ss->stream, data)
static inline void snd_soc_dai_dma_data_set(struct snd_soc_dai *dai, int stream, void *data)
{
dai->stream[stream].dma_data = data;
}
static inline void snd_soc_dai_init_dma_data(struct snd_soc_dai *dai, void *playback, void *capture)
{
snd_soc_dai_dma_data_set_playback(dai, playback);
snd_soc_dai_dma_data_set_capture(dai, capture);
}
static inline unsigned int snd_soc_dai_tdm_mask_get(struct snd_soc_dai *dai, int stream)
{
return dai->stream[stream].tdm_mask;
}
static inline void snd_soc_dai_tdm_mask_set(struct snd_soc_dai *dai, int stream,
unsigned int tdm_mask)
{
dai->stream[stream].tdm_mask = tdm_mask;
}
static inline unsigned int snd_soc_dai_stream_active(struct snd_soc_dai *dai, int stream)
{
/* see snd_soc_dai_action() for setup */
return dai->stream[stream].active;
}
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_stream() - Configures a DAI for stream operation
* @dai: DAI
* @stream: STREAM (opaque structure depending on DAI type)
* @direction: Stream direction(Playback/Capture)
* Some subsystems, such as SoundWire, don'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_stream(struct snd_soc_dai *dai,
void *stream, int direction)
{
if (dai->driver->ops->set_stream)
return dai->driver->ops->set_stream(dai, stream, direction);
else
return -ENOTSUPP;
}
/**
* snd_soc_dai_get_stream() - Retrieves stream from DAI
* @dai: DAI
* @direction: Stream direction(Playback/Capture)
*
* This routine only retrieves that was previously configured
* with snd_soc_dai_get_stream()
*
* Returns pointer to stream or an ERR_PTR value, e.g.
* ERR_PTR(-ENOTSUPP) if callback is not supported;
*/
static inline void *snd_soc_dai_get_stream(struct snd_soc_dai *dai,
int direction)
{
if (dai->driver->ops->get_stream)
return dai->driver->ops->get_stream(dai, direction);
else
return ERR_PTR(-ENOTSUPP);
}
#endif