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linux-stable/include/sound/soc-dai.h
Kuninori Morimoto 6b8ac43c33
ASoC: soc-dai: use bit field for bus_control 
.bus_control can be bit field.
this patch do it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/87v9uszazh.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
2019-08-22 18:30:26 +01:00

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/* 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);
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
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