362 строки
11 KiB
C
362 строки
11 KiB
C
/*
|
|
* linux/sound/soc-dai.h -- ALSA SoC Layer
|
|
*
|
|
* Copyright: 2005-2008 Wolfson Microelectronics. PLC.
|
|
*
|
|
* This program is free software; you can redistribute it and/or modify
|
|
* it under the terms of the GNU General Public License version 2 as
|
|
* published by the Free Software Foundation.
|
|
*
|
|
* 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_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 (*set_tristate)(struct snd_soc_dai *dai, int tristate);
|
|
|
|
/*
|
|
* 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:1; /* stream is in use */
|
|
unsigned int playback_active:1; /* stream is in use */
|
|
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_codec *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);
|
|
}
|
|
|
|
#endif
|