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android_kernel_xiaomi_sm8350/include/sound/soc.h
Liam Girdwood 8c6529dbf8 ALSA: asoc: core - add Digital Audio Interface (DAI) control functions. 
This patch adds several functions for DAI control and config
and replaces the current method of calling function pointers within
the DAI struct.

Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-07-10 09:32:50 +02:00

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/*
* linux/sound/soc.h -- ALSA SoC Layer
*
* Author: Liam Girdwood
* Created: Aug 11th 2005
* Copyright: 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.
*/
#ifndef __LINUX_SND_SOC_H
#define __LINUX_SND_SOC_H
#include <linux/platform_device.h>
#include <linux/types.h>
#include <linux/workqueue.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/control.h>
#include <sound/ac97_codec.h>
#define SND_SOC_VERSION "0.13.2"
/*
* Convenience kcontrol builders
*/
#define SOC_SINGLE_VALUE(reg, shift, max, invert) ((reg) | ((shift) << 8) |\
((shift) << 12) | ((max) << 16) | ((invert) << 24))
#define SOC_SINGLE_VALUE_EXT(reg, max, invert) ((reg) | ((max) << 16) |\
((invert) << 31))
#define SOC_SINGLE(xname, reg, shift, max, invert) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
.info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\
.put = snd_soc_put_volsw, \
.private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) }
#define SOC_SINGLE_TLV(xname, reg, shift, max, invert, tlv_array) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
SNDRV_CTL_ELEM_ACCESS_READWRITE,\
.tlv.p = (tlv_array), \
.info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\
.put = snd_soc_put_volsw, \
.private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) }
#define SOC_DOUBLE(xname, reg, shift_left, shift_right, max, invert) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
.info = snd_soc_info_volsw, .get = snd_soc_get_volsw, \
.put = snd_soc_put_volsw, \
.private_value = (reg) | ((shift_left) << 8) | \
((shift_right) << 12) | ((max) << 16) | ((invert) << 24) }
#define SOC_DOUBLE_R(xname, reg_left, reg_right, shift, max, invert) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
.info = snd_soc_info_volsw_2r, \
.get = snd_soc_get_volsw_2r, .put = snd_soc_put_volsw_2r, \
.private_value = (reg_left) | ((shift) << 8) | \
((max) << 12) | ((invert) << 20) | ((reg_right) << 24) }
#define SOC_DOUBLE_TLV(xname, reg, shift_left, shift_right, max, invert, tlv_array) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
SNDRV_CTL_ELEM_ACCESS_READWRITE,\
.tlv.p = (tlv_array), \
.info = snd_soc_info_volsw, .get = snd_soc_get_volsw, \
.put = snd_soc_put_volsw, \
.private_value = (reg) | ((shift_left) << 8) | \
((shift_right) << 12) | ((max) << 16) | ((invert) << 24) }
#define SOC_DOUBLE_R_TLV(xname, reg_left, reg_right, shift, max, invert, tlv_array) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
SNDRV_CTL_ELEM_ACCESS_READWRITE,\
.tlv.p = (tlv_array), \
.info = snd_soc_info_volsw_2r, \
.get = snd_soc_get_volsw_2r, .put = snd_soc_put_volsw_2r, \
.private_value = (reg_left) | ((shift) << 8) | \
((max) << 12) | ((invert) << 20) | ((reg_right) << 24) }
#define SOC_DOUBLE_S8_TLV(xname, reg, min, max, tlv_array) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \
SNDRV_CTL_ELEM_ACCESS_READWRITE, \
.tlv.p = (tlv_array), \
.info = snd_soc_info_volsw_s8, .get = snd_soc_get_volsw_s8, \
.put = snd_soc_put_volsw_s8, \
.private_value = (reg) | (((signed char)max) << 16) | \
(((signed char)min) << 24) }
#define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xtexts) \
{ .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \
.mask = xmask, .texts = xtexts }
#define SOC_ENUM_SINGLE(xreg, xshift, xmask, xtexts) \
SOC_ENUM_DOUBLE(xreg, xshift, xshift, xmask, xtexts)
#define SOC_ENUM_SINGLE_EXT(xmask, xtexts) \
{ .mask = xmask, .texts = xtexts }
#define SOC_ENUM(xname, xenum) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname,\
.info = snd_soc_info_enum_double, \
.get = snd_soc_get_enum_double, .put = snd_soc_put_enum_double, \
.private_value = (unsigned long)&xenum }
#define SOC_SINGLE_EXT(xname, xreg, xshift, xmask, xinvert,\
xhandler_get, xhandler_put) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
.info = snd_soc_info_volsw, \
.get = xhandler_get, .put = xhandler_put, \
.private_value = SOC_SINGLE_VALUE(xreg, xshift, xmask, xinvert) }
#define SOC_SINGLE_EXT_TLV(xname, xreg, xshift, xmask, xinvert,\
xhandler_get, xhandler_put, tlv_array) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
SNDRV_CTL_ELEM_ACCESS_READWRITE,\
.tlv.p = (tlv_array), \
.info = snd_soc_info_volsw, \
.get = xhandler_get, .put = xhandler_put, \
.private_value = SOC_SINGLE_VALUE(xreg, xshift, xmask, xinvert) }
#define SOC_SINGLE_BOOL_EXT(xname, xdata, xhandler_get, xhandler_put) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
.info = snd_soc_info_bool_ext, \
.get = xhandler_get, .put = xhandler_put, \
.private_value = xdata }
#define SOC_ENUM_EXT(xname, xenum, xhandler_get, xhandler_put) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
.info = snd_soc_info_enum_ext, \
.get = xhandler_get, .put = xhandler_put, \
.private_value = (unsigned long)&xenum }
/*
* Bias levels
*
* @ON: Bias is fully on for audio playback and capture operations.
* @PREPARE: Prepare for audio operations. Called before DAPM switching for
* stream start and stop operations.
* @STANDBY: Low power standby state when no playback/capture operations are
* in progress. NOTE: The transition time between STANDBY and ON
* should be as fast as possible and no longer than 10ms.
* @OFF: Power Off. No restrictions on transition times.
*/
enum snd_soc_bias_level {
SND_SOC_BIAS_ON,
SND_SOC_BIAS_PREPARE,
SND_SOC_BIAS_STANDBY,
SND_SOC_BIAS_OFF,
};
/*
* Digital Audio Interface (DAI) types
*/
#define SND_SOC_DAI_AC97 0x1
#define SND_SOC_DAI_I2S 0x2
#define SND_SOC_DAI_PCM 0x4
#define SND_SOC_DAI_AC97_BUS 0x8 /* for custom i.e. non ac97_codec.c */
/*
* DAI hardware audio formats
*/
#define SND_SOC_DAIFMT_I2S 0 /* I2S mode */
#define SND_SOC_DAIFMT_RIGHT_J 1 /* Right justified mode */
#define SND_SOC_DAIFMT_LEFT_J 2 /* Left Justified mode */
#define SND_SOC_DAIFMT_DSP_A 3 /* L data msb after FRM or LRC */
#define SND_SOC_DAIFMT_DSP_B 4 /* L data msb during FRM or LRC */
#define SND_SOC_DAIFMT_AC97 5 /* AC97 */
#define SND_SOC_DAIFMT_MSB SND_SOC_DAIFMT_LEFT_J
#define SND_SOC_DAIFMT_LSB SND_SOC_DAIFMT_RIGHT_J
/*
* DAI Gating
*/
#define SND_SOC_DAIFMT_CONT (0 << 4) /* continuous clock */
#define SND_SOC_DAIFMT_GATED (1 << 4) /* clock is gated when not Tx/Rx */
/*
* DAI Sync
* Synchronous LR (Left Right) clocks and Frame signals.
*/
#define SND_SOC_DAIFMT_SYNC (0 << 5) /* Tx FRM = Rx FRM */
#define SND_SOC_DAIFMT_ASYNC (1 << 5) /* Tx FRM ~ Rx FRM */
/*
* TDM
*/
#define SND_SOC_DAIFMT_TDM (1 << 6)
/*
* DAI hardware signal inversions
*/
#define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bclk + frm */
#define SND_SOC_DAIFMT_NB_IF (1 << 8) /* normal bclk + inv frm */
#define SND_SOC_DAIFMT_IB_NF (2 << 8) /* invert bclk + nor frm */
#define SND_SOC_DAIFMT_IB_IF (3 << 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 (0 << 12) /* codec clk & frm master */
#define SND_SOC_DAIFMT_CBS_CFM (1 << 12) /* codec clk slave & frm master */
#define SND_SOC_DAIFMT_CBM_CFS (2 << 12) /* codec clk master & frame slave */
#define SND_SOC_DAIFMT_CBS_CFS (3 << 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
/*
* AC97 codec ID's bitmask
*/
#define SND_SOC_DAI_AC97_ID0 (1 << 0)
#define SND_SOC_DAI_AC97_ID1 (1 << 1)
#define SND_SOC_DAI_AC97_ID2 (1 << 2)
#define SND_SOC_DAI_AC97_ID3 (1 << 3)
struct snd_soc_device;
struct snd_soc_pcm_stream;
struct snd_soc_ops;
struct snd_soc_dai_mode;
struct snd_soc_pcm_runtime;
struct snd_soc_dai;
struct snd_soc_codec;
struct snd_soc_machine_config;
struct soc_enum;
struct snd_soc_ac97_ops;
struct snd_soc_clock_info;
typedef int (*hw_write_t)(void *,const char* ,int);
typedef int (*hw_read_t)(void *,char* ,int);
extern struct snd_ac97_bus_ops soc_ac97_ops;
/* pcm <-> DAI connect */
void snd_soc_free_pcms(struct snd_soc_device *socdev);
int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid);
int snd_soc_register_card(struct snd_soc_device *socdev);
/* set runtime hw params */
int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream,
const struct snd_pcm_hardware *hw);
/* codec IO */
#define snd_soc_read(codec, reg) codec->read(codec, reg)
#define snd_soc_write(codec, reg, value) codec->write(codec, reg, value)
/* codec register bit access */
int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg,
unsigned short mask, unsigned short value);
int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned short reg,
unsigned short mask, unsigned short value);
int snd_soc_new_ac97_codec(struct snd_soc_codec *codec,
struct snd_ac97_bus_ops *ops, int num);
void snd_soc_free_ac97_codec(struct snd_soc_codec *codec);
/* 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, unsigned int freq_in, unsigned int freq_out);
/* 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 mask, int slots);
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);
/*
*Controls
*/
struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template,
void *data, char *long_name);
int snd_soc_info_enum_double(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo);
int snd_soc_info_enum_ext(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo);
int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol);
int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol);
int snd_soc_info_volsw(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo);
int snd_soc_info_volsw_ext(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo);
#define snd_soc_info_bool_ext snd_ctl_boolean_mono_info
int snd_soc_get_volsw(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol);
int snd_soc_put_volsw(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol);
int snd_soc_info_volsw_2r(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo);
int snd_soc_get_volsw_2r(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol);
int snd_soc_put_volsw_2r(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol);
int snd_soc_info_volsw_s8(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo);
int snd_soc_get_volsw_s8(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol);
int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol);
/* SoC PCM stream information */
struct snd_soc_pcm_stream {
char *stream_name;
u64 formats; /* SNDRV_PCM_FMTBIT_* */
unsigned int rates; /* SNDRV_PCM_RATE_* */
unsigned int rate_min; /* min rate */
unsigned int rate_max; /* max rate */
unsigned int channels_min; /* min channels */
unsigned int channels_max; /* max channels */
unsigned int active:1; /* stream is in use */
};
/* SoC audio ops */
struct snd_soc_ops {
int (*startup)(struct snd_pcm_substream *);
void (*shutdown)(struct snd_pcm_substream *);
int (*hw_params)(struct snd_pcm_substream *, struct snd_pcm_hw_params *);
int (*hw_free)(struct snd_pcm_substream *);
int (*prepare)(struct snd_pcm_substream *);
int (*trigger)(struct snd_pcm_substream *, int);
};
/* ASoC DAI ops */
struct snd_soc_dai_ops {
/* DAI clocking configuration */
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, unsigned int freq_in, unsigned int freq_out);
int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div);
/* DAI format configuration */
int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt);
int (*set_tdm_slot)(struct snd_soc_dai *dai,
unsigned int mask, int slots);
int (*set_tristate)(struct snd_soc_dai *dai, int tristate);
/* digital mute */
int (*digital_mute)(struct snd_soc_dai *dai, int mute);
};
/* SoC DAI (Digital Audio Interface) */
struct snd_soc_dai {
/* DAI description */
char *name;
unsigned int id;
unsigned char type;
/* DAI callbacks */
int (*probe)(struct platform_device *pdev,
struct snd_soc_dai *dai);
void (*remove)(struct platform_device *pdev,
struct snd_soc_dai *dai);
int (*suspend)(struct platform_device *pdev,
struct snd_soc_dai *dai);
int (*resume)(struct platform_device *pdev,
struct snd_soc_dai *dai);
/* ops */
struct snd_soc_ops ops;
struct snd_soc_dai_ops dai_ops;
/* DAI capabilities */
struct snd_soc_pcm_stream capture;
struct snd_soc_pcm_stream playback;
/* DAI runtime info */
struct snd_pcm_runtime *runtime;
struct snd_soc_codec *codec;
unsigned int active;
unsigned char pop_wait:1;
void *dma_data;
/* DAI private data */
void *private_data;
};
/* SoC Audio Codec */
struct snd_soc_codec {
char *name;
struct module *owner;
struct mutex mutex;
/* callbacks */
int (*set_bias_level)(struct snd_soc_codec *,
enum snd_soc_bias_level level);
/* runtime */
struct snd_card *card;
struct snd_ac97 *ac97; /* for ad-hoc ac97 devices */
unsigned int active;
unsigned int pcm_devs;
void *private_data;
/* codec IO */
void *control_data; /* codec control (i2c/3wire) data */
unsigned int (*read)(struct snd_soc_codec *, unsigned int);
int (*write)(struct snd_soc_codec *, unsigned int, unsigned int);
hw_write_t hw_write;
hw_read_t hw_read;
void *reg_cache;
short reg_cache_size;
short reg_cache_step;
/* dapm */
struct list_head dapm_widgets;
struct list_head dapm_paths;
enum snd_soc_bias_level bias_level;
enum snd_soc_bias_level suspend_bias_level;
struct delayed_work delayed_work;
/* codec DAI's */
struct snd_soc_dai *dai;
unsigned int num_dai;
};
/* codec device */
struct snd_soc_codec_device {
int (*probe)(struct platform_device *pdev);
int (*remove)(struct platform_device *pdev);
int (*suspend)(struct platform_device *pdev, pm_message_t state);
int (*resume)(struct platform_device *pdev);
};
/* SoC platform interface */
struct snd_soc_platform {
char *name;
int (*probe)(struct platform_device *pdev);
int (*remove)(struct platform_device *pdev);
int (*suspend)(struct platform_device *pdev,
struct snd_soc_dai *dai);
int (*resume)(struct platform_device *pdev,
struct snd_soc_dai *dai);
/* pcm creation and destruction */
int (*pcm_new)(struct snd_card *, struct snd_soc_dai *,
struct snd_pcm *);
void (*pcm_free)(struct snd_pcm *);
/* platform stream ops */
struct snd_pcm_ops *pcm_ops;
};
/* SoC machine DAI configuration, glues a codec and cpu DAI together */
struct snd_soc_dai_link {
char *name; /* Codec name */
char *stream_name; /* Stream name */
/* DAI */
struct snd_soc_dai *codec_dai;
struct snd_soc_dai *cpu_dai;
/* machine stream operations */
struct snd_soc_ops *ops;
/* codec/machine specific init - e.g. add machine controls */
int (*init)(struct snd_soc_codec *codec);
/* DAI pcm */
struct snd_pcm *pcm;
};
/* SoC machine */
struct snd_soc_machine {
char *name;
int (*probe)(struct platform_device *pdev);
int (*remove)(struct platform_device *pdev);
/* the pre and post PM functions are used to do any PM work before and
* after the codec and DAI's do any PM work. */
int (*suspend_pre)(struct platform_device *pdev, pm_message_t state);
int (*suspend_post)(struct platform_device *pdev, pm_message_t state);
int (*resume_pre)(struct platform_device *pdev);
int (*resume_post)(struct platform_device *pdev);
/* callbacks */
int (*set_bias_level)(struct snd_soc_machine *,
enum snd_soc_bias_level level);
/* CPU <--> Codec DAI links */
struct snd_soc_dai_link *dai_link;
int num_links;
};
/* SoC Device - the audio subsystem */
struct snd_soc_device {
struct device *dev;
struct snd_soc_machine *machine;
struct snd_soc_platform *platform;
struct snd_soc_codec *codec;
struct snd_soc_codec_device *codec_dev;
struct delayed_work delayed_work;
struct work_struct deferred_resume_work;
void *codec_data;
};
/* runtime channel data */
struct snd_soc_pcm_runtime {
struct snd_soc_dai_link *dai;
struct snd_soc_device *socdev;
};
/* enumerated kcontrol */
struct soc_enum {
unsigned short reg;
unsigned short reg2;
unsigned char shift_l;
unsigned char shift_r;
unsigned int mask;
const char **texts;
void *dapm;
};
#endif
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