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12ef193d58
linux/include/sound/soc.h
Troy Kisky 12ef193d58 ASoC: Allow setting codec register with debugfs filesystem 
i.e. echo 6 59 >/sys/kernel/debug/soc-audio.0/codec_reg
will set register 0x06 to a value of 0x59.
Also, pop_time debugfs interface setup is moved so that it
is setup in the same function as codec_reg

Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-10-30 14:34:02 +00:00

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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(xreg, xshift, xmax, xinvert) \
((unsigned long)&(struct soc_mixer_control) \
{.reg = xreg, .shift = xshift, .max = xmax, .invert = xinvert})
#define SOC_SINGLE_VALUE_EXT(xreg, xmax, xinvert) \
((unsigned long)&(struct soc_mixer_control) \
{.reg = xreg, .max = xmax, .invert = xinvert})
#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, xreg, shift_left, shift_right, xmax, xinvert) \
{ .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 = (unsigned long)&(struct soc_mixer_control) \
{.reg = xreg, .shift = shift_left, .rshift = shift_right, \
.max = xmax, .invert = xinvert} }
#define SOC_DOUBLE_R(xname, reg_left, reg_right, xshift, xmax, xinvert) \
{ .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 = (unsigned long)&(struct soc_mixer_control) \
{.reg = reg_left, .rreg = reg_right, .shift = xshift, \
.max = xmax, .invert = xinvert} }
#define SOC_DOUBLE_TLV(xname, xreg, shift_left, shift_right, xmax, xinvert, 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 = (unsigned long)&(struct soc_mixer_control) \
{.reg = xreg, .shift = shift_left, .rshift = shift_right,\
.max = xmax, .invert = xinvert} }
#define SOC_DOUBLE_R_TLV(xname, reg_left, reg_right, xshift, xmax, xinvert, 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 = (unsigned long)&(struct soc_mixer_control) \
{.reg = reg_left, .rreg = reg_right, .shift = xshift, \
.max = xmax, .invert = xinvert} }
#define SOC_DOUBLE_S8_TLV(xname, xreg, xmin, xmax, 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 = (unsigned long)&(struct soc_mixer_control) \
{.reg = xreg, .min = xmin, .max = xmax} }
#define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmax, xtexts) \
{ .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \
.max = xmax, .texts = xtexts }
#define SOC_ENUM_SINGLE(xreg, xshift, xmax, xtexts) \
SOC_ENUM_DOUBLE(xreg, xshift, xshift, xmax, xtexts)
#define SOC_ENUM_SINGLE_EXT(xmax, xtexts) \
{ .max = xmax, .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, xmax, 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, xmax, xinvert) }
#define SOC_SINGLE_EXT_TLV(xname, xreg, xshift, xmax, 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, xmax, 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);
int (*display_register)(struct snd_soc_codec *, char *,
size_t, unsigned int);
hw_write_t hw_write;
hw_read_t hw_read;
void *reg_cache;
short reg_cache_size;
short reg_cache_step;
/* dapm */
u32 pop_time;
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;
#ifdef CONFIG_DEBUG_FS
struct dentry *debugfs_root;
#endif
};
/* runtime channel data */
struct snd_soc_pcm_runtime {
struct snd_soc_dai_link *dai;
struct snd_soc_device *socdev;
};
/* mixer control */
struct soc_mixer_control {
int min, max;
unsigned int reg, rreg, shift, rshift, invert;
};
/* enumerated kcontrol */
struct soc_enum {
unsigned short reg;
unsigned short reg2;
unsigned char shift_l;
unsigned char shift_r;
unsigned int max;
const char **texts;
void *dapm;
};
#endif
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