mirror of
https://github.com/BobTheBlinker/android_kernel_motorola_sm6375.git
synced 2026-10-24 21:54:34 -04:00
Merge branch 'topic/asoc' into to-push
This commit is contained in:
Takashi Iwai
commit
5c8261e44e
139 changed files with 12650 additions and 4385 deletions
|
|
@ -1,7 +1,7 @@
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|||
/*
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||||
* audio.h -- Audio Driver for Wolfson WM8350 PMIC
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||||
*
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||||
* Copyright 2007 Wolfson Microelectronics PLC
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||||
* Copyright 2007, 2008 Wolfson Microelectronics PLC
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||||
*
|
||||
* This program is free software; you can redistribute it and/or modify it
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||||
* under the terms of the GNU General Public License as published by the
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||||
|
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@ -70,9 +70,9 @@
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|||
#define WM8350_CODEC_ISEL_0_5 3 /* x0.5 */
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#define WM8350_VMID_OFF 0
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#define WM8350_VMID_500K 1
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#define WM8350_VMID_100K 2
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#define WM8350_VMID_10K 3
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#define WM8350_VMID_300K 1
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#define WM8350_VMID_50K 2
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#define WM8350_VMID_5K 3
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||||
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||||
/*
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* R40 (0x28) - Clock Control 1
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@ -591,8 +591,38 @@
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#define WM8350_IRQ_CODEC_MICSCD 41
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#define WM8350_IRQ_CODEC_MICD 42
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/*
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* WM8350 Platform data.
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*
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* This must be initialised per platform for best audio performance.
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* Please see WM8350 datasheet for information.
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*/
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struct wm8350_audio_platform_data {
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int vmid_discharge_msecs; /* VMID --> OFF discharge time */
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int drain_msecs; /* OFF drain time */
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int cap_discharge_msecs; /* Cap ON (from OFF) discharge time */
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int vmid_charge_msecs; /* vmid power up time */
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u32 vmid_s_curve:2; /* vmid enable s curve speed */
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u32 dis_out4:2; /* out4 discharge speed */
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u32 dis_out3:2; /* out3 discharge speed */
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u32 dis_out2:2; /* out2 discharge speed */
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u32 dis_out1:2; /* out1 discharge speed */
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u32 vroi_out4:1; /* out4 tie off */
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u32 vroi_out3:1; /* out3 tie off */
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u32 vroi_out2:1; /* out2 tie off */
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||||
u32 vroi_out1:1; /* out1 tie off */
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u32 vroi_enable:1; /* enable tie off */
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u32 codec_current_on:2; /* current level ON */
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u32 codec_current_standby:2; /* current level STANDBY */
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u32 codec_current_charge:2; /* codec current @ vmid charge */
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};
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struct snd_soc_codec;
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struct wm8350_codec {
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struct platform_device *pdev;
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struct snd_soc_codec *codec;
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||||
struct wm8350_audio_platform_data *platform_data;
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||||
};
|
||||
|
||||
#endif
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||||
|
|
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|||
18
include/sound/l3.h
Normal file
18
include/sound/l3.h
Normal file
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@ -0,0 +1,18 @@
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|||
#ifndef _L3_H_
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#define _L3_H_ 1
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||||
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||||
struct l3_pins {
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void (*setdat)(int);
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void (*setclk)(int);
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||||
void (*setmode)(int);
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||||
int data_hold;
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||||
int data_setup;
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int clock_high;
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int mode_hold;
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int mode;
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||||
int mode_setup;
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||||
};
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||||
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||||
int l3_write(struct l3_pins *adap, u8 addr, u8 *data, int len);
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||||
|
||||
#endif
|
||||
14
include/sound/s3c24xx_uda134x.h
Normal file
14
include/sound/s3c24xx_uda134x.h
Normal file
|
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@ -0,0 +1,14 @@
|
|||
#ifndef _S3C24XX_UDA134X_H_
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||||
#define _S3C24XX_UDA134X_H_ 1
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||||
|
||||
#include <sound/uda134x.h>
|
||||
|
||||
struct s3c24xx_uda134x_platform_data {
|
||||
int l3_clk;
|
||||
int l3_mode;
|
||||
int l3_data;
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||||
void (*power) (int);
|
||||
int model;
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||||
};
|
||||
|
||||
#endif
|
||||
231
include/sound/soc-dai.h
Normal file
231
include/sound/soc-dai.h
Normal file
|
|
@ -0,0 +1,231 @@
|
|||
/*
|
||||
* 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
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#define __LINUX_SND_SOC_DAI_H
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||||
|
||||
|
||||
#include <linux/list.h>
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||||
|
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struct snd_pcm_substream;
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||||
|
||||
/*
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||||
* DAI hardware audio formats.
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||||
*
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||||
* Describes the physical PCM data formating and clocking. Add new formats
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||||
* to the end.
|
||||
*/
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||||
#define SND_SOC_DAIFMT_I2S 0 /* I2S mode */
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||||
#define SND_SOC_DAIFMT_RIGHT_J 1 /* Right Justified mode */
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||||
#define SND_SOC_DAIFMT_LEFT_J 2 /* Left Justified mode */
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||||
#define SND_SOC_DAIFMT_DSP_A 3 /* L data msb after FRM LRC */
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||||
#define SND_SOC_DAIFMT_DSP_B 4 /* L data msb during FRM LRC */
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||||
#define SND_SOC_DAIFMT_AC97 5 /* AC97 */
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||||
|
||||
/* left and right justified also known as MSB and LSB respectively */
|
||||
#define SND_SOC_DAIFMT_MSB SND_SOC_DAIFMT_LEFT_J
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||||
#define SND_SOC_DAIFMT_LSB SND_SOC_DAIFMT_RIGHT_J
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||||
|
||||
/*
|
||||
* DAI Clock gating.
|
||||
*
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||||
* DAI bit clocks can be be gated (disabled) when not the DAI is not
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* sending or receiving PCM data in a frame. This can be used to save power.
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||||
*/
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||||
#define SND_SOC_DAIFMT_CONT (0 << 4) /* continuous clock */
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||||
#define SND_SOC_DAIFMT_GATED (1 << 4) /* clock is gated */
|
||||
|
||||
/*
|
||||
* DAI Left/Right Clocks.
|
||||
*
|
||||
* Specifies whether the DAI can support different samples for similtanious
|
||||
* playback and capture. This usually requires a seperate physical frame
|
||||
* clock for playback and capture.
|
||||
*/
|
||||
#define SND_SOC_DAIFMT_SYNC (0 << 5) /* Tx FRM = Rx FRM */
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||||
#define SND_SOC_DAIFMT_ASYNC (1 << 5) /* Tx FRM ~ Rx FRM */
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||||
|
||||
/*
|
||||
* TDM
|
||||
*
|
||||
* Time Division Multiplexing. Allows PCM data to be multplexed with other
|
||||
* data on the DAI.
|
||||
*/
|
||||
#define SND_SOC_DAIFMT_TDM (1 << 6)
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||||
|
||||
/*
|
||||
* DAI hardware signal inversions.
|
||||
*
|
||||
* Specifies whether the DAI can also support inverted clocks for the specified
|
||||
* format.
|
||||
*/
|
||||
#define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bit clock + frame */
|
||||
#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 */
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||||
|
||||
/*
|
||||
* 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
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||||
#define SND_SOC_DAIFMT_MASTER_MASK 0xf000
|
||||
|
||||
/*
|
||||
* Master Clock Directions
|
||||
*/
|
||||
#define SND_SOC_CLOCK_IN 0
|
||||
#define SND_SOC_CLOCK_OUT 1
|
||||
|
||||
struct snd_soc_dai_ops;
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||||
struct snd_soc_dai;
|
||||
struct snd_ac97_bus_ops;
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||||
|
||||
/* Digital Audio Interface registration */
|
||||
int snd_soc_register_dai(struct snd_soc_dai *dai);
|
||||
void snd_soc_unregister_dai(struct snd_soc_dai *dai);
|
||||
int snd_soc_register_dais(struct snd_soc_dai *dai, size_t count);
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||||
void snd_soc_unregister_dais(struct snd_soc_dai *dai, size_t count);
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||||
|
||||
/* 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);
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||||
|
||||
/* 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);
|
||||
|
||||
/*
|
||||
* Digital Audio Interface.
|
||||
*
|
||||
* Describes the Digital Audio Interface in terms of it's ALSA, DAI and AC97
|
||||
* operations an capabilities. Codec and platfom drivers will register a this
|
||||
* structure for every DAI they have.
|
||||
*
|
||||
* This structure covers the clocking, formating and ALSA operations for each
|
||||
* interface a
|
||||
*/
|
||||
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, unsigned int freq_in, unsigned int freq_out);
|
||||
int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div);
|
||||
|
||||
/*
|
||||
* 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 (*set_tdm_slot)(struct snd_soc_dai *dai,
|
||||
unsigned int mask, int slots);
|
||||
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);
|
||||
|
||||
/*
|
||||
* 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 *);
|
||||
int (*trigger)(struct snd_pcm_substream *, int,
|
||||
struct snd_soc_dai *);
|
||||
};
|
||||
|
||||
/*
|
||||
* Digital Audio Interface runtime data.
|
||||
*
|
||||
* Holds runtime data for a DAI.
|
||||
*/
|
||||
struct snd_soc_dai {
|
||||
/* DAI description */
|
||||
char *name;
|
||||
unsigned int id;
|
||||
int ac97_control;
|
||||
|
||||
struct device *dev;
|
||||
|
||||
/* 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 snd_soc_dai *dai);
|
||||
int (*resume)(struct snd_soc_dai *dai);
|
||||
|
||||
/* ops */
|
||||
struct snd_soc_dai_ops 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;
|
||||
|
||||
/* parent codec/platform */
|
||||
union {
|
||||
struct snd_soc_codec *codec;
|
||||
struct snd_soc_platform *platform;
|
||||
};
|
||||
|
||||
struct list_head list;
|
||||
};
|
||||
|
||||
#endif
|
||||
|
|
@ -221,8 +221,6 @@ int snd_soc_dapm_new_controls(struct snd_soc_codec *codec,
|
|||
int num);
|
||||
|
||||
/* dapm path setup */
|
||||
int __deprecated snd_soc_dapm_connect_input(struct snd_soc_codec *codec,
|
||||
const char *sink_name, const char *control_name, const char *src_name);
|
||||
int snd_soc_dapm_new_widgets(struct snd_soc_codec *codec);
|
||||
void snd_soc_dapm_free(struct snd_soc_device *socdev);
|
||||
int snd_soc_dapm_add_routes(struct snd_soc_codec *codec,
|
||||
|
|
|
|||
|
|
@ -21,8 +21,6 @@
|
|||
#include <sound/control.h>
|
||||
#include <sound/ac97_codec.h>
|
||||
|
||||
#define SND_SOC_VERSION "0.13.2"
|
||||
|
||||
/*
|
||||
* Convenience kcontrol builders
|
||||
*/
|
||||
|
|
@ -145,105 +143,31 @@ enum snd_soc_bias_level {
|
|||
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_platform;
|
||||
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;
|
||||
|
||||
int snd_soc_register_platform(struct snd_soc_platform *platform);
|
||||
void snd_soc_unregister_platform(struct snd_soc_platform *platform);
|
||||
int snd_soc_register_codec(struct snd_soc_codec *codec);
|
||||
void snd_soc_unregister_codec(struct snd_soc_codec *codec);
|
||||
|
||||
/* 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);
|
||||
int snd_soc_init_card(struct snd_soc_device *socdev);
|
||||
|
||||
/* set runtime hw params */
|
||||
int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream,
|
||||
|
|
@ -263,27 +187,6 @@ 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
|
||||
*/
|
||||
|
|
@ -341,66 +244,14 @@ struct snd_soc_ops {
|
|||
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;
|
||||
struct device *dev;
|
||||
|
||||
struct list_head list;
|
||||
|
||||
/* callbacks */
|
||||
int (*set_bias_level)(struct snd_soc_codec *,
|
||||
|
|
@ -426,6 +277,7 @@ struct snd_soc_codec {
|
|||
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;
|
||||
|
|
@ -435,6 +287,11 @@ struct snd_soc_codec {
|
|||
/* codec DAI's */
|
||||
struct snd_soc_dai *dai;
|
||||
unsigned int num_dai;
|
||||
|
||||
#ifdef CONFIG_DEBUG_FS
|
||||
struct dentry *debugfs_reg;
|
||||
struct dentry *debugfs_pop_time;
|
||||
#endif
|
||||
};
|
||||
|
||||
/* codec device */
|
||||
|
|
@ -448,13 +305,12 @@ struct snd_soc_codec_device {
|
|||
/* SoC platform interface */
|
||||
struct snd_soc_platform {
|
||||
char *name;
|
||||
struct list_head list;
|
||||
|
||||
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);
|
||||
int (*suspend)(struct snd_soc_dai *dai);
|
||||
int (*resume)(struct snd_soc_dai *dai);
|
||||
|
||||
/* pcm creation and destruction */
|
||||
int (*pcm_new)(struct snd_card *, struct snd_soc_dai *,
|
||||
|
|
@ -484,9 +340,14 @@ struct snd_soc_dai_link {
|
|||
struct snd_pcm *pcm;
|
||||
};
|
||||
|
||||
/* SoC machine */
|
||||
struct snd_soc_machine {
|
||||
/* SoC card */
|
||||
struct snd_soc_card {
|
||||
char *name;
|
||||
struct device *dev;
|
||||
|
||||
struct list_head list;
|
||||
|
||||
int instantiated;
|
||||
|
||||
int (*probe)(struct platform_device *pdev);
|
||||
int (*remove)(struct platform_device *pdev);
|
||||
|
|
@ -499,23 +360,26 @@ struct snd_soc_machine {
|
|||
int (*resume_post)(struct platform_device *pdev);
|
||||
|
||||
/* callbacks */
|
||||
int (*set_bias_level)(struct snd_soc_machine *,
|
||||
int (*set_bias_level)(struct snd_soc_card *,
|
||||
enum snd_soc_bias_level level);
|
||||
|
||||
/* CPU <--> Codec DAI links */
|
||||
struct snd_soc_dai_link *dai_link;
|
||||
int num_links;
|
||||
|
||||
struct snd_soc_device *socdev;
|
||||
|
||||
struct snd_soc_platform *platform;
|
||||
struct delayed_work delayed_work;
|
||||
struct work_struct deferred_resume_work;
|
||||
};
|
||||
|
||||
/* 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_card *card;
|
||||
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;
|
||||
};
|
||||
|
||||
|
|
@ -542,4 +406,6 @@ struct soc_enum {
|
|||
void *dapm;
|
||||
};
|
||||
|
||||
#include <sound/soc-dai.h>
|
||||
|
||||
#endif
|
||||
|
|
|
|||
26
include/sound/uda134x.h
Normal file
26
include/sound/uda134x.h
Normal file
|
|
@ -0,0 +1,26 @@
|
|||
/*
|
||||
* uda134x.h -- UDA134x ALSA SoC Codec driver
|
||||
*
|
||||
* Copyright 2007 Dension Audio Systems Ltd.
|
||||
* Author: Zoltan Devai
|
||||
*
|
||||
* 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 _UDA134X_H
|
||||
#define _UDA134X_H
|
||||
|
||||
#include <sound/l3.h>
|
||||
|
||||
struct uda134x_platform_data {
|
||||
struct l3_pins l3;
|
||||
void (*power) (int);
|
||||
int model;
|
||||
#define UDA134X_UDA1340 1
|
||||
#define UDA134X_UDA1341 2
|
||||
#define UDA134X_UDA1344 3
|
||||
};
|
||||
|
||||
#endif /* _UDA134X_H */
|
||||
Loading…
Reference in a new issue