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sdk-hwV1.3/external/eyesee-mpp/middleware/sun8iw21/include/media/mm_comm_aio.h

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/******************************************************************************
Copyright (C), 2001-2016, Allwinner Tech. Co., Ltd.
******************************************************************************
File Name : mm_comm_aio.h
Version : Initial Draft
Author : Allwinner BU3-PD2 Team
Created : 2016/03/23
Last Modified :
Description : common struct definition for AENC
Function List :
History :
******************************************************************************/
#ifndef __IPCLINUX_COMM_AIO_H__
#define __IPCLINUX_COMM_AIO_H__
#include "plat_type.h"
#include "plat_errno.h"
#include "mm_common.h"
#ifdef __cplusplus
extern "C" {
#endif
#define MAX_AUDIO_FRAME_NUM 50 /*max count of audio frame in Buffer */
#define MAX_AUDIO_POINT_BYTES 4 /*max bytes of one sample point(now 32bit max)*/
#define MAX_VOICE_POINT_NUM 480 /*max sample per frame for voice encode */
#define MAX_AUDIO_POINT_NUM 2048 /*max sample per frame for all encoder(aacplus:2048)*/
#define MAX_AO_POINT_NUM 4096 /* from h3£»support 4096 framelen*/
#define MIN_AUDIO_POINT_NUM 80 /*min sample per frame*/
#define MAX_AI_POINT_NUM 2048 /*max sample per frame for all encoder(aacplus:2048)*/
/*max length of audio frame by bytes, one frame contain many sample point */
#define MAX_AUDIO_FRAME_LEN (MAX_AUDIO_POINT_BYTES*MAX_AO_POINT_NUM)
/*max length of audio stream by bytes */
#define MAX_AUDIO_STREAM_LEN MAX_AUDIO_FRAME_LEN
#define MAX_AI_USRFRM_DEPTH 30 /*max depth of user frame buf */
/*The VQE EQ Band num.*/
#define VQE_EQ_BAND_NUM 11
#define FILTER_CNT 6
typedef enum AUDIO_SAMPLE_RATE_E
{
AUDIO_SAMPLE_RATE_8000 = 8000, /* 8K samplerate*/
AUDIO_SAMPLE_RATE_12000 = 12000, /* 12K samplerate*/
AUDIO_SAMPLE_RATE_11025 = 11025, /* 11.025K samplerate*/
AUDIO_SAMPLE_RATE_16000 = 16000, /* 16K samplerate*/
AUDIO_SAMPLE_RATE_22050 = 22050, /* 22.050K samplerate*/
AUDIO_SAMPLE_RATE_24000 = 24000, /* 24K samplerate*/
AUDIO_SAMPLE_RATE_32000 = 32000, /* 32K samplerate*/
AUDIO_SAMPLE_RATE_44100 = 44100, /* 44.1K samplerate*/
AUDIO_SAMPLE_RATE_48000 = 48000, /* 48K samplerate*/
AUDIO_SAMPLE_RATE_BUTT,
} AUDIO_SAMPLE_RATE_E;
typedef enum AUDIO_BIT_WIDTH_E
{
AUDIO_BIT_WIDTH_8 = 0, /* 8bit width */
AUDIO_BIT_WIDTH_16 = 1, /* 16bit width*/
AUDIO_BIT_WIDTH_24 = 2, /* 24bit width*/
AUDIO_BIT_WIDTH_32 = 3, /* 32bit width*/
AUDIO_BIT_WIDTH_BUTT,
} AUDIO_BIT_WIDTH_E;
typedef enum AIO_MODE_E
{
AIO_MODE_I2S_MASTER = 0, /* AIO I2S master mode */
AIO_MODE_I2S_SLAVE, /* AIO I2S slave mode */
AIO_MODE_PCM_SLAVE_STD, /* AIO PCM slave standard mode */
AIO_MODE_PCM_SLAVE_NSTD, /* AIO PCM slave non-standard mode */
AIO_MODE_PCM_MASTER_STD, /* AIO PCM master standard mode */
AIO_MODE_PCM_MASTER_NSTD, /* AIO PCM master non-standard mode */
AIO_MODE_BUTT
} AIO_MODE_E;
typedef enum AIO_SOUND_MODE_E
{
AUDIO_SOUND_MODE_MONO =0,/*mono*/
AUDIO_SOUND_MODE_STEREO =1,/*stereo*/
AUDIO_SOUND_MODE_AW_1Chn1Ref = 100, //1 common channel and 1 ref channel used to aec.
AUDIO_SOUND_MODE_AW_2Chn1Ref = 101, //2 common channels and 1 ref channel used to aec.
AUDIO_SOUND_MODE_BUTT
} AUDIO_SOUND_MODE_E;
/*
An example of the packing scheme for G726-32 codewords is as shown, and bit A3 is the least significant bit of the first codeword:
RTP G726-32:
0 1
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-
|B B B B|A A A A|D D D D|C C C C| ...
|0 1 2 3|0 1 2 3|0 1 2 3|0 1 2 3|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-
MEDIA G726-32:
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-
|A A A A|B B B B|C C C C|D D D D| ...
|3 2 1 0|3 2 1 0|3 2 1 0|3 2 1 0|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-
*/
typedef enum G726_BPS_E
{
G726_16K = 0, /* G726 16kbps, see RFC3551.txt 4.5.4 G726-16 */
G726_24K, /* G726 24kbps, see RFC3551.txt 4.5.4 G726-24 */
G726_32K, /* G726 32kbps, see RFC3551.txt 4.5.4 G726-32 */
G726_40K, /* G726 40kbps, see RFC3551.txt 4.5.4 G726-40 */
MEDIA_G726_16K, /* G726 16kbps for ASF ... */
MEDIA_G726_24K, /* G726 24kbps for ASF ... */
MEDIA_G726_32K, /* G726 32kbps for ASF ... */
MEDIA_G726_40K, /* G726 40kbps for ASF ... */
G726_BUTT,
} G726_BPS_E;
typedef enum ADPCM_TYPE_E
{
/* see DVI4 diiffers in three respects from the IMA ADPCM at RFC3551.txt 4.5.1 DVI4 */
ADPCM_TYPE_DVI4 = 0, /* 32kbps ADPCM(DVI4) for RTP */
ADPCM_TYPE_IMA, /* 32kbps ADPCM(IMA),NOTICE:point num must be 161/241/321/481 */
ADPCM_TYPE_ORG_DVI4,
ADPCM_TYPE_BUTT,
} ADPCM_TYPE_E;
#define AI_EXPAND 0x01
#define AI_CUT 0x02
typedef enum PCM_CARD_TYPE_E
{
PCM_CARD_TYPE_AUDIOCODEC = 0, /* IC internal codec */
PCM_CARD_TYPE_SNDHDMI = 1, /* hdmi audio */
PCM_CARD_TYPE_UAC1 = 2,
PCM_CARD_TYPE_BUTT,
} PCM_CARD_TYPE_E;
//typedef struct AO_AGC_CONFIG_S
//{
// float fSample_rate; //iSample_rate : 8, 16, 32, 44.1, 48,
// int iSample_len; //default 10ms: the sample number processed,suggest value is 1024
// int iBytePerSample; //bps: 2,- short
// int iGain_level; //gain_level : 0,1,2; the higher the level is,the the gain will more power
// int iChannel; //channel mono:1 stereo:2
//
// int iReserverd; // used internally
// float ffilter[FILTER_CNT]; // used internally
// float fEnergy_Level; // used internally
//
//}AO_AGC_CONFIG_S;
//typedef struct AI_AGC_FLOAT_CONFIG_S
//{
// int iSampleRate;
// int iChannel; /* Support stereo and mono */
// int iSampleLen; /* sample number processed */
// int iBytePerSample; /* bps: 2,- short */
// float fTargetDb; /* range: [-30,0] */
// float fMaxGainDb; /* range: [0,30] */
//}AI_AGC_FLOAT_CONFIG_S;
typedef struct AGC_FLOAT_CONFIG_S
{
float fTargetDb; /* range: [-30,0] */
float fMaxGainDb; /* range: [0,30] */
}AGC_FLOAT_CONFIG_S;
typedef struct AIO_ATTR_S
{
AUDIO_SAMPLE_RATE_E enSamplerate; /**< sample rate */
AUDIO_BIT_WIDTH_E enBitwidth; /**< bitwidth */
AIO_MODE_E enWorkmode; /* master or slave mode */
AUDIO_SOUND_MODE_E enSoundmode; /**< mono or stereo. used by ai&ao, it is dst output attr of mpi_ai, input attr of mpi_ao */
unsigned int u32EXFlag; /* expand 8bit to 16bit,use AI_EXPAND(only valid for AI 8bit) */
unsigned int u32FrmNum; /* frame num in buf[2,MAX_AUDIO_FRAME_NUM] */
/**
point num per frame (80/160/240/320/480/1024/2048), i.e. period_size,
(ADPCM IMA should add 1 point, AMR only support 160)
used by ai&ao. ai get. ao set.
*/
unsigned int mPtNumPerFrm;
/**
channle number on FS, valid value:1/2/4/8, used by ai&ao. In mpi_ai, mChnCnt only include MicInChn, not include refChn.
for mpi_ai, mChnCnt may not be equal to enSoundmode as aec might merge two mic's data to one channel. mChnCnt is
input attr of mpi_ai, enSoundmode is output attr of mpi_ai.
for mpi_ao, mChnCnt == enSoundmode.
*/
unsigned int mChnCnt;
unsigned int u32ClkSel; /* 0: AI and AO clock is separate
1: AI and AO clock is inseparate, AI use AO's clock */
PCM_CARD_TYPE_E mPcmCardId; /**< audio card selection for play audio file */
/**
mic number. mpi_aenc will generate capture pcm identifier according to number of microphone.
0: use default number, i.d. 1
used by mpi_ai.
One device of card can has many MICs. i.d. hw:0,0 can has many MICs.
And in common, out hardware put many MICs in one audioCodec. So hw:0,0 owns all MICs.
*/
int mMicNum;
int ai_aec_en;
int aec_delay_ms;
/**
0:process aec in audio_hw
1:not process aec in audio_hw,user call AW_MPI_AI_GetFrame() to get capture frame and aec frame.
when bypassAec is set to true, we will transport pcm got from alsa-multi-plugin directly, channel count will +1.
only used for non-tunnel mode of ai-component output port,
i.d. ERRORTYPE AW_MPI_AI_GetFrame(AUDIO_DEV AudioDevId, AI_CHN AiChn, AUDIO_FRAME_S *pstFrm, AEC_FRAME_S *pstAecFrm, int s32MilliSec);
*/
int mbBypassAec;
int ai_ans_en;
int ai_ans_mode; /* four modes will be supported by ans,the value is 0 1 2 3,the greater te value is,the better of the noise suppression,and more side-effect.*/
int ai_agc_en; // enable or disable agc process in audio input module
//my_agc_handle ai_agc_cfg;
AGC_FLOAT_CONFIG_S ai_agc_float_cfg;
} AIO_ATTR_S;
typedef struct AI_CHN_PARAM_S
{
unsigned int u32UsrFrmDepth;
} AI_CHN_PARAM_S;
typedef struct AUDIO_FRAME_S
{
AUDIO_SAMPLE_RATE_E mSamplerate; /* sample rate, extended for ai&ao*/
AUDIO_BIT_WIDTH_E mBitwidth; /*audio frame bitwidth*/
AUDIO_SOUND_MODE_E mSoundmode; /*audio frame momo or stereo mode, used by ai&ao,adec*/
void *mpAddr;
unsigned long long mTimeStamp; /*audio frame timestamp, unit:us*/
unsigned int mSeq; /*audio frame seq*/
unsigned int mLen; /*data lenth per channel in frame*/
unsigned int mId;
} AUDIO_FRAME_S;
typedef struct AEC_FRAME_S
{
AUDIO_FRAME_S stRefFrame; /* AEC reference audio frame */
BOOL bValid; /* whether frame is valid */
BOOL bSysBind; /* whether is sysbind */
} AEC_FRAME_S;
typedef struct AUDIO_FRAME_COMBINE_S
{
AUDIO_FRAME_S stFrm; /* audio frame */
AEC_FRAME_S stRefFrm; /* AEC reference audio frame */
} AUDIO_FRAME_COMBINE_S;
typedef struct AUDIO_FRAME_INFO_S
{
AUDIO_FRAME_S *mpAFrame;/*frame ptr*/
unsigned int mId; /*frame id*/
} AUDIO_FRAME_INFO_S;
typedef struct AUDIO_STREAM_S
{
unsigned char *pStream; /* the virtual address of stream */
unsigned char *pStreamExtra;
unsigned int mLen; /* stream lenth, by bytes */
unsigned int mExtraLen;
unsigned long long mTimeStamp; /* frame time stamp */
int mId; /* frame id */
} AUDIO_STREAM_S;
typedef struct AUDIO_RESAMPLE_ATTR_S
{
unsigned int u32InPointNum; /* input point number of frame */
AUDIO_SAMPLE_RATE_E enInSampleRate; /* input sample rate */
AUDIO_SAMPLE_RATE_E enOutSampleRate; /* output sample rate */
} AUDIO_RESAMPLE_ATTR_S;
typedef struct AO_CHN_STATE_S
{
unsigned int u32ChnTotalNum; /* total number of channel buffer */
unsigned int u32ChnFreeNum; /* free number of channel buffer */
unsigned int u32ChnBusyNum; /* busy number of channel buffer */
} AO_CHN_STATE_S;
typedef struct AIO_RESMP_INFO_S
{
BOOL bReSmpEnable; /* resample enable or disable */
AUDIO_RESAMPLE_ATTR_S stResmpAttr;
} AIO_RESMP_INFO_S;
typedef enum AUDIO_TRACK_MODE_E
{
AUDIO_TRACK_NORMAL = 0,
AUDIO_TRACK_BOTH_LEFT = 1,
AUDIO_TRACK_BOTH_RIGHT = 2,
AUDIO_TRACK_EXCHANGE = 3,
AUDIO_TRACK_MIX = 4,
AUDIO_TRACK_LEFT_MUTE = 5,
AUDIO_TRACK_RIGHT_MUTE = 6,
AUDIO_TRACK_BOTH_MUTE = 7,
AUDIO_TRACK_BUTT
} AUDIO_TRACK_MODE_E;
typedef enum AUDIO_CLKDIR_E
{
AUDIO_CLKDIR_RISE = 0,
AUDIO_CLKDIR_FALL = 1,
AUDIO_CLKDIR_BUTT
} AUDIO_CLKDIR_E;
typedef enum AUDIO_FADE_RATE_E
{
AUDIO_FADE_RATE_1 = 0,
AUDIO_FADE_RATE_2 = 1,
AUDIO_FADE_RATE_4 = 2,
AUDIO_FADE_RATE_8 = 3,
AUDIO_FADE_RATE_16 = 4,
AUDIO_FADE_RATE_32 = 5,
AUDIO_FADE_RATE_64 = 6,
AUDIO_FADE_RATE_128 = 7,
AUDIO_FADE_RATE_BUTT
} AUDIO_FADE_RATE_E;
typedef struct AUDIO_FADE_S
{
BOOL bFade;
AUDIO_FADE_RATE_E enFadeInRate;
AUDIO_FADE_RATE_E enFadeOutRate;
} AUDIO_FADE_S;
typedef enum AUDIO_AEC_MODE_E
{
AUDIO_AEC_MODE_CLOSE = 0,
AUDIO_AEC_MODE_OPEN = 1,
AUDIO_AEC_MODE_BUTT
} AUDIO_AEC_MODE_E;
/**Defines the configure parameters of AGC.*/
typedef struct AUDIO_AGC_CONFIG_S
{
BOOL bUsrMode; /* mode 0: auto¡ê?mode 1: manual.*/
signed char s8TargetLevel; /* target voltage level, range: [-40, -1]dB */
signed char s8NoiseFloor; /* noise floor, range: [-65, -20]dB */
signed char s8MaxGain; /* max gain, range: [0, 30]dB */
signed char s8AdjustSpeed; /* adjustable speed, range: [0, 10]dB/s */
signed char s8ImproveSNR; /* switch for improving SNR, range: [0:close, 1:upper limit 3dB, 2:upper limit 6dB] */
signed char s8UseHighPassFilt; /* switch for using high pass filt, range: [0:close, 1:80Hz, 2:120Hz, 3:150:Hz, 4:300Hz: 5:500Hz] */
signed char s8OutputMode; /* output mode, mute when lower than noise floor, range: [0:close, 1:open] */
short s16NoiseSupSwitch; /* switch for noise suppression, range: [0:close, 1:open] */
int s32Reserved;
} AUDIO_AGC_CONFIG_S;
typedef struct AUDIO_GAIN_CONFIG_S
{
signed char s8GainValue; /*Value range: -20 -> 20 */
int s32Reserved;
} AUDIO_GAIN_CONFIG_S;
//////////////the following aec paras
/* dynamic range control parameters */
typedef struct
{
int sampling_rate; /* */
int target_level;
int max_gain;
int min_gain;
int attack_time;
int release_time;
int noise_threshold; /* */
}DRC_PARAMS_T;
/* bulk delay compensation parameters */
typedef struct
{
int sampling_rate; /* sampling rate, should be equal to the SR of echo canceller */
int block_size; /* processing block size, should be equal to the frame size of echo canceller */
int max_delay; /* max delay for the far end signal */
int delay_margin; /* delay magin, this is used for delay estimation variation
in some case, the variation could cause uncaulity
delay_margin shold be positive and litter than max_delay */
int near_delay; /* delay for the near end signal */
}BDC_PARAMS_T;
/* noise suppression parameters */
typedef struct
{
int sampling_rate; /* sampling rate */
int max_suppression; /* maximum amount of suppression allowed for noise reduction */
int overlap_percent; /* overlap of window, value from 0 ->1, 0:50%, 1:75%*/
int nonstat; /* stationarity of noise, value from 0 -> 3 */
}NS_PARAMS_T;
typedef struct
{
int G; /* boost/cut gain in dB */
int fc; /* cutoff/center frequency in Hz */
int Q; /* quality factor */
int type; /* filter type,value from 0->2 */
/* 0: low pass shelving filter */
/* 1: band pass peak filter */
/* 2: high pass shelving filter */
}EQ_CORE_PARAMS_T;
typedef struct
{
int biq_num; /*num of items(biquad)*/
int sampling_rate; /* sampling rate */
EQ_CORE_PARAMS_T* core_prms; /* eq parameters for generate IIR coeff*/
}EQ_PARAMS_T;
/* Nonlinear Processing parameters parameters */
typedef struct
{
/* nlp overlap of window */
int nlp_overlap_percent; //value from 0 -> 1
}NLP_PARAMS_T;
/* echo cancellation parameters */
typedef struct
{
/* sampling rate */
int sampling_rate;
/* echo path length, Unit ms, should be times of 8 */
int tail_length;
/* processing frame size, Unit ms, only 8ms is supported until now */
int frame_size;
/* Nonlinear Processing switch */
int enable_nlp;
/* Nonlinear Processing parameters */
NLP_PARAMS_T nlp_prms;
}AEC_PARAMS_T;
/* total init parameters for echo control */
typedef struct
{
/* echo canceller switch */
int enable_aec;
/* echo canceller parameters */
AEC_PARAMS_T aec_prms;
/* bulk delay compensator switch */
int enable_bdc;
/* bulk delay estimator parameters */
BDC_PARAMS_T bdc_prms;
/* clock drift compensator switch */
int enable_cdc;
/* dynamic range control switch */
int enable_txdrc;
/* transmitter drc parameters */
DRC_PARAMS_T txdrc_prms;
/* receiver drc */
int enable_rxdrc;
/*receiver drc parameters */
DRC_PARAMS_T rxdrc_prms;
/* transmitter equalizer */
int enable_txeq;
/* transmitter equalizer parameters */
EQ_PARAMS_T txeq_prms;
/* receiver equalizer */
int enable_rxeq;
/* receiver equalizer parameters */
EQ_PARAMS_T rxeq_prms;
/* noise suppressor switch */
int enable_ns;
/* noise suppression parameters */
NS_PARAMS_T ns_prms;
/* fade in to suppress the initial echo */
int enable_txfade;
}EC_PARAMS_T;
/////////////////////end aec struct
/**Defines the configure parameters of AEC.*/
typedef struct AI_AEC_CONFIG_S
{
EC_PARAMS_T prms;
} AI_AEC_CONFIG_S;
/**Defines the configure parameters of Pause.*/
typedef struct AISendDataInfo
{
unsigned int mLen;
unsigned int mbIgnore;
int64_t mPts; //unit:us
} AISendDataInfo;
/**Defines the configure parameters of ANR.*/
typedef struct AUDIO_ANR_CONFIG_S
{
BOOL bUsrMode; /* mode 0: auto¡ê?mode 1: manual.*/
short s16NrIntensity; /* noise reduce intensity, range: [0, 25] */
short s16NoiseDbThr; /* noise threshold, range: [30, 60] */
signed char s8SpProSwitch; /* switch for music probe, range: [0:close, 1:open] */
int s32Reserved;
} AUDIO_ANR_CONFIG_S;
/**Defines the configure parameters of HPF.*/
typedef enum AUDIO_HPF_FREQ_E
{
AUDIO_HPF_FREQ_80 = 80, /* 80Hz */
AUDIO_HPF_FREQ_120 = 120, /* 120Hz */
AUDIO_HPF_FREQ_150 = 150, /* 150Hz */
AUDIO_HPF_FREQ_BUTT,
} AUDIO_HPF_FREQ_E;
typedef struct AUDIO_HPF_CONFIG_S
{
BOOL bUsrMode; /* mode 0: auto mode 1: mannual.*/
AUDIO_HPF_FREQ_E enHpfFreq; /*freq to be processed*/
} AUDIO_HPF_CONFIG_S;
typedef struct AI_RNR_CONFIG_S
{
int sMaxNoiseSuppression; /*max noise depress threshold(dB), range:[-30, -12], default:-12*/
int sOverlapPercent; /*window overlap percent, range:[0, 1], default: 0*/
/*0:repeat 50% 1:repeat 75%*/
int sNonstat; /*noise smoothness, range:[0, 3], default:3*/
/*0:normal 1:weak unsmoothness 2:middle unsmoothness 3:strong unsmoothness*/
} AI_RNR_CONFIG_S;
typedef struct AI_DRC_CONFIG_S
{
int sTargetLevel;
int sMaxGain;
int sMinGain;
int sAttackTime;
int sReleaseTime;
int sNoiseThreshold;
} AI_DRC_CONFIG_S;
typedef enum AUDIO_EQ_MODE_S
{
AUDIO_EQ_TYPE_NORMAL=0, /* nature */
AUDIO_EQ_TYPE_DBB, /* dbb */
AUDIO_EQ_TYPE_POP, /* pop */
AUDIO_EQ_TYPE_ROCK, /* rock */
AUDIO_EQ_TYPE_CLASSIC, /* classic */
AUDIO_EQ_TYPE_JAZZ, /* jazz */
AUDIO_EQ_TYPE_VOCAL, /* vocal */
AUDIO_EQ_TYPE_DANCE, /* dance */
AUDIO_EQ_TYPE_SOFT, /* soft */
AUDIO_EQ_TYPE_USR_MODE=0xFF, /* user */
AUDIO_EQ_TYPE_MAX
}AUDIO_EQ_MODE_S;
typedef struct AUDIO_EQ_CONFIG_S
{
short s16GaindB[VQE_EQ_BAND_NUM]; /*only in usermode valid, from[1]->[10], frequncy include 31,62,125,250,500,1k,2k,4k,8k,16k, range:[-12, 12]*/
//s16GaindB[0]= 0xFF -> user mode, PostProInfo->s8GaindB[1]-PostProInfo->s8GaindB[10] data valid, range:[-12, 12], frequency difined above
//s16GaindB[0] <= AUDIO_EQ_TYPE_SOFT, typic type above defined: AUDIO_EQ_MODE_S {0: nature 1:dbb 2:pop ---> 8:soft}
int s32Reserved;
} AUDIO_EQ_CONFIG_S;
/**Defines the configure parameters of UPVQE work state.*/
typedef enum VQE_WORKSTATE_E
{
VQE_WORKSTATE_COMMON = 0, /* common environment, Applicable to the family of voice calls. */
VQE_WORKSTATE_MUSIC = 1, /* music environment , Applicable to the family of music environment. */
VQE_WORKSTATE_NOISY = 2, /* noisy environment , Applicable to the noisy voice calls. */
} VQE_WORKSTATE_E;
/**Defines the configure parameters of VQE.*/
typedef struct AI_VQE_CONFIG_S
{
int bHpfOpen;
int bAecOpen;
int bAnrOpen; /*Anr and Rnr can't enable at the same time,Anr used in voice noise reducing*/
int bRnrOpen;
int bAgcOpen;
int bEqOpen;
int bDrcOpen;
int s32WorkSampleRate; /* Sample Rate£º8KHz/16KHz¡£default: 8KHz*/
int s32FrameSample; /* VQE frame length£º 80-4096 */
VQE_WORKSTATE_E enWorkstate;
AUDIO_HPF_CONFIG_S stHpfCfg;
AI_AEC_CONFIG_S stAecCfg;
AUDIO_ANR_CONFIG_S stAnrCfg;
AI_RNR_CONFIG_S stRnrCfg;
AUDIO_AGC_CONFIG_S stAgcCfg;
AUDIO_EQ_CONFIG_S stEqCfg;
AI_DRC_CONFIG_S stDrcCfg;
} AI_VQE_CONFIG_S;
typedef struct AO_VQE_CONFIG_S
{
int bHpfOpen;
int bAnrOpen;
int bAgcOpen;
int bEqOpen;
int bGainOpen;
int s32WorkSampleRate; /* Sample Rate: 8KHz/16KHz/48KHz. default: 8KHz*/
int s32FrameSample; /* VQE frame length: 80-4096 */
VQE_WORKSTATE_E enWorkstate;
AUDIO_HPF_CONFIG_S stHpfCfg;
AUDIO_ANR_CONFIG_S stAnrCfg;
AUDIO_AGC_CONFIG_S stAgcCfg;
AUDIO_EQ_CONFIG_S stEqCfg;
AUDIO_GAIN_CONFIG_S stGainCfg;
} AO_VQE_CONFIG_S;
typedef struct AO_DRC_CONFIG_S
{
int sampling_rate;
int target_level;
int max_gain;
int min_gain;
int attack_time;
int release_time;
int noise_threshold;
}AO_DRC_CONFIG_S;
typedef struct AI_VQE_INFO_S
{
BOOL bVqeEnable; /* vqe enable or disable */
AI_VQE_CONFIG_S stAiVqeCfg;
} AI_VQE_INFO_S;
typedef struct AO_VQE_INFO_S
{
BOOL bVqeEnable; /* vqe enable or disable */
AO_VQE_CONFIG_S stAoVqeCfg;
} AO_VQE_INFO_S;
/**Defines the state of inner codec.*/
typedef struct AI_INNER_CODEC_STATE_S
{
BOOL bMicInl;
BOOL bMicInr;
} AI_INNER_CODEC_STATE_S;
/*Defines the configure parameters of AI saving file.*/
typedef struct AUDIO_SAVE_FILE_INFO_S
{
BOOL bCfg;
char mFilePath[256];
char mFileName[256];
unsigned int mFileSize; /*in Bytes*/
} AUDIO_SAVE_FILE_INFO_S;
typedef enum EN_AIO_ERR_CODE_E
{
AIO_ERR_CHN_NOT_ENABLE = 64,
} EN_ADEC_ERR_CODE_E;
typedef struct UvoiceLicenseParam
{
char license[128];
char license_path[128];
char uuid[128];
} UvoiceLicenseParam;
typedef struct UvoiceServerHandshake
{
const char *pPostBody;
char response[1024];
} UvoiceServerHandshake;
typedef enum AudioDevEventType
{
/**
if we want to use uvoice aec lib, we need get license, to get license, we need to know license code, uuid, filePath.
pEventData = UvoiceLicenseParam.
*/
AudioDevEvent_GetUvoiceLicenseParam = 0,
/**
use curl to post bodyMessage to uvoice server, get server response string.
pEventData = UvoiceServerHandshake
*/
AudioDevEvent_GetUvoiceServerResponse,
AudioDevEvent_Max = 0x7FFFFFFF
} AudioDevEventType;
typedef ERRORTYPE (*AudioDevCallbackFuncType)(void *cookie, AUDIO_DEV nAudioDevId, AudioDevEventType eEventType, int nPara0, void *pEventData);
/**
aec process callback.
@return
0: valid frame is ready
1: valid frame is not ready, process is successful.
-1:valid frame is not ready, process is failure.
*/
typedef int (*AecProcessFuncType)(void *cookie, AUDIO_FRAME_S *pFrm, BOOL bSuspendAec);
/**
ans process callback.
@param bSuspendAns
if suspend ans, copy data directly.
@return
0: valid frame is ready
1: valid frame is not ready, process is successful.
-1:valid frame is not ready, process is failure.
*/
typedef int (*AnsProcessFuncType)(void *cookie, AUDIO_FRAME_S *pFrm, const AIO_ATTR_S *pAttr, BOOL bSuspendAns);
typedef struct AI_CHN_ATTR_S
{
int nFrameSize; //unit:samples in one channel. 0:use default value, i.e. 1024.
} AI_CHN_ATTR_S;
/* invlalid device ID */
#define ERR_AI_INVALID_DEVID DEF_ERR(MOD_ID_AI, EN_ERR_LEVEL_ERROR, EN_ERR_INVALID_DEVID)
/* invlalid channel ID */
#define ERR_AI_INVALID_CHNID DEF_ERR(MOD_ID_AI, EN_ERR_LEVEL_ERROR, EN_ERR_INVALID_CHNID)
/* at lease one parameter is illagal ,eg, an illegal enumeration value */
#define ERR_AI_ILLEGAL_PARAM DEF_ERR(MOD_ID_AI, EN_ERR_LEVEL_ERROR, EN_ERR_ILLEGAL_PARAM)
/* channel exists */
#define ERR_AI_EXIST DEF_ERR(MOD_ID_AI, EN_ERR_LEVEL_ERROR, EN_ERR_EXIST)
/* channel exists */
#define ERR_AI_UNEXIST DEF_ERR(MOD_ID_AI, EN_ERR_LEVEL_ERROR, EN_ERR_UNEXIST)
/* using a NULL point */
#define ERR_AI_NULL_PTR DEF_ERR(MOD_ID_AI, EN_ERR_LEVEL_ERROR, EN_ERR_NULL_PTR)
/* try to enable or initialize system,device or channel, before configing attribute */
#define ERR_AI_NOT_CONFIG DEF_ERR(MOD_ID_AI, EN_ERR_LEVEL_ERROR, EN_ERR_NOT_CONFIG)
/* operation is not supported by NOW */
#define ERR_AI_NOT_SUPPORT DEF_ERR(MOD_ID_AI, EN_ERR_LEVEL_ERROR, EN_ERR_NOT_SUPPORT)
/* operation is not permitted ,eg, try to change stati attribute */
#define ERR_AI_NOT_PERM DEF_ERR(MOD_ID_AI, EN_ERR_LEVEL_ERROR, EN_ERR_NOT_PERM)
/* the devide is not enabled */
#define ERR_AI_NOT_ENABLED DEF_ERR(MOD_ID_AI, EN_ERR_LEVEL_ERROR, EN_ERR_UNEXIST)
/* failure caused by malloc memory */
#define ERR_AI_NOMEM DEF_ERR(MOD_ID_AI, EN_ERR_LEVEL_ERROR, EN_ERR_NOMEM)
/* failure caused by malloc buffer */
#define ERR_AI_NOBUF DEF_ERR(MOD_ID_AI, EN_ERR_LEVEL_ERROR, EN_ERR_NOBUF)
/* no data in buffer */
#define ERR_AI_BUF_EMPTY DEF_ERR(MOD_ID_AI, EN_ERR_LEVEL_ERROR, EN_ERR_BUF_EMPTY)
/* no buffer for new data */
#define ERR_AI_BUF_FULL DEF_ERR(MOD_ID_AI, EN_ERR_LEVEL_ERROR, EN_ERR_BUF_FULL)
/* system is not ready,had not initialed or loaded*/
#define ERR_AI_SYS_NOTREADY DEF_ERR(MOD_ID_AI, EN_ERR_LEVEL_ERROR, EN_ERR_SYS_NOTREADY)
#define ERR_AI_BUSY DEF_ERR(MOD_ID_AI, EN_ERR_LEVEL_ERROR, EN_ERR_BUSY)
/* component state is same as user wanted */
#define ERR_AI_SAMESTATE DEF_ERR(MOD_ID_AI, EN_ERR_LEVEL_ERROR, EN_ERR_SAMESTATE)
/* component state is transit to invalid state */
#define ERR_AI_INVALIDSTATE DEF_ERR(MOD_ID_AI, EN_ERR_LEVEL_ERROR, EN_ERR_INVALIDSTATE)
/* component current state can't transit to destination state */
#define ERR_AI_INCORRECT_STATE_TRANSITION DEF_ERR(MOD_ID_AI, EN_ERR_LEVEL_ERROR, EN_ERR_INCORRECT_STATE_TRANSITION)
/* Attempting a command that is not allowed during the present state. */
#define ERR_AI_INCORRECT_STATE_OPERATION DEF_ERR(MOD_ID_AI, EN_ERR_LEVEL_ERROR, EN_ERR_INCORRECT_STATE_OPERATION)
/* invlalid device ID */
#define ERR_AO_INVALID_DEVID DEF_ERR(MOD_ID_AO, EN_ERR_LEVEL_ERROR, EN_ERR_INVALID_DEVID)
/* invlalid channel ID */
#define ERR_AO_INVALID_CHNID DEF_ERR(MOD_ID_AO, EN_ERR_LEVEL_ERROR, EN_ERR_INVALID_CHNID)
/* at lease one parameter is illagal ,eg, an illegal enumeration value */
#define ERR_AO_ILLEGAL_PARAM DEF_ERR(MOD_ID_AO, EN_ERR_LEVEL_ERROR, EN_ERR_ILLEGAL_PARAM)
/* channel exists */
#define ERR_AO_EXIST DEF_ERR(MOD_ID_AO, EN_ERR_LEVEL_ERROR, EN_ERR_EXIST)
/* channel exists */
#define ERR_AO_UNEXIST DEF_ERR(MOD_ID_AO, EN_ERR_LEVEL_ERROR, EN_ERR_UNEXIST)
/* using a NULL point */
#define ERR_AO_NULL_PTR DEF_ERR(MOD_ID_AO, EN_ERR_LEVEL_ERROR, EN_ERR_NULL_PTR)
/* try to enable or initialize system,device or channel, before configing attribute */
#define ERR_AO_NOT_CONFIG DEF_ERR(MOD_ID_AO, EN_ERR_LEVEL_ERROR, EN_ERR_NOT_CONFIG)
/* operation is not supported by NOW */
#define ERR_AO_NOT_SUPPORT DEF_ERR(MOD_ID_AO, EN_ERR_LEVEL_ERROR, EN_ERR_NOT_SUPPORT)
/* operation is not permitted ,eg, try to change stati attribute */
#define ERR_AO_NOT_PERM DEF_ERR(MOD_ID_AO, EN_ERR_LEVEL_ERROR, EN_ERR_NOT_PERM)
/* the devide is not enabled */
#define ERR_AO_NOT_ENABLED DEF_ERR(MOD_ID_AO, EN_ERR_LEVEL_ERROR, EN_ERR_UNEXIST)
/* failure caused by malloc memory */
#define ERR_AO_NOMEM DEF_ERR(MOD_ID_AO, EN_ERR_LEVEL_ERROR, EN_ERR_NOMEM)
/* failure caused by malloc buffer */
#define ERR_AO_NOBUF DEF_ERR(MOD_ID_AO, EN_ERR_LEVEL_ERROR, EN_ERR_NOBUF)
/* no data in buffer */
#define ERR_AO_BUF_EMPTY DEF_ERR(MOD_ID_AO, EN_ERR_LEVEL_ERROR, EN_ERR_BUF_EMPTY)
/* no buffer for new data */
#define ERR_AO_BUF_FULL DEF_ERR(MOD_ID_AO, EN_ERR_LEVEL_ERROR, EN_ERR_BUF_FULL)
/* system is not ready,had not initialed or loaded*/
#define ERR_AO_SYS_NOTREADY DEF_ERR(MOD_ID_AO, EN_ERR_LEVEL_ERROR, EN_ERR_SYS_NOTREADY)
#define ERR_AO_BUSY DEF_ERR(MOD_ID_AO, EN_ERR_LEVEL_ERROR, EN_ERR_BUSY)
/* component state is same as user wanted */
#define ERR_AO_SAMESTATE DEF_ERR(MOD_ID_AO, EN_ERR_LEVEL_ERROR, EN_ERR_SAMESTATE)
/* component state is transit to invalid state */
#define ERR_AO_INVALIDSTATE DEF_ERR(MOD_ID_AO, EN_ERR_LEVEL_ERROR, EN_ERR_INVALIDSTATE)
/* component current state can't transit to destination state */
#define ERR_AO_INCORRECT_STATE_TRANSITION DEF_ERR(MOD_ID_AO, EN_ERR_LEVEL_ERROR, EN_ERR_INCORRECT_STATE_TRANSITION)
/* Attempting a command that is not allowed during the present state. */
#define ERR_AO_INCORRECT_STATE_OPERATION DEF_ERR(MOD_ID_AO, EN_ERR_LEVEL_ERROR, EN_ERR_INCORRECT_STATE_OPERATION)
#ifdef __cplusplus
}
#endif
#endif /* __IPCLINUX_COMM_AI_H__ */
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