AnotherFoxGuy/webrtc-audio-processing
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Update to current webrtc library

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This is from the upstream library commit id
3326535126e435f1ba647885ce43a8f0f3d317eb, corresponding to Chromium
88.0.4290.1.
This commit is contained in:
Arun Raghavan
2020-10-12 18:08:02 -04:00
parent b1b02581d3
commit bcec8b0b21
859 changed files with 76187 additions and 49580 deletions
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78
webrtc/modules/audio_coding/codecs/isac/main/source/codec.h
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@ -16,18 +16,22 @@
*
*/
#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CODEC_H_
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CODEC_H_
#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CODEC_H_
#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CODEC_H_
#include "structs.h"
#include <stddef.h>
#include "modules/audio_coding/codecs/isac/main/source/structs.h"
#include "modules/third_party/fft/fft.h"
void WebRtcIsac_ResetBitstream(Bitstr* bit_stream);
int WebRtcIsac_EstimateBandwidth(BwEstimatorstr* bwest_str, Bitstr* streamdata,
int WebRtcIsac_EstimateBandwidth(BwEstimatorstr* bwest_str,
Bitstr* streamdata,
size_t packet_size,
uint16_t rtp_seq_number,
uint32_t send_ts, uint32_t arr_ts,
uint32_t send_ts,
uint32_t arr_ts,
enum IsacSamplingRate encoderSampRate,
enum IsacSamplingRate decoderSampRate);
@ -37,7 +41,8 @@ int WebRtcIsac_DecodeLb(const TransformTables* transform_tables,
int16_t* current_framesamples,
int16_t isRCUPayload);
int WebRtcIsac_DecodeRcuLb(float* signal_out, ISACLBDecStruct* ISACdec_obj,
int WebRtcIsac_DecodeRcuLb(float* signal_out,
ISACLBDecStruct* ISACdec_obj,
int16_t* current_framesamples);
int WebRtcIsac_EncodeLb(const TransformTables* transform_tables,
@ -47,15 +52,20 @@ int WebRtcIsac_EncodeLb(const TransformTables* transform_tables,
int16_t bottleneckIndex);
int WebRtcIsac_EncodeStoredDataLb(const IsacSaveEncoderData* ISACSavedEnc_obj,
Bitstr* ISACBitStr_obj, int BWnumber,
Bitstr* ISACBitStr_obj,
int BWnumber,
float scale);
int WebRtcIsac_EncodeStoredDataUb(
const ISACUBSaveEncDataStruct* ISACSavedEnc_obj, Bitstr* bitStream,
int32_t jitterInfo, float scale, enum ISACBandwidth bandwidth);
const ISACUBSaveEncDataStruct* ISACSavedEnc_obj,
Bitstr* bitStream,
int32_t jitterInfo,
float scale,
enum ISACBandwidth bandwidth);
int16_t WebRtcIsac_GetRedPayloadUb(
const ISACUBSaveEncDataStruct* ISACSavedEncObj, Bitstr* bitStreamObj,
const ISACUBSaveEncDataStruct* ISACSavedEncObj,
Bitstr* bitStreamObj,
enum ISACBandwidth bandwidth);
/******************************************************************************
@ -81,7 +91,6 @@ int16_t WebRtcIsac_RateAllocation(int32_t inRateBitPerSec,
double* rateUBBitPerSec,
enum ISACBandwidth* bandwidthKHz);
/******************************************************************************
* WebRtcIsac_DecodeUb16()
*
@ -166,15 +175,8 @@ int WebRtcIsac_EncodeUb12(const TransformTables* transform_tables,
void WebRtcIsac_InitMasking(MaskFiltstr* maskdata);
void WebRtcIsac_InitPreFilterbank(PreFiltBankstr* prefiltdata);
void WebRtcIsac_InitPostFilterbank(PostFiltBankstr* postfiltdata);
void WebRtcIsac_InitPitchFilter(PitchFiltstr* pitchfiltdata);
void WebRtcIsac_InitPitchAnalysis(PitchAnalysisStruct* State);
/**************************** transform functions ****************************/
void WebRtcIsac_InitTransform(TransformTables* tables);
@ -193,41 +195,29 @@ void WebRtcIsac_Spec2time(const TransformTables* tables,
double* outre2,
FFTstr* fftstr_obj);
/******************************* filter functions ****************************/
void WebRtcIsac_AllPoleFilter(double* InOut, double* Coef, size_t lengthInOut,
int orderCoef);
void WebRtcIsac_AllZeroFilter(double* In, double* Coef, size_t lengthInOut,
int orderCoef, double* Out);
void WebRtcIsac_ZeroPoleFilter(double* In, double* ZeroCoef, double* PoleCoef,
size_t lengthInOut, int orderCoef, double* Out);
/***************************** filterbank functions **************************/
void WebRtcIsac_SplitAndFilterFloat(float* in, float* LP, float* HP,
double* LP_la, double* HP_la,
PreFiltBankstr* prefiltdata);
void WebRtcIsac_FilterAndCombineFloat(float* InLP, float* InHP, float* Out,
void WebRtcIsac_FilterAndCombineFloat(float* InLP,
float* InHP,
float* Out,
PostFiltBankstr* postfiltdata);
/************************* normalized lattice filters ************************/
void WebRtcIsac_NormLatticeFilterMa(int orderCoef, float* stateF, float* stateG,
float* lat_in, double* filtcoeflo,
void WebRtcIsac_NormLatticeFilterMa(int orderCoef,
float* stateF,
float* stateG,
float* lat_in,
double* filtcoeflo,
double* lat_out);
void WebRtcIsac_NormLatticeFilterAr(int orderCoef, float* stateF, float* stateG,
double* lat_in, double* lo_filt_coef,
void WebRtcIsac_NormLatticeFilterAr(int orderCoef,
float* stateF,
float* stateG,
double* lat_in,
double* lo_filt_coef,
float* lat_out);
void WebRtcIsac_Dir2Lat(double* a, int orderCoef, float* sth, float* cth);
void WebRtcIsac_AutoCorr(double* r, const double* x, size_t N, size_t order);
#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CODEC_H_ */
#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CODEC_H_ */