Update audio_processing module
Corresponds to upstream commit 524e9b043e7e86fd72353b987c9d5f6a1ebf83e1 Update notes: * Pull in third party license file * Replace .gypi files with BUILD.gn to keep track of what changes upstream * Bunch of new filse pulled in as dependencies * Won't build yet due to changes needed on top of these
This commit is contained in:
Arun Raghavan
233
webrtc/modules/audio_coding/codecs/isac/main/source/codec.h
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233
webrtc/modules/audio_coding/codecs/isac/main/source/codec.h
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/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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/*
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* codec.h
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*
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* This header file contains the calls to the internal encoder
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* and decoder functions.
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*
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*/
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#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CODEC_H_
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#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CODEC_H_
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#include "structs.h"
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void WebRtcIsac_ResetBitstream(Bitstr* bit_stream);
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int WebRtcIsac_EstimateBandwidth(BwEstimatorstr* bwest_str, Bitstr* streamdata,
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size_t packet_size,
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uint16_t rtp_seq_number,
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uint32_t send_ts, uint32_t arr_ts,
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enum IsacSamplingRate encoderSampRate,
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enum IsacSamplingRate decoderSampRate);
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int WebRtcIsac_DecodeLb(const TransformTables* transform_tables,
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float* signal_out,
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ISACLBDecStruct* ISACdec_obj,
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int16_t* current_framesamples,
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int16_t isRCUPayload);
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int WebRtcIsac_DecodeRcuLb(float* signal_out, ISACLBDecStruct* ISACdec_obj,
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int16_t* current_framesamples);
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int WebRtcIsac_EncodeLb(const TransformTables* transform_tables,
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float* in,
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ISACLBEncStruct* ISACencLB_obj,
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int16_t codingMode,
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int16_t bottleneckIndex);
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int WebRtcIsac_EncodeStoredDataLb(const IsacSaveEncoderData* ISACSavedEnc_obj,
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Bitstr* ISACBitStr_obj, int BWnumber,
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float scale);
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int WebRtcIsac_EncodeStoredDataUb(
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const ISACUBSaveEncDataStruct* ISACSavedEnc_obj, Bitstr* bitStream,
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int32_t jitterInfo, float scale, enum ISACBandwidth bandwidth);
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int16_t WebRtcIsac_GetRedPayloadUb(
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const ISACUBSaveEncDataStruct* ISACSavedEncObj, Bitstr* bitStreamObj,
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enum ISACBandwidth bandwidth);
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/******************************************************************************
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* WebRtcIsac_RateAllocation()
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* Internal function to perform a rate-allocation for upper and lower-band,
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* given a total rate.
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*
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* Input:
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* - inRateBitPerSec : a total bit-rate in bits/sec.
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*
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* Output:
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* - rateLBBitPerSec : a bit-rate allocated to the lower-band
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* in bits/sec.
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* - rateUBBitPerSec : a bit-rate allocated to the upper-band
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* in bits/sec.
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*
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* Return value : 0 if rate allocation has been successful.
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* -1 if failed to allocate rates.
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*/
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int16_t WebRtcIsac_RateAllocation(int32_t inRateBitPerSec,
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double* rateLBBitPerSec,
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double* rateUBBitPerSec,
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enum ISACBandwidth* bandwidthKHz);
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/******************************************************************************
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* WebRtcIsac_DecodeUb16()
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*
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* Decode the upper-band if the codec is in 0-16 kHz mode.
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*
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* Input/Output:
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* -ISACdec_obj : pointer to the upper-band decoder object. The
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* bit-stream is stored inside the decoder object.
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*
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* Output:
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* -signal_out : decoded audio, 480 samples 30 ms.
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*
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* Return value : >0 number of decoded bytes.
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* <0 if an error occurred.
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*/
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int WebRtcIsac_DecodeUb16(const TransformTables* transform_tables,
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float* signal_out,
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ISACUBDecStruct* ISACdec_obj,
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int16_t isRCUPayload);
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/******************************************************************************
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* WebRtcIsac_DecodeUb12()
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*
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* Decode the upper-band if the codec is in 0-12 kHz mode.
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*
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* Input/Output:
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* -ISACdec_obj : pointer to the upper-band decoder object. The
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* bit-stream is stored inside the decoder object.
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*
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* Output:
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* -signal_out : decoded audio, 480 samples 30 ms.
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*
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* Return value : >0 number of decoded bytes.
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* <0 if an error occurred.
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*/
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int WebRtcIsac_DecodeUb12(const TransformTables* transform_tables,
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float* signal_out,
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ISACUBDecStruct* ISACdec_obj,
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int16_t isRCUPayload);
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/******************************************************************************
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* WebRtcIsac_EncodeUb16()
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*
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* Encode the upper-band if the codec is in 0-16 kHz mode.
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*
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* Input:
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* -in : upper-band audio, 160 samples (10 ms).
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*
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* Input/Output:
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* -ISACdec_obj : pointer to the upper-band encoder object. The
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* bit-stream is stored inside the encoder object.
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*
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* Return value : >0 number of encoded bytes.
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* <0 if an error occurred.
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*/
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int WebRtcIsac_EncodeUb16(const TransformTables* transform_tables,
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float* in,
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ISACUBEncStruct* ISACenc_obj,
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int32_t jitterInfo);
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/******************************************************************************
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* WebRtcIsac_EncodeUb12()
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*
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* Encode the upper-band if the codec is in 0-12 kHz mode.
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*
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* Input:
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* -in : upper-band audio, 160 samples (10 ms).
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*
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* Input/Output:
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* -ISACdec_obj : pointer to the upper-band encoder object. The
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* bit-stream is stored inside the encoder object.
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*
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* Return value : >0 number of encoded bytes.
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* <0 if an error occurred.
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*/
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int WebRtcIsac_EncodeUb12(const TransformTables* transform_tables,
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float* in,
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ISACUBEncStruct* ISACenc_obj,
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int32_t jitterInfo);
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/************************** initialization functions *************************/
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void WebRtcIsac_InitMasking(MaskFiltstr* maskdata);
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void WebRtcIsac_InitPreFilterbank(PreFiltBankstr* prefiltdata);
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void WebRtcIsac_InitPostFilterbank(PostFiltBankstr* postfiltdata);
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void WebRtcIsac_InitPitchFilter(PitchFiltstr* pitchfiltdata);
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void WebRtcIsac_InitPitchAnalysis(PitchAnalysisStruct* State);
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/**************************** transform functions ****************************/
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void WebRtcIsac_InitTransform(TransformTables* tables);
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void WebRtcIsac_Time2Spec(const TransformTables* tables,
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double* inre1,
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double* inre2,
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int16_t* outre,
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int16_t* outim,
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FFTstr* fftstr_obj);
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void WebRtcIsac_Spec2time(const TransformTables* tables,
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double* inre,
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double* inim,
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double* outre1,
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double* outre2,
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FFTstr* fftstr_obj);
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/******************************* filter functions ****************************/
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void WebRtcIsac_AllPoleFilter(double* InOut, double* Coef, size_t lengthInOut,
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int orderCoef);
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void WebRtcIsac_AllZeroFilter(double* In, double* Coef, size_t lengthInOut,
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int orderCoef, double* Out);
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void WebRtcIsac_ZeroPoleFilter(double* In, double* ZeroCoef, double* PoleCoef,
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size_t lengthInOut, int orderCoef, double* Out);
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/***************************** filterbank functions **************************/
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void WebRtcIsac_SplitAndFilterFloat(float* in, float* LP, float* HP,
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double* LP_la, double* HP_la,
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PreFiltBankstr* prefiltdata);
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void WebRtcIsac_FilterAndCombineFloat(float* InLP, float* InHP, float* Out,
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PostFiltBankstr* postfiltdata);
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/************************* normalized lattice filters ************************/
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void WebRtcIsac_NormLatticeFilterMa(int orderCoef, float* stateF, float* stateG,
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float* lat_in, double* filtcoeflo,
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double* lat_out);
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void WebRtcIsac_NormLatticeFilterAr(int orderCoef, float* stateF, float* stateG,
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double* lat_in, double* lo_filt_coef,
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float* lat_out);
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void WebRtcIsac_Dir2Lat(double* a, int orderCoef, float* sth, float* cth);
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void WebRtcIsac_AutoCorr(double* r, const double* x, size_t N, size_t order);
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#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CODEC_H_ */
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