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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
/*
* codec.h
*
* This header file contains the calls to the internal encoder
* and decoder functions.
*
*/
#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CODEC_H_
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CODEC_H_
#include "structs.h"
void WebRtcIsac_ResetBitstream(Bitstr* bit_stream);
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,
enum IsacSamplingRate encoderSampRate,
enum IsacSamplingRate decoderSampRate);
int WebRtcIsac_DecodeLb(const TransformTables* transform_tables,
float* signal_out,
ISACLBDecStruct* ISACdec_obj,
int16_t* current_framesamples,
int16_t isRCUPayload);
int WebRtcIsac_DecodeRcuLb(float* signal_out, ISACLBDecStruct* ISACdec_obj,
int16_t* current_framesamples);
int WebRtcIsac_EncodeLb(const TransformTables* transform_tables,
float* in,
ISACLBEncStruct* ISACencLB_obj,
int16_t codingMode,
int16_t bottleneckIndex);
int WebRtcIsac_EncodeStoredDataLb(const IsacSaveEncoderData* ISACSavedEnc_obj,
Bitstr* ISACBitStr_obj, int BWnumber,
float scale);
int WebRtcIsac_EncodeStoredDataUb(
const ISACUBSaveEncDataStruct* ISACSavedEnc_obj, Bitstr* bitStream,
int32_t jitterInfo, float scale, enum ISACBandwidth bandwidth);
int16_t WebRtcIsac_GetRedPayloadUb(
const ISACUBSaveEncDataStruct* ISACSavedEncObj, Bitstr* bitStreamObj,
enum ISACBandwidth bandwidth);
/******************************************************************************
* WebRtcIsac_RateAllocation()
* Internal function to perform a rate-allocation for upper and lower-band,
* given a total rate.
*
* Input:
* - inRateBitPerSec : a total bit-rate in bits/sec.
*
* Output:
* - rateLBBitPerSec : a bit-rate allocated to the lower-band
* in bits/sec.
* - rateUBBitPerSec : a bit-rate allocated to the upper-band
* in bits/sec.
*
* Return value : 0 if rate allocation has been successful.
* -1 if failed to allocate rates.
*/
int16_t WebRtcIsac_RateAllocation(int32_t inRateBitPerSec,
double* rateLBBitPerSec,
double* rateUBBitPerSec,
enum ISACBandwidth* bandwidthKHz);
/******************************************************************************
* WebRtcIsac_DecodeUb16()
*
* Decode the upper-band if the codec is in 0-16 kHz mode.
*
* Input/Output:
* -ISACdec_obj : pointer to the upper-band decoder object. The
* bit-stream is stored inside the decoder object.
*
* Output:
* -signal_out : decoded audio, 480 samples 30 ms.
*
* Return value : >0 number of decoded bytes.
* <0 if an error occurred.
*/
int WebRtcIsac_DecodeUb16(const TransformTables* transform_tables,
float* signal_out,
ISACUBDecStruct* ISACdec_obj,
int16_t isRCUPayload);
/******************************************************************************
* WebRtcIsac_DecodeUb12()
*
* Decode the upper-band if the codec is in 0-12 kHz mode.
*
* Input/Output:
* -ISACdec_obj : pointer to the upper-band decoder object. The
* bit-stream is stored inside the decoder object.
*
* Output:
* -signal_out : decoded audio, 480 samples 30 ms.
*
* Return value : >0 number of decoded bytes.
* <0 if an error occurred.
*/
int WebRtcIsac_DecodeUb12(const TransformTables* transform_tables,
float* signal_out,
ISACUBDecStruct* ISACdec_obj,
int16_t isRCUPayload);
/******************************************************************************
* WebRtcIsac_EncodeUb16()
*
* Encode the upper-band if the codec is in 0-16 kHz mode.
*
* Input:
* -in : upper-band audio, 160 samples (10 ms).
*
* Input/Output:
* -ISACdec_obj : pointer to the upper-band encoder object. The
* bit-stream is stored inside the encoder object.
*
* Return value : >0 number of encoded bytes.
* <0 if an error occurred.
*/
int WebRtcIsac_EncodeUb16(const TransformTables* transform_tables,
float* in,
ISACUBEncStruct* ISACenc_obj,
int32_t jitterInfo);
/******************************************************************************
* WebRtcIsac_EncodeUb12()
*
* Encode the upper-band if the codec is in 0-12 kHz mode.
*
* Input:
* -in : upper-band audio, 160 samples (10 ms).
*
* Input/Output:
* -ISACdec_obj : pointer to the upper-band encoder object. The
* bit-stream is stored inside the encoder object.
*
* Return value : >0 number of encoded bytes.
* <0 if an error occurred.
*/
int WebRtcIsac_EncodeUb12(const TransformTables* transform_tables,
float* in,
ISACUBEncStruct* ISACenc_obj,
int32_t jitterInfo);
/************************** initialization functions *************************/
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);
void WebRtcIsac_Time2Spec(const TransformTables* tables,
double* inre1,
double* inre2,
int16_t* outre,
int16_t* outim,
FFTstr* fftstr_obj);
void WebRtcIsac_Spec2time(const TransformTables* tables,
double* inre,
double* inim,
double* outre1,
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,
PostFiltBankstr* postfiltdata);
/************************* normalized lattice filters ************************/
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,
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_ */
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