mirror of
https://github.com/signalwire/freeswitch.git
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413 lines
16 KiB
C
413 lines
16 KiB
C
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/*
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* Copyright (c) 2011 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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* entropy_coding.h
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*
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* This header file declares all of the functions used to arithmetically
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* encode the iSAC bistream
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*
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*/
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#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ENTROPY_CODING_H_
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#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ENTROPY_CODING_H_
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#include "structs.h"
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/* decode complex spectrum (return number of bytes in stream) */
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int WebRtcIsac_DecodeSpecLb(Bitstr *streamdata,
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double *fr,
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double *fi,
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WebRtc_Word16 AvgPitchGain_Q12);
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/******************************************************************************
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* WebRtcIsac_DecodeSpecUB16()
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* Decode real and imaginary part of the DFT coefficients, given a bit-stream.
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* This function is called when the codec is in 0-16 kHz bandwidth.
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* The decoded DFT coefficient can be transformed to time domain by
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* WebRtcIsac_Time2Spec().
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*
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* Input:
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* - streamdata : pointer to a stucture containg the encoded
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* data and theparameters needed for entropy
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* coding.
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*
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* Output:
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* -*fr : pointer to a buffer where the real part of DFT
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* coefficients are written to.
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* -*fi : pointer to a buffer where the imaginary part
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* of DFT coefficients are written to.
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*
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* Return value : < 0 if an error occures
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* 0 if succeeded.
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*/
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int WebRtcIsac_DecodeSpecUB16(
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Bitstr* streamdata,
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double* fr,
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double* fi);
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/******************************************************************************
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* WebRtcIsac_DecodeSpecUB12()
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* Decode real and imaginary part of the DFT coefficients, given a bit-stream.
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* This function is called when the codec is in 0-12 kHz bandwidth.
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* The decoded DFT coefficient can be transformed to time domain by
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* WebRtcIsac_Time2Spec().
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*
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* Input:
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* - streamdata : pointer to a stucture containg the encoded
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* data and theparameters needed for entropy
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* coding.
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*
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* Output:
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* -*fr : pointer to a buffer where the real part of DFT
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* coefficients are written to.
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* -*fi : pointer to a buffer where the imaginary part
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* of DFT coefficients are written to.
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*
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* Return value : < 0 if an error occures
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* 0 if succeeded.
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*/
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int WebRtcIsac_DecodeSpecUB12(
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Bitstr* streamdata,
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double* fr,
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double* fi);
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/* encode complex spectrum */
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int WebRtcIsac_EncodeSpecLb(const WebRtc_Word16* fr,
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const WebRtc_Word16* fi,
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Bitstr* streamdata,
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WebRtc_Word16 AvgPitchGain_Q12);
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/******************************************************************************
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* WebRtcIsac_EncodeSpecUB16()
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* Quantize and encode real and imaginary part of the DFT coefficients.
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* This function is called when the codec is in 0-16 kHz bandwidth.
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* The real and imaginary part are computed by calling WebRtcIsac_Time2Spec().
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*
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*
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* Input:
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* -*fr : pointer to a buffer where the real part of DFT
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* coefficients are stored.
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* -*fi : pointer to a buffer where the imaginary part
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* of DFT coefficients are stored.
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*
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* Output:
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* - streamdata : pointer to a stucture containg the encoded
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* data and theparameters needed for entropy
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* coding.
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*
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* Return value : < 0 if an error occures
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* 0 if succeeded.
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*/
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int WebRtcIsac_EncodeSpecUB16(
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const WebRtc_Word16* fr,
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const WebRtc_Word16* fi,
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Bitstr* streamdata);
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/******************************************************************************
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* WebRtcIsac_EncodeSpecUB12()
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* Quantize and encode real and imaginary part of the DFT coefficients.
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* This function is called when the codec is in 0-12 kHz bandwidth.
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* The real and imaginary part are computed by calling WebRtcIsac_Time2Spec().
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*
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*
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* Input:
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* -*fr : pointer to a buffer where the real part of DFT
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* coefficients are stored.
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* -*fi : pointer to a buffer where the imaginary part
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* of DFT coefficients are stored.
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*
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* Output:
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* - streamdata : pointer to a stucture containg the encoded
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* data and theparameters needed for entropy
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* coding.
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*
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* Return value : < 0 if an error occures
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* 0 if succeeded.
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*/
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int WebRtcIsac_EncodeSpecUB12(
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const WebRtc_Word16* fr,
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const WebRtc_Word16* fi,
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Bitstr* streamdata);
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/* decode & dequantize LPC Coef */
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int WebRtcIsac_DecodeLpcCoef(Bitstr *streamdata, double *LPCCoef, int *outmodel);
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int WebRtcIsac_DecodeLpcCoefUB(
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Bitstr* streamdata,
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double* lpcVecs,
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double* percepFilterGains,
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WebRtc_Word16 bandwidth);
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int WebRtcIsac_DecodeLpc(Bitstr *streamdata, double *LPCCoef_lo, double *LPCCoef_hi, int *outmodel);
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/* quantize & code LPC Coef */
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void WebRtcIsac_EncodeLpcLb(double *LPCCoef_lo, double *LPCCoef_hi, int *model, double *size, Bitstr *streamdata, ISAC_SaveEncData_t* encData);
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void WebRtcIsac_EncodeLpcGainLb(double *LPCCoef_lo, double *LPCCoef_hi, int model, Bitstr *streamdata, ISAC_SaveEncData_t* encData);
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/******************************************************************************
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* WebRtcIsac_EncodeLpcUB()
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* Encode LPC parameters, given as A-polynomial, of upper-band. The encoding
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* is performed in LAR domain.
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* For the upper-band, we compute and encode LPC of some sub-frames, LPC of
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* other sub-frames are computed by linear interpolation, in LAR domain. This
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* function performs the interpolation and returns the LPC of all sub-frames.
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*
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* Inputs:
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* - lpcCoef : a buffer containing A-polynomials of sub-frames
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* (excluding first coefficient that is 1).
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* - bandwidth : specifies if the codec is operating at 0-12 kHz
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* or 0-16 kHz mode.
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*
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* Input/output:
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* - streamdata : pointer to a stucture containg the encoded
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* data and theparameters needed for entropy
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* coding.
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*
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* Output:
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* - interpolLPCCoeff : Decoded and interpolated LPC (A-polynomial)
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* of all sub-frames.
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* If LP analysis is of order K, and there are N
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* sub-frames then this is a buffer of size
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* (k + 1) * N, each vector starts with the LPC gain
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* of the corresponding sub-frame. The LPC gains
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* are encoded and inserted after this function is
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* called. The first A-coefficient which is 1 is not
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* included.
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*
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* Return value : 0 if encoding is successful,
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* <0 if failed to encode.
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*/
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WebRtc_Word16 WebRtcIsac_EncodeLpcUB(
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double* lpcCoeff,
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Bitstr* streamdata,
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double* interpolLPCCoeff,
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WebRtc_Word16 bandwidth,
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ISACUBSaveEncDataStruct* encData);
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/******************************************************************************
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* WebRtcIsac_DecodeInterpolLpcUb()
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* Decode LPC coefficients and interpolate to get the coefficients fo all
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* sub-frmaes.
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*
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* Inputs:
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* - bandwidth : spepecifies if the codec is in 0-12 kHz or
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* 0-16 kHz mode.
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*
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* Input/output:
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* - streamdata : pointer to a stucture containg the encoded
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* data and theparameters needed for entropy
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* coding.
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*
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* Output:
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* - percepFilterParam : Decoded and interpolated LPC (A-polynomial) of
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* all sub-frames.
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* If LP analysis is of order K, and there are N
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* sub-frames then this is a buffer of size
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* (k + 1) * N, each vector starts with the LPC gain
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* of the corresponding sub-frame. The LPC gains
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* are encoded and inserted after this function is
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* called. The first A-coefficient which is 1 is not
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* included.
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*
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* Return value : 0 if encoding is successful,
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* <0 if failed to encode.
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*/
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WebRtc_Word16 WebRtcIsac_DecodeInterpolLpcUb(
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Bitstr* streamdata,
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double* percepFilterParam,
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WebRtc_Word16 bandwidth);
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/* decode & dequantize RC */
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int WebRtcIsac_DecodeRc(Bitstr *streamdata, WebRtc_Word16 *RCQ15);
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/* quantize & code RC */
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void WebRtcIsac_EncodeRc(WebRtc_Word16 *RCQ15, Bitstr *streamdata);
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/* decode & dequantize squared Gain */
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int WebRtcIsac_DecodeGain2(Bitstr *streamdata, WebRtc_Word32 *Gain2);
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/* quantize & code squared Gain (input is squared gain) */
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int WebRtcIsac_EncodeGain2(WebRtc_Word32 *gain2, Bitstr *streamdata);
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void WebRtcIsac_EncodePitchGain(WebRtc_Word16* PitchGains_Q12, Bitstr* streamdata, ISAC_SaveEncData_t* encData);
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void WebRtcIsac_EncodePitchLag(double* PitchLags, WebRtc_Word16* PitchGain_Q12, Bitstr* streamdata, ISAC_SaveEncData_t* encData);
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int WebRtcIsac_DecodePitchGain(Bitstr *streamdata, WebRtc_Word16 *PitchGain_Q12);
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int WebRtcIsac_DecodePitchLag(Bitstr *streamdata, WebRtc_Word16 *PitchGain_Q12, double *PitchLag);
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int WebRtcIsac_DecodeFrameLen(Bitstr *streamdata, WebRtc_Word16 *framelength);
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int WebRtcIsac_EncodeFrameLen(WebRtc_Word16 framelength, Bitstr *streamdata);
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int WebRtcIsac_DecodeSendBW(Bitstr *streamdata, WebRtc_Word16 *BWno);
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void WebRtcIsac_EncodeReceiveBw(int *BWno, Bitstr *streamdata);
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/* step-down */
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void WebRtcIsac_Poly2Rc(double *a, int N, double *RC);
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/* step-up */
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void WebRtcIsac_Rc2Poly(double *RC, int N, double *a);
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void WebRtcIsac_TranscodeLPCCoef(double *LPCCoef_lo, double *LPCCoef_hi, int model,
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int *index_g);
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/******************************************************************************
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* WebRtcIsac_EncodeLpcGainUb()
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* Encode LPC gains of sub-Frames.
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*
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* Input/outputs:
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* - lpGains : a buffer which contains 'SUBFRAME' number of
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* LP gains to be encoded. The input values are
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* overwritten by the quantized values.
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* - streamdata : pointer to a stucture containg the encoded
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* data and theparameters needed for entropy
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* coding.
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*
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* Output:
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* - lpcGainIndex : quantization indices for lpc gains, these will
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* be stored to be used for FEC.
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*/
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void WebRtcIsac_EncodeLpcGainUb(
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double* lpGains,
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Bitstr* streamdata,
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int* lpcGainIndex);
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/******************************************************************************
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* WebRtcIsac_EncodeLpcGainUb()
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* Store LPC gains of sub-Frames in 'streamdata'.
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*
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* Input:
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* - lpGains : a buffer which contains 'SUBFRAME' number of
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* LP gains to be encoded.
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* Input/outputs:
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* - streamdata : pointer to a stucture containg the encoded
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* data and theparameters needed for entropy
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* coding.
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*
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*/
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void WebRtcIsac_StoreLpcGainUb(
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double* lpGains,
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Bitstr* streamdata);
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/******************************************************************************
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* WebRtcIsac_DecodeLpcGainUb()
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* Decode the LPC gain of sub-frames.
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*
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* Input/output:
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* - streamdata : pointer to a stucture containg the encoded
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* data and theparameters needed for entropy
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* coding.
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*
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* Output:
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* - lpGains : a buffer where decoded LPC gians will be stored.
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*
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* Return value : 0 if succeeded.
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* <0 if failed.
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*/
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WebRtc_Word16 WebRtcIsac_DecodeLpcGainUb(
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double* lpGains,
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Bitstr* streamdata);
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/******************************************************************************
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* WebRtcIsac_EncodeBandwidth()
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* Encode if the bandwidth of encoded audio is 0-12 kHz or 0-16 kHz.
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*
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* Input:
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* - bandwidth : an enumerator specifying if the codec in is
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* 0-12 kHz or 0-16 kHz mode.
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*
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* Input/output:
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* - streamdata : pointer to a stucture containg the encoded
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* data and theparameters needed for entropy
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* coding.
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*
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* Return value : 0 if succeeded.
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* <0 if failed.
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*/
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WebRtc_Word16 WebRtcIsac_EncodeBandwidth(
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enum ISACBandwidth bandwidth,
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Bitstr* streamData);
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/******************************************************************************
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* WebRtcIsac_DecodeBandwidth()
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* Decode the bandwidth of the encoded audio, i.e. if the bandwidth is 0-12 kHz
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* or 0-16 kHz.
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*
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* Input/output:
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* - streamdata : pointer to a stucture containg the encoded
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* data and theparameters needed for entropy
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* coding.
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*
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* Output:
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* - bandwidth : an enumerator specifying if the codec is in
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* 0-12 kHz or 0-16 kHz mode.
|
||
|
|
*
|
||
|
|
* Return value : 0 if succeeded.
|
||
|
|
* <0 if failed.
|
||
|
|
*/
|
||
|
|
WebRtc_Word16 WebRtcIsac_DecodeBandwidth(
|
||
|
|
Bitstr* streamData,
|
||
|
|
enum ISACBandwidth* bandwidth);
|
||
|
|
|
||
|
|
|
||
|
|
/******************************************************************************
|
||
|
|
* WebRtcIsac_EncodeJitterInfo()
|
||
|
|
* Decode the jitter information.
|
||
|
|
*
|
||
|
|
* Input/output:
|
||
|
|
* - streamdata : pointer to a stucture containg the encoded
|
||
|
|
* data and theparameters needed for entropy
|
||
|
|
* coding.
|
||
|
|
*
|
||
|
|
* Input:
|
||
|
|
* - jitterInfo : one bit of info specifying if the channel is
|
||
|
|
* in high/low jitter. Zero indicates low jitter
|
||
|
|
* and one indicates high jitter.
|
||
|
|
*
|
||
|
|
* Return value : 0 if succeeded.
|
||
|
|
* <0 if failed.
|
||
|
|
*/
|
||
|
|
WebRtc_Word16 WebRtcIsac_EncodeJitterInfo(
|
||
|
|
WebRtc_Word32 jitterIndex,
|
||
|
|
Bitstr* streamData);
|
||
|
|
|
||
|
|
|
||
|
|
/******************************************************************************
|
||
|
|
* WebRtcIsac_DecodeJitterInfo()
|
||
|
|
* Decode the jitter information.
|
||
|
|
*
|
||
|
|
* Input/output:
|
||
|
|
* - streamdata : pointer to a stucture containg the encoded
|
||
|
|
* data and theparameters needed for entropy
|
||
|
|
* coding.
|
||
|
|
*
|
||
|
|
* Output:
|
||
|
|
* - jitterInfo : one bit of info specifying if the channel is
|
||
|
|
* in high/low jitter. Zero indicates low jitter
|
||
|
|
* and one indicates high jitter.
|
||
|
|
*
|
||
|
|
* Return value : 0 if succeeded.
|
||
|
|
* <0 if failed.
|
||
|
|
*/
|
||
|
|
WebRtc_Word16 WebRtcIsac_DecodeJitterInfo(
|
||
|
|
Bitstr* streamData,
|
||
|
|
WebRtc_Word32* jitterInfo);
|
||
|
|
|
||
|
|
#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ENTROPY_CODING_H_ */
|