git-svn-id: http://webrtc.googlecode.com/svn/trunk@4 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
niklase@google.com
292
modules/audio_coding/codecs/iSAC/main/source/codec.h
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292
modules/audio_coding/codecs/iSAC/main/source/codec.h
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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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* 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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int WebRtcIsac_EstimateBandwidth(
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BwEstimatorstr* bwest_str,
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Bitstr* streamdata,
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WebRtc_Word32 packet_size,
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WebRtc_UWord16 rtp_seq_number,
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WebRtc_UWord32 send_ts,
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WebRtc_UWord32 arr_ts,
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enum IsacSamplingRate encoderSampRate,
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enum IsacSamplingRate decoderSampRate);
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int WebRtcIsac_DecodeLb(
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float* signal_out,
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ISACLBDecStruct* ISACdec_obj,
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WebRtc_Word16* current_framesamples,
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WebRtc_Word16 isRCUPayload);
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int WebRtcIsac_DecodeRcuLb(
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float* signal_out,
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ISACLBDecStruct* ISACdec_obj,
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WebRtc_Word16* current_framesamples);
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int WebRtcIsac_EncodeLb(
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float* in,
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ISACLBEncStruct* ISACencLB_obj,
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WebRtc_Word16 codingMode,
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WebRtc_Word16 bottleneckIndex);
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int WebRtcIsac_EncodeStoredDataLb(
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const ISAC_SaveEncData_t* ISACSavedEnc_obj,
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Bitstr* ISACBitStr_obj,
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int BWnumber,
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float scale);
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int WebRtcIsac_EncodeStoredDataUb12(
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const ISACUBSaveEncDataStruct* ISACSavedEnc_obj,
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Bitstr* bitStream,
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WebRtc_Word32 jitterInfo,
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float scale);
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int WebRtcIsac_EncodeStoredDataUb16(
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const ISACUBSaveEncDataStruct* ISACSavedEnc_obj,
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Bitstr* bitStream,
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WebRtc_Word32 jitterInfo,
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float scale);
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WebRtc_Word16 WebRtcIsac_GetRedPayloadUb(
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const ISACUBSaveEncDataStruct* ISACSavedEncObj,
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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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WebRtc_Word16
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WebRtcIsac_RateAllocation(
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WebRtc_Word32 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(
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float* signal_out,
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ISACUBDecStruct* ISACdec_obj,
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WebRtc_Word16 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(
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float* signal_out,
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ISACUBDecStruct* ISACdec_obj,
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WebRtc_Word16 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(
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float* in,
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ISACUBEncStruct* ISACenc_obj,
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WebRtc_Word32 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(
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float* in,
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ISACUBEncStruct* ISACenc_obj,
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WebRtc_Word32 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();
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void WebRtcIsac_Time2Spec(double *inre1,
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double *inre2,
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WebRtc_Word16 *outre,
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WebRtc_Word16 *outim,
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FFTstr *fftstr_obj);
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void WebRtcIsac_Spec2time(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,
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double *Coef,
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int lengthInOut,
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int orderCoef);
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void WebRtcIsac_AllZeroFilter(double *In,
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double *Coef,
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int lengthInOut,
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int orderCoef,
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double *Out);
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void WebRtcIsac_ZeroPoleFilter(double *In,
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double *ZeroCoef,
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double *PoleCoef,
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int lengthInOut,
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int orderCoef,
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double *Out);
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/***************************** filterbank functions **************************/
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void WebRtcIsac_SplitAndFilter(double *in,
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double *LP,
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double *HP,
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double *LP_la,
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double *HP_la,
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PreFiltBankstr *prefiltdata);
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void WebRtcIsac_FilterAndCombine(double *InLP,
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double *InHP,
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double *Out,
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PostFiltBankstr *postfiltdata);
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void WebRtcIsac_SplitAndFilterFloat(float *in,
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float *LP,
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float *HP,
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double *LP_la,
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double *HP_la,
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PreFiltBankstr *prefiltdata);
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void WebRtcIsac_FilterAndCombineFloat(float *InLP,
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float *InHP,
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float *Out,
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PostFiltBankstr *postfiltdata);
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/************************* normalized lattice filters ************************/
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void WebRtcIsac_NormLatticeFilterMa(int orderCoef,
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float *stateF,
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float *stateG,
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float *lat_in,
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double *filtcoeflo,
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double *lat_out);
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void WebRtcIsac_NormLatticeFilterAr(int orderCoef,
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float *stateF,
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float *stateG,
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double *lat_in,
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double *lo_filt_coef,
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float *lat_out);
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void WebRtcIsac_Dir2Lat(double *a,
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int orderCoef,
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float *sth,
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float *cth);
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void WebRtcIsac_AutoCorr(double *r,
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const double *x,
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int N,
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int order);
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#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CODEC_H_ */
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