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7ba22b8eea
The audio processing code is using parts of the iSAC codec to do voice activity detection (VAD), but it's undesirable for it to pull in the entire iSAC codec as a dependency. So this CL factors out the parts of iSAC that's needed for VAD to a separate build target. Bug: webrtc:8396 Change-Id: I884e25d8fd0bc815fca664352b0573b4b173880e Reviewed-on: https://webrtc-review.googlesource.com/69640 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23110}
209 lines
7.9 KiB
C
209 lines
7.9 KiB
C
/*
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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 MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CODEC_H_
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#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CODEC_H_
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#include <stddef.h>
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#include "modules/audio_coding/codecs/isac/main/source/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_InitPostFilterbank(PostFiltBankstr* postfiltdata);
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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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/***************************** filterbank functions **************************/
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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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#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CODEC_H_ */
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