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Diffstat (limited to 'libSBRdec/src/lpp_tran.cpp')
| -rw-r--r-- | libSBRdec/src/lpp_tran.cpp | 1433 |
1 files changed, 949 insertions, 484 deletions
diff --git a/libSBRdec/src/lpp_tran.cpp b/libSBRdec/src/lpp_tran.cpp index 117e739..aa1fd5d 100644 --- a/libSBRdec/src/lpp_tran.cpp +++ b/libSBRdec/src/lpp_tran.cpp @@ -1,74 +1,85 @@ - -/* ----------------------------------------------------------------------------------------------------------- +/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. 2. COPYRIGHT LICENSE -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." 3. NO PATENT LICENSE -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION @@ -79,19 +90,30 @@ Am Wolfsmantel 33 www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ +----------------------------------------------------------------------------- */ + +/**************************** SBR decoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ /*! \file - \brief Low Power Profile Transposer, - This module provides the transposer. The main entry point is lppTransposer(). The function generates - high frequency content by copying data from the low band (provided by core codec) into the high band. - This process is also referred to as "patching". The function also implements spectral whitening by means of - inverse filtering based on LPC coefficients. - - Together with the QMF filterbank the transposer can be tested using a supplied test program. See main_audio.cpp for details. - This module does use fractional arithmetic and the accuracy of the computations has an impact on the overall sound quality. - The module also needs to take into account the different scaling of spectral data. + \brief Low Power Profile Transposer + This module provides the transposer. The main entry point is lppTransposer(). + The function generates high frequency content by copying data from the low + band (provided by core codec) into the high band. This process is also + referred to as "patching". The function also implements spectral whitening by + means of inverse filtering based on LPC coefficients. + + Together with the QMF filterbank the transposer can be tested using a supplied + test program. See main_audio.cpp for details. This module does use fractional + arithmetic and the accuracy of the computations has an impact on the overall + sound quality. The module also needs to take into account the different + scaling of spectral data. \sa lppTransposer(), main_audio.cpp, sbr_scale.h, \ref documentationOverview */ @@ -104,16 +126,13 @@ amm-info@iis.fraunhofer.de #include "genericStds.h" #include "autocorr2nd.h" - +#include "HFgen_preFlat.h" #if defined(__arm__) #include "arm/lpp_tran_arm.cpp" #endif - - -#define LPC_SCALE_FACTOR 2 - +#define LPC_SCALE_FACTOR 2 /*! * @@ -125,33 +144,28 @@ amm-info@iis.fraunhofer.de * level is being inserted to achieve a smooth transition. */ -#ifndef FUNCTION_mapInvfMode -static FIXP_DBL -mapInvfMode (INVF_MODE mode, - INVF_MODE prevMode, - WHITENING_FACTORS whFactors) -{ +static FIXP_DBL mapInvfMode(INVF_MODE mode, INVF_MODE prevMode, + WHITENING_FACTORS whFactors) { switch (mode) { - case INVF_LOW_LEVEL: - if(prevMode == INVF_OFF) - return whFactors.transitionLevel; - else - return whFactors.lowLevel; - - case INVF_MID_LEVEL: - return whFactors.midLevel; - - case INVF_HIGH_LEVEL: - return whFactors.highLevel; - - default: - if(prevMode == INVF_LOW_LEVEL) - return whFactors.transitionLevel; - else - return whFactors.off; + case INVF_LOW_LEVEL: + if (prevMode == INVF_OFF) + return whFactors.transitionLevel; + else + return whFactors.lowLevel; + + case INVF_MID_LEVEL: + return whFactors.midLevel; + + case INVF_HIGH_LEVEL: + return whFactors.highLevel; + + default: + if (prevMode == INVF_LOW_LEVEL) + return whFactors.transitionLevel; + else + return whFactors.off; } } -#endif /* #ifndef FUNCTION_mapInvfMode */ /*! * @@ -161,124 +175,148 @@ mapInvfMode (INVF_MODE mode, * */ -#ifndef FUNCTION_inverseFilteringLevelEmphasis -static void -inverseFilteringLevelEmphasis(HANDLE_SBR_LPP_TRANS hLppTrans,/*!< Handle of lpp transposer */ - UCHAR nInvfBands, /*!< Number of bands for inverse filtering */ - INVF_MODE *sbr_invf_mode, /*!< Current inverse filtering modes */ - INVF_MODE *sbr_invf_mode_prev, /*!< Previous inverse filtering modes */ - FIXP_DBL * bwVector /*!< Resulting filtering levels */ - ) -{ - for(int i = 0; i < nInvfBands; i++) { +static void inverseFilteringLevelEmphasis( + HANDLE_SBR_LPP_TRANS hLppTrans, /*!< Handle of lpp transposer */ + UCHAR nInvfBands, /*!< Number of bands for inverse filtering */ + INVF_MODE *sbr_invf_mode, /*!< Current inverse filtering modes */ + INVF_MODE *sbr_invf_mode_prev, /*!< Previous inverse filtering modes */ + FIXP_DBL *bwVector /*!< Resulting filtering levels */ +) { + for (int i = 0; i < nInvfBands; i++) { FIXP_DBL accu; - FIXP_DBL bwTmp = mapInvfMode (sbr_invf_mode[i], - sbr_invf_mode_prev[i], - hLppTrans->pSettings->whFactors); - - if(bwTmp < hLppTrans->bwVectorOld[i]) { - accu = fMultDiv2(FL2FXCONST_DBL(0.75f),bwTmp) + - fMultDiv2(FL2FXCONST_DBL(0.25f),hLppTrans->bwVectorOld[i]); - } - else { - accu = fMultDiv2(FL2FXCONST_DBL(0.90625f),bwTmp) + - fMultDiv2(FL2FXCONST_DBL(0.09375f),hLppTrans->bwVectorOld[i]); + FIXP_DBL bwTmp = mapInvfMode(sbr_invf_mode[i], sbr_invf_mode_prev[i], + hLppTrans->pSettings->whFactors); + + if (bwTmp < hLppTrans->bwVectorOld[i]) { + accu = fMultDiv2(FL2FXCONST_DBL(0.75f), bwTmp) + + fMultDiv2(FL2FXCONST_DBL(0.25f), hLppTrans->bwVectorOld[i]); + } else { + accu = fMultDiv2(FL2FXCONST_DBL(0.90625f), bwTmp) + + fMultDiv2(FL2FXCONST_DBL(0.09375f), hLppTrans->bwVectorOld[i]); } - if (accu < FL2FXCONST_DBL(0.015625f)>>1) + if (accu<FL2FXCONST_DBL(0.015625f)>> 1) { bwVector[i] = FL2FXCONST_DBL(0.0f); - else - bwVector[i] = fixMin(accu<<1,FL2FXCONST_DBL(0.99609375f)); + } else { + bwVector[i] = fixMin(accu << 1, FL2FXCONST_DBL(0.99609375f)); + } } } -#endif /* #ifndef FUNCTION_inverseFilteringLevelEmphasis */ /* Resulting autocorrelation determinant exponent */ -#define ACDET_EXP (2*(DFRACT_BITS+sbrScaleFactor->lb_scale+10-ac.det_scale)) -#define AC_EXP (-sbrScaleFactor->lb_scale+LPC_SCALE_FACTOR) -#define ALPHA_EXP (-sbrScaleFactor->lb_scale+LPC_SCALE_FACTOR+1) -/* Resulting transposed QMF values exponent 16 bit normalized samplebits assumed. */ -#define QMFOUT_EXP ((SAMPLE_BITS-15)-sbrScaleFactor->lb_scale) +#define ACDET_EXP \ + (2 * (DFRACT_BITS + sbrScaleFactor->lb_scale + 10 - ac.det_scale)) +#define AC_EXP (-sbrScaleFactor->lb_scale + LPC_SCALE_FACTOR) +#define ALPHA_EXP (-sbrScaleFactor->lb_scale + LPC_SCALE_FACTOR + 1) +/* Resulting transposed QMF values exponent 16 bit normalized samplebits + * assumed. */ +#define QMFOUT_EXP ((SAMPLE_BITS - 15) - sbrScaleFactor->lb_scale) + +static inline void calc_qmfBufferReal(FIXP_DBL **qmfBufferReal, + const FIXP_DBL *const lowBandReal, + const int startSample, + const int stopSample, const UCHAR hiBand, + const int dynamicScale, const int descale, + const FIXP_SGL a0r, const FIXP_SGL a1r) { + FIXP_DBL accu1, accu2; + int i; + + for (i = 0; i < stopSample - startSample; i++) { + accu1 = fMultDiv2(a1r, lowBandReal[i]); + accu1 = (fMultDiv2(a0r, lowBandReal[i + 1]) + accu1); + accu1 = accu1 >> dynamicScale; + + accu1 <<= 1; + accu2 = (lowBandReal[i + 2] >> descale); + qmfBufferReal[i + startSample][hiBand] = accu1 + accu2; + } +} /*! * * \brief Perform transposition by patching of subband samples. - * This function serves as the main entry point into the module. The function determines the areas for the - * patching process (these are the source range as well as the target range) and implements spectral whitening - * by means of inverse filtering. The function autoCorrelation2nd() is an auxiliary function for calculating the - * LPC coefficients for the filtering. The actual calculation of the LPC coefficients and the implementation - * of the filtering are done as part of lppTransposer(). + * This function serves as the main entry point into the module. The function + * determines the areas for the patching process (these are the source range as + * well as the target range) and implements spectral whitening by means of + * inverse filtering. The function autoCorrelation2nd() is an auxiliary function + * for calculating the LPC coefficients for the filtering. The actual + * calculation of the LPC coefficients and the implementation of the filtering + * are done as part of lppTransposer(). * - * Note that the filtering is done on all available QMF subsamples, whereas the patching is only done on those QMF - * subsamples that will be used in the next QMF synthesis. The filtering is also implemented before the patching - * includes further dependencies on parameters from the SBR data. + * Note that the filtering is done on all available QMF subsamples, whereas the + * patching is only done on those QMF subsamples that will be used in the next + * QMF synthesis. The filtering is also implemented before the patching includes + * further dependencies on parameters from the SBR data. * */ -void lppTransposer (HANDLE_SBR_LPP_TRANS hLppTrans, /*!< Handle of lpp transposer */ - QMF_SCALE_FACTOR *sbrScaleFactor, /*!< Scaling factors */ - FIXP_DBL **qmfBufferReal, /*!< Pointer to pointer to real part of subband samples (source) */ - - FIXP_DBL *degreeAlias, /*!< Vector for results of aliasing estimation */ - FIXP_DBL **qmfBufferImag, /*!< Pointer to pointer to imaginary part of subband samples (source) */ - const int useLP, - const int timeStep, /*!< Time step of envelope */ - const int firstSlotOffs, /*!< Start position in time */ - const int lastSlotOffs, /*!< Number of overlap-slots into next frame */ - const int nInvfBands, /*!< Number of bands for inverse filtering */ - INVF_MODE *sbr_invf_mode, /*!< Current inverse filtering modes */ - INVF_MODE *sbr_invf_mode_prev /*!< Previous inverse filtering modes */ - ) -{ - INT bwIndex[MAX_NUM_PATCHES]; - FIXP_DBL bwVector[MAX_NUM_PATCHES]; /*!< pole moving factors */ - - int i; - int loBand, start, stop; +void lppTransposer( + HANDLE_SBR_LPP_TRANS hLppTrans, /*!< Handle of lpp transposer */ + QMF_SCALE_FACTOR *sbrScaleFactor, /*!< Scaling factors */ + FIXP_DBL **qmfBufferReal, /*!< Pointer to pointer to real part of subband + samples (source) */ + + FIXP_DBL *degreeAlias, /*!< Vector for results of aliasing estimation */ + FIXP_DBL **qmfBufferImag, /*!< Pointer to pointer to imaginary part of + subband samples (source) */ + const int useLP, const int fPreWhitening, const int v_k_master0, + const int timeStep, /*!< Time step of envelope */ + const int firstSlotOffs, /*!< Start position in time */ + const int lastSlotOffs, /*!< Number of overlap-slots into next frame */ + const int nInvfBands, /*!< Number of bands for inverse filtering */ + INVF_MODE *sbr_invf_mode, /*!< Current inverse filtering modes */ + INVF_MODE *sbr_invf_mode_prev /*!< Previous inverse filtering modes */ +) { + INT bwIndex[MAX_NUM_PATCHES]; + FIXP_DBL bwVector[MAX_NUM_PATCHES]; /*!< pole moving factors */ + FIXP_DBL preWhiteningGains[(64) / 2]; + int preWhiteningGains_exp[(64) / 2]; + + int i; + int loBand, start, stop; TRANSPOSER_SETTINGS *pSettings = hLppTrans->pSettings; PATCH_PARAM *patchParam = pSettings->patchParam; - int patch; + int patch; - FIXP_SGL alphar[LPC_ORDER], a0r, a1r; - FIXP_SGL alphai[LPC_ORDER], a0i=0, a1i=0; - FIXP_SGL bw = FL2FXCONST_SGL(0.0f); + FIXP_SGL alphar[LPC_ORDER], a0r, a1r; + FIXP_SGL alphai[LPC_ORDER], a0i = 0, a1i = 0; + FIXP_SGL bw = FL2FXCONST_SGL(0.0f); - int autoCorrLength; + int autoCorrLength; - FIXP_DBL k1, k1_below=0, k1_below2=0; + FIXP_DBL k1, k1_below = 0, k1_below2 = 0; ACORR_COEFS ac; - int startSample; - int stopSample; - int stopSampleClear; + int startSample; + int stopSample; + int stopSampleClear; int comLowBandScale; int ovLowBandShift; int lowBandShift; -/* int ovHighBandShift;*/ + /* int ovHighBandShift;*/ int targetStopBand; - alphai[0] = FL2FXCONST_SGL(0.0f); alphai[1] = FL2FXCONST_SGL(0.0f); - startSample = firstSlotOffs * timeStep; - stopSample = pSettings->nCols + lastSlotOffs * timeStep; + stopSample = pSettings->nCols + lastSlotOffs * timeStep; + FDK_ASSERT((lastSlotOffs * timeStep) <= pSettings->overlap); - - inverseFilteringLevelEmphasis(hLppTrans, nInvfBands, sbr_invf_mode, sbr_invf_mode_prev, bwVector); + inverseFilteringLevelEmphasis(hLppTrans, nInvfBands, sbr_invf_mode, + sbr_invf_mode_prev, bwVector); stopSampleClear = stopSample; autoCorrLength = pSettings->nCols + pSettings->overlap; /* Set upper subbands to zero: - This is required in case that the patches do not cover the complete highband - (because the last patch would be too short). - Possible optimization: Clearing bands up to usb would be sufficient here. */ - targetStopBand = patchParam[pSettings->noOfPatches-1].targetStartBand - + patchParam[pSettings->noOfPatches-1].numBandsInPatch; + This is required in case that the patches do not cover the complete + highband (because the last patch would be too short). Possible + optimization: Clearing bands up to usb would be sufficient here. */ + targetStopBand = patchParam[pSettings->noOfPatches - 1].targetStartBand + + patchParam[pSettings->noOfPatches - 1].numBandsInPatch; int memSize = ((64) - targetStopBand) * sizeof(FIXP_DBL); @@ -287,67 +325,84 @@ void lppTransposer (HANDLE_SBR_LPP_TRANS hLppTrans, /*!< Handle of lpp transp FDKmemclear(&qmfBufferReal[i][targetStopBand], memSize); FDKmemclear(&qmfBufferImag[i][targetStopBand], memSize); } - } else - for (i = startSample; i < stopSampleClear; i++) { - FDKmemclear(&qmfBufferReal[i][targetStopBand], memSize); + } else { + for (i = startSample; i < stopSampleClear; i++) { + FDKmemclear(&qmfBufferReal[i][targetStopBand], memSize); + } } /* init bwIndex for each patch */ - FDKmemclear(bwIndex, pSettings->noOfPatches*sizeof(INT)); + FDKmemclear(bwIndex, sizeof(bwIndex)); /* Calc common low band scale factor */ - comLowBandScale = fixMin(sbrScaleFactor->ov_lb_scale,sbrScaleFactor->lb_scale); + comLowBandScale = + fixMin(sbrScaleFactor->ov_lb_scale, sbrScaleFactor->lb_scale); - ovLowBandShift = sbrScaleFactor->ov_lb_scale - comLowBandScale; - lowBandShift = sbrScaleFactor->lb_scale - comLowBandScale; + ovLowBandShift = sbrScaleFactor->ov_lb_scale - comLowBandScale; + lowBandShift = sbrScaleFactor->lb_scale - comLowBandScale; /* ovHighBandShift = firstSlotOffs == 0 ? ovLowBandShift:0;*/ + if (fPreWhitening) { + sbrDecoder_calculateGainVec( + qmfBufferReal, qmfBufferImag, + DFRACT_BITS - 1 - 16 - + sbrScaleFactor->ov_lb_scale, /* convert scale to exponent */ + DFRACT_BITS - 1 - 16 - + sbrScaleFactor->lb_scale, /* convert scale to exponent */ + pSettings->overlap, preWhiteningGains, preWhiteningGains_exp, + v_k_master0, startSample, stopSample); + } + /* outer loop over bands to do analysis only once for each band */ if (!useLP) { start = pSettings->lbStartPatching; stop = pSettings->lbStopPatching; - } else - { + } else { start = fixMax(1, pSettings->lbStartPatching - 2); stop = patchParam[0].targetStartBand; } - - for ( loBand = start; loBand < stop; loBand++ ) { - - FIXP_DBL lowBandReal[(((1024)/(32))+(6))+LPC_ORDER]; + for (loBand = start; loBand < stop; loBand++) { + FIXP_DBL lowBandReal[(((1024) / (32) * (4) / 2) + (3 * (4))) + LPC_ORDER]; FIXP_DBL *plowBandReal = lowBandReal; - FIXP_DBL **pqmfBufferReal = qmfBufferReal; - FIXP_DBL lowBandImag[(((1024)/(32))+(6))+LPC_ORDER]; + FIXP_DBL **pqmfBufferReal = + qmfBufferReal + firstSlotOffs * timeStep /* + pSettings->overlap */; + FIXP_DBL lowBandImag[(((1024) / (32) * (4) / 2) + (3 * (4))) + LPC_ORDER]; FIXP_DBL *plowBandImag = lowBandImag; - FIXP_DBL **pqmfBufferImag = qmfBufferImag; - int resetLPCCoeffs=0; - int dynamicScale = DFRACT_BITS-1-LPC_SCALE_FACTOR; + FIXP_DBL **pqmfBufferImag = + qmfBufferImag + firstSlotOffs * timeStep /* + pSettings->overlap */; + int resetLPCCoeffs = 0; + int dynamicScale = DFRACT_BITS - 1 - LPC_SCALE_FACTOR; int acDetScale = 0; /* scaling of autocorrelation determinant */ - for(i=0;i<LPC_ORDER;i++){ - *plowBandReal++ = hLppTrans->lpcFilterStatesReal[i][loBand]; + for (i = 0; + i < LPC_ORDER + firstSlotOffs * timeStep /*+pSettings->overlap*/; + i++) { + *plowBandReal++ = hLppTrans->lpcFilterStatesRealLegSBR[i][loBand]; if (!useLP) - *plowBandImag++ = hLppTrans->lpcFilterStatesImag[i][loBand]; + *plowBandImag++ = hLppTrans->lpcFilterStatesImagLegSBR[i][loBand]; } /* Take old slope length qmf slot source values out of (overlap)qmf buffer */ if (!useLP) { - for(i=0;i<pSettings->nCols+pSettings->overlap;i++){ + for (i = 0; + i < pSettings->nCols + pSettings->overlap - firstSlotOffs * timeStep; + i++) { *plowBandReal++ = (*pqmfBufferReal++)[loBand]; *plowBandImag++ = (*pqmfBufferImag++)[loBand]; } - } else - { + } else { /* pSettings->overlap is always even */ FDK_ASSERT((pSettings->overlap & 1) == 0); - - for(i=0;i<((pSettings->overlap+pSettings->nCols)>>1);i++) { + for (i = 0; i < ((pSettings->nCols + pSettings->overlap - + firstSlotOffs * timeStep) >> + 1); + i++) { *plowBandReal++ = (*pqmfBufferReal++)[loBand]; *plowBandReal++ = (*pqmfBufferReal++)[loBand]; } @@ -359,61 +414,77 @@ void lppTransposer (HANDLE_SBR_LPP_TRANS hLppTrans, /*!< Handle of lpp transp /* Determine dynamic scaling value. */ - dynamicScale = fixMin(dynamicScale, getScalefactor(lowBandReal, LPC_ORDER+pSettings->overlap) + ovLowBandShift); - dynamicScale = fixMin(dynamicScale, getScalefactor(&lowBandReal[LPC_ORDER+pSettings->overlap], pSettings->nCols) + lowBandShift); + dynamicScale = + fixMin(dynamicScale, + getScalefactor(lowBandReal, LPC_ORDER + pSettings->overlap) + + ovLowBandShift); + dynamicScale = + fixMin(dynamicScale, + getScalefactor(&lowBandReal[LPC_ORDER + pSettings->overlap], + pSettings->nCols) + + lowBandShift); if (!useLP) { - dynamicScale = fixMin(dynamicScale, getScalefactor(lowBandImag, LPC_ORDER+pSettings->overlap) + ovLowBandShift); - dynamicScale = fixMin(dynamicScale, getScalefactor(&lowBandImag[LPC_ORDER+pSettings->overlap], pSettings->nCols) + lowBandShift); + dynamicScale = + fixMin(dynamicScale, + getScalefactor(lowBandImag, LPC_ORDER + pSettings->overlap) + + ovLowBandShift); + dynamicScale = + fixMin(dynamicScale, + getScalefactor(&lowBandImag[LPC_ORDER + pSettings->overlap], + pSettings->nCols) + + lowBandShift); } - dynamicScale = fixMax(0, dynamicScale-1); /* one additional bit headroom to prevent -1.0 */ + dynamicScale = fixMax( + 0, dynamicScale - 1); /* one additional bit headroom to prevent -1.0 */ /* Scale temporal QMF buffer. */ - scaleValues(&lowBandReal[0], LPC_ORDER+pSettings->overlap, dynamicScale-ovLowBandShift); - scaleValues(&lowBandReal[LPC_ORDER+pSettings->overlap], pSettings->nCols, dynamicScale-lowBandShift); + scaleValues(&lowBandReal[0], LPC_ORDER + pSettings->overlap, + dynamicScale - ovLowBandShift); + scaleValues(&lowBandReal[LPC_ORDER + pSettings->overlap], pSettings->nCols, + dynamicScale - lowBandShift); if (!useLP) { - scaleValues(&lowBandImag[0], LPC_ORDER+pSettings->overlap, dynamicScale-ovLowBandShift); - scaleValues(&lowBandImag[LPC_ORDER+pSettings->overlap], pSettings->nCols, dynamicScale-lowBandShift); + scaleValues(&lowBandImag[0], LPC_ORDER + pSettings->overlap, + dynamicScale - ovLowBandShift); + scaleValues(&lowBandImag[LPC_ORDER + pSettings->overlap], + pSettings->nCols, dynamicScale - lowBandShift); } + if (!useLP) { + acDetScale += autoCorr2nd_cplx(&ac, lowBandReal + LPC_ORDER, + lowBandImag + LPC_ORDER, autoCorrLength); + } else { + acDetScale += + autoCorr2nd_real(&ac, lowBandReal + LPC_ORDER, autoCorrLength); + } - if (!useLP) { - acDetScale += autoCorr2nd_cplx(&ac, lowBandReal+LPC_ORDER, lowBandImag+LPC_ORDER, autoCorrLength); - } - else - { - acDetScale += autoCorr2nd_real(&ac, lowBandReal+LPC_ORDER, autoCorrLength); - } - - /* Examine dynamic of determinant in autocorrelation. */ - acDetScale += 2*(comLowBandScale + dynamicScale); - acDetScale *= 2; /* two times reflection coefficent scaling */ - acDetScale += ac.det_scale; /* ac scaling of determinant */ - - /* In case of determinant < 10^-38, resetLPCCoeffs=1 has to be enforced. */ - if (acDetScale>126 ) { - resetLPCCoeffs = 1; - } + /* Examine dynamic of determinant in autocorrelation. */ + acDetScale += 2 * (comLowBandScale + dynamicScale); + acDetScale *= 2; /* two times reflection coefficent scaling */ + acDetScale += ac.det_scale; /* ac scaling of determinant */ + /* In case of determinant < 10^-38, resetLPCCoeffs=1 has to be enforced. */ + if (acDetScale > 126) { + resetLPCCoeffs = 1; + } alphar[1] = FL2FXCONST_SGL(0.0f); - if (!useLP) - alphai[1] = FL2FXCONST_SGL(0.0f); + if (!useLP) alphai[1] = FL2FXCONST_SGL(0.0f); if (ac.det != FL2FXCONST_DBL(0.0f)) { - FIXP_DBL tmp,absTmp,absDet; + FIXP_DBL tmp, absTmp, absDet; absDet = fixp_abs(ac.det); if (!useLP) { - tmp = ( fMultDiv2(ac.r01r,ac.r12r) >> (LPC_SCALE_FACTOR-1) ) - - ( (fMultDiv2(ac.r01i,ac.r12i) + fMultDiv2(ac.r02r,ac.r11r)) >> (LPC_SCALE_FACTOR-1) ); - } else - { - tmp = ( fMultDiv2(ac.r01r,ac.r12r) >> (LPC_SCALE_FACTOR-1) ) - - ( fMultDiv2(ac.r02r,ac.r11r) >> (LPC_SCALE_FACTOR-1) ); + tmp = (fMultDiv2(ac.r01r, ac.r12r) >> (LPC_SCALE_FACTOR - 1)) - + ((fMultDiv2(ac.r01i, ac.r12i) + fMultDiv2(ac.r02r, ac.r11r)) >> + (LPC_SCALE_FACTOR - 1)); + } else { + tmp = (fMultDiv2(ac.r01r, ac.r12r) >> (LPC_SCALE_FACTOR - 1)) - + (fMultDiv2(ac.r02r, ac.r11r) >> (LPC_SCALE_FACTOR - 1)); } absTmp = fixp_abs(tmp); @@ -423,23 +494,23 @@ void lppTransposer (HANDLE_SBR_LPP_TRANS hLppTrans, /*!< Handle of lpp transp { INT scale; FIXP_DBL result = fDivNorm(absTmp, absDet, &scale); - scale = scale+ac.det_scale; + scale = scale + ac.det_scale; - if ( (scale > 0) && (result >= (FIXP_DBL)MAXVAL_DBL>>scale) ) { + if ((scale > 0) && (result >= (FIXP_DBL)MAXVAL_DBL >> scale)) { resetLPCCoeffs = 1; - } - else { - alphar[1] = FX_DBL2FX_SGL(scaleValue(result,scale)); - if((tmp<FL2FX_DBL(0.0f)) ^ (ac.det<FL2FX_DBL(0.0f))) { + } else { + alphar[1] = FX_DBL2FX_SGL(scaleValue(result, scale)); + if ((tmp < FL2FX_DBL(0.0f)) ^ (ac.det < FL2FX_DBL(0.0f))) { alphar[1] = -alphar[1]; } } } - if (!useLP) - { - tmp = ( fMultDiv2(ac.r01i,ac.r12r) >> (LPC_SCALE_FACTOR-1) ) + - ( (fMultDiv2(ac.r01r,ac.r12i) - (FIXP_DBL)fMultDiv2(ac.r02i,ac.r11r)) >> (LPC_SCALE_FACTOR-1) ) ; + if (!useLP) { + tmp = (fMultDiv2(ac.r01i, ac.r12r) >> (LPC_SCALE_FACTOR - 1)) + + ((fMultDiv2(ac.r01r, ac.r12i) - + (FIXP_DBL)fMultDiv2(ac.r02i, ac.r11r)) >> + (LPC_SCALE_FACTOR - 1)); absTmp = fixp_abs(tmp); @@ -449,14 +520,15 @@ void lppTransposer (HANDLE_SBR_LPP_TRANS hLppTrans, /*!< Handle of lpp transp { INT scale; FIXP_DBL result = fDivNorm(absTmp, absDet, &scale); - scale = scale+ac.det_scale; + scale = scale + ac.det_scale; - if ( (scale > 0) && (result >= /*FL2FXCONST_DBL(1.f)*/ (FIXP_DBL)MAXVAL_DBL>>scale) ) { + if ((scale > 0) && + (result >= /*FL2FXCONST_DBL(1.f)*/ (FIXP_DBL)MAXVAL_DBL >> + scale)) { resetLPCCoeffs = 1; - } - else { - alphai[1] = FX_DBL2FX_SGL(scaleValue(result,scale)); - if((tmp<FL2FX_DBL(0.0f)) ^ (ac.det<FL2FX_DBL(0.0f))) { + } else { + alphai[1] = FX_DBL2FX_SGL(scaleValue(result, scale)); + if ((tmp < FL2FX_DBL(0.0f)) ^ (ac.det < FL2FX_DBL(0.0f))) { alphai[1] = -alphai[1]; } } @@ -464,24 +536,23 @@ void lppTransposer (HANDLE_SBR_LPP_TRANS hLppTrans, /*!< Handle of lpp transp } } - alphar[0] = FL2FXCONST_SGL(0.0f); - if (!useLP) - alphai[0] = FL2FXCONST_SGL(0.0f); - - if ( ac.r11r != FL2FXCONST_DBL(0.0f) ) { + alphar[0] = FL2FXCONST_SGL(0.0f); + if (!useLP) alphai[0] = FL2FXCONST_SGL(0.0f); + if (ac.r11r != FL2FXCONST_DBL(0.0f)) { /* ac.r11r is always >=0 */ - FIXP_DBL tmp,absTmp; + FIXP_DBL tmp, absTmp; if (!useLP) { - tmp = (ac.r01r>>(LPC_SCALE_FACTOR+1)) + - (fMultDiv2(alphar[1],ac.r12r) + fMultDiv2(alphai[1],ac.r12i)); - } else - { - if(ac.r01r>=FL2FXCONST_DBL(0.0f)) - tmp = (ac.r01r>>(LPC_SCALE_FACTOR+1)) + fMultDiv2(alphar[1],ac.r12r); + tmp = (ac.r01r >> (LPC_SCALE_FACTOR + 1)) + + (fMultDiv2(alphar[1], ac.r12r) + fMultDiv2(alphai[1], ac.r12i)); + } else { + if (ac.r01r >= FL2FXCONST_DBL(0.0f)) + tmp = (ac.r01r >> (LPC_SCALE_FACTOR + 1)) + + fMultDiv2(alphar[1], ac.r12r); else - tmp = -((-ac.r01r)>>(LPC_SCALE_FACTOR+1)) + fMultDiv2(alphar[1],ac.r12r); + tmp = -((-ac.r01r) >> (LPC_SCALE_FACTOR + 1)) + + fMultDiv2(alphar[1], ac.r12r); } absTmp = fixp_abs(tmp); @@ -490,111 +561,108 @@ void lppTransposer (HANDLE_SBR_LPP_TRANS hLppTrans, /*!< Handle of lpp transp Quick check: is first filter coeff >= 1(4) */ - if (absTmp >= (ac.r11r>>1)) { - resetLPCCoeffs=1; - } - else { + if (absTmp >= (ac.r11r >> 1)) { + resetLPCCoeffs = 1; + } else { INT scale; FIXP_DBL result = fDivNorm(absTmp, fixp_abs(ac.r11r), &scale); - alphar[0] = FX_DBL2FX_SGL(scaleValue(result,scale+1)); + alphar[0] = FX_DBL2FX_SGL(scaleValue(result, scale + 1)); - if((tmp>FL2FX_DBL(0.0f)) ^ (ac.r11r<FL2FX_DBL(0.0f))) + if ((tmp > FL2FX_DBL(0.0f)) ^ (ac.r11r < FL2FX_DBL(0.0f))) alphar[0] = -alphar[0]; } - if (!useLP) - { - tmp = (ac.r01i>>(LPC_SCALE_FACTOR+1)) + - (fMultDiv2(alphai[1],ac.r12r) - fMultDiv2(alphar[1],ac.r12i)); + if (!useLP) { + tmp = (ac.r01i >> (LPC_SCALE_FACTOR + 1)) + + (fMultDiv2(alphai[1], ac.r12r) - fMultDiv2(alphar[1], ac.r12i)); absTmp = fixp_abs(tmp); /* Quick check: is second filter coeff >= 1(4) */ - if (absTmp >= (ac.r11r>>1)) { - resetLPCCoeffs=1; - } - else { + if (absTmp >= (ac.r11r >> 1)) { + resetLPCCoeffs = 1; + } else { INT scale; FIXP_DBL result = fDivNorm(absTmp, fixp_abs(ac.r11r), &scale); - alphai[0] = FX_DBL2FX_SGL(scaleValue(result,scale+1)); - if((tmp>FL2FX_DBL(0.0f)) ^ (ac.r11r<FL2FX_DBL(0.0f))) + alphai[0] = FX_DBL2FX_SGL(scaleValue(result, scale + 1)); + if ((tmp > FL2FX_DBL(0.0f)) ^ (ac.r11r < FL2FX_DBL(0.0f))) alphai[0] = -alphai[0]; } } } - - if (!useLP) - { + if (!useLP) { /* Now check the quadratic criteria */ - if( (fMultDiv2(alphar[0],alphar[0]) + fMultDiv2(alphai[0],alphai[0])) >= FL2FXCONST_DBL(0.5f) ) - resetLPCCoeffs=1; - if( (fMultDiv2(alphar[1],alphar[1]) + fMultDiv2(alphai[1],alphai[1])) >= FL2FXCONST_DBL(0.5f) ) - resetLPCCoeffs=1; + if ((fMultDiv2(alphar[0], alphar[0]) + fMultDiv2(alphai[0], alphai[0])) >= + FL2FXCONST_DBL(0.5f)) + resetLPCCoeffs = 1; + if ((fMultDiv2(alphar[1], alphar[1]) + fMultDiv2(alphai[1], alphai[1])) >= + FL2FXCONST_DBL(0.5f)) + resetLPCCoeffs = 1; } - if(resetLPCCoeffs){ + if (resetLPCCoeffs) { alphar[0] = FL2FXCONST_SGL(0.0f); alphar[1] = FL2FXCONST_SGL(0.0f); - if (!useLP) - { + if (!useLP) { alphai[0] = FL2FXCONST_SGL(0.0f); alphai[1] = FL2FXCONST_SGL(0.0f); } } - if (useLP) - { - + if (useLP) { /* Aliasing detection */ - if(ac.r11r==FL2FXCONST_DBL(0.0f)) { + if (ac.r11r == FL2FXCONST_DBL(0.0f)) { k1 = FL2FXCONST_DBL(0.0f); - } - else { - if ( fixp_abs(ac.r01r) >= fixp_abs(ac.r11r) ) { - if ( fMultDiv2(ac.r01r,ac.r11r) < FL2FX_DBL(0.0f)) { + } else { + if (fixp_abs(ac.r01r) >= fixp_abs(ac.r11r)) { + if (fMultDiv2(ac.r01r, ac.r11r) < FL2FX_DBL(0.0f)) { k1 = (FIXP_DBL)MAXVAL_DBL /*FL2FXCONST_SGL(1.0f)*/; - }else { - /* Since this value is squared later, it must not ever become -1.0f. */ - k1 = (FIXP_DBL)(MINVAL_DBL+1) /*FL2FXCONST_SGL(-1.0f)*/; + } else { + /* Since this value is squared later, it must not ever become -1.0f. + */ + k1 = (FIXP_DBL)(MINVAL_DBL + 1) /*FL2FXCONST_SGL(-1.0f)*/; } - } - else { + } else { INT scale; - FIXP_DBL result = fDivNorm(fixp_abs(ac.r01r), fixp_abs(ac.r11r), &scale); - k1 = scaleValue(result,scale); + FIXP_DBL result = + fDivNorm(fixp_abs(ac.r01r), fixp_abs(ac.r11r), &scale); + k1 = scaleValue(result, scale); - if(!((ac.r01r<FL2FX_DBL(0.0f)) ^ (ac.r11r<FL2FX_DBL(0.0f)))) { + if (!((ac.r01r < FL2FX_DBL(0.0f)) ^ (ac.r11r < FL2FX_DBL(0.0f)))) { k1 = -k1; } } } - if(loBand > 1){ + if ((loBand > 1) && (loBand < v_k_master0)) { /* Check if the gain should be locked */ - FIXP_DBL deg = /*FL2FXCONST_DBL(1.0f)*/ (FIXP_DBL)MAXVAL_DBL - fPow2(k1_below); + FIXP_DBL deg = + /*FL2FXCONST_DBL(1.0f)*/ (FIXP_DBL)MAXVAL_DBL - fPow2(k1_below); degreeAlias[loBand] = FL2FXCONST_DBL(0.0f); - if (((loBand & 1) == 0) && (k1 < FL2FXCONST_DBL(0.0f))){ - if (k1_below < FL2FXCONST_DBL(0.0f)) { /* 2-Ch Aliasing Detection */ + if (((loBand & 1) == 0) && (k1 < FL2FXCONST_DBL(0.0f))) { + if (k1_below < FL2FXCONST_DBL(0.0f)) { /* 2-Ch Aliasing Detection */ degreeAlias[loBand] = (FIXP_DBL)MAXVAL_DBL /*FL2FXCONST_DBL(1.0f)*/; - if ( k1_below2 > FL2FXCONST_DBL(0.0f) ) { /* 3-Ch Aliasing Detection */ - degreeAlias[loBand-1] = deg; + if (k1_below2 > + FL2FXCONST_DBL(0.0f)) { /* 3-Ch Aliasing Detection */ + degreeAlias[loBand - 1] = deg; } - } - else if ( k1_below2 > FL2FXCONST_DBL(0.0f) ) { /* 3-Ch Aliasing Detection */ - degreeAlias[loBand] = deg; + } else if (k1_below2 > + FL2FXCONST_DBL(0.0f)) { /* 3-Ch Aliasing Detection */ + degreeAlias[loBand] = deg; } } - if (((loBand & 1) == 1) && (k1 > FL2FXCONST_DBL(0.0f))){ - if (k1_below > FL2FXCONST_DBL(0.0f)) { /* 2-CH Aliasing Detection */ + if (((loBand & 1) == 1) && (k1 > FL2FXCONST_DBL(0.0f))) { + if (k1_below > FL2FXCONST_DBL(0.0f)) { /* 2-CH Aliasing Detection */ degreeAlias[loBand] = (FIXP_DBL)MAXVAL_DBL /*FL2FXCONST_DBL(1.0f)*/; - if ( k1_below2 < FL2FXCONST_DBL(0.0f) ) { /* 3-CH Aliasing Detection */ - degreeAlias[loBand-1] = deg; + if (k1_below2 < + FL2FXCONST_DBL(0.0f)) { /* 3-CH Aliasing Detection */ + degreeAlias[loBand - 1] = deg; } - } - else if ( k1_below2 < FL2FXCONST_DBL(0.0f) ) { /* 3-CH Aliasing Detection */ - degreeAlias[loBand] = deg; + } else if (k1_below2 < + FL2FXCONST_DBL(0.0f)) { /* 3-CH Aliasing Detection */ + degreeAlias[loBand] = deg; } } } @@ -605,25 +673,27 @@ void lppTransposer (HANDLE_SBR_LPP_TRANS hLppTrans, /*!< Handle of lpp transp patch = 0; - while ( patch < pSettings->noOfPatches ) { /* inner loop over every patch */ + while (patch < pSettings->noOfPatches) { /* inner loop over every patch */ int hiBand = loBand + patchParam[patch].targetBandOffs; - if ( loBand < patchParam[patch].sourceStartBand - || loBand >= patchParam[patch].sourceStopBand - //|| hiBand >= hLppTrans->pSettings->noChannels - ) { + if (loBand < patchParam[patch].sourceStartBand || + loBand >= patchParam[patch].sourceStopBand + //|| hiBand >= hLppTrans->pSettings->noChannels + ) { /* Lowband not in current patch - proceed */ patch++; continue; } - FDK_ASSERT( hiBand < (64) ); + FDK_ASSERT(hiBand < (64)); - /* bwIndex[patch] is already initialized with value from previous band inside this patch */ - while (hiBand >= pSettings->bwBorders[bwIndex[patch]]) + /* bwIndex[patch] is already initialized with value from previous band + * inside this patch */ + while (hiBand >= pSettings->bwBorders[bwIndex[patch]] && + bwIndex[patch] < MAX_NUM_PATCHES - 1) { bwIndex[patch]++; - + } /* Filter Step 2: add the left slope with the current filter to the buffer @@ -631,121 +701,498 @@ void lppTransposer (HANDLE_SBR_LPP_TRANS hLppTrans, /*!< Handle of lpp transp */ bw = FX_DBL2FX_SGL(bwVector[bwIndex[patch]]); - a0r = FX_DBL2FX_SGL(fMult(bw,alphar[0])); /* Apply current bandwidth expansion factor */ - - - if (!useLP) - a0i = FX_DBL2FX_SGL(fMult(bw,alphai[0])); - bw = FX_DBL2FX_SGL(fPow2(bw)); - a1r = FX_DBL2FX_SGL(fMult(bw,alphar[1])); - if (!useLP) - a1i = FX_DBL2FX_SGL(fMult(bw,alphai[1])); - + a0r = FX_DBL2FX_SGL( + fMult(bw, alphar[0])); /* Apply current bandwidth expansion factor */ + if (!useLP) a0i = FX_DBL2FX_SGL(fMult(bw, alphai[0])); + bw = FX_DBL2FX_SGL(fPow2(bw)); + a1r = FX_DBL2FX_SGL(fMult(bw, alphar[1])); + if (!useLP) a1i = FX_DBL2FX_SGL(fMult(bw, alphai[1])); /* Filter Step 3: insert the middle part which won't be windowed */ - - if ( bw <= FL2FXCONST_SGL(0.0f) ) { + if (bw <= FL2FXCONST_SGL(0.0f)) { if (!useLP) { - int descale = fixMin(DFRACT_BITS-1, (LPC_SCALE_FACTOR+dynamicScale)); - for(i = startSample; i < stopSample; i++ ) { - qmfBufferReal[i][hiBand] = lowBandReal[LPC_ORDER+i]>>descale; - qmfBufferImag[i][hiBand] = lowBandImag[LPC_ORDER+i]>>descale; + int descale = + fixMin(DFRACT_BITS - 1, (LPC_SCALE_FACTOR + dynamicScale)); + for (i = startSample; i < stopSample; i++) { + FIXP_DBL accu1, accu2; + accu1 = lowBandReal[LPC_ORDER + i] >> descale; + accu2 = lowBandImag[LPC_ORDER + i] >> descale; + if (fPreWhitening) { + accu1 = scaleValueSaturate( + fMultDiv2(accu1, preWhiteningGains[loBand]), + preWhiteningGains_exp[loBand] + 1); + accu2 = scaleValueSaturate( + fMultDiv2(accu2, preWhiteningGains[loBand]), + preWhiteningGains_exp[loBand] + 1); + } + qmfBufferReal[i][hiBand] = accu1; + qmfBufferImag[i][hiBand] = accu2; } - } else - { - int descale = fixMin(DFRACT_BITS-1, (LPC_SCALE_FACTOR+dynamicScale)); - for(i = startSample; i < stopSample; i++ ) { - qmfBufferReal[i][hiBand] = lowBandReal[LPC_ORDER+i]>>descale; + } else { + int descale = + fixMin(DFRACT_BITS - 1, (LPC_SCALE_FACTOR + dynamicScale)); + for (i = startSample; i < stopSample; i++) { + qmfBufferReal[i][hiBand] = lowBandReal[LPC_ORDER + i] >> descale; } } - } - else { /* bw <= 0 */ + } else { /* bw <= 0 */ if (!useLP) { - int descale = fixMin(DFRACT_BITS-1, (LPC_SCALE_FACTOR+dynamicScale)); + int descale = + fixMin(DFRACT_BITS - 1, (LPC_SCALE_FACTOR + dynamicScale)); #ifdef FUNCTION_LPPTRANSPOSER_func1 - lppTransposer_func1(lowBandReal+LPC_ORDER+startSample,lowBandImag+LPC_ORDER+startSample, - qmfBufferReal+startSample,qmfBufferImag+startSample, - stopSample-startSample, (int) hiBand, - dynamicScale,descale, - a0r, a0i, a1r, a1i); + lppTransposer_func1( + lowBandReal + LPC_ORDER + startSample, + lowBandImag + LPC_ORDER + startSample, + qmfBufferReal + startSample, qmfBufferImag + startSample, + stopSample - startSample, (int)hiBand, dynamicScale, descale, a0r, + a0i, a1r, a1i, fPreWhitening, preWhiteningGains[loBand], + preWhiteningGains_exp[loBand] + 1); #else - for(i = startSample; i < stopSample; i++ ) { + for (i = startSample; i < stopSample; i++) { FIXP_DBL accu1, accu2; - accu1 = (fMultDiv2(a0r,lowBandReal[LPC_ORDER+i-1]) - fMultDiv2(a0i,lowBandImag[LPC_ORDER+i-1]) + - fMultDiv2(a1r,lowBandReal[LPC_ORDER+i-2]) - fMultDiv2(a1i,lowBandImag[LPC_ORDER+i-2]))>>dynamicScale; - accu2 = (fMultDiv2(a0i,lowBandReal[LPC_ORDER+i-1]) + fMultDiv2(a0r,lowBandImag[LPC_ORDER+i-1]) + - fMultDiv2(a1i,lowBandReal[LPC_ORDER+i-2]) + fMultDiv2(a1r,lowBandImag[LPC_ORDER+i-2]))>>dynamicScale; - - qmfBufferReal[i][hiBand] = (lowBandReal[LPC_ORDER+i]>>descale) + (accu1<<1); - qmfBufferImag[i][hiBand] = (lowBandImag[LPC_ORDER+i]>>descale) + (accu2<<1); + accu1 = (fMultDiv2(a0r, lowBandReal[LPC_ORDER + i - 1]) - + fMultDiv2(a0i, lowBandImag[LPC_ORDER + i - 1]) + + fMultDiv2(a1r, lowBandReal[LPC_ORDER + i - 2]) - + fMultDiv2(a1i, lowBandImag[LPC_ORDER + i - 2])) >> + dynamicScale; + accu2 = (fMultDiv2(a0i, lowBandReal[LPC_ORDER + i - 1]) + + fMultDiv2(a0r, lowBandImag[LPC_ORDER + i - 1]) + + fMultDiv2(a1i, lowBandReal[LPC_ORDER + i - 2]) + + fMultDiv2(a1r, lowBandImag[LPC_ORDER + i - 2])) >> + dynamicScale; + + accu1 = (lowBandReal[LPC_ORDER + i] >> descale) + (accu1 << 1); + accu2 = (lowBandImag[LPC_ORDER + i] >> descale) + (accu2 << 1); + if (fPreWhitening) { + accu1 = scaleValueSaturate( + fMultDiv2(accu1, preWhiteningGains[loBand]), + preWhiteningGains_exp[loBand] + 1); + accu2 = scaleValueSaturate( + fMultDiv2(accu2, preWhiteningGains[loBand]), + preWhiteningGains_exp[loBand] + 1); + } + qmfBufferReal[i][hiBand] = accu1; + qmfBufferImag[i][hiBand] = accu2; } #endif - } else - { - int descale = fixMin(DFRACT_BITS-1, (LPC_SCALE_FACTOR+dynamicScale)); - + } else { FDK_ASSERT(dynamicScale >= 0); - for(i = startSample; i < stopSample; i++ ) { - FIXP_DBL accu1; - - accu1 = (fMultDiv2(a0r,lowBandReal[LPC_ORDER+i-1]) + fMultDiv2(a1r,lowBandReal[LPC_ORDER+i-2]))>>dynamicScale; - - qmfBufferReal[i][hiBand] = (lowBandReal[LPC_ORDER+i]>>descale) + (accu1<<1); - } + calc_qmfBufferReal( + qmfBufferReal, &(lowBandReal[LPC_ORDER + startSample - 2]), + startSample, stopSample, hiBand, dynamicScale, + fMin(DFRACT_BITS - 1, (LPC_SCALE_FACTOR + dynamicScale)), a0r, + a1r); } } /* bw <= 0 */ patch++; - } /* inner loop over patches */ + } /* inner loop over patches */ - /* + /* * store the unmodified filter coefficients if there is * an overlapping envelope *****************************************************************/ + } /* outer loop over bands (loBand) */ - } /* outer loop over bands (loBand) */ - - if (useLP) - { - for ( loBand = pSettings->lbStartPatching; loBand < pSettings->lbStopPatching; loBand++ ) { + if (useLP) { + for (loBand = pSettings->lbStartPatching; + loBand < pSettings->lbStopPatching; loBand++) { patch = 0; - while ( patch < pSettings->noOfPatches ) { - + while (patch < pSettings->noOfPatches) { UCHAR hiBand = loBand + patchParam[patch].targetBandOffs; - if ( loBand < patchParam[patch].sourceStartBand - || loBand >= patchParam[patch].sourceStopBand - || hiBand >= (64) /* Highband out of range (biterror) */ - ) { - /* Lowband not in current patch or highband out of range (might be caused by biterrors)- proceed */ + if (loBand < patchParam[patch].sourceStartBand || + loBand >= patchParam[patch].sourceStopBand || + hiBand >= (64) /* Highband out of range (biterror) */ + ) { + /* Lowband not in current patch or highband out of range (might be + * caused by biterrors)- proceed */ patch++; continue; } - if(hiBand != patchParam[patch].targetStartBand) + if (hiBand != patchParam[patch].targetStartBand) degreeAlias[hiBand] = degreeAlias[loBand]; patch++; } - }/* end for loop */ + } /* end for loop */ } - for (i = 0; i < nInvfBands; i++ ) { - hLppTrans->bwVectorOld[i] = bwVector[i]; - } + for (i = 0; i < nInvfBands; i++) { + hLppTrans->bwVectorOld[i] = bwVector[i]; + } /* set high band scale factor */ - sbrScaleFactor->hb_scale = comLowBandScale-(LPC_SCALE_FACTOR); + sbrScaleFactor->hb_scale = comLowBandScale - (LPC_SCALE_FACTOR); +} + +void lppTransposerHBE( + HANDLE_SBR_LPP_TRANS hLppTrans, /*!< Handle of lpp transposer */ + HANDLE_HBE_TRANSPOSER hQmfTransposer, + QMF_SCALE_FACTOR *sbrScaleFactor, /*!< Scaling factors */ + FIXP_DBL **qmfBufferReal, /*!< Pointer to pointer to real part of subband + samples (source) */ + FIXP_DBL **qmfBufferImag, /*!< Pointer to pointer to imaginary part of + subband samples (source) */ + const int timeStep, /*!< Time step of envelope */ + const int firstSlotOffs, /*!< Start position in time */ + const int lastSlotOffs, /*!< Number of overlap-slots into next frame */ + const int nInvfBands, /*!< Number of bands for inverse filtering */ + INVF_MODE *sbr_invf_mode, /*!< Current inverse filtering modes */ + INVF_MODE *sbr_invf_mode_prev /*!< Previous inverse filtering modes */ +) { + INT bwIndex; + FIXP_DBL bwVector[MAX_NUM_PATCHES_HBE]; /*!< pole moving factors */ + + int i; + int loBand, start, stop; + TRANSPOSER_SETTINGS *pSettings = hLppTrans->pSettings; + PATCH_PARAM *patchParam = pSettings->patchParam; + + FIXP_SGL alphar[LPC_ORDER], a0r, a1r; + FIXP_SGL alphai[LPC_ORDER], a0i = 0, a1i = 0; + FIXP_SGL bw = FL2FXCONST_SGL(0.0f); + + int autoCorrLength; + + ACORR_COEFS ac; + int startSample; + int stopSample; + int stopSampleClear; + + int comBandScale; + int ovLowBandShift; + int lowBandShift; + /* int ovHighBandShift;*/ + int targetStopBand; + + alphai[0] = FL2FXCONST_SGL(0.0f); + alphai[1] = FL2FXCONST_SGL(0.0f); + + startSample = firstSlotOffs * timeStep; + stopSample = pSettings->nCols + lastSlotOffs * timeStep; + + inverseFilteringLevelEmphasis(hLppTrans, nInvfBands, sbr_invf_mode, + sbr_invf_mode_prev, bwVector); + + stopSampleClear = stopSample; + + autoCorrLength = pSettings->nCols + pSettings->overlap; + + /* Set upper subbands to zero: + This is required in case that the patches do not cover the complete + highband (because the last patch would be too short). Possible + optimization: Clearing bands up to usb would be sufficient here. */ + targetStopBand = patchParam[pSettings->noOfPatches - 1].targetStartBand + + patchParam[pSettings->noOfPatches - 1].numBandsInPatch; + + int memSize = ((64) - targetStopBand) * sizeof(FIXP_DBL); + + for (i = startSample; i < stopSampleClear; i++) { + FDKmemclear(&qmfBufferReal[i][targetStopBand], memSize); + FDKmemclear(&qmfBufferImag[i][targetStopBand], memSize); + } + + /* + Calc common low band scale factor + */ + comBandScale = sbrScaleFactor->hb_scale; + ovLowBandShift = sbrScaleFactor->hb_scale - comBandScale; + lowBandShift = sbrScaleFactor->hb_scale - comBandScale; + /* ovHighBandShift = firstSlotOffs == 0 ? ovLowBandShift:0;*/ + + /* outer loop over bands to do analysis only once for each band */ + + start = hQmfTransposer->startBand; + stop = hQmfTransposer->stopBand; + + for (loBand = start; loBand < stop; loBand++) { + bwIndex = 0; + + FIXP_DBL lowBandReal[(((1024) / (32) * (4) / 2) + (3 * (4))) + LPC_ORDER]; + FIXP_DBL lowBandImag[(((1024) / (32) * (4) / 2) + (3 * (4))) + LPC_ORDER]; + + int resetLPCCoeffs = 0; + int dynamicScale = DFRACT_BITS - 1 - LPC_SCALE_FACTOR; + int acDetScale = 0; /* scaling of autocorrelation determinant */ + + for (i = 0; i < LPC_ORDER; i++) { + lowBandReal[i] = hLppTrans->lpcFilterStatesRealHBE[i][loBand]; + lowBandImag[i] = hLppTrans->lpcFilterStatesImagHBE[i][loBand]; + } + + for (; i < LPC_ORDER + firstSlotOffs * timeStep; i++) { + lowBandReal[i] = hLppTrans->lpcFilterStatesRealHBE[i][loBand]; + lowBandImag[i] = hLppTrans->lpcFilterStatesImagHBE[i][loBand]; + } + + /* + Take old slope length qmf slot source values out of (overlap)qmf buffer + */ + for (i = firstSlotOffs * timeStep; + i < pSettings->nCols + pSettings->overlap; i++) { + lowBandReal[i + LPC_ORDER] = qmfBufferReal[i][loBand]; + lowBandImag[i + LPC_ORDER] = qmfBufferImag[i][loBand]; + } + + /* store unmodified values to buffer */ + for (i = 0; i < LPC_ORDER + pSettings->overlap; i++) { + hLppTrans->lpcFilterStatesRealHBE[i][loBand] = + qmfBufferReal[pSettings->nCols - LPC_ORDER + i][loBand]; + hLppTrans->lpcFilterStatesImagHBE[i][loBand] = + qmfBufferImag[pSettings->nCols - LPC_ORDER + i][loBand]; + } + + /* + Determine dynamic scaling value. + */ + dynamicScale = + fixMin(dynamicScale, + getScalefactor(lowBandReal, LPC_ORDER + pSettings->overlap) + + ovLowBandShift); + dynamicScale = + fixMin(dynamicScale, + getScalefactor(&lowBandReal[LPC_ORDER + pSettings->overlap], + pSettings->nCols) + + lowBandShift); + dynamicScale = + fixMin(dynamicScale, + getScalefactor(lowBandImag, LPC_ORDER + pSettings->overlap) + + ovLowBandShift); + dynamicScale = + fixMin(dynamicScale, + getScalefactor(&lowBandImag[LPC_ORDER + pSettings->overlap], + pSettings->nCols) + + lowBandShift); + + dynamicScale = fixMax( + 0, dynamicScale - 1); /* one additional bit headroom to prevent -1.0 */ + + /* + Scale temporal QMF buffer. + */ + scaleValues(&lowBandReal[0], LPC_ORDER + pSettings->overlap, + dynamicScale - ovLowBandShift); + scaleValues(&lowBandReal[LPC_ORDER + pSettings->overlap], pSettings->nCols, + dynamicScale - lowBandShift); + scaleValues(&lowBandImag[0], LPC_ORDER + pSettings->overlap, + dynamicScale - ovLowBandShift); + scaleValues(&lowBandImag[LPC_ORDER + pSettings->overlap], pSettings->nCols, + dynamicScale - lowBandShift); + + acDetScale += autoCorr2nd_cplx(&ac, lowBandReal + LPC_ORDER, + lowBandImag + LPC_ORDER, autoCorrLength); + + /* Examine dynamic of determinant in autocorrelation. */ + acDetScale += 2 * (comBandScale + dynamicScale); + acDetScale *= 2; /* two times reflection coefficent scaling */ + acDetScale += ac.det_scale; /* ac scaling of determinant */ + + /* In case of determinant < 10^-38, resetLPCCoeffs=1 has to be enforced. */ + if (acDetScale > 126) { + resetLPCCoeffs = 1; + } + + alphar[1] = FL2FXCONST_SGL(0.0f); + alphai[1] = FL2FXCONST_SGL(0.0f); + + if (ac.det != FL2FXCONST_DBL(0.0f)) { + FIXP_DBL tmp, absTmp, absDet; + + absDet = fixp_abs(ac.det); + + tmp = (fMultDiv2(ac.r01r, ac.r12r) >> (LPC_SCALE_FACTOR - 1)) - + ((fMultDiv2(ac.r01i, ac.r12i) + fMultDiv2(ac.r02r, ac.r11r)) >> + (LPC_SCALE_FACTOR - 1)); + absTmp = fixp_abs(tmp); + + /* + Quick check: is first filter coeff >= 1(4) + */ + { + INT scale; + FIXP_DBL result = fDivNorm(absTmp, absDet, &scale); + scale = scale + ac.det_scale; + + if ((scale > 0) && (result >= (FIXP_DBL)MAXVAL_DBL >> scale)) { + resetLPCCoeffs = 1; + } else { + alphar[1] = FX_DBL2FX_SGL(scaleValue(result, scale)); + if ((tmp < FL2FX_DBL(0.0f)) ^ (ac.det < FL2FX_DBL(0.0f))) { + alphar[1] = -alphar[1]; + } + } + } + + tmp = (fMultDiv2(ac.r01i, ac.r12r) >> (LPC_SCALE_FACTOR - 1)) + + ((fMultDiv2(ac.r01r, ac.r12i) - + (FIXP_DBL)fMultDiv2(ac.r02i, ac.r11r)) >> + (LPC_SCALE_FACTOR - 1)); + + absTmp = fixp_abs(tmp); + + /* + Quick check: is second filter coeff >= 1(4) + */ + { + INT scale; + FIXP_DBL result = fDivNorm(absTmp, absDet, &scale); + scale = scale + ac.det_scale; + + if ((scale > 0) && + (result >= /*FL2FXCONST_DBL(1.f)*/ (FIXP_DBL)MAXVAL_DBL >> scale)) { + resetLPCCoeffs = 1; + } else { + alphai[1] = FX_DBL2FX_SGL(scaleValue(result, scale)); + if ((tmp < FL2FX_DBL(0.0f)) ^ (ac.det < FL2FX_DBL(0.0f))) { + alphai[1] = -alphai[1]; + } + } + } + } + + alphar[0] = FL2FXCONST_SGL(0.0f); + alphai[0] = FL2FXCONST_SGL(0.0f); + + if (ac.r11r != FL2FXCONST_DBL(0.0f)) { + /* ac.r11r is always >=0 */ + FIXP_DBL tmp, absTmp; + + tmp = (ac.r01r >> (LPC_SCALE_FACTOR + 1)) + + (fMultDiv2(alphar[1], ac.r12r) + fMultDiv2(alphai[1], ac.r12i)); + + absTmp = fixp_abs(tmp); + + /* + Quick check: is first filter coeff >= 1(4) + */ + + if (absTmp >= (ac.r11r >> 1)) { + resetLPCCoeffs = 1; + } else { + INT scale; + FIXP_DBL result = fDivNorm(absTmp, fixp_abs(ac.r11r), &scale); + alphar[0] = FX_DBL2FX_SGL(scaleValue(result, scale + 1)); + + if ((tmp > FL2FX_DBL(0.0f)) ^ (ac.r11r < FL2FX_DBL(0.0f))) + alphar[0] = -alphar[0]; + } + + tmp = (ac.r01i >> (LPC_SCALE_FACTOR + 1)) + + (fMultDiv2(alphai[1], ac.r12r) - fMultDiv2(alphar[1], ac.r12i)); + + absTmp = fixp_abs(tmp); + + /* + Quick check: is second filter coeff >= 1(4) + */ + if (absTmp >= (ac.r11r >> 1)) { + resetLPCCoeffs = 1; + } else { + INT scale; + FIXP_DBL result = fDivNorm(absTmp, fixp_abs(ac.r11r), &scale); + alphai[0] = FX_DBL2FX_SGL(scaleValue(result, scale + 1)); + if ((tmp > FL2FX_DBL(0.0f)) ^ (ac.r11r < FL2FX_DBL(0.0f))) { + alphai[0] = -alphai[0]; + } + } + } + + /* Now check the quadratic criteria */ + if ((fMultDiv2(alphar[0], alphar[0]) + fMultDiv2(alphai[0], alphai[0])) >= + FL2FXCONST_DBL(0.5f)) { + resetLPCCoeffs = 1; + } + if ((fMultDiv2(alphar[1], alphar[1]) + fMultDiv2(alphai[1], alphai[1])) >= + FL2FXCONST_DBL(0.5f)) { + resetLPCCoeffs = 1; + } + + if (resetLPCCoeffs) { + alphar[0] = FL2FXCONST_SGL(0.0f); + alphar[1] = FL2FXCONST_SGL(0.0f); + alphai[0] = FL2FXCONST_SGL(0.0f); + alphai[1] = FL2FXCONST_SGL(0.0f); + } + + while (bwIndex < MAX_NUM_PATCHES - 1 && + loBand >= pSettings->bwBorders[bwIndex]) { + bwIndex++; + } + + /* + Filter Step 2: add the left slope with the current filter to the buffer + pure source values are already in there + */ + bw = FX_DBL2FX_SGL(bwVector[bwIndex]); + + a0r = FX_DBL2FX_SGL( + fMult(bw, alphar[0])); /* Apply current bandwidth expansion factor */ + a0i = FX_DBL2FX_SGL(fMult(bw, alphai[0])); + bw = FX_DBL2FX_SGL(fPow2(bw)); + a1r = FX_DBL2FX_SGL(fMult(bw, alphar[1])); + a1i = FX_DBL2FX_SGL(fMult(bw, alphai[1])); + + /* + Filter Step 3: insert the middle part which won't be windowed + */ + if (bw <= FL2FXCONST_SGL(0.0f)) { + int descale = fixMin(DFRACT_BITS - 1, (LPC_SCALE_FACTOR + dynamicScale)); + for (i = startSample; i < stopSample; i++) { + qmfBufferReal[i][loBand] = lowBandReal[LPC_ORDER + i] >> descale; + qmfBufferImag[i][loBand] = lowBandImag[LPC_ORDER + i] >> descale; + } + } else { /* bw <= 0 */ + + int descale = fixMin(DFRACT_BITS - 1, (LPC_SCALE_FACTOR + dynamicScale)); + + for (i = startSample; i < stopSample; i++) { + FIXP_DBL accu1, accu2; + + accu1 = (fMultDiv2(a0r, lowBandReal[LPC_ORDER + i - 1]) - + fMultDiv2(a0i, lowBandImag[LPC_ORDER + i - 1]) + + fMultDiv2(a1r, lowBandReal[LPC_ORDER + i - 2]) - + fMultDiv2(a1i, lowBandImag[LPC_ORDER + i - 2])) >> + dynamicScale; + accu2 = (fMultDiv2(a0i, lowBandReal[LPC_ORDER + i - 1]) + + fMultDiv2(a0r, lowBandImag[LPC_ORDER + i - 1]) + + fMultDiv2(a1i, lowBandReal[LPC_ORDER + i - 2]) + + fMultDiv2(a1r, lowBandImag[LPC_ORDER + i - 2])) >> + dynamicScale; + + qmfBufferReal[i][loBand] = + (lowBandReal[LPC_ORDER + i] >> descale) + (accu1 << 1); + qmfBufferImag[i][loBand] = + (lowBandImag[LPC_ORDER + i] >> descale) + (accu2 << 1); + } + } /* bw <= 0 */ + + /* + * store the unmodified filter coefficients if there is + * an overlapping envelope + *****************************************************************/ + + } /* outer loop over bands (loBand) */ + + for (i = 0; i < nInvfBands; i++) { + hLppTrans->bwVectorOld[i] = bwVector[i]; + } + + /* + set high band scale factor + */ + sbrScaleFactor->hb_scale = comBandScale - (LPC_SCALE_FACTOR); } /*! @@ -755,21 +1202,20 @@ void lppTransposer (HANDLE_SBR_LPP_TRANS hLppTrans, /*!< Handle of lpp transp * */ SBR_ERROR -createLppTransposer (HANDLE_SBR_LPP_TRANS hs, /*!< Handle of low power transposer */ - TRANSPOSER_SETTINGS *pSettings, /*!< Pointer to settings */ - const int highBandStartSb, /*!< ? */ - UCHAR *v_k_master, /*!< Master table */ - const int numMaster, /*!< Valid entries in master table */ - const int usb, /*!< Highband area stop subband */ - const int timeSlots, /*!< Number of time slots */ - const int nCols, /*!< Number of colums (codec qmf bank) */ - UCHAR *noiseBandTable, /*!< Mapping of SBR noise bands to QMF bands */ - const int noNoiseBands, /*!< Number of noise bands */ - UINT fs, /*!< Sample Frequency */ - const int chan, /*!< Channel number */ - const int overlap - ) -{ +createLppTransposer( + HANDLE_SBR_LPP_TRANS hs, /*!< Handle of low power transposer */ + TRANSPOSER_SETTINGS *pSettings, /*!< Pointer to settings */ + const int highBandStartSb, /*!< ? */ + UCHAR *v_k_master, /*!< Master table */ + const int numMaster, /*!< Valid entries in master table */ + const int usb, /*!< Highband area stop subband */ + const int timeSlots, /*!< Number of time slots */ + const int nCols, /*!< Number of colums (codec qmf bank) */ + UCHAR *noiseBandTable, /*!< Mapping of SBR noise bands to QMF bands */ + const int noNoiseBands, /*!< Number of noise bands */ + UINT fs, /*!< Sample Frequency */ + const int chan, /*!< Channel number */ + const int overlap) { /* FB inverse filtering settings */ hs->pSettings = pSettings; @@ -777,50 +1223,40 @@ createLppTransposer (HANDLE_SBR_LPP_TRANS hs, /*!< Handle of low power transpose pSettings->overlap = overlap; switch (timeSlots) { + case 15: + case 16: + break; - case 15: - case 16: - break; - - default: - return SBRDEC_UNSUPPORTED_CONFIG; /* Unimplemented */ + default: + return SBRDEC_UNSUPPORTED_CONFIG; /* Unimplemented */ } - if (chan==0) { + if (chan == 0) { /* Init common data only once */ hs->pSettings->nCols = nCols; - return resetLppTransposer (hs, - highBandStartSb, - v_k_master, - numMaster, - noiseBandTable, - noNoiseBands, - usb, - fs); + return resetLppTransposer(hs, highBandStartSb, v_k_master, numMaster, + noiseBandTable, noNoiseBands, usb, fs); } return SBRDEC_OK; } - -static int findClosestEntry(UCHAR goalSb, UCHAR *v_k_master, UCHAR numMaster, UCHAR direction) -{ +static int findClosestEntry(UCHAR goalSb, UCHAR *v_k_master, UCHAR numMaster, + UCHAR direction) { int index; - if( goalSb <= v_k_master[0] ) - return v_k_master[0]; + if (goalSb <= v_k_master[0]) return v_k_master[0]; - if( goalSb >= v_k_master[numMaster] ) - return v_k_master[numMaster]; + if (goalSb >= v_k_master[numMaster]) return v_k_master[numMaster]; - if(direction) { + if (direction) { index = 0; - while( v_k_master[index] < goalSb ) { + while (v_k_master[index] < goalSb) { index++; } } else { index = numMaster; - while( v_k_master[index] > goalSb ) { + while (v_k_master[index] > goalSb) { index--; } } @@ -828,7 +1264,6 @@ static int findClosestEntry(UCHAR goalSb, UCHAR *v_k_master, UCHAR numMaster, UC return v_k_master[index]; } - /*! * * \brief Reset memory for one lpp transposer instance @@ -836,18 +1271,18 @@ static int findClosestEntry(UCHAR goalSb, UCHAR *v_k_master, UCHAR numMaster, UC * \return SBRDEC_OK on success, SBRDEC_UNSUPPORTED_CONFIG on error */ SBR_ERROR -resetLppTransposer (HANDLE_SBR_LPP_TRANS hLppTrans, /*!< Handle of lpp transposer */ - UCHAR highBandStartSb, /*!< High band area: start subband */ - UCHAR *v_k_master, /*!< Master table */ - UCHAR numMaster, /*!< Valid entries in master table */ - UCHAR *noiseBandTable, /*!< Mapping of SBR noise bands to QMF bands */ - UCHAR noNoiseBands, /*!< Number of noise bands */ - UCHAR usb, /*!< High band area: stop subband */ - UINT fs /*!< SBR output sampling frequency */ - ) -{ +resetLppTransposer( + HANDLE_SBR_LPP_TRANS hLppTrans, /*!< Handle of lpp transposer */ + UCHAR highBandStartSb, /*!< High band area: start subband */ + UCHAR *v_k_master, /*!< Master table */ + UCHAR numMaster, /*!< Valid entries in master table */ + UCHAR *noiseBandTable, /*!< Mapping of SBR noise bands to QMF bands */ + UCHAR noNoiseBands, /*!< Number of noise bands */ + UCHAR usb, /*!< High band area: stop subband */ + UINT fs /*!< SBR output sampling frequency */ +) { TRANSPOSER_SETTINGS *pSettings = hLppTrans->pSettings; - PATCH_PARAM *patchParam = pSettings->patchParam; + PATCH_PARAM *patchParam = pSettings->patchParam; int i, patch; int targetStopBand; @@ -855,43 +1290,52 @@ resetLppTransposer (HANDLE_SBR_LPP_TRANS hLppTrans, /*!< Handle of lpp transpos int patchDistance; int numBandsInPatch; - int lsb = v_k_master[0]; /* Start subband expressed in "non-critical" sampling terms*/ - int xoverOffset = highBandStartSb - lsb; /* Calculate distance in QMF bands between k0 and kx */ + int lsb = v_k_master[0]; /* Start subband expressed in "non-critical" sampling + terms*/ + int xoverOffset = highBandStartSb - + lsb; /* Calculate distance in QMF bands between k0 and kx */ int startFreqHz; int desiredBorder; - usb = fixMin(usb, v_k_master[numMaster]); /* Avoid endless loops (compare with float code). */ + usb = fixMin(usb, v_k_master[numMaster]); /* Avoid endless loops (compare with + float code). */ /* * Plausibility check */ - if ( lsb - SHIFT_START_SB < 4 ) { + if (pSettings->nCols == 64) { + if (lsb < 4) { + /* 4:1 SBR Requirement k0 >= 4 missed! */ + return SBRDEC_UNSUPPORTED_CONFIG; + } + } else if (lsb - SHIFT_START_SB < 4) { return SBRDEC_UNSUPPORTED_CONFIG; } - /* * Initialize the patching parameter */ /* ISO/IEC 14496-3 (Figure 4.48): goalSb = round( 2.048e6 / fs ) */ - desiredBorder = (((2048000*2) / fs) + 1) >> 1; + desiredBorder = (((2048000 * 2) / fs) + 1) >> 1; - desiredBorder = findClosestEntry(desiredBorder, v_k_master, numMaster, 1); /* Adapt region to master-table */ + desiredBorder = findClosestEntry(desiredBorder, v_k_master, numMaster, + 1); /* Adapt region to master-table */ /* First patch */ sourceStartBand = SHIFT_START_SB + xoverOffset; targetStopBand = lsb + xoverOffset; /* upperBand */ - /* Even (odd) numbered channel must be patched to even (odd) numbered channel */ + /* Even (odd) numbered channel must be patched to even (odd) numbered channel + */ patch = 0; - while(targetStopBand < usb) { - + while (targetStopBand < usb) { /* Too many patches? Allow MAX_NUM_PATCHES+1 patches here. we need to check later again, since patch might be the highest patch - AND contain less than 3 bands => actual number of patches will be reduced by 1. + AND contain less than 3 bands => actual number of patches will be reduced + by 1. */ if (patch > MAX_NUM_PATCHES) { return SBRDEC_UNSUPPORTED_CONFIG; @@ -900,26 +1344,42 @@ resetLppTransposer (HANDLE_SBR_LPP_TRANS hLppTrans, /*!< Handle of lpp transpos patchParam[patch].guardStartBand = targetStopBand; patchParam[patch].targetStartBand = targetStopBand; - numBandsInPatch = desiredBorder - targetStopBand; /* Get the desired range of the patch */ + numBandsInPatch = + desiredBorder - targetStopBand; /* Get the desired range of the patch */ - if ( numBandsInPatch >= lsb - sourceStartBand ) { + if (numBandsInPatch >= lsb - sourceStartBand) { /* Desired number bands are not available -> patch whole source range */ - patchDistance = targetStopBand - sourceStartBand; /* Get the targetOffset */ - patchDistance = patchDistance & ~1; /* Rounding off odd numbers and make all even */ - numBandsInPatch = lsb - (targetStopBand - patchDistance); /* Update number of bands to be patched */ - numBandsInPatch = findClosestEntry(targetStopBand + numBandsInPatch, v_k_master, numMaster, 0) - - targetStopBand; /* Adapt region to master-table */ + patchDistance = + targetStopBand - sourceStartBand; /* Get the targetOffset */ + patchDistance = + patchDistance & ~1; /* Rounding off odd numbers and make all even */ + numBandsInPatch = + lsb - (targetStopBand - + patchDistance); /* Update number of bands to be patched */ + numBandsInPatch = findClosestEntry(targetStopBand + numBandsInPatch, + v_k_master, numMaster, 0) - + targetStopBand; /* Adapt region to master-table */ + } + + if (pSettings->nCols == 64) { + if (numBandsInPatch == 0 && sourceStartBand == SHIFT_START_SB) { + return SBRDEC_UNSUPPORTED_CONFIG; + } } - /* Desired number bands are available -> get the minimal even patching distance */ - patchDistance = numBandsInPatch + targetStopBand - lsb; /* Get minimal distance */ - patchDistance = (patchDistance + 1) & ~1; /* Rounding up odd numbers and make all even */ + /* Desired number bands are available -> get the minimal even patching + * distance */ + patchDistance = + numBandsInPatch + targetStopBand - lsb; /* Get minimal distance */ + patchDistance = (patchDistance + 1) & + ~1; /* Rounding up odd numbers and make all even */ if (numBandsInPatch > 0) { patchParam[patch].sourceStartBand = targetStopBand - patchDistance; - patchParam[patch].targetBandOffs = patchDistance; + patchParam[patch].targetBandOffs = patchDistance; patchParam[patch].numBandsInPatch = numBandsInPatch; - patchParam[patch].sourceStopBand = patchParam[patch].sourceStartBand + numBandsInPatch; + patchParam[patch].sourceStopBand = + patchParam[patch].sourceStartBand + numBandsInPatch; targetStopBand += patchParam[patch].numBandsInPatch; patch++; @@ -929,19 +1389,19 @@ resetLppTransposer (HANDLE_SBR_LPP_TRANS hLppTrans, /*!< Handle of lpp transpos sourceStartBand = SHIFT_START_SB; /* Check if we are close to desiredBorder */ - if( desiredBorder - targetStopBand < 3) /* MPEG doc */ + if (desiredBorder - targetStopBand < 3) /* MPEG doc */ { desiredBorder = usb; } - } patch--; /* If highest patch contains less than three subband: skip it */ - if ( (patch>0) && (patchParam[patch].numBandsInPatch < 3) ) { + if ((patch > 0) && (patchParam[patch].numBandsInPatch < 3)) { patch--; - targetStopBand = patchParam[patch].targetStartBand + patchParam[patch].numBandsInPatch; + targetStopBand = + patchParam[patch].targetStartBand + patchParam[patch].numBandsInPatch; } /* now check if we don't have one too many */ @@ -953,31 +1413,36 @@ resetLppTransposer (HANDLE_SBR_LPP_TRANS hLppTrans, /*!< Handle of lpp transpos /* Check lowest and highest source subband */ pSettings->lbStartPatching = targetStopBand; - pSettings->lbStopPatching = 0; - for ( patch = 0; patch < pSettings->noOfPatches; patch++ ) { - pSettings->lbStartPatching = fixMin( pSettings->lbStartPatching, patchParam[patch].sourceStartBand ); - pSettings->lbStopPatching = fixMax( pSettings->lbStopPatching, patchParam[patch].sourceStopBand ); + pSettings->lbStopPatching = 0; + for (patch = 0; patch < pSettings->noOfPatches; patch++) { + pSettings->lbStartPatching = + fixMin(pSettings->lbStartPatching, patchParam[patch].sourceStartBand); + pSettings->lbStopPatching = + fixMax(pSettings->lbStopPatching, patchParam[patch].sourceStopBand); } - for(i = 0 ; i < noNoiseBands; i++){ - pSettings->bwBorders[i] = noiseBandTable[i+1]; + for (i = 0; i < noNoiseBands; i++) { + pSettings->bwBorders[i] = noiseBandTable[i + 1]; + } + for (; i < MAX_NUM_NOISE_VALUES; i++) { + pSettings->bwBorders[i] = 255; } /* * Choose whitening factors */ - startFreqHz = ( (lsb + xoverOffset)*fs ) >> 7; /* Shift does a division by 2*(64) */ + startFreqHz = + ((lsb + xoverOffset) * fs) >> 7; /* Shift does a division by 2*(64) */ - for( i = 1; i < NUM_WHFACTOR_TABLE_ENTRIES; i++ ) - { - if( startFreqHz < FDK_sbrDecoder_sbr_whFactorsIndex[i]) - break; + for (i = 1; i < NUM_WHFACTOR_TABLE_ENTRIES; i++) { + if (startFreqHz < FDK_sbrDecoder_sbr_whFactorsIndex[i]) break; } i--; pSettings->whFactors.off = FDK_sbrDecoder_sbr_whFactorsTable[i][0]; - pSettings->whFactors.transitionLevel = FDK_sbrDecoder_sbr_whFactorsTable[i][1]; + pSettings->whFactors.transitionLevel = + FDK_sbrDecoder_sbr_whFactorsTable[i][1]; pSettings->whFactors.lowLevel = FDK_sbrDecoder_sbr_whFactorsTable[i][2]; pSettings->whFactors.midLevel = FDK_sbrDecoder_sbr_whFactorsTable[i][3]; pSettings->whFactors.highLevel = FDK_sbrDecoder_sbr_whFactorsTable[i][4]; |