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Diffstat (limited to 'libSBRdec/src/lpp_tran.cpp')
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diff --git a/libSBRdec/src/lpp_tran.cpp b/libSBRdec/src/lpp_tran.cpp deleted file mode 100644 index 117e739..0000000 --- a/libSBRdec/src/lpp_tran.cpp +++ /dev/null @@ -1,986 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -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. - - 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. - -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: - -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 -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. - -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." - -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. - -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. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/*! - \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. - - \sa lppTransposer(), main_audio.cpp, sbr_scale.h, \ref documentationOverview -*/ - -#include "lpp_tran.h" - -#include "sbr_ram.h" -#include "sbr_rom.h" - -#include "genericStds.h" -#include "autocorr2nd.h" - - - -#if defined(__arm__) -#include "arm/lpp_tran_arm.cpp" -#endif - - - -#define LPC_SCALE_FACTOR 2 - - -/*! - * - * \brief Get bandwidth expansion factor from filtering level - * - * Returns a filter parameter (bandwidth expansion factor) depending on - * the desired filtering level signalled in the bitstream. - * When switching the filtering level from LOW to OFF, an additional - * 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) -{ - 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; - } -} -#endif /* #ifndef FUNCTION_mapInvfMode */ - -/*! - * - * \brief Perform inverse filtering level emphasis - * - * Retrieve bandwidth expansion factor and apply smoothing for each filter band - * - */ - -#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++) { - 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]); - } - - if (accu < FL2FXCONST_DBL(0.015625f)>>1) - bwVector[i] = FL2FXCONST_DBL(0.0f); - 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) - -/*! - * - * \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(). - * - * 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; - TRANSPOSER_SETTINGS *pSettings = hLppTrans->pSettings; - PATCH_PARAM *patchParam = pSettings->patchParam; - 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); - - int autoCorrLength; - - FIXP_DBL k1, k1_below=0, k1_below2=0; - - ACORR_COEFS ac; - int startSample; - int stopSample; - int stopSampleClear; - - int comLowBandScale; - 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); - - if (!useLP) { - for (i = startSample; i < stopSampleClear; i++) { - FDKmemclear(&qmfBufferReal[i][targetStopBand], memSize); - FDKmemclear(&qmfBufferImag[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)); - - /* - Calc common low band scale factor - */ - comLowBandScale = fixMin(sbrScaleFactor->ov_lb_scale,sbrScaleFactor->lb_scale); - - ovLowBandShift = sbrScaleFactor->ov_lb_scale - comLowBandScale; - lowBandShift = sbrScaleFactor->lb_scale - comLowBandScale; - /* ovHighBandShift = firstSlotOffs == 0 ? ovLowBandShift:0;*/ - - /* outer loop over bands to do analysis only once for each band */ - - if (!useLP) { - start = pSettings->lbStartPatching; - stop = pSettings->lbStopPatching; - } 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]; - FIXP_DBL *plowBandReal = lowBandReal; - FIXP_DBL **pqmfBufferReal = qmfBufferReal; - FIXP_DBL lowBandImag[(((1024)/(32))+(6))+LPC_ORDER]; - FIXP_DBL *plowBandImag = lowBandImag; - FIXP_DBL **pqmfBufferImag = qmfBufferImag; - 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]; - if (!useLP) - *plowBandImag++ = hLppTrans->lpcFilterStatesImag[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++){ - *plowBandReal++ = (*pqmfBufferReal++)[loBand]; - *plowBandImag++ = (*pqmfBufferImag++)[loBand]; - } - } else - { - /* pSettings->overlap is always even */ - FDK_ASSERT((pSettings->overlap & 1) == 0); - - for(i=0;i<((pSettings->overlap+pSettings->nCols)>>1);i++) { - *plowBandReal++ = (*pqmfBufferReal++)[loBand]; - *plowBandReal++ = (*pqmfBufferReal++)[loBand]; - } - if (pSettings->nCols & 1) { - *plowBandReal++ = (*pqmfBufferReal++)[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); - 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 = 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); - - if (!useLP) { - 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); - } - - /* 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 (ac.det != FL2FXCONST_DBL(0.0f)) { - 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) ); - } - 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]; - } - } - } - - 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); - - /* - 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); - if (!useLP) - alphai[0] = FL2FXCONST_SGL(0.0f); - - if ( ac.r11r != FL2FXCONST_DBL(0.0f) ) { - - /* ac.r11r is always >=0 */ - 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); - else - tmp = -((-ac.r01r)>>(LPC_SCALE_FACTOR+1)) + fMultDiv2(alphar[1],ac.r12r); - } - - 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]; - } - - 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 { - 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]; - } - } - } - - - 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(resetLPCCoeffs){ - alphar[0] = FL2FXCONST_SGL(0.0f); - alphar[1] = FL2FXCONST_SGL(0.0f); - if (!useLP) - { - alphai[0] = FL2FXCONST_SGL(0.0f); - alphai[1] = FL2FXCONST_SGL(0.0f); - } - } - - if (useLP) - { - - /* Aliasing detection */ - 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)) { - 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 { - INT 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)))) { - k1 = -k1; - } - } - } - if(loBand > 1){ - /* Check if the gain should be locked */ - 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 */ - 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; - } - } - 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 */ - 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; - } - } - else if ( k1_below2 < FL2FXCONST_DBL(0.0f) ) { /* 3-CH Aliasing Detection */ - degreeAlias[loBand] = deg; - } - } - } - /* remember k1 values of the 2 QMF channels below the current channel */ - k1_below2 = k1_below; - k1_below = k1; - } - - patch = 0; - - 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 - ) { - /* Lowband not in current patch - proceed */ - patch++; - continue; - } - - 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]++; - - - /* - 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[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])); - - - - /* - Filter Step 3: insert the middle part which won't be windowed - */ - - 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; - } - } 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 */ - - if (!useLP) { - 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); -#else - 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); - } -#endif - } else - { - int descale = fixMin(DFRACT_BITS-1, (LPC_SCALE_FACTOR+dynamicScale)); - - 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); - } - } - } /* bw <= 0 */ - - patch++; - - } /* inner loop over patches */ - - /* - * store the unmodified filter coefficients if there is - * an overlapping envelope - *****************************************************************/ - - - } /* outer loop over bands (loBand) */ - - if (useLP) - { - for ( loBand = pSettings->lbStartPatching; loBand < pSettings->lbStopPatching; loBand++ ) { - patch = 0; - 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 */ - patch++; - continue; - } - - if(hiBand != patchParam[patch].targetStartBand) - degreeAlias[hiBand] = degreeAlias[loBand]; - - patch++; - } - }/* end for loop */ - } - - for (i = 0; i < nInvfBands; i++ ) { - hLppTrans->bwVectorOld[i] = bwVector[i]; - } - - /* - set high band scale factor - */ - sbrScaleFactor->hb_scale = comLowBandScale-(LPC_SCALE_FACTOR); - -} - -/*! - * - * \brief Initialize one low power transposer instance - * - * - */ -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 - ) -{ - /* FB inverse filtering settings */ - hs->pSettings = pSettings; - - pSettings->nCols = nCols; - pSettings->overlap = overlap; - - switch (timeSlots) { - - case 15: - case 16: - break; - - default: - return SBRDEC_UNSUPPORTED_CONFIG; /* Unimplemented */ - } - - 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 SBRDEC_OK; -} - - -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[numMaster] ) - return v_k_master[numMaster]; - - if(direction) { - index = 0; - while( v_k_master[index] < goalSb ) { - index++; - } - } else { - index = numMaster; - while( v_k_master[index] > goalSb ) { - index--; - } - } - - return v_k_master[index]; -} - - -/*! - * - * \brief Reset memory for one lpp transposer instance - * - * \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 */ - ) -{ - TRANSPOSER_SETTINGS *pSettings = hLppTrans->pSettings; - PATCH_PARAM *patchParam = pSettings->patchParam; - - int i, patch; - int targetStopBand; - int sourceStartBand; - 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 startFreqHz; - - int desiredBorder; - - usb = fixMin(usb, v_k_master[numMaster]); /* Avoid endless loops (compare with float code). */ - - /* - * Plausibility check - */ - - 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 = 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 */ - patch = 0; - 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. - */ - if (patch > MAX_NUM_PATCHES) { - return SBRDEC_UNSUPPORTED_CONFIG; - } - - patchParam[patch].guardStartBand = targetStopBand; - patchParam[patch].targetStartBand = targetStopBand; - - numBandsInPatch = desiredBorder - targetStopBand; /* Get the desired range of the patch */ - - 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 */ - } - - /* 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].numBandsInPatch = numBandsInPatch; - patchParam[patch].sourceStopBand = patchParam[patch].sourceStartBand + numBandsInPatch; - - targetStopBand += patchParam[patch].numBandsInPatch; - patch++; - } - - /* All patches but first */ - sourceStartBand = SHIFT_START_SB; - - /* Check if we are close to desiredBorder */ - 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) ) { - patch--; - targetStopBand = patchParam[patch].targetStartBand + patchParam[patch].numBandsInPatch; - } - - /* now check if we don't have one too many */ - if (patch >= MAX_NUM_PATCHES) { - return SBRDEC_UNSUPPORTED_CONFIG; - } - - pSettings->noOfPatches = patch + 1; - - /* 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 ); - } - - for(i = 0 ; i < noNoiseBands; i++){ - pSettings->bwBorders[i] = noiseBandTable[i+1]; - } - - /* - * Choose whitening factors - */ - - 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; - } - i--; - - pSettings->whFactors.off = FDK_sbrDecoder_sbr_whFactorsTable[i][0]; - 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]; - - return SBRDEC_OK; -} |