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diff --git a/libSBRenc/src/ton_corr.cpp b/libSBRenc/src/ton_corr.cpp new file mode 100644 index 0000000..3142870 --- /dev/null +++ b/libSBRenc/src/ton_corr.cpp @@ -0,0 +1,883 @@ + +/* ----------------------------------------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2012 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 +----------------------------------------------------------------------------------------------------------- */ + +#include "ton_corr.h" + +#include "sbr_ram.h" +#include "sbr_misc.h" +#include "genericStds.h" +#include "autocorr2nd.h" + + + +/*************************************************************************** + + Send autoCorrSecondOrder to mlfile + +****************************************************************************/ + +/**************************************************************************/ +/*! + \brief Calculates the tonal to noise ration for different frequency bands + and time segments. + + The ratio between the predicted energy (tonal energy A) and the total + energy (A + B) is calculated. This is converted to the ratio between + the predicted energy (tonal energy A) and the non-predictable energy + (noise energy B). Hence the quota-matrix contains A/B = q/(1-q). + + The samples in nrgVector are scaled by 1.0/16.0 + The samples in pNrgVectorFreq are scaled by 1.0/2.0 + The samples in quotaMatrix are scaled by RELAXATION + + \return none. + +*/ +/**************************************************************************/ + +void +FDKsbrEnc_CalculateTonalityQuotas( HANDLE_SBR_TON_CORR_EST hTonCorr, /*!< Handle to SBR_TON_CORR struct. */ + FIXP_DBL **RESTRICT sourceBufferReal, /*!< The real part of the QMF-matrix. */ + FIXP_DBL **RESTRICT sourceBufferImag, /*!< The imaginary part of the QMF-matrix. */ + INT usb, /*!< upper side band, highest + 1 QMF band in the SBR range. */ + INT qmfScale /*!< sclefactor of QMF subsamples */ + ) +{ + INT i, k, r, r2, timeIndex, autoCorrScaling; + + INT startIndexMatrix = hTonCorr->startIndexMatrix; + INT totNoEst = hTonCorr->numberOfEstimates; + INT noEstPerFrame = hTonCorr->numberOfEstimatesPerFrame; + INT move = hTonCorr->move; + INT noQmfChannels = hTonCorr->noQmfChannels; /* Numer of Bands */ + INT buffLen = hTonCorr->bufferLength; /* Numer of Slots */ + INT stepSize = hTonCorr->stepSize; + INT *pBlockLength = hTonCorr->lpcLength; + INT** RESTRICT signMatrix = hTonCorr->signMatrix; + FIXP_DBL* RESTRICT nrgVector = hTonCorr->nrgVector; + FIXP_DBL** RESTRICT quotaMatrix = hTonCorr->quotaMatrix; + FIXP_DBL* RESTRICT pNrgVectorFreq = hTonCorr->nrgVectorFreq; + +#define BAND_V_SIZE QMF_MAX_TIME_SLOTS +#define NUM_V_COMBINE 8 /* Must be a divisor of 64 and fulfill the ASSERTs below */ + + FIXP_DBL *realBuf; + FIXP_DBL *imagBuf; + + FIXP_DBL alphar[2],alphai[2],fac; + + C_ALLOC_SCRATCH_START(ac, ACORR_COEFS, 1); + C_ALLOC_SCRATCH_START(realBufRef, FIXP_DBL, 2*BAND_V_SIZE*NUM_V_COMBINE); + + realBuf = realBufRef; + imagBuf = realBuf + BAND_V_SIZE*NUM_V_COMBINE; + + + FDK_ASSERT(buffLen <= BAND_V_SIZE); + FDK_ASSERT(sizeof(FIXP_DBL)*NUM_V_COMBINE*BAND_V_SIZE*2 < (1024*sizeof(FIXP_DBL)-sizeof(ACORR_COEFS)) ); + + /* + * Buffering of the quotaMatrix and the quotaMatrixTransp. + *********************************************************/ + for(i = 0 ; i < move; i++){ + FDKmemcpy(quotaMatrix[i],quotaMatrix[i + noEstPerFrame],noQmfChannels * sizeof(FIXP_DBL)); + FDKmemcpy(signMatrix[i],signMatrix[i + noEstPerFrame],noQmfChannels * sizeof(INT)); + } + + FDKmemmove(nrgVector,nrgVector+noEstPerFrame,move*sizeof(FIXP_DBL)); + FDKmemclear(nrgVector+startIndexMatrix,(totNoEst-startIndexMatrix)*sizeof(FIXP_DBL)); + FDKmemclear(pNrgVectorFreq,noQmfChannels * sizeof(FIXP_DBL)); + + /* + * Calculate the quotas for the current time steps. + **************************************************/ + + for (r = 0; r < usb; r++) + { + int blockLength; + + k = hTonCorr->nextSample; /* startSample */ + timeIndex = startIndexMatrix; + /* Copy as many as possible Band accross all Slots at once */ + if (realBuf != realBufRef) { + realBuf -= BAND_V_SIZE; + imagBuf -= BAND_V_SIZE; + } else { + realBuf += BAND_V_SIZE*(NUM_V_COMBINE-1); + imagBuf += BAND_V_SIZE*(NUM_V_COMBINE-1); + for (i = 0; i < buffLen; i++) { + int v; + FIXP_DBL *ptr; + ptr = realBuf+i; + for (v=0; v<NUM_V_COMBINE; v++) + { + ptr[0] = sourceBufferReal[i][r+v]; + ptr[0+BAND_V_SIZE*NUM_V_COMBINE] = sourceBufferImag[i][r+v]; + ptr -= BAND_V_SIZE; + } + } + } + + blockLength = pBlockLength[0]; + + while(k <= buffLen - blockLength) + { + autoCorrScaling = fixMin(getScalefactor(&realBuf[k-LPC_ORDER], LPC_ORDER+blockLength), getScalefactor(&imagBuf[k-LPC_ORDER], LPC_ORDER+blockLength)); + autoCorrScaling = fixMax(0, autoCorrScaling-1); + + scaleValues(&realBuf[k-LPC_ORDER], LPC_ORDER+blockLength, autoCorrScaling); + scaleValues(&imagBuf[k-LPC_ORDER], LPC_ORDER+blockLength, autoCorrScaling); + + autoCorrScaling <<= 1; /* consider qmf buffer scaling twice */ + autoCorrScaling += autoCorr2nd_cplx ( ac, realBuf+k, imagBuf+k, blockLength ); + + + if(ac->det == FL2FXCONST_DBL(0.0f)){ + alphar[1] = alphai[1] = FL2FXCONST_DBL(0.0f); + + alphar[0] = (ac->r01r)>>2; + alphai[0] = (ac->r01i)>>2; + + fac = fMultDiv2(ac->r00r, ac->r11r)>>1; + } + else{ + alphar[1] = (fMultDiv2(ac->r01r, ac->r12r)>>1) - (fMultDiv2(ac->r01i, ac->r12i)>>1) - (fMultDiv2(ac->r02r, ac->r11r)>>1); + alphai[1] = (fMultDiv2(ac->r01i, ac->r12r)>>1) + (fMultDiv2(ac->r01r, ac->r12i)>>1) - (fMultDiv2(ac->r02i, ac->r11r)>>1); + + alphar[0] = (fMultDiv2(ac->r01r, ac->det)>>(ac->det_scale+1)) + fMult(alphar[1], ac->r12r) + fMult(alphai[1], ac->r12i); + alphai[0] = (fMultDiv2(ac->r01i, ac->det)>>(ac->det_scale+1)) + fMult(alphai[1], ac->r12r) - fMult(alphar[1], ac->r12i); + + fac = fMultDiv2(ac->r00r, fMult(ac->det, ac->r11r))>>(ac->det_scale+1); + } + + if(fac == FL2FXCONST_DBL(0.0f)){ + quotaMatrix[timeIndex][r] = FL2FXCONST_DBL(0.0f); + signMatrix[timeIndex][r] = 0; + } + else { + /* quotaMatrix is scaled with the factor RELAXATION + parse RELAXATION in fractional part and shift factor: 1/(1/0.524288 * 2^RELAXATION_SHIFT) */ + FIXP_DBL tmp,num,denom; + INT numShift,denomShift,commonShift; + INT sign; + + num = fMultDiv2(alphar[0], ac->r01r) + fMultDiv2(alphai[0], ac->r01i) - fMultDiv2(alphar[1], fMult(ac->r02r, ac->r11r)) - fMultDiv2(alphai[1], fMult(ac->r02i, ac->r11r)); + num = fixp_abs(num); + + denom = (fac>>1) + (fMultDiv2(fac,RELAXATION_FRACT)>>RELAXATION_SHIFT) - num; + denom = fixp_abs(denom); + + num = fMult(num,RELAXATION_FRACT); + + numShift = CountLeadingBits(num) - 2; + num = scaleValue(num, numShift); + + denomShift = CountLeadingBits(denom); + denom = (FIXP_DBL)denom << denomShift; + + if ((num > FL2FXCONST_DBL(0.0f)) && (denom != FL2FXCONST_DBL(0.0f))) { + commonShift = fixMin(numShift - denomShift + RELAXATION_SHIFT, DFRACT_BITS-1); + if (commonShift < 0) { + commonShift = -commonShift; + tmp = schur_div(num,denom,16); + commonShift = fixMin(commonShift,CountLeadingBits(tmp)); + quotaMatrix[timeIndex][r] = tmp << commonShift; + } + else { + quotaMatrix[timeIndex][r] = schur_div(num,denom,16) >> commonShift; + } + } + else { + quotaMatrix[timeIndex][r] = FL2FXCONST_DBL(0.0f); + } + + if (ac->r11r != FL2FXCONST_DBL(0.0f)) { + if ( ( (ac->r01r >= FL2FXCONST_DBL(0.0f) ) && ( ac->r11r >= FL2FXCONST_DBL(0.0f) ) ) + ||( (ac->r01r < FL2FXCONST_DBL(0.0f) ) && ( ac->r11r < FL2FXCONST_DBL(0.0f) ) ) ) { + sign = 1; + } + else { + sign = -1; + } + } + else { + sign = 1; + } + + if(sign < 0) { + r2 = r; /* (INT) pow(-1, band); */ + } + else { + r2 = r + 1; /* (INT) pow(-1, band+1); */ + } + signMatrix[timeIndex][r] = 1 - 2*(r2 & 0x1); + } + + nrgVector[timeIndex] += ((ac->r00r) >> fixMin(DFRACT_BITS-1,(2*qmfScale+autoCorrScaling + SCALE_NRGVEC))); + /* pNrgVectorFreq[r] finally has to be divided by noEstPerFrame, replaced division by shifting with one */ + pNrgVectorFreq[r] = pNrgVectorFreq[r] + ((ac->r00r) >> fixMin(DFRACT_BITS-1,(2*qmfScale+autoCorrScaling + SCALE_NRGVEC))); + + blockLength = pBlockLength[1]; + k += stepSize; + timeIndex++; + } + } + + FDK_ASSERT(noEstPerFrame == 2); + + + C_ALLOC_SCRATCH_END(realBuf, FIXP_DBL, 2*BAND_V_SIZE*NUM_V_COMBINE); + C_ALLOC_SCRATCH_END(ac, ACORR_COEFS, 1); +} + +/**************************************************************************/ +/*! + \brief Extracts the parameters required in the decoder to obtain the + correct tonal to noise ratio after SBR. + + Estimates the tonal to noise ratio of the original signal (using LPC). + Predicts the tonal to noise ration of the SBR signal (in the decoder) by + patching the tonal to noise ratio values similar to the patching of the + lowband in the decoder. Given the tonal to noise ratio of the original + and the SBR signal, it estimates the required amount of inverse filtering, + additional noise as well as any additional sines. + + \return none. + +*/ +/**************************************************************************/ +void +FDKsbrEnc_TonCorrParamExtr(HANDLE_SBR_TON_CORR_EST hTonCorr,/*!< Handle to SBR_TON_CORR struct. */ + INVF_MODE* infVec, /*!< Vector where the inverse filtering levels will be stored. */ + FIXP_DBL * noiseLevels, /*!< Vector where the noise levels will be stored. */ + INT* missingHarmonicFlag, /*!< Flag set to one or zero, dependent on if any strong sines are missing.*/ + UCHAR * missingHarmonicsIndex, /*!< Vector indicating where sines are missing. */ + UCHAR * envelopeCompensation, /*!< Vector to store compensation values for the energies in. */ + const SBR_FRAME_INFO *frameInfo, /*!< Frame info struct, contains the time and frequency grid of the current frame.*/ + UCHAR* transientInfo, /*!< Transient info.*/ + UCHAR* freqBandTable, /*!< Frequency band tables for high-res.*/ + INT nSfb, /*!< Number of scalefactor bands for high-res. */ + XPOS_MODE xposType, /*!< Type of transposer used in the decoder.*/ + UINT sbrSyntaxFlags + ) +{ + INT band; + INT transientFlag = transientInfo[1] ; /*!< Flag indicating if a transient is present in the current frame. */ + INT transientPos = transientInfo[0]; /*!< Position of the transient.*/ + INT transientFrame, transientFrameInvfEst; + INVF_MODE* infVecPtr; + + + /* Determine if this is a frame where a transient starts... + + The detection of noise-floor, missing harmonics and invf_est, is not in sync for the + non-buf-opt decoder such as AAC. Hence we need to keep track on the transient in the + present frame as well as in the next. + */ + transientFrame = 0; + if(hTonCorr->transientNextFrame){ /* The transient was detected in the previous frame, but is actually */ + transientFrame = 1; + hTonCorr->transientNextFrame = 0; + + if(transientFlag){ + if(transientPos + hTonCorr->transientPosOffset >= frameInfo->borders[frameInfo->nEnvelopes]){ + hTonCorr->transientNextFrame = 1; + } + } + } + else{ + if(transientFlag){ + if(transientPos + hTonCorr->transientPosOffset < frameInfo->borders[frameInfo->nEnvelopes]){ + transientFrame = 1; + hTonCorr->transientNextFrame = 0; + } + else{ + hTonCorr->transientNextFrame = 1; + } + } + } + transientFrameInvfEst = transientFrame; + + + /* + Estimate the required invese filtereing level. + */ + if (hTonCorr->switchInverseFilt) + FDKsbrEnc_qmfInverseFilteringDetector(&hTonCorr->sbrInvFilt, + hTonCorr->quotaMatrix, + hTonCorr->nrgVector, + hTonCorr->indexVector, + hTonCorr->frameStartIndexInvfEst, + hTonCorr->numberOfEstimatesPerFrame + hTonCorr->frameStartIndexInvfEst, + transientFrameInvfEst, + infVec); + + /* + Detect what tones will be missing. + */ + if (xposType == XPOS_LC ){ + FDKsbrEnc_SbrMissingHarmonicsDetectorQmf(&hTonCorr->sbrMissingHarmonicsDetector, + hTonCorr->quotaMatrix, + hTonCorr->signMatrix, + hTonCorr->indexVector, + frameInfo, + transientInfo, + missingHarmonicFlag, + missingHarmonicsIndex, + freqBandTable, + nSfb, + envelopeCompensation, + hTonCorr->nrgVectorFreq); + } + else{ + *missingHarmonicFlag = 0; + FDKmemclear(missingHarmonicsIndex,nSfb*sizeof(UCHAR)); + } + + + + /* + Noise floor estimation + */ + + infVecPtr = hTonCorr->sbrInvFilt.prevInvfMode; + + FDKsbrEnc_sbrNoiseFloorEstimateQmf(&hTonCorr->sbrNoiseFloorEstimate, + frameInfo, + noiseLevels, + hTonCorr->quotaMatrix, + hTonCorr->indexVector, + *missingHarmonicFlag, + hTonCorr->frameStartIndex, + hTonCorr->numberOfEstimatesPerFrame, + transientFrame, + infVecPtr, + sbrSyntaxFlags); + + + /* Store the invfVec data for the next frame...*/ + for(band = 0 ; band < hTonCorr->sbrInvFilt.noDetectorBands; band++){ + hTonCorr->sbrInvFilt.prevInvfMode[band] = infVec[band]; + } +} + +/**************************************************************************/ +/*! + \brief Searches for the closest match in the frequency master table. + + + + \return closest entry. + +*/ +/**************************************************************************/ +static INT +findClosestEntry(INT goalSb, + UCHAR *v_k_master, + INT numMaster, + INT 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 resets the patch + + + + \return errorCode, noError if successful. + +*/ +/**************************************************************************/ +static INT +resetPatch(HANDLE_SBR_TON_CORR_EST hTonCorr, /*!< Handle to SBR_TON_CORR struct. */ + INT xposctrl, /*!< Different patch modes. */ + INT highBandStartSb, /*!< Start band of the SBR range. */ + UCHAR *v_k_master, /*!< Master frequency table from which all other table are derived.*/ + INT numMaster, /*!< Number of elements in the master table. */ + INT fs, /*!< Sampling frequency. */ + INT noChannels) /*!< Number of QMF-channels. */ +{ + INT patch,k,i; + INT targetStopBand; + + PATCH_PARAM *patchParam = hTonCorr->patchParam; + + INT sbGuard = hTonCorr->guard; + INT sourceStartBand; + INT patchDistance; + INT numBandsInPatch; + + INT lsb = v_k_master[0]; /* Lowest subband related to the synthesis filterbank */ + INT usb = v_k_master[numMaster]; /* Stop subband related to the synthesis filterbank */ + INT xoverOffset = highBandStartSb - v_k_master[0]; /* Calculate distance in subbands between k0 and kx */ + + INT goalSb; + + + /* + * Initialize the patching parameter + */ + + if (xposctrl == 1) { + lsb += xoverOffset; + xoverOffset = 0; + } + + goalSb = (INT)( (2 * noChannels * 16000 + (fs>>1)) / fs ); /* 16 kHz band */ + goalSb = findClosestEntry(goalSb, v_k_master, numMaster, 1); /* Adapt region to master-table */ + + /* First patch */ + sourceStartBand = hTonCorr->shiftStartSb + xoverOffset; + targetStopBand = lsb + xoverOffset; + + /* even (odd) numbered channel must be patched to even (odd) numbered channel */ + patch = 0; + while(targetStopBand < usb) { + + /* To many patches */ + if (patch >= MAX_NUM_PATCHES) + return(1); /*Number of patches to high */ + + patchParam[patch].guardStartBand = targetStopBand; + targetStopBand += sbGuard; + patchParam[patch].targetStartBand = targetStopBand; + + numBandsInPatch = goalSb - 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); + 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) { + patch--; + } else { + patchParam[patch].sourceStartBand = targetStopBand - patchDistance; + patchParam[patch].targetBandOffs = patchDistance; + patchParam[patch].numBandsInPatch = numBandsInPatch; + patchParam[patch].sourceStopBand = patchParam[patch].sourceStartBand + numBandsInPatch; + + targetStopBand += patchParam[patch].numBandsInPatch; + } + + /* All patches but first */ + sourceStartBand = hTonCorr->shiftStartSb; + + /* Check if we are close to goalSb */ + if( fixp_abs(targetStopBand - goalSb) < 3) { + goalSb = usb; + } + + patch++; + + } + + patch--; + + /* if highest patch contains less than three subband: skip it */ + if ( patchParam[patch].numBandsInPatch < 3 && patch > 0 ) { + patch--; + targetStopBand = patchParam[patch].targetStartBand + patchParam[patch].numBandsInPatch; + } + + hTonCorr->noOfPatches = patch + 1; + + + /* Assign the index-vector, so we know where to look for the high-band. + -1 represents a guard-band. */ + for(k = 0; k < hTonCorr->patchParam[0].guardStartBand; k++) + hTonCorr->indexVector[k] = k; + + for(i = 0; i < hTonCorr->noOfPatches; i++) + { + INT sourceStart = hTonCorr->patchParam[i].sourceStartBand; + INT targetStart = hTonCorr->patchParam[i].targetStartBand; + INT numberOfBands = hTonCorr->patchParam[i].numBandsInPatch; + INT startGuardBand = hTonCorr->patchParam[i].guardStartBand; + + for(k = 0; k < (targetStart- startGuardBand); k++) + hTonCorr->indexVector[startGuardBand+k] = -1; + + for(k = 0; k < numberOfBands; k++) + hTonCorr->indexVector[targetStart+k] = sourceStart+k; + } + + return (0); +} + +/**************************************************************************/ +/*! + \brief Creates an instance of the tonality correction parameter module. + + The module includes modules for inverse filtering level estimation, + missing harmonics detection and noise floor level estimation. + + \return errorCode, noError if successful. +*/ +/**************************************************************************/ +INT +FDKsbrEnc_CreateTonCorrParamExtr(HANDLE_SBR_TON_CORR_EST hTonCorr, /*!< Pointer to handle to SBR_TON_CORR struct. */ + INT chan) /*!< Channel index, needed for mem allocation */ +{ + INT i; + FIXP_DBL* quotaMatrix = GetRam_Sbr_quotaMatrix(chan); + INT* signMatrix = GetRam_Sbr_signMatrix(chan); + + FDKmemclear(hTonCorr, sizeof(SBR_TON_CORR_EST)); + + for (i=0; i<MAX_NO_OF_ESTIMATES; i++) { + hTonCorr->quotaMatrix[i] = quotaMatrix + (i*QMF_CHANNELS); + hTonCorr->signMatrix[i] = signMatrix + (i*QMF_CHANNELS); + } + + FDKsbrEnc_CreateSbrMissingHarmonicsDetector (&hTonCorr->sbrMissingHarmonicsDetector, chan); + + return 0; +} + + + +/**************************************************************************/ +/*! + \brief Initialize an instance of the tonality correction parameter module. + + The module includes modules for inverse filtering level estimation, + missing harmonics detection and noise floor level estimation. + + \return errorCode, noError if successful. +*/ +/**************************************************************************/ +INT +FDKsbrEnc_InitTonCorrParamExtr (INT frameSize, /*!< Current SBR frame size. */ + HANDLE_SBR_TON_CORR_EST hTonCorr, /*!< Pointer to handle to SBR_TON_CORR struct. */ + HANDLE_SBR_CONFIG_DATA sbrCfg, /*!< Pointer to SBR configuration parameters. */ + INT timeSlots, /*!< Number of time-slots per frame */ + INT xposCtrl, /*!< Different patch modes. */ + INT ana_max_level, /*!< Maximum level of the adaptive noise. */ + INT noiseBands, /*!< Number of noise bands per octave. */ + INT noiseFloorOffset, /*!< Noise floor offset. */ + UINT useSpeechConfig) /*!< Speech or music tuning. */ +{ + INT nCols = sbrCfg->noQmfSlots; + INT fs = sbrCfg->sampleFreq; + INT noQmfChannels = sbrCfg->noQmfBands; + + INT highBandStartSb = sbrCfg->freqBandTable[LOW_RES][0]; + UCHAR *v_k_master = sbrCfg->v_k_master; + INT numMaster = sbrCfg->num_Master; + + UCHAR **freqBandTable = sbrCfg->freqBandTable; + INT *nSfb = sbrCfg->nSfb; + + INT i; + + /* + Reset the patching and allocate memory for the quota matrix. + Assing parameters for the LPC analysis. + */ + if (sbrCfg->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) { + switch (timeSlots) { + case NUMBER_TIME_SLOTS_1920: + hTonCorr->lpcLength[0] = 8 - LPC_ORDER; + hTonCorr->lpcLength[1] = 7 - LPC_ORDER; + hTonCorr->numberOfEstimates = NO_OF_ESTIMATES_LD; + hTonCorr->numberOfEstimatesPerFrame = sbrCfg->noQmfSlots / 7; + hTonCorr->frameStartIndexInvfEst = 0; + hTonCorr->transientPosOffset = FRAME_MIDDLE_SLOT_512LD; + break; + case NUMBER_TIME_SLOTS_2048: + hTonCorr->lpcLength[0] = 8 - LPC_ORDER; + hTonCorr->lpcLength[1] = 8 - LPC_ORDER; + hTonCorr->numberOfEstimates = NO_OF_ESTIMATES_LD; + hTonCorr->numberOfEstimatesPerFrame = sbrCfg->noQmfSlots / 8; + hTonCorr->frameStartIndexInvfEst = 0; + hTonCorr->transientPosOffset = FRAME_MIDDLE_SLOT_512LD; + break; + } + } else + switch (timeSlots) { + case NUMBER_TIME_SLOTS_2048: + hTonCorr->lpcLength[0] = 16 - LPC_ORDER; /* blockLength[0] */ + hTonCorr->lpcLength[1] = 16 - LPC_ORDER; /* blockLength[0] */ + hTonCorr->numberOfEstimates = NO_OF_ESTIMATES_LC; + hTonCorr->numberOfEstimatesPerFrame = sbrCfg->noQmfSlots / 16; + hTonCorr->frameStartIndexInvfEst = 0; + hTonCorr->transientPosOffset = FRAME_MIDDLE_SLOT_2048; + break; + case NUMBER_TIME_SLOTS_1920: + hTonCorr->lpcLength[0] = 15 - LPC_ORDER; /* blockLength[0] */ + hTonCorr->lpcLength[1] = 15 - LPC_ORDER; /* blockLength[0] */ + hTonCorr->numberOfEstimates = NO_OF_ESTIMATES_LC; + hTonCorr->numberOfEstimatesPerFrame = sbrCfg->noQmfSlots / 15; + hTonCorr->frameStartIndexInvfEst = 0; + hTonCorr->transientPosOffset = FRAME_MIDDLE_SLOT_1920; + break; + default: + return -1; + } + + hTonCorr->bufferLength = nCols; + hTonCorr->stepSize = hTonCorr->lpcLength[0] + LPC_ORDER; /* stepSize[0] implicitly 0. */ + + hTonCorr->nextSample = LPC_ORDER; /* firstSample */ + hTonCorr->move = hTonCorr->numberOfEstimates - hTonCorr->numberOfEstimatesPerFrame; /* Number of estimates to move when buffering.*/ + hTonCorr->startIndexMatrix = hTonCorr->numberOfEstimates - hTonCorr->numberOfEstimatesPerFrame; /* Where to store the latest estimations in the tonality Matrix.*/ + hTonCorr->frameStartIndex = 0; /* Where in the tonality matrix the current frame (to be sent to the decoder) starts. */ + hTonCorr->prevTransientFlag = 0; + hTonCorr->transientNextFrame = 0; + + hTonCorr->noQmfChannels = noQmfChannels; + + for (i=0; i<hTonCorr->numberOfEstimates; i++) { + FDKmemclear (hTonCorr->quotaMatrix[i] , sizeof(FIXP_DBL)*noQmfChannels); + FDKmemclear (hTonCorr->signMatrix[i] , sizeof(INT)*noQmfChannels); + } + + /* Reset the patch.*/ + hTonCorr->guard = 0; + hTonCorr->shiftStartSb = 1; + + if(resetPatch(hTonCorr, + xposCtrl, + highBandStartSb, + v_k_master, + numMaster, + fs, + noQmfChannels)) + return(1); + + if(FDKsbrEnc_InitSbrNoiseFloorEstimate (&hTonCorr->sbrNoiseFloorEstimate, + ana_max_level, + freqBandTable[LO], + nSfb[LO], + noiseBands, + noiseFloorOffset, + timeSlots, + useSpeechConfig)) + return(1); + + + if(FDKsbrEnc_initInvFiltDetector(&hTonCorr->sbrInvFilt, + hTonCorr->sbrNoiseFloorEstimate.freqBandTableQmf, + hTonCorr->sbrNoiseFloorEstimate.noNoiseBands, + useSpeechConfig)) + return(1); + + + + if(FDKsbrEnc_InitSbrMissingHarmonicsDetector( + &hTonCorr->sbrMissingHarmonicsDetector, + fs, + frameSize, + nSfb[HI], + noQmfChannels, + hTonCorr->numberOfEstimates, + hTonCorr->move, + hTonCorr->numberOfEstimatesPerFrame, + sbrCfg->sbrSyntaxFlags)) + return(1); + + + + return (0); +} + + + +/**************************************************************************/ +/*! + \brief resets tonality correction parameter module. + + + + \return errorCode, noError if successful. + +*/ +/**************************************************************************/ +INT +FDKsbrEnc_ResetTonCorrParamExtr(HANDLE_SBR_TON_CORR_EST hTonCorr, /*!< Handle to SBR_TON_CORR struct. */ + INT xposctrl, /*!< Different patch modes. */ + INT highBandStartSb, /*!< Start band of the SBR range. */ + UCHAR *v_k_master, /*!< Master frequency table from which all other table are derived.*/ + INT numMaster, /*!< Number of elements in the master table. */ + INT fs, /*!< Sampling frequency (of the SBR part). */ + UCHAR ** freqBandTable, /*!< Frequency band table for low-res and high-res. */ + INT* nSfb, /*!< Number of frequency bands (hig-res and low-res). */ + INT noQmfChannels /*!< Number of QMF channels. */ + ) +{ + + /* Reset the patch.*/ + hTonCorr->guard = 0; + hTonCorr->shiftStartSb = 1; + + if(resetPatch(hTonCorr, + xposctrl, + highBandStartSb, + v_k_master, + numMaster, + fs, + noQmfChannels)) + return(1); + + + + /* Reset the noise floor estimate.*/ + if(FDKsbrEnc_resetSbrNoiseFloorEstimate (&hTonCorr->sbrNoiseFloorEstimate, + freqBandTable[LO], + nSfb[LO])) + return(1); + + /* + Reset the inveerse filtereing detector. + */ + if(FDKsbrEnc_resetInvFiltDetector(&hTonCorr->sbrInvFilt, + hTonCorr->sbrNoiseFloorEstimate.freqBandTableQmf, + hTonCorr->sbrNoiseFloorEstimate.noNoiseBands)) + return(1); +/* Reset the missing harmonics detector. */ + if(FDKsbrEnc_ResetSbrMissingHarmonicsDetector (&hTonCorr->sbrMissingHarmonicsDetector, + nSfb[HI])) + return(1); + + return (0); +} + + + + + +/**************************************************************************/ +/*! + \brief Deletes the tonality correction paramtere module. + + + + \return none + +*/ +/**************************************************************************/ +void +FDKsbrEnc_DeleteTonCorrParamExtr (HANDLE_SBR_TON_CORR_EST hTonCorr) /*!< Handle to SBR_TON_CORR struct. */ +{ + + if (hTonCorr) { + + FreeRam_Sbr_quotaMatrix(hTonCorr->quotaMatrix); + + FreeRam_Sbr_signMatrix(hTonCorr->signMatrix); + + FDKsbrEnc_DeleteSbrMissingHarmonicsDetector (&hTonCorr->sbrMissingHarmonicsDetector); + } +} |