diff options
Diffstat (limited to 'libSBRenc/src/ton_corr.cpp')
| -rw-r--r-- | libSBRenc/src/ton_corr.cpp | 881 |
1 files changed, 0 insertions, 881 deletions
diff --git a/libSBRenc/src/ton_corr.cpp b/libSBRenc/src/ton_corr.cpp deleted file mode 100644 index 224da11..0000000 --- a/libSBRenc/src/ton_corr.cpp +++ /dev/null @@ -1,881 +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 ------------------------------------------------------------------------------------------------------------ */ - -#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++; - } - } - - - 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); - } -} |