/* ----------------------------------------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2015 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 ----------------------------------------------------------------------------------------------------------- */ /******************************** MPEG Audio Encoder ************************** Initial author: Alex Groeschel, Tobias Chalupka contents/description: Temporal noise shaping ******************************************************************************/ #include "aacenc_tns.h" #include "psy_const.h" #include "psy_configuration.h" #include "tns_func.h" #include "aacEnc_rom.h" #include "aacenc_tns.h" #define FILTER_DIRECTION 0 /* 0 = up, 1 = down */ static const FIXP_DBL acfWindowLong[12+3+1] = { 0x7fffffff,0x7fb80000,0x7ee00000,0x7d780000,0x7b800000,0x78f80000,0x75e00000,0x72380000, 0x6e000000,0x69380000,0x63e00000,0x5df80000,0x57800000,0x50780000,0x48e00000,0x40b80000 }; static const FIXP_DBL acfWindowShort[4+3+1] = { 0x7fffffff,0x7e000000,0x78000000,0x6e000000,0x60000000,0x4e000000,0x38000000,0x1e000000 }; typedef struct{ INT bitRateFrom[2]; /* noneSbr=0, useSbr=1 */ INT bitRateTo[2]; /* noneSbr=0, useSbr=1 */ TNS_PARAMETER_TABULATED paramTab[2]; /* mono=0, stereo=1 */ } TNS_INFO_TAB; #define TNS_TIMERES_SCALE (1) #define FL2_TIMERES_FIX(a) ( FL2FXCONST_DBL(a/(float)(1<= tnsInfoTab[i].bitRateFrom[sbrLd?1:0]) && bitRate <= tnsInfoTab[i].bitRateTo[sbrLd?1:0]) { tnsConfigTab = &tnsInfoTab[i].paramTab[(channels==1)?0:1]; } } return tnsConfigTab; } static INT getTnsMaxBands( const INT sampleRate, const INT granuleLength, const INT isShortBlock ) { int i; INT numBands = -1; const TNS_MAX_TAB_ENTRY *pMaxBandsTab = NULL; int maxBandsTabSize = 0; switch (granuleLength) { case 960: pMaxBandsTab = tnsMaxBandsTab960; maxBandsTabSize = sizeof(tnsMaxBandsTab960)/sizeof(TNS_MAX_TAB_ENTRY); break; case 1024: pMaxBandsTab = tnsMaxBandsTab1024; maxBandsTabSize = sizeof(tnsMaxBandsTab1024)/sizeof(TNS_MAX_TAB_ENTRY); break; case 480: pMaxBandsTab = tnsMaxBandsTab480; maxBandsTabSize = sizeof(tnsMaxBandsTab480)/sizeof(TNS_MAX_TAB_ENTRY); break; case 512: pMaxBandsTab = tnsMaxBandsTab512; maxBandsTabSize = sizeof(tnsMaxBandsTab512)/sizeof(TNS_MAX_TAB_ENTRY); break; default: numBands = -1; } if (pMaxBandsTab!=NULL) { for (i=0; i= pMaxBandsTab[i].samplingRate) { break; } } } return numBands; } /***************************************************************************/ /*! \brief FDKaacEnc_FreqToBandWithRounding Returns index of nearest band border \param frequency \param sampling frequency \param total number of bands \param pointer to table of band borders \return band border ****************************************************************************/ INT FDKaacEnc_FreqToBandWithRounding( const INT freq, const INT fs, const INT numOfBands, const INT *bandStartOffset ) { INT lineNumber, band; /* assert(freq >= 0); */ lineNumber = (freq*bandStartOffset[numOfBands]*4/fs+1)/2; /* freq > fs/2 */ if (lineNumber >= bandStartOffset[numOfBands]) return numOfBands; /* find band the line number lies in */ for (band=0; bandlineNumber) break; } /* round to nearest band border */ if (lineNumber - bandStartOffset[band] > bandStartOffset[band+1] - lineNumber ) { band++; } return(band); } /***************************************************************************** functionname: FDKaacEnc_InitTnsConfiguration description: fill TNS_CONFIG structure with sensible content returns: input: bitrate, samplerate, number of channels, blocktype (long or short), TNS Config struct (modified), psy config struct, tns active flag output: *****************************************************************************/ AAC_ENCODER_ERROR FDKaacEnc_InitTnsConfiguration(INT bitRate, INT sampleRate, INT channels, INT blockType, INT granuleLength, INT isLowDelay, INT ldSbrPresent, TNS_CONFIG *tC, PSY_CONFIGURATION *pC, INT active, INT useTnsPeak) { int i; //float acfTimeRes = (blockType == SHORT_WINDOW) ? 0.125f : 0.046875f; if (channels <= 0) return (AAC_ENCODER_ERROR)1; tC->isLowDelay = isLowDelay; /* initialize TNS filter flag, order, and coefficient resolution (in bits per coeff) */ tC->tnsActive = (active) ? TRUE : FALSE; tC->maxOrder = (blockType == SHORT_WINDOW) ? 5 : 12; /* maximum: 7, 20 */ if (bitRate < 16000) tC->maxOrder -= 2; tC->coefRes = (blockType == SHORT_WINDOW) ? 3 : 4; /* LPC stop line: highest MDCT line to be coded, but do not go beyond TNS_MAX_BANDS! */ tC->lpcStopBand = getTnsMaxBands(sampleRate, granuleLength, (blockType == SHORT_WINDOW) ? 1 : 0); if (tC->lpcStopBand < 0) { return (AAC_ENCODER_ERROR)1; } tC->lpcStopBand = FDKmin(tC->lpcStopBand, pC->sfbActive); tC->lpcStopLine = pC->sfbOffset[tC->lpcStopBand]; switch (granuleLength) { case 960: case 1024: /* TNS start line: skip lower MDCT lines to prevent artifacts due to filter mismatch */ tC->lpcStartBand[LOFILT] = (blockType == SHORT_WINDOW) ? 0 : ((sampleRate <= 8000) ? 2 : ((sampleRate < 18783) ? 4 : 8)); tC->lpcStartLine[LOFILT] = pC->sfbOffset[tC->lpcStartBand[LOFILT]]; i = tC->lpcStopBand; while (pC->sfbOffset[i] > (tC->lpcStartLine[LOFILT] + (tC->lpcStopLine - tC->lpcStartLine[LOFILT]) / 4)) i--; tC->lpcStartBand[HIFILT] = i; tC->lpcStartLine[HIFILT] = pC->sfbOffset[i]; tC->confTab.threshOn[HIFILT] = 1437; tC->confTab.threshOn[LOFILT] = 1500; tC->confTab.tnsLimitOrder[HIFILT] = tC->maxOrder; tC->confTab.tnsLimitOrder[LOFILT] = tC->maxOrder - 7; tC->confTab.tnsFilterDirection[HIFILT] = FILTER_DIRECTION; tC->confTab.tnsFilterDirection[LOFILT] = FILTER_DIRECTION; tC->confTab.acfSplit[HIFILT] = -1; /* signal Merged4to2QuartersAutoCorrelation in FDKaacEnc_MergedAutoCorrelation*/ tC->confTab.acfSplit[LOFILT] = -1; /* signal Merged4to2QuartersAutoCorrelation in FDKaacEnc_MergedAutoCorrelation */ tC->confTab.filterEnabled[HIFILT] = 1; tC->confTab.filterEnabled[LOFILT] = 1; tC->confTab.seperateFiltersAllowed = 1; /* compute autocorrelation window based on maximum filter order for given block type */ /* for (i = 0; i <= tC->maxOrder + 3; i++) { float acfWinTemp = acfTimeRes * i; acfWindow[i] = FL2FXCONST_DBL(1.0f - acfWinTemp * acfWinTemp); } */ if (blockType == SHORT_WINDOW) { FDKmemcpy(tC->acfWindow[HIFILT], acfWindowShort, FDKmin(sizeof(acfWindowShort), sizeof(tC->acfWindow[HIFILT]))); FDKmemcpy(tC->acfWindow[LOFILT], acfWindowShort, FDKmin(sizeof(acfWindowShort), sizeof(tC->acfWindow[HIFILT]))); } else { FDKmemcpy(tC->acfWindow[HIFILT], acfWindowLong, FDKmin(sizeof(acfWindowLong), sizeof(tC->acfWindow[HIFILT]))); FDKmemcpy(tC->acfWindow[LOFILT], acfWindowLong, FDKmin(sizeof(acfWindowLong), sizeof(tC->acfWindow[HIFILT]))); } break; case 480: case 512: { const TNS_PARAMETER_TABULATED* pCfg = FDKaacEnc_GetTnsParam(bitRate, channels, ldSbrPresent); if ( pCfg != NULL ) { FDKmemcpy(&(tC->confTab), pCfg, sizeof(tC->confTab)); tC->lpcStartBand[HIFILT] = FDKaacEnc_FreqToBandWithRounding(pCfg->filterStartFreq[HIFILT], sampleRate, pC->sfbCnt, pC->sfbOffset); tC->lpcStartLine[HIFILT] = pC->sfbOffset[tC->lpcStartBand[HIFILT]]; tC->lpcStartBand[LOFILT] = FDKaacEnc_FreqToBandWithRounding(pCfg->filterStartFreq[LOFILT], sampleRate, pC->sfbCnt, pC->sfbOffset); tC->lpcStartLine[LOFILT] = pC->sfbOffset[tC->lpcStartBand[LOFILT]]; FDKaacEnc_CalcGaussWindow(tC->acfWindow[HIFILT], tC->maxOrder+1, sampleRate, granuleLength, pCfg->tnsTimeResolution[HIFILT], TNS_TIMERES_SCALE); FDKaacEnc_CalcGaussWindow(tC->acfWindow[LOFILT], tC->maxOrder+1, sampleRate, granuleLength, pCfg->tnsTimeResolution[LOFILT], TNS_TIMERES_SCALE); } else { tC->tnsActive = FALSE; /* no configuration available, disable tns tool */ } } break; default: tC->tnsActive = FALSE; /* no configuration available, disable tns tool */ } return AAC_ENC_OK; } /***************************************************************************/ /*! \brief FDKaacEnc_ScaleUpSpectrum Scales up spectrum lines in a given frequency section \param scaled spectrum \param original spectrum \param frequency line to start scaling \param frequency line to enc scaling \return scale factor ****************************************************************************/ static inline INT FDKaacEnc_ScaleUpSpectrum( FIXP_DBL *dest, const FIXP_DBL *src, const INT startLine, const INT stopLine ) { INT i, scale; FIXP_DBL maxVal = FL2FXCONST_DBL(0.f); /* Get highest value in given spectrum */ for (i=startLine; i>scale); } } else { for (i=startLine; i<(stopLine-lag); i++) { result += (fMult(spectrum[i], spectrum[i+lag])>>scale); } } return result; } /***************************************************************************/ /*! \brief FDKaacEnc_AutoCorrNormFac Autocorrelation function for 1st and 2nd half of the spectrum \param pointer to spectrum \param pointer to autocorrelation window \param filter start line ****************************************************************************/ static inline FIXP_DBL FDKaacEnc_AutoCorrNormFac( const FIXP_DBL value, const INT scale, INT *sc ) { #define HLM_MIN_NRG 0.0000000037252902984619140625f /* 2^-28 */ #define MAX_INV_NRGFAC (1.f/HLM_MIN_NRG) FIXP_DBL retValue; FIXP_DBL A, B; if (scale>=0) { A = value; B = FL2FXCONST_DBL(HLM_MIN_NRG)>>fixMin(DFRACT_BITS-1,scale); } else { A = value>>fixMin(DFRACT_BITS-1,(-scale)); B = FL2FXCONST_DBL(HLM_MIN_NRG); } if (A > B) { int shift = 0; FIXP_DBL tmp = invSqrtNorm2(value,&shift); retValue = fMult(tmp,tmp); *sc += (2*shift); } else { /* MAX_INV_NRGFAC*FDKpow(2,-28) = 1/2^-28 * 2^-28 = 1.0 */ retValue = /*FL2FXCONST_DBL(MAX_INV_NRGFAC*FDKpow(2,-28))*/ (FIXP_DBL)MAXVAL_DBL; *sc += scale+28; } return retValue; } static void FDKaacEnc_MergedAutoCorrelation( const FIXP_DBL *spectrum, const INT isLowDelay, const FIXP_DBL acfWindow[MAX_NUM_OF_FILTERS][TNS_MAX_ORDER+3+1], const INT lpcStartLine[MAX_NUM_OF_FILTERS], const INT lpcStopLine, const INT maxOrder, const INT acfSplit[MAX_NUM_OF_FILTERS], FIXP_DBL *_rxx1, FIXP_DBL *_rxx2 ) { int i, idx0, idx1, idx2, idx3, idx4, lag; FIXP_DBL rxx1_0, rxx2_0, rxx3_0, rxx4_0; /* buffer for temporal spectrum */ C_ALLOC_SCRATCH_START(pSpectrum, FIXP_DBL, (1024)); /* pre-initialization output */ FDKmemclear(&_rxx1[0], sizeof(FIXP_DBL)*(maxOrder+1)); FDKmemclear(&_rxx2[0], sizeof(FIXP_DBL)*(maxOrder+1)); idx0 = idx1 = idx2 = idx3 = idx4 = 0; /* MDCT line indices separating the 1st, 2nd, 3rd, and 4th analysis quarters */ if ( (acfSplit[LOFILT]==-1) || (acfSplit[HIFILT]==-1) ) { /* autocorrelation function for 1st, 2nd, 3rd, and 4th quarter of the spectrum */ idx0 = lpcStartLine[LOFILT]; i = lpcStopLine - lpcStartLine[LOFILT]; idx1 = idx0 + i / 4; idx2 = idx0 + i / 2; idx3 = idx0 + i * 3 / 4; idx4 = lpcStopLine; } else { FDK_ASSERT(acfSplit[LOFILT]==1); FDK_ASSERT(acfSplit[HIFILT]==3); i = (lpcStopLine - lpcStartLine[HIFILT]) / 3; idx0 = lpcStartLine[LOFILT]; idx1 = lpcStartLine[HIFILT]; idx2 = idx1 + i; idx3 = idx2 + i; idx4 = lpcStopLine; } /* copy spectrum to temporal buffer and scale up as much as possible */ INT sc1 = FDKaacEnc_ScaleUpSpectrum(pSpectrum, spectrum, idx0, idx1); INT sc2 = FDKaacEnc_ScaleUpSpectrum(pSpectrum, spectrum, idx1, idx2); INT sc3 = FDKaacEnc_ScaleUpSpectrum(pSpectrum, spectrum, idx2, idx3); INT sc4 = FDKaacEnc_ScaleUpSpectrum(pSpectrum, spectrum, idx3, idx4); /* get scaling values for summation */ INT nsc1, nsc2, nsc3, nsc4; for (nsc1=1; (1<dataRaw.Short.subBlockInfo[subBlockNumber] : &tnsData->dataRaw.Long.subBlockInfo; tnsData->filtersMerged = FALSE; tsbi->tnsActive[HIFILT] = FALSE; tsbi->predictionGain[HIFILT] = 1000; tsbi->tnsActive[LOFILT] = FALSE; tsbi->predictionGain[LOFILT] = 1000; tnsInfo->numOfFilters[subBlockNumber] = 0; tnsInfo->coefRes[subBlockNumber] = tC->coefRes; for (i = 0; i < tC->maxOrder; i++) { tnsInfo->coef[subBlockNumber][HIFILT][i] = tnsInfo->coef[subBlockNumber][LOFILT][i] = 0; } tnsInfo->length[subBlockNumber][HIFILT] = tnsInfo->length[subBlockNumber][LOFILT] = 0; tnsInfo->order [subBlockNumber][HIFILT] = tnsInfo->order [subBlockNumber][LOFILT] = 0; if ( (tC->tnsActive) && (tC->maxOrder>0) ) { int sumSqrCoef; FDKaacEnc_MergedAutoCorrelation( spectrum, tC->isLowDelay, tC->acfWindow, tC->lpcStartLine, tC->lpcStopLine, tC->maxOrder, tC->confTab.acfSplit, rxx1, rxx2); /* compute higher TNS filter in lattice (ParCor) form with LeRoux-Gueguen algorithm */ tsbi->predictionGain[HIFILT] = FDKaacEnc_AutoToParcor(rxx2, parcor_tmp, tC->confTab.tnsLimitOrder[HIFILT]); /* non-linear quantization of TNS lattice coefficients with given resolution */ FDKaacEnc_Parcor2Index( parcor_tmp, tnsInfo->coef[subBlockNumber][HIFILT], tC->confTab.tnsLimitOrder[HIFILT], tC->coefRes); /* reduce filter order by truncating trailing zeros, compute sum(abs(coefs)) */ for (i = tC->confTab.tnsLimitOrder[HIFILT] - 1; i >= 0; i--) { if (tnsInfo->coef[subBlockNumber][HIFILT][i] != 0) { break; } } tnsInfo->order[subBlockNumber][HIFILT] = i + 1; sumSqrCoef = 0; for (; i >= 0; i--) { sumSqrCoef += tnsInfo->coef[subBlockNumber][HIFILT][i] * tnsInfo->coef[subBlockNumber][HIFILT][i]; } tnsInfo->direction[subBlockNumber][HIFILT] = tC->confTab.tnsFilterDirection[HIFILT]; tnsInfo->length[subBlockNumber][HIFILT] = sfbCnt - tC->lpcStartBand[HIFILT]; /* disable TNS if predictionGain is less than 3dB or sumSqrCoef is too small */ if ((tsbi->predictionGain[HIFILT] > tC->confTab.threshOn[HIFILT]) || (sumSqrCoef > (tC->confTab.tnsLimitOrder[HIFILT]/2 + 2))) { tsbi->tnsActive[HIFILT] = TRUE; tnsInfo->numOfFilters[subBlockNumber]++; /* compute second filter for lower quarter; only allowed for long windows! */ if ( (blockType != SHORT_WINDOW) && (tC->confTab.filterEnabled[LOFILT]) && (tC->confTab.seperateFiltersAllowed) ) { /* compute second filter for lower frequencies */ /* compute TNS filter in lattice (ParCor) form with LeRoux-Gueguen algorithm */ INT predGain = FDKaacEnc_AutoToParcor(rxx1, parcor_tmp, tC->confTab.tnsLimitOrder[LOFILT]); /* non-linear quantization of TNS lattice coefficients with given resolution */ FDKaacEnc_Parcor2Index( parcor_tmp, tnsInfo->coef[subBlockNumber][LOFILT], tC->confTab.tnsLimitOrder[LOFILT], tC->coefRes); /* reduce filter order by truncating trailing zeros, compute sum(abs(coefs)) */ for (i = tC->confTab.tnsLimitOrder[LOFILT] - 1; i >= 0; i--) { if (tnsInfo->coef[subBlockNumber][LOFILT][i] != 0) { break; } } tnsInfo->order[subBlockNumber][LOFILT] = i + 1; sumSqrCoef = 0; for (; i >= 0; i--) { sumSqrCoef += tnsInfo->coef[subBlockNumber][LOFILT][i] * tnsInfo->coef[subBlockNumber][LOFILT][i]; } tnsInfo->direction[subBlockNumber][LOFILT] = tC->confTab.tnsFilterDirection[LOFILT]; tnsInfo->length[subBlockNumber][LOFILT] = tC->lpcStartBand[HIFILT] - tC->lpcStartBand[LOFILT]; /* filter lower quarter if gain is high enough, but not if it's too high */ if ( ( (predGain > tC->confTab.threshOn[LOFILT]) && (predGain < (16000 * tC->confTab.tnsLimitOrder[LOFILT])) ) || ( (sumSqrCoef > 9) && (sumSqrCoef < 22 * tC->confTab.tnsLimitOrder[LOFILT]) ) ) { /* compare lower to upper filter; if they are very similar, merge them */ tsbi->tnsActive[LOFILT] = TRUE; sumSqrCoef = 0; for (i = 0; i < tC->confTab.tnsLimitOrder[LOFILT]; i++) { sumSqrCoef += FDKabs(tnsInfo->coef[subBlockNumber][HIFILT][i] - tnsInfo->coef[subBlockNumber][LOFILT][i]); } if ( (sumSqrCoef < 2) && (tnsInfo->direction[subBlockNumber][LOFILT] == tnsInfo->direction[subBlockNumber][HIFILT]) ) { tnsData->filtersMerged = TRUE; tnsInfo->length[subBlockNumber][HIFILT] = sfbCnt - tC->lpcStartBand[LOFILT]; for (; i < tnsInfo->order[subBlockNumber][HIFILT]; i++) { if (FDKabs(tnsInfo->coef[subBlockNumber][HIFILT][i]) > 1) { break; } } for (i--; i >= 0; i--) { if (tnsInfo->coef[subBlockNumber][HIFILT][i] != 0) { break; } } if (i < tnsInfo->order[subBlockNumber][HIFILT]) { tnsInfo->order[subBlockNumber][HIFILT] = i + 1; } } else { tnsInfo->numOfFilters[subBlockNumber]++; } } /* filter lower part */ tsbi->predictionGain[LOFILT]=predGain; } /* second filter allowed */ } /* if predictionGain > 1437 ... */ } /* maxOrder > 0 && tnsActive */ return 0; } /***************************************************************************/ /*! \brief FDKaacLdEnc_TnsSync synchronize TNS parameters when TNS gain difference small (relative) \param pointer to TNS data structure (destination) \param pointer to TNS data structure (source) \param pointer to TNS config structure \param number of sub-block \param block type \return void ****************************************************************************/ void FDKaacEnc_TnsSync( TNS_DATA *tnsDataDest, const TNS_DATA *tnsDataSrc, TNS_INFO *tnsInfoDest, TNS_INFO *tnsInfoSrc, const INT blockTypeDest, const INT blockTypeSrc, const TNS_CONFIG *tC ) { int i, w, absDiff, nWindows; TNS_SUBBLOCK_INFO *sbInfoDest; const TNS_SUBBLOCK_INFO *sbInfoSrc; /* if one channel contains short blocks and the other not, do not synchronize */ if ( (blockTypeSrc == SHORT_WINDOW && blockTypeDest != SHORT_WINDOW) || (blockTypeDest == SHORT_WINDOW && blockTypeSrc != SHORT_WINDOW) ) { return; } if (blockTypeDest != SHORT_WINDOW) { sbInfoDest = &tnsDataDest->dataRaw.Long.subBlockInfo; sbInfoSrc = &tnsDataSrc->dataRaw.Long.subBlockInfo; nWindows = 1; } else { sbInfoDest = &tnsDataDest->dataRaw.Short.subBlockInfo[0]; sbInfoSrc = &tnsDataSrc->dataRaw.Short.subBlockInfo[0]; nWindows = 8; } for (w=0; wtnsActive[HIFILT] || pSbInfoSrcW->tnsActive[HIFILT]) { for (i = 0; i < tC->maxOrder; i++) { absDiff = FDKabs(tnsInfoDest->coef[w][HIFILT][i] - tnsInfoSrc->coef[w][HIFILT][i]); absDiffSum += absDiff; /* if coefficients diverge too much between channels, do not synchronize */ if ((absDiff > 1) || (absDiffSum > 2)) { doSync = 0; break; } } if (doSync) { /* if no significant difference was detected, synchronize coefficient sets */ if (pSbInfoSrcW->tnsActive[HIFILT]) { /* no dest filter, or more dest than source filters: use one dest filter */ if ((!pSbInfoDestW->tnsActive[HIFILT]) || ((pSbInfoDestW->tnsActive[HIFILT]) && (tnsInfoDest->numOfFilters[w] > tnsInfoSrc->numOfFilters[w]))) { pSbInfoDestW->tnsActive[HIFILT] = tnsInfoDest->numOfFilters[w] = 1; } tnsDataDest->filtersMerged = tnsDataSrc->filtersMerged; tnsInfoDest->order [w][HIFILT] = tnsInfoSrc->order [w][HIFILT]; tnsInfoDest->length [w][HIFILT] = tnsInfoSrc->length [w][HIFILT]; tnsInfoDest->direction [w][HIFILT] = tnsInfoSrc->direction [w][HIFILT]; tnsInfoDest->coefCompress[w][HIFILT] = tnsInfoSrc->coefCompress[w][HIFILT]; for (i = 0; i < tC->maxOrder; i++) { tnsInfoDest->coef[w][HIFILT][i] = tnsInfoSrc->coef[w][HIFILT][i]; } } else pSbInfoDestW->tnsActive[HIFILT] = tnsInfoDest->numOfFilters[w] = 0; } } } } /***************************************************************************/ /*! \brief FDKaacEnc_TnsEncode perform TNS encoding \param pointer to TNS info structure \param pointer to TNS data structure \param number of sfbs \param pointer to TNS config structure \param low-pass line \param pointer to spectrum \param number of sub-block \param block type \return ERROR STATUS ****************************************************************************/ INT FDKaacEnc_TnsEncode( TNS_INFO* tnsInfo, TNS_DATA* tnsData, const INT numOfSfb, const TNS_CONFIG *tC, const INT lowPassLine, FIXP_DBL* spectrum, const INT subBlockNumber, const INT blockType ) { INT i, startLine, stopLine; if ( ( (blockType == SHORT_WINDOW) && (!tnsData->dataRaw.Short.subBlockInfo[subBlockNumber].tnsActive[HIFILT]) ) || ( (blockType != SHORT_WINDOW) && (!tnsData->dataRaw.Long.subBlockInfo.tnsActive[HIFILT]) ) ) { return 1; } startLine = (tnsData->filtersMerged) ? tC->lpcStartLine[LOFILT] : tC->lpcStartLine[HIFILT]; stopLine = tC->lpcStopLine; for (i=0; inumOfFilters[subBlockNumber]; i++) { INT lpcGainFactor; FIXP_DBL LpcCoeff[TNS_MAX_ORDER]; FIXP_DBL workBuffer[TNS_MAX_ORDER]; FIXP_DBL parcor_tmp[TNS_MAX_ORDER]; FDKaacEnc_Index2Parcor( tnsInfo->coef[subBlockNumber][i], parcor_tmp, tnsInfo->order[subBlockNumber][i], tC->coefRes); lpcGainFactor = FDKaacEnc_ParcorToLpc( parcor_tmp, LpcCoeff, tnsInfo->order[subBlockNumber][i], workBuffer); FDKaacEnc_AnalysisFilter( &spectrum[startLine], stopLine - startLine, LpcCoeff, tnsInfo->order[subBlockNumber][i], lpcGainFactor); /* update for second filter */ startLine = tC->lpcStartLine[LOFILT]; stopLine = tC->lpcStartLine[HIFILT]; } return(0); } static void FDKaacEnc_CalcGaussWindow( FIXP_DBL *win, const int winSize, const INT samplingRate, const INT transformResolution, const FIXP_DBL timeResolution, const INT timeResolution_e ) { #define PI_E (2) #define PI_M FL2FXCONST_DBL(3.1416f/(float)(1<> (DFRACT_BITS-1)); tmp = (FIXP_DBL)((LONG)workBuffer[0]^sign); if(input[0]=0; j--) { FIXP_DBL accu1 = fMult(tmp, input[j]); FIXP_DBL accu2 = fMult(tmp, workBuffer[j]); workBuffer[j] += accu1; input[j] += accu2; } workBuffer++; } if (input[0] == 0) input[0] = 1; tmp = fMult((FIXP_DBL)((LONG)TNS_PREDGAIN_SCALE<<21), fDivNorm(fAbs(autoCorr_0), fAbs(input[0]), &scale)); if ( fMultDiv2(autoCorr_0, input[0]) FDKaacEnc_tnsCoeff3Borders[i]) index=i; } return(index-4); } static INT FDKaacEnc_Search4(FIXP_DBL parcor) { INT i, index=0; for(i=0;i<16;i++){ if(parcor > FDKaacEnc_tnsCoeff4Borders[i]) index=i; } return(index-8); } /***************************************************************************** functionname: FDKaacEnc_Parcor2Index *****************************************************************************/ static void FDKaacEnc_Parcor2Index( const FIXP_DBL *parcor, INT *RESTRICT index, const INT order, const INT bitsPerCoeff ) { INT i; for(i=0; i, ptr. to work buffer (required size: order) output: LPC coefficients *****************************************************************************/ static INT FDKaacEnc_ParcorToLpc( const FIXP_DBL *reflCoeff, FIXP_DBL *RESTRICT LpcCoeff, const INT numOfCoeff, FIXP_DBL *RESTRICT workBuffer ) { INT i, j; INT shiftval, par2LpcShiftVal = 6; /* 6 should be enough, bec. max(numOfCoeff) = 20 */ FIXP_DBL maxVal = FL2FXCONST_DBL(0.0f); LpcCoeff[0] = reflCoeff[0] >> par2LpcShiftVal; for(i=1; i> par2LpcShiftVal; } /* normalize LpcCoeff and calc shiftfactor */ for(i=0; i=par2LpcShiftVal) ? par2LpcShiftVal : shiftval; for(i=0; i0) { INT idx = 0; /* keep filter coefficients twice and save memory copy operation in modulo state buffer */ #if defined(ARCH_PREFER_MULT_32x16) FIXP_SGL coeff[2*TNS_MAX_ORDER]; const FIXP_SGL *pCoeff; for(i=0;i=0); signal[j] = (tmp<