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diff --git a/libSBRenc/src/nf_est.cpp b/libSBRenc/src/nf_est.cpp new file mode 100644 index 0000000..62bcc79 --- /dev/null +++ b/libSBRenc/src/nf_est.cpp @@ -0,0 +1,575 @@ + +/* ----------------------------------------------------------------------------------------------------------- +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 "nf_est.h" + +#include "sbr_misc.h" + +#include "genericStds.h" + +/* smoothFilter[4] = {0.05857864376269f, 0.2f, 0.34142135623731f, 0.4f}; */ +static const FIXP_DBL smoothFilter[4] = { 0x077f813d, 0x19999995, 0x2bb3b1f5, 0x33333335 }; + +/* static const INT smoothFilterLength = 4; */ + +static const FIXP_DBL QuantOffset = (INT)0xfc000000; /* ld64(0.25) */ + +#ifndef min +#define min(a,b) ( a < b ? a:b) +#endif + +#ifndef max +#define max(a,b) ( a > b ? a:b) +#endif + +#define NOISE_FLOOR_OFFSET_SCALING (3) + + + +/**************************************************************************/ +/*! + \brief The function applies smoothing to the noise levels. + + + + \return none + +*/ +/**************************************************************************/ +static void +smoothingOfNoiseLevels(FIXP_DBL *NoiseLevels, /*!< pointer to noise-floor levels.*/ + INT nEnvelopes, /*!< Number of noise floor envelopes.*/ + INT noNoiseBands, /*!< Number of noise bands for every noise floor envelope. */ + FIXP_DBL prevNoiseLevels[NF_SMOOTHING_LENGTH][MAX_NUM_NOISE_VALUES],/*!< Previous noise floor envelopes. */ + const FIXP_DBL *smoothFilter, /*!< filter used for smoothing the noise floor levels. */ + INT transientFlag) /*!< flag indicating if a transient is present*/ + +{ + INT i,band,env; + FIXP_DBL accu; + + for(env = 0; env < nEnvelopes; env++){ + if(transientFlag){ + for (i = 0; i < NF_SMOOTHING_LENGTH; i++){ + FDKmemcpy(prevNoiseLevels[i],NoiseLevels+env*noNoiseBands,noNoiseBands*sizeof(FIXP_DBL)); + } + } + else { + for (i = 1; i < NF_SMOOTHING_LENGTH; i++){ + FDKmemcpy(prevNoiseLevels[i - 1],prevNoiseLevels[i],noNoiseBands*sizeof(FIXP_DBL)); + } + FDKmemcpy(prevNoiseLevels[NF_SMOOTHING_LENGTH - 1],NoiseLevels+env*noNoiseBands,noNoiseBands*sizeof(FIXP_DBL)); + } + + for (band = 0; band < noNoiseBands; band++){ + accu = FL2FXCONST_DBL(0.0f); + for (i = 0; i < NF_SMOOTHING_LENGTH; i++){ + accu += fMultDiv2(smoothFilter[i], prevNoiseLevels[i][band]); + } + FDK_ASSERT( (band + env*noNoiseBands) < MAX_NUM_NOISE_VALUES); + NoiseLevels[band+ env*noNoiseBands] = accu<<1; + } + } +} + +/**************************************************************************/ +/*! + \brief Does the noise floor level estiamtion. + + The noiseLevel samples are scaled by the factor 0.25 + + \return none + +*/ +/**************************************************************************/ +static void +qmfBasedNoiseFloorDetection(FIXP_DBL *noiseLevel, /*!< Pointer to vector to store the noise levels in.*/ + FIXP_DBL ** quotaMatrixOrig, /*!< Matrix holding the quota values of the original. */ + SCHAR *indexVector, /*!< Index vector to obtain the patched data. */ + INT startIndex, /*!< Start index. */ + INT stopIndex, /*!< Stop index. */ + INT startChannel, /*!< Start channel of the current noise floor band.*/ + INT stopChannel, /*!< Stop channel of the current noise floor band. */ + FIXP_DBL ana_max_level, /*!< Maximum level of the adaptive noise.*/ + FIXP_DBL noiseFloorOffset, /*!< Noise floor offset. */ + INT missingHarmonicFlag, /*!< Flag indicating if a strong tonal component is missing.*/ + FIXP_DBL weightFac, /*!< Weightening factor for the difference between orig and sbr. */ + INVF_MODE diffThres, /*!< Threshold value to control the inverse filtering decision.*/ + INVF_MODE inverseFilteringLevel) /*!< Inverse filtering level of the current band.*/ +{ + INT scale, l, k; + FIXP_DBL meanOrig=FL2FXCONST_DBL(0.0f), meanSbr=FL2FXCONST_DBL(0.0f), diff; + FIXP_DBL invIndex = GetInvInt(stopIndex-startIndex); + FIXP_DBL invChannel = GetInvInt(stopChannel-startChannel); + FIXP_DBL accu; + + /* + Calculate the mean value, over the current time segment, for the original, the HFR + and the difference, over all channels in the current frequency range. + */ + + if(missingHarmonicFlag == 1){ + for(l = startChannel; l < stopChannel;l++){ + /* tonalityOrig */ + accu = FL2FXCONST_DBL(0.0f); + for(k = startIndex ; k < stopIndex; k++){ + accu += fMultDiv2(quotaMatrixOrig[k][l], invIndex); + } + meanOrig = fixMax(meanOrig,(accu<<1)); + + /* tonalitySbr */ + accu = FL2FXCONST_DBL(0.0f); + for(k = startIndex ; k < stopIndex; k++){ + accu += fMultDiv2(quotaMatrixOrig[k][indexVector[l]], invIndex); + } + meanSbr = fixMax(meanSbr,(accu<<1)); + + } + } + else{ + for(l = startChannel; l < stopChannel;l++){ + /* tonalityOrig */ + accu = FL2FXCONST_DBL(0.0f); + for(k = startIndex ; k < stopIndex; k++){ + accu += fMultDiv2(quotaMatrixOrig[k][l], invIndex); + } + meanOrig += fMult((accu<<1), invChannel); + + /* tonalitySbr */ + accu = FL2FXCONST_DBL(0.0f); + for(k = startIndex ; k < stopIndex; k++){ + accu += fMultDiv2(quotaMatrixOrig[k][indexVector[l]], invIndex); + } + meanSbr += fMult((accu<<1), invChannel); + } + } + + /* Small fix to avoid noise during silent passages.*/ + if( meanOrig <= FL2FXCONST_DBL(0.000976562f*RELAXATION_FLOAT) && + meanSbr <= FL2FXCONST_DBL(0.000976562f*RELAXATION_FLOAT) ) + { + meanOrig = FL2FXCONST_DBL(101.5936673f*RELAXATION_FLOAT); + meanSbr = FL2FXCONST_DBL(101.5936673f*RELAXATION_FLOAT); + } + + meanOrig = fixMax(meanOrig,RELAXATION); + meanSbr = fixMax(meanSbr,RELAXATION); + + if (missingHarmonicFlag == 1 || + inverseFilteringLevel == INVF_MID_LEVEL || + inverseFilteringLevel == INVF_LOW_LEVEL || + inverseFilteringLevel == INVF_OFF || + inverseFilteringLevel <= diffThres) + { + diff = RELAXATION; + } + else { + accu = fDivNorm(meanSbr, meanOrig, &scale); + + diff = fixMax( RELAXATION, + fMult(RELAXATION_FRACT,fMult(weightFac,accu)) >>( RELAXATION_SHIFT-scale ) ) ; + } + + /* + * noise Level is now a positive value, i.e. + * the more harmonic the signal is the higher noise level, + * this makes no sense so we change the sign. + *********************************************************/ + accu = fDivNorm(diff, meanOrig, &scale); + scale -= 2; + + if ( (scale>0) && (accu > ((FIXP_DBL)MAXVAL_DBL)>>scale) ) { + *noiseLevel = (FIXP_DBL)MAXVAL_DBL; + } + else { + *noiseLevel = scaleValue(accu, scale); + } + + /* + * Add a noise floor offset to compensate for bias in the detector + *****************************************************************/ + if(!missingHarmonicFlag) + *noiseLevel = fMult(*noiseLevel, noiseFloorOffset)<<(NOISE_FLOOR_OFFSET_SCALING); + + /* + * check to see that we don't exceed the maximum allowed level + **************************************************************/ + *noiseLevel = fixMin(*noiseLevel, ana_max_level); /* ana_max_level is scaled with factor 0.25 */ +} + +/**************************************************************************/ +/*! + \brief Does the noise floor level estiamtion. + The function calls the Noisefloor estimation function + for the time segments decided based upon the transient + information. The block is always divided into one or two segments. + + + \return none + +*/ +/**************************************************************************/ +void +FDKsbrEnc_sbrNoiseFloorEstimateQmf(HANDLE_SBR_NOISE_FLOOR_ESTIMATE h_sbrNoiseFloorEstimate, /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct */ + const SBR_FRAME_INFO *frame_info, /*!< Time frequency grid of the current frame. */ + FIXP_DBL *noiseLevels, /*!< Pointer to vector to store the noise levels in.*/ + FIXP_DBL **quotaMatrixOrig, /*!< Matrix holding the quota values of the original. */ + SCHAR *indexVector, /*!< Index vector to obtain the patched data. */ + INT missingHarmonicsFlag, /*!< Flag indicating if a strong tonal component will be missing. */ + INT startIndex, /*!< Start index. */ + int numberOfEstimatesPerFrame, /*!< The number of tonality estimates per frame. */ + int transientFrame, /*!< A flag indicating if a transient is present. */ + INVF_MODE* pInvFiltLevels, /*!< Pointer to the vector holding the inverse filtering levels. */ + UINT sbrSyntaxFlags + ) + +{ + + INT nNoiseEnvelopes, startPos[2], stopPos[2], env, band; + + INT noNoiseBands = h_sbrNoiseFloorEstimate->noNoiseBands; + INT *freqBandTable = h_sbrNoiseFloorEstimate->freqBandTableQmf; + + nNoiseEnvelopes = frame_info->nNoiseEnvelopes; + + if (sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) { + nNoiseEnvelopes = 1; + startPos[0] = startIndex; + stopPos[0] = startIndex + min(numberOfEstimatesPerFrame,2); + } else + if(nNoiseEnvelopes == 1){ + startPos[0] = startIndex; + stopPos[0] = startIndex + 2; + } + else{ + startPos[0] = startIndex; + stopPos[0] = startIndex + 1; + startPos[1] = startIndex + 1; + stopPos[1] = startIndex + 2; + } + + /* + * Estimate the noise floor. + **************************************/ + for(env = 0; env < nNoiseEnvelopes; env++){ + for(band = 0; band < noNoiseBands; band++){ + FDK_ASSERT( (band + env*noNoiseBands) < MAX_NUM_NOISE_VALUES); + qmfBasedNoiseFloorDetection(&noiseLevels[band + env*noNoiseBands], + quotaMatrixOrig, + indexVector, + startPos[env], + stopPos[env], + freqBandTable[band], + freqBandTable[band+1], + h_sbrNoiseFloorEstimate->ana_max_level, + h_sbrNoiseFloorEstimate->noiseFloorOffset[band], + missingHarmonicsFlag, + h_sbrNoiseFloorEstimate->weightFac, + h_sbrNoiseFloorEstimate->diffThres, + pInvFiltLevels[band]); + } + } + + + /* + * Smoothing of the values. + **************************/ + smoothingOfNoiseLevels(noiseLevels, + nNoiseEnvelopes, + h_sbrNoiseFloorEstimate->noNoiseBands, + h_sbrNoiseFloorEstimate->prevNoiseLevels, + h_sbrNoiseFloorEstimate->smoothFilter, + transientFrame); + + + /* quantisation*/ + for(env = 0; env < nNoiseEnvelopes; env++){ + for(band = 0; band < noNoiseBands; band++){ + FDK_ASSERT( (band + env*noNoiseBands) < MAX_NUM_NOISE_VALUES); + noiseLevels[band + env*noNoiseBands] = + (FIXP_DBL)NOISE_FLOOR_OFFSET_64 - (FIXP_DBL)CalcLdData(noiseLevels[band + env*noNoiseBands]+(FIXP_DBL)1) + QuantOffset; + } + } +} + +/**************************************************************************/ +/*! + \brief + + + \return errorCode, noError if successful + +*/ +/**************************************************************************/ +static INT +downSampleLoRes(INT *v_result, /*!< */ + INT num_result, /*!< */ + const UCHAR *freqBandTableRef,/*!< */ + INT num_Ref) /*!< */ +{ + INT step; + INT i,j; + INT org_length,result_length; + INT v_index[MAX_FREQ_COEFFS/2]; + + /* init */ + org_length=num_Ref; + result_length=num_result; + + v_index[0]=0; /* Always use left border */ + i=0; + while(org_length > 0) /* Create downsample vector */ + { + i++; + step=org_length/result_length; /* floor; */ + org_length=org_length - step; + result_length--; + v_index[i]=v_index[i-1]+step; + } + + if(i != num_result ) /* Should never happen */ + return (1);/* error downsampling */ + + for(j=0;j<=i;j++) /* Use downsample vector to index LoResolution vector. */ + { + v_result[j]=freqBandTableRef[v_index[j]]; + } + + return (0); +} + +/**************************************************************************/ +/*! + \brief Initialize an instance of the noise floor level estimation module. + + + \return errorCode, noError if successful + +*/ +/**************************************************************************/ +INT +FDKsbrEnc_InitSbrNoiseFloorEstimate (HANDLE_SBR_NOISE_FLOOR_ESTIMATE h_sbrNoiseFloorEstimate, /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct */ + INT ana_max_level, /*!< Maximum level of the adaptive noise. */ + const UCHAR *freqBandTable, /*!< Frequany band table. */ + INT nSfb, /*!< Number of frequency bands. */ + INT noiseBands, /*!< Number of noise bands per octave. */ + INT noiseFloorOffset, /*!< Noise floor offset. */ + INT timeSlots, /*!< Number of time slots in a frame. */ + UINT useSpeechConfig /*!< Flag: adapt tuning parameters according to speech */ + ) +{ + INT i, qexp, qtmp; + FIXP_DBL tmp, exp; + + FDKmemclear(h_sbrNoiseFloorEstimate,sizeof(SBR_NOISE_FLOOR_ESTIMATE)); + + h_sbrNoiseFloorEstimate->smoothFilter = smoothFilter; + if (useSpeechConfig) { + h_sbrNoiseFloorEstimate->weightFac = (FIXP_DBL)MAXVAL_DBL; + h_sbrNoiseFloorEstimate->diffThres = INVF_LOW_LEVEL; + } + else { + h_sbrNoiseFloorEstimate->weightFac = FL2FXCONST_DBL(0.25f); + h_sbrNoiseFloorEstimate->diffThres = INVF_MID_LEVEL; + } + + h_sbrNoiseFloorEstimate->timeSlots = timeSlots; + h_sbrNoiseFloorEstimate->noiseBands = noiseBands; + + /* h_sbrNoiseFloorEstimate->ana_max_level is scaled by 0.25 */ + switch(ana_max_level) + { + case 6: + h_sbrNoiseFloorEstimate->ana_max_level = (FIXP_DBL)MAXVAL_DBL; + break; + case 3: + h_sbrNoiseFloorEstimate->ana_max_level = FL2FXCONST_DBL(0.5); + break; + case -3: + h_sbrNoiseFloorEstimate->ana_max_level = FL2FXCONST_DBL(0.125); + break; + default: + /* Should not enter here */ + h_sbrNoiseFloorEstimate->ana_max_level = (FIXP_DBL)MAXVAL_DBL; + break; + } + + /* + calculate number of noise bands and allocate + */ + if(FDKsbrEnc_resetSbrNoiseFloorEstimate(h_sbrNoiseFloorEstimate,freqBandTable,nSfb)) + return(1); + + if(noiseFloorOffset == 0) { + tmp = ((FIXP_DBL)MAXVAL_DBL)>>NOISE_FLOOR_OFFSET_SCALING; + } + else { + FDK_ASSERT(noiseFloorOffset<=8); /* because of NOISE_FLOOR_OFFSET_SCALING */ + + /* Assumes the noise floor offset in tuning table are in q31 */ + /* Currently the table contains only 0 for noise floor offset */ + /* Change the qformat here when non-zero values would be filled */ + exp = fDivNorm((FIXP_DBL)noiseFloorOffset, 3, &qexp); + tmp = fPow(2, DFRACT_BITS-1, exp, qexp, &qtmp); + tmp = scaleValue(tmp, qtmp-NOISE_FLOOR_OFFSET_SCALING); + } + + for(i=0;i<h_sbrNoiseFloorEstimate->noNoiseBands;i++) { + h_sbrNoiseFloorEstimate->noiseFloorOffset[i] = tmp; + } + + return (0); +} + +/**************************************************************************/ +/*! + \brief Resets the current instance of the noise floor estiamtion + module. + + + \return errorCode, noError if successful + +*/ +/**************************************************************************/ +INT +FDKsbrEnc_resetSbrNoiseFloorEstimate (HANDLE_SBR_NOISE_FLOOR_ESTIMATE h_sbrNoiseFloorEstimate, /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct */ + const UCHAR *freqBandTable, /*!< Frequany band table. */ + INT nSfb) /*!< Number of bands in the frequency band table. */ +{ + INT k2,kx; + + /* + * Calculate number of noise bands + ***********************************/ + k2=freqBandTable[nSfb]; + kx=freqBandTable[0]; + if(h_sbrNoiseFloorEstimate->noiseBands == 0){ + h_sbrNoiseFloorEstimate->noNoiseBands = 1; + } + else{ + /* + * Calculate number of noise bands 1,2 or 3 bands/octave + ********************************************************/ + FIXP_DBL tmp, ratio, lg2; + INT ratio_e, qlg2; + + ratio = fDivNorm(k2, kx, &ratio_e); + lg2 = fLog2(ratio, ratio_e, &qlg2); + tmp = fMult((FIXP_DBL)(h_sbrNoiseFloorEstimate->noiseBands<<24), lg2); + tmp = scaleValue(tmp, qlg2-23); + + h_sbrNoiseFloorEstimate->noNoiseBands = (INT)((tmp + (FIXP_DBL)1) >> 1); + + if (h_sbrNoiseFloorEstimate->noNoiseBands > MAX_NUM_NOISE_COEFFS) + h_sbrNoiseFloorEstimate->noNoiseBands = MAX_NUM_NOISE_COEFFS; + + if( h_sbrNoiseFloorEstimate->noNoiseBands==0) + h_sbrNoiseFloorEstimate->noNoiseBands=1; + } + + + return(downSampleLoRes(h_sbrNoiseFloorEstimate->freqBandTableQmf, + h_sbrNoiseFloorEstimate->noNoiseBands, + freqBandTable,nSfb)); +} + +/**************************************************************************/ +/*! + \brief Deletes the current instancce of the noise floor level + estimation module. + + + \return none + +*/ +/**************************************************************************/ +void +FDKsbrEnc_deleteSbrNoiseFloorEstimate (HANDLE_SBR_NOISE_FLOOR_ESTIMATE h_sbrNoiseFloorEstimate) /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct */ +{ + + if (h_sbrNoiseFloorEstimate) { + /* + nothing to do + */ + } +} |