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-
-/* -----------------------------------------------------------------------------------------------------------
-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 "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 (4)
-
-
-
-/**************************************************************************/
-/*!
- \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 {
- /* noiseFloorOffset has to be smaller than 12, because
- the result of the calculation below must be smaller than 1:
- (2^(noiseFloorOffset/3))*2^4<1 */
- FDK_ASSERT(noiseFloorOffset<12);
-
- /* Assumes the noise floor offset in tuning table are in q31 */
- /* 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, nNoiseBands;
-
- 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);
-
- nNoiseBands = (INT)((tmp + (FIXP_DBL)1) >> 1);
-
-
- if (nNoiseBands > MAX_NUM_NOISE_COEFFS ) {
- nNoiseBands = MAX_NUM_NOISE_COEFFS;
- }
-
- if( nNoiseBands == 0 ) {
- nNoiseBands = 1;
- }
-
- h_sbrNoiseFloorEstimate->noNoiseBands = nNoiseBands;
-
- }
-
-
- 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
- */
- }
-}