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      1 
      2 /* -----------------------------------------------------------------------------------------------------------
      3 Software License for The Fraunhofer FDK AAC Codec Library for Android
      4 
      5  Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Frderung der angewandten Forschung e.V.
      6   All rights reserved.
      7 
      8  1.    INTRODUCTION
      9 The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
     10 the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
     11 This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
     12 
     13 AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
     14 audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
     15 independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
     16 of the MPEG specifications.
     17 
     18 Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
     19 may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
     20 individually for the purpose of encoding or decoding bit streams in products that are compliant with
     21 the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
     22 these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
     23 software may already be covered under those patent licenses when it is used for those licensed purposes only.
     24 
     25 Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
     26 are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
     27 applications information and documentation.
     28 
     29 2.    COPYRIGHT LICENSE
     30 
     31 Redistribution and use in source and binary forms, with or without modification, are permitted without
     32 payment of copyright license fees provided that you satisfy the following conditions:
     33 
     34 You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
     35 your modifications thereto in source code form.
     36 
     37 You must retain the complete text of this software license in the documentation and/or other materials
     38 provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
     39 You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
     40 modifications thereto to recipients of copies in binary form.
     41 
     42 The name of Fraunhofer may not be used to endorse or promote products derived from this library without
     43 prior written permission.
     44 
     45 You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
     46 software or your modifications thereto.
     47 
     48 Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
     49 and the date of any change. For modified versions of the FDK AAC Codec, the term
     50 "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
     51 "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
     52 
     53 3.    NO PATENT LICENSE
     54 
     55 NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
     56 ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
     57 respect to this software.
     58 
     59 You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
     60 by appropriate patent licenses.
     61 
     62 4.    DISCLAIMER
     63 
     64 This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
     65 "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
     66 of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
     67 CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
     68 including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
     69 or business interruption, however caused and on any theory of liability, whether in contract, strict
     70 liability, or tort (including negligence), arising in any way out of the use of this software, even if
     71 advised of the possibility of such damage.
     72 
     73 5.    CONTACT INFORMATION
     74 
     75 Fraunhofer Institute for Integrated Circuits IIS
     76 Attention: Audio and Multimedia Departments - FDK AAC LL
     77 Am Wolfsmantel 33
     78 91058 Erlangen, Germany
     79 
     80 www.iis.fraunhofer.de/amm
     81 amm-info (at) iis.fraunhofer.de
     82 ----------------------------------------------------------------------------------------------------------- */
     83 
     84 #include "nf_est.h"
     85 
     86 #include "sbr_misc.h"
     87 
     88 #include "genericStds.h"
     89 
     90 /* smoothFilter[4]  = {0.05857864376269f, 0.2f, 0.34142135623731f, 0.4f}; */
     91 static const FIXP_DBL smoothFilter[4]  = { 0x077f813d, 0x19999995, 0x2bb3b1f5, 0x33333335 };
     92 
     93 /* static const INT smoothFilterLength = 4; */
     94 
     95 static const FIXP_DBL QuantOffset = (INT)0xfc000000;  /* ld64(0.25) */
     96 
     97 #ifndef min
     98 #define min(a,b) ( a < b ? a:b)
     99 #endif
    100 
    101 #ifndef max
    102 #define max(a,b) ( a > b ? a:b)
    103 #endif
    104 
    105 #define NOISE_FLOOR_OFFSET_SCALING  (4)
    106 
    107 
    108 
    109 /**************************************************************************/
    110 /*!
    111   \brief     The function applies smoothing to the noise levels.
    112 
    113 
    114 
    115   \return    none
    116 
    117 */
    118 /**************************************************************************/
    119 static void
    120 smoothingOfNoiseLevels(FIXP_DBL *NoiseLevels,        /*!< pointer to noise-floor levels.*/
    121                        INT nEnvelopes,               /*!< Number of noise floor envelopes.*/
    122                        INT noNoiseBands,             /*!< Number of noise bands for every noise floor envelope. */
    123                        FIXP_DBL prevNoiseLevels[NF_SMOOTHING_LENGTH][MAX_NUM_NOISE_VALUES],/*!< Previous noise floor envelopes. */
    124                        const FIXP_DBL *smoothFilter, /*!< filter used for smoothing the noise floor levels. */
    125                        INT transientFlag)            /*!< flag indicating if a transient is present*/
    126 
    127 {
    128   INT i,band,env;
    129   FIXP_DBL accu;
    130 
    131   for(env = 0; env < nEnvelopes; env++){
    132     if(transientFlag){
    133       for (i = 0; i < NF_SMOOTHING_LENGTH; i++){
    134         FDKmemcpy(prevNoiseLevels[i],NoiseLevels+env*noNoiseBands,noNoiseBands*sizeof(FIXP_DBL));
    135       }
    136     }
    137     else {
    138       for (i = 1; i < NF_SMOOTHING_LENGTH; i++){
    139         FDKmemcpy(prevNoiseLevels[i - 1],prevNoiseLevels[i],noNoiseBands*sizeof(FIXP_DBL));
    140       }
    141       FDKmemcpy(prevNoiseLevels[NF_SMOOTHING_LENGTH - 1],NoiseLevels+env*noNoiseBands,noNoiseBands*sizeof(FIXP_DBL));
    142     }
    143 
    144     for (band = 0; band < noNoiseBands; band++){
    145       accu = FL2FXCONST_DBL(0.0f);
    146       for (i = 0; i < NF_SMOOTHING_LENGTH; i++){
    147         accu += fMultDiv2(smoothFilter[i], prevNoiseLevels[i][band]);
    148       }
    149       FDK_ASSERT( (band + env*noNoiseBands) < MAX_NUM_NOISE_VALUES);
    150       NoiseLevels[band+ env*noNoiseBands] = accu<<1;
    151     }
    152   }
    153 }
    154 
    155 /**************************************************************************/
    156 /*!
    157   \brief     Does the noise floor level estiamtion.
    158 
    159   The noiseLevel samples are scaled by the factor 0.25
    160 
    161   \return    none
    162 
    163 */
    164 /**************************************************************************/
    165 static void
    166 qmfBasedNoiseFloorDetection(FIXP_DBL *noiseLevel,              /*!< Pointer to vector to store the noise levels in.*/
    167                             FIXP_DBL ** quotaMatrixOrig,       /*!< Matrix holding the quota values of the original. */
    168                             SCHAR *indexVector,                /*!< Index vector to obtain the patched data. */
    169                             INT startIndex,                    /*!< Start index. */
    170                             INT stopIndex,                     /*!< Stop index. */
    171                             INT startChannel,                  /*!< Start channel of the current noise floor band.*/
    172                             INT stopChannel,                   /*!< Stop channel of the current noise floor band. */
    173                             FIXP_DBL ana_max_level,            /*!< Maximum level of the adaptive noise.*/
    174                             FIXP_DBL noiseFloorOffset,         /*!< Noise floor offset. */
    175                             INT missingHarmonicFlag,           /*!< Flag indicating if a strong tonal component is missing.*/
    176                             FIXP_DBL weightFac,                /*!< Weightening factor for the difference between orig and sbr. */
    177                             INVF_MODE diffThres,               /*!< Threshold value to control the inverse filtering decision.*/
    178                             INVF_MODE inverseFilteringLevel)   /*!< Inverse filtering level of the current band.*/
    179 {
    180   INT scale, l, k;
    181   FIXP_DBL meanOrig=FL2FXCONST_DBL(0.0f), meanSbr=FL2FXCONST_DBL(0.0f), diff;
    182   FIXP_DBL invIndex = GetInvInt(stopIndex-startIndex);
    183   FIXP_DBL invChannel = GetInvInt(stopChannel-startChannel);
    184   FIXP_DBL accu;
    185 
    186    /*
    187    Calculate the mean value, over the current time segment, for the original, the HFR
    188    and the difference, over all channels in the current frequency range.
    189    */
    190 
    191   if(missingHarmonicFlag == 1){
    192     for(l = startChannel; l < stopChannel;l++){
    193       /* tonalityOrig */
    194       accu = FL2FXCONST_DBL(0.0f);
    195       for(k = startIndex ; k < stopIndex; k++){
    196         accu += fMultDiv2(quotaMatrixOrig[k][l], invIndex);
    197       }
    198       meanOrig = fixMax(meanOrig,(accu<<1));
    199 
    200       /* tonalitySbr */
    201       accu = FL2FXCONST_DBL(0.0f);
    202       for(k = startIndex ; k < stopIndex; k++){
    203         accu += fMultDiv2(quotaMatrixOrig[k][indexVector[l]], invIndex);
    204       }
    205       meanSbr  = fixMax(meanSbr,(accu<<1));
    206 
    207     }
    208   }
    209   else{
    210     for(l = startChannel; l < stopChannel;l++){
    211       /* tonalityOrig */
    212       accu = FL2FXCONST_DBL(0.0f);
    213       for(k = startIndex ; k < stopIndex; k++){
    214         accu += fMultDiv2(quotaMatrixOrig[k][l], invIndex);
    215       }
    216       meanOrig += fMult((accu<<1), invChannel);
    217 
    218       /* tonalitySbr */
    219       accu = FL2FXCONST_DBL(0.0f);
    220       for(k = startIndex ; k < stopIndex; k++){
    221         accu += fMultDiv2(quotaMatrixOrig[k][indexVector[l]], invIndex);
    222       }
    223       meanSbr  += fMult((accu<<1), invChannel);
    224     }
    225   }
    226 
    227   /* Small fix to avoid noise during silent passages.*/
    228   if( meanOrig <= FL2FXCONST_DBL(0.000976562f*RELAXATION_FLOAT) &&
    229       meanSbr <= FL2FXCONST_DBL(0.000976562f*RELAXATION_FLOAT) )
    230   {
    231     meanOrig = FL2FXCONST_DBL(101.5936673f*RELAXATION_FLOAT);
    232     meanSbr  = FL2FXCONST_DBL(101.5936673f*RELAXATION_FLOAT);
    233   }
    234 
    235   meanOrig = fixMax(meanOrig,RELAXATION);
    236   meanSbr  = fixMax(meanSbr,RELAXATION);
    237 
    238   if (missingHarmonicFlag == 1 ||
    239       inverseFilteringLevel == INVF_MID_LEVEL ||
    240       inverseFilteringLevel == INVF_LOW_LEVEL ||
    241       inverseFilteringLevel == INVF_OFF ||
    242       inverseFilteringLevel <= diffThres)
    243   {
    244     diff = RELAXATION;
    245   }
    246   else {
    247     accu = fDivNorm(meanSbr, meanOrig, &scale);
    248 
    249     diff = fixMax( RELAXATION,
    250                    fMult(RELAXATION_FRACT,fMult(weightFac,accu)) >>( RELAXATION_SHIFT-scale ) ) ;
    251   }
    252 
    253   /*
    254    * noise Level is now a positive value, i.e.
    255    * the more harmonic the signal is the higher noise level,
    256    * this makes no sense so we change the sign.
    257    *********************************************************/
    258   accu = fDivNorm(diff, meanOrig, &scale);
    259   scale -= 2;
    260 
    261   if ( (scale>0) && (accu > ((FIXP_DBL)MAXVAL_DBL)>>scale) ) {
    262     *noiseLevel = (FIXP_DBL)MAXVAL_DBL;
    263   }
    264   else {
    265     *noiseLevel = scaleValue(accu, scale);
    266   }
    267 
    268   /*
    269    * Add a noise floor offset to compensate for bias in the detector
    270    *****************************************************************/
    271   if(!missingHarmonicFlag)
    272     *noiseLevel = fMult(*noiseLevel, noiseFloorOffset)<<(NOISE_FLOOR_OFFSET_SCALING);
    273 
    274   /*
    275    * check to see that we don't exceed the maximum allowed level
    276    **************************************************************/
    277   *noiseLevel = fixMin(*noiseLevel, ana_max_level);     /* ana_max_level is scaled with factor 0.25 */
    278 }
    279 
    280 /**************************************************************************/
    281 /*!
    282   \brief     Does the noise floor level estiamtion.
    283   The function calls the Noisefloor estimation function
    284   for the time segments decided based upon the transient
    285   information. The block is always divided into one or two segments.
    286 
    287 
    288   \return    none
    289 
    290 */
    291 /**************************************************************************/
    292 void
    293 FDKsbrEnc_sbrNoiseFloorEstimateQmf(HANDLE_SBR_NOISE_FLOOR_ESTIMATE h_sbrNoiseFloorEstimate, /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct */
    294                          const SBR_FRAME_INFO *frame_info,   /*!< Time frequency grid of the current frame. */
    295                          FIXP_DBL *noiseLevels,              /*!< Pointer to vector to store the noise levels in.*/
    296                          FIXP_DBL **quotaMatrixOrig,         /*!< Matrix holding the quota values of the original. */
    297                          SCHAR    *indexVector,              /*!< Index vector to obtain the patched data. */
    298                          INT missingHarmonicsFlag,           /*!< Flag indicating if a strong tonal component will be missing. */
    299                          INT startIndex,                     /*!< Start index. */
    300                          int numberOfEstimatesPerFrame,      /*!< The number of tonality estimates per frame. */
    301                          int transientFrame,                 /*!< A flag indicating if a transient is present. */
    302                          INVF_MODE* pInvFiltLevels,          /*!< Pointer to the vector holding the inverse filtering levels. */
    303                          UINT sbrSyntaxFlags
    304                          )
    305 
    306 {
    307 
    308   INT nNoiseEnvelopes, startPos[2], stopPos[2], env, band;
    309 
    310   INT noNoiseBands      = h_sbrNoiseFloorEstimate->noNoiseBands;
    311   INT *freqBandTable    = h_sbrNoiseFloorEstimate->freqBandTableQmf;
    312 
    313   nNoiseEnvelopes = frame_info->nNoiseEnvelopes;
    314 
    315   if (sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) {
    316     nNoiseEnvelopes = 1;
    317     startPos[0] = startIndex;
    318     stopPos[0]  = startIndex + min(numberOfEstimatesPerFrame,2);
    319   } else
    320   if(nNoiseEnvelopes == 1){
    321     startPos[0] = startIndex;
    322     stopPos[0]  = startIndex + 2;
    323   }
    324   else{
    325     startPos[0] = startIndex;
    326     stopPos[0]  = startIndex + 1;
    327     startPos[1] = startIndex + 1;
    328     stopPos[1]  = startIndex + 2;
    329   }
    330 
    331   /*
    332    * Estimate the noise floor.
    333    **************************************/
    334   for(env = 0; env < nNoiseEnvelopes; env++){
    335     for(band = 0; band < noNoiseBands; band++){
    336       FDK_ASSERT( (band + env*noNoiseBands) < MAX_NUM_NOISE_VALUES);
    337       qmfBasedNoiseFloorDetection(&noiseLevels[band + env*noNoiseBands],
    338                                   quotaMatrixOrig,
    339                                   indexVector,
    340                                   startPos[env],
    341                                   stopPos[env],
    342                                   freqBandTable[band],
    343                                   freqBandTable[band+1],
    344                                   h_sbrNoiseFloorEstimate->ana_max_level,
    345                                   h_sbrNoiseFloorEstimate->noiseFloorOffset[band],
    346                                   missingHarmonicsFlag,
    347                                   h_sbrNoiseFloorEstimate->weightFac,
    348                                   h_sbrNoiseFloorEstimate->diffThres,
    349                                   pInvFiltLevels[band]);
    350     }
    351   }
    352 
    353 
    354   /*
    355    * Smoothing of the values.
    356    **************************/
    357   smoothingOfNoiseLevels(noiseLevels,
    358                          nNoiseEnvelopes,
    359                          h_sbrNoiseFloorEstimate->noNoiseBands,
    360                          h_sbrNoiseFloorEstimate->prevNoiseLevels,
    361                          h_sbrNoiseFloorEstimate->smoothFilter,
    362                          transientFrame);
    363 
    364 
    365   /* quantisation*/
    366   for(env = 0; env < nNoiseEnvelopes; env++){
    367     for(band = 0; band < noNoiseBands; band++){
    368       FDK_ASSERT( (band + env*noNoiseBands) < MAX_NUM_NOISE_VALUES);
    369       noiseLevels[band + env*noNoiseBands] =
    370          (FIXP_DBL)NOISE_FLOOR_OFFSET_64 - (FIXP_DBL)CalcLdData(noiseLevels[band + env*noNoiseBands]+(FIXP_DBL)1) + QuantOffset;
    371     }
    372   }
    373 }
    374 
    375 /**************************************************************************/
    376 /*!
    377   \brief
    378 
    379 
    380   \return    errorCode, noError if successful
    381 
    382 */
    383 /**************************************************************************/
    384 static INT
    385 downSampleLoRes(INT *v_result,              /*!<    */
    386                 INT num_result,             /*!<    */
    387                 const UCHAR *freqBandTableRef,/*!<    */
    388                 INT num_Ref)                /*!<    */
    389 {
    390   INT step;
    391   INT i,j;
    392   INT org_length,result_length;
    393   INT v_index[MAX_FREQ_COEFFS/2];
    394 
    395   /* init */
    396   org_length=num_Ref;
    397   result_length=num_result;
    398 
    399   v_index[0]=0;	/* Always use left border */
    400   i=0;
    401   while(org_length > 0)	/* Create downsample vector */
    402     {
    403       i++;
    404       step=org_length/result_length; /* floor; */
    405       org_length=org_length - step;
    406       result_length--;
    407       v_index[i]=v_index[i-1]+step;
    408     }
    409 
    410   if(i != num_result )	/* Should never happen */
    411     return (1);/* error downsampling */
    412 
    413   for(j=0;j<=i;j++)	/* Use downsample vector to index LoResolution vector. */
    414     {
    415       v_result[j]=freqBandTableRef[v_index[j]];
    416     }
    417 
    418   return (0);
    419 }
    420 
    421 /**************************************************************************/
    422 /*!
    423   \brief    Initialize an instance of the noise floor level estimation module.
    424 
    425 
    426   \return    errorCode, noError if successful
    427 
    428 */
    429 /**************************************************************************/
    430 INT
    431 FDKsbrEnc_InitSbrNoiseFloorEstimate (HANDLE_SBR_NOISE_FLOOR_ESTIMATE  h_sbrNoiseFloorEstimate,   /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct */
    432                              INT ana_max_level,                       /*!< Maximum level of the adaptive noise. */
    433                              const UCHAR *freqBandTable,      /*!< Frequany band table. */
    434                              INT nSfb,                                /*!< Number of frequency bands. */
    435                              INT noiseBands,                          /*!< Number of noise bands per octave. */
    436                              INT noiseFloorOffset,                    /*!< Noise floor offset. */
    437                              INT timeSlots,                           /*!< Number of time slots in a frame. */
    438                              UINT useSpeechConfig             /*!< Flag: adapt tuning parameters according to speech */
    439                             )
    440 {
    441   INT i, qexp, qtmp;
    442   FIXP_DBL tmp, exp;
    443 
    444   FDKmemclear(h_sbrNoiseFloorEstimate,sizeof(SBR_NOISE_FLOOR_ESTIMATE));
    445 
    446   h_sbrNoiseFloorEstimate->smoothFilter = smoothFilter;
    447   if (useSpeechConfig) {
    448     h_sbrNoiseFloorEstimate->weightFac = (FIXP_DBL)MAXVAL_DBL;
    449     h_sbrNoiseFloorEstimate->diffThres = INVF_LOW_LEVEL;
    450   }
    451   else {
    452     h_sbrNoiseFloorEstimate->weightFac = FL2FXCONST_DBL(0.25f);
    453     h_sbrNoiseFloorEstimate->diffThres = INVF_MID_LEVEL;
    454   }
    455 
    456   h_sbrNoiseFloorEstimate->timeSlots     = timeSlots;
    457   h_sbrNoiseFloorEstimate->noiseBands    = noiseBands;
    458 
    459   /* h_sbrNoiseFloorEstimate->ana_max_level is scaled by 0.25  */
    460   switch(ana_max_level)
    461   {
    462   case 6:
    463       h_sbrNoiseFloorEstimate->ana_max_level = (FIXP_DBL)MAXVAL_DBL;
    464       break;
    465   case 3:
    466       h_sbrNoiseFloorEstimate->ana_max_level = FL2FXCONST_DBL(0.5);
    467       break;
    468   case -3:
    469       h_sbrNoiseFloorEstimate->ana_max_level = FL2FXCONST_DBL(0.125);
    470       break;
    471   default:
    472       /* Should not enter here */
    473       h_sbrNoiseFloorEstimate->ana_max_level = (FIXP_DBL)MAXVAL_DBL;
    474       break;
    475   }
    476 
    477   /*
    478     calculate number of noise bands and allocate
    479   */
    480   if(FDKsbrEnc_resetSbrNoiseFloorEstimate(h_sbrNoiseFloorEstimate,freqBandTable,nSfb))
    481     return(1);
    482 
    483   if(noiseFloorOffset == 0) {
    484     tmp = ((FIXP_DBL)MAXVAL_DBL)>>NOISE_FLOOR_OFFSET_SCALING;
    485   }
    486   else {
    487     /* noiseFloorOffset has to be smaller than 12, because
    488        the result of the calculation below must be smaller than 1:
    489        (2^(noiseFloorOffset/3))*2^4<1                                        */
    490     FDK_ASSERT(noiseFloorOffset<12);
    491 
    492     /* Assumes the noise floor offset in tuning table are in q31    */
    493     /* Change the qformat here when non-zero values would be filled */
    494     exp = fDivNorm((FIXP_DBL)noiseFloorOffset, 3, &qexp);
    495     tmp = fPow(2, DFRACT_BITS-1, exp, qexp, &qtmp);
    496     tmp = scaleValue(tmp, qtmp-NOISE_FLOOR_OFFSET_SCALING);
    497   }
    498 
    499   for(i=0;i<h_sbrNoiseFloorEstimate->noNoiseBands;i++) {
    500     h_sbrNoiseFloorEstimate->noiseFloorOffset[i] = tmp;
    501   }
    502 
    503   return (0);
    504 }
    505 
    506 /**************************************************************************/
    507 /*!
    508   \brief     Resets the current instance of the noise floor estiamtion
    509           module.
    510 
    511 
    512   \return    errorCode, noError if successful
    513 
    514 */
    515 /**************************************************************************/
    516 INT
    517 FDKsbrEnc_resetSbrNoiseFloorEstimate (HANDLE_SBR_NOISE_FLOOR_ESTIMATE h_sbrNoiseFloorEstimate, /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct */
    518                             const UCHAR *freqBandTable,             /*!< Frequany band table. */
    519                             INT nSfb)                             /*!< Number of bands in the frequency band table. */
    520 {
    521     INT k2,kx;
    522 
    523     /*
    524     * Calculate number of noise bands
    525     ***********************************/
    526     k2=freqBandTable[nSfb];
    527     kx=freqBandTable[0];
    528     if(h_sbrNoiseFloorEstimate->noiseBands == 0){
    529         h_sbrNoiseFloorEstimate->noNoiseBands = 1;
    530     }
    531     else{
    532         /*
    533         * Calculate number of noise bands 1,2 or 3 bands/octave
    534         ********************************************************/
    535         FIXP_DBL tmp, ratio, lg2;
    536         INT ratio_e, qlg2, nNoiseBands;
    537 
    538         ratio = fDivNorm(k2, kx, &ratio_e);
    539         lg2 = fLog2(ratio, ratio_e, &qlg2);
    540         tmp = fMult((FIXP_DBL)(h_sbrNoiseFloorEstimate->noiseBands<<24), lg2);
    541         tmp = scaleValue(tmp, qlg2-23);
    542 
    543         nNoiseBands = (INT)((tmp + (FIXP_DBL)1) >> 1);
    544 
    545 
    546         if (nNoiseBands > MAX_NUM_NOISE_COEFFS ) {
    547           nNoiseBands = MAX_NUM_NOISE_COEFFS;
    548         }
    549 
    550         if( nNoiseBands == 0 ) {
    551           nNoiseBands = 1;
    552         }
    553 
    554         h_sbrNoiseFloorEstimate->noNoiseBands = nNoiseBands;
    555 
    556     }
    557 
    558 
    559     return(downSampleLoRes(h_sbrNoiseFloorEstimate->freqBandTableQmf,
    560         h_sbrNoiseFloorEstimate->noNoiseBands,
    561         freqBandTable,nSfb));
    562 }
    563 
    564 /**************************************************************************/
    565 /*!
    566   \brief     Deletes the current instancce of the noise floor level
    567   estimation module.
    568 
    569 
    570   \return    none
    571 
    572 */
    573 /**************************************************************************/
    574 void
    575 FDKsbrEnc_deleteSbrNoiseFloorEstimate (HANDLE_SBR_NOISE_FLOOR_ESTIMATE h_sbrNoiseFloorEstimate)  /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct */
    576 {
    577 
    578   if (h_sbrNoiseFloorEstimate) {
    579     /*
    580       nothing to do
    581     */
    582   }
    583 }
    584