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+
+/* -----------------------------------------------------------------------------------------------------------
+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
+    */
+  }
+}