Include patched FDK-AAC in the repository

The initial idea was to get the DAB+ patch into upstream, but since
that follows the android source releases, there is no place for a custom
DAB+ patch there.

So instead of having to maintain a patched fdk-aac that has to have the
same .so version as the distribution package on which it is installed,
we prefer having a separate fdk-aac-dab library to avoid collision.

At that point, there's no reason to keep fdk-aac in a separate
repository, as odr-audioenc is the only tool that needs DAB+ encoding
support. Including it here simplifies installation, and makes it
consistent with toolame-dab, also shipped in this repository.

DAB+ decoding support (needed by ODR-SourceCompanion, dablin, etisnoop,
welle.io and others) can be done using upstream FDK-AAC.

1 files changed, 612 insertions, 0 deletions

diff --git a/fdk-aac/libSBRenc/src/nf_est.cpp b/fdk-aac/libSBRenc/src/nf_est.cpp
new file mode 100644
index 0000000..290ec35
--- /dev/null
+++ b/fdk-aac/libSBRenc/src/nf_est.cpp
@@ -0,0 +1,612 @@
+/* -----------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2018 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
+----------------------------------------------------------------------------- */
+
+/**************************** SBR encoder library ******************************
+
+   Author(s):
+
+   Description:
+
+*******************************************************************************/
+
+#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 *
+        pSmoothFilter, /*!< 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(pSmoothFilter[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 = fixMin(fMult(*noiseLevel, noiseFloorOffset),
+                         (FIXP_DBL)MAXVAL_DBL >> NOISE_FLOOR_OFFSET_SCALING)
+                  << 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. */
+    UINT 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;
+
+  startPos[0] = startIndex;
+
+  if (nNoiseEnvelopes == 1) {
+    stopPos[0] = startIndex + min(numberOfEstimatesPerFrame, 2);
+  } else {
+    stopPos[0] = startIndex + 1;
+    startPos[1] = startIndex + 1;
+    stopPos[1] = startIndex + min(numberOfEstimatesPerFrame, 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,  /*!< Frequency 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,  /*!< Frequency 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
+    */
+  }
+}