android-4.3_r1.1/s

/* -----------------------------------------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android

� Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur F�rderung 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
----------------------------------------------------------------------------------------------------------- */

/*!
  \file
  \brief  Frequency scale calculation
*/

#include "sbrdec_freq_sca.h"

#include "transcendent.h"
#include "sbr_rom.h"
#include "env_extr.h"

#include "genericStds.h"      /* need log() for debug-code only */

#define MAX_OCTAVE         29
#define MAX_SECOND_REGION  50


static int  numberOfBands(FIXP_SGL bpo_div16, int start, int stop, int warpFlag);
static void CalcBands(UCHAR * diff, UCHAR start, UCHAR stop, UCHAR num_bands);
static SBR_ERROR modifyBands(UCHAR max_band, UCHAR * diff, UCHAR length);
static void cumSum(UCHAR start_value, UCHAR* diff, UCHAR length, UCHAR *start_adress);


/*!
  \brief     Retrieve QMF-band where the SBR range starts

  Convert startFreq which was read from the bitstream into a
  QMF-channel number.

  \return  Number of start band
*/
static UCHAR
getStartBand(UINT   fs,               /*!< Output sampling frequency */
             UCHAR  startFreq,        /*!< Index to table of possible start bands */
             UINT   headerDataFlags)  /*!< Info to SBR mode */
{
  INT  band;
  UINT fsMapped;

    fsMapped = fs;

  switch (fsMapped) {
    case 48000:
      band = FDK_sbrDecoder_sbr_start_freq_48[startFreq];
      break;
    case 44100:
      band = FDK_sbrDecoder_sbr_start_freq_44[startFreq];
      break;
    case 32000:
      band = FDK_sbrDecoder_sbr_start_freq_32[startFreq];
      break;
    case 24000:
      band = FDK_sbrDecoder_sbr_start_freq_24[startFreq];
      break;
    case 22050:
      band = FDK_sbrDecoder_sbr_start_freq_22[startFreq];
      break;
    case 16000:
      band = FDK_sbrDecoder_sbr_start_freq_16[startFreq];
      break;
    default:
      band = 255;
  }

  return band;
}


/*!
  \brief     Retrieve QMF-band where the SBR range starts

  Convert startFreq which was read from the bitstream into a
  QMF-channel number.

  \return  Number of start band
*/
static UCHAR
getStopBand(UINT   fs,               /*!< Output sampling frequency */
            UCHAR  stopFreq,         /*!< Index to table of possible start bands */
            UINT   headerDataFlags,  /*!< Info to SBR mode */
            UCHAR  k0)               /*!< Start freq index */
{
  UCHAR k2;

  if (stopFreq < 14) {
    INT    stopMin;
    UCHAR  diff_tot[MAX_OCTAVE + MAX_SECOND_REGION];
    UCHAR *diff0 = diff_tot;
    UCHAR *diff1 = diff_tot+MAX_OCTAVE;

    if (fs < 32000) {
      stopMin = (((2*6000*2*(64)) / fs) + 1) >> 1;
    }
    else {
      if (fs < 64000) {
        stopMin = (((2*8000*2*(64)) / fs) + 1) >> 1;
      }
      else {
        stopMin = (((2*10000*2*(64)) / fs) + 1) >> 1;
      }
    }

    /*
      Choose a stop band between k1 and 64 depending on stopFreq (0..13),
      based on a logarithmic scale.
      The vectors diff0 and diff1 are used temporarily here.
    */
    CalcBands( diff0, stopMin, 64, 13);
    shellsort( diff0, 13);
    cumSum(stopMin, diff0, 13, diff1);
    k2 = diff1[stopFreq];
  }
  else if (stopFreq==14)
    k2 = 2*k0;
  else
    k2 = 3*k0;

  /* Limit to Nyquist */
  if (k2 > (64))
    k2 = (64);


  /* Range checks */
  /* 1 <= difference <= 48; 1 <= fs <= 96000 */
  if ( ((k2 - k0) > MAX_FREQ_COEFFS) || (k2 <= k0) ) {
    return 255;
  }

  if (headerDataFlags & (SBRDEC_SYNTAX_USAC|SBRDEC_SYNTAX_RSVD50)) {
    /* 1 <= difference <= 35; 42000 <= fs <= 96000 */
    if ( (fs >= 42000) && ( (k2 - k0) > MAX_FREQ_COEFFS_FS44100 ) ) {
      return 255;
    }
    /* 1 <= difference <= 32; 46009 <= fs <= 96000 */
    if ( (fs >= 46009) && ( (k2 - k0) > MAX_FREQ_COEFFS_FS48000 ) ) {
      return 255;
    }
  }
  else {
    /* 1 <= difference <= 35; fs == 44100 */
    if ( (fs == 44100) && ( (k2 - k0) > MAX_FREQ_COEFFS_FS44100 ) ) {
      return 255;
    }
    /* 1 <= difference <= 32; 48000 <= fs <= 96000 */
    if ( (fs >= 48000) && ( (k2 - k0) > MAX_FREQ_COEFFS_FS48000 ) ) {
      return 255;
    }
  }

  return k2;
}


/*!
  \brief     Generates master frequency tables

  Frequency tables are calculated according to the selected domain
  (linear/logarithmic) and granularity.
  IEC 14496-3 4.6.18.3.2.1

  \return  errorCode, 0 if successful
*/
SBR_ERROR
sbrdecUpdateFreqScale(UCHAR * v_k_master,    /*!< Master table to be created */
                      UCHAR *numMaster,      /*!< Number of entries in master table */
                      UINT   fs,             /*!< SBR working sampling rate */
                      HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Control data from bitstream */
                      UINT flags)
{
  FIXP_SGL bpo_div16;        /* bands_per_octave divided by 16 */
  INT      dk=0;

  /* Internal variables */
  UCHAR  k0, k2, i;
  UCHAR  num_bands0 = 0;
  UCHAR  num_bands1 = 0;
  UCHAR  diff_tot[MAX_OCTAVE + MAX_SECOND_REGION];
  UCHAR *diff0 = diff_tot;
  UCHAR *diff1 = diff_tot+MAX_OCTAVE;
  INT    k2_achived;
  INT    k2_diff;
  INT    incr=0;

  /*
    Determine start band
  */
  k0 = getStartBand(fs, hHeaderData->bs_data.startFreq, flags);
  if (k0 == 255) {
    return SBRDEC_UNSUPPORTED_CONFIG;
  }

  /*
    Determine stop band
  */
  k2 = getStopBand(fs, hHeaderData->bs_data.stopFreq, flags, k0);
  if (k2 == 255) {
    return SBRDEC_UNSUPPORTED_CONFIG;
  }

  if(hHeaderData->bs_data.freqScale>0) { /* Bark */
    INT k1;

    if(hHeaderData->bs_data.freqScale==1) {
      bpo_div16 = FL2FXCONST_SGL(12.0f/16.0f);
    }
    else if(hHeaderData->bs_data.freqScale==2) {
      bpo_div16 = FL2FXCONST_SGL(10.0f/16.0f);
    }
    else {
      bpo_div16 =  FL2FXCONST_SGL(8.0f/16.0f);
    }


    if( 1000 * k2 > 2245 * k0 ) { /* Two or more regions */
      k1 = 2*k0;

      num_bands0 = numberOfBands(bpo_div16, k0, k1, 0);
      num_bands1 = numberOfBands(bpo_div16, k1, k2, hHeaderData->bs_data.alterScale );
      if ( num_bands0 < 1) {
        return SBRDEC_UNSUPPORTED_CONFIG;
      }
      if ( num_bands1 < 1 ) {
        return SBRDEC_UNSUPPORTED_CONFIG;
      }

      CalcBands(diff0, k0, k1, num_bands0);
      shellsort( diff0, num_bands0);
      if (diff0[0] == 0) {
#ifdef DEBUG_TOOLS
#endif
        return SBRDEC_UNSUPPORTED_CONFIG;
      }

      cumSum(k0, diff0, num_bands0, v_k_master);

      CalcBands(diff1, k1, k2, num_bands1);
      shellsort( diff1, num_bands1);
      if(diff0[num_bands0-1] > diff1[0]) {
        SBR_ERROR err;

        err = modifyBands(diff0[num_bands0-1],diff1, num_bands1);
        if (err)
          return SBRDEC_UNSUPPORTED_CONFIG;
      }

      /* Add 2nd region */
      cumSum(k1, diff1, num_bands1, &v_k_master[num_bands0]);
      *numMaster = num_bands0 + num_bands1;     /* Output nr of bands */

    }
    else { /* Only one region */
      k1=k2;

      num_bands0 = numberOfBands(bpo_div16, k0, k1, 0);
      if ( num_bands0 < 1) {
        return SBRDEC_UNSUPPORTED_CONFIG;
      }
      CalcBands(diff0, k0, k1, num_bands0);
      shellsort(diff0, num_bands0);
      if (diff0[0] == 0) {
#ifdef DEBUG_TOOLS
#endif
        return SBRDEC_UNSUPPORTED_CONFIG;
      }

      cumSum(k0, diff0, num_bands0, v_k_master);
      *numMaster = num_bands0;        /* Output nr of bands */

    }
  }
  else { /* Linear mode */
     if (hHeaderData->bs_data.alterScale==0) {
        dk = 1;
        /* FLOOR to get to few number of bands (next lower even number) */
        num_bands0 = (k2 - k0) & 254;
      } else {
        dk = 2;
        num_bands0 = ( ((k2 - k0) >> 1) + 1 ) & 254; /* ROUND to the closest fit */
      }

      if (num_bands0 < 1) {
        return SBRDEC_UNSUPPORTED_CONFIG;
        /* We must return already here because 'i' can become negative below. */
      }

      k2_achived = k0 + num_bands0*dk;
      k2_diff = k2 - k2_achived;

      for(i=0;i<num_bands0;i++)
        diff_tot[i] = dk;

      /* If linear scale wasn't achieved */
      /* and we got too wide SBR area */
      if (k2_diff < 0) {
          incr = 1;
          i = 0;
      }

      /* If linear scale wasn't achieved */
      /* and we got too small SBR area */
      if (k2_diff > 0) {
          incr = -1;
          i = num_bands0-1;
      }

      /* Adjust diff vector to get sepc. SBR range */
      while (k2_diff != 0) {
        diff_tot[i] = diff_tot[i] - incr;
        i = i + incr;
        k2_diff = k2_diff + incr;
      }

      cumSum(k0, diff_tot, num_bands0, v_k_master);/* cumsum */
    *numMaster = num_bands0;  /* Output nr of bands */
  }

  if (*numMaster < 1) {
    return SBRDEC_UNSUPPORTED_CONFIG;
  }


  /*
    Print out the calculated table
  */

  return SBRDEC_OK;
}


/*!
  \brief     Calculate frequency ratio of one SBR band

  All SBR bands should span a constant frequency range in the logarithmic
  domain. This function calculates the ratio of any SBR band's upper and lower
  frequency.

 \return    num_band-th root of k_start/k_stop
*/
static FIXP_SGL calcFactorPerBand(int k_start, int k_stop, int num_bands)
{
/* Scaled bandfactor and step 1 bit right to avoid overflow
 * use double data type */
  FIXP_DBL bandfactor = FL2FXCONST_DBL(0.25f); /* Start value */
  FIXP_DBL step = FL2FXCONST_DBL(0.125f);      /* Initial increment for factor */

  int    direction = 1;

/* Because saturation can't be done in INT IIS,
 * changed start and stop data type from FIXP_SGL to FIXP_DBL */
  FIXP_DBL start = k_start << (DFRACT_BITS-8);
  FIXP_DBL stop = k_stop << (DFRACT_BITS-8);

  FIXP_DBL temp;

  int   j, i=0;

  while ( step > FL2FXCONST_DBL(0.0f)) {
    i++;
    temp = stop;

    /* Calculate temp^num_bands: */
    for (j=0; j<num_bands; j++)
      //temp = fMult(temp,bandfactor);
      temp = fMultDiv2(temp,bandfactor)<<2;

    if (temp<start) { /* Factor too strong, make it weaker */
      if (direction == 0)
        /* Halfen step. Right shift is not done as fract because otherwise the
           lowest bit cannot be cleared due to rounding */
        step = (FIXP_DBL)((LONG)step >> 1);
      direction = 1;
      bandfactor = bandfactor + step;
    }
    else {  /* Factor is too weak: make it stronger */
      if (direction == 1)
        step = (FIXP_DBL)((LONG)step >> 1);
      direction = 0;
      bandfactor = bandfactor - step;
    }

    if (i>100) {
      step = FL2FXCONST_DBL(0.0f);
    }
  }
  return FX_DBL2FX_SGL(bandfactor<<1);
}


/*!
  \brief     Calculate number of SBR bands between start and stop band

  Given the number of bands per octave, this function calculates how many
  bands fit in the given frequency range.
  When the warpFlag is set, the 'band density' is decreased by a factor
  of 1/1.3

  \return    number of bands
*/
static int
numberOfBands(FIXP_SGL bpo_div16, /*!< Input: number of bands per octave divided by 16 */
              int    start,     /*!< First QMF band of SBR frequency range */
              int    stop,      /*!< Last QMF band of SBR frequency range + 1 */
              int    warpFlag)  /*!< Stretching flag */
{
  FIXP_SGL num_bands_div128;
  int    num_bands;

  num_bands_div128 = FX_DBL2FX_SGL(fMult(FDK_getNumOctavesDiv8(start,stop),bpo_div16));

  if (warpFlag) {
    /* Apply the warp factor of 1.3 to get wider bands.  We use a value
       of 32768/25200 instead of the exact value to avoid critical cases
       of rounding.
    */
    num_bands_div128 = FX_DBL2FX_SGL(fMult(num_bands_div128, FL2FXCONST_SGL(25200.0/32768.0)));
  }

  /* add scaled 1 for rounding to even numbers: */
  num_bands_div128 = num_bands_div128 + FL2FXCONST_SGL( 1.0f/128.0f );
  /* scale back to right aligned integer and double the value: */
  num_bands = 2 * ((LONG)num_bands_div128 >> (FRACT_BITS - 7));

  return(num_bands);
}


/*!
  \brief     Calculate width of SBR bands

  Given the desired number of bands within the SBR frequency range,
  this function calculates the width of each SBR band in QMF channels.
  The bands get wider from start to stop (bark scale).
*/
static void
CalcBands(UCHAR * diff,    /*!< Vector of widths to be calculated */
          UCHAR start,     /*!< Lower end of subband range */
          UCHAR stop,      /*!< Upper end of subband range */
          UCHAR num_bands) /*!< Desired number of bands */
{
  int i;
  int previous;
  int current;
  FIXP_SGL exact, temp;
  FIXP_SGL bandfactor = calcFactorPerBand(start, stop, num_bands);

  previous = stop; /* Start with highest QMF channel */
  exact = (FIXP_SGL)(stop << (FRACT_BITS-8)); /* Shift left to gain some accuracy */

  for(i=num_bands-1; i>=0; i--) {
    /* Calculate border of next lower sbr band */
    exact = FX_DBL2FX_SGL(fMult(exact,bandfactor));

    /* Add scaled 0.5 for rounding:
       We use a value 128/256 instead of 0.5 to avoid some critical cases of rounding. */
    temp = exact +  FL2FXCONST_SGL(128.0/32768.0);

    /* scale back to right alinged integer: */
    current = (LONG)temp >> (FRACT_BITS-8);

    /* Save width of band i */
    diff[i] = previous - current;
    previous = current;
  }
}


/*!
  \brief     Calculate cumulated sum vector from delta vector
*/
static void
cumSum(UCHAR start_value, UCHAR* diff, UCHAR length, UCHAR *start_adress)
{
  int i;
  start_adress[0]=start_value;
  for(i=1; i<=length; i++)
    start_adress[i] = start_adress[i-1] + diff[i-1];
}


/*!
  \brief     Adapt width of frequency bands in the second region

  If SBR spans more than 2 octaves, the upper part of a bark-frequency-scale
  is calculated separately. This function tries to avoid that the second region
  starts with a band smaller than the highest band of the first region.
*/
static SBR_ERROR
modifyBands(UCHAR max_band_previous, UCHAR * diff, UCHAR length)
{
  int change = max_band_previous - diff[0];

  /* Limit the change so that the last band cannot get narrower than the first one */
  if ( change > (diff[length-1]-diff[0])>>1 )
    change = (diff[length-1]-diff[0])>>1;

  diff[0] += change;
  diff[length-1] -= change;
  shellsort(diff, length);

  return SBRDEC_OK;
}


/*!
  \brief   Update high resolution frequency band table
*/
static void
sbrdecUpdateHiRes(UCHAR * h_hires,
                  UCHAR * num_hires,
                  UCHAR * v_k_master,
                  UCHAR num_bands,
                  UCHAR xover_band)
{
  UCHAR i;

  *num_hires = num_bands-xover_band;

  for(i=xover_band; i<=num_bands; i++) {
    h_hires[i-xover_band] = v_k_master[i];
  }
}


/*!
  \brief  Build low resolution table out of high resolution table
*/
static void
sbrdecUpdateLoRes(UCHAR * h_lores,
                  UCHAR * num_lores,
                  UCHAR * h_hires,
                  UCHAR num_hires)
{
  UCHAR i;

  if( (num_hires & 1) == 0) {
    /* If even number of hires bands */
    *num_lores = num_hires >> 1;
    /* Use every second lores=hires[0,2,4...] */
    for(i=0; i<=*num_lores; i++)
      h_lores[i] = h_hires[i*2];
  }
  else {
    /* Odd number of hires, which means xover is odd */
    *num_lores = (num_hires+1) >> 1;
    /* Use lores=hires[0,1,3,5 ...] */
    h_lores[0] = h_hires[0];
    for(i=1; i<=*num_lores; i++) {
      h_lores[i] = h_hires[i*2-1];
    }
  }
}


/*!
  \brief   Derive a low-resolution frequency-table from the master frequency table
*/
void
sbrdecDownSampleLoRes(UCHAR *v_result,
                      UCHAR num_result,
                      UCHAR *freqBandTableRef,
                      UCHAR num_Ref)
{
  int step;
  int i,j;
  int org_length,result_length;
  int v_index[MAX_FREQ_COEFFS>>1];

  /* 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;
    org_length = org_length - step;
    result_length--;
    v_index[i] = v_index[i-1] + step;
  }

  for(j=0;j<=i;j++) {
    /* Use downsample vector to index LoResolution vector */
    v_result[j]=freqBandTableRef[v_index[j]];
  }

}


/*!
  \brief   Sorting routine
*/
void shellsort(UCHAR *in, UCHAR n)
{

  int i, j, v, w;
  int inc = 1;

  do
    inc = 3 * inc + 1;
  while (inc <= n);

  do {
    inc = inc / 3;
    for (i = inc; i < n; i++) {
      v = in[i];
      j = i;
      while ((w=in[j-inc]) > v) {
        in[j] = w;
        j -= inc;
        if (j < inc)
          break;
      }
      in[j] = v;
    }
  } while (inc > 1);

}


/*!
  \brief   Reset frequency band tables
  \return  errorCode, 0 if successful
*/
SBR_ERROR
resetFreqBandTables(HANDLE_SBR_HEADER_DATA hHeaderData, const UINT flags)
{
  SBR_ERROR err = SBRDEC_OK;
  int k2,kx, lsb, usb;
  int     intTemp;
  UCHAR    nBandsLo, nBandsHi;
  HANDLE_FREQ_BAND_DATA hFreq = &hHeaderData->freqBandData;

  /* Calculate master frequency function */
  err = sbrdecUpdateFreqScale(hFreq->v_k_master,
                              &hFreq->numMaster,
                              hHeaderData->sbrProcSmplRate,
                              hHeaderData,
                              flags);

  if ( err || (hHeaderData->bs_info.xover_band > hFreq->numMaster) ) {
    return SBRDEC_UNSUPPORTED_CONFIG;
  }

  /* Derive Hiresolution from master frequency function */
  sbrdecUpdateHiRes(hFreq->freqBandTable[1], &nBandsHi, hFreq->v_k_master, hFreq->numMaster, hHeaderData->bs_info.xover_band );
  /* Derive Loresolution from Hiresolution */
  sbrdecUpdateLoRes(hFreq->freqBandTable[0], &nBandsLo, hFreq->freqBandTable[1], nBandsHi);


  hFreq->nSfb[0] = nBandsLo;
  hFreq->nSfb[1] = nBandsHi;

  /* Check index to freqBandTable[0] */
  if ( !(nBandsLo > 0) || (nBandsLo > (MAX_FREQ_COEFFS>>1)) ) {
    return SBRDEC_UNSUPPORTED_CONFIG;
  }

  lsb = hFreq->freqBandTable[0][0];
  usb = hFreq->freqBandTable[0][nBandsLo];

  /* Additional check for lsb */
  if ( (lsb > (32)) || (lsb >= usb) ) {
    return SBRDEC_UNSUPPORTED_CONFIG;
  }


  /* Calculate number of noise bands */

  k2 = hFreq->freqBandTable[1][nBandsHi];
  kx = hFreq->freqBandTable[1][0];

  if (hHeaderData->bs_data.noise_bands == 0)
  {
    hFreq->nNfb = 1;
  }
  else /* Calculate no of noise bands 1,2 or 3 bands/octave */
  {
    /* Fetch number of octaves divided by 32 */
    intTemp = (LONG)FDK_getNumOctavesDiv8(kx,k2) >> 2;

    /* Integer-Multiplication with number of bands: */
    intTemp = intTemp * hHeaderData->bs_data.noise_bands;

    /* Add scaled 0.5 for rounding: */
    intTemp = intTemp + (LONG)FL2FXCONST_SGL(0.5f/32.0f);

    /* Convert to right-aligned integer: */
    intTemp = intTemp >> (FRACT_BITS - 1 /*sign*/ - 5 /* rescale */);

    /* Compare with float calculation */
    FDK_ASSERT( intTemp ==  (int)((hHeaderData->bs_data.noise_bands * FDKlog( (float)k2/kx) / (float)(FDKlog(2.0)))+0.5) );

    if( intTemp==0)
      intTemp=1;

    hFreq->nNfb = intTemp;
  }

  hFreq->nInvfBands = hFreq->nNfb;

  if( hFreq->nNfb > MAX_NOISE_COEFFS ) {
    return SBRDEC_UNSUPPORTED_CONFIG;
  }

  /* Get noise bands */
  sbrdecDownSampleLoRes(hFreq->freqBandTableNoise,
                        hFreq->nNfb,
                        hFreq->freqBandTable[0],
                        nBandsLo);


  hFreq->lowSubband  = lsb;
  hFreq->highSubband = usb;

  return SBRDEC_OK;
}