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

© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
  All rights reserved.

 1.    INTRODUCTION
The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
This FDK AAC Codec software is intended to be used on a wide variety of Android devices.

AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
of the MPEG specifications.

Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
individually for the purpose of encoding or decoding bit streams in products that are compliant with
the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
software may already be covered under those patent licenses when it is used for those licensed purposes only.

Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
applications information and documentation.

2.    COPYRIGHT LICENSE

Redistribution and use in source and binary forms, with or without modification, are permitted without
payment of copyright license fees provided that you satisfy the following conditions:

You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
your modifications thereto in source code form.

You must retain the complete text of this software license in the documentation and/or other materials
provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
modifications thereto to recipients of copies in binary form.

The name of Fraunhofer may not be used to endorse or promote products derived from this library without
prior written permission.

You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
software or your modifications thereto.

Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
and the date of any change. For modified versions of the FDK AAC Codec, the term
"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."

3.    NO PATENT LICENSE

NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
respect to this software.

You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
by appropriate patent licenses.

4.    DISCLAIMER

This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
or business interruption, however caused and on any theory of liability, whether in contract, strict
liability, or tort (including negligence), arising in any way out of the use of this software, even if
advised of the possibility of such damage.

5.    CONTACT INFORMATION

Fraunhofer Institute for Integrated Circuits IIS
Attention: Audio and Multimedia Departments - FDK AAC LL
Am Wolfsmantel 33
91058 Erlangen, Germany

www.iis.fraunhofer.de/amm
amm-info@iis.fraunhofer.de
----------------------------------------------------------------------------------------------------------- */

/*!
  \file
  \brief  Envelope calculation  

  The envelope adjustor compares the energies present in the transposed
  highband to the reference energies conveyed with the bitstream.
  The highband is amplified (sometimes) or attenuated (mostly) to the
  desired level.

  The spectral shape of the reference energies can be changed several times per
  frame if necessary. Each set of energy values corresponding to a certain range
  in time will be called an <em>envelope</em> here.
  The bitstream supports several frequency scales and two resolutions. Normally,
  one or more QMF-subbands are grouped to one SBR-band. An envelope contains
  reference energies for each SBR-band.
  In addition to the energy envelopes, noise envelopes are transmitted that
  define the ratio of energy which is generated by adding noise instead of
  transposing the lowband. The noise envelopes are given in a coarser time
  and frequency resolution.
  If a signal contains strong tonal components, synthetic sines can be
  generated in individual SBR bands.

  An overlap buffer of 6 QMF-timeslots is used to allow a more
  flexible alignment of the envelopes in time that is not restricted to the
  core codec's frame borders.
  Therefore the envelope adjustor has access to the spectral data of the
  current frame as well as the last 6 QMF-timeslots of the previous frame.
  However, in average only the data of 1 frame is being processed as
  the adjustor is called once per frame.

  Depending on the frequency range set in the bitstream, only QMF-subbands between
  <em>lowSubband</em> and <em>highSubband</em> are adjusted.

  Scaling of spectral data to maximize SNR (see #QMF_SCALE_FACTOR) as well as a special Mantissa-Exponent format
  ( see  calculateSbrEnvelope() ) are being used. The main entry point for this modules is calculateSbrEnvelope().

  \sa sbr_scale.h, #QMF_SCALE_FACTOR, calculateSbrEnvelope(), \ref documentationOverview
*/


#include "env_calc.h"

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

#include "genericStds.h"           /* need FDKpow() for debug outputs */

#if defined(__arm__)
#include "arm/env_calc_arm.cpp"
#endif

typedef struct
{
    FIXP_DBL nrgRef[MAX_FREQ_COEFFS];
    FIXP_DBL nrgEst[MAX_FREQ_COEFFS];
    FIXP_DBL nrgGain[MAX_FREQ_COEFFS];
    FIXP_DBL noiseLevel[MAX_FREQ_COEFFS];
    FIXP_DBL nrgSine[MAX_FREQ_COEFFS];

    SCHAR   nrgRef_e[MAX_FREQ_COEFFS];
    SCHAR   nrgEst_e[MAX_FREQ_COEFFS];
    SCHAR   nrgGain_e[MAX_FREQ_COEFFS];
    SCHAR   noiseLevel_e[MAX_FREQ_COEFFS];
    SCHAR   nrgSine_e[MAX_FREQ_COEFFS];
}
ENV_CALC_NRGS;

/*static*/ void equalizeFiltBufferExp(FIXP_DBL *filtBuffer,
                                  SCHAR   *filtBuffer_e,
                                  FIXP_DBL *NrgGain,
                                  SCHAR   *NrgGain_e,
                                  int    subbands);

/*static*/ void calcNrgPerSubband(FIXP_DBL  **analysBufferReal,
                              FIXP_DBL  **analysBufferImag,
                              int       lowSubband, int highSubband,
                              int       start_pos,  int next_pos,
                              SCHAR     frameExp,
                              FIXP_DBL *nrgEst,
                              SCHAR    *nrgEst_e );

/*static*/ void calcNrgPerSfb(FIXP_DBL  **analysBufferReal,
                          FIXP_DBL  **analysBufferImag,
                          int       nSfb,
                          UCHAR    *freqBandTable,
                          int       start_pos,  int next_pos,
                          SCHAR     input_e,
                          FIXP_DBL *nrg_est,
                          SCHAR    *nrg_est_e );

/*static*/ void calcSubbandGain(FIXP_DBL  nrgRef, SCHAR nrgRef_e, ENV_CALC_NRGS* nrgs, int c,
                            FIXP_DBL  tmpNoise, SCHAR tmpNoise_e,
                            UCHAR     sinePresentFlag,
                            UCHAR     sineMapped,
                            int       noNoiseFlag);

/*static*/ void calcAvgGain(ENV_CALC_NRGS* nrgs,
                        int        lowSubband,
                        int        highSubband,
                        FIXP_DBL  *sumRef_m,
                        SCHAR     *sumRef_e,
                        FIXP_DBL  *ptrAvgGain_m,
                        SCHAR     *ptrAvgGain_e);

/*static*/ void adjustTimeSlotLC(FIXP_DBL  *ptrReal,
                           ENV_CALC_NRGS* nrgs,
                           UCHAR *ptrHarmIndex,
                           int    lowSubbands,
                           int    noSubbands,
                           int    scale_change,
                           int    noNoiseFlag,
                           int   *ptrPhaseIndex,
                           int    fCldfb);
/*static*/ void adjustTimeSlotHQ(FIXP_DBL  *ptrReal,
                           FIXP_DBL  *ptrImag,
                           HANDLE_SBR_CALCULATE_ENVELOPE h_sbr_cal_env,
                           ENV_CALC_NRGS* nrgs,
                           int    lowSubbands,
                           int    noSubbands,
                           int    scale_change,
                           FIXP_SGL smooth_ratio,
                           int    noNoiseFlag,
                           int    filtBufferNoiseShift);


/*!
  \brief     Map sine flags from bitstream to QMF bands

  The bitstream carries only 1 sine flag per band and frame.
  This function maps every sine flag from the bitstream to a specific QMF subband
  and to a specific envelope where the sine shall start.
  The result is stored in the vector sineMapped which contains one entry per
  QMF subband. The value of an entry specifies the envelope where a sine
  shall start. A value of #MAX_ENVELOPES indicates that no sine is present
  in the subband.
  The missing harmonics flags from the previous frame (harmFlagsPrev) determine
  if a sine starts at the beginning of the frame or at the transient position.
  Additionally, the flags in harmFlagsPrev are being updated by this function
  for the next frame.
*/
/*static*/ void mapSineFlags(UCHAR *freqBandTable, /*!< Band borders (there's only 1 flag per band) */
                         int nSfb,                     /*!< Number of bands in the table */
                         UCHAR *addHarmonics,           /*!< vector with 1 flag per sfb */
                         int *harmFlagsPrev,           /*!< Packed 'addHarmonics' */
                         int tranEnv,                  /*!< Transient position */
                         SCHAR *sineMapped)            /*!< Resulting vector of sine start positions for each QMF band */

{
  int i;
  int lowSubband2 = freqBandTable[0]<<1;
  int bitcount = 0;
  int oldflags = *harmFlagsPrev;
  int newflags = 0;

  /*
    Format of harmFlagsPrev:

    first word = flags for highest 16 sfb bands in use
    second word = flags for next lower 16 sfb bands (if present)
    third word = flags for lowest 16 sfb bands (if present)

    Up to MAX_FREQ_COEFFS sfb bands can be flagged for a sign.
    The lowest bit of the first word corresponds to the _highest_ sfb band in use.
    This is ensures that each flag is  mapped to the same QMF band even after a
    change of the crossover-frequency.
  */


  /* Reset the output vector first */
  FDKmemset(sineMapped, MAX_ENVELOPES,MAX_FREQ_COEFFS); /* MAX_ENVELOPES means 'no sine' */

  freqBandTable += nSfb;
  addHarmonics  += nSfb-1;

  for (i=nSfb; i!=0; i--) {
    int ui = *freqBandTable--;                 /* Upper limit of the current scale factor band. */
    int li = *freqBandTable;                   /* Lower limit of the current scale factor band. */

    if ( *addHarmonics-- ) {                   /* There is a sine in this band */

      unsigned int mask = 1 << bitcount;
      newflags |= mask;                        /* Set flag */

      /*
        If there was a sine in the last frame, let it continue from the first envelope on
        else start at the transient position.
      */
      sineMapped[(ui+li-lowSubband2) >> 1] = ( oldflags & mask ) ? 0 : tranEnv;
    }

    if ((++bitcount == 16) || i==1) {
      bitcount = 0;
      *harmFlagsPrev++ = newflags;
      oldflags = *harmFlagsPrev;               /* Fetch 16 of the old flags */
      newflags = 0;
    }
  }
}


/*!
  \brief     Reduce gain-adjustment induced aliasing for real valued filterbank.
*/
/*static*/ void
aliasingReduction(FIXP_DBL* degreeAlias,       /*!< estimated aliasing for each QMF channel */
                  ENV_CALC_NRGS* nrgs,
                  int*      useAliasReduction, /*!< synthetic sine engergy for each subband, used as flag */
                  int       noSubbands)        /*!< number of QMF channels to process */
{
  FIXP_DBL* nrgGain   = nrgs->nrgGain;          /*!< subband gains to be modified */
  SCHAR*    nrgGain_e = nrgs->nrgGain_e;        /*!< subband gains to be modified (exponents) */
  FIXP_DBL* nrgEst    = nrgs->nrgEst;           /*!< subband energy before amplification */
  SCHAR*    nrgEst_e  = nrgs->nrgEst_e;         /*!< subband energy before amplification (exponents) */
  int grouping = 0, index = 0, noGroups, k;
  int groupVector[MAX_FREQ_COEFFS];

  /* Calculate grouping*/
  for (k = 0; k < noSubbands-1; k++ ){
    if ( (degreeAlias[k + 1] != FL2FXCONST_DBL(0.0f)) && useAliasReduction[k] ) {
      if(grouping==0){
        groupVector[index++] = k;
        grouping = 1;
      }
      else{
        if(groupVector[index-1] + 3 == k){
          groupVector[index++] = k + 1;
          grouping = 0;
        }
      }
    }
    else{
      if(grouping){
        if(useAliasReduction[k])
          groupVector[index++] = k + 1;
        else
          groupVector[index++] = k;
        grouping = 0;
      }
    }
  }

  if(grouping){
    groupVector[index++] = noSubbands;
  }
  noGroups = index >> 1;


  /*Calculate new gain*/
  for (int group = 0; group < noGroups; group ++) {
    FIXP_DBL nrgOrig = FL2FXCONST_DBL(0.0f);    /* Original signal energy in current group of bands */
    SCHAR    nrgOrig_e = 0;
    FIXP_DBL nrgAmp = FL2FXCONST_DBL(0.0f);     /* Amplified signal energy in group (using current gains) */
    SCHAR    nrgAmp_e = 0;
    FIXP_DBL nrgMod = FL2FXCONST_DBL(0.0f);   /* Signal energy in group when applying modified gains */
    SCHAR    nrgMod_e = 0;
    FIXP_DBL groupGain;         /* Total energy gain in group */
    SCHAR    groupGain_e;
    FIXP_DBL compensation;      /* Compensation factor for the energy change when applying modified gains */
    SCHAR    compensation_e;

    int startGroup = groupVector[2*group];
    int stopGroup  = groupVector[2*group+1];

    /* Calculate total energy in group before and after amplification with current gains: */
    for(k = startGroup; k < stopGroup; k++){
      /* Get original band energy */
      FIXP_DBL tmp = nrgEst[k];
      SCHAR    tmp_e = nrgEst_e[k];

      FDK_add_MantExp(tmp, tmp_e, nrgOrig, nrgOrig_e, &nrgOrig, &nrgOrig_e);

      /* Multiply band energy with current gain */
      tmp = fMult(tmp,nrgGain[k]);
      tmp_e = tmp_e + nrgGain_e[k];

      FDK_add_MantExp(tmp, tmp_e, nrgAmp, nrgAmp_e, &nrgAmp, &nrgAmp_e);
    }

    /* Calculate total energy gain in group */
    FDK_divide_MantExp(nrgAmp, nrgAmp_e,
                       nrgOrig, nrgOrig_e,
                       &groupGain, &groupGain_e);

    for(k = startGroup; k < stopGroup; k++){
      FIXP_DBL tmp;
      SCHAR    tmp_e;

      FIXP_DBL alpha = degreeAlias[k];
      if (k < noSubbands - 1) {
        if (degreeAlias[k + 1] > alpha)
          alpha = degreeAlias[k + 1];
      }

      /* Modify gain depending on the degree of aliasing */
      FDK_add_MantExp( fMult(alpha,groupGain), groupGain_e,
                       fMult(/*FL2FXCONST_DBL(1.0f)*/ (FIXP_DBL)MAXVAL_DBL - alpha,nrgGain[k]), nrgGain_e[k],
                       &nrgGain[k], &nrgGain_e[k] );

      /* Apply modified gain to original energy */
      tmp = fMult(nrgGain[k],nrgEst[k]);
      tmp_e = nrgGain_e[k] + nrgEst_e[k];

      /* Accumulate energy with modified gains applied */
      FDK_add_MantExp( tmp, tmp_e,
                       nrgMod, nrgMod_e,
                       &nrgMod, &nrgMod_e );
    }

    /* Calculate compensation factor to retain the energy of the amplified signal */
    FDK_divide_MantExp(nrgAmp, nrgAmp_e,
                       nrgMod, nrgMod_e,
                       &compensation, &compensation_e);

    /* Apply compensation factor to all gains of the group */
    for(k = startGroup; k < stopGroup; k++){
      nrgGain[k] = fMult(nrgGain[k],compensation);
      nrgGain_e[k] = nrgGain_e[k] + compensation_e;
    }
  }
}


 /* Convert headroom bits to exponent */
#define SCALE2EXP(s) (15-(s))
#define EXP2SCALE(e) (15-(e))

/*!
  \brief  Apply spectral envelope to subband samples

  This function is called from sbr_dec.cpp in each frame.

  To enhance accuracy and due to the usage of tables for squareroots and
  inverse, some calculations are performed with the operands being split
  into mantissa and exponent. The variable names in the source code carry
  the suffixes <em>_m</em> and  <em>_e</em> respectively. The control data
  in #hFrameData containts envelope data which is represented by this format but
  stored in single words. (See requantizeEnvelopeData() for details). This data
  is unpacked within calculateSbrEnvelope() to follow the described suffix convention.

  The actual value (comparable to the corresponding float-variable in the
  research-implementation) of a mantissa/exponent-pair can be calculated as

  \f$ value = value\_m * 2^{value\_e} \f$

  All energies and noise levels decoded from the bitstream suit for an
  original signal magnitude of \f$\pm 32768 \f$ rather than \f$ \pm 1\f$. Therefore,
  the scale factor <em>hb_scale</em> passed into this function will be converted
  to an 'input exponent' (#input_e), which fits the internal representation.

  Before the actual processing, an exponent #adj_e for resulting adjusted
  samples is derived from the maximum reference energy.

  Then, for each envelope, the following steps are performed:

  \li Calculate energy in the signal to be adjusted. Depending on the the value of
      #interpolFreq (interpolation mode), this is either done seperately
      for each QMF-subband or for each SBR-band.
      The resulting energies are stored in #nrgEst_m[#MAX_FREQ_COEFFS] (mantissas)
      and #nrgEst_e[#MAX_FREQ_COEFFS] (exponents).
  \li Calculate gain and noise level for each subband:<br>
      \f$ gain  = \sqrt{ \frac{nrgRef}{nrgEst} \cdot (1 - noiseRatio) }
          \hspace{2cm}
          noise = \sqrt{ nrgRef \cdot noiseRatio }
      \f$<br>
      where <em>noiseRatio</em> and <em>nrgRef</em> are extracted from the
      bitstream and <em>nrgEst</em> is the subband energy before adjustment.
      The resulting gains are stored in #nrgGain_m[#MAX_FREQ_COEFFS]
      (mantissas) and #nrgGain_e[#MAX_FREQ_COEFFS] (exponents), the noise levels
      are stored in #noiseLevel_m[#MAX_FREQ_COEFFS] and #noiseLevel_e[#MAX_FREQ_COEFFS]
      (exponents).
      The sine levels are stored in #nrgSine_m[#MAX_FREQ_COEFFS]
      and #nrgSine_e[#MAX_FREQ_COEFFS].
  \li Noise limiting: The gain for each subband is limited both absolutely
      and relatively compared to the total gain over all subbands.
  \li Boost gain: Calculate and apply boost factor for each limiter band
      in order to compensate for the energy loss imposed by the limiting.
  \li Apply gains and add noise: The gains and noise levels are applied
      to all timeslots of the current envelope. A short FIR-filter (length 4
      QMF-timeslots) can be used to smooth the sudden change at the envelope borders.
      Each complex subband sample of the current timeslot is multiplied by the
      smoothed gain, then random noise with the calculated level is added.

  \note
  To reduce the stack size, some of the local arrays could be located within
  the time output buffer. Of the 512 samples temporarily available there,
  about half the size is already used by #SBR_FRAME_DATA. A pointer to the
  remaining free memory could be supplied by an additional argument to calculateSbrEnvelope()
  in sbr_dec:

  \par
  \code
    calculateSbrEnvelope (&hSbrDec->sbrScaleFactor,
                          &hSbrDec->SbrCalculateEnvelope,
                          hHeaderData,
                          hFrameData,
                          QmfBufferReal,
                          QmfBufferImag,
                          timeOutPtr + sizeof(SBR_FRAME_DATA)/sizeof(Float) + 1);
  \endcode

  \par
  Within calculateSbrEnvelope(), some pointers could be defined instead of the arrays
  #nrgRef_m, #nrgRef_e, #nrgEst_m, #nrgEst_e, #noiseLevel_m:

  \par
  \code
    fract*        nrgRef_m = timeOutPtr;
    SCHAR*        nrgRef_e = nrgRef_m + MAX_FREQ_COEFFS;
    fract*        nrgEst_m = nrgRef_e + MAX_FREQ_COEFFS;
    SCHAR*        nrgEst_e = nrgEst_m + MAX_FREQ_COEFFS;
    fract*        noiseLevel_m = nrgEst_e + MAX_FREQ_COEFFS;
  \endcode

  <br>
*/
void
calculateSbrEnvelope (QMF_SCALE_FACTOR  *sbrScaleFactor,           /*!< Scaling factors */
                      HANDLE_SBR_CALCULATE_ENVELOPE h_sbr_cal_env, /*!< Handle to struct filled by the create-function */
                      HANDLE_SBR_HEADER_DATA hHeaderData,          /*!< Static control data */
                      HANDLE_SBR_FRAME_DATA  hFrameData,           /*!< Control data of current frame */
                      FIXP_DBL **analysBufferReal,                 /*!< Real part of subband samples to be processed */
                      FIXP_DBL **analysBufferImag,                 /*!< Imag part of subband samples to be processed */
                      const int useLP,
                      FIXP_DBL *degreeAlias,                       /*!< Estimated aliasing for each QMF channel */
                      const UINT flags,
                      const int frameErrorFlag
                      )
{
  int c, i, j, envNoise = 0;
  UCHAR*   borders = hFrameData->frameInfo.borders;

  FIXP_SGL *noiseLevels       = hFrameData->sbrNoiseFloorLevel;
  HANDLE_FREQ_BAND_DATA hFreq = &hHeaderData->freqBandData;

  int lowSubband  = hFreq->lowSubband;
  int highSubband = hFreq->highSubband;
  int noSubbands  = highSubband - lowSubband;

  int    noNoiseBands = hFreq->nNfb;
  int    no_cols      = hHeaderData->numberTimeSlots * hHeaderData->timeStep;
  UCHAR  first_start  = borders[0] * hHeaderData->timeStep;

  SCHAR  sineMapped[MAX_FREQ_COEFFS];
  SCHAR  ov_adj_e = SCALE2EXP(sbrScaleFactor->ov_hb_scale);
  SCHAR  adj_e = 0;
  SCHAR  output_e;
  SCHAR  final_e = 0;

  SCHAR  maxGainLimit_e = (frameErrorFlag) ? MAX_GAIN_CONCEAL_EXP : MAX_GAIN_EXP;

  int useAliasReduction[64];
  UCHAR smooth_length = 0;

  FIXP_SGL * pIenv = hFrameData->iEnvelope;

  /*
    Extract sine flags for all QMF bands
  */
  mapSineFlags(hFreq->freqBandTable[1],
               hFreq->nSfb[1],
               hFrameData->addHarmonics,
               h_sbr_cal_env->harmFlagsPrev,
               hFrameData->frameInfo.tranEnv,
               sineMapped);


  /*
    Scan for maximum in bufferd noise levels.
    This is needed in case that we had strong noise in the previous frame
    which is smoothed into the current frame.
    The resulting exponent is used as start value for the maximum search
    in reference energies
  */
  if (!useLP)
    adj_e = h_sbr_cal_env->filtBufferNoise_e - getScalefactor(h_sbr_cal_env->filtBufferNoise, noSubbands);

  /*
    Scan for maximum reference energy to be able
    to select appropriate values for adj_e and final_e.
  */

  for (i = 0; i < hFrameData->frameInfo.nEnvelopes; i++) {
    INT maxSfbNrg_e = -FRACT_BITS+NRG_EXP_OFFSET; /* start value for maximum search */

    /* Fetch frequency resolution for current envelope: */
    for (j=hFreq->nSfb[hFrameData->frameInfo.freqRes[i]]; j!=0; j--) {
      maxSfbNrg_e = fixMax(maxSfbNrg_e,(INT)((LONG)(*pIenv++) & MASK_E));
    }
    maxSfbNrg_e -= NRG_EXP_OFFSET;

    /* Energy -> magnitude (sqrt halfens exponent) */
    maxSfbNrg_e = (maxSfbNrg_e+1) >> 1;  /* +1 to go safe (round to next higher int) */

    /* Some safety margin is needed for 2 reasons:
       - The signal energy is not equally spread over all subband samples in
         a specific sfb of an envelope (Nrg could be too high by a factor of
         envWidth * sfbWidth)
       - Smoothing can smear high gains of the previous envelope into the current
    */
    maxSfbNrg_e += 6;

    if (borders[i] < hHeaderData->numberTimeSlots)
      /* This envelope affects timeslots that belong to the output frame */
      adj_e = (maxSfbNrg_e > adj_e) ? maxSfbNrg_e : adj_e;

    if (borders[i+1] > hHeaderData->numberTimeSlots)
      /* This envelope affects timeslots after the output frame */
      final_e =  (maxSfbNrg_e > final_e) ? maxSfbNrg_e : final_e;

  }

  /*
    Calculate adjustment factors and apply them for every envelope.
  */
  pIenv = hFrameData->iEnvelope;

  for (i = 0; i < hFrameData->frameInfo.nEnvelopes; i++) {

    int k, noNoiseFlag;
    SCHAR  noise_e, input_e = SCALE2EXP(sbrScaleFactor->hb_scale);
    C_ALLOC_SCRATCH_START(pNrgs, ENV_CALC_NRGS, 1);

    /*
      Helper variables.
    */
    UCHAR start_pos = hHeaderData->timeStep * borders[i];  /* Start-position in time (subband sample) for current envelope. */
    UCHAR stop_pos = hHeaderData->timeStep * borders[i+1]; /* Stop-position in time (subband sample) for current envelope. */
    UCHAR freq_res = hFrameData->frameInfo.freqRes[i];     /* Frequency resolution for current envelope. */


    /* Always do fully initialize the temporary energy table. This prevents negative energies and extreme gain factors in
       cases where the number of limiter bands exceeds the number of subbands. The latter can be caused by undetected bit
       errors and is tested by some streams from the certification set. */
    FDKmemclear(pNrgs, sizeof(ENV_CALC_NRGS));

    /* If the start-pos of the current envelope equals the stop pos of the current
       noise envelope, increase the pointer (i.e. choose the next noise-floor).*/
    if (borders[i] == hFrameData->frameInfo.bordersNoise[envNoise+1]){
      noiseLevels += noNoiseBands;   /* The noise floor data is stored in a row [noiseFloor1 noiseFloor2...].*/
      envNoise++;
    }

    if(i==hFrameData->frameInfo.tranEnv || i==h_sbr_cal_env->prevTranEnv) /* attack */
    {
      noNoiseFlag = 1;
      if (!useLP)
        smooth_length = 0;  /* No smoothing on attacks! */
    }
    else {
      noNoiseFlag = 0;
      if (!useLP)
        smooth_length = (1 - hHeaderData->bs_data.smoothingLength) << 2;  /* can become either 0 or 4 */
    }


    /*
      Energy estimation in transposed highband.
    */
    if (hHeaderData->bs_data.interpolFreq)
      calcNrgPerSubband(analysBufferReal,
                        (useLP) ? NULL : analysBufferImag,
                        lowSubband, highSubband,
                        start_pos, stop_pos,
                        input_e,
                        pNrgs->nrgEst,
                        pNrgs->nrgEst_e);
    else
      calcNrgPerSfb(analysBufferReal,
                    (useLP) ? NULL : analysBufferImag,
                    hFreq->nSfb[freq_res],
                    hFreq->freqBandTable[freq_res],
                    start_pos, stop_pos,
                    input_e,
                    pNrgs->nrgEst,
                    pNrgs->nrgEst_e);

    /*
      Calculate subband gains
    */
    {
      UCHAR * table = hFreq->freqBandTable[freq_res];
      UCHAR * pUiNoise = &hFreq->freqBandTableNoise[1]; /*! Upper limit of the current noise floor band. */

      FIXP_SGL * pNoiseLevels = noiseLevels;

      FIXP_DBL tmpNoise = FX_SGL2FX_DBL((FIXP_SGL)((LONG)(*pNoiseLevels) & MASK_M));
      SCHAR    tmpNoise_e = (UCHAR)((LONG)(*pNoiseLevels++) & MASK_E) - NOISE_EXP_OFFSET;

      int cc = 0;
      c = 0;
      for (j = 0; j < hFreq->nSfb[freq_res]; j++) {

        FIXP_DBL refNrg   = FX_SGL2FX_DBL((FIXP_SGL)((LONG)(*pIenv) & MASK_M));
        SCHAR    refNrg_e = (SCHAR)((LONG)(*pIenv) & MASK_E) - NRG_EXP_OFFSET;

        UCHAR sinePresentFlag = 0;
        int li = table[j];
        int ui = table[j+1];

        for (k=li; k<ui; k++) {
          sinePresentFlag |= (i >= sineMapped[cc]);
          cc++;
        }

        for (k=li; k<ui; k++) {
          if (k >= *pUiNoise) {
            tmpNoise = FX_SGL2FX_DBL((FIXP_SGL)((LONG)(*pNoiseLevels) & MASK_M));
            tmpNoise_e = (SCHAR)((LONG)(*pNoiseLevels++) & MASK_E) - NOISE_EXP_OFFSET;

            pUiNoise++;
          }

          FDK_ASSERT(k >= lowSubband);

          if (useLP)
            useAliasReduction[k-lowSubband] = !sinePresentFlag;

          pNrgs->nrgSine[c] = FL2FXCONST_DBL(0.0f);
          pNrgs->nrgSine_e[c] = 0;

          calcSubbandGain(refNrg, refNrg_e, pNrgs, c,
                          tmpNoise, tmpNoise_e,
                          sinePresentFlag, i >= sineMapped[c],
                          noNoiseFlag);

          pNrgs->nrgRef[c]   = refNrg;
          pNrgs->nrgRef_e[c] = refNrg_e;

          c++;
        }
        pIenv++;
      }
    }

    /*
      Noise limiting
    */

    for (c = 0; c < hFreq->noLimiterBands; c++) {

      FIXP_DBL sumRef, boostGain, maxGain;
      FIXP_DBL accu = FL2FXCONST_DBL(0.0f);
      SCHAR   sumRef_e, boostGain_e, maxGain_e, accu_e = 0;

      calcAvgGain(pNrgs,
                  hFreq->limiterBandTable[c], hFreq->limiterBandTable[c+1],
                  &sumRef, &sumRef_e,
                  &maxGain, &maxGain_e);

      /* Multiply maxGain with limiterGain: */
      maxGain = fMult(maxGain, FDK_sbrDecoder_sbr_limGains_m[hHeaderData->bs_data.limiterGains]);
      maxGain_e += FDK_sbrDecoder_sbr_limGains_e[hHeaderData->bs_data.limiterGains];

      /* Scale mantissa of MaxGain into range between 0.5 and 1: */
      if (maxGain == FL2FXCONST_DBL(0.0f))
        maxGain_e = -FRACT_BITS;
      else {
        SCHAR charTemp = CountLeadingBits(maxGain);
        maxGain_e -= charTemp;
        maxGain  <<= (int)charTemp;
      }

      if (maxGain_e >= maxGainLimit_e) { /* upper limit (e.g. 96 dB) */
        maxGain = FL2FXCONST_DBL(0.5f);
        maxGain_e = maxGainLimit_e;
      }


      /* Every subband gain is compared to the scaled "average gain"
         and limited if necessary: */
      for (k = hFreq->limiterBandTable[c]; k < hFreq->limiterBandTable[c+1]; k++) {
        if ( (pNrgs->nrgGain_e[k] > maxGain_e) || (pNrgs->nrgGain_e[k] == maxGain_e && pNrgs->nrgGain[k]>maxGain) ) {

          FIXP_DBL noiseAmp;
          SCHAR    noiseAmp_e;

          FDK_divide_MantExp(maxGain, maxGain_e, pNrgs->nrgGain[k], pNrgs->nrgGain_e[k], &noiseAmp, &noiseAmp_e);
          pNrgs->noiseLevel[k]    = fMult(pNrgs->noiseLevel[k],noiseAmp);
          pNrgs->noiseLevel_e[k] += noiseAmp_e;
          pNrgs->nrgGain[k]       = maxGain;
          pNrgs->nrgGain_e[k]     = maxGain_e;
        }
      }

      /* -- Boost gain
        Calculate and apply boost factor for each limiter band:
        1. Check how much energy would be present when using the limited gain
        2. Calculate boost factor by comparison with reference energy
        3. Apply boost factor to compensate for the energy loss due to limiting
      */
      for (k = hFreq->limiterBandTable[c]; k < hFreq->limiterBandTable[c + 1]; k++) {

        /* 1.a  Add energy of adjusted signal (using preliminary gain) */
        FIXP_DBL  tmp   = fMult(pNrgs->nrgGain[k],pNrgs->nrgEst[k]);
        SCHAR     tmp_e = pNrgs->nrgGain_e[k] + pNrgs->nrgEst_e[k];
        FDK_add_MantExp(tmp, tmp_e, accu, accu_e, &accu, &accu_e);

        /* 1.b  Add sine energy (if present) */
        if(pNrgs->nrgSine[k] != FL2FXCONST_DBL(0.0f)) {
          FDK_add_MantExp(pNrgs->nrgSine[k], pNrgs->nrgSine_e[k], accu, accu_e, &accu, &accu_e);
        }
        else {
          /* 1.c  Add noise energy (if present) */
          if(noNoiseFlag == 0) {
            FDK_add_MantExp(pNrgs->noiseLevel[k], pNrgs->noiseLevel_e[k], accu, accu_e, &accu, &accu_e);
          }
        }
      }

      /* 2.a  Calculate ratio of wanted energy and accumulated energy */
      if (accu == (FIXP_DBL)0) { /* If divisor is 0, limit quotient to +4 dB */
        boostGain = FL2FXCONST_DBL(0.6279716f);
        boostGain_e = 2;
      } else {
        INT div_e;
        boostGain = fDivNorm(sumRef, accu, &div_e);
        boostGain_e = sumRef_e - accu_e + div_e;
      }


      /* 2.b Result too high? --> Limit the boost factor to +4 dB */
      if((boostGain_e  > 3) ||
         (boostGain_e == 2 && boostGain > FL2FXCONST_DBL(0.6279716f)) ||
         (boostGain_e == 3 && boostGain > FL2FXCONST_DBL(0.3139858f)) )
      {
        boostGain = FL2FXCONST_DBL(0.6279716f);
        boostGain_e = 2;
      }
      /* 3.  Multiply all signal components with the boost factor */
      for (k = hFreq->limiterBandTable[c]; k < hFreq->limiterBandTable[c + 1]; k++) {
        pNrgs->nrgGain[k]   = fMultDiv2(pNrgs->nrgGain[k],boostGain);
        pNrgs->nrgGain_e[k] = pNrgs->nrgGain_e[k] + boostGain_e + 1;

        pNrgs->nrgSine[k]   = fMultDiv2(pNrgs->nrgSine[k],boostGain);
        pNrgs->nrgSine_e[k] = pNrgs->nrgSine_e[k] + boostGain_e + 1;

        pNrgs->noiseLevel[k]   = fMultDiv2(pNrgs->noiseLevel[k],boostGain);
        pNrgs->noiseLevel_e[k] = pNrgs->noiseLevel_e[k] + boostGain_e + 1;
      }
    }
    /* End of noise limiting */

    if (useLP)
      aliasingReduction(degreeAlias+lowSubband,
                        pNrgs,
                        useAliasReduction,
                        noSubbands);

    /* For the timeslots within the range for the output frame,
       use the same scale for the noise levels.
       Drawback: If the envelope exceeds the frame border, the noise levels
                 will have to be rescaled later to fit final_e of
                 the gain-values.
    */
    noise_e = (start_pos < no_cols) ? adj_e : final_e;

    /*
      Convert energies to amplitude levels
    */
    for (k=0; k<noSubbands; k++) {
      FDK_sqrt_MantExp(&pNrgs->nrgSine[k],    &pNrgs->nrgSine_e[k],    &noise_e);
      FDK_sqrt_MantExp(&pNrgs->nrgGain[k],    &pNrgs->nrgGain_e[k],    &pNrgs->nrgGain_e[k]);
      FDK_sqrt_MantExp(&pNrgs->noiseLevel[k], &pNrgs->noiseLevel_e[k], &noise_e);
    }



    /*
      Apply calculated gains and adaptive noise
    */

    /* assembleHfSignals() */
    {
      int scale_change, sc_change;
      FIXP_SGL smooth_ratio;
      int filtBufferNoiseShift=0;

      /* Initialize smoothing buffers with the first valid values */
      if (h_sbr_cal_env->startUp)
      {
        if (!useLP) {
          h_sbr_cal_env->filtBufferNoise_e = noise_e;

          FDKmemcpy(h_sbr_cal_env->filtBuffer_e,    pNrgs->nrgGain_e,  noSubbands*sizeof(SCHAR));
          FDKmemcpy(h_sbr_cal_env->filtBufferNoise, pNrgs->noiseLevel, noSubbands*sizeof(FIXP_DBL));
          FDKmemcpy(h_sbr_cal_env->filtBuffer,      pNrgs->nrgGain,    noSubbands*sizeof(FIXP_DBL));

        }
        h_sbr_cal_env->startUp = 0;
      }

      if (!useLP) {

        equalizeFiltBufferExp(h_sbr_cal_env->filtBuffer,    /* buffered */
                              h_sbr_cal_env->filtBuffer_e,  /* buffered */
                              pNrgs->nrgGain,               /* current  */
                              pNrgs->nrgGain_e,             /* current  */
                              noSubbands);

        /* Adapt exponent of buffered noise levels to the current exponent
           so they can easily be smoothed */
        if((h_sbr_cal_env->filtBufferNoise_e - noise_e)>=0) {
          int shift = fixMin(DFRACT_BITS-1,(int)(h_sbr_cal_env->filtBufferNoise_e - noise_e));
          for (k=0; k<noSubbands; k++)
            h_sbr_cal_env->filtBufferNoise[k] <<= shift;
        }
        else {
          int shift = fixMin(DFRACT_BITS-1,-(int)(h_sbr_cal_env->filtBufferNoise_e - noise_e));
          for (k=0; k<noSubbands; k++)
            h_sbr_cal_env->filtBufferNoise[k] >>= shift;
        }

        h_sbr_cal_env->filtBufferNoise_e = noise_e;
      }

      /* find best scaling! */
      scale_change = -(DFRACT_BITS-1);
      for(k=0;k<noSubbands;k++) {
          scale_change = fixMax(scale_change,(int)pNrgs->nrgGain_e[k]);
      }
      sc_change = (start_pos<no_cols)? adj_e - input_e : final_e - input_e;

      if ((scale_change-sc_change+1)<0)
          scale_change-=(scale_change-sc_change+1);

      scale_change = (scale_change-sc_change)+1;

      for(k=0;k<noSubbands;k++) {
          int sc = scale_change-pNrgs->nrgGain_e[k] + (sc_change-1);
          pNrgs->nrgGain[k]  >>= sc;
          pNrgs->nrgGain_e[k] += sc;
      }

      if (!useLP) {
        for(k=0;k<noSubbands;k++) {
          int sc = scale_change-h_sbr_cal_env->filtBuffer_e[k] + (sc_change-1);
          h_sbr_cal_env->filtBuffer[k] >>= sc;
        }
      }

      for (j = start_pos; j < stop_pos; j++)
      {
        /* This timeslot is located within the first part of the processing buffer
           and will be fed into the QMF-synthesis for the current frame.
               adj_e - input_e
           This timeslot will not yet be fed into the QMF so we do not care
           about the adj_e.
               sc_change = final_e - input_e
        */
        if ( (j==no_cols) && (start_pos<no_cols) )
        {
          int shift = (int) (noise_e - final_e);
          if (!useLP)
            filtBufferNoiseShift = shift;               /* shifting of h_sbr_cal_env->filtBufferNoise[k] will be applied in function adjustTimeSlotHQ() */
          if (shift>=0) {
            shift = fixMin(DFRACT_BITS-1,shift);
            for (k=0; k<noSubbands; k++) {
              pNrgs->nrgSine[k] <<= shift;
              pNrgs->noiseLevel[k]  <<= shift;
              /*
              if (!useLP)
                h_sbr_cal_env->filtBufferNoise[k]  <<= shift;
              */
            }
          }
          else {
            shift = fixMin(DFRACT_BITS-1,-shift);
            for (k=0; k<noSubbands; k++) {
              pNrgs->nrgSine[k] >>= shift;
              pNrgs->noiseLevel[k]  >>= shift;
              /*
              if (!useLP)
                h_sbr_cal_env->filtBufferNoise[k]  >>= shift;
              */
            }
          }

          /* update noise scaling */
          noise_e = final_e;
          if (!useLP)
            h_sbr_cal_env->filtBufferNoise_e = noise_e;  /* scaling value unused! */

          /* update gain buffer*/
          sc_change -= (final_e - input_e);

          if (sc_change<0) {
            for(k=0;k<noSubbands;k++) {
                pNrgs->nrgGain[k]  >>= -sc_change;
                pNrgs->nrgGain_e[k] += -sc_change;
            }
            if (!useLP) {
              for(k=0;k<noSubbands;k++) {
                    h_sbr_cal_env->filtBuffer[k] >>= -sc_change;
              }
            }
          } else {
            scale_change+=sc_change;
          }

        } // if

        if (!useLP) {

          /* Prevent the smoothing filter from running on constant levels */
          if (j-start_pos < smooth_length)
            smooth_ratio = FDK_sbrDecoder_sbr_smoothFilter[j-start_pos];

          else
            smooth_ratio = FL2FXCONST_SGL(0.0f);

          adjustTimeSlotHQ(&analysBufferReal[j][lowSubband],
                           &analysBufferImag[j][lowSubband],
                           h_sbr_cal_env,
                           pNrgs,
                           lowSubband,
                           noSubbands,
                           scale_change,
                           smooth_ratio,
                           noNoiseFlag,
                           filtBufferNoiseShift);
        }
        else
        {
          adjustTimeSlotLC(&analysBufferReal[j][lowSubband],
                           pNrgs,
                          &h_sbr_cal_env->harmIndex,
                           lowSubband,
                           noSubbands,
                           scale_change,
                           noNoiseFlag,
                          &h_sbr_cal_env->phaseIndex,
                           (flags & SBRDEC_ELD_GRID));
        }
      } // for

      if (!useLP) {
        /* Update time-smoothing-buffers for gains and noise levels
           The gains and the noise values of the current envelope are copied into the buffer.
           This has to be done at the end of each envelope as the values are required for
           a smooth transition to the next envelope. */
        FDKmemcpy(h_sbr_cal_env->filtBuffer,      pNrgs->nrgGain,    noSubbands*sizeof(FIXP_DBL));
        FDKmemcpy(h_sbr_cal_env->filtBuffer_e,    pNrgs->nrgGain_e,  noSubbands*sizeof(SCHAR));
        FDKmemcpy(h_sbr_cal_env->filtBufferNoise, pNrgs->noiseLevel, noSubbands*sizeof(FIXP_DBL));
      }

    }
    C_ALLOC_SCRATCH_END(pNrgs, ENV_CALC_NRGS, 1);
  }

  /* Rescale output samples */
  {
    FIXP_DBL maxVal;
    int ov_reserve, reserve;

    /* Determine headroom in old adjusted samples */
    maxVal = maxSubbandSample( analysBufferReal,
                              (useLP) ? NULL : analysBufferImag,
                               lowSubband,
                               highSubband,
                               0,
                               first_start);

    ov_reserve = fNorm(maxVal);

    /* Determine headroom in new adjusted samples */
    maxVal = maxSubbandSample( analysBufferReal,
                               (useLP) ? NULL : analysBufferImag,
                               lowSubband,
                               highSubband,
                               first_start,
                               no_cols);

    reserve = fNorm(maxVal);

    /* Determine common output exponent */
    if (ov_adj_e - ov_reserve  >  adj_e - reserve ) /* set output_e to the maximum */
      output_e = ov_adj_e - ov_reserve;
    else
      output_e = adj_e - reserve;

    /* Rescale old samples */
    rescaleSubbandSamples( analysBufferReal,
                           (useLP) ? NULL : analysBufferImag,
                           lowSubband, highSubband,
                           0, first_start,
                           ov_adj_e - output_e);

    /* Rescale new samples */
    rescaleSubbandSamples( analysBufferReal,
                           (useLP) ? NULL : analysBufferImag,
                           lowSubband, highSubband,
                           first_start, no_cols,
                           adj_e - output_e);
  }

  /* Update hb_scale */
  sbrScaleFactor->hb_scale = EXP2SCALE(output_e);

  /* Save the current final exponent for the next frame: */
  sbrScaleFactor->ov_hb_scale = EXP2SCALE(final_e);


  /* We need to remeber to the next frame that the transient
     will occur in the first envelope (if tranEnv == nEnvelopes). */
  if(hFrameData->frameInfo.tranEnv == hFrameData->frameInfo.nEnvelopes)
    h_sbr_cal_env->prevTranEnv = 0;
  else
    h_sbr_cal_env->prevTranEnv = -1;

}


/*!
  \brief   Create envelope instance

  Must be called once for each channel before calculateSbrEnvelope() can be used.

  \return  errorCode, 0 if successful
*/
SBR_ERROR
createSbrEnvelopeCalc (HANDLE_SBR_CALCULATE_ENVELOPE hs,   /*!< pointer to envelope instance */
                       HANDLE_SBR_HEADER_DATA hHeaderData, /*!< static SBR control data, initialized with defaults */
                       const int chan,                     /*!< Channel for which to assign buffers */
                       const UINT flags)
{
  SBR_ERROR err = SBRDEC_OK;
  int i;

  /* Clear previous missing harmonics flags */
  for (i=0; i<(MAX_FREQ_COEFFS+15)>>4; i++) {
    hs->harmFlagsPrev[i] = 0;
  }
  hs->harmIndex = 0;

  /*
    Setup pointers for time smoothing.
    The buffer itself will be initialized later triggered by the startUp-flag.
  */
  hs->prevTranEnv = -1;


  /* initialization */
  resetSbrEnvelopeCalc(hs);

  if (chan==0) { /* do this only once */
    err = resetFreqBandTables(hHeaderData, flags);
  }

  return err;
}

/*!
  \brief   Create envelope instance

  Must be called once for each channel before calculateSbrEnvelope() can be used.

  \return  errorCode, 0 if successful
*/
int
deleteSbrEnvelopeCalc (HANDLE_SBR_CALCULATE_ENVELOPE hs)
{
  return 0;
}


/*!
  \brief   Reset envelope instance

  This function must be called for each channel on a change of configuration.
  Note that resetFreqBandTables should also be called in this case.

  \return  errorCode, 0 if successful
*/
void
resetSbrEnvelopeCalc (HANDLE_SBR_CALCULATE_ENVELOPE hCalEnv) /*!< pointer to envelope instance */
{
  hCalEnv->phaseIndex = 0;

  /* Noise exponent needs to be reset because the output exponent for the next frame depends on it */
  hCalEnv->filtBufferNoise_e = 0;

  hCalEnv->startUp = 1;
}


/*!
  \brief  Equalize exponents of the buffered gain values and the new ones

  After equalization of exponents, the FIR-filter addition for smoothing
  can be performed.
  This function is called once for each envelope before adjusting.
*/
/*static*/ void equalizeFiltBufferExp(FIXP_DBL *filtBuffer,     /*!< bufferd gains */
                                  SCHAR    *filtBuffer_e,   /*!< exponents of bufferd gains */
                                  FIXP_DBL *nrgGain,        /*!< gains for current envelope */
                                  SCHAR    *nrgGain_e,      /*!< exponents of gains for current envelope */
                                  int       subbands)       /*!< Number of QMF subbands */
{
  int   band;
  int  diff;

  for (band=0; band<subbands; band++){
    diff = (int) (nrgGain_e[band] - filtBuffer_e[band]);
    if (diff>0) {
      filtBuffer[band] >>= diff;   /* Compensate for the scale change by shifting the mantissa. */
      filtBuffer_e[band] += diff;  /* New gain is bigger, use its exponent */
    }
    else if (diff<0) {
      /* The buffered gains seem to be larger, but maybe there
         are some unused bits left in the mantissa */

      int reserve = CntLeadingZeros(fixp_abs(filtBuffer[band]))-1;

      if ((-diff) <= reserve) {
        /* There is enough space in the buffered mantissa so
           that we can take the new exponent as common.
        */
        filtBuffer[band] <<= (-diff);
        filtBuffer_e[band] += diff;  /* becomes equal to *ptrNewExp */
      }
      else {
        filtBuffer[band] <<= reserve;   /* Shift the mantissa as far as possible: */
        filtBuffer_e[band] -= reserve;  /* Compensate in the exponent: */

        /* For the remaining difference, change the new gain value */
        diff = fixMin(-(reserve + diff),DFRACT_BITS-1);
        nrgGain[band] >>= diff;
        nrgGain_e[band] += diff;
      }
    }
  }
}

/*!
  \brief  Shift left the mantissas of all subband samples
          in the giventime and frequency range by the specified number of bits.

  This function is used to rescale the audio data in the overlap buffer
  which has already been envelope adjusted with the last frame.
*/
void rescaleSubbandSamples(FIXP_DBL ** re,   /*!< Real part of input and output subband samples */
                           FIXP_DBL ** im,   /*!< Imaginary part of input and output subband samples */
                           int lowSubband,   /*!< Begin of frequency range to process */
                           int highSubband,  /*!< End of frequency range to process */
                           int start_pos,    /*!< Begin of time rage (QMF-timeslot) */
                           int next_pos,     /*!< End of time rage (QMF-timeslot) */
                           int shift)        /*!< number of bits to shift */
{
  int width = highSubband-lowSubband;

  if ( (width > 0) && (shift!=0) ) {
    if (im!=NULL) {
      for (int l=start_pos; l<next_pos; l++) {
          scaleValues(&re[l][lowSubband], width, shift);
          scaleValues(&im[l][lowSubband], width, shift);
      }
    } else
    {
      for (int l=start_pos; l<next_pos; l++) {
          scaleValues(&re[l][lowSubband], width, shift);
      }
    }
  }
}


/*!
  \brief   Determine headroom for shifting

  Determine by how much the spectrum can be shifted left
  for better accuracy in later processing.

  \return  Number of free bits in the biggest spectral value
*/

FIXP_DBL maxSubbandSample( FIXP_DBL ** re,   /*!< Real part of input and output subband samples */
                           FIXP_DBL ** im,   /*!< Real part of input and output subband samples */
                           int lowSubband,   /*!< Begin of frequency range to process */
                           int highSubband,  /*!< Number of QMF bands to process */
                           int start_pos,    /*!< Begin of time rage (QMF-timeslot) */
                           int next_pos      /*!< End of time rage (QMF-timeslot) */
                          )
{
  FIXP_DBL maxVal = FL2FX_DBL(0.0f);
  unsigned int width = highSubband - lowSubband;

  FDK_ASSERT(width <= (64));

  if ( width > 0 ) {
    if (im!=NULL) 
    {
      for (int l=start_pos; l<next_pos; l++) 
      {
#ifdef FUNCTION_FDK_get_maxval
        maxVal = FDK_get_maxval(maxVal, &re[l][lowSubband], &im[l][lowSubband], width);
#else
        int k=width;
        FIXP_DBL *reTmp = &re[l][lowSubband];
        FIXP_DBL *imTmp = &im[l][lowSubband];
        do{
          FIXP_DBL tmp1 = *(reTmp++);
          FIXP_DBL tmp2 = *(imTmp++);
          maxVal |= (FIXP_DBL)((LONG)(tmp1)^((LONG)tmp1>>(DFRACT_BITS-1)));
          maxVal |= (FIXP_DBL)((LONG)(tmp2)^((LONG)tmp2>>(DFRACT_BITS-1)));
        } while(--k!=0);
#endif
      }
    } else
    {
      for (int l=start_pos; l<next_pos; l++) {
        int k=width;
        FIXP_DBL *reTmp = &re[l][lowSubband];
        do{
          FIXP_DBL tmp = *(reTmp++);
          maxVal |= (FIXP_DBL)((LONG)(tmp)^((LONG)tmp>>(DFRACT_BITS-1)));
        }while(--k!=0);
      }
    }
  }

  return(maxVal);
}

#define SHIFT_BEFORE_SQUARE (3) /* (7/2) */
/*!<
  If the accumulator does not provide enough overflow bits or
  does not provide a high dynamic range, the below energy calculation
  requires an additional shift operation for each sample.
  On the other hand, doing the shift allows using a single-precision
  multiplication for the square (at least 16bit x 16bit).
  For even values of OVRFLW_BITS (0, 2, 4, 6), saturated arithmetic
  is required for the energy accumulation.
  Theoretically, the sample-squares can sum up to a value of 76,
  requiring 7 overflow bits. However since such situations are *very*
  rare, accu can be limited to 64.
  In case native saturated arithmetic is not available, overflows
  can be prevented by replacing the above #define by
    #define SHIFT_BEFORE_SQUARE ((8 - OVRFLW_BITS) / 2)
  which will result in slightly reduced accuracy.
*/

/*!
  \brief  Estimates the mean energy of each filter-bank channel for the
          duration of the current envelope

  This function is used when interpolFreq is true.
*/
/*static*/ void calcNrgPerSubband(FIXP_DBL  **analysBufferReal, /*!< Real part of subband samples */
                              FIXP_DBL  **analysBufferImag, /*!< Imaginary part of subband samples */
                              int       lowSubband,           /*!< Begin of the SBR frequency range */
                              int       highSubband,          /*!< High end of the SBR frequency range */
                              int       start_pos,            /*!< First QMF-slot of current envelope */
                              int       next_pos,             /*!< Last QMF-slot of current envelope + 1 */
                              SCHAR     frameExp,             /*!< Common exponent for all input samples */
                              FIXP_DBL *nrgEst,               /*!< resulting Energy (0..1) */
                              SCHAR    *nrgEst_e )            /*!< Exponent of resulting Energy */
{
  FIXP_SGL invWidth;
  SCHAR  preShift;
  SCHAR  shift;
  FIXP_DBL sum;
  int k,l;

  /* Divide by width of envelope later: */
  invWidth = FX_DBL2FX_SGL(GetInvInt(next_pos - start_pos));
  /* The common exponent needs to be doubled because all mantissas are squared: */
  frameExp = frameExp << 1;

  for (k=lowSubband; k<highSubband; k++) {
    FIXP_DBL  bufferReal[(((1024)/(32))+(6))];
    FIXP_DBL  bufferImag[(((1024)/(32))+(6))];
    FIXP_DBL maxVal = FL2FX_DBL(0.0f);

    if (analysBufferImag!=NULL)
    {
      for (l=start_pos;l<next_pos;l++) 
      {
        bufferImag[l] = analysBufferImag[l][k];
        maxVal |= (FIXP_DBL)((LONG)(bufferImag[l])^((LONG)bufferImag[l]>>(DFRACT_BITS-1)));
        bufferReal[l] = analysBufferReal[l][k];
        maxVal |= (FIXP_DBL)((LONG)(bufferReal[l])^((LONG)bufferReal[l]>>(DFRACT_BITS-1)));
      }
    }
    else
    {
      for (l=start_pos;l<next_pos;l++)
      {
        bufferReal[l] = analysBufferReal[l][k];
        maxVal |= (FIXP_DBL)((LONG)(bufferReal[l])^((LONG)bufferReal[l]>>(DFRACT_BITS-1)));
      }
    }

    if (maxVal!=FL2FXCONST_DBL(0.f)) {


      /* If the accu does not provide enough overflow bits, we cannot
         shift the samples up to the limit.
         Instead, keep up to 3 free bits in each sample, i.e. up to
         6 bits after calculation of square.
         Please note the comment on saturated arithmetic above!
      */
      FIXP_DBL accu = FL2FXCONST_DBL(0.0f);
      preShift = CntLeadingZeros(maxVal)-1;
      preShift -= SHIFT_BEFORE_SQUARE;

      if (preShift>=0) {
        if (analysBufferImag!=NULL) {
          for (l=start_pos; l<next_pos; l++) {
            FIXP_DBL temp1 = bufferReal[l] << (int)preShift;
            FIXP_DBL temp2 = bufferImag[l] << (int)preShift;
            accu = fPow2AddDiv2(accu, temp1);
            accu = fPow2AddDiv2(accu, temp2);
          }
        } else
        {
          for (l=start_pos; l<next_pos; l++) {
            FIXP_DBL temp = bufferReal[l] << (int)preShift;
            accu = fPow2AddDiv2(accu, temp);
          }
        }
      }
      else {    /* if negative shift value */
        int negpreShift = -preShift;
        if (analysBufferImag!=NULL) {
          for (l=start_pos; l<next_pos; l++) {
            FIXP_DBL temp1 = bufferReal[l] >> (int)negpreShift;
            FIXP_DBL temp2 = bufferImag[l] >> (int)negpreShift;
            accu = fPow2AddDiv2(accu, temp1);
            accu = fPow2AddDiv2(accu, temp2);
          }
        } else
        {
          for (l=start_pos; l<next_pos; l++) {
            FIXP_DBL temp = bufferReal[l] >> (int)negpreShift;
            accu = fPow2AddDiv2(accu, temp);
          }
        }
      }
      accu <<= 1;

      /* Convert double precision to Mantissa/Exponent: */
      shift = fNorm(accu);
      sum = accu << (int)shift;

      /* Divide by width of envelope and apply frame scale: */
      *nrgEst++ = fMult(sum, invWidth);
      shift += 2 * preShift;
      if (analysBufferImag!=NULL)
        *nrgEst_e++ = frameExp - shift;
      else
        *nrgEst_e++ = frameExp - shift + 1;  /* +1 due to missing imag. part */
    } /* maxVal!=0 */
    else {

      /* Prevent a zero-mantissa-number from being misinterpreted
         due to its exponent. */
      *nrgEst++ = FL2FXCONST_DBL(0.0f);
      *nrgEst_e++ = 0;
    }
  }
}

/*!
  \brief   Estimates the mean energy of each Scale factor band for the
           duration of the current envelope.

  This function is used when interpolFreq is false.
*/
/*static*/ void calcNrgPerSfb(FIXP_DBL  **analysBufferReal,  /*!< Real part of subband samples */
                          FIXP_DBL  **analysBufferImag,  /*!< Imaginary part of subband samples */
                          int       nSfb,                /*!< Number of scale factor bands */
                          UCHAR    *freqBandTable,       /*!< First Subband for each Sfb */
                          int       start_pos,           /*!< First QMF-slot of current envelope */
                          int       next_pos,            /*!< Last QMF-slot of current envelope + 1 */
                          SCHAR     input_e,             /*!< Common exponent for all input samples */
                          FIXP_DBL *nrgEst,              /*!< resulting Energy (0..1) */
                          SCHAR    *nrgEst_e )           /*!< Exponent of resulting Energy */
{
  FIXP_SGL  invWidth;
  FIXP_DBL  temp;
  SCHAR  preShift;
  SCHAR   shift, sum_e;
  FIXP_DBL  sum;

  int j,k,l,li,ui;
  FIXP_DBL sumAll, sumLine; /* Single precision would be sufficient,
                             but overflow bits are required for accumulation */

  /* Divide by width of envelope later: */
  invWidth = FX_DBL2FX_SGL(GetInvInt(next_pos - start_pos));
  /* The common exponent needs to be doubled because all mantissas are squared: */
  input_e = input_e << 1;

  for(j=0; j<nSfb; j++) {
    li = freqBandTable[j];
    ui = freqBandTable[j+1];

    FIXP_DBL maxVal = maxSubbandSample( analysBufferReal,
                                        analysBufferImag,
                                        li,
                                        ui,
                                        start_pos,
                                        next_pos );

    if (maxVal!=FL2FXCONST_DBL(0.f)) {

      preShift = CntLeadingZeros(maxVal)-1;

      /* If the accu does not provide enough overflow bits, we cannot
         shift the samples up to the limit.
         Instead, keep up to 3 free bits in each sample, i.e. up to
         6 bits after calculation of square.
         Please note the comment on saturated arithmetic above!
      */
      preShift -= SHIFT_BEFORE_SQUARE;

      sumAll = FL2FXCONST_DBL(0.0f);


      for (k=li; k<ui; k++) {

        sumLine = FL2FXCONST_DBL(0.0f);

        if (analysBufferImag!=NULL) {
          if (preShift>=0) {
            for (l=start_pos; l<next_pos; l++) {
              temp   = analysBufferReal[l][k] << (int)preShift;
              sumLine += fPow2Div2(temp);
              temp   = analysBufferImag[l][k] << (int)preShift;
              sumLine += fPow2Div2(temp);

            }
          } else {
            for (l=start_pos; l<next_pos; l++) {
              temp   = analysBufferReal[l][k] >> -(int)preShift;
              sumLine += fPow2Div2(temp);
              temp   = analysBufferImag[l][k] >> -(int)preShift;
              sumLine += fPow2Div2(temp);
            }
          }
        } else
        {
          if (preShift>=0) {
            for (l=start_pos; l<next_pos; l++) {
              temp   = analysBufferReal[l][k] << (int)preShift;
              sumLine += fPow2Div2(temp);
            }
          } else {
            for (l=start_pos; l<next_pos; l++) {
              temp   = analysBufferReal[l][k] >> -(int)preShift;
              sumLine += fPow2Div2(temp);
            }
          }
        }

        /* The number of QMF-channels per SBR bands may be up to 15.
           Shift right to avoid overflows in sum over all channels. */
        sumLine = sumLine >> (4-1);
        sumAll  += sumLine;
      }

      /* Convert double precision to Mantissa/Exponent: */
      shift = fNorm(sumAll);
      sum = sumAll << (int)shift;

      /* Divide by width of envelope: */
      sum = fMult(sum,invWidth);

      /* Divide by width of Sfb: */
      sum = fMult(sum, FX_DBL2FX_SGL(GetInvInt(ui-li)));

      /* Set all Subband energies in the Sfb to the average energy: */
      if (analysBufferImag!=NULL)
        sum_e = input_e + 4 - shift;  /* -4 to compensate right-shift */
      else
        sum_e = input_e + 4 + 1 - shift;  /* -4 to compensate right-shift; +1 due to missing imag. part */

      sum_e -= 2 * preShift;
    } /* maxVal!=0 */
    else {

      /* Prevent a zero-mantissa-number from being misinterpreted
         due to its exponent. */
      sum = FL2FXCONST_DBL(0.0f);
      sum_e = 0;
    }

    for (k=li; k<ui; k++)
    {
      *nrgEst++   = sum;
      *nrgEst_e++ = sum_e;
    }
  }
}


/*!
  \brief  Calculate gain, noise, and additional sine level for one subband.

  The resulting energy gain is given by mantissa and exponent.
*/
/*static*/ void calcSubbandGain(FIXP_DBL  nrgRef,            /*!< Reference Energy according to envelope data */
                            SCHAR     nrgRef_e,          /*!< Reference Energy according to envelope data (exponent) */
                            ENV_CALC_NRGS* nrgs,
                            int       i,
                            FIXP_DBL  tmpNoise,          /*!< Relative noise level */
                            SCHAR     tmpNoise_e,        /*!< Relative noise level (exponent) */
                            UCHAR     sinePresentFlag,   /*!< Indicates if sine is present on band */
                            UCHAR     sineMapped,        /*!< Indicates if sine must be added */
                            int       noNoiseFlag)       /*!< Flag to suppress noise addition */
{
  FIXP_DBL  nrgEst          = nrgs->nrgEst[i];            /*!< Energy in transposed signal */
  SCHAR     nrgEst_e        = nrgs->nrgEst_e[i];          /*!< Energy in transposed signal (exponent) */
  FIXP_DBL *ptrNrgGain      = &nrgs->nrgGain[i];          /*!< Resulting energy gain */
  SCHAR    *ptrNrgGain_e    = &nrgs->nrgGain_e[i];        /*!< Resulting energy gain (exponent) */
  FIXP_DBL *ptrNoiseLevel   = &nrgs->noiseLevel[i];       /*!< Resulting absolute noise energy */
  SCHAR    *ptrNoiseLevel_e = &nrgs->noiseLevel_e[i];     /*!< Resulting absolute noise energy (exponent) */
  FIXP_DBL *ptrNrgSine      = &nrgs->nrgSine[i];          /*!< Additional sine energy */
  SCHAR    *ptrNrgSine_e    = &nrgs->nrgSine_e[i];        /*!< Additional sine energy (exponent) */

  FIXP_DBL a, b, c;
  SCHAR    a_e, b_e, c_e;

  /*
     This addition of 1 prevents divisions by zero in the reference code.
     For very small energies in nrgEst, it prevents the gains from becoming
     very high which could cause some trouble due to the smoothing.
  */
  b_e = (int)(nrgEst_e - 1);
  if (b_e>=0) {
    nrgEst = (FL2FXCONST_DBL(0.5f) >> (INT)fixMin(b_e+1,DFRACT_BITS-1)) + (nrgEst >> 1);
    nrgEst_e += 1;  /* shift by 1 bit to avoid overflow */

  } else {
    nrgEst = (nrgEst >> (INT)(fixMin(-b_e+1,DFRACT_BITS-1))) + (FL2FXCONST_DBL(0.5f) >> 1);
    nrgEst_e = 2;  /* shift by 1 bit to avoid overflow */
  }

  /*  A = NrgRef * TmpNoise */
  a = fMult(nrgRef,tmpNoise);
  a_e = nrgRef_e + tmpNoise_e;

  /*  B = 1 + TmpNoise */
  b_e = (int)(tmpNoise_e - 1);
  if (b_e>=0) {
    b = (FL2FXCONST_DBL(0.5f) >> (INT)fixMin(b_e+1,DFRACT_BITS-1)) + (tmpNoise >> 1);
    b_e = tmpNoise_e + 1;  /* shift by 1 bit to avoid overflow */
  } else {
    b = (tmpNoise >> (INT)(fixMin(-b_e+1,DFRACT_BITS-1))) + (FL2FXCONST_DBL(0.5f) >> 1);
    b_e = 2;  /* shift by 1 bit to avoid overflow */
  }

  /*  noiseLevel = A / B = (NrgRef * TmpNoise) / (1 + TmpNoise) */
  FDK_divide_MantExp( a,  a_e,
                      b,  b_e,
                      ptrNoiseLevel, ptrNoiseLevel_e);

  if (sinePresentFlag) {

    /*  C = (1 + TmpNoise) * NrgEst */
    c = fMult(b,nrgEst);
    c_e = b_e + nrgEst_e;

    /*  gain = A / C = (NrgRef * TmpNoise) / (1 + TmpNoise) * NrgEst */
    FDK_divide_MantExp( a,  a_e,
                        c,  c_e,
                        ptrNrgGain, ptrNrgGain_e);

    if (sineMapped) {

      /*  sineLevel = nrgRef/ (1 + TmpNoise) */
      FDK_divide_MantExp( nrgRef,  nrgRef_e,
                          b,  b_e,
                          ptrNrgSine, ptrNrgSine_e);
    }
  }
  else {
    if (noNoiseFlag) {
      /*  B = NrgEst */
      b = nrgEst;
      b_e = nrgEst_e;
    }
    else {
      /*  B = NrgEst * (1 + TmpNoise) */
      b = fMult(b,nrgEst);
      b_e = b_e + nrgEst_e;
    }


    /*  gain = nrgRef / B */
    FDK_divide_MantExp( nrgRef,  nrgRef_e,
                        b,  b_e,
                        ptrNrgGain, ptrNrgGain_e);
  }
}


/*!
  \brief  Calculate "average gain" for the specified subband range.

  This is rather a gain of the average magnitude than the average
  of gains!
  The result is used as a relative limit for all gains within the
  current "limiter band" (a certain frequency range).
*/
/*static*/ void calcAvgGain(ENV_CALC_NRGS* nrgs,
                        int        lowSubband,    /*!< Begin of the limiter band */
                        int        highSubband,   /*!< High end of the limiter band */
                        FIXP_DBL  *ptrSumRef,
                        SCHAR     *ptrSumRef_e,
                        FIXP_DBL  *ptrAvgGain,  /*!< Resulting overall gain (mantissa) */
                        SCHAR     *ptrAvgGain_e)  /*!< Resulting overall gain (exponent) */
{
  FIXP_DBL  *nrgRef   = nrgs->nrgRef;       /*!< Reference Energy according to envelope data */
  SCHAR     *nrgRef_e = nrgs->nrgRef_e;     /*!< Reference Energy according to envelope data (exponent) */
  FIXP_DBL  *nrgEst   = nrgs->nrgEst;       /*!< Energy in transposed signal */
  SCHAR     *nrgEst_e = nrgs->nrgEst_e;     /*!< Energy in transposed signal (exponent) */

  FIXP_DBL sumRef = 1;
  FIXP_DBL sumEst = 1;
  SCHAR    sumRef_e = -FRACT_BITS;
  SCHAR    sumEst_e = -FRACT_BITS;
  int      k;

  for (k=lowSubband; k<highSubband; k++){
    /* Add nrgRef[k] to sumRef: */
    FDK_add_MantExp( sumRef, sumRef_e,
                     nrgRef[k], nrgRef_e[k],
                     &sumRef, &sumRef_e );

    /* Add nrgEst[k] to sumEst: */
    FDK_add_MantExp( sumEst, sumEst_e,
                     nrgEst[k], nrgEst_e[k],
                     &sumEst, &sumEst_e );
  }

  FDK_divide_MantExp(sumRef, sumRef_e,
                     sumEst, sumEst_e,
                     ptrAvgGain, ptrAvgGain_e);

  *ptrSumRef = sumRef;
  *ptrSumRef_e = sumRef_e;
}


/*!
  \brief   Amplify one timeslot of the signal with the calculated gains
           and add the noisefloor.
*/

/*static*/ void adjustTimeSlotLC(FIXP_DBL *ptrReal,       /*!< Subband samples to be adjusted, real part */
                             ENV_CALC_NRGS* nrgs,
                             UCHAR    *ptrHarmIndex,  /*!< Harmonic index */
                             int       lowSubband,    /*!< Lowest QMF-channel in the currently used SBR range. */
                             int       noSubbands,    /*!< Number of QMF subbands */
                             int       scale_change,  /*!< Number of bits to shift adjusted samples */
                             int       noNoiseFlag,   /*!< Flag to suppress noise addition */
                             int      *ptrPhaseIndex, /*!< Start index to random number array */
                             int       fCldfb)        /*!< CLDFB 80 flag */
{
  FIXP_DBL *pGain       = nrgs->nrgGain;     /*!< Gains of current envelope */
  FIXP_DBL *pNoiseLevel = nrgs->noiseLevel;  /*!< Noise levels of current envelope */
  FIXP_DBL *pSineLevel  = nrgs->nrgSine;     /*!< Sine levels */

  int    k;
  int    index = *ptrPhaseIndex;
  UCHAR  harmIndex = *ptrHarmIndex;
  UCHAR  freqInvFlag = (lowSubband & 1);
  FIXP_DBL  signalReal, sineLevel, sineLevelNext, sineLevelPrev;
  int    tone_count = 0;
  int    sineSign = 1;

  #define C1   ((FIXP_SGL)FL2FXCONST_SGL(2.f*0.00815f))
  #define C1_CLDFB ((FIXP_SGL)FL2FXCONST_SGL(2.f*0.16773f))

  /*
    First pass for k=0 pulled out of the loop:
  */

  index = (index + 1) & (SBR_NF_NO_RANDOM_VAL - 1);

  /*
    The next multiplication constitutes the actual envelope adjustment
    of the signal and should be carried out with full accuracy
    (supplying #FRACT_BITS valid bits).
  */
  signalReal    = fMultDiv2(*ptrReal,*pGain++) << ((int)scale_change);
  sineLevel     = *pSineLevel++;
  sineLevelNext = (noSubbands > 1) ? pSineLevel[0] : FL2FXCONST_DBL(0.0f);

  if (sineLevel!=FL2FXCONST_DBL(0.0f)) tone_count++;

  else if (!noNoiseFlag)
        /* Add noisefloor to the amplified signal */
        signalReal += (fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][0], pNoiseLevel[0])<<4);

  if (fCldfb) {

    if (!(harmIndex&0x1)) {
      /* harmIndex 0,2 */
      signalReal += (harmIndex&0x2) ? -sineLevel : sineLevel;
      *ptrReal++ = signalReal;
    }
    else {
      /* harmIndex 1,3 in combination with freqInvFlag */
      int shift = (int) (scale_change+1);
      shift = (shift>=0) ? fixMin(DFRACT_BITS-1,shift) : fixMax(-(DFRACT_BITS-1),shift);

      FIXP_DBL tmp1 = scaleValue( fMultDiv2(C1_CLDFB, sineLevel), -shift );

      FIXP_DBL tmp2 = fMultDiv2(C1_CLDFB, sineLevelNext);


      /* save switch and compare operations and reduce to XOR statement */
      if ( ((harmIndex>>1)&0x1)^freqInvFlag) {
          *(ptrReal-1) += tmp1;
          signalReal   -= tmp2;
      } else {
          *(ptrReal-1) -= tmp1;
          signalReal   += tmp2;
      }
      *ptrReal++ = signalReal;
      freqInvFlag = !freqInvFlag;
    }

  } else
  {
    if (!(harmIndex&0x1)) {
      /* harmIndex 0,2 */
      signalReal += (harmIndex&0x2) ? -sineLevel : sineLevel;
      *ptrReal++ = signalReal;
    }
    else {
      /* harmIndex 1,3 in combination with freqInvFlag */
      int shift = (int) (scale_change+1);
      shift = (shift>=0) ? fixMin(DFRACT_BITS-1,shift) : fixMax(-(DFRACT_BITS-1),shift);

      FIXP_DBL tmp1 = (shift>=0) ? ( fMultDiv2(C1, sineLevel) >> shift )
                                 : ( fMultDiv2(C1, sineLevel) << (-shift) );
      FIXP_DBL tmp2 = fMultDiv2(C1, sineLevelNext);


      /* save switch and compare operations and reduce to XOR statement */
      if ( ((harmIndex>>1)&0x1)^freqInvFlag) {
          *(ptrReal-1) += tmp1;
          signalReal   -= tmp2;
      } else {
          *(ptrReal-1) -= tmp1;
          signalReal   += tmp2;
      }
      *ptrReal++ = signalReal;
      freqInvFlag = !freqInvFlag;
    }
  }

  pNoiseLevel++;

  if ( noSubbands > 2 ) {
    if (!(harmIndex&0x1)) {
      /* harmIndex 0,2 */
      if(!harmIndex) 
      {
        sineSign = 0;
      }

      for (k=noSubbands-2; k!=0; k--) {
        FIXP_DBL sinelevel = *pSineLevel++;
        index++;
        if (((signalReal = (sineSign ? -sinelevel : sinelevel)) == FL2FXCONST_DBL(0.0f))  && !noNoiseFlag) 
        {
          /* Add noisefloor to the amplified signal */
          index &= (SBR_NF_NO_RANDOM_VAL - 1);
          signalReal += (fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][0], pNoiseLevel[0])<<4);
        }
        
        /* The next multiplication constitutes the actual envelope adjustment of the signal. */
        signalReal += fMultDiv2(*ptrReal,*pGain++) << ((int)scale_change);

        pNoiseLevel++;
        *ptrReal++ = signalReal;
      } /* for ... */
    }
    else {
      /* harmIndex 1,3 in combination with freqInvFlag */
      if (harmIndex==1) freqInvFlag = !freqInvFlag;

      for (k=noSubbands-2; k!=0; k--) {
        index++;
        /* The next multiplication constitutes the actual envelope adjustment of the signal. */
        signalReal = fMultDiv2(*ptrReal,*pGain++) << ((int)scale_change);

        if (*pSineLevel++!=FL2FXCONST_DBL(0.0f)) tone_count++;
        else if (!noNoiseFlag) {
          /* Add noisefloor to the amplified signal */
          index &= (SBR_NF_NO_RANDOM_VAL - 1);
          signalReal += (fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][0], pNoiseLevel[0])<<4);
        }

        pNoiseLevel++;

        if (tone_count <= 16) {
          FIXP_DBL addSine = fMultDiv2((pSineLevel[-2] - pSineLevel[0]), C1);
          signalReal += (freqInvFlag) ? (-addSine) : (addSine);
        }

        *ptrReal++ = signalReal;
        freqInvFlag = !freqInvFlag;
      } /* for ... */
    }
  }

  if (noSubbands > -1) {
    index++;
    /* The next multiplication constitutes the actual envelope adjustment of the signal. */
    signalReal    = fMultDiv2(*ptrReal,*pGain) << ((int)scale_change);
    sineLevelPrev = fMultDiv2(pSineLevel[-1],FL2FX_SGL(0.0163f));
    sineLevel     = pSineLevel[0];

    if (pSineLevel[0]!=FL2FXCONST_DBL(0.0f)) tone_count++;
    else if (!noNoiseFlag) {
        /* Add noisefloor to the amplified signal */
        index &= (SBR_NF_NO_RANDOM_VAL - 1);
        signalReal = signalReal + (fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][0], pNoiseLevel[0])<<4);
    }

    if (!(harmIndex&0x1)) {
      /* harmIndex 0,2 */
      *ptrReal = signalReal + ( (sineSign) ? -sineLevel : sineLevel);
    }
    else {
      /* harmIndex 1,3 in combination with freqInvFlag */
      if(tone_count <= 16){
        if (freqInvFlag) {
          *ptrReal++   = signalReal - sineLevelPrev;
          if (noSubbands + lowSubband < 63)
            *ptrReal = *ptrReal + fMultDiv2(C1, sineLevel);
        }
        else {
          *ptrReal++ = signalReal + sineLevelPrev;
          if (noSubbands + lowSubband < 63)
            *ptrReal = *ptrReal - fMultDiv2(C1, sineLevel);
        }
      }
      else *ptrReal = signalReal;
    }
  }
  *ptrHarmIndex = (harmIndex + 1) & 3;
  *ptrPhaseIndex = index & (SBR_NF_NO_RANDOM_VAL - 1);
}
void adjustTimeSlotHQ(FIXP_DBL *RESTRICT ptrReal,           /*!< Subband samples to be adjusted, real part */
                      FIXP_DBL *RESTRICT ptrImag,               /*!< Subband samples to be adjusted, imag part */
                      HANDLE_SBR_CALCULATE_ENVELOPE h_sbr_cal_env,
                      ENV_CALC_NRGS* nrgs,
                      int       lowSubband,            /*!< Lowest QMF-channel in the currently used SBR range. */
                      int       noSubbands,            /*!< Number of QMF subbands */
                      int       scale_change,          /*!< Number of bits to shift adjusted samples */
                      FIXP_SGL  smooth_ratio,          /*!< Impact of last envelope */
                      int       noNoiseFlag,           /*!< Start index to random number array */
                      int       filtBufferNoiseShift)  /*!< Shift factor of filtBufferNoise */
{

  FIXP_DBL *RESTRICT gain       = nrgs->nrgGain;        /*!< Gains of current envelope */
  FIXP_DBL *RESTRICT noiseLevel = nrgs->noiseLevel;     /*!< Noise levels of current envelope */
  FIXP_DBL *RESTRICT pSineLevel = nrgs->nrgSine;        /*!< Sine levels */

  FIXP_DBL *RESTRICT filtBuffer      = h_sbr_cal_env->filtBuffer;      /*!< Gains of last envelope */
  FIXP_DBL *RESTRICT filtBufferNoise = h_sbr_cal_env->filtBufferNoise; /*!< Noise levels of last envelope */
  UCHAR    *RESTRICT ptrHarmIndex    =&h_sbr_cal_env->harmIndex;       /*!< Harmonic index */
  int      *RESTRICT ptrPhaseIndex   =&h_sbr_cal_env->phaseIndex;      /*!< Start index to random number array */

  int    k;
  FIXP_DBL signalReal, signalImag;
  FIXP_DBL noiseReal,  noiseImag;
  FIXP_DBL  smoothedGain, smoothedNoise;
  FIXP_SGL direct_ratio = /*FL2FXCONST_SGL(1.0f) */ (FIXP_SGL)MAXVAL_SGL - smooth_ratio;
  int    index = *ptrPhaseIndex;
  UCHAR   harmIndex = *ptrHarmIndex;
  register int freqInvFlag = (lowSubband & 1);
  FIXP_DBL sineLevel;
  int shift;

  *ptrPhaseIndex = (index+noSubbands) & (SBR_NF_NO_RANDOM_VAL - 1);
  *ptrHarmIndex = (harmIndex + 1) & 3;

  /*
    Possible optimization:
    smooth_ratio and harmIndex stay constant during the loop.
    It might be faster to include a separate loop in each path.

    the check for smooth_ratio is now outside the loop and the workload
    of the whole function decreased by about 20 %
  */

  filtBufferNoiseShift += 1;      /* due to later use of fMultDiv2 instead of fMult */
  if (filtBufferNoiseShift<0)
    shift = fixMin(DFRACT_BITS-1,-filtBufferNoiseShift);
  else
    shift = fixMin(DFRACT_BITS-1, filtBufferNoiseShift);

  if (smooth_ratio > FL2FXCONST_SGL(0.0f)) {

    for (k=0; k<noSubbands; k++) {
      /*
        Smoothing: The old envelope has been bufferd and a certain ratio
        of the old gains and noise levels is used.
      */

      smoothedGain = fMult(smooth_ratio,filtBuffer[k]) +
                     fMult(direct_ratio,gain[k]);

      if (filtBufferNoiseShift<0) {
        smoothedNoise = (fMultDiv2(smooth_ratio,filtBufferNoise[k])>>shift) +
                         fMult(direct_ratio,noiseLevel[k]);
      }
      else {
        smoothedNoise = (fMultDiv2(smooth_ratio,filtBufferNoise[k])<<shift) +
                         fMult(direct_ratio,noiseLevel[k]);
      }

      /*
        The next 2 multiplications constitute the actual envelope adjustment
        of the signal and should be carried out with full accuracy
        (supplying #DFRACT_BITS valid bits).
      */
      signalReal = fMultDiv2(*ptrReal,smoothedGain)<<((int)scale_change);
      signalImag = fMultDiv2(*ptrImag,smoothedGain)<<((int)scale_change);

      index++;

      if (pSineLevel[k] != FL2FXCONST_DBL(0.0f)) {
        sineLevel = pSineLevel[k];

        switch(harmIndex) {
        case 0:
          *ptrReal++ = (signalReal + sineLevel);
          *ptrImag++ = (signalImag);
          break;
        case 2:
          *ptrReal++ = (signalReal - sineLevel);
          *ptrImag++ = (signalImag);
          break;
        case 1:
          *ptrReal++ = (signalReal);
          if (freqInvFlag)
            *ptrImag++ = (signalImag - sineLevel);
          else
            *ptrImag++ = (signalImag + sineLevel);
          break;
        case 3:
          *ptrReal++ = signalReal;
          if (freqInvFlag)
            *ptrImag++ = (signalImag + sineLevel);
          else
            *ptrImag++ = (signalImag - sineLevel);
          break;
        }
      }
      else {
        if (noNoiseFlag) {
          /* Just the amplified signal is saved */
          *ptrReal++ = (signalReal);
          *ptrImag++ = (signalImag);
        }
        else {
          /* Add noisefloor to the amplified signal */
          index &= (SBR_NF_NO_RANDOM_VAL - 1);
          noiseReal = fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][0], smoothedNoise)<<4;
          noiseImag = fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][1], smoothedNoise)<<4;
          *ptrReal++ = (signalReal + noiseReal);
          *ptrImag++ = (signalImag + noiseImag);
        }
      }
      freqInvFlag ^= 1;
    }

  }
  else 
  {
    for (k=0; k<noSubbands; k++) 
    {
      smoothedGain  = gain[k];
      signalReal = fMultDiv2(*ptrReal, smoothedGain) << scale_change;
      signalImag = fMultDiv2(*ptrImag, smoothedGain) << scale_change;

      index++;

      if ((sineLevel = pSineLevel[k]) != FL2FXCONST_DBL(0.0f)) 
      {
        switch (harmIndex) 
        {
        case 0:
          signalReal += sineLevel;
          break;
        case 1:
          if (freqInvFlag)
            signalImag -= sineLevel;
          else
            signalImag += sineLevel;
          break;
        case 2:
          signalReal -= sineLevel;
          break;
        case 3:
          if (freqInvFlag)
            signalImag += sineLevel;
          else
            signalImag -= sineLevel;
          break;
        }
      }
      else 
      {
        if (noNoiseFlag == 0)
        {
          /* Add noisefloor to the amplified signal */
          smoothedNoise = noiseLevel[k];
          index &= (SBR_NF_NO_RANDOM_VAL - 1);
          noiseReal = fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][0], smoothedNoise);
          noiseImag = fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][1], smoothedNoise);
          signalReal += noiseReal<<4;
          signalImag += noiseImag<<4;
        }
      }
      *ptrReal++ = signalReal;
      *ptrImag++ = signalImag;

      freqInvFlag ^= 1;
    }
  }
}


/*!
  \brief   Reset limiter bands.

  Build frequency band table for the gain limiter dependent on
  the previously generated transposer patch areas.

  \return  SBRDEC_OK if ok,  SBRDEC_UNSUPPORTED_CONFIG on error
*/
SBR_ERROR
ResetLimiterBands ( UCHAR *limiterBandTable,   /*!< Resulting band borders in QMF channels */
                    UCHAR *noLimiterBands,     /*!< Resulting number of limiter band */
                    UCHAR *freqBandTable,      /*!< Table with possible band borders */
                    int noFreqBands,                   /*!< Number of bands in freqBandTable */
                    const PATCH_PARAM *patchParam,     /*!< Transposer patch parameters */
                    int noPatches,                     /*!< Number of transposer patches */
                    int limiterBands)                  /*!< Selected 'band density' from bitstream */
{
  int i, k, isPatchBorder[2], loLimIndex, hiLimIndex, tempNoLim, nBands;
  UCHAR workLimiterBandTable[MAX_FREQ_COEFFS / 2 + MAX_NUM_PATCHES + 1];
  int patchBorders[MAX_NUM_PATCHES + 1];
  int kx, k2;
  FIXP_DBL temp;

  int lowSubband = freqBandTable[0];
  int highSubband = freqBandTable[noFreqBands];

  /* 1 limiter band. */
  if(limiterBands == 0) {
    limiterBandTable[0] = 0;
    limiterBandTable[1] = highSubband - lowSubband;
    nBands = 1;
  } else {
    for (i = 0; i < noPatches; i++) {
      patchBorders[i] = patchParam[i].guardStartBand - lowSubband;
    }
    patchBorders[i] = highSubband - lowSubband;

    /* 1.2, 2, or 3 limiter bands/octave plus bandborders at patchborders. */
    for (k = 0; k <= noFreqBands; k++) {
      workLimiterBandTable[k] = freqBandTable[k] - lowSubband;
    }
    for (k = 1; k < noPatches; k++) {
      workLimiterBandTable[noFreqBands + k] = patchBorders[k];
    }

    tempNoLim = nBands = noFreqBands + noPatches - 1;
    shellsort(workLimiterBandTable, tempNoLim + 1);

    loLimIndex = 0;
    hiLimIndex = 1;


    while (hiLimIndex <= tempNoLim) {
      k2 = workLimiterBandTable[hiLimIndex] + lowSubband;
      kx = workLimiterBandTable[loLimIndex] + lowSubband;

      temp = FX_SGL2FX_DBL(FDK_getNumOctavesDiv8(kx,k2)); /* Number of octaves */
      temp = fMult(temp, FDK_sbrDecoder_sbr_limiterBandsPerOctaveDiv4[limiterBands]);

      if (temp < FL2FXCONST_DBL (0.49f)>>5) {
        if (workLimiterBandTable[hiLimIndex] == workLimiterBandTable[loLimIndex]) {
          workLimiterBandTable[hiLimIndex] = highSubband;
          nBands--;
          hiLimIndex++;
          continue;
        }
        isPatchBorder[0] = isPatchBorder[1] = 0;
        for (k = 0; k <= noPatches; k++) {
          if (workLimiterBandTable[hiLimIndex] == patchBorders[k]) {
            isPatchBorder[1] = 1;
            break;
          }
        }
        if (!isPatchBorder[1]) {
          workLimiterBandTable[hiLimIndex] = highSubband;
          nBands--;
          hiLimIndex++;
          continue;
        }
        for (k = 0; k <= noPatches; k++) {
          if (workLimiterBandTable[loLimIndex] == patchBorders[k]) {
            isPatchBorder[0] = 1;
            break;
          }
        }
        if (!isPatchBorder[0]) {
          workLimiterBandTable[loLimIndex] = highSubband;
          nBands--;
        }
      }
      loLimIndex = hiLimIndex;
      hiLimIndex++;

    }
    shellsort(workLimiterBandTable, tempNoLim + 1);

    /* Test if algorithm exceeded maximum allowed limiterbands */
    if( nBands > MAX_NUM_LIMITERS || nBands <= 0) {
      return SBRDEC_UNSUPPORTED_CONFIG;
    }

    /* Copy limiterbands from working buffer into final destination */
    for (k = 0; k <= nBands; k++) {
      limiterBandTable[k] = workLimiterBandTable[k];
    }
  }
  *noLimiterBands = nBands;

  return SBRDEC_OK;
}

