/* -----------------------------------------------------------------------------

Software License for The Fraunhofer FDK AAC Codec Library for Android

© Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten

Forschung e.V. All rights reserved.

 1.    INTRODUCTION

The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software

that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding

scheme for digital audio. This FDK AAC Codec software is intended to be used on

a wide variety of Android devices.

AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient

general perceptual audio codecs. AAC-ELD is considered the best-performing

full-bandwidth communications codec by independent studies and is widely

deployed. AAC has been standardized by ISO and IEC as part of the MPEG

specifications.

Patent licenses for necessary patent claims for the FDK AAC Codec (including

those of Fraunhofer) may be obtained through Via Licensing

(www.vialicensing.com) or through the respective patent owners individually for

the purpose of encoding or decoding bit streams in products that are compliant

with the ISO/IEC MPEG audio standards. Please note that most manufacturers of

Android devices already license these patent claims through Via Licensing or

directly from the patent owners, and therefore FDK AAC Codec software may

already be covered under those patent licenses when it is used for those

licensed purposes only.

Commercially-licensed AAC software libraries, including floating-point versions

with enhanced sound quality, are also available from Fraunhofer. Users are

encouraged to check the Fraunhofer website for additional applications

information and documentation.

2.    COPYRIGHT LICENSE

Redistribution and use in source and binary forms, with or without modification,

are permitted without payment of copyright license fees provided that you

satisfy the following conditions:

You must retain the complete text of this software license in redistributions of

the FDK AAC Codec or your modifications thereto in source code form.

You must retain the complete text of this software license in the documentation

and/or other materials provided with redistributions of the FDK AAC Codec or

your modifications thereto in binary form. You must make available free of

charge copies of the complete source code of the FDK AAC Codec and your

modifications thereto to recipients of copies in binary form.

The name of Fraunhofer may not be used to endorse or promote products derived

from this library without prior written permission.

You may not charge copyright license fees for anyone to use, copy or distribute

the FDK AAC Codec software or your modifications thereto.

Your modified versions of the FDK AAC Codec must carry prominent notices stating

that you changed the software and the date of any change. For modified versions

of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android"

must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK

AAC Codec Library for Android."

3.    NO PATENT LICENSE

NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without

limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE.

Fraunhofer provides no warranty of patent non-infringement with respect to this

software.

You may use this FDK AAC Codec software or modifications thereto only for

purposes that are authorized by appropriate patent licenses.

4.    DISCLAIMER

This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright

holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES,

including but not limited to the implied warranties of merchantability and

fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR

CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary,

or consequential damages, including but not limited to procurement of substitute

goods or services; loss of use, data, or profits, or business interruption,

however caused and on any theory of liability, whether in contract, strict

liability, or tort (including negligence), arising in any way out of the use of

this software, even if advised of the possibility of such damage.

5.    CONTACT INFORMATION

Fraunhofer Institute for Integrated Circuits IIS

Attention: Audio and Multimedia Departments - FDK AAC LL

Am Wolfsmantel 33

91058 Erlangen, Germany

www.iis.fraunhofer.de/amm

amm-info@iis.fraunhofer.de

----------------------------------------------------------------------------- */

/**************************** SBR decoder library ******************************

   Author(s):

   Description:

*******************************************************************************/

/*!

  \file

  \brief  parametric stereo decoder

*/

#include "psdec.h"

#include "FDK_bitbuffer.h"

#include "sbr_rom.h"

#include "sbr_ram.h"

#include "FDK_tools_rom.h"

#include "genericStds.h"

#include "FDK_trigFcts.h"

/********************************************************************/

/*                       MLQUAL DEFINES                             */

/********************************************************************/

#define FRACT_ZERO FRACT_BITS - 1

/********************************************************************/

SBR_ERROR ResetPsDec(HANDLE_PS_DEC h_ps_d);

/***** HELPERS *****/

/***************************************************************************/

/*!

  \brief  Creates one instance of the PS_DEC struct

  \return Error info

****************************************************************************/

int CreatePsDec(HANDLE_PS_DEC *h_PS_DEC, /*!< pointer to the module state */

                int aacSamplesPerFrame) {

  SBR_ERROR errorInfo = SBRDEC_OK;

  HANDLE_PS_DEC h_ps_d;

  int i;

  if (*h_PS_DEC == NULL) {

    /* Get ps dec ram */

    h_ps_d = GetRam_ps_dec();

    if (h_ps_d == NULL) {

      goto bail;

    }

  } else {

    /* Reset an open instance */

    h_ps_d = *h_PS_DEC;

  }

  /*

   * Create Analysis Hybrid filterbank.

   */

  FDKhybridAnalysisOpen(&h_ps_d->specificTo.mpeg.hybridAnalysis,

                        h_ps_d->specificTo.mpeg.pHybridAnaStatesLFdmx,

                        sizeof(h_ps_d->specificTo.mpeg.pHybridAnaStatesLFdmx),

                        NULL, 0);

  /* initialisation */

  switch (aacSamplesPerFrame) {

    case 960:

      h_ps_d->noSubSamples = 30; /* col */

      break;

    case 1024:

      h_ps_d->noSubSamples = 32; /* col */

      break;

    default:

      h_ps_d->noSubSamples = -1;

      break;

  }

  if (h_ps_d->noSubSamples > MAX_NUM_COL || h_ps_d->noSubSamples <= 0) {

    goto bail;

  }

  h_ps_d->noChannels = NO_QMF_CHANNELS; /* row */

  h_ps_d->psDecodedPrv = 0;

  h_ps_d->procFrameBased = -1;

  for (i = 0; i < (1) + 1; i++) {

    h_ps_d->bPsDataAvail[i] = ppt_none;

  }

  {

    int error;

    error = FDKdecorrelateOpen(&(h_ps_d->specificTo.mpeg.apDecor),

                               h_ps_d->specificTo.mpeg.decorrBufferCplx,

                               (2 * ((825) + (373))));

    if (error) goto bail;

  }

  for (i = 0; i < (1) + 1; i++) {

    FDKmemclear(&h_ps_d->bsData[i].mpeg, sizeof(MPEG_PS_BS_DATA));

  }

  errorInfo = ResetPsDec(h_ps_d);

  if (errorInfo != SBRDEC_OK) goto bail;

  *h_PS_DEC = h_ps_d;

  return 0;

bail:

  if (h_ps_d != NULL) {

    DeletePsDec(&h_ps_d);

  }

  return -1;

} /*END CreatePsDec */

/***************************************************************************/

/*!

  \brief  Delete one instance of the PS_DEC struct

  \return Error info

****************************************************************************/

int DeletePsDec(HANDLE_PS_DEC *h_PS_DEC) /*!< pointer to the module state */

{

  if (*h_PS_DEC == NULL) {

    return -1;

  }

  {

    HANDLE_PS_DEC h_ps_d = *h_PS_DEC;

    FDKdecorrelateClose(&(h_ps_d->specificTo.mpeg.apDecor));

  }

  FreeRam_ps_dec(h_PS_DEC);

  return 0;

} /*END DeletePsDec */

/***************************************************************************/

/*!

  \brief resets some values of the PS handle to default states

  \return

****************************************************************************/

SBR_ERROR ResetPsDec(HANDLE_PS_DEC h_ps_d) /*!< pointer to the module state */

{

  SBR_ERROR errorInfo = SBRDEC_OK;

  INT i;

  /* explicitly init state variables to safe values (until first ps header

   * arrives) */

  h_ps_d->specificTo.mpeg.lastUsb = 0;

  /*

   * Initialize Analysis Hybrid filterbank.

   */

  FDKhybridAnalysisInit(&h_ps_d->specificTo.mpeg.hybridAnalysis, THREE_TO_TEN,

                        NO_QMF_BANDS_HYBRID20, NO_QMF_BANDS_HYBRID20, 1);

  /*

   * Initialize Synthesis Hybrid filterbank.

   */

  for (i = 0; i < 2; i++) {

    FDKhybridSynthesisInit(&h_ps_d->specificTo.mpeg.hybridSynthesis[i],

                           THREE_TO_TEN, NO_QMF_CHANNELS, NO_QMF_CHANNELS);

  }

  {

    INT error;

    error = FDKdecorrelateInit(&h_ps_d->specificTo.mpeg.apDecor, 71, DECORR_PS,

                               DUCKER_AUTOMATIC, 0, 0, 0, 0, 1, /* isLegacyPS */

                               1);

    if (error) return SBRDEC_NOT_INITIALIZED;

  }

  for (i = 0; i < NO_IID_GROUPS; i++) {

    h_ps_d->specificTo.mpeg.h11rPrev[i] = FL2FXCONST_DBL(0.5f);

    h_ps_d->specificTo.mpeg.h12rPrev[i] = FL2FXCONST_DBL(0.5f);

  }

  FDKmemclear(h_ps_d->specificTo.mpeg.h21rPrev,

              sizeof(h_ps_d->specificTo.mpeg.h21rPrev));

  FDKmemclear(h_ps_d->specificTo.mpeg.h22rPrev,

              sizeof(h_ps_d->specificTo.mpeg.h22rPrev));

  return errorInfo;

}

/***************************************************************************/

/*!

  \brief  Feed delaylines when parametric stereo is switched on.

  \return

****************************************************************************/

void PreparePsProcessing(HANDLE_PS_DEC h_ps_d,

                         const FIXP_DBL *const *const rIntBufferLeft,

                         const FIXP_DBL *const *const iIntBufferLeft,

                         const int scaleFactorLowBand) {

  if (h_ps_d->procFrameBased ==

      1) /* If we have switched from frame to slot based processing  */

  {      /* fill hybrid delay buffer.                                */

    int i, j;

    for (i = 0; i < HYBRID_FILTER_DELAY; i++) {

      FIXP_DBL qmfInputData[2][NO_QMF_BANDS_HYBRID20];

      FIXP_DBL hybridOutputData[2][NO_SUB_QMF_CHANNELS];

      for (j = 0; j < NO_QMF_BANDS_HYBRID20; j++) {

        qmfInputData[0][j] =

            scaleValue(rIntBufferLeft[i][j], scaleFactorLowBand);

        qmfInputData[1][j] =

            scaleValue(iIntBufferLeft[i][j], scaleFactorLowBand);

      }

      FDKhybridAnalysisApply(&h_ps_d->specificTo.mpeg.hybridAnalysis,

                             qmfInputData[0], qmfInputData[1],

                             hybridOutputData[0], hybridOutputData[1]);

    }

    h_ps_d->procFrameBased = 0; /* switch to slot based processing. */

  } /* procFrameBased==1 */

}

void initSlotBasedRotation(

    HANDLE_PS_DEC h_ps_d, /*!< pointer to the module state */

    int env, int usb) {

  INT group = 0;

  INT bin = 0;

  INT noIidSteps;

  FIXP_SGL invL;

  FIXP_DBL ScaleL, ScaleR;

  FIXP_DBL Alpha, Beta;

  FIXP_DBL h11r, h12r, h21r, h22r;

  const FIXP_DBL *PScaleFactors;

  if (h_ps_d->bsData[h_ps_d->processSlot].mpeg.bFineIidQ) {

    PScaleFactors = ScaleFactorsFine; /* values are shiftet right by one */

    noIidSteps = NO_IID_STEPS_FINE;

  } else {

    PScaleFactors = ScaleFactors; /* values are shiftet right by one */

    noIidSteps = NO_IID_STEPS;

  }

  /* dequantize and decode */

  for (group = 0; group < NO_IID_GROUPS; group++) {

    bin = bins2groupMap20[group];

    /*!

    <h3> type 'A' rotation </h3>

    mixing procedure R_a, used in baseline version<br>

     Scale-factor vectors c1 and c2 are precalculated in initPsTables () and

    stored in scaleFactors[] and scaleFactorsFine[] = pScaleFactors []. From the

    linearized IID parameters (intensity differences), two scale factors are

     calculated. They are used to obtain the coefficients h11... h22.

    */

    /* ScaleR and ScaleL are scaled by 1 shift right */

    ScaleR = PScaleFactors[noIidSteps + h_ps_d->specificTo.mpeg.pCoef

                                            ->aaIidIndexMapped[env][bin]];

    ScaleL = PScaleFactors[noIidSteps - h_ps_d->specificTo.mpeg.pCoef

                                            ->aaIidIndexMapped[env][bin]];

    Beta = fMult(

        fMult(Alphas[h_ps_d->specificTo.mpeg.pCoef->aaIccIndexMapped[env][bin]],

              (ScaleR - ScaleL)),

        FIXP_SQRT05);

    Alpha =

        Alphas[h_ps_d->specificTo.mpeg.pCoef->aaIccIndexMapped[env][bin]] >> 1;

    /* Alpha and Beta are now both scaled by 2 shifts right */

    /* calculate the coefficients h11... h22 from scale-factors and ICC

     * parameters */

    /* h values are scaled by 1 shift right */

    {

      FIXP_DBL trigData[4];

      inline_fixp_cos_sin(Beta + Alpha, Beta - Alpha, 2, trigData);

      h11r = fMult(ScaleL, trigData[0]);

      h12r = fMult(ScaleR, trigData[2]);

      h21r = fMult(ScaleL, trigData[1]);

      h22r = fMult(ScaleR, trigData[3]);

    }

    /*****************************************************************************************/

    /* Interpolation of the matrices H11... H22: */

    /*                                                                                       */

    /* H11(k,n) = H11(k,n[e]) + (n-n[e]) * (H11(k,n[e+1] - H11(k,n[e])) /

     * (n[e+1] - n[e])    */

    /* ... */

    /*****************************************************************************************/

    /* invL = 1/(length of envelope) */

    invL = FX_DBL2FX_SGL(GetInvInt(

        h_ps_d->bsData[h_ps_d->processSlot].mpeg.aEnvStartStop[env + 1] -

        h_ps_d->bsData[h_ps_d->processSlot].mpeg.aEnvStartStop[env]));

    h_ps_d->specificTo.mpeg.pCoef->H11r[group] =

        h_ps_d->specificTo.mpeg.h11rPrev[group];

    h_ps_d->specificTo.mpeg.pCoef->H12r[group] =

        h_ps_d->specificTo.mpeg.h12rPrev[group];

    h_ps_d->specificTo.mpeg.pCoef->H21r[group] =

        h_ps_d->specificTo.mpeg.h21rPrev[group];

    h_ps_d->specificTo.mpeg.pCoef->H22r[group] =

        h_ps_d->specificTo.mpeg.h22rPrev[group];

    h_ps_d->specificTo.mpeg.pCoef->DeltaH11r[group] =

        fMult(h11r - h_ps_d->specificTo.mpeg.pCoef->H11r[group], invL);

    h_ps_d->specificTo.mpeg.pCoef->DeltaH12r[group] =

        fMult(h12r - h_ps_d->specificTo.mpeg.pCoef->H12r[group], invL);

    h_ps_d->specificTo.mpeg.pCoef->DeltaH21r[group] =

        fMult(h21r - h_ps_d->specificTo.mpeg.pCoef->H21r[group], invL);

    h_ps_d->specificTo.mpeg.pCoef->DeltaH22r[group] =

        fMult(h22r - h_ps_d->specificTo.mpeg.pCoef->H22r[group], invL);

    /* update prev coefficients for interpolation in next envelope */

    h_ps_d->specificTo.mpeg.h11rPrev[group] = h11r;

    h_ps_d->specificTo.mpeg.h12rPrev[group] = h12r;

    h_ps_d->specificTo.mpeg.h21rPrev[group] = h21r;

    h_ps_d->specificTo.mpeg.h22rPrev[group] = h22r;

  } /* group loop */

}

static const UCHAR groupTable[NO_IID_GROUPS + 1] = {

    0,  1,  2,  3,  4,  5,  6,  7,  8,  9,  10, 11,

    12, 13, 14, 15, 16, 18, 21, 25, 30, 42, 71};

static void applySlotBasedRotation(

    HANDLE_PS_DEC h_ps_d, /*!< pointer to the module state */

    FIXP_DBL *mHybridRealLeft, /*!< hybrid values real left  */

    FIXP_DBL *mHybridImagLeft, /*!< hybrid values imag left  */

    FIXP_DBL *mHybridRealRight, /*!< hybrid values real right  */

    FIXP_DBL *mHybridImagRight  /*!< hybrid values imag right  */

) {

  INT group;

  INT subband;

  /**********************************************************************************************/

  /*!

  <h2> Mapping </h2>

  The number of stereo bands that is actually used depends on the number of

  availble parameters for IID and ICC: <pre> nr. of IID para.| nr. of ICC para.

  | nr. of Stereo bands

   ----------------|------------------|-------------------

     10,20         |     10,20        |        20

     10,20         |     34           |        34

     34            |     10,20        |        34

     34            |     34           |        34

  </pre>

  In the case the number of parameters for IIS and ICC differs from the number

  of stereo bands, a mapping from the lower number to the higher number of

  parameters is applied. Index mapping of IID and ICC parameters is already done

  in psbitdec.cpp. Further mapping is not needed here in baseline version.

  **********************************************************************************************/

  /************************************************************************************************/

  /*!

  <h2> Mixing </h2>

  To generate the QMF subband signals for the subband samples n = n[e]+1 ,,,

  n_[e+1] the parameters at position n[e] and n[e+1] are required as well as the

  subband domain signals s_k(n) and d_k(n) for n = n[e]+1... n_[e+1]. n[e]

  represents the start position for envelope e. The border positions n[e] are

  handled in DecodePS().

  The stereo sub subband signals are constructed as:

  <pre>

  l_k(n) = H11(k,n) s_k(n) + H21(k,n) d_k(n)

  r_k(n) = H21(k,n) s_k(n) + H22(k,n) d_k(n)

  </pre>

  In order to obtain the matrices H11(k,n)... H22 (k,n), the vectors h11(b)...

  h22(b) need to be calculated first (b: parameter index). Depending on ICC mode

  either mixing procedure R_a or R_b is used for that. For both procedures, the

  parameters for parameter position n[e+1] is used.

  ************************************************************************************************/

  /************************************************************************************************/

  /*!

  <h2>Phase parameters </h2>

  With disabled phase parameters (which is the case in baseline version), the

  H-matrices are just calculated by:

  <pre>

  H11(k,n[e+1] = h11(b(k))

  (...)

  b(k): parameter index according to mapping table

  </pre>

  <h2>Processing of the samples in the sub subbands </h2>

  this loop includes the interpolation of the coefficients Hxx

  ************************************************************************************************/

  /******************************************************/

  /* construct stereo sub subband signals according to: */

  /*                                                    */

  /* l_k(n) = H11(k,n) s_k(n) + H21(k,n) d_k(n)         */

  /* r_k(n) = H12(k,n) s_k(n) + H22(k,n) d_k(n)         */

  /******************************************************/

  PS_DEC_COEFFICIENTS *pCoef = h_ps_d->specificTo.mpeg.pCoef;

  for (group = 0; group < NO_IID_GROUPS; group++) {

    pCoef->H11r[group] += pCoef->DeltaH11r[group];

    pCoef->H12r[group] += pCoef->DeltaH12r[group];

    pCoef->H21r[group] += pCoef->DeltaH21r[group];

    pCoef->H22r[group] += pCoef->DeltaH22r[group];

    const int start = groupTable[group];

    const int stop = groupTable[group + 1];

    for (subband = start; subband < stop; subband++) {

      FIXP_DBL tmpLeft =

          fMultAdd(fMultDiv2(pCoef->H11r[group], mHybridRealLeft[subband]),

                   pCoef->H21r[group], mHybridRealRight[subband]);

      FIXP_DBL tmpRight =

          fMultAdd(fMultDiv2(pCoef->H12r[group], mHybridRealLeft[subband]),

                   pCoef->H22r[group], mHybridRealRight[subband]);

      mHybridRealLeft[subband] = tmpLeft;

      mHybridRealRight[subband] = tmpRight;

      tmpLeft =

          fMultAdd(fMultDiv2(pCoef->H11r[group], mHybridImagLeft[subband]),

                   pCoef->H21r[group], mHybridImagRight[subband]);

      tmpRight =

          fMultAdd(fMultDiv2(pCoef->H12r[group], mHybridImagLeft[subband]),

                   pCoef->H22r[group], mHybridImagRight[subband]);

      mHybridImagLeft[subband] = tmpLeft;

      mHybridImagRight[subband] = tmpRight;

    } /* subband */

  }

}

/***************************************************************************/

/*!

  \brief  Applies IID, ICC, IPD and OPD parameters to the current frame.

  \return none

****************************************************************************/

void ApplyPsSlot(

    HANDLE_PS_DEC h_ps_d,      /*!< handle PS_DEC*/

    FIXP_DBL **rIntBufferLeft, /*!< real bands left qmf channel (38x64)  */

    FIXP_DBL **iIntBufferLeft, /*!< imag bands left qmf channel (38x64)  */

    FIXP_DBL *rIntBufferRight, /*!< real bands right qmf channel (38x64) */

    FIXP_DBL *iIntBufferRight, /*!< imag bands right qmf channel (38x64) */

    const int scaleFactorLowBand_no_ov, const int scaleFactorLowBand,

    const int scaleFactorHighBand, const int lsb, const int usb) {

/*!

The 64-band QMF representation of the monaural signal generated by the SBR tool

is used as input of the PS tool. After the PS processing, the outputs of the

left and right hybrid synthesis filterbanks are used to generate the stereo

output signal.

<pre>

           -------------            ----------            -------------

          | Hybrid      | M_n[k,m] |          | L_n[k,m] | Hybrid      | l[n]

 m[n] --->| analysis    |--------->|          |--------->| synthesis   |----->

           -------------           | Stereo   |           -------------

                 |                 | recon-   |

                 |                 | stuction |

                \|/                |          |

           -------------           |          |

          | De-         | D_n[k,m] |          |

          | correlation |--------->|          |

           -------------           |          |           -------------

                                   |          | R_n[k,m] | Hybrid      | r[n]

                                   |          |--------->| synthesis   |----->

 IID, ICC ------------------------>|          |          | filter bank |

(IPD, OPD)                          ----------            -------------

m[n]:      QMF represantation of the mono input

M_n[k,m]:  (sub-)sub-band domain signals of the mono input

D_n[k,m]:  decorrelated (sub-)sub-band domain signals

L_n[k,m]:  (sub-)sub-band domain signals of the left output

R_n[k,m]:  (sub-)sub-band domain signals of the right output

l[n],r[n]: left/right output signals

</pre>

*/

#define NO_HYBRID_DATA_BANDS (71)

  int i;

  FIXP_DBL qmfInputData[2][NO_QMF_BANDS_HYBRID20];

  FIXP_DBL *hybridData[2][2];

  C_ALLOC_SCRATCH_START(pHybridData, FIXP_DBL, 4 * NO_HYBRID_DATA_BANDS);

  hybridData[0][0] =

      pHybridData + 0 * NO_HYBRID_DATA_BANDS; /* left real hybrid data */

  hybridData[0][1] =

      pHybridData + 1 * NO_HYBRID_DATA_BANDS; /* left imag hybrid data */

  hybridData[1][0] =

      pHybridData + 2 * NO_HYBRID_DATA_BANDS; /* right real hybrid data */

  hybridData[1][1] =

      pHybridData + 3 * NO_HYBRID_DATA_BANDS; /* right imag hybrid data */

  /*!

  Hybrid analysis filterbank:

  The lower 3 (5) of the 64 QMF subbands are further split to provide better

  frequency resolution. for PS processing. For the 10 and 20 stereo bands

  configuration, the QMF band H_0(w) is split up into 8 (sub-) sub-bands and the

  QMF bands H_1(w) and H_2(w) are spit into 2 (sub-) 4th. (See figures 8.20

  and 8.22 of ISO/IEC 14496-3:2001/FDAM 2:2004(E) )

  */

  /*

   * Hybrid analysis.

   */

  /* Get qmf input data and apply descaling */

  for (i = 0; i < NO_QMF_BANDS_HYBRID20; i++) {

    qmfInputData[0][i] = scaleValue(rIntBufferLeft[HYBRID_FILTER_DELAY][i],

                                    scaleFactorLowBand_no_ov);

    qmfInputData[1][i] = scaleValue(iIntBufferLeft[HYBRID_FILTER_DELAY][i],

                                    scaleFactorLowBand_no_ov);

  }

  /* LF - part */

  FDKhybridAnalysisApply(&h_ps_d->specificTo.mpeg.hybridAnalysis,

                         qmfInputData[0], qmfInputData[1], hybridData[0][0],

                         hybridData[0][1]);

  /* HF - part */

  /* bands up to lsb */

  scaleValues(&hybridData[0][0][NO_SUB_QMF_CHANNELS - 2],

              &rIntBufferLeft[0][NO_QMF_BANDS_HYBRID20],

              lsb - NO_QMF_BANDS_HYBRID20, scaleFactorLowBand);

  scaleValues(&hybridData[0][1][NO_SUB_QMF_CHANNELS - 2],

              &iIntBufferLeft[0][NO_QMF_BANDS_HYBRID20],

              lsb - NO_QMF_BANDS_HYBRID20, scaleFactorLowBand);

  /* bands from lsb to usb */

  scaleValues(&hybridData[0][0][lsb + (NO_SUB_QMF_CHANNELS - 2 -

                                       NO_QMF_BANDS_HYBRID20)],

              &rIntBufferLeft[0][lsb], usb - lsb, scaleFactorHighBand);

  scaleValues(&hybridData[0][1][lsb + (NO_SUB_QMF_CHANNELS - 2 -

                                       NO_QMF_BANDS_HYBRID20)],

              &iIntBufferLeft[0][lsb], usb - lsb, scaleFactorHighBand);

  /* bands from usb to NO_SUB_QMF_CHANNELS which should be zero for non-overlap

     slots but can be non-zero for overlap slots */

  FDKmemcpy(

      &hybridData[0][0]

                 [usb + (NO_SUB_QMF_CHANNELS - 2 - NO_QMF_BANDS_HYBRID20)],

      &rIntBufferLeft[0][usb], sizeof(FIXP_DBL) * (NO_QMF_CHANNELS - usb));

  FDKmemcpy(

      &hybridData[0][1]

                 [usb + (NO_SUB_QMF_CHANNELS - 2 - NO_QMF_BANDS_HYBRID20)],

      &iIntBufferLeft[0][usb], sizeof(FIXP_DBL) * (NO_QMF_CHANNELS - usb));

  /*!

  Decorrelation:

  By means of all-pass filtering and delaying, the (sub-)sub-band samples s_k(n)

  are converted into de-correlated (sub-)sub-band samples d_k(n).

  - k: frequency in hybrid spectrum

  - n: time index

  */

  FDKdecorrelateApply(&h_ps_d->specificTo.mpeg.apDecor,

                      &hybridData[0][0][0], /* left real hybrid data */

                      &hybridData[0][1][0], /* left imag hybrid data */

                      &hybridData[1][0][0], /* right real hybrid data */

                      &hybridData[1][1][0], /* right imag hybrid data */

                      0                     /* startHybBand */

  );

  /*!

  Stereo Processing:

  The sets of (sub-)sub-band samples s_k(n) and d_k(n) are processed according

  to the stereo cues which are defined per stereo band.

  */

  applySlotBasedRotation(h_ps_d,

                         &hybridData[0][0][0], /* left real hybrid data */

                         &hybridData[0][1][0], /* left imag hybrid data */

                         &hybridData[1][0][0], /* right real hybrid data */

                         &hybridData[1][1][0]  /* right imag hybrid data */

  );

  /*!

  Hybrid synthesis filterbank:

  The stereo processed hybrid subband signals l_k(n) and r_k(n) are fed into the

  hybrid synthesis filterbanks which are identical to the 64 complex synthesis

  filterbank of the SBR tool. The input to the filterbank are slots of 64 QMF

  samples. For each slot the filterbank outputs one block of 64 samples of one

  reconstructed stereo channel. The hybrid synthesis filterbank is computed

  seperatly for the left and right channel.

  */

  /*

   * Hybrid synthesis.

   */

  for (i = 0; i < 2; i++) {

    FDKhybridSynthesisApply(

        &h_ps_d->specificTo.mpeg.hybridSynthesis[i],

        hybridData[i][0], /* real hybrid data */

        hybridData[i][1], /* imag hybrid data */

        (i == 0) ? rIntBufferLeft[0]

                 : rIntBufferRight, /* output real qmf buffer */

        (i == 0) ? iIntBufferLeft[0]

                 : iIntBufferRight /* output imag qmf buffer */

    );

  }

  /* free temporary hybrid qmf values of one timeslot */

  C_ALLOC_SCRATCH_END(pHybridData, FIXP_DBL, 4 * NO_HYBRID_DATA_BANDS);

} /* END ApplyPsSlot */