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+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/********************************  Fraunhofer IIS  ***************************
+
+   Author(s):   Markus Lohwasser, Josef Hoepfl, Manuel Jander
+   Description: QMF filterbank
+
+******************************************************************************/
+
+/*!
+  \file
+  \brief  Complex qmf analysis/synthesis,  
+  This module contains the qmf filterbank for analysis [ cplxAnalysisQmfFiltering() ] and
+  synthesis [ cplxSynthesisQmfFiltering() ]. It is a polyphase implementation of a complex
+  exponential modulated filter bank. The analysis part usually runs at half the sample rate
+  than the synthesis part. (So called "dual-rate" mode.)
+
+  The coefficients of the prototype filter are specified in #sbr_qmf_64_640 (in sbr_rom.cpp).
+  Thus only a 64 channel version (32 on the analysis side) with a 640 tap prototype filter
+  are used.
+
+  \anchor PolyphaseFiltering <h2>About polyphase filtering</h2>
+  The polyphase implementation of a filterbank requires filtering at the input and output.
+  This is implemented as part of cplxAnalysisQmfFiltering() and cplxSynthesisQmfFiltering().
+  The implementation requires the filter coefficients in a specific structure as described in
+  #sbr_qmf_64_640_qmf (in sbr_rom.cpp).
+
+  This module comprises the computationally most expensive functions of the SBR decoder. The accuracy of
+  computations is also important and has a direct impact on the overall sound quality. Therefore a special
+  test program is available which can be used to only test the filterbank: main_audio.cpp
+
+  This modules also uses scaling of data to provide better SNR on fixed-point processors. See #QMF_SCALE_FACTOR (in sbr_scale.h) for details.
+  An interesting note: The function getScalefactor() can constitute a significant amount of computational complexity - very much depending on the
+  bitrate. Since it is a rather small function, effective assembler optimization might be possible.
+
+*/
+
+#include "qmf.h"
+
+
+#include "fixpoint_math.h"
+#include "dct.h"
+
+#ifdef QMFSYN_STATES_16BIT
+#define QSSCALE (7)
+#define FX_DBL2FX_QSS(x) ((FIXP_QSS) ((x)>>(DFRACT_BITS-QSS_BITS-QSSCALE) ))
+#define FX_QSS2FX_DBL(x) ((FIXP_DBL)((LONG)x)<<(DFRACT_BITS-QSS_BITS-QSSCALE))
+#else
+#define QSSCALE (0)
+#define FX_DBL2FX_QSS(x) (x)
+#define FX_QSS2FX_DBL(x) (x)
+#endif
+
+
+#if defined(__arm__)
+#include "arm/qmf_arm.cpp"
+
+#endif
+
+/*!
+ * \brief Algorithmic scaling in sbrForwardModulation()
+ *
+ * The scaling in sbrForwardModulation() is caused by:
+ *
+ *   \li 1 R_SHIFT in sbrForwardModulation()
+ *   \li 5/6 R_SHIFT in dct3() if using 32/64 Bands
+ *   \li 1 ommited gain of 2.0 in qmfForwardModulation()
+ */
+#define ALGORITHMIC_SCALING_IN_ANALYSIS_FILTERBANK 7
+
+/*!
+ * \brief Algorithmic scaling in cplxSynthesisQmfFiltering()
+ *
+ * The scaling in cplxSynthesisQmfFiltering() is caused by:
+ *
+ *   \li  5/6 R_SHIFT in dct2() if using 32/64 Bands
+ *   \li  1 ommited gain of 2.0 in qmfInverseModulation()
+ *   \li -6 division by 64 in synthesis filterbank
+ *   \li x bits external influence
+ */
+#define ALGORITHMIC_SCALING_IN_SYNTHESIS_FILTERBANK 1
+
+
+/*!
+  \brief Perform Synthesis Prototype Filtering on a single slot of input data.
+
+  The filter takes 2 * qmf->no_channels of input data and
+  generates qmf->no_channels time domain output samples.
+*/
+static
+#ifndef FUNCTION_qmfSynPrototypeFirSlot
+void qmfSynPrototypeFirSlot(
+#else
+void qmfSynPrototypeFirSlot_fallback(
+#endif
+                             HANDLE_QMF_FILTER_BANK qmf,
+                             FIXP_QMF *RESTRICT realSlot,            /*!< Input: Pointer to real Slot */
+                             FIXP_QMF *RESTRICT imagSlot,            /*!< Input: Pointer to imag Slot */
+                             INT_PCM  *RESTRICT timeOut,             /*!< Time domain data */
+                             int       stride
+                            )
+{
+  FIXP_QSS* FilterStates = (FIXP_QSS*)qmf->FilterStates;
+  int       no_channels = qmf->no_channels;
+  const FIXP_PFT *p_Filter = qmf->p_filter;
+  int p_stride = qmf->p_stride;
+  int j;
+  FIXP_QSS *RESTRICT sta = FilterStates;
+  const FIXP_PFT *RESTRICT p_flt, *RESTRICT p_fltm;
+  int scale = ((DFRACT_BITS-SAMPLE_BITS)-1-qmf->outScalefactor);
+
+  p_flt  = p_Filter+p_stride*QMF_NO_POLY;          /*                     5-ter von 330 */
+  p_fltm = p_Filter+(qmf->FilterSize/2)-p_stride*QMF_NO_POLY;  /* 5 + (320 - 2*5) = 315-ter von 330 */
+
+  FDK_ASSERT(SAMPLE_BITS-1-qmf->outScalefactor >= 0); //   (DFRACT_BITS-SAMPLE_BITS)-1-qmf->outScalefactor >= 0);
+
+  for (j = no_channels-1; j >= 0; j--) {  /* ---- läuft ueber alle Linien eines Slots ---- */
+    FIXP_QMF imag  =  imagSlot[j];  // no_channels-1 .. 0
+    FIXP_QMF real  =  realSlot[j];  // ~~"~~
+    {
+      INT_PCM tmp;
+      FIXP_DBL Are = FX_QSS2FX_DBL(sta[0]) + fMultDiv2( p_fltm[0] , real);
+
+      if (qmf->outGain!=(FIXP_DBL)0x80000000) {
+        Are = fMult(Are,qmf->outGain);
+      }
+
+  #if SAMPLE_BITS > 16
+      tmp = (INT_PCM)(SATURATE_SHIFT(fAbs(Are), scale, SAMPLE_BITS));
+  #else
+      tmp = (INT_PCM)(SATURATE_RIGHT_SHIFT(fAbs(Are), scale, SAMPLE_BITS));
+  #endif
+      if (Are < (FIXP_QMF)0) {
+        tmp = -tmp;
+      }
+      timeOut[ (j)*stride ] = tmp;
+    }
+
+    sta[0] = sta[1] + FX_DBL2FX_QSS(fMultDiv2( p_flt [4] , imag ));
+    sta[1] = sta[2] + FX_DBL2FX_QSS(fMultDiv2( p_fltm[1] , real ));
+    sta[2] = sta[3] + FX_DBL2FX_QSS(fMultDiv2( p_flt [3] , imag ));
+    sta[3] = sta[4] + FX_DBL2FX_QSS(fMultDiv2( p_fltm[2] , real ));
+    sta[4] = sta[5] + FX_DBL2FX_QSS(fMultDiv2( p_flt [2] , imag ));
+    sta[5] = sta[6] + FX_DBL2FX_QSS(fMultDiv2( p_fltm[3] , real ));
+    sta[6] = sta[7] + FX_DBL2FX_QSS(fMultDiv2( p_flt [1] , imag ));
+    sta[7] = sta[8] + FX_DBL2FX_QSS(fMultDiv2( p_fltm[4] , real ));
+    sta[8] =          FX_DBL2FX_QSS(fMultDiv2( p_flt [0] , imag ));
+
+    p_flt  += (p_stride*QMF_NO_POLY);
+    p_fltm -= (p_stride*QMF_NO_POLY);
+    sta    += 9; // = (2*QMF_NO_POLY-1);
+  }
+}
+
+#ifndef FUNCTION_qmfSynPrototypeFirSlot_NonSymmetric
+/*!
+  \brief Perform Synthesis Prototype Filtering on a single slot of input data.
+
+  The filter takes 2 * qmf->no_channels of input data and
+  generates qmf->no_channels time domain output samples.
+*/
+static
+void qmfSynPrototypeFirSlot_NonSymmetric(
+                             HANDLE_QMF_FILTER_BANK qmf,
+                             FIXP_QMF *RESTRICT realSlot,            /*!< Input: Pointer to real Slot */
+                             FIXP_QMF *RESTRICT imagSlot,            /*!< Input: Pointer to imag Slot */
+                             INT_PCM  *RESTRICT timeOut,             /*!< Time domain data */
+                             int       stride
+                            )
+{
+  FIXP_QSS* FilterStates = (FIXP_QSS*)qmf->FilterStates;
+  int       no_channels = qmf->no_channels;
+  const FIXP_PFT *p_Filter = qmf->p_filter;
+  int p_stride = qmf->p_stride;
+  int j;
+  FIXP_QSS *RESTRICT sta = FilterStates;
+  const FIXP_PFT *RESTRICT p_flt, *RESTRICT p_fltm;
+  int scale = ((DFRACT_BITS-SAMPLE_BITS)-1-qmf->outScalefactor);
+
+  p_flt  = p_Filter;                           /*!< Pointer to first half of filter coefficients */
+  p_fltm = &p_flt[qmf->FilterSize/2];  /* at index 320, overall 640 coefficients */
+
+  FDK_ASSERT(SAMPLE_BITS-1-qmf->outScalefactor >= 0); //   (DFRACT_BITS-SAMPLE_BITS)-1-qmf->outScalefactor >= 0);
+
+  for (j = no_channels-1; j >= 0; j--) {  /* ---- läuft ueber alle Linien eines Slots ---- */
+
+    FIXP_QMF imag  =  imagSlot[j];  // no_channels-1 .. 0
+    FIXP_QMF real  =  realSlot[j];  // ~~"~~
+    {
+      INT_PCM tmp;
+      FIXP_QMF Are = sta[0] + FX_DBL2FX_QSS(fMultDiv2( p_fltm[4] , real ));
+
+  #if SAMPLE_BITS > 16
+      tmp = (INT_PCM)(SATURATE_SHIFT(fAbs(Are), scale, SAMPLE_BITS));
+  #else
+      tmp = (INT_PCM)(SATURATE_RIGHT_SHIFT(fAbs(Are), scale, SAMPLE_BITS));
+  #endif
+      if (Are < (FIXP_QMF)0) {
+        tmp = -tmp;
+      }
+      timeOut[j*stride] = tmp;
+    }
+
+    sta[0] = sta[1] + FX_DBL2FX_QSS(fMultDiv2( p_flt [4] , imag ));
+    sta[1] = sta[2] + FX_DBL2FX_QSS(fMultDiv2( p_fltm[3] , real ));
+    sta[2] = sta[3] + FX_DBL2FX_QSS(fMultDiv2( p_flt [3] , imag ));
+
+    sta[3] = sta[4] + FX_DBL2FX_QSS(fMultDiv2( p_fltm[2] , real ));
+    sta[4] = sta[5] + FX_DBL2FX_QSS(fMultDiv2( p_flt [2] , imag ));
+    sta[5] = sta[6] + FX_DBL2FX_QSS(fMultDiv2( p_fltm[1] , real ));
+    sta[6] = sta[7] + FX_DBL2FX_QSS(fMultDiv2( p_flt [1] , imag ));
+
+    sta[7] = sta[8] + FX_DBL2FX_QSS(fMultDiv2( p_fltm[0] , real ));
+    sta[8] =          FX_DBL2FX_QSS(fMultDiv2( p_flt [0] , imag ));
+
+    p_flt  += (p_stride*QMF_NO_POLY);
+    p_fltm += (p_stride*QMF_NO_POLY);
+    sta    += 9; // = (2*QMF_NO_POLY-1);
+  }
+
+}
+#endif /* FUNCTION_qmfSynPrototypeFirSlot_NonSymmetric */
+
+#ifndef FUNCTION_qmfAnaPrototypeFirSlot
+/*!
+  \brief Perform Analysis Prototype Filtering on a single slot of input data.
+*/
+static
+void qmfAnaPrototypeFirSlot( FIXP_QMF *analysisBuffer,
+                             int       no_channels,             /*!< Number channels of analysis filter */
+                             const FIXP_PFT *p_filter,
+                             int       p_stride,                /*!< Stide of analysis filter    */
+                             FIXP_QAS *RESTRICT pFilterStates
+                            )
+{
+    int k;
+
+    FIXP_DBL accu;
+    const FIXP_PFT *RESTRICT p_flt = p_filter;
+    FIXP_QMF *RESTRICT pData_0 = analysisBuffer + 2*no_channels - 1;
+    FIXP_QMF *RESTRICT pData_1 = analysisBuffer;
+
+    FIXP_QAS *RESTRICT sta_0 = (FIXP_QAS *)pFilterStates;
+    FIXP_QAS *RESTRICT sta_1 = (FIXP_QAS *)pFilterStates + (2*QMF_NO_POLY*no_channels) - 1;
+    int pfltStep = QMF_NO_POLY * (p_stride);
+    int staStep1 = no_channels<<1;
+    int staStep2 = (no_channels<<3) - 1; /* Rewind one less */
+
+    /* FIR filter 0 */
+    accu =   fMultDiv2( p_flt[0], *sta_1);  sta_1 -= staStep1;
+    accu +=  fMultDiv2( p_flt[1], *sta_1);  sta_1 -= staStep1;
+    accu +=  fMultDiv2( p_flt[2], *sta_1);  sta_1 -= staStep1;
+    accu +=  fMultDiv2( p_flt[3], *sta_1);  sta_1 -= staStep1;
+    accu +=  fMultDiv2( p_flt[4], *sta_1);
+    *pData_1++ = FX_DBL2FX_QMF(accu<<1);
+    sta_1 += staStep2;
+
+    p_flt += pfltStep;
+
+    /* FIR filters 1..63 127..65 */
+    for (k=0; k<no_channels-1; k++)
+    {
+      accu =  fMultDiv2( p_flt[0], *sta_0);  sta_0 += staStep1;
+      accu += fMultDiv2( p_flt[1], *sta_0);  sta_0 += staStep1;
+      accu += fMultDiv2( p_flt[2], *sta_0);  sta_0 += staStep1;
+      accu += fMultDiv2( p_flt[3], *sta_0);  sta_0 += staStep1;
+      accu += fMultDiv2( p_flt[4], *sta_0);
+      *pData_0-- = FX_DBL2FX_QMF(accu<<1);
+      sta_0 -= staStep2;
+
+      accu =   fMultDiv2( p_flt[0], *sta_1);  sta_1 -= staStep1;
+      accu +=  fMultDiv2( p_flt[1], *sta_1);  sta_1 -= staStep1;
+      accu +=  fMultDiv2( p_flt[2], *sta_1);  sta_1 -= staStep1;
+      accu +=  fMultDiv2( p_flt[3], *sta_1);  sta_1 -= staStep1;
+      accu +=  fMultDiv2( p_flt[4], *sta_1);
+      *pData_1++ = FX_DBL2FX_QMF(accu<<1);
+      sta_1 += staStep2;
+
+      p_flt += pfltStep;
+    }
+
+    /* FIR filter 64 */
+    accu =  fMultDiv2( p_flt[0], *sta_0);  sta_0 += staStep1;
+    accu += fMultDiv2( p_flt[1], *sta_0);  sta_0 += staStep1;
+    accu += fMultDiv2( p_flt[2], *sta_0);  sta_0 += staStep1;
+    accu += fMultDiv2( p_flt[3], *sta_0);  sta_0 += staStep1;
+    accu += fMultDiv2( p_flt[4], *sta_0);
+    *pData_0-- = FX_DBL2FX_QMF(accu<<1);
+    sta_0 -= staStep2;
+}
+#endif /* !defined(FUNCTION_qmfAnaPrototypeFirSlot) */
+
+
+#ifndef FUNCTION_qmfAnaPrototypeFirSlot_NonSymmetric
+/*!
+  \brief Perform Analysis Prototype Filtering on a single slot of input data.
+*/
+static
+void qmfAnaPrototypeFirSlot_NonSymmetric(
+                                        FIXP_QMF *analysisBuffer,
+                                        int       no_channels,             /*!< Number channels of analysis filter */
+                                        const FIXP_PFT *p_filter,
+                                        int       p_stride,                /*!< Stide of analysis filter    */
+                                        FIXP_QAS *RESTRICT pFilterStates
+                                       )
+{
+  const FIXP_PFT *RESTRICT p_flt = p_filter;
+  int  p, k;
+
+  for (k = 0; k < 2*no_channels; k++)
+  {
+    FIXP_DBL accu = (FIXP_DBL)0;
+
+    p_flt += QMF_NO_POLY * (p_stride - 1);
+
+    /*
+      Perform FIR-Filter
+    */
+    for (p = 0; p < QMF_NO_POLY; p++) {
+      accu +=  fMultDiv2(*p_flt++, pFilterStates[2*no_channels * p]);
+    }
+    analysisBuffer[2*no_channels - 1 - k] = FX_DBL2FX_QMF(accu<<1);
+    pFilterStates++;
+  }
+}
+#endif /* FUNCTION_qmfAnaPrototypeFirSlot_NonSymmetric */
+
+/*!
+ *
+ * \brief Perform real-valued forward modulation of the time domain
+ *        data of timeIn and stores the real part of the subband
+ *        samples in rSubband
+ *
+ */
+static void
+qmfForwardModulationLP_even( HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Analysis Bank  */
+                             FIXP_QMF *timeIn,              /*!< Time Signal */
+                             FIXP_QMF *rSubband )           /*!< Real Output */
+{
+  int i;
+  int L = anaQmf->no_channels;
+  int M = L>>1;
+  int scale;
+  FIXP_QMF accu;
+
+  const FIXP_QMF *timeInTmp1 = (FIXP_QMF *) &timeIn[3 * M];
+  const FIXP_QMF *timeInTmp2 = timeInTmp1;
+  FIXP_QMF *rSubbandTmp = rSubband;
+
+  rSubband[0] = timeIn[3 * M] >> 1;
+
+  for (i = M-1; i != 0; i--) {
+    accu = ((*--timeInTmp1) >> 1) + ((*++timeInTmp2) >> 1);
+    *++rSubbandTmp = accu;
+  }
+
+  timeInTmp1 = &timeIn[2 * M];
+  timeInTmp2 = &timeIn[0];
+  rSubbandTmp = &rSubband[M];
+
+  for (i = L-M; i != 0; i--) {
+    accu = ((*timeInTmp1--) >> 1) - ((*timeInTmp2++) >> 1);
+    *rSubbandTmp++ = accu;
+  }
+
+  dct_III(rSubband, timeIn, L, &scale);
+}
+
+#if !defined(FUNCTION_qmfForwardModulationLP_odd)
+static void
+qmfForwardModulationLP_odd( HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Analysis Bank  */
+                            const FIXP_QMF *timeIn,        /*!< Time Signal */
+                            FIXP_QMF *rSubband )           /*!< Real Output */
+{
+  int i;
+  int L = anaQmf->no_channels;
+  int M = L>>1;
+  int shift = (anaQmf->no_channels>>6) + 1;
+
+  for (i = 0; i < M; i++) {
+    rSubband[M + i]     = (timeIn[L - 1 - i]>>1) - (timeIn[i]>>shift);
+    rSubband[M - 1 - i] = (timeIn[L + i]>>1)     + (timeIn[2 * L - 1 - i]>>shift);
+  }
+
+  dct_IV(rSubband, L, &shift);
+}
+#endif /* !defined(FUNCTION_qmfForwardModulationLP_odd) */
+
+
+
+/*!
+ *
+ * \brief Perform complex-valued forward modulation of the time domain
+ *        data of timeIn and stores the real part of the subband
+ *        samples in rSubband, and the imaginary part in iSubband
+ *
+ *        Only the lower bands are obtained (upto anaQmf->lsb). For
+ *        a full bandwidth analysis it is required to set both anaQmf->lsb
+ *        and anaQmf->usb to the amount of QMF bands.
+ *
+ */
+static void
+qmfForwardModulationHQ( HANDLE_QMF_FILTER_BANK anaQmf,     /*!< Handle of Qmf Analysis Bank  */
+                        const FIXP_QMF *RESTRICT timeIn,   /*!< Time Signal */
+                        FIXP_QMF *RESTRICT rSubband,       /*!< Real Output */
+                        FIXP_QMF *RESTRICT iSubband        /*!< Imaginary Output */
+                       )
+{
+  int i;
+  int L = anaQmf->no_channels;
+  int L2 = L<<1;
+  int shift = 0;
+
+  for (i = 0; i < L; i+=2) {
+    FIXP_QMF x0, x1, y0, y1;
+
+    x0 = timeIn[i] >> 1;
+    x1 = timeIn[i+1] >> 1;
+    y0 = timeIn[L2 - 1 - i] >> 1;
+    y1 = timeIn[L2 - 2 - i] >> 1;
+
+    rSubband[i] = x0 - y0;
+    rSubband[i+1] = x1 - y1;
+    iSubband[i] = x0 + y0;
+    iSubband[i+1] = x1 + y1;
+  }
+
+  dct_IV(rSubband, L, &shift);
+  dst_IV(iSubband, L, &shift);
+
+  {
+    {
+      const FIXP_QTW *RESTRICT sbr_t_cos;
+      const FIXP_QTW *RESTRICT sbr_t_sin;
+      sbr_t_cos = anaQmf->t_cos;
+      sbr_t_sin = anaQmf->t_sin;
+
+      for (i = 0; i < anaQmf->lsb; i++) {
+        cplxMult(&iSubband[i], &rSubband[i], iSubband[i], rSubband[i], sbr_t_cos[i], sbr_t_sin[i]);
+      }
+    }
+  }
+}
+
+/*
+ * \brief Perform one QMF slot analysis of the time domain data of timeIn
+ *        with specified stride and stores the real part of the subband
+ *        samples in rSubband, and the imaginary part in iSubband
+ *
+ *        Only the lower bands are obtained (upto anaQmf->lsb). For
+ *        a full bandwidth analysis it is required to set both anaQmf->lsb
+ *        and anaQmf->usb to the amount of QMF bands.
+ */
+void
+qmfAnalysisFilteringSlot( HANDLE_QMF_FILTER_BANK anaQmf,  /*!< Handle of Qmf Synthesis Bank  */
+                          FIXP_QMF      *qmfReal,         /*!< Low and High band, real */
+                          FIXP_QMF      *qmfImag,         /*!< Low and High band, imag */
+                          const INT_PCM *RESTRICT timeIn, /*!< Pointer to input */
+                          const int      stride,          /*!< stride factor of input */
+                          FIXP_QMF      *pWorkBuffer      /*!< pointer to temporal working buffer */
+                         )
+{
+    int i;
+    int offset = anaQmf->no_channels*(QMF_NO_POLY*2-1);
+    /*
+      Feed time signal into oldest anaQmf->no_channels states
+    */
+    {
+      FIXP_QAS *RESTRICT FilterStatesAnaTmp = ((FIXP_QAS*)anaQmf->FilterStates)+offset;
+
+      /* Feed and scale actual time in slot */
+      for(i=anaQmf->no_channels>>1; i!=0; i--) {
+        /* Place INT_PCM value left aligned in scaledTimeIn */
+#if (QAS_BITS==SAMPLE_BITS)
+        *FilterStatesAnaTmp++ = (FIXP_QAS)*timeIn; timeIn += stride;
+        *FilterStatesAnaTmp++ = (FIXP_QAS)*timeIn; timeIn += stride;
+#elif (QAS_BITS>SAMPLE_BITS)
+        *FilterStatesAnaTmp++ = (FIXP_QAS)((*timeIn)<<(QAS_BITS-SAMPLE_BITS)); timeIn += stride;
+        *FilterStatesAnaTmp++ = (FIXP_QAS)((*timeIn)<<(QAS_BITS-SAMPLE_BITS)); timeIn += stride;
+#else
+        *FilterStatesAnaTmp++ = (FIXP_QAS)((*timeIn)>>(SAMPLE_BITS-QAS_BITS)); timeIn += stride;
+        *FilterStatesAnaTmp++ = (FIXP_QAS)((*timeIn)>>(SAMPLE_BITS-QAS_BITS)); timeIn += stride;
+#endif
+      }
+    }
+
+    if (anaQmf->flags & QMF_FLAG_NONSYMMETRIC) {
+      qmfAnaPrototypeFirSlot_NonSymmetric(
+                              pWorkBuffer,
+                              anaQmf->no_channels,
+                              anaQmf->p_filter,
+                              anaQmf->p_stride,
+                              (FIXP_QAS*)anaQmf->FilterStates
+                            );
+    } else {
+      qmfAnaPrototypeFirSlot( pWorkBuffer,
+                              anaQmf->no_channels,
+                              anaQmf->p_filter,
+                              anaQmf->p_stride,
+                              (FIXP_QAS*)anaQmf->FilterStates
+                            );
+    }
+
+    if (anaQmf->flags & QMF_FLAG_LP) {
+      if (anaQmf->flags & QMF_FLAG_CLDFB)
+        qmfForwardModulationLP_odd( anaQmf,
+                                    pWorkBuffer,
+                                    qmfReal );
+      else
+        qmfForwardModulationLP_even( anaQmf,
+                                     pWorkBuffer,
+                                     qmfReal );
+
+    } else {
+      qmfForwardModulationHQ( anaQmf,
+                              pWorkBuffer,
+                              qmfReal,
+                              qmfImag
+                             );
+    }
+    /*
+      Shift filter states
+
+      Should be realized with modulo adressing on a DSP instead of a true buffer shift
+    */
+    FDKmemmove ((FIXP_QAS*)anaQmf->FilterStates, (FIXP_QAS*)anaQmf->FilterStates+anaQmf->no_channels, offset*sizeof(FIXP_QAS));
+}
+
+
+/*!
+ *
+ * \brief Perform complex-valued subband filtering of the time domain
+ *        data of timeIn and stores the real part of the subband
+ *        samples in rAnalysis, and the imaginary part in iAnalysis
+ * The qmf coefficient table is symmetric. The symmetry is expoited by
+ * shrinking the coefficient table to half the size. The addressing mode
+ * takes care of the symmetries.
+ *
+ * Only the lower bands are obtained (upto anaQmf->lsb). For
+ * a full bandwidth analysis it is required to set both anaQmf->lsb
+ * and anaQmf->usb to the amount of QMF bands.
+ *
+ * \sa PolyphaseFiltering
+ */
+
+void
+qmfAnalysisFiltering( HANDLE_QMF_FILTER_BANK anaQmf,    /*!< Handle of Qmf Analysis Bank */
+                      FIXP_QMF **qmfReal,               /*!< Pointer to real subband slots */
+                      FIXP_QMF **qmfImag,               /*!< Pointer to imag subband slots */
+                      QMF_SCALE_FACTOR *scaleFactor,
+                      const INT_PCM *timeIn,            /*!< Time signal */
+                      const int  stride,
+                      FIXP_QMF  *pWorkBuffer            /*!< pointer to temporal working buffer */
+                      )
+{
+  int i;
+  int no_channels = anaQmf->no_channels;
+
+  scaleFactor->lb_scale = -ALGORITHMIC_SCALING_IN_ANALYSIS_FILTERBANK;
+  scaleFactor->lb_scale -= anaQmf->filterScale;
+
+  for (i = 0; i < anaQmf->no_col; i++)
+  {
+      FIXP_QMF *qmfImagSlot = NULL;
+
+      if (!(anaQmf->flags & QMF_FLAG_LP)) {
+        qmfImagSlot = qmfImag[i];
+      }
+
+      qmfAnalysisFilteringSlot( anaQmf, qmfReal[i], qmfImagSlot, timeIn , stride, pWorkBuffer );
+
+      timeIn += no_channels*stride;
+
+  } /* no_col loop  i  */
+}
+
+/*!
+ *
+ * \brief Perform low power inverse modulation of the subband
+ *        samples stored in rSubband (real part) and iSubband (imaginary
+ *        part) and stores the result in pWorkBuffer.
+ *
+ */
+inline
+static void
+qmfInverseModulationLP_even( HANDLE_QMF_FILTER_BANK synQmf,   /*!< Handle of Qmf Synthesis Bank  */
+                             const FIXP_QMF *qmfReal,         /*!< Pointer to qmf real subband slot (input) */
+                             const int   scaleFactorLowBand,  /*!< Scalefactor for Low band */
+                             const int   scaleFactorHighBand, /*!< Scalefactor for High band */
+                             FIXP_QMF *pTimeOut               /*!< Pointer to qmf subband slot (output)*/
+                           )
+{
+  int i;
+  int L = synQmf->no_channels;
+  int M = L>>1;
+  int scale;
+  FIXP_QMF tmp;
+  FIXP_QMF *RESTRICT tReal = pTimeOut;
+  FIXP_QMF *RESTRICT tImag = pTimeOut + L;
+
+  /* Move input to output vector with offset */
+  scaleValues(&tReal[0],             &qmfReal[0],             synQmf->lsb,             scaleFactorLowBand);
+  scaleValues(&tReal[0+synQmf->lsb], &qmfReal[0+synQmf->lsb], synQmf->usb-synQmf->lsb, scaleFactorHighBand);
+  FDKmemclear(&tReal[0+synQmf->usb], (L-synQmf->usb)*sizeof(FIXP_QMF));
+
+  /* Dct type-2 transform */
+  dct_II(tReal, tImag, L, &scale);
+
+  /* Expand output and replace inplace the output buffers */
+  tImag[0] = tReal[M];
+  tImag[M] = (FIXP_QMF)0;
+  tmp = tReal [0];
+  tReal [0] = tReal[M];
+  tReal [M] = tmp;
+
+  for (i = 1; i < M/2; i++) {
+    /* Imag */
+    tmp = tReal[L - i];
+    tImag[M - i] =  tmp;
+    tImag[i + M] = -tmp;
+
+    tmp = tReal[M + i];
+    tImag[i] =  tmp;
+    tImag[L - i] = -tmp;
+
+    /* Real */
+    tReal [M + i] = tReal[i];
+    tReal [L - i] = tReal[M - i];
+    tmp = tReal[i];
+    tReal[i] = tReal [M - i];
+    tReal [M - i] = tmp;
+
+  }
+  /* Remaining odd terms */
+  tmp = tReal[M + M/2];
+  tImag[M/2]     =  tmp;
+  tImag[M/2 + M] = -tmp;
+
+  tReal [M + M/2] = tReal[M/2];
+}
+
+inline
+static void
+qmfInverseModulationLP_odd( HANDLE_QMF_FILTER_BANK synQmf,   /*!< Handle of Qmf Synthesis Bank  */
+                            const FIXP_QMF *qmfReal,         /*!< Pointer to qmf real subband slot (input) */
+                            const int scaleFactorLowBand,    /*!< Scalefactor for Low band */
+                            const int scaleFactorHighBand,   /*!< Scalefactor for High band */
+                            FIXP_QMF *pTimeOut               /*!< Pointer to qmf subband slot (output)*/
+                          )
+{
+  int i;
+  int L = synQmf->no_channels;
+  int M = L>>1;
+  int shift = 0;
+
+  /* Move input to output vector with offset */
+  scaleValues(pTimeOut+M,              qmfReal,             synQmf->lsb,             scaleFactorLowBand);
+  scaleValues(pTimeOut+M+synQmf->lsb,  qmfReal+synQmf->lsb, synQmf->usb-synQmf->lsb, scaleFactorHighBand);
+  FDKmemclear(pTimeOut+M+synQmf->usb, (L-synQmf->usb)*sizeof(FIXP_QMF));
+
+  dct_IV(pTimeOut+M, L, &shift);
+  for (i = 0; i < M; i++) {
+    pTimeOut[i]             =  pTimeOut[L - 1 - i];
+    pTimeOut[2 * L - 1 - i] = -pTimeOut[L + i];
+  }
+}
+
+
+/*!
+ *
+ * \brief Perform complex-valued inverse modulation of the subband
+ *        samples stored in rSubband (real part) and iSubband (imaginary
+ *        part) and stores the result in pWorkBuffer.
+ *
+ */
+inline
+static void
+qmfInverseModulationHQ( HANDLE_QMF_FILTER_BANK synQmf,  /*!< Handle of Qmf Synthesis Bank     */
+                        const FIXP_QMF *qmfReal,        /*!< Pointer to qmf real subband slot */
+                        const FIXP_QMF *qmfImag,        /*!< Pointer to qmf imag subband slot */
+                        const int   scaleFactorLowBand, /*!< Scalefactor for Low band         */
+                        const int   scaleFactorHighBand,/*!< Scalefactor for High band        */
+                        FIXP_QMF  *pWorkBuffer          /*!< WorkBuffer (output)              */
+                      )
+{
+  int i;
+  int L = synQmf->no_channels;
+  int M = L>>1;
+  int shift = 0;
+  FIXP_QMF *RESTRICT tReal = pWorkBuffer;
+  FIXP_QMF *RESTRICT tImag = pWorkBuffer+L;
+
+  if (synQmf->flags & QMF_FLAG_CLDFB){
+    for (i = 0; i < synQmf->lsb; i++) {
+      cplxMult(&tImag[i], &tReal[i],
+                scaleValue(qmfImag[i],scaleFactorLowBand), scaleValue(qmfReal[i],scaleFactorLowBand),
+                synQmf->t_cos[i], synQmf->t_sin[i]);
+    }
+    for (; i < synQmf->usb; i++) {
+      cplxMult(&tImag[i], &tReal[i],
+                scaleValue(qmfImag[i],scaleFactorHighBand), scaleValue(qmfReal[i],scaleFactorHighBand),
+                synQmf->t_cos[i], synQmf->t_sin[i]);
+    }
+  }
+
+  if ( (synQmf->flags & QMF_FLAG_CLDFB) == 0) {
+    scaleValues(&tReal[0],             &qmfReal[0],             synQmf->lsb,             scaleFactorLowBand);
+    scaleValues(&tReal[0+synQmf->lsb], &qmfReal[0+synQmf->lsb], synQmf->usb-synQmf->lsb, scaleFactorHighBand);
+    scaleValues(&tImag[0],             &qmfImag[0],             synQmf->lsb,             scaleFactorLowBand);
+    scaleValues(&tImag[0+synQmf->lsb], &qmfImag[0+synQmf->lsb], synQmf->usb-synQmf->lsb, scaleFactorHighBand);
+  }
+
+  FDKmemclear(&tReal[synQmf->usb], (synQmf->no_channels-synQmf->usb)*sizeof(FIXP_QMF));
+  FDKmemclear(&tImag[synQmf->usb], (synQmf->no_channels-synQmf->usb)*sizeof(FIXP_QMF));
+
+  dct_IV(tReal, L, &shift);
+  dst_IV(tImag, L, &shift);
+
+  if (synQmf->flags & QMF_FLAG_CLDFB){
+    for (i = 0; i < M; i++) {
+      FIXP_QMF r1, i1, r2, i2;
+      r1 = tReal[i];
+      i2 = tImag[L - 1 - i];
+      r2 = tReal[L - i - 1];
+      i1 = tImag[i];
+
+      tReal[i] = (r1 - i1)>>1;
+      tImag[L - 1 - i] = -(r1 + i1)>>1;
+      tReal[L - i - 1] =  (r2 - i2)>>1;
+      tImag[i] = -(r2 + i2)>>1;
+    }
+  } else
+  {
+    /* The array accesses are negative to compensate the missing minus sign in the low and hi band gain. */
+    /* 26 cycles on ARM926 */
+    for (i = 0; i < M; i++) {
+      FIXP_QMF r1, i1, r2, i2;
+      r1 = -tReal[i];
+      i2 = -tImag[L - 1 - i];
+      r2 = -tReal[L - i - 1];
+      i1 = -tImag[i];
+
+      tReal[i] = (r1 - i1)>>1;
+      tImag[L - 1 - i] = -(r1 + i1)>>1;
+      tReal[L - i - 1] =  (r2 - i2)>>1;
+      tImag[i] = -(r2 + i2)>>1;
+    }
+  }
+}
+
+void qmfSynthesisFilteringSlot( HANDLE_QMF_FILTER_BANK  synQmf,
+                                const FIXP_QMF  *realSlot,
+                                const FIXP_QMF  *imagSlot,
+                                const int        scaleFactorLowBand,
+                                const int        scaleFactorHighBand,
+                                INT_PCM         *timeOut,
+                                const int        stride,
+                                FIXP_QMF        *pWorkBuffer)
+{
+    if (!(synQmf->flags & QMF_FLAG_LP))
+      qmfInverseModulationHQ ( synQmf,
+                               realSlot,
+                               imagSlot,
+                               scaleFactorLowBand,
+                               scaleFactorHighBand,
+                               pWorkBuffer
+                             );
+    else
+    {
+      if (synQmf->flags & QMF_FLAG_CLDFB) {
+        qmfInverseModulationLP_odd ( synQmf,
+                                 realSlot,
+                                 scaleFactorLowBand,
+                                 scaleFactorHighBand,
+                                 pWorkBuffer
+                               );
+      } else {
+        qmfInverseModulationLP_even ( synQmf,
+                                 realSlot,
+                                 scaleFactorLowBand,
+                                 scaleFactorHighBand,
+                                 pWorkBuffer
+                               );
+      }
+    }
+
+    if (synQmf->flags & QMF_FLAG_NONSYMMETRIC) {
+        qmfSynPrototypeFirSlot_NonSymmetric (
+                                 synQmf,
+                                 pWorkBuffer,
+                                 pWorkBuffer+synQmf->no_channels,
+                                 timeOut,
+                                 stride
+                               );
+    } else {
+        qmfSynPrototypeFirSlot ( synQmf,
+                                 pWorkBuffer,
+                                 pWorkBuffer+synQmf->no_channels,
+                                 timeOut,
+                                 stride
+                               );
+    }
+}
+
+
+/*!
+ *
+ *
+ * \brief Perform complex-valued subband synthesis of the
+ *        low band and the high band and store the
+ *        time domain data in timeOut
+ *
+ * First step: Calculate the proper scaling factor of current
+ * spectral data in qmfReal/qmfImag, old spectral data in the overlap
+ * range and filter states.
+ *
+ * Second step: Perform Frequency-to-Time mapping with inverse
+ * Modulation slot-wise.
+ *
+ * Third step: Perform FIR-filter slot-wise. To save space for filter
+ * states, the MAC operations are executed directly on the filter states
+ * instead of accumulating several products in the accumulator. The
+ * buffer shift at the end of the function should be replaced by a
+ * modulo operation, which is available on some DSPs.
+ *
+ * Last step: Copy the upper part of the spectral data to the overlap buffer.
+ *
+ * The qmf coefficient table is symmetric. The symmetry is exploited by
+ * shrinking the coefficient table to half the size. The addressing mode
+ * takes care of the symmetries.  If the #define #QMFTABLE_FULL is set,
+ * coefficient addressing works on the full table size. The code will be
+ * slightly faster and slightly more compact.
+ *
+ * Workbuffer requirement: 2 x sizeof(**QmfBufferReal) * synQmf->no_channels
+ */
+void
+qmfSynthesisFiltering( HANDLE_QMF_FILTER_BANK synQmf,       /*!< Handle of Qmf Synthesis Bank  */
+                       FIXP_QMF  **QmfBufferReal,           /*!< Low and High band, real */
+                       FIXP_QMF  **QmfBufferImag,           /*!< Low and High band, imag */
+                       const QMF_SCALE_FACTOR *scaleFactor,
+                       const INT   ov_len,                  /*!< split Slot of overlap and actual slots */
+                       INT_PCM    *timeOut,                 /*!< Pointer to output */
+                       const INT   stride,                  /*!< stride factor of output */
+                       FIXP_QMF   *pWorkBuffer              /*!< pointer to temporal working buffer */
+                      )
+{
+  int i;
+  int L = synQmf->no_channels;
+  SCHAR scaleFactorHighBand;
+  SCHAR scaleFactorLowBand_ov, scaleFactorLowBand_no_ov;
+
+  /* adapt scaling */
+  scaleFactorHighBand = -ALGORITHMIC_SCALING_IN_ANALYSIS_FILTERBANK - scaleFactor->hb_scale;
+  scaleFactorLowBand_ov = - ALGORITHMIC_SCALING_IN_ANALYSIS_FILTERBANK - scaleFactor->ov_lb_scale;
+  scaleFactorLowBand_no_ov = - ALGORITHMIC_SCALING_IN_ANALYSIS_FILTERBANK - scaleFactor->lb_scale;
+
+  for (i = 0; i < synQmf->no_col; i++)  /* ----- no_col loop ----- */
+  {
+    const FIXP_DBL *QmfBufferImagSlot = NULL;
+
+    SCHAR scaleFactorLowBand = (i<ov_len) ? scaleFactorLowBand_ov : scaleFactorLowBand_no_ov;
+
+    if (!(synQmf->flags & QMF_FLAG_LP))
+        QmfBufferImagSlot = QmfBufferImag[i];
+
+    qmfSynthesisFilteringSlot(  synQmf,
+                                QmfBufferReal[i],
+                                QmfBufferImagSlot,
+                                scaleFactorLowBand,
+                                scaleFactorHighBand,
+                                timeOut+(i*L*stride),
+                                stride,
+                                pWorkBuffer);
+  } /* no_col loop  i  */
+
+}
+
+
+/*!
+ *
+ * \brief Create QMF filter bank instance
+ *
+ * \return 0 if successful
+ *
+ */
+static int
+qmfInitFilterBank (HANDLE_QMF_FILTER_BANK h_Qmf,     /*!< Handle to return */
+                   void *pFilterStates,              /*!< Handle to filter states */
+                   int noCols,                       /*!< Number of timeslots per frame */
+                   int lsb,                          /*!< Lower end of QMF frequency range */
+                   int usb,                          /*!< Upper end of QMF frequency range */
+                   int no_channels,                  /*!< Number of channels (bands) */
+                   UINT flags)                       /*!< flags */
+{
+  FDKmemclear(h_Qmf,sizeof(QMF_FILTER_BANK));
+
+  if (flags & QMF_FLAG_MPSLDFB)
+  {
+    return -1;
+  }
+
+  if ( !(flags & QMF_FLAG_MPSLDFB) && (flags & QMF_FLAG_CLDFB) )
+  {
+    flags |= QMF_FLAG_NONSYMMETRIC;
+    h_Qmf->filterScale = QMF_CLDFB_PFT_SCALE;
+
+    h_Qmf->p_stride = 1;
+    switch (no_channels) {
+      case 64:
+        h_Qmf->t_cos = qmf_phaseshift_cos64_cldfb;
+        h_Qmf->t_sin = qmf_phaseshift_sin64_cldfb;
+        h_Qmf->p_filter = qmf_cldfb_640;
+        h_Qmf->FilterSize = 640;
+        break;
+      case 32:
+        h_Qmf->t_cos = qmf_phaseshift_cos32_cldfb;
+        h_Qmf->t_sin = qmf_phaseshift_sin32_cldfb;
+        h_Qmf->p_filter = qmf_cldfb_320;
+        h_Qmf->FilterSize = 320;
+        break;
+      default:
+        return -1;
+    }
+  }
+
+  if ( !(flags & QMF_FLAG_MPSLDFB) && ((flags & QMF_FLAG_CLDFB) == 0) )
+  {
+    switch (no_channels) {
+      case 64:
+        h_Qmf->p_filter = qmf_64;
+        h_Qmf->t_cos = qmf_phaseshift_cos64;
+        h_Qmf->t_sin = qmf_phaseshift_sin64;
+        h_Qmf->p_stride = 1;
+        h_Qmf->FilterSize = 640;
+        h_Qmf->filterScale = 0;
+        break;
+      case 32:
+        h_Qmf->p_filter = qmf_64;
+        h_Qmf->t_cos = qmf_phaseshift_cos32;
+        h_Qmf->t_sin = qmf_phaseshift_sin32;
+        h_Qmf->p_stride = 2;
+        h_Qmf->FilterSize = 640;
+        h_Qmf->filterScale = 0;
+        break;
+      default:
+        return -1;
+    }
+  }
+
+  h_Qmf->flags = flags;
+
+  h_Qmf->no_channels = no_channels;
+  h_Qmf->no_col = noCols;
+
+  h_Qmf->lsb = lsb;
+  h_Qmf->usb = fMin(usb, h_Qmf->no_channels);
+
+  h_Qmf->FilterStates = (void*)pFilterStates;
+
+  h_Qmf->outScalefactor = ALGORITHMIC_SCALING_IN_ANALYSIS_FILTERBANK + ALGORITHMIC_SCALING_IN_SYNTHESIS_FILTERBANK + h_Qmf->filterScale;
+
+  if (h_Qmf->p_stride == 2) {
+    h_Qmf->outScalefactor -= 1;
+  }
+  h_Qmf->outGain = (FIXP_DBL)0x80000000; /* default init value will be not applied */
+
+  return (0);
+}
+
+/*!
+ *
+ * \brief Adjust synthesis qmf filter states
+ *
+ * \return void
+ *
+ */
+static inline void
+qmfAdaptFilterStates (HANDLE_QMF_FILTER_BANK synQmf,     /*!< Handle of Qmf Filter Bank */
+                      int scaleFactorDiff)               /*!< Scale factor difference to be applied */
+{
+  if (synQmf == NULL || synQmf->FilterStates == NULL) {
+    return;
+  }
+  scaleValues((FIXP_QSS*)synQmf->FilterStates, synQmf->no_channels*(QMF_NO_POLY*2 - 1), scaleFactorDiff);
+}
+
+/*!
+ *
+ * \brief Create QMF filter bank instance
+ *
+ * Only the lower bands are obtained (upto anaQmf->lsb). For
+ * a full bandwidth analysis it is required to set both anaQmf->lsb
+ * and anaQmf->usb to the amount of QMF bands.
+ *
+ * \return 0 if succesful
+ *
+ */
+int
+qmfInitAnalysisFilterBank (HANDLE_QMF_FILTER_BANK h_Qmf,   /*!< Returns handle */
+                           FIXP_QAS *pFilterStates,        /*!< Handle to filter states */
+                           int noCols,                     /*!< Number of timeslots per frame */
+                           int lsb,                        /*!< lower end of QMF */
+                           int usb,                        /*!< upper end of QMF */
+                           int no_channels,                /*!< Number of channels (bands) */
+                           int flags)                      /*!< Low Power flag */
+{
+  int err = qmfInitFilterBank(h_Qmf, pFilterStates, noCols, lsb, usb, no_channels, flags);
+  if ( !(flags & QMF_FLAG_KEEP_STATES) && (h_Qmf->FilterStates != NULL) ) {
+    FDKmemclear(h_Qmf->FilterStates, (2*QMF_NO_POLY-1)*h_Qmf->no_channels*sizeof(FIXP_QAS));
+  }
+
+  return err;
+}
+
+/*!
+ *
+ * \brief Create QMF filter bank instance
+ *
+ * Only the lower bands are obtained (upto anaQmf->lsb). For
+ * a full bandwidth analysis it is required to set both anaQmf->lsb
+ * and anaQmf->usb to the amount of QMF bands.
+ *
+ * \return 0 if succesful
+ *
+ */
+int
+qmfInitSynthesisFilterBank (HANDLE_QMF_FILTER_BANK h_Qmf,   /*!< Returns handle */
+                            FIXP_QSS *pFilterStates,        /*!< Handle to filter states */
+                            int noCols,                     /*!< Number of timeslots per frame */
+                            int lsb,                        /*!< lower end of QMF */
+                            int usb,                        /*!< upper end of QMF */
+                            int no_channels,                /*!< Number of channels (bands) */
+                            int flags)                      /*!< Low Power flag */
+{
+  int oldOutScale = h_Qmf->outScalefactor;
+  int err = qmfInitFilterBank(h_Qmf, pFilterStates, noCols, lsb, usb, no_channels, flags);
+  if ( h_Qmf->FilterStates != NULL ) {
+    if ( !(flags & QMF_FLAG_KEEP_STATES) ) {
+      FDKmemclear(h_Qmf->FilterStates, (2*QMF_NO_POLY-1)*h_Qmf->no_channels*sizeof(FIXP_QSS));
+    } else {
+      qmfAdaptFilterStates(h_Qmf, oldOutScale-h_Qmf->outScalefactor);
+    }
+  }
+  return err;
+}
+
+
+
+
+/*!
+ *
+ * \brief Change scale factor for output data and adjust qmf filter states
+ *
+ * \return void
+ *
+ */
+void
+qmfChangeOutScalefactor (HANDLE_QMF_FILTER_BANK synQmf,     /*!< Handle of Qmf Synthesis Bank */
+                         int outScalefactor                 /*!< New scaling factor for output data */
+                        )
+{
+  if (synQmf == NULL || synQmf->FilterStates == NULL) {
+    return;
+  }
+
+  /* Add internal filterbank scale */
+  outScalefactor += ALGORITHMIC_SCALING_IN_ANALYSIS_FILTERBANK + ALGORITHMIC_SCALING_IN_SYNTHESIS_FILTERBANK + synQmf->filterScale;
+
+  if (synQmf->p_stride == 2) {
+    outScalefactor -= 1;
+  }
+
+  /* adjust filter states when scale factor has been changed */
+  if (synQmf->outScalefactor != outScalefactor)
+  {
+    int diff;
+
+    if (outScalefactor > (SAMPLE_BITS - 1)) {
+      outScalefactor = SAMPLE_BITS - 1;
+    } else if (outScalefactor < (1 - SAMPLE_BITS)) {
+      outScalefactor = 1 - SAMPLE_BITS;
+    }
+
+    diff = synQmf->outScalefactor - outScalefactor;
+
+    qmfAdaptFilterStates(synQmf, diff);
+
+    /* save new scale factor */
+    synQmf->outScalefactor = outScalefactor;
+  }
+}
+
+/*!
+ *
+ * \brief Change gain for output data
+ *
+ * \return void
+ *
+ */
+void
+qmfChangeOutGain (HANDLE_QMF_FILTER_BANK synQmf,     /*!< Handle of Qmf Synthesis Bank */
+                  FIXP_DBL outputGain                /*!< New gain for output data */
+                 )
+{
+  synQmf->outGain = outputGain;
+}
+