Snapshot 2bda038c163298531d47394bc2c09e1409c5d0db

Change-Id: If584e579464f28b97d50e51fc76ba654a5536c54

1 files changed, 1000 insertions, 0 deletions

diff --git a/libSBRdec/src/lpp_tran.cpp b/libSBRdec/src/lpp_tran.cpp
new file mode 100644
index 0000000..2d51831
--- /dev/null
+++ b/libSBRdec/src/lpp_tran.cpp
@@ -0,0 +1,1000 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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
+----------------------------------------------------------------------------------------------------------- */
+
+/*!
+  \file
+  \brief  Low Power Profile Transposer,  
+  This module provides the transposer. The main entry point is lppTransposer(). The function generates
+  high frequency content by copying data from the low band (provided by core codec) into the high band.
+  This process is also referred to as "patching". The function also implements spectral whitening by means of
+  inverse filtering based on LPC coefficients.
+
+  Together with the QMF filterbank the transposer can be tested using a supplied test program. See main_audio.cpp for details.
+  This module does use fractional arithmetic and the accuracy of the computations has an impact on the overall sound quality.
+  The module also needs to take into account the different scaling of spectral data.
+
+  \sa lppTransposer(), main_audio.cpp, sbr_scale.h, \ref documentationOverview
+*/
+
+#include "lpp_tran.h"
+
+#include "sbr_ram.h"
+#include "sbr_rom.h"
+
+#include "genericStds.h"
+#include "autocorr2nd.h"
+
+
+
+#if defined(__arm__)
+#include "arm/lpp_tran_arm.cpp"
+#endif
+
+
+
+#define LPC_SCALE_FACTOR  2
+
+
+/*!
+ *
+ * \brief Get bandwidth expansion factor from filtering level
+ *
+ * Returns a filter parameter (bandwidth expansion factor) depending on
+ * the desired filtering level signalled in the bitstream.
+ * When switching the filtering level from LOW to OFF, an additional
+ * level is being inserted to achieve a smooth transition.
+ */
+
+#ifndef FUNCTION_mapInvfMode
+static FIXP_DBL
+mapInvfMode (INVF_MODE mode,
+             INVF_MODE prevMode,
+             WHITENING_FACTORS whFactors)
+{
+  switch (mode) {
+  case INVF_LOW_LEVEL:
+    if(prevMode == INVF_OFF)
+      return whFactors.transitionLevel;
+    else
+      return whFactors.lowLevel;
+
+  case INVF_MID_LEVEL:
+    return whFactors.midLevel;
+
+  case INVF_HIGH_LEVEL:
+    return whFactors.highLevel;
+
+  default:
+    if(prevMode == INVF_LOW_LEVEL)
+      return whFactors.transitionLevel;
+    else
+      return whFactors.off;
+  }
+}
+#endif /* #ifndef FUNCTION_mapInvfMode */
+
+/*!
+ *
+ * \brief Perform inverse filtering level emphasis
+ *
+ * Retrieve bandwidth expansion factor and apply smoothing for each filter band
+ *
+ */
+
+#ifndef FUNCTION_inverseFilteringLevelEmphasis
+static void
+inverseFilteringLevelEmphasis(HANDLE_SBR_LPP_TRANS hLppTrans,/*!< Handle of lpp transposer  */
+                              UCHAR nInvfBands,              /*!< Number of bands for inverse filtering */
+                              INVF_MODE *sbr_invf_mode,      /*!< Current inverse filtering modes */
+                              INVF_MODE *sbr_invf_mode_prev, /*!< Previous inverse filtering modes */
+                              FIXP_DBL * bwVector            /*!< Resulting filtering levels */
+                              )
+{
+  for(int i = 0; i < nInvfBands; i++) {
+    FIXP_DBL accu;
+    FIXP_DBL bwTmp = mapInvfMode (sbr_invf_mode[i],
+                                  sbr_invf_mode_prev[i],
+                                  hLppTrans->pSettings->whFactors);
+
+    if(bwTmp < hLppTrans->bwVectorOld[i]) {
+      accu = fMultDiv2(FL2FXCONST_DBL(0.75f),bwTmp) +
+             fMultDiv2(FL2FXCONST_DBL(0.25f),hLppTrans->bwVectorOld[i]);
+    }
+    else {
+      accu = fMultDiv2(FL2FXCONST_DBL(0.90625f),bwTmp) +
+             fMultDiv2(FL2FXCONST_DBL(0.09375f),hLppTrans->bwVectorOld[i]);
+    }
+
+    if (accu <  FL2FXCONST_DBL(0.015625f)>>1)
+      bwVector[i] = FL2FXCONST_DBL(0.0f);
+    else
+      bwVector[i] = fixMin(accu<<1,FL2FXCONST_DBL(0.99609375f));
+  }
+}
+#endif /* #ifndef FUNCTION_inverseFilteringLevelEmphasis */
+
+/* Resulting autocorrelation determinant exponent */
+#define ACDET_EXP (2*(DFRACT_BITS+sbrScaleFactor->lb_scale+10-ac.det_scale))
+#define AC_EXP (-sbrScaleFactor->lb_scale+LPC_SCALE_FACTOR)
+#define ALPHA_EXP (-sbrScaleFactor->lb_scale+LPC_SCALE_FACTOR+1)
+/* Resulting transposed QMF values exponent 16 bit normalized samplebits assumed. */
+#define QMFOUT_EXP ((SAMPLE_BITS-15)-sbrScaleFactor->lb_scale)
+
+/*!
+ *
+ * \brief Perform transposition by patching of subband samples.
+ * This function serves as the main entry point into the module. The function determines the areas for the
+ * patching process (these are the source range as well as the target range) and implements spectral whitening
+ * by means of inverse filtering. The function autoCorrelation2nd() is an auxiliary function for calculating the
+ * LPC coefficients for the filtering.  The actual calculation of the LPC coefficients and the implementation
+ * of the filtering are done as part of lppTransposer().
+ *
+ * Note that the filtering is done on all available QMF subsamples, whereas the patching is only done on those QMF
+ * subsamples that will be used in the next QMF synthesis. The filtering is also implemented before the patching
+ * includes further dependencies on parameters from the SBR data.
+ *
+ */
+
+void lppTransposer (HANDLE_SBR_LPP_TRANS hLppTrans,    /*!< Handle of lpp transposer  */
+                    QMF_SCALE_FACTOR  *sbrScaleFactor, /*!< Scaling factors */
+                    FIXP_DBL **qmfBufferReal,          /*!< Pointer to pointer to real part of subband samples (source) */
+
+                    FIXP_DBL *degreeAlias,             /*!< Vector for results of aliasing estimation */
+                    FIXP_DBL **qmfBufferImag,          /*!< Pointer to pointer to imaginary part of subband samples (source) */
+                    const int useLP,
+                    const int timeStep,                /*!< Time step of envelope */
+                    const int firstSlotOffs,           /*!< Start position in time */
+                    const int lastSlotOffs,            /*!< Number of overlap-slots into next frame */
+                    const int nInvfBands,              /*!< Number of bands for inverse filtering */
+                    INVF_MODE *sbr_invf_mode,          /*!< Current inverse filtering modes */
+                    INVF_MODE *sbr_invf_mode_prev      /*!< Previous inverse filtering modes */
+                    )
+{
+  INT    bwIndex[MAX_NUM_PATCHES];
+  FIXP_DBL  bwVector[MAX_NUM_PATCHES];       /*!< pole moving factors */
+
+  int    i;
+  int    loBand, start, stop;
+  TRANSPOSER_SETTINGS *pSettings = hLppTrans->pSettings;
+  PATCH_PARAM *patchParam = pSettings->patchParam;
+  int    patch;
+
+  FIXP_SGL  alphar[LPC_ORDER], a0r, a1r;
+  FIXP_SGL  alphai[LPC_ORDER], a0i=0, a1i=0;
+  FIXP_SGL  bw = FL2FXCONST_SGL(0.0f);
+
+  int    autoCorrLength;
+
+  FIXP_DBL k1, k1_below=0, k1_below2=0;
+
+  ACORR_COEFS ac;
+  int    startSample;
+  int    stopSample;
+  int    stopSampleClear;
+
+  int comLowBandScale;
+  int ovLowBandShift;
+  int lowBandShift;
+/*  int ovHighBandShift;*/
+  int targetStopBand;
+
+
+  alphai[0] = FL2FXCONST_SGL(0.0f);
+  alphai[1] = FL2FXCONST_SGL(0.0f);
+
+
+  startSample = firstSlotOffs * timeStep;
+  stopSample  = pSettings->nCols + lastSlotOffs * timeStep;
+
+
+  inverseFilteringLevelEmphasis(hLppTrans, nInvfBands, sbr_invf_mode, sbr_invf_mode_prev, bwVector);
+
+  stopSampleClear = stopSample;
+
+  autoCorrLength = pSettings->nCols + pSettings->overlap;
+
+  /* Set upper subbands to zero:
+     This is required in case that the patches do not cover the complete highband
+     (because the last patch would be too short).
+     Possible optimization: Clearing bands up to usb would be sufficient here. */
+  targetStopBand = patchParam[pSettings->noOfPatches-1].targetStartBand
+                 + patchParam[pSettings->noOfPatches-1].numBandsInPatch;
+
+  int memSize = ((64) - targetStopBand) * sizeof(FIXP_DBL);
+
+  if (!useLP) {
+    for (i = startSample; i < stopSampleClear; i++) {
+      FDKmemclear(&qmfBufferReal[i][targetStopBand], memSize);
+      FDKmemclear(&qmfBufferImag[i][targetStopBand], memSize);
+    }
+  } else
+  for (i = startSample; i < stopSampleClear; i++) {
+    FDKmemclear(&qmfBufferReal[i][targetStopBand], memSize);
+  }
+
+  /* init bwIndex for each patch */
+  FDKmemclear(bwIndex, pSettings->noOfPatches*sizeof(INT));
+
+  /*
+    Calc common low band scale factor
+  */
+  comLowBandScale = fixMin(sbrScaleFactor->ov_lb_scale,sbrScaleFactor->lb_scale);
+
+  ovLowBandShift =  sbrScaleFactor->ov_lb_scale - comLowBandScale;
+  lowBandShift   =  sbrScaleFactor->lb_scale - comLowBandScale;
+  /*  ovHighBandShift = firstSlotOffs == 0 ? ovLowBandShift:0;*/
+
+  /* outer loop over bands to do analysis only once for each band */
+
+  if (!useLP) {
+    start = pSettings->lbStartPatching;
+    stop = pSettings->lbStopPatching;
+  } else
+  {
+    start = fixMax(1, pSettings->lbStartPatching - 2);
+    stop = patchParam[0].targetStartBand;
+  }
+
+
+  for ( loBand = start; loBand <  stop; loBand++ ) {
+
+    FIXP_DBL  lowBandReal[(((1024)/(32))+(6))+LPC_ORDER];
+    FIXP_DBL *plowBandReal = lowBandReal;
+    FIXP_DBL **pqmfBufferReal = qmfBufferReal;
+    FIXP_DBL  lowBandImag[(((1024)/(32))+(6))+LPC_ORDER];
+    FIXP_DBL *plowBandImag = lowBandImag;
+    FIXP_DBL **pqmfBufferImag = qmfBufferImag;
+    int resetLPCCoeffs=0;
+    int dynamicScale = DFRACT_BITS-1-LPC_SCALE_FACTOR;
+    int acDetScale = 0; /* scaling of autocorrelation determinant */
+
+    for(i=0;i<LPC_ORDER;i++){
+      *plowBandReal++ = hLppTrans->lpcFilterStatesReal[i][loBand];
+      if (!useLP)
+        *plowBandImag++ = hLppTrans->lpcFilterStatesImag[i][loBand];
+    }
+
+    /*
+      Take old slope length qmf slot source values out of (overlap)qmf buffer
+    */
+    if (!useLP) {
+      for(i=0;i<pSettings->nCols+pSettings->overlap;i++){
+        *plowBandReal++ = (*pqmfBufferReal++)[loBand];
+        *plowBandImag++ = (*pqmfBufferImag++)[loBand];
+      }
+    } else
+    {
+      /* pSettings->overlap is always even */
+      FDK_ASSERT((pSettings->overlap & 1) == 0);
+
+      for(i=0;i<((pSettings->overlap+pSettings->nCols)>>1);i++) {
+        *plowBandReal++ = (*pqmfBufferReal++)[loBand];
+        *plowBandReal++ = (*pqmfBufferReal++)[loBand];
+      }
+      if (pSettings->nCols & 1) {
+        *plowBandReal++ = (*pqmfBufferReal++)[loBand];
+      }
+    }
+
+    /*
+      Determine dynamic scaling value.
+     */
+    dynamicScale = fixMin(dynamicScale, getScalefactor(lowBandReal, LPC_ORDER+pSettings->overlap) + ovLowBandShift);
+    dynamicScale = fixMin(dynamicScale, getScalefactor(&lowBandReal[LPC_ORDER+pSettings->overlap], pSettings->nCols) + lowBandShift);
+    if (!useLP) {
+      dynamicScale = fixMin(dynamicScale, getScalefactor(lowBandImag, LPC_ORDER+pSettings->overlap) + ovLowBandShift);
+      dynamicScale = fixMin(dynamicScale, getScalefactor(&lowBandImag[LPC_ORDER+pSettings->overlap], pSettings->nCols) + lowBandShift);
+    }
+    dynamicScale = fixMax(0, dynamicScale-1); /* one additional bit headroom to prevent -1.0 */
+
+    /*
+      Scale temporal QMF buffer.
+     */
+    scaleValues(&lowBandReal[0], LPC_ORDER+pSettings->overlap, dynamicScale-ovLowBandShift);
+    scaleValues(&lowBandReal[LPC_ORDER+pSettings->overlap], pSettings->nCols, dynamicScale-lowBandShift);
+
+    if (!useLP) {
+      scaleValues(&lowBandImag[0], LPC_ORDER+pSettings->overlap, dynamicScale-ovLowBandShift);
+      scaleValues(&lowBandImag[LPC_ORDER+pSettings->overlap], pSettings->nCols, dynamicScale-lowBandShift);
+    }
+
+
+      if (!useLP) {
+        acDetScale += autoCorr2nd_cplx(&ac, lowBandReal+LPC_ORDER, lowBandImag+LPC_ORDER, autoCorrLength);
+      }
+      else
+      {
+        acDetScale += autoCorr2nd_real(&ac, lowBandReal+LPC_ORDER, autoCorrLength);
+      }
+
+      /* Examine dynamic of determinant in autocorrelation. */
+      acDetScale += 2*(comLowBandScale + dynamicScale);
+      acDetScale *= 2;              /* two times reflection coefficent scaling */
+      acDetScale += ac.det_scale;   /* ac scaling of determinant */
+
+      /* In case of determinant < 10^-38, resetLPCCoeffs=1 has to be enforced. */
+      if (acDetScale>126 ) {
+        resetLPCCoeffs = 1;
+      }
+
+
+    alphar[1] = FL2FXCONST_SGL(0.0f);
+    if (!useLP)
+      alphai[1] = FL2FXCONST_SGL(0.0f);
+
+    if (ac.det != FL2FXCONST_DBL(0.0f)) {
+      FIXP_DBL tmp,absTmp,absDet;
+
+      absDet = fixp_abs(ac.det);
+
+      if (!useLP) {
+        tmp = ( fMultDiv2(ac.r01r,ac.r12r) >> (LPC_SCALE_FACTOR-1) ) -
+              ( (fMultDiv2(ac.r01i,ac.r12i) + fMultDiv2(ac.r02r,ac.r11r)) >> (LPC_SCALE_FACTOR-1) );
+      } else
+      {
+        tmp = ( fMultDiv2(ac.r01r,ac.r12r) >> (LPC_SCALE_FACTOR-1) ) -
+              ( fMultDiv2(ac.r02r,ac.r11r) >> (LPC_SCALE_FACTOR-1) );
+      }
+      absTmp = fixp_abs(tmp);
+
+      /*
+        Quick check: is first filter coeff >= 1(4)
+       */
+      {
+        INT scale;
+        FIXP_DBL result = fDivNorm(absTmp, absDet, &scale);
+        scale = scale+ac.det_scale;
+
+        if ( (scale > 0) && (result >= (FIXP_DBL)MAXVAL_DBL>>scale) ) {
+          resetLPCCoeffs = 1;
+        }
+        else {
+          alphar[1] = FX_DBL2FX_SGL(scaleValue(result,scale));
+          if((tmp<FL2FX_DBL(0.0f)) ^ (ac.det<FL2FX_DBL(0.0f))) {
+            alphar[1] = -alphar[1];
+          }
+        }
+      }
+
+      if (!useLP)
+      {
+        tmp =  ( fMultDiv2(ac.r01i,ac.r12r) >> (LPC_SCALE_FACTOR-1) ) +
+               ( (fMultDiv2(ac.r01r,ac.r12i) - (FIXP_DBL)fMultDiv2(ac.r02i,ac.r11r)) >> (LPC_SCALE_FACTOR-1) ) ;
+
+        absTmp = fixp_abs(tmp);
+
+        /*
+        Quick check: is second filter coeff >= 1(4)
+        */
+        {
+          INT scale;
+          FIXP_DBL result = fDivNorm(absTmp, absDet, &scale);
+          scale = scale+ac.det_scale;
+
+          if ( (scale > 0) && (result >= /*FL2FXCONST_DBL(1.f)*/ (FIXP_DBL)MAXVAL_DBL>>scale) ) {
+            resetLPCCoeffs = 1;
+          }
+          else {
+            alphai[1] = FX_DBL2FX_SGL(scaleValue(result,scale));
+            if((tmp<FL2FX_DBL(0.0f)) ^ (ac.det<FL2FX_DBL(0.0f))) {
+              alphai[1] = -alphai[1];
+            }
+          }
+        }
+      }
+    }
+
+    alphar[0] =  FL2FXCONST_SGL(0.0f);
+    if (!useLP)
+      alphai[0] = FL2FXCONST_SGL(0.0f);
+
+    if ( ac.r11r != FL2FXCONST_DBL(0.0f) ) {
+
+      /* ac.r11r is always >=0 */
+      FIXP_DBL tmp,absTmp;
+
+      if (!useLP) {
+        tmp = (ac.r01r>>(LPC_SCALE_FACTOR+1)) +
+              (fMultDiv2(alphar[1],ac.r12r) + fMultDiv2(alphai[1],ac.r12i));
+      } else
+      {
+        if(ac.r01r>=FL2FXCONST_DBL(0.0f))
+          tmp = (ac.r01r>>(LPC_SCALE_FACTOR+1)) + fMultDiv2(alphar[1],ac.r12r);
+        else
+          tmp = -((-ac.r01r)>>(LPC_SCALE_FACTOR+1)) + fMultDiv2(alphar[1],ac.r12r);
+      }
+
+      absTmp = fixp_abs(tmp);
+
+      /*
+        Quick check: is first filter coeff >= 1(4)
+      */
+
+      if (absTmp >= (ac.r11r>>1)) {
+        resetLPCCoeffs=1;
+      }
+      else {
+        INT scale;
+        FIXP_DBL result = fDivNorm(absTmp, fixp_abs(ac.r11r), &scale);
+        alphar[0] =  FX_DBL2FX_SGL(scaleValue(result,scale+1));
+
+        if((tmp>FL2FX_DBL(0.0f)) ^ (ac.r11r<FL2FX_DBL(0.0f)))
+          alphar[0] = -alphar[0];
+      }
+
+      if (!useLP)
+      {
+        tmp = (ac.r01i>>(LPC_SCALE_FACTOR+1)) +
+              (fMultDiv2(alphai[1],ac.r12r) - fMultDiv2(alphar[1],ac.r12i));
+
+        absTmp = fixp_abs(tmp);
+
+        /*
+        Quick check: is second filter coeff >= 1(4)
+        */
+        if (absTmp >= (ac.r11r>>1)) {
+          resetLPCCoeffs=1;
+        }
+        else {
+          INT scale;
+          FIXP_DBL result = fDivNorm(absTmp, fixp_abs(ac.r11r), &scale);
+          alphai[0] = FX_DBL2FX_SGL(scaleValue(result,scale+1));
+          if((tmp>FL2FX_DBL(0.0f)) ^ (ac.r11r<FL2FX_DBL(0.0f)))
+            alphai[0] = -alphai[0];
+        }
+      }
+    }
+
+
+    if (!useLP)
+    {
+      /* Now check the quadratic criteria */
+      if( (fMultDiv2(alphar[0],alphar[0]) + fMultDiv2(alphai[0],alphai[0])) >= FL2FXCONST_DBL(0.5f) )
+        resetLPCCoeffs=1;
+      if( (fMultDiv2(alphar[1],alphar[1]) + fMultDiv2(alphai[1],alphai[1])) >= FL2FXCONST_DBL(0.5f) )
+        resetLPCCoeffs=1;
+    }
+
+    if(resetLPCCoeffs){
+      alphar[0] = FL2FXCONST_SGL(0.0f);
+      alphar[1] = FL2FXCONST_SGL(0.0f);
+      if (!useLP)
+      {
+        alphai[0] = FL2FXCONST_SGL(0.0f);
+        alphai[1] = FL2FXCONST_SGL(0.0f);
+      }
+    }
+
+    if (useLP)
+    {
+
+      /* Aliasing detection */
+      if(ac.r11r==FL2FXCONST_DBL(0.0f)) {
+        k1 = FL2FXCONST_DBL(0.0f);
+      }
+      else {
+        if ( fixp_abs(ac.r01r) >= fixp_abs(ac.r11r) ) {
+          if ( fMultDiv2(ac.r01r,ac.r11r) < FL2FX_DBL(0.0f)) {
+            k1 = (FIXP_DBL)MAXVAL_DBL /*FL2FXCONST_SGL(1.0f)*/;
+          }else {
+            /* Since this value is squared later, it must not ever become -1.0f. */
+            k1 = (FIXP_DBL)(MINVAL_DBL+1) /*FL2FXCONST_SGL(-1.0f)*/;
+          }
+        }
+        else {
+          INT scale;
+          FIXP_DBL result = fDivNorm(fixp_abs(ac.r01r), fixp_abs(ac.r11r), &scale);
+          k1 = scaleValue(result,scale);
+
+          if(!((ac.r01r<FL2FX_DBL(0.0f)) ^ (ac.r11r<FL2FX_DBL(0.0f)))) {
+            k1 = -k1;
+          }
+        }
+      }
+      if(loBand > 1){
+        /* Check if the gain should be locked */
+        FIXP_DBL deg = /*FL2FXCONST_DBL(1.0f)*/ (FIXP_DBL)MAXVAL_DBL - fPow2(k1_below);
+        degreeAlias[loBand] = FL2FXCONST_DBL(0.0f);
+        if (((loBand & 1) == 0) && (k1 < FL2FXCONST_DBL(0.0f))){
+          if (k1_below < FL2FXCONST_DBL(0.0f)) {         /* 2-Ch Aliasing Detection */
+            degreeAlias[loBand] = (FIXP_DBL)MAXVAL_DBL /*FL2FXCONST_DBL(1.0f)*/;
+            if ( k1_below2 > FL2FXCONST_DBL(0.0f) ) {    /* 3-Ch Aliasing Detection */
+              degreeAlias[loBand-1] = deg;
+            }
+          }
+          else if ( k1_below2 > FL2FXCONST_DBL(0.0f) ) { /* 3-Ch Aliasing Detection */
+            degreeAlias[loBand]   = deg;
+          }
+        }
+        if (((loBand & 1) == 1) && (k1 > FL2FXCONST_DBL(0.0f))){
+          if (k1_below > FL2FXCONST_DBL(0.0f)) {         /* 2-CH Aliasing Detection */
+            degreeAlias[loBand] = (FIXP_DBL)MAXVAL_DBL /*FL2FXCONST_DBL(1.0f)*/;
+            if ( k1_below2 < FL2FXCONST_DBL(0.0f) ) {    /* 3-CH Aliasing Detection */
+              degreeAlias[loBand-1] = deg;
+            }
+          }
+          else if ( k1_below2 < FL2FXCONST_DBL(0.0f) ) { /* 3-CH Aliasing Detection */
+            degreeAlias[loBand]   = deg;
+          }
+        }
+      }
+      /* remember k1 values of the 2 QMF channels below the current channel */
+      k1_below2 = k1_below;
+      k1_below = k1;
+    }
+
+    patch = 0;
+
+    while ( patch < pSettings->noOfPatches ) { /* inner loop over every patch */
+
+      int hiBand = loBand + patchParam[patch].targetBandOffs;
+
+      if ( loBand < patchParam[patch].sourceStartBand
+           || loBand >= patchParam[patch].sourceStopBand
+           //|| hiBand >= hLppTrans->pSettings->noChannels
+           ) {
+        /* Lowband not in current patch - proceed */
+        patch++;
+        continue;
+      }
+
+      FDK_ASSERT( hiBand < (64) );
+
+      /* bwIndex[patch] is already initialized with value from previous band inside this patch */
+      while (hiBand >= pSettings->bwBorders[bwIndex[patch]])
+        bwIndex[patch]++;
+
+
+      /*
+        Filter Step 2: add the left slope with the current filter to the buffer
+                       pure source values are already in there
+      */
+      bw = FX_DBL2FX_SGL(bwVector[bwIndex[patch]]);
+
+      a0r = FX_DBL2FX_SGL(fMult(bw,alphar[0])); /* Apply current bandwidth expansion factor */
+
+
+      if (!useLP)
+        a0i = FX_DBL2FX_SGL(fMult(bw,alphai[0]));
+      bw =  FX_DBL2FX_SGL(fPow2(bw));
+      a1r = FX_DBL2FX_SGL(fMult(bw,alphar[1]));
+      if (!useLP)
+        a1i = FX_DBL2FX_SGL(fMult(bw,alphai[1]));
+
+
+
+      /*
+        Filter Step 3: insert the middle part which won't be windowed
+      */
+
+      if ( bw <= FL2FXCONST_SGL(0.0f) ) {
+        if (!useLP) {
+          int descale = fixMin(DFRACT_BITS-1, (LPC_SCALE_FACTOR+dynamicScale));
+          for(i = startSample; i < stopSample; i++ ) {
+            qmfBufferReal[i][hiBand] = lowBandReal[LPC_ORDER+i]>>descale;
+            qmfBufferImag[i][hiBand] = lowBandImag[LPC_ORDER+i]>>descale;
+          }
+        } else
+        {
+          int descale = fixMin(DFRACT_BITS-1, (LPC_SCALE_FACTOR+dynamicScale));
+          for(i = startSample; i < stopSample; i++ ) {
+            qmfBufferReal[i][hiBand] = lowBandReal[LPC_ORDER+i]>>descale;
+          }
+        }
+      }
+      else {  /* bw <= 0 */
+
+        if (!useLP) {
+          int descale = fixMin(DFRACT_BITS-1, (LPC_SCALE_FACTOR+dynamicScale));
+#ifdef FUNCTION_LPPTRANSPOSER_func1
+          lppTransposer_func1(lowBandReal+LPC_ORDER+startSample,lowBandImag+LPC_ORDER+startSample,
+                              qmfBufferReal+startSample,qmfBufferImag+startSample,
+                              stopSample-startSample, (int) hiBand,
+                              dynamicScale,descale,
+                              a0r, a0i, a1r, a1i);
+#else
+          for(i = startSample; i < stopSample; i++ ) {
+            FIXP_DBL accu1, accu2;
+
+            accu1 = (fMultDiv2(a0r,lowBandReal[LPC_ORDER+i-1]) - fMultDiv2(a0i,lowBandImag[LPC_ORDER+i-1]) +
+                     fMultDiv2(a1r,lowBandReal[LPC_ORDER+i-2]) - fMultDiv2(a1i,lowBandImag[LPC_ORDER+i-2]))>>dynamicScale;
+            accu2 = (fMultDiv2(a0i,lowBandReal[LPC_ORDER+i-1]) + fMultDiv2(a0r,lowBandImag[LPC_ORDER+i-1]) +
+                     fMultDiv2(a1i,lowBandReal[LPC_ORDER+i-2]) + fMultDiv2(a1r,lowBandImag[LPC_ORDER+i-2]))>>dynamicScale;
+
+            qmfBufferReal[i][hiBand] = (lowBandReal[LPC_ORDER+i]>>descale) + (accu1<<1);
+            qmfBufferImag[i][hiBand] = (lowBandImag[LPC_ORDER+i]>>descale) + (accu2<<1);
+          }
+#endif
+        } else
+        {
+          int descale = fixMin(DFRACT_BITS-1, (LPC_SCALE_FACTOR+dynamicScale));
+
+          FDK_ASSERT(dynamicScale >= 0);
+          for(i = startSample; i < stopSample; i++ ) {
+            FIXP_DBL accu1;
+
+            accu1 = (fMultDiv2(a0r,lowBandReal[LPC_ORDER+i-1]) + fMultDiv2(a1r,lowBandReal[LPC_ORDER+i-2]))>>dynamicScale;
+
+            qmfBufferReal[i][hiBand] = (lowBandReal[LPC_ORDER+i]>>descale) + (accu1<<1);
+          }
+        }
+      } /* bw <= 0 */
+
+      patch++;
+
+    }  /* inner loop over patches */
+
+     /*
+     * store the unmodified filter coefficients if there is
+     * an overlapping envelope
+     *****************************************************************/
+
+
+  }  /* outer loop over bands (loBand) */
+
+  if (useLP)
+  {
+    for ( loBand = pSettings->lbStartPatching; loBand <  pSettings->lbStopPatching; loBand++ ) {
+      patch = 0;
+      while ( patch < pSettings->noOfPatches ) {
+
+        UCHAR hiBand = loBand + patchParam[patch].targetBandOffs;
+
+        if ( loBand < patchParam[patch].sourceStartBand
+          || loBand >= patchParam[patch].sourceStopBand
+          || hiBand >= (64)              /* Highband out of range (biterror) */
+          ) {
+          /* Lowband not in current patch or highband out of range (might be caused by biterrors)- proceed */
+          patch++;
+          continue;
+        }
+
+        if(hiBand != patchParam[patch].targetStartBand)
+          degreeAlias[hiBand] = degreeAlias[loBand];
+
+        patch++;
+      }
+    }/* end  for loop */
+  }
+
+ for (i = 0; i < nInvfBands; i++ ) {
+   hLppTrans->bwVectorOld[i] = bwVector[i];
+ }
+
+  /*
+    set high band scale factor
+  */
+  sbrScaleFactor->hb_scale = comLowBandScale-(LPC_SCALE_FACTOR);
+
+}
+
+/*!
+ *
+ * \brief Initialize one low power transposer instance
+ *
+ *
+ */
+SBR_ERROR
+createLppTransposer (HANDLE_SBR_LPP_TRANS hs, /*!< Handle of low power transposer  */
+                     TRANSPOSER_SETTINGS *pSettings, /*!< Pointer to settings */
+                     const int  highBandStartSb, /*!< ? */
+                     UCHAR *v_k_master,         /*!< Master table */
+                     const int numMaster,       /*!< Valid entries in master table */
+                     const int usb,             /*!< Highband area stop subband */
+                     const int timeSlots,       /*!< Number of time slots */
+                     const int nCols,           /*!< Number of colums (codec qmf bank) */
+                     UCHAR *noiseBandTable,     /*!< Mapping of SBR noise bands to QMF bands */
+                     const int  noNoiseBands,   /*!< Number of noise bands */
+                     UINT   fs,                 /*!< Sample Frequency */
+                     const int chan,            /*!< Channel number */
+                     const int overlap
+                     )
+{
+  /* FB inverse filtering settings */
+  hs->pSettings = pSettings;
+
+  pSettings->nCols = nCols;
+  pSettings->overlap = overlap;
+
+  switch (timeSlots) {
+
+  case 15:
+  case 16:
+    break;
+
+  default:
+    return SBRDEC_UNSUPPORTED_CONFIG; /* Unimplemented */
+  }
+
+  if (chan==0) {
+    /* Init common data only once */
+    hs->pSettings->nCols = nCols;
+
+    return resetLppTransposer (hs,
+                               highBandStartSb,
+                               v_k_master,
+                               numMaster,
+                               noiseBandTable,
+                               noNoiseBands,
+                               usb,
+                               fs);
+  }
+  return SBRDEC_OK;
+}
+
+
+static int findClosestEntry(UCHAR goalSb, UCHAR *v_k_master, UCHAR numMaster, UCHAR direction)
+{
+  int index;
+
+  if( goalSb <= v_k_master[0] )
+    return v_k_master[0];
+
+  if( goalSb >= v_k_master[numMaster] )
+    return v_k_master[numMaster];
+
+  if(direction) {
+    index = 0;
+    while( v_k_master[index] < goalSb ) {
+      index++;
+    }
+  } else {
+    index = numMaster;
+    while( v_k_master[index] > goalSb ) {
+      index--;
+    }
+  }
+
+  return v_k_master[index];
+}
+
+
+/*!
+ *
+ * \brief Reset memory for one lpp transposer instance
+ *
+ * \return SBRDEC_OK on success, SBRDEC_UNSUPPORTED_CONFIG on error
+ */
+SBR_ERROR
+resetLppTransposer (HANDLE_SBR_LPP_TRANS hLppTrans,  /*!< Handle of lpp transposer  */
+                    UCHAR  highBandStartSb,          /*!< High band area: start subband */
+                    UCHAR *v_k_master,               /*!< Master table */
+                    UCHAR  numMaster,                /*!< Valid entries in master table */
+                    UCHAR *noiseBandTable,           /*!< Mapping of SBR noise bands to QMF bands */
+                    UCHAR  noNoiseBands,             /*!< Number of noise bands */
+                    UCHAR  usb,                      /*!< High band area: stop subband */
+                    UINT   fs                        /*!< SBR output sampling frequency */
+                    )
+{
+  TRANSPOSER_SETTINGS *pSettings = hLppTrans->pSettings;
+  PATCH_PARAM  *patchParam = pSettings->patchParam;
+
+  int i, patch;
+  int targetStopBand;
+  int sourceStartBand;
+  int patchDistance;
+  int numBandsInPatch;
+
+  int lsb = v_k_master[0];                 /* Start subband expressed in "non-critical" sampling terms*/
+  int xoverOffset = highBandStartSb - lsb; /* Calculate distance in QMF bands between k0 and kx */
+  int startFreqHz;
+
+  int desiredBorder;
+
+  usb = fixMin(usb, v_k_master[numMaster]); /* Avoid endless loops (compare with float code). */
+
+  /*
+   * Plausibility check
+   */
+
+  if ( lsb - SHIFT_START_SB < 4 ) {
+    return SBRDEC_UNSUPPORTED_CONFIG;
+  }
+
+
+  /*
+   * Initialize the patching parameter
+   */
+ desiredBorder = 21;
+ if (fs < 92017) {
+    desiredBorder = 23;
+  }
+  if (fs < 75132) {
+    desiredBorder = 32;
+  }
+  if (fs < 55426) {
+    desiredBorder = 43;
+  }
+  if (fs < 46009) {
+    desiredBorder = 46;
+  }
+  if (fs < 35777) {
+    desiredBorder = 64;
+  }
+
+  desiredBorder = findClosestEntry(desiredBorder, v_k_master, numMaster, 1); /* Adapt region to master-table */
+
+  /* First patch */
+  sourceStartBand = SHIFT_START_SB + xoverOffset;
+  targetStopBand = lsb + xoverOffset; /* upperBand */
+
+  /* Even (odd) numbered channel must be patched to even (odd) numbered channel */
+  patch = 0;
+  while(targetStopBand < usb) {
+
+    /* Too many patches?
+       Allow MAX_NUM_PATCHES+1 patches here.
+       we need to check later again, since patch might be the highest patch
+       AND contain less than 3 bands => actual number of patches will be reduced by 1.
+    */
+    if (patch > MAX_NUM_PATCHES) {
+      return SBRDEC_UNSUPPORTED_CONFIG;
+    }
+
+    patchParam[patch].guardStartBand = targetStopBand;
+    patchParam[patch].targetStartBand = targetStopBand;
+
+    numBandsInPatch = desiredBorder - targetStopBand;                   /* Get the desired range of the patch */
+
+    if ( numBandsInPatch >= lsb - sourceStartBand ) {
+      /* Desired number bands are not available -> patch whole source range */
+      patchDistance   = targetStopBand - sourceStartBand;        /* Get the targetOffset */
+      patchDistance   = patchDistance & ~1;                      /* Rounding off odd numbers and make all even */
+      numBandsInPatch = lsb - (targetStopBand - patchDistance);  /* Update number of bands to be patched */
+      numBandsInPatch = findClosestEntry(targetStopBand + numBandsInPatch, v_k_master, numMaster, 0) -
+                        targetStopBand;  /* Adapt region to master-table */
+    }
+
+    /* Desired number bands are available -> get the minimal even patching distance */
+    patchDistance   = numBandsInPatch + targetStopBand - lsb;  /* Get minimal distance */
+    patchDistance   = (patchDistance + 1) & ~1;                /* Rounding up odd numbers and make all even */
+
+    if (numBandsInPatch > 0) {
+      patchParam[patch].sourceStartBand = targetStopBand - patchDistance;
+      patchParam[patch].targetBandOffs  = patchDistance;
+      patchParam[patch].numBandsInPatch = numBandsInPatch;
+      patchParam[patch].sourceStopBand  = patchParam[patch].sourceStartBand + numBandsInPatch;
+
+      targetStopBand += patchParam[patch].numBandsInPatch;
+      patch++;
+    }
+
+    /* All patches but first */
+    sourceStartBand = SHIFT_START_SB;
+
+    /* Check if we are close to desiredBorder */
+    if( desiredBorder - targetStopBand < 3)  /* MPEG doc */
+    {
+      desiredBorder = usb;
+    }
+
+  }
+
+  patch--;
+
+  /* If highest patch contains less than three subband: skip it */
+  if ( (patch>0) && (patchParam[patch].numBandsInPatch < 3) ) {
+    patch--;
+    targetStopBand = patchParam[patch].targetStartBand + patchParam[patch].numBandsInPatch;
+  }
+
+  /* now check if we don't have one too many */
+  if (patch >= MAX_NUM_PATCHES) {
+    return SBRDEC_UNSUPPORTED_CONFIG;
+  }
+
+  pSettings->noOfPatches = patch + 1;
+
+  /* Check lowest and highest source subband */
+  pSettings->lbStartPatching = targetStopBand;
+  pSettings->lbStopPatching  = 0;
+  for ( patch = 0; patch < pSettings->noOfPatches; patch++ ) {
+    pSettings->lbStartPatching = fixMin( pSettings->lbStartPatching, patchParam[patch].sourceStartBand );
+    pSettings->lbStopPatching  = fixMax( pSettings->lbStopPatching, patchParam[patch].sourceStopBand );
+  }
+
+  for(i = 0 ; i < noNoiseBands; i++){
+    pSettings->bwBorders[i] = noiseBandTable[i+1];
+  }
+
+  /*
+   * Choose whitening factors
+   */
+
+  startFreqHz = ( (lsb + xoverOffset)*fs ) >> 7;  /* Shift does a division by 2*(64) */
+
+  for( i = 1; i < NUM_WHFACTOR_TABLE_ENTRIES; i++ )
+  {
+    if( startFreqHz < FDK_sbrDecoder_sbr_whFactorsIndex[i])
+      break;
+  }
+  i--;
+
+  pSettings->whFactors.off = FDK_sbrDecoder_sbr_whFactorsTable[i][0];
+  pSettings->whFactors.transitionLevel = FDK_sbrDecoder_sbr_whFactorsTable[i][1];
+  pSettings->whFactors.lowLevel = FDK_sbrDecoder_sbr_whFactorsTable[i][2];
+  pSettings->whFactors.midLevel = FDK_sbrDecoder_sbr_whFactorsTable[i][3];
+  pSettings->whFactors.highLevel = FDK_sbrDecoder_sbr_whFactorsTable[i][4];
+
+  return SBRDEC_OK;
+}