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diff --git a/fdk-aac/libSBRdec/src/lpp_tran.cpp b/fdk-aac/libSBRdec/src/lpp_tran.cpp
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+/* -----------------------------------------------------------------------------
+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  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
+*/
+
+#ifdef __ANDROID__
+#include "log/log.h"
+#endif
+
+#include "lpp_tran.h"
+
+#include "sbr_ram.h"
+#include "sbr_rom.h"
+
+#include "genericStds.h"
+#include "autocorr2nd.h"
+
+#include "HFgen_preFlat.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.
+ */
+
+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;
+  }
+}
+
+/*!
+ *
+ * \brief Perform inverse filtering level emphasis
+ *
+ * Retrieve bandwidth expansion factor and apply smoothing for each filter band
+ *
+ */
+
+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));
+    }
+  }
+}
+
+/* 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)
+
+static inline void calc_qmfBufferReal(FIXP_DBL **qmfBufferReal,
+                                      const FIXP_DBL *const lowBandReal,
+                                      const int startSample,
+                                      const int stopSample, const UCHAR hiBand,
+                                      const int dynamicScale, const int descale,
+                                      const FIXP_SGL a0r, const FIXP_SGL a1r) {
+  FIXP_DBL accu1, accu2;
+  int i;
+
+  for (i = 0; i < stopSample - startSample; i++) {
+    accu1 = fMultDiv2(a1r, lowBandReal[i]);
+    accu1 = (fMultDiv2(a0r, lowBandReal[i + 1]) + accu1);
+    accu1 = accu1 >> dynamicScale;
+
+    accu1 <<= 1;
+    accu2 = (lowBandReal[i + 2] >> descale);
+    qmfBufferReal[i + startSample][hiBand] = accu1 + accu2;
+  }
+}
+
+/*!
+ *
+ * \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 fPreWhitening, const int v_k_master0,
+    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 */
+  FIXP_DBL preWhiteningGains[(64) / 2];
+  int preWhiteningGains_exp[(64) / 2];
+
+  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;*/
+
+  alphai[0] = FL2FXCONST_SGL(0.0f);
+  alphai[1] = FL2FXCONST_SGL(0.0f);
+
+  startSample = firstSlotOffs * timeStep;
+  stopSample = pSettings->nCols + lastSlotOffs * timeStep;
+  FDK_ASSERT((lastSlotOffs * timeStep) <= pSettings->overlap);
+
+  inverseFilteringLevelEmphasis(hLppTrans, nInvfBands, sbr_invf_mode,
+                                sbr_invf_mode_prev, bwVector);
+
+  stopSampleClear = stopSample;
+
+  autoCorrLength = pSettings->nCols + pSettings->overlap;
+
+  if (pSettings->noOfPatches > 0) {
+    /* 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. */
+    int 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);
+      }
+    }
+  }
+#ifdef __ANDROID__
+  else {
+    // Safetynet logging
+    android_errorWriteLog(0x534e4554, "112160868");
+  }
+#endif
+
+  /* init bwIndex for each patch */
+  FDKmemclear(bwIndex, sizeof(bwIndex));
+
+  /*
+    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;*/
+
+  if (fPreWhitening) {
+    sbrDecoder_calculateGainVec(
+        qmfBufferReal, qmfBufferImag,
+        DFRACT_BITS - 1 - 16 -
+            sbrScaleFactor->ov_lb_scale, /* convert scale to exponent */
+        DFRACT_BITS - 1 - 16 -
+            sbrScaleFactor->lb_scale, /* convert scale to exponent */
+        pSettings->overlap, preWhiteningGains, preWhiteningGains_exp,
+        v_k_master0, startSample, stopSample);
+  }
+
+  /* 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) * (4) / 2) + (3 * (4))) + LPC_ORDER];
+    FIXP_DBL *plowBandReal = lowBandReal;
+    FIXP_DBL **pqmfBufferReal =
+        qmfBufferReal + firstSlotOffs * timeStep /* + pSettings->overlap */;
+    FIXP_DBL lowBandImag[(((1024) / (32) * (4) / 2) + (3 * (4))) + LPC_ORDER];
+    FIXP_DBL *plowBandImag = lowBandImag;
+    FIXP_DBL **pqmfBufferImag =
+        qmfBufferImag + firstSlotOffs * timeStep /* + pSettings->overlap */;
+    int resetLPCCoeffs = 0;
+    int dynamicScale = DFRACT_BITS - 1 - LPC_SCALE_FACTOR;
+    int acDetScale = 0; /* scaling of autocorrelation determinant */
+
+    for (i = 0;
+         i < LPC_ORDER + firstSlotOffs * timeStep /*+pSettings->overlap*/;
+         i++) {
+      *plowBandReal++ = hLppTrans->lpcFilterStatesRealLegSBR[i][loBand];
+      if (!useLP)
+        *plowBandImag++ = hLppTrans->lpcFilterStatesImagLegSBR[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 - firstSlotOffs * timeStep;
+           i++) {
+        *plowBandReal++ = (*pqmfBufferReal++)[loBand];
+        *plowBandImag++ = (*pqmfBufferImag++)[loBand];
+      }
+    } else {
+      /* pSettings->overlap is always even */
+      FDK_ASSERT((pSettings->overlap & 1) == 0);
+      for (i = 0; i < ((pSettings->nCols + pSettings->overlap -
+                        firstSlotOffs * timeStep) >>
+                       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) && (loBand < v_k_master0)) {
+        /* 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] < MAX_NUM_PATCHES - 1) {
+        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++) {
+            FIXP_DBL accu1, accu2;
+            accu1 = lowBandReal[LPC_ORDER + i] >> descale;
+            accu2 = lowBandImag[LPC_ORDER + i] >> descale;
+            if (fPreWhitening) {
+              accu1 = scaleValueSaturate(
+                  fMultDiv2(accu1, preWhiteningGains[loBand]),
+                  preWhiteningGains_exp[loBand] + 1);
+              accu2 = scaleValueSaturate(
+                  fMultDiv2(accu2, preWhiteningGains[loBand]),
+                  preWhiteningGains_exp[loBand] + 1);
+            }
+            qmfBufferReal[i][hiBand] = accu1;
+            qmfBufferImag[i][hiBand] = accu2;
+          }
+        } 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, fPreWhitening, preWhiteningGains[loBand],
+              preWhiteningGains_exp[loBand] + 1);
+#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;
+
+            accu1 = (lowBandReal[LPC_ORDER + i] >> descale) + (accu1 << 1);
+            accu2 = (lowBandImag[LPC_ORDER + i] >> descale) + (accu2 << 1);
+            if (fPreWhitening) {
+              accu1 = scaleValueSaturate(
+                  fMultDiv2(accu1, preWhiteningGains[loBand]),
+                  preWhiteningGains_exp[loBand] + 1);
+              accu2 = scaleValueSaturate(
+                  fMultDiv2(accu2, preWhiteningGains[loBand]),
+                  preWhiteningGains_exp[loBand] + 1);
+            }
+            qmfBufferReal[i][hiBand] = accu1;
+            qmfBufferImag[i][hiBand] = accu2;
+          }
+#endif
+        } else {
+          FDK_ASSERT(dynamicScale >= 0);
+          calc_qmfBufferReal(
+              qmfBufferReal, &(lowBandReal[LPC_ORDER + startSample - 2]),
+              startSample, stopSample, hiBand, dynamicScale,
+              fMin(DFRACT_BITS - 1, (LPC_SCALE_FACTOR + dynamicScale)), a0r,
+              a1r);
+        }
+      } /* 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);
+}
+
+void lppTransposerHBE(
+    HANDLE_SBR_LPP_TRANS hLppTrans, /*!< Handle of lpp transposer  */
+    HANDLE_HBE_TRANSPOSER hQmfTransposer,
+    QMF_SCALE_FACTOR *sbrScaleFactor, /*!< Scaling factors */
+    FIXP_DBL **qmfBufferReal, /*!< Pointer to pointer to real part of subband
+                                 samples (source) */
+    FIXP_DBL **qmfBufferImag, /*!< Pointer to pointer to imaginary part of
+                                 subband samples (source) */
+    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;
+  FIXP_DBL bwVector[MAX_NUM_PATCHES_HBE]; /*!< pole moving factors */
+
+  int i;
+  int loBand, start, stop;
+  TRANSPOSER_SETTINGS *pSettings = hLppTrans->pSettings;
+  PATCH_PARAM *patchParam = pSettings->patchParam;
+
+  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;
+
+  ACORR_COEFS ac;
+  int startSample;
+  int stopSample;
+  int stopSampleClear;
+
+  int comBandScale;
+  int ovLowBandShift;
+  int lowBandShift;
+  /*  int ovHighBandShift;*/
+
+  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;
+
+  if (pSettings->noOfPatches > 0) {
+    /* 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. */
+    int targetStopBand =
+        patchParam[pSettings->noOfPatches - 1].targetStartBand +
+        patchParam[pSettings->noOfPatches - 1].numBandsInPatch;
+
+    int memSize = ((64) - targetStopBand) * sizeof(FIXP_DBL);
+
+    for (i = startSample; i < stopSampleClear; i++) {
+      FDKmemclear(&qmfBufferReal[i][targetStopBand], memSize);
+      FDKmemclear(&qmfBufferImag[i][targetStopBand], memSize);
+    }
+  }
+#ifdef __ANDROID__
+  else {
+    // Safetynet logging
+    android_errorWriteLog(0x534e4554, "112160868");
+  }
+#endif
+
+  /*
+  Calc common low band scale factor
+  */
+  comBandScale = sbrScaleFactor->hb_scale;
+
+  ovLowBandShift = sbrScaleFactor->hb_scale - comBandScale;
+  lowBandShift = sbrScaleFactor->hb_scale - comBandScale;
+  /*  ovHighBandShift = firstSlotOffs == 0 ? ovLowBandShift:0;*/
+
+  /* outer loop over bands to do analysis only once for each band */
+
+  start = hQmfTransposer->startBand;
+  stop = hQmfTransposer->stopBand;
+
+  for (loBand = start; loBand < stop; loBand++) {
+    bwIndex = 0;
+
+    FIXP_DBL lowBandReal[(((1024) / (32) * (4) / 2) + (3 * (4))) + LPC_ORDER];
+    FIXP_DBL lowBandImag[(((1024) / (32) * (4) / 2) + (3 * (4))) + LPC_ORDER];
+
+    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++) {
+      lowBandReal[i] = hLppTrans->lpcFilterStatesRealHBE[i][loBand];
+      lowBandImag[i] = hLppTrans->lpcFilterStatesImagHBE[i][loBand];
+    }
+
+    for (; i < LPC_ORDER + firstSlotOffs * timeStep; i++) {
+      lowBandReal[i] = hLppTrans->lpcFilterStatesRealHBE[i][loBand];
+      lowBandImag[i] = hLppTrans->lpcFilterStatesImagHBE[i][loBand];
+    }
+
+    /*
+    Take old slope length qmf slot source values out of (overlap)qmf buffer
+    */
+    for (i = firstSlotOffs * timeStep;
+         i < pSettings->nCols + pSettings->overlap; i++) {
+      lowBandReal[i + LPC_ORDER] = qmfBufferReal[i][loBand];
+      lowBandImag[i + LPC_ORDER] = qmfBufferImag[i][loBand];
+    }
+
+    /* store unmodified values to buffer */
+    for (i = 0; i < LPC_ORDER + pSettings->overlap; i++) {
+      hLppTrans->lpcFilterStatesRealHBE[i][loBand] =
+          qmfBufferReal[pSettings->nCols - LPC_ORDER + i][loBand];
+      hLppTrans->lpcFilterStatesImagHBE[i][loBand] =
+          qmfBufferImag[pSettings->nCols - LPC_ORDER + i][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);
+    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);
+    scaleValues(&lowBandImag[0], LPC_ORDER + pSettings->overlap,
+                dynamicScale - ovLowBandShift);
+    scaleValues(&lowBandImag[LPC_ORDER + pSettings->overlap], pSettings->nCols,
+                dynamicScale - lowBandShift);
+
+    acDetScale += autoCorr2nd_cplx(&ac, lowBandReal + LPC_ORDER,
+                                   lowBandImag + LPC_ORDER, autoCorrLength);
+
+    /* Examine dynamic of determinant in autocorrelation. */
+    acDetScale += 2 * (comBandScale + 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);
+    alphai[1] = FL2FXCONST_SGL(0.0f);
+
+    if (ac.det != FL2FXCONST_DBL(0.0f)) {
+      FIXP_DBL tmp, absTmp, absDet;
+
+      absDet = fixp_abs(ac.det);
+
+      tmp = (fMultDiv2(ac.r01r, ac.r12r) >> (LPC_SCALE_FACTOR - 1)) -
+            ((fMultDiv2(ac.r01i, ac.r12i) + 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];
+          }
+        }
+      }
+
+      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);
+    alphai[0] = FL2FXCONST_SGL(0.0f);
+
+    if (ac.r11r != FL2FXCONST_DBL(0.0f)) {
+      /* ac.r11r is always >=0 */
+      FIXP_DBL tmp, absTmp;
+
+      tmp = (ac.r01r >> (LPC_SCALE_FACTOR + 1)) +
+            (fMultDiv2(alphar[1], ac.r12r) + fMultDiv2(alphai[1], ac.r12i));
+
+      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];
+      }
+
+      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];
+        }
+      }
+    }
+
+    /* 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);
+      alphai[0] = FL2FXCONST_SGL(0.0f);
+      alphai[1] = FL2FXCONST_SGL(0.0f);
+    }
+
+    while (bwIndex < MAX_NUM_PATCHES - 1 &&
+           loBand >= pSettings->bwBorders[bwIndex]) {
+      bwIndex++;
+    }
+
+    /*
+    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]);
+
+    a0r = FX_DBL2FX_SGL(
+        fMult(bw, alphar[0])); /* Apply current bandwidth expansion factor */
+    a0i = FX_DBL2FX_SGL(fMult(bw, alphai[0]));
+    bw = FX_DBL2FX_SGL(fPow2(bw));
+    a1r = FX_DBL2FX_SGL(fMult(bw, alphar[1]));
+    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)) {
+      int descale = fixMin(DFRACT_BITS - 1, (LPC_SCALE_FACTOR + dynamicScale));
+      for (i = startSample; i < stopSample; i++) {
+        qmfBufferReal[i][loBand] = lowBandReal[LPC_ORDER + i] >> descale;
+        qmfBufferImag[i][loBand] = lowBandImag[LPC_ORDER + i] >> descale;
+      }
+    } else { /* bw <= 0 */
+
+      int descale = fixMin(DFRACT_BITS - 1, (LPC_SCALE_FACTOR + dynamicScale));
+
+      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][loBand] =
+            (lowBandReal[LPC_ORDER + i] >> descale) + (accu1 << 1);
+        qmfBufferImag[i][loBand] =
+            (lowBandImag[LPC_ORDER + i] >> descale) + (accu2 << 1);
+      }
+    } /* bw <= 0 */
+
+    /*
+     * store the unmodified filter coefficients if there is
+     * an overlapping envelope
+     *****************************************************************/
+
+  } /* outer loop over bands (loBand) */
+
+  for (i = 0; i < nInvfBands; i++) {
+    hLppTrans->bwVectorOld[i] = bwVector[i];
+  }
+
+  /*
+  set high band scale factor
+  */
+  sbrScaleFactor->hb_scale = comBandScale - (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 (pSettings->nCols == 64) {
+    if (lsb < 4) {
+      /* 4:1 SBR Requirement k0 >= 4 missed! */
+      return SBRDEC_UNSUPPORTED_CONFIG;
+    }
+  } else if (lsb - SHIFT_START_SB < 4) {
+    return SBRDEC_UNSUPPORTED_CONFIG;
+  }
+
+  /*
+   * Initialize the patching parameter
+   */
+  /* ISO/IEC 14496-3 (Figure 4.48): goalSb = round( 2.048e6 / fs ) */
+  desiredBorder = (((2048000 * 2) / fs) + 1) >> 1;
+
+  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 */
+    }
+
+    if (pSettings->nCols == 64) {
+      if (numBandsInPatch == 0 && sourceStartBand == SHIFT_START_SB) {
+        return SBRDEC_UNSUPPORTED_CONFIG;
+      }
+    }
+
+    /* 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];
+  }
+  for (; i < MAX_NUM_NOISE_VALUES; i++) {
+    pSettings->bwBorders[i] = 255;
+  }
+
+  /*
+   * 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;
+}