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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
+----------------------------------------------------------------------------------------------------------- */
+
+/*****************************  MPEG-4 AAC Decoder  **************************
+
+   Author(s):   Josef Hoepfl
+   Description: long/short-block decoding
+
+******************************************************************************/
+
+#include "block.h"
+
+#include "aac_rom.h"
+#include "FDK_bitstream.h"
+#include "FDK_tools_rom.h"
+
+
+
+
+#include "aacdec_hcr.h"
+#include "rvlc.h"
+
+
+#if defined(__arm__)
+#include "arm/block_arm.cpp"
+#endif
+
+/*!
+  \brief Read escape sequence of codeword
+
+  The function reads the escape sequence from the bitstream,
+  if the absolute value of the quantized coefficient has the
+  value 16.
+
+  \return  quantized coefficient
+*/
+LONG CBlock_GetEscape(HANDLE_FDK_BITSTREAM bs, /*!< pointer to bitstream */
+                     const LONG q)        /*!< quantized coefficient */
+{
+  LONG i, off, neg ;
+
+  if (q < 0)
+  {
+    if (q != -16) return q;
+    neg = 1;
+  }
+  else
+  {
+    if (q != +16) return q;
+    neg = 0;
+  }
+
+  for (i=4; ; i++)
+  {
+    if (FDKreadBits(bs,1) == 0)
+      break;
+  }
+
+  if (i > 16)
+  {
+    if (i - 16 > CACHE_BITS) { /* cannot read more than "CACHE_BITS" bits at once in the function FDKreadBits() */
+      return (MAX_QUANTIZED_VALUE + 1); /* returning invalid value that will be captured later */
+    }
+
+    off = FDKreadBits(bs,i-16) << 16;
+    off |= FDKreadBits(bs,16);
+  }
+  else
+  {
+    off = FDKreadBits(bs,i);
+  }
+
+  i = off + (1 << i);
+
+  if (neg) i = -i;
+
+  return i;
+}
+
+AAC_DECODER_ERROR CBlock_ReadScaleFactorData(
+        CAacDecoderChannelInfo *pAacDecoderChannelInfo,
+        HANDLE_FDK_BITSTREAM bs,
+        UINT flags
+        )
+{
+  int temp;
+  int band;
+  int group;
+  int position = 0; /* accu for intensity delta coding */
+  int factor = pAacDecoderChannelInfo->pDynData->RawDataInfo.GlobalGain; /* accu for scale factor delta coding */
+  UCHAR *pCodeBook = pAacDecoderChannelInfo->pDynData->aCodeBook;
+  SHORT *pScaleFactor = pAacDecoderChannelInfo->pDynData->aScaleFactor;
+  const CodeBookDescription *hcb =&AACcodeBookDescriptionTable[BOOKSCL];
+
+  int ScaleFactorBandsTransmitted = GetScaleFactorBandsTransmitted(&pAacDecoderChannelInfo->icsInfo);
+  for (group=0; group < GetWindowGroups(&pAacDecoderChannelInfo->icsInfo); group++)
+  {
+    for (band=0; band < ScaleFactorBandsTransmitted; band++)
+    {
+      switch (pCodeBook[group*16+band]) {
+
+      case ZERO_HCB: /* zero book */
+        pScaleFactor[group*16+band] = 0;
+        break;
+
+      default: /* decode scale factor */
+        {
+          temp = CBlock_DecodeHuffmanWord(bs,hcb);
+          factor += temp - 60; /* MIDFAC 1.5 dB */
+        }
+        pScaleFactor[group*16+band] = factor - 100;
+        break;
+
+      case INTENSITY_HCB: /* intensity steering */
+      case INTENSITY_HCB2:
+        temp = CBlock_DecodeHuffmanWord(bs,hcb);
+        position += temp - 60;
+        pScaleFactor[group*16+band] = position - 100;
+        break;
+
+      case NOISE_HCB: /* PNS */
+        if (flags & (AC_MPS_RES|AC_USAC|AC_RSVD50)) {
+          return AAC_DEC_PARSE_ERROR;
+        }
+        CPns_Read( &pAacDecoderChannelInfo->data.aac.PnsData, bs, hcb, pAacDecoderChannelInfo->pDynData->aScaleFactor, pAacDecoderChannelInfo->pDynData->RawDataInfo.GlobalGain, band, group);
+        break;
+      }
+    }
+  }
+
+  return AAC_DEC_OK;
+}
+
+void CBlock_ScaleSpectralData(CAacDecoderChannelInfo *pAacDecoderChannelInfo, SamplingRateInfo *pSamplingRateInfo)
+{
+  int band;
+  int window;
+  const SHORT * RESTRICT pSfbScale  = pAacDecoderChannelInfo->pDynData->aSfbScale;
+  SHORT * RESTRICT pSpecScale = pAacDecoderChannelInfo->specScale;
+  int groupwin,group;
+  const SHORT * RESTRICT BandOffsets = GetScaleFactorBandOffsets(&pAacDecoderChannelInfo->icsInfo, pSamplingRateInfo);
+  SPECTRAL_PTR RESTRICT pSpectralCoefficient = pAacDecoderChannelInfo->pSpectralCoefficient;
+
+
+  FDKmemclear(pSpecScale, 8*sizeof(SHORT));
+
+  int max_band = GetScaleFactorBandsTransmitted(&pAacDecoderChannelInfo->icsInfo);
+  for (window=0, group=0; group < GetWindowGroups(&pAacDecoderChannelInfo->icsInfo); group++)
+  {
+    for (groupwin=0; groupwin < GetWindowGroupLength(&pAacDecoderChannelInfo->icsInfo,group); groupwin++, window++)
+    {
+      int SpecScale_window = pSpecScale[window];
+      FIXP_DBL *pSpectrum = SPEC(pSpectralCoefficient, window,  pAacDecoderChannelInfo->granuleLength);
+
+      /* find scaling for current window */
+      for (band=0; band < max_band; band++)
+      {
+        SpecScale_window = fMax(SpecScale_window, (int)pSfbScale[window*16+band]);
+      }
+
+      if (pAacDecoderChannelInfo->pDynData->TnsData.Active) {
+        SpecScale_window += TNS_SCALE;
+      }
+
+      /* store scaling of current window */
+      pSpecScale[window] = SpecScale_window;
+
+#ifdef FUNCTION_CBlock_ScaleSpectralData_func1
+
+      CBlock_ScaleSpectralData_func1(pSpectrum, max_band, BandOffsets, SpecScale_window, pSfbScale, window);
+
+#else /* FUNCTION_CBlock_ScaleSpectralData_func1 */
+      for (band=0; band < max_band; band++)
+      {
+        int scale = SpecScale_window - pSfbScale[window*16+band];
+        if (scale)
+        {
+          /* following relation can be used for optimizations: (BandOffsets[i]%4) == 0 for all i */
+          int max_index = BandOffsets[band+1];
+          for (int index = BandOffsets[band]; index < max_index; index++)
+          {
+            pSpectrum[index] >>= scale;
+          }
+        }
+      }
+#endif  /* FUNCTION_CBlock_ScaleSpectralData_func1 */
+    }
+  }
+
+}
+
+AAC_DECODER_ERROR CBlock_ReadSectionData(HANDLE_FDK_BITSTREAM bs,
+                                         CAacDecoderChannelInfo *pAacDecoderChannelInfo,
+                                         const SamplingRateInfo *pSamplingRateInfo,
+                                         const UINT  flags)
+{
+  int top, band;
+  int sect_len, sect_len_incr;
+  int group;
+  UCHAR sect_cb;
+  UCHAR *pCodeBook = pAacDecoderChannelInfo->pDynData->aCodeBook;
+  /* HCR input (long) */
+  SHORT *pNumLinesInSec    = pAacDecoderChannelInfo->pDynData->specificTo.aac.aNumLineInSec4Hcr;
+  int    numLinesInSecIdx  = 0;
+  UCHAR *pHcrCodeBook      = pAacDecoderChannelInfo->pDynData->specificTo.aac.aCodeBooks4Hcr;
+  const SHORT *BandOffsets = GetScaleFactorBandOffsets(&pAacDecoderChannelInfo->icsInfo, pSamplingRateInfo);
+  pAacDecoderChannelInfo->pDynData->specificTo.aac.numberSection = 0;
+  AAC_DECODER_ERROR ErrorStatus = AAC_DEC_OK;
+
+  FDKmemclear(pCodeBook, sizeof(UCHAR)*(8*16));
+
+  const int nbits = (IsLongBlock(&pAacDecoderChannelInfo->icsInfo) == 1) ? 5 : 3;
+
+  int sect_esc_val = (1 << nbits) - 1 ;
+
+  UCHAR ScaleFactorBandsTransmitted = GetScaleFactorBandsTransmitted(&pAacDecoderChannelInfo->icsInfo);
+  for (group=0; group<GetWindowGroups(&pAacDecoderChannelInfo->icsInfo); group++)
+  {
+    for (band=0; band < ScaleFactorBandsTransmitted; )
+    {
+      sect_len = 0;
+      if ( flags & AC_ER_VCB11 )  {
+        sect_cb = (UCHAR) FDKreadBits(bs,5);
+      }
+      else
+        sect_cb = (UCHAR) FDKreadBits(bs,4);
+
+      if ( ((flags & AC_ER_VCB11) == 0) || ( sect_cb < 11 ) || ((sect_cb > 11) && (sect_cb < 16)) ) {
+        sect_len_incr = FDKreadBits(bs, nbits);
+        while (sect_len_incr == sect_esc_val)
+        {
+          sect_len += sect_esc_val;
+          sect_len_incr = FDKreadBits(bs, nbits);
+        }
+      }
+      else {
+        sect_len_incr = 1;
+      }
+
+      sect_len += sect_len_incr;
+
+
+      top = band + sect_len;
+
+      if (flags & AC_ER_HCR) {
+        /* HCR input (long) -- collecting sideinfo (for HCR-_long_ only) */
+        pNumLinesInSec[numLinesInSecIdx] = BandOffsets[top] - BandOffsets[band];
+        numLinesInSecIdx++;
+        if (numLinesInSecIdx >= MAX_SFB_HCR) {
+          return AAC_DEC_PARSE_ERROR;
+        }
+        if (
+             (sect_cb == BOOKSCL) )
+        {
+          return AAC_DEC_INVALID_CODE_BOOK;
+        } else {
+          *pHcrCodeBook++ = sect_cb;
+        }
+        pAacDecoderChannelInfo->pDynData->specificTo.aac.numberSection++;
+      }
+
+      /* Check spectral line limits */
+      if (IsLongBlock( &(pAacDecoderChannelInfo->icsInfo) ))
+      {
+        if (top > 64) {
+          return AAC_DEC_DECODE_FRAME_ERROR;
+        }
+      } else { /* short block */
+        if (top + group*16 > (8 * 16)) {
+          return AAC_DEC_DECODE_FRAME_ERROR;
+        }
+      }
+
+      /* Check if decoded codebook index is feasible */
+      if ( (sect_cb == BOOKSCL)
+       || ( (sect_cb == INTENSITY_HCB || sect_cb == INTENSITY_HCB2) && pAacDecoderChannelInfo->pDynData->RawDataInfo.CommonWindow == 0)
+         )
+      {
+        return AAC_DEC_INVALID_CODE_BOOK;
+      }
+
+      /* Store codebook index */
+      for (; band < top; band++)
+      {
+        pCodeBook[group*16+band] = sect_cb;
+      }
+    }
+  }
+
+
+  return ErrorStatus;
+}
+
+/* mso: provides a faster way to i-quantize a whole band in one go */
+
+/**
+ * \brief inverse quantize one sfb. Each value of the sfb is processed according to the
+ *        formula: spectrum[i] = Sign(spectrum[i]) * Matissa(spectrum[i])^(4/3) * 2^(lsb/4).
+ * \param spectrum pointer to first line of the sfb to be inverse quantized.
+ * \param noLines number of lines belonging to the sfb.
+ * \param lsb last 2 bits of the scale factor of the sfb.
+ * \param scale max allowed shift scale for the sfb.
+ */
+static
+void InverseQuantizeBand( FIXP_DBL * RESTRICT spectrum,
+                              INT noLines,
+                              INT lsb,
+                              INT scale )
+{
+    const FIXP_DBL * RESTRICT InverseQuantTabler=(FIXP_DBL *)InverseQuantTable;
+    const FIXP_DBL * RESTRICT MantissaTabler=(FIXP_DBL *)MantissaTable[lsb];
+    const SCHAR* RESTRICT ExponentTabler=(SCHAR*)ExponentTable[lsb];
+
+    FIXP_DBL *ptr = spectrum;
+    FIXP_DBL signedValue;
+
+    FDK_ASSERT(noLines>2);
+    for (INT i=noLines; i--; )
+    {
+        if ((signedValue = *ptr++) != FL2FXCONST_DBL(0))
+        {
+          FIXP_DBL value = fAbs(signedValue);
+          UINT freeBits = CntLeadingZeros(value);
+          UINT exponent = 32 - freeBits;
+
+          UINT x = (UINT) (LONG)value << (INT) freeBits;
+          x <<= 1;                                  /* shift out sign bit to avoid masking later on */
+          UINT tableIndex = x >> 24;
+          x = (x >> 20) &  0x0F;
+
+          UINT r0=(UINT)(LONG)InverseQuantTabler[tableIndex+0];
+          UINT r1=(UINT)(LONG)InverseQuantTabler[tableIndex+1];
+          UINT temp= (r1 - r0)*x + (r0 << 4);
+
+          value = fMultDiv2((FIXP_DBL)temp, MantissaTabler[exponent]);
+
+          /* + 1 compensates fMultDiv2() */
+          scaleValueInPlace(&value, scale + ExponentTabler[exponent] + 1);
+
+          signedValue = (signedValue < (FIXP_DBL)0) ? -value : value;
+          ptr[-1] = signedValue;
+        }
+    }
+}
+
+AAC_DECODER_ERROR CBlock_InverseQuantizeSpectralData(CAacDecoderChannelInfo *pAacDecoderChannelInfo, SamplingRateInfo *pSamplingRateInfo)
+{
+  int window, group, groupwin, band;
+  int ScaleFactorBandsTransmitted = GetScaleFactorBandsTransmitted(&pAacDecoderChannelInfo->icsInfo);
+  UCHAR *RESTRICT pCodeBook = pAacDecoderChannelInfo->pDynData->aCodeBook;
+  SHORT *RESTRICT pSfbScale = pAacDecoderChannelInfo->pDynData->aSfbScale;
+  SHORT *RESTRICT pScaleFactor = pAacDecoderChannelInfo->pDynData->aScaleFactor;
+  const SHORT *RESTRICT BandOffsets = GetScaleFactorBandOffsets(&pAacDecoderChannelInfo->icsInfo, pSamplingRateInfo);
+
+  FDKmemclear(pAacDecoderChannelInfo->pDynData->aSfbScale, (8*16)*sizeof(SHORT));
+
+  for (window=0, group=0; group < GetWindowGroups(&pAacDecoderChannelInfo->icsInfo); group++)
+  {
+    for (groupwin=0; groupwin < GetWindowGroupLength(&pAacDecoderChannelInfo->icsInfo,group); groupwin++, window++)
+    {
+      /* inverse quantization */
+      for (band=0; band < ScaleFactorBandsTransmitted; band++)
+      {
+        FIXP_DBL *pSpectralCoefficient = SPEC(pAacDecoderChannelInfo->pSpectralCoefficient, window, pAacDecoderChannelInfo->granuleLength) + BandOffsets[band];
+
+        int noLines = BandOffsets[band+1] - BandOffsets[band];
+        int bnds = group*16+band;
+        int i;
+
+        if ((pCodeBook[bnds] == ZERO_HCB)
+         || (pCodeBook[bnds] == INTENSITY_HCB)
+         || (pCodeBook[bnds] == INTENSITY_HCB2)
+           )
+          continue;
+
+        if (pCodeBook[bnds] == NOISE_HCB)
+        {
+          /* Leave headroom for PNS values. + 1 because ceil(log2(2^(0.25*3))) = 1,
+             worst case of additional headroom required because of the scalefactor. */
+          pSfbScale[window*16+band] = (pScaleFactor [bnds] >> 2) + 1 ;
+          continue;
+        }
+
+        /* Find max spectral line value of the current sfb */
+        FIXP_DBL locMax = (FIXP_DBL)0;
+
+        for (i = noLines; i-- ; ) {
+          /* Expensive memory access */
+          locMax = fMax(fixp_abs(pSpectralCoefficient[i]), locMax);
+        }
+
+        /* Cheap robustness improvement - Do not remove!!! */
+        if (fixp_abs(locMax) > (FIXP_DBL)MAX_QUANTIZED_VALUE) {
+          return AAC_DEC_DECODE_FRAME_ERROR;
+        }
+
+        /*
+           The inverse quantized spectral lines are defined by:
+        pSpectralCoefficient[i] = Sign(pSpectralCoefficient[i]) * 2^(0.25*pScaleFactor[bnds]) * pSpectralCoefficient[i]^(4/3)
+           This is equivalent to:
+        pSpectralCoefficient[i]    = Sign(pSpectralCoefficient[i]) * (2^(pScaleFactor[bnds] % 4) * pSpectralCoefficient[i]^(4/3))
+        pSpectralCoefficient_e[i] += pScaleFactor[bnds]/4
+        */
+        {
+          int msb = pScaleFactor [bnds] >> 2 ;
+          int lsb = pScaleFactor [bnds] & 0x03 ;
+
+          int scale = GetScaleFromValue(locMax, lsb);
+
+          pSfbScale[window*16+band] = msb - scale;
+          InverseQuantizeBand(pSpectralCoefficient, noLines, lsb, scale);
+        }
+      }
+    }
+  }
+
+
+  return AAC_DEC_OK;
+}
+
+
+AAC_DECODER_ERROR  CBlock_ReadSpectralData(HANDLE_FDK_BITSTREAM bs,
+                                           CAacDecoderChannelInfo *pAacDecoderChannelInfo,
+                                           const SamplingRateInfo *pSamplingRateInfo,
+                                           const UINT  flags)
+{
+  int i,index;
+  int window,group,groupwin,groupoffset,band;
+  UCHAR *RESTRICT pCodeBook = pAacDecoderChannelInfo->pDynData->aCodeBook;
+  const SHORT *RESTRICT BandOffsets = GetScaleFactorBandOffsets(&pAacDecoderChannelInfo->icsInfo, pSamplingRateInfo);
+
+  SPECTRAL_PTR pSpectralCoefficient = pAacDecoderChannelInfo->pSpectralCoefficient;
+  FIXP_DBL locMax;
+
+  int ScaleFactorBandsTransmitted = GetScaleFactorBandsTransmitted(&pAacDecoderChannelInfo->icsInfo);
+
+  FDK_ASSERT(BandOffsets != NULL);
+
+  FDKmemclear(pSpectralCoefficient, sizeof(SPECTRUM));
+
+  if ( (flags & AC_ER_HCR) == 0 )
+  {
+    groupoffset = 0;
+
+    /* plain huffman decoder  short */
+    for (group=0; group < GetWindowGroups(&pAacDecoderChannelInfo->icsInfo); group++)
+    {
+      for (band=0; band < ScaleFactorBandsTransmitted; band++)
+      {
+        int bnds = group*16+band;
+        UCHAR currentCB = pCodeBook[bnds];
+
+        /* patch to run plain-huffman-decoder with vcb11 input codebooks (LAV-checking might be possible below using the virtual cb and a LAV-table) */
+        if ((currentCB >= 16) && (currentCB <= 31)) {
+          pCodeBook[bnds] = currentCB = 11;
+        }
+        if ( !((currentCB == ZERO_HCB)
+            || (currentCB == NOISE_HCB)
+            || (currentCB == INTENSITY_HCB)
+            || (currentCB == INTENSITY_HCB2)) )
+        {
+          const CodeBookDescription *hcb = &AACcodeBookDescriptionTable[currentCB];
+          int step = hcb->Dimension;
+          int offset = hcb->Offset;
+          int bits = hcb->numBits;
+          int mask = (1<<bits)-1;
+
+          for (groupwin=0; groupwin < GetWindowGroupLength(&pAacDecoderChannelInfo->icsInfo,group); groupwin++)
+          {
+            window = groupoffset + groupwin;
+
+            FIXP_DBL *mdctSpectrum = SPEC(pSpectralCoefficient, window, pAacDecoderChannelInfo->granuleLength);
+
+            locMax = (FIXP_DBL)0 ;
+
+            for (index=BandOffsets[band]; index < BandOffsets[band+1]; index+=step)
+            {
+              int idx = CBlock_DecodeHuffmanWord(bs,hcb);
+
+              for (i=0; i<step; i++) {
+                FIXP_DBL tmp;
+
+                tmp = (FIXP_DBL)((idx & mask)-offset);
+                idx >>= bits;
+
+                if (offset == 0) {
+                  if (tmp != FIXP_DBL(0))
+                    tmp = (FDKreadBits(bs,1))? -tmp : tmp;
+                }
+                mdctSpectrum[index+i] = tmp;
+              }
+
+              if (currentCB == ESCBOOK)
+              {
+                mdctSpectrum[index+0] = (FIXP_DBL)CBlock_GetEscape(bs, (LONG)mdctSpectrum[index+0]);
+                mdctSpectrum[index+1] = (FIXP_DBL)CBlock_GetEscape(bs, (LONG)mdctSpectrum[index+1]);
+
+              }
+            }
+          }
+        }
+      }
+      groupoffset += GetWindowGroupLength(&pAacDecoderChannelInfo->icsInfo,group);
+    }
+    /* plain huffman decoding (short) finished */
+  }
+  /* HCR - Huffman Codeword Reordering  short */
+  else  /* if ( flags & AC_ER_HCR ) */
+  {
+    H_HCR_INFO hHcr = &pAacDecoderChannelInfo->pComData->overlay.aac.erHcrInfo;
+    int hcrStatus = 0;
+    int hcrConcealWholeFrame = 0;
+
+    /* advanced Huffman decoding starts here (HCR decoding :) */
+    if ( pAacDecoderChannelInfo->pDynData->specificTo.aac.lenOfReorderedSpectralData != 0 ) {
+
+      /* HCR initialization short */
+      hcrStatus = HcrInit(hHcr, pAacDecoderChannelInfo, pSamplingRateInfo, bs);
+
+      if (hcrStatus != 0) {
+#if HCR_ERROR_CONCEALMENT
+        hcrConcealWholeFrame = 1;
+        return AAC_DEC_DECODE_FRAME_ERROR;  /* concealment is muting in the first step, therefore return now */
+        // hcr decoding is not skipped because of returning above
+#else
+        return AAC_DEC_DECODE_FRAME_ERROR;
+#endif
+      }
+
+      /* HCR decoding short */
+      hcrStatus = HcrDecoder(hHcr, pAacDecoderChannelInfo, pSamplingRateInfo, bs);
+
+
+#if HCR_ERROR_CONCEALMENT
+      HcrMuteErroneousLines(hHcr);
+#else
+      return AAC_DEC_DECODE_FRAME_ERROR;
+#endif /* HCR_ERROR_CONCEALMENT */
+
+      FDKpushFor (bs, pAacDecoderChannelInfo->pDynData->specificTo.aac.lenOfReorderedSpectralData);
+    }
+  }
+  /* HCR - Huffman Codeword Reordering short finished */
+
+
+
+  if ( IsLongBlock(&pAacDecoderChannelInfo->icsInfo) && !(flags & (AC_ELD|AC_SCALABLE)) )
+  {
+    /* apply pulse data */
+    CPulseData_Apply(&pAacDecoderChannelInfo->pDynData->specificTo.aac.PulseData,
+                      GetScaleFactorBandOffsets(&pAacDecoderChannelInfo->icsInfo, pSamplingRateInfo),
+                      SPEC_LONG(pSpectralCoefficient));
+  }
+
+
+  return AAC_DEC_OK;
+}
+
+
+
+void ApplyTools ( CAacDecoderChannelInfo *pAacDecoderChannelInfo[],
+                  const SamplingRateInfo *pSamplingRateInfo,
+                  const UINT flags,
+                  const int channel )
+{
+
+  if ( !(flags & (AC_USAC|AC_RSVD50|AC_MPS_RES)) ) {
+    CPns_Apply(
+           &pAacDecoderChannelInfo[channel]->data.aac.PnsData,
+           &pAacDecoderChannelInfo[channel]->icsInfo,
+            pAacDecoderChannelInfo[channel]->pSpectralCoefficient,
+            pAacDecoderChannelInfo[channel]->specScale,
+            pAacDecoderChannelInfo[channel]->pDynData->aScaleFactor,
+            pSamplingRateInfo,
+            pAacDecoderChannelInfo[channel]->granuleLength,
+            channel
+            );
+  }
+
+  CTns_Apply (
+         &pAacDecoderChannelInfo[channel]->pDynData->TnsData,
+         &pAacDecoderChannelInfo[channel]->icsInfo,
+          pAacDecoderChannelInfo[channel]->pSpectralCoefficient,
+          pSamplingRateInfo,
+          pAacDecoderChannelInfo[channel]->granuleLength
+          );
+}
+
+static
+int getWindow2Nr(int length, int shape)
+{
+  int nr = 0;
+
+  if (shape == 2) {
+    /* Low Overlap, 3/4 zeroed */
+    nr = (length * 3)>>2;
+  }
+
+  return nr;
+}
+
+void CBlock_FrequencyToTime(CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo,
+                            CAacDecoderChannelInfo *pAacDecoderChannelInfo,
+                            INT_PCM outSamples[],
+                            const SHORT frameLen,
+                            const int stride,
+                            const int frameOk,
+                            FIXP_DBL *pWorkBuffer1 )
+{
+  int fr, fl, tl, nSamples, nSpec;
+
+  /* Determine left slope length (fl), right slope length (fr) and transform length (tl).
+     USAC: The slope length may mismatch with the previous frame in case of LPD / FD
+           transitions. The adjustment is handled by the imdct implementation.
+  */
+  tl = frameLen;
+  nSpec = 1;
+
+  switch( pAacDecoderChannelInfo->icsInfo.WindowSequence ) {
+    default:
+    case OnlyLongSequence:
+      fl = frameLen;
+      fr = frameLen - getWindow2Nr(frameLen, GetWindowShape(&pAacDecoderChannelInfo->icsInfo));
+      break;
+    case LongStopSequence:
+      fl = frameLen >> 3;
+      fr = frameLen;
+      break;
+    case LongStartSequence: /* or StopStartSequence */
+      fl = frameLen;
+      fr = frameLen >> 3;
+      break;
+    case EightShortSequence:
+      fl = fr = frameLen >> 3;
+      tl >>= 3;
+      nSpec = 8;
+      break;
+  }
+
+  {
+    int i;
+
+    {
+      FIXP_DBL *tmp = pAacDecoderChannelInfo->pComData->workBufferCore1->mdctOutTemp;
+
+      nSamples = imdct_block(
+             &pAacDecoderStaticChannelInfo->IMdct,
+              tmp,
+              SPEC_LONG(pAacDecoderChannelInfo->pSpectralCoefficient),
+              pAacDecoderChannelInfo->specScale,
+              nSpec,
+              frameLen,
+              tl,
+              FDKgetWindowSlope(fl, GetWindowShape(&pAacDecoderChannelInfo->icsInfo)),
+              fl,
+              FDKgetWindowSlope(fr, GetWindowShape(&pAacDecoderChannelInfo->icsInfo)),
+              fr,
+              (FIXP_DBL)0 );
+
+      for (i=0; i<frameLen; i++) {
+        outSamples[i*stride] = IMDCT_SCALE(tmp[i]);
+      }
+    }
+  }
+
+  FDK_ASSERT(nSamples == frameLen);
+
+}
+
+#include "ldfiltbank.h"
+void CBlock_FrequencyToTimeLowDelay( CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo,
+                                     CAacDecoderChannelInfo *pAacDecoderChannelInfo,
+                                     INT_PCM outSamples[],
+                                     const short frameLen,
+                                     const char stride )
+{
+  InvMdctTransformLowDelay_fdk (
+          SPEC_LONG(pAacDecoderChannelInfo->pSpectralCoefficient),
+          pAacDecoderChannelInfo->specScale[0],
+          outSamples,
+          pAacDecoderStaticChannelInfo->pOverlapBuffer,
+          stride,
+          frameLen
+          );
+}