Include patched FDK-AAC in the repository

The initial idea was to get the DAB+ patch into upstream, but since
that follows the android source releases, there is no place for a custom
DAB+ patch there.

So instead of having to maintain a patched fdk-aac that has to have the
same .so version as the distribution package on which it is installed,
we prefer having a separate fdk-aac-dab library to avoid collision.

At that point, there's no reason to keep fdk-aac in a separate
repository, as odr-audioenc is the only tool that needs DAB+ encoding
support. Including it here simplifies installation, and makes it
consistent with toolame-dab, also shipped in this repository.

DAB+ decoding support (needed by ODR-SourceCompanion, dablin, etisnoop,
welle.io and others) can be done using upstream FDK-AAC.

41 files changed, 22125 insertions, 0 deletions

diff --git a/fdk-aac/libSBRenc/src/bit_sbr.cpp b/fdk-aac/libSBRenc/src/bit_sbr.cpp
new file mode 100644
index 0000000..5a65e98
--- /dev/null
+++ b/fdk-aac/libSBRenc/src/bit_sbr.cpp
@@ -0,0 +1,1049 @@
+/* -----------------------------------------------------------------------------
+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 encoder library ******************************
+
+   Author(s):
+
+   Description:
+
+*******************************************************************************/
+
+/*!
+  \file
+  \brief  SBR bit writing routines $Revision: 93300 $
+*/
+
+#include "bit_sbr.h"
+
+#include "code_env.h"
+#include "cmondata.h"
+#include "sbr.h"
+
+#include "ps_main.h"
+
+typedef enum { SBR_ID_SCE = 1, SBR_ID_CPE } SBR_ELEMENT_TYPE;
+
+static INT encodeSbrData(HANDLE_SBR_ENV_DATA sbrEnvDataLeft,
+                         HANDLE_SBR_ENV_DATA sbrEnvDataRight,
+                         HANDLE_PARAMETRIC_STEREO hParametricStereo,
+                         HANDLE_COMMON_DATA cmonData, SBR_ELEMENT_TYPE sbrElem,
+                         INT coupling, UINT sbrSyntaxFlags);
+
+static INT encodeSbrHeader(HANDLE_SBR_HEADER_DATA sbrHeaderData,
+                           HANDLE_SBR_BITSTREAM_DATA sbrBitstreamData,
+                           HANDLE_COMMON_DATA cmonData);
+
+static INT encodeSbrHeaderData(HANDLE_SBR_HEADER_DATA sbrHeaderData,
+                               HANDLE_FDK_BITSTREAM hBitStream);
+
+static INT encodeSbrSingleChannelElement(
+    HANDLE_SBR_ENV_DATA sbrEnvData, HANDLE_FDK_BITSTREAM hBitStream,
+    HANDLE_PARAMETRIC_STEREO hParametricStereo, const UINT sbrSyntaxFlags);
+
+static INT encodeSbrChannelPairElement(
+    HANDLE_SBR_ENV_DATA sbrEnvDataLeft, HANDLE_SBR_ENV_DATA sbrEnvDataRight,
+    HANDLE_PARAMETRIC_STEREO hParametricStereo, HANDLE_FDK_BITSTREAM hBitStream,
+    const INT coupling, const UINT sbrSyntaxFlags);
+
+static INT encodeSbrGrid(HANDLE_SBR_ENV_DATA sbrEnvData,
+                         HANDLE_FDK_BITSTREAM hBitStream);
+
+static int encodeLowDelaySbrGrid(HANDLE_SBR_ENV_DATA sbrEnvData,
+                                 HANDLE_FDK_BITSTREAM hBitStream,
+                                 const int transmitFreqs,
+                                 const UINT sbrSyntaxFlags);
+
+static INT encodeSbrDtdf(HANDLE_SBR_ENV_DATA sbrEnvData,
+                         HANDLE_FDK_BITSTREAM hBitStream);
+
+static INT writeNoiseLevelData(HANDLE_SBR_ENV_DATA sbrEnvData,
+                               HANDLE_FDK_BITSTREAM hBitStream, INT coupling);
+
+static INT writeEnvelopeData(HANDLE_SBR_ENV_DATA sbrEnvData,
+                             HANDLE_FDK_BITSTREAM hBitStream, INT coupling);
+
+static INT writeSyntheticCodingData(HANDLE_SBR_ENV_DATA sbrEnvData,
+                                    HANDLE_FDK_BITSTREAM hBitStream);
+
+static INT encodeExtendedData(HANDLE_PARAMETRIC_STEREO hParametricStereo,
+                              HANDLE_FDK_BITSTREAM hBitStream);
+
+static INT getSbrExtendedDataSize(HANDLE_PARAMETRIC_STEREO hParametricStereo);
+
+/*****************************************************************************
+
+    functionname: FDKsbrEnc_WriteEnvSingleChannelElement
+    description:  writes pure SBR single channel data element
+    returns:      number of bits written
+    input:
+    output:
+
+*****************************************************************************/
+INT FDKsbrEnc_WriteEnvSingleChannelElement(
+    HANDLE_SBR_HEADER_DATA sbrHeaderData,
+    HANDLE_PARAMETRIC_STEREO hParametricStereo,
+    HANDLE_SBR_BITSTREAM_DATA sbrBitstreamData, HANDLE_SBR_ENV_DATA sbrEnvData,
+    HANDLE_COMMON_DATA cmonData, UINT sbrSyntaxFlags)
+
+{
+  INT payloadBits = 0;
+
+  cmonData->sbrHdrBits = 0;
+  cmonData->sbrDataBits = 0;
+
+  /* write pure sbr data */
+  if (sbrEnvData != NULL) {
+    /* write header */
+    payloadBits += encodeSbrHeader(sbrHeaderData, sbrBitstreamData, cmonData);
+
+    /* write data */
+    payloadBits += encodeSbrData(sbrEnvData, NULL, hParametricStereo, cmonData,
+                                 SBR_ID_SCE, 0, sbrSyntaxFlags);
+  }
+  return payloadBits;
+}
+
+/*****************************************************************************
+
+    functionname: FDKsbrEnc_WriteEnvChannelPairElement
+    description:  writes pure SBR channel pair data element
+    returns:      number of bits written
+    input:
+    output:
+
+*****************************************************************************/
+INT FDKsbrEnc_WriteEnvChannelPairElement(
+    HANDLE_SBR_HEADER_DATA sbrHeaderData,
+    HANDLE_PARAMETRIC_STEREO hParametricStereo,
+    HANDLE_SBR_BITSTREAM_DATA sbrBitstreamData,
+    HANDLE_SBR_ENV_DATA sbrEnvDataLeft, HANDLE_SBR_ENV_DATA sbrEnvDataRight,
+    HANDLE_COMMON_DATA cmonData, UINT sbrSyntaxFlags)
+
+{
+  INT payloadBits = 0;
+  cmonData->sbrHdrBits = 0;
+  cmonData->sbrDataBits = 0;
+
+  /* write pure sbr data */
+  if ((sbrEnvDataLeft != NULL) && (sbrEnvDataRight != NULL)) {
+    /* write header */
+    payloadBits += encodeSbrHeader(sbrHeaderData, sbrBitstreamData, cmonData);
+
+    /* write data */
+    payloadBits += encodeSbrData(sbrEnvDataLeft, sbrEnvDataRight,
+                                 hParametricStereo, cmonData, SBR_ID_CPE,
+                                 sbrHeaderData->coupling, sbrSyntaxFlags);
+  }
+  return payloadBits;
+}
+
+INT FDKsbrEnc_CountSbrChannelPairElement(
+    HANDLE_SBR_HEADER_DATA sbrHeaderData,
+    HANDLE_PARAMETRIC_STEREO hParametricStereo,
+    HANDLE_SBR_BITSTREAM_DATA sbrBitstreamData,
+    HANDLE_SBR_ENV_DATA sbrEnvDataLeft, HANDLE_SBR_ENV_DATA sbrEnvDataRight,
+    HANDLE_COMMON_DATA cmonData, UINT sbrSyntaxFlags) {
+  INT payloadBits;
+  INT bitPos = FDKgetValidBits(&cmonData->sbrBitbuf);
+
+  payloadBits = FDKsbrEnc_WriteEnvChannelPairElement(
+      sbrHeaderData, hParametricStereo, sbrBitstreamData, sbrEnvDataLeft,
+      sbrEnvDataRight, cmonData, sbrSyntaxFlags);
+
+  FDKpushBack(&cmonData->sbrBitbuf,
+              (FDKgetValidBits(&cmonData->sbrBitbuf) - bitPos));
+
+  return payloadBits;
+}
+
+void sbrEncoder_GetHeader(SBR_ENCODER *sbrEncoder, HANDLE_FDK_BITSTREAM hBs,
+                          INT element_index, int fSendHeaders) {
+  encodeSbrHeaderData(&sbrEncoder->sbrElement[element_index]->sbrHeaderData,
+                      hBs);
+
+  if (fSendHeaders == 0) {
+    /* Prevent header being embedded into the SBR payload. */
+    sbrEncoder->sbrElement[element_index]->sbrBitstreamData.NrSendHeaderData =
+        -1;
+    sbrEncoder->sbrElement[element_index]->sbrBitstreamData.HeaderActive = 0;
+    sbrEncoder->sbrElement[element_index]
+        ->sbrBitstreamData.CountSendHeaderData = -1;
+  }
+}
+
+/*****************************************************************************
+
+    functionname: encodeSbrHeader
+    description:  encodes SBR Header information
+    returns:      number of bits written
+    input:
+    output:
+
+*****************************************************************************/
+static INT encodeSbrHeader(HANDLE_SBR_HEADER_DATA sbrHeaderData,
+                           HANDLE_SBR_BITSTREAM_DATA sbrBitstreamData,
+                           HANDLE_COMMON_DATA cmonData) {
+  INT payloadBits = 0;
+
+  if (sbrBitstreamData->HeaderActive) {
+    payloadBits += FDKwriteBits(&cmonData->sbrBitbuf, 1, 1);
+    payloadBits += encodeSbrHeaderData(sbrHeaderData, &cmonData->sbrBitbuf);
+  } else {
+    payloadBits += FDKwriteBits(&cmonData->sbrBitbuf, 0, 1);
+  }
+
+  cmonData->sbrHdrBits = payloadBits;
+
+  return payloadBits;
+}
+
+/*****************************************************************************
+
+    functionname: encodeSbrHeaderData
+    description:  writes sbr_header()
+                  bs_protocol_version through bs_header_extra_2
+    returns:      number of bits written
+    input:
+    output:
+
+*****************************************************************************/
+static INT encodeSbrHeaderData(HANDLE_SBR_HEADER_DATA sbrHeaderData,
+                               HANDLE_FDK_BITSTREAM hBitStream)
+
+{
+  INT payloadBits = 0;
+  if (sbrHeaderData != NULL) {
+    payloadBits += FDKwriteBits(hBitStream, sbrHeaderData->sbr_amp_res,
+                                SI_SBR_AMP_RES_BITS);
+    payloadBits += FDKwriteBits(hBitStream, sbrHeaderData->sbr_start_frequency,
+                                SI_SBR_START_FREQ_BITS);
+    payloadBits += FDKwriteBits(hBitStream, sbrHeaderData->sbr_stop_frequency,
+                                SI_SBR_STOP_FREQ_BITS);
+    payloadBits += FDKwriteBits(hBitStream, sbrHeaderData->sbr_xover_band,
+                                SI_SBR_XOVER_BAND_BITS);
+
+    payloadBits += FDKwriteBits(hBitStream, 0, SI_SBR_RESERVED_BITS);
+
+    payloadBits += FDKwriteBits(hBitStream, sbrHeaderData->header_extra_1,
+                                SI_SBR_HEADER_EXTRA_1_BITS);
+    payloadBits += FDKwriteBits(hBitStream, sbrHeaderData->header_extra_2,
+                                SI_SBR_HEADER_EXTRA_2_BITS);
+
+    if (sbrHeaderData->header_extra_1) {
+      payloadBits += FDKwriteBits(hBitStream, sbrHeaderData->freqScale,
+                                  SI_SBR_FREQ_SCALE_BITS);
+      payloadBits += FDKwriteBits(hBitStream, sbrHeaderData->alterScale,
+                                  SI_SBR_ALTER_SCALE_BITS);
+      payloadBits += FDKwriteBits(hBitStream, sbrHeaderData->sbr_noise_bands,
+                                  SI_SBR_NOISE_BANDS_BITS);
+    } /* sbrHeaderData->header_extra_1 */
+
+    if (sbrHeaderData->header_extra_2) {
+      payloadBits += FDKwriteBits(hBitStream, sbrHeaderData->sbr_limiter_bands,
+                                  SI_SBR_LIMITER_BANDS_BITS);
+      payloadBits += FDKwriteBits(hBitStream, sbrHeaderData->sbr_limiter_gains,
+                                  SI_SBR_LIMITER_GAINS_BITS);
+      payloadBits += FDKwriteBits(hBitStream, sbrHeaderData->sbr_interpol_freq,
+                                  SI_SBR_INTERPOL_FREQ_BITS);
+      payloadBits +=
+          FDKwriteBits(hBitStream, sbrHeaderData->sbr_smoothing_length,
+                       SI_SBR_SMOOTHING_LENGTH_BITS);
+
+    } /* sbrHeaderData->header_extra_2 */
+  }   /* sbrHeaderData != NULL */
+
+  return payloadBits;
+}
+
+/*****************************************************************************
+
+    functionname: encodeSbrData
+    description:  encodes sbr Data information
+    returns:      number of bits written
+    input:
+    output:
+
+*****************************************************************************/
+static INT encodeSbrData(HANDLE_SBR_ENV_DATA sbrEnvDataLeft,
+                         HANDLE_SBR_ENV_DATA sbrEnvDataRight,
+                         HANDLE_PARAMETRIC_STEREO hParametricStereo,
+                         HANDLE_COMMON_DATA cmonData, SBR_ELEMENT_TYPE sbrElem,
+                         INT coupling, UINT sbrSyntaxFlags) {
+  INT payloadBits = 0;
+
+  switch (sbrElem) {
+    case SBR_ID_SCE:
+      payloadBits +=
+          encodeSbrSingleChannelElement(sbrEnvDataLeft, &cmonData->sbrBitbuf,
+                                        hParametricStereo, sbrSyntaxFlags);
+      break;
+    case SBR_ID_CPE:
+      payloadBits += encodeSbrChannelPairElement(
+          sbrEnvDataLeft, sbrEnvDataRight, hParametricStereo,
+          &cmonData->sbrBitbuf, coupling, sbrSyntaxFlags);
+      break;
+    default:
+      /* we never should apply SBR to any other element type */
+      FDK_ASSERT(0);
+  }
+
+  cmonData->sbrDataBits = payloadBits;
+
+  return payloadBits;
+}
+
+#define MODE_FREQ_TANS 1
+#define MODE_NO_FREQ_TRAN 0
+#define LD_TRANSMISSION MODE_FREQ_TANS
+static int encodeFreqs(int mode) { return ((mode & MODE_FREQ_TANS) ? 1 : 0); }
+
+/*****************************************************************************
+
+    functionname: encodeSbrSingleChannelElement
+    description:  encodes sbr SCE information
+    returns:      number of bits written
+    input:
+    output:
+
+*****************************************************************************/
+static INT encodeSbrSingleChannelElement(
+    HANDLE_SBR_ENV_DATA sbrEnvData, HANDLE_FDK_BITSTREAM hBitStream,
+    HANDLE_PARAMETRIC_STEREO hParametricStereo, const UINT sbrSyntaxFlags) {
+  INT i, payloadBits = 0;
+
+  payloadBits += FDKwriteBits(hBitStream, 0,
+                              SI_SBR_DATA_EXTRA_BITS); /* no reserved bits */
+
+  if (sbrEnvData->ldGrid) {
+    if (sbrEnvData->hSbrBSGrid->frameClass != FIXFIXonly) {
+      /* encode normal SbrGrid */
+      payloadBits += encodeSbrGrid(sbrEnvData, hBitStream);
+    } else {
+      /* use FIXFIXonly frame Grid */
+      payloadBits += encodeLowDelaySbrGrid(
+          sbrEnvData, hBitStream, encodeFreqs(LD_TRANSMISSION), sbrSyntaxFlags);
+    }
+  } else {
+    if (sbrSyntaxFlags & SBR_SYNTAX_SCALABLE) {
+      payloadBits += FDKwriteBits(hBitStream, 1, SI_SBR_COUPLING_BITS);
+    }
+    payloadBits += encodeSbrGrid(sbrEnvData, hBitStream);
+  }
+
+  payloadBits += encodeSbrDtdf(sbrEnvData, hBitStream);
+
+  for (i = 0; i < sbrEnvData->noOfnoisebands; i++) {
+    payloadBits += FDKwriteBits(hBitStream, sbrEnvData->sbr_invf_mode_vec[i],
+                                SI_SBR_INVF_MODE_BITS);
+  }
+
+  payloadBits += writeEnvelopeData(sbrEnvData, hBitStream, 0);
+  payloadBits += writeNoiseLevelData(sbrEnvData, hBitStream, 0);
+
+  payloadBits += writeSyntheticCodingData(sbrEnvData, hBitStream);
+
+  payloadBits += encodeExtendedData(hParametricStereo, hBitStream);
+
+  return payloadBits;
+}
+
+/*****************************************************************************
+
+    functionname: encodeSbrChannelPairElement
+    description:  encodes sbr CPE information
+    returns:
+    input:
+    output:
+
+*****************************************************************************/
+static INT encodeSbrChannelPairElement(
+    HANDLE_SBR_ENV_DATA sbrEnvDataLeft, HANDLE_SBR_ENV_DATA sbrEnvDataRight,
+    HANDLE_PARAMETRIC_STEREO hParametricStereo, HANDLE_FDK_BITSTREAM hBitStream,
+    const INT coupling, const UINT sbrSyntaxFlags) {
+  INT payloadBits = 0;
+  INT i = 0;
+
+  payloadBits += FDKwriteBits(hBitStream, 0,
+                              SI_SBR_DATA_EXTRA_BITS); /* no reserved bits */
+
+  payloadBits += FDKwriteBits(hBitStream, coupling, SI_SBR_COUPLING_BITS);
+
+  if (coupling) {
+    if (sbrEnvDataLeft->ldGrid) {
+      if (sbrEnvDataLeft->hSbrBSGrid->frameClass != FIXFIXonly) {
+        /* normal SbrGrid */
+        payloadBits += encodeSbrGrid(sbrEnvDataLeft, hBitStream);
+
+      } else {
+        /* FIXFIXonly frame Grid */
+        payloadBits +=
+            encodeLowDelaySbrGrid(sbrEnvDataLeft, hBitStream,
+                                  encodeFreqs(LD_TRANSMISSION), sbrSyntaxFlags);
+      }
+    } else
+      payloadBits += encodeSbrGrid(sbrEnvDataLeft, hBitStream);
+
+    payloadBits += encodeSbrDtdf(sbrEnvDataLeft, hBitStream);
+    payloadBits += encodeSbrDtdf(sbrEnvDataRight, hBitStream);
+
+    for (i = 0; i < sbrEnvDataLeft->noOfnoisebands; i++) {
+      payloadBits +=
+          FDKwriteBits(hBitStream, sbrEnvDataLeft->sbr_invf_mode_vec[i],
+                       SI_SBR_INVF_MODE_BITS);
+    }
+
+    payloadBits += writeEnvelopeData(sbrEnvDataLeft, hBitStream, 1);
+    payloadBits += writeNoiseLevelData(sbrEnvDataLeft, hBitStream, 1);
+    payloadBits += writeEnvelopeData(sbrEnvDataRight, hBitStream, 1);
+    payloadBits += writeNoiseLevelData(sbrEnvDataRight, hBitStream, 1);
+
+    payloadBits += writeSyntheticCodingData(sbrEnvDataLeft, hBitStream);
+    payloadBits += writeSyntheticCodingData(sbrEnvDataRight, hBitStream);
+
+  } else { /* no coupling */
+    FDK_ASSERT(sbrEnvDataLeft->ldGrid == sbrEnvDataRight->ldGrid);
+
+    if (sbrEnvDataLeft->ldGrid || sbrEnvDataRight->ldGrid) {
+      /* sbrEnvDataLeft (left channel) */
+      if (sbrEnvDataLeft->hSbrBSGrid->frameClass != FIXFIXonly) {
+        /* no FIXFIXonly Frame so we dont need encodeLowDelaySbrGrid */
+        /* normal SbrGrid */
+        payloadBits += encodeSbrGrid(sbrEnvDataLeft, hBitStream);
+
+      } else {
+        /* FIXFIXonly frame Grid */
+        payloadBits +=
+            encodeLowDelaySbrGrid(sbrEnvDataLeft, hBitStream,
+                                  encodeFreqs(LD_TRANSMISSION), sbrSyntaxFlags);
+      }
+
+      /* sbrEnvDataRight (right channel) */
+      if (sbrEnvDataRight->hSbrBSGrid->frameClass != FIXFIXonly) {
+        /* no FIXFIXonly Frame so we dont need encodeLowDelaySbrGrid */
+        /* normal SbrGrid */
+        payloadBits += encodeSbrGrid(sbrEnvDataRight, hBitStream);
+
+      } else {
+        /* FIXFIXonly frame Grid */
+        payloadBits +=
+            encodeLowDelaySbrGrid(sbrEnvDataRight, hBitStream,
+                                  encodeFreqs(LD_TRANSMISSION), sbrSyntaxFlags);
+      }
+    } else {
+      payloadBits += encodeSbrGrid(sbrEnvDataLeft, hBitStream);
+      payloadBits += encodeSbrGrid(sbrEnvDataRight, hBitStream);
+    }
+    payloadBits += encodeSbrDtdf(sbrEnvDataLeft, hBitStream);
+    payloadBits += encodeSbrDtdf(sbrEnvDataRight, hBitStream);
+
+    for (i = 0; i < sbrEnvDataLeft->noOfnoisebands; i++) {
+      payloadBits +=
+          FDKwriteBits(hBitStream, sbrEnvDataLeft->sbr_invf_mode_vec[i],
+                       SI_SBR_INVF_MODE_BITS);
+    }
+    for (i = 0; i < sbrEnvDataRight->noOfnoisebands; i++) {
+      payloadBits +=
+          FDKwriteBits(hBitStream, sbrEnvDataRight->sbr_invf_mode_vec[i],
+                       SI_SBR_INVF_MODE_BITS);
+    }
+
+    payloadBits += writeEnvelopeData(sbrEnvDataLeft, hBitStream, 0);
+    payloadBits += writeEnvelopeData(sbrEnvDataRight, hBitStream, 0);
+    payloadBits += writeNoiseLevelData(sbrEnvDataLeft, hBitStream, 0);
+    payloadBits += writeNoiseLevelData(sbrEnvDataRight, hBitStream, 0);
+
+    payloadBits += writeSyntheticCodingData(sbrEnvDataLeft, hBitStream);
+    payloadBits += writeSyntheticCodingData(sbrEnvDataRight, hBitStream);
+
+  } /* coupling */
+
+  payloadBits += encodeExtendedData(hParametricStereo, hBitStream);
+
+  return payloadBits;
+}
+
+static INT ceil_ln2(INT x) {
+  INT tmp = -1;
+  while ((1 << ++tmp) < x)
+    ;
+  return (tmp);
+}
+
+/*****************************************************************************
+
+    functionname: encodeSbrGrid
+    description:  if hBitStream != NULL writes bits that describes the
+                  time/frequency grouping of a frame; else counts them only
+    returns:      number of bits written or counted
+    input:
+    output:
+
+*****************************************************************************/
+static INT encodeSbrGrid(HANDLE_SBR_ENV_DATA sbrEnvData,
+                         HANDLE_FDK_BITSTREAM hBitStream) {
+  INT payloadBits = 0;
+  INT i, temp;
+  INT bufferFrameStart = sbrEnvData->hSbrBSGrid->bufferFrameStart;
+  INT numberTimeSlots = sbrEnvData->hSbrBSGrid->numberTimeSlots;
+
+  if (sbrEnvData->ldGrid)
+    payloadBits += FDKwriteBits(hBitStream, sbrEnvData->hSbrBSGrid->frameClass,
+                                SBR_CLA_BITS_LD);
+  else
+    payloadBits += FDKwriteBits(hBitStream, sbrEnvData->hSbrBSGrid->frameClass,
+                                SBR_CLA_BITS);
+
+  switch (sbrEnvData->hSbrBSGrid->frameClass) {
+    case FIXFIXonly:
+      FDK_ASSERT(0 /* Fatal error in encodeSbrGrid! */);
+      break;
+    case FIXFIX:
+      temp = ceil_ln2(sbrEnvData->hSbrBSGrid->bs_num_env);
+      payloadBits += FDKwriteBits(hBitStream, temp, SBR_ENV_BITS);
+      if ((sbrEnvData->ldGrid) && (sbrEnvData->hSbrBSGrid->bs_num_env == 1))
+        payloadBits += FDKwriteBits(hBitStream, sbrEnvData->currentAmpResFF,
+                                    SI_SBR_AMP_RES_BITS);
+      payloadBits += FDKwriteBits(hBitStream, sbrEnvData->hSbrBSGrid->v_f[0],
+                                  SBR_RES_BITS);
+
+      break;
+
+    case FIXVAR:
+    case VARFIX:
+      if (sbrEnvData->hSbrBSGrid->frameClass == FIXVAR)
+        temp = sbrEnvData->hSbrBSGrid->bs_abs_bord -
+               (bufferFrameStart + numberTimeSlots);
+      else
+        temp = sbrEnvData->hSbrBSGrid->bs_abs_bord - bufferFrameStart;
+
+      payloadBits += FDKwriteBits(hBitStream, temp, SBR_ABS_BITS);
+      payloadBits +=
+          FDKwriteBits(hBitStream, sbrEnvData->hSbrBSGrid->n, SBR_NUM_BITS);
+
+      for (i = 0; i < sbrEnvData->hSbrBSGrid->n; i++) {
+        temp = (sbrEnvData->hSbrBSGrid->bs_rel_bord[i] - 2) >> 1;
+        payloadBits += FDKwriteBits(hBitStream, temp, SBR_REL_BITS);
+      }
+
+      temp = ceil_ln2(sbrEnvData->hSbrBSGrid->n + 2);
+      payloadBits += FDKwriteBits(hBitStream, sbrEnvData->hSbrBSGrid->p, temp);
+
+      for (i = 0; i < sbrEnvData->hSbrBSGrid->n + 1; i++) {
+        payloadBits += FDKwriteBits(hBitStream, sbrEnvData->hSbrBSGrid->v_f[i],
+                                    SBR_RES_BITS);
+      }
+      break;
+
+    case VARVAR:
+      temp = sbrEnvData->hSbrBSGrid->bs_abs_bord_0 - bufferFrameStart;
+      payloadBits += FDKwriteBits(hBitStream, temp, SBR_ABS_BITS);
+      temp = sbrEnvData->hSbrBSGrid->bs_abs_bord_1 -
+             (bufferFrameStart + numberTimeSlots);
+      payloadBits += FDKwriteBits(hBitStream, temp, SBR_ABS_BITS);
+
+      payloadBits += FDKwriteBits(
+          hBitStream, sbrEnvData->hSbrBSGrid->bs_num_rel_0, SBR_NUM_BITS);
+      payloadBits += FDKwriteBits(
+          hBitStream, sbrEnvData->hSbrBSGrid->bs_num_rel_1, SBR_NUM_BITS);
+
+      for (i = 0; i < sbrEnvData->hSbrBSGrid->bs_num_rel_0; i++) {
+        temp = (sbrEnvData->hSbrBSGrid->bs_rel_bord_0[i] - 2) >> 1;
+        payloadBits += FDKwriteBits(hBitStream, temp, SBR_REL_BITS);
+      }
+
+      for (i = 0; i < sbrEnvData->hSbrBSGrid->bs_num_rel_1; i++) {
+        temp = (sbrEnvData->hSbrBSGrid->bs_rel_bord_1[i] - 2) >> 1;
+        payloadBits += FDKwriteBits(hBitStream, temp, SBR_REL_BITS);
+      }
+
+      temp = ceil_ln2(sbrEnvData->hSbrBSGrid->bs_num_rel_0 +
+                      sbrEnvData->hSbrBSGrid->bs_num_rel_1 + 2);
+      payloadBits += FDKwriteBits(hBitStream, sbrEnvData->hSbrBSGrid->p, temp);
+
+      temp = sbrEnvData->hSbrBSGrid->bs_num_rel_0 +
+             sbrEnvData->hSbrBSGrid->bs_num_rel_1 + 1;
+
+      for (i = 0; i < temp; i++) {
+        payloadBits += FDKwriteBits(
+            hBitStream, sbrEnvData->hSbrBSGrid->v_fLR[i], SBR_RES_BITS);
+      }
+      break;
+  }
+
+  return payloadBits;
+}
+
+#define SBR_CLA_BITS_LD 1
+/*****************************************************************************
+
+    functionname: encodeLowDelaySbrGrid
+    description:  if hBitStream != NULL writes bits that describes the
+                  time/frequency grouping of a frame;
+                  else counts them only
+                  (this function only write the FIXFIXonly Bitstream data)
+    returns:      number of bits written or counted
+    input:
+    output:
+
+*****************************************************************************/
+static int encodeLowDelaySbrGrid(HANDLE_SBR_ENV_DATA sbrEnvData,
+                                 HANDLE_FDK_BITSTREAM hBitStream,
+                                 const int transmitFreqs,
+                                 const UINT sbrSyntaxFlags) {
+  int payloadBits = 0;
+  int i;
+
+  /* write FIXFIXonly Grid */
+  /* write frameClass [1 bit] for FIXFIXonly Grid */
+  payloadBits += FDKwriteBits(hBitStream, 1, SBR_CLA_BITS_LD);
+
+  /* absolute Borders are fix: 0,X,X,X,nTimeSlots; so we dont have to transmit
+   * them */
+  /* only transmit the transient position! */
+  /* with this info (b1) we can reconstruct the Frame on Decoder side : */
+  /* border[0] = 0; border[1] = b1; border[2]=b1+2; border[3] = nrTimeSlots */
+
+  /* use 3 or 4bits for transient border (border) */
+  if (sbrEnvData->hSbrBSGrid->numberTimeSlots == 8)
+    payloadBits +=
+        FDKwriteBits(hBitStream, sbrEnvData->hSbrBSGrid->bs_abs_bord, 3);
+  else
+    payloadBits +=
+        FDKwriteBits(hBitStream, sbrEnvData->hSbrBSGrid->bs_abs_bord, 4);
+
+  if (transmitFreqs) {
+    /* write FreqRes grid */
+    for (i = 0; i < sbrEnvData->hSbrBSGrid->bs_num_env; i++) {
+      payloadBits += FDKwriteBits(hBitStream, sbrEnvData->hSbrBSGrid->v_f[i],
+                                  SBR_RES_BITS);
+    }
+  }
+
+  return payloadBits;
+}
+
+/*****************************************************************************
+
+    functionname: encodeSbrDtdf
+    description:  writes bits that describes the direction of the envelopes of a
+frame returns:      number of bits written input: output:
+
+*****************************************************************************/
+static INT encodeSbrDtdf(HANDLE_SBR_ENV_DATA sbrEnvData,
+                         HANDLE_FDK_BITSTREAM hBitStream) {
+  INT i, payloadBits = 0, noOfNoiseEnvelopes;
+
+  noOfNoiseEnvelopes = sbrEnvData->noOfEnvelopes > 1 ? 2 : 1;
+
+  for (i = 0; i < sbrEnvData->noOfEnvelopes; ++i) {
+    payloadBits +=
+        FDKwriteBits(hBitStream, sbrEnvData->domain_vec[i], SBR_DIR_BITS);
+  }
+  for (i = 0; i < noOfNoiseEnvelopes; ++i) {
+    payloadBits +=
+        FDKwriteBits(hBitStream, sbrEnvData->domain_vec_noise[i], SBR_DIR_BITS);
+  }
+
+  return payloadBits;
+}
+
+/*****************************************************************************
+
+    functionname: writeNoiseLevelData
+    description:  writes bits corresponding to the noise-floor-level
+    returns:      number of bits written
+    input:
+    output:
+
+*****************************************************************************/
+static INT writeNoiseLevelData(HANDLE_SBR_ENV_DATA sbrEnvData,
+                               HANDLE_FDK_BITSTREAM hBitStream, INT coupling) {
+  INT j, i, payloadBits = 0;
+  INT nNoiseEnvelopes = sbrEnvData->noOfEnvelopes > 1 ? 2 : 1;
+
+  for (i = 0; i < nNoiseEnvelopes; i++) {
+    switch (sbrEnvData->domain_vec_noise[i]) {
+      case FREQ:
+        if (coupling && sbrEnvData->balance) {
+          payloadBits += FDKwriteBits(
+              hBitStream,
+              sbrEnvData->sbr_noise_levels[i * sbrEnvData->noOfnoisebands],
+              sbrEnvData->si_sbr_start_noise_bits_balance);
+        } else {
+          payloadBits += FDKwriteBits(
+              hBitStream,
+              sbrEnvData->sbr_noise_levels[i * sbrEnvData->noOfnoisebands],
+              sbrEnvData->si_sbr_start_noise_bits);
+        }
+
+        for (j = 1 + i * sbrEnvData->noOfnoisebands;
+             j < (sbrEnvData->noOfnoisebands * (1 + i)); j++) {
+          if (coupling) {
+            if (sbrEnvData->balance) {
+              /* coupling && balance */
+              payloadBits += FDKwriteBits(hBitStream,
+                                          sbrEnvData->hufftableNoiseBalanceFreqC
+                                              [sbrEnvData->sbr_noise_levels[j] +
+                                               CODE_BOOK_SCF_LAV_BALANCE11],
+                                          sbrEnvData->hufftableNoiseBalanceFreqL
+                                              [sbrEnvData->sbr_noise_levels[j] +
+                                               CODE_BOOK_SCF_LAV_BALANCE11]);
+            } else {
+              /* coupling && !balance */
+              payloadBits += FDKwriteBits(
+                  hBitStream,
+                  sbrEnvData->hufftableNoiseLevelFreqC
+                      [sbrEnvData->sbr_noise_levels[j] + CODE_BOOK_SCF_LAV11],
+                  sbrEnvData->hufftableNoiseLevelFreqL
+                      [sbrEnvData->sbr_noise_levels[j] + CODE_BOOK_SCF_LAV11]);
+            }
+          } else {
+            /* !coupling */
+            payloadBits += FDKwriteBits(
+                hBitStream,
+                sbrEnvData
+                    ->hufftableNoiseFreqC[sbrEnvData->sbr_noise_levels[j] +
+                                          CODE_BOOK_SCF_LAV11],
+                sbrEnvData
+                    ->hufftableNoiseFreqL[sbrEnvData->sbr_noise_levels[j] +
+                                          CODE_BOOK_SCF_LAV11]);
+          }
+        }
+        break;
+
+      case TIME:
+        for (j = i * sbrEnvData->noOfnoisebands;
+             j < (sbrEnvData->noOfnoisebands * (1 + i)); j++) {
+          if (coupling) {
+            if (sbrEnvData->balance) {
+              /* coupling && balance */
+              payloadBits += FDKwriteBits(hBitStream,
+                                          sbrEnvData->hufftableNoiseBalanceTimeC
+                                              [sbrEnvData->sbr_noise_levels[j] +
+                                               CODE_BOOK_SCF_LAV_BALANCE11],
+                                          sbrEnvData->hufftableNoiseBalanceTimeL
+                                              [sbrEnvData->sbr_noise_levels[j] +
+                                               CODE_BOOK_SCF_LAV_BALANCE11]);
+            } else {
+              /* coupling && !balance */
+              payloadBits += FDKwriteBits(
+                  hBitStream,
+                  sbrEnvData->hufftableNoiseLevelTimeC
+                      [sbrEnvData->sbr_noise_levels[j] + CODE_BOOK_SCF_LAV11],
+                  sbrEnvData->hufftableNoiseLevelTimeL
+                      [sbrEnvData->sbr_noise_levels[j] + CODE_BOOK_SCF_LAV11]);
+            }
+          } else {
+            /* !coupling */
+            payloadBits += FDKwriteBits(
+                hBitStream,
+                sbrEnvData
+                    ->hufftableNoiseLevelTimeC[sbrEnvData->sbr_noise_levels[j] +
+                                               CODE_BOOK_SCF_LAV11],
+                sbrEnvData
+                    ->hufftableNoiseLevelTimeL[sbrEnvData->sbr_noise_levels[j] +
+                                               CODE_BOOK_SCF_LAV11]);
+          }
+        }
+        break;
+    }
+  }
+  return payloadBits;
+}
+
+/*****************************************************************************
+
+    functionname: writeEnvelopeData
+    description:  writes bits corresponding to the envelope
+    returns:      number of bits written
+    input:
+    output:
+
+*****************************************************************************/
+static INT writeEnvelopeData(HANDLE_SBR_ENV_DATA sbrEnvData,
+                             HANDLE_FDK_BITSTREAM hBitStream, INT coupling) {
+  INT payloadBits = 0, j, i, delta;
+
+  for (j = 0; j < sbrEnvData->noOfEnvelopes;
+       j++) { /* loop over all envelopes */
+    if (sbrEnvData->domain_vec[j] == FREQ) {
+      if (coupling && sbrEnvData->balance) {
+        payloadBits += FDKwriteBits(hBitStream, sbrEnvData->ienvelope[j][0],
+                                    sbrEnvData->si_sbr_start_env_bits_balance);
+      } else {
+        payloadBits += FDKwriteBits(hBitStream, sbrEnvData->ienvelope[j][0],
+                                    sbrEnvData->si_sbr_start_env_bits);
+      }
+    }
+
+    for (i = 1 - sbrEnvData->domain_vec[j]; i < sbrEnvData->noScfBands[j];
+         i++) {
+      delta = sbrEnvData->ienvelope[j][i];
+      if (coupling && sbrEnvData->balance) {
+        FDK_ASSERT(fixp_abs(delta) <= sbrEnvData->codeBookScfLavBalance);
+      } else {
+        FDK_ASSERT(fixp_abs(delta) <= sbrEnvData->codeBookScfLav);
+      }
+      if (coupling) {
+        if (sbrEnvData->balance) {
+          if (sbrEnvData->domain_vec[j]) {
+            /* coupling && balance && TIME */
+            payloadBits += FDKwriteBits(
+                hBitStream,
+                sbrEnvData
+                    ->hufftableBalanceTimeC[delta +
+                                            sbrEnvData->codeBookScfLavBalance],
+                sbrEnvData
+                    ->hufftableBalanceTimeL[delta +
+                                            sbrEnvData->codeBookScfLavBalance]);
+          } else {
+            /* coupling && balance && FREQ */
+            payloadBits += FDKwriteBits(
+                hBitStream,
+                sbrEnvData
+                    ->hufftableBalanceFreqC[delta +
+                                            sbrEnvData->codeBookScfLavBalance],
+                sbrEnvData
+                    ->hufftableBalanceFreqL[delta +
+                                            sbrEnvData->codeBookScfLavBalance]);
+          }
+        } else {
+          if (sbrEnvData->domain_vec[j]) {
+            /* coupling && !balance && TIME */
+            payloadBits += FDKwriteBits(
+                hBitStream,
+                sbrEnvData
+                    ->hufftableLevelTimeC[delta + sbrEnvData->codeBookScfLav],
+                sbrEnvData
+                    ->hufftableLevelTimeL[delta + sbrEnvData->codeBookScfLav]);
+          } else {
+            /* coupling && !balance && FREQ */
+            payloadBits += FDKwriteBits(
+                hBitStream,
+                sbrEnvData
+                    ->hufftableLevelFreqC[delta + sbrEnvData->codeBookScfLav],
+                sbrEnvData
+                    ->hufftableLevelFreqL[delta + sbrEnvData->codeBookScfLav]);
+          }
+        }
+      } else {
+        if (sbrEnvData->domain_vec[j]) {
+          /* !coupling && TIME */
+          payloadBits += FDKwriteBits(
+              hBitStream,
+              sbrEnvData->hufftableTimeC[delta + sbrEnvData->codeBookScfLav],
+              sbrEnvData->hufftableTimeL[delta + sbrEnvData->codeBookScfLav]);
+        } else {
+          /* !coupling && FREQ */
+          payloadBits += FDKwriteBits(
+              hBitStream,
+              sbrEnvData->hufftableFreqC[delta + sbrEnvData->codeBookScfLav],
+              sbrEnvData->hufftableFreqL[delta + sbrEnvData->codeBookScfLav]);
+        }
+      }
+    }
+  }
+  return payloadBits;
+}
+
+/*****************************************************************************
+
+    functionname: encodeExtendedData
+    description:  writes bits corresponding to the extended data
+    returns:      number of bits written
+    input:
+    output:
+
+*****************************************************************************/
+static INT encodeExtendedData(HANDLE_PARAMETRIC_STEREO hParametricStereo,
+                              HANDLE_FDK_BITSTREAM hBitStream) {
+  INT extDataSize;
+  INT payloadBits = 0;
+
+  extDataSize = getSbrExtendedDataSize(hParametricStereo);
+
+  if (extDataSize != 0) {
+    INT maxExtSize = (1 << SI_SBR_EXTENSION_SIZE_BITS) - 1;
+    INT writtenNoBits = 0; /* needed to byte align the extended data */
+
+    payloadBits += FDKwriteBits(hBitStream, 1, SI_SBR_EXTENDED_DATA_BITS);
+    FDK_ASSERT(extDataSize <= SBR_EXTENDED_DATA_MAX_CNT);
+
+    if (extDataSize < maxExtSize) {
+      payloadBits +=
+          FDKwriteBits(hBitStream, extDataSize, SI_SBR_EXTENSION_SIZE_BITS);
+    } else {
+      payloadBits +=
+          FDKwriteBits(hBitStream, maxExtSize, SI_SBR_EXTENSION_SIZE_BITS);
+      payloadBits += FDKwriteBits(hBitStream, extDataSize - maxExtSize,
+                                  SI_SBR_EXTENSION_ESC_COUNT_BITS);
+    }
+
+    /* parametric coding signalled here? */
+    if (hParametricStereo) {
+      writtenNoBits += FDKwriteBits(hBitStream, EXTENSION_ID_PS_CODING,
+                                    SI_SBR_EXTENSION_ID_BITS);
+      writtenNoBits +=
+          FDKsbrEnc_PSEnc_WritePSData(hParametricStereo, hBitStream);
+    }
+
+    payloadBits += writtenNoBits;
+
+    /* byte alignment */
+    writtenNoBits = writtenNoBits % 8;
+    if (writtenNoBits)
+      payloadBits += FDKwriteBits(hBitStream, 0, (8 - writtenNoBits));
+  } else {
+    payloadBits += FDKwriteBits(hBitStream, 0, SI_SBR_EXTENDED_DATA_BITS);
+  }
+
+  return payloadBits;
+}
+
+/*****************************************************************************
+
+    functionname: writeSyntheticCodingData
+    description:  writes bits corresponding to the "synthetic-coding"-extension
+    returns:      number of bits written
+    input:
+    output:
+
+*****************************************************************************/
+static INT writeSyntheticCodingData(HANDLE_SBR_ENV_DATA sbrEnvData,
+                                    HANDLE_FDK_BITSTREAM hBitStream)
+
+{
+  INT i;
+  INT payloadBits = 0;
+
+  payloadBits += FDKwriteBits(hBitStream, sbrEnvData->addHarmonicFlag, 1);
+
+  if (sbrEnvData->addHarmonicFlag) {
+    for (i = 0; i < sbrEnvData->noHarmonics; i++) {
+      payloadBits += FDKwriteBits(hBitStream, sbrEnvData->addHarmonic[i], 1);
+    }
+  }
+
+  return payloadBits;
+}
+
+/*****************************************************************************
+
+    functionname: getSbrExtendedDataSize
+    description:  counts the number of bits needed for encoding the
+                  extended data (including extension id)
+
+    returns:      number of bits needed for the extended data
+    input:
+    output:
+
+*****************************************************************************/
+static INT getSbrExtendedDataSize(HANDLE_PARAMETRIC_STEREO hParametricStereo) {
+  INT extDataBits = 0;
+
+  /* add your new extended data counting methods here */
+
+  /*
+    no extended data
+  */
+
+  if (hParametricStereo) {
+    /* PS extended data */
+    extDataBits += SI_SBR_EXTENSION_ID_BITS;
+    extDataBits += FDKsbrEnc_PSEnc_WritePSData(hParametricStereo, NULL);
+  }
+
+  return (extDataBits + 7) >> 3;
+}
diff --git a/fdk-aac/libSBRenc/src/bit_sbr.h b/fdk-aac/libSBRenc/src/bit_sbr.h
new file mode 100644
index 0000000..e90f52c
--- /dev/null
+++ b/fdk-aac/libSBRenc/src/bit_sbr.h
@@ -0,0 +1,267 @@
+/* -----------------------------------------------------------------------------
+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 encoder library ******************************
+
+   Author(s):
+
+   Description:
+
+*******************************************************************************/
+
+/*!
+  \file
+  \brief  SBR bit writing $Revision: 92790 $
+*/
+#ifndef BIT_SBR_H
+#define BIT_SBR_H
+
+#include "sbr_def.h"
+#include "cmondata.h"
+#include "fram_gen.h"
+
+struct SBR_ENV_DATA;
+
+struct SBR_BITSTREAM_DATA {
+  INT TotalBits;
+  INT PayloadBits;
+  INT FillBits;
+  INT HeaderActive;
+  INT HeaderActiveDelay; /**< sbr payload and its header is delayed depending on
+                            encoder configuration*/
+  INT NrSendHeaderData;  /**< input from commandline */
+  INT CountSendHeaderData; /**< modulo count. If < 0 then no counting is done
+                              (no SBR headers) */
+  INT rightBorderFIX;      /**< force VARFIX or FIXFIX frames */
+};
+
+typedef struct SBR_BITSTREAM_DATA *HANDLE_SBR_BITSTREAM_DATA;
+
+struct SBR_HEADER_DATA {
+  AMP_RES sbr_amp_res;
+  INT sbr_start_frequency;
+  INT sbr_stop_frequency;
+  INT sbr_xover_band;
+  INT sbr_noise_bands;
+  INT sbr_data_extra;
+  INT header_extra_1;
+  INT header_extra_2;
+  INT sbr_lc_stereo_mode;
+  INT sbr_limiter_bands;
+  INT sbr_limiter_gains;
+  INT sbr_interpol_freq;
+  INT sbr_smoothing_length;
+  INT alterScale;
+  INT freqScale;
+
+  /*
+    element of channelpairelement
+  */
+  INT coupling;
+  INT prev_coupling;
+
+  /*
+    element of singlechannelelement
+  */
+};
+typedef struct SBR_HEADER_DATA *HANDLE_SBR_HEADER_DATA;
+
+struct SBR_ENV_DATA {
+  INT sbr_xpos_ctrl;
+  FREQ_RES freq_res_fixfix[2];
+  UCHAR fResTransIsLow;
+
+  INVF_MODE sbr_invf_mode;
+  INVF_MODE sbr_invf_mode_vec[MAX_NUM_NOISE_VALUES];
+
+  XPOS_MODE sbr_xpos_mode;
+
+  INT ienvelope[MAX_ENVELOPES][MAX_FREQ_COEFFS];
+
+  INT codeBookScfLavBalance;
+  INT codeBookScfLav;
+  const INT *hufftableTimeC;
+  const INT *hufftableFreqC;
+  const UCHAR *hufftableTimeL;
+  const UCHAR *hufftableFreqL;
+
+  const INT *hufftableLevelTimeC;
+  const INT *hufftableBalanceTimeC;
+  const INT *hufftableLevelFreqC;
+  const INT *hufftableBalanceFreqC;
+  const UCHAR *hufftableLevelTimeL;
+  const UCHAR *hufftableBalanceTimeL;
+  const UCHAR *hufftableLevelFreqL;
+  const UCHAR *hufftableBalanceFreqL;
+
+  const UCHAR *hufftableNoiseTimeL;
+  const INT *hufftableNoiseTimeC;
+  const UCHAR *hufftableNoiseFreqL;
+  const INT *hufftableNoiseFreqC;
+
+  const UCHAR *hufftableNoiseLevelTimeL;
+  const INT *hufftableNoiseLevelTimeC;
+  const UCHAR *hufftableNoiseBalanceTimeL;
+  const INT *hufftableNoiseBalanceTimeC;
+  const UCHAR *hufftableNoiseLevelFreqL;
+  const INT *hufftableNoiseLevelFreqC;
+  const UCHAR *hufftableNoiseBalanceFreqL;
+  const INT *hufftableNoiseBalanceFreqC;
+
+  HANDLE_SBR_GRID hSbrBSGrid;
+
+  INT noHarmonics;
+  INT addHarmonicFlag;
+  UCHAR addHarmonic[MAX_FREQ_COEFFS];
+
+  /* calculated helper vars */
+  INT si_sbr_start_env_bits_balance;
+  INT si_sbr_start_env_bits;
+  INT si_sbr_start_noise_bits_balance;
+  INT si_sbr_start_noise_bits;
+
+  INT noOfEnvelopes;
+  INT noScfBands[MAX_ENVELOPES];
+  INT domain_vec[MAX_ENVELOPES];
+  INT domain_vec_noise[MAX_ENVELOPES];
+  SCHAR sbr_noise_levels[MAX_FREQ_COEFFS];
+  INT noOfnoisebands;
+
+  INT balance;
+  AMP_RES init_sbr_amp_res;
+  AMP_RES currentAmpResFF;
+  FIXP_DBL
+  ton_HF[SBR_GLOBAL_TONALITY_VALUES]; /* tonality is scaled by
+                                         2^19/0.524288f (fract part of
+                                         RELAXATION) */
+  FIXP_DBL global_tonality;
+
+  /* extended data */
+  INT extended_data;
+  INT extension_size;
+  INT extension_id;
+  UCHAR extended_data_buffer[SBR_EXTENDED_DATA_MAX_CNT];
+
+  UCHAR ldGrid;
+};
+typedef struct SBR_ENV_DATA *HANDLE_SBR_ENV_DATA;
+
+INT FDKsbrEnc_WriteEnvSingleChannelElement(
+    struct SBR_HEADER_DATA *sbrHeaderData,
+    struct T_PARAMETRIC_STEREO *hParametricStereo,
+    struct SBR_BITSTREAM_DATA *sbrBitstreamData,
+    struct SBR_ENV_DATA *sbrEnvData, struct COMMON_DATA *cmonData,
+    UINT sbrSyntaxFlags);
+
+INT FDKsbrEnc_WriteEnvChannelPairElement(
+    struct SBR_HEADER_DATA *sbrHeaderData,
+    struct T_PARAMETRIC_STEREO *hParametricStereo,
+    struct SBR_BITSTREAM_DATA *sbrBitstreamData,
+    struct SBR_ENV_DATA *sbrEnvDataLeft, struct SBR_ENV_DATA *sbrEnvDataRight,
+    struct COMMON_DATA *cmonData, UINT sbrSyntaxFlags);
+
+INT FDKsbrEnc_CountSbrChannelPairElement(
+    struct SBR_HEADER_DATA *sbrHeaderData,
+    struct T_PARAMETRIC_STEREO *hParametricStereo,
+    struct SBR_BITSTREAM_DATA *sbrBitstreamData,
+    struct SBR_ENV_DATA *sbrEnvDataLeft, struct SBR_ENV_DATA *sbrEnvDataRight,
+    struct COMMON_DATA *cmonData, UINT sbrSyntaxFlags);
+
+/* debugging and tuning functions */
+
+/*#define SBR_ENV_STATISTICS */
+
+/*#define SBR_PAYLOAD_MONITOR*/
+
+#endif
diff --git a/fdk-aac/libSBRenc/src/cmondata.h b/fdk-aac/libSBRenc/src/cmondata.h
new file mode 100644
index 0000000..0779b4d
--- /dev/null
+++ b/fdk-aac/libSBRenc/src/cmondata.h
@@ -0,0 +1,127 @@
+/* -----------------------------------------------------------------------------
+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 encoder library ******************************
+
+   Author(s):
+
+   Description:
+
+*******************************************************************************/
+
+/*!
+  \file
+  \brief  Core Coder's and SBR's shared data structure definition $Revision:
+  92790 $
+*/
+#ifndef CMONDATA_H
+#define CMONDATA_H
+
+#include "FDK_bitstream.h"
+
+struct COMMON_DATA {
+  INT sbrHdrBits;               /**< number of SBR header bits */
+  INT sbrDataBits;              /**< number of SBR data bits */
+  INT sbrFillBits;              /**< number of SBR fill bits */
+  FDK_BITSTREAM sbrBitbuf;      /**< the SBR data bitbuffer */
+  FDK_BITSTREAM tmpWriteBitbuf; /**< helper var for writing header*/
+  INT xOverFreq;                /**< the SBR crossover frequency */
+  INT dynBwEnabled;    /**< indicates if dynamic bandwidth is enabled */
+  INT sbrNumChannels;  /**< number of channels (meaning mono or stereo) */
+  INT dynXOverFreqEnc; /**< encoder dynamic crossover frequency */
+};
+
+typedef struct COMMON_DATA *HANDLE_COMMON_DATA;
+
+#endif
diff --git a/fdk-aac/libSBRenc/src/code_env.cpp b/fdk-aac/libSBRenc/src/code_env.cpp
new file mode 100644
index 0000000..fb0f6a4
--- /dev/null
+++ b/fdk-aac/libSBRenc/src/code_env.cpp
@@ -0,0 +1,602 @@
+/* -----------------------------------------------------------------------------
+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 encoder library ******************************
+
+   Author(s):
+
+   Description:
+
+*******************************************************************************/
+
+#include "code_env.h"
+#include "sbrenc_rom.h"
+
+/*****************************************************************************
+
+ functionname: FDKsbrEnc_InitSbrHuffmanTables
+ description:  initializes Huffman Tables dependent on chosen amp_res
+ returns:      error handle
+ input:
+ output:
+
+*****************************************************************************/
+INT FDKsbrEnc_InitSbrHuffmanTables(HANDLE_SBR_ENV_DATA sbrEnvData,
+                                   HANDLE_SBR_CODE_ENVELOPE henv,
+                                   HANDLE_SBR_CODE_ENVELOPE hnoise,
+                                   AMP_RES amp_res) {
+  if ((!henv) || (!hnoise) || (!sbrEnvData)) return (1); /* not init. */
+
+  sbrEnvData->init_sbr_amp_res = amp_res;
+
+  switch (amp_res) {
+    case SBR_AMP_RES_3_0:
+      /*envelope data*/
+
+      /*Level/Pan - coding */
+      sbrEnvData->hufftableLevelTimeC = v_Huff_envelopeLevelC11T;
+      sbrEnvData->hufftableLevelTimeL = v_Huff_envelopeLevelL11T;
+      sbrEnvData->hufftableBalanceTimeC = bookSbrEnvBalanceC11T;
+      sbrEnvData->hufftableBalanceTimeL = bookSbrEnvBalanceL11T;
+
+      sbrEnvData->hufftableLevelFreqC = v_Huff_envelopeLevelC11F;
+      sbrEnvData->hufftableLevelFreqL = v_Huff_envelopeLevelL11F;
+      sbrEnvData->hufftableBalanceFreqC = bookSbrEnvBalanceC11F;
+      sbrEnvData->hufftableBalanceFreqL = bookSbrEnvBalanceL11F;
+
+      /*Right/Left - coding */
+      sbrEnvData->hufftableTimeC = v_Huff_envelopeLevelC11T;
+      sbrEnvData->hufftableTimeL = v_Huff_envelopeLevelL11T;
+      sbrEnvData->hufftableFreqC = v_Huff_envelopeLevelC11F;
+      sbrEnvData->hufftableFreqL = v_Huff_envelopeLevelL11F;
+
+      sbrEnvData->codeBookScfLavBalance = CODE_BOOK_SCF_LAV_BALANCE11;
+      sbrEnvData->codeBookScfLav = CODE_BOOK_SCF_LAV11;
+
+      sbrEnvData->si_sbr_start_env_bits = SI_SBR_START_ENV_BITS_AMP_RES_3_0;
+      sbrEnvData->si_sbr_start_env_bits_balance =
+          SI_SBR_START_ENV_BITS_BALANCE_AMP_RES_3_0;
+      break;
+
+    case SBR_AMP_RES_1_5:
+      /*envelope data*/
+
+      /*Level/Pan - coding */
+      sbrEnvData->hufftableLevelTimeC = v_Huff_envelopeLevelC10T;
+      sbrEnvData->hufftableLevelTimeL = v_Huff_envelopeLevelL10T;
+      sbrEnvData->hufftableBalanceTimeC = bookSbrEnvBalanceC10T;
+      sbrEnvData->hufftableBalanceTimeL = bookSbrEnvBalanceL10T;
+
+      sbrEnvData->hufftableLevelFreqC = v_Huff_envelopeLevelC10F;
+      sbrEnvData->hufftableLevelFreqL = v_Huff_envelopeLevelL10F;
+      sbrEnvData->hufftableBalanceFreqC = bookSbrEnvBalanceC10F;
+      sbrEnvData->hufftableBalanceFreqL = bookSbrEnvBalanceL10F;
+
+      /*Right/Left - coding */
+      sbrEnvData->hufftableTimeC = v_Huff_envelopeLevelC10T;
+      sbrEnvData->hufftableTimeL = v_Huff_envelopeLevelL10T;
+      sbrEnvData->hufftableFreqC = v_Huff_envelopeLevelC10F;
+      sbrEnvData->hufftableFreqL = v_Huff_envelopeLevelL10F;
+
+      sbrEnvData->codeBookScfLavBalance = CODE_BOOK_SCF_LAV_BALANCE10;
+      sbrEnvData->codeBookScfLav = CODE_BOOK_SCF_LAV10;
+
+      sbrEnvData->si_sbr_start_env_bits = SI_SBR_START_ENV_BITS_AMP_RES_1_5;
+      sbrEnvData->si_sbr_start_env_bits_balance =
+          SI_SBR_START_ENV_BITS_BALANCE_AMP_RES_1_5;
+      break;
+
+    default:
+      return (1); /* undefined amp_res mode */
+  }
+
+  /* these are common to both amp_res values */
+  /*Noise data*/
+
+  /*Level/Pan - coding */
+  sbrEnvData->hufftableNoiseLevelTimeC = v_Huff_NoiseLevelC11T;
+  sbrEnvData->hufftableNoiseLevelTimeL = v_Huff_NoiseLevelL11T;
+  sbrEnvData->hufftableNoiseBalanceTimeC = bookSbrNoiseBalanceC11T;
+  sbrEnvData->hufftableNoiseBalanceTimeL = bookSbrNoiseBalanceL11T;
+
+  sbrEnvData->hufftableNoiseLevelFreqC = v_Huff_envelopeLevelC11F;
+  sbrEnvData->hufftableNoiseLevelFreqL = v_Huff_envelopeLevelL11F;
+  sbrEnvData->hufftableNoiseBalanceFreqC = bookSbrEnvBalanceC11F;
+  sbrEnvData->hufftableNoiseBalanceFreqL = bookSbrEnvBalanceL11F;
+
+  /*Right/Left - coding */
+  sbrEnvData->hufftableNoiseTimeC = v_Huff_NoiseLevelC11T;
+  sbrEnvData->hufftableNoiseTimeL = v_Huff_NoiseLevelL11T;
+  sbrEnvData->hufftableNoiseFreqC = v_Huff_envelopeLevelC11F;
+  sbrEnvData->hufftableNoiseFreqL = v_Huff_envelopeLevelL11F;
+
+  sbrEnvData->si_sbr_start_noise_bits = SI_SBR_START_NOISE_BITS_AMP_RES_3_0;
+  sbrEnvData->si_sbr_start_noise_bits_balance =
+      SI_SBR_START_NOISE_BITS_BALANCE_AMP_RES_3_0;
+
+  /* init envelope tables and codebooks */
+  henv->codeBookScfLavBalanceTime = sbrEnvData->codeBookScfLavBalance;
+  henv->codeBookScfLavBalanceFreq = sbrEnvData->codeBookScfLavBalance;
+  henv->codeBookScfLavLevelTime = sbrEnvData->codeBookScfLav;
+  henv->codeBookScfLavLevelFreq = sbrEnvData->codeBookScfLav;
+  henv->codeBookScfLavTime = sbrEnvData->codeBookScfLav;
+  henv->codeBookScfLavFreq = sbrEnvData->codeBookScfLav;
+
+  henv->hufftableLevelTimeL = sbrEnvData->hufftableLevelTimeL;
+  henv->hufftableBalanceTimeL = sbrEnvData->hufftableBalanceTimeL;
+  henv->hufftableTimeL = sbrEnvData->hufftableTimeL;
+  henv->hufftableLevelFreqL = sbrEnvData->hufftableLevelFreqL;
+  henv->hufftableBalanceFreqL = sbrEnvData->hufftableBalanceFreqL;
+  henv->hufftableFreqL = sbrEnvData->hufftableFreqL;
+
+  henv->codeBookScfLavFreq = sbrEnvData->codeBookScfLav;
+  henv->codeBookScfLavTime = sbrEnvData->codeBookScfLav;
+
+  henv->start_bits = sbrEnvData->si_sbr_start_env_bits;
+  henv->start_bits_balance = sbrEnvData->si_sbr_start_env_bits_balance;
+
+  /* init noise tables and codebooks */
+
+  hnoise->codeBookScfLavBalanceTime = CODE_BOOK_SCF_LAV_BALANCE11;
+  hnoise->codeBookScfLavBalanceFreq = CODE_BOOK_SCF_LAV_BALANCE11;
+  hnoise->codeBookScfLavLevelTime = CODE_BOOK_SCF_LAV11;
+  hnoise->codeBookScfLavLevelFreq = CODE_BOOK_SCF_LAV11;
+  hnoise->codeBookScfLavTime = CODE_BOOK_SCF_LAV11;
+  hnoise->codeBookScfLavFreq = CODE_BOOK_SCF_LAV11;
+
+  hnoise->hufftableLevelTimeL = sbrEnvData->hufftableNoiseLevelTimeL;
+  hnoise->hufftableBalanceTimeL = sbrEnvData->hufftableNoiseBalanceTimeL;
+  hnoise->hufftableTimeL = sbrEnvData->hufftableNoiseTimeL;
+  hnoise->hufftableLevelFreqL = sbrEnvData->hufftableNoiseLevelFreqL;
+  hnoise->hufftableBalanceFreqL = sbrEnvData->hufftableNoiseBalanceFreqL;
+  hnoise->hufftableFreqL = sbrEnvData->hufftableNoiseFreqL;
+
+  hnoise->start_bits = sbrEnvData->si_sbr_start_noise_bits;
+  hnoise->start_bits_balance = sbrEnvData->si_sbr_start_noise_bits_balance;
+
+  /* No delta coding in time from the previous frame due to 1.5dB FIx-FIX rule
+   */
+  henv->upDate = 0;
+  hnoise->upDate = 0;
+  return (0);
+}
+
+/*******************************************************************************
+ Functionname:  indexLow2High
+ *******************************************************************************
+
+ Description:   Nice small patch-functions in order to cope with non-factor-2
+                ratios between high-res and low-res
+
+ Arguments:     INT offset, INT index, FREQ_RES res
+
+ Return:        INT
+
+*******************************************************************************/
+static INT indexLow2High(INT offset, INT index, FREQ_RES res) {
+  if (res == FREQ_RES_LOW) {
+    if (offset >= 0) {
+      if (index < offset)
+        return (index);
+      else
+        return (2 * index - offset);
+    } else {
+      offset = -offset;
+      if (index < offset)
+        return (2 * index + index);
+      else
+        return (2 * index + offset);
+    }
+  } else
+    return (index);
+}
+
+/*******************************************************************************
+ Functionname:  mapLowResEnergyVal
+ *******************************************************************************
+
+ Description:
+
+ Arguments:     INT currVal,INT* prevData, INT offset, INT index, FREQ_RES res
+
+ Return:        none
+
+*******************************************************************************/
+static void mapLowResEnergyVal(SCHAR currVal, SCHAR *prevData, INT offset,
+                               INT index, FREQ_RES res) {
+  if (res == FREQ_RES_LOW) {
+    if (offset >= 0) {
+      if (index < offset)
+        prevData[index] = currVal;
+      else {
+        prevData[2 * index - offset] = currVal;
+        prevData[2 * index + 1 - offset] = currVal;
+      }
+    } else {
+      offset = -offset;
+      if (index < offset) {
+        prevData[3 * index] = currVal;
+        prevData[3 * index + 1] = currVal;
+        prevData[3 * index + 2] = currVal;
+      } else {
+        prevData[2 * index + offset] = currVal;
+        prevData[2 * index + 1 + offset] = currVal;
+      }
+    }
+  } else
+    prevData[index] = currVal;
+}
+
+/*******************************************************************************
+ Functionname:  computeBits
+ *******************************************************************************
+
+ Description:
+
+ Arguments:     INT delta,
+                INT codeBookScfLavLevel,
+                INT codeBookScfLavBalance,
+                const UCHAR * hufftableLevel,
+                const UCHAR * hufftableBalance, INT coupling, INT channel)
+
+ Return:        INT
+
+*******************************************************************************/
+static INT computeBits(SCHAR *delta, INT codeBookScfLavLevel,
+                       INT codeBookScfLavBalance, const UCHAR *hufftableLevel,
+                       const UCHAR *hufftableBalance, INT coupling,
+                       INT channel) {
+  INT index;
+  INT delta_bits = 0;
+
+  if (coupling) {
+    if (channel == 1) {
+      if (*delta < 0)
+        index = fixMax(*delta, -codeBookScfLavBalance);
+      else
+        index = fixMin(*delta, codeBookScfLavBalance);
+
+      if (index != *delta) {
+        *delta = index;
+        return (10000);
+      }
+
+      delta_bits = hufftableBalance[index + codeBookScfLavBalance];
+    } else {
+      if (*delta < 0)
+        index = fixMax(*delta, -codeBookScfLavLevel);
+      else
+        index = fixMin(*delta, codeBookScfLavLevel);
+
+      if (index != *delta) {
+        *delta = index;
+        return (10000);
+      }
+      delta_bits = hufftableLevel[index + codeBookScfLavLevel];
+    }
+  } else {
+    if (*delta < 0)
+      index = fixMax(*delta, -codeBookScfLavLevel);
+    else
+      index = fixMin(*delta, codeBookScfLavLevel);
+
+    if (index != *delta) {
+      *delta = index;
+      return (10000);
+    }
+    delta_bits = hufftableLevel[index + codeBookScfLavLevel];
+  }
+
+  return (delta_bits);
+}
+
+/*******************************************************************************
+ Functionname:  FDKsbrEnc_codeEnvelope
+ *******************************************************************************
+
+ Description:
+
+ Arguments:     INT *sfb_nrg,
+                const FREQ_RES *freq_res,
+                SBR_CODE_ENVELOPE * h_sbrCodeEnvelope,
+                INT *directionVec, INT scalable, INT nEnvelopes, INT channel,
+                INT headerActive)
+
+ Return:        none
+                h_sbrCodeEnvelope->sfb_nrg_prev is modified !
+                sfb_nrg is modified
+                h_sbrCodeEnvelope->update is modfied !
+                *directionVec is modified
+
+*******************************************************************************/
+void FDKsbrEnc_codeEnvelope(SCHAR *sfb_nrg, const FREQ_RES *freq_res,
+                            SBR_CODE_ENVELOPE *h_sbrCodeEnvelope,
+                            INT *directionVec, INT coupling, INT nEnvelopes,
+                            INT channel, INT headerActive) {
+  INT i, no_of_bands, band;
+  FIXP_DBL tmp1, tmp2, tmp3, dF_edge_1stEnv;
+  SCHAR *ptr_nrg;
+
+  INT codeBookScfLavLevelTime;
+  INT codeBookScfLavLevelFreq;
+  INT codeBookScfLavBalanceTime;
+  INT codeBookScfLavBalanceFreq;
+  const UCHAR *hufftableLevelTimeL;
+  const UCHAR *hufftableBalanceTimeL;
+  const UCHAR *hufftableLevelFreqL;
+  const UCHAR *hufftableBalanceFreqL;
+
+  INT offset = h_sbrCodeEnvelope->offset;
+  INT envDataTableCompFactor;
+
+  INT delta_F_bits = 0, delta_T_bits = 0;
+  INT use_dT;
+
+  SCHAR delta_F[MAX_FREQ_COEFFS];
+  SCHAR delta_T[MAX_FREQ_COEFFS];
+  SCHAR last_nrg, curr_nrg;
+
+  tmp1 = FL2FXCONST_DBL(0.5f) >> (DFRACT_BITS - 16 - 1);
+  tmp2 = h_sbrCodeEnvelope->dF_edge_1stEnv >> (DFRACT_BITS - 16);
+  tmp3 = (FIXP_DBL)fMult(h_sbrCodeEnvelope->dF_edge_incr,
+                         ((FIXP_DBL)h_sbrCodeEnvelope->dF_edge_incr_fac) << 15);
+
+  dF_edge_1stEnv = tmp1 + tmp2 + tmp3;
+
+  if (coupling) {
+    codeBookScfLavLevelTime = h_sbrCodeEnvelope->codeBookScfLavLevelTime;
+    codeBookScfLavLevelFreq = h_sbrCodeEnvelope->codeBookScfLavLevelFreq;
+    codeBookScfLavBalanceTime = h_sbrCodeEnvelope->codeBookScfLavBalanceTime;
+    codeBookScfLavBalanceFreq = h_sbrCodeEnvelope->codeBookScfLavBalanceFreq;
+    hufftableLevelTimeL = h_sbrCodeEnvelope->hufftableLevelTimeL;
+    hufftableBalanceTimeL = h_sbrCodeEnvelope->hufftableBalanceTimeL;
+    hufftableLevelFreqL = h_sbrCodeEnvelope->hufftableLevelFreqL;
+    hufftableBalanceFreqL = h_sbrCodeEnvelope->hufftableBalanceFreqL;
+  } else {
+    codeBookScfLavLevelTime = h_sbrCodeEnvelope->codeBookScfLavTime;
+    codeBookScfLavLevelFreq = h_sbrCodeEnvelope->codeBookScfLavFreq;
+    codeBookScfLavBalanceTime = h_sbrCodeEnvelope->codeBookScfLavTime;
+    codeBookScfLavBalanceFreq = h_sbrCodeEnvelope->codeBookScfLavFreq;
+    hufftableLevelTimeL = h_sbrCodeEnvelope->hufftableTimeL;
+    hufftableBalanceTimeL = h_sbrCodeEnvelope->hufftableTimeL;
+    hufftableLevelFreqL = h_sbrCodeEnvelope->hufftableFreqL;
+    hufftableBalanceFreqL = h_sbrCodeEnvelope->hufftableFreqL;
+  }
+
+  if (coupling == 1 && channel == 1)
+    envDataTableCompFactor =
+        1; /*should be one when the new huffman-tables are ready*/
+  else
+    envDataTableCompFactor = 0;
+
+  if (h_sbrCodeEnvelope->deltaTAcrossFrames == 0) h_sbrCodeEnvelope->upDate = 0;
+
+  /* no delta coding in time in case of a header */
+  if (headerActive) h_sbrCodeEnvelope->upDate = 0;
+
+  for (i = 0; i < nEnvelopes; i++) {
+    if (freq_res[i] == FREQ_RES_HIGH)
+      no_of_bands = h_sbrCodeEnvelope->nSfb[FREQ_RES_HIGH];
+    else
+      no_of_bands = h_sbrCodeEnvelope->nSfb[FREQ_RES_LOW];
+
+    ptr_nrg = sfb_nrg;
+    curr_nrg = *ptr_nrg;
+
+    delta_F[0] = curr_nrg >> envDataTableCompFactor;
+
+    if (coupling && channel == 1)
+      delta_F_bits = h_sbrCodeEnvelope->start_bits_balance;
+    else
+      delta_F_bits = h_sbrCodeEnvelope->start_bits;
+
+    if (h_sbrCodeEnvelope->upDate != 0) {
+      delta_T[0] = (curr_nrg - h_sbrCodeEnvelope->sfb_nrg_prev[0]) >>
+                   envDataTableCompFactor;
+
+      delta_T_bits = computeBits(&delta_T[0], codeBookScfLavLevelTime,
+                                 codeBookScfLavBalanceTime, hufftableLevelTimeL,
+                                 hufftableBalanceTimeL, coupling, channel);
+    }
+
+    mapLowResEnergyVal(curr_nrg, h_sbrCodeEnvelope->sfb_nrg_prev, offset, 0,
+                       freq_res[i]);
+
+    /* ensure that nrg difference is not higher than codeBookScfLavXXXFreq */
+    if (coupling && channel == 1) {
+      for (band = no_of_bands - 1; band > 0; band--) {
+        if (ptr_nrg[band] - ptr_nrg[band - 1] > codeBookScfLavBalanceFreq) {
+          ptr_nrg[band - 1] = ptr_nrg[band] - codeBookScfLavBalanceFreq;
+        }
+      }
+      for (band = 1; band < no_of_bands; band++) {
+        if (ptr_nrg[band - 1] - ptr_nrg[band] > codeBookScfLavBalanceFreq) {
+          ptr_nrg[band] = ptr_nrg[band - 1] - codeBookScfLavBalanceFreq;
+        }
+      }
+    } else {
+      for (band = no_of_bands - 1; band > 0; band--) {
+        if (ptr_nrg[band] - ptr_nrg[band - 1] > codeBookScfLavLevelFreq) {
+          ptr_nrg[band - 1] = ptr_nrg[band] - codeBookScfLavLevelFreq;
+        }
+      }
+      for (band = 1; band < no_of_bands; band++) {
+        if (ptr_nrg[band - 1] - ptr_nrg[band] > codeBookScfLavLevelFreq) {
+          ptr_nrg[band] = ptr_nrg[band - 1] - codeBookScfLavLevelFreq;
+        }
+      }
+    }
+
+    /* Coding loop*/
+    for (band = 1; band < no_of_bands; band++) {
+      last_nrg = (*ptr_nrg);
+      ptr_nrg++;
+      curr_nrg = (*ptr_nrg);
+
+      delta_F[band] = (curr_nrg - last_nrg) >> envDataTableCompFactor;
+
+      delta_F_bits += computeBits(
+          &delta_F[band], codeBookScfLavLevelFreq, codeBookScfLavBalanceFreq,
+          hufftableLevelFreqL, hufftableBalanceFreqL, coupling, channel);
+
+      if (h_sbrCodeEnvelope->upDate != 0) {
+        delta_T[band] =
+            curr_nrg -
+            h_sbrCodeEnvelope
+                ->sfb_nrg_prev[indexLow2High(offset, band, freq_res[i])];
+        delta_T[band] = delta_T[band] >> envDataTableCompFactor;
+      }
+
+      mapLowResEnergyVal(curr_nrg, h_sbrCodeEnvelope->sfb_nrg_prev, offset,
+                         band, freq_res[i]);
+
+      if (h_sbrCodeEnvelope->upDate != 0) {
+        delta_T_bits += computeBits(
+            &delta_T[band], codeBookScfLavLevelTime, codeBookScfLavBalanceTime,
+            hufftableLevelTimeL, hufftableBalanceTimeL, coupling, channel);
+      }
+    }
+
+    /* Replace sfb_nrg with deltacoded samples and set flag */
+    if (i == 0) {
+      INT tmp_bits;
+      tmp_bits = (((delta_T_bits * dF_edge_1stEnv) >> (DFRACT_BITS - 18)) +
+                  (FIXP_DBL)1) >>
+                 1;
+      use_dT = (h_sbrCodeEnvelope->upDate != 0 && (delta_F_bits > tmp_bits));
+    } else
+      use_dT = (delta_T_bits < delta_F_bits && h_sbrCodeEnvelope->upDate != 0);
+
+    if (use_dT) {
+      directionVec[i] = TIME;
+      FDKmemcpy(sfb_nrg, delta_T, no_of_bands * sizeof(SCHAR));
+    } else {
+      h_sbrCodeEnvelope->upDate = 0;
+      directionVec[i] = FREQ;
+      FDKmemcpy(sfb_nrg, delta_F, no_of_bands * sizeof(SCHAR));
+    }
+    sfb_nrg += no_of_bands;
+    h_sbrCodeEnvelope->upDate = 1;
+  }
+}
+
+/*******************************************************************************
+ Functionname:  FDKsbrEnc_InitSbrCodeEnvelope
+ *******************************************************************************
+
+ Description:
+
+ Arguments:
+
+ Return:
+
+*******************************************************************************/
+INT FDKsbrEnc_InitSbrCodeEnvelope(HANDLE_SBR_CODE_ENVELOPE h_sbrCodeEnvelope,
+                                  INT *nSfb, INT deltaTAcrossFrames,
+                                  FIXP_DBL dF_edge_1stEnv,
+                                  FIXP_DBL dF_edge_incr) {
+  FDKmemclear(h_sbrCodeEnvelope, sizeof(SBR_CODE_ENVELOPE));
+
+  h_sbrCodeEnvelope->deltaTAcrossFrames = deltaTAcrossFrames;
+  h_sbrCodeEnvelope->dF_edge_1stEnv = dF_edge_1stEnv;
+  h_sbrCodeEnvelope->dF_edge_incr = dF_edge_incr;
+  h_sbrCodeEnvelope->dF_edge_incr_fac = 0;
+  h_sbrCodeEnvelope->upDate = 0;
+  h_sbrCodeEnvelope->nSfb[FREQ_RES_LOW] = nSfb[FREQ_RES_LOW];
+  h_sbrCodeEnvelope->nSfb[FREQ_RES_HIGH] = nSfb[FREQ_RES_HIGH];
+  h_sbrCodeEnvelope->offset = 2 * h_sbrCodeEnvelope->nSfb[FREQ_RES_LOW] -
+                              h_sbrCodeEnvelope->nSfb[FREQ_RES_HIGH];
+
+  return (0);
+}
diff --git a/fdk-aac/libSBRenc/src/code_env.h b/fdk-aac/libSBRenc/src/code_env.h
new file mode 100644
index 0000000..673a783
--- /dev/null
+++ b/fdk-aac/libSBRenc/src/code_env.h
@@ -0,0 +1,161 @@
+/* -----------------------------------------------------------------------------
+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 encoder library ******************************
+
+   Author(s):
+
+   Description:
+
+*******************************************************************************/
+
+/*!
+  \file
+  \brief  DPCM Envelope coding $Revision: 92790 $
+*/
+
+#ifndef CODE_ENV_H
+#define CODE_ENV_H
+
+#include "sbr_def.h"
+#include "bit_sbr.h"
+#include "fram_gen.h"
+
+typedef struct {
+  INT offset;
+  INT upDate;
+  INT nSfb[2];
+  SCHAR sfb_nrg_prev[MAX_FREQ_COEFFS];
+  INT deltaTAcrossFrames;
+  FIXP_DBL dF_edge_1stEnv;
+  FIXP_DBL dF_edge_incr;
+  INT dF_edge_incr_fac;
+
+  INT codeBookScfLavTime;
+  INT codeBookScfLavFreq;
+
+  INT codeBookScfLavLevelTime;
+  INT codeBookScfLavLevelFreq;
+  INT codeBookScfLavBalanceTime;
+  INT codeBookScfLavBalanceFreq;
+
+  INT start_bits;
+  INT start_bits_balance;
+
+  const UCHAR *hufftableTimeL;
+  const UCHAR *hufftableFreqL;
+
+  const UCHAR *hufftableLevelTimeL;
+  const UCHAR *hufftableBalanceTimeL;
+  const UCHAR *hufftableLevelFreqL;
+  const UCHAR *hufftableBalanceFreqL;
+} SBR_CODE_ENVELOPE;
+typedef SBR_CODE_ENVELOPE *HANDLE_SBR_CODE_ENVELOPE;
+
+void FDKsbrEnc_codeEnvelope(SCHAR *sfb_nrg, const FREQ_RES *freq_res,
+                            SBR_CODE_ENVELOPE *h_sbrCodeEnvelope,
+                            INT *directionVec, INT coupling, INT nEnvelopes,
+                            INT channel, INT headerActive);
+
+INT FDKsbrEnc_InitSbrCodeEnvelope(HANDLE_SBR_CODE_ENVELOPE h_sbrCodeEnvelope,
+                                  INT *nSfb, INT deltaTAcrossFrames,
+                                  FIXP_DBL dF_edge_1stEnv,
+                                  FIXP_DBL dF_edge_incr);
+
+INT FDKsbrEnc_InitSbrHuffmanTables(struct SBR_ENV_DATA *sbrEnvData,
+                                   HANDLE_SBR_CODE_ENVELOPE henv,
+                                   HANDLE_SBR_CODE_ENVELOPE hnoise,
+                                   AMP_RES amp_res);
+
+#endif
diff --git a/fdk-aac/libSBRenc/src/env_bit.cpp b/fdk-aac/libSBRenc/src/env_bit.cpp
new file mode 100644
index 0000000..41812ac
--- /dev/null
+++ b/fdk-aac/libSBRenc/src/env_bit.cpp
@@ -0,0 +1,257 @@
+/* -----------------------------------------------------------------------------
+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 encoder library ******************************
+
+   Author(s):
+
+   Description:
+
+*******************************************************************************/
+
+/*!
+  \file
+  \brief  Remaining SBR Bit Writing Routines
+*/
+
+#include "env_bit.h"
+#include "cmondata.h"
+
+#ifndef min
+#define min(a, b) (a < b ? a : b)
+#endif
+
+#ifndef max
+#define max(a, b) (a > b ? a : b)
+#endif
+
+/* ***************************** crcAdvance **********************************/
+/**
+ * @fn
+ * @brief    updates crc data register
+ * @return   none
+ *
+ * This function updates the crc register
+ *
+ */
+static void crcAdvance(USHORT crcPoly, USHORT crcMask, USHORT *crc,
+                       ULONG bValue, INT bBits) {
+  INT i;
+  USHORT flag;
+
+  for (i = bBits - 1; i >= 0; i--) {
+    flag = ((*crc) & crcMask) ? (1) : (0);
+    flag ^= (bValue & (1 << i)) ? (1) : (0);
+
+    (*crc) <<= 1;
+    if (flag) (*crc) ^= crcPoly;
+  }
+}
+
+/* ***************************** FDKsbrEnc_InitSbrBitstream
+ * **********************************/
+/**
+ * @fn
+ * @brief    Inittialisation of sbr bitstream, write of dummy header and CRC
+ * @return   none
+ *
+ *
+ *
+ */
+
+INT FDKsbrEnc_InitSbrBitstream(
+    HANDLE_COMMON_DATA hCmonData,
+    UCHAR *memoryBase, /*!< Pointer to bitstream buffer */
+    INT memorySize,    /*!< Length of bitstream buffer in bytes */
+    HANDLE_FDK_CRCINFO hCrcInfo, UINT sbrSyntaxFlags) /*!< SBR syntax flags */
+{
+  INT crcRegion = 0;
+
+  /* reset bit buffer */
+  FDKresetBitbuffer(&hCmonData->sbrBitbuf, BS_WRITER);
+
+  FDKinitBitStream(&hCmonData->tmpWriteBitbuf, memoryBase, memorySize, 0,
+                   BS_WRITER);
+
+  if (sbrSyntaxFlags & SBR_SYNTAX_CRC) {
+    if (sbrSyntaxFlags & SBR_SYNTAX_DRM_CRC) { /* Init and start CRC region */
+      FDKwriteBits(&hCmonData->sbrBitbuf, 0x0, SI_SBR_DRM_CRC_BITS);
+      FDKcrcInit(hCrcInfo, 0x001d, 0xFFFF, SI_SBR_DRM_CRC_BITS);
+      crcRegion = FDKcrcStartReg(hCrcInfo, &hCmonData->sbrBitbuf, 0);
+    } else {
+      FDKwriteBits(&hCmonData->sbrBitbuf, 0x0, SI_SBR_CRC_BITS);
+    }
+  }
+
+  return (crcRegion);
+}
+
+/* ************************** FDKsbrEnc_AssembleSbrBitstream
+ * *******************************/
+/**
+ * @fn
+ * @brief    Formats the SBR payload
+ * @return   nothing
+ *
+ * Also the CRC will be calculated here.
+ *
+ */
+
+void FDKsbrEnc_AssembleSbrBitstream(HANDLE_COMMON_DATA hCmonData,
+                                    HANDLE_FDK_CRCINFO hCrcInfo, INT crcRegion,
+                                    UINT sbrSyntaxFlags) {
+  USHORT crcReg = SBR_CRCINIT;
+  INT numCrcBits, i;
+
+  /* check if SBR is present */
+  if (hCmonData == NULL) return;
+
+  hCmonData->sbrFillBits = 0; /* Fill bits are written only for GA streams */
+
+  if (sbrSyntaxFlags & SBR_SYNTAX_DRM_CRC) {
+    /*
+     * Calculate and write DRM CRC
+     */
+    FDKcrcEndReg(hCrcInfo, &hCmonData->sbrBitbuf, crcRegion);
+    FDKwriteBits(&hCmonData->tmpWriteBitbuf, FDKcrcGetCRC(hCrcInfo) ^ 0xFF,
+                 SI_SBR_DRM_CRC_BITS);
+  } else {
+    if (!(sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY)) {
+      /* Do alignment here, because its defined as part of the
+       * sbr_extension_data */
+      int sbrLoad = hCmonData->sbrHdrBits + hCmonData->sbrDataBits;
+
+      if (sbrSyntaxFlags & SBR_SYNTAX_CRC) {
+        sbrLoad += SI_SBR_CRC_BITS;
+      }
+
+      sbrLoad += 4; /* Do byte Align with 4 bit offset. ISO/IEC 14496-3:2005(E)
+                       page 39. */
+
+      hCmonData->sbrFillBits = (8 - (sbrLoad % 8)) % 8;
+
+      /*
+        append fill bits
+      */
+      FDKwriteBits(&hCmonData->sbrBitbuf, 0, hCmonData->sbrFillBits);
+
+      FDK_ASSERT(FDKgetValidBits(&hCmonData->sbrBitbuf) % 8 == 4);
+    }
+
+    /*
+      calculate crc
+    */
+    if (sbrSyntaxFlags & SBR_SYNTAX_CRC) {
+      FDK_BITSTREAM tmpCRCBuf = hCmonData->sbrBitbuf;
+      FDKresetBitbuffer(&tmpCRCBuf, BS_READER);
+
+      numCrcBits = hCmonData->sbrHdrBits + hCmonData->sbrDataBits +
+                   hCmonData->sbrFillBits;
+
+      for (i = 0; i < numCrcBits; i++) {
+        INT bit;
+        bit = FDKreadBits(&tmpCRCBuf, 1);
+        crcAdvance(SBR_CRC_POLY, SBR_CRC_MASK, &crcReg, bit, 1);
+      }
+      crcReg &= (SBR_CRC_RANGE);
+
+      /*
+       * Write CRC data.
+       */
+      FDKwriteBits(&hCmonData->tmpWriteBitbuf, crcReg, SI_SBR_CRC_BITS);
+    }
+  }
+
+  FDKsyncCache(&hCmonData->tmpWriteBitbuf);
+}
diff --git a/fdk-aac/libSBRenc/src/env_bit.h b/fdk-aac/libSBRenc/src/env_bit.h
new file mode 100644
index 0000000..b91802c
--- /dev/null
+++ b/fdk-aac/libSBRenc/src/env_bit.h
@@ -0,0 +1,135 @@
+/* -----------------------------------------------------------------------------
+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 encoder library ******************************
+
+   Author(s):
+
+   Description:
+
+*******************************************************************************/
+
+/*!
+  \file
+  \brief  Remaining SBR Bit Writing Routines
+*/
+
+#ifndef ENV_BIT_H
+#define ENV_BIT_H
+
+#include "sbr_encoder.h"
+#include "FDK_crc.h"
+
+/* G(x) = x^10 + x^9 + x^5 + x^4 + x + 1 */
+#define SBR_CRC_POLY (0x0233)
+#define SBR_CRC_MASK (0x0200)
+#define SBR_CRC_RANGE (0x03FF)
+#define SBR_CRC_MAXREGS 1
+#define SBR_CRCINIT (0x0)
+
+#define SI_SBR_CRC_ENABLE_BITS 0
+#define SI_SBR_CRC_BITS 10
+#define SI_SBR_DRM_CRC_BITS 8
+
+struct COMMON_DATA;
+
+INT FDKsbrEnc_InitSbrBitstream(struct COMMON_DATA *hCmonData, UCHAR *memoryBase,
+                               INT memorySize, HANDLE_FDK_CRCINFO hCrcInfo,
+                               UINT sbrSyntaxFlags);
+
+void FDKsbrEnc_AssembleSbrBitstream(struct COMMON_DATA *hCmonData,
+                                    HANDLE_FDK_CRCINFO hCrcInfo, INT crcRegion,
+                                    UINT sbrSyntaxFlags);
+
+#endif /* #ifndef ENV_BIT_H */
diff --git a/fdk-aac/libSBRenc/src/env_est.cpp b/fdk-aac/libSBRenc/src/env_est.cpp
new file mode 100644
index 0000000..4af561b
--- /dev/null
+++ b/fdk-aac/libSBRenc/src/env_est.cpp
@@ -0,0 +1,1986 @@
+/* -----------------------------------------------------------------------------
+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 encoder library ******************************
+
+   Author(s):
+
+   Description:
+
+*******************************************************************************/
+
+#include "env_est.h"
+#include "tran_det.h"
+
+#include "qmf.h"
+
+#include "fram_gen.h"
+#include "bit_sbr.h"
+#include "cmondata.h"
+#include "sbrenc_ram.h"
+
+#include "genericStds.h"
+
+#define QUANT_ERROR_THRES 200
+#define Y_NRG_SCALE 5 /* noCols = 32 -> shift(5) */
+#define MAX_NRG_SLOTS_LD 16
+
+static const UCHAR panTable[2][10] = {{0, 2, 4, 6, 8, 12, 16, 20, 24},
+                                      {0, 2, 4, 8, 12, 0, 0, 0, 0}};
+static const UCHAR maxIndex[2] = {9, 5};
+
+/******************************************************************************
+ Functionname:  FDKsbrEnc_GetTonality
+******************************************************************************/
+/***************************************************************************/
+/*!
+
+  \brief      Calculates complete energy per band from the energy values
+              of the QMF subsamples.
+
+  \brief      quotaMatrix - calculated in FDKsbrEnc_CalculateTonalityQuotas()
+  \brief      noEstPerFrame - number of estimations per frame
+  \brief      startIndex - start index for the quota matrix
+  \brief      Energies - energy matrix
+  \brief      startBand - start band
+  \brief      stopBand - number of QMF bands
+  \brief      numberCols - number of QMF subsamples
+
+  \return     mean tonality of the 5 bands with the highest energy
+              scaled by 2^(RELAXATION_SHIFT+2)*RELAXATION_FRACT
+
+****************************************************************************/
+static FIXP_DBL FDKsbrEnc_GetTonality(const FIXP_DBL *const *quotaMatrix,
+                                      const INT noEstPerFrame,
+                                      const INT startIndex,
+                                      const FIXP_DBL *const *Energies,
+                                      const UCHAR startBand, const INT stopBand,
+                                      const INT numberCols) {
+  UCHAR b, e, k;
+  INT no_enMaxBand[SBR_MAX_ENERGY_VALUES] = {-1, -1, -1, -1, -1};
+  FIXP_DBL energyMax[SBR_MAX_ENERGY_VALUES] = {
+      FL2FXCONST_DBL(0.0f), FL2FXCONST_DBL(0.0f), FL2FXCONST_DBL(0.0f),
+      FL2FXCONST_DBL(0.0f), FL2FXCONST_DBL(0.0f)};
+  FIXP_DBL energyMaxMin = MAXVAL_DBL; /* min. energy in energyMax array */
+  UCHAR posEnergyMaxMin = 0; /* min. energy in energyMax array position */
+  FIXP_DBL tonalityBand[SBR_MAX_ENERGY_VALUES] = {
+      FL2FXCONST_DBL(0.0f), FL2FXCONST_DBL(0.0f), FL2FXCONST_DBL(0.0f),
+      FL2FXCONST_DBL(0.0f), FL2FXCONST_DBL(0.0f)};
+  FIXP_DBL globalTonality = FL2FXCONST_DBL(0.0f);
+  FIXP_DBL energyBand[64];
+  INT maxNEnergyValues; /* max. number of max. energy values */
+
+  /*** Sum up energies for each band ***/
+  FDK_ASSERT(numberCols == 15 || numberCols == 16);
+  /* numberCols is always 15 or 16 for ELD. In case of 16 bands, the
+      energyBands are initialized with the [15]th column.
+      The rest of the column energies are added in the next step.   */
+  if (numberCols == 15) {
+    for (b = startBand; b < stopBand; b++) {
+      energyBand[b] = FL2FXCONST_DBL(0.0f);
+    }
+  } else {
+    for (b = startBand; b < stopBand; b++) {
+      energyBand[b] = Energies[15][b] >> 4;
+    }
+  }
+
+  for (k = 0; k < 15; k++) {
+    for (b = startBand; b < stopBand; b++) {
+      energyBand[b] += Energies[k][b] >> 4;
+    }
+  }
+
+  /*** Determine 5 highest band-energies ***/
+  maxNEnergyValues = fMin(SBR_MAX_ENERGY_VALUES, stopBand - startBand);
+
+  /* Get min. value in energyMax array */
+  energyMaxMin = energyMax[0] = energyBand[startBand];
+  no_enMaxBand[0] = startBand;
+  posEnergyMaxMin = 0;
+  for (k = 1; k < maxNEnergyValues; k++) {
+    energyMax[k] = energyBand[startBand + k];
+    no_enMaxBand[k] = startBand + k;
+    if (energyMaxMin > energyMax[k]) {
+      energyMaxMin = energyMax[k];
+      posEnergyMaxMin = k;
+    }
+  }
+
+  for (b = startBand + maxNEnergyValues; b < stopBand; b++) {
+    if (energyBand[b] > energyMaxMin) {
+      energyMax[posEnergyMaxMin] = energyBand[b];
+      no_enMaxBand[posEnergyMaxMin] = b;
+
+      /* Again, get min. value in energyMax array */
+      energyMaxMin = energyMax[0];
+      posEnergyMaxMin = 0;
+      for (k = 1; k < maxNEnergyValues; k++) {
+        if (energyMaxMin > energyMax[k]) {
+          energyMaxMin = energyMax[k];
+          posEnergyMaxMin = k;
+        }
+      }
+    }
+  }
+  /*** End determine 5 highest band-energies ***/
+
+  /* Get tonality values for 5 highest energies */
+  for (e = 0; e < maxNEnergyValues; e++) {
+    tonalityBand[e] = FL2FXCONST_DBL(0.0f);
+    for (k = 0; k < noEstPerFrame; k++) {
+      tonalityBand[e] += quotaMatrix[startIndex + k][no_enMaxBand[e]] >> 1;
+    }
+    globalTonality +=
+        tonalityBand[e] >> 2; /* headroom of 2+1 (max. 5 additions) */
+  }
+
+  return globalTonality;
+}
+
+/***************************************************************************/
+/*!
+
+  \brief      Calculates energy form real and imaginary part of
+              the QMF subsamples
+
+  \return     none
+
+****************************************************************************/
+LNK_SECTION_CODE_L1
+static void FDKsbrEnc_getEnergyFromCplxQmfData(
+    FIXP_DBL **RESTRICT energyValues, /*!< the result of the operation */
+    FIXP_DBL **RESTRICT realValues, /*!< the real part of the QMF subsamples */
+    FIXP_DBL **RESTRICT
+        imagValues,   /*!< the imaginary part of the QMF subsamples */
+    INT numberBands,  /*!< number of QMF bands */
+    INT numberCols,   /*!< number of QMF subsamples */
+    INT *qmfScale,    /*!< sclefactor of QMF subsamples */
+    INT *energyScale) /*!< scalefactor of energies */
+{
+  int j, k;
+  int scale;
+  FIXP_DBL max_val = FL2FXCONST_DBL(0.0f);
+
+  /* Get Scratch buffer */
+  C_ALLOC_SCRATCH_START(tmpNrg, FIXP_DBL, 32 * 64 / 2)
+
+  /* Get max possible scaling of QMF data */
+  scale = DFRACT_BITS;
+  for (k = 0; k < numberCols; k++) {
+    scale = fixMin(scale, fixMin(getScalefactor(realValues[k], numberBands),
+                                 getScalefactor(imagValues[k], numberBands)));
+  }
+
+  /* Tweak scaling stability for zero signal to non-zero signal transitions */
+  if (scale >= DFRACT_BITS - 1) {
+    scale = (FRACT_BITS - 1 - *qmfScale);
+  }
+  /* prevent scaling of QMF values to -1.f */
+  scale = fixMax(0, scale - 1);
+
+  /* Update QMF scale */
+  *qmfScale += scale;
+
+  /*
+     Calculate energy of each time slot pair, max energy
+     and shift QMF values as far as possible to the left.
+   */
+  {
+    FIXP_DBL *nrgValues = tmpNrg;
+    for (k = 0; k < numberCols; k += 2) {
+      /* Load band vector addresses of 2 consecutive timeslots */
+      FIXP_DBL *RESTRICT r0 = realValues[k];
+      FIXP_DBL *RESTRICT i0 = imagValues[k];
+      FIXP_DBL *RESTRICT r1 = realValues[k + 1];
+      FIXP_DBL *RESTRICT i1 = imagValues[k + 1];
+      for (j = 0; j < numberBands; j++) {
+        FIXP_DBL energy;
+        FIXP_DBL tr0, tr1, ti0, ti1;
+
+        /* Read QMF values of 2 timeslots */
+        tr0 = r0[j];
+        tr1 = r1[j];
+        ti0 = i0[j];
+        ti1 = i1[j];
+
+        /* Scale QMF Values and Calc Energy average of both timeslots */
+        tr0 <<= scale;
+        ti0 <<= scale;
+        energy = fPow2AddDiv2(fPow2Div2(tr0), ti0) >> 1;
+
+        tr1 <<= scale;
+        ti1 <<= scale;
+        energy += fPow2AddDiv2(fPow2Div2(tr1), ti1) >> 1;
+
+        /* Write timeslot pair energy to scratch */
+        *nrgValues++ = energy;
+        max_val = fixMax(max_val, energy);
+
+        /* Write back scaled QMF values */
+        r0[j] = tr0;
+        r1[j] = tr1;
+        i0[j] = ti0;
+        i1[j] = ti1;
+      }
+    }
+  }
+  /* energyScale: scalefactor energies of current frame */
+  *energyScale =
+      2 * (*qmfScale) -
+      1; /* if qmfScale > 0: nr of right shifts otherwise nr of left shifts */
+
+  /* Scale timeslot pair energies and write to output buffer */
+  scale = CountLeadingBits(max_val);
+  {
+    FIXP_DBL *nrgValues = tmpNrg;
+    for (k = 0; k<numberCols>> 1; k++) {
+      scaleValues(energyValues[k], nrgValues, numberBands, scale);
+      nrgValues += numberBands;
+    }
+    *energyScale += scale;
+  }
+
+  /* Free Scratch buffer */
+  C_ALLOC_SCRATCH_END(tmpNrg, FIXP_DBL, 32 * 64 / 2)
+}
+
+LNK_SECTION_CODE_L1
+static void FDKsbrEnc_getEnergyFromCplxQmfDataFull(
+    FIXP_DBL **RESTRICT energyValues, /*!< the result of the operation */
+    FIXP_DBL **RESTRICT realValues, /*!< the real part of the QMF subsamples */
+    FIXP_DBL **RESTRICT
+        imagValues,   /*!< the imaginary part of the QMF subsamples */
+    int numberBands,  /*!< number of QMF bands */
+    int numberCols,   /*!< number of QMF subsamples */
+    int *qmfScale,    /*!< scalefactor of QMF subsamples */
+    int *energyScale) /*!< scalefactor of energies */
+{
+  int j, k;
+  int scale;
+  FIXP_DBL max_val = FL2FXCONST_DBL(0.0f);
+
+  /* Get Scratch buffer */
+  C_ALLOC_SCRATCH_START(tmpNrg, FIXP_DBL, MAX_NRG_SLOTS_LD * 64)
+
+  FDK_ASSERT(numberCols <= MAX_NRG_SLOTS_LD);
+  FDK_ASSERT(numberBands <= 64);
+
+  /* Get max possible scaling of QMF data */
+  scale = DFRACT_BITS;
+  for (k = 0; k < numberCols; k++) {
+    scale = fixMin(scale, fixMin(getScalefactor(realValues[k], numberBands),
+                                 getScalefactor(imagValues[k], numberBands)));
+  }
+
+  /* Tweak scaling stability for zero signal to non-zero signal transitions */
+  if (scale >= DFRACT_BITS - 1) {
+    scale = (FRACT_BITS - 1 - *qmfScale);
+  }
+  /* prevent scaling of QFM values to -1.f */
+  scale = fixMax(0, scale - 1);
+
+  /* Update QMF scale */
+  *qmfScale += scale;
+
+  /*
+     Calculate energy of each time slot pair, max energy
+     and shift QMF values as far as possible to the left.
+   */
+  {
+    FIXP_DBL *nrgValues = tmpNrg;
+    for (k = 0; k < numberCols; k++) {
+      /* Load band vector addresses of 1 timeslot */
+      FIXP_DBL *RESTRICT r0 = realValues[k];
+      FIXP_DBL *RESTRICT i0 = imagValues[k];
+      for (j = 0; j < numberBands; j++) {
+        FIXP_DBL energy;
+        FIXP_DBL tr0, ti0;
+
+        /* Read QMF values of 1 timeslot */
+        tr0 = r0[j];
+        ti0 = i0[j];
+
+        /* Scale QMF Values and Calc Energy */
+        tr0 <<= scale;
+        ti0 <<= scale;
+        energy = fPow2AddDiv2(fPow2Div2(tr0), ti0);
+        *nrgValues++ = energy;
+
+        max_val = fixMax(max_val, energy);
+
+        /* Write back scaled QMF values */
+        r0[j] = tr0;
+        i0[j] = ti0;
+      }
+    }
+  }
+  /* energyScale: scalefactor energies of current frame */
+  *energyScale =
+      2 * (*qmfScale) -
+      1; /* if qmfScale > 0: nr of right shifts otherwise nr of left shifts */
+
+  /* Scale timeslot pair energies and write to output buffer */
+  scale = CountLeadingBits(max_val);
+  {
+    FIXP_DBL *nrgValues = tmpNrg;
+    for (k = 0; k < numberCols; k++) {
+      scaleValues(energyValues[k], nrgValues, numberBands, scale);
+      nrgValues += numberBands;
+    }
+    *energyScale += scale;
+  }
+
+  /* Free Scratch buffer */
+  C_ALLOC_SCRATCH_END(tmpNrg, FIXP_DBL, MAX_NRG_SLOTS_LD * 64)
+}
+
+/***************************************************************************/
+/*!
+
+  \brief  Quantisation of the panorama value (balance)
+
+  \return the quantized pan value
+
+****************************************************************************/
+static INT mapPanorama(INT nrgVal,     /*! integer value of the energy */
+                       INT ampRes,     /*! amplitude resolution [1.5/3dB] */
+                       INT *quantError /*! quantization error of energy val*/
+) {
+  int i;
+  INT min_val, val;
+  UCHAR panIndex;
+  INT sign;
+
+  sign = nrgVal > 0 ? 1 : -1;
+
+  nrgVal *= sign;
+
+  min_val = FDK_INT_MAX;
+  panIndex = 0;
+  for (i = 0; i < maxIndex[ampRes]; i++) {
+    val = fixp_abs((nrgVal - (INT)panTable[ampRes][i]));
+
+    if (val < min_val) {
+      min_val = val;
+      panIndex = i;
+    }
+  }
+
+  *quantError = min_val;
+
+  return panTable[ampRes][maxIndex[ampRes] - 1] +
+         sign * panTable[ampRes][panIndex];
+}
+
+/***************************************************************************/
+/*!
+
+  \brief  Quantisation of the noise floor levels
+
+  \return void
+
+****************************************************************************/
+static void sbrNoiseFloorLevelsQuantisation(
+    SCHAR *RESTRICT iNoiseLevels, /*! quantized noise levels */
+    FIXP_DBL *RESTRICT
+        NoiseLevels, /*! the noise levels. Exponent = LD_DATA_SHIFT  */
+    INT coupling     /*! the coupling flag */
+) {
+  INT i;
+  INT tmp, dummy;
+
+  /* Quantisation, similar to sfb quant... */
+  for (i = 0; i < MAX_NUM_NOISE_VALUES; i++) {
+    /* tmp = NoiseLevels[i] > (PFLOAT)30.0f ? 30: (INT) (NoiseLevels[i] +
+     * (PFLOAT)0.5); */
+    /* 30>>LD_DATA_SHIFT = 0.46875 */
+    if ((FIXP_DBL)NoiseLevels[i] > FL2FXCONST_DBL(0.46875f)) {
+      tmp = 30;
+    } else {
+      /* tmp = (INT)((FIXP_DBL)NoiseLevels[i] + (FL2FXCONST_DBL(0.5f)>>(*/
+      /* FRACT_BITS+ */                                 /* 6-1)));*/
+      /* tmp = tmp >> (DFRACT_BITS-1-LD_DATA_SHIFT); */ /* conversion to integer
+                                                           happens here */
+      /* rounding is done by shifting one bit less than necessary to the right,
+       * adding '1' and then shifting the final bit */
+      tmp = ((((INT)NoiseLevels[i]) >>
+              (DFRACT_BITS - 1 - LD_DATA_SHIFT))); /* conversion to integer */
+      if (tmp != 0) tmp += 1;
+    }
+
+    if (coupling) {
+      tmp = tmp < -30 ? -30 : tmp;
+      tmp = mapPanorama(tmp, 1, &dummy);
+    }
+    iNoiseLevels[i] = tmp;
+  }
+}
+
+/***************************************************************************/
+/*!
+
+  \brief  Calculation of noise floor for coupling
+
+  \return void
+
+****************************************************************************/
+static void coupleNoiseFloor(
+    FIXP_DBL *RESTRICT noise_level_left, /*! noise level left  (modified)*/
+    FIXP_DBL *RESTRICT noise_level_right /*! noise level right (modified)*/
+) {
+  FIXP_DBL cmpValLeft, cmpValRight;
+  INT i;
+  FIXP_DBL temp1, temp2;
+
+  for (i = 0; i < MAX_NUM_NOISE_VALUES; i++) {
+    /* Calculation of the power function using ld64:
+       z  = x^y;
+       z' = CalcLd64(z) = y*CalcLd64(x)/64;
+       z  = CalcInvLd64(z');
+    */
+    cmpValLeft = NOISE_FLOOR_OFFSET_64 - noise_level_left[i];
+    cmpValRight = NOISE_FLOOR_OFFSET_64 - noise_level_right[i];
+
+    if (cmpValRight < FL2FXCONST_DBL(0.0f)) {
+      temp1 = CalcInvLdData(NOISE_FLOOR_OFFSET_64 - noise_level_right[i]);
+    } else {
+      temp1 = CalcInvLdData(NOISE_FLOOR_OFFSET_64 - noise_level_right[i]);
+      temp1 = temp1 << (DFRACT_BITS - 1 - LD_DATA_SHIFT -
+                        1); /* INT to fract conversion of result, if input of
+                               CalcInvLdData is positiv */
+    }
+
+    if (cmpValLeft < FL2FXCONST_DBL(0.0f)) {
+      temp2 = CalcInvLdData(NOISE_FLOOR_OFFSET_64 - noise_level_left[i]);
+    } else {
+      temp2 = CalcInvLdData(NOISE_FLOOR_OFFSET_64 - noise_level_left[i]);
+      temp2 = temp2 << (DFRACT_BITS - 1 - LD_DATA_SHIFT -
+                        1); /* INT to fract conversion of result, if input of
+                               CalcInvLdData is positiv */
+    }
+
+    if ((cmpValLeft < FL2FXCONST_DBL(0.0f)) &&
+        (cmpValRight < FL2FXCONST_DBL(0.0f))) {
+      noise_level_left[i] =
+          NOISE_FLOOR_OFFSET_64 -
+          (CalcLdData(
+              ((temp1 >> 1) +
+               (temp2 >> 1)))); /* no scaling needed! both values are dfract */
+      noise_level_right[i] = CalcLdData(temp2) - CalcLdData(temp1);
+    }
+
+    if ((cmpValLeft >= FL2FXCONST_DBL(0.0f)) &&
+        (cmpValRight >= FL2FXCONST_DBL(0.0f))) {
+      noise_level_left[i] = NOISE_FLOOR_OFFSET_64 -
+                            (CalcLdData(((temp1 >> 1) + (temp2 >> 1))) +
+                             FL2FXCONST_DBL(0.109375f)); /* scaled with 7/64 */
+      noise_level_right[i] = CalcLdData(temp2) - CalcLdData(temp1);
+    }
+
+    if ((cmpValLeft >= FL2FXCONST_DBL(0.0f)) &&
+        (cmpValRight < FL2FXCONST_DBL(0.0f))) {
+      noise_level_left[i] = NOISE_FLOOR_OFFSET_64 -
+                            (CalcLdData(((temp1 >> (7 + 1)) + (temp2 >> 1))) +
+                             FL2FXCONST_DBL(0.109375f)); /* scaled with 7/64 */
+      noise_level_right[i] =
+          (CalcLdData(temp2) + FL2FXCONST_DBL(0.109375f)) - CalcLdData(temp1);
+    }
+
+    if ((cmpValLeft < FL2FXCONST_DBL(0.0f)) &&
+        (cmpValRight >= FL2FXCONST_DBL(0.0f))) {
+      noise_level_left[i] = NOISE_FLOOR_OFFSET_64 -
+                            (CalcLdData(((temp1 >> 1) + (temp2 >> (7 + 1)))) +
+                             FL2FXCONST_DBL(0.109375f)); /* scaled with 7/64 */
+      noise_level_right[i] = CalcLdData(temp2) -
+                             (CalcLdData(temp1) +
+                              FL2FXCONST_DBL(0.109375f)); /* scaled with 7/64 */
+    }
+  }
+}
+
+/***************************************************************************/
+/*!
+
+  \brief  Calculation of energy starting in lower band (li) up to upper band
+(ui) over slots (start_pos) to (stop_pos)
+
+  \return void
+
+****************************************************************************/
+
+static FIXP_DBL getEnvSfbEnergy(
+    INT li,             /*! lower band */
+    INT ui,             /*! upper band */
+    INT start_pos,      /*! start slot */
+    INT stop_pos,       /*! stop slot */
+    INT border_pos,     /*! slots scaling border */
+    FIXP_DBL **YBuffer, /*! sfb energy buffer */
+    INT YBufferSzShift, /*! Energy buffer index scale */
+    INT scaleNrg0,      /*! scaling of lower slots */
+    INT scaleNrg1)      /*! scaling of upper slots */
+{
+  /* use dynamic scaling for outer energy loop;
+     energies are critical and every bit is important */
+  int sc0, sc1, k, l;
+
+  FIXP_DBL nrgSum, nrg1, nrg2, accu1, accu2;
+  INT dynScale, dynScale1, dynScale2;
+  if (ui - li == 0)
+    dynScale = DFRACT_BITS - 1;
+  else
+    dynScale = CalcLdInt(ui - li) >> (DFRACT_BITS - 1 - LD_DATA_SHIFT);
+
+  sc0 = fixMin(scaleNrg0, Y_NRG_SCALE);
+  sc1 = fixMin(scaleNrg1, Y_NRG_SCALE);
+  /* dynScale{1,2} is set such that the right shift below is positive */
+  dynScale1 = fixMin((scaleNrg0 - sc0), dynScale);
+  dynScale2 = fixMin((scaleNrg1 - sc1), dynScale);
+  nrgSum = accu1 = accu2 = (FIXP_DBL)0;
+
+  for (k = li; k < ui; k++) {
+    nrg1 = nrg2 = (FIXP_DBL)0;
+    for (l = start_pos; l < border_pos; l++) {
+      nrg1 += YBuffer[l >> YBufferSzShift][k] >> sc0;
+    }
+    for (; l < stop_pos; l++) {
+      nrg2 += YBuffer[l >> YBufferSzShift][k] >> sc1;
+    }
+    accu1 += (nrg1 >> dynScale1);
+    accu2 += (nrg2 >> dynScale2);
+  }
+  /* This shift factor is always positive. See comment above. */
+  nrgSum +=
+      (accu1 >> fixMin((scaleNrg0 - sc0 - dynScale1), (DFRACT_BITS - 1))) +
+      (accu2 >> fixMin((scaleNrg1 - sc1 - dynScale2), (DFRACT_BITS - 1)));
+
+  return nrgSum;
+}
+
+/***************************************************************************/
+/*!
+
+  \brief  Energy compensation in missing harmonic mode
+
+  \return void
+
+****************************************************************************/
+static FIXP_DBL mhLoweringEnergy(FIXP_DBL nrg, INT M) {
+  /*
+     Compensating for the fact that we in the decoder map the "average energy to
+     every QMF band, and use this when we calculate the boost-factor. Since the
+     mapped energy isn't the average energy but the maximum energy in case of
+     missing harmonic creation, we will in the boost function calculate that too
+     much limiting has been applied and hence we will boost the signal although
+     it isn't called for. Hence we need to compensate for this by lowering the
+     transmitted energy values for the sines so they will get the correct level
+     after the boost is applied.
+  */
+  if (M > 2) {
+    INT tmpScale;
+    tmpScale = CountLeadingBits(nrg);
+    nrg <<= tmpScale;
+    nrg = fMult(nrg, FL2FXCONST_DBL(0.398107267f)); /* The maximum boost
+                                                       is 1.584893, so the
+                                                       maximum attenuation
+                                                       should be
+                                                       square(1/1.584893) =
+                                                       0.398107267 */
+    nrg >>= tmpScale;
+  } else {
+    if (M > 1) {
+      nrg >>= 1;
+    }
+  }
+
+  return nrg;
+}
+
+/***************************************************************************/
+/*!
+
+  \brief  Energy compensation in none missing harmonic mode
+
+  \return void
+
+****************************************************************************/
+static FIXP_DBL nmhLoweringEnergy(FIXP_DBL nrg, const FIXP_DBL nrgSum,
+                                  const INT nrgSum_scale, const INT M) {
+  if (nrg > FL2FXCONST_DBL(0)) {
+    int sc = 0;
+    /* gain = nrgSum / (nrg*(M+1)) */
+    FIXP_DBL gain = fMult(fDivNorm(nrgSum, nrg, &sc), GetInvInt(M + 1));
+    sc += nrgSum_scale;
+
+    /* reduce nrg if gain smaller 1.f */
+    if (!((sc >= 0) && (gain > ((FIXP_DBL)MAXVAL_DBL >> sc)))) {
+      nrg = fMult(scaleValue(gain, sc), nrg);
+    }
+  }
+  return nrg;
+}
+
+/***************************************************************************/
+/*!
+
+  \brief  calculates the envelope values from the energies, depending on
+          framing and stereo mode
+
+  \return void
+
+****************************************************************************/
+static void calculateSbrEnvelope(
+    FIXP_DBL **RESTRICT YBufferLeft,  /*! energy buffer left */
+    FIXP_DBL **RESTRICT YBufferRight, /*! energy buffer right */
+    int *RESTRICT YBufferScaleLeft,   /*! scale energy buffer left */
+    int *RESTRICT YBufferScaleRight,  /*! scale energy buffer right */
+    const SBR_FRAME_INFO *frame_info, /*! frame info vector */
+    SCHAR *RESTRICT sfb_nrgLeft,      /*! sfb energy buffer left */
+    SCHAR *RESTRICT sfb_nrgRight,     /*! sfb energy buffer right */
+    HANDLE_SBR_CONFIG_DATA h_con,     /*! handle to config data   */
+    HANDLE_ENV_CHANNEL h_sbr,         /*! envelope channel handle */
+    SBR_STEREO_MODE stereoMode,       /*! stereo coding mode */
+    INT *maxQuantError, /*! maximum quantization error, for panorama. */
+    int YBufferSzShift) /*! Energy buffer index scale */
+
+{
+  int env, j, m = 0;
+  INT no_of_bands, start_pos, stop_pos, li, ui;
+  FREQ_RES freq_res;
+
+  INT ca = 2 - h_sbr->encEnvData.init_sbr_amp_res;
+  INT oneBitLess = 0;
+  if (ca == 2)
+    oneBitLess =
+        1; /* LD_DATA_SHIFT => ld64 scaling; one bit less for rounding */
+
+  INT quantError;
+  INT nEnvelopes = frame_info->nEnvelopes;
+  INT short_env = frame_info->shortEnv - 1;
+  INT timeStep = h_sbr->sbrExtractEnvelope.time_step;
+  INT commonScale, scaleLeft0, scaleLeft1;
+  INT scaleRight0 = 0, scaleRight1 = 0;
+
+  commonScale = fixMin(YBufferScaleLeft[0], YBufferScaleLeft[1]);
+
+  if (stereoMode == SBR_COUPLING) {
+    commonScale = fixMin(commonScale, YBufferScaleRight[0]);
+    commonScale = fixMin(commonScale, YBufferScaleRight[1]);
+  }
+
+  commonScale = commonScale - 7;
+
+  scaleLeft0 = YBufferScaleLeft[0] - commonScale;
+  scaleLeft1 = YBufferScaleLeft[1] - commonScale;
+  FDK_ASSERT((scaleLeft0 >= 0) && (scaleLeft1 >= 0));
+
+  if (stereoMode == SBR_COUPLING) {
+    scaleRight0 = YBufferScaleRight[0] - commonScale;
+    scaleRight1 = YBufferScaleRight[1] - commonScale;
+    FDK_ASSERT((scaleRight0 >= 0) && (scaleRight1 >= 0));
+    *maxQuantError = 0;
+  }
+
+  for (env = 0; env < nEnvelopes; env++) {
+    FIXP_DBL pNrgLeft[32];
+    FIXP_DBL pNrgRight[32];
+    int envNrg_scale;
+    FIXP_DBL envNrgLeft = FL2FXCONST_DBL(0.0f);
+    FIXP_DBL envNrgRight = FL2FXCONST_DBL(0.0f);
+    int missingHarmonic[32];
+    int count[32];
+
+    start_pos = timeStep * frame_info->borders[env];
+    stop_pos = timeStep * frame_info->borders[env + 1];
+    freq_res = frame_info->freqRes[env];
+    no_of_bands = h_con->nSfb[freq_res];
+    envNrg_scale = DFRACT_BITS - fNormz((FIXP_DBL)no_of_bands);
+    if (env == short_env) {
+      j = fMax(2, timeStep); /* consider at least 2 QMF slots less for short
+                                envelopes (envelopes just before transients) */
+      if ((stop_pos - start_pos - j) > 0) {
+        stop_pos = stop_pos - j;
+      }
+    }
+    for (j = 0; j < no_of_bands; j++) {
+      FIXP_DBL nrgLeft = FL2FXCONST_DBL(0.0f);
+      FIXP_DBL nrgRight = FL2FXCONST_DBL(0.0f);
+
+      li = h_con->freqBandTable[freq_res][j];
+      ui = h_con->freqBandTable[freq_res][j + 1];
+
+      if (freq_res == FREQ_RES_HIGH) {
+        if (j == 0 && ui - li > 1) {
+          li++;
+        }
+      } else {
+        if (j == 0 && ui - li > 2) {
+          li++;
+        }
+      }
+
+      /*
+        Find out whether a sine will be missing in the scale-factor
+        band that we're currently processing.
+      */
+      missingHarmonic[j] = 0;
+
+      if (h_sbr->encEnvData.addHarmonicFlag) {
+        if (freq_res == FREQ_RES_HIGH) {
+          if (h_sbr->encEnvData
+                  .addHarmonic[j]) { /*A missing sine in the current band*/
+            missingHarmonic[j] = 1;
+          }
+        } else {
+          INT i;
+          INT startBandHigh = 0;
+          INT stopBandHigh = 0;
+
+          while (h_con->freqBandTable[FREQ_RES_HIGH][startBandHigh] <
+                 h_con->freqBandTable[FREQ_RES_LOW][j])
+            startBandHigh++;
+          while (h_con->freqBandTable[FREQ_RES_HIGH][stopBandHigh] <
+                 h_con->freqBandTable[FREQ_RES_LOW][j + 1])
+            stopBandHigh++;
+
+          for (i = startBandHigh; i < stopBandHigh; i++) {
+            if (h_sbr->encEnvData.addHarmonic[i]) {
+              missingHarmonic[j] = 1;
+            }
+          }
+        }
+      }
+
+      /*
+        If a sine is missing in a scalefactorband, with more than one qmf
+        channel use the nrg from the channel with the largest nrg rather than
+        the mean. Compensate for the boost calculation in the decdoder.
+      */
+      int border_pos =
+          fixMin(stop_pos, h_sbr->sbrExtractEnvelope.YBufferWriteOffset
+                               << YBufferSzShift);
+
+      if (missingHarmonic[j]) {
+        int k;
+        count[j] = stop_pos - start_pos;
+        nrgLeft = FL2FXCONST_DBL(0.0f);
+
+        for (k = li; k < ui; k++) {
+          FIXP_DBL tmpNrg;
+          tmpNrg = getEnvSfbEnergy(k, k + 1, start_pos, stop_pos, border_pos,
+                                   YBufferLeft, YBufferSzShift, scaleLeft0,
+                                   scaleLeft1);
+
+          nrgLeft = fixMax(nrgLeft, tmpNrg);
+        }
+
+        /* Energy lowering compensation */
+        nrgLeft = mhLoweringEnergy(nrgLeft, ui - li);
+
+        if (stereoMode == SBR_COUPLING) {
+          nrgRight = FL2FXCONST_DBL(0.0f);
+
+          for (k = li; k < ui; k++) {
+            FIXP_DBL tmpNrg;
+            tmpNrg = getEnvSfbEnergy(k, k + 1, start_pos, stop_pos, border_pos,
+                                     YBufferRight, YBufferSzShift, scaleRight0,
+                                     scaleRight1);
+
+            nrgRight = fixMax(nrgRight, tmpNrg);
+          }
+
+          /* Energy lowering compensation */
+          nrgRight = mhLoweringEnergy(nrgRight, ui - li);
+        }
+      } /* end missingHarmonic */
+      else {
+        count[j] = (stop_pos - start_pos) * (ui - li);
+
+        nrgLeft = getEnvSfbEnergy(li, ui, start_pos, stop_pos, border_pos,
+                                  YBufferLeft, YBufferSzShift, scaleLeft0,
+                                  scaleLeft1);
+
+        if (stereoMode == SBR_COUPLING) {
+          nrgRight = getEnvSfbEnergy(li, ui, start_pos, stop_pos, border_pos,
+                                     YBufferRight, YBufferSzShift, scaleRight0,
+                                     scaleRight1);
+        }
+      } /* !missingHarmonic */
+
+      /* save energies */
+      pNrgLeft[j] = nrgLeft;
+      pNrgRight[j] = nrgRight;
+      envNrgLeft += (nrgLeft >> envNrg_scale);
+      envNrgRight += (nrgRight >> envNrg_scale);
+    } /* j */
+
+    for (j = 0; j < no_of_bands; j++) {
+      FIXP_DBL nrgLeft2 = FL2FXCONST_DBL(0.0f);
+      FIXP_DBL nrgLeft = pNrgLeft[j];
+      FIXP_DBL nrgRight = pNrgRight[j];
+
+      /* None missing harmonic Energy lowering compensation */
+      if (!missingHarmonic[j] && h_sbr->fLevelProtect) {
+        /* in case of missing energy in base band,
+           reduce reference energy to prevent overflows in decoder output */
+        nrgLeft =
+            nmhLoweringEnergy(nrgLeft, envNrgLeft, envNrg_scale, no_of_bands);
+        if (stereoMode == SBR_COUPLING) {
+          nrgRight = nmhLoweringEnergy(nrgRight, envNrgRight, envNrg_scale,
+                                       no_of_bands);
+        }
+      }
+
+      if (stereoMode == SBR_COUPLING) {
+        /* calc operation later with log */
+        nrgLeft2 = nrgLeft;
+        nrgLeft = (nrgRight + nrgLeft) >> 1;
+      }
+
+      /* nrgLeft = f20_log2(nrgLeft / (PFLOAT)(count * 64))+(PFLOAT)44; */
+      /* If nrgLeft == 0 then the Log calculations below do fail. */
+      if (nrgLeft > FL2FXCONST_DBL(0.0f)) {
+        FIXP_DBL tmp0, tmp1, tmp2, tmp3;
+        INT tmpScale;
+
+        tmpScale = CountLeadingBits(nrgLeft);
+        nrgLeft = nrgLeft << tmpScale;
+
+        tmp0 = CalcLdData(nrgLeft); /* scaled by 1/64 */
+        tmp1 = ((FIXP_DBL)(commonScale + tmpScale))
+               << (DFRACT_BITS - 1 - LD_DATA_SHIFT - 1); /* scaled by 1/64 */
+        tmp2 = ((FIXP_DBL)(count[j] * 64)) << (DFRACT_BITS - 1 - 14 - 1);
+        tmp2 = CalcLdData(tmp2); /* scaled by 1/64 */
+        tmp3 = FL2FXCONST_DBL(0.6875f - 0.21875f - 0.015625f) >>
+               1; /* scaled by 1/64 */
+
+        nrgLeft = ((tmp0 - tmp2) >> 1) + (tmp3 - tmp1);
+      } else {
+        nrgLeft = FL2FXCONST_DBL(-1.0f);
+      }
+
+      /* ld64 to integer conversion */
+      nrgLeft = fixMin(fixMax(nrgLeft, FL2FXCONST_DBL(0.0f)),
+                       (FL2FXCONST_DBL(0.5f) >> oneBitLess));
+      nrgLeft = (FIXP_DBL)(LONG)nrgLeft >>
+                (DFRACT_BITS - 1 - LD_DATA_SHIFT - 1 - oneBitLess - 1);
+      sfb_nrgLeft[m] = ((INT)nrgLeft + 1) >> 1; /* rounding */
+
+      if (stereoMode == SBR_COUPLING) {
+        FIXP_DBL scaleFract;
+        int sc0, sc1;
+
+        nrgLeft2 = fixMax((FIXP_DBL)0x1, nrgLeft2);
+        nrgRight = fixMax((FIXP_DBL)0x1, nrgRight);
+
+        sc0 = CountLeadingBits(nrgLeft2);
+        sc1 = CountLeadingBits(nrgRight);
+
+        scaleFract =
+            ((FIXP_DBL)(sc0 - sc1))
+            << (DFRACT_BITS - 1 -
+                LD_DATA_SHIFT); /* scale value in ld64 representation */
+        nrgRight = CalcLdData(nrgLeft2 << sc0) - CalcLdData(nrgRight << sc1) -
+                   scaleFract;
+
+        /* ld64 to integer conversion */
+        nrgRight = (FIXP_DBL)(LONG)(nrgRight) >>
+                   (DFRACT_BITS - 1 - LD_DATA_SHIFT - 1 - oneBitLess);
+        nrgRight = (nrgRight + (FIXP_DBL)1) >> 1; /* rounding */
+
+        sfb_nrgRight[m] = mapPanorama(
+            nrgRight, h_sbr->encEnvData.init_sbr_amp_res, &quantError);
+
+        *maxQuantError = fixMax(quantError, *maxQuantError);
+      }
+
+      m++;
+    } /* j */
+
+    /* Do energy compensation for sines that are present in two
+        QMF-bands in the original, but will only occur in one band in
+        the decoder due to the synthetic sine coding.*/
+    if (h_con->useParametricCoding) {
+      m -= no_of_bands;
+      for (j = 0; j < no_of_bands; j++) {
+        if (freq_res == FREQ_RES_HIGH &&
+            h_sbr->sbrExtractEnvelope.envelopeCompensation[j]) {
+          sfb_nrgLeft[m] -=
+              (ca *
+               fixp_abs(
+                   (INT)h_sbr->sbrExtractEnvelope.envelopeCompensation[j]));
+        }
+        sfb_nrgLeft[m] = fixMax(0, sfb_nrgLeft[m]);
+        m++;
+      }
+    } /* useParametricCoding */
+
+  } /* env loop */
+}
+
+/***************************************************************************/
+/*!
+
+  \brief  calculates the noise floor and the envelope values from the
+          energies, depending on framing and stereo mode
+
+  FDKsbrEnc_extractSbrEnvelope is the main function for encoding and writing the
+  envelope and the noise floor. The function includes the following processes:
+
+  -Analysis subband filtering.
+  -Encoding SA and pan parameters (if enabled).
+  -Transient detection.
+
+****************************************************************************/
+
+LNK_SECTION_CODE_L1
+void FDKsbrEnc_extractSbrEnvelope1(
+    HANDLE_SBR_CONFIG_DATA h_con, /*! handle to config data   */
+    HANDLE_SBR_HEADER_DATA sbrHeaderData,
+    HANDLE_SBR_BITSTREAM_DATA sbrBitstreamData, HANDLE_ENV_CHANNEL hEnvChan,
+    HANDLE_COMMON_DATA hCmonData, SBR_ENV_TEMP_DATA *eData,
+    SBR_FRAME_TEMP_DATA *fData) {
+  HANDLE_SBR_EXTRACT_ENVELOPE sbrExtrEnv = &hEnvChan->sbrExtractEnvelope;
+
+  if (sbrExtrEnv->YBufferSzShift == 0)
+    FDKsbrEnc_getEnergyFromCplxQmfDataFull(
+        &sbrExtrEnv->YBuffer[sbrExtrEnv->YBufferWriteOffset],
+        sbrExtrEnv->rBuffer + sbrExtrEnv->rBufferReadOffset,
+        sbrExtrEnv->iBuffer + sbrExtrEnv->rBufferReadOffset, h_con->noQmfBands,
+        sbrExtrEnv->no_cols, &hEnvChan->qmfScale, &sbrExtrEnv->YBufferScale[1]);
+  else
+    FDKsbrEnc_getEnergyFromCplxQmfData(
+        &sbrExtrEnv->YBuffer[sbrExtrEnv->YBufferWriteOffset],
+        sbrExtrEnv->rBuffer + sbrExtrEnv->rBufferReadOffset,
+        sbrExtrEnv->iBuffer + sbrExtrEnv->rBufferReadOffset, h_con->noQmfBands,
+        sbrExtrEnv->no_cols, &hEnvChan->qmfScale, &sbrExtrEnv->YBufferScale[1]);
+
+  /* Energie values =
+   * sbrExtrEnv->YBuffer[sbrExtrEnv->YBufferWriteOffset][x].floatVal *
+   * (1<<2*7-sbrExtrEnv->YBufferScale[1]) */
+
+  /*
+    Precalculation of Tonality Quotas  COEFF Transform OK
+  */
+  FDKsbrEnc_CalculateTonalityQuotas(
+      &hEnvChan->TonCorr, sbrExtrEnv->rBuffer, sbrExtrEnv->iBuffer,
+      h_con->freqBandTable[HI][h_con->nSfb[HI]], hEnvChan->qmfScale);
+
+  if (h_con->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) {
+    FIXP_DBL tonality = FDKsbrEnc_GetTonality(
+        hEnvChan->TonCorr.quotaMatrix,
+        hEnvChan->TonCorr.numberOfEstimatesPerFrame,
+        hEnvChan->TonCorr.startIndexMatrix,
+        sbrExtrEnv->YBuffer + sbrExtrEnv->YBufferWriteOffset,
+        h_con->freqBandTable[HI][0] + 1, h_con->noQmfBands,
+        sbrExtrEnv->no_cols);
+
+    hEnvChan->encEnvData.ton_HF[1] = hEnvChan->encEnvData.ton_HF[0];
+    hEnvChan->encEnvData.ton_HF[0] = tonality;
+
+    /* tonality is scaled by 2^19/0.524288f (fract part of RELAXATION) */
+    hEnvChan->encEnvData.global_tonality =
+        (hEnvChan->encEnvData.ton_HF[0] >> 1) +
+        (hEnvChan->encEnvData.ton_HF[1] >> 1);
+  }
+
+  /*
+    Transient detection COEFF Transform OK
+  */
+
+  if (h_con->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) {
+    FDKsbrEnc_fastTransientDetect(&hEnvChan->sbrFastTransientDetector,
+                                  sbrExtrEnv->YBuffer, sbrExtrEnv->YBufferScale,
+                                  sbrExtrEnv->YBufferWriteOffset,
+                                  eData->transient_info);
+
+  } else {
+    FDKsbrEnc_transientDetect(
+        &hEnvChan->sbrTransientDetector, sbrExtrEnv->YBuffer,
+        sbrExtrEnv->YBufferScale, eData->transient_info,
+        sbrExtrEnv->YBufferWriteOffset, sbrExtrEnv->YBufferSzShift,
+        sbrExtrEnv->time_step, hEnvChan->SbrEnvFrame.frameMiddleSlot);
+  }
+
+  /*
+    Generate flags for 2 env in a FIXFIX-frame.
+    Remove this function to get always 1 env per FIXFIX-frame.
+  */
+
+  /*
+    frame Splitter COEFF Transform OK
+  */
+  FDKsbrEnc_frameSplitter(
+      sbrExtrEnv->YBuffer, sbrExtrEnv->YBufferScale,
+      &hEnvChan->sbrTransientDetector, h_con->freqBandTable[1],
+      eData->transient_info, sbrExtrEnv->YBufferWriteOffset,
+      sbrExtrEnv->YBufferSzShift, h_con->nSfb[1], sbrExtrEnv->time_step,
+      sbrExtrEnv->no_cols, &hEnvChan->encEnvData.global_tonality);
+}
+
+/***************************************************************************/
+/*!
+
+  \brief  calculates the noise floor and the envelope values from the
+          energies, depending on framing and stereo mode
+
+  FDKsbrEnc_extractSbrEnvelope is the main function for encoding and writing the
+  envelope and the noise floor. The function includes the following processes:
+
+  -Determine time/frequency division of current granule.
+  -Sending transient info to bitstream.
+  -Set amp_res to 1.5 dB if the current frame contains only one envelope.
+  -Lock dynamic bandwidth frequency change if the next envelope not starts on a
+  frame boundary.
+  -MDCT transposer (needed to detect where harmonics will be missing).
+  -Spectrum Estimation (used for pulse train and missing harmonics detection).
+  -Pulse train detection.
+  -Inverse Filtering detection.
+  -Waveform Coding.
+  -Missing Harmonics detection.
+  -Extract envelope of current frame.
+  -Noise floor estimation.
+  -Noise floor quantisation and coding.
+  -Encode envelope of current frame.
+  -Send the encoded data to the bitstream.
+  -Write to bitstream.
+
+****************************************************************************/
+
+LNK_SECTION_CODE_L1
+void FDKsbrEnc_extractSbrEnvelope2(
+    HANDLE_SBR_CONFIG_DATA h_con, /*! handle to config data   */
+    HANDLE_SBR_HEADER_DATA sbrHeaderData,
+    HANDLE_PARAMETRIC_STEREO hParametricStereo,
+    HANDLE_SBR_BITSTREAM_DATA sbrBitstreamData, HANDLE_ENV_CHANNEL h_envChan0,
+    HANDLE_ENV_CHANNEL h_envChan1, HANDLE_COMMON_DATA hCmonData,
+    SBR_ENV_TEMP_DATA *eData, SBR_FRAME_TEMP_DATA *fData, int clearOutput) {
+  HANDLE_ENV_CHANNEL h_envChan[MAX_NUM_CHANNELS] = {h_envChan0, h_envChan1};
+  int ch, i, j, c, YSzShift = h_envChan[0]->sbrExtractEnvelope.YBufferSzShift;
+
+  SBR_STEREO_MODE stereoMode = h_con->stereoMode;
+  int nChannels = h_con->nChannels;
+  FDK_ASSERT(nChannels <= MAX_NUM_CHANNELS);
+  const int *v_tuning;
+  static const int v_tuningHEAAC[6] = {0, 2, 4, 0, 0, 0};
+
+  static const int v_tuningELD[6] = {0, 2, 3, 0, 0, 0};
+
+  if (h_con->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY)
+    v_tuning = v_tuningELD;
+  else
+    v_tuning = v_tuningHEAAC;
+
+  /*
+    Select stereo mode.
+  */
+  if (stereoMode == SBR_COUPLING) {
+    if (eData[0].transient_info[1] && eData[1].transient_info[1]) {
+      eData[0].transient_info[0] =
+          fixMin(eData[1].transient_info[0], eData[0].transient_info[0]);
+      eData[1].transient_info[0] = eData[0].transient_info[0];
+    } else {
+      if (eData[0].transient_info[1] && !eData[1].transient_info[1]) {
+        eData[1].transient_info[0] = eData[0].transient_info[0];
+      } else {
+        if (!eData[0].transient_info[1] && eData[1].transient_info[1])
+          eData[0].transient_info[0] = eData[1].transient_info[0];
+        else {
+          eData[0].transient_info[0] =
+              fixMax(eData[1].transient_info[0], eData[0].transient_info[0]);
+          eData[1].transient_info[0] = eData[0].transient_info[0];
+        }
+      }
+    }
+  }
+
+  /*
+    Determine time/frequency division of current granule
+  */
+  eData[0].frame_info = FDKsbrEnc_frameInfoGenerator(
+      &h_envChan[0]->SbrEnvFrame, eData[0].transient_info,
+      sbrBitstreamData->rightBorderFIX,
+      h_envChan[0]->sbrExtractEnvelope.pre_transient_info,
+      h_envChan[0]->encEnvData.ldGrid, v_tuning);
+
+  h_envChan[0]->encEnvData.hSbrBSGrid = &h_envChan[0]->SbrEnvFrame.SbrGrid;
+
+  /* AAC LD patch for transient prediction */
+  if (h_envChan[0]->encEnvData.ldGrid && eData[0].transient_info[2]) {
+    /* if next frame will start with transient, set shortEnv to
+     * numEnvelopes(shortend Envelope = shortEnv-1)*/
+    h_envChan[0]->SbrEnvFrame.SbrFrameInfo.shortEnv =
+        h_envChan[0]->SbrEnvFrame.SbrFrameInfo.nEnvelopes;
+  }
+
+  switch (stereoMode) {
+    case SBR_LEFT_RIGHT:
+    case SBR_SWITCH_LRC:
+      eData[1].frame_info = FDKsbrEnc_frameInfoGenerator(
+          &h_envChan[1]->SbrEnvFrame, eData[1].transient_info,
+          sbrBitstreamData->rightBorderFIX,
+          h_envChan[1]->sbrExtractEnvelope.pre_transient_info,
+          h_envChan[1]->encEnvData.ldGrid, v_tuning);
+
+      h_envChan[1]->encEnvData.hSbrBSGrid = &h_envChan[1]->SbrEnvFrame.SbrGrid;
+
+      if (h_envChan[1]->encEnvData.ldGrid && eData[1].transient_info[2]) {
+        /* if next frame will start with transient, set shortEnv to
+         * numEnvelopes(shortend Envelope = shortEnv-1)*/
+        h_envChan[1]->SbrEnvFrame.SbrFrameInfo.shortEnv =
+            h_envChan[1]->SbrEnvFrame.SbrFrameInfo.nEnvelopes;
+      }
+
+      /* compare left and right frame_infos */
+      if (eData[0].frame_info->nEnvelopes != eData[1].frame_info->nEnvelopes) {
+        stereoMode = SBR_LEFT_RIGHT;
+      } else {
+        for (i = 0; i < eData[0].frame_info->nEnvelopes + 1; i++) {
+          if (eData[0].frame_info->borders[i] !=
+              eData[1].frame_info->borders[i]) {
+            stereoMode = SBR_LEFT_RIGHT;
+            break;
+          }
+        }
+        for (i = 0; i < eData[0].frame_info->nEnvelopes; i++) {
+          if (eData[0].frame_info->freqRes[i] !=
+              eData[1].frame_info->freqRes[i]) {
+            stereoMode = SBR_LEFT_RIGHT;
+            break;
+          }
+        }
+        if (eData[0].frame_info->shortEnv != eData[1].frame_info->shortEnv) {
+          stereoMode = SBR_LEFT_RIGHT;
+        }
+      }
+      break;
+    case SBR_COUPLING:
+      eData[1].frame_info = eData[0].frame_info;
+      h_envChan[1]->encEnvData.hSbrBSGrid = &h_envChan[0]->SbrEnvFrame.SbrGrid;
+      break;
+    case SBR_MONO:
+      /* nothing to do */
+      break;
+    default:
+      FDK_ASSERT(0);
+  }
+
+  for (ch = 0; ch < nChannels; ch++) {
+    HANDLE_ENV_CHANNEL hEnvChan = h_envChan[ch];
+    HANDLE_SBR_EXTRACT_ENVELOPE sbrExtrEnv = &hEnvChan->sbrExtractEnvelope;
+    SBR_ENV_TEMP_DATA *ed = &eData[ch];
+
+    /*
+       Send transient info to bitstream and store for next call
+    */
+    sbrExtrEnv->pre_transient_info[0] = ed->transient_info[0]; /* tran_pos */
+    sbrExtrEnv->pre_transient_info[1] = ed->transient_info[1]; /* tran_flag */
+    hEnvChan->encEnvData.noOfEnvelopes = ed->nEnvelopes =
+        ed->frame_info->nEnvelopes; /* number of envelopes of current frame */
+
+    /*
+      Check if the current frame is divided into one envelope only. If so, set
+      the amplitude resolution to 1.5 dB, otherwise may set back to chosen value
+    */
+    if ((hEnvChan->encEnvData.hSbrBSGrid->frameClass == FIXFIX) &&
+        (ed->nEnvelopes == 1)) {
+      if (h_con->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) {
+        /* Note: global_tonaliy_float_value ==
+           ((float)hEnvChan->encEnvData.global_tonality/((INT64)(1)<<(31-(19+2)))/0.524288*(2.0/3.0)));
+                 threshold_float_value ==
+           ((float)h_con->thresholdAmpResFF_m/((INT64)(1)<<(31-(h_con->thresholdAmpResFF_e)))/0.524288*(2.0/3.0)));
+         */
+        /* decision of SBR_AMP_RES */
+        if (fIsLessThan(/* global_tonality > threshold ? */
+                        h_con->thresholdAmpResFF_m, h_con->thresholdAmpResFF_e,
+                        hEnvChan->encEnvData.global_tonality,
+                        RELAXATION_SHIFT + 2)) {
+          hEnvChan->encEnvData.currentAmpResFF = SBR_AMP_RES_1_5;
+        } else {
+          hEnvChan->encEnvData.currentAmpResFF = SBR_AMP_RES_3_0;
+        }
+      } else
+        hEnvChan->encEnvData.currentAmpResFF = SBR_AMP_RES_1_5;
+
+      if (hEnvChan->encEnvData.currentAmpResFF !=
+          hEnvChan->encEnvData.init_sbr_amp_res) {
+        FDKsbrEnc_InitSbrHuffmanTables(
+            &hEnvChan->encEnvData, &hEnvChan->sbrCodeEnvelope,
+            &hEnvChan->sbrCodeNoiseFloor, hEnvChan->encEnvData.currentAmpResFF);
+      }
+    } else {
+      if (sbrHeaderData->sbr_amp_res != hEnvChan->encEnvData.init_sbr_amp_res) {
+        FDKsbrEnc_InitSbrHuffmanTables(
+            &hEnvChan->encEnvData, &hEnvChan->sbrCodeEnvelope,
+            &hEnvChan->sbrCodeNoiseFloor, sbrHeaderData->sbr_amp_res);
+      }
+    }
+
+    if (!clearOutput) {
+      /*
+        Tonality correction parameter extraction (inverse filtering level, noise
+        floor additional sines).
+      */
+      FDKsbrEnc_TonCorrParamExtr(
+          &hEnvChan->TonCorr, hEnvChan->encEnvData.sbr_invf_mode_vec,
+          ed->noiseFloor, &hEnvChan->encEnvData.addHarmonicFlag,
+          hEnvChan->encEnvData.addHarmonic, sbrExtrEnv->envelopeCompensation,
+          ed->frame_info, ed->transient_info, h_con->freqBandTable[HI],
+          h_con->nSfb[HI], hEnvChan->encEnvData.sbr_xpos_mode,
+          h_con->sbrSyntaxFlags);
+    }
+
+    /* Low energy in low band fix */
+    if (hEnvChan->sbrTransientDetector.prevLowBandEnergy <
+            hEnvChan->sbrTransientDetector.prevHighBandEnergy &&
+        hEnvChan->sbrTransientDetector.prevHighBandEnergy > FL2FX_DBL(0.03)
+        /* The fix needs the non-fast transient detector running.
+           It sets prevLowBandEnergy and prevHighBandEnergy.      */
+        && !(h_con->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY)) {
+      hEnvChan->fLevelProtect = 1;
+
+      for (i = 0; i < MAX_NUM_NOISE_VALUES; i++)
+        hEnvChan->encEnvData.sbr_invf_mode_vec[i] = INVF_HIGH_LEVEL;
+    } else {
+      hEnvChan->fLevelProtect = 0;
+    }
+
+    hEnvChan->encEnvData.sbr_invf_mode =
+        hEnvChan->encEnvData.sbr_invf_mode_vec[0];
+
+    hEnvChan->encEnvData.noOfnoisebands =
+        hEnvChan->TonCorr.sbrNoiseFloorEstimate.noNoiseBands;
+
+  } /* ch */
+
+  /*
+     Save number of scf bands per envelope
+   */
+  for (ch = 0; ch < nChannels; ch++) {
+    for (i = 0; i < eData[ch].nEnvelopes; i++) {
+      h_envChan[ch]->encEnvData.noScfBands[i] =
+          (eData[ch].frame_info->freqRes[i] == FREQ_RES_HIGH
+               ? h_con->nSfb[FREQ_RES_HIGH]
+               : h_con->nSfb[FREQ_RES_LOW]);
+    }
+  }
+
+  /*
+    Extract envelope of current frame.
+  */
+  switch (stereoMode) {
+    case SBR_MONO:
+      calculateSbrEnvelope(h_envChan[0]->sbrExtractEnvelope.YBuffer, NULL,
+                           h_envChan[0]->sbrExtractEnvelope.YBufferScale, NULL,
+                           eData[0].frame_info, eData[0].sfb_nrg, NULL, h_con,
+                           h_envChan[0], SBR_MONO, NULL, YSzShift);
+      break;
+    case SBR_LEFT_RIGHT:
+      calculateSbrEnvelope(h_envChan[0]->sbrExtractEnvelope.YBuffer, NULL,
+                           h_envChan[0]->sbrExtractEnvelope.YBufferScale, NULL,
+                           eData[0].frame_info, eData[0].sfb_nrg, NULL, h_con,
+                           h_envChan[0], SBR_MONO, NULL, YSzShift);
+      calculateSbrEnvelope(h_envChan[1]->sbrExtractEnvelope.YBuffer, NULL,
+                           h_envChan[1]->sbrExtractEnvelope.YBufferScale, NULL,
+                           eData[1].frame_info, eData[1].sfb_nrg, NULL, h_con,
+                           h_envChan[1], SBR_MONO, NULL, YSzShift);
+      break;
+    case SBR_COUPLING:
+      calculateSbrEnvelope(h_envChan[0]->sbrExtractEnvelope.YBuffer,
+                           h_envChan[1]->sbrExtractEnvelope.YBuffer,
+                           h_envChan[0]->sbrExtractEnvelope.YBufferScale,
+                           h_envChan[1]->sbrExtractEnvelope.YBufferScale,
+                           eData[0].frame_info, eData[0].sfb_nrg,
+                           eData[1].sfb_nrg, h_con, h_envChan[0], SBR_COUPLING,
+                           &fData->maxQuantError, YSzShift);
+      break;
+    case SBR_SWITCH_LRC:
+      calculateSbrEnvelope(h_envChan[0]->sbrExtractEnvelope.YBuffer, NULL,
+                           h_envChan[0]->sbrExtractEnvelope.YBufferScale, NULL,
+                           eData[0].frame_info, eData[0].sfb_nrg, NULL, h_con,
+                           h_envChan[0], SBR_MONO, NULL, YSzShift);
+      calculateSbrEnvelope(h_envChan[1]->sbrExtractEnvelope.YBuffer, NULL,
+                           h_envChan[1]->sbrExtractEnvelope.YBufferScale, NULL,
+                           eData[1].frame_info, eData[1].sfb_nrg, NULL, h_con,
+                           h_envChan[1], SBR_MONO, NULL, YSzShift);
+      calculateSbrEnvelope(h_envChan[0]->sbrExtractEnvelope.YBuffer,
+                           h_envChan[1]->sbrExtractEnvelope.YBuffer,
+                           h_envChan[0]->sbrExtractEnvelope.YBufferScale,
+                           h_envChan[1]->sbrExtractEnvelope.YBufferScale,
+                           eData[0].frame_info, eData[0].sfb_nrg_coupling,
+                           eData[1].sfb_nrg_coupling, h_con, h_envChan[0],
+                           SBR_COUPLING, &fData->maxQuantError, YSzShift);
+      break;
+  }
+
+  /*
+    Noise floor quantisation and coding.
+  */
+
+  switch (stereoMode) {
+    case SBR_MONO:
+      sbrNoiseFloorLevelsQuantisation(eData[0].noise_level, eData[0].noiseFloor,
+                                      0);
+
+      FDKsbrEnc_codeEnvelope(eData[0].noise_level, fData->res,
+                             &h_envChan[0]->sbrCodeNoiseFloor,
+                             h_envChan[0]->encEnvData.domain_vec_noise, 0,
+                             (eData[0].frame_info->nEnvelopes > 1 ? 2 : 1), 0,
+                             sbrBitstreamData->HeaderActive);
+
+      break;
+    case SBR_LEFT_RIGHT:
+      sbrNoiseFloorLevelsQuantisation(eData[0].noise_level, eData[0].noiseFloor,
+                                      0);
+
+      FDKsbrEnc_codeEnvelope(eData[0].noise_level, fData->res,
+                             &h_envChan[0]->sbrCodeNoiseFloor,
+                             h_envChan[0]->encEnvData.domain_vec_noise, 0,
+                             (eData[0].frame_info->nEnvelopes > 1 ? 2 : 1), 0,
+                             sbrBitstreamData->HeaderActive);
+
+      sbrNoiseFloorLevelsQuantisation(eData[1].noise_level, eData[1].noiseFloor,
+                                      0);
+
+      FDKsbrEnc_codeEnvelope(eData[1].noise_level, fData->res,
+                             &h_envChan[1]->sbrCodeNoiseFloor,
+                             h_envChan[1]->encEnvData.domain_vec_noise, 0,
+                             (eData[1].frame_info->nEnvelopes > 1 ? 2 : 1), 0,
+                             sbrBitstreamData->HeaderActive);
+
+      break;
+
+    case SBR_COUPLING:
+      coupleNoiseFloor(eData[0].noiseFloor, eData[1].noiseFloor);
+
+      sbrNoiseFloorLevelsQuantisation(eData[0].noise_level, eData[0].noiseFloor,
+                                      0);
+
+      FDKsbrEnc_codeEnvelope(eData[0].noise_level, fData->res,
+                             &h_envChan[0]->sbrCodeNoiseFloor,
+                             h_envChan[0]->encEnvData.domain_vec_noise, 1,
+                             (eData[0].frame_info->nEnvelopes > 1 ? 2 : 1), 0,
+                             sbrBitstreamData->HeaderActive);
+
+      sbrNoiseFloorLevelsQuantisation(eData[1].noise_level, eData[1].noiseFloor,
+                                      1);
+
+      FDKsbrEnc_codeEnvelope(eData[1].noise_level, fData->res,
+                             &h_envChan[1]->sbrCodeNoiseFloor,
+                             h_envChan[1]->encEnvData.domain_vec_noise, 1,
+                             (eData[1].frame_info->nEnvelopes > 1 ? 2 : 1), 1,
+                             sbrBitstreamData->HeaderActive);
+
+      break;
+    case SBR_SWITCH_LRC:
+      sbrNoiseFloorLevelsQuantisation(eData[0].noise_level, eData[0].noiseFloor,
+                                      0);
+      sbrNoiseFloorLevelsQuantisation(eData[1].noise_level, eData[1].noiseFloor,
+                                      0);
+      coupleNoiseFloor(eData[0].noiseFloor, eData[1].noiseFloor);
+      sbrNoiseFloorLevelsQuantisation(eData[0].noise_level_coupling,
+                                      eData[0].noiseFloor, 0);
+      sbrNoiseFloorLevelsQuantisation(eData[1].noise_level_coupling,
+                                      eData[1].noiseFloor, 1);
+      break;
+  }
+
+  /*
+    Encode envelope of current frame.
+  */
+  switch (stereoMode) {
+    case SBR_MONO:
+      sbrHeaderData->coupling = 0;
+      h_envChan[0]->encEnvData.balance = 0;
+      FDKsbrEnc_codeEnvelope(
+          eData[0].sfb_nrg, eData[0].frame_info->freqRes,
+          &h_envChan[0]->sbrCodeEnvelope, h_envChan[0]->encEnvData.domain_vec,
+          sbrHeaderData->coupling, eData[0].frame_info->nEnvelopes, 0,
+          sbrBitstreamData->HeaderActive);
+      break;
+    case SBR_LEFT_RIGHT:
+      sbrHeaderData->coupling = 0;
+
+      h_envChan[0]->encEnvData.balance = 0;
+      h_envChan[1]->encEnvData.balance = 0;
+
+      FDKsbrEnc_codeEnvelope(
+          eData[0].sfb_nrg, eData[0].frame_info->freqRes,
+          &h_envChan[0]->sbrCodeEnvelope, h_envChan[0]->encEnvData.domain_vec,
+          sbrHeaderData->coupling, eData[0].frame_info->nEnvelopes, 0,
+          sbrBitstreamData->HeaderActive);
+      FDKsbrEnc_codeEnvelope(
+          eData[1].sfb_nrg, eData[1].frame_info->freqRes,
+          &h_envChan[1]->sbrCodeEnvelope, h_envChan[1]->encEnvData.domain_vec,
+          sbrHeaderData->coupling, eData[1].frame_info->nEnvelopes, 0,
+          sbrBitstreamData->HeaderActive);
+      break;
+    case SBR_COUPLING:
+      sbrHeaderData->coupling = 1;
+      h_envChan[0]->encEnvData.balance = 0;
+      h_envChan[1]->encEnvData.balance = 1;
+
+      FDKsbrEnc_codeEnvelope(
+          eData[0].sfb_nrg, eData[0].frame_info->freqRes,
+          &h_envChan[0]->sbrCodeEnvelope, h_envChan[0]->encEnvData.domain_vec,
+          sbrHeaderData->coupling, eData[0].frame_info->nEnvelopes, 0,
+          sbrBitstreamData->HeaderActive);
+      FDKsbrEnc_codeEnvelope(
+          eData[1].sfb_nrg, eData[1].frame_info->freqRes,
+          &h_envChan[1]->sbrCodeEnvelope, h_envChan[1]->encEnvData.domain_vec,
+          sbrHeaderData->coupling, eData[1].frame_info->nEnvelopes, 1,
+          sbrBitstreamData->HeaderActive);
+      break;
+    case SBR_SWITCH_LRC: {
+      INT payloadbitsLR;
+      INT payloadbitsCOUPLING;
+
+      SCHAR sfbNrgPrevTemp[MAX_NUM_CHANNELS][MAX_FREQ_COEFFS];
+      SCHAR noisePrevTemp[MAX_NUM_CHANNELS][MAX_NUM_NOISE_COEFFS];
+      INT upDateNrgTemp[MAX_NUM_CHANNELS];
+      INT upDateNoiseTemp[MAX_NUM_CHANNELS];
+      INT domainVecTemp[MAX_NUM_CHANNELS][MAX_ENVELOPES];
+      INT domainVecNoiseTemp[MAX_NUM_CHANNELS][MAX_ENVELOPES];
+
+      INT tempFlagRight = 0;
+      INT tempFlagLeft = 0;
+
+      /*
+         Store previous values, in order to be able to "undo" what is being
+         done.
+      */
+
+      for (ch = 0; ch < nChannels; ch++) {
+        FDKmemcpy(sfbNrgPrevTemp[ch],
+                  h_envChan[ch]->sbrCodeEnvelope.sfb_nrg_prev,
+                  MAX_FREQ_COEFFS * sizeof(SCHAR));
+
+        FDKmemcpy(noisePrevTemp[ch],
+                  h_envChan[ch]->sbrCodeNoiseFloor.sfb_nrg_prev,
+                  MAX_NUM_NOISE_COEFFS * sizeof(SCHAR));
+
+        upDateNrgTemp[ch] = h_envChan[ch]->sbrCodeEnvelope.upDate;
+        upDateNoiseTemp[ch] = h_envChan[ch]->sbrCodeNoiseFloor.upDate;
+
+        /*
+          forbid time coding in the first envelope in case of a different
+          previous stereomode
+        */
+        if (sbrHeaderData->prev_coupling) {
+          h_envChan[ch]->sbrCodeEnvelope.upDate = 0;
+          h_envChan[ch]->sbrCodeNoiseFloor.upDate = 0;
+        }
+      } /* ch */
+
+      /*
+         Code ordinary Left/Right stereo
+      */
+      FDKsbrEnc_codeEnvelope(eData[0].sfb_nrg, eData[0].frame_info->freqRes,
+                             &h_envChan[0]->sbrCodeEnvelope,
+                             h_envChan[0]->encEnvData.domain_vec, 0,
+                             eData[0].frame_info->nEnvelopes, 0,
+                             sbrBitstreamData->HeaderActive);
+      FDKsbrEnc_codeEnvelope(eData[1].sfb_nrg, eData[1].frame_info->freqRes,
+                             &h_envChan[1]->sbrCodeEnvelope,
+                             h_envChan[1]->encEnvData.domain_vec, 0,
+                             eData[1].frame_info->nEnvelopes, 0,
+                             sbrBitstreamData->HeaderActive);
+
+      c = 0;
+      for (i = 0; i < eData[0].nEnvelopes; i++) {
+        for (j = 0; j < h_envChan[0]->encEnvData.noScfBands[i]; j++) {
+          h_envChan[0]->encEnvData.ienvelope[i][j] = eData[0].sfb_nrg[c];
+          h_envChan[1]->encEnvData.ienvelope[i][j] = eData[1].sfb_nrg[c];
+          c++;
+        }
+      }
+
+      FDKsbrEnc_codeEnvelope(eData[0].noise_level, fData->res,
+                             &h_envChan[0]->sbrCodeNoiseFloor,
+                             h_envChan[0]->encEnvData.domain_vec_noise, 0,
+                             (eData[0].frame_info->nEnvelopes > 1 ? 2 : 1), 0,
+                             sbrBitstreamData->HeaderActive);
+
+      for (i = 0; i < MAX_NUM_NOISE_VALUES; i++)
+        h_envChan[0]->encEnvData.sbr_noise_levels[i] = eData[0].noise_level[i];
+
+      FDKsbrEnc_codeEnvelope(eData[1].noise_level, fData->res,
+                             &h_envChan[1]->sbrCodeNoiseFloor,
+                             h_envChan[1]->encEnvData.domain_vec_noise, 0,
+                             (eData[1].frame_info->nEnvelopes > 1 ? 2 : 1), 0,
+                             sbrBitstreamData->HeaderActive);
+
+      for (i = 0; i < MAX_NUM_NOISE_VALUES; i++)
+        h_envChan[1]->encEnvData.sbr_noise_levels[i] = eData[1].noise_level[i];
+
+      sbrHeaderData->coupling = 0;
+      h_envChan[0]->encEnvData.balance = 0;
+      h_envChan[1]->encEnvData.balance = 0;
+
+      payloadbitsLR = FDKsbrEnc_CountSbrChannelPairElement(
+          sbrHeaderData, hParametricStereo, sbrBitstreamData,
+          &h_envChan[0]->encEnvData, &h_envChan[1]->encEnvData, hCmonData,
+          h_con->sbrSyntaxFlags);
+
+      /*
+        swap saved stored with current values
+      */
+      for (ch = 0; ch < nChannels; ch++) {
+        INT itmp;
+        for (i = 0; i < MAX_FREQ_COEFFS; i++) {
+          /*
+            swap sfb energies
+          */
+          itmp = h_envChan[ch]->sbrCodeEnvelope.sfb_nrg_prev[i];
+          h_envChan[ch]->sbrCodeEnvelope.sfb_nrg_prev[i] =
+              sfbNrgPrevTemp[ch][i];
+          sfbNrgPrevTemp[ch][i] = itmp;
+        }
+        for (i = 0; i < MAX_NUM_NOISE_COEFFS; i++) {
+          /*
+            swap noise energies
+          */
+          itmp = h_envChan[ch]->sbrCodeNoiseFloor.sfb_nrg_prev[i];
+          h_envChan[ch]->sbrCodeNoiseFloor.sfb_nrg_prev[i] =
+              noisePrevTemp[ch][i];
+          noisePrevTemp[ch][i] = itmp;
+        }
+        /* swap update flags */
+        itmp = h_envChan[ch]->sbrCodeEnvelope.upDate;
+        h_envChan[ch]->sbrCodeEnvelope.upDate = upDateNrgTemp[ch];
+        upDateNrgTemp[ch] = itmp;
+
+        itmp = h_envChan[ch]->sbrCodeNoiseFloor.upDate;
+        h_envChan[ch]->sbrCodeNoiseFloor.upDate = upDateNoiseTemp[ch];
+        upDateNoiseTemp[ch] = itmp;
+
+        /*
+            save domain vecs
+        */
+        FDKmemcpy(domainVecTemp[ch], h_envChan[ch]->encEnvData.domain_vec,
+                  sizeof(INT) * MAX_ENVELOPES);
+        FDKmemcpy(domainVecNoiseTemp[ch],
+                  h_envChan[ch]->encEnvData.domain_vec_noise,
+                  sizeof(INT) * MAX_ENVELOPES);
+
+        /*
+          forbid time coding in the first envelope in case of a different
+          previous stereomode
+        */
+
+        if (!sbrHeaderData->prev_coupling) {
+          h_envChan[ch]->sbrCodeEnvelope.upDate = 0;
+          h_envChan[ch]->sbrCodeNoiseFloor.upDate = 0;
+        }
+      } /* ch */
+
+      /*
+         Coupling
+       */
+
+      FDKsbrEnc_codeEnvelope(
+          eData[0].sfb_nrg_coupling, eData[0].frame_info->freqRes,
+          &h_envChan[0]->sbrCodeEnvelope, h_envChan[0]->encEnvData.domain_vec,
+          1, eData[0].frame_info->nEnvelopes, 0,
+          sbrBitstreamData->HeaderActive);
+
+      FDKsbrEnc_codeEnvelope(
+          eData[1].sfb_nrg_coupling, eData[1].frame_info->freqRes,
+          &h_envChan[1]->sbrCodeEnvelope, h_envChan[1]->encEnvData.domain_vec,
+          1, eData[1].frame_info->nEnvelopes, 1,
+          sbrBitstreamData->HeaderActive);
+
+      c = 0;
+      for (i = 0; i < eData[0].nEnvelopes; i++) {
+        for (j = 0; j < h_envChan[0]->encEnvData.noScfBands[i]; j++) {
+          h_envChan[0]->encEnvData.ienvelope[i][j] =
+              eData[0].sfb_nrg_coupling[c];
+          h_envChan[1]->encEnvData.ienvelope[i][j] =
+              eData[1].sfb_nrg_coupling[c];
+          c++;
+        }
+      }
+
+      FDKsbrEnc_codeEnvelope(eData[0].noise_level_coupling, fData->res,
+                             &h_envChan[0]->sbrCodeNoiseFloor,
+                             h_envChan[0]->encEnvData.domain_vec_noise, 1,
+                             (eData[0].frame_info->nEnvelopes > 1 ? 2 : 1), 0,
+                             sbrBitstreamData->HeaderActive);
+
+      for (i = 0; i < MAX_NUM_NOISE_VALUES; i++)
+        h_envChan[0]->encEnvData.sbr_noise_levels[i] =
+            eData[0].noise_level_coupling[i];
+
+      FDKsbrEnc_codeEnvelope(eData[1].noise_level_coupling, fData->res,
+                             &h_envChan[1]->sbrCodeNoiseFloor,
+                             h_envChan[1]->encEnvData.domain_vec_noise, 1,
+                             (eData[1].frame_info->nEnvelopes > 1 ? 2 : 1), 1,
+                             sbrBitstreamData->HeaderActive);
+
+      for (i = 0; i < MAX_NUM_NOISE_VALUES; i++)
+        h_envChan[1]->encEnvData.sbr_noise_levels[i] =
+            eData[1].noise_level_coupling[i];
+
+      sbrHeaderData->coupling = 1;
+
+      h_envChan[0]->encEnvData.balance = 0;
+      h_envChan[1]->encEnvData.balance = 1;
+
+      tempFlagLeft = h_envChan[0]->encEnvData.addHarmonicFlag;
+      tempFlagRight = h_envChan[1]->encEnvData.addHarmonicFlag;
+
+      payloadbitsCOUPLING = FDKsbrEnc_CountSbrChannelPairElement(
+          sbrHeaderData, hParametricStereo, sbrBitstreamData,
+          &h_envChan[0]->encEnvData, &h_envChan[1]->encEnvData, hCmonData,
+          h_con->sbrSyntaxFlags);
+
+      h_envChan[0]->encEnvData.addHarmonicFlag = tempFlagLeft;
+      h_envChan[1]->encEnvData.addHarmonicFlag = tempFlagRight;
+
+      if (payloadbitsCOUPLING < payloadbitsLR) {
+        /*
+          copy coded coupling envelope and noise data to l/r
+        */
+        for (ch = 0; ch < nChannels; ch++) {
+          SBR_ENV_TEMP_DATA *ed = &eData[ch];
+          FDKmemcpy(ed->sfb_nrg, ed->sfb_nrg_coupling,
+                    MAX_NUM_ENVELOPE_VALUES * sizeof(SCHAR));
+          FDKmemcpy(ed->noise_level, ed->noise_level_coupling,
+                    MAX_NUM_NOISE_VALUES * sizeof(SCHAR));
+        }
+
+        sbrHeaderData->coupling = 1;
+        h_envChan[0]->encEnvData.balance = 0;
+        h_envChan[1]->encEnvData.balance = 1;
+      } else {
+        /*
+          restore saved l/r items
+        */
+        for (ch = 0; ch < nChannels; ch++) {
+          FDKmemcpy(h_envChan[ch]->sbrCodeEnvelope.sfb_nrg_prev,
+                    sfbNrgPrevTemp[ch], MAX_FREQ_COEFFS * sizeof(SCHAR));
+
+          h_envChan[ch]->sbrCodeEnvelope.upDate = upDateNrgTemp[ch];
+
+          FDKmemcpy(h_envChan[ch]->sbrCodeNoiseFloor.sfb_nrg_prev,
+                    noisePrevTemp[ch], MAX_NUM_NOISE_COEFFS * sizeof(SCHAR));
+
+          FDKmemcpy(h_envChan[ch]->encEnvData.domain_vec, domainVecTemp[ch],
+                    sizeof(INT) * MAX_ENVELOPES);
+          FDKmemcpy(h_envChan[ch]->encEnvData.domain_vec_noise,
+                    domainVecNoiseTemp[ch], sizeof(INT) * MAX_ENVELOPES);
+
+          h_envChan[ch]->sbrCodeNoiseFloor.upDate = upDateNoiseTemp[ch];
+        }
+
+        sbrHeaderData->coupling = 0;
+        h_envChan[0]->encEnvData.balance = 0;
+        h_envChan[1]->encEnvData.balance = 0;
+      }
+    } break;
+  } /* switch */
+
+  /* tell the envelope encoders how long it has been, since we last sent
+     a frame starting with a dF-coded envelope */
+  if (stereoMode == SBR_MONO) {
+    if (h_envChan[0]->encEnvData.domain_vec[0] == TIME)
+      h_envChan[0]->sbrCodeEnvelope.dF_edge_incr_fac++;
+    else
+      h_envChan[0]->sbrCodeEnvelope.dF_edge_incr_fac = 0;
+  } else {
+    if (h_envChan[0]->encEnvData.domain_vec[0] == TIME ||
+        h_envChan[1]->encEnvData.domain_vec[0] == TIME) {
+      h_envChan[0]->sbrCodeEnvelope.dF_edge_incr_fac++;
+      h_envChan[1]->sbrCodeEnvelope.dF_edge_incr_fac++;
+    } else {
+      h_envChan[0]->sbrCodeEnvelope.dF_edge_incr_fac = 0;
+      h_envChan[1]->sbrCodeEnvelope.dF_edge_incr_fac = 0;
+    }
+  }
+
+  /*
+    Send the encoded data to the bitstream
+  */
+  for (ch = 0; ch < nChannels; ch++) {
+    SBR_ENV_TEMP_DATA *ed = &eData[ch];
+    c = 0;
+    for (i = 0; i < ed->nEnvelopes; i++) {
+      for (j = 0; j < h_envChan[ch]->encEnvData.noScfBands[i]; j++) {
+        h_envChan[ch]->encEnvData.ienvelope[i][j] = ed->sfb_nrg[c];
+
+        c++;
+      }
+    }
+    for (i = 0; i < MAX_NUM_NOISE_VALUES; i++) {
+      h_envChan[ch]->encEnvData.sbr_noise_levels[i] = ed->noise_level[i];
+    }
+  } /* ch */
+
+  /*
+    Write bitstream
+  */
+  if (nChannels == 2) {
+    FDKsbrEnc_WriteEnvChannelPairElement(
+        sbrHeaderData, hParametricStereo, sbrBitstreamData,
+        &h_envChan[0]->encEnvData, &h_envChan[1]->encEnvData, hCmonData,
+        h_con->sbrSyntaxFlags);
+  } else {
+    FDKsbrEnc_WriteEnvSingleChannelElement(
+        sbrHeaderData, hParametricStereo, sbrBitstreamData,
+        &h_envChan[0]->encEnvData, hCmonData, h_con->sbrSyntaxFlags);
+  }
+
+  /*
+   * Update buffers.
+   */
+  for (ch = 0; ch < nChannels; ch++) {
+    int YBufferLength = h_envChan[ch]->sbrExtractEnvelope.no_cols >>
+                        h_envChan[ch]->sbrExtractEnvelope.YBufferSzShift;
+    for (i = 0; i < h_envChan[ch]->sbrExtractEnvelope.YBufferWriteOffset; i++) {
+      FDKmemcpy(h_envChan[ch]->sbrExtractEnvelope.YBuffer[i],
+                h_envChan[ch]->sbrExtractEnvelope.YBuffer[i + YBufferLength],
+                sizeof(FIXP_DBL) * 64);
+    }
+    h_envChan[ch]->sbrExtractEnvelope.YBufferScale[0] =
+        h_envChan[ch]->sbrExtractEnvelope.YBufferScale[1];
+  }
+
+  sbrHeaderData->prev_coupling = sbrHeaderData->coupling;
+}
+
+/***************************************************************************/
+/*!
+
+  \brief  creates an envelope extractor handle
+
+  \return error status
+
+****************************************************************************/
+INT FDKsbrEnc_CreateExtractSbrEnvelope(HANDLE_SBR_EXTRACT_ENVELOPE hSbrCut,
+                                       INT channel, INT chInEl,
+                                       UCHAR *dynamic_RAM) {
+  INT i;
+  FIXP_DBL *rBuffer, *iBuffer;
+  INT n;
+  FIXP_DBL *YBufferDyn;
+
+  FDKmemclear(hSbrCut, sizeof(SBR_EXTRACT_ENVELOPE));
+
+  if (NULL == (hSbrCut->p_YBuffer = GetRam_Sbr_envYBuffer(channel))) {
+    goto bail;
+  }
+
+  for (i = 0; i < (32 >> 1); i++) {
+    hSbrCut->YBuffer[i] = hSbrCut->p_YBuffer + (i * 64);
+  }
+  YBufferDyn = GetRam_Sbr_envYBuffer(chInEl, dynamic_RAM);
+  for (n = 0; i < 32; i++, n++) {
+    hSbrCut->YBuffer[i] = YBufferDyn + (n * 64);
+  }
+
+  rBuffer = GetRam_Sbr_envRBuffer(0, dynamic_RAM);
+  iBuffer = GetRam_Sbr_envIBuffer(0, dynamic_RAM);
+
+  for (i = 0; i < 32; i++) {
+    hSbrCut->rBuffer[i] = rBuffer + (i * 64);
+    hSbrCut->iBuffer[i] = iBuffer + (i * 64);
+  }
+
+  return 0;
+
+bail:
+  FDKsbrEnc_deleteExtractSbrEnvelope(hSbrCut);
+
+  return -1;
+}
+
+/***************************************************************************/
+/*!
+
+  \brief  Initialize an envelope extractor instance.
+
+  \return error status
+
+****************************************************************************/
+INT FDKsbrEnc_InitExtractSbrEnvelope(HANDLE_SBR_EXTRACT_ENVELOPE hSbrCut,
+                                     int no_cols, int no_rows, int start_index,
+                                     int time_slots, int time_step,
+                                     int tran_off, ULONG statesInitFlag,
+                                     int chInEl, UCHAR *dynamic_RAM,
+                                     UINT sbrSyntaxFlags) {
+  int YBufferLength, rBufferLength;
+  int i;
+
+  if (sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) {
+    int off = TRANSIENT_OFFSET_LD;
+    hSbrCut->YBufferWriteOffset = (no_cols >> 1) + off * time_step;
+  } else {
+    hSbrCut->YBufferWriteOffset = tran_off * time_step;
+  }
+  hSbrCut->rBufferReadOffset = 0;
+
+  YBufferLength = hSbrCut->YBufferWriteOffset + no_cols;
+  rBufferLength = no_cols;
+
+  hSbrCut->pre_transient_info[0] = 0;
+  hSbrCut->pre_transient_info[1] = 0;
+
+  hSbrCut->no_cols = no_cols;
+  hSbrCut->no_rows = no_rows;
+  hSbrCut->start_index = start_index;
+
+  hSbrCut->time_slots = time_slots;
+  hSbrCut->time_step = time_step;
+
+  FDK_ASSERT(no_rows <= 64);
+
+  /* Use half the Energy values if time step is 2 or greater */
+  if (time_step >= 2)
+    hSbrCut->YBufferSzShift = 1;
+  else
+    hSbrCut->YBufferSzShift = 0;
+
+  YBufferLength >>= hSbrCut->YBufferSzShift;
+  hSbrCut->YBufferWriteOffset >>= hSbrCut->YBufferSzShift;
+
+  FDK_ASSERT(YBufferLength <= 32);
+
+  FIXP_DBL *YBufferDyn = GetRam_Sbr_envYBuffer(chInEl, dynamic_RAM);
+  INT n = 0;
+  for (i = (32 >> 1); i < 32; i++, n++) {
+    hSbrCut->YBuffer[i] = YBufferDyn + (n * 64);
+  }
+
+  if (statesInitFlag) {
+    for (i = 0; i < YBufferLength; i++) {
+      FDKmemclear(hSbrCut->YBuffer[i], 64 * sizeof(FIXP_DBL));
+    }
+  }
+
+  for (i = 0; i < rBufferLength; i++) {
+    FDKmemclear(hSbrCut->rBuffer[i], 64 * sizeof(FIXP_DBL));
+    FDKmemclear(hSbrCut->iBuffer[i], 64 * sizeof(FIXP_DBL));
+  }
+
+  FDKmemclear(hSbrCut->envelopeCompensation, sizeof(UCHAR) * MAX_FREQ_COEFFS);
+
+  if (statesInitFlag) {
+    hSbrCut->YBufferScale[0] = hSbrCut->YBufferScale[1] = FRACT_BITS - 1;
+  }
+
+  return (0);
+}
+
+/***************************************************************************/
+/*!
+
+  \brief  deinitializes an envelope extractor handle
+
+  \return void
+
+****************************************************************************/
+
+void FDKsbrEnc_deleteExtractSbrEnvelope(HANDLE_SBR_EXTRACT_ENVELOPE hSbrCut) {
+  if (hSbrCut) {
+    FreeRam_Sbr_envYBuffer(&hSbrCut->p_YBuffer);
+  }
+}
+
+INT FDKsbrEnc_GetEnvEstDelay(HANDLE_SBR_EXTRACT_ENVELOPE hSbr) {
+  return hSbr->no_rows *
+         ((hSbr->YBufferWriteOffset) *
+              2 /* mult 2 because nrg's are grouped half */
+          - hSbr->rBufferReadOffset); /* in reference hold half spec and calc
+                                         nrg's on overlapped spec */
+}
diff --git a/fdk-aac/libSBRenc/src/env_est.h b/fdk-aac/libSBRenc/src/env_est.h
new file mode 100644
index 0000000..006f55b
--- /dev/null
+++ b/fdk-aac/libSBRenc/src/env_est.h
@@ -0,0 +1,223 @@
+/* -----------------------------------------------------------------------------
+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 encoder library ******************************
+
+   Author(s):
+
+   Description:
+
+*******************************************************************************/
+
+/*!
+  \file
+  \brief  Envelope estimation structs and prototypes $Revision: 92790 $
+*/
+#ifndef ENV_EST_H
+#define ENV_EST_H
+
+#include "sbr_def.h"
+#include "sbr_encoder.h" /* SBR econfig structs */
+#include "ps_main.h"
+#include "bit_sbr.h"
+#include "fram_gen.h"
+#include "tran_det.h"
+#include "code_env.h"
+#include "ton_corr.h"
+
+typedef struct {
+  FIXP_DBL *rBuffer[32];
+  FIXP_DBL *iBuffer[32];
+
+  FIXP_DBL *p_YBuffer;
+
+  FIXP_DBL *YBuffer[32];
+  int YBufferScale[2];
+
+  UCHAR envelopeCompensation[MAX_FREQ_COEFFS];
+  UCHAR pre_transient_info[2];
+
+  int YBufferWriteOffset;
+  int YBufferSzShift;
+  int rBufferReadOffset;
+
+  int no_cols;
+  int no_rows;
+  int start_index;
+
+  int time_slots;
+  int time_step;
+} SBR_EXTRACT_ENVELOPE;
+typedef SBR_EXTRACT_ENVELOPE *HANDLE_SBR_EXTRACT_ENVELOPE;
+
+struct ENV_CHANNEL {
+  FAST_TRAN_DETECTOR sbrFastTransientDetector;
+  SBR_TRANSIENT_DETECTOR sbrTransientDetector;
+  SBR_CODE_ENVELOPE sbrCodeEnvelope;
+  SBR_CODE_ENVELOPE sbrCodeNoiseFloor;
+  SBR_EXTRACT_ENVELOPE sbrExtractEnvelope;
+
+  SBR_ENVELOPE_FRAME SbrEnvFrame;
+  SBR_TON_CORR_EST TonCorr;
+
+  struct SBR_ENV_DATA encEnvData;
+
+  int qmfScale;
+  UCHAR fLevelProtect;
+};
+typedef struct ENV_CHANNEL *HANDLE_ENV_CHANNEL;
+
+/************  Function Declarations ***************/
+
+INT FDKsbrEnc_CreateExtractSbrEnvelope(HANDLE_SBR_EXTRACT_ENVELOPE hSbrCut,
+                                       INT channel, INT chInEl,
+                                       UCHAR *dynamic_RAM);
+
+INT FDKsbrEnc_InitExtractSbrEnvelope(HANDLE_SBR_EXTRACT_ENVELOPE hSbr,
+                                     int no_cols, int no_rows, int start_index,
+                                     int time_slots, int time_step,
+                                     int tran_off, ULONG statesInitFlag,
+                                     int chInEl, UCHAR *dynamic_RAM,
+                                     UINT sbrSyntaxFlags);
+
+void FDKsbrEnc_deleteExtractSbrEnvelope(HANDLE_SBR_EXTRACT_ENVELOPE hSbrCut);
+
+typedef struct {
+  FREQ_RES res[MAX_NUM_NOISE_VALUES];
+  int maxQuantError;
+
+} SBR_FRAME_TEMP_DATA;
+
+typedef struct {
+  const SBR_FRAME_INFO *frame_info;
+  FIXP_DBL noiseFloor[MAX_NUM_NOISE_VALUES];
+  SCHAR sfb_nrg_coupling
+      [MAX_NUM_ENVELOPE_VALUES]; /* only used if stereomode = SWITCH_L_R_C */
+  SCHAR sfb_nrg[MAX_NUM_ENVELOPE_VALUES];
+  SCHAR noise_level_coupling
+      [MAX_NUM_NOISE_VALUES]; /* only used if stereomode = SWITCH_L_R_C */
+  SCHAR noise_level[MAX_NUM_NOISE_VALUES];
+  UCHAR transient_info[3];
+  UCHAR nEnvelopes;
+} SBR_ENV_TEMP_DATA;
+
+/*
+ * Extract features from QMF data. Afterwards, the QMF data is not required
+ * anymore.
+ */
+void FDKsbrEnc_extractSbrEnvelope1(HANDLE_SBR_CONFIG_DATA h_con,
+                                   HANDLE_SBR_HEADER_DATA sbrHeaderData,
+                                   HANDLE_SBR_BITSTREAM_DATA sbrBitstreamData,
+                                   HANDLE_ENV_CHANNEL h_envChan,
+                                   HANDLE_COMMON_DATA cmonData,
+                                   SBR_ENV_TEMP_DATA *eData,
+                                   SBR_FRAME_TEMP_DATA *fData);
+
+/*
+ * Process the previously features extracted by FDKsbrEnc_extractSbrEnvelope1
+ * and create/encode SBR envelopes.
+ */
+void FDKsbrEnc_extractSbrEnvelope2(HANDLE_SBR_CONFIG_DATA h_con,
+                                   HANDLE_SBR_HEADER_DATA sbrHeaderData,
+                                   HANDLE_PARAMETRIC_STEREO hParametricStereo,
+                                   HANDLE_SBR_BITSTREAM_DATA sbrBitstreamData,
+                                   HANDLE_ENV_CHANNEL sbrEnvChannel0,
+                                   HANDLE_ENV_CHANNEL sbrEnvChannel1,
+                                   HANDLE_COMMON_DATA cmonData,
+                                   SBR_ENV_TEMP_DATA *eData,
+                                   SBR_FRAME_TEMP_DATA *fData, int clearOutput);
+
+INT FDKsbrEnc_GetEnvEstDelay(HANDLE_SBR_EXTRACT_ENVELOPE hSbr);
+
+#endif
diff --git a/fdk-aac/libSBRenc/src/fram_gen.cpp b/fdk-aac/libSBRenc/src/fram_gen.cpp
new file mode 100644
index 0000000..7ed6e79
--- /dev/null
+++ b/fdk-aac/libSBRenc/src/fram_gen.cpp
@@ -0,0 +1,1965 @@
+/* -----------------------------------------------------------------------------
+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 encoder library ******************************
+
+   Author(s):
+
+   Description:
+
+*******************************************************************************/
+
+#include "fram_gen.h"
+#include "sbr_misc.h"
+
+#include "genericStds.h"
+
+static const SBR_FRAME_INFO frameInfo1_2048 = {1, {0, 16}, {FREQ_RES_HIGH},
+                                               0, 1,       {0, 16}};
+
+static const SBR_FRAME_INFO frameInfo2_2048 = {
+    2, {0, 8, 16}, {FREQ_RES_HIGH, FREQ_RES_HIGH}, 0, 2, {0, 8, 16}};
+
+static const SBR_FRAME_INFO frameInfo4_2048 = {
+    4,
+    {0, 4, 8, 12, 16},
+    {FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH},
+    0,
+    2,
+    {0, 8, 16}};
+
+static const SBR_FRAME_INFO frameInfo1_2304 = {1, {0, 18}, {FREQ_RES_HIGH},
+                                               0, 1,       {0, 18}};
+
+static const SBR_FRAME_INFO frameInfo2_2304 = {
+    2, {0, 9, 18}, {FREQ_RES_HIGH, FREQ_RES_HIGH}, 0, 2, {0, 9, 18}};
+
+static const SBR_FRAME_INFO frameInfo4_2304 = {
+    4,
+    {0, 5, 9, 14, 18},
+    {FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH},
+    0,
+    2,
+    {0, 9, 18}};
+
+static const SBR_FRAME_INFO frameInfo1_1920 = {1, {0, 15}, {FREQ_RES_HIGH},
+                                               0, 1,       {0, 15}};
+
+static const SBR_FRAME_INFO frameInfo2_1920 = {
+    2, {0, 8, 15}, {FREQ_RES_HIGH, FREQ_RES_HIGH}, 0, 2, {0, 8, 15}};
+
+static const SBR_FRAME_INFO frameInfo4_1920 = {
+    4,
+    {0, 4, 8, 12, 15},
+    {FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH},
+    0,
+    2,
+    {0, 8, 15}};
+
+static const SBR_FRAME_INFO frameInfo1_1152 = {1, {0, 9}, {FREQ_RES_HIGH},
+                                               0, 1,      {0, 9}};
+
+static const SBR_FRAME_INFO frameInfo2_1152 = {
+    2, {0, 5, 9}, {FREQ_RES_HIGH, FREQ_RES_HIGH}, 0, 2, {0, 5, 9}};
+
+static const SBR_FRAME_INFO frameInfo4_1152 = {
+    4,
+    {0, 2, 5, 7, 9},
+    {FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH},
+    0,
+    2,
+    {0, 5, 9}};
+
+/* AACLD frame info */
+static const SBR_FRAME_INFO frameInfo1_512LD = {1, {0, 8}, {FREQ_RES_HIGH},
+                                                0, 1,      {0, 8}};
+
+static const SBR_FRAME_INFO frameInfo2_512LD = {
+    2, {0, 4, 8}, {FREQ_RES_HIGH, FREQ_RES_HIGH}, 0, 2, {0, 4, 8}};
+
+static const SBR_FRAME_INFO frameInfo4_512LD = {
+    4,
+    {0, 2, 4, 6, 8},
+    {FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH},
+    0,
+    2,
+    {0, 4, 8}};
+
+static int calcFillLengthMax(
+    int tranPos,        /*!< input : transient position (ref: tran det) */
+    int numberTimeSlots /*!< input : number of timeslots */
+);
+
+static void fillFrameTran(
+    const int *v_tuningSegm, /*!< tuning: desired segment lengths */
+    const int *v_tuningFreq, /*!< tuning: desired frequency resolutions */
+    int tran,                /*!< input : position of transient */
+    int *v_bord,             /*!< memNew: borders */
+    int *length_v_bord,      /*!< memNew: # borders */
+    int *v_freq,             /*!< memNew: frequency resolutions */
+    int *length_v_freq,      /*!< memNew: # frequency resolutions */
+    int *bmin,               /*!< hlpNew: first mandatory border */
+    int *bmax                /*!< hlpNew: last  mandatory border */
+);
+
+static void fillFramePre(INT dmax, INT *v_bord, INT *length_v_bord, INT *v_freq,
+                         INT *length_v_freq, INT bmin, INT rest);
+
+static void fillFramePost(INT *parts, INT *d, INT dmax, INT *v_bord,
+                          INT *length_v_bord, INT *v_freq, INT *length_v_freq,
+                          INT bmax, INT bufferFrameStart, INT numberTimeSlots,
+                          INT fmax);
+
+static void fillFrameInter(INT *nL, const int *v_tuningSegm, INT *v_bord,
+                           INT *length_v_bord, INT bmin, INT *v_freq,
+                           INT *length_v_freq, INT *v_bordFollow,
+                           INT *length_v_bordFollow, INT *v_freqFollow,
+                           INT *length_v_freqFollow, INT i_fillFollow, INT dmin,
+                           INT dmax, INT numberTimeSlots);
+
+static void calcFrameClass(FRAME_CLASS *frameClass, FRAME_CLASS *frameClassOld,
+                           INT tranFlag, INT *spreadFlag);
+
+static void specialCase(INT *spreadFlag, INT allowSpread, INT *v_bord,
+                        INT *length_v_bord, INT *v_freq, INT *length_v_freq,
+                        INT *parts, INT d);
+
+static void calcCmonBorder(INT *i_cmon, INT *i_tran, INT *v_bord,
+                           INT *length_v_bord, INT tran, INT bufferFrameStart,
+                           INT numberTimeSlots);
+
+static void keepForFollowUp(INT *v_bordFollow, INT *length_v_bordFollow,
+                            INT *v_freqFollow, INT *length_v_freqFollow,
+                            INT *i_tranFollow, INT *i_fillFollow, INT *v_bord,
+                            INT *length_v_bord, INT *v_freq, INT i_cmon,
+                            INT i_tran, INT parts, INT numberTimeSlots);
+
+static void calcCtrlSignal(HANDLE_SBR_GRID hSbrGrid, FRAME_CLASS frameClass,
+                           INT *v_bord, INT length_v_bord, INT *v_freq,
+                           INT length_v_freq, INT i_cmon, INT i_tran,
+                           INT spreadFlag, INT nL);
+
+static void ctrlSignal2FrameInfo(HANDLE_SBR_GRID hSbrGrid,
+                                 HANDLE_SBR_FRAME_INFO hFrameInfo,
+                                 FREQ_RES *freq_res_fixfix);
+
+/* table for 8 time slot index */
+static const int envelopeTable_8[8][5] = {
+    /* transientIndex  nEnv, tranIdx, shortEnv, border1, border2, ... */
+    /* borders from left to right side; -1 = not in use */
+    /*[|T-|------]*/ {2, 0, 0, 1, -1},
+    /*[|-T-|-----]*/ {2, 0, 0, 2, -1},
+    /*[--|T-|----]*/ {3, 1, 1, 2, 4},
+    /*[---|T-|---]*/ {3, 1, 1, 3, 5},
+    /*[----|T-|--]*/ {3, 1, 1, 4, 6},
+    /*[-----|T--|]*/ {2, 1, 1, 5, -1},
+    /*[------|T-|]*/ {2, 1, 1, 6, -1},
+    /*[-------|T|]*/ {2, 1, 1, 7, -1},
+};
+
+/* table for 16 time slot index */
+static const int envelopeTable_16[16][6] = {
+    /* transientIndex  nEnv, tranIdx, shortEnv, border1, border2, ... */
+    /* length from left to right side; -1 = not in use */
+    /*[|T---|------------|]*/ {2, 0, 0, 4, -1, -1},
+    /*[|-T---|-----------|]*/ {2, 0, 0, 5, -1, -1},
+    /*[|--|T---|----------]*/ {3, 1, 1, 2, 6, -1},
+    /*[|---|T---|---------]*/ {3, 1, 1, 3, 7, -1},
+    /*[|----|T---|--------]*/ {3, 1, 1, 4, 8, -1},
+    /*[|-----|T---|-------]*/ {3, 1, 1, 5, 9, -1},
+    /*[|------|T---|------]*/ {3, 1, 1, 6, 10, -1},
+    /*[|-------|T---|-----]*/ {3, 1, 1, 7, 11, -1},
+    /*[|--------|T---|----]*/ {3, 1, 1, 8, 12, -1},
+    /*[|---------|T---|---]*/ {3, 1, 1, 9, 13, -1},
+    /*[|----------|T---|--]*/ {3, 1, 1, 10, 14, -1},
+    /*[|-----------|T----|]*/ {2, 1, 1, 11, -1, -1},
+    /*[|------------|T---|]*/ {2, 1, 1, 12, -1, -1},
+    /*[|-------------|T--|]*/ {2, 1, 1, 13, -1, -1},
+    /*[|--------------|T-|]*/ {2, 1, 1, 14, -1, -1},
+    /*[|---------------|T|]*/ {2, 1, 1, 15, -1, -1},
+};
+
+/* table for 15 time slot index */
+static const int envelopeTable_15[15][6] = {
+    /* transientIndex  nEnv, tranIdx, shortEnv, border1, border2, ... */
+    /* length from left to right side; -1 = not in use */
+    /*[|T---|------------]*/ {2, 0, 0, 4, -1, -1},
+    /*[|-T---|-----------]*/ {2, 0, 0, 5, -1, -1},
+    /*[|--|T---|---------]*/ {3, 1, 1, 2, 6, -1},
+    /*[|---|T---|--------]*/ {3, 1, 1, 3, 7, -1},
+    /*[|----|T---|-------]*/ {3, 1, 1, 4, 8, -1},
+    /*[|-----|T---|------]*/ {3, 1, 1, 5, 9, -1},
+    /*[|------|T---|-----]*/ {3, 1, 1, 6, 10, -1},
+    /*[|-------|T---|----]*/ {3, 1, 1, 7, 11, -1},
+    /*[|--------|T---|---]*/ {3, 1, 1, 8, 12, -1},
+    /*[|---------|T---|--]*/ {3, 1, 1, 9, 13, -1},
+    /*[|----------|T----|]*/ {2, 1, 1, 10, -1, -1},
+    /*[|-----------|T---|]*/ {2, 1, 1, 11, -1, -1},
+    /*[|------------|T--|]*/ {2, 1, 1, 12, -1, -1},
+    /*[|-------------|T-|]*/ {2, 1, 1, 13, -1, -1},
+    /*[|--------------|T|]*/ {2, 1, 1, 14, -1, -1},
+};
+
+static const int minFrameTranDistance = 4;
+
+static const FREQ_RES freqRes_table_8[] = {
+    FREQ_RES_LOW,  FREQ_RES_LOW,  FREQ_RES_LOW,  FREQ_RES_LOW, FREQ_RES_LOW,
+    FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH};
+
+static const FREQ_RES freqRes_table_16[16] = {
+    /* size of envelope */
+    /* 0-4 */ FREQ_RES_LOW,
+    FREQ_RES_LOW,
+    FREQ_RES_LOW,
+    FREQ_RES_LOW,
+    FREQ_RES_LOW,
+    /* 5-9 */ FREQ_RES_LOW,
+    FREQ_RES_HIGH,
+    FREQ_RES_HIGH,
+    FREQ_RES_HIGH,
+    FREQ_RES_HIGH,
+    /* 10-16 */ FREQ_RES_HIGH,
+    FREQ_RES_HIGH,
+    FREQ_RES_HIGH,
+    FREQ_RES_HIGH,
+    FREQ_RES_HIGH,
+    FREQ_RES_HIGH};
+
+static void generateFixFixOnly(HANDLE_SBR_FRAME_INFO hSbrFrameInfo,
+                               HANDLE_SBR_GRID hSbrGrid, int tranPosInternal,
+                               int numberTimeSlots, UCHAR fResTransIsLow);
+
+/*!
+  Functionname: FDKsbrEnc_frameInfoGenerator
+
+  Description:  produces the FRAME_INFO struct for the current frame
+
+  Arguments:    hSbrEnvFrame          - pointer to sbr envelope handle
+                v_pre_transient_info  - pointer to transient info vector
+                v_transient_info      - pointer to previous transient info
+vector v_tuning              - pointer to tuning vector
+
+ Return:      frame_info        - pointer to SBR_FRAME_INFO struct
+
+*******************************************************************************/
+HANDLE_SBR_FRAME_INFO
+FDKsbrEnc_frameInfoGenerator(HANDLE_SBR_ENVELOPE_FRAME hSbrEnvFrame,
+                             UCHAR *v_transient_info, const INT rightBorderFIX,
+                             UCHAR *v_transient_info_pre, int ldGrid,
+                             const int *v_tuning) {
+  INT numEnv, tranPosInternal = 0, bmin = 0, bmax = 0, parts, d, i_cmon = 0,
+              i_tran = 0, nL;
+  INT fmax = 0;
+
+  INT *v_bord = hSbrEnvFrame->v_bord;
+  INT *v_freq = hSbrEnvFrame->v_freq;
+  INT *v_bordFollow = hSbrEnvFrame->v_bordFollow;
+  INT *v_freqFollow = hSbrEnvFrame->v_freqFollow;
+
+  INT *length_v_bordFollow = &hSbrEnvFrame->length_v_bordFollow;
+  INT *length_v_freqFollow = &hSbrEnvFrame->length_v_freqFollow;
+  INT *length_v_bord = &hSbrEnvFrame->length_v_bord;
+  INT *length_v_freq = &hSbrEnvFrame->length_v_freq;
+  INT *spreadFlag = &hSbrEnvFrame->spreadFlag;
+  INT *i_tranFollow = &hSbrEnvFrame->i_tranFollow;
+  INT *i_fillFollow = &hSbrEnvFrame->i_fillFollow;
+  FRAME_CLASS *frameClassOld = &hSbrEnvFrame->frameClassOld;
+  FRAME_CLASS frameClass = FIXFIX;
+
+  INT allowSpread = hSbrEnvFrame->allowSpread;
+  INT numEnvStatic = hSbrEnvFrame->numEnvStatic;
+  INT staticFraming = hSbrEnvFrame->staticFraming;
+  INT dmin = hSbrEnvFrame->dmin;
+  INT dmax = hSbrEnvFrame->dmax;
+
+  INT bufferFrameStart = hSbrEnvFrame->SbrGrid.bufferFrameStart;
+  INT numberTimeSlots = hSbrEnvFrame->SbrGrid.numberTimeSlots;
+  INT frameMiddleSlot = hSbrEnvFrame->frameMiddleSlot;
+
+  INT tranPos = v_transient_info[0];
+  INT tranFlag = v_transient_info[1];
+
+  const int *v_tuningSegm = v_tuning;
+  const int *v_tuningFreq = v_tuning + 3;
+
+  hSbrEnvFrame->v_tuningSegm = v_tuningSegm;
+
+  if (ldGrid) {
+    /* in case there was a transient at the very end of the previous frame,
+     * start with a transient envelope */
+    if (!tranFlag && v_transient_info_pre[1] &&
+        (numberTimeSlots - v_transient_info_pre[0] < minFrameTranDistance)) {
+      tranFlag = 1;
+      tranPos = 0;
+    }
+  }
+
+  /*
+   * Synopsis:
+   *
+   * The frame generator creates the time-/frequency-grid for one SBR frame.
+   * Input signals are provided by the transient detector and the frame
+   * splitter (transientDetectNew() & FrameSplitter() in tran_det.c).  The
+   * framing is controlled by adjusting tuning parameters stored in
+   * FRAME_GEN_TUNING.  The parameter values are dependent on frame lengths
+   * and bitrates, and may in the future be signal dependent.
+   *
+   * The envelope borders are stored for frame generator internal use in
+   * aBorders.  The contents of aBorders represent positions along the time
+   * axis given in the figures in fram_gen.h (the "frame-generator" rows).
+   * The unit is "time slot".  The figures in fram_gen.h also define the
+   * detection ranges for the transient detector.  For every border in
+   * aBorders, there is a corresponding entry in aFreqRes, which defines the
+   * frequency resolution of the envelope following (delimited by) the
+   * border.
+   *
+   * When no transients are present, FIXFIX class frames are used.  The
+   * frame splitter decides whether to use one or two envelopes in the
+   * FIXFIX frame.  "Sparse transients" (separated by a few frames without
+   * transients) are handeled by [FIXVAR, VARFIX] pairs or (depending on
+   * tuning and transient position relative the nominal frame boundaries)
+   * by [FIXVAR, VARVAR, VARFIX] triples.  "Tight transients" (in
+   * consecutive frames) are handeled by [..., VARVAR, VARVAR, ...]
+   * sequences.
+   *
+   * The generator assumes that transients are "sparse", and designs
+   * borders for [FIXVAR, VARFIX] pairs right away, where the first frame
+   * corresponds to the present frame.  At the next call of the generator
+   * it is known whether the transient actually is "sparse" or not.  If
+   * 'yes', the already calculated VARFIX borders are used.  If 'no', new
+   * borders, meeting the requirements of the "tight" transient, are
+   * calculated.
+   *
+   * The generator produces two outputs:  A "clear-text bitstream" stored in
+   * SBR_GRID, and a straight-forward representation of the grid stored in
+   * SBR_FRAME_INFO.  The former is subsequently converted to the actual
+   * bitstream sbr_grid() (encodeSbrGrid() in bit_sbr.c).  The latter is
+   * used by other encoder functions, such as the envelope estimator
+   * (calculateSbrEnvelope() in env_est.c) and the noise floor and missing
+   * harmonics detector (TonCorrParamExtr() in nf_est.c).
+   */
+
+  if (staticFraming) {
+    /*--------------------------------------------------------------------------
+      Ignore transient detector
+    ---------------------------------------------------------------------------*/
+
+    frameClass = FIXFIX;
+    numEnv = numEnvStatic;   /* {1,2,4,8} */
+    *frameClassOld = FIXFIX; /* for change to dyn */
+    hSbrEnvFrame->SbrGrid.bs_num_env = numEnv;
+    hSbrEnvFrame->SbrGrid.frameClass = frameClass;
+  } else {
+    /*--------------------------------------------------------------------------
+      Calculate frame class to use
+    ---------------------------------------------------------------------------*/
+    if (rightBorderFIX) {
+      tranFlag = 0;
+      *spreadFlag = 0;
+    }
+    calcFrameClass(&frameClass, frameClassOld, tranFlag, spreadFlag);
+
+    /* patch for new frame class FIXFIXonly for AAC LD */
+    if (tranFlag && ldGrid) {
+      frameClass = FIXFIXonly;
+      *frameClassOld = FIXFIX;
+    }
+
+    /*
+     * every transient is processed below by inserting
+     *
+     * - one border at the onset of the transient
+     * - one or more "decay borders" (after the onset of the transient)
+     * - optionally one "attack border" (before the onset of the transient)
+     *
+     * those borders are referred to as "mandatory borders" and are
+     * defined by the 'segmentLength' array in FRAME_GEN_TUNING
+     *
+     * the frequency resolutions of the corresponding envelopes are
+     * defined by the 'segmentRes' array in FRAME_GEN_TUNING
+     */
+
+    /*--------------------------------------------------------------------------
+      Design frame (or follow-up old design)
+    ---------------------------------------------------------------------------*/
+    if (tranFlag) {
+      /* Always for FixVar, often but not always for VarVar */
+
+      /*--------------------------------------------------------------------------
+        Design part of T/F-grid around the new transient
+      ---------------------------------------------------------------------------*/
+
+      tranPosInternal =
+          frameMiddleSlot + tranPos + bufferFrameStart; /* FH 00-06-26 */
+      /*
+        add mandatory borders around transient
+      */
+
+      fillFrameTran(v_tuningSegm, v_tuningFreq, tranPosInternal, v_bord,
+                    length_v_bord, v_freq, length_v_freq, &bmin, &bmax);
+
+      /* make sure we stay within the maximum SBR frame overlap */
+      fmax = calcFillLengthMax(tranPos, numberTimeSlots);
+    }
+
+    switch (frameClass) {
+      case FIXFIXonly:
+        FDK_ASSERT(ldGrid);
+        tranPosInternal = tranPos;
+        generateFixFixOnly(&(hSbrEnvFrame->SbrFrameInfo),
+                           &(hSbrEnvFrame->SbrGrid), tranPosInternal,
+                           numberTimeSlots, hSbrEnvFrame->fResTransIsLow);
+
+        return &(hSbrEnvFrame->SbrFrameInfo);
+
+      case FIXVAR:
+
+        /*--------------------------------------------------------------------------
+           Design remaining parts of T/F-grid (assuming next frame is VarFix)
+        ---------------------------------------------------------------------------*/
+
+        /*--------------------------------------------------------------------------
+          Fill region before new transient:
+        ---------------------------------------------------------------------------*/
+        fillFramePre(dmax, v_bord, length_v_bord, v_freq, length_v_freq, bmin,
+                     bmin - bufferFrameStart); /* FH 00-06-26 */
+
+        /*--------------------------------------------------------------------------
+          Fill region after new transient:
+        ---------------------------------------------------------------------------*/
+        fillFramePost(&parts, &d, dmax, v_bord, length_v_bord, v_freq,
+                      length_v_freq, bmax, bufferFrameStart, numberTimeSlots,
+                      fmax);
+
+        /*--------------------------------------------------------------------------
+          Take care of special case:
+        ---------------------------------------------------------------------------*/
+        if (parts == 1 && d < dmin) /* no fill, short last envelope */
+          specialCase(spreadFlag, allowSpread, v_bord, length_v_bord, v_freq,
+                      length_v_freq, &parts, d);
+
+        /*--------------------------------------------------------------------------
+          Calculate common border (split-point)
+        ---------------------------------------------------------------------------*/
+        calcCmonBorder(&i_cmon, &i_tran, v_bord, length_v_bord, tranPosInternal,
+                       bufferFrameStart, numberTimeSlots); /* FH 00-06-26 */
+
+        /*--------------------------------------------------------------------------
+          Extract data for proper follow-up in next frame
+        ---------------------------------------------------------------------------*/
+        keepForFollowUp(v_bordFollow, length_v_bordFollow, v_freqFollow,
+                        length_v_freqFollow, i_tranFollow, i_fillFollow, v_bord,
+                        length_v_bord, v_freq, i_cmon, i_tran, parts,
+                        numberTimeSlots); /* FH 00-06-26 */
+
+        /*--------------------------------------------------------------------------
+          Calculate control signal
+        ---------------------------------------------------------------------------*/
+        calcCtrlSignal(&hSbrEnvFrame->SbrGrid, frameClass, v_bord,
+                       *length_v_bord, v_freq, *length_v_freq, i_cmon, i_tran,
+                       *spreadFlag, DC);
+        break;
+      case VARFIX:
+        /*--------------------------------------------------------------------------
+          Follow-up old transient - calculate control signal
+        ---------------------------------------------------------------------------*/
+        calcCtrlSignal(&hSbrEnvFrame->SbrGrid, frameClass, v_bordFollow,
+                       *length_v_bordFollow, v_freqFollow, *length_v_freqFollow,
+                       DC, *i_tranFollow, *spreadFlag, DC);
+        break;
+      case VARVAR:
+        if (*spreadFlag) { /* spread across three frames */
+          /*--------------------------------------------------------------------------
+            Follow-up old transient - calculate control signal
+          ---------------------------------------------------------------------------*/
+          calcCtrlSignal(&hSbrEnvFrame->SbrGrid, frameClass, v_bordFollow,
+                         *length_v_bordFollow, v_freqFollow,
+                         *length_v_freqFollow, DC, *i_tranFollow, *spreadFlag,
+                         DC);
+
+          *spreadFlag = 0;
+
+          /*--------------------------------------------------------------------------
+            Extract data for proper follow-up in next frame
+          ---------------------------------------------------------------------------*/
+          v_bordFollow[0] = hSbrEnvFrame->SbrGrid.bs_abs_bord_1 -
+                            numberTimeSlots; /* FH 00-06-26 */
+          v_freqFollow[0] = 1;
+          *length_v_bordFollow = 1;
+          *length_v_freqFollow = 1;
+
+          *i_tranFollow = -DC;
+          *i_fillFollow = -DC;
+        } else {
+          /*--------------------------------------------------------------------------
+            Design remaining parts of T/F-grid (assuming next frame is VarFix)
+            adapt or fill region before new transient:
+          ---------------------------------------------------------------------------*/
+          fillFrameInter(&nL, v_tuningSegm, v_bord, length_v_bord, bmin, v_freq,
+                         length_v_freq, v_bordFollow, length_v_bordFollow,
+                         v_freqFollow, length_v_freqFollow, *i_fillFollow, dmin,
+                         dmax, numberTimeSlots);
+
+          /*--------------------------------------------------------------------------
+            Fill after transient:
+          ---------------------------------------------------------------------------*/
+          fillFramePost(&parts, &d, dmax, v_bord, length_v_bord, v_freq,
+                        length_v_freq, bmax, bufferFrameStart, numberTimeSlots,
+                        fmax);
+
+          /*--------------------------------------------------------------------------
+            Take care of special case:
+          ---------------------------------------------------------------------------*/
+          if (parts == 1 && d < dmin) /*% no fill, short last envelope */
+            specialCase(spreadFlag, allowSpread, v_bord, length_v_bord, v_freq,
+                        length_v_freq, &parts, d);
+
+          /*--------------------------------------------------------------------------
+            Calculate common border (split-point)
+          ---------------------------------------------------------------------------*/
+          calcCmonBorder(&i_cmon, &i_tran, v_bord, length_v_bord,
+                         tranPosInternal, bufferFrameStart, numberTimeSlots);
+
+          /*--------------------------------------------------------------------------
+            Extract data for proper follow-up in next frame
+          ---------------------------------------------------------------------------*/
+          keepForFollowUp(v_bordFollow, length_v_bordFollow, v_freqFollow,
+                          length_v_freqFollow, i_tranFollow, i_fillFollow,
+                          v_bord, length_v_bord, v_freq, i_cmon, i_tran, parts,
+                          numberTimeSlots);
+
+          /*--------------------------------------------------------------------------
+            Calculate control signal
+          ---------------------------------------------------------------------------*/
+          calcCtrlSignal(&hSbrEnvFrame->SbrGrid, frameClass, v_bord,
+                         *length_v_bord, v_freq, *length_v_freq, i_cmon, i_tran,
+                         0, nL);
+        }
+        break;
+      case FIXFIX:
+        if (tranPos == 0)
+          numEnv = 1;
+        else
+          numEnv = 2;
+
+        hSbrEnvFrame->SbrGrid.bs_num_env = numEnv;
+        hSbrEnvFrame->SbrGrid.frameClass = frameClass;
+
+        break;
+      default:
+        FDK_ASSERT(0);
+    }
+  }
+
+  /*-------------------------------------------------------------------------
+    Convert control signal to frame info struct
+  ---------------------------------------------------------------------------*/
+  ctrlSignal2FrameInfo(&hSbrEnvFrame->SbrGrid, &hSbrEnvFrame->SbrFrameInfo,
+                       hSbrEnvFrame->freq_res_fixfix);
+
+  return &hSbrEnvFrame->SbrFrameInfo;
+}
+
+/***************************************************************************/
+/*!
+  \brief    Gnerates frame info for FIXFIXonly frame class used for low delay
+ version
+
+  \return   nothing
+ ****************************************************************************/
+static void generateFixFixOnly(HANDLE_SBR_FRAME_INFO hSbrFrameInfo,
+                               HANDLE_SBR_GRID hSbrGrid, int tranPosInternal,
+                               int numberTimeSlots, UCHAR fResTransIsLow) {
+  int nEnv, i, k = 0, tranIdx;
+  const int *pTable = NULL;
+  const FREQ_RES *freqResTable = NULL;
+
+  switch (numberTimeSlots) {
+    case 8: {
+      pTable = envelopeTable_8[tranPosInternal];
+    }
+      freqResTable = freqRes_table_8;
+      break;
+    case 15:
+      pTable = envelopeTable_15[tranPosInternal];
+      freqResTable = freqRes_table_16;
+      break;
+    case 16:
+      pTable = envelopeTable_16[tranPosInternal];
+      freqResTable = freqRes_table_16;
+      break;
+  }
+
+  /* look number of envolpes in table */
+  nEnv = pTable[0];
+  /* look up envolpe distribution in table */
+  for (i = 1; i < nEnv; i++) hSbrFrameInfo->borders[i] = pTable[i + 2];
+
+  /* open and close frame border */
+  hSbrFrameInfo->borders[0] = 0;
+  hSbrFrameInfo->borders[nEnv] = numberTimeSlots;
+
+  /* adjust segment-frequency-resolution according to the segment-length */
+  for (i = 0; i < nEnv; i++) {
+    k = hSbrFrameInfo->borders[i + 1] - hSbrFrameInfo->borders[i];
+    if (!fResTransIsLow)
+      hSbrFrameInfo->freqRes[i] = freqResTable[k];
+    else
+      hSbrFrameInfo->freqRes[i] = FREQ_RES_LOW;
+
+    hSbrGrid->v_f[i] = hSbrFrameInfo->freqRes[i];
+  }
+
+  hSbrFrameInfo->nEnvelopes = nEnv;
+  hSbrFrameInfo->shortEnv = pTable[2];
+  /* transient idx */
+  tranIdx = pTable[1];
+
+  /* add noise floors */
+  hSbrFrameInfo->bordersNoise[0] = 0;
+  hSbrFrameInfo->bordersNoise[1] =
+      hSbrFrameInfo->borders[tranIdx ? tranIdx : 1];
+  hSbrFrameInfo->bordersNoise[2] = numberTimeSlots;
+  hSbrFrameInfo->nNoiseEnvelopes = 2;
+
+  hSbrGrid->frameClass = FIXFIXonly;
+  hSbrGrid->bs_abs_bord = tranPosInternal;
+  hSbrGrid->bs_num_env = nEnv;
+}
+
+/*******************************************************************************
+ Functionname:  FDKsbrEnc_initFrameInfoGenerator
+ *******************************************************************************
+
+ Description:
+
+ Arguments:   hSbrEnvFrame  - pointer to sbr envelope handle
+              allowSpread   - commandline parameter
+              numEnvStatic  - commandline parameter
+              staticFraming - commandline parameter
+
+ Return:      none
+
+*******************************************************************************/
+void FDKsbrEnc_initFrameInfoGenerator(HANDLE_SBR_ENVELOPE_FRAME hSbrEnvFrame,
+                                      INT allowSpread, INT numEnvStatic,
+                                      INT staticFraming, INT timeSlots,
+                                      const FREQ_RES *freq_res_fixfix,
+                                      UCHAR fResTransIsLow,
+                                      INT ldGrid) { /* FH 00-06-26 */
+
+  FDKmemclear(hSbrEnvFrame, sizeof(SBR_ENVELOPE_FRAME));
+
+  /* Initialisation */
+  hSbrEnvFrame->frameClassOld = FIXFIX;
+  hSbrEnvFrame->spreadFlag = 0;
+
+  hSbrEnvFrame->allowSpread = allowSpread;
+  hSbrEnvFrame->numEnvStatic = numEnvStatic;
+  hSbrEnvFrame->staticFraming = staticFraming;
+  hSbrEnvFrame->freq_res_fixfix[0] = freq_res_fixfix[0];
+  hSbrEnvFrame->freq_res_fixfix[1] = freq_res_fixfix[1];
+  hSbrEnvFrame->fResTransIsLow = fResTransIsLow;
+
+  hSbrEnvFrame->length_v_bord = 0;
+  hSbrEnvFrame->length_v_bordFollow = 0;
+
+  hSbrEnvFrame->length_v_freq = 0;
+  hSbrEnvFrame->length_v_freqFollow = 0;
+
+  hSbrEnvFrame->i_tranFollow = 0;
+  hSbrEnvFrame->i_fillFollow = 0;
+
+  hSbrEnvFrame->SbrGrid.numberTimeSlots = timeSlots;
+
+  if (ldGrid) {
+    /*case CODEC_AACLD:*/
+    hSbrEnvFrame->dmin = 2;
+    hSbrEnvFrame->dmax = 16;
+    hSbrEnvFrame->frameMiddleSlot = FRAME_MIDDLE_SLOT_512LD;
+    hSbrEnvFrame->SbrGrid.bufferFrameStart = 0;
+  } else
+    switch (timeSlots) {
+      case NUMBER_TIME_SLOTS_1920:
+        hSbrEnvFrame->dmin = 4;
+        hSbrEnvFrame->dmax = 12;
+        hSbrEnvFrame->SbrGrid.bufferFrameStart = 0;
+        hSbrEnvFrame->frameMiddleSlot = FRAME_MIDDLE_SLOT_1920;
+        break;
+      case NUMBER_TIME_SLOTS_2048:
+        hSbrEnvFrame->dmin = 4;
+        hSbrEnvFrame->dmax = 12;
+        hSbrEnvFrame->SbrGrid.bufferFrameStart = 0;
+        hSbrEnvFrame->frameMiddleSlot = FRAME_MIDDLE_SLOT_2048;
+        break;
+      case NUMBER_TIME_SLOTS_1152:
+        hSbrEnvFrame->dmin = 2;
+        hSbrEnvFrame->dmax = 8;
+        hSbrEnvFrame->SbrGrid.bufferFrameStart = 0;
+        hSbrEnvFrame->frameMiddleSlot = FRAME_MIDDLE_SLOT_1152;
+        break;
+      case NUMBER_TIME_SLOTS_2304:
+        hSbrEnvFrame->dmin = 4;
+        hSbrEnvFrame->dmax = 15;
+        hSbrEnvFrame->SbrGrid.bufferFrameStart = 0;
+        hSbrEnvFrame->frameMiddleSlot = FRAME_MIDDLE_SLOT_2304;
+        break;
+      default:
+        FDK_ASSERT(0);
+    }
+}
+
+/*******************************************************************************
+ Functionname:  fillFrameTran
+ *******************************************************************************
+
+ Description:  Add mandatory borders, as described by the tuning vector
+               and the current transient position
+
+ Arguments:
+      modified:
+              v_bord        - int pointer to v_bord vector
+              length_v_bord - length of v_bord vector
+              v_freq        - int pointer to v_freq vector
+              length_v_freq - length of v_freq vector
+              bmin          - int pointer to bmin (call by reference)
+              bmax          - int pointer to bmax (call by reference)
+      not modified:
+              tran          - position of transient
+              v_tuningSegm  - int pointer to v_tuningSegm vector
+              v_tuningFreq  - int pointer to v_tuningFreq vector
+
+ Return:      none
+
+*******************************************************************************/
+static void fillFrameTran(
+    const int *v_tuningSegm, /*!< tuning: desired segment lengths */
+    const int *v_tuningFreq, /*!< tuning: desired frequency resolutions */
+    int tran,                /*!< input : position of transient */
+    int *v_bord,             /*!< memNew: borders */
+    int *length_v_bord,      /*!< memNew: # borders */
+    int *v_freq,             /*!< memNew: frequency resolutions */
+    int *length_v_freq,      /*!< memNew: # frequency resolutions */
+    int *bmin,               /*!< hlpNew: first mandatory border */
+    int *bmax                /*!< hlpNew: last  mandatory border */
+) {
+  int bord, i;
+
+  *length_v_bord = 0;
+  *length_v_freq = 0;
+
+  /* add attack env leading border (optional) */
+  if (v_tuningSegm[0]) {
+    /* v_bord = [(Ba)] start of attack env */
+    FDKsbrEnc_AddRight(v_bord, length_v_bord, (tran - v_tuningSegm[0]));
+
+    /* v_freq = [(Fa)] res of attack env */
+    FDKsbrEnc_AddRight(v_freq, length_v_freq, v_tuningFreq[0]);
+  }
+
+  /* add attack env trailing border/first decay env leading border */
+  bord = tran;
+  FDKsbrEnc_AddRight(v_bord, length_v_bord, tran); /* v_bord = [(Ba),Bd1] */
+
+  /* add first decay env trailing border/2:nd decay env leading border */
+  if (v_tuningSegm[1]) {
+    bord += v_tuningSegm[1];
+
+    /* v_bord = [(Ba),Bd1,Bd2] */
+    FDKsbrEnc_AddRight(v_bord, length_v_bord, bord);
+
+    /* v_freq = [(Fa),Fd1] */
+    FDKsbrEnc_AddRight(v_freq, length_v_freq, v_tuningFreq[1]);
+  }
+
+  /* add 2:nd decay env trailing border (optional) */
+  if (v_tuningSegm[2] != 0) {
+    bord += v_tuningSegm[2];
+
+    /* v_bord = [(Ba),Bd1, Bd2,(Bd3)] */
+    FDKsbrEnc_AddRight(v_bord, length_v_bord, bord);
+
+    /* v_freq = [(Fa),Fd1,(Fd2)] */
+    FDKsbrEnc_AddRight(v_freq, length_v_freq, v_tuningFreq[2]);
+  }
+
+  /*  v_freq = [(Fa),Fd1,(Fd2),1] */
+  FDKsbrEnc_AddRight(v_freq, length_v_freq, 1);
+
+  /*  calc min and max values of mandatory borders */
+  *bmin = v_bord[0];
+  for (i = 0; i < *length_v_bord; i++)
+    if (v_bord[i] < *bmin) *bmin = v_bord[i];
+
+  *bmax = v_bord[0];
+  for (i = 0; i < *length_v_bord; i++)
+    if (v_bord[i] > *bmax) *bmax = v_bord[i];
+}
+
+/*******************************************************************************
+ Functionname:  fillFramePre
+ *******************************************************************************
+
+ Description: Add borders before mandatory borders, if needed
+
+ Arguments:
+       modified:
+              v_bord        - int pointer to v_bord vector
+              length_v_bord - length of v_bord vector
+              v_freq        - int pointer to v_freq vector
+              length_v_freq - length of v_freq vector
+       not modified:
+              dmax          - int value
+              bmin          - int value
+              rest          - int value
+
+ Return:      none
+
+*******************************************************************************/
+static void fillFramePre(INT dmax, INT *v_bord, INT *length_v_bord, INT *v_freq,
+                         INT *length_v_freq, INT bmin, INT rest) {
+  /*
+    input state:
+    v_bord = [(Ba),Bd1, Bd2 ,(Bd3)]
+    v_freq = [(Fa),Fd1,(Fd2),1 ]
+  */
+
+  INT parts, d, j, S, s = 0, segm, bord;
+
+  /*
+    start with one envelope
+  */
+
+  parts = 1;
+  d = rest;
+
+  /*
+    calc # of additional envelopes and corresponding lengths
+  */
+
+  while (d > dmax) {
+    parts++;
+
+    segm = rest / parts;
+    S = (segm - 2) >> 1;
+    s = fixMin(8, 2 * S + 2);
+    d = rest - (parts - 1) * s;
+  }
+
+  /*
+    add borders before mandatory borders
+  */
+
+  bord = bmin;
+
+  for (j = 0; j <= parts - 2; j++) {
+    bord = bord - s;
+
+    /* v_bord = [...,(Bf),(Ba),Bd1, Bd2 ,(Bd3)] */
+    FDKsbrEnc_AddLeft(v_bord, length_v_bord, bord);
+
+    /* v_freq = [...,(1 ),(Fa),Fd1,(Fd2), 1   ] */
+    FDKsbrEnc_AddLeft(v_freq, length_v_freq, 1);
+  }
+}
+
+/***************************************************************************/
+/*!
+  \brief Overlap control
+
+  Calculate max length of trailing fill segments, such that we always get a
+  border within the frame overlap region
+
+  \return void
+
+****************************************************************************/
+static int calcFillLengthMax(
+    int tranPos,        /*!< input : transient position (ref: tran det) */
+    int numberTimeSlots /*!< input : number of timeslots */
+) {
+  int fmax;
+
+  /*
+    calculate transient position within envelope buffer
+  */
+  switch (numberTimeSlots) {
+    case NUMBER_TIME_SLOTS_2048:
+      if (tranPos < 4)
+        fmax = 6;
+      else if (tranPos == 4 || tranPos == 5)
+        fmax = 4;
+      else
+        fmax = 8;
+      break;
+
+    case NUMBER_TIME_SLOTS_1920:
+      if (tranPos < 4)
+        fmax = 5;
+      else if (tranPos == 4 || tranPos == 5)
+        fmax = 3;
+      else
+        fmax = 7;
+      break;
+
+    default:
+      fmax = 8;
+      break;
+  }
+
+  return fmax;
+}
+
+/*******************************************************************************
+ Functionname:  fillFramePost
+ *******************************************************************************
+
+ Description: -Add borders after mandatory borders, if needed
+               Make a preliminary design of next frame,
+               assuming no transient is present there
+
+ Arguments:
+       modified:
+              parts         - int pointer to parts (call by reference)
+              d             - int pointer to d (call by reference)
+              v_bord        - int pointer to v_bord vector
+              length_v_bord - length of v_bord vector
+              v_freq        - int pointer to v_freq vector
+              length_v_freq - length of v_freq vector
+        not modified:
+              bmax          - int value
+              dmax          - int value
+
+ Return:      none
+
+*******************************************************************************/
+static void fillFramePost(INT *parts, INT *d, INT dmax, INT *v_bord,
+                          INT *length_v_bord, INT *v_freq, INT *length_v_freq,
+                          INT bmax, INT bufferFrameStart, INT numberTimeSlots,
+                          INT fmax) {
+  INT j, rest, segm, S, s = 0, bord;
+
+  /*
+    input state:
+    v_bord = [...,(Bf),(Ba),Bd1, Bd2 ,(Bd3)]
+    v_freq = [...,(1 ),(Fa),Fd1,(Fd2),1    ]
+  */
+
+  rest = bufferFrameStart + 2 * numberTimeSlots - bmax;
+  *d = rest;
+
+  if (*d > 0) {
+    *parts = 1; /* start with one envelope */
+
+    /* calc # of additional envelopes and corresponding lengths */
+
+    while (*d > dmax) {
+      *parts = *parts + 1;
+
+      segm = rest / (*parts);
+      S = (segm - 2) >> 1;
+      s = fixMin(fmax, 2 * S + 2);
+      *d = rest - (*parts - 1) * s;
+    }
+
+    /* add borders after mandatory borders */
+
+    bord = bmax;
+    for (j = 0; j <= *parts - 2; j++) {
+      bord += s;
+
+      /* v_bord =  [...,(Bf),(Ba),Bd1, Bd2 ,(Bd3),(Bf)] */
+      FDKsbrEnc_AddRight(v_bord, length_v_bord, bord);
+
+      /* v_freq =  [...,(1 ),(Fa),Fd1,(Fd2), 1   , 1! ,1] */
+      FDKsbrEnc_AddRight(v_freq, length_v_freq, 1);
+    }
+  } else {
+    *parts = 1;
+
+    /* remove last element from v_bord and v_freq */
+
+    *length_v_bord = *length_v_bord - 1;
+    *length_v_freq = *length_v_freq - 1;
+  }
+}
+
+/*******************************************************************************
+ Functionname:  fillFrameInter
+ *******************************************************************************
+
+ Description:
+
+ Arguments:   nL                  -
+              v_tuningSegm        -
+              v_bord              -
+              length_v_bord       -
+              bmin                -
+              v_freq              -
+              length_v_freq       -
+              v_bordFollow        -
+              length_v_bordFollow -
+              v_freqFollow        -
+              length_v_freqFollow -
+              i_fillFollow        -
+              dmin                -
+              dmax                -
+
+ Return:      none
+
+*******************************************************************************/
+static void fillFrameInter(INT *nL, const int *v_tuningSegm, INT *v_bord,
+                           INT *length_v_bord, INT bmin, INT *v_freq,
+                           INT *length_v_freq, INT *v_bordFollow,
+                           INT *length_v_bordFollow, INT *v_freqFollow,
+                           INT *length_v_freqFollow, INT i_fillFollow, INT dmin,
+                           INT dmax, INT numberTimeSlots) {
+  INT middle, b_new, numBordFollow, bordMaxFollow, i;
+
+  if (numberTimeSlots != NUMBER_TIME_SLOTS_1152) {
+    /* % remove fill borders: */
+    if (i_fillFollow >= 1) {
+      *length_v_bordFollow = i_fillFollow;
+      *length_v_freqFollow = i_fillFollow;
+    }
+
+    numBordFollow = *length_v_bordFollow;
+    bordMaxFollow = v_bordFollow[numBordFollow - 1];
+
+    /* remove even more borders if needed */
+    middle = bmin - bordMaxFollow;
+    while (middle < 0) {
+      numBordFollow--;
+      bordMaxFollow = v_bordFollow[numBordFollow - 1];
+      middle = bmin - bordMaxFollow;
+    }
+
+    *length_v_bordFollow = numBordFollow;
+    *length_v_freqFollow = numBordFollow;
+    *nL = numBordFollow - 1;
+
+    b_new = *length_v_bord;
+
+    if (middle <= dmax) {
+      if (middle >= dmin) { /* concatenate */
+        FDKsbrEnc_AddVecLeft(v_bord, length_v_bord, v_bordFollow,
+                             *length_v_bordFollow);
+        FDKsbrEnc_AddVecLeft(v_freq, length_v_freq, v_freqFollow,
+                             *length_v_freqFollow);
+      }
+
+      else {
+        if (v_tuningSegm[0] != 0) { /* remove one new border and concatenate */
+          *length_v_bord = b_new - 1;
+          FDKsbrEnc_AddVecLeft(v_bord, length_v_bord, v_bordFollow,
+                               *length_v_bordFollow);
+
+          *length_v_freq = b_new - 1;
+          FDKsbrEnc_AddVecLeft(v_freq + 1, length_v_freq, v_freqFollow,
+                               *length_v_freqFollow);
+        } else {
+          if (*length_v_bordFollow >
+              1) { /* remove one old border and concatenate */
+            FDKsbrEnc_AddVecLeft(v_bord, length_v_bord, v_bordFollow,
+                                 *length_v_bordFollow - 1);
+            FDKsbrEnc_AddVecLeft(v_freq, length_v_freq, v_freqFollow,
+                                 *length_v_bordFollow - 1);
+
+            *nL = *nL - 1;
+          } else { /* remove new "transient" border and concatenate */
+
+            for (i = 0; i < *length_v_bord - 1; i++) v_bord[i] = v_bord[i + 1];
+
+            for (i = 0; i < *length_v_freq - 1; i++) v_freq[i] = v_freq[i + 1];
+
+            *length_v_bord = b_new - 1;
+            *length_v_freq = b_new - 1;
+
+            FDKsbrEnc_AddVecLeft(v_bord, length_v_bord, v_bordFollow,
+                                 *length_v_bordFollow);
+            FDKsbrEnc_AddVecLeft(v_freq, length_v_freq, v_freqFollow,
+                                 *length_v_freqFollow);
+          }
+        }
+      }
+    } else { /* middle > dmax */
+
+      fillFramePre(dmax, v_bord, length_v_bord, v_freq, length_v_freq, bmin,
+                   middle);
+      FDKsbrEnc_AddVecLeft(v_bord, length_v_bord, v_bordFollow,
+                           *length_v_bordFollow);
+      FDKsbrEnc_AddVecLeft(v_freq, length_v_freq, v_freqFollow,
+                           *length_v_freqFollow);
+    }
+
+  } else { /* numberTimeSlots==NUMBER_TIME_SLOTS_1152 */
+
+    INT l, m;
+
+    /*------------------------------------------------------------------------
+      remove fill borders
+      ------------------------------------------------------------------------*/
+    if (i_fillFollow >= 1) {
+      *length_v_bordFollow = i_fillFollow;
+      *length_v_freqFollow = i_fillFollow;
+    }
+
+    numBordFollow = *length_v_bordFollow;
+    bordMaxFollow = v_bordFollow[numBordFollow - 1];
+
+    /*------------------------------------------------------------------------
+      remove more borders if necessary to eliminate overlap
+      ------------------------------------------------------------------------*/
+
+    /* check for overlap */
+    middle = bmin - bordMaxFollow;
+
+    /* intervals:
+       i)             middle <  0     : overlap, must remove borders
+       ii)       0 <= middle <  dmin  : no overlap but too tight, must remove
+       borders iii)   dmin <= middle <= dmax  : ok, just concatenate iv)    dmax
+       <= middle          : too wide, must add borders
+     */
+
+    /* first remove old non-fill-borders... */
+    while (middle < 0) {
+      /* ...but don't remove all of them */
+      if (numBordFollow == 1) break;
+
+      numBordFollow--;
+      bordMaxFollow = v_bordFollow[numBordFollow - 1];
+      middle = bmin - bordMaxFollow;
+    }
+
+    /* if this isn't enough, remove new non-fill borders */
+    if (middle < 0) {
+      for (l = 0, m = 0; l < *length_v_bord; l++) {
+        if (v_bord[l] > bordMaxFollow) {
+          v_bord[m] = v_bord[l];
+          v_freq[m] = v_freq[l];
+          m++;
+        }
+      }
+
+      *length_v_bord = l;
+      *length_v_freq = l;
+
+      bmin = v_bord[0];
+    }
+
+    /*------------------------------------------------------------------------
+      update modified follow-up data
+      ------------------------------------------------------------------------*/
+
+    *length_v_bordFollow = numBordFollow;
+    *length_v_freqFollow = numBordFollow;
+
+    /* left relative borders correspond to follow-up */
+    *nL = numBordFollow - 1;
+
+    /*------------------------------------------------------------------------
+      take care of intervals ii through iv
+      ------------------------------------------------------------------------*/
+
+    /* now middle should be >= 0 */
+    middle = bmin - bordMaxFollow;
+
+    if (middle <= dmin) /* (ii) */
+    {
+      b_new = *length_v_bord;
+
+      if (v_tuningSegm[0] != 0) {
+        /* remove new "luxury" border and concatenate */
+        *length_v_bord = b_new - 1;
+        FDKsbrEnc_AddVecLeft(v_bord, length_v_bord, v_bordFollow,
+                             *length_v_bordFollow);
+
+        *length_v_freq = b_new - 1;
+        FDKsbrEnc_AddVecLeft(v_freq + 1, length_v_freq, v_freqFollow,
+                             *length_v_freqFollow);
+
+      } else if (*length_v_bordFollow > 1) {
+        /* remove old border and concatenate */
+        FDKsbrEnc_AddVecLeft(v_bord, length_v_bord, v_bordFollow,
+                             *length_v_bordFollow - 1);
+        FDKsbrEnc_AddVecLeft(v_freq, length_v_freq, v_freqFollow,
+                             *length_v_bordFollow - 1);
+
+        *nL = *nL - 1;
+      } else {
+        /* remove new border and concatenate */
+        for (i = 0; i < *length_v_bord - 1; i++) v_bord[i] = v_bord[i + 1];
+
+        for (i = 0; i < *length_v_freq - 1; i++) v_freq[i] = v_freq[i + 1];
+
+        *length_v_bord = b_new - 1;
+        *length_v_freq = b_new - 1;
+
+        FDKsbrEnc_AddVecLeft(v_bord, length_v_bord, v_bordFollow,
+                             *length_v_bordFollow);
+        FDKsbrEnc_AddVecLeft(v_freq, length_v_freq, v_freqFollow,
+                             *length_v_freqFollow);
+      }
+    } else if ((middle >= dmin) && (middle <= dmax)) /* (iii) */
+    {
+      /* concatenate */
+      FDKsbrEnc_AddVecLeft(v_bord, length_v_bord, v_bordFollow,
+                           *length_v_bordFollow);
+      FDKsbrEnc_AddVecLeft(v_freq, length_v_freq, v_freqFollow,
+                           *length_v_freqFollow);
+
+    } else /* (iv) */
+    {
+      fillFramePre(dmax, v_bord, length_v_bord, v_freq, length_v_freq, bmin,
+                   middle);
+      FDKsbrEnc_AddVecLeft(v_bord, length_v_bord, v_bordFollow,
+                           *length_v_bordFollow);
+      FDKsbrEnc_AddVecLeft(v_freq, length_v_freq, v_freqFollow,
+                           *length_v_freqFollow);
+    }
+  }
+}
+
+/*******************************************************************************
+ Functionname:  calcFrameClass
+ *******************************************************************************
+
+ Description:
+
+ Arguments:  INT* frameClass, INT* frameClassOld, INT tranFlag, INT* spreadFlag)
+
+ Return:      none
+
+*******************************************************************************/
+static void calcFrameClass(FRAME_CLASS *frameClass, FRAME_CLASS *frameClassOld,
+                           INT tranFlag, INT *spreadFlag) {
+  switch (*frameClassOld) {
+    case FIXFIXonly:
+    case FIXFIX:
+      if (tranFlag)
+        *frameClass = FIXVAR;
+      else
+        *frameClass = FIXFIX;
+      break;
+    case FIXVAR:
+      if (tranFlag) {
+        *frameClass = VARVAR;
+        *spreadFlag = 0;
+      } else {
+        if (*spreadFlag)
+          *frameClass = VARVAR;
+        else
+          *frameClass = VARFIX;
+      }
+      break;
+    case VARFIX:
+      if (tranFlag)
+        *frameClass = FIXVAR;
+      else
+        *frameClass = FIXFIX;
+      break;
+    case VARVAR:
+      if (tranFlag) {
+        *frameClass = VARVAR;
+        *spreadFlag = 0;
+      } else {
+        if (*spreadFlag)
+          *frameClass = VARVAR;
+        else
+          *frameClass = VARFIX;
+      }
+      break;
+  };
+
+  *frameClassOld = *frameClass;
+}
+
+/*******************************************************************************
+ Functionname:  specialCase
+ *******************************************************************************
+
+ Description:
+
+ Arguments:   spreadFlag
+              allowSpread
+              v_bord
+              length_v_bord
+              v_freq
+              length_v_freq
+              parts
+              d
+
+ Return:      none
+
+*******************************************************************************/
+static void specialCase(INT *spreadFlag, INT allowSpread, INT *v_bord,
+                        INT *length_v_bord, INT *v_freq, INT *length_v_freq,
+                        INT *parts, INT d) {
+  INT L;
+
+  L = *length_v_bord;
+
+  if (allowSpread) { /* add one "step 8" */
+    *spreadFlag = 1;
+    FDKsbrEnc_AddRight(v_bord, length_v_bord, v_bord[L - 1] + 8);
+    FDKsbrEnc_AddRight(v_freq, length_v_freq, 1);
+    (*parts)++;
+  } else {
+    if (d == 1) { /*  stretch one slot */
+      *length_v_bord = L - 1;
+      *length_v_freq = L - 1;
+    } else {
+      if ((v_bord[L - 1] - v_bord[L - 2]) > 2) { /* compress one quant step */
+        v_bord[L - 1] = v_bord[L - 1] - 2;
+        v_freq[*length_v_freq - 1] = 0; /* use low res for short segment */
+      }
+    }
+  }
+}
+
+/*******************************************************************************
+ Functionname:  calcCmonBorder
+ *******************************************************************************
+
+ Description:
+
+ Arguments:   i_cmon
+              i_tran
+              v_bord
+              length_v_bord
+              tran
+
+ Return:      none
+
+*******************************************************************************/
+static void calcCmonBorder(INT *i_cmon, INT *i_tran, INT *v_bord,
+                           INT *length_v_bord, INT tran, INT bufferFrameStart,
+                           INT numberTimeSlots) { /* FH 00-06-26 */
+  INT i;
+
+  for (i = 0; i < *length_v_bord; i++)
+    if (v_bord[i] >= bufferFrameStart + numberTimeSlots) { /* FH 00-06-26 */
+      *i_cmon = i;
+      break;
+    }
+
+  /* keep track of transient: */
+  for (i = 0; i < *length_v_bord; i++)
+    if (v_bord[i] >= tran) {
+      *i_tran = i;
+      break;
+    } else
+      *i_tran = EMPTY;
+}
+
+/*******************************************************************************
+ Functionname:  keepForFollowUp
+ *******************************************************************************
+
+ Description:
+
+ Arguments:   v_bordFollow
+              length_v_bordFollow
+              v_freqFollow
+              length_v_freqFollow
+              i_tranFollow
+              i_fillFollow
+              v_bord
+              length_v_bord
+              v_freq
+              i_cmon
+              i_tran
+              parts)
+
+ Return:      none
+
+*******************************************************************************/
+static void keepForFollowUp(INT *v_bordFollow, INT *length_v_bordFollow,
+                            INT *v_freqFollow, INT *length_v_freqFollow,
+                            INT *i_tranFollow, INT *i_fillFollow, INT *v_bord,
+                            INT *length_v_bord, INT *v_freq, INT i_cmon,
+                            INT i_tran, INT parts,
+                            INT numberTimeSlots) { /* FH 00-06-26 */
+  INT L, i, j;
+
+  L = *length_v_bord;
+
+  (*length_v_bordFollow) = 0;
+  (*length_v_freqFollow) = 0;
+
+  for (j = 0, i = i_cmon; i < L; i++, j++) {
+    v_bordFollow[j] = v_bord[i] - numberTimeSlots; /* FH 00-06-26 */
+    v_freqFollow[j] = v_freq[i];
+    (*length_v_bordFollow)++;
+    (*length_v_freqFollow)++;
+  }
+  if (i_tran != EMPTY)
+    *i_tranFollow = i_tran - i_cmon;
+  else
+    *i_tranFollow = EMPTY;
+  *i_fillFollow = L - (parts - 1) - i_cmon;
+}
+
+/*******************************************************************************
+ Functionname:  calcCtrlSignal
+ *******************************************************************************
+
+ Description:
+
+ Arguments:   hSbrGrid
+              frameClass
+              v_bord
+              length_v_bord
+              v_freq
+              length_v_freq
+              i_cmon
+              i_tran
+              spreadFlag
+              nL
+
+ Return:      none
+
+*******************************************************************************/
+static void calcCtrlSignal(HANDLE_SBR_GRID hSbrGrid, FRAME_CLASS frameClass,
+                           INT *v_bord, INT length_v_bord, INT *v_freq,
+                           INT length_v_freq, INT i_cmon, INT i_tran,
+                           INT spreadFlag, INT nL) {
+  INT i, r, a, n, p, b, aL, aR, ntot, nmax, nR;
+
+  INT *v_f = hSbrGrid->v_f;
+  INT *v_fLR = hSbrGrid->v_fLR;
+  INT *v_r = hSbrGrid->bs_rel_bord;
+  INT *v_rL = hSbrGrid->bs_rel_bord_0;
+  INT *v_rR = hSbrGrid->bs_rel_bord_1;
+
+  INT length_v_r = 0;
+  INT length_v_rR = 0;
+  INT length_v_rL = 0;
+
+  switch (frameClass) {
+    case FIXVAR:
+      /* absolute border: */
+
+      a = v_bord[i_cmon];
+
+      /* relative borders: */
+      length_v_r = 0;
+      i = i_cmon;
+
+      while (i >= 1) {
+        r = v_bord[i] - v_bord[i - 1];
+        FDKsbrEnc_AddRight(v_r, &length_v_r, r);
+        i--;
+      }
+
+      /*  number of relative borders: */
+      n = length_v_r;
+
+      /* freq res: */
+      for (i = 0; i < i_cmon; i++) v_f[i] = v_freq[i_cmon - 1 - i];
+      v_f[i_cmon] = 1;
+
+      /* pointer: */
+      p = (i_cmon >= i_tran && i_tran != EMPTY) ? (i_cmon - i_tran + 1) : (0);
+
+      hSbrGrid->frameClass = frameClass;
+      hSbrGrid->bs_abs_bord = a;
+      hSbrGrid->n = n;
+      hSbrGrid->p = p;
+
+      break;
+    case VARFIX:
+      /* absolute border: */
+      a = v_bord[0];
+
+      /* relative borders: */
+      length_v_r = 0;
+
+      for (i = 1; i < length_v_bord; i++) {
+        r = v_bord[i] - v_bord[i - 1];
+        FDKsbrEnc_AddRight(v_r, &length_v_r, r);
+      }
+
+      /* number of relative borders: */
+      n = length_v_r;
+
+      /* freq res: */
+      FDKmemcpy(v_f, v_freq, length_v_freq * sizeof(INT));
+
+      /* pointer: */
+      p = (i_tran >= 0 && i_tran != EMPTY) ? (i_tran + 1) : (0);
+
+      hSbrGrid->frameClass = frameClass;
+      hSbrGrid->bs_abs_bord = a;
+      hSbrGrid->n = n;
+      hSbrGrid->p = p;
+
+      break;
+    case VARVAR:
+      if (spreadFlag) {
+        /* absolute borders: */
+        b = length_v_bord;
+
+        aL = v_bord[0];
+        aR = v_bord[b - 1];
+
+        /* number of relative borders:    */
+        ntot = b - 2;
+
+        nmax = 2; /* n: {0,1,2} */
+        if (ntot > nmax) {
+          nL = nmax;
+          nR = ntot - nmax;
+        } else {
+          nL = ntot;
+          nR = 0;
+        }
+
+        /* relative borders: */
+        length_v_rL = 0;
+        for (i = 1; i <= nL; i++) {
+          r = v_bord[i] - v_bord[i - 1];
+          FDKsbrEnc_AddRight(v_rL, &length_v_rL, r);
+        }
+
+        length_v_rR = 0;
+        i = b - 1;
+        while (i >= b - nR) {
+          r = v_bord[i] - v_bord[i - 1];
+          FDKsbrEnc_AddRight(v_rR, &length_v_rR, r);
+          i--;
+        }
+
+        /* pointer (only one due to constraint in frame info): */
+        p = (i_tran > 0 && i_tran != EMPTY) ? (b - i_tran) : (0);
+
+        /* freq res: */
+
+        for (i = 0; i < b - 1; i++) v_fLR[i] = v_freq[i];
+      } else {
+        length_v_bord = i_cmon + 1;
+
+        /* absolute borders: */
+        b = length_v_bord;
+
+        aL = v_bord[0];
+        aR = v_bord[b - 1];
+
+        /* number of relative borders:   */
+        ntot = b - 2;
+        nR = ntot - nL;
+
+        /* relative borders: */
+        length_v_rL = 0;
+        for (i = 1; i <= nL; i++) {
+          r = v_bord[i] - v_bord[i - 1];
+          FDKsbrEnc_AddRight(v_rL, &length_v_rL, r);
+        }
+
+        length_v_rR = 0;
+        i = b - 1;
+        while (i >= b - nR) {
+          r = v_bord[i] - v_bord[i - 1];
+          FDKsbrEnc_AddRight(v_rR, &length_v_rR, r);
+          i--;
+        }
+
+        /* pointer (only one due to constraint in frame info): */
+        p = (i_cmon >= i_tran && i_tran != EMPTY) ? (i_cmon - i_tran + 1) : (0);
+
+        /* freq res: */
+        for (i = 0; i < b - 1; i++) v_fLR[i] = v_freq[i];
+      }
+
+      hSbrGrid->frameClass = frameClass;
+      hSbrGrid->bs_abs_bord_0 = aL;
+      hSbrGrid->bs_abs_bord_1 = aR;
+      hSbrGrid->bs_num_rel_0 = nL;
+      hSbrGrid->bs_num_rel_1 = nR;
+      hSbrGrid->p = p;
+
+      break;
+
+    default:
+      /* do nothing */
+      break;
+  }
+}
+
+/*******************************************************************************
+ Functionname:  createDefFrameInfo
+ *******************************************************************************
+
+ Description: Copies the default (static) frameInfo structs to the frameInfo
+              passed by reference; only used for FIXFIX frames
+
+ Arguments:   hFrameInfo             - HANLDE_SBR_FRAME_INFO
+              nEnv                   - INT
+              nTimeSlots             - INT
+
+ Return:      none; hSbrFrameInfo contains a copy of the default frameInfo
+
+ Written:     Andreas Schneider
+ Revised:
+*******************************************************************************/
+static void createDefFrameInfo(HANDLE_SBR_FRAME_INFO hSbrFrameInfo, INT nEnv,
+                               INT nTimeSlots) {
+  switch (nEnv) {
+    case 1:
+      switch (nTimeSlots) {
+        case NUMBER_TIME_SLOTS_1920:
+          FDKmemcpy(hSbrFrameInfo, &frameInfo1_1920, sizeof(SBR_FRAME_INFO));
+          break;
+        case NUMBER_TIME_SLOTS_2048:
+          FDKmemcpy(hSbrFrameInfo, &frameInfo1_2048, sizeof(SBR_FRAME_INFO));
+          break;
+        case NUMBER_TIME_SLOTS_1152:
+          FDKmemcpy(hSbrFrameInfo, &frameInfo1_1152, sizeof(SBR_FRAME_INFO));
+          break;
+        case NUMBER_TIME_SLOTS_2304:
+          FDKmemcpy(hSbrFrameInfo, &frameInfo1_2304, sizeof(SBR_FRAME_INFO));
+          break;
+        case NUMBER_TIME_SLOTS_512LD:
+          FDKmemcpy(hSbrFrameInfo, &frameInfo1_512LD, sizeof(SBR_FRAME_INFO));
+          break;
+        default:
+          FDK_ASSERT(0);
+      }
+      break;
+    case 2:
+      switch (nTimeSlots) {
+        case NUMBER_TIME_SLOTS_1920:
+          FDKmemcpy(hSbrFrameInfo, &frameInfo2_1920, sizeof(SBR_FRAME_INFO));
+          break;
+        case NUMBER_TIME_SLOTS_2048:
+          FDKmemcpy(hSbrFrameInfo, &frameInfo2_2048, sizeof(SBR_FRAME_INFO));
+          break;
+        case NUMBER_TIME_SLOTS_1152:
+          FDKmemcpy(hSbrFrameInfo, &frameInfo2_1152, sizeof(SBR_FRAME_INFO));
+          break;
+        case NUMBER_TIME_SLOTS_2304:
+          FDKmemcpy(hSbrFrameInfo, &frameInfo2_2304, sizeof(SBR_FRAME_INFO));
+          break;
+        case NUMBER_TIME_SLOTS_512LD:
+          FDKmemcpy(hSbrFrameInfo, &frameInfo2_512LD, sizeof(SBR_FRAME_INFO));
+          break;
+        default:
+          FDK_ASSERT(0);
+      }
+      break;
+    case 4:
+      switch (nTimeSlots) {
+        case NUMBER_TIME_SLOTS_1920:
+          FDKmemcpy(hSbrFrameInfo, &frameInfo4_1920, sizeof(SBR_FRAME_INFO));
+          break;
+        case NUMBER_TIME_SLOTS_2048:
+          FDKmemcpy(hSbrFrameInfo, &frameInfo4_2048, sizeof(SBR_FRAME_INFO));
+          break;
+        case NUMBER_TIME_SLOTS_1152:
+          FDKmemcpy(hSbrFrameInfo, &frameInfo4_1152, sizeof(SBR_FRAME_INFO));
+          break;
+        case NUMBER_TIME_SLOTS_2304:
+          FDKmemcpy(hSbrFrameInfo, &frameInfo4_2304, sizeof(SBR_FRAME_INFO));
+          break;
+        case NUMBER_TIME_SLOTS_512LD:
+          FDKmemcpy(hSbrFrameInfo, &frameInfo4_512LD, sizeof(SBR_FRAME_INFO));
+          break;
+        default:
+          FDK_ASSERT(0);
+      }
+      break;
+    default:
+      FDK_ASSERT(0);
+  }
+}
+
+/*******************************************************************************
+ Functionname:  ctrlSignal2FrameInfo
+ *******************************************************************************
+
+ Description: Convert "clear-text" sbr_grid() to "frame info" used by the
+              envelope and noise floor estimators.
+              This is basically (except for "low level" calculations) the
+              bitstream decoder defined in the MPEG-4 standard, sub clause
+              4.6.18.3.3, Time / Frequency Grid.  See inline comments for
+              explanation of the shorten and noise border algorithms.
+
+ Arguments:   hSbrGrid - source
+              hSbrFrameInfo - destination
+              freq_res_fixfix - frequency resolution for FIXFIX frames
+
+ Return:      void; hSbrFrameInfo contains the updated FRAME_INFO struct
+
+*******************************************************************************/
+static void ctrlSignal2FrameInfo(
+    HANDLE_SBR_GRID hSbrGrid,            /* input : the grid handle       */
+    HANDLE_SBR_FRAME_INFO hSbrFrameInfo, /* output: the frame info handle */
+    FREQ_RES
+        *freq_res_fixfix /* in/out: frequency resolution for FIXFIX frames */
+) {
+  INT frameSplit = 0;
+  INT nEnv = 0, border = 0, i, k, p /*?*/;
+  INT *v_r = hSbrGrid->bs_rel_bord;
+  INT *v_f = hSbrGrid->v_f;
+
+  FRAME_CLASS frameClass = hSbrGrid->frameClass;
+  INT bufferFrameStart = hSbrGrid->bufferFrameStart;
+  INT numberTimeSlots = hSbrGrid->numberTimeSlots;
+
+  switch (frameClass) {
+    case FIXFIX:
+      createDefFrameInfo(hSbrFrameInfo, hSbrGrid->bs_num_env, numberTimeSlots);
+
+      frameSplit = (hSbrFrameInfo->nEnvelopes > 1);
+      for (i = 0; i < hSbrFrameInfo->nEnvelopes; i++) {
+        hSbrGrid->v_f[i] = hSbrFrameInfo->freqRes[i] =
+            freq_res_fixfix[frameSplit];
+      }
+      break;
+
+    case FIXVAR:
+    case VARFIX:
+      nEnv = hSbrGrid->n + 1; /* read n [SBR_NUM_BITS bits] */ /*? snd*/
+      FDK_ASSERT(nEnv <= MAX_ENVELOPES_FIXVAR_VARFIX);
+
+      hSbrFrameInfo->nEnvelopes = nEnv;
+
+      border = hSbrGrid->bs_abs_bord; /* read the absolute border */
+
+      if (nEnv == 1)
+        hSbrFrameInfo->nNoiseEnvelopes = 1;
+      else
+        hSbrFrameInfo->nNoiseEnvelopes = 2;
+
+      break;
+
+    default:
+      /* do nothing */
+      break;
+  }
+
+  switch (frameClass) {
+    case FIXVAR:
+      hSbrFrameInfo->borders[0] =
+          bufferFrameStart; /* start-position of 1st envelope */
+
+      hSbrFrameInfo->borders[nEnv] = border;
+
+      for (k = 0, i = nEnv - 1; k < nEnv - 1; k++, i--) {
+        border -= v_r[k];
+        hSbrFrameInfo->borders[i] = border;
+      }
+
+      /* make either envelope nr. nEnv + 1 - p short; or don't shorten if p == 0
+       */
+      p = hSbrGrid->p;
+      if (p == 0) {
+        hSbrFrameInfo->shortEnv = 0;
+      } else {
+        hSbrFrameInfo->shortEnv = nEnv + 1 - p;
+      }
+
+      for (k = 0, i = nEnv - 1; k < nEnv; k++, i--) {
+        hSbrFrameInfo->freqRes[i] = (FREQ_RES)v_f[k];
+      }
+
+      /* if either there is no short envelope or the last envelope is short...
+       */
+      if (p == 0 || p == 1) {
+        hSbrFrameInfo->bordersNoise[1] = hSbrFrameInfo->borders[nEnv - 1];
+      } else {
+        hSbrFrameInfo->bordersNoise[1] =
+            hSbrFrameInfo->borders[hSbrFrameInfo->shortEnv];
+      }
+
+      break;
+
+    case VARFIX:
+      /* in this case 'border' indicates the start of the 1st envelope */
+      hSbrFrameInfo->borders[0] = border;
+
+      for (k = 0; k < nEnv - 1; k++) {
+        border += v_r[k];
+        hSbrFrameInfo->borders[k + 1] = border;
+      }
+
+      hSbrFrameInfo->borders[nEnv] = bufferFrameStart + numberTimeSlots;
+
+      p = hSbrGrid->p;
+      if (p == 0 || p == 1) {
+        hSbrFrameInfo->shortEnv = 0;
+      } else {
+        hSbrFrameInfo->shortEnv = p - 1;
+      }
+
+      for (k = 0; k < nEnv; k++) {
+        hSbrFrameInfo->freqRes[k] = (FREQ_RES)v_f[k];
+      }
+
+      switch (p) {
+        case 0:
+          hSbrFrameInfo->bordersNoise[1] = hSbrFrameInfo->borders[1];
+          break;
+        case 1:
+          hSbrFrameInfo->bordersNoise[1] = hSbrFrameInfo->borders[nEnv - 1];
+          break;
+        default:
+          hSbrFrameInfo->bordersNoise[1] =
+              hSbrFrameInfo->borders[hSbrFrameInfo->shortEnv];
+          break;
+      }
+      break;
+
+    case VARVAR:
+      nEnv = hSbrGrid->bs_num_rel_0 + hSbrGrid->bs_num_rel_1 + 1;
+      FDK_ASSERT(nEnv <= MAX_ENVELOPES_VARVAR); /* just to be sure */
+      hSbrFrameInfo->nEnvelopes = nEnv;
+
+      hSbrFrameInfo->borders[0] = border = hSbrGrid->bs_abs_bord_0;
+
+      for (k = 0, i = 1; k < hSbrGrid->bs_num_rel_0; k++, i++) {
+        border += hSbrGrid->bs_rel_bord_0[k];
+        hSbrFrameInfo->borders[i] = border;
+      }
+
+      border = hSbrGrid->bs_abs_bord_1;
+      hSbrFrameInfo->borders[nEnv] = border;
+
+      for (k = 0, i = nEnv - 1; k < hSbrGrid->bs_num_rel_1; k++, i--) {
+        border -= hSbrGrid->bs_rel_bord_1[k];
+        hSbrFrameInfo->borders[i] = border;
+      }
+
+      p = hSbrGrid->p;
+      if (p == 0) {
+        hSbrFrameInfo->shortEnv = 0;
+      } else {
+        hSbrFrameInfo->shortEnv = nEnv + 1 - p;
+      }
+
+      for (k = 0; k < nEnv; k++) {
+        hSbrFrameInfo->freqRes[k] = (FREQ_RES)hSbrGrid->v_fLR[k];
+      }
+
+      if (nEnv == 1) {
+        hSbrFrameInfo->nNoiseEnvelopes = 1;
+        hSbrFrameInfo->bordersNoise[0] = hSbrGrid->bs_abs_bord_0;
+        hSbrFrameInfo->bordersNoise[1] = hSbrGrid->bs_abs_bord_1;
+      } else {
+        hSbrFrameInfo->nNoiseEnvelopes = 2;
+        hSbrFrameInfo->bordersNoise[0] = hSbrGrid->bs_abs_bord_0;
+
+        if (p == 0 || p == 1) {
+          hSbrFrameInfo->bordersNoise[1] = hSbrFrameInfo->borders[nEnv - 1];
+        } else {
+          hSbrFrameInfo->bordersNoise[1] =
+              hSbrFrameInfo->borders[hSbrFrameInfo->shortEnv];
+        }
+        hSbrFrameInfo->bordersNoise[2] = hSbrGrid->bs_abs_bord_1;
+      }
+      break;
+
+    default:
+      /* do nothing */
+      break;
+  }
+
+  if (frameClass == VARFIX || frameClass == FIXVAR) {
+    hSbrFrameInfo->bordersNoise[0] = hSbrFrameInfo->borders[0];
+    if (nEnv == 1) {
+      hSbrFrameInfo->bordersNoise[1] = hSbrFrameInfo->borders[nEnv];
+    } else {
+      hSbrFrameInfo->bordersNoise[2] = hSbrFrameInfo->borders[nEnv];
+    }
+  }
+}
diff --git a/fdk-aac/libSBRenc/src/fram_gen.h b/fdk-aac/libSBRenc/src/fram_gen.h
new file mode 100644
index 0000000..0c5edc3
--- /dev/null
+++ b/fdk-aac/libSBRenc/src/fram_gen.h
@@ -0,0 +1,343 @@
+/* -----------------------------------------------------------------------------
+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 encoder library ******************************
+
+   Author(s):
+
+   Description:
+
+*******************************************************************************/
+
+/*!
+  \file
+  \brief  Framing generator prototypes and structs $Revision: 92790 $
+*/
+#ifndef FRAM_GEN_H
+#define FRAM_GEN_H
+
+#include "sbr_def.h" /* for MAX_ENVELOPES and MAX_NOISE_ENVELOPES in struct FRAME_INFO and CODEC_TYPE */
+#include "sbr_encoder.h" /* for FREQ_RES */
+
+#define MAX_ENVELOPES_VARVAR \
+  MAX_ENVELOPES /*!< worst case number of envelopes in a VARVAR frame */
+#define MAX_ENVELOPES_FIXVAR_VARFIX \
+  4 /*!< worst case number of envelopes in VARFIX and FIXVAR frames */
+#define MAX_NUM_REL \
+  3 /*!< maximum number of relative borders in any VAR frame */
+
+/* SBR frame class definitions */
+typedef enum {
+  FIXFIX =
+      0,  /*!< bs_frame_class: leading and trailing frame borders are fixed */
+  FIXVAR, /*!< bs_frame_class: leading frame border is fixed, trailing frame
+             border is variable */
+  VARFIX, /*!< bs_frame_class: leading frame border is variable, trailing frame
+             border is fixed */
+  VARVAR /*!< bs_frame_class: leading and trailing frame borders are variable */
+  ,
+  FIXFIXonly /*!< bs_frame_class: leading border fixed (0), trailing border
+                fixed (nrTimeSlots) and encased borders are dynamically derived
+                from the tranPos */
+} FRAME_CLASS;
+
+/* helper constants */
+#define DC 4711     /*!< helper constant: don't care */
+#define EMPTY (-99) /*!< helper constant: empty */
+
+/* system constants: AAC+SBR, DRM Frame-Length */
+#define FRAME_MIDDLE_SLOT_1920 4
+#define NUMBER_TIME_SLOTS_1920 15
+
+#define LD_PRETRAN_OFF 3
+#define FRAME_MIDDLE_SLOT_512LD 4
+#define NUMBER_TIME_SLOTS_512LD 8
+#define TRANSIENT_OFFSET_LD 0
+
+/*
+system constants: AAC+SBR or aacPRO (hybrid format), Standard Frame-Length,
+Multi-Rate
+---------------------------------------------------------------------------
+Number of slots (numberTimeSlots): 16  (NUMBER_TIME_SLOTS_2048)
+Detector-offset (frameMiddleSlot):  4
+Overlap                          :  3
+Buffer-offset                    :  8  (BUFFER_FRAME_START_2048 = 0)
+
+
+                        |<------------tranPos---------->|
+                |c|d|e|f|0|1|2|3|4|5|6|7|8|9|a|b|c|d|e|f|
+        FixFix  |                               |
+        FixVar  |                               :<- ->:
+        VarFix  :<- ->:                         |
+        VarVar  :<- ->:                         :<- ->:
+                0 1 2 3 4 5 6 7 8 9 a b c d e f 0 1 2 3
+................................................................................
+
+|-|-|-|-|-|-|-|-B-|-|-|-|-|-|-|-|-|-|-|-|-|-|-|-B-|-|-|-|-|-|-|-|-|-|-|-|-|-|-|-|
+
+frame-generator:0                               16              24 32
+analysis-buffer:8                               24              32 40
+*/
+#define FRAME_MIDDLE_SLOT_2048 4
+#define NUMBER_TIME_SLOTS_2048 16
+
+/*
+system constants: mp3PRO, Multi-Rate & Single-Rate
+--------------------------------------------------
+Number of slots (numberTimeSlots):  9    (NUMBER_TIME_SLOTS_1152)
+Detector-offset (frameMiddleSlot):  4    (FRAME_MIDDLE_SLOT_1152)
+Overlap                          :  3
+Buffer-offset                    :  4.5  (BUFFER_FRAME_START_1152 = 0)
+
+
+                         |<----tranPos---->|
+                 |5|6|7|8|0|1|2|3|4|5|6|7|8|
+         FixFix  |                 |
+         FixVar  |                 :<- ->:
+         VarFix  :<- ->:           |
+         VarVar  :<- ->:           :<- ->:
+                 0 1 2 3 4 5 6 7 8 0 1 2 3
+        .............................................
+
+        -|-|-|-|-B-|-|-|-|-|-|-|-|-B-|-|-|-|-|-|-|-|-|
+
+frame-generator: 0                 9       13        18
+analysis-buffer: 4.5               13.5              22.5
+*/
+#define FRAME_MIDDLE_SLOT_1152 4
+#define NUMBER_TIME_SLOTS_1152 9
+
+/* system constants: Layer2+SBR */
+#define FRAME_MIDDLE_SLOT_2304 8
+#define NUMBER_TIME_SLOTS_2304 18
+
+/*!
+  \struct SBR_GRID
+  \brief  sbr_grid() signals to be converted to bitstream elements
+
+  The variables hold the signals (e.g. lengths and numbers) in "clear text"
+*/
+
+typedef struct {
+  /* system constants */
+  INT bufferFrameStart; /*!< frame generator vs analysis buffer time alignment
+                           (currently set to 0, offset added elsewhere) */
+  INT numberTimeSlots;  /*!< number of SBR timeslots per frame */
+
+  /* will be adjusted for every frame */
+  FRAME_CLASS frameClass; /*!< SBR frame class  */
+  INT bs_num_env;         /*!< bs_num_env, number of envelopes for FIXFIX */
+  INT bs_abs_bord; /*!< bs_abs_bord, absolute border for VARFIX and FIXVAR */
+  INT n;           /*!< number of relative borders for VARFIX and FIXVAR   */
+  INT p;           /*!< pointer-to-transient-border  */
+  INT bs_rel_bord[MAX_NUM_REL]; /*!< bs_rel_bord, relative borders for all VAR
+                                 */
+  INT v_f[MAX_ENVELOPES_FIXVAR_VARFIX]; /*!< envelope frequency resolutions for
+                                           FIXVAR and VARFIX  */
+
+  INT bs_abs_bord_0; /*!< bs_abs_bord_0, leading absolute border for VARVAR */
+  INT bs_abs_bord_1; /*!< bs_abs_bord_1, trailing absolute border for VARVAR */
+  INT bs_num_rel_0;  /*!< bs_num_rel_0, number of relative borders associated
+                        with leading absolute border for VARVAR */
+  INT bs_num_rel_1;  /*!< bs_num_rel_1, number of relative borders associated
+                        with trailing absolute border for VARVAR */
+  INT bs_rel_bord_0[MAX_NUM_REL];  /*!< bs_rel_bord_0, relative borders
+                                      associated with  leading absolute border
+                                      for  VARVAR */
+  INT bs_rel_bord_1[MAX_NUM_REL];  /*!< bs_rel_bord_1, relative borders
+                                      associated with trailing absolute border
+                                      for VARVAR */
+  INT v_fLR[MAX_ENVELOPES_VARVAR]; /*!< envelope frequency resolutions for
+                                      VARVAR  */
+
+} SBR_GRID;
+typedef SBR_GRID *HANDLE_SBR_GRID;
+
+/*!
+  \struct SBR_FRAME_INFO
+  \brief  time/frequency grid description for one frame
+*/
+typedef struct {
+  INT nEnvelopes;                  /*!< number of envelopes */
+  INT borders[MAX_ENVELOPES + 1];  /*!< envelope borders in SBR timeslots */
+  FREQ_RES freqRes[MAX_ENVELOPES]; /*!< frequency resolution of each envelope */
+  INT shortEnv; /*!< number of an envelope to be shortened (starting at 1) or 0
+                   for no shortened envelope */
+  INT nNoiseEnvelopes; /*!< number of noise floors */
+  INT bordersNoise[MAX_NOISE_ENVELOPES +
+                   1]; /*!< noise floor borders in SBR timeslots */
+} SBR_FRAME_INFO;
+/* WARNING: When rearranging the elements of this struct keep in mind that the
+ * static initializations in the corresponding C-file have to be rearranged as
+ * well! snd 2002/01/23
+ */
+typedef SBR_FRAME_INFO *HANDLE_SBR_FRAME_INFO;
+
+/*!
+  \struct SBR_ENVELOPE_FRAME
+  \brief  frame generator main struct
+
+  Contains tuning parameters, time/frequency grid description, sbr_grid()
+  bitstream elements, and generator internal signals
+*/
+typedef struct {
+  /* system constants */
+  INT frameMiddleSlot; /*!< transient detector offset in SBR timeslots */
+
+  /* basic tuning parameters */
+  INT staticFraming; /*!< 1: run static framing in time, i.e. exclusive use of
+                        bs_frame_class = FIXFIX */
+  INT numEnvStatic;  /*!< number of envelopes per frame for static framing */
+  FREQ_RES
+  freq_res_fixfix[2]; /*!< envelope frequency resolution to use for
+                         bs_frame_class = FIXFIX; single env and split */
+  UCHAR
+  fResTransIsLow; /*!< frequency resolution for transient frames - always
+                     low (0) or according to table (1) */
+
+  /* expert tuning parameters */
+  const int *v_tuningSegm; /*!< segment lengths to use around transient */
+  const int *v_tuningFreq; /*!< frequency resolutions to use around transient */
+  INT dmin;                /*!< minimum length of dependent segments */
+  INT dmax;                /*!< maximum length of dependent segments */
+  INT allowSpread; /*!< 1: allow isolated transient to influence grid of 3
+                      consecutive frames */
+
+  /* internally used signals */
+  FRAME_CLASS frameClassOld; /*!< frame class used for previous frame */
+  INT spreadFlag; /*!< 1: use VARVAR instead of VARFIX to follow up old
+                     transient */
+
+  INT v_bord[2 * MAX_ENVELOPES_VARVAR + 1]; /*!< borders for current frame and
+                                               preliminary borders for next
+                                               frame (fixed borders excluded) */
+  INT length_v_bord; /*!< helper variable: length of v_bord */
+  INT v_freq[2 * MAX_ENVELOPES_VARVAR + 1]; /*!< frequency resolutions for
+                                               current frame and preliminary
+                                               resolutions for next frame */
+  INT length_v_freq; /*!< helper variable: length of v_freq */
+
+  INT v_bordFollow[MAX_ENVELOPES_VARVAR]; /*!< preliminary borders for current
+                                             frame (calculated during previous
+                                             frame) */
+  INT length_v_bordFollow; /*!< helper variable: length of v_bordFollow */
+  INT i_tranFollow; /*!< points to transient border in v_bordFollow (may be
+                       negative, see keepForFollowUp()) */
+  INT i_fillFollow; /*!< points to first fill border in v_bordFollow */
+  INT v_freqFollow[MAX_ENVELOPES_VARVAR]; /*!< preliminary frequency resolutions
+                                             for current frame (calculated
+                                             during previous frame) */
+  INT length_v_freqFollow; /*!< helper variable: length of v_freqFollow */
+
+  /* externally needed signals */
+  SBR_GRID
+  SbrGrid; /*!< sbr_grid() signals to be converted to bitstream elements */
+  SBR_FRAME_INFO
+  SbrFrameInfo; /*!< time/frequency grid description for one frame */
+} SBR_ENVELOPE_FRAME;
+typedef SBR_ENVELOPE_FRAME *HANDLE_SBR_ENVELOPE_FRAME;
+
+void FDKsbrEnc_initFrameInfoGenerator(HANDLE_SBR_ENVELOPE_FRAME hSbrEnvFrame,
+                                      INT allowSpread, INT numEnvStatic,
+                                      INT staticFraming, INT timeSlots,
+                                      const FREQ_RES *freq_res_fixfix,
+                                      UCHAR fResTransIsLow, INT ldGrid);
+
+HANDLE_SBR_FRAME_INFO
+FDKsbrEnc_frameInfoGenerator(HANDLE_SBR_ENVELOPE_FRAME hSbrEnvFrame,
+                             UCHAR *v_transient_info, const INT rightBorderFIX,
+                             UCHAR *v_transient_info_pre, int ldGrid,
+                             const int *v_tuning);
+
+#endif
diff --git a/fdk-aac/libSBRenc/src/invf_est.cpp b/fdk-aac/libSBRenc/src/invf_est.cpp
new file mode 100644
index 0000000..53b47ac
--- /dev/null
+++ b/fdk-aac/libSBRenc/src/invf_est.cpp
@@ -0,0 +1,610 @@
+/* -----------------------------------------------------------------------------
+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 encoder library ******************************
+
+   Author(s):
+
+   Description:
+
+*******************************************************************************/
+
+#include "invf_est.h"
+#include "sbr_misc.h"
+
+#include "genericStds.h"
+
+#define MAX_NUM_REGIONS 10
+#define SCALE_FAC_QUO 512.0f
+#define SCALE_FAC_NRG 256.0f
+
+#ifndef min
+#define min(a, b) (a < b ? a : b)
+#endif
+
+#ifndef max
+#define max(a, b) (a > b ? a : b)
+#endif
+
+static const FIXP_DBL quantStepsSbr[4] = {
+    0x00400000, 0x02800000, 0x03800000,
+    0x04c00000}; /* table scaled with SCALE_FAC_QUO */
+static const FIXP_DBL quantStepsOrig[4] = {
+    0x00000000, 0x00c00000, 0x01c00000,
+    0x02800000}; /* table scaled with SCALE_FAC_QUO */
+static const FIXP_DBL nrgBorders[4] = {
+    0x0c800000, 0x0f000000, 0x11800000,
+    0x14000000}; /* table scaled with SCALE_FAC_NRG */
+
+static const DETECTOR_PARAMETERS detectorParamsAAC = {
+    quantStepsSbr,
+    quantStepsOrig,
+    nrgBorders,
+    4, /* Number of borders SBR. */
+    4, /* Number of borders orig. */
+    4, /* Number of borders Nrg. */
+    {
+        /* Region space. */
+        {INVF_MID_LEVEL, INVF_LOW_LEVEL, INVF_OFF, INVF_OFF,
+         INVF_OFF}, /*    |      */
+        {INVF_MID_LEVEL, INVF_LOW_LEVEL, INVF_OFF, INVF_OFF,
+         INVF_OFF}, /*    |      */
+        {INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_LOW_LEVEL, INVF_OFF,
+         INVF_OFF}, /* regionSbr */
+        {INVF_HIGH_LEVEL, INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_OFF,
+         INVF_OFF}, /*    |      */
+        {INVF_HIGH_LEVEL, INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_OFF,
+         INVF_OFF} /*    |      */
+    }, /*------------------------ regionOrig ---------------------------------*/
+    {
+        /* Region space transient. */
+        {INVF_LOW_LEVEL, INVF_LOW_LEVEL, INVF_LOW_LEVEL, INVF_OFF,
+         INVF_OFF}, /*    |      */
+        {INVF_LOW_LEVEL, INVF_LOW_LEVEL, INVF_LOW_LEVEL, INVF_OFF,
+         INVF_OFF}, /*    |      */
+        {INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_MID_LEVEL, INVF_OFF,
+         INVF_OFF}, /* regionSbr */
+        {INVF_HIGH_LEVEL, INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_OFF,
+         INVF_OFF}, /*    |      */
+        {INVF_HIGH_LEVEL, INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_OFF,
+         INVF_OFF} /*    |      */
+    }, /*------------------------ regionOrig ---------------------------------*/
+    {-4, -3, -2, -1,
+     0} /* Reduction factor of the inverse filtering for low energies.*/
+};
+
+static const FIXP_DBL hysteresis =
+    0x00400000; /* Delta value for hysteresis. scaled with SCALE_FAC_QUO */
+
+/*
+ * AAC+SBR PARAMETERS for Speech
+ *********************************/
+static const DETECTOR_PARAMETERS detectorParamsAACSpeech = {
+    quantStepsSbr,
+    quantStepsOrig,
+    nrgBorders,
+    4, /* Number of borders SBR. */
+    4, /* Number of borders orig. */
+    4, /* Number of borders Nrg. */
+    {
+        /* Region space. */
+        {INVF_MID_LEVEL, INVF_MID_LEVEL, INVF_LOW_LEVEL, INVF_OFF,
+         INVF_OFF}, /*    |      */
+        {INVF_MID_LEVEL, INVF_MID_LEVEL, INVF_LOW_LEVEL, INVF_OFF,
+         INVF_OFF}, /*    |      */
+        {INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_MID_LEVEL, INVF_OFF,
+         INVF_OFF}, /* regionSbr */
+        {INVF_HIGH_LEVEL, INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_OFF,
+         INVF_OFF}, /*    |      */
+        {INVF_HIGH_LEVEL, INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_OFF,
+         INVF_OFF} /*    |      */
+    }, /*------------------------ regionOrig ---------------------------------*/
+    {
+        /* Region space transient. */
+        {INVF_MID_LEVEL, INVF_MID_LEVEL, INVF_LOW_LEVEL, INVF_OFF,
+         INVF_OFF}, /*    |      */
+        {INVF_MID_LEVEL, INVF_MID_LEVEL, INVF_LOW_LEVEL, INVF_OFF,
+         INVF_OFF}, /*    |      */
+        {INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_MID_LEVEL, INVF_OFF,
+         INVF_OFF}, /* regionSbr */
+        {INVF_HIGH_LEVEL, INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_OFF,
+         INVF_OFF}, /*    |      */
+        {INVF_HIGH_LEVEL, INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_OFF,
+         INVF_OFF} /*    |      */
+    }, /*------------------------ regionOrig ---------------------------------*/
+    {-4, -3, -2, -1,
+     0} /* Reduction factor of the inverse filtering for low energies.*/
+};
+
+/*
+ * Smoothing filters.
+ ************************/
+typedef const FIXP_DBL FIR_FILTER[5];
+
+static const FIR_FILTER fir_0 = {0x7fffffff, 0x00000000, 0x00000000, 0x00000000,
+                                 0x00000000};
+static const FIR_FILTER fir_1 = {0x2aaaaa80, 0x555554ff, 0x00000000, 0x00000000,
+                                 0x00000000};
+static const FIR_FILTER fir_2 = {0x10000000, 0x30000000, 0x40000000, 0x00000000,
+                                 0x00000000};
+static const FIR_FILTER fir_3 = {0x077f80e8, 0x199999a0, 0x2bb3b240, 0x33333340,
+                                 0x00000000};
+static const FIR_FILTER fir_4 = {0x04130598, 0x0ebdb000, 0x1becfa60, 0x2697a4c0,
+                                 0x2aaaaa80};
+
+static const FIR_FILTER *const fir_table[5] = {&fir_0, &fir_1, &fir_2, &fir_3,
+                                               &fir_4};
+
+/**************************************************************************/
+/*!
+  \brief     Calculates the values used for the detector.
+
+
+  \return    none
+
+*/
+/**************************************************************************/
+static void calculateDetectorValues(
+    FIXP_DBL **quotaMatrixOrig, /*!< Matrix holding the tonality values of the
+                                   original. */
+    SCHAR *indexVector,         /*!< Index vector to obtain the patched data. */
+    FIXP_DBL *nrgVector,        /*!< Energy vector. */
+    DETECTOR_VALUES *detectorValues, /*!< pointer to DETECTOR_VALUES struct. */
+    INT startChannel,                /*!< Start channel. */
+    INT stopChannel,                 /*!< Stop channel. */
+    INT startIndex,                  /*!< Start index. */
+    INT stopIndex,                   /*!< Stop index. */
+    INT numberOfStrongest /*!< The number of sorted tonal components to be
+                             considered. */
+) {
+  INT i, temp, j;
+
+  const FIXP_DBL *filter = *fir_table[INVF_SMOOTHING_LENGTH];
+  FIXP_DBL origQuotaMeanStrongest, sbrQuotaMeanStrongest;
+  FIXP_DBL origQuota, sbrQuota;
+  FIXP_DBL invIndex, invChannel, invTemp;
+  FIXP_DBL quotaVecOrig[64], quotaVecSbr[64];
+
+  FDKmemclear(quotaVecOrig, 64 * sizeof(FIXP_DBL));
+  FDKmemclear(quotaVecSbr, 64 * sizeof(FIXP_DBL));
+
+  invIndex = GetInvInt(stopIndex - startIndex);
+  invChannel = GetInvInt(stopChannel - startChannel);
+
+  /*
+   Calculate the mean value, over the current time segment, for the original,
+   the HFR and the difference, over all channels in the current frequency range.
+   NOTE: the averaging is done on the values quota/(1 - quota + RELAXATION).
+   */
+
+  /* The original, the sbr signal and the total energy */
+  detectorValues->avgNrg = FL2FXCONST_DBL(0.0f);
+  for (j = startIndex; j < stopIndex; j++) {
+    for (i = startChannel; i < stopChannel; i++) {
+      quotaVecOrig[i] += fMult(quotaMatrixOrig[j][i], invIndex);
+
+      if (indexVector[i] != -1)
+        quotaVecSbr[i] += fMult(quotaMatrixOrig[j][indexVector[i]], invIndex);
+    }
+    detectorValues->avgNrg += fMult(nrgVector[j], invIndex);
+  }
+
+  /*
+   Calculate the mean value, over the current frequency range, for the original,
+   the HFR and the difference. Also calculate the same mean values for the three
+   vectors, but only includeing the x strongest copmponents.
+   */
+
+  origQuota = FL2FXCONST_DBL(0.0f);
+  sbrQuota = FL2FXCONST_DBL(0.0f);
+  for (i = startChannel; i < stopChannel; i++) {
+    origQuota += fMultDiv2(quotaVecOrig[i], invChannel);
+    sbrQuota += fMultDiv2(quotaVecSbr[i], invChannel);
+  }
+
+  /*
+   Calculate the mean value for the x strongest components
+  */
+  FDKsbrEnc_Shellsort_fract(quotaVecOrig + startChannel,
+                            stopChannel - startChannel);
+  FDKsbrEnc_Shellsort_fract(quotaVecSbr + startChannel,
+                            stopChannel - startChannel);
+
+  origQuotaMeanStrongest = FL2FXCONST_DBL(0.0f);
+  sbrQuotaMeanStrongest = FL2FXCONST_DBL(0.0f);
+
+  temp = min(stopChannel - startChannel, numberOfStrongest);
+  invTemp = GetInvInt(temp);
+
+  for (i = 0; i < temp; i++) {
+    origQuotaMeanStrongest +=
+        fMultDiv2(quotaVecOrig[i + stopChannel - temp], invTemp);
+    sbrQuotaMeanStrongest +=
+        fMultDiv2(quotaVecSbr[i + stopChannel - temp], invTemp);
+  }
+
+  /*
+   The value for the strongest component
+  */
+  detectorValues->origQuotaMax = quotaVecOrig[stopChannel - 1];
+  detectorValues->sbrQuotaMax = quotaVecSbr[stopChannel - 1];
+
+  /*
+   Buffer values
+  */
+  FDKmemmove(detectorValues->origQuotaMean, detectorValues->origQuotaMean + 1,
+             INVF_SMOOTHING_LENGTH * sizeof(FIXP_DBL));
+  FDKmemmove(detectorValues->sbrQuotaMean, detectorValues->sbrQuotaMean + 1,
+             INVF_SMOOTHING_LENGTH * sizeof(FIXP_DBL));
+  FDKmemmove(detectorValues->origQuotaMeanStrongest,
+             detectorValues->origQuotaMeanStrongest + 1,
+             INVF_SMOOTHING_LENGTH * sizeof(FIXP_DBL));
+  FDKmemmove(detectorValues->sbrQuotaMeanStrongest,
+             detectorValues->sbrQuotaMeanStrongest + 1,
+             INVF_SMOOTHING_LENGTH * sizeof(FIXP_DBL));
+
+  detectorValues->origQuotaMean[INVF_SMOOTHING_LENGTH] = origQuota << 1;
+  detectorValues->sbrQuotaMean[INVF_SMOOTHING_LENGTH] = sbrQuota << 1;
+  detectorValues->origQuotaMeanStrongest[INVF_SMOOTHING_LENGTH] =
+      origQuotaMeanStrongest << 1;
+  detectorValues->sbrQuotaMeanStrongest[INVF_SMOOTHING_LENGTH] =
+      sbrQuotaMeanStrongest << 1;
+
+  /*
+   Filter values
+  */
+  detectorValues->origQuotaMeanFilt = FL2FXCONST_DBL(0.0f);
+  detectorValues->sbrQuotaMeanFilt = FL2FXCONST_DBL(0.0f);
+  detectorValues->origQuotaMeanStrongestFilt = FL2FXCONST_DBL(0.0f);
+  detectorValues->sbrQuotaMeanStrongestFilt = FL2FXCONST_DBL(0.0f);
+
+  for (i = 0; i < INVF_SMOOTHING_LENGTH + 1; i++) {
+    detectorValues->origQuotaMeanFilt +=
+        fMult(detectorValues->origQuotaMean[i], filter[i]);
+    detectorValues->sbrQuotaMeanFilt +=
+        fMult(detectorValues->sbrQuotaMean[i], filter[i]);
+    detectorValues->origQuotaMeanStrongestFilt +=
+        fMult(detectorValues->origQuotaMeanStrongest[i], filter[i]);
+    detectorValues->sbrQuotaMeanStrongestFilt +=
+        fMult(detectorValues->sbrQuotaMeanStrongest[i], filter[i]);
+  }
+}
+
+/**************************************************************************/
+/*!
+  \brief     Returns the region in which the input value belongs.
+
+
+
+  \return    region.
+
+*/
+/**************************************************************************/
+static INT findRegion(
+    FIXP_DBL currVal,        /*!< The current value. */
+    const FIXP_DBL *borders, /*!< The border of the regions. */
+    const INT numBorders     /*!< The number of borders. */
+) {
+  INT i;
+
+  if (currVal < borders[0]) {
+    return 0;
+  }
+
+  for (i = 1; i < numBorders; i++) {
+    if (currVal >= borders[i - 1] && currVal < borders[i]) {
+      return i;
+    }
+  }
+
+  if (currVal >= borders[numBorders - 1]) {
+    return numBorders;
+  }
+
+  return 0; /* We never get here, it's just to avoid compiler warnings.*/
+}
+
+/**************************************************************************/
+/*!
+  \brief     Makes a clever decision based on the quota vector.
+
+
+  \return     decision on which invf mode to use
+
+*/
+/**************************************************************************/
+static INVF_MODE decisionAlgorithm(
+    const DETECTOR_PARAMETERS
+        *detectorParams, /*!< Struct with the detector parameters. */
+    DETECTOR_VALUES *detectorValues, /*!< Struct with the detector values. */
+    INT transientFlag,  /*!< Flag indicating if there is a transient present.*/
+    INT *prevRegionSbr, /*!< The previous region in which the Sbr value was. */
+    INT *prevRegionOrig /*!< The previous region in which the Orig value was. */
+) {
+  INT invFiltLevel, regionSbr, regionOrig, regionNrg;
+
+  /*
+   Current thresholds.
+   */
+  const INT numRegionsSbr = detectorParams->numRegionsSbr;
+  const INT numRegionsOrig = detectorParams->numRegionsOrig;
+  const INT numRegionsNrg = detectorParams->numRegionsNrg;
+
+  FIXP_DBL quantStepsSbrTmp[MAX_NUM_REGIONS];
+  FIXP_DBL quantStepsOrigTmp[MAX_NUM_REGIONS];
+
+  /*
+   Current detector values.
+   */
+  FIXP_DBL origQuotaMeanFilt;
+  FIXP_DBL sbrQuotaMeanFilt;
+  FIXP_DBL nrg;
+
+  /* 0.375 = 3.0 / 8.0; 0.31143075889 = log2(RELAXATION)/64.0; 0.625 =
+   * log(16)/64.0; 0.6875 = 44/64.0 */
+  origQuotaMeanFilt =
+      (fMultDiv2(FL2FXCONST_DBL(2.f * 0.375f),
+                 (FIXP_DBL)(CalcLdData(max(detectorValues->origQuotaMeanFilt,
+                                           (FIXP_DBL)1)) +
+                            FL2FXCONST_DBL(0.31143075889f))))
+      << 0; /* scaled by 1/2^9 */
+  sbrQuotaMeanFilt =
+      (fMultDiv2(FL2FXCONST_DBL(2.f * 0.375f),
+                 (FIXP_DBL)(CalcLdData(max(detectorValues->sbrQuotaMeanFilt,
+                                           (FIXP_DBL)1)) +
+                            FL2FXCONST_DBL(0.31143075889f))))
+      << 0; /* scaled by 1/2^9 */
+  /* If energy is zero then we will get different results for different word
+   * lengths. */
+  nrg =
+      (fMultDiv2(FL2FXCONST_DBL(2.f * 0.375f),
+                 (FIXP_DBL)(CalcLdData(detectorValues->avgNrg + (FIXP_DBL)1) +
+                            FL2FXCONST_DBL(0.0625f) + FL2FXCONST_DBL(0.6875f))))
+      << 0; /* scaled by 1/2^8; 2^44 -> qmf energy scale */
+
+  FDKmemcpy(quantStepsSbrTmp, detectorParams->quantStepsSbr,
+            numRegionsSbr * sizeof(FIXP_DBL));
+  FDKmemcpy(quantStepsOrigTmp, detectorParams->quantStepsOrig,
+            numRegionsOrig * sizeof(FIXP_DBL));
+
+  if (*prevRegionSbr < numRegionsSbr)
+    quantStepsSbrTmp[*prevRegionSbr] =
+        detectorParams->quantStepsSbr[*prevRegionSbr] + hysteresis;
+  if (*prevRegionSbr > 0)
+    quantStepsSbrTmp[*prevRegionSbr - 1] =
+        detectorParams->quantStepsSbr[*prevRegionSbr - 1] - hysteresis;
+
+  if (*prevRegionOrig < numRegionsOrig)
+    quantStepsOrigTmp[*prevRegionOrig] =
+        detectorParams->quantStepsOrig[*prevRegionOrig] + hysteresis;
+  if (*prevRegionOrig > 0)
+    quantStepsOrigTmp[*prevRegionOrig - 1] =
+        detectorParams->quantStepsOrig[*prevRegionOrig - 1] - hysteresis;
+
+  regionSbr = findRegion(sbrQuotaMeanFilt, quantStepsSbrTmp, numRegionsSbr);
+  regionOrig = findRegion(origQuotaMeanFilt, quantStepsOrigTmp, numRegionsOrig);
+  regionNrg = findRegion(nrg, detectorParams->nrgBorders, numRegionsNrg);
+
+  *prevRegionSbr = regionSbr;
+  *prevRegionOrig = regionOrig;
+
+  /* Use different settings if a transient is present*/
+  invFiltLevel =
+      (transientFlag == 1)
+          ? detectorParams->regionSpaceTransient[regionSbr][regionOrig]
+          : detectorParams->regionSpace[regionSbr][regionOrig];
+
+  /* Compensate for low energy.*/
+  invFiltLevel =
+      max(invFiltLevel + detectorParams->EnergyCompFactor[regionNrg], 0);
+
+  return (INVF_MODE)(invFiltLevel);
+}
+
+/**************************************************************************/
+/*!
+  \brief     Estiamtion of the inverse filtering level required
+             in the decoder.
+
+   A second order LPC is calculated for every filterbank channel, using
+   the covariance method. THe ratio between the energy of the predicted
+   signal and the energy of the non-predictable signal is calcualted.
+
+  \return    none.
+
+*/
+/**************************************************************************/
+void FDKsbrEnc_qmfInverseFilteringDetector(
+    HANDLE_SBR_INV_FILT_EST
+        hInvFilt,           /*!< Handle to the SBR_INV_FILT_EST struct. */
+    FIXP_DBL **quotaMatrix, /*!< The matrix holding the tonality values of the
+                               original. */
+    FIXP_DBL *nrgVector,    /*!< The energy vector. */
+    SCHAR *indexVector,     /*!< Index vector to obtain the patched data. */
+    INT startIndex,         /*!< Start index. */
+    INT stopIndex,          /*!< Stop index. */
+    INT transientFlag, /*!< Flag indicating if a transient is present or not.*/
+    INVF_MODE *infVec  /*!< Vector holding the inverse filtering levels. */
+) {
+  INT band;
+
+  /*
+   * Do the inverse filtering level estimation.
+   *****************************************************/
+  for (band = 0; band < hInvFilt->noDetectorBands; band++) {
+    INT startChannel = hInvFilt->freqBandTableInvFilt[band];
+    INT stopChannel = hInvFilt->freqBandTableInvFilt[band + 1];
+
+    calculateDetectorValues(quotaMatrix, indexVector, nrgVector,
+                            &hInvFilt->detectorValues[band], startChannel,
+                            stopChannel, startIndex, stopIndex,
+                            hInvFilt->numberOfStrongest);
+
+    infVec[band] = decisionAlgorithm(
+        hInvFilt->detectorParams, &hInvFilt->detectorValues[band],
+        transientFlag, &hInvFilt->prevRegionSbr[band],
+        &hInvFilt->prevRegionOrig[band]);
+  }
+}
+
+/**************************************************************************/
+/*!
+  \brief     Initialize an instance of the inverse filtering level estimator.
+
+
+  \return   errorCode, noError if successful.
+
+*/
+/**************************************************************************/
+INT FDKsbrEnc_initInvFiltDetector(
+    HANDLE_SBR_INV_FILT_EST
+        hInvFilt, /*!< Pointer to a handle to the SBR_INV_FILT_EST struct. */
+    INT *freqBandTableDetector, /*!< Frequency band table for the inverse
+                                   filtering. */
+    INT numDetectorBands,       /*!< Number of inverse filtering bands. */
+    UINT
+        useSpeechConfig /*!< Flag: adapt tuning parameters according to speech*/
+) {
+  INT i;
+
+  FDKmemclear(hInvFilt, sizeof(SBR_INV_FILT_EST));
+
+  hInvFilt->detectorParams =
+      (useSpeechConfig) ? &detectorParamsAACSpeech : &detectorParamsAAC;
+
+  hInvFilt->noDetectorBandsMax = numDetectorBands;
+
+  /*
+     Memory initialisation
+  */
+  for (i = 0; i < hInvFilt->noDetectorBandsMax; i++) {
+    FDKmemclear(&hInvFilt->detectorValues[i], sizeof(DETECTOR_VALUES));
+    hInvFilt->prevInvfMode[i] = INVF_OFF;
+    hInvFilt->prevRegionOrig[i] = 0;
+    hInvFilt->prevRegionSbr[i] = 0;
+  }
+
+  /*
+  Reset the inverse fltering detector.
+  */
+  FDKsbrEnc_resetInvFiltDetector(hInvFilt, freqBandTableDetector,
+                                 hInvFilt->noDetectorBandsMax);
+
+  return (0);
+}
+
+/**************************************************************************/
+/*!
+  \brief     resets sbr inverse filtering structure.
+
+
+
+  \return   errorCode, noError if successful.
+
+*/
+/**************************************************************************/
+INT FDKsbrEnc_resetInvFiltDetector(
+    HANDLE_SBR_INV_FILT_EST
+        hInvFilt,               /*!< Handle to the SBR_INV_FILT_EST struct. */
+    INT *freqBandTableDetector, /*!< Frequency band table for the inverse
+                                   filtering. */
+    INT numDetectorBands)       /*!< Number of inverse filtering bands. */
+{
+  hInvFilt->numberOfStrongest = 1;
+  FDKmemcpy(hInvFilt->freqBandTableInvFilt, freqBandTableDetector,
+            (numDetectorBands + 1) * sizeof(INT));
+  hInvFilt->noDetectorBands = numDetectorBands;
+
+  return (0);
+}
diff --git a/fdk-aac/libSBRenc/src/invf_est.h b/fdk-aac/libSBRenc/src/invf_est.h
new file mode 100644
index 0000000..3ab6726
--- /dev/null
+++ b/fdk-aac/libSBRenc/src/invf_est.h
@@ -0,0 +1,181 @@
+/* -----------------------------------------------------------------------------
+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 encoder library ******************************
+
+   Author(s):
+
+   Description:
+
+*******************************************************************************/
+
+/*!
+  \file
+  \brief  Inverse Filtering detection prototypes $Revision: 92790 $
+*/
+#ifndef INVF_EST_H
+#define INVF_EST_H
+
+#include "sbr_encoder.h"
+#include "sbr_def.h"
+
+#define INVF_SMOOTHING_LENGTH 2
+
+typedef struct {
+  const FIXP_DBL *quantStepsSbr;
+  const FIXP_DBL *quantStepsOrig;
+  const FIXP_DBL *nrgBorders;
+  INT numRegionsSbr;
+  INT numRegionsOrig;
+  INT numRegionsNrg;
+  INVF_MODE regionSpace[5][5];
+  INVF_MODE regionSpaceTransient[5][5];
+  INT EnergyCompFactor[5];
+
+} DETECTOR_PARAMETERS;
+
+typedef struct {
+  FIXP_DBL origQuotaMean[INVF_SMOOTHING_LENGTH + 1];
+  FIXP_DBL sbrQuotaMean[INVF_SMOOTHING_LENGTH + 1];
+  FIXP_DBL origQuotaMeanStrongest[INVF_SMOOTHING_LENGTH + 1];
+  FIXP_DBL sbrQuotaMeanStrongest[INVF_SMOOTHING_LENGTH + 1];
+
+  FIXP_DBL origQuotaMeanFilt;
+  FIXP_DBL sbrQuotaMeanFilt;
+  FIXP_DBL origQuotaMeanStrongestFilt;
+  FIXP_DBL sbrQuotaMeanStrongestFilt;
+
+  FIXP_DBL origQuotaMax;
+  FIXP_DBL sbrQuotaMax;
+
+  FIXP_DBL avgNrg;
+} DETECTOR_VALUES;
+
+typedef struct {
+  INT numberOfStrongest;
+
+  INT prevRegionSbr[MAX_NUM_NOISE_VALUES];
+  INT prevRegionOrig[MAX_NUM_NOISE_VALUES];
+
+  INT freqBandTableInvFilt[MAX_NUM_NOISE_VALUES];
+  INT noDetectorBands;
+  INT noDetectorBandsMax;
+
+  const DETECTOR_PARAMETERS *detectorParams;
+
+  INVF_MODE prevInvfMode[MAX_NUM_NOISE_VALUES];
+  DETECTOR_VALUES detectorValues[MAX_NUM_NOISE_VALUES];
+
+  FIXP_DBL nrgAvg;
+  FIXP_DBL wmQmf[MAX_NUM_NOISE_VALUES];
+} SBR_INV_FILT_EST;
+
+typedef SBR_INV_FILT_EST *HANDLE_SBR_INV_FILT_EST;
+
+void FDKsbrEnc_qmfInverseFilteringDetector(HANDLE_SBR_INV_FILT_EST hInvFilt,
+                                           FIXP_DBL **quotaMatrix,
+                                           FIXP_DBL *nrgVector,
+                                           SCHAR *indexVector, INT startIndex,
+                                           INT stopIndex, INT transientFlag,
+                                           INVF_MODE *infVec);
+
+INT FDKsbrEnc_initInvFiltDetector(HANDLE_SBR_INV_FILT_EST hInvFilt,
+                                  INT *freqBandTableDetector,
+                                  INT numDetectorBands, UINT useSpeechConfig);
+
+INT FDKsbrEnc_resetInvFiltDetector(HANDLE_SBR_INV_FILT_EST hInvFilt,
+                                   INT *freqBandTableDetector,
+                                   INT numDetectorBands);
+
+#endif /* _QMF_INV_FILT_H */
diff --git a/fdk-aac/libSBRenc/src/mh_det.cpp b/fdk-aac/libSBRenc/src/mh_det.cpp
new file mode 100644
index 0000000..2f3b386
--- /dev/null
+++ b/fdk-aac/libSBRenc/src/mh_det.cpp
@@ -0,0 +1,1396 @@
+/* -----------------------------------------------------------------------------
+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 encoder library ******************************
+
+   Author(s):
+
+   Description:
+
+*******************************************************************************/
+
+#include "mh_det.h"
+
+#include "sbrenc_ram.h"
+#include "sbr_misc.h"
+
+#include "genericStds.h"
+
+#define SFM_SHIFT 2 /* Attention: SFM_SCALE depends on SFM_SHIFT */
+#define SFM_SCALE (MAXVAL_DBL >> SFM_SHIFT) /* 1.0 >> SFM_SHIFT */
+
+/*!< Detector Parameters for AAC core codec. */
+static const DETECTOR_PARAMETERS_MH paramsAac = {
+    9, /*!< deltaTime */
+    {
+        FL2FXCONST_DBL(20.0f * RELAXATION_FLOAT), /*!< thresHoldDiff */
+        FL2FXCONST_DBL(1.26f * RELAXATION_FLOAT), /*!< thresHoldDiffGuide */
+        FL2FXCONST_DBL(15.0f * RELAXATION_FLOAT), /*!< thresHoldTone */
+        FL2FXCONST_DBL((1.0f / 15.0f) *
+                       RELAXATION_FLOAT),         /*!< invThresHoldTone */
+        FL2FXCONST_DBL(1.26f * RELAXATION_FLOAT), /*!< thresHoldToneGuide */
+        FL2FXCONST_DBL(0.3f) >> SFM_SHIFT,        /*!< sfmThresSbr */
+        FL2FXCONST_DBL(0.1f) >> SFM_SHIFT,        /*!< sfmThresOrig */
+        FL2FXCONST_DBL(0.3f),                     /*!< decayGuideOrig */
+        FL2FXCONST_DBL(0.5f),                     /*!< decayGuideDiff */
+        FL2FXCONST_DBL(-0.000112993269),
+        /* LD64(FL2FXCONST_DBL(0.995f)) */ /*!< derivThresMaxLD64 */
+        FL2FXCONST_DBL(-0.000112993269),
+        /* LD64(FL2FXCONST_DBL(0.995f)) */ /*!< derivThresBelowLD64 */
+        FL2FXCONST_DBL(
+            -0.005030126483f) /* LD64(FL2FXCONST_DBL(0.8f)) */ /*!<
+                                                                  derivThresAboveLD64
+                                                                */
+    },
+    50 /*!< maxComp */
+};
+
+/*!< Detector Parameters for AAC LD core codec. */
+static const DETECTOR_PARAMETERS_MH paramsAacLd = {
+    16, /*!< Delta time. */
+    {
+        FL2FXCONST_DBL(25.0f * RELAXATION_FLOAT), /*!< thresHoldDiff */
+        FL2FXCONST_DBL(1.26f * RELAXATION_FLOAT), /*!< tresHoldDiffGuide */
+        FL2FXCONST_DBL(15.0f * RELAXATION_FLOAT), /*!< thresHoldTone */
+        FL2FXCONST_DBL((1.0f / 15.0f) *
+                       RELAXATION_FLOAT),         /*!< invThresHoldTone */
+        FL2FXCONST_DBL(1.26f * RELAXATION_FLOAT), /*!< thresHoldToneGuide */
+        FL2FXCONST_DBL(0.3f) >> SFM_SHIFT,        /*!< sfmThresSbr */
+        FL2FXCONST_DBL(0.1f) >> SFM_SHIFT,        /*!< sfmThresOrig */
+        FL2FXCONST_DBL(0.3f),                     /*!< decayGuideOrig */
+        FL2FXCONST_DBL(0.2f),                     /*!< decayGuideDiff */
+        FL2FXCONST_DBL(-0.000112993269),
+        /* LD64(FL2FXCONST_DBL(0.995f)) */ /*!< derivThresMaxLD64 */
+        FL2FXCONST_DBL(-0.000112993269),
+        /* LD64(FL2FXCONST_DBL(0.995f)) */ /*!< derivThresBelowLD64 */
+        FL2FXCONST_DBL(
+            -0.005030126483f) /* LD64(FL2FXCONST_DBL(0.8f)) */ /*!<
+                                                                  derivThresAboveLD64
+                                                                */
+    },
+    50 /*!< maxComp */
+};
+
+/**************************************************************************/
+/*!
+  \brief     Calculates the difference in tonality between original and SBR
+             for a given time and frequency region.
+
+             The values for pDiffMapped2Scfb are scaled by RELAXATION
+
+  \return    none.
+
+*/
+/**************************************************************************/
+static void diff(FIXP_DBL *RESTRICT pTonalityOrig, FIXP_DBL *pDiffMapped2Scfb,
+                 const UCHAR *RESTRICT pFreqBandTable, INT nScfb,
+                 SCHAR *indexVector) {
+  UCHAR i, ll, lu, k;
+  FIXP_DBL maxValOrig, maxValSbr, tmp;
+  INT scale;
+
+  for (i = 0; i < nScfb; i++) {
+    ll = pFreqBandTable[i];
+    lu = pFreqBandTable[i + 1];
+
+    maxValOrig = FL2FXCONST_DBL(0.0f);
+    maxValSbr = FL2FXCONST_DBL(0.0f);
+
+    for (k = ll; k < lu; k++) {
+      maxValOrig = fixMax(maxValOrig, pTonalityOrig[k]);
+      maxValSbr = fixMax(maxValSbr, pTonalityOrig[indexVector[k]]);
+    }
+
+    if ((maxValSbr >= RELAXATION)) {
+      tmp = fDivNorm(maxValOrig, maxValSbr, &scale);
+      pDiffMapped2Scfb[i] =
+          scaleValue(fMult(tmp, RELAXATION_FRACT),
+                     fixMax(-(DFRACT_BITS - 1), (scale - RELAXATION_SHIFT)));
+    } else {
+      pDiffMapped2Scfb[i] = maxValOrig;
+    }
+  }
+}
+
+/**************************************************************************/
+/*!
+  \brief     Calculates a flatness measure of the tonality measures.
+
+  Calculation of the power function and using scalefactor for basis:
+    Using log2:
+    z  = (2^k * x)^y;
+    z' = CalcLd(z) = y*CalcLd(x) + y*k;
+    z  = CalcInvLd(z');
+
+    Using ld64:
+    z  = (2^k * x)^y;
+    z' = CalcLd64(z) = y*CalcLd64(x)/64 + y*k/64;
+    z  = CalcInvLd64(z');
+
+  The values pSfmOrigVec and pSfmSbrVec are scaled by the factor 1/4.0
+
+  \return    none.
+
+*/
+/**************************************************************************/
+static void calculateFlatnessMeasure(FIXP_DBL *pQuotaBuffer, SCHAR *indexVector,
+                                     FIXP_DBL *pSfmOrigVec,
+                                     FIXP_DBL *pSfmSbrVec,
+                                     const UCHAR *pFreqBandTable, INT nSfb) {
+  INT i, j;
+  FIXP_DBL invBands, tmp1, tmp2;
+  INT shiftFac0, shiftFacSum0;
+  INT shiftFac1, shiftFacSum1;
+  FIXP_DBL accu;
+
+  for (i = 0; i < nSfb; i++) {
+    INT ll = pFreqBandTable[i];
+    INT lu = pFreqBandTable[i + 1];
+    pSfmOrigVec[i] = (FIXP_DBL)(MAXVAL_DBL >> 2);
+    pSfmSbrVec[i] = (FIXP_DBL)(MAXVAL_DBL >> 2);
+
+    if (lu - ll > 1) {
+      FIXP_DBL amOrig, amTransp, gmOrig, gmTransp, sfmOrig, sfmTransp;
+      invBands = GetInvInt(lu - ll);
+      shiftFacSum0 = 0;
+      shiftFacSum1 = 0;
+      amOrig = amTransp = FL2FXCONST_DBL(0.0f);
+      gmOrig = gmTransp = (FIXP_DBL)MAXVAL_DBL;
+
+      for (j = ll; j < lu; j++) {
+        sfmOrig = pQuotaBuffer[j];
+        sfmTransp = pQuotaBuffer[indexVector[j]];
+
+        amOrig += fMult(sfmOrig, invBands);
+        amTransp += fMult(sfmTransp, invBands);
+
+        shiftFac0 = CountLeadingBits(sfmOrig);
+        shiftFac1 = CountLeadingBits(sfmTransp);
+
+        gmOrig = fMult(gmOrig, sfmOrig << shiftFac0);
+        gmTransp = fMult(gmTransp, sfmTransp << shiftFac1);
+
+        shiftFacSum0 += shiftFac0;
+        shiftFacSum1 += shiftFac1;
+      }
+
+      if (gmOrig > FL2FXCONST_DBL(0.0f)) {
+        tmp1 = CalcLdData(gmOrig);    /* CalcLd64(x)/64 */
+        tmp1 = fMult(invBands, tmp1); /* y*CalcLd64(x)/64 */
+
+        /* y*k/64 */
+        accu = (FIXP_DBL)-shiftFacSum0 << (DFRACT_BITS - 1 - 8);
+        tmp2 = fMultDiv2(invBands, accu) << (2 + 1);
+
+        tmp2 = tmp1 + tmp2;           /* y*CalcLd64(x)/64 + y*k/64 */
+        gmOrig = CalcInvLdData(tmp2); /* CalcInvLd64(z'); */
+      } else {
+        gmOrig = FL2FXCONST_DBL(0.0f);
+      }
+
+      if (gmTransp > FL2FXCONST_DBL(0.0f)) {
+        tmp1 = CalcLdData(gmTransp);  /* CalcLd64(x)/64 */
+        tmp1 = fMult(invBands, tmp1); /* y*CalcLd64(x)/64 */
+
+        /* y*k/64 */
+        accu = (FIXP_DBL)-shiftFacSum1 << (DFRACT_BITS - 1 - 8);
+        tmp2 = fMultDiv2(invBands, accu) << (2 + 1);
+
+        tmp2 = tmp1 + tmp2;             /* y*CalcLd64(x)/64 + y*k/64 */
+        gmTransp = CalcInvLdData(tmp2); /* CalcInvLd64(z'); */
+      } else {
+        gmTransp = FL2FXCONST_DBL(0.0f);
+      }
+      if (amOrig != FL2FXCONST_DBL(0.0f))
+        pSfmOrigVec[i] =
+            FDKsbrEnc_LSI_divide_scale_fract(gmOrig, amOrig, SFM_SCALE);
+
+      if (amTransp != FL2FXCONST_DBL(0.0f))
+        pSfmSbrVec[i] =
+            FDKsbrEnc_LSI_divide_scale_fract(gmTransp, amTransp, SFM_SCALE);
+    }
+  }
+}
+
+/**************************************************************************/
+/*!
+  \brief     Calculates the input to the missing harmonics detection.
+
+
+  \return    none.
+
+*/
+/**************************************************************************/
+static void calculateDetectorInput(
+    FIXP_DBL **RESTRICT pQuotaBuffer, /*!< Pointer to tonality matrix. */
+    SCHAR *RESTRICT indexVector, FIXP_DBL **RESTRICT tonalityDiff,
+    FIXP_DBL **RESTRICT pSfmOrig, FIXP_DBL **RESTRICT pSfmSbr,
+    const UCHAR *freqBandTable, INT nSfb, INT noEstPerFrame, INT move) {
+  INT est;
+
+  /*
+  New estimate.
+  */
+  for (est = 0; est < noEstPerFrame; est++) {
+    diff(pQuotaBuffer[est + move], tonalityDiff[est + move], freqBandTable,
+         nSfb, indexVector);
+
+    calculateFlatnessMeasure(pQuotaBuffer[est + move], indexVector,
+                             pSfmOrig[est + move], pSfmSbr[est + move],
+                             freqBandTable, nSfb);
+  }
+}
+
+/**************************************************************************/
+/*!
+  \brief     Checks that the detection is not due to a LP filter
+
+  This function determines if a newly detected missing harmonics is not
+  in fact just a low-pass filtere input signal. If so, the detection is
+  removed.
+
+  \return    none.
+
+*/
+/**************************************************************************/
+static void removeLowPassDetection(UCHAR *RESTRICT pAddHarmSfb,
+                                   UCHAR **RESTRICT pDetectionVectors,
+                                   INT start, INT stop, INT nSfb,
+                                   const UCHAR *RESTRICT pFreqBandTable,
+                                   FIXP_DBL *RESTRICT pNrgVector,
+                                   THRES_HOLDS mhThresh)
+
+{
+  INT i, est;
+  INT maxDerivPos = pFreqBandTable[nSfb];
+  INT numBands = pFreqBandTable[nSfb];
+  FIXP_DBL nrgLow, nrgHigh;
+  FIXP_DBL nrgLD64, nrgLowLD64, nrgHighLD64, nrgDiffLD64;
+  FIXP_DBL valLD64, maxValLD64, maxValAboveLD64;
+  INT bLPsignal = 0;
+
+  maxValLD64 = FL2FXCONST_DBL(-1.0f);
+  for (i = numBands - 1 - 2; i > pFreqBandTable[0]; i--) {
+    nrgLow = pNrgVector[i];
+    nrgHigh = pNrgVector[i + 2];
+
+    if (nrgLow != FL2FXCONST_DBL(0.0f) && nrgLow > nrgHigh) {
+      nrgLowLD64 = CalcLdData(nrgLow >> 1);
+      nrgDiffLD64 = CalcLdData((nrgLow >> 1) - (nrgHigh >> 1));
+      valLD64 = nrgDiffLD64 - nrgLowLD64;
+      if (valLD64 > maxValLD64) {
+        maxDerivPos = i;
+        maxValLD64 = valLD64;
+      }
+      if (maxValLD64 > mhThresh.derivThresMaxLD64) {
+        break;
+      }
+    }
+  }
+
+  /* Find the largest "gradient" above. (should be relatively flat, hence we
+     expect a low value if the signal is LP.*/
+  maxValAboveLD64 = FL2FXCONST_DBL(-1.0f);
+  for (i = numBands - 1 - 2; i > maxDerivPos + 2; i--) {
+    nrgLow = pNrgVector[i];
+    nrgHigh = pNrgVector[i + 2];
+
+    if (nrgLow != FL2FXCONST_DBL(0.0f) && nrgLow > nrgHigh) {
+      nrgLowLD64 = CalcLdData(nrgLow >> 1);
+      nrgDiffLD64 = CalcLdData((nrgLow >> 1) - (nrgHigh >> 1));
+      valLD64 = nrgDiffLD64 - nrgLowLD64;
+      if (valLD64 > maxValAboveLD64) {
+        maxValAboveLD64 = valLD64;
+      }
+    } else {
+      if (nrgHigh != FL2FXCONST_DBL(0.0f) && nrgHigh > nrgLow) {
+        nrgHighLD64 = CalcLdData(nrgHigh >> 1);
+        nrgDiffLD64 = CalcLdData((nrgHigh >> 1) - (nrgLow >> 1));
+        valLD64 = nrgDiffLD64 - nrgHighLD64;
+        if (valLD64 > maxValAboveLD64) {
+          maxValAboveLD64 = valLD64;
+        }
+      }
+    }
+  }
+
+  if (maxValLD64 > mhThresh.derivThresMaxLD64 &&
+      maxValAboveLD64 < mhThresh.derivThresAboveLD64) {
+    bLPsignal = 1;
+
+    for (i = maxDerivPos - 1; i > maxDerivPos - 5 && i >= 0; i--) {
+      if (pNrgVector[i] != FL2FXCONST_DBL(0.0f) &&
+          pNrgVector[i] > pNrgVector[maxDerivPos + 2]) {
+        nrgDiffLD64 = CalcLdData((pNrgVector[i] >> 1) -
+                                 (pNrgVector[maxDerivPos + 2] >> 1));
+        nrgLD64 = CalcLdData(pNrgVector[i] >> 1);
+        valLD64 = nrgDiffLD64 - nrgLD64;
+        if (valLD64 < mhThresh.derivThresBelowLD64) {
+          bLPsignal = 0;
+          break;
+        }
+      } else {
+        bLPsignal = 0;
+        break;
+      }
+    }
+  }
+
+  if (bLPsignal) {
+    for (i = 0; i < nSfb; i++) {
+      if (maxDerivPos >= pFreqBandTable[i] &&
+          maxDerivPos < pFreqBandTable[i + 1])
+        break;
+    }
+
+    if (pAddHarmSfb[i]) {
+      pAddHarmSfb[i] = 0;
+      for (est = start; est < stop; est++) {
+        pDetectionVectors[est][i] = 0;
+      }
+    }
+  }
+}
+
+/**************************************************************************/
+/*!
+  \brief     Checks if it is allowed to detect a missing tone, that wasn't
+             detected previously.
+
+
+  \return    newDetectionAllowed flag.
+
+*/
+/**************************************************************************/
+static INT isDetectionOfNewToneAllowed(
+    const SBR_FRAME_INFO *pFrameInfo, INT *pDetectionStartPos,
+    INT noEstPerFrame, INT prevTransientFrame, INT prevTransientPos,
+    INT prevTransientFlag, INT transientPosOffset, INT transientFlag,
+    INT transientPos, INT deltaTime,
+    HANDLE_SBR_MISSING_HARMONICS_DETECTOR h_sbrMissingHarmonicsDetector) {
+  INT transientFrame, newDetectionAllowed;
+
+  /* Determine if this is a frame where a transient starts...
+   * If the transient flag was set the previous frame but not the
+   * transient frame flag, the transient frame flag is set in the current frame.
+   *****************************************************************************/
+  transientFrame = 0;
+  if (transientFlag) {
+    if (transientPos + transientPosOffset <
+        pFrameInfo->borders[pFrameInfo->nEnvelopes]) {
+      transientFrame = 1;
+      if (noEstPerFrame > 1) {
+        if (transientPos + transientPosOffset >
+            h_sbrMissingHarmonicsDetector->timeSlots >> 1) {
+          *pDetectionStartPos = noEstPerFrame;
+        } else {
+          *pDetectionStartPos = noEstPerFrame >> 1;
+        }
+
+      } else {
+        *pDetectionStartPos = noEstPerFrame;
+      }
+    }
+  } else {
+    if (prevTransientFlag && !prevTransientFrame) {
+      transientFrame = 1;
+      *pDetectionStartPos = 0;
+    }
+  }
+
+  /*
+   * Determine if detection of new missing harmonics are allowed.
+   * If the frame contains a transient it's ok. If the previous
+   * frame contained a transient it needs to be sufficiently close
+   * to the start of the current frame.
+   ****************************************************************/
+  newDetectionAllowed = 0;
+  if (transientFrame) {
+    newDetectionAllowed = 1;
+  } else {
+    if (prevTransientFrame &&
+        fixp_abs(pFrameInfo->borders[0] -
+                 (prevTransientPos + transientPosOffset -
+                  h_sbrMissingHarmonicsDetector->timeSlots)) < deltaTime) {
+      newDetectionAllowed = 1;
+      *pDetectionStartPos = 0;
+    }
+  }
+
+  h_sbrMissingHarmonicsDetector->previousTransientFlag = transientFlag;
+  h_sbrMissingHarmonicsDetector->previousTransientFrame = transientFrame;
+  h_sbrMissingHarmonicsDetector->previousTransientPos = transientPos;
+
+  return (newDetectionAllowed);
+}
+
+/**************************************************************************/
+/*!
+  \brief     Cleans up the detection after a transient.
+
+
+  \return    none.
+
+*/
+/**************************************************************************/
+static void transientCleanUp(FIXP_DBL **quotaBuffer, INT nSfb,
+                             UCHAR **detectionVectors, UCHAR *pAddHarmSfb,
+                             UCHAR *pPrevAddHarmSfb, INT **signBuffer,
+                             const UCHAR *pFreqBandTable, INT start, INT stop,
+                             INT newDetectionAllowed, FIXP_DBL *pNrgVector,
+                             THRES_HOLDS mhThresh) {
+  INT i, j, est;
+
+  for (est = start; est < stop; est++) {
+    for (i = 0; i < nSfb; i++) {
+      pAddHarmSfb[i] = pAddHarmSfb[i] || detectionVectors[est][i];
+    }
+  }
+
+  if (newDetectionAllowed == 1) {
+    /*
+     * Check for duplication of sines located
+     * on the border of two scf-bands.
+     *************************************************/
+    for (i = 0; i < nSfb - 1; i++) {
+      /* detection in adjacent channels.*/
+      if (pAddHarmSfb[i] && pAddHarmSfb[i + 1]) {
+        FIXP_DBL maxVal1, maxVal2;
+        INT maxPos1, maxPos2, maxPosTime1, maxPosTime2;
+
+        INT li = pFreqBandTable[i];
+        INT ui = pFreqBandTable[i + 1];
+
+        /* Find maximum tonality in the the two scf bands.*/
+        maxPosTime1 = start;
+        maxPos1 = li;
+        maxVal1 = quotaBuffer[start][li];
+        for (est = start; est < stop; est++) {
+          for (j = li; j < ui; j++) {
+            if (quotaBuffer[est][j] > maxVal1) {
+              maxVal1 = quotaBuffer[est][j];
+              maxPos1 = j;
+              maxPosTime1 = est;
+            }
+          }
+        }
+
+        li = pFreqBandTable[i + 1];
+        ui = pFreqBandTable[i + 2];
+
+        /* Find maximum tonality in the the two scf bands.*/
+        maxPosTime2 = start;
+        maxPos2 = li;
+        maxVal2 = quotaBuffer[start][li];
+        for (est = start; est < stop; est++) {
+          for (j = li; j < ui; j++) {
+            if (quotaBuffer[est][j] > maxVal2) {
+              maxVal2 = quotaBuffer[est][j];
+              maxPos2 = j;
+              maxPosTime2 = est;
+            }
+          }
+        }
+
+        /* If the maximum values are in adjacent QMF-channels, we need to remove
+           the lowest of the two.*/
+        if (maxPos2 - maxPos1 < 2) {
+          if (pPrevAddHarmSfb[i] == 1 && pPrevAddHarmSfb[i + 1] == 0) {
+            /* Keep the lower, remove the upper.*/
+            pAddHarmSfb[i + 1] = 0;
+            for (est = start; est < stop; est++) {
+              detectionVectors[est][i + 1] = 0;
+            }
+          } else {
+            if (pPrevAddHarmSfb[i] == 0 && pPrevAddHarmSfb[i + 1] == 1) {
+              /* Keep the upper, remove the lower.*/
+              pAddHarmSfb[i] = 0;
+              for (est = start; est < stop; est++) {
+                detectionVectors[est][i] = 0;
+              }
+            } else {
+              /* If the maximum values are in adjacent QMF-channels, and if the
+                 signs indicate that it is the same sine, we need to remove the
+                 lowest of the two.*/
+              if (maxVal1 > maxVal2) {
+                if (signBuffer[maxPosTime1][maxPos2] < 0 &&
+                    signBuffer[maxPosTime1][maxPos1] > 0) {
+                  /* Keep the lower, remove the upper.*/
+                  pAddHarmSfb[i + 1] = 0;
+                  for (est = start; est < stop; est++) {
+                    detectionVectors[est][i + 1] = 0;
+                  }
+                }
+              } else {
+                if (signBuffer[maxPosTime2][maxPos2] < 0 &&
+                    signBuffer[maxPosTime2][maxPos1] > 0) {
+                  /* Keep the upper, remove the lower.*/
+                  pAddHarmSfb[i] = 0;
+                  for (est = start; est < stop; est++) {
+                    detectionVectors[est][i] = 0;
+                  }
+                }
+              }
+            }
+          }
+        }
+      }
+    }
+
+    /* Make sure that the detection is not the cut-off of a low pass filter. */
+    removeLowPassDetection(pAddHarmSfb, detectionVectors, start, stop, nSfb,
+                           pFreqBandTable, pNrgVector, mhThresh);
+  } else {
+    /*
+     * If a missing harmonic wasn't missing the previous frame
+     * the transient-flag needs to be set in order to be allowed to detect it.
+     *************************************************************************/
+    for (i = 0; i < nSfb; i++) {
+      if (pAddHarmSfb[i] - pPrevAddHarmSfb[i] > 0) pAddHarmSfb[i] = 0;
+    }
+  }
+}
+
+/*****************************************************************************/
+/*!
+  \brief     Detection for one tonality estimate.
+
+  This is the actual missing harmonics detection, using information from the
+  previous detection.
+
+  If a missing harmonic was detected (in a previous frame) due to too high
+  tonality differences, but there was not enough tonality difference in the
+  current frame, the detection algorithm still continues to trace the strongest
+  tone in the scalefactor band (assuming that this is the tone that is going to
+  be replaced in the decoder). This is done to avoid abrupt endings of sines
+  fading out (e.g. in the glockenspiel).
+
+  The function also tries to estimate where one sine is going to be replaced
+  with multiple sines (due to the patching). This is done by comparing the
+  tonality flatness measure of the original and the SBR signal.
+
+  The function also tries to estimate (for the scalefactor bands only
+  containing one qmf subband) when a strong tone in the original will be
+  replaced by a strong tone in the adjacent QMF subband.
+
+  \return    none.
+
+*/
+/**************************************************************************/
+static void detection(FIXP_DBL *quotaBuffer, FIXP_DBL *pDiffVecScfb, INT nSfb,
+                      UCHAR *pHarmVec, const UCHAR *pFreqBandTable,
+                      FIXP_DBL *sfmOrig, FIXP_DBL *sfmSbr,
+                      GUIDE_VECTORS guideVectors, GUIDE_VECTORS newGuideVectors,
+                      THRES_HOLDS mhThresh) {
+  INT i, j, ll, lu;
+  FIXP_DBL thresTemp, thresOrig;
+
+  /*
+   * Do detection on the difference vector, i.e. the difference between
+   * the original and the transposed.
+   *********************************************************************/
+  for (i = 0; i < nSfb; i++) {
+    thresTemp = (guideVectors.guideVectorDiff[i] != FL2FXCONST_DBL(0.0f))
+                    ? fMax(fMult(mhThresh.decayGuideDiff,
+                                 guideVectors.guideVectorDiff[i]),
+                           mhThresh.thresHoldDiffGuide)
+                    : mhThresh.thresHoldDiff;
+
+    thresTemp = fMin(thresTemp, mhThresh.thresHoldDiff);
+
+    if (pDiffVecScfb[i] > thresTemp) {
+      pHarmVec[i] = 1;
+      newGuideVectors.guideVectorDiff[i] = pDiffVecScfb[i];
+    } else {
+      /* If the guide wasn't zero, but the current level is to low,
+         start tracking the decay on the tone in the original rather
+         than the difference.*/
+      if (guideVectors.guideVectorDiff[i] != FL2FXCONST_DBL(0.0f)) {
+        guideVectors.guideVectorOrig[i] = mhThresh.thresHoldToneGuide;
+      }
+    }
+  }
+
+  /*
+   * Trace tones in the original signal that at one point
+   * have been detected because they will be replaced by
+   * multiple tones in the sbr signal.
+   ****************************************************/
+
+  for (i = 0; i < nSfb; i++) {
+    ll = pFreqBandTable[i];
+    lu = pFreqBandTable[i + 1];
+
+    thresOrig =
+        fixMax(fMult(guideVectors.guideVectorOrig[i], mhThresh.decayGuideOrig),
+               mhThresh.thresHoldToneGuide);
+    thresOrig = fixMin(thresOrig, mhThresh.thresHoldTone);
+
+    if (guideVectors.guideVectorOrig[i] != FL2FXCONST_DBL(0.0f)) {
+      for (j = ll; j < lu; j++) {
+        if (quotaBuffer[j] > thresOrig) {
+          pHarmVec[i] = 1;
+          newGuideVectors.guideVectorOrig[i] = quotaBuffer[j];
+        }
+      }
+    }
+  }
+
+  /*
+   * Check for multiple sines in the transposed signal,
+   * where there is only one in the original.
+   ****************************************************/
+  thresOrig = mhThresh.thresHoldTone;
+
+  for (i = 0; i < nSfb; i++) {
+    ll = pFreqBandTable[i];
+    lu = pFreqBandTable[i + 1];
+
+    if (pHarmVec[i] == 0) {
+      if (lu - ll > 1) {
+        for (j = ll; j < lu; j++) {
+          if (quotaBuffer[j] > thresOrig &&
+              (sfmSbr[i] > mhThresh.sfmThresSbr &&
+               sfmOrig[i] < mhThresh.sfmThresOrig)) {
+            pHarmVec[i] = 1;
+            newGuideVectors.guideVectorOrig[i] = quotaBuffer[j];
+          }
+        }
+      } else {
+        if (i < nSfb - 1) {
+          ll = pFreqBandTable[i];
+
+          if (i > 0) {
+            if (quotaBuffer[ll] > mhThresh.thresHoldTone &&
+                (pDiffVecScfb[i + 1] < mhThresh.invThresHoldTone ||
+                 pDiffVecScfb[i - 1] < mhThresh.invThresHoldTone)) {
+              pHarmVec[i] = 1;
+              newGuideVectors.guideVectorOrig[i] = quotaBuffer[ll];
+            }
+          } else {
+            if (quotaBuffer[ll] > mhThresh.thresHoldTone &&
+                pDiffVecScfb[i + 1] < mhThresh.invThresHoldTone) {
+              pHarmVec[i] = 1;
+              newGuideVectors.guideVectorOrig[i] = quotaBuffer[ll];
+            }
+          }
+        }
+      }
+    }
+  }
+}
+
+/**************************************************************************/
+/*!
+  \brief     Do detection for every tonality estimate, using forward prediction.
+
+
+  \return    none.
+
+*/
+/**************************************************************************/
+static void detectionWithPrediction(
+    FIXP_DBL **quotaBuffer, FIXP_DBL **pDiffVecScfb, INT **signBuffer, INT nSfb,
+    const UCHAR *pFreqBandTable, FIXP_DBL **sfmOrig, FIXP_DBL **sfmSbr,
+    UCHAR **detectionVectors, UCHAR *pPrevAddHarmSfb,
+    GUIDE_VECTORS *guideVectors, INT noEstPerFrame, INT detectionStart,
+    INT totNoEst, INT newDetectionAllowed, INT *pAddHarmFlag,
+    UCHAR *pAddHarmSfb, FIXP_DBL *pNrgVector,
+    const DETECTOR_PARAMETERS_MH *mhParams) {
+  INT est = 0, i;
+  INT start;
+
+  FDKmemclear(pAddHarmSfb, nSfb * sizeof(UCHAR));
+
+  if (newDetectionAllowed) {
+    /* Since we don't want to use the transient region for detection (since the
+       tonality values tend to be a bit unreliable for this region) the
+       guide-values are copied to the current starting point. */
+    if (totNoEst > 1) {
+      start = detectionStart + 1;
+
+      if (start != 0) {
+        FDKmemcpy(guideVectors[start].guideVectorDiff,
+                  guideVectors[0].guideVectorDiff, nSfb * sizeof(FIXP_DBL));
+        FDKmemcpy(guideVectors[start].guideVectorOrig,
+                  guideVectors[0].guideVectorOrig, nSfb * sizeof(FIXP_DBL));
+        FDKmemclear(guideVectors[start - 1].guideVectorDetected,
+                    nSfb * sizeof(UCHAR));
+      }
+    } else {
+      start = 0;
+    }
+  } else {
+    start = 0;
+  }
+
+  for (est = start; est < totNoEst; est++) {
+    /*
+     * Do detection on the current frame using
+     * guide-info from the previous.
+     *******************************************/
+    if (est > 0) {
+      FDKmemcpy(guideVectors[est].guideVectorDetected,
+                detectionVectors[est - 1], nSfb * sizeof(UCHAR));
+    }
+
+    FDKmemclear(detectionVectors[est], nSfb * sizeof(UCHAR));
+
+    if (est < totNoEst - 1) {
+      FDKmemclear(guideVectors[est + 1].guideVectorDiff,
+                  nSfb * sizeof(FIXP_DBL));
+      FDKmemclear(guideVectors[est + 1].guideVectorOrig,
+                  nSfb * sizeof(FIXP_DBL));
+      FDKmemclear(guideVectors[est + 1].guideVectorDetected,
+                  nSfb * sizeof(UCHAR));
+
+      detection(quotaBuffer[est], pDiffVecScfb[est], nSfb,
+                detectionVectors[est], pFreqBandTable, sfmOrig[est],
+                sfmSbr[est], guideVectors[est], guideVectors[est + 1],
+                mhParams->thresHolds);
+    } else {
+      FDKmemclear(guideVectors[est].guideVectorDiff, nSfb * sizeof(FIXP_DBL));
+      FDKmemclear(guideVectors[est].guideVectorOrig, nSfb * sizeof(FIXP_DBL));
+      FDKmemclear(guideVectors[est].guideVectorDetected, nSfb * sizeof(UCHAR));
+
+      detection(quotaBuffer[est], pDiffVecScfb[est], nSfb,
+                detectionVectors[est], pFreqBandTable, sfmOrig[est],
+                sfmSbr[est], guideVectors[est], guideVectors[est],
+                mhParams->thresHolds);
+    }
+  }
+
+  /* Clean up the detection.*/
+  transientCleanUp(quotaBuffer, nSfb, detectionVectors, pAddHarmSfb,
+                   pPrevAddHarmSfb, signBuffer, pFreqBandTable, start, totNoEst,
+                   newDetectionAllowed, pNrgVector, mhParams->thresHolds);
+
+  /* Set flag... */
+  *pAddHarmFlag = 0;
+  for (i = 0; i < nSfb; i++) {
+    if (pAddHarmSfb[i]) {
+      *pAddHarmFlag = 1;
+      break;
+    }
+  }
+
+  FDKmemcpy(pPrevAddHarmSfb, pAddHarmSfb, nSfb * sizeof(UCHAR));
+  FDKmemcpy(guideVectors[0].guideVectorDetected, pAddHarmSfb,
+            nSfb * sizeof(INT));
+
+  for (i = 0; i < nSfb; i++) {
+    guideVectors[0].guideVectorDiff[i] = FL2FXCONST_DBL(0.0f);
+    guideVectors[0].guideVectorOrig[i] = FL2FXCONST_DBL(0.0f);
+
+    if (pAddHarmSfb[i] == 1) {
+      /* If we had a detection use the guide-value in the next frame from the
+      last estimate were the detection was done.*/
+      for (est = start; est < totNoEst; est++) {
+        if (guideVectors[est].guideVectorDiff[i] != FL2FXCONST_DBL(0.0f)) {
+          guideVectors[0].guideVectorDiff[i] =
+              guideVectors[est].guideVectorDiff[i];
+        }
+        if (guideVectors[est].guideVectorOrig[i] != FL2FXCONST_DBL(0.0f)) {
+          guideVectors[0].guideVectorOrig[i] =
+              guideVectors[est].guideVectorOrig[i];
+        }
+      }
+    }
+  }
+}
+
+/**************************************************************************/
+/*!
+  \brief     Calculates a compensation vector for the energy data.
+
+  This function calculates a compensation vector for the energy data (i.e.
+  envelope data) that is calculated elsewhere. This is since, one sine on
+  the border of two scalefactor bands, will be replace by one sine in the
+  middle of either scalefactor band. However, since the sine that is replaced
+  will influence the energy estimate in both scalefactor bands (in the envelops
+  calculation function) a compensation value is required in order to avoid
+  noise substitution in the decoder next to the synthetic sine.
+
+  \return    none.
+
+*/
+/**************************************************************************/
+static void calculateCompVector(UCHAR *pAddHarmSfb, FIXP_DBL **pTonalityMatrix,
+                                INT **pSignMatrix, UCHAR *pEnvComp, INT nSfb,
+                                const UCHAR *freqBandTable, INT totNoEst,
+                                INT maxComp, UCHAR *pPrevEnvComp,
+                                INT newDetectionAllowed) {
+  INT scfBand, est, l, ll, lu, maxPosF, maxPosT;
+  FIXP_DBL maxVal;
+  INT compValue;
+  FIXP_DBL tmp;
+
+  FDKmemclear(pEnvComp, nSfb * sizeof(UCHAR));
+
+  for (scfBand = 0; scfBand < nSfb; scfBand++) {
+    if (pAddHarmSfb[scfBand]) { /* A missing sine was detected */
+      ll = freqBandTable[scfBand];
+      lu = freqBandTable[scfBand + 1];
+
+      maxPosF = 0; /* First find the maximum*/
+      maxPosT = 0;
+      maxVal = FL2FXCONST_DBL(0.0f);
+
+      for (est = 0; est < totNoEst; est++) {
+        for (l = ll; l < lu; l++) {
+          if (pTonalityMatrix[est][l] > maxVal) {
+            maxVal = pTonalityMatrix[est][l];
+            maxPosF = l;
+            maxPosT = est;
+          }
+        }
+      }
+
+      /*
+       * If the maximum tonality is at the lower border of the
+       * scalefactor band, we check the sign of the adjacent channels
+       * to see if this sine is shared by the lower channel. If so, the
+       * energy of the single sine will be present in two scalefactor bands
+       * in the SBR data, which will cause problems in the decoder, when we
+       * add a sine to just one of the channels.
+       *********************************************************************/
+      if (maxPosF == ll && scfBand) {
+        if (!pAddHarmSfb[scfBand - 1]) { /* No detection below*/
+          if (pSignMatrix[maxPosT][maxPosF - 1] > 0 &&
+              pSignMatrix[maxPosT][maxPosF] < 0) {
+            /* The comp value is calulated as the tonallity value, i.e we want
+               to reduce the envelope data for this channel with as much as the
+               tonality that is spread from the channel above. (ld64(RELAXATION)
+               = 0.31143075889) */
+            tmp = fixp_abs(
+                (FIXP_DBL)CalcLdData(pTonalityMatrix[maxPosT][maxPosF - 1]) +
+                RELAXATION_LD64);
+            tmp = (tmp >> (DFRACT_BITS - 1 - LD_DATA_SHIFT - 1)) +
+                  (FIXP_DBL)1; /* shift one bit less for rounding */
+            compValue = ((INT)(LONG)tmp) >> 1;
+
+            /* limit the comp-value*/
+            if (compValue > maxComp) compValue = maxComp;
+
+            pEnvComp[scfBand - 1] = compValue;
+          }
+        }
+      }
+
+      /*
+       * Same as above, but for the upper end of the scalefactor-band.
+       ***************************************************************/
+      if (maxPosF == lu - 1 && scfBand + 1 < nSfb) { /* Upper border*/
+        if (!pAddHarmSfb[scfBand + 1]) {
+          if (pSignMatrix[maxPosT][maxPosF] > 0 &&
+              pSignMatrix[maxPosT][maxPosF + 1] < 0) {
+            tmp = fixp_abs(
+                (FIXP_DBL)CalcLdData(pTonalityMatrix[maxPosT][maxPosF + 1]) +
+                RELAXATION_LD64);
+            tmp = (tmp >> (DFRACT_BITS - 1 - LD_DATA_SHIFT - 1)) +
+                  (FIXP_DBL)1; /* shift one bit less for rounding */
+            compValue = ((INT)(LONG)tmp) >> 1;
+
+            if (compValue > maxComp) compValue = maxComp;
+
+            pEnvComp[scfBand + 1] = compValue;
+          }
+        }
+      }
+    }
+  }
+
+  if (newDetectionAllowed == 0) {
+    for (scfBand = 0; scfBand < nSfb; scfBand++) {
+      if (pEnvComp[scfBand] != 0 && pPrevEnvComp[scfBand] == 0)
+        pEnvComp[scfBand] = 0;
+    }
+  }
+
+  /* remember the value for the next frame.*/
+  FDKmemcpy(pPrevEnvComp, pEnvComp, nSfb * sizeof(UCHAR));
+}
+
+/**************************************************************************/
+/*!
+  \brief     Detects where strong tonal components will be missing after
+             HFR in the decoder.
+
+
+  \return    none.
+
+*/
+/**************************************************************************/
+void FDKsbrEnc_SbrMissingHarmonicsDetectorQmf(
+    HANDLE_SBR_MISSING_HARMONICS_DETECTOR h_sbrMHDet, FIXP_DBL **pQuotaBuffer,
+    INT **pSignBuffer, SCHAR *indexVector, const SBR_FRAME_INFO *pFrameInfo,
+    const UCHAR *pTranInfo, INT *pAddHarmonicsFlag,
+    UCHAR *pAddHarmonicsScaleFactorBands, const UCHAR *freqBandTable, INT nSfb,
+    UCHAR *envelopeCompensation, FIXP_DBL *pNrgVector) {
+  INT transientFlag = pTranInfo[1];
+  INT transientPos = pTranInfo[0];
+  INT newDetectionAllowed;
+  INT transientDetStart = 0;
+
+  UCHAR **detectionVectors = h_sbrMHDet->detectionVectors;
+  INT move = h_sbrMHDet->move;
+  INT noEstPerFrame = h_sbrMHDet->noEstPerFrame;
+  INT totNoEst = h_sbrMHDet->totNoEst;
+  INT prevTransientFlag = h_sbrMHDet->previousTransientFlag;
+  INT prevTransientFrame = h_sbrMHDet->previousTransientFrame;
+  INT transientPosOffset = h_sbrMHDet->transientPosOffset;
+  INT prevTransientPos = h_sbrMHDet->previousTransientPos;
+  GUIDE_VECTORS *guideVectors = h_sbrMHDet->guideVectors;
+  INT deltaTime = h_sbrMHDet->mhParams->deltaTime;
+  INT maxComp = h_sbrMHDet->mhParams->maxComp;
+
+  int est;
+
+  /*
+  Buffer values.
+  */
+  FDK_ASSERT(move <= (MAX_NO_OF_ESTIMATES >> 1));
+  FDK_ASSERT(noEstPerFrame <= (MAX_NO_OF_ESTIMATES >> 1));
+
+  FIXP_DBL *sfmSbr[MAX_NO_OF_ESTIMATES];
+  FIXP_DBL *sfmOrig[MAX_NO_OF_ESTIMATES];
+  FIXP_DBL *tonalityDiff[MAX_NO_OF_ESTIMATES];
+
+  for (est = 0; est < MAX_NO_OF_ESTIMATES / 2; est++) {
+    sfmSbr[est] = h_sbrMHDet->sfmSbr[est];
+    sfmOrig[est] = h_sbrMHDet->sfmOrig[est];
+    tonalityDiff[est] = h_sbrMHDet->tonalityDiff[est];
+  }
+
+  C_ALLOC_SCRATCH_START(_scratch, FIXP_DBL,
+                        3 * MAX_NO_OF_ESTIMATES / 2 * MAX_FREQ_COEFFS)
+  FIXP_DBL *scratch = _scratch;
+  for (; est < MAX_NO_OF_ESTIMATES; est++) {
+    sfmSbr[est] = scratch;
+    scratch += MAX_FREQ_COEFFS;
+    sfmOrig[est] = scratch;
+    scratch += MAX_FREQ_COEFFS;
+    tonalityDiff[est] = scratch;
+    scratch += MAX_FREQ_COEFFS;
+  }
+
+  /* Determine if we're allowed to detect "missing harmonics" that wasn't
+     detected before. In order to be allowed to do new detection, there must be
+     a transient in the current frame, or a transient in the previous frame
+     sufficiently close to the current frame. */
+  newDetectionAllowed = isDetectionOfNewToneAllowed(
+      pFrameInfo, &transientDetStart, noEstPerFrame, prevTransientFrame,
+      prevTransientPos, prevTransientFlag, transientPosOffset, transientFlag,
+      transientPos, deltaTime, h_sbrMHDet);
+
+  /* Calulate the variables that will be used subsequently for the actual
+   * detection */
+  calculateDetectorInput(pQuotaBuffer, indexVector, tonalityDiff, sfmOrig,
+                         sfmSbr, freqBandTable, nSfb, noEstPerFrame, move);
+
+  /* Do the actual detection using information from previous detections */
+  detectionWithPrediction(pQuotaBuffer, tonalityDiff, pSignBuffer, nSfb,
+                          freqBandTable, sfmOrig, sfmSbr, detectionVectors,
+                          h_sbrMHDet->guideScfb, guideVectors, noEstPerFrame,
+                          transientDetStart, totNoEst, newDetectionAllowed,
+                          pAddHarmonicsFlag, pAddHarmonicsScaleFactorBands,
+                          pNrgVector, h_sbrMHDet->mhParams);
+
+  /* Calculate the comp vector, so that the energy can be
+     compensated for a sine between two QMF-bands. */
+  calculateCompVector(pAddHarmonicsScaleFactorBands, pQuotaBuffer, pSignBuffer,
+                      envelopeCompensation, nSfb, freqBandTable, totNoEst,
+                      maxComp, h_sbrMHDet->prevEnvelopeCompensation,
+                      newDetectionAllowed);
+
+  for (est = 0; est < move; est++) {
+    FDKmemcpy(tonalityDiff[est], tonalityDiff[est + noEstPerFrame],
+              sizeof(FIXP_DBL) * MAX_FREQ_COEFFS);
+    FDKmemcpy(sfmOrig[est], sfmOrig[est + noEstPerFrame],
+              sizeof(FIXP_DBL) * MAX_FREQ_COEFFS);
+    FDKmemcpy(sfmSbr[est], sfmSbr[est + noEstPerFrame],
+              sizeof(FIXP_DBL) * MAX_FREQ_COEFFS);
+  }
+  C_ALLOC_SCRATCH_END(_scratch, FIXP_DBL,
+                      3 * MAX_NO_OF_ESTIMATES / 2 * MAX_FREQ_COEFFS)
+}
+
+/**************************************************************************/
+/*!
+  \brief     Initialize an instance of the missing harmonics detector.
+
+
+  \return    errorCode, noError if OK.
+
+*/
+/**************************************************************************/
+INT FDKsbrEnc_CreateSbrMissingHarmonicsDetector(
+    HANDLE_SBR_MISSING_HARMONICS_DETECTOR hSbrMHDet, INT chan) {
+  HANDLE_SBR_MISSING_HARMONICS_DETECTOR hs = hSbrMHDet;
+  INT i;
+
+  UCHAR *detectionVectors = GetRam_Sbr_detectionVectors(chan);
+  UCHAR *guideVectorDetected = GetRam_Sbr_guideVectorDetected(chan);
+  FIXP_DBL *guideVectorDiff = GetRam_Sbr_guideVectorDiff(chan);
+  FIXP_DBL *guideVectorOrig = GetRam_Sbr_guideVectorOrig(chan);
+
+  FDKmemclear(hs, sizeof(SBR_MISSING_HARMONICS_DETECTOR));
+
+  hs->prevEnvelopeCompensation = GetRam_Sbr_prevEnvelopeCompensation(chan);
+  hs->guideScfb = GetRam_Sbr_guideScfb(chan);
+
+  if ((NULL == detectionVectors) || (NULL == guideVectorDetected) ||
+      (NULL == guideVectorDiff) || (NULL == guideVectorOrig) ||
+      (NULL == hs->prevEnvelopeCompensation) || (NULL == hs->guideScfb)) {
+    goto bail;
+  }
+
+  for (i = 0; i < MAX_NO_OF_ESTIMATES; i++) {
+    hs->guideVectors[i].guideVectorDiff =
+        guideVectorDiff + (i * MAX_FREQ_COEFFS);
+    hs->guideVectors[i].guideVectorOrig =
+        guideVectorOrig + (i * MAX_FREQ_COEFFS);
+    hs->detectionVectors[i] = detectionVectors + (i * MAX_FREQ_COEFFS);
+    hs->guideVectors[i].guideVectorDetected =
+        guideVectorDetected + (i * MAX_FREQ_COEFFS);
+  }
+
+  return 0;
+
+bail:
+  hs->guideVectors[0].guideVectorDiff = guideVectorDiff;
+  hs->guideVectors[0].guideVectorOrig = guideVectorOrig;
+  hs->detectionVectors[0] = detectionVectors;
+  hs->guideVectors[0].guideVectorDetected = guideVectorDetected;
+
+  FDKsbrEnc_DeleteSbrMissingHarmonicsDetector(hs);
+  return -1;
+}
+
+/**************************************************************************/
+/*!
+  \brief     Initialize an instance of the missing harmonics detector.
+
+
+  \return    errorCode, noError if OK.
+
+*/
+/**************************************************************************/
+INT FDKsbrEnc_InitSbrMissingHarmonicsDetector(
+    HANDLE_SBR_MISSING_HARMONICS_DETECTOR hSbrMHDet, INT sampleFreq,
+    INT frameSize, INT nSfb, INT qmfNoChannels, INT totNoEst, INT move,
+    INT noEstPerFrame, UINT sbrSyntaxFlags) {
+  HANDLE_SBR_MISSING_HARMONICS_DETECTOR hs = hSbrMHDet;
+  int i;
+
+  FDK_ASSERT(totNoEst <= MAX_NO_OF_ESTIMATES);
+
+  if (sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) {
+    switch (frameSize) {
+      case 1024:
+      case 512:
+        hs->transientPosOffset = FRAME_MIDDLE_SLOT_512LD;
+        hs->timeSlots = 16;
+        break;
+      case 960:
+      case 480:
+        hs->transientPosOffset = FRAME_MIDDLE_SLOT_512LD;
+        hs->timeSlots = 15;
+        break;
+      default:
+        return -1;
+    }
+  } else {
+    switch (frameSize) {
+      case 2048:
+      case 1024:
+        hs->transientPosOffset = FRAME_MIDDLE_SLOT_2048;
+        hs->timeSlots = NUMBER_TIME_SLOTS_2048;
+        break;
+      case 1920:
+      case 960:
+        hs->transientPosOffset = FRAME_MIDDLE_SLOT_1920;
+        hs->timeSlots = NUMBER_TIME_SLOTS_1920;
+        break;
+      default:
+        return -1;
+    }
+  }
+
+  if (sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) {
+    hs->mhParams = &paramsAacLd;
+  } else
+    hs->mhParams = &paramsAac;
+
+  hs->qmfNoChannels = qmfNoChannels;
+  hs->sampleFreq = sampleFreq;
+  hs->nSfb = nSfb;
+
+  hs->totNoEst = totNoEst;
+  hs->move = move;
+  hs->noEstPerFrame = noEstPerFrame;
+
+  for (i = 0; i < totNoEst; i++) {
+    FDKmemclear(hs->guideVectors[i].guideVectorDiff,
+                sizeof(FIXP_DBL) * MAX_FREQ_COEFFS);
+    FDKmemclear(hs->guideVectors[i].guideVectorOrig,
+                sizeof(FIXP_DBL) * MAX_FREQ_COEFFS);
+    FDKmemclear(hs->detectionVectors[i], sizeof(UCHAR) * MAX_FREQ_COEFFS);
+    FDKmemclear(hs->guideVectors[i].guideVectorDetected,
+                sizeof(UCHAR) * MAX_FREQ_COEFFS);
+  }
+
+  // for(i=0; i<totNoEst/2; i++) {
+  for (i = 0; i < MAX_NO_OF_ESTIMATES / 2; i++) {
+    FDKmemclear(hs->tonalityDiff[i], sizeof(FIXP_DBL) * MAX_FREQ_COEFFS);
+    FDKmemclear(hs->sfmOrig[i], sizeof(FIXP_DBL) * MAX_FREQ_COEFFS);
+    FDKmemclear(hs->sfmSbr[i], sizeof(FIXP_DBL) * MAX_FREQ_COEFFS);
+  }
+
+  FDKmemclear(hs->prevEnvelopeCompensation, sizeof(UCHAR) * MAX_FREQ_COEFFS);
+  FDKmemclear(hs->guideScfb, sizeof(UCHAR) * MAX_FREQ_COEFFS);
+
+  hs->previousTransientFlag = 0;
+  hs->previousTransientFrame = 0;
+  hs->previousTransientPos = 0;
+
+  return (0);
+}
+
+/**************************************************************************/
+/*!
+  \brief     Deletes an instance of the missing harmonics detector.
+
+
+  \return    none.
+
+*/
+/**************************************************************************/
+void FDKsbrEnc_DeleteSbrMissingHarmonicsDetector(
+    HANDLE_SBR_MISSING_HARMONICS_DETECTOR hSbrMHDet) {
+  if (hSbrMHDet) {
+    HANDLE_SBR_MISSING_HARMONICS_DETECTOR hs = hSbrMHDet;
+
+    FreeRam_Sbr_detectionVectors(&hs->detectionVectors[0]);
+    FreeRam_Sbr_guideVectorDetected(&hs->guideVectors[0].guideVectorDetected);
+    FreeRam_Sbr_guideVectorDiff(&hs->guideVectors[0].guideVectorDiff);
+    FreeRam_Sbr_guideVectorOrig(&hs->guideVectors[0].guideVectorOrig);
+    FreeRam_Sbr_prevEnvelopeCompensation(&hs->prevEnvelopeCompensation);
+    FreeRam_Sbr_guideScfb(&hs->guideScfb);
+  }
+}
+
+/**************************************************************************/
+/*!
+  \brief     Resets an instance of the missing harmonics detector.
+
+
+  \return    error code, noError if OK.
+
+*/
+/**************************************************************************/
+INT FDKsbrEnc_ResetSbrMissingHarmonicsDetector(
+    HANDLE_SBR_MISSING_HARMONICS_DETECTOR hSbrMissingHarmonicsDetector,
+    INT nSfb) {
+  int i;
+  FIXP_DBL tempGuide[MAX_FREQ_COEFFS];
+  UCHAR tempGuideInt[MAX_FREQ_COEFFS];
+  INT nSfbPrev;
+
+  nSfbPrev = hSbrMissingHarmonicsDetector->nSfb;
+  hSbrMissingHarmonicsDetector->nSfb = nSfb;
+
+  FDKmemcpy(tempGuideInt, hSbrMissingHarmonicsDetector->guideScfb,
+            nSfbPrev * sizeof(UCHAR));
+
+  if (nSfb > nSfbPrev) {
+    for (i = 0; i < (nSfb - nSfbPrev); i++) {
+      hSbrMissingHarmonicsDetector->guideScfb[i] = 0;
+    }
+
+    for (i = 0; i < nSfbPrev; i++) {
+      hSbrMissingHarmonicsDetector->guideScfb[i + (nSfb - nSfbPrev)] =
+          tempGuideInt[i];
+    }
+  } else {
+    for (i = 0; i < nSfb; i++) {
+      hSbrMissingHarmonicsDetector->guideScfb[i] =
+          tempGuideInt[i + (nSfbPrev - nSfb)];
+    }
+  }
+
+  FDKmemcpy(tempGuide,
+            hSbrMissingHarmonicsDetector->guideVectors[0].guideVectorDiff,
+            nSfbPrev * sizeof(FIXP_DBL));
+
+  if (nSfb > nSfbPrev) {
+    for (i = 0; i < (nSfb - nSfbPrev); i++) {
+      hSbrMissingHarmonicsDetector->guideVectors[0].guideVectorDiff[i] =
+          FL2FXCONST_DBL(0.0f);
+    }
+
+    for (i = 0; i < nSfbPrev; i++) {
+      hSbrMissingHarmonicsDetector->guideVectors[0]
+          .guideVectorDiff[i + (nSfb - nSfbPrev)] = tempGuide[i];
+    }
+  } else {
+    for (i = 0; i < nSfb; i++) {
+      hSbrMissingHarmonicsDetector->guideVectors[0].guideVectorDiff[i] =
+          tempGuide[i + (nSfbPrev - nSfb)];
+    }
+  }
+
+  FDKmemcpy(tempGuide,
+            hSbrMissingHarmonicsDetector->guideVectors[0].guideVectorOrig,
+            nSfbPrev * sizeof(FIXP_DBL));
+
+  if (nSfb > nSfbPrev) {
+    for (i = 0; i < (nSfb - nSfbPrev); i++) {
+      hSbrMissingHarmonicsDetector->guideVectors[0].guideVectorOrig[i] =
+          FL2FXCONST_DBL(0.0f);
+    }
+
+    for (i = 0; i < nSfbPrev; i++) {
+      hSbrMissingHarmonicsDetector->guideVectors[0]
+          .guideVectorOrig[i + (nSfb - nSfbPrev)] = tempGuide[i];
+    }
+  } else {
+    for (i = 0; i < nSfb; i++) {
+      hSbrMissingHarmonicsDetector->guideVectors[0].guideVectorOrig[i] =
+          tempGuide[i + (nSfbPrev - nSfb)];
+    }
+  }
+
+  FDKmemcpy(tempGuideInt,
+            hSbrMissingHarmonicsDetector->guideVectors[0].guideVectorDetected,
+            nSfbPrev * sizeof(UCHAR));
+
+  if (nSfb > nSfbPrev) {
+    for (i = 0; i < (nSfb - nSfbPrev); i++) {
+      hSbrMissingHarmonicsDetector->guideVectors[0].guideVectorDetected[i] = 0;
+    }
+
+    for (i = 0; i < nSfbPrev; i++) {
+      hSbrMissingHarmonicsDetector->guideVectors[0]
+          .guideVectorDetected[i + (nSfb - nSfbPrev)] = tempGuideInt[i];
+    }
+  } else {
+    for (i = 0; i < nSfb; i++) {
+      hSbrMissingHarmonicsDetector->guideVectors[0].guideVectorDetected[i] =
+          tempGuideInt[i + (nSfbPrev - nSfb)];
+    }
+  }
+
+  FDKmemcpy(tempGuideInt,
+            hSbrMissingHarmonicsDetector->prevEnvelopeCompensation,
+            nSfbPrev * sizeof(UCHAR));
+
+  if (nSfb > nSfbPrev) {
+    for (i = 0; i < (nSfb - nSfbPrev); i++) {
+      hSbrMissingHarmonicsDetector->prevEnvelopeCompensation[i] = 0;
+    }
+
+    for (i = 0; i < nSfbPrev; i++) {
+      hSbrMissingHarmonicsDetector
+          ->prevEnvelopeCompensation[i + (nSfb - nSfbPrev)] = tempGuideInt[i];
+    }
+  } else {
+    for (i = 0; i < nSfb; i++) {
+      hSbrMissingHarmonicsDetector->prevEnvelopeCompensation[i] =
+          tempGuideInt[i + (nSfbPrev - nSfb)];
+    }
+  }
+
+  return 0;
+}
diff --git a/fdk-aac/libSBRenc/src/mh_det.h b/fdk-aac/libSBRenc/src/mh_det.h
new file mode 100644
index 0000000..89d81b5
--- /dev/null
+++ b/fdk-aac/libSBRenc/src/mh_det.h
@@ -0,0 +1,204 @@
+/* -----------------------------------------------------------------------------
+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 encoder library ******************************
+
+   Author(s):
+
+   Description:
+
+*******************************************************************************/
+
+/*!
+  \file
+  \brief  missing harmonics detection header file $Revision: 92790 $
+*/
+
+#ifndef MH_DET_H
+#define MH_DET_H
+
+#include "sbr_encoder.h"
+#include "fram_gen.h"
+
+typedef struct {
+  FIXP_DBL thresHoldDiff;      /*!< threshold for tonality difference */
+  FIXP_DBL thresHoldDiffGuide; /*!< threshold for tonality difference for the
+                                  guide */
+  FIXP_DBL thresHoldTone;      /*!< threshold for tonality for a sine */
+  FIXP_DBL invThresHoldTone;
+  FIXP_DBL thresHoldToneGuide; /*!< threshold for tonality for a sine for the
+                                  guide */
+  FIXP_DBL sfmThresSbr;    /*!< tonality flatness measure threshold for the SBR
+                              signal.*/
+  FIXP_DBL sfmThresOrig;   /*!< tonality flatness measure threshold for the
+                              original signal.*/
+  FIXP_DBL decayGuideOrig; /*!< decay value of the tonality value of the guide
+                              for the tone. */
+  FIXP_DBL decayGuideDiff; /*!< decay value of the tonality value of the guide
+                              for the tonality difference. */
+  FIXP_DBL derivThresMaxLD64;   /*!< threshold for detecting LP character in a
+                                   signal. */
+  FIXP_DBL derivThresBelowLD64; /*!< threshold for detecting LP character in a
+                                   signal. */
+  FIXP_DBL derivThresAboveLD64; /*!< threshold for detecting LP character in a
+                                   signal. */
+} THRES_HOLDS;
+
+typedef struct {
+  INT deltaTime; /*!< maximum allowed transient distance (from frame border in
+                    number of qmf subband sample) for a frame to be considered a
+                    transient frame.*/
+  THRES_HOLDS thresHolds; /*!< the thresholds used for detection. */
+  INT maxComp; /*!< maximum alllowed compensation factor for the envelope data.
+                */
+} DETECTOR_PARAMETERS_MH;
+
+typedef struct {
+  FIXP_DBL *guideVectorDiff;
+  FIXP_DBL *guideVectorOrig;
+  UCHAR *guideVectorDetected;
+} GUIDE_VECTORS;
+
+typedef struct {
+  INT qmfNoChannels;
+  INT nSfb;
+  INT sampleFreq;
+  INT previousTransientFlag;
+  INT previousTransientFrame;
+  INT previousTransientPos;
+
+  INT noVecPerFrame;
+  INT transientPosOffset;
+
+  INT move;
+  INT totNoEst;
+  INT noEstPerFrame;
+  INT timeSlots;
+
+  UCHAR *guideScfb;
+  UCHAR *prevEnvelopeCompensation;
+  UCHAR *detectionVectors[MAX_NO_OF_ESTIMATES];
+  FIXP_DBL tonalityDiff[MAX_NO_OF_ESTIMATES / 2][MAX_FREQ_COEFFS];
+  FIXP_DBL sfmOrig[MAX_NO_OF_ESTIMATES / 2][MAX_FREQ_COEFFS];
+  FIXP_DBL sfmSbr[MAX_NO_OF_ESTIMATES / 2][MAX_FREQ_COEFFS];
+  const DETECTOR_PARAMETERS_MH *mhParams;
+  GUIDE_VECTORS guideVectors[MAX_NO_OF_ESTIMATES];
+} SBR_MISSING_HARMONICS_DETECTOR;
+
+typedef SBR_MISSING_HARMONICS_DETECTOR *HANDLE_SBR_MISSING_HARMONICS_DETECTOR;
+
+void FDKsbrEnc_SbrMissingHarmonicsDetectorQmf(
+    HANDLE_SBR_MISSING_HARMONICS_DETECTOR h_sbrMissingHarmonicsDetector,
+    FIXP_DBL **pQuotaBuffer, INT **pSignBuffer, SCHAR *indexVector,
+    const SBR_FRAME_INFO *pFrameInfo, const UCHAR *pTranInfo,
+    INT *pAddHarmonicsFlag, UCHAR *pAddHarmonicsScaleFactorBands,
+    const UCHAR *freqBandTable, INT nSfb, UCHAR *envelopeCompensation,
+    FIXP_DBL *pNrgVector);
+
+INT FDKsbrEnc_CreateSbrMissingHarmonicsDetector(
+    HANDLE_SBR_MISSING_HARMONICS_DETECTOR hSbrMHDet, INT chan);
+
+INT FDKsbrEnc_InitSbrMissingHarmonicsDetector(
+    HANDLE_SBR_MISSING_HARMONICS_DETECTOR h_sbrMissingHarmonicsDetector,
+    INT sampleFreq, INT frameSize, INT nSfb, INT qmfNoChannels, INT totNoEst,
+    INT move, INT noEstPerFrame, UINT sbrSyntaxFlags);
+
+void FDKsbrEnc_DeleteSbrMissingHarmonicsDetector(
+    HANDLE_SBR_MISSING_HARMONICS_DETECTOR h_sbrMissingHarmonicsDetector);
+
+INT FDKsbrEnc_ResetSbrMissingHarmonicsDetector(
+    HANDLE_SBR_MISSING_HARMONICS_DETECTOR hSbrMissingHarmonicsDetector,
+    INT nSfb);
+
+#endif
diff --git a/fdk-aac/libSBRenc/src/nf_est.cpp b/fdk-aac/libSBRenc/src/nf_est.cpp
new file mode 100644
index 0000000..290ec35
--- /dev/null
+++ b/fdk-aac/libSBRenc/src/nf_est.cpp
@@ -0,0 +1,612 @@
+/* -----------------------------------------------------------------------------
+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 encoder library ******************************
+
+   Author(s):
+
+   Description:
+
+*******************************************************************************/
+
+#include "nf_est.h"
+
+#include "sbr_misc.h"
+
+#include "genericStds.h"
+
+/* smoothFilter[4]  = {0.05857864376269f, 0.2f, 0.34142135623731f, 0.4f}; */
+static const FIXP_DBL smoothFilter[4] = {0x077f813d, 0x19999995, 0x2bb3b1f5,
+                                         0x33333335};
+
+/* static const INT smoothFilterLength = 4; */
+
+static const FIXP_DBL QuantOffset = (INT)0xfc000000; /* ld64(0.25) */
+
+#ifndef min
+#define min(a, b) (a < b ? a : b)
+#endif
+
+#ifndef max
+#define max(a, b) (a > b ? a : b)
+#endif
+
+#define NOISE_FLOOR_OFFSET_SCALING (4)
+
+/**************************************************************************/
+/*!
+  \brief     The function applies smoothing to the noise levels.
+
+
+
+  \return    none
+
+*/
+/**************************************************************************/
+static void smoothingOfNoiseLevels(
+    FIXP_DBL *NoiseLevels, /*!< pointer to noise-floor levels.*/
+    INT nEnvelopes,        /*!< Number of noise floor envelopes.*/
+    INT noNoiseBands, /*!< Number of noise bands for every noise floor envelope.
+                       */
+    FIXP_DBL prevNoiseLevels[NF_SMOOTHING_LENGTH]
+                            [MAX_NUM_NOISE_VALUES], /*!< Previous noise floor
+                                                       envelopes. */
+    const FIXP_DBL *
+        pSmoothFilter, /*!< filter used for smoothing the noise floor levels. */
+    INT transientFlag) /*!< flag indicating if a transient is present*/
+
+{
+  INT i, band, env;
+  FIXP_DBL accu;
+
+  for (env = 0; env < nEnvelopes; env++) {
+    if (transientFlag) {
+      for (i = 0; i < NF_SMOOTHING_LENGTH; i++) {
+        FDKmemcpy(prevNoiseLevels[i], NoiseLevels + env * noNoiseBands,
+                  noNoiseBands * sizeof(FIXP_DBL));
+      }
+    } else {
+      for (i = 1; i < NF_SMOOTHING_LENGTH; i++) {
+        FDKmemcpy(prevNoiseLevels[i - 1], prevNoiseLevels[i],
+                  noNoiseBands * sizeof(FIXP_DBL));
+      }
+      FDKmemcpy(prevNoiseLevels[NF_SMOOTHING_LENGTH - 1],
+                NoiseLevels + env * noNoiseBands,
+                noNoiseBands * sizeof(FIXP_DBL));
+    }
+
+    for (band = 0; band < noNoiseBands; band++) {
+      accu = FL2FXCONST_DBL(0.0f);
+      for (i = 0; i < NF_SMOOTHING_LENGTH; i++) {
+        accu += fMultDiv2(pSmoothFilter[i], prevNoiseLevels[i][band]);
+      }
+      FDK_ASSERT((band + env * noNoiseBands) < MAX_NUM_NOISE_VALUES);
+      NoiseLevels[band + env * noNoiseBands] = accu << 1;
+    }
+  }
+}
+
+/**************************************************************************/
+/*!
+  \brief     Does the noise floor level estiamtion.
+
+  The noiseLevel samples are scaled by the factor 0.25
+
+  \return    none
+
+*/
+/**************************************************************************/
+static void qmfBasedNoiseFloorDetection(
+    FIXP_DBL *noiseLevel,            /*!< Pointer to vector to
+                                        store the noise levels
+                                        in.*/
+    FIXP_DBL **quotaMatrixOrig,      /*!< Matrix holding the quota
+                                        values of the original. */
+    SCHAR *indexVector,              /*!< Index vector to obtain the
+                                        patched data. */
+    INT startIndex,                  /*!< Start index. */
+    INT stopIndex,                   /*!< Stop index. */
+    INT startChannel,                /*!< Start channel of the current
+                                        noise floor band.*/
+    INT stopChannel,                 /*!< Stop channel of the current
+                                        noise floor band. */
+    FIXP_DBL ana_max_level,          /*!< Maximum level of the
+                                        adaptive noise.*/
+    FIXP_DBL noiseFloorOffset,       /*!< Noise floor offset. */
+    INT missingHarmonicFlag,         /*!< Flag indicating if a
+                                        strong tonal component
+                                        is missing.*/
+    FIXP_DBL weightFac,              /*!< Weightening factor for the
+                                        difference between orig and sbr.
+                                      */
+    INVF_MODE diffThres,             /*!< Threshold value to control the
+                                        inverse filtering decision.*/
+    INVF_MODE inverseFilteringLevel) /*!< Inverse filtering
+                                        level of the current
+                                        band.*/
+{
+  INT scale, l, k;
+  FIXP_DBL meanOrig = FL2FXCONST_DBL(0.0f), meanSbr = FL2FXCONST_DBL(0.0f),
+           diff;
+  FIXP_DBL invIndex = GetInvInt(stopIndex - startIndex);
+  FIXP_DBL invChannel = GetInvInt(stopChannel - startChannel);
+  FIXP_DBL accu;
+
+  /*
+  Calculate the mean value, over the current time segment, for the original, the
+  HFR and the difference, over all channels in the current frequency range.
+  */
+
+  if (missingHarmonicFlag == 1) {
+    for (l = startChannel; l < stopChannel; l++) {
+      /* tonalityOrig */
+      accu = FL2FXCONST_DBL(0.0f);
+      for (k = startIndex; k < stopIndex; k++) {
+        accu += fMultDiv2(quotaMatrixOrig[k][l], invIndex);
+      }
+      meanOrig = fixMax(meanOrig, (accu << 1));
+
+      /* tonalitySbr */
+      accu = FL2FXCONST_DBL(0.0f);
+      for (k = startIndex; k < stopIndex; k++) {
+        accu += fMultDiv2(quotaMatrixOrig[k][indexVector[l]], invIndex);
+      }
+      meanSbr = fixMax(meanSbr, (accu << 1));
+    }
+  } else {
+    for (l = startChannel; l < stopChannel; l++) {
+      /* tonalityOrig */
+      accu = FL2FXCONST_DBL(0.0f);
+      for (k = startIndex; k < stopIndex; k++) {
+        accu += fMultDiv2(quotaMatrixOrig[k][l], invIndex);
+      }
+      meanOrig += fMult((accu << 1), invChannel);
+
+      /* tonalitySbr */
+      accu = FL2FXCONST_DBL(0.0f);
+      for (k = startIndex; k < stopIndex; k++) {
+        accu += fMultDiv2(quotaMatrixOrig[k][indexVector[l]], invIndex);
+      }
+      meanSbr += fMult((accu << 1), invChannel);
+    }
+  }
+
+  /* Small fix to avoid noise during silent passages.*/
+  if (meanOrig <= FL2FXCONST_DBL(0.000976562f * RELAXATION_FLOAT) &&
+      meanSbr <= FL2FXCONST_DBL(0.000976562f * RELAXATION_FLOAT)) {
+    meanOrig = FL2FXCONST_DBL(101.5936673f * RELAXATION_FLOAT);
+    meanSbr = FL2FXCONST_DBL(101.5936673f * RELAXATION_FLOAT);
+  }
+
+  meanOrig = fixMax(meanOrig, RELAXATION);
+  meanSbr = fixMax(meanSbr, RELAXATION);
+
+  if (missingHarmonicFlag == 1 || inverseFilteringLevel == INVF_MID_LEVEL ||
+      inverseFilteringLevel == INVF_LOW_LEVEL ||
+      inverseFilteringLevel == INVF_OFF || inverseFilteringLevel <= diffThres) {
+    diff = RELAXATION;
+  } else {
+    accu = fDivNorm(meanSbr, meanOrig, &scale);
+
+    diff = fixMax(RELAXATION, fMult(RELAXATION_FRACT, fMult(weightFac, accu)) >>
+                                  (RELAXATION_SHIFT - scale));
+  }
+
+  /*
+   * noise Level is now a positive value, i.e.
+   * the more harmonic the signal is the higher noise level,
+   * this makes no sense so we change the sign.
+   *********************************************************/
+  accu = fDivNorm(diff, meanOrig, &scale);
+  scale -= 2;
+
+  if ((scale > 0) && (accu > ((FIXP_DBL)MAXVAL_DBL) >> scale)) {
+    *noiseLevel = (FIXP_DBL)MAXVAL_DBL;
+  } else {
+    *noiseLevel = scaleValue(accu, scale);
+  }
+
+  /*
+   * Add a noise floor offset to compensate for bias in the detector
+   *****************************************************************/
+  if (!missingHarmonicFlag) {
+    *noiseLevel = fixMin(fMult(*noiseLevel, noiseFloorOffset),
+                         (FIXP_DBL)MAXVAL_DBL >> NOISE_FLOOR_OFFSET_SCALING)
+                  << NOISE_FLOOR_OFFSET_SCALING;
+  }
+
+  /*
+   * check to see that we don't exceed the maximum allowed level
+   **************************************************************/
+  *noiseLevel =
+      fixMin(*noiseLevel,
+             ana_max_level); /* ana_max_level is scaled with factor 0.25 */
+}
+
+/**************************************************************************/
+/*!
+  \brief     Does the noise floor level estiamtion.
+  The function calls the Noisefloor estimation function
+  for the time segments decided based upon the transient
+  information. The block is always divided into one or two segments.
+
+
+  \return    none
+
+*/
+/**************************************************************************/
+void FDKsbrEnc_sbrNoiseFloorEstimateQmf(
+    HANDLE_SBR_NOISE_FLOOR_ESTIMATE
+        h_sbrNoiseFloorEstimate, /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct
+                                  */
+    const SBR_FRAME_INFO
+        *frame_info, /*!< Time frequency grid of the current frame. */
+    FIXP_DBL
+        *noiseLevels, /*!< Pointer to vector to store the noise levels in.*/
+    FIXP_DBL **quotaMatrixOrig, /*!< Matrix holding the quota values of the
+                                   original. */
+    SCHAR *indexVector,         /*!< Index vector to obtain the patched data. */
+    INT missingHarmonicsFlag,   /*!< Flag indicating if a strong tonal component
+                                   will be missing. */
+    INT startIndex,             /*!< Start index. */
+    UINT numberOfEstimatesPerFrame, /*!< The number of tonality estimates per
+                                       frame. */
+    int transientFrame, /*!< A flag indicating if a transient is present. */
+    INVF_MODE *pInvFiltLevels, /*!< Pointer to the vector holding the inverse
+                                  filtering levels. */
+    UINT sbrSyntaxFlags)
+
+{
+  INT nNoiseEnvelopes, startPos[2], stopPos[2], env, band;
+
+  INT noNoiseBands = h_sbrNoiseFloorEstimate->noNoiseBands;
+  INT *freqBandTable = h_sbrNoiseFloorEstimate->freqBandTableQmf;
+
+  nNoiseEnvelopes = frame_info->nNoiseEnvelopes;
+
+  startPos[0] = startIndex;
+
+  if (nNoiseEnvelopes == 1) {
+    stopPos[0] = startIndex + min(numberOfEstimatesPerFrame, 2);
+  } else {
+    stopPos[0] = startIndex + 1;
+    startPos[1] = startIndex + 1;
+    stopPos[1] = startIndex + min(numberOfEstimatesPerFrame, 2);
+  }
+
+  /*
+   * Estimate the noise floor.
+   **************************************/
+  for (env = 0; env < nNoiseEnvelopes; env++) {
+    for (band = 0; band < noNoiseBands; band++) {
+      FDK_ASSERT((band + env * noNoiseBands) < MAX_NUM_NOISE_VALUES);
+      qmfBasedNoiseFloorDetection(
+          &noiseLevels[band + env * noNoiseBands], quotaMatrixOrig, indexVector,
+          startPos[env], stopPos[env], freqBandTable[band],
+          freqBandTable[band + 1], h_sbrNoiseFloorEstimate->ana_max_level,
+          h_sbrNoiseFloorEstimate->noiseFloorOffset[band], missingHarmonicsFlag,
+          h_sbrNoiseFloorEstimate->weightFac,
+          h_sbrNoiseFloorEstimate->diffThres, pInvFiltLevels[band]);
+    }
+  }
+
+  /*
+   * Smoothing of the values.
+   **************************/
+  smoothingOfNoiseLevels(noiseLevels, nNoiseEnvelopes,
+                         h_sbrNoiseFloorEstimate->noNoiseBands,
+                         h_sbrNoiseFloorEstimate->prevNoiseLevels,
+                         h_sbrNoiseFloorEstimate->smoothFilter, transientFrame);
+
+  /* quantisation*/
+  for (env = 0; env < nNoiseEnvelopes; env++) {
+    for (band = 0; band < noNoiseBands; band++) {
+      FDK_ASSERT((band + env * noNoiseBands) < MAX_NUM_NOISE_VALUES);
+      noiseLevels[band + env * noNoiseBands] =
+          (FIXP_DBL)NOISE_FLOOR_OFFSET_64 -
+          (FIXP_DBL)CalcLdData(noiseLevels[band + env * noNoiseBands] +
+                               (FIXP_DBL)1) +
+          QuantOffset;
+    }
+  }
+}
+
+/**************************************************************************/
+/*!
+  \brief
+
+
+  \return    errorCode, noError if successful
+
+*/
+/**************************************************************************/
+static INT downSampleLoRes(INT *v_result,                 /*!<    */
+                           INT num_result,                /*!<    */
+                           const UCHAR *freqBandTableRef, /*!<    */
+                           INT num_Ref)                   /*!<    */
+{
+  INT step;
+  INT i, j;
+  INT org_length, result_length;
+  INT v_index[MAX_FREQ_COEFFS / 2];
+
+  /* init */
+  org_length = num_Ref;
+  result_length = num_result;
+
+  v_index[0] = 0; /* Always use left border */
+  i = 0;
+  while (org_length > 0) /* Create downsample vector */
+  {
+    i++;
+    step = org_length / result_length; /* floor; */
+    org_length = org_length - step;
+    result_length--;
+    v_index[i] = v_index[i - 1] + step;
+  }
+
+  if (i != num_result) /* Should never happen */
+    return (1);        /* error downsampling */
+
+  for (j = 0; j <= i;
+       j++) /* Use downsample vector to index LoResolution vector. */
+  {
+    v_result[j] = freqBandTableRef[v_index[j]];
+  }
+
+  return (0);
+}
+
+/**************************************************************************/
+/*!
+  \brief    Initialize an instance of the noise floor level estimation module.
+
+
+  \return    errorCode, noError if successful
+
+*/
+/**************************************************************************/
+INT FDKsbrEnc_InitSbrNoiseFloorEstimate(
+    HANDLE_SBR_NOISE_FLOOR_ESTIMATE
+        h_sbrNoiseFloorEstimate, /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct
+                                  */
+    INT ana_max_level,           /*!< Maximum level of the adaptive noise. */
+    const UCHAR *freqBandTable,  /*!< Frequency band table. */
+    INT nSfb,                    /*!< Number of frequency bands. */
+    INT noiseBands,              /*!< Number of noise bands per octave. */
+    INT noiseFloorOffset,        /*!< Noise floor offset. */
+    INT timeSlots,               /*!< Number of time slots in a frame. */
+    UINT useSpeechConfig /*!< Flag: adapt tuning parameters according to speech
+                          */
+) {
+  INT i, qexp, qtmp;
+  FIXP_DBL tmp, exp;
+
+  FDKmemclear(h_sbrNoiseFloorEstimate, sizeof(SBR_NOISE_FLOOR_ESTIMATE));
+
+  h_sbrNoiseFloorEstimate->smoothFilter = smoothFilter;
+  if (useSpeechConfig) {
+    h_sbrNoiseFloorEstimate->weightFac = (FIXP_DBL)MAXVAL_DBL;
+    h_sbrNoiseFloorEstimate->diffThres = INVF_LOW_LEVEL;
+  } else {
+    h_sbrNoiseFloorEstimate->weightFac = FL2FXCONST_DBL(0.25f);
+    h_sbrNoiseFloorEstimate->diffThres = INVF_MID_LEVEL;
+  }
+
+  h_sbrNoiseFloorEstimate->timeSlots = timeSlots;
+  h_sbrNoiseFloorEstimate->noiseBands = noiseBands;
+
+  /* h_sbrNoiseFloorEstimate->ana_max_level is scaled by 0.25  */
+  switch (ana_max_level) {
+    case 6:
+      h_sbrNoiseFloorEstimate->ana_max_level = (FIXP_DBL)MAXVAL_DBL;
+      break;
+    case 3:
+      h_sbrNoiseFloorEstimate->ana_max_level = FL2FXCONST_DBL(0.5);
+      break;
+    case -3:
+      h_sbrNoiseFloorEstimate->ana_max_level = FL2FXCONST_DBL(0.125);
+      break;
+    default:
+      /* Should not enter here */
+      h_sbrNoiseFloorEstimate->ana_max_level = (FIXP_DBL)MAXVAL_DBL;
+      break;
+  }
+
+  /*
+    calculate number of noise bands and allocate
+  */
+  if (FDKsbrEnc_resetSbrNoiseFloorEstimate(h_sbrNoiseFloorEstimate,
+                                           freqBandTable, nSfb))
+    return (1);
+
+  if (noiseFloorOffset == 0) {
+    tmp = ((FIXP_DBL)MAXVAL_DBL) >> NOISE_FLOOR_OFFSET_SCALING;
+  } else {
+    /* noiseFloorOffset has to be smaller than 12, because
+       the result of the calculation below must be smaller than 1:
+       (2^(noiseFloorOffset/3))*2^4<1                                        */
+    FDK_ASSERT(noiseFloorOffset < 12);
+
+    /* Assumes the noise floor offset in tuning table are in q31    */
+    /* Change the qformat here when non-zero values would be filled */
+    exp = fDivNorm((FIXP_DBL)noiseFloorOffset, 3, &qexp);
+    tmp = fPow(2, DFRACT_BITS - 1, exp, qexp, &qtmp);
+    tmp = scaleValue(tmp, qtmp - NOISE_FLOOR_OFFSET_SCALING);
+  }
+
+  for (i = 0; i < h_sbrNoiseFloorEstimate->noNoiseBands; i++) {
+    h_sbrNoiseFloorEstimate->noiseFloorOffset[i] = tmp;
+  }
+
+  return (0);
+}
+
+/**************************************************************************/
+/*!
+  \brief     Resets the current instance of the noise floor estiamtion
+          module.
+
+
+  \return    errorCode, noError if successful
+
+*/
+/**************************************************************************/
+INT FDKsbrEnc_resetSbrNoiseFloorEstimate(
+    HANDLE_SBR_NOISE_FLOOR_ESTIMATE
+        h_sbrNoiseFloorEstimate, /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct
+                                  */
+    const UCHAR *freqBandTable,  /*!< Frequency band table. */
+    INT nSfb /*!< Number of bands in the frequency band table. */
+) {
+  INT k2, kx;
+
+  /*
+   * Calculate number of noise bands
+   ***********************************/
+  k2 = freqBandTable[nSfb];
+  kx = freqBandTable[0];
+  if (h_sbrNoiseFloorEstimate->noiseBands == 0) {
+    h_sbrNoiseFloorEstimate->noNoiseBands = 1;
+  } else {
+    /*
+     * Calculate number of noise bands 1,2 or 3 bands/octave
+     ********************************************************/
+    FIXP_DBL tmp, ratio, lg2;
+    INT ratio_e, qlg2, nNoiseBands;
+
+    ratio = fDivNorm(k2, kx, &ratio_e);
+    lg2 = fLog2(ratio, ratio_e, &qlg2);
+    tmp = fMult((FIXP_DBL)(h_sbrNoiseFloorEstimate->noiseBands << 24), lg2);
+    tmp = scaleValue(tmp, qlg2 - 23);
+
+    nNoiseBands = (INT)((tmp + (FIXP_DBL)1) >> 1);
+
+    if (nNoiseBands > MAX_NUM_NOISE_COEFFS) {
+      nNoiseBands = MAX_NUM_NOISE_COEFFS;
+    }
+
+    if (nNoiseBands == 0) {
+      nNoiseBands = 1;
+    }
+
+    h_sbrNoiseFloorEstimate->noNoiseBands = nNoiseBands;
+  }
+
+  return (downSampleLoRes(h_sbrNoiseFloorEstimate->freqBandTableQmf,
+                          h_sbrNoiseFloorEstimate->noNoiseBands, freqBandTable,
+                          nSfb));
+}
+
+/**************************************************************************/
+/*!
+  \brief     Deletes the current instancce of the noise floor level
+  estimation module.
+
+
+  \return    none
+
+*/
+/**************************************************************************/
+void FDKsbrEnc_deleteSbrNoiseFloorEstimate(
+    HANDLE_SBR_NOISE_FLOOR_ESTIMATE
+        h_sbrNoiseFloorEstimate) /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct
+                                  */
+{
+  if (h_sbrNoiseFloorEstimate) {
+    /*
+      nothing to do
+    */
+  }
+}
diff --git a/fdk-aac/libSBRenc/src/nf_est.h b/fdk-aac/libSBRenc/src/nf_est.h
new file mode 100644
index 0000000..c2f16e9
--- /dev/null
+++ b/fdk-aac/libSBRenc/src/nf_est.h
@@ -0,0 +1,185 @@
+/* -----------------------------------------------------------------------------
+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 encoder library ******************************
+
+   Author(s):
+
+   Description:
+
+*******************************************************************************/
+
+/*!
+  \file
+  \brief  Noise floor estimation structs and prototypes $Revision: 92790 $
+*/
+
+#ifndef NF_EST_H
+#define NF_EST_H
+
+#include "sbr_encoder.h"
+#include "fram_gen.h"
+
+#define NF_SMOOTHING_LENGTH 4 /*!< Smoothing length of the noise floors. */
+
+typedef struct {
+  FIXP_DBL
+  prevNoiseLevels[NF_SMOOTHING_LENGTH]
+                 [MAX_NUM_NOISE_VALUES]; /*!< The previous noise levels. */
+  FIXP_DBL noiseFloorOffset
+      [MAX_NUM_NOISE_VALUES];   /*!< Noise floor offset, scaled with
+                                   NOISE_FLOOR_OFFSET_SCALING */
+  const FIXP_DBL *smoothFilter; /*!< Smoothing filter to use. */
+  FIXP_DBL ana_max_level;       /*!< Max level allowed.   */
+  FIXP_DBL weightFac; /*!< Weightening factor for the difference between orig
+                         and sbr. */
+  INT freqBandTableQmf[MAX_NUM_NOISE_VALUES +
+                       1]; /*!< Frequncy band table for the noise floor bands.*/
+  INT noNoiseBands;        /*!< Number of noisebands. */
+  INT noiseBands;          /*!< NoiseBands switch 4 bit.*/
+  INT timeSlots;           /*!< Number of timeslots in a frame. */
+  INVF_MODE diffThres;     /*!< Threshold value to control the inverse filtering
+                              decision */
+} SBR_NOISE_FLOOR_ESTIMATE;
+
+typedef SBR_NOISE_FLOOR_ESTIMATE *HANDLE_SBR_NOISE_FLOOR_ESTIMATE;
+
+void FDKsbrEnc_sbrNoiseFloorEstimateQmf(
+    HANDLE_SBR_NOISE_FLOOR_ESTIMATE
+        h_sbrNoiseFloorEstimate, /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct
+                                  */
+    const SBR_FRAME_INFO
+        *frame_info, /*!< Time frequency grid of the current frame. */
+    FIXP_DBL
+        *noiseLevels, /*!< Pointer to vector to store the noise levels in.*/
+    FIXP_DBL **quotaMatrixOrig, /*!< Matrix holding the quota values of the
+                                   original. */
+    SCHAR *indexVector,         /*!< Index vector to obtain the patched data. */
+    INT missingHarmonicsFlag,   /*!< Flag indicating if a strong tonal component
+                                   will be missing. */
+    INT startIndex,             /*!< Start index. */
+    UINT numberOfEstimatesPerFrame, /*!< The number of tonality estimates per
+                                       frame. */
+    INT transientFrame, /*!< A flag indicating if a transient is present. */
+    INVF_MODE *pInvFiltLevels, /*!< Pointer to the vector holding the inverse
+                                  filtering levels. */
+    UINT sbrSyntaxFlags);
+
+INT FDKsbrEnc_InitSbrNoiseFloorEstimate(
+    HANDLE_SBR_NOISE_FLOOR_ESTIMATE
+        h_sbrNoiseFloorEstimate, /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct
+                                  */
+    INT ana_max_level,           /*!< Maximum level of the adaptive noise. */
+    const UCHAR *freqBandTable,  /*!< Frequany band table. */
+    INT nSfb,                    /*!< Number of frequency bands. */
+    INT noiseBands,              /*!< Number of noise bands per octave. */
+    INT noiseFloorOffset,        /*!< Noise floor offset. */
+    INT timeSlots,               /*!< Number of time slots in a frame. */
+    UINT useSpeechConfig /*!< Flag: adapt tuning parameters according to speech
+                          */
+);
+
+INT FDKsbrEnc_resetSbrNoiseFloorEstimate(
+    HANDLE_SBR_NOISE_FLOOR_ESTIMATE
+        h_sbrNoiseFloorEstimate, /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct
+                                  */
+    const UCHAR *freqBandTable,  /*!< Frequany band table. */
+    INT nSfb); /*!< Number of bands in the frequency band table. */
+
+void FDKsbrEnc_deleteSbrNoiseFloorEstimate(
+    HANDLE_SBR_NOISE_FLOOR_ESTIMATE
+        h_sbrNoiseFloorEstimate); /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct
+                                   */
+
+#endif
diff --git a/fdk-aac/libSBRenc/src/ps_bitenc.cpp b/fdk-aac/libSBRenc/src/ps_bitenc.cpp
new file mode 100644
index 0000000..e30af2a
--- /dev/null
+++ b/fdk-aac/libSBRenc/src/ps_bitenc.cpp
@@ -0,0 +1,624 @@
+/* -----------------------------------------------------------------------------
+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 encoder library ******************************
+
+   Author(s):   N. Rettelbach
+
+   Description: Parametric Stereo bitstream encoder
+
+*******************************************************************************/
+
+#include "ps_bitenc.h"
+
+#include "ps_main.h"
+
+static inline UCHAR FDKsbrEnc_WriteBits_ps(HANDLE_FDK_BITSTREAM hBitStream,
+                                           UINT value,
+                                           const UINT numberOfBits) {
+  /* hBitStream == NULL happens here intentionally */
+  if (hBitStream != NULL) {
+    FDKwriteBits(hBitStream, value, numberOfBits);
+  }
+  return numberOfBits;
+}
+
+#define SI_SBR_EXTENSION_SIZE_BITS 4
+#define SI_SBR_EXTENSION_ESC_COUNT_BITS 8
+#define SI_SBR_EXTENSION_ID_BITS 2
+#define EXTENSION_ID_PS_CODING 2
+#define PS_EXT_ID_V0 0
+
+static const INT iidDeltaCoarse_Offset = 14;
+static const INT iidDeltaCoarse_MaxVal = 28;
+static const INT iidDeltaFine_Offset = 30;
+static const INT iidDeltaFine_MaxVal = 60;
+
+/* PS Stereo Huffmantable: iidDeltaFreqCoarse */
+static const UINT iidDeltaFreqCoarse_Length[] = {
+    17, 17, 17, 17, 16, 15, 13, 10, 9,  7,  6,  5,  4,  3, 1,
+    3,  4,  5,  6,  6,  8,  11, 13, 14, 14, 15, 17, 18, 18};
+static const UINT iidDeltaFreqCoarse_Code[] = {
+    0x0001fffb, 0x0001fffc, 0x0001fffd, 0x0001fffa, 0x0000fffc, 0x00007ffc,
+    0x00001ffd, 0x000003fe, 0x000001fe, 0x0000007e, 0x0000003c, 0x0000001d,
+    0x0000000d, 0x00000005, 0000000000, 0x00000004, 0x0000000c, 0x0000001c,
+    0x0000003d, 0x0000003e, 0x000000fe, 0x000007fe, 0x00001ffc, 0x00003ffc,
+    0x00003ffd, 0x00007ffd, 0x0001fffe, 0x0003fffe, 0x0003ffff};
+
+/* PS Stereo Huffmantable: iidDeltaFreqFine */
+static const UINT iidDeltaFreqFine_Length[] = {
+    18, 18, 18, 18, 18, 18, 18, 18, 18, 17, 18, 17, 17, 16, 16, 15,
+    14, 14, 13, 12, 12, 11, 10, 10, 8,  7,  6,  5,  4,  3,  1,  3,
+    4,  5,  6,  7,  8,  9,  10, 11, 11, 12, 13, 14, 14, 15, 16, 16,
+    17, 17, 18, 17, 18, 18, 18, 18, 18, 18, 18, 18, 18};
+static const UINT iidDeltaFreqFine_Code[] = {
+    0x0001feb4, 0x0001feb5, 0x0001fd76, 0x0001fd77, 0x0001fd74, 0x0001fd75,
+    0x0001fe8a, 0x0001fe8b, 0x0001fe88, 0x0000fe80, 0x0001feb6, 0x0000fe82,
+    0x0000feb8, 0x00007f42, 0x00007fae, 0x00003faf, 0x00001fd1, 0x00001fe9,
+    0x00000fe9, 0x000007ea, 0x000007fb, 0x000003fb, 0x000001fb, 0x000001ff,
+    0x0000007c, 0x0000003c, 0x0000001c, 0x0000000c, 0000000000, 0x00000001,
+    0x00000001, 0x00000002, 0x00000001, 0x0000000d, 0x0000001d, 0x0000003d,
+    0x0000007d, 0x000000fc, 0x000001fc, 0x000003fc, 0x000003f4, 0x000007eb,
+    0x00000fea, 0x00001fea, 0x00001fd6, 0x00003fd0, 0x00007faf, 0x00007f43,
+    0x0000feb9, 0x0000fe83, 0x0001feb7, 0x0000fe81, 0x0001fe89, 0x0001fe8e,
+    0x0001fe8f, 0x0001fe8c, 0x0001fe8d, 0x0001feb2, 0x0001feb3, 0x0001feb0,
+    0x0001feb1};
+
+/* PS Stereo Huffmantable: iidDeltaTimeCoarse */
+static const UINT iidDeltaTimeCoarse_Length[] = {
+    19, 19, 19, 20, 20, 20, 17, 15, 12, 10, 8,  6,  4,  2, 1,
+    3,  5,  7,  9,  11, 13, 14, 17, 19, 20, 20, 20, 20, 20};
+static const UINT iidDeltaTimeCoarse_Code[] = {
+    0x0007fff9, 0x0007fffa, 0x0007fffb, 0x000ffff8, 0x000ffff9, 0x000ffffa,
+    0x0001fffd, 0x00007ffe, 0x00000ffe, 0x000003fe, 0x000000fe, 0x0000003e,
+    0x0000000e, 0x00000002, 0000000000, 0x00000006, 0x0000001e, 0x0000007e,
+    0x000001fe, 0x000007fe, 0x00001ffe, 0x00003ffe, 0x0001fffc, 0x0007fff8,
+    0x000ffffb, 0x000ffffc, 0x000ffffd, 0x000ffffe, 0x000fffff};
+
+/* PS Stereo Huffmantable: iidDeltaTimeFine */
+static const UINT iidDeltaTimeFine_Length[] = {
+    16, 16, 16, 16, 16, 16, 16, 16, 16, 15, 15, 15, 15, 15, 15, 14,
+    14, 13, 13, 13, 12, 12, 11, 10, 9,  9,  7,  6,  5,  3,  1,  2,
+    5,  6,  7,  8,  9,  10, 11, 11, 12, 12, 13, 13, 14, 14, 15, 15,
+    15, 15, 16, 16, 16, 16, 16, 16, 16, 16, 16, 16, 16};
+static const UINT iidDeltaTimeFine_Code[] = {
+    0x00004ed4, 0x00004ed5, 0x00004ece, 0x00004ecf, 0x00004ecc, 0x00004ed6,
+    0x00004ed8, 0x00004f46, 0x00004f60, 0x00002718, 0x00002719, 0x00002764,
+    0x00002765, 0x0000276d, 0x000027b1, 0x000013b7, 0x000013d6, 0x000009c7,
+    0x000009e9, 0x000009ed, 0x000004ee, 0x000004f7, 0x00000278, 0x00000139,
+    0x0000009a, 0x0000009f, 0x00000020, 0x00000011, 0x0000000a, 0x00000003,
+    0x00000001, 0000000000, 0x0000000b, 0x00000012, 0x00000021, 0x0000004c,
+    0x0000009b, 0x0000013a, 0x00000279, 0x00000270, 0x000004ef, 0x000004e2,
+    0x000009ea, 0x000009d8, 0x000013d7, 0x000013d0, 0x000027b2, 0x000027a2,
+    0x0000271a, 0x0000271b, 0x00004f66, 0x00004f67, 0x00004f61, 0x00004f47,
+    0x00004ed9, 0x00004ed7, 0x00004ecd, 0x00004ed2, 0x00004ed3, 0x00004ed0,
+    0x00004ed1};
+
+static const INT iccDelta_Offset = 7;
+static const INT iccDelta_MaxVal = 14;
+/* PS Stereo Huffmantable: iccDeltaFreq */
+static const UINT iccDeltaFreq_Length[] = {14, 14, 12, 10, 7, 5,  3, 1,
+                                           2,  4,  6,  8,  9, 11, 13};
+static const UINT iccDeltaFreq_Code[] = {
+    0x00003fff, 0x00003ffe, 0x00000ffe, 0x000003fe, 0x0000007e,
+    0x0000001e, 0x00000006, 0000000000, 0x00000002, 0x0000000e,
+    0x0000003e, 0x000000fe, 0x000001fe, 0x000007fe, 0x00001ffe};
+
+/* PS Stereo Huffmantable: iccDeltaTime */
+static const UINT iccDeltaTime_Length[] = {14, 13, 11, 9, 7,  5,  3, 1,
+                                           2,  4,  6,  8, 10, 12, 14};
+static const UINT iccDeltaTime_Code[] = {
+    0x00003ffe, 0x00001ffe, 0x000007fe, 0x000001fe, 0x0000007e,
+    0x0000001e, 0x00000006, 0000000000, 0x00000002, 0x0000000e,
+    0x0000003e, 0x000000fe, 0x000003fe, 0x00000ffe, 0x00003fff};
+
+static const INT ipdDelta_Offset = 0;
+static const INT ipdDelta_MaxVal = 7;
+/* PS Stereo Huffmantable: ipdDeltaFreq */
+static const UINT ipdDeltaFreq_Length[] = {1, 3, 4, 4, 4, 4, 4, 4};
+static const UINT ipdDeltaFreq_Code[] = {0x00000001, 0000000000, 0x00000006,
+                                         0x00000004, 0x00000002, 0x00000003,
+                                         0x00000005, 0x00000007};
+
+/* PS Stereo Huffmantable: ipdDeltaTime */
+static const UINT ipdDeltaTime_Length[] = {1, 3, 4, 5, 5, 4, 4, 3};
+static const UINT ipdDeltaTime_Code[] = {0x00000001, 0x00000002, 0x00000002,
+                                         0x00000003, 0x00000002, 0000000000,
+                                         0x00000003, 0x00000003};
+
+static const INT opdDelta_Offset = 0;
+static const INT opdDelta_MaxVal = 7;
+/* PS Stereo Huffmantable: opdDeltaFreq */
+static const UINT opdDeltaFreq_Length[] = {1, 3, 4, 4, 5, 5, 4, 3};
+static const UINT opdDeltaFreq_Code[] = {
+    0x00000001, 0x00000001, 0x00000006, 0x00000004,
+    0x0000000f, 0x0000000e, 0x00000005, 0000000000,
+};
+
+/* PS Stereo Huffmantable: opdDeltaTime */
+static const UINT opdDeltaTime_Length[] = {1, 3, 4, 5, 5, 4, 4, 3};
+static const UINT opdDeltaTime_Code[] = {0x00000001, 0x00000002, 0x00000001,
+                                         0x00000007, 0x00000006, 0000000000,
+                                         0x00000002, 0x00000003};
+
+static INT getNoBands(const INT mode) {
+  INT noBands = 0;
+
+  switch (mode) {
+    case 0:
+    case 3: /* coarse */
+      noBands = PS_BANDS_COARSE;
+      break;
+    case 1:
+    case 4: /* mid */
+      noBands = PS_BANDS_MID;
+      break;
+    case 2:
+    case 5:  /* fine not supported */
+    default: /* coarse as default */
+      noBands = PS_BANDS_COARSE;
+  }
+
+  return noBands;
+}
+
+static INT getIIDRes(INT iidMode) {
+  if (iidMode < 3)
+    return PS_IID_RES_COARSE;
+  else
+    return PS_IID_RES_FINE;
+}
+
+static INT encodeDeltaFreq(HANDLE_FDK_BITSTREAM hBitBuf, const INT *val,
+                           const INT nBands, const UINT *codeTable,
+                           const UINT *lengthTable, const INT tableOffset,
+                           const INT maxVal, INT *error) {
+  INT bitCnt = 0;
+  INT lastVal = 0;
+  INT band;
+
+  for (band = 0; band < nBands; band++) {
+    INT delta = (val[band] - lastVal) + tableOffset;
+    lastVal = val[band];
+    if ((delta > maxVal) || (delta < 0)) {
+      *error = 1;
+      delta = delta > 0 ? maxVal : 0;
+    }
+    bitCnt +=
+        FDKsbrEnc_WriteBits_ps(hBitBuf, codeTable[delta], lengthTable[delta]);
+  }
+
+  return bitCnt;
+}
+
+static INT encodeDeltaTime(HANDLE_FDK_BITSTREAM hBitBuf, const INT *val,
+                           const INT *valLast, const INT nBands,
+                           const UINT *codeTable, const UINT *lengthTable,
+                           const INT tableOffset, const INT maxVal,
+                           INT *error) {
+  INT bitCnt = 0;
+  INT band;
+
+  for (band = 0; band < nBands; band++) {
+    INT delta = (val[band] - valLast[band]) + tableOffset;
+    if ((delta > maxVal) || (delta < 0)) {
+      *error = 1;
+      delta = delta > 0 ? maxVal : 0;
+    }
+    bitCnt +=
+        FDKsbrEnc_WriteBits_ps(hBitBuf, codeTable[delta], lengthTable[delta]);
+  }
+
+  return bitCnt;
+}
+
+INT FDKsbrEnc_EncodeIid(HANDLE_FDK_BITSTREAM hBitBuf, const INT *iidVal,
+                        const INT *iidValLast, const INT nBands,
+                        const PS_IID_RESOLUTION res, const PS_DELTA mode,
+                        INT *error) {
+  const UINT *codeTable;
+  const UINT *lengthTable;
+  INT bitCnt = 0;
+
+  bitCnt = 0;
+
+  switch (mode) {
+    case PS_DELTA_FREQ:
+      switch (res) {
+        case PS_IID_RES_COARSE:
+          codeTable = iidDeltaFreqCoarse_Code;
+          lengthTable = iidDeltaFreqCoarse_Length;
+          bitCnt += encodeDeltaFreq(hBitBuf, iidVal, nBands, codeTable,
+                                    lengthTable, iidDeltaCoarse_Offset,
+                                    iidDeltaCoarse_MaxVal, error);
+          break;
+        case PS_IID_RES_FINE:
+          codeTable = iidDeltaFreqFine_Code;
+          lengthTable = iidDeltaFreqFine_Length;
+          bitCnt +=
+              encodeDeltaFreq(hBitBuf, iidVal, nBands, codeTable, lengthTable,
+                              iidDeltaFine_Offset, iidDeltaFine_MaxVal, error);
+          break;
+        default:
+          *error = 1;
+      }
+      break;
+
+    case PS_DELTA_TIME:
+      switch (res) {
+        case PS_IID_RES_COARSE:
+          codeTable = iidDeltaTimeCoarse_Code;
+          lengthTable = iidDeltaTimeCoarse_Length;
+          bitCnt += encodeDeltaTime(
+              hBitBuf, iidVal, iidValLast, nBands, codeTable, lengthTable,
+              iidDeltaCoarse_Offset, iidDeltaCoarse_MaxVal, error);
+          break;
+        case PS_IID_RES_FINE:
+          codeTable = iidDeltaTimeFine_Code;
+          lengthTable = iidDeltaTimeFine_Length;
+          bitCnt += encodeDeltaTime(hBitBuf, iidVal, iidValLast, nBands,
+                                    codeTable, lengthTable, iidDeltaFine_Offset,
+                                    iidDeltaFine_MaxVal, error);
+          break;
+        default:
+          *error = 1;
+      }
+      break;
+
+    default:
+      *error = 1;
+  }
+
+  return bitCnt;
+}
+
+INT FDKsbrEnc_EncodeIcc(HANDLE_FDK_BITSTREAM hBitBuf, const INT *iccVal,
+                        const INT *iccValLast, const INT nBands,
+                        const PS_DELTA mode, INT *error) {
+  const UINT *codeTable;
+  const UINT *lengthTable;
+  INT bitCnt = 0;
+
+  switch (mode) {
+    case PS_DELTA_FREQ:
+      codeTable = iccDeltaFreq_Code;
+      lengthTable = iccDeltaFreq_Length;
+      bitCnt += encodeDeltaFreq(hBitBuf, iccVal, nBands, codeTable, lengthTable,
+                                iccDelta_Offset, iccDelta_MaxVal, error);
+      break;
+
+    case PS_DELTA_TIME:
+      codeTable = iccDeltaTime_Code;
+      lengthTable = iccDeltaTime_Length;
+
+      bitCnt +=
+          encodeDeltaTime(hBitBuf, iccVal, iccValLast, nBands, codeTable,
+                          lengthTable, iccDelta_Offset, iccDelta_MaxVal, error);
+      break;
+
+    default:
+      *error = 1;
+  }
+
+  return bitCnt;
+}
+
+INT FDKsbrEnc_EncodeIpd(HANDLE_FDK_BITSTREAM hBitBuf, const INT *ipdVal,
+                        const INT *ipdValLast, const INT nBands,
+                        const PS_DELTA mode, INT *error) {
+  const UINT *codeTable;
+  const UINT *lengthTable;
+  INT bitCnt = 0;
+
+  switch (mode) {
+    case PS_DELTA_FREQ:
+      codeTable = ipdDeltaFreq_Code;
+      lengthTable = ipdDeltaFreq_Length;
+      bitCnt += encodeDeltaFreq(hBitBuf, ipdVal, nBands, codeTable, lengthTable,
+                                ipdDelta_Offset, ipdDelta_MaxVal, error);
+      break;
+
+    case PS_DELTA_TIME:
+      codeTable = ipdDeltaTime_Code;
+      lengthTable = ipdDeltaTime_Length;
+
+      bitCnt +=
+          encodeDeltaTime(hBitBuf, ipdVal, ipdValLast, nBands, codeTable,
+                          lengthTable, ipdDelta_Offset, ipdDelta_MaxVal, error);
+      break;
+
+    default:
+      *error = 1;
+  }
+
+  return bitCnt;
+}
+
+INT FDKsbrEnc_EncodeOpd(HANDLE_FDK_BITSTREAM hBitBuf, const INT *opdVal,
+                        const INT *opdValLast, const INT nBands,
+                        const PS_DELTA mode, INT *error) {
+  const UINT *codeTable;
+  const UINT *lengthTable;
+  INT bitCnt = 0;
+
+  switch (mode) {
+    case PS_DELTA_FREQ:
+      codeTable = opdDeltaFreq_Code;
+      lengthTable = opdDeltaFreq_Length;
+      bitCnt += encodeDeltaFreq(hBitBuf, opdVal, nBands, codeTable, lengthTable,
+                                opdDelta_Offset, opdDelta_MaxVal, error);
+      break;
+
+    case PS_DELTA_TIME:
+      codeTable = opdDeltaTime_Code;
+      lengthTable = opdDeltaTime_Length;
+
+      bitCnt +=
+          encodeDeltaTime(hBitBuf, opdVal, opdValLast, nBands, codeTable,
+                          lengthTable, opdDelta_Offset, opdDelta_MaxVal, error);
+      break;
+
+    default:
+      *error = 1;
+  }
+
+  return bitCnt;
+}
+
+static INT encodeIpdOpd(HANDLE_PS_OUT psOut, HANDLE_FDK_BITSTREAM hBitBuf) {
+  INT bitCnt = 0;
+  INT error = 0;
+  INT env;
+
+  FDKsbrEnc_WriteBits_ps(hBitBuf, psOut->enableIpdOpd, 1);
+
+  if (psOut->enableIpdOpd == 1) {
+    INT *ipdLast = psOut->ipdLast;
+    INT *opdLast = psOut->opdLast;
+
+    for (env = 0; env < psOut->nEnvelopes; env++) {
+      bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, psOut->deltaIPD[env], 1);
+      bitCnt += FDKsbrEnc_EncodeIpd(hBitBuf, psOut->ipd[env], ipdLast,
+                                    getNoBands(psOut->iidMode),
+                                    psOut->deltaIPD[env], &error);
+
+      bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, psOut->deltaOPD[env], 1);
+      bitCnt += FDKsbrEnc_EncodeOpd(hBitBuf, psOut->opd[env], opdLast,
+                                    getNoBands(psOut->iidMode),
+                                    psOut->deltaOPD[env], &error);
+    }
+    /* reserved bit */
+    bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, 0, 1);
+  }
+
+  return bitCnt;
+}
+
+static INT getEnvIdx(const INT nEnvelopes, const INT frameClass) {
+  INT envIdx = 0;
+
+  switch (nEnvelopes) {
+    case 0:
+      envIdx = 0;
+      break;
+
+    case 1:
+      if (frameClass == 0)
+        envIdx = 1;
+      else
+        envIdx = 0;
+      break;
+
+    case 2:
+      if (frameClass == 0)
+        envIdx = 2;
+      else
+        envIdx = 1;
+      break;
+
+    case 3:
+      envIdx = 2;
+      break;
+
+    case 4:
+      envIdx = 3;
+      break;
+
+    default:
+      /* unsupported number of envelopes */
+      envIdx = 0;
+  }
+
+  return envIdx;
+}
+
+static INT encodePSExtension(const HANDLE_PS_OUT psOut,
+                             HANDLE_FDK_BITSTREAM hBitBuf) {
+  INT bitCnt = 0;
+
+  if (psOut->enableIpdOpd == 1) {
+    INT ipdOpdBits = 0;
+    INT extSize = (2 + encodeIpdOpd(psOut, NULL) + 7) >> 3;
+
+    if (extSize < 15) {
+      bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, extSize, 4);
+    } else {
+      bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, 15, 4);
+      bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, (extSize - 15), 8);
+    }
+
+    /* write ipd opd data */
+    ipdOpdBits += FDKsbrEnc_WriteBits_ps(hBitBuf, PS_EXT_ID_V0, 2);
+    ipdOpdBits += encodeIpdOpd(psOut, hBitBuf);
+
+    /* byte align the ipd opd data  */
+    if (ipdOpdBits % 8)
+      ipdOpdBits += FDKsbrEnc_WriteBits_ps(hBitBuf, 0, (8 - (ipdOpdBits % 8)));
+
+    bitCnt += ipdOpdBits;
+  }
+
+  return (bitCnt);
+}
+
+INT FDKsbrEnc_WritePSBitstream(const HANDLE_PS_OUT psOut,
+                               HANDLE_FDK_BITSTREAM hBitBuf) {
+  INT psExtEnable = 0;
+  INT bitCnt = 0;
+  INT error = 0;
+  INT env;
+
+  if (psOut != NULL) {
+    /* PS HEADER */
+    bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, psOut->enablePSHeader, 1);
+
+    if (psOut->enablePSHeader) {
+      bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, psOut->enableIID, 1);
+      if (psOut->enableIID) {
+        bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, psOut->iidMode, 3);
+      }
+      bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, psOut->enableICC, 1);
+      if (psOut->enableICC) {
+        bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, psOut->iccMode, 3);
+      }
+      if (psOut->enableIpdOpd) {
+        psExtEnable = 1;
+      }
+      bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, psExtEnable, 1);
+    }
+
+    /* Frame class, number of envelopes */
+    bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, psOut->frameClass, 1);
+    bitCnt += FDKsbrEnc_WriteBits_ps(
+        hBitBuf, getEnvIdx(psOut->nEnvelopes, psOut->frameClass), 2);
+
+    if (psOut->frameClass == 1) {
+      for (env = 0; env < psOut->nEnvelopes; env++) {
+        bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, psOut->frameBorder[env], 5);
+      }
+    }
+
+    if (psOut->enableIID == 1) {
+      INT *iidLast = psOut->iidLast;
+      for (env = 0; env < psOut->nEnvelopes; env++) {
+        bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, psOut->deltaIID[env], 1);
+        bitCnt += FDKsbrEnc_EncodeIid(
+            hBitBuf, psOut->iid[env], iidLast, getNoBands(psOut->iidMode),
+            (PS_IID_RESOLUTION)getIIDRes(psOut->iidMode), psOut->deltaIID[env],
+            &error);
+
+        iidLast = psOut->iid[env];
+      }
+    }
+
+    if (psOut->enableICC == 1) {
+      INT *iccLast = psOut->iccLast;
+      for (env = 0; env < psOut->nEnvelopes; env++) {
+        bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, psOut->deltaICC[env], 1);
+        bitCnt += FDKsbrEnc_EncodeIcc(hBitBuf, psOut->icc[env], iccLast,
+                                      getNoBands(psOut->iccMode),
+                                      psOut->deltaICC[env], &error);
+
+        iccLast = psOut->icc[env];
+      }
+    }
+
+    if (psExtEnable != 0) {
+      bitCnt += encodePSExtension(psOut, hBitBuf);
+    }
+
+  } /* if(psOut != NULL) */
+
+  return bitCnt;
+}
diff --git a/fdk-aac/libSBRenc/src/ps_bitenc.h b/fdk-aac/libSBRenc/src/ps_bitenc.h
new file mode 100644
index 0000000..1d383e3
--- /dev/null
+++ b/fdk-aac/libSBRenc/src/ps_bitenc.h
@@ -0,0 +1,173 @@
+/* -----------------------------------------------------------------------------
+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 encoder library ******************************
+
+   Author(s):   N. Rettelbach
+
+   Description: Parametric Stereo bitstream encoder
+
+*******************************************************************************/
+
+#include "ps_main.h"
+#include "ps_const.h"
+#include "FDK_bitstream.h"
+
+#ifndef PS_BITENC_H
+#define PS_BITENC_H
+
+typedef struct T_PS_OUT {
+  INT enablePSHeader;
+  INT enableIID;
+  INT iidMode;
+  INT enableICC;
+  INT iccMode;
+  INT enableIpdOpd;
+
+  INT frameClass;
+  INT nEnvelopes;
+  /* ENV data */
+  INT frameBorder[PS_MAX_ENVELOPES];
+
+  /* iid data  */
+  PS_DELTA deltaIID[PS_MAX_ENVELOPES];
+  INT iid[PS_MAX_ENVELOPES][PS_MAX_BANDS];
+  INT iidLast[PS_MAX_BANDS];
+
+  /* icc data  */
+  PS_DELTA deltaICC[PS_MAX_ENVELOPES];
+  INT icc[PS_MAX_ENVELOPES][PS_MAX_BANDS];
+  INT iccLast[PS_MAX_BANDS];
+
+  /* ipd data  */
+  PS_DELTA deltaIPD[PS_MAX_ENVELOPES];
+  INT ipd[PS_MAX_ENVELOPES][PS_MAX_BANDS];
+  INT ipdLast[PS_MAX_BANDS];
+
+  /* opd data  */
+  PS_DELTA deltaOPD[PS_MAX_ENVELOPES];
+  INT opd[PS_MAX_ENVELOPES][PS_MAX_BANDS];
+  INT opdLast[PS_MAX_BANDS];
+
+} PS_OUT, *HANDLE_PS_OUT;
+
+#ifdef __cplusplus
+extern "C" {
+#endif /* __cplusplus */
+
+INT FDKsbrEnc_EncodeIid(HANDLE_FDK_BITSTREAM hBitBuf, const INT *iidVal,
+                        const INT *iidValLast, const INT nBands,
+                        const PS_IID_RESOLUTION res, const PS_DELTA mode,
+                        INT *error);
+
+INT FDKsbrEnc_EncodeIcc(HANDLE_FDK_BITSTREAM hBitBuf, const INT *iccVal,
+                        const INT *iccValLast, const INT nBands,
+                        const PS_DELTA mode, INT *error);
+
+INT FDKsbrEnc_EncodeIpd(HANDLE_FDK_BITSTREAM hBitBuf, const INT *ipdVal,
+                        const INT *ipdValLast, const INT nBands,
+                        const PS_DELTA mode, INT *error);
+
+INT FDKsbrEnc_EncodeOpd(HANDLE_FDK_BITSTREAM hBitBuf, const INT *opdVal,
+                        const INT *opdValLast, const INT nBands,
+                        const PS_DELTA mode, INT *error);
+
+INT FDKsbrEnc_WritePSBitstream(const HANDLE_PS_OUT psOut,
+                               HANDLE_FDK_BITSTREAM hBitBuf);
+
+#ifdef __cplusplus
+}
+#endif /* __cplusplus */
+
+#endif /* defined(PSENC_ENABLE) */
diff --git a/fdk-aac/libSBRenc/src/ps_const.h b/fdk-aac/libSBRenc/src/ps_const.h
new file mode 100644
index 0000000..b9a33f9
--- /dev/null
+++ b/fdk-aac/libSBRenc/src/ps_const.h
@@ -0,0 +1,150 @@
+/* -----------------------------------------------------------------------------
+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 encoder library ******************************
+
+   Author(s):   N. Rettelbach
+
+   Description: Parametric Stereo constants
+
+*******************************************************************************/
+
+#ifndef PS_CONST_H
+#define PS_CONST_H
+
+#define MAX_PS_CHANNELS (2)
+#define HYBRID_MAX_QMF_BANDS (3)
+#define HYBRID_FILTER_LENGTH (13)
+#define HYBRID_FILTER_DELAY ((HYBRID_FILTER_LENGTH - 1) / 2)
+
+#define HYBRID_FRAMESIZE (32)
+#define HYBRID_READ_OFFSET (10)
+
+#define MAX_HYBRID_BANDS ((64 - HYBRID_MAX_QMF_BANDS + 10))
+
+typedef enum {
+  PS_RES_COARSE = 0,
+  PS_RES_MID = 1,
+  PS_RES_FINE = 2
+} PS_RESOLUTION;
+
+typedef enum {
+  PS_BANDS_COARSE = 10,
+  PS_BANDS_MID = 20,
+  PS_MAX_BANDS = PS_BANDS_MID
+} PS_BANDS;
+
+typedef enum { PS_IID_RES_COARSE = 0, PS_IID_RES_FINE } PS_IID_RESOLUTION;
+
+typedef enum { PS_ICC_ROT_A = 0, PS_ICC_ROT_B } PS_ICC_ROTATION_MODE;
+
+typedef enum { PS_DELTA_FREQ, PS_DELTA_TIME } PS_DELTA;
+
+typedef enum {
+  PS_MAX_ENVELOPES = 4
+
+} PS_CONSTS;
+
+typedef enum {
+  PSENC_OK = 0x0000, /*!< No error happened. All fine. */
+  PSENC_INVALID_HANDLE =
+      0x0020, /*!< Handle passed to function call was invalid. */
+  PSENC_MEMORY_ERROR = 0x0021, /*!< Memory allocation failed. */
+  PSENC_INIT_ERROR = 0x0040,   /*!< General initialization error. */
+  PSENC_ENCODE_ERROR = 0x0060  /*!< The encoding process was interrupted by an
+                                  unexpected error. */
+
+} FDK_PSENC_ERROR;
+
+#endif
diff --git a/fdk-aac/libSBRenc/src/ps_encode.cpp b/fdk-aac/libSBRenc/src/ps_encode.cpp
new file mode 100644
index 0000000..88d3131
--- /dev/null
+++ b/fdk-aac/libSBRenc/src/ps_encode.cpp
@@ -0,0 +1,1031 @@
+/* -----------------------------------------------------------------------------
+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 encoder library ******************************
+
+   Author(s):   M. Neuendorf, N. Rettelbach, M. Multrus
+
+   Description: PS parameter extraction, encoding
+
+*******************************************************************************/
+
+/*!
+  \file
+  \brief  PS parameter extraction, encoding functions $Revision: 96441 $
+*/
+
+#include "ps_main.h"
+#include "ps_encode.h"
+#include "qmf.h"
+#include "sbr_misc.h"
+#include "sbrenc_ram.h"
+
+#include "genericStds.h"
+
+inline void FDKsbrEnc_addFIXP_DBL(const FIXP_DBL *X, const FIXP_DBL *Y,
+                                  FIXP_DBL *Z, INT n) {
+  for (INT i = 0; i < n; i++) Z[i] = (X[i] >> 1) + (Y[i] >> 1);
+}
+
+#define LOG10_2_10 3.01029995664f /* 10.0f*log10(2.f) */
+
+static const INT
+    iidGroupBordersLoRes[QMF_GROUPS_LO_RES + SUBQMF_GROUPS_LO_RES + 1] = {
+        0,  1,  2,  3,  4,  5, /* 6 subqmf subbands - 0th qmf subband */
+        6,  7,                 /* 2 subqmf subbands - 1st qmf subband */
+        8,  9,                 /* 2 subqmf subbands - 2nd qmf subband */
+        10, 11, 12, 13, 14, 15, 16, 18, 21, 25, 30, 42, 71};
+
+static const UCHAR
+    iidGroupWidthLdLoRes[QMF_GROUPS_LO_RES + SUBQMF_GROUPS_LO_RES] = {
+        0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 1, 2, 2, 3, 4, 5};
+
+static const INT subband2parameter20[QMF_GROUPS_LO_RES + SUBQMF_GROUPS_LO_RES] =
+    {1, 0, 0,  1,  2,  3, /* 6 subqmf subbands - 0th qmf subband */
+     4, 5,                /* 2 subqmf subbands - 1st qmf subband */
+     6, 7,                /* 2 subqmf subbands - 2nd qmf subband */
+     8, 9, 10, 11, 12, 13, 14, 15, 16, 17, 18, 19};
+
+typedef enum {
+  MAX_TIME_DIFF_FRAMES = 20,
+  MAX_PS_NOHEADER_CNT = 10,
+  MAX_NOENV_CNT = 10,
+  DO_NOT_USE_THIS_MODE = 0x7FFFFF
+} __PS_CONSTANTS;
+
+static const FIXP_DBL iidQuant_fx[15] = {
+    (FIXP_DBL)0xce000000, (FIXP_DBL)0xdc000000, (FIXP_DBL)0xe4000000,
+    (FIXP_DBL)0xec000000, (FIXP_DBL)0xf2000000, (FIXP_DBL)0xf8000000,
+    (FIXP_DBL)0xfc000000, (FIXP_DBL)0x00000000, (FIXP_DBL)0x04000000,
+    (FIXP_DBL)0x08000000, (FIXP_DBL)0x0e000000, (FIXP_DBL)0x14000000,
+    (FIXP_DBL)0x1c000000, (FIXP_DBL)0x24000000, (FIXP_DBL)0x32000000};
+
+static const FIXP_DBL iidQuantFine_fx[31] = {
+    (FIXP_DBL)0x9c000001, (FIXP_DBL)0xa6000001, (FIXP_DBL)0xb0000001,
+    (FIXP_DBL)0xba000001, (FIXP_DBL)0xc4000000, (FIXP_DBL)0xce000000,
+    (FIXP_DBL)0xd4000000, (FIXP_DBL)0xda000000, (FIXP_DBL)0xe0000000,
+    (FIXP_DBL)0xe6000000, (FIXP_DBL)0xec000000, (FIXP_DBL)0xf0000000,
+    (FIXP_DBL)0xf4000000, (FIXP_DBL)0xf8000000, (FIXP_DBL)0xfc000000,
+    (FIXP_DBL)0x00000000, (FIXP_DBL)0x04000000, (FIXP_DBL)0x08000000,
+    (FIXP_DBL)0x0c000000, (FIXP_DBL)0x10000000, (FIXP_DBL)0x14000000,
+    (FIXP_DBL)0x1a000000, (FIXP_DBL)0x20000000, (FIXP_DBL)0x26000000,
+    (FIXP_DBL)0x2c000000, (FIXP_DBL)0x32000000, (FIXP_DBL)0x3c000000,
+    (FIXP_DBL)0x45ffffff, (FIXP_DBL)0x4fffffff, (FIXP_DBL)0x59ffffff,
+    (FIXP_DBL)0x63ffffff};
+
+static const FIXP_DBL iccQuant[8] = {
+    (FIXP_DBL)0x7fffffff, (FIXP_DBL)0x77ef9d7f, (FIXP_DBL)0x6babc97f,
+    (FIXP_DBL)0x4ceaf27f, (FIXP_DBL)0x2f0ed3c0, (FIXP_DBL)0x00000000,
+    (FIXP_DBL)0xb49ba601, (FIXP_DBL)0x80000000};
+
+static FDK_PSENC_ERROR InitPSData(HANDLE_PS_DATA hPsData) {
+  FDK_PSENC_ERROR error = PSENC_OK;
+
+  if (hPsData == NULL) {
+    error = PSENC_INVALID_HANDLE;
+  } else {
+    int i, env;
+    FDKmemclear(hPsData, sizeof(PS_DATA));
+
+    for (i = 0; i < PS_MAX_BANDS; i++) {
+      hPsData->iidIdxLast[i] = 0;
+      hPsData->iccIdxLast[i] = 0;
+    }
+
+    hPsData->iidEnable = hPsData->iidEnableLast = 0;
+    hPsData->iccEnable = hPsData->iccEnableLast = 0;
+    hPsData->iidQuantMode = hPsData->iidQuantModeLast = PS_IID_RES_COARSE;
+    hPsData->iccQuantMode = hPsData->iccQuantModeLast = PS_ICC_ROT_A;
+
+    for (env = 0; env < PS_MAX_ENVELOPES; env++) {
+      hPsData->iccDiffMode[env] = PS_DELTA_FREQ;
+      hPsData->iccDiffMode[env] = PS_DELTA_FREQ;
+
+      for (i = 0; i < PS_MAX_BANDS; i++) {
+        hPsData->iidIdx[env][i] = 0;
+        hPsData->iccIdx[env][i] = 0;
+      }
+    }
+
+    hPsData->nEnvelopesLast = 0;
+
+    hPsData->headerCnt = MAX_PS_NOHEADER_CNT;
+    hPsData->iidTimeCnt = MAX_TIME_DIFF_FRAMES;
+    hPsData->iccTimeCnt = MAX_TIME_DIFF_FRAMES;
+    hPsData->noEnvCnt = MAX_NOENV_CNT;
+  }
+
+  return error;
+}
+
+static FIXP_DBL quantizeCoef(const FIXP_DBL *RESTRICT input, const INT nBands,
+                             const FIXP_DBL *RESTRICT quantTable,
+                             const INT idxOffset, const INT nQuantSteps,
+                             INT *RESTRICT quantOut) {
+  INT idx, band;
+  FIXP_DBL quantErr = FL2FXCONST_DBL(0.f);
+
+  for (band = 0; band < nBands; band++) {
+    for (idx = 0; idx < nQuantSteps - 1; idx++) {
+      if (fixp_abs((input[band] >> 1) - (quantTable[idx + 1] >> 1)) >
+          fixp_abs((input[band] >> 1) - (quantTable[idx] >> 1))) {
+        break;
+      }
+    }
+    quantErr += (fixp_abs(input[band] - quantTable[idx]) >>
+                 PS_QUANT_SCALE); /* don't scale before subtraction; diff
+                                     smaller (64-25)/64 */
+    quantOut[band] = idx - idxOffset;
+  }
+
+  return quantErr;
+}
+
+static INT getICCMode(const INT nBands, const INT rotType) {
+  INT mode = 0;
+
+  switch (nBands) {
+    case PS_BANDS_COARSE:
+      mode = PS_RES_COARSE;
+      break;
+    case PS_BANDS_MID:
+      mode = PS_RES_MID;
+      break;
+    default:
+      mode = 0;
+  }
+  if (rotType == PS_ICC_ROT_B) {
+    mode += 3;
+  }
+
+  return mode;
+}
+
+static INT getIIDMode(const INT nBands, const INT iidRes) {
+  INT mode = 0;
+
+  switch (nBands) {
+    case PS_BANDS_COARSE:
+      mode = PS_RES_COARSE;
+      break;
+    case PS_BANDS_MID:
+      mode = PS_RES_MID;
+      break;
+    default:
+      mode = 0;
+      break;
+  }
+
+  if (iidRes == PS_IID_RES_FINE) {
+    mode += 3;
+  }
+
+  return mode;
+}
+
+static INT envelopeReducible(FIXP_DBL iid[PS_MAX_ENVELOPES][PS_MAX_BANDS],
+                             FIXP_DBL icc[PS_MAX_ENVELOPES][PS_MAX_BANDS],
+                             INT psBands, INT nEnvelopes) {
+#define THRESH_SCALE 7
+
+  INT reducible = 1; /* true */
+  INT e = 0, b = 0;
+  FIXP_DBL dIid = FL2FXCONST_DBL(0.f);
+  FIXP_DBL dIcc = FL2FXCONST_DBL(0.f);
+
+  FIXP_DBL iidErrThreshold, iccErrThreshold;
+  FIXP_DBL iidMeanError, iccMeanError;
+
+  /* square values to prevent sqrt,
+     multiply bands to prevent division; bands shifted DFRACT_BITS instead
+     (DFRACT_BITS-1) because fMultDiv2 used*/
+  iidErrThreshold =
+      fMultDiv2(FL2FXCONST_DBL(6.5f * 6.5f / (IID_SCALE_FT * IID_SCALE_FT)),
+                (FIXP_DBL)(psBands << ((DFRACT_BITS)-THRESH_SCALE)));
+  iccErrThreshold =
+      fMultDiv2(FL2FXCONST_DBL(0.75f * 0.75f),
+                (FIXP_DBL)(psBands << ((DFRACT_BITS)-THRESH_SCALE)));
+
+  if (nEnvelopes <= 1) {
+    reducible = 0;
+  } else {
+    /* mean error criterion */
+    for (e = 0; (e < nEnvelopes / 2) && (reducible != 0); e++) {
+      iidMeanError = iccMeanError = FL2FXCONST_DBL(0.f);
+      for (b = 0; b < psBands; b++) {
+        dIid = (iid[2 * e][b] >> 1) -
+               (iid[2 * e + 1][b] >> 1); /* scale 1 bit; squared -> 2 bit */
+        dIcc = (icc[2 * e][b] >> 1) - (icc[2 * e + 1][b] >> 1);
+        iidMeanError += fPow2Div2(dIid) >> (5 - 1); /* + (bands=20) scale = 5 */
+        iccMeanError += fPow2Div2(dIcc) >> (5 - 1);
+      } /* --> scaling = 7 bit = THRESH_SCALE !! */
+
+      /* instead sqrt values are squared!
+         instead of division, multiply threshold with psBands
+         scaling necessary!! */
+
+      /* quit as soon as threshold is reached */
+      if ((iidMeanError > (iidErrThreshold)) ||
+          (iccMeanError > (iccErrThreshold))) {
+        reducible = 0;
+      }
+    }
+  } /* nEnvelopes != 1 */
+
+  return reducible;
+}
+
+static void processIidData(PS_DATA *psData,
+                           FIXP_DBL iid[PS_MAX_ENVELOPES][PS_MAX_BANDS],
+                           const INT psBands, const INT nEnvelopes,
+                           const FIXP_DBL quantErrorThreshold) {
+  INT iidIdxFine[PS_MAX_ENVELOPES][PS_MAX_BANDS];
+  INT iidIdxCoarse[PS_MAX_ENVELOPES][PS_MAX_BANDS];
+
+  FIXP_DBL errIID = FL2FXCONST_DBL(0.f);
+  FIXP_DBL errIIDFine = FL2FXCONST_DBL(0.f);
+  INT bitsIidFreq = 0;
+  INT bitsIidTime = 0;
+  INT bitsFineTot = 0;
+  INT bitsCoarseTot = 0;
+  INT error = 0;
+  INT env, band;
+  INT diffMode[PS_MAX_ENVELOPES], diffModeFine[PS_MAX_ENVELOPES];
+  INT loudnDiff = 0;
+  INT iidTransmit = 0;
+
+  /* Quantize IID coefficients */
+  for (env = 0; env < nEnvelopes; env++) {
+    errIID +=
+        quantizeCoef(iid[env], psBands, iidQuant_fx, 7, 15, iidIdxCoarse[env]);
+    errIIDFine += quantizeCoef(iid[env], psBands, iidQuantFine_fx, 15, 31,
+                               iidIdxFine[env]);
+  }
+
+  /* normalize error to number of envelopes, ps bands
+     errIID /= psBands*nEnvelopes;
+     errIIDFine /= psBands*nEnvelopes; */
+
+  /* Check if IID coefficients should be used in this frame */
+  psData->iidEnable = 0;
+  for (env = 0; env < nEnvelopes; env++) {
+    for (band = 0; band < psBands; band++) {
+      loudnDiff += fixp_abs(iidIdxCoarse[env][band]);
+      iidTransmit++;
+    }
+  }
+
+  if (loudnDiff >
+      fMultI(FL2FXCONST_DBL(0.7f), iidTransmit)) { /* 0.7f empiric value */
+    psData->iidEnable = 1;
+  }
+
+  /* if iid not active -> RESET data */
+  if (psData->iidEnable == 0) {
+    psData->iidTimeCnt = MAX_TIME_DIFF_FRAMES;
+    for (env = 0; env < nEnvelopes; env++) {
+      psData->iidDiffMode[env] = PS_DELTA_FREQ;
+      FDKmemclear(psData->iidIdx[env], sizeof(INT) * psBands);
+    }
+    return;
+  }
+
+  /* count COARSE quantization bits for first envelope*/
+  bitsIidFreq = FDKsbrEnc_EncodeIid(NULL, iidIdxCoarse[0], NULL, psBands,
+                                    PS_IID_RES_COARSE, PS_DELTA_FREQ, &error);
+
+  if ((psData->iidTimeCnt >= MAX_TIME_DIFF_FRAMES) ||
+      (psData->iidQuantModeLast == PS_IID_RES_FINE)) {
+    bitsIidTime = DO_NOT_USE_THIS_MODE;
+  } else {
+    bitsIidTime =
+        FDKsbrEnc_EncodeIid(NULL, iidIdxCoarse[0], psData->iidIdxLast, psBands,
+                            PS_IID_RES_COARSE, PS_DELTA_TIME, &error);
+  }
+
+  /* decision DELTA_FREQ vs DELTA_TIME */
+  if (bitsIidTime > bitsIidFreq) {
+    diffMode[0] = PS_DELTA_FREQ;
+    bitsCoarseTot = bitsIidFreq;
+  } else {
+    diffMode[0] = PS_DELTA_TIME;
+    bitsCoarseTot = bitsIidTime;
+  }
+
+  /* count COARSE quantization bits for following envelopes*/
+  for (env = 1; env < nEnvelopes; env++) {
+    bitsIidFreq = FDKsbrEnc_EncodeIid(NULL, iidIdxCoarse[env], NULL, psBands,
+                                      PS_IID_RES_COARSE, PS_DELTA_FREQ, &error);
+    bitsIidTime =
+        FDKsbrEnc_EncodeIid(NULL, iidIdxCoarse[env], iidIdxCoarse[env - 1],
+                            psBands, PS_IID_RES_COARSE, PS_DELTA_TIME, &error);
+
+    /* decision DELTA_FREQ vs DELTA_TIME */
+    if (bitsIidTime > bitsIidFreq) {
+      diffMode[env] = PS_DELTA_FREQ;
+      bitsCoarseTot += bitsIidFreq;
+    } else {
+      diffMode[env] = PS_DELTA_TIME;
+      bitsCoarseTot += bitsIidTime;
+    }
+  }
+
+  /* count FINE quantization bits for first envelope*/
+  bitsIidFreq = FDKsbrEnc_EncodeIid(NULL, iidIdxFine[0], NULL, psBands,
+                                    PS_IID_RES_FINE, PS_DELTA_FREQ, &error);
+
+  if ((psData->iidTimeCnt >= MAX_TIME_DIFF_FRAMES) ||
+      (psData->iidQuantModeLast == PS_IID_RES_COARSE)) {
+    bitsIidTime = DO_NOT_USE_THIS_MODE;
+  } else {
+    bitsIidTime =
+        FDKsbrEnc_EncodeIid(NULL, iidIdxFine[0], psData->iidIdxLast, psBands,
+                            PS_IID_RES_FINE, PS_DELTA_TIME, &error);
+  }
+
+  /* decision DELTA_FREQ vs DELTA_TIME */
+  if (bitsIidTime > bitsIidFreq) {
+    diffModeFine[0] = PS_DELTA_FREQ;
+    bitsFineTot = bitsIidFreq;
+  } else {
+    diffModeFine[0] = PS_DELTA_TIME;
+    bitsFineTot = bitsIidTime;
+  }
+
+  /* count FINE quantization bits for following envelopes*/
+  for (env = 1; env < nEnvelopes; env++) {
+    bitsIidFreq = FDKsbrEnc_EncodeIid(NULL, iidIdxFine[env], NULL, psBands,
+                                      PS_IID_RES_FINE, PS_DELTA_FREQ, &error);
+    bitsIidTime =
+        FDKsbrEnc_EncodeIid(NULL, iidIdxFine[env], iidIdxFine[env - 1], psBands,
+                            PS_IID_RES_FINE, PS_DELTA_TIME, &error);
+
+    /* decision DELTA_FREQ vs DELTA_TIME */
+    if (bitsIidTime > bitsIidFreq) {
+      diffModeFine[env] = PS_DELTA_FREQ;
+      bitsFineTot += bitsIidFreq;
+    } else {
+      diffModeFine[env] = PS_DELTA_TIME;
+      bitsFineTot += bitsIidTime;
+    }
+  }
+
+  if (bitsFineTot == bitsCoarseTot) {
+    /* if same number of bits is needed, use the quantization with lower error
+     */
+    if (errIIDFine < errIID) {
+      bitsCoarseTot = DO_NOT_USE_THIS_MODE;
+    } else {
+      bitsFineTot = DO_NOT_USE_THIS_MODE;
+    }
+  } else {
+    /* const FIXP_DBL minThreshold =
+     * FL2FXCONST_DBL(0.2f/(IID_SCALE_FT*PS_QUANT_SCALE_FT)*(psBands*nEnvelopes));
+     */
+    const FIXP_DBL minThreshold =
+        (FIXP_DBL)((LONG)0x00019999 * (psBands * nEnvelopes));
+
+    /* decision RES_FINE vs RES_COARSE                 */
+    /* test if errIIDFine*quantErrorThreshold < errIID */
+    /* shiftVal 2 comes from scaling of quantErrorThreshold */
+    if (fixMax(((errIIDFine >> 1) + (minThreshold >> 1)) >> 1,
+               fMult(quantErrorThreshold, errIIDFine)) < (errIID >> 2)) {
+      bitsCoarseTot = DO_NOT_USE_THIS_MODE;
+    } else if (fixMax(((errIID >> 1) + (minThreshold >> 1)) >> 1,
+                      fMult(quantErrorThreshold, errIID)) < (errIIDFine >> 2)) {
+      bitsFineTot = DO_NOT_USE_THIS_MODE;
+    }
+  }
+
+  /* decision RES_FINE vs RES_COARSE */
+  if (bitsFineTot < bitsCoarseTot) {
+    psData->iidQuantMode = PS_IID_RES_FINE;
+    for (env = 0; env < nEnvelopes; env++) {
+      psData->iidDiffMode[env] = diffModeFine[env];
+      FDKmemcpy(psData->iidIdx[env], iidIdxFine[env], psBands * sizeof(INT));
+    }
+  } else {
+    psData->iidQuantMode = PS_IID_RES_COARSE;
+    for (env = 0; env < nEnvelopes; env++) {
+      psData->iidDiffMode[env] = diffMode[env];
+      FDKmemcpy(psData->iidIdx[env], iidIdxCoarse[env], psBands * sizeof(INT));
+    }
+  }
+
+  /* Count DELTA_TIME encoding streaks */
+  for (env = 0; env < nEnvelopes; env++) {
+    if (psData->iidDiffMode[env] == PS_DELTA_TIME)
+      psData->iidTimeCnt++;
+    else
+      psData->iidTimeCnt = 0;
+  }
+}
+
+static INT similarIid(PS_DATA *psData, const INT psBands,
+                      const INT nEnvelopes) {
+  const INT diffThr = (psData->iidQuantMode == PS_IID_RES_COARSE) ? 2 : 3;
+  const INT sumDiffThr = diffThr * psBands / 4;
+  INT similar = 0;
+  INT diff = 0;
+  INT sumDiff = 0;
+  INT env = 0;
+  INT b = 0;
+  if ((nEnvelopes == psData->nEnvelopesLast) && (nEnvelopes == 1)) {
+    similar = 1;
+    for (env = 0; env < nEnvelopes; env++) {
+      sumDiff = 0;
+      b = 0;
+      do {
+        diff = fixp_abs(psData->iidIdx[env][b] - psData->iidIdxLast[b]);
+        sumDiff += diff;
+        if ((diff > diffThr) /* more than x quantization steps in any band */
+            || (sumDiff > sumDiffThr)) { /* more than x quantisations steps
+                                            overall difference */
+          similar = 0;
+        }
+        b++;
+      } while ((b < psBands) && (similar > 0));
+    }
+  } /* nEnvelopes==1  */
+
+  return similar;
+}
+
+static INT similarIcc(PS_DATA *psData, const INT psBands,
+                      const INT nEnvelopes) {
+  const INT diffThr = 2;
+  const INT sumDiffThr = diffThr * psBands / 4;
+  INT similar = 0;
+  INT diff = 0;
+  INT sumDiff = 0;
+  INT env = 0;
+  INT b = 0;
+  if ((nEnvelopes == psData->nEnvelopesLast) && (nEnvelopes == 1)) {
+    similar = 1;
+    for (env = 0; env < nEnvelopes; env++) {
+      sumDiff = 0;
+      b = 0;
+      do {
+        diff = fixp_abs(psData->iccIdx[env][b] - psData->iccIdxLast[b]);
+        sumDiff += diff;
+        if ((diff > diffThr) /* more than x quantisation step in any band */
+            || (sumDiff > sumDiffThr)) { /* more than x quantisations steps
+                                            overall difference */
+          similar = 0;
+        }
+        b++;
+      } while ((b < psBands) && (similar > 0));
+    }
+  } /* nEnvelopes==1  */
+
+  return similar;
+}
+
+static void processIccData(
+    PS_DATA *psData,
+    FIXP_DBL icc[PS_MAX_ENVELOPES][PS_MAX_BANDS], /* const input values:
+                                                     unable to declare as
+                                                     const, since it does
+                                                     not poINT to const
+                                                     memory */
+    const INT psBands, const INT nEnvelopes) {
+  FIXP_DBL errICC = FL2FXCONST_DBL(0.f);
+  INT env, band;
+  INT bitsIccFreq, bitsIccTime;
+  INT error = 0;
+  INT inCoherence = 0, iccTransmit = 0;
+  INT *iccIdxLast;
+
+  iccIdxLast = psData->iccIdxLast;
+
+  /* Quantize ICC coefficients */
+  for (env = 0; env < nEnvelopes; env++) {
+    errICC +=
+        quantizeCoef(icc[env], psBands, iccQuant, 0, 8, psData->iccIdx[env]);
+  }
+
+  /* Check if ICC coefficients should be used */
+  psData->iccEnable = 0;
+  for (env = 0; env < nEnvelopes; env++) {
+    for (band = 0; band < psBands; band++) {
+      inCoherence += psData->iccIdx[env][band];
+      iccTransmit++;
+    }
+  }
+  if (inCoherence >
+      fMultI(FL2FXCONST_DBL(0.5f), iccTransmit)) { /* 0.5f empiric value */
+    psData->iccEnable = 1;
+  }
+
+  if (psData->iccEnable == 0) {
+    psData->iccTimeCnt = MAX_TIME_DIFF_FRAMES;
+    for (env = 0; env < nEnvelopes; env++) {
+      psData->iccDiffMode[env] = PS_DELTA_FREQ;
+      FDKmemclear(psData->iccIdx[env], sizeof(INT) * psBands);
+    }
+    return;
+  }
+
+  for (env = 0; env < nEnvelopes; env++) {
+    bitsIccFreq = FDKsbrEnc_EncodeIcc(NULL, psData->iccIdx[env], NULL, psBands,
+                                      PS_DELTA_FREQ, &error);
+
+    if (psData->iccTimeCnt < MAX_TIME_DIFF_FRAMES) {
+      bitsIccTime = FDKsbrEnc_EncodeIcc(NULL, psData->iccIdx[env], iccIdxLast,
+                                        psBands, PS_DELTA_TIME, &error);
+    } else {
+      bitsIccTime = DO_NOT_USE_THIS_MODE;
+    }
+
+    if (bitsIccFreq > bitsIccTime) {
+      psData->iccDiffMode[env] = PS_DELTA_TIME;
+      psData->iccTimeCnt++;
+    } else {
+      psData->iccDiffMode[env] = PS_DELTA_FREQ;
+      psData->iccTimeCnt = 0;
+    }
+    iccIdxLast = psData->iccIdx[env];
+  }
+}
+
+static void calculateIID(FIXP_DBL ldPwrL[PS_MAX_ENVELOPES][PS_MAX_BANDS],
+                         FIXP_DBL ldPwrR[PS_MAX_ENVELOPES][PS_MAX_BANDS],
+                         FIXP_DBL iid[PS_MAX_ENVELOPES][PS_MAX_BANDS],
+                         INT nEnvelopes, INT psBands) {
+  INT i = 0;
+  INT env = 0;
+  for (env = 0; env < nEnvelopes; env++) {
+    for (i = 0; i < psBands; i++) {
+      /* iid[env][i] = 10.0f*(float)log10(pwrL[env][i]/pwrR[env][i]);
+       */
+      FIXP_DBL IID = fMultDiv2(FL2FXCONST_DBL(LOG10_2_10 / IID_SCALE_FT),
+                               (ldPwrL[env][i] - ldPwrR[env][i]));
+
+      IID = fixMin(IID, (FIXP_DBL)(MAXVAL_DBL >> (LD_DATA_SHIFT + 1)));
+      IID = fixMax(IID, (FIXP_DBL)(MINVAL_DBL >> (LD_DATA_SHIFT + 1)));
+      iid[env][i] = IID << (LD_DATA_SHIFT + 1);
+    }
+  }
+}
+
+static void calculateICC(FIXP_DBL pwrL[PS_MAX_ENVELOPES][PS_MAX_BANDS],
+                         FIXP_DBL pwrR[PS_MAX_ENVELOPES][PS_MAX_BANDS],
+                         FIXP_DBL pwrCr[PS_MAX_ENVELOPES][PS_MAX_BANDS],
+                         FIXP_DBL pwrCi[PS_MAX_ENVELOPES][PS_MAX_BANDS],
+                         FIXP_DBL icc[PS_MAX_ENVELOPES][PS_MAX_BANDS],
+                         INT nEnvelopes, INT psBands) {
+  INT i = 0;
+  INT env = 0;
+  INT border = psBands;
+
+  switch (psBands) {
+    case PS_BANDS_COARSE:
+      border = 5;
+      break;
+    case PS_BANDS_MID:
+      border = 11;
+      break;
+    default:
+      break;
+  }
+
+  for (env = 0; env < nEnvelopes; env++) {
+    for (i = 0; i < border; i++) {
+      /* icc[env][i] = min( pwrCr[env][i] / (float) sqrt(pwrL[env][i] *
+       * pwrR[env][i]) , 1.f);
+       */
+      int scale;
+      FIXP_DBL invNrg = invSqrtNorm2(
+          fMax(fMult(pwrL[env][i], pwrR[env][i]), (FIXP_DBL)1), &scale);
+      icc[env][i] =
+          SATURATE_LEFT_SHIFT(fMult(pwrCr[env][i], invNrg), scale, DFRACT_BITS);
+    }
+
+    for (; i < psBands; i++) {
+      int denom_e;
+      FIXP_DBL denom_m = fMultNorm(pwrL[env][i], pwrR[env][i], &denom_e);
+
+      if (denom_m == (FIXP_DBL)0) {
+        icc[env][i] = (FIXP_DBL)MAXVAL_DBL;
+      } else {
+        int num_e, result_e;
+        FIXP_DBL num_m, result_m;
+
+        num_e = CountLeadingBits(
+            fixMax(fixp_abs(pwrCr[env][i]), fixp_abs(pwrCi[env][i])));
+        num_m = fPow2Div2((pwrCr[env][i] << num_e)) +
+                fPow2Div2((pwrCi[env][i] << num_e));
+
+        result_m = fDivNorm(num_m, denom_m, &result_e);
+        result_e += (-2 * num_e + 1) - denom_e;
+        icc[env][i] = scaleValueSaturate(sqrtFixp(result_m >> (result_e & 1)),
+                                         (result_e + (result_e & 1)) >> 1);
+      }
+    }
+  }
+}
+
+void FDKsbrEnc_initPsBandNrgScale(HANDLE_PS_ENCODE hPsEncode) {
+  INT group, bin;
+  INT nIidGroups = hPsEncode->nQmfIidGroups + hPsEncode->nSubQmfIidGroups;
+
+  FDKmemclear(hPsEncode->psBandNrgScale, PS_MAX_BANDS * sizeof(SCHAR));
+
+  for (group = 0; group < nIidGroups; group++) {
+    /* Translate group to bin */
+    bin = hPsEncode->subband2parameterIndex[group];
+
+    /* Translate from 20 bins to 10 bins */
+    if (hPsEncode->psEncMode == PS_BANDS_COARSE) {
+      bin = bin >> 1;
+    }
+
+    hPsEncode->psBandNrgScale[bin] =
+        (hPsEncode->psBandNrgScale[bin] == 0)
+            ? (hPsEncode->iidGroupWidthLd[group] + 5)
+            : (fixMax(hPsEncode->iidGroupWidthLd[group],
+                      hPsEncode->psBandNrgScale[bin]) +
+               1);
+  }
+}
+
+FDK_PSENC_ERROR FDKsbrEnc_CreatePSEncode(HANDLE_PS_ENCODE *phPsEncode) {
+  FDK_PSENC_ERROR error = PSENC_OK;
+
+  if (phPsEncode == NULL) {
+    error = PSENC_INVALID_HANDLE;
+  } else {
+    HANDLE_PS_ENCODE hPsEncode = NULL;
+    if (NULL == (hPsEncode = GetRam_PsEncode())) {
+      error = PSENC_MEMORY_ERROR;
+      goto bail;
+    }
+    FDKmemclear(hPsEncode, sizeof(PS_ENCODE));
+    *phPsEncode = hPsEncode; /* return allocated handle */
+  }
+bail:
+  return error;
+}
+
+FDK_PSENC_ERROR FDKsbrEnc_InitPSEncode(HANDLE_PS_ENCODE hPsEncode,
+                                       const PS_BANDS psEncMode,
+                                       const FIXP_DBL iidQuantErrorThreshold) {
+  FDK_PSENC_ERROR error = PSENC_OK;
+
+  if (NULL == hPsEncode) {
+    error = PSENC_INVALID_HANDLE;
+  } else {
+    if (PSENC_OK != (InitPSData(&hPsEncode->psData))) {
+      goto bail;
+    }
+
+    switch (psEncMode) {
+      case PS_BANDS_COARSE:
+      case PS_BANDS_MID:
+        hPsEncode->nQmfIidGroups = QMF_GROUPS_LO_RES;
+        hPsEncode->nSubQmfIidGroups = SUBQMF_GROUPS_LO_RES;
+        FDKmemcpy(hPsEncode->iidGroupBorders, iidGroupBordersLoRes,
+                  (hPsEncode->nQmfIidGroups + hPsEncode->nSubQmfIidGroups + 1) *
+                      sizeof(INT));
+        FDKmemcpy(hPsEncode->subband2parameterIndex, subband2parameter20,
+                  (hPsEncode->nQmfIidGroups + hPsEncode->nSubQmfIidGroups) *
+                      sizeof(INT));
+        FDKmemcpy(hPsEncode->iidGroupWidthLd, iidGroupWidthLdLoRes,
+                  (hPsEncode->nQmfIidGroups + hPsEncode->nSubQmfIidGroups) *
+                      sizeof(UCHAR));
+        break;
+      default:
+        error = PSENC_INIT_ERROR;
+        goto bail;
+    }
+
+    hPsEncode->psEncMode = psEncMode;
+    hPsEncode->iidQuantErrorThreshold = iidQuantErrorThreshold;
+    FDKsbrEnc_initPsBandNrgScale(hPsEncode);
+  }
+bail:
+  return error;
+}
+
+FDK_PSENC_ERROR FDKsbrEnc_DestroyPSEncode(HANDLE_PS_ENCODE *phPsEncode) {
+  FDK_PSENC_ERROR error = PSENC_OK;
+
+  if (NULL != phPsEncode) {
+    FreeRam_PsEncode(phPsEncode);
+  }
+
+  return error;
+}
+
+typedef struct {
+  FIXP_DBL pwrL[PS_MAX_ENVELOPES][PS_MAX_BANDS];
+  FIXP_DBL pwrR[PS_MAX_ENVELOPES][PS_MAX_BANDS];
+  FIXP_DBL ldPwrL[PS_MAX_ENVELOPES][PS_MAX_BANDS];
+  FIXP_DBL ldPwrR[PS_MAX_ENVELOPES][PS_MAX_BANDS];
+  FIXP_DBL pwrCr[PS_MAX_ENVELOPES][PS_MAX_BANDS];
+  FIXP_DBL pwrCi[PS_MAX_ENVELOPES][PS_MAX_BANDS];
+
+} PS_PWR_DATA;
+
+FDK_PSENC_ERROR FDKsbrEnc_PSEncode(
+    HANDLE_PS_ENCODE hPsEncode, HANDLE_PS_OUT hPsOut, UCHAR *dynBandScale,
+    UINT maxEnvelopes,
+    FIXP_DBL *hybridData[HYBRID_FRAMESIZE][MAX_PS_CHANNELS][2],
+    const INT frameSize, const INT sendHeader) {
+  FDK_PSENC_ERROR error = PSENC_OK;
+
+  HANDLE_PS_DATA hPsData = &hPsEncode->psData;
+  FIXP_DBL iid[PS_MAX_ENVELOPES][PS_MAX_BANDS];
+  FIXP_DBL icc[PS_MAX_ENVELOPES][PS_MAX_BANDS];
+  int envBorder[PS_MAX_ENVELOPES + 1];
+
+  int group, bin, col, subband, band;
+  int i = 0;
+
+  int env = 0;
+  int psBands = (int)hPsEncode->psEncMode;
+  int nIidGroups = hPsEncode->nQmfIidGroups + hPsEncode->nSubQmfIidGroups;
+  int nEnvelopes = fixMin(maxEnvelopes, (UINT)PS_MAX_ENVELOPES);
+
+  C_ALLOC_SCRATCH_START(pwrData, PS_PWR_DATA, 1)
+
+  for (env = 0; env < nEnvelopes + 1; env++) {
+    envBorder[env] = fMultI(GetInvInt(nEnvelopes), frameSize * env);
+  }
+
+  for (env = 0; env < nEnvelopes; env++) {
+    /* clear energy array */
+    for (band = 0; band < psBands; band++) {
+      pwrData->pwrL[env][band] = pwrData->pwrR[env][band] =
+          pwrData->pwrCr[env][band] = pwrData->pwrCi[env][band] = FIXP_DBL(1);
+    }
+
+    /**** calculate energies and correlation ****/
+
+    /* start with hybrid data */
+    for (group = 0; group < nIidGroups; group++) {
+      /* Translate group to bin */
+      bin = hPsEncode->subband2parameterIndex[group];
+
+      /* Translate from 20 bins to 10 bins */
+      if (hPsEncode->psEncMode == PS_BANDS_COARSE) {
+        bin >>= 1;
+      }
+
+      /* determine group border */
+      int bScale = hPsEncode->psBandNrgScale[bin];
+
+      FIXP_DBL pwrL_env_bin = pwrData->pwrL[env][bin];
+      FIXP_DBL pwrR_env_bin = pwrData->pwrR[env][bin];
+      FIXP_DBL pwrCr_env_bin = pwrData->pwrCr[env][bin];
+      FIXP_DBL pwrCi_env_bin = pwrData->pwrCi[env][bin];
+
+      int scale = (int)dynBandScale[bin];
+      for (col = envBorder[env]; col < envBorder[env + 1]; col++) {
+        for (subband = hPsEncode->iidGroupBorders[group];
+             subband < hPsEncode->iidGroupBorders[group + 1]; subband++) {
+          FIXP_DBL l_real = (hybridData[col][0][0][subband]) << scale;
+          FIXP_DBL l_imag = (hybridData[col][0][1][subband]) << scale;
+          FIXP_DBL r_real = (hybridData[col][1][0][subband]) << scale;
+          FIXP_DBL r_imag = (hybridData[col][1][1][subband]) << scale;
+
+          pwrL_env_bin += (fPow2Div2(l_real) + fPow2Div2(l_imag)) >> bScale;
+          pwrR_env_bin += (fPow2Div2(r_real) + fPow2Div2(r_imag)) >> bScale;
+          pwrCr_env_bin +=
+              (fMultDiv2(l_real, r_real) + fMultDiv2(l_imag, r_imag)) >> bScale;
+          pwrCi_env_bin +=
+              (fMultDiv2(r_real, l_imag) - fMultDiv2(l_real, r_imag)) >> bScale;
+        }
+      }
+      /* assure, nrg's of left and right channel are not negative; necessary on
+       * 16 bit multiply units */
+      pwrData->pwrL[env][bin] = fixMax((FIXP_DBL)0, pwrL_env_bin);
+      pwrData->pwrR[env][bin] = fixMax((FIXP_DBL)0, pwrR_env_bin);
+
+      pwrData->pwrCr[env][bin] = pwrCr_env_bin;
+      pwrData->pwrCi[env][bin] = pwrCi_env_bin;
+
+    } /* nIidGroups */
+
+    /* calc logarithmic energy */
+    LdDataVector(pwrData->pwrL[env], pwrData->ldPwrL[env], psBands);
+    LdDataVector(pwrData->pwrR[env], pwrData->ldPwrR[env], psBands);
+
+  } /* nEnvelopes */
+
+  /* calculate iid and icc */
+  calculateIID(pwrData->ldPwrL, pwrData->ldPwrR, iid, nEnvelopes, psBands);
+  calculateICC(pwrData->pwrL, pwrData->pwrR, pwrData->pwrCr, pwrData->pwrCi,
+               icc, nEnvelopes, psBands);
+
+  /*** Envelope Reduction ***/
+  while (envelopeReducible(iid, icc, psBands, nEnvelopes)) {
+    int e = 0;
+    /* sum energies of two neighboring envelopes */
+    nEnvelopes >>= 1;
+    for (e = 0; e < nEnvelopes; e++) {
+      FDKsbrEnc_addFIXP_DBL(pwrData->pwrL[2 * e], pwrData->pwrL[2 * e + 1],
+                            pwrData->pwrL[e], psBands);
+      FDKsbrEnc_addFIXP_DBL(pwrData->pwrR[2 * e], pwrData->pwrR[2 * e + 1],
+                            pwrData->pwrR[e], psBands);
+      FDKsbrEnc_addFIXP_DBL(pwrData->pwrCr[2 * e], pwrData->pwrCr[2 * e + 1],
+                            pwrData->pwrCr[e], psBands);
+      FDKsbrEnc_addFIXP_DBL(pwrData->pwrCi[2 * e], pwrData->pwrCi[2 * e + 1],
+                            pwrData->pwrCi[e], psBands);
+
+      /* calc logarithmic energy */
+      LdDataVector(pwrData->pwrL[e], pwrData->ldPwrL[e], psBands);
+      LdDataVector(pwrData->pwrR[e], pwrData->ldPwrR[e], psBands);
+
+      /* reduce number of envelopes and adjust borders */
+      envBorder[e] = envBorder[2 * e];
+    }
+    envBorder[nEnvelopes] = envBorder[2 * nEnvelopes];
+
+    /* re-calculate iid and icc */
+    calculateIID(pwrData->ldPwrL, pwrData->ldPwrR, iid, nEnvelopes, psBands);
+    calculateICC(pwrData->pwrL, pwrData->pwrR, pwrData->pwrCr, pwrData->pwrCi,
+                 icc, nEnvelopes, psBands);
+  }
+
+  /*  */
+  if (sendHeader) {
+    hPsData->headerCnt = MAX_PS_NOHEADER_CNT;
+    hPsData->iidTimeCnt = MAX_TIME_DIFF_FRAMES;
+    hPsData->iccTimeCnt = MAX_TIME_DIFF_FRAMES;
+    hPsData->noEnvCnt = MAX_NOENV_CNT;
+  }
+
+  /*** Parameter processing, quantisation etc ***/
+  processIidData(hPsData, iid, psBands, nEnvelopes,
+                 hPsEncode->iidQuantErrorThreshold);
+  processIccData(hPsData, icc, psBands, nEnvelopes);
+
+  /*** Initialize output struct ***/
+
+  /* PS Header on/off ? */
+  if ((hPsData->headerCnt < MAX_PS_NOHEADER_CNT) &&
+      ((hPsData->iidQuantMode == hPsData->iidQuantModeLast) &&
+       (hPsData->iccQuantMode == hPsData->iccQuantModeLast)) &&
+      ((hPsData->iidEnable == hPsData->iidEnableLast) &&
+       (hPsData->iccEnable == hPsData->iccEnableLast))) {
+    hPsOut->enablePSHeader = 0;
+  } else {
+    hPsOut->enablePSHeader = 1;
+    hPsData->headerCnt = 0;
+  }
+
+  /* nEnvelopes = 0 ? */
+  if ((hPsData->noEnvCnt < MAX_NOENV_CNT) &&
+      (similarIid(hPsData, psBands, nEnvelopes)) &&
+      (similarIcc(hPsData, psBands, nEnvelopes))) {
+    hPsOut->nEnvelopes = nEnvelopes = 0;
+    hPsData->noEnvCnt++;
+  } else {
+    hPsData->noEnvCnt = 0;
+  }
+
+  if (nEnvelopes > 0) {
+    hPsOut->enableIID = hPsData->iidEnable;
+    hPsOut->iidMode = getIIDMode(psBands, hPsData->iidQuantMode);
+
+    hPsOut->enableICC = hPsData->iccEnable;
+    hPsOut->iccMode = getICCMode(psBands, hPsData->iccQuantMode);
+
+    hPsOut->enableIpdOpd = 0;
+    hPsOut->frameClass = 0;
+    hPsOut->nEnvelopes = nEnvelopes;
+
+    for (env = 0; env < nEnvelopes; env++) {
+      hPsOut->frameBorder[env] = envBorder[env + 1];
+      hPsOut->deltaIID[env] = (PS_DELTA)hPsData->iidDiffMode[env];
+      hPsOut->deltaICC[env] = (PS_DELTA)hPsData->iccDiffMode[env];
+      for (band = 0; band < psBands; band++) {
+        hPsOut->iid[env][band] = hPsData->iidIdx[env][band];
+        hPsOut->icc[env][band] = hPsData->iccIdx[env][band];
+      }
+    }
+
+    /* IPD OPD not supported right now */
+    FDKmemclear(hPsOut->ipd,
+                PS_MAX_ENVELOPES * PS_MAX_BANDS * sizeof(PS_DELTA));
+    for (env = 0; env < PS_MAX_ENVELOPES; env++) {
+      hPsOut->deltaIPD[env] = PS_DELTA_FREQ;
+      hPsOut->deltaOPD[env] = PS_DELTA_FREQ;
+    }
+
+    FDKmemclear(hPsOut->ipdLast, PS_MAX_BANDS * sizeof(INT));
+    FDKmemclear(hPsOut->opdLast, PS_MAX_BANDS * sizeof(INT));
+
+    for (band = 0; band < PS_MAX_BANDS; band++) {
+      hPsOut->iidLast[band] = hPsData->iidIdxLast[band];
+      hPsOut->iccLast[band] = hPsData->iccIdxLast[band];
+    }
+
+    /* save iids and iccs for differential time coding in the next frame */
+    hPsData->nEnvelopesLast = nEnvelopes;
+    hPsData->iidEnableLast = hPsData->iidEnable;
+    hPsData->iccEnableLast = hPsData->iccEnable;
+    hPsData->iidQuantModeLast = hPsData->iidQuantMode;
+    hPsData->iccQuantModeLast = hPsData->iccQuantMode;
+    for (i = 0; i < psBands; i++) {
+      hPsData->iidIdxLast[i] = hPsData->iidIdx[nEnvelopes - 1][i];
+      hPsData->iccIdxLast[i] = hPsData->iccIdx[nEnvelopes - 1][i];
+    }
+  } /* Envelope > 0 */
+
+  C_ALLOC_SCRATCH_END(pwrData, PS_PWR_DATA, 1)
+
+  return error;
+}
diff --git a/fdk-aac/libSBRenc/src/ps_encode.h b/fdk-aac/libSBRenc/src/ps_encode.h
new file mode 100644
index 0000000..4237a00
--- /dev/null
+++ b/fdk-aac/libSBRenc/src/ps_encode.h
@@ -0,0 +1,185 @@
+/* -----------------------------------------------------------------------------
+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 encoder library ******************************
+
+   Author(s):   M. Neuendorf, N. Rettelbach, M. Multrus
+
+   Description: PS Parameter extraction, encoding
+
+*******************************************************************************/
+
+/*!
+  \file
+  \brief  PS parameter extraction, encoding functions $Revision: 92790 $
+*/
+
+#ifndef PS_ENCODE_H
+#define PS_ENCODE_H
+
+#include "ps_const.h"
+#include "ps_bitenc.h"
+
+#define IID_SCALE_FT (64.f) /* maxVal in Quant tab is +/- 50 */
+#define IID_SCALE 6         /* maxVal in Quant tab is +/- 50 */
+#define IID_MAXVAL (1 << IID_SCALE)
+
+#define PS_QUANT_SCALE_FT \
+  (64.f) /* error smaller (64-25)/64 * 20 bands * 4 env -> QuantScale 64 */
+#define PS_QUANT_SCALE \
+  6 /* error smaller (64-25)/64 * 20 bands * 4 env -> QuantScale 6 bit */
+
+#define QMF_GROUPS_LO_RES 12
+#define SUBQMF_GROUPS_LO_RES 10
+#define QMF_GROUPS_HI_RES 18
+#define SUBQMF_GROUPS_HI_RES 30
+
+typedef struct T_PS_DATA {
+  INT iidEnable;
+  INT iidEnableLast;
+  INT iidQuantMode;
+  INT iidQuantModeLast;
+  INT iidDiffMode[PS_MAX_ENVELOPES];
+  INT iidIdx[PS_MAX_ENVELOPES][PS_MAX_BANDS];
+  INT iidIdxLast[PS_MAX_BANDS];
+
+  INT iccEnable;
+  INT iccEnableLast;
+  INT iccQuantMode;
+  INT iccQuantModeLast;
+  INT iccDiffMode[PS_MAX_ENVELOPES];
+  INT iccIdx[PS_MAX_ENVELOPES][PS_MAX_BANDS];
+  INT iccIdxLast[PS_MAX_BANDS];
+
+  INT nEnvelopesLast;
+
+  INT headerCnt;
+  INT iidTimeCnt;
+  INT iccTimeCnt;
+  INT noEnvCnt;
+
+} PS_DATA, *HANDLE_PS_DATA;
+
+typedef struct T_PS_ENCODE {
+  PS_DATA psData;
+
+  PS_BANDS psEncMode;
+  INT nQmfIidGroups;
+  INT nSubQmfIidGroups;
+  INT iidGroupBorders[QMF_GROUPS_HI_RES + SUBQMF_GROUPS_HI_RES + 1];
+  INT subband2parameterIndex[QMF_GROUPS_HI_RES + SUBQMF_GROUPS_HI_RES];
+  UCHAR iidGroupWidthLd[QMF_GROUPS_HI_RES + SUBQMF_GROUPS_HI_RES];
+  FIXP_DBL iidQuantErrorThreshold;
+
+  UCHAR psBandNrgScale[PS_MAX_BANDS];
+
+} PS_ENCODE;
+
+typedef struct T_PS_ENCODE *HANDLE_PS_ENCODE;
+
+FDK_PSENC_ERROR FDKsbrEnc_CreatePSEncode(HANDLE_PS_ENCODE *phPsEncode);
+
+FDK_PSENC_ERROR FDKsbrEnc_InitPSEncode(HANDLE_PS_ENCODE hPsEncode,
+                                       const PS_BANDS psEncMode,
+                                       const FIXP_DBL iidQuantErrorThreshold);
+
+FDK_PSENC_ERROR FDKsbrEnc_DestroyPSEncode(HANDLE_PS_ENCODE *phPsEncode);
+
+FDK_PSENC_ERROR FDKsbrEnc_PSEncode(
+    HANDLE_PS_ENCODE hPsEncode, HANDLE_PS_OUT hPsOut, UCHAR *dynBandScale,
+    UINT maxEnvelopes,
+    FIXP_DBL *hybridData[HYBRID_FRAMESIZE][MAX_PS_CHANNELS][2],
+    const INT frameSize, const INT sendHeader);
+
+#endif
diff --git a/fdk-aac/libSBRenc/src/ps_main.cpp b/fdk-aac/libSBRenc/src/ps_main.cpp
new file mode 100644
index 0000000..4d7a7a5
--- /dev/null
+++ b/fdk-aac/libSBRenc/src/ps_main.cpp
@@ -0,0 +1,606 @@
+/* -----------------------------------------------------------------------------
+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 encoder library ******************************
+
+   Author(s):   M. Multrus
+
+   Description: PS Wrapper, Downmix
+
+*******************************************************************************/
+
+#include "ps_main.h"
+
+/* Includes ******************************************************************/
+#include "ps_bitenc.h"
+#include "sbrenc_ram.h"
+
+/*--------------- function declarations --------------------*/
+static void psFindBestScaling(
+    HANDLE_PARAMETRIC_STEREO hParametricStereo,
+    FIXP_DBL *hybridData[HYBRID_FRAMESIZE][MAX_PS_CHANNELS][2],
+    UCHAR *dynBandScale, FIXP_DBL *maxBandValue, SCHAR *dmxScale);
+
+/*------------- function definitions ----------------*/
+FDK_PSENC_ERROR PSEnc_Create(HANDLE_PARAMETRIC_STEREO *phParametricStereo) {
+  FDK_PSENC_ERROR error = PSENC_OK;
+  HANDLE_PARAMETRIC_STEREO hParametricStereo = NULL;
+
+  if (phParametricStereo == NULL) {
+    error = PSENC_INVALID_HANDLE;
+  } else {
+    int i;
+
+    if (NULL == (hParametricStereo = GetRam_ParamStereo())) {
+      error = PSENC_MEMORY_ERROR;
+      goto bail;
+    }
+    FDKmemclear(hParametricStereo, sizeof(PARAMETRIC_STEREO));
+
+    if (PSENC_OK !=
+        (error = FDKsbrEnc_CreatePSEncode(&hParametricStereo->hPsEncode))) {
+      error = PSENC_MEMORY_ERROR;
+      goto bail;
+    }
+
+    for (i = 0; i < MAX_PS_CHANNELS; i++) {
+      if (FDKhybridAnalysisOpen(
+              &hParametricStereo->fdkHybAnaFilter[i],
+              hParametricStereo->__staticHybAnaStatesLF[i],
+              sizeof(hParametricStereo->__staticHybAnaStatesLF[i]),
+              hParametricStereo->__staticHybAnaStatesHF[i],
+              sizeof(hParametricStereo->__staticHybAnaStatesHF[i])) != 0) {
+        error = PSENC_MEMORY_ERROR;
+        goto bail;
+      }
+    }
+  }
+
+bail:
+  if (phParametricStereo != NULL) {
+    *phParametricStereo = hParametricStereo; /* return allocated handle */
+  }
+
+  if (error != PSENC_OK) {
+    PSEnc_Destroy(phParametricStereo);
+  }
+  return error;
+}
+
+FDK_PSENC_ERROR PSEnc_Init(HANDLE_PARAMETRIC_STEREO hParametricStereo,
+                           const HANDLE_PSENC_CONFIG hPsEncConfig,
+                           INT noQmfSlots, INT noQmfBands, UCHAR *dynamic_RAM) {
+  FDK_PSENC_ERROR error = PSENC_OK;
+
+  if ((NULL == hParametricStereo) || (NULL == hPsEncConfig)) {
+    error = PSENC_INVALID_HANDLE;
+  } else {
+    int ch, i;
+
+    hParametricStereo->initPS = 1;
+    hParametricStereo->noQmfSlots = noQmfSlots;
+    hParametricStereo->noQmfBands = noQmfBands;
+
+    /* clear delay lines */
+    FDKmemclear(hParametricStereo->qmfDelayLines,
+                sizeof(hParametricStereo->qmfDelayLines));
+
+    hParametricStereo->qmfDelayScale = FRACT_BITS - 1;
+
+    /* create configuration for hybrid filter bank */
+    for (ch = 0; ch < MAX_PS_CHANNELS; ch++) {
+      FDKhybridAnalysisInit(&hParametricStereo->fdkHybAnaFilter[ch],
+                            THREE_TO_TEN, 64, 64, 1);
+    } /* ch */
+
+    FDKhybridSynthesisInit(&hParametricStereo->fdkHybSynFilter, THREE_TO_TEN,
+                           64, 64);
+
+    /* determine average delay */
+    hParametricStereo->psDelay =
+        (HYBRID_FILTER_DELAY * hParametricStereo->noQmfBands);
+
+    if ((hPsEncConfig->maxEnvelopes < PSENC_NENV_1) ||
+        (hPsEncConfig->maxEnvelopes > PSENC_NENV_MAX)) {
+      hPsEncConfig->maxEnvelopes = PSENC_NENV_DEFAULT;
+    }
+    hParametricStereo->maxEnvelopes = hPsEncConfig->maxEnvelopes;
+
+    if (PSENC_OK !=
+        (error = FDKsbrEnc_InitPSEncode(
+             hParametricStereo->hPsEncode, (PS_BANDS)hPsEncConfig->nStereoBands,
+             hPsEncConfig->iidQuantErrorThreshold))) {
+      goto bail;
+    }
+
+    for (ch = 0; ch < MAX_PS_CHANNELS; ch++) {
+      FIXP_DBL *pDynReal = GetRam_Sbr_envRBuffer(ch, dynamic_RAM);
+      FIXP_DBL *pDynImag = GetRam_Sbr_envIBuffer(ch, dynamic_RAM);
+
+      for (i = 0; i < HYBRID_FRAMESIZE; i++) {
+        hParametricStereo->pHybridData[i + HYBRID_READ_OFFSET][ch][0] =
+            &pDynReal[i * MAX_HYBRID_BANDS];
+        hParametricStereo->pHybridData[i + HYBRID_READ_OFFSET][ch][1] =
+            &pDynImag[i * MAX_HYBRID_BANDS];
+        ;
+      }
+
+      for (i = 0; i < HYBRID_READ_OFFSET; i++) {
+        hParametricStereo->pHybridData[i][ch][0] =
+            hParametricStereo->__staticHybridData[i][ch][0];
+        hParametricStereo->pHybridData[i][ch][1] =
+            hParametricStereo->__staticHybridData[i][ch][1];
+      }
+    } /* ch */
+
+    /* clear static hybrid buffer */
+    FDKmemclear(hParametricStereo->__staticHybridData,
+                sizeof(hParametricStereo->__staticHybridData));
+
+    /* clear bs buffer */
+    FDKmemclear(hParametricStereo->psOut, sizeof(hParametricStereo->psOut));
+
+    hParametricStereo->psOut[0].enablePSHeader =
+        1; /* write ps header in first frame */
+
+    /* clear scaling buffer */
+    FDKmemclear(hParametricStereo->dynBandScale, sizeof(UCHAR) * PS_MAX_BANDS);
+    FDKmemclear(hParametricStereo->maxBandValue,
+                sizeof(FIXP_DBL) * PS_MAX_BANDS);
+
+  } /* valid handle */
+bail:
+  return error;
+}
+
+FDK_PSENC_ERROR PSEnc_Destroy(HANDLE_PARAMETRIC_STEREO *phParametricStereo) {
+  FDK_PSENC_ERROR error = PSENC_OK;
+
+  if (NULL != phParametricStereo) {
+    HANDLE_PARAMETRIC_STEREO hParametricStereo = *phParametricStereo;
+    if (hParametricStereo != NULL) {
+      FDKsbrEnc_DestroyPSEncode(&hParametricStereo->hPsEncode);
+      FreeRam_ParamStereo(phParametricStereo);
+    }
+  }
+
+  return error;
+}
+
+static FDK_PSENC_ERROR ExtractPSParameters(
+    HANDLE_PARAMETRIC_STEREO hParametricStereo, const int sendHeader,
+    FIXP_DBL *hybridData[HYBRID_FRAMESIZE][MAX_PS_CHANNELS][2]) {
+  FDK_PSENC_ERROR error = PSENC_OK;
+
+  if (hParametricStereo == NULL) {
+    error = PSENC_INVALID_HANDLE;
+  } else {
+    /* call ps encode function */
+    if (hParametricStereo->initPS) {
+      hParametricStereo->psOut[1] = hParametricStereo->psOut[0];
+    }
+    hParametricStereo->psOut[0] = hParametricStereo->psOut[1];
+
+    if (PSENC_OK !=
+        (error = FDKsbrEnc_PSEncode(
+             hParametricStereo->hPsEncode, &hParametricStereo->psOut[1],
+             hParametricStereo->dynBandScale, hParametricStereo->maxEnvelopes,
+             hybridData, hParametricStereo->noQmfSlots, sendHeader))) {
+      goto bail;
+    }
+
+    if (hParametricStereo->initPS) {
+      hParametricStereo->psOut[0] = hParametricStereo->psOut[1];
+      hParametricStereo->initPS = 0;
+    }
+  }
+bail:
+  return error;
+}
+
+static FDK_PSENC_ERROR DownmixPSQmfData(
+    HANDLE_PARAMETRIC_STEREO hParametricStereo,
+    HANDLE_QMF_FILTER_BANK sbrSynthQmf, FIXP_DBL **RESTRICT mixRealQmfData,
+    FIXP_DBL **RESTRICT mixImagQmfData, INT_PCM *downsampledOutSignal,
+    const UINT downsampledOutSignalBufSize,
+    FIXP_DBL *hybridData[HYBRID_FRAMESIZE][MAX_PS_CHANNELS][2],
+    const INT noQmfSlots, const INT psQmfScale[MAX_PS_CHANNELS],
+    SCHAR *qmfScale) {
+  FDK_PSENC_ERROR error = PSENC_OK;
+
+  if (hParametricStereo == NULL) {
+    error = PSENC_INVALID_HANDLE;
+  } else {
+    int n, k;
+    C_AALLOC_SCRATCH_START(pWorkBuffer, FIXP_DBL, 2 * 64)
+
+    /* define scalings */
+    int dynQmfScale = fixMax(
+        0, hParametricStereo->dmxScale -
+               1); /* scale one bit more for addition of left and right */
+    int downmixScale = psQmfScale[0] - dynQmfScale;
+    const FIXP_DBL maxStereoScaleFactor = MAXVAL_DBL; /* 2.f/2.f */
+
+    for (n = 0; n < noQmfSlots; n++) {
+      FIXP_DBL tmpHybrid[2][MAX_HYBRID_BANDS];
+
+      for (k = 0; k < 71; k++) {
+        int dynScale, sc; /* scaling */
+        FIXP_DBL tmpLeftReal, tmpRightReal, tmpLeftImag, tmpRightImag;
+        FIXP_DBL tmpScaleFactor, stereoScaleFactor;
+
+        tmpLeftReal = hybridData[n][0][0][k];
+        tmpLeftImag = hybridData[n][0][1][k];
+        tmpRightReal = hybridData[n][1][0][k];
+        tmpRightImag = hybridData[n][1][1][k];
+
+        sc = fixMax(
+            0, CntLeadingZeros(fixMax(
+                   fixMax(fixp_abs(tmpLeftReal), fixp_abs(tmpLeftImag)),
+                   fixMax(fixp_abs(tmpRightReal), fixp_abs(tmpRightImag)))) -
+                   2);
+
+        tmpLeftReal <<= sc;
+        tmpLeftImag <<= sc;
+        tmpRightReal <<= sc;
+        tmpRightImag <<= sc;
+        dynScale = fixMin(sc - dynQmfScale, DFRACT_BITS - 1);
+
+        /* calc stereo scale factor to avoid loss of energy in bands */
+        /* stereo scale factor = min(2.0f, sqrt( (abs(l(k, n)^2 + abs(r(k, n)^2
+         * )))/(0.5f*abs(l(k, n) + r(k, n))) )) */
+        stereoScaleFactor = fPow2Div2(tmpLeftReal) + fPow2Div2(tmpLeftImag) +
+                            fPow2Div2(tmpRightReal) + fPow2Div2(tmpRightImag);
+
+        /* might be that tmpScaleFactor becomes negative, so fabs(.) */
+        tmpScaleFactor =
+            fixp_abs(stereoScaleFactor + fMult(tmpLeftReal, tmpRightReal) +
+                     fMult(tmpLeftImag, tmpRightImag));
+
+        /* min(2.0f, sqrt(stereoScaleFactor/(0.5f*tmpScaleFactor)))  */
+        if ((stereoScaleFactor >> 1) <
+            fMult(maxStereoScaleFactor, tmpScaleFactor)) {
+          int sc_num = CountLeadingBits(stereoScaleFactor);
+          int sc_denum = CountLeadingBits(tmpScaleFactor);
+          sc = -(sc_num - sc_denum);
+
+          tmpScaleFactor = schur_div((stereoScaleFactor << (sc_num)) >> 1,
+                                     tmpScaleFactor << sc_denum, 16);
+
+          /* prevent odd scaling for next sqrt calculation */
+          if (sc & 0x1) {
+            sc++;
+            tmpScaleFactor >>= 1;
+          }
+          stereoScaleFactor = sqrtFixp(tmpScaleFactor);
+          stereoScaleFactor <<= (sc >> 1);
+        } else {
+          stereoScaleFactor = maxStereoScaleFactor;
+        }
+
+        /* write data to hybrid output */
+        tmpHybrid[0][k] = fMultDiv2(stereoScaleFactor,
+                                    (FIXP_DBL)(tmpLeftReal + tmpRightReal)) >>
+                          dynScale;
+        tmpHybrid[1][k] = fMultDiv2(stereoScaleFactor,
+                                    (FIXP_DBL)(tmpLeftImag + tmpRightImag)) >>
+                          dynScale;
+
+      } /* hybrid bands - k */
+
+      FDKhybridSynthesisApply(&hParametricStereo->fdkHybSynFilter, tmpHybrid[0],
+                              tmpHybrid[1], mixRealQmfData[n],
+                              mixImagQmfData[n]);
+
+      qmfSynthesisFilteringSlot(
+          sbrSynthQmf, mixRealQmfData[n], mixImagQmfData[n], downmixScale - 7,
+          downmixScale - 7,
+          downsampledOutSignal + (n * sbrSynthQmf->no_channels), 1,
+          pWorkBuffer);
+
+    } /* slots */
+
+    *qmfScale = -downmixScale + 7;
+
+    C_AALLOC_SCRATCH_END(pWorkBuffer, FIXP_DBL, 2 * 64)
+
+    {
+      const INT noQmfSlots2 = hParametricStereo->noQmfSlots >> 1;
+      const int noQmfBands = hParametricStereo->noQmfBands;
+
+      INT scale, i, j, slotOffset;
+
+      FIXP_DBL tmp[2][64];
+
+      for (i = 0; i < noQmfSlots2; i++) {
+        FDKmemcpy(tmp[0], hParametricStereo->qmfDelayLines[0][i],
+                  noQmfBands * sizeof(FIXP_DBL));
+        FDKmemcpy(tmp[1], hParametricStereo->qmfDelayLines[1][i],
+                  noQmfBands * sizeof(FIXP_DBL));
+
+        FDKmemcpy(hParametricStereo->qmfDelayLines[0][i],
+                  mixRealQmfData[i + noQmfSlots2],
+                  noQmfBands * sizeof(FIXP_DBL));
+        FDKmemcpy(hParametricStereo->qmfDelayLines[1][i],
+                  mixImagQmfData[i + noQmfSlots2],
+                  noQmfBands * sizeof(FIXP_DBL));
+
+        FDKmemcpy(mixRealQmfData[i + noQmfSlots2], mixRealQmfData[i],
+                  noQmfBands * sizeof(FIXP_DBL));
+        FDKmemcpy(mixImagQmfData[i + noQmfSlots2], mixImagQmfData[i],
+                  noQmfBands * sizeof(FIXP_DBL));
+
+        FDKmemcpy(mixRealQmfData[i], tmp[0], noQmfBands * sizeof(FIXP_DBL));
+        FDKmemcpy(mixImagQmfData[i], tmp[1], noQmfBands * sizeof(FIXP_DBL));
+      }
+
+      if (hParametricStereo->qmfDelayScale > *qmfScale) {
+        scale = hParametricStereo->qmfDelayScale - *qmfScale;
+        slotOffset = 0;
+      } else {
+        scale = *qmfScale - hParametricStereo->qmfDelayScale;
+        slotOffset = noQmfSlots2;
+      }
+
+      for (i = 0; i < noQmfSlots2; i++) {
+        for (j = 0; j < noQmfBands; j++) {
+          mixRealQmfData[i + slotOffset][j] >>= scale;
+          mixImagQmfData[i + slotOffset][j] >>= scale;
+        }
+      }
+
+      scale = *qmfScale;
+      *qmfScale = fMin(*qmfScale, hParametricStereo->qmfDelayScale);
+      hParametricStereo->qmfDelayScale = scale;
+    }
+
+  } /* valid handle */
+
+  return error;
+}
+
+INT FDKsbrEnc_PSEnc_WritePSData(HANDLE_PARAMETRIC_STEREO hParametricStereo,
+                                HANDLE_FDK_BITSTREAM hBitstream) {
+  return (
+      (hParametricStereo != NULL)
+          ? FDKsbrEnc_WritePSBitstream(&hParametricStereo->psOut[0], hBitstream)
+          : 0);
+}
+
+FDK_PSENC_ERROR FDKsbrEnc_PSEnc_ParametricStereoProcessing(
+    HANDLE_PARAMETRIC_STEREO hParametricStereo, INT_PCM *samples[2],
+    UINT samplesBufSize, QMF_FILTER_BANK **hQmfAnalysis,
+    FIXP_DBL **RESTRICT downmixedRealQmfData,
+    FIXP_DBL **RESTRICT downmixedImagQmfData, INT_PCM *downsampledOutSignal,
+    HANDLE_QMF_FILTER_BANK sbrSynthQmf, SCHAR *qmfScale, const int sendHeader) {
+  FDK_PSENC_ERROR error = PSENC_OK;
+  INT psQmfScale[MAX_PS_CHANNELS] = {0};
+  int psCh, i;
+  C_AALLOC_SCRATCH_START(pWorkBuffer, FIXP_DBL, 4 * 64)
+
+  for (psCh = 0; psCh < MAX_PS_CHANNELS; psCh++) {
+    for (i = 0; i < hQmfAnalysis[psCh]->no_col; i++) {
+      qmfAnalysisFilteringSlot(
+          hQmfAnalysis[psCh], &pWorkBuffer[2 * 64], /* qmfReal[64] */
+          &pWorkBuffer[3 * 64],                     /* qmfImag[64] */
+          samples[psCh] + i * hQmfAnalysis[psCh]->no_channels, 1,
+          &pWorkBuffer[0 * 64] /* qmf workbuffer 2*64 */
+      );
+
+      FDKhybridAnalysisApply(
+          &hParametricStereo->fdkHybAnaFilter[psCh],
+          &pWorkBuffer[2 * 64], /* qmfReal[64] */
+          &pWorkBuffer[3 * 64], /* qmfImag[64] */
+          hParametricStereo->pHybridData[i + HYBRID_READ_OFFSET][psCh][0],
+          hParametricStereo->pHybridData[i + HYBRID_READ_OFFSET][psCh][1]);
+
+    } /* no_col loop  i  */
+
+    psQmfScale[psCh] = hQmfAnalysis[psCh]->outScalefactor;
+
+  } /* for psCh */
+
+  C_AALLOC_SCRATCH_END(pWorkBuffer, FIXP_DBL, 4 * 64)
+
+  /* find best scaling in new QMF and Hybrid data */
+  psFindBestScaling(
+      hParametricStereo, &hParametricStereo->pHybridData[HYBRID_READ_OFFSET],
+      hParametricStereo->dynBandScale, hParametricStereo->maxBandValue,
+      &hParametricStereo->dmxScale);
+
+  /* extract the ps parameters */
+  if (PSENC_OK !=
+      (error = ExtractPSParameters(hParametricStereo, sendHeader,
+                                   &hParametricStereo->pHybridData[0]))) {
+    goto bail;
+  }
+
+  /* save hybrid date for next frame */
+  for (i = 0; i < HYBRID_READ_OFFSET; i++) {
+    FDKmemcpy(
+        hParametricStereo->pHybridData[i][0][0],
+        hParametricStereo->pHybridData[hParametricStereo->noQmfSlots + i][0][0],
+        MAX_HYBRID_BANDS * sizeof(FIXP_DBL)); /* left, real */
+    FDKmemcpy(
+        hParametricStereo->pHybridData[i][0][1],
+        hParametricStereo->pHybridData[hParametricStereo->noQmfSlots + i][0][1],
+        MAX_HYBRID_BANDS * sizeof(FIXP_DBL)); /* left, imag */
+    FDKmemcpy(
+        hParametricStereo->pHybridData[i][1][0],
+        hParametricStereo->pHybridData[hParametricStereo->noQmfSlots + i][1][0],
+        MAX_HYBRID_BANDS * sizeof(FIXP_DBL)); /* right, real */
+    FDKmemcpy(
+        hParametricStereo->pHybridData[i][1][1],
+        hParametricStereo->pHybridData[hParametricStereo->noQmfSlots + i][1][1],
+        MAX_HYBRID_BANDS * sizeof(FIXP_DBL)); /* right, imag */
+  }
+
+  /* downmix and hybrid synthesis */
+  if (PSENC_OK !=
+      (error = DownmixPSQmfData(
+           hParametricStereo, sbrSynthQmf, downmixedRealQmfData,
+           downmixedImagQmfData, downsampledOutSignal, samplesBufSize,
+           &hParametricStereo->pHybridData[HYBRID_READ_OFFSET],
+           hParametricStereo->noQmfSlots, psQmfScale, qmfScale))) {
+    goto bail;
+  }
+
+bail:
+
+  return error;
+}
+
+static void psFindBestScaling(
+    HANDLE_PARAMETRIC_STEREO hParametricStereo,
+    FIXP_DBL *hybridData[HYBRID_FRAMESIZE][MAX_PS_CHANNELS][2],
+    UCHAR *dynBandScale, FIXP_DBL *maxBandValue, SCHAR *dmxScale) {
+  HANDLE_PS_ENCODE hPsEncode = hParametricStereo->hPsEncode;
+
+  INT group, bin, col, band;
+  const INT frameSize = hParametricStereo->noQmfSlots;
+  const INT psBands = (INT)hPsEncode->psEncMode;
+  const INT nIidGroups = hPsEncode->nQmfIidGroups + hPsEncode->nSubQmfIidGroups;
+
+  /* group wise scaling */
+  FIXP_DBL maxVal[2][PS_MAX_BANDS];
+  FIXP_DBL maxValue = FL2FXCONST_DBL(0.f);
+
+  FDKmemclear(maxVal, sizeof(maxVal));
+
+  /* start with hybrid data */
+  for (group = 0; group < nIidGroups; group++) {
+    /* Translate group to bin */
+    bin = hPsEncode->subband2parameterIndex[group];
+
+    /* Translate from 20 bins to 10 bins */
+    if (hPsEncode->psEncMode == PS_BANDS_COARSE) {
+      bin >>= 1;
+    }
+
+    /* QMF downmix scaling */
+    for (col = 0; col < frameSize; col++) {
+      int i, section = (col < frameSize - HYBRID_READ_OFFSET) ? 0 : 1;
+      FIXP_DBL tmp = maxVal[section][bin];
+      for (i = hPsEncode->iidGroupBorders[group];
+           i < hPsEncode->iidGroupBorders[group + 1]; i++) {
+        tmp = fixMax(tmp, (FIXP_DBL)fixp_abs(hybridData[col][0][0][i]));
+        tmp = fixMax(tmp, (FIXP_DBL)fixp_abs(hybridData[col][0][1][i]));
+        tmp = fixMax(tmp, (FIXP_DBL)fixp_abs(hybridData[col][1][0][i]));
+        tmp = fixMax(tmp, (FIXP_DBL)fixp_abs(hybridData[col][1][1][i]));
+      }
+      maxVal[section][bin] = tmp;
+    }
+  } /* nIidGroups */
+
+  /* convert maxSpec to maxScaling, find scaling space */
+  for (band = 0; band < psBands; band++) {
+#ifndef MULT_16x16
+    dynBandScale[band] =
+        CountLeadingBits(fixMax(maxVal[0][band], maxBandValue[band]));
+#else
+    dynBandScale[band] = fixMax(
+        0, CountLeadingBits(fixMax(maxVal[0][band], maxBandValue[band])) -
+               FRACT_BITS);
+#endif
+    maxValue = fixMax(maxValue, fixMax(maxVal[0][band], maxVal[1][band]));
+    maxBandValue[band] = fixMax(maxVal[0][band], maxVal[1][band]);
+  }
+
+    /* calculate maximal scaling for QMF downmix */
+#ifndef MULT_16x16
+  *dmxScale = fixMin(DFRACT_BITS, CountLeadingBits(maxValue));
+#else
+  *dmxScale = fixMax(0, fixMin(FRACT_BITS, CountLeadingBits((maxValue))));
+#endif
+}
diff --git a/fdk-aac/libSBRenc/src/ps_main.h b/fdk-aac/libSBRenc/src/ps_main.h
new file mode 100644
index 0000000..88b2993
--- /dev/null
+++ b/fdk-aac/libSBRenc/src/ps_main.h
@@ -0,0 +1,270 @@
+/* -----------------------------------------------------------------------------
+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 encoder library ******************************
+
+   Author(s):   Markus Multrus
+
+   Description: PS Wrapper, Downmix header file
+
+*******************************************************************************/
+
+#ifndef PS_MAIN_H
+#define PS_MAIN_H
+
+/* Includes ******************************************************************/
+
+#include "sbr_def.h"
+#include "qmf.h"
+#include "ps_encode.h"
+#include "FDK_bitstream.h"
+#include "FDK_hybrid.h"
+
+/* Data Types ****************************************************************/
+typedef enum {
+  PSENC_STEREO_BANDS_INVALID = 0,
+  PSENC_STEREO_BANDS_10 = 10,
+  PSENC_STEREO_BANDS_20 = 20
+
+} PSENC_STEREO_BANDS_CONFIG;
+
+typedef enum {
+  PSENC_NENV_1 = 1,
+  PSENC_NENV_2 = 2,
+  PSENC_NENV_4 = 4,
+  PSENC_NENV_DEFAULT = PSENC_NENV_2,
+  PSENC_NENV_MAX = PSENC_NENV_4
+
+} PSENC_NENV_CONFIG;
+
+typedef struct {
+  UINT bitrateFrom; /* inclusive */
+  UINT bitrateTo;   /* exclusive */
+  PSENC_STEREO_BANDS_CONFIG nStereoBands;
+  PSENC_NENV_CONFIG nEnvelopes;
+  LONG iidQuantErrorThreshold; /* quantization threshold to switch between
+                                  coarse and fine iid quantization */
+
+} psTuningTable_t;
+
+/* Function / Class Declarations *********************************************/
+
+typedef struct T_PARAMETRIC_STEREO {
+  HANDLE_PS_ENCODE hPsEncode;
+  PS_OUT psOut[2];
+
+  FIXP_DBL __staticHybridData[HYBRID_READ_OFFSET][MAX_PS_CHANNELS][2]
+                             [MAX_HYBRID_BANDS];
+  FIXP_DBL
+  *pHybridData[HYBRID_READ_OFFSET + HYBRID_FRAMESIZE][MAX_PS_CHANNELS][2];
+
+  FIXP_DBL qmfDelayLines[2][32 >> 1][64];
+  int qmfDelayScale;
+
+  INT psDelay;
+  UINT maxEnvelopes;
+  UCHAR dynBandScale[PS_MAX_BANDS];
+  FIXP_DBL maxBandValue[PS_MAX_BANDS];
+  SCHAR dmxScale;
+  INT initPS;
+  INT noQmfSlots;
+  INT noQmfBands;
+
+  FIXP_DBL __staticHybAnaStatesLF[MAX_PS_CHANNELS][2 * HYBRID_FILTER_LENGTH *
+                                                   HYBRID_MAX_QMF_BANDS];
+  FIXP_DBL __staticHybAnaStatesHF[MAX_PS_CHANNELS][2 * HYBRID_FILTER_DELAY *
+                                                   (64 - HYBRID_MAX_QMF_BANDS)];
+  FDK_ANA_HYB_FILTER fdkHybAnaFilter[MAX_PS_CHANNELS];
+  FDK_SYN_HYB_FILTER fdkHybSynFilter;
+
+} PARAMETRIC_STEREO;
+
+typedef struct T_PSENC_CONFIG {
+  INT frameSize;
+  INT qmfFilterMode;
+  INT sbrPsDelay;
+  PSENC_STEREO_BANDS_CONFIG nStereoBands;
+  PSENC_NENV_CONFIG maxEnvelopes;
+  FIXP_DBL iidQuantErrorThreshold;
+
+} PSENC_CONFIG, *HANDLE_PSENC_CONFIG;
+
+typedef struct T_PARAMETRIC_STEREO *HANDLE_PARAMETRIC_STEREO;
+
+/**
+ * \brief  Create a parametric stereo encoder instance.
+ *
+ * \param phParametricStereo    A pointer to a parametric stereo handle to be
+ * allocated. Initialized on return.
+ *
+ * \return
+ *          - PSENC_OK, on succes.
+ *          - PSENC_INVALID_HANDLE, PSENC_MEMORY_ERROR, on failure.
+ */
+FDK_PSENC_ERROR PSEnc_Create(HANDLE_PARAMETRIC_STEREO *phParametricStereo);
+
+/**
+ * \brief  Initialize a parametric stereo encoder instance.
+ *
+ * \param hParametricStereo     Meta Data handle.
+ * \param hPsEncConfig          Filled parametric stereo configuration
+ * structure.
+ * \param noQmfSlots            Number of slots within one audio frame.
+ * \param noQmfBands            Number of QMF bands.
+ * \param dynamic_RAM           Pointer to preallocated workbuffer.
+ *
+ * \return
+ *          - PSENC_OK, on succes.
+ *          - PSENC_INVALID_HANDLE, PSENC_INIT_ERROR, on failure.
+ */
+FDK_PSENC_ERROR PSEnc_Init(HANDLE_PARAMETRIC_STEREO hParametricStereo,
+                           const HANDLE_PSENC_CONFIG hPsEncConfig,
+                           INT noQmfSlots, INT noQmfBands, UCHAR *dynamic_RAM);
+
+/**
+ * \brief  Destroy parametric stereo encoder instance.
+ *
+ * Deallocate instance and free whole memory.
+ *
+ * \param phParametricStereo    Pointer to the parametric stereo handle to be
+ * deallocated.
+ *
+ * \return
+ *          - PSENC_OK, on succes.
+ *          - PSENC_INVALID_HANDLE, on failure.
+ */
+FDK_PSENC_ERROR PSEnc_Destroy(HANDLE_PARAMETRIC_STEREO *phParametricStereo);
+
+/**
+ * \brief  Apply parametric stereo processing.
+ *
+ * \param hParametricStereo     Meta Data handle.
+ * \param samples               Pointer to 2 channel audio input signal.
+ * \param timeInStride,         Stride factor of input buffer.
+ * \param hQmfAnalysis,         Pointer to QMF analysis filterbanks.
+ * \param downmixedRealQmfData  Pointer to real QMF buffer to be written to.
+ * \param downmixedImagQmfData  Pointer to imag QMF buffer to be written to.
+ * \param downsampledOutSignal  Pointer to buffer where to write downmixed
+ * timesignal.
+ * \param sbrSynthQmf           Pointer to QMF synthesis filterbank.
+ * \param qmfScale              Return scaling factor of the qmf data.
+ * \param sendHeader            Signal whether to write header data.
+ *
+ * \return
+ *          - PSENC_OK, on succes.
+ *          - PSENC_INVALID_HANDLE, PSENC_ENCODE_ERROR, on failure.
+ */
+FDK_PSENC_ERROR FDKsbrEnc_PSEnc_ParametricStereoProcessing(
+    HANDLE_PARAMETRIC_STEREO hParametricStereo, INT_PCM *samples[2],
+    UINT timeInStride, QMF_FILTER_BANK **hQmfAnalysis,
+    FIXP_DBL **RESTRICT downmixedRealQmfData,
+    FIXP_DBL **RESTRICT downmixedImagQmfData, INT_PCM *downsampledOutSignal,
+    HANDLE_QMF_FILTER_BANK sbrSynthQmf, SCHAR *qmfScale, const int sendHeader);
+
+/**
+ * \brief  Write parametric stereo bitstream.
+ *
+ * Write ps_data() element to bitstream and return number of written bits.
+ * Returns number of written bits only, if hBitstream == NULL.
+ *
+ * \param hParametricStereo     Meta Data handle.
+ * \param hBitstream            Bitstream buffer handle.
+ *
+ * \return
+ *          - number of written bits.
+ */
+INT FDKsbrEnc_PSEnc_WritePSData(HANDLE_PARAMETRIC_STEREO hParametricStereo,
+                                HANDLE_FDK_BITSTREAM hBitstream);
+
+#endif /* PS_MAIN_H */
diff --git a/fdk-aac/libSBRenc/src/resampler.cpp b/fdk-aac/libSBRenc/src/resampler.cpp
new file mode 100644
index 0000000..b1781a7
--- /dev/null
+++ b/fdk-aac/libSBRenc/src/resampler.cpp
@@ -0,0 +1,444 @@
+/* -----------------------------------------------------------------------------
+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 encoder library ******************************
+
+   Author(s):
+
+   Description:
+
+*******************************************************************************/
+
+/*!
+  \file
+  \brief  FDK resampler tool box:$Revision: 91655 $
+  \author M. Werner
+*/
+
+#include "resampler.h"
+
+#include "genericStds.h"
+
+/**************************************************************************/
+/*                   BIQUAD Filter Specifications                         */
+/**************************************************************************/
+
+#define B1 0
+#define B2 1
+#define A1 2
+#define A2 3
+
+#define BQC(x) FL2FXCONST_SGL(x / 2)
+
+struct FILTER_PARAM {
+  const FIXP_SGL *coeffa; /*! SOS matrix One row/section. Scaled using BQC().
+                             Order of coefficients: B1,B2,A1,A2. B0=A0=1.0 */
+  FIXP_DBL g;             /*! overall gain */
+  int Wc;       /*! normalized passband bandwidth at input samplerate * 1000 */
+  int noCoeffs; /*! number of filter coeffs */
+  int delay;    /*! delay in samples at input samplerate */
+};
+
+#define BIQUAD_COEFSTEP 4
+
+/**
+ *\brief Low Pass
+ Wc = 0,5, order 30, Stop Band -96dB. Wc criteria is "almost 0dB passband", not
+ the usual -3db gain point. [b,a]=cheby2(30,96,0.505) [sos,g]=tf2sos(b,a)
+ bandwidth 0.48
+ */
+static const FIXP_SGL sos48[] = {
+    BQC(1.98941075681938),      BQC(0.999999996890811),
+    BQC(0.863264527201963),     BQC(0.189553799960663),
+    BQC(1.90733804822445),      BQC(1.00000001736189),
+    BQC(0.836321575841691),     BQC(0.203505809266564),
+    BQC(1.75616665495325),      BQC(0.999999946079721),
+    BQC(0.784699225121588),     BQC(0.230471265506986),
+    BQC(1.55727745512726),      BQC(1.00000011737815),
+    BQC(0.712515423588351),     BQC(0.268752723900498),
+    BQC(1.33407591943643),      BQC(0.999999795953228),
+    BQC(0.625059117330989),     BQC(0.316194685288965),
+    BQC(1.10689898412458),      BQC(1.00000035057114),
+    BQC(0.52803514366398),      BQC(0.370517843224669),
+    BQC(0.89060371078454),      BQC(0.999999343962822),
+    BQC(0.426920462165257),     BQC(0.429608200207746),
+    BQC(0.694438261209433),     BQC(1.0000008629792),
+    BQC(0.326530699561716),     BQC(0.491714450654174),
+    BQC(0.523237800935322),     BQC(1.00000101349782),
+    BQC(0.230829556274851),     BQC(0.555559034843281),
+    BQC(0.378631165929563),     BQC(0.99998986482665),
+    BQC(0.142906422036095),     BQC(0.620338874442411),
+    BQC(0.260786911308437),     BQC(1.00003261460178),
+    BQC(0.0651008576256505),    BQC(0.685759923926262),
+    BQC(0.168409429188098),     BQC(0.999933049695828),
+    BQC(-0.000790067789975562), BQC(0.751905896602325),
+    BQC(0.100724533818628),     BQC(1.00009472669872),
+    BQC(-0.0533772830257041),   BQC(0.81930744384525),
+    BQC(0.0561434357867363),    BQC(0.999911636304276),
+    BQC(-0.0913550299236405),   BQC(0.88883625875915),
+    BQC(0.0341680678662057),    BQC(1.00003667508676),
+    BQC(-0.113405185536697),    BQC(0.961756638268446)};
+
+static const FIXP_DBL g48 =
+    FL2FXCONST_DBL(0.002712866530047) - (FIXP_DBL)0x8000;
+
+static const struct FILTER_PARAM param_set48 = {
+    sos48, g48, 480, 15, 4 /* LF 2 */
+};
+
+/**
+ *\brief Low Pass
+ Wc = 0,5, order 24, Stop Band -96dB. Wc criteria is "almost 0dB passband", not
+ the usual -3db gain point. [b,a]=cheby2(24,96,0.5) [sos,g]=tf2sos(b,a)
+ bandwidth 0.45
+ */
+static const FIXP_SGL sos45[] = {
+    BQC(1.982962601444),     BQC(1.00000000007504),    BQC(0.646113303737836),
+    BQC(0.10851149979981),   BQC(1.85334094281111),    BQC(0.999999999677192),
+    BQC(0.612073220102006),  BQC(0.130022141698044),   BQC(1.62541051415425),
+    BQC(1.00000000080398),   BQC(0.547879702855959),   BQC(0.171165825133192),
+    BQC(1.34554656923247),   BQC(0.9999999980169),     BQC(0.460373914508491),
+    BQC(0.228677463376354),  BQC(1.05656568503116),    BQC(1.00000000569363),
+    BQC(0.357891894038287),  BQC(0.298676843912185),   BQC(0.787967587877312),
+    BQC(0.999999984415017),  BQC(0.248826893211877),   BQC(0.377441803512978),
+    BQC(0.555480971120497),  BQC(1.00000003583307),    BQC(0.140614263345315),
+    BQC(0.461979302213679),  BQC(0.364986207070964),   BQC(0.999999932084303),
+    BQC(0.0392669446074516), BQC(0.55033451180825),    BQC(0.216827267631558),
+    BQC(1.00000010534682),   BQC(-0.0506232228865103), BQC(0.641691581560946),
+    BQC(0.108951672277119),  BQC(0.999999871167516),   BQC(-0.125584840183225),
+    BQC(0.736367748771803),  BQC(0.0387988607229035),  BQC(1.00000011205574),
+    BQC(-0.182814849097974), BQC(0.835802108714964),   BQC(0.0042866175809225),
+    BQC(0.999999954830813),  BQC(-0.21965740617151),   BQC(0.942623047782363)};
+
+static const FIXP_DBL g45 =
+    FL2FXCONST_DBL(0.00242743980909524) - (FIXP_DBL)0x8000;
+
+static const struct FILTER_PARAM param_set45 = {
+    sos45, g45, 450, 12, 4 /* LF 2 */
+};
+
+/*
+ Created by Octave 2.1.73, Mon Oct 13 17:31:32 2008 CEST
+ Wc = 0,5, order 16, Stop Band -96dB damping.
+ [b,a]=cheby2(16,96,0.5)
+ [sos,g]=tf2sos(b,a)
+ bandwidth = 0.41
+ */
+
+static const FIXP_SGL sos41[] = {
+    BQC(1.96193625292),      BQC(0.999999999999964),   BQC(0.169266178786789),
+    BQC(0.0128823300475907), BQC(1.68913437662092),    BQC(1.00000000000053),
+    BQC(0.124751503206552),  BQC(0.0537472273950989),  BQC(1.27274692366017),
+    BQC(0.999999999995674),  BQC(0.0433108625178357),  BQC(0.131015753236317),
+    BQC(0.85214175088395),   BQC(1.00000000001813),    BQC(-0.0625658152550408),
+    BQC(0.237763778993806),  BQC(0.503841579939009),   BQC(0.999999999953223),
+    BQC(-0.179176128722865), BQC(0.367475236424474),   BQC(0.249990711986162),
+    BQC(1.00000000007952),   BQC(-0.294425165824676),  BQC(0.516594857170212),
+    BQC(0.087971668680286),  BQC(0.999999999915528),   BQC(-0.398956566777928),
+    BQC(0.686417767801123),  BQC(0.00965373325350294), BQC(1.00000000003744),
+    BQC(-0.48579173764817),  BQC(0.884931534239068)};
+
+static const FIXP_DBL g41 = FL2FXCONST_DBL(0.00155956951169248);
+
+static const struct FILTER_PARAM param_set41 = {
+    sos41, g41, 410, 8, 5 /* LF 3 */
+};
+
+/*
+ # Created by Octave 2.1.73, Mon Oct 13 17:55:33 2008 CEST
+ Wc = 0,5, order 12, Stop Band -96dB damping.
+ [b,a]=cheby2(12,96,0.5);
+ [sos,g]=tf2sos(b,a)
+*/
+static const FIXP_SGL sos35[] = {
+    BQC(1.93299325235762),   BQC(0.999999999999985),  BQC(-0.140733187246596),
+    BQC(0.0124139497836062), BQC(1.4890416764109),    BQC(1.00000000000011),
+    BQC(-0.198215402588504), BQC(0.0746730616584138), BQC(0.918450161309795),
+    BQC(0.999999999999619),  BQC(-0.30133912791941),  BQC(0.192276468839529),
+    BQC(0.454877024246818),  BQC(1.00000000000086),   BQC(-0.432337328809815),
+    BQC(0.356852933642815),  BQC(0.158017147118507),  BQC(0.999999999998876),
+    BQC(-0.574817494249777), BQC(0.566380436970833),  BQC(0.0171834649478749),
+    BQC(1.00000000000055),   BQC(-0.718581178041165), BQC(0.83367484487889)};
+
+static const FIXP_DBL g35 = FL2FXCONST_DBL(0.00162580994125131);
+
+static const struct FILTER_PARAM param_set35 = {sos35, g35, 350, 6, 4};
+
+/*
+ # Created by Octave 2.1.73, Mon Oct 13 18:15:38 2008 CEST
+ Wc = 0,5, order 8, Stop Band -96dB damping.
+ [b,a]=cheby2(8,96,0.5);
+ [sos,g]=tf2sos(b,a)
+*/
+static const FIXP_SGL sos25[] = {
+    BQC(1.85334094301225),   BQC(1.0),
+    BQC(-0.702127214212663), BQC(0.132452403998767),
+    BQC(1.056565682167),     BQC(0.999999999999997),
+    BQC(-0.789503667880785), BQC(0.236328693569128),
+    BQC(0.364986307455489),  BQC(0.999999999999996),
+    BQC(-0.955191189843375), BQC(0.442966457936379),
+    BQC(0.0387985751642125), BQC(1.0),
+    BQC(-1.19817786088084),  BQC(0.770493895456328)};
+
+static const FIXP_DBL g25 = FL2FXCONST_DBL(0.000945182835294559);
+
+static const struct FILTER_PARAM param_set25 = {sos25, g25, 250, 4, 5};
+
+/* Must be sorted in descending order */
+static const struct FILTER_PARAM *const filter_paramSet[] = {
+    &param_set48, &param_set45, &param_set41, &param_set35, &param_set25};
+
+/**************************************************************************/
+/*                         Resampler Functions                            */
+/**************************************************************************/
+
+/*!
+  \brief   Reset downsampler instance and clear delay lines
+
+  \return  success of operation
+*/
+
+INT FDKaacEnc_InitDownsampler(
+    DOWNSAMPLER *DownSampler, /*!< pointer to downsampler instance */
+    int Wc,                   /*!< normalized cutoff freq * 1000*  */
+    int ratio)                /*!< downsampler ratio */
+
+{
+  UINT i;
+  const struct FILTER_PARAM *currentSet = NULL;
+
+  FDKmemclear(DownSampler->downFilter.states,
+              sizeof(DownSampler->downFilter.states));
+  DownSampler->downFilter.ptr = 0;
+
+  /*
+    find applicable parameter set
+  */
+  currentSet = filter_paramSet[0];
+  for (i = 1; i < sizeof(filter_paramSet) / sizeof(struct FILTER_PARAM *);
+       i++) {
+    if (filter_paramSet[i]->Wc <= Wc) {
+      break;
+    }
+    currentSet = filter_paramSet[i];
+  }
+
+  DownSampler->downFilter.coeffa = currentSet->coeffa;
+
+  DownSampler->downFilter.gain = currentSet->g;
+  FDK_ASSERT(currentSet->noCoeffs <= MAXNR_SECTIONS * 2);
+
+  DownSampler->downFilter.noCoeffs = currentSet->noCoeffs;
+  DownSampler->delay = currentSet->delay;
+  DownSampler->downFilter.Wc = currentSet->Wc;
+
+  DownSampler->ratio = ratio;
+  DownSampler->pending = ratio - 1;
+  return (1);
+}
+
+/*!
+  \brief   faster simple folding operation
+           Filter:
+           H(z) = A(z)/B(z)
+           with
+           A(z) = a[0]*z^0 + a[1]*z^1 + a[2]*z^2 ... a[n]*z^n
+
+  \return  filtered value
+*/
+
+static inline INT_PCM AdvanceFilter(
+    LP_FILTER *downFilter, /*!< pointer to iir filter instance */
+    INT_PCM *pInput,       /*!< input of filter                */
+    int downRatio) {
+  INT_PCM output;
+  int i, n;
+
+#define BIQUAD_SCALE 12
+
+  FIXP_DBL y = FL2FXCONST_DBL(0.0f);
+  FIXP_DBL input;
+
+  for (n = 0; n < downRatio; n++) {
+    FIXP_BQS(*states)[2] = downFilter->states;
+    const FIXP_SGL *coeff = downFilter->coeffa;
+    int s1, s2;
+
+    s1 = downFilter->ptr;
+    s2 = s1 ^ 1;
+
+#if (SAMPLE_BITS == 16)
+    input = ((FIXP_DBL)pInput[n]) << (DFRACT_BITS - SAMPLE_BITS - BIQUAD_SCALE);
+#elif (SAMPLE_BITS == 32)
+    input = pInput[n] >> BIQUAD_SCALE;
+#else
+#error NOT IMPLEMENTED
+#endif
+
+    FIXP_BQS state1, state2, state1b, state2b;
+
+    state1 = states[0][s1];
+    state2 = states[0][s2];
+
+    /* Loop over sections */
+    for (i = 0; i < downFilter->noCoeffs; i++) {
+      FIXP_DBL state0;
+
+      /* Load merged states (from next section) */
+      state1b = states[i + 1][s1];
+      state2b = states[i + 1][s2];
+
+      state0 = input + fMult(state1, coeff[B1]) + fMult(state2, coeff[B2]);
+      y = state0 - fMult(state1b, coeff[A1]) - fMult(state2b, coeff[A2]);
+
+      /* Store new feed forward merge state */
+      states[i + 1][s2] = y << 1;
+      /* Store new feed backward state */
+      states[i][s2] = input << 1;
+
+      /* Feedback output to next section. */
+      input = y;
+
+      /* Transfer merged states */
+      state1 = state1b;
+      state2 = state2b;
+
+      /* Step to next coef set */
+      coeff += BIQUAD_COEFSTEP;
+    }
+    downFilter->ptr ^= 1;
+  }
+  /* Apply global gain */
+  y = fMult(y, downFilter->gain);
+
+  /* Apply final gain/scaling to output */
+#if (SAMPLE_BITS == 16)
+  output = (INT_PCM)SATURATE_RIGHT_SHIFT(
+      y + (FIXP_DBL)(1 << (DFRACT_BITS - SAMPLE_BITS - BIQUAD_SCALE - 1)),
+      DFRACT_BITS - SAMPLE_BITS - BIQUAD_SCALE, SAMPLE_BITS);
+  // output = (INT_PCM) SATURATE_RIGHT_SHIFT(y,
+  // DFRACT_BITS-SAMPLE_BITS-BIQUAD_SCALE, SAMPLE_BITS);
+#else
+  output = SATURATE_LEFT_SHIFT(y, BIQUAD_SCALE, SAMPLE_BITS);
+#endif
+
+  return output;
+}
+
+/*!
+  \brief   FDKaacEnc_Downsample numInSamples of type INT_PCM
+           Returns number of output samples in numOutSamples
+
+  \return  success of operation
+*/
+
+INT FDKaacEnc_Downsample(
+    DOWNSAMPLER *DownSampler, /*!< pointer to downsampler instance */
+    INT_PCM *inSamples,       /*!< pointer to input samples */
+    INT numInSamples,         /*!< number  of input samples  */
+    INT_PCM *outSamples,      /*!< pointer to output samples */
+    INT *numOutSamples        /*!< pointer tp number of output samples */
+) {
+  INT i;
+  *numOutSamples = 0;
+
+  for (i = 0; i < numInSamples; i += DownSampler->ratio) {
+    *outSamples = AdvanceFilter(&(DownSampler->downFilter), &inSamples[i],
+                                DownSampler->ratio);
+    outSamples++;
+  }
+  *numOutSamples = numInSamples / DownSampler->ratio;
+
+  return 0;
+}
diff --git a/fdk-aac/libSBRenc/src/resampler.h b/fdk-aac/libSBRenc/src/resampler.h
new file mode 100644
index 0000000..7aa1cae
--- /dev/null
+++ b/fdk-aac/libSBRenc/src/resampler.h
@@ -0,0 +1,159 @@
+/* -----------------------------------------------------------------------------
+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 encoder library ******************************
+
+   Author(s):
+
+   Description:
+
+*******************************************************************************/
+
+#ifndef RESAMPLER_H
+#define RESAMPLER_H
+/*!
+  \file
+  \brief  Fixed Point Resampler Tool Box $Revision: 92790 $
+*/
+
+#include "common_fix.h"
+
+/**************************************************************************/
+/*                         BIQUAD Filter Structure                           */
+/**************************************************************************/
+
+#define MAXNR_SECTIONS (15)
+
+typedef FIXP_DBL FIXP_BQS;
+
+typedef struct {
+  FIXP_BQS states[MAXNR_SECTIONS + 1][2]; /*! state buffer */
+  const FIXP_SGL *coeffa;                 /*! pointer to filter coeffs */
+  FIXP_DBL gain;                          /*! overall gain factor */
+  int Wc;                                 /*! normalized cutoff freq * 1000 */
+  int noCoeffs;                           /*! number of filter coeffs sets */
+  int ptr;                                /*! index to rinbuffers */
+} LP_FILTER;
+
+/**************************************************************************/
+/*                        Downsampler Structure                           */
+/**************************************************************************/
+
+typedef struct {
+  LP_FILTER downFilter; /*! filter instance */
+  int ratio;            /*! downsampling ration */
+  int delay;            /*! downsampling delay (source fs)   */
+  int pending;          /*! number of pending output samples */
+} DOWNSAMPLER;
+
+/**
+ * \brief Initialized a given downsampler structure.
+ */
+INT FDKaacEnc_InitDownsampler(
+    DOWNSAMPLER *DownSampler, /*!< pointer to downsampler instance */
+    INT Wc,                   /*!< normalized cutoff freq * 1000 */
+    INT ratio);               /*!< downsampler ratio */
+
+/**
+ * \brief Downsample a set of audio samples. numInSamples must be at least equal
+ * to the downsampler ratio.
+ */
+INT FDKaacEnc_Downsample(
+    DOWNSAMPLER *DownSampler, /*!< pointer to downsampler instance */
+    INT_PCM *inSamples,       /*!< pointer to input samples */
+    INT numInSamples,         /*!< number  of input samples  */
+    INT_PCM *outSamples,      /*!< pointer to output samples */
+    INT *numOutSamples);      /*!< pointer tp number of output samples */
+
+#endif /* RESAMPLER_H */
diff --git a/fdk-aac/libSBRenc/src/sbr.h b/fdk-aac/libSBRenc/src/sbr.h
new file mode 100644
index 0000000..341dcab
--- /dev/null
+++ b/fdk-aac/libSBRenc/src/sbr.h
@@ -0,0 +1,194 @@
+/* -----------------------------------------------------------------------------
+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 encoder library ******************************
+
+   Author(s):
+
+   Description:
+
+*******************************************************************************/
+
+/*!
+  \file
+  \brief  Main SBR structs definitions $Revision: 92790 $
+*/
+
+#ifndef SBR_H
+#define SBR_H
+
+#include "fram_gen.h"
+#include "bit_sbr.h"
+#include "tran_det.h"
+#include "code_env.h"
+#include "env_est.h"
+#include "cmondata.h"
+
+#include "qmf.h"
+#include "resampler.h"
+
+#include "ton_corr.h"
+
+/* SBR bitstream delay */
+#define MAX_DELAY_FRAMES 2
+
+/* sbr encoder downsampling type */
+typedef enum { SBRENC_DS_NONE, SBRENC_DS_TIME, SBRENC_DS_QMF } SBRENC_DS_TYPE;
+
+typedef struct SBR_CHANNEL {
+  struct ENV_CHANNEL hEnvChannel;
+  // INT_PCM                  *pDSOutBuffer;            /**< Pointer to
+  // downsampled audio output of SBR encoder */
+  DOWNSAMPLER downSampler;
+
+} SBR_CHANNEL;
+typedef SBR_CHANNEL* HANDLE_SBR_CHANNEL;
+
+typedef struct SBR_ELEMENT {
+  HANDLE_SBR_CHANNEL sbrChannel[2];
+  QMF_FILTER_BANK* hQmfAnalysis[2];
+  SBR_CONFIG_DATA sbrConfigData;
+  SBR_HEADER_DATA sbrHeaderData;
+  SBR_BITSTREAM_DATA sbrBitstreamData;
+  COMMON_DATA CmonData;
+  INT dynXOverFreqDelay[5]; /**< to delay a frame (I don't like it that much
+                               that way - hrc) */
+  SBR_ELEMENT_INFO elInfo;
+
+  UCHAR payloadDelayLine[1 + MAX_DELAY_FRAMES][MAX_PAYLOAD_SIZE];
+  UINT payloadDelayLineSize[1 + MAX_DELAY_FRAMES]; /* Sizes in bits */
+
+} SBR_ELEMENT, *HANDLE_SBR_ELEMENT;
+
+typedef struct SBR_ENCODER {
+  HANDLE_SBR_ELEMENT sbrElement[(8)];
+  HANDLE_SBR_CHANNEL pSbrChannel[(8)];
+  QMF_FILTER_BANK QmfAnalysis[(8)];
+  DOWNSAMPLER lfeDownSampler;
+  int lfeChIdx;     /* -1 default for no lfe, else assign channel index. */
+  int noElements;   /* Number of elements. */
+  int nChannels;    /* Total channel count across all elements. */
+  int frameSize;    /* SBR framelength. */
+  int bufferOffset; /* Offset for SBR parameter extraction in time domain input
+                       buffer. */
+  int downsampledOffset; /* Offset of downsampled/mixed output for core encoder.
+                          */
+  int downmixSize;       /* Size in samples of downsampled/mixed output for core
+                            encoder. */
+  INT downSampleFactor;  /* Sampling rate relation between the SBR and the core
+                            encoder. */
+  SBRENC_DS_TYPE
+  downsamplingMethod; /* Method of downsmapling, time-domain, QMF or none.
+                       */
+  int nBitstrDelay;   /* Amount of SBR frames to be delayed in bitstream domain.
+                       */
+  int sbrDecDelay;    /* SBR decoder delay in samples */
+  INT estimateBitrate; /* Estimate bitrate of SBR encoder. */
+  INT inputDataDelay;  /* Delay caused by downsampler, in/out buffer at
+                          sbrEncoder_EncodeFrame. */
+
+  UCHAR* dynamicRam;
+  UCHAR* pSBRdynamic_RAM;
+
+  HANDLE_PARAMETRIC_STEREO hParametricStereo;
+  QMF_FILTER_BANK qmfSynthesisPS;
+
+  /* parameters describing allocation volume of present instance */
+  INT maxElements;
+  INT maxChannels;
+  INT supportPS;
+
+} SBR_ENCODER;
+
+#endif /* SBR_H */
diff --git a/fdk-aac/libSBRenc/src/sbr_def.h b/fdk-aac/libSBRenc/src/sbr_def.h
new file mode 100644
index 0000000..53eba71
--- /dev/null
+++ b/fdk-aac/libSBRenc/src/sbr_def.h
@@ -0,0 +1,276 @@
+/* -----------------------------------------------------------------------------
+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 encoder library ******************************
+
+   Author(s):
+
+   Description:
+
+*******************************************************************************/
+
+/*!
+  \file
+  \brief  SBR main definitions $Revision: 92790 $
+*/
+#ifndef SBR_DEF_H
+#define SBR_DEF_H
+
+#include "common_fix.h"
+
+#define noError 0
+#define HANDLE_ERROR_INFO INT
+#define ERROR(a, b) 1
+
+/* #define SBR_ENV_STATISTICS_BITRATE */
+#undef SBR_ENV_STATISTICS_BITRATE
+
+/* #define SBR_ENV_STATISTICS */
+#undef SBR_ENV_STATISTICS
+
+/* #define SBR_PAYLOAD_MONITOR */
+#undef SBR_PAYLOAD_MONITOR
+
+#define SWAP(a, b) tempr = a, a = b, b = tempr
+#define TRUE 1
+#define FALSE 0
+
+/* Constants */
+#define EPS 1e-12
+#define LOG2 0.69314718056f /* natural logarithm of 2 */
+#define ILOG2 1.442695041f  /* 1/LOG2 */
+#define RELAXATION_FLOAT (1e-6f)
+#define RELAXATION (FL2FXCONST_DBL(RELAXATION_FLOAT))
+#define RELAXATION_FRACT \
+  (FL2FXCONST_DBL(0.524288f)) /* 0.524288f is fractional part of RELAXATION */
+#define RELAXATION_SHIFT (19)
+#define RELAXATION_LD64                                 \
+  (FL2FXCONST_DBL(0.31143075889f)) /* (ld64(RELAXATION) \
+                                    */
+
+/************  Definitions ***************/
+#define SBR_COMP_MODE_DELTA 0
+#define SBR_COMP_MODE_CTS 1
+#define SBR_MAX_ENERGY_VALUES 5
+#define SBR_GLOBAL_TONALITY_VALUES 2
+
+#define MAX_NUM_CHANNELS 2
+
+#define MAX_NOISE_ENVELOPES 2
+#define MAX_NUM_NOISE_COEFFS 5
+#define MAX_NUM_NOISE_VALUES (MAX_NUM_NOISE_COEFFS * MAX_NOISE_ENVELOPES)
+
+#define MAX_NUM_ENVELOPE_VALUES (MAX_ENVELOPES * MAX_FREQ_COEFFS)
+#define MAX_ENVELOPES 5
+#define MAX_FREQ_COEFFS 48
+
+#define MAX_FREQ_COEFFS_FS44100 35
+#define MAX_FREQ_COEFFS_FS48000 32
+
+#define NO_OF_ESTIMATES_LC 4
+#define NO_OF_ESTIMATES_LD 3
+#define MAX_NO_OF_ESTIMATES 4
+
+#define NOISE_FLOOR_OFFSET 6
+#define NOISE_FLOOR_OFFSET_64 (FL2FXCONST_DBL(0.09375f))
+
+#define LOW_RES 0
+#define HIGH_RES 1
+
+#define LO 0
+#define HI 1
+
+#define LENGTH_SBR_FRAME_INFO 35 /* 19 */
+
+#define SBR_NSFB_LOW_RES 9   /*  8 */
+#define SBR_NSFB_HIGH_RES 18 /* 16 */
+
+#define SBR_XPOS_CTRL_DEFAULT 2
+
+#define SBR_FREQ_SCALE_DEFAULT 2
+#define SBR_ALTER_SCALE_DEFAULT 1
+#define SBR_NOISE_BANDS_DEFAULT 2
+
+#define SBR_LIMITER_BANDS_DEFAULT 2
+#define SBR_LIMITER_GAINS_DEFAULT 2
+#define SBR_LIMITER_GAINS_INFINITE 3
+#define SBR_INTERPOL_FREQ_DEFAULT 1
+#define SBR_SMOOTHING_LENGTH_DEFAULT 0
+
+/* sbr_header */
+#define SI_SBR_AMP_RES_BITS 1
+#define SI_SBR_COUPLING_BITS 1
+#define SI_SBR_START_FREQ_BITS 4
+#define SI_SBR_STOP_FREQ_BITS 4
+#define SI_SBR_XOVER_BAND_BITS 3
+#define SI_SBR_RESERVED_BITS 2
+#define SI_SBR_DATA_EXTRA_BITS 1
+#define SI_SBR_HEADER_EXTRA_1_BITS 1
+#define SI_SBR_HEADER_EXTRA_2_BITS 1
+
+/* sbr_header extra 1 */
+#define SI_SBR_FREQ_SCALE_BITS 2
+#define SI_SBR_ALTER_SCALE_BITS 1
+#define SI_SBR_NOISE_BANDS_BITS 2
+
+/* sbr_header extra 2 */
+#define SI_SBR_LIMITER_BANDS_BITS 2
+#define SI_SBR_LIMITER_GAINS_BITS 2
+#define SI_SBR_INTERPOL_FREQ_BITS 1
+#define SI_SBR_SMOOTHING_LENGTH_BITS 1
+
+/* sbr_grid */
+#define SBR_CLA_BITS 2    /*!< size of bs_frame_class */
+#define SBR_CLA_BITS_LD 1 /*!< size of bs_frame_class */
+#define SBR_ENV_BITS 2    /*!< size of bs_num_env_raw */
+#define SBR_ABS_BITS 2    /*!< size of bs_abs_bord_raw for HE-AAC */
+#define SBR_NUM_BITS 2    /*!< size of bs_num_rel */
+#define SBR_REL_BITS 2    /*!< size of bs_rel_bord_raw */
+#define SBR_RES_BITS 1    /*!< size of bs_freq_res_flag */
+#define SBR_DIR_BITS 1    /*!< size of bs_df_flag */
+
+/* sbr_data */
+#define SI_SBR_INVF_MODE_BITS 2
+
+#define SI_SBR_START_ENV_BITS_AMP_RES_3_0 6
+#define SI_SBR_START_ENV_BITS_BALANCE_AMP_RES_3_0 5
+#define SI_SBR_START_NOISE_BITS_AMP_RES_3_0 5
+#define SI_SBR_START_NOISE_BITS_BALANCE_AMP_RES_3_0 5
+
+#define SI_SBR_START_ENV_BITS_AMP_RES_1_5 7
+#define SI_SBR_START_ENV_BITS_BALANCE_AMP_RES_1_5 6
+
+#define SI_SBR_EXTENDED_DATA_BITS 1
+#define SI_SBR_EXTENSION_SIZE_BITS 4
+#define SI_SBR_EXTENSION_ESC_COUNT_BITS 8
+#define SI_SBR_EXTENSION_ID_BITS 2
+
+#define SBR_EXTENDED_DATA_MAX_CNT (15 + 255)
+
+#define EXTENSION_ID_PS_CODING 2
+
+/* Envelope coding constants */
+#define FREQ 0
+#define TIME 1
+
+/* qmf data scaling */
+#define QMF_SCALE_OFFSET 7
+
+/* huffman tables */
+#define CODE_BOOK_SCF_LAV00 60
+#define CODE_BOOK_SCF_LAV01 31
+#define CODE_BOOK_SCF_LAV10 60
+#define CODE_BOOK_SCF_LAV11 31
+#define CODE_BOOK_SCF_LAV_BALANCE11 12
+#define CODE_BOOK_SCF_LAV_BALANCE10 24
+
+typedef enum { SBR_AMP_RES_1_5 = 0, SBR_AMP_RES_3_0 } AMP_RES;
+
+typedef enum {
+  XPOS_MDCT,
+  XPOS_MDCT_CROSS,
+  XPOS_LC,
+  XPOS_RESERVED,
+  XPOS_SWITCHED /* not a real choice but used here to control behaviour */
+} XPOS_MODE;
+
+typedef enum {
+  INVF_OFF = 0,
+  INVF_LOW_LEVEL,
+  INVF_MID_LEVEL,
+  INVF_HIGH_LEVEL,
+  INVF_SWITCHED /* not a real choice but used here to control behaviour */
+} INVF_MODE;
+
+#endif
diff --git a/fdk-aac/libSBRenc/src/sbr_encoder.cpp b/fdk-aac/libSBRenc/src/sbr_encoder.cpp
new file mode 100644
index 0000000..26257a1
--- /dev/null
+++ b/fdk-aac/libSBRenc/src/sbr_encoder.cpp
@@ -0,0 +1,2577 @@
+/* -----------------------------------------------------------------------------
+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 encoder library ******************************
+
+   Author(s):   Andreas Ehret, Tobias Chalupka
+
+   Description: SBR encoder top level processing.
+
+*******************************************************************************/
+
+#include "sbr_encoder.h"
+
+#include "sbrenc_ram.h"
+#include "sbrenc_rom.h"
+#include "sbrenc_freq_sca.h"
+#include "env_bit.h"
+#include "cmondata.h"
+#include "sbr_misc.h"
+#include "sbr.h"
+#include "qmf.h"
+
+#include "ps_main.h"
+
+#define SBRENCODER_LIB_VL0 4
+#define SBRENCODER_LIB_VL1 0
+#define SBRENCODER_LIB_VL2 0
+
+/***************************************************************************/
+/*
+ * SBR Delay balancing definitions.
+ */
+
+/*
+      input buffer (1ch)
+
+      |------------ 1537   -------------|-----|---------- 2048 -------------|
+           (core2sbr delay     )          ds     (read, core and ds area)
+*/
+
+#define SFB(dwnsmp) \
+  (32 << (dwnsmp -  \
+          1)) /* SBR Frequency bands: 64 for dual-rate, 32 for single-rate */
+#define STS(fl)                                                              \
+  (((fl) == 1024) ? 32                                                       \
+                  : 30) /* SBR Time Slots: 32 for core frame length 1024, 30 \
+                           for core frame length 960 */
+
+#define DELAY_QMF_ANA(dwnsmp) \
+  ((320 << ((dwnsmp)-1)) - (32 << ((dwnsmp)-1))) /* Full bandwidth */
+#define DELAY_HYB_ANA (10 * 64) /* + 0.5 */      /*  */
+#define DELAY_HYB_SYN (6 * 64 - 32)              /*  */
+#define DELAY_QMF_POSTPROC(dwnsmp) \
+  (32 * (dwnsmp))                               /* QMF postprocessing delay */
+#define DELAY_DEC_QMF(dwnsmp) (6 * SFB(dwnsmp)) /* Decoder QMF overlap */
+#define DELAY_QMF_SYN(dwnsmp) \
+  (1 << (dwnsmp -             \
+         1)) /* QMF_NO_POLY/2=2.5, rounded down to 2, half for single-rate */
+#define DELAY_QMF_DS (32) /* QMF synthesis for downsampled time signal */
+
+/* Delay in QMF paths */
+#define DELAY_SBR(fl, dwnsmp) \
+  (DELAY_QMF_ANA(dwnsmp) + (SFB(dwnsmp) * STS(fl) - 1) + DELAY_QMF_SYN(dwnsmp))
+#define DELAY_PS(fl, dwnsmp)                                       \
+  (DELAY_QMF_ANA(dwnsmp) + DELAY_HYB_ANA + DELAY_DEC_QMF(dwnsmp) + \
+   (SFB(dwnsmp) * STS(fl) - 1) + DELAY_HYB_SYN + DELAY_QMF_SYN(dwnsmp))
+#define DELAY_ELDSBR(fl, dwnsmp) \
+  ((((fl) / 2) * (dwnsmp)) - 1 + DELAY_QMF_POSTPROC(dwnsmp))
+#define DELAY_ELDv2SBR(fl, dwnsmp)                                        \
+  ((((fl) / 2) * (dwnsmp)) - 1 + 80 * (dwnsmp)) /* 80 is the delay caused \
+                                                   by the sum of the CLD  \
+                                                   analysis and the MPSLD \
+                                                   synthesis filterbank */
+
+/* Delay in core path (core and downsampler not taken into account) */
+#define DELAY_COREPATH_SBR(fl, dwnsmp) \
+  ((DELAY_QMF_ANA(dwnsmp) + DELAY_DEC_QMF(dwnsmp) + DELAY_QMF_SYN(dwnsmp)))
+#define DELAY_COREPATH_ELDSBR(fl, dwnsmp) ((DELAY_QMF_POSTPROC(dwnsmp)))
+#define DELAY_COREPATH_ELDv2SBR(fl, dwnsmp) (128 * (dwnsmp)) /* 4 slots */
+#define DELAY_COREPATH_PS(fl, dwnsmp)                                        \
+  ((DELAY_QMF_ANA(dwnsmp) + DELAY_QMF_DS +                                   \
+    /*(DELAY_AAC(fl)*2) + */ DELAY_QMF_ANA(dwnsmp) + DELAY_DEC_QMF(dwnsmp) + \
+    DELAY_HYB_SYN + DELAY_QMF_SYN(dwnsmp))) /* 2048 - 463*2 */
+
+/* Delay differences between SBR- and downsampled path for SBR and SBR+PS */
+#define DELAY_AAC2SBR(fl, dwnsmp) \
+  ((DELAY_COREPATH_SBR(fl, dwnsmp)) - DELAY_SBR((fl), (dwnsmp)))
+#define DELAY_ELD2SBR(fl, dwnsmp) \
+  ((DELAY_COREPATH_ELDSBR(fl, dwnsmp)) - DELAY_ELDSBR(fl, dwnsmp))
+#define DELAY_AAC2PS(fl, dwnsmp) \
+  ((DELAY_COREPATH_PS(fl, dwnsmp)) - DELAY_PS(fl, dwnsmp)) /* 2048 - 463*2 */
+
+/* Assumption: The sample delay resulting of of DELAY_AAC2PS is always smaller
+ * than the sample delay implied by DELAY_AAC2SBR */
+#define MAX_DS_FILTER_DELAY \
+  (5) /* the additional max downsampler filter delay (source fs) */
+#define MAX_SAMPLE_DELAY                                                 \
+  (DELAY_AAC2SBR(1024, 2) + MAX_DS_FILTER_DELAY) /* maximum delay: frame \
+                                                    length of 1024 and   \
+                                                    dual-rate sbr */
+
+/***************************************************************************/
+
+/*************** Delay parameters for sbrEncoder_Init_delay() **************/
+typedef struct {
+  int dsDelay;        /* the delay of the (time-domain) downsampler itself */
+  int delay;          /* overall delay / samples  */
+  int sbrDecDelay;    /* SBR decoder's delay */
+  int corePathOffset; /* core path offset / samples; added by
+                         sbrEncoder_Init_delay() */
+  int sbrPathOffset;  /* SBR path offset / samples; added by
+                         sbrEncoder_Init_delay() */
+  int bitstrDelay; /* bitstream delay / frames; added by sbrEncoder_Init_delay()
+                    */
+  int delayInput2Core; /* delay of the input to the core / samples */
+} DELAY_PARAM;
+/***************************************************************************/
+
+#define INVALID_TABLE_IDX -1
+
+/***************************************************************************/
+/*!
+
+  \brief  Selects the SBR tuning settings to use dependent on number of
+          channels, bitrate, sample rate and core coder
+
+  \return Index to the appropriate table
+
+****************************************************************************/
+#define DISTANCE_CEIL_VALUE 5000000
+static INT getSbrTuningTableIndex(
+    UINT bitrate,     /*! the total bitrate in bits/sec */
+    UINT numChannels, /*! the number of channels for the core coder */
+    UINT sampleRate,  /*! the sampling rate of the core coder */
+    AUDIO_OBJECT_TYPE core, UINT *pBitRateClosest) {
+  int i, bitRateClosestLowerIndex = -1, bitRateClosestUpperIndex = -1,
+         found = 0;
+  UINT bitRateClosestUpper = 0, bitRateClosestLower = DISTANCE_CEIL_VALUE;
+
+#define isForThisCore(i)                                                     \
+  ((sbrTuningTable[i].coreCoder == CODEC_AACLD && core == AOT_ER_AAC_ELD) || \
+   (sbrTuningTable[i].coreCoder == CODEC_AAC && core != AOT_ER_AAC_ELD))
+
+  for (i = 0; i < sbrTuningTableSize; i++) {
+    if (isForThisCore(i)) /* tuning table is for this core codec */
+    {
+      if (numChannels == sbrTuningTable[i].numChannels &&
+          sampleRate == sbrTuningTable[i].sampleRate) {
+        found = 1;
+        if ((bitrate >= sbrTuningTable[i].bitrateFrom) &&
+            (bitrate < sbrTuningTable[i].bitrateTo)) {
+          return i;
+        } else {
+          if (sbrTuningTable[i].bitrateFrom > bitrate) {
+            if (sbrTuningTable[i].bitrateFrom < bitRateClosestLower) {
+              bitRateClosestLower = sbrTuningTable[i].bitrateFrom;
+              bitRateClosestLowerIndex = i;
+            }
+          }
+          if (sbrTuningTable[i].bitrateTo <= bitrate) {
+            if (sbrTuningTable[i].bitrateTo > bitRateClosestUpper) {
+              bitRateClosestUpper = sbrTuningTable[i].bitrateTo - 1;
+              bitRateClosestUpperIndex = i;
+            }
+          }
+        }
+      }
+    }
+  }
+
+  if (bitRateClosestUpperIndex >= 0) {
+    return bitRateClosestUpperIndex;
+  }
+
+  if (pBitRateClosest != NULL) {
+    /* If there was at least one matching tuning entry pick the least distance
+     * bit rate */
+    if (found) {
+      int distanceUpper = DISTANCE_CEIL_VALUE,
+          distanceLower = DISTANCE_CEIL_VALUE;
+      if (bitRateClosestLowerIndex >= 0) {
+        distanceLower =
+            sbrTuningTable[bitRateClosestLowerIndex].bitrateFrom - bitrate;
+      }
+      if (bitRateClosestUpperIndex >= 0) {
+        distanceUpper =
+            bitrate - sbrTuningTable[bitRateClosestUpperIndex].bitrateTo;
+      }
+      if (distanceUpper < distanceLower) {
+        *pBitRateClosest = bitRateClosestUpper;
+      } else {
+        *pBitRateClosest = bitRateClosestLower;
+      }
+    } else {
+      *pBitRateClosest = 0;
+    }
+  }
+
+  return INVALID_TABLE_IDX;
+}
+
+/***************************************************************************/
+/*!
+
+  \brief  Selects the PS tuning settings to use dependent on bitrate
+  and core coder
+
+  \return Index to the appropriate table
+
+****************************************************************************/
+static INT getPsTuningTableIndex(UINT bitrate, UINT *pBitRateClosest) {
+  INT i, paramSets = sizeof(psTuningTable) / sizeof(psTuningTable[0]);
+  int bitRateClosestLowerIndex = -1, bitRateClosestUpperIndex = -1;
+  UINT bitRateClosestUpper = 0, bitRateClosestLower = DISTANCE_CEIL_VALUE;
+
+  for (i = 0; i < paramSets; i++) {
+    if ((bitrate >= psTuningTable[i].bitrateFrom) &&
+        (bitrate < psTuningTable[i].bitrateTo)) {
+      return i;
+    } else {
+      if (psTuningTable[i].bitrateFrom > bitrate) {
+        if (psTuningTable[i].bitrateFrom < bitRateClosestLower) {
+          bitRateClosestLower = psTuningTable[i].bitrateFrom;
+          bitRateClosestLowerIndex = i;
+        }
+      }
+      if (psTuningTable[i].bitrateTo <= bitrate) {
+        if (psTuningTable[i].bitrateTo > bitRateClosestUpper) {
+          bitRateClosestUpper = psTuningTable[i].bitrateTo - 1;
+          bitRateClosestUpperIndex = i;
+        }
+      }
+    }
+  }
+
+  if (bitRateClosestUpperIndex >= 0) {
+    return bitRateClosestUpperIndex;
+  }
+
+  if (pBitRateClosest != NULL) {
+    int distanceUpper = DISTANCE_CEIL_VALUE,
+        distanceLower = DISTANCE_CEIL_VALUE;
+    if (bitRateClosestLowerIndex >= 0) {
+      distanceLower =
+          sbrTuningTable[bitRateClosestLowerIndex].bitrateFrom - bitrate;
+    }
+    if (bitRateClosestUpperIndex >= 0) {
+      distanceUpper =
+          bitrate - sbrTuningTable[bitRateClosestUpperIndex].bitrateTo;
+    }
+    if (distanceUpper < distanceLower) {
+      *pBitRateClosest = bitRateClosestUpper;
+    } else {
+      *pBitRateClosest = bitRateClosestLower;
+    }
+  }
+
+  return INVALID_TABLE_IDX;
+}
+
+/***************************************************************************/
+/*!
+
+  \brief  In case of downsampled SBR we may need to lower the stop freq
+          of a tuning setting to fit into the lower half of the
+          spectrum ( which is sampleRate/4 )
+
+  \return the adapted stop frequency index (-1 -> error)
+
+  \ingroup SbrEncCfg
+
+****************************************************************************/
+static INT FDKsbrEnc_GetDownsampledStopFreq(const INT sampleRateCore,
+                                            const INT startFreq, INT stopFreq,
+                                            const INT downSampleFactor) {
+  INT maxStopFreqRaw = sampleRateCore / 2;
+  INT startBand, stopBand;
+  HANDLE_ERROR_INFO err;
+
+  while (stopFreq > 0 && FDKsbrEnc_getSbrStopFreqRAW(stopFreq, sampleRateCore) >
+                             maxStopFreqRaw) {
+    stopFreq--;
+  }
+
+  if (FDKsbrEnc_getSbrStopFreqRAW(stopFreq, sampleRateCore) > maxStopFreqRaw)
+    return -1;
+
+  err = FDKsbrEnc_FindStartAndStopBand(
+      sampleRateCore << (downSampleFactor - 1), sampleRateCore,
+      32 << (downSampleFactor - 1), startFreq, stopFreq, &startBand, &stopBand);
+  if (err) return -1;
+
+  return stopFreq;
+}
+
+/***************************************************************************/
+/*!
+
+  \brief  tells us, if for the given coreCoder, bitrate, number of channels
+          and input sampling rate an SBR setting is available. If yes, it
+          tells us also the core sampling rate we would need to run with
+
+  \return a flag indicating success: yes (1) or no (0)
+
+****************************************************************************/
+static UINT FDKsbrEnc_IsSbrSettingAvail(
+    UINT bitrate,           /*! the total bitrate in bits/sec */
+    UINT vbrMode,           /*! the vbr paramter, 0 means constant bitrate */
+    UINT numOutputChannels, /*! the number of channels for the core coder */
+    UINT sampleRateInput,   /*! the input sample rate [in Hz] */
+    UINT sampleRateCore,    /*! the core's sampling rate */
+    AUDIO_OBJECT_TYPE core) {
+  INT idx = INVALID_TABLE_IDX;
+
+  if (sampleRateInput < 16000) return 0;
+
+  if (bitrate == 0) {
+    /* map vbr quality to bitrate */
+    if (vbrMode < 30)
+      bitrate = 24000;
+    else if (vbrMode < 40)
+      bitrate = 28000;
+    else if (vbrMode < 60)
+      bitrate = 32000;
+    else if (vbrMode < 75)
+      bitrate = 40000;
+    else
+      bitrate = 48000;
+    bitrate *= numOutputChannels;
+  }
+
+  idx = getSbrTuningTableIndex(bitrate, numOutputChannels, sampleRateCore, core,
+                               NULL);
+
+  return (idx == INVALID_TABLE_IDX ? 0 : 1);
+}
+
+/***************************************************************************/
+/*!
+
+  \brief  Adjusts the SBR settings according to the chosen core coder
+          settings which are accessible via config->codecSettings
+
+  \return A flag indicating success: yes (1) or no (0)
+
+****************************************************************************/
+static UINT FDKsbrEnc_AdjustSbrSettings(
+    const sbrConfigurationPtr config, /*! output, modified */
+    UINT bitRate,                     /*! the total bitrate in bits/sec */
+    UINT numChannels,                 /*! the core coder number of channels */
+    UINT sampleRateCore,              /*! the core coder sampling rate in Hz */
+    UINT sampleRateSbr,               /*! the sbr coder sampling rate in Hz */
+    UINT transFac,                    /*! the short block to long block ratio */
+    UINT standardBitrate, /*! the standard bitrate per channel in bits/sec */
+    UINT vbrMode, /*! the vbr paramter, 0 poor quality .. 100 high quality*/
+    UINT useSpeechConfig,   /*!< adapt tuning parameters for speech ? */
+    UINT lcsMode,           /*! the low complexity stereo mode */
+    UINT bParametricStereo, /*!< use parametric stereo */
+    AUDIO_OBJECT_TYPE core) /* Core audio codec object type */
+{
+  INT idx = INVALID_TABLE_IDX;
+  /* set the core codec settings */
+  config->codecSettings.bitRate = bitRate;
+  config->codecSettings.nChannels = numChannels;
+  config->codecSettings.sampleFreq = sampleRateCore;
+  config->codecSettings.transFac = transFac;
+  config->codecSettings.standardBitrate = standardBitrate;
+
+  if (bitRate < 28000) {
+    config->threshold_AmpRes_FF_m = (FIXP_DBL)MAXVAL_DBL;
+    config->threshold_AmpRes_FF_e = 7;
+  } else if (bitRate >= 28000 && bitRate <= 48000) {
+    /* The float threshold is 75
+       0.524288f is fractional part of RELAXATION, the quotaMatrix and therefore
+       tonality are scaled by this 2/3 is because the original implementation
+       divides the tonality values by 3, here it's divided by 2 128 compensates
+       the necessary shiftfactor of 7 */
+    config->threshold_AmpRes_FF_m =
+        FL2FXCONST_DBL(75.0f * 0.524288f / (2.0f / 3.0f) / 128.0f);
+    config->threshold_AmpRes_FF_e = 7;
+  } else if (bitRate > 48000) {
+    config->threshold_AmpRes_FF_m = FL2FXCONST_DBL(0);
+    config->threshold_AmpRes_FF_e = 0;
+  }
+
+  if (bitRate == 0) {
+    /* map vbr quality to bitrate */
+    if (vbrMode < 30)
+      bitRate = 24000;
+    else if (vbrMode < 40)
+      bitRate = 28000;
+    else if (vbrMode < 60)
+      bitRate = 32000;
+    else if (vbrMode < 75)
+      bitRate = 40000;
+    else
+      bitRate = 48000;
+    bitRate *= numChannels;
+    /* fix to enable mono vbrMode<40 @ 44.1 of 48kHz */
+    if (numChannels == 1) {
+      if (sampleRateSbr == 44100 || sampleRateSbr == 48000) {
+        if (vbrMode < 40) bitRate = 32000;
+      }
+    }
+  }
+
+  idx =
+      getSbrTuningTableIndex(bitRate, numChannels, sampleRateCore, core, NULL);
+
+  if (idx != INVALID_TABLE_IDX) {
+    config->startFreq = sbrTuningTable[idx].startFreq;
+    config->stopFreq = sbrTuningTable[idx].stopFreq;
+    if (useSpeechConfig) {
+      config->startFreq = sbrTuningTable[idx].startFreqSpeech;
+      config->stopFreq = sbrTuningTable[idx].stopFreqSpeech;
+    }
+
+    /* Adapt stop frequency in case of downsampled SBR - only 32 bands then */
+    if (1 == config->downSampleFactor) {
+      INT dsStopFreq = FDKsbrEnc_GetDownsampledStopFreq(
+          sampleRateCore, config->startFreq, config->stopFreq,
+          config->downSampleFactor);
+      if (dsStopFreq < 0) {
+        return 0;
+      }
+
+      config->stopFreq = dsStopFreq;
+    }
+
+    config->sbr_noise_bands = sbrTuningTable[idx].numNoiseBands;
+    if (core == AOT_ER_AAC_ELD) config->init_amp_res_FF = SBR_AMP_RES_1_5;
+    config->noiseFloorOffset = sbrTuningTable[idx].noiseFloorOffset;
+
+    config->ana_max_level = sbrTuningTable[idx].noiseMaxLevel;
+    config->stereoMode = sbrTuningTable[idx].stereoMode;
+    config->freqScale = sbrTuningTable[idx].freqScale;
+
+    if (numChannels == 1) {
+      /* stereo case */
+      switch (core) {
+        case AOT_AAC_LC:
+          if (bitRate <= (useSpeechConfig ? 24000U : 20000U)) {
+            config->freq_res_fixfix[0] = FREQ_RES_LOW; /* set low frequency
+                                                          resolution for
+                                                          non-split frames */
+            config->freq_res_fixfix[1] = FREQ_RES_LOW; /* set low frequency
+                                                          resolution for split
+                                                          frames */
+          }
+          break;
+        case AOT_ER_AAC_ELD:
+          if (bitRate < 36000)
+            config->freq_res_fixfix[1] = FREQ_RES_LOW; /* set low frequency
+                                                          resolution for split
+                                                          frames */
+          if (bitRate < 26000) {
+            config->freq_res_fixfix[0] = FREQ_RES_LOW; /* set low frequency
+                                                          resolution for
+                                                          non-split frames */
+            config->fResTransIsLow =
+                1; /* for transient frames, set low frequency resolution */
+          }
+          break;
+        default:
+          break;
+      }
+    } else {
+      /* stereo case */
+      switch (core) {
+        case AOT_AAC_LC:
+          if (bitRate <= 28000) {
+            config->freq_res_fixfix[0] = FREQ_RES_LOW; /* set low frequency
+                                                          resolution for
+                                                          non-split frames */
+            config->freq_res_fixfix[1] = FREQ_RES_LOW; /* set low frequency
+                                                          resolution for split
+                                                          frames */
+          }
+          break;
+        case AOT_ER_AAC_ELD:
+          if (bitRate < 72000) {
+            config->freq_res_fixfix[1] = FREQ_RES_LOW; /* set low frequency
+                                                          resolution for split
+                                                          frames */
+          }
+          if (bitRate < 52000) {
+            config->freq_res_fixfix[0] = FREQ_RES_LOW; /* set low frequency
+                                                          resolution for
+                                                          non-split frames */
+            config->fResTransIsLow =
+                1; /* for transient frames, set low frequency resolution */
+          }
+          break;
+        default:
+          break;
+      }
+      if (bitRate <= 28000) {
+        /*
+          additionally restrict frequency resolution in FIXFIX frames
+          to further reduce SBR payload size */
+        config->freq_res_fixfix[0] = FREQ_RES_LOW;
+        config->freq_res_fixfix[1] = FREQ_RES_LOW;
+      }
+    }
+
+    /* adjust usage of parametric coding dependent on bitrate and speech config
+     * flag */
+    if (useSpeechConfig) config->parametricCoding = 0;
+
+    if (core == AOT_ER_AAC_ELD) {
+      if (bitRate < 28000) config->init_amp_res_FF = SBR_AMP_RES_3_0;
+      config->SendHeaderDataTime = -1;
+    }
+
+    if (numChannels == 1) {
+      if (bitRate < 16000) {
+        config->parametricCoding = 0;
+      }
+    } else {
+      if (bitRate < 20000) {
+        config->parametricCoding = 0;
+      }
+    }
+
+    config->useSpeechConfig = useSpeechConfig;
+
+    /* PS settings */
+    config->bParametricStereo = bParametricStereo;
+
+    return 1;
+  } else {
+    return 0;
+  }
+}
+
+/*****************************************************************************
+
+ functionname: FDKsbrEnc_InitializeSbrDefaults
+ description:  initializes the SBR configuration
+ returns:      error status
+ input:        - core codec type,
+               - factor of SBR to core frame length,
+               - core frame length
+ output:       initialized SBR configuration
+
+*****************************************************************************/
+static UINT FDKsbrEnc_InitializeSbrDefaults(sbrConfigurationPtr config,
+                                            INT downSampleFactor,
+                                            UINT codecGranuleLen,
+                                            const INT isLowDelay) {
+  if ((downSampleFactor < 1 || downSampleFactor > 2) ||
+      (codecGranuleLen * downSampleFactor > 64 * 32))
+    return (0); /* error */
+
+  config->SendHeaderDataTime = 1000;
+  config->useWaveCoding = 0;
+  config->crcSbr = 0;
+  config->dynBwSupported = 1;
+  if (isLowDelay)
+    config->tran_thr = 6000;
+  else
+    config->tran_thr = 13000;
+
+  config->parametricCoding = 1;
+
+  config->sbrFrameSize = codecGranuleLen * downSampleFactor;
+  config->downSampleFactor = downSampleFactor;
+
+  /* sbr default parameters */
+  config->sbr_data_extra = 0;
+  config->amp_res = SBR_AMP_RES_3_0;
+  config->tran_fc = 0;
+  config->tran_det_mode = 1;
+  config->spread = 1;
+  config->stat = 0;
+  config->e = 1;
+  config->deltaTAcrossFrames = 1;
+  config->dF_edge_1stEnv = FL2FXCONST_DBL(0.3f);
+  config->dF_edge_incr = FL2FXCONST_DBL(0.3f);
+
+  config->sbr_invf_mode = INVF_SWITCHED;
+  config->sbr_xpos_mode = XPOS_LC;
+  config->sbr_xpos_ctrl = SBR_XPOS_CTRL_DEFAULT;
+  config->sbr_xpos_level = 0;
+  config->useSaPan = 0;
+  config->dynBwEnabled = 0;
+
+  /* the following parameters are overwritten by the
+     FDKsbrEnc_AdjustSbrSettings() function since they are included in the
+     tuning table */
+  config->stereoMode = SBR_SWITCH_LRC;
+  config->ana_max_level = 6;
+  config->noiseFloorOffset = 0;
+  config->startFreq = 5; /*  5.9 respectively  6.0 kHz at fs = 44.1/48 kHz */
+  config->stopFreq = 9;  /* 16.2 respectively 16.8 kHz at fs = 44.1/48 kHz */
+  config->freq_res_fixfix[0] = FREQ_RES_HIGH; /* non-split case */
+  config->freq_res_fixfix[1] = FREQ_RES_HIGH; /* split case */
+  config->fResTransIsLow = 0; /* for transient frames, set variable frequency
+                                 resolution according to freqResTable */
+
+  /* header_extra_1 */
+  config->freqScale = SBR_FREQ_SCALE_DEFAULT;
+  config->alterScale = SBR_ALTER_SCALE_DEFAULT;
+  config->sbr_noise_bands = SBR_NOISE_BANDS_DEFAULT;
+
+  /* header_extra_2 */
+  config->sbr_limiter_bands = SBR_LIMITER_BANDS_DEFAULT;
+  config->sbr_limiter_gains = SBR_LIMITER_GAINS_DEFAULT;
+  config->sbr_interpol_freq = SBR_INTERPOL_FREQ_DEFAULT;
+  config->sbr_smoothing_length = SBR_SMOOTHING_LENGTH_DEFAULT;
+
+  return 1;
+}
+
+/*****************************************************************************
+
+ functionname: DeleteEnvChannel
+ description:  frees memory of one SBR channel
+ returns:      -
+ input:        handle of channel
+ output:       released handle
+
+*****************************************************************************/
+static void deleteEnvChannel(HANDLE_ENV_CHANNEL hEnvCut) {
+  if (hEnvCut) {
+    FDKsbrEnc_DeleteTonCorrParamExtr(&hEnvCut->TonCorr);
+
+    FDKsbrEnc_deleteExtractSbrEnvelope(&hEnvCut->sbrExtractEnvelope);
+  }
+}
+
+/*****************************************************************************
+
+ functionname: sbrEncoder_ChannelClose
+ description:  close the channel coding handle
+ returns:
+ input:        phSbrChannel
+ output:
+
+*****************************************************************************/
+static void sbrEncoder_ChannelClose(HANDLE_SBR_CHANNEL hSbrChannel) {
+  if (hSbrChannel != NULL) {
+    deleteEnvChannel(&hSbrChannel->hEnvChannel);
+  }
+}
+
+/*****************************************************************************
+
+ functionname: sbrEncoder_ElementClose
+ description:  close the channel coding handle
+ returns:
+ input:        phSbrChannel
+ output:
+
+*****************************************************************************/
+static void sbrEncoder_ElementClose(HANDLE_SBR_ELEMENT *phSbrElement) {
+  HANDLE_SBR_ELEMENT hSbrElement = *phSbrElement;
+
+  if (hSbrElement != NULL) {
+    if (hSbrElement->sbrConfigData.v_k_master)
+      FreeRam_Sbr_v_k_master(&hSbrElement->sbrConfigData.v_k_master);
+    if (hSbrElement->sbrConfigData.freqBandTable[LO])
+      FreeRam_Sbr_freqBandTableLO(
+          &hSbrElement->sbrConfigData.freqBandTable[LO]);
+    if (hSbrElement->sbrConfigData.freqBandTable[HI])
+      FreeRam_Sbr_freqBandTableHI(
+          &hSbrElement->sbrConfigData.freqBandTable[HI]);
+
+    FreeRam_SbrElement(phSbrElement);
+  }
+  return;
+}
+
+void sbrEncoder_Close(HANDLE_SBR_ENCODER *phSbrEncoder) {
+  HANDLE_SBR_ENCODER hSbrEncoder = *phSbrEncoder;
+
+  if (hSbrEncoder != NULL) {
+    int el, ch;
+
+    for (el = 0; el < (8); el++) {
+      if (hSbrEncoder->sbrElement[el] != NULL) {
+        sbrEncoder_ElementClose(&hSbrEncoder->sbrElement[el]);
+      }
+    }
+
+    /* Close sbr Channels */
+    for (ch = 0; ch < (8); ch++) {
+      if (hSbrEncoder->pSbrChannel[ch]) {
+        sbrEncoder_ChannelClose(hSbrEncoder->pSbrChannel[ch]);
+        FreeRam_SbrChannel(&hSbrEncoder->pSbrChannel[ch]);
+      }
+
+      if (hSbrEncoder->QmfAnalysis[ch].FilterStates)
+        FreeRam_Sbr_QmfStatesAnalysis(
+            (FIXP_QAS **)&hSbrEncoder->QmfAnalysis[ch].FilterStates);
+    }
+
+    if (hSbrEncoder->hParametricStereo)
+      PSEnc_Destroy(&hSbrEncoder->hParametricStereo);
+    if (hSbrEncoder->qmfSynthesisPS.FilterStates)
+      FreeRam_PsQmfStatesSynthesis(
+          (FIXP_DBL **)&hSbrEncoder->qmfSynthesisPS.FilterStates);
+
+    /* Release Overlay */
+    if (hSbrEncoder->pSBRdynamic_RAM)
+      FreeRam_SbrDynamic_RAM((FIXP_DBL **)&hSbrEncoder->pSBRdynamic_RAM);
+
+    FreeRam_SbrEncoder(phSbrEncoder);
+  }
+}
+
+/*****************************************************************************
+
+ functionname: updateFreqBandTable
+ description:  updates vk_master
+ returns:      -
+ input:        config handle
+ output:       error info
+
+*****************************************************************************/
+static INT updateFreqBandTable(HANDLE_SBR_CONFIG_DATA sbrConfigData,
+                               HANDLE_SBR_HEADER_DATA sbrHeaderData,
+                               const INT downSampleFactor) {
+  INT k0, k2;
+
+  if (FDKsbrEnc_FindStartAndStopBand(
+          sbrConfigData->sampleFreq,
+          sbrConfigData->sampleFreq >> (downSampleFactor - 1),
+          sbrConfigData->noQmfBands, sbrHeaderData->sbr_start_frequency,
+          sbrHeaderData->sbr_stop_frequency, &k0, &k2))
+    return (1);
+
+  if (FDKsbrEnc_UpdateFreqScale(
+          sbrConfigData->v_k_master, &sbrConfigData->num_Master, k0, k2,
+          sbrHeaderData->freqScale, sbrHeaderData->alterScale))
+    return (1);
+
+  sbrHeaderData->sbr_xover_band = 0;
+
+  if (FDKsbrEnc_UpdateHiRes(sbrConfigData->freqBandTable[HI],
+                            &sbrConfigData->nSfb[HI], sbrConfigData->v_k_master,
+                            sbrConfigData->num_Master,
+                            &sbrHeaderData->sbr_xover_band))
+    return (1);
+
+  FDKsbrEnc_UpdateLoRes(
+      sbrConfigData->freqBandTable[LO], &sbrConfigData->nSfb[LO],
+      sbrConfigData->freqBandTable[HI], sbrConfigData->nSfb[HI]);
+
+  sbrConfigData->xOverFreq =
+      (sbrConfigData->freqBandTable[LOW_RES][0] * sbrConfigData->sampleFreq /
+           sbrConfigData->noQmfBands +
+       1) >>
+      1;
+
+  return (0);
+}
+
+/*****************************************************************************
+
+ functionname: resetEnvChannel
+ description:  resets parameters and allocates memory
+ returns:      error status
+ input:
+ output:       hEnv
+
+*****************************************************************************/
+static INT resetEnvChannel(HANDLE_SBR_CONFIG_DATA sbrConfigData,
+                           HANDLE_SBR_HEADER_DATA sbrHeaderData,
+                           HANDLE_ENV_CHANNEL hEnv) {
+  /* note !!! hEnv->encEnvData.noOfnoisebands will be updated later in function
+   * FDKsbrEnc_extractSbrEnvelope !!!*/
+  hEnv->TonCorr.sbrNoiseFloorEstimate.noiseBands =
+      sbrHeaderData->sbr_noise_bands;
+
+  if (FDKsbrEnc_ResetTonCorrParamExtr(
+          &hEnv->TonCorr, sbrConfigData->xposCtrlSwitch,
+          sbrConfigData->freqBandTable[HI][0], sbrConfigData->v_k_master,
+          sbrConfigData->num_Master, sbrConfigData->sampleFreq,
+          sbrConfigData->freqBandTable, sbrConfigData->nSfb,
+          sbrConfigData->noQmfBands))
+    return (1);
+
+  hEnv->sbrCodeNoiseFloor.nSfb[LO] =
+      hEnv->TonCorr.sbrNoiseFloorEstimate.noNoiseBands;
+  hEnv->sbrCodeNoiseFloor.nSfb[HI] =
+      hEnv->TonCorr.sbrNoiseFloorEstimate.noNoiseBands;
+
+  hEnv->sbrCodeEnvelope.nSfb[LO] = sbrConfigData->nSfb[LO];
+  hEnv->sbrCodeEnvelope.nSfb[HI] = sbrConfigData->nSfb[HI];
+
+  hEnv->encEnvData.noHarmonics = sbrConfigData->nSfb[HI];
+
+  hEnv->sbrCodeEnvelope.upDate = 0;
+  hEnv->sbrCodeNoiseFloor.upDate = 0;
+
+  return (0);
+}
+
+/* ****************************** FDKsbrEnc_SbrGetXOverFreq
+ * ******************************/
+/**
+ * @fn
+ * @brief       calculates the closest possible crossover frequency
+ * @return      the crossover frequency SBR accepts
+ *
+ */
+static INT FDKsbrEnc_SbrGetXOverFreq(
+    HANDLE_SBR_ELEMENT hEnv, /*!< handle to SBR encoder instance */
+    INT xoverFreq) /*!< from core coder suggested crossover frequency */
+{
+  INT band;
+  INT lastDiff, newDiff;
+  INT cutoffSb;
+
+  UCHAR *RESTRICT pVKMaster = hEnv->sbrConfigData.v_k_master;
+
+  /* Check if there is a matching cutoff frequency in the master table */
+  cutoffSb = (4 * xoverFreq * hEnv->sbrConfigData.noQmfBands /
+                  hEnv->sbrConfigData.sampleFreq +
+              1) >>
+             1;
+  lastDiff = cutoffSb;
+  for (band = 0; band < hEnv->sbrConfigData.num_Master; band++) {
+    newDiff = fixp_abs((INT)pVKMaster[band] - cutoffSb);
+
+    if (newDiff >= lastDiff) {
+      band--;
+      break;
+    }
+
+    lastDiff = newDiff;
+  }
+
+  return ((pVKMaster[band] * hEnv->sbrConfigData.sampleFreq /
+               hEnv->sbrConfigData.noQmfBands +
+           1) >>
+          1);
+}
+
+/*****************************************************************************
+
+ functionname: FDKsbrEnc_EnvEncodeFrame
+ description: performs the sbr envelope calculation for one element
+ returns:
+ input:
+ output:
+
+*****************************************************************************/
+INT FDKsbrEnc_EnvEncodeFrame(
+    HANDLE_SBR_ENCODER hEnvEncoder, int iElement,
+    INT_PCM *samples,    /*!< time samples, always deinterleaved */
+    UINT samplesBufSize, /*!< time buffer channel stride */
+    UINT *sbrDataBits,   /*!< Size of SBR payload  */
+    UCHAR *sbrData,      /*!< SBR payload  */
+    int clearOutput      /*!< Do not consider any input signal */
+) {
+  HANDLE_SBR_ELEMENT hSbrElement = NULL;
+  FDK_CRCINFO crcInfo;
+  INT crcReg;
+  INT ch;
+  INT band;
+  INT cutoffSb;
+  INT newXOver;
+
+  if (hEnvEncoder == NULL) return -1;
+
+  hSbrElement = hEnvEncoder->sbrElement[iElement];
+
+  if (hSbrElement == NULL) return -1;
+
+  /* header bitstream handling */
+  HANDLE_SBR_BITSTREAM_DATA sbrBitstreamData = &hSbrElement->sbrBitstreamData;
+
+  INT psHeaderActive = 0;
+  sbrBitstreamData->HeaderActive = 0;
+
+  /* Anticipate PS header because of internal PS bitstream delay in order to be
+   * in sync with SBR header. */
+  if (sbrBitstreamData->CountSendHeaderData ==
+      (sbrBitstreamData->NrSendHeaderData - 1)) {
+    psHeaderActive = 1;
+  }
+
+  /* Signal SBR header to be written into bitstream */
+  if (sbrBitstreamData->CountSendHeaderData == 0) {
+    sbrBitstreamData->HeaderActive = 1;
+  }
+
+  /* Increment header interval counter */
+  if (sbrBitstreamData->NrSendHeaderData == 0) {
+    sbrBitstreamData->CountSendHeaderData = 1;
+  } else {
+    if (sbrBitstreamData->CountSendHeaderData >= 0) {
+      sbrBitstreamData->CountSendHeaderData++;
+      sbrBitstreamData->CountSendHeaderData %=
+          sbrBitstreamData->NrSendHeaderData;
+    }
+  }
+
+  if (hSbrElement->CmonData.dynBwEnabled) {
+    INT i;
+    for (i = 4; i > 0; i--)
+      hSbrElement->dynXOverFreqDelay[i] = hSbrElement->dynXOverFreqDelay[i - 1];
+
+    hSbrElement->dynXOverFreqDelay[0] = hSbrElement->CmonData.dynXOverFreqEnc;
+    if (hSbrElement->dynXOverFreqDelay[1] > hSbrElement->dynXOverFreqDelay[2])
+      newXOver = hSbrElement->dynXOverFreqDelay[2];
+    else
+      newXOver = hSbrElement->dynXOverFreqDelay[1];
+
+    /* has the crossover frequency changed? */
+    if (hSbrElement->sbrConfigData.dynXOverFreq != newXOver) {
+      /* get corresponding master band */
+      cutoffSb = ((4 * newXOver * hSbrElement->sbrConfigData.noQmfBands /
+                   hSbrElement->sbrConfigData.sampleFreq) +
+                  1) >>
+                 1;
+
+      for (band = 0; band < hSbrElement->sbrConfigData.num_Master; band++) {
+        if (cutoffSb == hSbrElement->sbrConfigData.v_k_master[band]) break;
+      }
+      FDK_ASSERT(band < hSbrElement->sbrConfigData.num_Master);
+
+      hSbrElement->sbrConfigData.dynXOverFreq = newXOver;
+      hSbrElement->sbrHeaderData.sbr_xover_band = band;
+      hSbrElement->sbrBitstreamData.HeaderActive = 1;
+      psHeaderActive = 1; /* ps header is one frame delayed */
+
+      /*
+        update vk_master table
+      */
+      if (updateFreqBandTable(&hSbrElement->sbrConfigData,
+                              &hSbrElement->sbrHeaderData,
+                              hEnvEncoder->downSampleFactor))
+        return (1);
+
+      /* reset SBR channels */
+      INT nEnvCh = hSbrElement->sbrConfigData.nChannels;
+      for (ch = 0; ch < nEnvCh; ch++) {
+        if (resetEnvChannel(&hSbrElement->sbrConfigData,
+                            &hSbrElement->sbrHeaderData,
+                            &hSbrElement->sbrChannel[ch]->hEnvChannel))
+          return (1);
+      }
+    }
+  }
+
+  /*
+    allocate space for dummy header and crc
+  */
+  crcReg = FDKsbrEnc_InitSbrBitstream(
+      &hSbrElement->CmonData,
+      hSbrElement->payloadDelayLine[hEnvEncoder->nBitstrDelay],
+      MAX_PAYLOAD_SIZE * sizeof(UCHAR), &crcInfo,
+      hSbrElement->sbrConfigData.sbrSyntaxFlags);
+
+  /* Temporal Envelope Data */
+  SBR_FRAME_TEMP_DATA _fData;
+  SBR_FRAME_TEMP_DATA *fData = &_fData;
+  SBR_ENV_TEMP_DATA eData[MAX_NUM_CHANNELS];
+
+  /* Init Temporal Envelope Data */
+  {
+    int i;
+
+    FDKmemclear(&eData[0], sizeof(SBR_ENV_TEMP_DATA));
+    FDKmemclear(&eData[1], sizeof(SBR_ENV_TEMP_DATA));
+    FDKmemclear(fData, sizeof(SBR_FRAME_TEMP_DATA));
+
+    for (i = 0; i < MAX_NUM_NOISE_VALUES; i++) fData->res[i] = FREQ_RES_HIGH;
+  }
+
+  if (!clearOutput) {
+    /*
+     * Transform audio data into QMF domain
+     */
+    for (ch = 0; ch < hSbrElement->sbrConfigData.nChannels; ch++) {
+      HANDLE_ENV_CHANNEL h_envChan = &hSbrElement->sbrChannel[ch]->hEnvChannel;
+      HANDLE_SBR_EXTRACT_ENVELOPE sbrExtrEnv = &h_envChan->sbrExtractEnvelope;
+
+      if (hSbrElement->elInfo.fParametricStereo == 0) {
+        QMF_SCALE_FACTOR tmpScale;
+        FIXP_DBL **pQmfReal, **pQmfImag;
+        C_AALLOC_SCRATCH_START(qmfWorkBuffer, FIXP_DBL, 64 * 2)
+
+        /* Obtain pointers to QMF buffers. */
+        pQmfReal = sbrExtrEnv->rBuffer;
+        pQmfImag = sbrExtrEnv->iBuffer;
+
+        qmfAnalysisFiltering(
+            hSbrElement->hQmfAnalysis[ch], pQmfReal, pQmfImag, &tmpScale,
+            samples + hSbrElement->elInfo.ChannelIndex[ch] * samplesBufSize, 0,
+            1, qmfWorkBuffer);
+
+        h_envChan->qmfScale = tmpScale.lb_scale + 7;
+
+        C_AALLOC_SCRATCH_END(qmfWorkBuffer, FIXP_DBL, 64 * 2)
+
+      } /* fParametricStereo == 0 */
+
+      /*
+        Parametric Stereo processing
+      */
+      if (hSbrElement->elInfo.fParametricStereo) {
+        INT error = noError;
+
+        /* Limit Parametric Stereo to one instance */
+        FDK_ASSERT(ch == 0);
+
+        if (error == noError) {
+          /* parametric stereo processing:
+             - input:
+               o left and right time domain samples
+             - processing:
+               o stereo qmf analysis
+               o stereo hybrid analysis
+               o ps parameter extraction
+               o downmix + hybrid synthesis
+             - output:
+               o downmixed qmf data is written to sbrExtrEnv->rBuffer and
+             sbrExtrEnv->iBuffer
+          */
+          SCHAR qmfScale;
+          INT_PCM *pSamples[2] = {
+              samples + hSbrElement->elInfo.ChannelIndex[0] * samplesBufSize,
+              samples + hSbrElement->elInfo.ChannelIndex[1] * samplesBufSize};
+          error = FDKsbrEnc_PSEnc_ParametricStereoProcessing(
+              hEnvEncoder->hParametricStereo, pSamples, samplesBufSize,
+              hSbrElement->hQmfAnalysis, sbrExtrEnv->rBuffer,
+              sbrExtrEnv->iBuffer,
+              samples + hSbrElement->elInfo.ChannelIndex[ch] * samplesBufSize,
+              &hEnvEncoder->qmfSynthesisPS, &qmfScale, psHeaderActive);
+          h_envChan->qmfScale = (int)qmfScale;
+        }
+
+      } /* if (hEnvEncoder->hParametricStereo) */
+
+      /*
+
+         Extract Envelope relevant things from QMF data
+
+      */
+      FDKsbrEnc_extractSbrEnvelope1(&hSbrElement->sbrConfigData,
+                                    &hSbrElement->sbrHeaderData,
+                                    &hSbrElement->sbrBitstreamData, h_envChan,
+                                    &hSbrElement->CmonData, &eData[ch], fData);
+
+    } /* hEnvEncoder->sbrConfigData.nChannels */
+  }
+
+  /*
+     Process Envelope relevant things and calculate envelope data and write
+     payload
+  */
+  FDKsbrEnc_extractSbrEnvelope2(
+      &hSbrElement->sbrConfigData, &hSbrElement->sbrHeaderData,
+      (hSbrElement->elInfo.fParametricStereo) ? hEnvEncoder->hParametricStereo
+                                              : NULL,
+      &hSbrElement->sbrBitstreamData, &hSbrElement->sbrChannel[0]->hEnvChannel,
+      (hSbrElement->sbrConfigData.stereoMode != SBR_MONO)
+          ? &hSbrElement->sbrChannel[1]->hEnvChannel
+          : NULL,
+      &hSbrElement->CmonData, eData, fData, clearOutput);
+
+  hSbrElement->sbrBitstreamData.rightBorderFIX = 0;
+
+  /*
+    format payload, calculate crc
+  */
+  FDKsbrEnc_AssembleSbrBitstream(&hSbrElement->CmonData, &crcInfo, crcReg,
+                                 hSbrElement->sbrConfigData.sbrSyntaxFlags);
+
+  /*
+    save new payload, set to zero length if greater than MAX_PAYLOAD_SIZE
+  */
+  hSbrElement->payloadDelayLineSize[hEnvEncoder->nBitstrDelay] =
+      FDKgetValidBits(&hSbrElement->CmonData.sbrBitbuf);
+
+  if (hSbrElement->payloadDelayLineSize[hEnvEncoder->nBitstrDelay] >
+      (MAX_PAYLOAD_SIZE << 3))
+    hSbrElement->payloadDelayLineSize[hEnvEncoder->nBitstrDelay] = 0;
+
+  /* While filling the Delay lines, sbrData is NULL */
+  if (sbrData) {
+    *sbrDataBits = hSbrElement->payloadDelayLineSize[0];
+    FDKmemcpy(sbrData, hSbrElement->payloadDelayLine[0],
+              (hSbrElement->payloadDelayLineSize[0] + 7) >> 3);
+  }
+
+  /* delay header active flag */
+  if (hSbrElement->sbrBitstreamData.HeaderActive == 1) {
+    hSbrElement->sbrBitstreamData.HeaderActiveDelay =
+        1 + hEnvEncoder->nBitstrDelay;
+  } else {
+    if (hSbrElement->sbrBitstreamData.HeaderActiveDelay > 0) {
+      hSbrElement->sbrBitstreamData.HeaderActiveDelay--;
+    }
+  }
+
+  return (0);
+}
+
+/*****************************************************************************
+
+ functionname: FDKsbrEnc_Downsample
+ description: performs downsampling and delay compensation of the core path
+ returns:
+ input:
+ output:
+
+*****************************************************************************/
+INT FDKsbrEnc_Downsample(
+    HANDLE_SBR_ENCODER hSbrEncoder,
+    INT_PCM *samples,    /*!< time samples, always deinterleaved */
+    UINT samplesBufSize, /*!< time buffer size per channel */
+    UINT numChannels,    /*!< number of channels */
+    UINT *sbrDataBits,   /*!< Size of SBR payload  */
+    UCHAR *sbrData,      /*!< SBR payload  */
+    int clearOutput      /*!< Do not consider any input signal */
+) {
+  HANDLE_SBR_ELEMENT hSbrElement = NULL;
+  INT nOutSamples;
+  int el;
+  if (hSbrEncoder->downSampleFactor > 1) {
+    /* Do downsampling */
+
+    /* Loop over elements (LFE is handled later) */
+    for (el = 0; el < hSbrEncoder->noElements; el++) {
+      hSbrElement = hSbrEncoder->sbrElement[el];
+      if (hSbrEncoder->sbrElement[el] != NULL) {
+        if (hSbrEncoder->downsamplingMethod == SBRENC_DS_TIME) {
+          int ch;
+          int nChannels = hSbrElement->sbrConfigData.nChannels;
+
+          for (ch = 0; ch < nChannels; ch++) {
+            FDKaacEnc_Downsample(
+                &hSbrElement->sbrChannel[ch]->downSampler,
+                samples +
+                    hSbrElement->elInfo.ChannelIndex[ch] * samplesBufSize +
+                    hSbrEncoder->bufferOffset / numChannels,
+                hSbrElement->sbrConfigData.frameSize,
+                samples + hSbrElement->elInfo.ChannelIndex[ch] * samplesBufSize,
+                &nOutSamples);
+          }
+        }
+      }
+    }
+
+    /* Handle LFE (if existing) */
+    if (hSbrEncoder->lfeChIdx != -1) { /* lfe downsampler */
+      FDKaacEnc_Downsample(&hSbrEncoder->lfeDownSampler,
+                           samples + hSbrEncoder->lfeChIdx * samplesBufSize +
+                               hSbrEncoder->bufferOffset / numChannels,
+                           hSbrEncoder->frameSize,
+                           samples + hSbrEncoder->lfeChIdx * samplesBufSize,
+                           &nOutSamples);
+    }
+  } else {
+    /* No downsampling. Still, some buffer shifting for correct delay */
+    int samples2Copy = hSbrEncoder->frameSize;
+    if (hSbrEncoder->bufferOffset / (int)numChannels < samples2Copy) {
+      for (int c = 0; c < (int)numChannels; c++) {
+        /* Do memmove while taking care of overlapping memory areas. (memcpy
+           does not necessarily take care) Distinguish between oeverlapping and
+           non overlapping version due to reasons of complexity. */
+        FDKmemmove(samples + c * samplesBufSize,
+                   samples + c * samplesBufSize +
+                       hSbrEncoder->bufferOffset / numChannels,
+                   samples2Copy * sizeof(INT_PCM));
+      }
+    } else {
+      for (int c = 0; c < (int)numChannels; c++) {
+        /* Simple memcpy since the memory areas are not overlapping */
+        FDKmemcpy(samples + c * samplesBufSize,
+                  samples + c * samplesBufSize +
+                      hSbrEncoder->bufferOffset / numChannels,
+                  samples2Copy * sizeof(INT_PCM));
+      }
+    }
+  }
+
+  return 0;
+}
+
+/*****************************************************************************
+
+ functionname: createEnvChannel
+ description:  initializes parameters and allocates memory
+ returns:      error status
+ input:
+ output:       hEnv
+
+*****************************************************************************/
+
+static INT createEnvChannel(HANDLE_ENV_CHANNEL hEnv, INT channel,
+                            UCHAR *dynamic_RAM) {
+  FDKmemclear(hEnv, sizeof(struct ENV_CHANNEL));
+
+  if (FDKsbrEnc_CreateTonCorrParamExtr(&hEnv->TonCorr, channel)) {
+    return (1);
+  }
+
+  if (FDKsbrEnc_CreateExtractSbrEnvelope(&hEnv->sbrExtractEnvelope, channel,
+                                         /*chan*/ 0, dynamic_RAM)) {
+    return (1);
+  }
+
+  return 0;
+}
+
+/*****************************************************************************
+
+ functionname: initEnvChannel
+ description:  initializes parameters
+ returns:      error status
+ input:
+ output:
+
+*****************************************************************************/
+static INT initEnvChannel(HANDLE_SBR_CONFIG_DATA sbrConfigData,
+                          HANDLE_SBR_HEADER_DATA sbrHeaderData,
+                          HANDLE_ENV_CHANNEL hEnv, sbrConfigurationPtr params,
+                          ULONG statesInitFlag, INT chanInEl,
+                          UCHAR *dynamic_RAM) {
+  int frameShift, tran_off = 0;
+  INT e;
+  INT tran_fc;
+  INT timeSlots, timeStep, startIndex;
+  INT noiseBands[2] = {3, 3};
+
+  e = 1 << params->e;
+
+  FDK_ASSERT(params->e >= 0);
+
+  hEnv->encEnvData.freq_res_fixfix[0] = params->freq_res_fixfix[0];
+  hEnv->encEnvData.freq_res_fixfix[1] = params->freq_res_fixfix[1];
+  hEnv->encEnvData.fResTransIsLow = params->fResTransIsLow;
+
+  hEnv->fLevelProtect = 0;
+
+  hEnv->encEnvData.ldGrid =
+      (sbrConfigData->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) ? 1 : 0;
+
+  hEnv->encEnvData.sbr_xpos_mode = (XPOS_MODE)params->sbr_xpos_mode;
+
+  if (hEnv->encEnvData.sbr_xpos_mode == XPOS_SWITCHED) {
+    /*
+       no other type than XPOS_MDCT or XPOS_SPEECH allowed,
+       but enable switching
+    */
+    sbrConfigData->switchTransposers = TRUE;
+    hEnv->encEnvData.sbr_xpos_mode = XPOS_MDCT;
+  } else {
+    sbrConfigData->switchTransposers = FALSE;
+  }
+
+  hEnv->encEnvData.sbr_xpos_ctrl = params->sbr_xpos_ctrl;
+
+  /* extended data */
+  if (params->parametricCoding) {
+    hEnv->encEnvData.extended_data = 1;
+  } else {
+    hEnv->encEnvData.extended_data = 0;
+  }
+
+  hEnv->encEnvData.extension_size = 0;
+
+  startIndex = QMF_FILTER_PROTOTYPE_SIZE - sbrConfigData->noQmfBands;
+
+  switch (params->sbrFrameSize) {
+    case 2304:
+      timeSlots = 18;
+      break;
+    case 2048:
+    case 1024:
+    case 512:
+      timeSlots = 16;
+      break;
+    case 1920:
+    case 960:
+    case 480:
+      timeSlots = 15;
+      break;
+    case 1152:
+      timeSlots = 9;
+      break;
+    default:
+      return (1); /* Illegal frame size */
+  }
+
+  timeStep = sbrConfigData->noQmfSlots / timeSlots;
+
+  if (FDKsbrEnc_InitTonCorrParamExtr(
+          params->sbrFrameSize, &hEnv->TonCorr, sbrConfigData, timeSlots,
+          params->sbr_xpos_ctrl, params->ana_max_level,
+          sbrHeaderData->sbr_noise_bands, params->noiseFloorOffset,
+          params->useSpeechConfig))
+    return (1);
+
+  hEnv->encEnvData.noOfnoisebands =
+      hEnv->TonCorr.sbrNoiseFloorEstimate.noNoiseBands;
+
+  noiseBands[0] = hEnv->encEnvData.noOfnoisebands;
+  noiseBands[1] = hEnv->encEnvData.noOfnoisebands;
+
+  hEnv->encEnvData.sbr_invf_mode = (INVF_MODE)params->sbr_invf_mode;
+
+  if (hEnv->encEnvData.sbr_invf_mode == INVF_SWITCHED) {
+    hEnv->encEnvData.sbr_invf_mode = INVF_MID_LEVEL;
+    hEnv->TonCorr.switchInverseFilt = TRUE;
+  } else {
+    hEnv->TonCorr.switchInverseFilt = FALSE;
+  }
+
+  tran_fc = params->tran_fc;
+
+  if (tran_fc == 0) {
+    tran_fc = fixMin(
+        5000, FDKsbrEnc_getSbrStartFreqRAW(sbrHeaderData->sbr_start_frequency,
+                                           params->codecSettings.sampleFreq));
+  }
+
+  tran_fc =
+      (tran_fc * 4 * sbrConfigData->noQmfBands / sbrConfigData->sampleFreq +
+       1) >>
+      1;
+
+  if (sbrConfigData->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) {
+    frameShift = LD_PRETRAN_OFF;
+    tran_off = LD_PRETRAN_OFF + FRAME_MIDDLE_SLOT_512LD * timeStep;
+  } else {
+    frameShift = 0;
+    switch (timeSlots) {
+      /* The factor of 2 is by definition. */
+      case NUMBER_TIME_SLOTS_2048:
+        tran_off = 8 + FRAME_MIDDLE_SLOT_2048 * timeStep;
+        break;
+      case NUMBER_TIME_SLOTS_1920:
+        tran_off = 7 + FRAME_MIDDLE_SLOT_1920 * timeStep;
+        break;
+      default:
+        return 1;
+    }
+  }
+  if (FDKsbrEnc_InitExtractSbrEnvelope(
+          &hEnv->sbrExtractEnvelope, sbrConfigData->noQmfSlots,
+          sbrConfigData->noQmfBands, startIndex, timeSlots, timeStep, tran_off,
+          statesInitFlag, chanInEl, dynamic_RAM, sbrConfigData->sbrSyntaxFlags))
+    return (1);
+
+  if (FDKsbrEnc_InitSbrCodeEnvelope(&hEnv->sbrCodeEnvelope, sbrConfigData->nSfb,
+                                    params->deltaTAcrossFrames,
+                                    params->dF_edge_1stEnv,
+                                    params->dF_edge_incr))
+    return (1);
+
+  if (FDKsbrEnc_InitSbrCodeEnvelope(&hEnv->sbrCodeNoiseFloor, noiseBands,
+                                    params->deltaTAcrossFrames, 0, 0))
+    return (1);
+
+  sbrConfigData->initAmpResFF = params->init_amp_res_FF;
+
+  if (FDKsbrEnc_InitSbrHuffmanTables(&hEnv->encEnvData, &hEnv->sbrCodeEnvelope,
+                                     &hEnv->sbrCodeNoiseFloor,
+                                     sbrHeaderData->sbr_amp_res))
+    return (1);
+
+  FDKsbrEnc_initFrameInfoGenerator(
+      &hEnv->SbrEnvFrame, params->spread, e, params->stat, timeSlots,
+      hEnv->encEnvData.freq_res_fixfix, hEnv->encEnvData.fResTransIsLow,
+      hEnv->encEnvData.ldGrid);
+
+  if (sbrConfigData->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY)
+
+  {
+    INT bandwidth_qmf_slot =
+        (sbrConfigData->sampleFreq >> 1) / (sbrConfigData->noQmfBands);
+    if (FDKsbrEnc_InitSbrFastTransientDetector(
+            &hEnv->sbrFastTransientDetector, sbrConfigData->noQmfSlots,
+            bandwidth_qmf_slot, sbrConfigData->noQmfBands,
+            sbrConfigData->freqBandTable[0][0]))
+      return (1);
+  }
+
+  /* The transient detector has to be initialized also if the fast transient
+     detector was active, because the values from the transient detector
+     structure are used. */
+  if (FDKsbrEnc_InitSbrTransientDetector(
+          &hEnv->sbrTransientDetector, sbrConfigData->sbrSyntaxFlags,
+          sbrConfigData->frameSize, sbrConfigData->sampleFreq, params, tran_fc,
+          sbrConfigData->noQmfSlots, sbrConfigData->noQmfBands,
+          hEnv->sbrExtractEnvelope.YBufferWriteOffset,
+          hEnv->sbrExtractEnvelope.YBufferSzShift, frameShift, tran_off))
+    return (1);
+
+  sbrConfigData->xposCtrlSwitch = params->sbr_xpos_ctrl;
+
+  hEnv->encEnvData.noHarmonics = sbrConfigData->nSfb[HI];
+  hEnv->encEnvData.addHarmonicFlag = 0;
+
+  return (0);
+}
+
+INT sbrEncoder_Open(HANDLE_SBR_ENCODER *phSbrEncoder, INT nElements,
+                    INT nChannels, INT supportPS) {
+  INT i;
+  INT errorStatus = 1;
+  HANDLE_SBR_ENCODER hSbrEncoder = NULL;
+
+  if (phSbrEncoder == NULL) {
+    goto bail;
+  }
+
+  hSbrEncoder = GetRam_SbrEncoder();
+  if (hSbrEncoder == NULL) {
+    goto bail;
+  }
+  FDKmemclear(hSbrEncoder, sizeof(SBR_ENCODER));
+
+  if (NULL ==
+      (hSbrEncoder->pSBRdynamic_RAM = (UCHAR *)GetRam_SbrDynamic_RAM())) {
+    goto bail;
+  }
+  hSbrEncoder->dynamicRam = hSbrEncoder->pSBRdynamic_RAM;
+
+  /* Create SBR elements */
+  for (i = 0; i < nElements; i++) {
+    hSbrEncoder->sbrElement[i] = GetRam_SbrElement(i);
+    if (hSbrEncoder->sbrElement[i] == NULL) {
+      goto bail;
+    }
+    FDKmemclear(hSbrEncoder->sbrElement[i], sizeof(SBR_ELEMENT));
+    hSbrEncoder->sbrElement[i]->sbrConfigData.freqBandTable[LO] =
+        GetRam_Sbr_freqBandTableLO(i);
+    hSbrEncoder->sbrElement[i]->sbrConfigData.freqBandTable[HI] =
+        GetRam_Sbr_freqBandTableHI(i);
+    hSbrEncoder->sbrElement[i]->sbrConfigData.v_k_master =
+        GetRam_Sbr_v_k_master(i);
+    if ((hSbrEncoder->sbrElement[i]->sbrConfigData.freqBandTable[LO] == NULL) ||
+        (hSbrEncoder->sbrElement[i]->sbrConfigData.freqBandTable[HI] == NULL) ||
+        (hSbrEncoder->sbrElement[i]->sbrConfigData.v_k_master == NULL)) {
+      goto bail;
+    }
+  }
+
+  /* Create SBR channels */
+  for (i = 0; i < nChannels; i++) {
+    hSbrEncoder->pSbrChannel[i] = GetRam_SbrChannel(i);
+    if (hSbrEncoder->pSbrChannel[i] == NULL) {
+      goto bail;
+    }
+
+    if (createEnvChannel(&hSbrEncoder->pSbrChannel[i]->hEnvChannel, i,
+                         hSbrEncoder->dynamicRam)) {
+      goto bail;
+    }
+  }
+
+  /* Create QMF States */
+  for (i = 0; i < fixMax(nChannels, (supportPS) ? 2 : 0); i++) {
+    hSbrEncoder->QmfAnalysis[i].FilterStates = GetRam_Sbr_QmfStatesAnalysis(i);
+    if (hSbrEncoder->QmfAnalysis[i].FilterStates == NULL) {
+      goto bail;
+    }
+  }
+
+  /* Create Parametric Stereo handle */
+  if (supportPS) {
+    if (PSEnc_Create(&hSbrEncoder->hParametricStereo)) {
+      goto bail;
+    }
+
+    hSbrEncoder->qmfSynthesisPS.FilterStates = GetRam_PsQmfStatesSynthesis();
+    if (hSbrEncoder->qmfSynthesisPS.FilterStates == NULL) {
+      goto bail;
+    }
+  } /* supportPS */
+
+  *phSbrEncoder = hSbrEncoder;
+
+  errorStatus = 0;
+  return errorStatus;
+
+bail:
+  /* Close SBR encoder instance */
+  sbrEncoder_Close(&hSbrEncoder);
+  return errorStatus;
+}
+
+static INT FDKsbrEnc_Reallocate(HANDLE_SBR_ENCODER hSbrEncoder,
+                                SBR_ELEMENT_INFO elInfo[(8)],
+                                const INT noElements) {
+  INT totalCh = 0;
+  INT totalQmf = 0;
+  INT coreEl;
+  INT el = -1;
+
+  hSbrEncoder->lfeChIdx = -1; /* default value, until lfe found */
+
+  for (coreEl = 0; coreEl < noElements; coreEl++) {
+    /* SBR only handles SCE and CPE's */
+    if (elInfo[coreEl].elType == ID_SCE || elInfo[coreEl].elType == ID_CPE) {
+      el++;
+    } else {
+      if (elInfo[coreEl].elType == ID_LFE) {
+        hSbrEncoder->lfeChIdx = elInfo[coreEl].ChannelIndex[0];
+      }
+      continue;
+    }
+
+    SBR_ELEMENT_INFO *pelInfo = &elInfo[coreEl];
+    HANDLE_SBR_ELEMENT hSbrElement = hSbrEncoder->sbrElement[el];
+
+    int ch;
+    for (ch = 0; ch < pelInfo->nChannelsInEl; ch++) {
+      hSbrElement->sbrChannel[ch] = hSbrEncoder->pSbrChannel[totalCh];
+      totalCh++;
+    }
+    /* analysis QMF */
+    for (ch = 0;
+         ch < ((pelInfo->fParametricStereo) ? 2 : pelInfo->nChannelsInEl);
+         ch++) {
+      hSbrElement->elInfo.ChannelIndex[ch] = pelInfo->ChannelIndex[ch];
+      hSbrElement->hQmfAnalysis[ch] = &hSbrEncoder->QmfAnalysis[totalQmf++];
+    }
+
+    /* Copy Element info */
+    hSbrElement->elInfo.elType = pelInfo->elType;
+    hSbrElement->elInfo.instanceTag = pelInfo->instanceTag;
+    hSbrElement->elInfo.nChannelsInEl = pelInfo->nChannelsInEl;
+    hSbrElement->elInfo.fParametricStereo = pelInfo->fParametricStereo;
+    hSbrElement->elInfo.fDualMono = pelInfo->fDualMono;
+  } /* coreEl */
+
+  return 0;
+}
+
+/*****************************************************************************
+
+ functionname: FDKsbrEnc_bsBufInit
+ description:  initializes bitstream buffer
+ returns:      initialized bitstream buffer in env encoder
+ input:
+ output:       hEnv
+
+*****************************************************************************/
+static INT FDKsbrEnc_bsBufInit(HANDLE_SBR_ELEMENT hSbrElement,
+                               int nBitstrDelay) {
+  UCHAR *bitstreamBuffer;
+
+  /* initialize the bitstream buffer */
+  bitstreamBuffer = hSbrElement->payloadDelayLine[nBitstrDelay];
+  FDKinitBitStream(&hSbrElement->CmonData.sbrBitbuf, bitstreamBuffer,
+                   MAX_PAYLOAD_SIZE * sizeof(UCHAR), 0, BS_WRITER);
+
+  return (0);
+}
+
+/*****************************************************************************
+
+ functionname: FDKsbrEnc_EnvInit
+ description:  initializes parameters
+ returns:      error status
+ input:
+ output:       hEnv
+
+*****************************************************************************/
+static INT FDKsbrEnc_EnvInit(HANDLE_SBR_ELEMENT hSbrElement,
+                             sbrConfigurationPtr params, INT *coreBandWith,
+                             AUDIO_OBJECT_TYPE aot, int nElement,
+                             const int headerPeriod, ULONG statesInitFlag,
+                             const SBRENC_DS_TYPE downsamplingMethod,
+                             UCHAR *dynamic_RAM) {
+  int ch, i;
+
+  if ((params->codecSettings.nChannels < 1) ||
+      (params->codecSettings.nChannels > MAX_NUM_CHANNELS)) {
+    return (1);
+  }
+
+  /* init and set syntax flags */
+  hSbrElement->sbrConfigData.sbrSyntaxFlags = 0;
+
+  switch (aot) {
+    case AOT_ER_AAC_ELD:
+      hSbrElement->sbrConfigData.sbrSyntaxFlags |= SBR_SYNTAX_LOW_DELAY;
+      break;
+    default:
+      break;
+  }
+  if (params->crcSbr) {
+    hSbrElement->sbrConfigData.sbrSyntaxFlags |= SBR_SYNTAX_CRC;
+  }
+
+  hSbrElement->sbrConfigData.noQmfBands = 64 >> (2 - params->downSampleFactor);
+  switch (hSbrElement->sbrConfigData.noQmfBands) {
+    case 64:
+      hSbrElement->sbrConfigData.noQmfSlots = params->sbrFrameSize >> 6;
+      break;
+    case 32:
+      hSbrElement->sbrConfigData.noQmfSlots = params->sbrFrameSize >> 5;
+      break;
+    default:
+      hSbrElement->sbrConfigData.noQmfSlots = params->sbrFrameSize >> 6;
+      return (2);
+  }
+
+  /*
+    now initialize sbrConfigData, sbrHeaderData and sbrBitstreamData,
+  */
+  hSbrElement->sbrConfigData.nChannels = params->codecSettings.nChannels;
+
+  if (params->codecSettings.nChannels == 2) {
+    if ((hSbrElement->elInfo.elType == ID_CPE) &&
+        ((hSbrElement->elInfo.fDualMono == 1))) {
+      hSbrElement->sbrConfigData.stereoMode = SBR_LEFT_RIGHT;
+    } else {
+      hSbrElement->sbrConfigData.stereoMode = params->stereoMode;
+    }
+  } else {
+    hSbrElement->sbrConfigData.stereoMode = SBR_MONO;
+  }
+
+  hSbrElement->sbrConfigData.frameSize = params->sbrFrameSize;
+
+  hSbrElement->sbrConfigData.sampleFreq =
+      params->downSampleFactor * params->codecSettings.sampleFreq;
+
+  hSbrElement->sbrBitstreamData.CountSendHeaderData = 0;
+  if (params->SendHeaderDataTime > 0) {
+    if (headerPeriod == -1) {
+      hSbrElement->sbrBitstreamData.NrSendHeaderData = (INT)(
+          params->SendHeaderDataTime * hSbrElement->sbrConfigData.sampleFreq /
+          (1000 * hSbrElement->sbrConfigData.frameSize));
+      hSbrElement->sbrBitstreamData.NrSendHeaderData =
+          fixMax(hSbrElement->sbrBitstreamData.NrSendHeaderData, 1);
+    } else {
+      /* assure header period at least once per second */
+      hSbrElement->sbrBitstreamData.NrSendHeaderData = fixMin(
+          fixMax(headerPeriod, 1), (hSbrElement->sbrConfigData.sampleFreq /
+                                    hSbrElement->sbrConfigData.frameSize));
+    }
+  } else {
+    hSbrElement->sbrBitstreamData.NrSendHeaderData = 0;
+  }
+
+  hSbrElement->sbrHeaderData.sbr_data_extra = params->sbr_data_extra;
+  hSbrElement->sbrBitstreamData.HeaderActive = 0;
+  hSbrElement->sbrBitstreamData.rightBorderFIX = 0;
+  hSbrElement->sbrHeaderData.sbr_start_frequency = params->startFreq;
+  hSbrElement->sbrHeaderData.sbr_stop_frequency = params->stopFreq;
+  hSbrElement->sbrHeaderData.sbr_xover_band = 0;
+  hSbrElement->sbrHeaderData.sbr_lc_stereo_mode = 0;
+
+  /* data_extra */
+  if (params->sbr_xpos_ctrl != SBR_XPOS_CTRL_DEFAULT)
+    hSbrElement->sbrHeaderData.sbr_data_extra = 1;
+
+  hSbrElement->sbrHeaderData.sbr_amp_res = (AMP_RES)params->amp_res;
+
+  /* header_extra_1 */
+  hSbrElement->sbrHeaderData.freqScale = params->freqScale;
+  hSbrElement->sbrHeaderData.alterScale = params->alterScale;
+  hSbrElement->sbrHeaderData.sbr_noise_bands = params->sbr_noise_bands;
+  hSbrElement->sbrHeaderData.header_extra_1 = 0;
+
+  if ((params->freqScale != SBR_FREQ_SCALE_DEFAULT) ||
+      (params->alterScale != SBR_ALTER_SCALE_DEFAULT) ||
+      (params->sbr_noise_bands != SBR_NOISE_BANDS_DEFAULT)) {
+    hSbrElement->sbrHeaderData.header_extra_1 = 1;
+  }
+
+  /* header_extra_2 */
+  hSbrElement->sbrHeaderData.sbr_limiter_bands = params->sbr_limiter_bands;
+  hSbrElement->sbrHeaderData.sbr_limiter_gains = params->sbr_limiter_gains;
+
+  if ((hSbrElement->sbrConfigData.sampleFreq > 48000) &&
+      (hSbrElement->sbrHeaderData.sbr_start_frequency >= 9)) {
+    hSbrElement->sbrHeaderData.sbr_limiter_gains = SBR_LIMITER_GAINS_INFINITE;
+  }
+
+  hSbrElement->sbrHeaderData.sbr_interpol_freq = params->sbr_interpol_freq;
+  hSbrElement->sbrHeaderData.sbr_smoothing_length =
+      params->sbr_smoothing_length;
+  hSbrElement->sbrHeaderData.header_extra_2 = 0;
+
+  if ((params->sbr_limiter_bands != SBR_LIMITER_BANDS_DEFAULT) ||
+      (params->sbr_limiter_gains != SBR_LIMITER_GAINS_DEFAULT) ||
+      (params->sbr_interpol_freq != SBR_INTERPOL_FREQ_DEFAULT) ||
+      (params->sbr_smoothing_length != SBR_SMOOTHING_LENGTH_DEFAULT)) {
+    hSbrElement->sbrHeaderData.header_extra_2 = 1;
+  }
+
+  /* other switches */
+  hSbrElement->sbrConfigData.useWaveCoding = params->useWaveCoding;
+  hSbrElement->sbrConfigData.useParametricCoding = params->parametricCoding;
+  hSbrElement->sbrConfigData.thresholdAmpResFF_m =
+      params->threshold_AmpRes_FF_m;
+  hSbrElement->sbrConfigData.thresholdAmpResFF_e =
+      params->threshold_AmpRes_FF_e;
+
+  /* init freq band table */
+  if (updateFreqBandTable(&hSbrElement->sbrConfigData,
+                          &hSbrElement->sbrHeaderData,
+                          params->downSampleFactor)) {
+    return (1);
+  }
+
+  /* now create envelope ext and QMF for each available channel */
+  for (ch = 0; ch < hSbrElement->sbrConfigData.nChannels; ch++) {
+    if (initEnvChannel(&hSbrElement->sbrConfigData, &hSbrElement->sbrHeaderData,
+                       &hSbrElement->sbrChannel[ch]->hEnvChannel, params,
+                       statesInitFlag, ch, dynamic_RAM)) {
+      return (1);
+    }
+
+  } /* nChannels */
+
+  /* reset and intialize analysis qmf */
+  for (ch = 0; ch < ((hSbrElement->elInfo.fParametricStereo)
+                         ? 2
+                         : hSbrElement->sbrConfigData.nChannels);
+       ch++) {
+    int err;
+    UINT qmfFlags =
+        (hSbrElement->sbrConfigData.sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY)
+            ? QMF_FLAG_CLDFB
+            : 0;
+    if (statesInitFlag)
+      qmfFlags &= ~QMF_FLAG_KEEP_STATES;
+    else
+      qmfFlags |= QMF_FLAG_KEEP_STATES;
+
+    err = qmfInitAnalysisFilterBank(
+        hSbrElement->hQmfAnalysis[ch],
+        (FIXP_QAS *)hSbrElement->hQmfAnalysis[ch]->FilterStates,
+        hSbrElement->sbrConfigData.noQmfSlots,
+        hSbrElement->sbrConfigData.noQmfBands,
+        hSbrElement->sbrConfigData.noQmfBands,
+        hSbrElement->sbrConfigData.noQmfBands, qmfFlags);
+    if (0 != err) {
+      return err;
+    }
+  }
+
+  /*  */
+  hSbrElement->CmonData.xOverFreq = hSbrElement->sbrConfigData.xOverFreq;
+  hSbrElement->CmonData.dynBwEnabled =
+      (params->dynBwSupported && params->dynBwEnabled);
+  hSbrElement->CmonData.dynXOverFreqEnc =
+      FDKsbrEnc_SbrGetXOverFreq(hSbrElement, hSbrElement->CmonData.xOverFreq);
+  for (i = 0; i < 5; i++)
+    hSbrElement->dynXOverFreqDelay[i] = hSbrElement->CmonData.dynXOverFreqEnc;
+  hSbrElement->CmonData.sbrNumChannels = hSbrElement->sbrConfigData.nChannels;
+  hSbrElement->sbrConfigData.dynXOverFreq = hSbrElement->CmonData.xOverFreq;
+
+  /* Update Bandwith to be passed to the core encoder */
+  *coreBandWith = hSbrElement->CmonData.xOverFreq;
+
+  return (0);
+}
+
+INT sbrEncoder_GetInBufferSize(int noChannels) {
+  INT temp;
+
+  temp = (2048);
+  temp += 1024 + MAX_SAMPLE_DELAY;
+  temp *= noChannels;
+  temp *= sizeof(INT_PCM);
+  return temp;
+}
+
+/*
+ * Encode Dummy SBR payload frames to fill the delay lines.
+ */
+static INT FDKsbrEnc_DelayCompensation(HANDLE_SBR_ENCODER hEnvEnc,
+                                       INT_PCM *timeBuffer,
+                                       UINT timeBufferBufSize) {
+  int n, el;
+
+  for (n = hEnvEnc->nBitstrDelay; n > 0; n--) {
+    for (el = 0; el < hEnvEnc->noElements; el++) {
+      if (FDKsbrEnc_EnvEncodeFrame(
+              hEnvEnc, el,
+              timeBuffer + hEnvEnc->downsampledOffset / hEnvEnc->nChannels,
+              timeBufferBufSize, NULL, NULL, 1))
+        return -1;
+    }
+    sbrEncoder_UpdateBuffers(hEnvEnc, timeBuffer, timeBufferBufSize);
+  }
+  return 0;
+}
+
+UINT sbrEncoder_LimitBitRate(UINT bitRate, UINT numChannels,
+                             UINT coreSampleRate, AUDIO_OBJECT_TYPE aot) {
+  UINT newBitRate = bitRate;
+  INT index;
+
+  FDK_ASSERT(numChannels > 0 && numChannels <= 2);
+  if (aot == AOT_PS) {
+    if (numChannels == 1) {
+      index = getPsTuningTableIndex(bitRate, &newBitRate);
+      if (index == INVALID_TABLE_IDX) {
+        bitRate = newBitRate;
+      }
+    } else {
+      return 0;
+    }
+  }
+  index = getSbrTuningTableIndex(bitRate, numChannels, coreSampleRate, aot,
+                                 &newBitRate);
+  if (index != INVALID_TABLE_IDX) {
+    newBitRate = bitRate;
+  }
+
+  return newBitRate;
+}
+
+UINT sbrEncoder_IsSingleRatePossible(AUDIO_OBJECT_TYPE aot) {
+  UINT isPossible = (AOT_PS == aot) ? 0 : 1;
+  return isPossible;
+}
+
+/*****************************************************************************/
+/*                                                                           */
+/*functionname: sbrEncoder_Init_delay                                        */
+/*description:  Determine Delay balancing and new encoder delay              */
+/*                                                                           */
+/*returns:      - error status                                               */
+/*input:        - frame length of the core (i.e. e.g. AAC)                   */
+/*              - number of channels                                         */
+/*              - downsample factor (1 for downsampled, 2 for dual-rate SBR) */
+/*              - low delay presence                                         */
+/*              - ps presence                                                */
+/*              - downsampling method: QMF-, time domain or no downsampling  */
+/*              - various delay values (see DELAY_PARAM struct description)  */
+/*                                                                           */
+/*Example: Delay balancing for a HE-AACv1 encoder (time-domain downsampling) */
+/*========================================================================== */
+/*                                                                           */
+/*    +--------+            +--------+ +--------+ +--------+ +--------+      */
+/*    |core    |            |ds 2:1  | |AAC     | |QMF     | |QMF     |      */
+/*  +-+path    +------------+        +-+core    +-+analysis+-+overlap +-+    */
+/*  | |offset  |            |        | |        | |32 bands| |        | |    */
+/*  | +--------+            +--------+ +--------+ +--------+ +--------+ |    */
+/*  |                              core path                    +-------++   */
+/*  |                                                           |QMF     |   */
+/*->+                                                           +synth.  +-> */
+/*  |                                                           |64 bands|   */
+/*  |                                                           +-------++   */
+/*  | +--------+ +--------+ +--------+ +--------+                       |    */
+/*  | |SBR path| |QMF     | |subband | |bs delay|                       |    */
+/*  +-+offset  +-+analysis+-+sample  +-+(full   +-----------------------+    */
+/*    |        | |64 bands| |buffer  | | frames)|                            */
+/*    +--------+ +--------+ +--------+ +--------+                            */
+/*                                 SBR path                                  */
+/*                                                                           */
+/*****************************************************************************/
+static INT sbrEncoder_Init_delay(
+    const int coreFrameLength,               /* input */
+    const int numChannels,                   /* input */
+    const int downSampleFactor,              /* input */
+    const int lowDelay,                      /* input */
+    const int usePs,                         /* input */
+    const int is212,                         /* input */
+    const SBRENC_DS_TYPE downsamplingMethod, /* input */
+    DELAY_PARAM *hDelayParam                 /* input/output */
+) {
+  int delayCorePath = 0;   /* delay in core path */
+  int delaySbrPath = 0;    /* delay difference in QMF aka SBR path */
+  int delayInput2Core = 0; /* delay from the input to the core */
+  int delaySbrDec = 0;     /* delay of the decoder's SBR module */
+
+  int delayCore = hDelayParam->delay; /* delay of the core */
+
+  /* Added delay by the SBR delay initialization */
+  int corePathOffset = 0; /* core path */
+  int sbrPathOffset = 0;  /* sbr path */
+  int bitstreamDelay = 0; /* sbr path, framewise */
+
+  int flCore = coreFrameLength; /* core frame length */
+
+  int returnValue = 0; /* return value - 0 means: no error */
+
+  /* 1) Calculate actual delay  for core and SBR path */
+  if (is212) {
+    delayCorePath = DELAY_COREPATH_ELDv2SBR(flCore, downSampleFactor);
+    delaySbrPath = DELAY_ELDv2SBR(flCore, downSampleFactor);
+    delaySbrDec = ((flCore) / 2) * (downSampleFactor);
+  } else if (lowDelay) {
+    delayCorePath = DELAY_COREPATH_ELDSBR(flCore, downSampleFactor);
+    delaySbrPath = DELAY_ELDSBR(flCore, downSampleFactor);
+    delaySbrDec = DELAY_QMF_POSTPROC(downSampleFactor);
+  } else if (usePs) {
+    delayCorePath = DELAY_COREPATH_PS(flCore, downSampleFactor);
+    delaySbrPath = DELAY_PS(flCore, downSampleFactor);
+    delaySbrDec = DELAY_COREPATH_SBR(flCore, downSampleFactor);
+  } else {
+    delayCorePath = DELAY_COREPATH_SBR(flCore, downSampleFactor);
+    delaySbrPath = DELAY_SBR(flCore, downSampleFactor);
+    delaySbrDec = DELAY_COREPATH_SBR(flCore, downSampleFactor);
+  }
+  delayCorePath += delayCore * downSampleFactor;
+  delayCorePath +=
+      (downsamplingMethod == SBRENC_DS_TIME) ? hDelayParam->dsDelay : 0;
+
+  /* 2) Manage coupling of paths */
+  if (downsamplingMethod == SBRENC_DS_QMF && delayCorePath > delaySbrPath) {
+    /* In case of QMF downsampling, both paths are coupled, i.e. the SBR path
+       offset would be added to both the SBR path and to the core path
+       as well, thus making it impossible to achieve delay balancing.
+       To overcome that problem, a framewise delay is added to the SBR path
+       first, until the overall delay of the core path is shorter than
+       the delay of the SBR path. When this is achieved, the missing delay
+       difference can be added as downsampled offset to the core path.
+    */
+    while (delayCorePath > delaySbrPath) {
+      /* Add one frame delay to SBR path */
+      delaySbrPath += flCore * downSampleFactor;
+      bitstreamDelay += 1;
+    }
+  }
+
+  /* 3) Calculate necessary additional delay to balance the paths */
+  if (delayCorePath > delaySbrPath) {
+    /* Delay QMF input */
+    while (delayCorePath > delaySbrPath + (int)flCore * (int)downSampleFactor) {
+      /* Do bitstream frame-wise delay balancing if there are
+         more than SBR framelength samples delay difference */
+      delaySbrPath += flCore * downSampleFactor;
+      bitstreamDelay += 1;
+    }
+    /* Multiply input offset by input channels */
+    corePathOffset = 0;
+    sbrPathOffset = (delayCorePath - delaySbrPath) * numChannels;
+  } else {
+    /* Delay AAC data */
+    /* Multiply downsampled offset by AAC core channels. Divide by 2 because of
+       half samplerate of downsampled data. */
+    corePathOffset = ((delaySbrPath - delayCorePath) * numChannels) >>
+                     (downSampleFactor - 1);
+    sbrPathOffset = 0;
+  }
+
+  /* 4) Calculate delay from input to core */
+  if (usePs) {
+    delayInput2Core =
+        (DELAY_QMF_ANA(downSampleFactor) + DELAY_QMF_DS + DELAY_HYB_SYN) +
+        (downSampleFactor * corePathOffset) + 1;
+  } else if (downsamplingMethod == SBRENC_DS_TIME) {
+    delayInput2Core = corePathOffset + hDelayParam->dsDelay;
+  } else {
+    delayInput2Core = corePathOffset;
+  }
+
+  /* 6) Set output parameters */
+  hDelayParam->delay = FDKmax(delayCorePath, delaySbrPath); /* overall delay */
+  hDelayParam->sbrDecDelay = delaySbrDec;         /* SBR decoder delay */
+  hDelayParam->delayInput2Core = delayInput2Core; /* delay input - core */
+  hDelayParam->bitstrDelay = bitstreamDelay;    /* bitstream delay, in frames */
+  hDelayParam->corePathOffset = corePathOffset; /* offset added to core path */
+  hDelayParam->sbrPathOffset = sbrPathOffset;   /* offset added to SBR path */
+
+  return returnValue;
+}
+
+/*****************************************************************************
+
+ functionname: sbrEncoder_Init
+ description:  initializes the SBR encoder
+ returns:      error status
+
+*****************************************************************************/
+INT sbrEncoder_Init(HANDLE_SBR_ENCODER hSbrEncoder,
+                    SBR_ELEMENT_INFO elInfo[(8)], int noElements,
+                    INT_PCM *inputBuffer, UINT inputBufferBufSize,
+                    INT *coreBandwidth, INT *inputBufferOffset,
+                    INT *numChannels, const UINT syntaxFlags,
+                    INT *coreSampleRate, UINT *downSampleFactor,
+                    INT *frameLength, AUDIO_OBJECT_TYPE aot, int *delay,
+                    int transformFactor, const int headerPeriod,
+                    ULONG statesInitFlag) {
+  HANDLE_ERROR_INFO errorInfo = noError;
+  sbrConfiguration sbrConfig[(8)];
+  INT error = 0;
+  INT lowestBandwidth;
+  /* Save input parameters */
+  INT inputSampleRate = *coreSampleRate;
+  int coreFrameLength = *frameLength;
+  int inputBandWidth = *coreBandwidth;
+  int inputChannels = *numChannels;
+
+  SBRENC_DS_TYPE downsamplingMethod = SBRENC_DS_NONE;
+  int highestSbrStartFreq, highestSbrStopFreq;
+  int lowDelay = 0;
+  int usePs = 0;
+  int is212 = 0;
+
+  DELAY_PARAM delayParam;
+
+  /* check whether SBR setting is available for the current encoder
+   * configuration (bitrate, samplerate) */
+  if (!sbrEncoder_IsSingleRatePossible(aot)) {
+    *downSampleFactor = 2;
+  }
+
+  if (aot == AOT_PS || aot == AOT_DABPLUS_PS) {
+    usePs = 1;
+  }
+  if (aot == AOT_ER_AAC_ELD) {
+    lowDelay = 1;
+  } else if (aot == AOT_ER_AAC_LD) {
+    error = 1;
+    goto bail;
+  }
+
+  /* Parametric Stereo */
+  if (usePs) {
+    if (*numChannels == 2 && noElements == 1) {
+      /* Override Element type in case of Parametric stereo */
+      elInfo[0].elType = ID_SCE;
+      elInfo[0].fParametricStereo = 1;
+      elInfo[0].nChannelsInEl = 1;
+      /* core encoder gets downmixed mono signal */
+      *numChannels = 1;
+    } else {
+      error = 1;
+      goto bail;
+    }
+  } /* usePs */
+
+  /* set the core's sample rate */
+  switch (*downSampleFactor) {
+    case 1:
+      *coreSampleRate = inputSampleRate;
+      downsamplingMethod = SBRENC_DS_NONE;
+      break;
+    case 2:
+      *coreSampleRate = inputSampleRate >> 1;
+      downsamplingMethod = usePs ? SBRENC_DS_QMF : SBRENC_DS_TIME;
+      break;
+    default:
+      *coreSampleRate = inputSampleRate >> 1;
+      return 0; /* return error */
+  }
+
+  /* check whether SBR setting is available for the current encoder
+   * configuration (bitrate, coreSampleRate) */
+  {
+    int el, coreEl;
+
+    /* Check if every element config is feasible */
+    for (coreEl = 0; coreEl < noElements; coreEl++) {
+      /* SBR only handles SCE and CPE's */
+      if (elInfo[coreEl].elType != ID_SCE && elInfo[coreEl].elType != ID_CPE) {
+        continue;
+      }
+      /* check if desired configuration is available */
+      if (!FDKsbrEnc_IsSbrSettingAvail(elInfo[coreEl].bitRate, 0,
+                                       elInfo[coreEl].nChannelsInEl,
+                                       inputSampleRate, *coreSampleRate, aot)) {
+        error = 1;
+        goto bail;
+      }
+    }
+
+    hSbrEncoder->nChannels = *numChannels;
+    hSbrEncoder->frameSize = coreFrameLength * *downSampleFactor;
+    hSbrEncoder->downsamplingMethod = downsamplingMethod;
+    hSbrEncoder->downSampleFactor = *downSampleFactor;
+    hSbrEncoder->estimateBitrate = 0;
+    hSbrEncoder->inputDataDelay = 0;
+    is212 = ((aot == AOT_ER_AAC_ELD) && (syntaxFlags & AC_LD_MPS)) ? 1 : 0;
+
+    /* Open SBR elements */
+    el = -1;
+    highestSbrStartFreq = highestSbrStopFreq = 0;
+    lowestBandwidth = 99999;
+
+    /* Loop through each core encoder element and get a matching SBR element
+     * config */
+    for (coreEl = 0; coreEl < noElements; coreEl++) {
+      /* SBR only handles SCE and CPE's */
+      if (elInfo[coreEl].elType == ID_SCE || elInfo[coreEl].elType == ID_CPE) {
+        el++;
+      } else {
+        continue;
+      }
+
+      /* Set parametric Stereo Flag. */
+      if (usePs) {
+        elInfo[coreEl].fParametricStereo = 1;
+      } else {
+        elInfo[coreEl].fParametricStereo = 0;
+      }
+
+      /*
+       * Init sbrConfig structure
+       */
+      if (!FDKsbrEnc_InitializeSbrDefaults(&sbrConfig[el], *downSampleFactor,
+                                           coreFrameLength, IS_LOWDELAY(aot))) {
+        error = 1;
+        goto bail;
+      }
+
+      /*
+       * Modify sbrConfig structure according to Element parameters
+       */
+      if (!FDKsbrEnc_AdjustSbrSettings(
+              &sbrConfig[el], elInfo[coreEl].bitRate,
+              elInfo[coreEl].nChannelsInEl, *coreSampleRate, inputSampleRate,
+              transformFactor, 24000, 0, 0, /* useSpeechConfig */
+              0,                            /* lcsMode */
+              usePs,                        /* bParametricStereo */
+              aot)) {
+        error = 1;
+        goto bail;
+      }
+
+      /* Find common frequency border for all SBR elements */
+      highestSbrStartFreq =
+          fixMax(highestSbrStartFreq, sbrConfig[el].startFreq);
+      highestSbrStopFreq = fixMax(highestSbrStopFreq, sbrConfig[el].stopFreq);
+
+    } /* first element loop */
+
+    /* Set element count (can be less than core encoder element count) */
+    hSbrEncoder->noElements = el + 1;
+
+    FDKsbrEnc_Reallocate(hSbrEncoder, elInfo, noElements);
+
+    for (el = 0; el < hSbrEncoder->noElements; el++) {
+      int bandwidth = *coreBandwidth;
+
+      /* Use lowest common bandwidth */
+      sbrConfig[el].startFreq = highestSbrStartFreq;
+      sbrConfig[el].stopFreq = highestSbrStopFreq;
+
+      /* initialize SBR element, and get core bandwidth */
+      error = FDKsbrEnc_EnvInit(hSbrEncoder->sbrElement[el], &sbrConfig[el],
+                                &bandwidth, aot, el, headerPeriod,
+                                statesInitFlag, hSbrEncoder->downsamplingMethod,
+                                hSbrEncoder->dynamicRam);
+
+      if (error != 0) {
+        error = 2;
+        goto bail;
+      }
+
+      /* Get lowest core encoder bandwidth to be returned later. */
+      lowestBandwidth = fixMin(lowestBandwidth, bandwidth);
+
+    } /* second element loop */
+
+    /* Initialize a downsampler for each channel in each SBR element */
+    if (hSbrEncoder->downsamplingMethod == SBRENC_DS_TIME) {
+      for (el = 0; el < hSbrEncoder->noElements; el++) {
+        HANDLE_SBR_ELEMENT hSbrEl = hSbrEncoder->sbrElement[el];
+        INT Wc, ch;
+
+        Wc = 500; /* Cutoff frequency with full bandwidth */
+
+        for (ch = 0; ch < hSbrEl->elInfo.nChannelsInEl; ch++) {
+          FDKaacEnc_InitDownsampler(&hSbrEl->sbrChannel[ch]->downSampler, Wc,
+                                    *downSampleFactor);
+          FDK_ASSERT(hSbrEl->sbrChannel[ch]->downSampler.delay <=
+                     MAX_DS_FILTER_DELAY);
+        }
+      } /* third element loop */
+
+      /* lfe */
+      FDKaacEnc_InitDownsampler(&hSbrEncoder->lfeDownSampler, 0,
+                                *downSampleFactor);
+    }
+
+    /* Get delay information */
+    delayParam.dsDelay =
+        hSbrEncoder->sbrElement[0]->sbrChannel[0]->downSampler.delay;
+    delayParam.delay = *delay;
+
+    error = sbrEncoder_Init_delay(coreFrameLength, *numChannels,
+                                  *downSampleFactor, lowDelay, usePs, is212,
+                                  downsamplingMethod, &delayParam);
+
+    if (error != 0) {
+      error = 3;
+      goto bail;
+    }
+
+    hSbrEncoder->nBitstrDelay = delayParam.bitstrDelay;
+    hSbrEncoder->sbrDecDelay = delayParam.sbrDecDelay;
+    hSbrEncoder->inputDataDelay = delayParam.delayInput2Core;
+
+    /* Assign core encoder Bandwidth */
+    *coreBandwidth = lowestBandwidth;
+
+    /* Estimate sbr bitrate, 2.5 kBit/s per sbr channel */
+    hSbrEncoder->estimateBitrate += 2500 * (*numChannels);
+
+    /* Initialize bitstream buffer for each element */
+    for (el = 0; el < hSbrEncoder->noElements; el++) {
+      FDKsbrEnc_bsBufInit(hSbrEncoder->sbrElement[el], delayParam.bitstrDelay);
+    }
+
+    /* initialize parametric stereo */
+    if (usePs) {
+      PSENC_CONFIG psEncConfig;
+      FDK_ASSERT(hSbrEncoder->noElements == 1);
+      INT psTuningTableIdx = getPsTuningTableIndex(elInfo[0].bitRate, NULL);
+
+      psEncConfig.frameSize = coreFrameLength;  // sbrConfig.sbrFrameSize;
+      psEncConfig.qmfFilterMode = 0;
+      psEncConfig.sbrPsDelay = 0;
+
+      /* tuning parameters */
+      if (psTuningTableIdx != INVALID_TABLE_IDX) {
+        psEncConfig.nStereoBands = psTuningTable[psTuningTableIdx].nStereoBands;
+        psEncConfig.maxEnvelopes = psTuningTable[psTuningTableIdx].nEnvelopes;
+        psEncConfig.iidQuantErrorThreshold =
+            (FIXP_DBL)psTuningTable[psTuningTableIdx].iidQuantErrorThreshold;
+
+        /* calculation is not quite linear, increased number of envelopes causes
+         * more bits */
+        /* assume avg. 50 bits per frame for 10 stereo bands / 1 envelope
+         * configuration */
+        hSbrEncoder->estimateBitrate +=
+            ((((*coreSampleRate) * 5 * psEncConfig.nStereoBands *
+               psEncConfig.maxEnvelopes) /
+              hSbrEncoder->frameSize));
+
+      } else {
+        error = ERROR(CDI, "Invalid ps tuning table index.");
+        goto bail;
+      }
+
+      qmfInitSynthesisFilterBank(
+          &hSbrEncoder->qmfSynthesisPS,
+          (FIXP_DBL *)hSbrEncoder->qmfSynthesisPS.FilterStates,
+          hSbrEncoder->sbrElement[0]->sbrConfigData.noQmfSlots,
+          hSbrEncoder->sbrElement[0]->sbrConfigData.noQmfBands >> 1,
+          hSbrEncoder->sbrElement[0]->sbrConfigData.noQmfBands >> 1,
+          hSbrEncoder->sbrElement[0]->sbrConfigData.noQmfBands >> 1,
+          (statesInitFlag) ? 0 : QMF_FLAG_KEEP_STATES);
+
+      if (errorInfo == noError) {
+        /* update delay */
+        psEncConfig.sbrPsDelay =
+            FDKsbrEnc_GetEnvEstDelay(&hSbrEncoder->sbrElement[0]
+                                          ->sbrChannel[0]
+                                          ->hEnvChannel.sbrExtractEnvelope);
+
+        errorInfo =
+            PSEnc_Init(hSbrEncoder->hParametricStereo, &psEncConfig,
+                       hSbrEncoder->sbrElement[0]->sbrConfigData.noQmfSlots,
+                       hSbrEncoder->sbrElement[0]->sbrConfigData.noQmfBands,
+                       hSbrEncoder->dynamicRam);
+      }
+    }
+
+    hSbrEncoder->downsampledOffset = delayParam.corePathOffset;
+    hSbrEncoder->bufferOffset = delayParam.sbrPathOffset;
+    *delay = delayParam.delay;
+
+    { hSbrEncoder->downmixSize = coreFrameLength * (*numChannels); }
+
+    /* Delay Compensation: fill bitstream delay buffer with zero input signal */
+    if (hSbrEncoder->nBitstrDelay > 0) {
+      error = FDKsbrEnc_DelayCompensation(hSbrEncoder, inputBuffer,
+                                          inputBufferBufSize);
+      if (error != 0) goto bail;
+    }
+
+    /* Set Output frame length */
+    *frameLength = coreFrameLength * *downSampleFactor;
+    /* Input buffer offset */
+    *inputBufferOffset =
+        fixMax(delayParam.sbrPathOffset, delayParam.corePathOffset);
+  }
+
+  return error;
+
+bail:
+  /* Restore input settings */
+  *coreSampleRate = inputSampleRate;
+  *frameLength = coreFrameLength;
+  *numChannels = inputChannels;
+  *coreBandwidth = inputBandWidth;
+
+  return error;
+}
+
+INT sbrEncoder_EncodeFrame(HANDLE_SBR_ENCODER hSbrEncoder, INT_PCM *samples,
+                           UINT samplesBufSize, UINT sbrDataBits[(8)],
+                           UCHAR sbrData[(8)][MAX_PAYLOAD_SIZE]) {
+  INT error;
+  int el;
+
+  for (el = 0; el < hSbrEncoder->noElements; el++) {
+    if (hSbrEncoder->sbrElement[el] != NULL) {
+      error = FDKsbrEnc_EnvEncodeFrame(
+          hSbrEncoder, el,
+          samples + hSbrEncoder->downsampledOffset / hSbrEncoder->nChannels,
+          samplesBufSize, &sbrDataBits[el], sbrData[el], 0);
+      if (error) return error;
+    }
+  }
+
+  error = FDKsbrEnc_Downsample(
+      hSbrEncoder,
+      samples + hSbrEncoder->downsampledOffset / hSbrEncoder->nChannels,
+      samplesBufSize, hSbrEncoder->nChannels, &sbrDataBits[el], sbrData[el], 0);
+  if (error) return error;
+
+  return 0;
+}
+
+INT sbrEncoder_UpdateBuffers(HANDLE_SBR_ENCODER hSbrEncoder,
+                             INT_PCM *timeBuffer, UINT timeBufferBufSize) {
+  if (hSbrEncoder->downsampledOffset > 0) {
+    int c;
+    int nd = hSbrEncoder->downmixSize / hSbrEncoder->nChannels;
+
+    for (c = 0; c < hSbrEncoder->nChannels; c++) {
+      /* Move delayed downsampled data */
+      FDKmemcpy(timeBuffer + timeBufferBufSize * c,
+                timeBuffer + timeBufferBufSize * c + nd,
+                sizeof(INT_PCM) *
+                    (hSbrEncoder->downsampledOffset / hSbrEncoder->nChannels));
+    }
+  } else {
+    int c;
+
+    for (c = 0; c < hSbrEncoder->nChannels; c++) {
+      /* Move delayed input data */
+      FDKmemcpy(
+          timeBuffer + timeBufferBufSize * c,
+          timeBuffer + timeBufferBufSize * c + hSbrEncoder->frameSize,
+          sizeof(INT_PCM) * hSbrEncoder->bufferOffset / hSbrEncoder->nChannels);
+    }
+  }
+  if (hSbrEncoder->nBitstrDelay > 0) {
+    int el;
+
+    for (el = 0; el < hSbrEncoder->noElements; el++) {
+      FDKmemmove(
+          hSbrEncoder->sbrElement[el]->payloadDelayLine[0],
+          hSbrEncoder->sbrElement[el]->payloadDelayLine[1],
+          sizeof(UCHAR) * (hSbrEncoder->nBitstrDelay * MAX_PAYLOAD_SIZE));
+
+      FDKmemmove(&hSbrEncoder->sbrElement[el]->payloadDelayLineSize[0],
+                 &hSbrEncoder->sbrElement[el]->payloadDelayLineSize[1],
+                 sizeof(UINT) * (hSbrEncoder->nBitstrDelay));
+    }
+  }
+  return 0;
+}
+
+INT sbrEncoder_SendHeader(HANDLE_SBR_ENCODER hSbrEncoder) {
+  INT error = -1;
+  if (hSbrEncoder) {
+    int el;
+    for (el = 0; el < hSbrEncoder->noElements; el++) {
+      if ((hSbrEncoder->noElements == 1) &&
+          (hSbrEncoder->sbrElement[0]->elInfo.fParametricStereo == 1)) {
+        hSbrEncoder->sbrElement[el]->sbrBitstreamData.CountSendHeaderData =
+            hSbrEncoder->sbrElement[el]->sbrBitstreamData.NrSendHeaderData - 1;
+      } else {
+        hSbrEncoder->sbrElement[el]->sbrBitstreamData.CountSendHeaderData = 0;
+      }
+    }
+    error = 0;
+  }
+  return error;
+}
+
+INT sbrEncoder_ContainsHeader(HANDLE_SBR_ENCODER hSbrEncoder) {
+  INT sbrHeader = 1;
+  if (hSbrEncoder) {
+    int el;
+    for (el = 0; el < hSbrEncoder->noElements; el++) {
+      sbrHeader &=
+          (hSbrEncoder->sbrElement[el]->sbrBitstreamData.HeaderActiveDelay == 1)
+              ? 1
+              : 0;
+    }
+  }
+  return sbrHeader;
+}
+
+INT sbrEncoder_GetHeaderDelay(HANDLE_SBR_ENCODER hSbrEncoder) {
+  INT delay = -1;
+
+  if (hSbrEncoder) {
+    if ((hSbrEncoder->noElements == 1) &&
+        (hSbrEncoder->sbrElement[0]->elInfo.fParametricStereo == 1)) {
+      delay = hSbrEncoder->nBitstrDelay + 1;
+    } else {
+      delay = hSbrEncoder->nBitstrDelay;
+    }
+  }
+  return delay;
+}
+INT sbrEncoder_GetBsDelay(HANDLE_SBR_ENCODER hSbrEncoder) {
+  INT delay = -1;
+
+  if (hSbrEncoder) {
+    delay = hSbrEncoder->nBitstrDelay;
+  }
+  return delay;
+}
+
+INT sbrEncoder_SAPPrepare(HANDLE_SBR_ENCODER hSbrEncoder) {
+  INT error = -1;
+  if (hSbrEncoder) {
+    int el;
+    for (el = 0; el < hSbrEncoder->noElements; el++) {
+      hSbrEncoder->sbrElement[el]->sbrBitstreamData.rightBorderFIX = 1;
+    }
+    error = 0;
+  }
+  return error;
+}
+
+INT sbrEncoder_GetEstimateBitrate(HANDLE_SBR_ENCODER hSbrEncoder) {
+  INT estimateBitrate = 0;
+
+  if (hSbrEncoder) {
+    estimateBitrate += hSbrEncoder->estimateBitrate;
+  }
+
+  return estimateBitrate;
+}
+
+INT sbrEncoder_GetInputDataDelay(HANDLE_SBR_ENCODER hSbrEncoder) {
+  INT delay = -1;
+
+  if (hSbrEncoder) {
+    delay = hSbrEncoder->inputDataDelay;
+  }
+  return delay;
+}
+
+INT sbrEncoder_GetSbrDecDelay(HANDLE_SBR_ENCODER hSbrEncoder) {
+  INT delay = -1;
+
+  if (hSbrEncoder) {
+    delay = hSbrEncoder->sbrDecDelay;
+  }
+  return delay;
+}
+
+INT sbrEncoder_GetLibInfo(LIB_INFO *info) {
+  int i;
+
+  if (info == NULL) {
+    return -1;
+  }
+  /* search for next free tab */
+  for (i = 0; i < FDK_MODULE_LAST; i++) {
+    if (info[i].module_id == FDK_NONE) break;
+  }
+  if (i == FDK_MODULE_LAST) {
+    return -1;
+  }
+  info += i;
+
+  info->module_id = FDK_SBRENC;
+  info->version =
+      LIB_VERSION(SBRENCODER_LIB_VL0, SBRENCODER_LIB_VL1, SBRENCODER_LIB_VL2);
+  LIB_VERSION_STRING(info);
+#ifdef __ANDROID__
+  info->build_date = "";
+  info->build_time = "";
+#else
+  info->build_date = __DATE__;
+  info->build_time = __TIME__;
+#endif
+  info->title = "SBR Encoder";
+
+  /* Set flags */
+  info->flags = 0 | CAPF_SBR_HQ | CAPF_SBR_PS_MPEG;
+  /* End of flags */
+
+  return 0;
+}
diff --git a/fdk-aac/libSBRenc/src/sbr_misc.cpp b/fdk-aac/libSBRenc/src/sbr_misc.cpp
new file mode 100644
index 0000000..83d7e36
--- /dev/null
+++ b/fdk-aac/libSBRenc/src/sbr_misc.cpp
@@ -0,0 +1,265 @@
+/* -----------------------------------------------------------------------------
+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 encoder library ******************************
+
+   Author(s):
+
+   Description:
+
+*******************************************************************************/
+
+/*!
+  \file
+  \brief  Sbr miscellaneous helper functions $Revision: 36750 $
+*/
+#include "sbr_misc.h"
+
+void FDKsbrEnc_Shellsort_fract(FIXP_DBL *in, INT n) {
+  FIXP_DBL v;
+  INT i, j;
+  INT inc = 1;
+
+  do
+    inc = 3 * inc + 1;
+  while (inc <= n);
+
+  do {
+    inc = inc / 3;
+    for (i = inc + 1; i <= n; i++) {
+      v = in[i - 1];
+      j = i;
+      while (in[j - inc - 1] > v) {
+        in[j - 1] = in[j - inc - 1];
+        j -= inc;
+        if (j <= inc) break;
+      }
+      in[j - 1] = v;
+    }
+  } while (inc > 1);
+}
+
+/* Sorting routine */
+void FDKsbrEnc_Shellsort_int(INT *in, INT n) {
+  INT i, j, v;
+  INT inc = 1;
+
+  do
+    inc = 3 * inc + 1;
+  while (inc <= n);
+
+  do {
+    inc = inc / 3;
+    for (i = inc + 1; i <= n; i++) {
+      v = in[i - 1];
+      j = i;
+      while (in[j - inc - 1] > v) {
+        in[j - 1] = in[j - inc - 1];
+        j -= inc;
+        if (j <= inc) break;
+      }
+      in[j - 1] = v;
+    }
+  } while (inc > 1);
+}
+
+/*******************************************************************************
+ Functionname:  FDKsbrEnc_AddVecLeft
+ *******************************************************************************
+
+ Description:
+
+ Arguments:   INT* dst, INT* length_dst, INT* src, INT length_src
+
+ Return:      none
+
+*******************************************************************************/
+void FDKsbrEnc_AddVecLeft(INT *dst, INT *length_dst, INT *src, INT length_src) {
+  INT i;
+
+  for (i = length_src - 1; i >= 0; i--)
+    FDKsbrEnc_AddLeft(dst, length_dst, src[i]);
+}
+
+/*******************************************************************************
+ Functionname:  FDKsbrEnc_AddLeft
+ *******************************************************************************
+
+ Description:
+
+ Arguments:   INT* vector, INT* length_vector, INT value
+
+ Return:      none
+
+*******************************************************************************/
+void FDKsbrEnc_AddLeft(INT *vector, INT *length_vector, INT value) {
+  INT i;
+
+  for (i = *length_vector; i > 0; i--) vector[i] = vector[i - 1];
+  vector[0] = value;
+  (*length_vector)++;
+}
+
+/*******************************************************************************
+ Functionname:  FDKsbrEnc_AddRight
+ *******************************************************************************
+
+ Description:
+
+ Arguments:   INT* vector, INT* length_vector, INT value
+
+ Return:      none
+
+*******************************************************************************/
+void FDKsbrEnc_AddRight(INT *vector, INT *length_vector, INT value) {
+  vector[*length_vector] = value;
+  (*length_vector)++;
+}
+
+/*******************************************************************************
+ Functionname:  FDKsbrEnc_AddVecRight
+ *******************************************************************************
+
+ Description:
+
+ Arguments:   INT* dst, INT* length_dst, INT* src, INT length_src)
+
+ Return:      none
+
+*******************************************************************************/
+void FDKsbrEnc_AddVecRight(INT *dst, INT *length_dst, INT *src,
+                           INT length_src) {
+  INT i;
+  for (i = 0; i < length_src; i++) FDKsbrEnc_AddRight(dst, length_dst, src[i]);
+}
+
+/*****************************************************************************
+
+  functionname: FDKsbrEnc_LSI_divide_scale_fract
+
+  description:  Calculates division with best precision and scales the result.
+
+  return:       num*scale/denom
+
+*****************************************************************************/
+FIXP_DBL FDKsbrEnc_LSI_divide_scale_fract(FIXP_DBL num, FIXP_DBL denom,
+                                          FIXP_DBL scale) {
+  FIXP_DBL tmp = FL2FXCONST_DBL(0.0f);
+  if (num != FL2FXCONST_DBL(0.0f)) {
+    INT shiftCommon;
+    INT shiftNum = CountLeadingBits(num);
+    INT shiftDenom = CountLeadingBits(denom);
+    INT shiftScale = CountLeadingBits(scale);
+
+    num = num << shiftNum;
+    scale = scale << shiftScale;
+
+    tmp = fMultDiv2(num, scale);
+
+    if (denom > (tmp >> fixMin(shiftNum + shiftScale - 1, (DFRACT_BITS - 1)))) {
+      denom = denom << shiftDenom;
+      tmp = schur_div(tmp, denom, 15);
+      shiftCommon =
+          fixMin((shiftNum - shiftDenom + shiftScale - 1), (DFRACT_BITS - 1));
+      if (shiftCommon < 0)
+        tmp <<= -shiftCommon;
+      else
+        tmp >>= shiftCommon;
+    } else {
+      tmp = /*FL2FXCONST_DBL(1.0)*/ (FIXP_DBL)MAXVAL_DBL;
+    }
+  }
+
+  return (tmp);
+}
diff --git a/fdk-aac/libSBRenc/src/sbr_misc.h b/fdk-aac/libSBRenc/src/sbr_misc.h
new file mode 100644
index 0000000..fad853f
--- /dev/null
+++ b/fdk-aac/libSBRenc/src/sbr_misc.h
@@ -0,0 +1,127 @@
+/* -----------------------------------------------------------------------------
+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 encoder library ******************************
+
+   Author(s):
+
+   Description:
+
+*******************************************************************************/
+
+/*!
+  \file
+  \brief  Sbr miscellaneous helper functions prototypes $Revision: 92790 $
+  \author
+*/
+
+#ifndef SBR_MISC_H
+#define SBR_MISC_H
+
+#include "sbr_encoder.h"
+
+/* Sorting routines */
+void FDKsbrEnc_Shellsort_fract(FIXP_DBL *in, INT n);
+void FDKsbrEnc_Shellsort_int(INT *in, INT n);
+
+void FDKsbrEnc_AddLeft(INT *vector, INT *length_vector, INT value);
+void FDKsbrEnc_AddRight(INT *vector, INT *length_vector, INT value);
+void FDKsbrEnc_AddVecLeft(INT *dst, INT *length_dst, INT *src, INT length_src);
+void FDKsbrEnc_AddVecRight(INT *dst, INT *length_vector_dst, INT *src,
+                           INT length_src);
+
+FIXP_DBL FDKsbrEnc_LSI_divide_scale_fract(FIXP_DBL num, FIXP_DBL denom,
+                                          FIXP_DBL scale);
+
+#endif
diff --git a/fdk-aac/libSBRenc/src/sbrenc_freq_sca.cpp b/fdk-aac/libSBRenc/src/sbrenc_freq_sca.cpp
new file mode 100644
index 0000000..c86e047
--- /dev/null
+++ b/fdk-aac/libSBRenc/src/sbrenc_freq_sca.cpp
@@ -0,0 +1,674 @@
+/* -----------------------------------------------------------------------------
+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 encoder library ******************************
+
+   Author(s):
+
+   Description:
+
+*******************************************************************************/
+
+/*!
+  \file
+  \brief  frequency scale $Revision: 95225 $
+*/
+
+#include "sbrenc_freq_sca.h"
+#include "sbr_misc.h"
+
+#include "genericStds.h"
+
+/*  StartFreq */
+static INT getStartFreq(INT fsCore, const INT start_freq);
+
+/* StopFreq */
+static INT getStopFreq(INT fsCore, const INT stop_freq);
+
+static INT numberOfBands(INT b_p_o, INT start, INT stop, FIXP_DBL warp_factor);
+static void CalcBands(INT *diff, INT start, INT stop, INT num_bands);
+static INT modifyBands(INT max_band, INT *diff, INT length);
+static void cumSum(INT start_value, INT *diff, INT length, UCHAR *start_adress);
+
+/*******************************************************************************
+ Functionname:  FDKsbrEnc_getSbrStartFreqRAW
+ *******************************************************************************
+ Description:
+
+ Arguments:
+
+ Return:
+ *******************************************************************************/
+
+INT FDKsbrEnc_getSbrStartFreqRAW(INT startFreq, INT fsCore) {
+  INT result;
+
+  if (startFreq < 0 || startFreq > 15) {
+    return -1;
+  }
+  /* Update startFreq struct */
+  result = getStartFreq(fsCore, startFreq);
+
+  result =
+      (result * (fsCore >> 5) + 1) >> 1; /* (result*fsSBR/QMFbands+1)>>1; */
+
+  return (result);
+
+} /* End FDKsbrEnc_getSbrStartFreqRAW */
+
+/*******************************************************************************
+ Functionname:  getSbrStopFreq
+ *******************************************************************************
+ Description:
+
+ Arguments:
+
+ Return:
+ *******************************************************************************/
+INT FDKsbrEnc_getSbrStopFreqRAW(INT stopFreq, INT fsCore) {
+  INT result;
+
+  if (stopFreq < 0 || stopFreq > 13) return -1;
+
+  /* Uppdate stopFreq struct */
+  result = getStopFreq(fsCore, stopFreq);
+  result =
+      (result * (fsCore >> 5) + 1) >> 1; /* (result*fsSBR/QMFbands+1)>>1; */
+
+  return (result);
+} /* End getSbrStopFreq */
+
+/*******************************************************************************
+ Functionname:  getStartFreq
+ *******************************************************************************
+ Description:
+
+ Arguments:  fsCore - core sampling rate
+
+
+ Return:
+ *******************************************************************************/
+static INT getStartFreq(INT fsCore, const INT start_freq) {
+  INT k0_min;
+
+  switch (fsCore) {
+    case 8000:
+      k0_min = 24; /* (3000 * nQmfChannels / fsSBR ) + 0.5 */
+      break;
+    case 11025:
+      k0_min = 17; /* (3000 * nQmfChannels / fsSBR ) + 0.5 */
+      break;
+    case 12000:
+      k0_min = 16; /* (3000 * nQmfChannels / fsSBR ) + 0.5 */
+      break;
+    case 16000:
+      k0_min = 16; /* (4000 * nQmfChannels / fsSBR ) + 0.5 */
+      break;
+    case 22050:
+      k0_min = 12; /* (4000 * nQmfChannels / fsSBR ) + 0.5 */
+      break;
+    case 24000:
+      k0_min = 11; /* (4000 * nQmfChannels / fsSBR ) + 0.5 */
+      break;
+    case 32000:
+      k0_min = 10; /* (5000 * nQmfChannels / fsSBR ) + 0.5 */
+      break;
+    case 44100:
+      k0_min = 7; /* (5000 * nQmfChannels / fsSBR ) + 0.5 */
+      break;
+    case 48000:
+      k0_min = 7; /* (5000 * nQmfChannels / fsSBR ) + 0.5 */
+      break;
+    case 96000:
+      k0_min = 3; /* (5000 * nQmfChannels / fsSBR ) + 0.5 */
+      break;
+    default:
+      k0_min = 11; /* illegal fs */
+  }
+
+  switch (fsCore) {
+    case 8000: {
+      INT v_offset[] = {-8, -7, -6, -5, -4, -3, -2, -1, 0, 1, 2, 3, 4, 5, 6, 7};
+      return (k0_min + v_offset[start_freq]);
+    }
+    case 11025: {
+      INT v_offset[] = {-5, -4, -3, -2, -1, 0, 1, 2, 3, 4, 5, 6, 7, 9, 11, 13};
+      return (k0_min + v_offset[start_freq]);
+    }
+    case 12000: {
+      INT v_offset[] = {-5, -3, -2, -1, 0, 1, 2, 3, 4, 5, 6, 7, 9, 11, 13, 16};
+      return (k0_min + v_offset[start_freq]);
+    }
+    case 16000: {
+      INT v_offset[] = {-6, -4, -2, -1, 0, 1, 2, 3, 4, 5, 6, 7, 9, 11, 13, 16};
+      return (k0_min + v_offset[start_freq]);
+    }
+    case 22050:
+    case 24000:
+    case 32000: {
+      INT v_offset[] = {-4, -2, -1, 0, 1, 2, 3, 4, 5, 6, 7, 9, 11, 13, 16, 20};
+      return (k0_min + v_offset[start_freq]);
+    }
+    case 44100:
+    case 48000:
+    case 96000: {
+      INT v_offset[] = {-2, -1, 0, 1, 2, 3, 4, 5, 6, 7, 9, 11, 13, 16, 20, 24};
+      return (k0_min + v_offset[start_freq]);
+    }
+    default: {
+      INT v_offset[] = {0, 1, 2, 3, 4, 5, 6, 7, 9, 11, 13, 16, 20, 24, 28, 33};
+      return (k0_min + v_offset[start_freq]);
+    }
+  }
+} /* End getStartFreq */
+
+/*******************************************************************************
+ Functionname:  getStopFreq
+ *******************************************************************************
+ Description:
+
+ Arguments:
+
+ Return:
+ *******************************************************************************/
+static INT getStopFreq(INT fsCore, const INT stop_freq) {
+  INT result, i;
+  INT k1_min;
+  INT v_dstop[13];
+
+  INT *v_stop_freq = NULL;
+  INT v_stop_freq_16[14] = {48, 49, 50, 51, 52, 54, 55,
+                            56, 57, 59, 60, 61, 63, 64};
+  INT v_stop_freq_22[14] = {35, 37, 38, 40, 42, 44, 46,
+                            48, 51, 53, 56, 58, 61, 64};
+  INT v_stop_freq_24[14] = {32, 34, 36, 38, 40, 42, 44,
+                            46, 49, 52, 55, 58, 61, 64};
+  INT v_stop_freq_32[14] = {32, 34, 36, 38, 40, 42, 44,
+                            46, 49, 52, 55, 58, 61, 64};
+  INT v_stop_freq_44[14] = {23, 25, 27, 29, 32, 34, 37,
+                            40, 43, 47, 51, 55, 59, 64};
+  INT v_stop_freq_48[14] = {21, 23, 25, 27, 30, 32, 35,
+                            38, 42, 45, 49, 54, 59, 64};
+  INT v_stop_freq_64[14] = {20, 22, 24, 26, 29, 31, 34,
+                            37, 41, 45, 49, 54, 59, 64};
+  INT v_stop_freq_88[14] = {15, 17, 19, 21, 23, 26, 29,
+                            33, 37, 41, 46, 51, 57, 64};
+  INT v_stop_freq_96[14] = {13, 15, 17, 19, 21, 24, 27,
+                            31, 35, 39, 44, 50, 57, 64};
+  INT v_stop_freq_192[14] = {7,  8,  10, 12, 14, 16, 19,
+                             23, 27, 32, 38, 46, 54, 64};
+
+  switch (fsCore) {
+    case 8000:
+      k1_min = 48;
+      v_stop_freq = v_stop_freq_16;
+      break;
+    case 11025:
+      k1_min = 35;
+      v_stop_freq = v_stop_freq_22;
+      break;
+    case 12000:
+      k1_min = 32;
+      v_stop_freq = v_stop_freq_24;
+      break;
+    case 16000:
+      k1_min = 32;
+      v_stop_freq = v_stop_freq_32;
+      break;
+    case 22050:
+      k1_min = 23;
+      v_stop_freq = v_stop_freq_44;
+      break;
+    case 24000:
+      k1_min = 21;
+      v_stop_freq = v_stop_freq_48;
+      break;
+    case 32000:
+      k1_min = 20;
+      v_stop_freq = v_stop_freq_64;
+      break;
+    case 44100:
+      k1_min = 15;
+      v_stop_freq = v_stop_freq_88;
+      break;
+    case 48000:
+      k1_min = 13;
+      v_stop_freq = v_stop_freq_96;
+      break;
+    case 96000:
+      k1_min = 7;
+      v_stop_freq = v_stop_freq_192;
+      break;
+    default:
+      k1_min = 21; /* illegal fs  */
+  }
+
+  /* Ensure increasing bandwidth */
+  for (i = 0; i <= 12; i++) {
+    v_dstop[i] = v_stop_freq[i + 1] - v_stop_freq[i];
+  }
+
+  FDKsbrEnc_Shellsort_int(v_dstop, 13); /* Sort bandwidth changes */
+
+  result = k1_min;
+  for (i = 0; i < stop_freq; i++) {
+    result = result + v_dstop[i];
+  }
+
+  return (result);
+
+} /* End getStopFreq */
+
+/*******************************************************************************
+ Functionname:  FDKsbrEnc_FindStartAndStopBand
+ *******************************************************************************
+ Description:
+
+ Arguments:     srSbr            SBR sampling freqency
+                srCore           AAC core sampling freqency
+                noChannels       Number of QMF channels
+                startFreq        SBR start frequency in QMF bands
+                stopFreq         SBR start frequency in QMF bands
+
+               *k0               Output parameter
+               *k2               Output parameter
+
+ Return:       Error code (0 is OK)
+ *******************************************************************************/
+INT FDKsbrEnc_FindStartAndStopBand(const INT srSbr, const INT srCore,
+                                   const INT noChannels, const INT startFreq,
+                                   const INT stopFreq, INT *k0, INT *k2) {
+  /* Update startFreq struct */
+  *k0 = getStartFreq(srCore, startFreq);
+
+  /* Test if start freq is outside corecoder range */
+  if (srSbr * noChannels < *k0 * srCore) {
+    return (
+        1); /* raise the cross-over frequency and/or lower the number
+               of target bands per octave (or lower the sampling frequency) */
+  }
+
+  /*Update stopFreq struct */
+  if (stopFreq < 14) {
+    *k2 = getStopFreq(srCore, stopFreq);
+  } else if (stopFreq == 14) {
+    *k2 = 2 * *k0;
+  } else {
+    *k2 = 3 * *k0;
+  }
+
+  /* limit to Nyqvist */
+  if (*k2 > noChannels) {
+    *k2 = noChannels;
+  }
+
+  /* Test for invalid  k0 k2 combinations */
+  if ((srCore == 22050) && ((*k2 - *k0) > MAX_FREQ_COEFFS_FS44100))
+    return (1); /* Number of bands exceeds valid range of MAX_FREQ_COEFFS for
+                   fs=44.1kHz */
+
+  if ((srCore >= 24000) && ((*k2 - *k0) > MAX_FREQ_COEFFS_FS48000))
+    return (1); /* Number of bands exceeds valid range of MAX_FREQ_COEFFS for
+                   fs>=48kHz */
+
+  if ((*k2 - *k0) > MAX_FREQ_COEFFS)
+    return (1); /*Number of bands exceeds valid range of MAX_FREQ_COEFFS */
+
+  if ((*k2 - *k0) < 0) return (1); /* Number of bands is negative */
+
+  return (0);
+}
+
+/*******************************************************************************
+ Functionname:  FDKsbrEnc_UpdateFreqScale
+ *******************************************************************************
+ Description:
+
+ Arguments:
+
+ Return:
+ *******************************************************************************/
+INT FDKsbrEnc_UpdateFreqScale(UCHAR *v_k_master, INT *h_num_bands, const INT k0,
+                              const INT k2, const INT freqScale,
+                              const INT alterScale)
+
+{
+  INT b_p_o = 0; /* bands_per_octave */
+  FIXP_DBL warp = FL2FXCONST_DBL(0.0f);
+  INT dk = 0;
+
+  /* Internal variables */
+  INT k1 = 0, i;
+  INT num_bands0;
+  INT num_bands1;
+  INT diff_tot[MAX_OCTAVE + MAX_SECOND_REGION];
+  INT *diff0 = diff_tot;
+  INT *diff1 = diff_tot + MAX_OCTAVE;
+  INT k2_achived;
+  INT k2_diff;
+  INT incr = 0;
+
+  /* Init */
+  if (freqScale == 1) b_p_o = 12;
+  if (freqScale == 2) b_p_o = 10;
+  if (freqScale == 3) b_p_o = 8;
+
+  if (freqScale > 0) /*Bark*/
+  {
+    if (alterScale == 0)
+      warp = FL2FXCONST_DBL(0.5f); /* 1.0/(1.0*2.0) */
+    else
+      warp = FL2FXCONST_DBL(1.0f / 2.6f); /* 1.0/(1.3*2.0); */
+
+    if (4 * k2 >= 9 * k0) /*two or more regions (how many times the basis band
+                             is copied)*/
+    {
+      k1 = 2 * k0;
+
+      num_bands0 = numberOfBands(b_p_o, k0, k1, FL2FXCONST_DBL(0.5f));
+      num_bands1 = numberOfBands(b_p_o, k1, k2, warp);
+
+      CalcBands(diff0, k0, k1, num_bands0);       /*CalcBands1 => diff0 */
+      FDKsbrEnc_Shellsort_int(diff0, num_bands0); /*SortBands sort diff0 */
+
+      if (diff0[0] == 0) /* too wide FB bands for target tuning */
+      {
+        return (1); /* raise the cross-over frequency and/or lower the number
+                       of target bands per octave (or lower the sampling
+                       frequency */
+      }
+
+      cumSum(k0, diff0, num_bands0, v_k_master); /* cumsum */
+
+      CalcBands(diff1, k1, k2, num_bands1);       /* CalcBands2 => diff1 */
+      FDKsbrEnc_Shellsort_int(diff1, num_bands1); /* SortBands sort diff1 */
+      if (diff0[num_bands0 - 1] > diff1[0])       /* max(1) > min(2) */
+      {
+        if (modifyBands(diff0[num_bands0 - 1], diff1, num_bands1)) return (1);
+      }
+
+      /* Add 2'nd region */
+      cumSum(k1, diff1, num_bands1, &v_k_master[num_bands0]);
+      *h_num_bands = num_bands0 + num_bands1; /* Output nr of bands */
+
+    } else /* one region */
+    {
+      k1 = k2;
+
+      num_bands0 = numberOfBands(b_p_o, k0, k1, FL2FXCONST_DBL(0.5f));
+      CalcBands(diff0, k0, k1, num_bands0);       /* CalcBands1 => diff0 */
+      FDKsbrEnc_Shellsort_int(diff0, num_bands0); /* SortBands sort diff0 */
+
+      if (diff0[0] == 0) /* too wide FB bands for target tuning */
+      {
+        return (1); /* raise the cross-over frequency and/or lower the number
+                       of target bands per octave (or lower the sampling
+                       frequency */
+      }
+
+      cumSum(k0, diff0, num_bands0, v_k_master); /* cumsum */
+      *h_num_bands = num_bands0;                 /* Output nr of bands */
+    }
+  } else /* Linear mode */
+  {
+    if (alterScale == 0) {
+      dk = 1;
+      num_bands0 = 2 * ((k2 - k0) / 2); /* FLOOR to get to few number of bands*/
+    } else {
+      dk = 2;
+      num_bands0 =
+          2 * (((k2 - k0) / dk + 1) / 2); /* ROUND to get closest fit */
+    }
+
+    k2_achived = k0 + num_bands0 * dk;
+    k2_diff = k2 - k2_achived;
+
+    for (i = 0; i < num_bands0; i++) diff_tot[i] = dk;
+
+    /* If linear scale wasn't achived */
+    /* and we got wide SBR are */
+    if (k2_diff < 0) {
+      incr = 1;
+      i = 0;
+    }
+
+    /* If linear scale wasn't achived */
+    /* and we got small SBR are */
+    if (k2_diff > 0) {
+      incr = -1;
+      i = num_bands0 - 1;
+    }
+
+    /* Adjust diff vector to get sepc. SBR range */
+    while (k2_diff != 0) {
+      diff_tot[i] = diff_tot[i] - incr;
+      i = i + incr;
+      k2_diff = k2_diff + incr;
+    }
+
+    cumSum(k0, diff_tot, num_bands0, v_k_master); /* cumsum */
+    *h_num_bands = num_bands0;                    /* Output nr of bands */
+  }
+
+  if (*h_num_bands < 1) return (1); /*To small sbr area */
+
+  return (0);
+} /* End FDKsbrEnc_UpdateFreqScale */
+
+static INT numberOfBands(INT b_p_o, INT start, INT stop, FIXP_DBL warp_factor) {
+  INT result = 0;
+  /* result = 2* (INT) ( (double)b_p_o *
+   * (double)(FDKlog((double)stop/(double)start)/FDKlog((double)2)) *
+   * (double)FX_DBL2FL(warp_factor) + 0.5); */
+  result = ((b_p_o * fMult((CalcLdInt(stop) - CalcLdInt(start)), warp_factor) +
+             (FL2FX_DBL(0.5f) >> LD_DATA_SHIFT)) >>
+            ((DFRACT_BITS - 1) - LD_DATA_SHIFT))
+           << 1; /* do not optimize anymore (rounding!!) */
+
+  return (result);
+}
+
+static void CalcBands(INT *diff, INT start, INT stop, INT num_bands) {
+  INT i, qb, qe, qtmp;
+  INT previous;
+  INT current;
+  FIXP_DBL base, exp, tmp;
+
+  previous = start;
+  for (i = 1; i <= num_bands; i++) {
+    base = fDivNorm((FIXP_DBL)stop, (FIXP_DBL)start, &qb);
+    exp = fDivNorm((FIXP_DBL)i, (FIXP_DBL)num_bands, &qe);
+    tmp = fPow(base, qb, exp, qe, &qtmp);
+    tmp = fMult(tmp, (FIXP_DBL)(start << 24));
+    current = (INT)scaleValue(tmp, qtmp - 23);
+    current = (current + 1) >> 1; /* rounding*/
+    diff[i - 1] = current - previous;
+    previous = current;
+  }
+
+} /* End CalcBands */
+
+static void cumSum(INT start_value, INT *diff, INT length,
+                   UCHAR *start_adress) {
+  INT i;
+  start_adress[0] = start_value;
+  for (i = 1; i <= length; i++)
+    start_adress[i] = start_adress[i - 1] + diff[i - 1];
+} /* End cumSum */
+
+static INT modifyBands(INT max_band_previous, INT *diff, INT length) {
+  INT change = max_band_previous - diff[0];
+
+  /* Limit the change so that the last band cannot get narrower than the first
+   * one */
+  if (change > (diff[length - 1] - diff[0]) / 2)
+    change = (diff[length - 1] - diff[0]) / 2;
+
+  diff[0] += change;
+  diff[length - 1] -= change;
+  FDKsbrEnc_Shellsort_int(diff, length);
+
+  return (0);
+} /* End modifyBands */
+
+/*******************************************************************************
+ Functionname:  FDKsbrEnc_UpdateHiRes
+ *******************************************************************************
+ Description:
+
+
+ Arguments:
+
+ Return:
+ *******************************************************************************/
+INT FDKsbrEnc_UpdateHiRes(UCHAR *h_hires, INT *num_hires, UCHAR *v_k_master,
+                          INT num_master, INT *xover_band) {
+  INT i;
+  INT max1, max2;
+
+  if ((v_k_master[*xover_band] >
+       32) || /* v_k_master[*xover_band] > noQMFChannels(dualRate)/divider */
+      (*xover_band > num_master)) {
+    /* xover_band error, too big for this startFreq. Will be clipped */
+
+    /* Calculate maximum value for xover_band */
+    max1 = 0;
+    max2 = num_master;
+    while ((v_k_master[max1 + 1] < 32) && /* noQMFChannels(dualRate)/divider */
+           ((max1 + 1) < max2)) {
+      max1++;
+    }
+
+    *xover_band = max1;
+  }
+
+  *num_hires = num_master - *xover_band;
+  for (i = *xover_band; i <= num_master; i++) {
+    h_hires[i - *xover_band] = v_k_master[i];
+  }
+
+  return (0);
+} /* End FDKsbrEnc_UpdateHiRes */
+
+/*******************************************************************************
+ Functionname:  FDKsbrEnc_UpdateLoRes
+ *******************************************************************************
+ Description:
+
+ Arguments:
+
+ Return:
+ *******************************************************************************/
+void FDKsbrEnc_UpdateLoRes(UCHAR *h_lores, INT *num_lores, UCHAR *h_hires,
+                           INT num_hires) {
+  INT i;
+
+  if (num_hires % 2 == 0) /* if even number of hires bands */
+  {
+    *num_lores = num_hires / 2;
+    /* Use every second lores=hires[0,2,4...] */
+    for (i = 0; i <= *num_lores; i++) h_lores[i] = h_hires[i * 2];
+
+  } else /* odd number of hires which means xover is odd */
+  {
+    *num_lores = (num_hires + 1) / 2;
+
+    /* Use lores=hires[0,1,3,5 ...] */
+    h_lores[0] = h_hires[0];
+    for (i = 1; i <= *num_lores; i++) {
+      h_lores[i] = h_hires[i * 2 - 1];
+    }
+  }
+
+} /* End FDKsbrEnc_UpdateLoRes */
diff --git a/fdk-aac/libSBRenc/src/sbrenc_freq_sca.h b/fdk-aac/libSBRenc/src/sbrenc_freq_sca.h
new file mode 100644
index 0000000..9b8d360
--- /dev/null
+++ b/fdk-aac/libSBRenc/src/sbrenc_freq_sca.h
@@ -0,0 +1,132 @@
+/* -----------------------------------------------------------------------------
+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 encoder library ******************************
+
+   Author(s):
+
+   Description:
+
+*******************************************************************************/
+
+/*!
+  \file
+  \brief  frequency scale prototypes $Revision: 92790 $
+*/
+#ifndef SBRENC_FREQ_SCA_H
+#define SBRENC_FREQ_SCA_H
+
+#include "sbr_encoder.h"
+#include "sbr_def.h"
+
+#define MAX_OCTAVE 29
+#define MAX_SECOND_REGION 50
+
+INT FDKsbrEnc_UpdateFreqScale(UCHAR *v_k_master, INT *h_num_bands, const INT k0,
+                              const INT k2, const INT freq_scale,
+                              const INT alter_scale);
+
+INT FDKsbrEnc_UpdateHiRes(UCHAR *h_hires, INT *num_hires, UCHAR *v_k_master,
+                          INT num_master, INT *xover_band);
+
+void FDKsbrEnc_UpdateLoRes(UCHAR *v_lores, INT *num_lores, UCHAR *v_hires,
+                           INT num_hires);
+
+INT FDKsbrEnc_FindStartAndStopBand(const INT srSbr, const INT srCore,
+                                   const INT noChannels, const INT startFreq,
+                                   const INT stop_freq, INT *k0, INT *k2);
+
+INT FDKsbrEnc_getSbrStartFreqRAW(INT startFreq, INT fsCore);
+INT FDKsbrEnc_getSbrStopFreqRAW(INT stopFreq, INT fsCore);
+#endif
diff --git a/fdk-aac/libSBRenc/src/sbrenc_ram.cpp b/fdk-aac/libSBRenc/src/sbrenc_ram.cpp
new file mode 100644
index 0000000..fb30fa2
--- /dev/null
+++ b/fdk-aac/libSBRenc/src/sbrenc_ram.cpp
@@ -0,0 +1,249 @@
+/* -----------------------------------------------------------------------------
+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 encoder library ******************************
+
+   Author(s):
+
+   Description:
+
+*******************************************************************************/
+
+/*!
+  \file
+  \brief Memory layout
+  $Revision: 92864 $
+
+  This module declares all static and dynamic memory spaces
+*/
+#include "sbrenc_ram.h"
+
+#include "sbr.h"
+#include "genericStds.h"
+
+C_AALLOC_MEM(Ram_SbrDynamic_RAM, FIXP_DBL,
+             ((SBR_ENC_DYN_RAM_SIZE) / sizeof(FIXP_DBL)))
+
+/*!
+  \name StaticSbrData
+
+  Static memory areas, must not be overwritten in other sections of the encoder
+*/
+/* @{ */
+
+/*! static sbr encoder instance for one encoder (2 channels)
+  all major static and dynamic memory areas are located
+  in module sbr_ram and sbr rom
+*/
+C_ALLOC_MEM(Ram_SbrEncoder, SBR_ENCODER, 1)
+C_ALLOC_MEM2(Ram_SbrChannel, SBR_CHANNEL, 1, (8))
+C_ALLOC_MEM2(Ram_SbrElement, SBR_ELEMENT, 1, (8))
+
+/*! Filter states for QMF-analysis. <br>
+  Dimension: #MAXNRSBRCHANNELS * #SBR_QMF_FILTER_LENGTH
+*/
+C_AALLOC_MEM2_L(Ram_Sbr_QmfStatesAnalysis, FIXP_QAS, 640, (8), SECT_DATA_L1)
+
+/*! Matrix holding the quota values for all estimates, all channels
+  Dimension #MAXNRSBRCHANNELS * +#SBR_QMF_CHANNELS* #MAX_NO_OF_ESTIMATES
+*/
+C_ALLOC_MEM2_L(Ram_Sbr_quotaMatrix, FIXP_DBL, (MAX_NO_OF_ESTIMATES * 64), (8),
+               SECT_DATA_L1)
+
+/*! Matrix holding the sign values for all estimates, all channels
+  Dimension #MAXNRSBRCHANNELS * +#SBR_QMF_CHANNELS* #MAX_NO_OF_ESTIMATES
+*/
+C_ALLOC_MEM2(Ram_Sbr_signMatrix, INT, (MAX_NO_OF_ESTIMATES * 64), (8))
+
+/*! Frequency band table (low res) <br>
+  Dimension #MAX_FREQ_COEFFS/2+1
+*/
+C_ALLOC_MEM2(Ram_Sbr_freqBandTableLO, UCHAR, (MAX_FREQ_COEFFS / 2 + 1), (8))
+
+/*! Frequency band table (high res) <br>
+  Dimension #MAX_FREQ_COEFFS +1
+*/
+C_ALLOC_MEM2(Ram_Sbr_freqBandTableHI, UCHAR, (MAX_FREQ_COEFFS + 1), (8))
+
+/*! vk matser table <br>
+  Dimension #MAX_FREQ_COEFFS +1
+*/
+C_ALLOC_MEM2(Ram_Sbr_v_k_master, UCHAR, (MAX_FREQ_COEFFS + 1), (8))
+
+/*
+  Missing harmonics detection
+*/
+
+/*! sbr_detectionVectors <br>
+  Dimension #MAX_NUM_CHANNELS*#MAX_NO_OF_ESTIMATES*#MAX_FREQ_COEFFS]
+*/
+C_ALLOC_MEM2(Ram_Sbr_detectionVectors, UCHAR,
+             (MAX_NO_OF_ESTIMATES * MAX_FREQ_COEFFS), (8))
+
+/*! sbr_prevCompVec[ <br>
+  Dimension #MAX_NUM_CHANNELS*#MAX_FREQ_COEFFS]
+*/
+C_ALLOC_MEM2(Ram_Sbr_prevEnvelopeCompensation, UCHAR, MAX_FREQ_COEFFS, (8))
+/*! sbr_guideScfb[ <br>
+  Dimension #MAX_NUM_CHANNELS*#MAX_FREQ_COEFFS]
+*/
+C_ALLOC_MEM2(Ram_Sbr_guideScfb, UCHAR, MAX_FREQ_COEFFS, (8))
+
+/*! sbr_guideVectorDetected <br>
+  Dimension #MAX_NUM_CHANNELS*#MAX_NO_OF_ESTIMATES*#MAX_FREQ_COEFFS]
+*/
+C_ALLOC_MEM2(Ram_Sbr_guideVectorDetected, UCHAR,
+             (MAX_NO_OF_ESTIMATES * MAX_FREQ_COEFFS), (8))
+C_ALLOC_MEM2(Ram_Sbr_guideVectorDiff, FIXP_DBL,
+             (MAX_NO_OF_ESTIMATES * MAX_FREQ_COEFFS), (8))
+C_ALLOC_MEM2(Ram_Sbr_guideVectorOrig, FIXP_DBL,
+             (MAX_NO_OF_ESTIMATES * MAX_FREQ_COEFFS), (8))
+
+/*
+  Static Parametric Stereo memory
+*/
+C_AALLOC_MEM_L(Ram_PsQmfStatesSynthesis, FIXP_DBL, 640 / 2, SECT_DATA_L1)
+
+C_ALLOC_MEM_L(Ram_PsEncode, PS_ENCODE, 1, SECT_DATA_L1)
+C_ALLOC_MEM(Ram_ParamStereo, PARAMETRIC_STEREO, 1)
+
+/* @} */
+
+/*!
+  \name DynamicSbrData
+
+  Dynamic memory areas, might be reused in other algorithm sections,
+  e.g. the core encoder.
+*/
+/* @{ */
+
+/*! Energy buffer for envelope extraction <br>
+  Dimension #MAXNRSBRCHANNELS * +#SBR_QMF_SLOTS *  #SBR_QMF_CHANNELS
+*/
+C_ALLOC_MEM2(Ram_Sbr_envYBuffer, FIXP_DBL, (32 / 2 * 64), (8))
+
+FIXP_DBL* GetRam_Sbr_envYBuffer(int n, UCHAR* dynamic_RAM) {
+  FDK_ASSERT(dynamic_RAM != 0);
+  /* The reinterpret_cast is used to suppress a compiler warning. We know that
+   * (dynamic_RAM + OFFSET_NRG + (n*Y_2_BUF_BYTE)) is sufficiently aligned, so
+   * the cast is safe */
+  return reinterpret_cast<FIXP_DBL*>(
+      reinterpret_cast<void*>(dynamic_RAM + OFFSET_NRG + (n * Y_2_BUF_BYTE)));
+}
+
+/*
+ * QMF data
+ */
+/* The SBR encoder uses a single channel overlapping buffer set (always n=0),
+ * but PS does not. */
+FIXP_DBL* GetRam_Sbr_envRBuffer(int n, UCHAR* dynamic_RAM) {
+  FDK_ASSERT(dynamic_RAM != 0);
+  /* The reinterpret_cast is used to suppress a compiler warning. We know that
+   * (dynamic_RAM + OFFSET_QMF + (n*(ENV_R_BUFF_BYTE+ENV_I_BUFF_BYTE))) is
+   * sufficiently aligned, so the cast is safe */
+  return reinterpret_cast<FIXP_DBL*>(reinterpret_cast<void*>(
+      dynamic_RAM + OFFSET_QMF + (n * (ENV_R_BUFF_BYTE + ENV_I_BUFF_BYTE))));
+}
+FIXP_DBL* GetRam_Sbr_envIBuffer(int n, UCHAR* dynamic_RAM) {
+  FDK_ASSERT(dynamic_RAM != 0);
+  /* The reinterpret_cast is used to suppress a compiler warning. We know that
+   * (dynamic_RAM + OFFSET_QMF + (ENV_R_BUFF_BYTE) +
+   * (n*(ENV_R_BUFF_BYTE+ENV_I_BUFF_BYTE))) is sufficiently aligned, so the cast
+   * is safe */
+  return reinterpret_cast<FIXP_DBL*>(
+      reinterpret_cast<void*>(dynamic_RAM + OFFSET_QMF + (ENV_R_BUFF_BYTE) +
+                              (n * (ENV_R_BUFF_BYTE + ENV_I_BUFF_BYTE))));
+}
+
+/* @} */
diff --git a/fdk-aac/libSBRenc/src/sbrenc_ram.h b/fdk-aac/libSBRenc/src/sbrenc_ram.h
new file mode 100644
index 0000000..cf23378
--- /dev/null
+++ b/fdk-aac/libSBRenc/src/sbrenc_ram.h
@@ -0,0 +1,199 @@
+/* -----------------------------------------------------------------------------
+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 encoder library ******************************
+
+   Author(s):
+
+   Description:
+
+*******************************************************************************/
+
+/*!
+\file
+\brief Memory layout
+$Revision: 92790 $
+*/
+#ifndef SBRENC_RAM_H
+#define SBRENC_RAM_H
+
+#include "sbr_def.h"
+#include "env_est.h"
+#include "sbr_encoder.h"
+#include "sbr.h"
+
+#include "ps_main.h"
+#include "ps_encode.h"
+
+#define ENV_TRANSIENTS_BYTE ((sizeof(FIXP_DBL) * (MAX_NUM_CHANNELS * 3 * 32)))
+
+#define ENV_R_BUFF_BYTE ((sizeof(FIXP_DBL) * ((32) * MAX_HYBRID_BANDS)))
+#define ENV_I_BUFF_BYTE ((sizeof(FIXP_DBL) * ((32) * MAX_HYBRID_BANDS)))
+#define Y_BUF_CH_BYTE \
+  ((2 * sizeof(FIXP_DBL) * (((32) - (32 / 2)) * MAX_HYBRID_BANDS)))
+
+#define ENV_R_BUF_PS_BYTE ((sizeof(FIXP_DBL) * 32 * 64 / 2))
+#define ENV_I_BUF_PS_BYTE ((sizeof(FIXP_DBL) * 32 * 64 / 2))
+
+#define TON_BUF_CH_BYTE \
+  ((sizeof(FIXP_DBL) * (MAX_NO_OF_ESTIMATES * MAX_FREQ_COEFFS)))
+
+#define Y_2_BUF_BYTE (Y_BUF_CH_BYTE)
+
+/* Workbuffer RAM - Allocation */
+/*
+ ++++++++++++++++++++++++++++++++++++++++++++++++++++
+ |        OFFSET_QMF       |        OFFSET_NRG      |
+ ++++++++++++++++++++++++++++++++++++++++++++++++++++
+  ------------------------- -------------------------
+ |                         |         0.5 *          |
+ |     sbr_envRBuffer      | sbr_envYBuffer_size    |
+ |     sbr_envIBuffer      |                        |
+  ------------------------- -------------------------
+
+*/
+#define BUF_NRG_SIZE ((MAX_NUM_CHANNELS * Y_2_BUF_BYTE))
+#define BUF_QMF_SIZE (ENV_R_BUFF_BYTE + ENV_I_BUFF_BYTE)
+
+/* Size of the shareable memory region than can be reused */
+#define SBR_ENC_DYN_RAM_SIZE (BUF_QMF_SIZE + BUF_NRG_SIZE)
+
+#define OFFSET_QMF (0)
+#define OFFSET_NRG (OFFSET_QMF + BUF_QMF_SIZE)
+
+/*
+ *****************************************************************************************************
+ */
+
+H_ALLOC_MEM(Ram_SbrDynamic_RAM, FIXP_DBL)
+
+H_ALLOC_MEM(Ram_SbrEncoder, SBR_ENCODER)
+H_ALLOC_MEM(Ram_SbrChannel, SBR_CHANNEL)
+H_ALLOC_MEM(Ram_SbrElement, SBR_ELEMENT)
+
+H_ALLOC_MEM(Ram_Sbr_quotaMatrix, FIXP_DBL)
+H_ALLOC_MEM(Ram_Sbr_signMatrix, INT)
+
+H_ALLOC_MEM(Ram_Sbr_QmfStatesAnalysis, FIXP_QAS)
+
+H_ALLOC_MEM(Ram_Sbr_freqBandTableLO, UCHAR)
+H_ALLOC_MEM(Ram_Sbr_freqBandTableHI, UCHAR)
+H_ALLOC_MEM(Ram_Sbr_v_k_master, UCHAR)
+
+H_ALLOC_MEM(Ram_Sbr_detectionVectors, UCHAR)
+H_ALLOC_MEM(Ram_Sbr_prevEnvelopeCompensation, UCHAR)
+H_ALLOC_MEM(Ram_Sbr_guideScfb, UCHAR)
+H_ALLOC_MEM(Ram_Sbr_guideVectorDetected, UCHAR)
+
+/* Dynamic Memory Allocation */
+
+H_ALLOC_MEM(Ram_Sbr_envYBuffer, FIXP_DBL)
+FIXP_DBL* GetRam_Sbr_envYBuffer(int n, UCHAR* dynamic_RAM);
+FIXP_DBL* GetRam_Sbr_envRBuffer(int n, UCHAR* dynamic_RAM);
+FIXP_DBL* GetRam_Sbr_envIBuffer(int n, UCHAR* dynamic_RAM);
+
+H_ALLOC_MEM(Ram_Sbr_guideVectorDiff, FIXP_DBL)
+H_ALLOC_MEM(Ram_Sbr_guideVectorOrig, FIXP_DBL)
+
+H_ALLOC_MEM(Ram_PsQmfStatesSynthesis, FIXP_DBL)
+
+H_ALLOC_MEM(Ram_PsEncode, PS_ENCODE)
+
+FIXP_DBL* FDKsbrEnc_SliceRam_PsRqmf(FIXP_DBL* rQmfData, UCHAR* dynamic_RAM,
+                                    int n, int i, int qmfSlots);
+FIXP_DBL* FDKsbrEnc_SliceRam_PsIqmf(FIXP_DBL* iQmfData, UCHAR* dynamic_RAM,
+                                    int n, int i, int qmfSlots);
+
+H_ALLOC_MEM(Ram_ParamStereo, PARAMETRIC_STEREO)
+#endif
diff --git a/fdk-aac/libSBRenc/src/sbrenc_rom.cpp b/fdk-aac/libSBRenc/src/sbrenc_rom.cpp
new file mode 100644
index 0000000..737afaf
--- /dev/null
+++ b/fdk-aac/libSBRenc/src/sbrenc_rom.cpp
@@ -0,0 +1,910 @@
+/* -----------------------------------------------------------------------------
+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 encoder library ******************************
+
+   Author(s):   Tobias Chalupka
+
+   Description: Definition of constant tables
+
+*******************************************************************************/
+
+/*!
+  \file
+  \brief  Definition of constant tables
+  $Revision: 95404 $
+
+  This module contains most of the constant data that can be stored in ROM.
+*/
+
+#include "sbrenc_rom.h"
+#include "genericStds.h"
+
+//@{
+/*******************************************************************************
+
+   Table Overview:
+
+ o envelope level,   1.5 dB:
+    1a)  v_Huff_envelopeLevelC10T[121]
+    1b)  v_Huff_envelopeLevelL10T[121]
+    2a)  v_Huff_envelopeLevelC10F[121]
+    2b)  v_Huff_envelopeLevelL10F[121]
+
+ o envelope balance, 1.5 dB:
+    3a)  bookSbrEnvBalanceC10T[49]
+    3b)  bookSbrEnvBalanceL10T[49]
+    4a)  bookSbrEnvBalanceC10F[49]
+    4b)  bookSbrEnvBalanceL10F[49]
+
+ o envelope level,   3.0 dB:
+    5a)  v_Huff_envelopeLevelC11T[63]
+    5b)  v_Huff_envelopeLevelL11T[63]
+    6a)  v_Huff_envelopeLevelC11F[63]
+    6b)  v_Huff_envelopeLevelC11F[63]
+
+ o envelope balance, 3.0 dB:
+    7a)  bookSbrEnvBalanceC11T[25]
+    7b)  bookSbrEnvBalanceL11T[25]
+    8a)  bookSbrEnvBalanceC11F[25]
+    8b)  bookSbrEnvBalanceL11F[25]
+
+ o noise level,      3.0 dB:
+    9a)  v_Huff_NoiseLevelC11T[63]
+    9b)  v_Huff_NoiseLevelL11T[63]
+    - ) (v_Huff_envelopeLevelC11F[63] is used for freq dir)
+    - ) (v_Huff_envelopeLevelL11F[63] is used for freq dir)
+
+ o noise balance,    3.0 dB:
+   10a)  bookSbrNoiseBalanceC11T[25]
+   10b)  bookSbrNoiseBalanceL11T[25]
+    - ) (bookSbrEnvBalanceC11F[25] is used for freq dir)
+    - ) (bookSbrEnvBalanceL11F[25] is used for freq dir)
+
+
+  (1.5 dB is never used for noise)
+
+********************************************************************************/
+
+/*******************************************************************************/
+/* table       : envelope level, 1.5 dB */
+/* theor range : [-58,58], CODE_BOOK_SCF_LAV   = 58 */
+/* implem range: [-60,60], CODE_BOOK_SCF_LAV10 = 60 */
+/* raw stats   : envelopeLevel_00 (yes, wrong suffix in name)  KK 01-03-09 */
+/*******************************************************************************/
+
+/* direction: time
+   contents : codewords
+   raw table: HuffCode3C2FIX.m/envelopeLevel_00T_cF.mat/v_nChex_cF
+   built by : FH 01-07-05 */
+
+const INT v_Huff_envelopeLevelC10T[121] = {
+    0x0003FFD6, 0x0003FFD7, 0x0003FFD8, 0x0003FFD9, 0x0003FFDA, 0x0003FFDB,
+    0x0007FFB8, 0x0007FFB9, 0x0007FFBA, 0x0007FFBB, 0x0007FFBC, 0x0007FFBD,
+    0x0007FFBE, 0x0007FFBF, 0x0007FFC0, 0x0007FFC1, 0x0007FFC2, 0x0007FFC3,
+    0x0007FFC4, 0x0007FFC5, 0x0007FFC6, 0x0007FFC7, 0x0007FFC8, 0x0007FFC9,
+    0x0007FFCA, 0x0007FFCB, 0x0007FFCC, 0x0007FFCD, 0x0007FFCE, 0x0007FFCF,
+    0x0007FFD0, 0x0007FFD1, 0x0007FFD2, 0x0007FFD3, 0x0001FFE6, 0x0003FFD4,
+    0x0000FFF0, 0x0001FFE9, 0x0003FFD5, 0x0001FFE7, 0x0000FFF1, 0x0000FFEC,
+    0x0000FFED, 0x0000FFEE, 0x00007FF4, 0x00003FF9, 0x00003FF7, 0x00001FFA,
+    0x00001FF9, 0x00000FFB, 0x000007FC, 0x000003FC, 0x000001FD, 0x000000FD,
+    0x0000007D, 0x0000003D, 0x0000001D, 0x0000000D, 0x00000005, 0x00000001,
+    0x00000000, 0x00000004, 0x0000000C, 0x0000001C, 0x0000003C, 0x0000007C,
+    0x000000FC, 0x000001FC, 0x000003FD, 0x00000FFA, 0x00001FF8, 0x00003FF6,
+    0x00003FF8, 0x00007FF5, 0x0000FFEF, 0x0001FFE8, 0x0000FFF2, 0x0007FFD4,
+    0x0007FFD5, 0x0007FFD6, 0x0007FFD7, 0x0007FFD8, 0x0007FFD9, 0x0007FFDA,
+    0x0007FFDB, 0x0007FFDC, 0x0007FFDD, 0x0007FFDE, 0x0007FFDF, 0x0007FFE0,
+    0x0007FFE1, 0x0007FFE2, 0x0007FFE3, 0x0007FFE4, 0x0007FFE5, 0x0007FFE6,
+    0x0007FFE7, 0x0007FFE8, 0x0007FFE9, 0x0007FFEA, 0x0007FFEB, 0x0007FFEC,
+    0x0007FFED, 0x0007FFEE, 0x0007FFEF, 0x0007FFF0, 0x0007FFF1, 0x0007FFF2,
+    0x0007FFF3, 0x0007FFF4, 0x0007FFF5, 0x0007FFF6, 0x0007FFF7, 0x0007FFF8,
+    0x0007FFF9, 0x0007FFFA, 0x0007FFFB, 0x0007FFFC, 0x0007FFFD, 0x0007FFFE,
+    0x0007FFFF};
+
+/* direction: time
+   contents : codeword lengths
+   raw table: HuffCode3C2FIX.m/envelopeLevel_00T_cF.mat/v_nLhex_cF
+   built by : FH 01-07-05 */
+
+const UCHAR v_Huff_envelopeLevelL10T[121] = {
+    0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x13, 0x13, 0x13, 0x13, 0x13,
+    0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13,
+    0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13,
+    0x13, 0x11, 0x12, 0x10, 0x11, 0x12, 0x11, 0x10, 0x10, 0x10, 0x10,
+    0x0F, 0x0E, 0x0E, 0x0D, 0x0D, 0x0C, 0x0B, 0x0A, 0x09, 0x08, 0x07,
+    0x06, 0x05, 0x04, 0x03, 0x02, 0x02, 0x03, 0x04, 0x05, 0x06, 0x07,
+    0x08, 0x09, 0x0A, 0x0C, 0x0D, 0x0E, 0x0E, 0x0F, 0x10, 0x11, 0x10,
+    0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13,
+    0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13,
+    0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13,
+    0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13};
+
+/* direction: freq
+   contents : codewords
+   raw table: HuffCode3C2FIX.m/envelopeLevel_00F_cF.mat/v_nChex_cF
+   built by : FH 01-07-05 */
+
+const INT v_Huff_envelopeLevelC10F[121] = {
+    0x0007FFE7, 0x0007FFE8, 0x000FFFD2, 0x000FFFD3, 0x000FFFD4, 0x000FFFD5,
+    0x000FFFD6, 0x000FFFD7, 0x000FFFD8, 0x0007FFDA, 0x000FFFD9, 0x000FFFDA,
+    0x000FFFDB, 0x000FFFDC, 0x0007FFDB, 0x000FFFDD, 0x0007FFDC, 0x0007FFDD,
+    0x000FFFDE, 0x0003FFE4, 0x000FFFDF, 0x000FFFE0, 0x000FFFE1, 0x0007FFDE,
+    0x000FFFE2, 0x000FFFE3, 0x000FFFE4, 0x0007FFDF, 0x000FFFE5, 0x0007FFE0,
+    0x0003FFE8, 0x0007FFE1, 0x0003FFE0, 0x0003FFE9, 0x0001FFEF, 0x0003FFE5,
+    0x0001FFEC, 0x0001FFED, 0x0001FFEE, 0x0000FFF4, 0x0000FFF3, 0x0000FFF0,
+    0x00007FF7, 0x00007FF6, 0x00003FFA, 0x00001FFA, 0x00001FF9, 0x00000FFA,
+    0x00000FF8, 0x000007F9, 0x000003FB, 0x000001FC, 0x000001FA, 0x000000FB,
+    0x0000007C, 0x0000003C, 0x0000001C, 0x0000000C, 0x00000005, 0x00000001,
+    0x00000000, 0x00000004, 0x0000000D, 0x0000001D, 0x0000003D, 0x000000FA,
+    0x000000FC, 0x000001FB, 0x000003FA, 0x000007F8, 0x000007FA, 0x000007FB,
+    0x00000FF9, 0x00000FFB, 0x00001FF8, 0x00001FFB, 0x00003FF8, 0x00003FF9,
+    0x0000FFF1, 0x0000FFF2, 0x0001FFEA, 0x0001FFEB, 0x0003FFE1, 0x0003FFE2,
+    0x0003FFEA, 0x0003FFE3, 0x0003FFE6, 0x0003FFE7, 0x0003FFEB, 0x000FFFE6,
+    0x0007FFE2, 0x000FFFE7, 0x000FFFE8, 0x000FFFE9, 0x000FFFEA, 0x000FFFEB,
+    0x000FFFEC, 0x0007FFE3, 0x000FFFED, 0x000FFFEE, 0x000FFFEF, 0x000FFFF0,
+    0x0007FFE4, 0x000FFFF1, 0x0003FFEC, 0x000FFFF2, 0x000FFFF3, 0x0007FFE5,
+    0x0007FFE6, 0x000FFFF4, 0x000FFFF5, 0x000FFFF6, 0x000FFFF7, 0x000FFFF8,
+    0x000FFFF9, 0x000FFFFA, 0x000FFFFB, 0x000FFFFC, 0x000FFFFD, 0x000FFFFE,
+    0x000FFFFF};
+
+/* direction: freq
+   contents : codeword lengths
+   raw table: HuffCode3C2FIX.m/envelopeLevel_00F_cF.mat/v_nLhex_cF
+   built by : FH 01-07-05 */
+
+const UCHAR v_Huff_envelopeLevelL10F[121] = {
+    0x13, 0x13, 0x14, 0x14, 0x14, 0x14, 0x14, 0x14, 0x14, 0x13, 0x14,
+    0x14, 0x14, 0x14, 0x13, 0x14, 0x13, 0x13, 0x14, 0x12, 0x14, 0x14,
+    0x14, 0x13, 0x14, 0x14, 0x14, 0x13, 0x14, 0x13, 0x12, 0x13, 0x12,
+    0x12, 0x11, 0x12, 0x11, 0x11, 0x11, 0x10, 0x10, 0x10, 0x0F, 0x0F,
+    0x0E, 0x0D, 0x0D, 0x0C, 0x0C, 0x0B, 0x0A, 0x09, 0x09, 0x08, 0x07,
+    0x06, 0x05, 0x04, 0x03, 0x02, 0x02, 0x03, 0x04, 0x05, 0x06, 0x08,
+    0x08, 0x09, 0x0A, 0x0B, 0x0B, 0x0B, 0x0C, 0x0C, 0x0D, 0x0D, 0x0E,
+    0x0E, 0x10, 0x10, 0x11, 0x11, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12,
+    0x12, 0x14, 0x13, 0x14, 0x14, 0x14, 0x14, 0x14, 0x14, 0x13, 0x14,
+    0x14, 0x14, 0x14, 0x13, 0x14, 0x12, 0x14, 0x14, 0x13, 0x13, 0x14,
+    0x14, 0x14, 0x14, 0x14, 0x14, 0x14, 0x14, 0x14, 0x14, 0x14, 0x14};
+
+/*******************************************************************************/
+/* table       : envelope balance, 1.5 dB */
+/* theor range : [-48,48], CODE_BOOK_SCF_LAV = 48 */
+/* implem range: same but mapped to [-24,24], CODE_BOOK_SCF_LAV_BALANCE10 = 24
+ */
+/* raw stats   : envelopePan_00 (yes, wrong suffix in name)  KK 01-03-09 */
+/*******************************************************************************/
+
+/* direction: time
+   contents : codewords
+   raw table: HuffCode3C.m/envelopePan_00T.mat/v_nBhex
+   built by : FH 01-05-15 */
+
+const INT bookSbrEnvBalanceC10T[49] = {
+    0x0000FFE4, 0x0000FFE5, 0x0000FFE6, 0x0000FFE7, 0x0000FFE8, 0x0000FFE9,
+    0x0000FFEA, 0x0000FFEB, 0x0000FFEC, 0x0000FFED, 0x0000FFEE, 0x0000FFEF,
+    0x0000FFF0, 0x0000FFF1, 0x0000FFF2, 0x0000FFF3, 0x0000FFF4, 0x0000FFE2,
+    0x00000FFC, 0x000007FC, 0x000001FE, 0x0000007E, 0x0000001E, 0x00000006,
+    0x00000000, 0x00000002, 0x0000000E, 0x0000003E, 0x000000FE, 0x000007FD,
+    0x00000FFD, 0x00007FF0, 0x0000FFE3, 0x0000FFF5, 0x0000FFF6, 0x0000FFF7,
+    0x0000FFF8, 0x0000FFF9, 0x0000FFFA, 0x0001FFF6, 0x0001FFF7, 0x0001FFF8,
+    0x0001FFF9, 0x0001FFFA, 0x0001FFFB, 0x0001FFFC, 0x0001FFFD, 0x0001FFFE,
+    0x0001FFFF};
+
+/* direction: time
+   contents : codeword lengths
+   raw table: HuffCode3C.m/envelopePan_00T.mat/v_nLhex
+   built by : FH 01-05-15 */
+
+const UCHAR bookSbrEnvBalanceL10T[49] = {
+    0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10,
+    0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x0C, 0x0B,
+    0x09, 0x07, 0x05, 0x03, 0x01, 0x02, 0x04, 0x06, 0x08, 0x0B,
+    0x0C, 0x0F, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x11,
+    0x11, 0x11, 0x11, 0x11, 0x11, 0x11, 0x11, 0x11, 0x11};
+
+/* direction: freq
+   contents : codewords
+   raw table: HuffCode3C.m/envelopePan_00F.mat/v_nBhex
+   built by : FH 01-05-15 */
+
+const INT bookSbrEnvBalanceC10F[49] = {
+    0x0003FFE2, 0x0003FFE3, 0x0003FFE4, 0x0003FFE5, 0x0003FFE6, 0x0003FFE7,
+    0x0003FFE8, 0x0003FFE9, 0x0003FFEA, 0x0003FFEB, 0x0003FFEC, 0x0003FFED,
+    0x0003FFEE, 0x0003FFEF, 0x0003FFF0, 0x0000FFF7, 0x0001FFF0, 0x00003FFC,
+    0x000007FE, 0x000007FC, 0x000000FE, 0x0000007E, 0x0000000E, 0x00000002,
+    0x00000000, 0x00000006, 0x0000001E, 0x0000003E, 0x000001FE, 0x000007FD,
+    0x00000FFE, 0x00007FFA, 0x0000FFF6, 0x0003FFF1, 0x0003FFF2, 0x0003FFF3,
+    0x0003FFF4, 0x0003FFF5, 0x0003FFF6, 0x0003FFF7, 0x0003FFF8, 0x0003FFF9,
+    0x0003FFFA, 0x0003FFFB, 0x0003FFFC, 0x0003FFFD, 0x0003FFFE, 0x0007FFFE,
+    0x0007FFFF};
+
+/* direction: freq
+   contents : codeword lengths
+   raw table: HuffCode3C.m/envelopePan_00F.mat/v_nLhex
+   built by : FH 01-05-15 */
+
+const UCHAR bookSbrEnvBalanceL10F[49] = {
+    0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12,
+    0x12, 0x12, 0x12, 0x12, 0x12, 0x10, 0x11, 0x0E, 0x0B, 0x0B,
+    0x08, 0x07, 0x04, 0x02, 0x01, 0x03, 0x05, 0x06, 0x09, 0x0B,
+    0x0C, 0x0F, 0x10, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12,
+    0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x13, 0x13};
+
+/*******************************************************************************/
+/* table       : envelope level, 3.0 dB */
+/* theor range : [-29,29], CODE_BOOK_SCF_LAV   = 29 */
+/* implem range: [-31,31], CODE_BOOK_SCF_LAV11 = 31 */
+/* raw stats   : envelopeLevel_11  KK 00-02-03 */
+/*******************************************************************************/
+
+/* direction: time
+   contents : codewords
+   raw table: HuffCode2.m
+   built by : FH 00-02-04 */
+
+const INT v_Huff_envelopeLevelC11T[63] = {
+    0x0003FFED, 0x0003FFEE, 0x0007FFDE, 0x0007FFDF, 0x0007FFE0, 0x0007FFE1,
+    0x0007FFE2, 0x0007FFE3, 0x0007FFE4, 0x0007FFE5, 0x0007FFE6, 0x0007FFE7,
+    0x0007FFE8, 0x0007FFE9, 0x0007FFEA, 0x0007FFEB, 0x0007FFEC, 0x0001FFF4,
+    0x0000FFF7, 0x0000FFF9, 0x0000FFF8, 0x00003FFB, 0x00003FFA, 0x00003FF8,
+    0x00001FFA, 0x00000FFC, 0x000007FC, 0x000000FE, 0x0000003E, 0x0000000E,
+    0x00000002, 0x00000000, 0x00000006, 0x0000001E, 0x0000007E, 0x000001FE,
+    0x000007FD, 0x00001FFB, 0x00003FF9, 0x00003FFC, 0x00007FFA, 0x0000FFF6,
+    0x0001FFF5, 0x0003FFEC, 0x0007FFED, 0x0007FFEE, 0x0007FFEF, 0x0007FFF0,
+    0x0007FFF1, 0x0007FFF2, 0x0007FFF3, 0x0007FFF4, 0x0007FFF5, 0x0007FFF6,
+    0x0007FFF7, 0x0007FFF8, 0x0007FFF9, 0x0007FFFA, 0x0007FFFB, 0x0007FFFC,
+    0x0007FFFD, 0x0007FFFE, 0x0007FFFF};
+
+/* direction: time
+   contents : codeword lengths
+   raw table: HuffCode2.m
+   built by : FH 00-02-04 */
+
+const UCHAR v_Huff_envelopeLevelL11T[63] = {
+    0x12, 0x12, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13,
+    0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x11, 0x10, 0x10, 0x10, 0x0E,
+    0x0E, 0x0E, 0x0D, 0x0C, 0x0B, 0x08, 0x06, 0x04, 0x02, 0x01, 0x03,
+    0x05, 0x07, 0x09, 0x0B, 0x0D, 0x0E, 0x0E, 0x0F, 0x10, 0x11, 0x12,
+    0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13,
+    0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13};
+
+/* direction: freq
+   contents : codewords
+   raw table: HuffCode2.m
+   built by : FH 00-02-04 */
+
+const INT v_Huff_envelopeLevelC11F[63] = {
+    0x000FFFF0, 0x000FFFF1, 0x000FFFF2, 0x000FFFF3, 0x000FFFF4, 0x000FFFF5,
+    0x000FFFF6, 0x0003FFF3, 0x0007FFF5, 0x0007FFEE, 0x0007FFEF, 0x0007FFF6,
+    0x0003FFF4, 0x0003FFF2, 0x000FFFF7, 0x0007FFF0, 0x0001FFF5, 0x0003FFF0,
+    0x0001FFF4, 0x0000FFF7, 0x0000FFF6, 0x00007FF8, 0x00003FFB, 0x00000FFD,
+    0x000007FD, 0x000003FD, 0x000001FD, 0x000000FD, 0x0000003E, 0x0000000E,
+    0x00000002, 0x00000000, 0x00000006, 0x0000001E, 0x000000FC, 0x000001FC,
+    0x000003FC, 0x000007FC, 0x00000FFC, 0x00001FFC, 0x00003FFA, 0x00007FF9,
+    0x00007FFA, 0x0000FFF8, 0x0000FFF9, 0x0001FFF6, 0x0001FFF7, 0x0003FFF5,
+    0x0003FFF6, 0x0003FFF1, 0x000FFFF8, 0x0007FFF1, 0x0007FFF2, 0x0007FFF3,
+    0x000FFFF9, 0x0007FFF7, 0x0007FFF4, 0x000FFFFA, 0x000FFFFB, 0x000FFFFC,
+    0x000FFFFD, 0x000FFFFE, 0x000FFFFF};
+
+/* direction: freq
+   contents : codeword lengths
+   raw table: HuffCode2.m
+   built by : FH 00-02-04 */
+
+const UCHAR v_Huff_envelopeLevelL11F[63] = {
+    0x14, 0x14, 0x14, 0x14, 0x14, 0x14, 0x14, 0x12, 0x13, 0x13, 0x13,
+    0x13, 0x12, 0x12, 0x14, 0x13, 0x11, 0x12, 0x11, 0x10, 0x10, 0x0F,
+    0x0E, 0x0C, 0x0B, 0x0A, 0x09, 0x08, 0x06, 0x04, 0x02, 0x01, 0x03,
+    0x05, 0x08, 0x09, 0x0A, 0x0B, 0x0C, 0x0D, 0x0E, 0x0F, 0x0F, 0x10,
+    0x10, 0x11, 0x11, 0x12, 0x12, 0x12, 0x14, 0x13, 0x13, 0x13, 0x14,
+    0x13, 0x13, 0x14, 0x14, 0x14, 0x14, 0x14, 0x14};
+
+/*******************************************************************************/
+/* table       : envelope balance, 3.0 dB */
+/* theor range : [-24,24], CODE_BOOK_SCF_LAV = 24 */
+/* implem range: same but mapped to [-12,12], CODE_BOOK_SCF_LAV_BALANCE11 = 12
+ */
+/* raw stats   : envelopeBalance_11  KK 00-02-03 */
+/*******************************************************************************/
+
+/* direction: time
+   contents : codewords
+   raw table: HuffCode3C.m/envelopeBalance_11T.mat/v_nBhex
+   built by : FH 01-05-15 */
+
+const INT bookSbrEnvBalanceC11T[25] = {
+    0x00001FF2, 0x00001FF3, 0x00001FF4, 0x00001FF5, 0x00001FF6,
+    0x00001FF7, 0x00001FF8, 0x00000FF8, 0x000000FE, 0x0000007E,
+    0x0000000E, 0x00000006, 0x00000000, 0x00000002, 0x0000001E,
+    0x0000003E, 0x000001FE, 0x00001FF9, 0x00001FFA, 0x00001FFB,
+    0x00001FFC, 0x00001FFD, 0x00001FFE, 0x00003FFE, 0x00003FFF};
+
+/* direction: time
+   contents : codeword lengths
+   raw table: HuffCode3C.m/envelopeBalance_11T.mat/v_nLhex
+   built by : FH 01-05-15 */
+
+const UCHAR bookSbrEnvBalanceL11T[25] = {
+    0x0D, 0x0D, 0x0D, 0x0D, 0x0D, 0x0D, 0x0D, 0x0C, 0x08,
+    0x07, 0x04, 0x03, 0x01, 0x02, 0x05, 0x06, 0x09, 0x0D,
+    0x0D, 0x0D, 0x0D, 0x0D, 0x0D, 0x0E, 0x0E};
+
+/* direction: freq
+   contents : codewords
+   raw table: HuffCode3C.m/envelopeBalance_11F.mat/v_nBhex
+   built by : FH 01-05-15 */
+
+const INT bookSbrEnvBalanceC11F[25] = {
+    0x00001FF7, 0x00001FF8, 0x00001FF9, 0x00001FFA, 0x00001FFB,
+    0x00003FF8, 0x00003FF9, 0x000007FC, 0x000000FE, 0x0000007E,
+    0x0000000E, 0x00000002, 0x00000000, 0x00000006, 0x0000001E,
+    0x0000003E, 0x000001FE, 0x00000FFA, 0x00001FF6, 0x00003FFA,
+    0x00003FFB, 0x00003FFC, 0x00003FFD, 0x00003FFE, 0x00003FFF};
+
+/* direction: freq
+   contents : codeword lengths
+   raw table: HuffCode3C.m/envelopeBalance_11F.mat/v_nLhex
+   built by : FH 01-05-15 */
+
+const UCHAR bookSbrEnvBalanceL11F[25] = {
+    0x0D, 0x0D, 0x0D, 0x0D, 0x0D, 0x0E, 0x0E, 0x0B, 0x08,
+    0x07, 0x04, 0x02, 0x01, 0x03, 0x05, 0x06, 0x09, 0x0C,
+    0x0D, 0x0E, 0x0E, 0x0E, 0x0E, 0x0E, 0x0E};
+
+/*******************************************************************************/
+/* table       : noise level, 3.0 dB */
+/* theor range : [-29,29], CODE_BOOK_SCF_LAV   = 29 */
+/* implem range: [-31,31], CODE_BOOK_SCF_LAV11 = 31 */
+/* raw stats   : noiseLevel_11  KK 00-02-03 */
+/*******************************************************************************/
+
+/* direction: time
+   contents : codewords
+   raw table: HuffCode2.m
+   built by : FH 00-02-04 */
+
+const INT v_Huff_NoiseLevelC11T[63] = {
+    0x00001FCE, 0x00001FCF, 0x00001FD0, 0x00001FD1, 0x00001FD2, 0x00001FD3,
+    0x00001FD4, 0x00001FD5, 0x00001FD6, 0x00001FD7, 0x00001FD8, 0x00001FD9,
+    0x00001FDA, 0x00001FDB, 0x00001FDC, 0x00001FDD, 0x00001FDE, 0x00001FDF,
+    0x00001FE0, 0x00001FE1, 0x00001FE2, 0x00001FE3, 0x00001FE4, 0x00001FE5,
+    0x00001FE6, 0x00001FE7, 0x000007F2, 0x000000FD, 0x0000003E, 0x0000000E,
+    0x00000006, 0x00000000, 0x00000002, 0x0000001E, 0x000000FC, 0x000003F8,
+    0x00001FCC, 0x00001FE8, 0x00001FE9, 0x00001FEA, 0x00001FEB, 0x00001FEC,
+    0x00001FCD, 0x00001FED, 0x00001FEE, 0x00001FEF, 0x00001FF0, 0x00001FF1,
+    0x00001FF2, 0x00001FF3, 0x00001FF4, 0x00001FF5, 0x00001FF6, 0x00001FF7,
+    0x00001FF8, 0x00001FF9, 0x00001FFA, 0x00001FFB, 0x00001FFC, 0x00001FFD,
+    0x00001FFE, 0x00003FFE, 0x00003FFF};
+
+/* direction: time
+   contents : codeword lengths
+   raw table: HuffCode2.m
+   built by : FH 00-02-04 */
+
+const UCHAR v_Huff_NoiseLevelL11T[63] = {
+    0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D,
+    0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D,
+    0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D,
+    0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D,
+    0x0000000D, 0x0000000D, 0x0000000B, 0x00000008, 0x00000006, 0x00000004,
+    0x00000003, 0x00000001, 0x00000002, 0x00000005, 0x00000008, 0x0000000A,
+    0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D,
+    0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D,
+    0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D,
+    0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D,
+    0x0000000D, 0x0000000E, 0x0000000E};
+
+/*******************************************************************************/
+/* table       : noise balance, 3.0 dB */
+/* theor range : [-24,24], CODE_BOOK_SCF_LAV = 24 */
+/* implem range: same but mapped to [-12,12], CODE_BOOK_SCF_LAV_BALANCE11 = 12
+ */
+/* raw stats   : noiseBalance_11  KK 00-02-03 */
+/*******************************************************************************/
+
+/* direction: time
+   contents : codewords
+   raw table: HuffCode3C.m/noiseBalance_11.mat/v_nBhex
+   built by : FH 01-05-15 */
+
+const INT bookSbrNoiseBalanceC11T[25] = {
+    0x000000EC, 0x000000ED, 0x000000EE, 0x000000EF, 0x000000F0,
+    0x000000F1, 0x000000F2, 0x000000F3, 0x000000F4, 0x000000F5,
+    0x0000001C, 0x00000002, 0x00000000, 0x00000006, 0x0000003A,
+    0x000000F6, 0x000000F7, 0x000000F8, 0x000000F9, 0x000000FA,
+    0x000000FB, 0x000000FC, 0x000000FD, 0x000000FE, 0x000000FF};
+
+/* direction: time
+   contents : codeword lengths
+   raw table: HuffCode3C.m/noiseBalance_11.mat/v_nLhex
+   built by : FH 01-05-15 */
+
+const UCHAR bookSbrNoiseBalanceL11T[25] = {
+    0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08,
+    0x08, 0x05, 0x02, 0x01, 0x03, 0x06, 0x08, 0x08, 0x08,
+    0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08};
+
+/*
+   tuningTable
+*/
+const sbrTuningTable_t sbrTuningTable[] = {
+    /* Some of the low bitrates are commented out here, this is because the
+       encoder could lose frames at those bitrates and throw an error
+       because it has insufficient bits to encode for some test items.
+    */
+
+    /*** HE-AAC section ***/
+    /*                        sf,sfsp,sf,sfsp,nnb,nfo,saml,SM,FS*/
+
+    /*** mono ***/
+
+    /* 8/16 kHz dual rate */
+    {CODEC_AAC, 8000, 10000, 8000, 1, 7, 6, 11, 10, 1, 0, 6, SBR_MONO, 3},
+    {CODEC_AAC, 10000, 12000, 8000, 1, 11, 7, 13, 12, 1, 0, 6, SBR_MONO, 3},
+    {CODEC_AAC, 12000, 16001, 8000, 1, 14, 10, 13, 13, 1, 0, 6, SBR_MONO, 3},
+    {CODEC_AAC, 16000, 24000, 8000, 1, 14, 10, 14, 14, 2, 0, 3, SBR_MONO, 2},
+    {CODEC_AAC, 24000, 32000, 8000, 1, 14, 10, 14, 14, 2, 0, 3, SBR_MONO, 2},
+    {CODEC_AAC, 32000, 48001, 8000, 1, 14, 11, 15, 15, 2, 0, 3, SBR_MONO, 2},
+
+    /* 11/22 kHz dual rate */
+    {CODEC_AAC, 8000, 10000, 11025, 1, 5, 4, 6, 6, 1, 0, 6, SBR_MONO, 3},
+    {CODEC_AAC, 10000, 12000, 11025, 1, 8, 5, 12, 9, 1, 0, 6, SBR_MONO, 3},
+    {CODEC_AAC, 12000, 16000, 11025, 1, 12, 8, 13, 8, 1, 0, 6, SBR_MONO, 3},
+    {CODEC_AAC, 16000, 20000, 11025, 1, 12, 8, 13, 8, 1, 0, 6, SBR_MONO, 3},
+    {CODEC_AAC, 20000, 24001, 11025, 1, 13, 9, 13, 8, 1, 0, 6, SBR_MONO, 3},
+    {CODEC_AAC, 24000, 32000, 11025, 1, 14, 10, 14, 9, 2, 0, 3, SBR_MONO, 2},
+    {CODEC_AAC, 32000, 48000, 11025, 1, 15, 11, 15, 10, 2, 0, 3, SBR_MONO, 2},
+    {CODEC_AAC, 48000, 64001, 11025, 1, 15, 11, 15, 10, 2, 0, 3, SBR_MONO, 1},
+
+    /* 12/24 kHz dual rate */
+    {CODEC_AAC, 8000, 10000, 12000, 1, 4, 3, 6, 6, 1, 0, 6, SBR_MONO, 3},
+    {CODEC_AAC, 10000, 12000, 12000, 1, 7, 4, 11, 8, 1, 0, 6, SBR_MONO, 3},
+    {CODEC_AAC, 12000, 16000, 12000, 1, 11, 7, 12, 8, 1, 0, 6, SBR_MONO, 3},
+    {CODEC_AAC, 16000, 20000, 12000, 1, 11, 7, 12, 8, 1, 0, 6, SBR_MONO, 3},
+    {CODEC_AAC, 20000, 24001, 12000, 1, 12, 8, 12, 8, 1, 0, 6, SBR_MONO, 3},
+    {CODEC_AAC, 24000, 32000, 12000, 1, 13, 9, 13, 9, 2, 0, 3, SBR_MONO, 2},
+    {CODEC_AAC, 32000, 48000, 12000, 1, 14, 10, 14, 10, 2, 0, 3, SBR_MONO, 2},
+    {CODEC_AAC, 48000, 64001, 12000, 1, 14, 11, 15, 11, 2, 0, 3, SBR_MONO, 1},
+
+    /* 16/32 kHz dual rate */
+    {CODEC_AAC, 8000, 10000, 16000, 1, 1, 1, 0, 0, 1, 0, 6, SBR_MONO, 3},
+    {CODEC_AAC, 10000, 12000, 16000, 1, 2, 1, 6, 0, 1, 0, 6, SBR_MONO, 3},
+    {CODEC_AAC, 12000, 16000, 16000, 1, 4, 2, 6, 0, 1, 0, 6, SBR_MONO, 3},
+    {CODEC_AAC, 16000, 18000, 16000, 1, 4, 2, 8, 3, 1, 0, 6, SBR_MONO, 3},
+    {CODEC_AAC, 18000, 22000, 16000, 1, 6, 5, 11, 7, 2, 0, 6, SBR_MONO, 2},
+    {CODEC_AAC, 22000, 28000, 16000, 1, 10, 9, 12, 8, 2, 0, 6, SBR_MONO, 2},
+    {CODEC_AAC, 28000, 36000, 16000, 1, 12, 12, 13, 13, 2, 0, 3, SBR_MONO, 2},
+    {CODEC_AAC, 36000, 44000, 16000, 1, 14, 14, 13, 13, 2, 0, 3, SBR_MONO, 1},
+    {CODEC_AAC, 44000, 64001, 16000, 1, 14, 14, 13, 13, 2, 0, 3, SBR_MONO, 1},
+
+    /* 22.05/44.1 kHz dual rate */
+    /* { CODEC_AAC,   8000, 11369,  22050, 1,  1, 1, 1, 1,  1, 0, 6,
+       SBR_MONO, 3 }, */
+    {CODEC_AAC, 11369, 16000, 22050, 1, 3, 1, 4, 4, 1, 0, 6, SBR_MONO, 3},
+    {CODEC_AAC, 16000, 18000, 22050, 1, 3, 1, 5, 4, 1, 0, 6, SBR_MONO, 3},
+    {CODEC_AAC, 18000, 22000, 22050, 1, 4, 4, 8, 5, 2, 0, 6, SBR_MONO, 2},
+    {CODEC_AAC, 22000, 28000, 22050, 1, 7, 6, 8, 6, 2, 0, 6, SBR_MONO, 2},
+    {CODEC_AAC, 28000, 36000, 22050, 1, 10, 10, 9, 9, 2, 0, 3, SBR_MONO, 2},
+    {CODEC_AAC, 36000, 44000, 22050, 1, 11, 11, 10, 10, 2, 0, 3, SBR_MONO, 1},
+    {CODEC_AAC, 44000, 64001, 22050, 1, 13, 13, 12, 12, 2, 0, 3, SBR_MONO, 1},
+
+    /* 24/48 kHz dual rate */
+    /* { CODEC_AAC,   8000, 12000,  24000, 1,  1, 1, 1, 1,  1, 0, 6,
+       SBR_MONO, 3 }, */
+    {CODEC_AAC, 12000, 16000, 24000, 1, 3, 1, 4, 4, 1, 0, 6, SBR_MONO, 3},
+    {CODEC_AAC, 16000, 18000, 24000, 1, 3, 1, 5, 4, 1, 0, 6, SBR_MONO, 3},
+    {CODEC_AAC, 18000, 22000, 24000, 1, 4, 3, 8, 5, 2, 0, 6, SBR_MONO, 2},
+    {CODEC_AAC, 22000, 28000, 24000, 1, 7, 6, 8, 6, 2, 0, 6, SBR_MONO, 2},
+    {CODEC_AAC, 28000, 36000, 24000, 1, 10, 10, 9, 9, 2, 0, 3, SBR_MONO, 2},
+    {CODEC_AAC, 36000, 44000, 24000, 1, 11, 11, 10, 10, 2, 0, 3, SBR_MONO, 1},
+    {CODEC_AAC, 44000, 64001, 24000, 1, 13, 13, 11, 11, 2, 0, 3, SBR_MONO, 1},
+
+    /* 32/64 kHz dual rate */
+    {CODEC_AAC, 24000, 36000, 32000, 1, 4, 4, 4, 4, 2, 0, 3, SBR_MONO, 3},
+    {CODEC_AAC, 36000, 60000, 32000, 1, 7, 7, 6, 6, 2, 0, 3, SBR_MONO, 2},
+    {CODEC_AAC, 60000, 72000, 32000, 1, 9, 9, 8, 8, 2, 0, 3, SBR_MONO, 1},
+    {CODEC_AAC, 72000, 100000, 32000, 1, 11, 11, 10, 10, 2, 0, 3, SBR_MONO, 1},
+    {CODEC_AAC, 100000, 160001, 32000, 1, 13, 13, 11, 11, 2, 0, 3, SBR_MONO, 1},
+
+    /* 44.1/88.2 kHz dual rate */
+    {CODEC_AAC, 24000, 36000, 44100, 1, 4, 4, 4, 4, 2, 0, 3, SBR_MONO, 3},
+    {CODEC_AAC, 36000, 60000, 44100, 1, 7, 7, 6, 6, 2, 0, 3, SBR_MONO, 2},
+    {CODEC_AAC, 60000, 72000, 44100, 1, 9, 9, 8, 8, 2, 0, 3, SBR_MONO, 1},
+    {CODEC_AAC, 72000, 100000, 44100, 1, 11, 11, 10, 10, 2, 0, 3, SBR_MONO, 1},
+    {CODEC_AAC, 100000, 160001, 44100, 1, 13, 13, 11, 11, 2, 0, 3, SBR_MONO, 1},
+
+    /* 48/96 kHz dual rate */
+    {CODEC_AAC, 32000, 36000, 48000, 1, 4, 4, 9, 9, 2, 0, 3, SBR_MONO, 3},
+    {CODEC_AAC, 36000, 60000, 48000, 1, 7, 7, 10, 10, 2, 0, 3, SBR_MONO, 2},
+    {CODEC_AAC, 60000, 72000, 48000, 1, 9, 9, 10, 10, 2, 0, 3, SBR_MONO, 1},
+    {CODEC_AAC, 72000, 100000, 48000, 1, 11, 11, 11, 11, 2, 0, 3, SBR_MONO, 1},
+    {CODEC_AAC, 100000, 160001, 48000, 1, 13, 13, 11, 11, 2, 0, 3, SBR_MONO, 1},
+
+    /*** stereo ***/
+    /* 08/16 kHz dual rate */
+    {CODEC_AAC, 16000, 24000, 8000, 2, 6, 6, 9, 7, 1, 0, -3, SBR_SWITCH_LRC, 3},
+    {CODEC_AAC, 24000, 28000, 8000, 2, 9, 9, 11, 9, 1, 0, -3, SBR_SWITCH_LRC,
+     3},
+    {CODEC_AAC, 28000, 36000, 8000, 2, 11, 9, 11, 9, 2, 0, -3, SBR_SWITCH_LRC,
+     2},
+    {CODEC_AAC, 36000, 44000, 8000, 2, 13, 11, 13, 11, 2, 0, -3, SBR_SWITCH_LRC,
+     2},
+    {CODEC_AAC, 44000, 52000, 8000, 2, 14, 12, 13, 12, 2, 0, -3, SBR_SWITCH_LRC,
+     2},
+    {CODEC_AAC, 52000, 60000, 8000, 2, 14, 14, 13, 13, 3, 0, -3, SBR_SWITCH_LRC,
+     1},
+    {CODEC_AAC, 60000, 76000, 8000, 2, 14, 14, 13, 13, 3, 0, -3, SBR_LEFT_RIGHT,
+     1},
+    {CODEC_AAC, 76000, 128001, 8000, 2, 14, 14, 13, 13, 3, 0, -3,
+     SBR_LEFT_RIGHT, 1},
+
+    /* 11/22 kHz dual rate */
+    {CODEC_AAC, 16000, 24000, 11025, 2, 7, 5, 9, 7, 1, 0, -3, SBR_SWITCH_LRC,
+     3},
+    {CODEC_AAC, 24000, 28000, 11025, 2, 10, 8, 10, 8, 1, 0, -3, SBR_SWITCH_LRC,
+     3},
+    {CODEC_AAC, 28000, 36000, 11025, 2, 12, 8, 12, 8, 2, 0, -3, SBR_SWITCH_LRC,
+     2},
+    {CODEC_AAC, 36000, 44000, 11025, 2, 13, 9, 13, 9, 2, 0, -3, SBR_SWITCH_LRC,
+     2},
+    {CODEC_AAC, 44000, 52000, 11025, 2, 14, 11, 13, 11, 2, 0, -3,
+     SBR_SWITCH_LRC, 2},
+    {CODEC_AAC, 52000, 60000, 11025, 2, 15, 15, 13, 13, 3, 0, -3,
+     SBR_SWITCH_LRC, 1},
+    {CODEC_AAC, 60000, 76000, 11025, 2, 15, 15, 13, 13, 3, 0, -3,
+     SBR_LEFT_RIGHT, 1},
+    {CODEC_AAC, 76000, 128001, 11025, 2, 15, 15, 13, 13, 3, 0, -3,
+     SBR_LEFT_RIGHT, 1},
+
+    /* 12/24 kHz dual rate */
+    {CODEC_AAC, 16000, 24000, 12000, 2, 6, 4, 9, 7, 1, 0, -3, SBR_SWITCH_LRC,
+     3},
+    {CODEC_AAC, 24000, 28000, 12000, 2, 9, 7, 10, 8, 1, 0, -3, SBR_SWITCH_LRC,
+     3},
+    {CODEC_AAC, 28000, 36000, 12000, 2, 11, 7, 12, 8, 2, 0, -3, SBR_SWITCH_LRC,
+     2},
+    {CODEC_AAC, 36000, 44000, 12000, 2, 12, 9, 12, 9, 2, 0, -3, SBR_SWITCH_LRC,
+     2},
+    {CODEC_AAC, 44000, 52000, 12000, 2, 13, 12, 13, 12, 2, 0, -3,
+     SBR_SWITCH_LRC, 2},
+    {CODEC_AAC, 52000, 60000, 12000, 2, 14, 14, 13, 13, 3, 0, -3,
+     SBR_SWITCH_LRC, 1},
+    {CODEC_AAC, 60000, 76000, 12000, 2, 14, 14, 13, 13, 3, 0, -3,
+     SBR_LEFT_RIGHT, 1},
+    {CODEC_AAC, 76000, 128001, 12000, 2, 14, 14, 13, 13, 3, 0, -3,
+     SBR_LEFT_RIGHT, 1},
+
+    /* 16/32 kHz dual rate */
+    {CODEC_AAC, 16000, 24000, 16000, 2, 4, 2, 1, 0, 1, 0, -3, SBR_SWITCH_LRC,
+     3},
+    {CODEC_AAC, 24000, 28000, 16000, 2, 8, 7, 10, 8, 1, 0, -3, SBR_SWITCH_LRC,
+     3},
+    {CODEC_AAC, 28000, 36000, 16000, 2, 10, 9, 12, 11, 2, 0, -3, SBR_SWITCH_LRC,
+     2},
+    {CODEC_AAC, 36000, 44000, 16000, 2, 13, 13, 13, 13, 2, 0, -3,
+     SBR_SWITCH_LRC, 2},
+    {CODEC_AAC, 44000, 52000, 16000, 2, 14, 14, 13, 13, 2, 0, -3,
+     SBR_SWITCH_LRC, 2},
+    {CODEC_AAC, 52000, 60000, 16000, 2, 14, 14, 13, 13, 3, 0, -3,
+     SBR_SWITCH_LRC, 1},
+    {CODEC_AAC, 60000, 76000, 16000, 2, 14, 14, 13, 13, 3, 0, -3,
+     SBR_LEFT_RIGHT, 1},
+    {CODEC_AAC, 76000, 128001, 16000, 2, 14, 14, 13, 13, 3, 0, -3,
+     SBR_LEFT_RIGHT, 1},
+
+    /* 22.05/44.1 kHz dual rate */
+    {CODEC_AAC, 16000, 24000, 22050, 2, 2, 1, 1, 0, 1, 0, -3, SBR_SWITCH_LRC,
+     3},
+    {CODEC_AAC, 24000, 28000, 22050, 2, 5, 4, 6, 5, 1, 0, -3, SBR_SWITCH_LRC,
+     3},
+    {CODEC_AAC, 28000, 32000, 22050, 2, 5, 4, 8, 7, 2, 0, -3, SBR_SWITCH_LRC,
+     2},
+    {CODEC_AAC, 32000, 36000, 22050, 2, 7, 6, 8, 7, 2, 0, -3, SBR_SWITCH_LRC,
+     2},
+    {CODEC_AAC, 36000, 44000, 22050, 2, 10, 10, 9, 9, 2, 0, -3, SBR_SWITCH_LRC,
+     2},
+    {CODEC_AAC, 44000, 52000, 22050, 2, 12, 12, 9, 9, 3, 0, -3, SBR_SWITCH_LRC,
+     2},
+    {CODEC_AAC, 52000, 60000, 22050, 2, 13, 13, 10, 10, 3, 0, -3,
+     SBR_SWITCH_LRC, 1},
+    {CODEC_AAC, 60000, 76000, 22050, 2, 14, 14, 12, 12, 3, 0, -3,
+     SBR_LEFT_RIGHT, 1},
+    {CODEC_AAC, 76000, 128001, 22050, 2, 14, 14, 12, 12, 3, 0, -3,
+     SBR_LEFT_RIGHT, 1},
+
+    /* 24/48 kHz dual rate */
+    {CODEC_AAC, 16000, 24000, 24000, 2, 2, 1, 1, 0, 1, 0, -3, SBR_SWITCH_LRC,
+     3},
+    {CODEC_AAC, 24000, 28000, 24000, 2, 5, 5, 6, 6, 1, 0, -3, SBR_SWITCH_LRC,
+     3},
+    {CODEC_AAC, 28000, 36000, 24000, 2, 7, 6, 8, 7, 2, 0, -3, SBR_SWITCH_LRC,
+     2},
+    {CODEC_AAC, 36000, 44000, 24000, 2, 10, 10, 9, 9, 2, 0, -3, SBR_SWITCH_LRC,
+     2},
+    {CODEC_AAC, 44000, 52000, 24000, 2, 12, 12, 9, 9, 3, 0, -3, SBR_SWITCH_LRC,
+     2},
+    {CODEC_AAC, 52000, 60000, 24000, 2, 13, 13, 10, 10, 3, 0, -3,
+     SBR_SWITCH_LRC, 1},
+    {CODEC_AAC, 60000, 76000, 24000, 2, 14, 14, 12, 12, 3, 0, -3,
+     SBR_LEFT_RIGHT, 1},
+    {CODEC_AAC, 76000, 128001, 24000, 2, 14, 14, 12, 12, 3, 0, -3,
+     SBR_LEFT_RIGHT, 1},
+
+    /* 32/64 kHz dual rate */
+    {CODEC_AAC, 32000, 60000, 32000, 2, 4, 4, 4, 4, 2, 0, -3, SBR_SWITCH_LRC,
+     3},
+    {CODEC_AAC, 60000, 80000, 32000, 2, 7, 7, 6, 6, 3, 0, -3, SBR_SWITCH_LRC,
+     2},
+    {CODEC_AAC, 80000, 112000, 32000, 2, 9, 9, 8, 8, 3, 0, -3, SBR_LEFT_RIGHT,
+     1},
+    {CODEC_AAC, 112000, 144000, 32000, 2, 11, 11, 10, 10, 3, 0, -3,
+     SBR_LEFT_RIGHT, 1},
+    {CODEC_AAC, 144000, 256001, 32000, 2, 13, 13, 11, 11, 3, 0, -3,
+     SBR_LEFT_RIGHT, 1},
+
+    /* 44.1/88.2 kHz dual rate */
+    {CODEC_AAC, 32000, 60000, 44100, 2, 4, 4, 4, 4, 2, 0, -3, SBR_SWITCH_LRC,
+     3},
+    {CODEC_AAC, 60000, 80000, 44100, 2, 7, 7, 6, 6, 3, 0, -3, SBR_SWITCH_LRC,
+     2},
+    {CODEC_AAC, 80000, 112000, 44100, 2, 9, 9, 8, 8, 3, 0, -3, SBR_LEFT_RIGHT,
+     1},
+    {CODEC_AAC, 112000, 144000, 44100, 2, 11, 11, 10, 10, 3, 0, -3,
+     SBR_LEFT_RIGHT, 1},
+    {CODEC_AAC, 144000, 256001, 44100, 2, 13, 13, 11, 11, 3, 0, -3,
+     SBR_LEFT_RIGHT, 1},
+
+    /* 48/96 kHz dual rate */
+    {CODEC_AAC, 36000, 60000, 48000, 2, 4, 4, 9, 9, 2, 0, -3, SBR_SWITCH_LRC,
+     3},
+    {CODEC_AAC, 60000, 80000, 48000, 2, 7, 7, 9, 9, 3, 0, -3, SBR_SWITCH_LRC,
+     2},
+    {CODEC_AAC, 80000, 112000, 48000, 2, 9, 9, 10, 10, 3, 0, -3, SBR_LEFT_RIGHT,
+     1},
+    {CODEC_AAC, 112000, 144000, 48000, 2, 11, 11, 11, 11, 3, 0, -3,
+     SBR_LEFT_RIGHT, 1},
+    {CODEC_AAC, 144000, 256001, 48000, 2, 13, 13, 11, 11, 3, 0, -3,
+     SBR_LEFT_RIGHT, 1},
+
+    /** AAC LOW DELAY SECTION **/
+
+    /* 24 kHz dual rate - 12kHz singlerate is not allowed (deactivated in
+       FDKsbrEnc_IsSbrSettingAvail()) */
+    {CODEC_AACLD, 8000, 32000, 12000, 1, 1, 1, 0, 0, 1, 0, 6, SBR_MONO, 3},
+
+    /*** mono ***/
+    /* 16/32 kHz dual rate */
+    {CODEC_AACLD, 16000, 18000, 16000, 1, 4, 5, 9, 7, 1, 0, 6, SBR_MONO, 3},
+    {CODEC_AACLD, 18000, 22000, 16000, 1, 7, 7, 12, 12, 1, 6, 9, SBR_MONO, 3},
+    {CODEC_AACLD, 22000, 28000, 16000, 1, 6, 6, 9, 9, 2, 3, 6, SBR_MONO, 3},
+    {CODEC_AACLD, 28000, 36000, 16000, 1, 8, 8, 12, 7, 2, 9, 12, SBR_MONO, 3},
+    {CODEC_AACLD, 36000, 44000, 16000, 1, 10, 14, 12, 13, 2, 0, 3, SBR_MONO, 1},
+    {CODEC_AACLD, 44000, 64001, 16000, 1, 11, 14, 13, 13, 2, 0, 3, SBR_MONO, 1},
+
+    /* 22.05/44.1 kHz dual rate */
+    {CODEC_AACLD, 18000, 22000, 22050, 1, 4, 4, 5, 5, 2, 0, 6, SBR_MONO, 3},
+    {CODEC_AACLD, 22000, 28000, 22050, 1, 5, 5, 6, 6, 2, 0, 6, SBR_MONO, 2},
+    {CODEC_AACLD, 28000, 36000, 22050, 1, 7, 8, 8, 8, 2, 0, 3, SBR_MONO, 2},
+    {CODEC_AACLD, 36000, 44000, 22050, 1, 9, 9, 9, 9, 2, 0, 3, SBR_MONO, 1},
+    {CODEC_AACLD, 44000, 52000, 22050, 1, 12, 11, 11, 11, 2, 0, 3, SBR_MONO, 1},
+    {CODEC_AACLD, 52000, 64001, 22050, 1, 13, 11, 11, 10, 2, 0, 3, SBR_MONO, 1},
+
+    /* 24/48 kHz dual rate */
+    {CODEC_AACLD, 20000, 22000, 24000, 1, 3, 4, 8, 8, 2, 0, 6, SBR_MONO, 2},
+    {CODEC_AACLD, 22000, 28000, 24000, 1, 3, 8, 8, 7, 2, 0, 3, SBR_MONO, 2},
+    {CODEC_AACLD, 28000, 36000, 24000, 1, 4, 8, 8, 7, 2, 0, 3, SBR_MONO, 2},
+    {CODEC_AACLD, 36000, 56000, 24000, 1, 8, 9, 9, 8, 2, 0, 3, SBR_MONO, 1},
+    {CODEC_AACLD, 56000, 64001, 24000, 1, 13, 11, 11, 10, 2, 0, 3, SBR_MONO, 1},
+
+    /* 32/64 kHz dual rate */
+    {CODEC_AACLD, 24000, 36000, 32000, 1, 4, 4, 4, 4, 2, 0, 3, SBR_MONO, 3},
+    {CODEC_AACLD, 36000, 60000, 32000, 1, 7, 7, 6, 6, 2, 0, 3, SBR_MONO, 2},
+    {CODEC_AACLD, 60000, 72000, 32000, 1, 9, 9, 8, 8, 2, 0, 3, SBR_MONO, 1},
+    {CODEC_AACLD, 72000, 100000, 32000, 1, 11, 11, 10, 10, 2, 0, 3, SBR_MONO,
+     1},
+    {CODEC_AACLD, 100000, 160001, 32000, 1, 13, 13, 11, 11, 2, 0, 3, SBR_MONO,
+     1},
+
+    /* 44/88 kHz dual rate */
+    {CODEC_AACLD, 36000, 60000, 44100, 1, 8, 7, 6, 9, 2, 0, 3, SBR_MONO, 2},
+    {CODEC_AACLD, 60000, 72000, 44100, 1, 9, 9, 10, 10, 2, 0, 3, SBR_MONO, 1},
+    {CODEC_AACLD, 72000, 100000, 44100, 1, 11, 11, 11, 11, 2, 0, 3, SBR_MONO,
+     1},
+    {CODEC_AACLD, 100000, 160001, 44100, 1, 13, 13, 11, 11, 2, 0, 3, SBR_MONO,
+     1},
+
+    /* 48/96 kHz dual rate */ /* 32 and 40kbps line tuned for dual-rate SBR
+                               */
+    {CODEC_AACLD, 36000, 60000, 48000, 1, 4, 7, 4, 4, 2, 0, 3, SBR_MONO, 3},
+    {CODEC_AACLD, 60000, 72000, 48000, 1, 9, 9, 10, 10, 2, 0, 3, SBR_MONO, 1},
+    {CODEC_AACLD, 72000, 100000, 48000, 1, 11, 11, 11, 11, 2, 0, 3, SBR_MONO,
+     1},
+    {CODEC_AACLD, 100000, 160001, 48000, 1, 13, 13, 11, 11, 2, 0, 3, SBR_MONO,
+     1},
+
+    /*** stereo ***/
+    /* 16/32 kHz dual rate */
+    {CODEC_AACLD, 32000, 36000, 16000, 2, 10, 9, 12, 11, 2, 0, -3,
+     SBR_SWITCH_LRC, 2},
+    {CODEC_AACLD, 36000, 44000, 16000, 2, 13, 13, 13, 13, 2, 0, -3,
+     SBR_SWITCH_LRC, 2},
+    {CODEC_AACLD, 44000, 52000, 16000, 2, 10, 9, 11, 9, 2, 0, -3,
+     SBR_SWITCH_LRC, 2},
+    {CODEC_AACLD, 52000, 60000, 16000, 2, 14, 14, 13, 13, 3, 0, -3,
+     SBR_SWITCH_LRC, 1},
+    {CODEC_AACLD, 60000, 76000, 16000, 2, 14, 14, 13, 13, 3, 0, -3,
+     SBR_LEFT_RIGHT, 1},
+    {CODEC_AACLD, 76000, 128001, 16000, 2, 14, 14, 13, 13, 3, 0, -3,
+     SBR_LEFT_RIGHT, 1},
+
+    /* 22.05/44.1 kHz dual rate */
+    {CODEC_AACLD, 32000, 36000, 22050, 2, 5, 4, 7, 6, 2, 0, -3, SBR_SWITCH_LRC,
+     2},
+    {CODEC_AACLD, 36000, 44000, 22050, 2, 5, 8, 8, 8, 2, 0, -3, SBR_SWITCH_LRC,
+     2},
+    {CODEC_AACLD, 44000, 52000, 22050, 2, 7, 10, 8, 8, 3, 0, -3, SBR_SWITCH_LRC,
+     2},
+    {CODEC_AACLD, 52000, 60000, 22050, 2, 9, 11, 9, 9, 3, 0, -3, SBR_SWITCH_LRC,
+     1},
+    {CODEC_AACLD, 60000, 76000, 22050, 2, 10, 12, 10, 11, 3, 0, -3,
+     SBR_LEFT_RIGHT, 1},
+    {CODEC_AACLD, 76000, 82000, 22050, 2, 12, 12, 11, 11, 3, 0, -3,
+     SBR_LEFT_RIGHT, 1},
+    {CODEC_AACLD, 82000, 128001, 22050, 2, 13, 12, 11, 11, 3, 0, -3,
+     SBR_LEFT_RIGHT, 1},
+
+    /* 24/48 kHz dual rate */
+    {CODEC_AACLD, 32000, 36000, 24000, 2, 5, 4, 7, 6, 2, 0, -3, SBR_SWITCH_LRC,
+     2},
+    {CODEC_AACLD, 36000, 44000, 24000, 2, 4, 8, 8, 8, 2, 0, -3, SBR_SWITCH_LRC,
+     2},
+    {CODEC_AACLD, 44000, 52000, 24000, 2, 6, 10, 8, 8, 3, 0, -3, SBR_SWITCH_LRC,
+     2},
+    {CODEC_AACLD, 52000, 60000, 24000, 2, 9, 11, 9, 9, 3, 0, -3, SBR_SWITCH_LRC,
+     1},
+    {CODEC_AACLD, 60000, 76000, 24000, 2, 11, 12, 10, 11, 3, 0, -3,
+     SBR_LEFT_RIGHT, 1},
+    {CODEC_AACLD, 76000, 88000, 24000, 2, 12, 13, 11, 11, 3, 0, -3,
+     SBR_LEFT_RIGHT, 1},
+    {CODEC_AACLD, 88000, 128001, 24000, 2, 13, 13, 11, 11, 3, 0, -3,
+     SBR_LEFT_RIGHT, 1},
+
+    /* 32/64 kHz dual rate */
+    {CODEC_AACLD, 60000, 80000, 32000, 2, 7, 7, 6, 6, 3, 0, -3, SBR_SWITCH_LRC,
+     2},
+    {CODEC_AACLD, 80000, 112000, 32000, 2, 9, 9, 8, 8, 3, 0, -3, SBR_LEFT_RIGHT,
+     1},
+    {CODEC_AACLD, 112000, 144000, 32000, 2, 11, 11, 10, 10, 3, 0, -3,
+     SBR_LEFT_RIGHT, 1},
+    {CODEC_AACLD, 144000, 256001, 32000, 2, 13, 13, 11, 11, 3, 0, -3,
+     SBR_LEFT_RIGHT, 1},
+
+    /* 44.1/88.2 kHz dual rate */
+    {CODEC_AACLD, 60000, 80000, 44100, 2, 7, 7, 6, 6, 3, 0, -3, SBR_SWITCH_LRC,
+     2},
+    {CODEC_AACLD, 80000, 112000, 44100, 2, 10, 10, 8, 8, 3, 0, -3,
+     SBR_LEFT_RIGHT, 1},
+    {CODEC_AACLD, 112000, 144000, 44100, 2, 12, 12, 10, 10, 3, 0, -3,
+     SBR_LEFT_RIGHT, 1},
+    {CODEC_AACLD, 144000, 256001, 44100, 2, 13, 13, 11, 11, 3, 0, -3,
+     SBR_LEFT_RIGHT, 1},
+
+    /* 48/96 kHz dual rate */
+    {CODEC_AACLD, 60000, 80000, 48000, 2, 7, 7, 10, 10, 2, 0, -3,
+     SBR_SWITCH_LRC, 2},
+    {CODEC_AACLD, 80000, 112000, 48000, 2, 9, 9, 10, 10, 3, 0, -3,
+     SBR_LEFT_RIGHT, 1},
+    {CODEC_AACLD, 112000, 144000, 48000, 2, 11, 11, 11, 11, 3, 0, -3,
+     SBR_LEFT_RIGHT, 1},
+    {CODEC_AACLD, 144000, 176000, 48000, 2, 12, 12, 11, 11, 3, 0, -3,
+     SBR_LEFT_RIGHT, 1},
+    {CODEC_AACLD, 176000, 256001, 48000, 2, 13, 13, 11, 11, 3, 0, -3,
+     SBR_LEFT_RIGHT, 1},
+
+};
+
+const int sbrTuningTableSize =
+    sizeof(sbrTuningTable) / sizeof(sbrTuningTable[0]);
+
+const psTuningTable_t psTuningTable[4] = {
+    {8000, 22000, PSENC_STEREO_BANDS_10, PSENC_NENV_1,
+     FL2FXCONST_DBL(3.0f / 4.0f)},
+    {22000, 28000, PSENC_STEREO_BANDS_20, PSENC_NENV_1,
+     FL2FXCONST_DBL(2.0f / 4.0f)},
+    {28000, 36000, PSENC_STEREO_BANDS_20, PSENC_NENV_2,
+     FL2FXCONST_DBL(1.5f / 4.0f)},
+    {36000, 160001, PSENC_STEREO_BANDS_20, PSENC_NENV_4,
+     FL2FXCONST_DBL(1.1f / 4.0f)},
+};
+
+//@}
diff --git a/fdk-aac/libSBRenc/src/sbrenc_rom.h b/fdk-aac/libSBRenc/src/sbrenc_rom.h
new file mode 100644
index 0000000..18c1fb9
--- /dev/null
+++ b/fdk-aac/libSBRenc/src/sbrenc_rom.h
@@ -0,0 +1,145 @@
+/* -----------------------------------------------------------------------------
+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 encoder library ******************************
+
+   Author(s):
+
+   Description:
+
+*******************************************************************************/
+
+/*!
+\file
+\brief Declaration of constant tables
+$Revision: 92790 $
+*/
+#ifndef SBRENC_ROM_H
+#define SBRENC_ROM_H
+
+#include "sbr_def.h"
+#include "sbr_encoder.h"
+
+#include "ps_main.h"
+
+/*
+  huffman tables
+*/
+extern const INT v_Huff_envelopeLevelC10T[121];
+extern const UCHAR v_Huff_envelopeLevelL10T[121];
+extern const INT v_Huff_envelopeLevelC10F[121];
+extern const UCHAR v_Huff_envelopeLevelL10F[121];
+extern const INT bookSbrEnvBalanceC10T[49];
+extern const UCHAR bookSbrEnvBalanceL10T[49];
+extern const INT bookSbrEnvBalanceC10F[49];
+extern const UCHAR bookSbrEnvBalanceL10F[49];
+extern const INT v_Huff_envelopeLevelC11T[63];
+extern const UCHAR v_Huff_envelopeLevelL11T[63];
+extern const INT v_Huff_envelopeLevelC11F[63];
+extern const UCHAR v_Huff_envelopeLevelL11F[63];
+extern const INT bookSbrEnvBalanceC11T[25];
+extern const UCHAR bookSbrEnvBalanceL11T[25];
+extern const INT bookSbrEnvBalanceC11F[25];
+extern const UCHAR bookSbrEnvBalanceL11F[25];
+extern const INT v_Huff_NoiseLevelC11T[63];
+extern const UCHAR v_Huff_NoiseLevelL11T[63];
+extern const INT bookSbrNoiseBalanceC11T[25];
+extern const UCHAR bookSbrNoiseBalanceL11T[25];
+
+extern const sbrTuningTable_t sbrTuningTable[];
+extern const int sbrTuningTableSize;
+
+extern const psTuningTable_t psTuningTable[4];
+
+#endif
diff --git a/fdk-aac/libSBRenc/src/ton_corr.cpp b/fdk-aac/libSBRenc/src/ton_corr.cpp
new file mode 100644
index 0000000..1c050e2
--- /dev/null
+++ b/fdk-aac/libSBRenc/src/ton_corr.cpp
@@ -0,0 +1,891 @@
+/* -----------------------------------------------------------------------------
+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 encoder library ******************************
+
+   Author(s):
+
+   Description:
+
+*******************************************************************************/
+
+#include "ton_corr.h"
+
+#include "sbrenc_ram.h"
+#include "sbr_misc.h"
+#include "genericStds.h"
+#include "autocorr2nd.h"
+
+#define BAND_V_SIZE 32
+#define NUM_V_COMBINE \
+  8 /* Must be a divisor of 64 and fulfill the ASSERTs below */
+
+/**************************************************************************/
+/*!
+  \brief Calculates the tonal to noise ration for different frequency bands
+   and time segments.
+
+   The ratio between the predicted energy (tonal energy A) and the total
+   energy (A + B) is calculated. This is converted to the ratio between
+   the predicted energy (tonal energy A) and the non-predictable energy
+   (noise energy B). Hence the quota-matrix contains A/B = q/(1-q).
+
+   The samples in nrgVector are scaled by 1.0/16.0
+   The samples in pNrgVectorFreq  are scaled by 1.0/2.0
+   The samples in quotaMatrix are scaled by RELAXATION
+
+  \return none.
+
+*/
+/**************************************************************************/
+
+void FDKsbrEnc_CalculateTonalityQuotas(
+    HANDLE_SBR_TON_CORR_EST hTonCorr, /*!< Handle to SBR_TON_CORR struct. */
+    FIXP_DBL **RESTRICT
+        sourceBufferReal, /*!< The real part of the QMF-matrix.  */
+    FIXP_DBL **RESTRICT
+        sourceBufferImag, /*!< The imaginary part of the QMF-matrix. */
+    INT usb,     /*!< upper side band, highest + 1 QMF band in the SBR range. */
+    INT qmfScale /*!< sclefactor of QMF subsamples */
+) {
+  INT i, k, r, r2, timeIndex, autoCorrScaling;
+
+  INT startIndexMatrix = hTonCorr->startIndexMatrix;
+  INT totNoEst = hTonCorr->numberOfEstimates;
+  INT noEstPerFrame = hTonCorr->numberOfEstimatesPerFrame;
+  INT move = hTonCorr->move;
+  INT noQmfChannels = hTonCorr->noQmfChannels; /* Number of Bands */
+  INT buffLen = hTonCorr->bufferLength;        /* Number of Slots */
+  INT stepSize = hTonCorr->stepSize;
+  INT *pBlockLength = hTonCorr->lpcLength;
+  INT **RESTRICT signMatrix = hTonCorr->signMatrix;
+  FIXP_DBL *RESTRICT nrgVector = hTonCorr->nrgVector;
+  FIXP_DBL **RESTRICT quotaMatrix = hTonCorr->quotaMatrix;
+  FIXP_DBL *RESTRICT pNrgVectorFreq = hTonCorr->nrgVectorFreq;
+
+  FIXP_DBL *realBuf;
+  FIXP_DBL *imagBuf;
+
+  FIXP_DBL alphar[2], alphai[2], fac;
+
+  C_ALLOC_SCRATCH_START(ac, ACORR_COEFS, 1)
+  C_ALLOC_SCRATCH_START(realBufRef, FIXP_DBL, 2 * BAND_V_SIZE * NUM_V_COMBINE)
+  realBuf = realBufRef;
+  imagBuf = realBuf + BAND_V_SIZE * NUM_V_COMBINE;
+
+  FDK_ASSERT(buffLen <= BAND_V_SIZE);
+  FDK_ASSERT(sizeof(FIXP_DBL) * NUM_V_COMBINE * BAND_V_SIZE * 2 <
+             (1024 * sizeof(FIXP_DBL) - sizeof(ACORR_COEFS)));
+
+  /*
+   * Buffering of the quotaMatrix and the quotaMatrixTransp.
+   *********************************************************/
+  for (i = 0; i < move; i++) {
+    FDKmemcpy(quotaMatrix[i], quotaMatrix[i + noEstPerFrame],
+              noQmfChannels * sizeof(FIXP_DBL));
+    FDKmemcpy(signMatrix[i], signMatrix[i + noEstPerFrame],
+              noQmfChannels * sizeof(INT));
+  }
+
+  FDKmemmove(nrgVector, nrgVector + noEstPerFrame, move * sizeof(FIXP_DBL));
+  FDKmemclear(nrgVector + startIndexMatrix,
+              (totNoEst - startIndexMatrix) * sizeof(FIXP_DBL));
+  FDKmemclear(pNrgVectorFreq, noQmfChannels * sizeof(FIXP_DBL));
+
+  /*
+   * Calculate the quotas for the current time steps.
+   **************************************************/
+
+  for (r = 0; r < usb; r++) {
+    int blockLength;
+
+    k = hTonCorr->nextSample; /* startSample */
+    timeIndex = startIndexMatrix;
+    /* Copy as many as possible Band across all Slots at once */
+    if (realBuf != realBufRef) {
+      realBuf -= BAND_V_SIZE;
+      imagBuf -= BAND_V_SIZE;
+    } else {
+      realBuf += BAND_V_SIZE * (NUM_V_COMBINE - 1);
+      imagBuf += BAND_V_SIZE * (NUM_V_COMBINE - 1);
+
+      for (i = 0; i < buffLen; i++) {
+        int v;
+        FIXP_DBL *ptr;
+        ptr = realBuf + i;
+        for (v = 0; v < NUM_V_COMBINE; v++) {
+          ptr[0] = sourceBufferReal[i][r + v];
+          ptr[0 + BAND_V_SIZE * NUM_V_COMBINE] = sourceBufferImag[i][r + v];
+          ptr -= BAND_V_SIZE;
+        }
+      }
+    }
+
+    blockLength = pBlockLength[0];
+
+    while (k <= buffLen - blockLength) {
+      autoCorrScaling = fixMin(
+          getScalefactor(&realBuf[k - LPC_ORDER], LPC_ORDER + blockLength),
+          getScalefactor(&imagBuf[k - LPC_ORDER], LPC_ORDER + blockLength));
+      autoCorrScaling = fixMax(0, autoCorrScaling - 1);
+
+      scaleValues(&realBuf[k - LPC_ORDER], LPC_ORDER + blockLength,
+                  autoCorrScaling);
+      scaleValues(&imagBuf[k - LPC_ORDER], LPC_ORDER + blockLength,
+                  autoCorrScaling);
+
+      autoCorrScaling <<= 1; /* consider qmf buffer scaling twice */
+      autoCorrScaling +=
+          autoCorr2nd_cplx(ac, realBuf + k, imagBuf + k, blockLength);
+
+      if (ac->det == FL2FXCONST_DBL(0.0f)) {
+        alphar[1] = alphai[1] = FL2FXCONST_DBL(0.0f);
+
+        alphar[0] = (ac->r01r) >> 2;
+        alphai[0] = (ac->r01i) >> 2;
+
+        fac = fMultDiv2(ac->r00r, ac->r11r) >> 1;
+      } else {
+        alphar[1] = (fMultDiv2(ac->r01r, ac->r12r) >> 1) -
+                    (fMultDiv2(ac->r01i, ac->r12i) >> 1) -
+                    (fMultDiv2(ac->r02r, ac->r11r) >> 1);
+        alphai[1] = (fMultDiv2(ac->r01i, ac->r12r) >> 1) +
+                    (fMultDiv2(ac->r01r, ac->r12i) >> 1) -
+                    (fMultDiv2(ac->r02i, ac->r11r) >> 1);
+
+        alphar[0] = (fMultDiv2(ac->r01r, ac->det) >> (ac->det_scale + 1)) +
+                    fMult(alphar[1], ac->r12r) + fMult(alphai[1], ac->r12i);
+        alphai[0] = (fMultDiv2(ac->r01i, ac->det) >> (ac->det_scale + 1)) +
+                    fMult(alphai[1], ac->r12r) - fMult(alphar[1], ac->r12i);
+
+        fac = fMultDiv2(ac->r00r, fMult(ac->det, ac->r11r)) >>
+              (ac->det_scale + 1);
+      }
+
+      if (fac == FL2FXCONST_DBL(0.0f)) {
+        quotaMatrix[timeIndex][r] = FL2FXCONST_DBL(0.0f);
+        signMatrix[timeIndex][r] = 0;
+      } else {
+        /* quotaMatrix is scaled with the factor RELAXATION
+           parse RELAXATION in fractional part and shift factor: 1/(1/0.524288 *
+           2^RELAXATION_SHIFT) */
+        FIXP_DBL tmp, num, denom;
+        INT numShift, denomShift, commonShift;
+        INT sign;
+
+        num = fMultDiv2(alphar[0], ac->r01r) + fMultDiv2(alphai[0], ac->r01i) -
+              fMultDiv2(alphar[1], fMult(ac->r02r, ac->r11r)) -
+              fMultDiv2(alphai[1], fMult(ac->r02i, ac->r11r));
+        num = fixp_abs(num);
+
+        denom = (fac >> 1) +
+                (fMultDiv2(fac, RELAXATION_FRACT) >> RELAXATION_SHIFT) - num;
+        denom = fixp_abs(denom);
+
+        num = fMult(num, RELAXATION_FRACT);
+
+        numShift = CountLeadingBits(num) - 2;
+        num = scaleValue(num, numShift);
+
+        denomShift = CountLeadingBits(denom);
+        denom = (FIXP_DBL)denom << denomShift;
+
+        if ((num > FL2FXCONST_DBL(0.0f)) && (denom != FL2FXCONST_DBL(0.0f))) {
+          commonShift =
+              fixMin(numShift - denomShift + RELAXATION_SHIFT, DFRACT_BITS - 1);
+          if (commonShift < 0) {
+            commonShift = -commonShift;
+            tmp = schur_div(num, denom, 16);
+            commonShift = fixMin(commonShift, CountLeadingBits(tmp));
+            quotaMatrix[timeIndex][r] = tmp << commonShift;
+          } else {
+            quotaMatrix[timeIndex][r] =
+                schur_div(num, denom, 16) >> commonShift;
+          }
+        } else {
+          quotaMatrix[timeIndex][r] = FL2FXCONST_DBL(0.0f);
+        }
+
+        if (ac->r11r != FL2FXCONST_DBL(0.0f)) {
+          if (((ac->r01r >= FL2FXCONST_DBL(0.0f)) &&
+               (ac->r11r >= FL2FXCONST_DBL(0.0f))) ||
+              ((ac->r01r < FL2FXCONST_DBL(0.0f)) &&
+               (ac->r11r < FL2FXCONST_DBL(0.0f)))) {
+            sign = 1;
+          } else {
+            sign = -1;
+          }
+        } else {
+          sign = 1;
+        }
+
+        if (sign < 0) {
+          r2 = r; /* (INT) pow(-1, band); */
+        } else {
+          r2 = r + 1; /* (INT) pow(-1, band+1); */
+        }
+        signMatrix[timeIndex][r] = 1 - 2 * (r2 & 0x1);
+      }
+
+      nrgVector[timeIndex] +=
+          ((ac->r00r) >>
+           fixMin(DFRACT_BITS - 1,
+                  (2 * qmfScale + autoCorrScaling + SCALE_NRGVEC)));
+      /* pNrgVectorFreq[r] finally has to be divided by noEstPerFrame, replaced
+       * division by shifting with one */
+      pNrgVectorFreq[r] =
+          pNrgVectorFreq[r] +
+          ((ac->r00r) >>
+           fixMin(DFRACT_BITS - 1,
+                  (2 * qmfScale + autoCorrScaling + SCALE_NRGVEC)));
+
+      blockLength = pBlockLength[1];
+      k += stepSize;
+      timeIndex++;
+    }
+  }
+
+  C_ALLOC_SCRATCH_END(realBufRef, FIXP_DBL, 2 * BAND_V_SIZE * NUM_V_COMBINE)
+  C_ALLOC_SCRATCH_END(ac, ACORR_COEFS, 1)
+}
+
+/**************************************************************************/
+/*!
+  \brief Extracts the parameters required in the decoder to obtain the
+  correct tonal to noise ratio after SBR.
+
+  Estimates the tonal to noise ratio of the original signal (using LPC).
+  Predicts the tonal to noise ration of the SBR signal (in the decoder) by
+  patching the tonal to noise ratio values similar to the patching of the
+  lowband in the decoder. Given the tonal to noise ratio of the original
+  and the SBR signal, it estimates the required amount of inverse filtering,
+  additional noise as well as any additional sines.
+
+  \return none.
+
+*/
+/**************************************************************************/
+void FDKsbrEnc_TonCorrParamExtr(
+    HANDLE_SBR_TON_CORR_EST hTonCorr, /*!< Handle to SBR_TON_CORR struct. */
+    INVF_MODE *infVec, /*!< Vector where the inverse filtering levels will be
+                          stored. */
+    FIXP_DBL *noiseLevels, /*!< Vector where the noise levels will be stored. */
+    INT *missingHarmonicFlag, /*!< Flag set to one or zero, dependent on if any
+                                 strong sines are missing.*/
+    UCHAR *missingHarmonicsIndex, /*!< Vector indicating where sines are
+                                     missing. */
+    UCHAR *envelopeCompensation,  /*!< Vector to store compensation values for
+                                     the energies in. */
+    const SBR_FRAME_INFO *frameInfo, /*!< Frame info struct, contains the time
+                                        and frequency grid of the current
+                                        frame.*/
+    UCHAR *transientInfo,            /*!< Transient info.*/
+    UCHAR *freqBandTable,            /*!< Frequency band tables for high-res.*/
+    INT nSfb,           /*!< Number of scalefactor bands for high-res. */
+    XPOS_MODE xposType, /*!< Type of transposer used in the decoder.*/
+    UINT sbrSyntaxFlags) {
+  INT band;
+  INT transientFlag = transientInfo[1]; /*!< Flag indicating if a transient is
+                                           present in the current frame. */
+  INT transientPos = transientInfo[0];  /*!< Position of the transient.*/
+  INT transientFrame, transientFrameInvfEst;
+  INVF_MODE *infVecPtr;
+
+  /* Determine if this is a frame where a transient starts...
+
+  The detection of noise-floor, missing harmonics and invf_est, is not in sync
+  for the non-buf-opt decoder such as AAC. Hence we need to keep track on the
+  transient in the present frame as well as in the next.
+  */
+  transientFrame = 0;
+  if (hTonCorr->transientNextFrame) { /* The transient was detected in the
+                                         previous frame, but is actually */
+    transientFrame = 1;
+    hTonCorr->transientNextFrame = 0;
+
+    if (transientFlag) {
+      if (transientPos + hTonCorr->transientPosOffset >=
+          frameInfo->borders[frameInfo->nEnvelopes]) {
+        hTonCorr->transientNextFrame = 1;
+      }
+    }
+  } else {
+    if (transientFlag) {
+      if (transientPos + hTonCorr->transientPosOffset <
+          frameInfo->borders[frameInfo->nEnvelopes]) {
+        transientFrame = 1;
+        hTonCorr->transientNextFrame = 0;
+      } else {
+        hTonCorr->transientNextFrame = 1;
+      }
+    }
+  }
+  transientFrameInvfEst = transientFrame;
+
+  /*
+    Estimate the required invese filtereing level.
+  */
+  if (hTonCorr->switchInverseFilt)
+    FDKsbrEnc_qmfInverseFilteringDetector(
+        &hTonCorr->sbrInvFilt, hTonCorr->quotaMatrix, hTonCorr->nrgVector,
+        hTonCorr->indexVector, hTonCorr->frameStartIndexInvfEst,
+        hTonCorr->numberOfEstimatesPerFrame + hTonCorr->frameStartIndexInvfEst,
+        transientFrameInvfEst, infVec);
+
+  /*
+      Detect what tones will be missing.
+   */
+  if (xposType == XPOS_LC) {
+    FDKsbrEnc_SbrMissingHarmonicsDetectorQmf(
+        &hTonCorr->sbrMissingHarmonicsDetector, hTonCorr->quotaMatrix,
+        hTonCorr->signMatrix, hTonCorr->indexVector, frameInfo, transientInfo,
+        missingHarmonicFlag, missingHarmonicsIndex, freqBandTable, nSfb,
+        envelopeCompensation, hTonCorr->nrgVectorFreq);
+  } else {
+    *missingHarmonicFlag = 0;
+    FDKmemclear(missingHarmonicsIndex, nSfb * sizeof(UCHAR));
+  }
+
+  /*
+    Noise floor estimation
+  */
+
+  infVecPtr = hTonCorr->sbrInvFilt.prevInvfMode;
+
+  FDKsbrEnc_sbrNoiseFloorEstimateQmf(
+      &hTonCorr->sbrNoiseFloorEstimate, frameInfo, noiseLevels,
+      hTonCorr->quotaMatrix, hTonCorr->indexVector, *missingHarmonicFlag,
+      hTonCorr->frameStartIndex, hTonCorr->numberOfEstimatesPerFrame,
+      transientFrame, infVecPtr, sbrSyntaxFlags);
+
+  /* Store the invfVec data for the next frame...*/
+  for (band = 0; band < hTonCorr->sbrInvFilt.noDetectorBands; band++) {
+    hTonCorr->sbrInvFilt.prevInvfMode[band] = infVec[band];
+  }
+}
+
+/**************************************************************************/
+/*!
+  \brief     Searches for the closest match in the frequency master table.
+
+
+
+  \return   closest entry.
+
+*/
+/**************************************************************************/
+static INT findClosestEntry(INT goalSb, UCHAR *v_k_master, INT numMaster,
+                            INT 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     resets the patch
+
+
+
+  \return   errorCode, noError if successful.
+
+*/
+/**************************************************************************/
+static INT resetPatch(
+    HANDLE_SBR_TON_CORR_EST hTonCorr, /*!< Handle to SBR_TON_CORR struct. */
+    INT xposctrl,                     /*!< Different patch modes. */
+    INT highBandStartSb,              /*!< Start band of the SBR range. */
+    UCHAR *v_k_master, /*!< Master frequency table from which all other table
+                          are derived.*/
+    INT numMaster,     /*!< Number of elements in the master table. */
+    INT fs,            /*!< Sampling frequency. */
+    INT noChannels)    /*!< Number of QMF-channels. */
+{
+  INT patch, k, i;
+  INT targetStopBand;
+
+  PATCH_PARAM *patchParam = hTonCorr->patchParam;
+
+  INT sbGuard = hTonCorr->guard;
+  INT sourceStartBand;
+  INT patchDistance;
+  INT numBandsInPatch;
+
+  INT lsb =
+      v_k_master[0]; /* Lowest subband related to the synthesis filterbank */
+  INT usb = v_k_master[numMaster]; /* Stop subband related to the synthesis
+                                      filterbank */
+  INT xoverOffset =
+      highBandStartSb -
+      v_k_master[0]; /* Calculate distance in subbands between k0 and kx */
+
+  INT goalSb;
+
+  /*
+   * Initialize the patching parameter
+   */
+
+  if (xposctrl == 1) {
+    lsb += xoverOffset;
+    xoverOffset = 0;
+  }
+
+  goalSb = (INT)((2 * noChannels * 16000 + (fs >> 1)) / fs); /* 16 kHz band */
+  goalSb = findClosestEntry(goalSb, v_k_master, numMaster,
+                            1); /* Adapt region to master-table */
+
+  /* First patch */
+  sourceStartBand = hTonCorr->shiftStartSb + xoverOffset;
+  targetStopBand = lsb + xoverOffset;
+
+  /* even (odd) numbered channel must be patched to even (odd) numbered channel
+   */
+  patch = 0;
+  while (targetStopBand < usb) {
+    /* To many patches */
+    if (patch >= MAX_NUM_PATCHES) return (1); /*Number of patches to high */
+
+    patchParam[patch].guardStartBand = targetStopBand;
+    targetStopBand += sbGuard;
+    patchParam[patch].targetStartBand = targetStopBand;
+
+    numBandsInPatch =
+        goalSb - 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);
+      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) {
+      patch--;
+    } else {
+      patchParam[patch].sourceStartBand = targetStopBand - patchDistance;
+      patchParam[patch].targetBandOffs = patchDistance;
+      patchParam[patch].numBandsInPatch = numBandsInPatch;
+      patchParam[patch].sourceStopBand =
+          patchParam[patch].sourceStartBand + numBandsInPatch;
+
+      targetStopBand += patchParam[patch].numBandsInPatch;
+    }
+
+    /* All patches but first */
+    sourceStartBand = hTonCorr->shiftStartSb;
+
+    /* Check if we are close to goalSb */
+    if (fixp_abs(targetStopBand - goalSb) < 3) {
+      goalSb = usb;
+    }
+
+    patch++;
+  }
+
+  patch--;
+
+  /* if highest patch contains less than three subband: skip it */
+  if (patchParam[patch].numBandsInPatch < 3 && patch > 0) {
+    patch--;
+  }
+
+  hTonCorr->noOfPatches = patch + 1;
+
+  /* Assign the index-vector, so we know where to look for the high-band.
+     -1 represents a guard-band. */
+  for (k = 0; k < hTonCorr->patchParam[0].guardStartBand; k++)
+    hTonCorr->indexVector[k] = k;
+
+  for (i = 0; i < hTonCorr->noOfPatches; i++) {
+    INT sourceStart = hTonCorr->patchParam[i].sourceStartBand;
+    INT targetStart = hTonCorr->patchParam[i].targetStartBand;
+    INT numberOfBands = hTonCorr->patchParam[i].numBandsInPatch;
+    INT startGuardBand = hTonCorr->patchParam[i].guardStartBand;
+
+    for (k = 0; k < (targetStart - startGuardBand); k++)
+      hTonCorr->indexVector[startGuardBand + k] = -1;
+
+    for (k = 0; k < numberOfBands; k++)
+      hTonCorr->indexVector[targetStart + k] = sourceStart + k;
+  }
+
+  return (0);
+}
+
+/**************************************************************************/
+/*!
+  \brief     Creates an instance of the tonality correction parameter module.
+
+  The module includes modules for inverse filtering level estimation,
+  missing harmonics detection and noise floor level estimation.
+
+  \return   errorCode, noError if successful.
+*/
+/**************************************************************************/
+INT FDKsbrEnc_CreateTonCorrParamExtr(
+    HANDLE_SBR_TON_CORR_EST
+        hTonCorr, /*!< Pointer to handle to SBR_TON_CORR struct. */
+    INT chan)     /*!< Channel index, needed for mem allocation */
+{
+  INT i;
+  FIXP_DBL *quotaMatrix = GetRam_Sbr_quotaMatrix(chan);
+  INT *signMatrix = GetRam_Sbr_signMatrix(chan);
+
+  if ((NULL == quotaMatrix) || (NULL == signMatrix)) {
+    goto bail;
+  }
+
+  FDKmemclear(hTonCorr, sizeof(SBR_TON_CORR_EST));
+
+  for (i = 0; i < MAX_NO_OF_ESTIMATES; i++) {
+    hTonCorr->quotaMatrix[i] = quotaMatrix + (i * 64);
+    hTonCorr->signMatrix[i] = signMatrix + (i * 64);
+  }
+
+  if (0 != FDKsbrEnc_CreateSbrMissingHarmonicsDetector(
+               &hTonCorr->sbrMissingHarmonicsDetector, chan)) {
+    goto bail;
+  }
+
+  return 0;
+
+bail:
+  hTonCorr->quotaMatrix[0] = quotaMatrix;
+  hTonCorr->signMatrix[0] = signMatrix;
+
+  FDKsbrEnc_DeleteTonCorrParamExtr(hTonCorr);
+
+  return -1;
+}
+
+/**************************************************************************/
+/*!
+  \brief     Initialize an instance of the tonality correction parameter module.
+
+  The module includes modules for inverse filtering level estimation,
+  missing harmonics detection and noise floor level estimation.
+
+  \return   errorCode, noError if successful.
+*/
+/**************************************************************************/
+INT FDKsbrEnc_InitTonCorrParamExtr(
+    INT frameSize, /*!< Current SBR frame size. */
+    HANDLE_SBR_TON_CORR_EST
+        hTonCorr, /*!< Pointer to handle to SBR_TON_CORR struct. */
+    HANDLE_SBR_CONFIG_DATA
+        sbrCfg,           /*!< Pointer to SBR configuration parameters. */
+    INT timeSlots,        /*!< Number of time-slots per frame */
+    INT xposCtrl,         /*!< Different patch modes. */
+    INT ana_max_level,    /*!< Maximum level of the adaptive noise. */
+    INT noiseBands,       /*!< Number of noise bands per octave. */
+    INT noiseFloorOffset, /*!< Noise floor offset. */
+    UINT useSpeechConfig) /*!< Speech or music tuning. */
+{
+  INT nCols = sbrCfg->noQmfSlots;
+  INT fs = sbrCfg->sampleFreq;
+  INT noQmfChannels = sbrCfg->noQmfBands;
+
+  INT highBandStartSb = sbrCfg->freqBandTable[LOW_RES][0];
+  UCHAR *v_k_master = sbrCfg->v_k_master;
+  INT numMaster = sbrCfg->num_Master;
+
+  UCHAR **freqBandTable = sbrCfg->freqBandTable;
+  INT *nSfb = sbrCfg->nSfb;
+
+  INT i;
+
+  /*
+  Reset the patching and allocate memory for the quota matrix.
+  Assuming parameters for the LPC analysis.
+  */
+  if (sbrCfg->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) {
+    switch (timeSlots) {
+      case NUMBER_TIME_SLOTS_1920:
+        hTonCorr->lpcLength[0] = 8 - LPC_ORDER;
+        hTonCorr->lpcLength[1] = 7 - LPC_ORDER;
+        hTonCorr->numberOfEstimates = NO_OF_ESTIMATES_LD;
+        hTonCorr->numberOfEstimatesPerFrame = 2; /* sbrCfg->noQmfSlots / 7 */
+        hTonCorr->frameStartIndexInvfEst = 0;
+        hTonCorr->transientPosOffset = FRAME_MIDDLE_SLOT_512LD;
+        break;
+      case NUMBER_TIME_SLOTS_2048:
+        hTonCorr->lpcLength[0] = 8 - LPC_ORDER;
+        hTonCorr->lpcLength[1] = 8 - LPC_ORDER;
+        hTonCorr->numberOfEstimates = NO_OF_ESTIMATES_LD;
+        hTonCorr->numberOfEstimatesPerFrame = 2; /* sbrCfg->noQmfSlots / 8 */
+        hTonCorr->frameStartIndexInvfEst = 0;
+        hTonCorr->transientPosOffset = FRAME_MIDDLE_SLOT_512LD;
+        break;
+    }
+  } else
+    switch (timeSlots) {
+      case NUMBER_TIME_SLOTS_2048:
+        hTonCorr->lpcLength[0] = 16 - LPC_ORDER; /* blockLength[0] */
+        hTonCorr->lpcLength[1] = 16 - LPC_ORDER; /* blockLength[0] */
+        hTonCorr->numberOfEstimates = NO_OF_ESTIMATES_LC;
+        hTonCorr->numberOfEstimatesPerFrame = sbrCfg->noQmfSlots / 16;
+        hTonCorr->frameStartIndexInvfEst = 0;
+        hTonCorr->transientPosOffset = FRAME_MIDDLE_SLOT_2048;
+        break;
+      case NUMBER_TIME_SLOTS_1920:
+        hTonCorr->lpcLength[0] = 15 - LPC_ORDER; /* blockLength[0] */
+        hTonCorr->lpcLength[1] = 15 - LPC_ORDER; /* blockLength[0] */
+        hTonCorr->numberOfEstimates = NO_OF_ESTIMATES_LC;
+        hTonCorr->numberOfEstimatesPerFrame = sbrCfg->noQmfSlots / 15;
+        hTonCorr->frameStartIndexInvfEst = 0;
+        hTonCorr->transientPosOffset = FRAME_MIDDLE_SLOT_1920;
+        break;
+      default:
+        return -1;
+    }
+
+  hTonCorr->bufferLength = nCols;
+  hTonCorr->stepSize =
+      hTonCorr->lpcLength[0] + LPC_ORDER; /* stepSize[0] implicitly 0. */
+
+  hTonCorr->nextSample = LPC_ORDER; /* firstSample */
+  hTonCorr->move = hTonCorr->numberOfEstimates -
+                   hTonCorr->numberOfEstimatesPerFrame; /* Number of estimates
+                                                           to move when
+                                                           buffering.*/
+  if (hTonCorr->move < 0) {
+    return -1;
+  }
+  hTonCorr->startIndexMatrix =
+      hTonCorr->numberOfEstimates -
+      hTonCorr->numberOfEstimatesPerFrame; /* Where to store the latest
+                                              estimations in the tonality
+                                              Matrix.*/
+  hTonCorr->frameStartIndex = 0; /* Where in the tonality matrix the current
+                                    frame (to be sent to the decoder) starts. */
+  hTonCorr->prevTransientFlag = 0;
+  hTonCorr->transientNextFrame = 0;
+
+  hTonCorr->noQmfChannels = noQmfChannels;
+
+  for (i = 0; i < hTonCorr->numberOfEstimates; i++) {
+    FDKmemclear(hTonCorr->quotaMatrix[i], sizeof(FIXP_DBL) * noQmfChannels);
+    FDKmemclear(hTonCorr->signMatrix[i], sizeof(INT) * noQmfChannels);
+  }
+
+  /* Reset the patch.*/
+  hTonCorr->guard = 0;
+  hTonCorr->shiftStartSb = 1;
+
+  if (resetPatch(hTonCorr, xposCtrl, highBandStartSb, v_k_master, numMaster, fs,
+                 noQmfChannels))
+    return (1);
+
+  if (FDKsbrEnc_InitSbrNoiseFloorEstimate(
+          &hTonCorr->sbrNoiseFloorEstimate, ana_max_level, freqBandTable[LO],
+          nSfb[LO], noiseBands, noiseFloorOffset, timeSlots, useSpeechConfig))
+    return (1);
+
+  if (FDKsbrEnc_initInvFiltDetector(
+          &hTonCorr->sbrInvFilt,
+          hTonCorr->sbrNoiseFloorEstimate.freqBandTableQmf,
+          hTonCorr->sbrNoiseFloorEstimate.noNoiseBands, useSpeechConfig))
+    return (1);
+
+  if (FDKsbrEnc_InitSbrMissingHarmonicsDetector(
+          &hTonCorr->sbrMissingHarmonicsDetector, fs, frameSize, nSfb[HI],
+          noQmfChannels, hTonCorr->numberOfEstimates, hTonCorr->move,
+          hTonCorr->numberOfEstimatesPerFrame, sbrCfg->sbrSyntaxFlags))
+    return (1);
+
+  return (0);
+}
+
+/**************************************************************************/
+/*!
+  \brief     resets tonality correction parameter module.
+
+
+
+  \return   errorCode, noError if successful.
+
+*/
+/**************************************************************************/
+INT FDKsbrEnc_ResetTonCorrParamExtr(
+    HANDLE_SBR_TON_CORR_EST hTonCorr, /*!< Handle to SBR_TON_CORR struct. */
+    INT xposctrl,                     /*!< Different patch modes. */
+    INT highBandStartSb,              /*!< Start band of the SBR range. */
+    UCHAR *v_k_master, /*!< Master frequency table from which all other table
+                          are derived.*/
+    INT numMaster,     /*!< Number of elements in the master table. */
+    INT fs,            /*!< Sampling frequency (of the SBR part). */
+    UCHAR *
+        *freqBandTable, /*!< Frequency band table for low-res and high-res. */
+    INT *nSfb,          /*!< Number of frequency bands (hig-res and low-res). */
+    INT noQmfChannels   /*!< Number of QMF channels. */
+) {
+  /* Reset the patch.*/
+  hTonCorr->guard = 0;
+  hTonCorr->shiftStartSb = 1;
+
+  if (resetPatch(hTonCorr, xposctrl, highBandStartSb, v_k_master, numMaster, fs,
+                 noQmfChannels))
+    return (1);
+
+  /* Reset the noise floor estimate.*/
+  if (FDKsbrEnc_resetSbrNoiseFloorEstimate(&hTonCorr->sbrNoiseFloorEstimate,
+                                           freqBandTable[LO], nSfb[LO]))
+    return (1);
+
+  /*
+  Reset the inveerse filtereing detector.
+  */
+  if (FDKsbrEnc_resetInvFiltDetector(
+          &hTonCorr->sbrInvFilt,
+          hTonCorr->sbrNoiseFloorEstimate.freqBandTableQmf,
+          hTonCorr->sbrNoiseFloorEstimate.noNoiseBands))
+    return (1);
+  /* Reset the missing harmonics detector. */
+  if (FDKsbrEnc_ResetSbrMissingHarmonicsDetector(
+          &hTonCorr->sbrMissingHarmonicsDetector, nSfb[HI]))
+    return (1);
+
+  return (0);
+}
+
+/**************************************************************************/
+/*!
+  \brief  Deletes the tonality correction paramtere module.
+
+
+
+  \return   none
+
+*/
+/**************************************************************************/
+void FDKsbrEnc_DeleteTonCorrParamExtr(
+    HANDLE_SBR_TON_CORR_EST hTonCorr) /*!< Handle to SBR_TON_CORR struct. */
+{
+  if (hTonCorr) {
+    FreeRam_Sbr_quotaMatrix(hTonCorr->quotaMatrix);
+
+    FreeRam_Sbr_signMatrix(hTonCorr->signMatrix);
+
+    FDKsbrEnc_DeleteSbrMissingHarmonicsDetector(
+        &hTonCorr->sbrMissingHarmonicsDetector);
+  }
+}
diff --git a/fdk-aac/libSBRenc/src/ton_corr.h b/fdk-aac/libSBRenc/src/ton_corr.h
new file mode 100644
index 0000000..91aa278
--- /dev/null
+++ b/fdk-aac/libSBRenc/src/ton_corr.h
@@ -0,0 +1,258 @@
+/* -----------------------------------------------------------------------------
+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 encoder library ******************************
+
+   Author(s):
+
+   Description:
+
+*******************************************************************************/
+
+/*!
+  \file
+  \brief  General tonality correction detector module.
+*/
+#ifndef TON_CORR_H
+#define TON_CORR_H
+
+#include "sbr_encoder.h"
+#include "mh_det.h"
+#include "nf_est.h"
+#include "invf_est.h"
+
+#define MAX_NUM_PATCHES 6
+#define SCALE_NRGVEC 4
+
+/** parameter set for one single patch */
+typedef struct {
+  INT sourceStartBand; /*!< first band in lowbands where to take the samples
+                          from */
+  INT sourceStopBand;  /*!< first band in lowbands which is not included in the
+                          patch anymore */
+  INT guardStartBand;  /*!< first band in highbands to be filled with zeros in
+                          order to  reduce interferences between patches */
+  INT targetStartBand; /*!< first band in highbands to be filled with whitened
+                          lowband signal */
+  INT targetBandOffs;  /*!< difference between 'startTargetBand' and
+                          'startSourceBand' */
+  INT numBandsInPatch; /*!< number of consecutive bands in this one patch */
+} PATCH_PARAM;
+
+typedef struct {
+  INT switchInverseFilt; /*!< Flag to enable dynamic adaption of invf. detection
+                          */
+  INT noQmfChannels;
+  INT bufferLength;      /*!< Length of the r and i buffers. */
+  INT stepSize;          /*!< Stride for the lpc estimate. */
+  INT numberOfEstimates; /*!< The total number of estiamtes, available in the
+                            quotaMatrix.*/
+  UINT numberOfEstimatesPerFrame; /*!< The number of estimates per frame
+                                     available in the quotaMatrix.*/
+  INT lpcLength[2]; /*!< Segment length used for second order LPC analysis.*/
+  INT nextSample;   /*!< Where to start the LPC analysis of the current frame.*/
+  INT move; /*!< How many estimates to move in the quotaMatrix, when buffering.
+             */
+  INT frameStartIndex; /*!< The start index for the current frame in the r and i
+                          buffers. */
+  INT startIndexMatrix;       /*!< The start index for the current frame in the
+                                 quotaMatrix. */
+  INT frameStartIndexInvfEst; /*!< The start index of the inverse filtering, not
+                                 the same as the others, dependent on what
+                                 decoder is used (buffer opt, or no buffer opt).
+                               */
+  INT prevTransientFlag;  /*!< The transisent flag (from the transient detector)
+                             for the previous frame. */
+  INT transientNextFrame; /*!< Flag to indicate that the transient will show up
+                             in the next frame. */
+  INT transientPosOffset; /*!< An offset value to match the transient pos as
+                             calculated by the transient detector with the
+                             actual position in the frame.*/
+
+  INT* signMatrix[MAX_NO_OF_ESTIMATES]; /*!< Matrix holding the sign of each
+                                           channe, i.e. indicating in what part
+                                           of a QMF channel a possible sine is.
+                                         */
+
+  FIXP_DBL* quotaMatrix[MAX_NO_OF_ESTIMATES]; /*!< Matrix holding the quota
+                                                 values for all estimates, all
+                                                 channels. */
+
+  FIXP_DBL nrgVector[MAX_NO_OF_ESTIMATES]; /*!< Vector holding the averaged
+                                              energies for every QMF band. */
+  FIXP_DBL nrgVectorFreq[64]; /*!< Vector holding the averaged energies for
+                                 every QMF channel */
+
+  SCHAR indexVector[64]; /*!< Index vector poINTing to the correct lowband
+                            channel, when indexing a highband channel, -1
+                            represents a guard band */
+  PATCH_PARAM
+  patchParam[MAX_NUM_PATCHES]; /*!< new parameter set for patching */
+  INT guard;                   /*!< number of guardbands between every patch */
+  INT shiftStartSb; /*!< lowest subband of source range to be included in the
+                       patches */
+  INT noOfPatches;  /*!< number of patches */
+
+  SBR_MISSING_HARMONICS_DETECTOR
+  sbrMissingHarmonicsDetector; /*!< SBR_MISSING_HARMONICS_DETECTOR struct.
+                                */
+  SBR_NOISE_FLOOR_ESTIMATE
+  sbrNoiseFloorEstimate;       /*!< SBR_NOISE_FLOOR_ESTIMATE struct. */
+  SBR_INV_FILT_EST sbrInvFilt; /*!< SBR_INV_FILT_EST struct. */
+} SBR_TON_CORR_EST;
+
+typedef SBR_TON_CORR_EST* HANDLE_SBR_TON_CORR_EST;
+
+void FDKsbrEnc_TonCorrParamExtr(
+    HANDLE_SBR_TON_CORR_EST hTonCorr, /*!< Handle to SBR_TON_CORR struct. */
+    INVF_MODE* infVec, /*!< Vector where the inverse filtering levels will be
+                          stored. */
+    FIXP_DBL* noiseLevels, /*!< Vector where the noise levels will be stored. */
+    INT* missingHarmonicFlag, /*!< Flag set to one or zero, dependent on if any
+                                 strong sines are missing.*/
+    UCHAR* missingHarmonicsIndex, /*!< Vector indicating where sines are
+                                     missing. */
+    UCHAR* envelopeCompensation,  /*!< Vector to store compensation values for
+                                     the energies in. */
+    const SBR_FRAME_INFO* frameInfo, /*!< Frame info struct, contains the time
+                                        and frequency grid of the current
+                                        frame.*/
+    UCHAR* transientInfo,            /*!< Transient info.*/
+    UCHAR* freqBandTable,            /*!< Frequency band tables for high-res.*/
+    INT nSfb,           /*!< Number of scalefactor bands for high-res. */
+    XPOS_MODE xposType, /*!< Type of transposer used in the decoder.*/
+    UINT sbrSyntaxFlags);
+
+INT FDKsbrEnc_CreateTonCorrParamExtr(
+    HANDLE_SBR_TON_CORR_EST
+        hTonCorr, /*!< Pointer to handle to SBR_TON_CORR struct. */
+    INT chan);    /*!< Channel index, needed for mem allocation */
+
+INT FDKsbrEnc_InitTonCorrParamExtr(
+    INT frameSize, /*!< Current SBR frame size. */
+    HANDLE_SBR_TON_CORR_EST
+        hTonCorr, /*!< Pointer to handle to SBR_TON_CORR struct. */
+    HANDLE_SBR_CONFIG_DATA
+        sbrCfg,           /*!< Pointer to SBR configuration parameters. */
+    INT timeSlots,        /*!< Number of time-slots per frame */
+    INT xposCtrl,         /*!< Different patch modes. */
+    INT ana_max_level,    /*!< Maximum level of the adaptive noise. */
+    INT noiseBands,       /*!< Number of noise bands per octave. */
+    INT noiseFloorOffset, /*!< Noise floor offset. */
+    UINT useSpeechConfig  /*!< Speech or music tuning. */
+);
+
+void FDKsbrEnc_DeleteTonCorrParamExtr(
+    HANDLE_SBR_TON_CORR_EST hTonCorr); /*!< Handle to SBR_TON_CORR struct. */
+
+void FDKsbrEnc_CalculateTonalityQuotas(
+    HANDLE_SBR_TON_CORR_EST hTonCorr, FIXP_DBL** sourceBufferReal,
+    FIXP_DBL** sourceBufferImag, INT usb,
+    INT qmfScale /*!< sclefactor of QMF subsamples */
+);
+
+INT FDKsbrEnc_ResetTonCorrParamExtr(
+    HANDLE_SBR_TON_CORR_EST hTonCorr, /*!< Handle to SBR_TON_CORR struct. */
+    INT xposctrl,                     /*!< Different patch modes. */
+    INT highBandStartSb,              /*!< Start band of the SBR range. */
+    UCHAR* v_k_master, /*!< Master frequency table from which all other table
+                          are derived.*/
+    INT numMaster,     /*!< Number of elements in the master table. */
+    INT fs,            /*!< Sampling frequency (of the SBR part). */
+    UCHAR**
+        freqBandTable, /*!< Frequency band table for low-res and high-res. */
+    INT* nSfb,         /*!< Number of frequency bands (hig-res and low-res). */
+    INT noQmfChannels  /*!< Number of QMF channels. */
+);
+#endif
diff --git a/fdk-aac/libSBRenc/src/tran_det.cpp b/fdk-aac/libSBRenc/src/tran_det.cpp
new file mode 100644
index 0000000..3b6765a
--- /dev/null
+++ b/fdk-aac/libSBRenc/src/tran_det.cpp
@@ -0,0 +1,1092 @@
+/* -----------------------------------------------------------------------------
+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 encoder library ******************************
+
+   Author(s):   Tobias Chalupka
+
+   Description: SBR encoder transient detector
+
+*******************************************************************************/
+
+#include "tran_det.h"
+
+#include "fram_gen.h"
+#include "sbrenc_ram.h"
+#include "sbr_misc.h"
+
+#include "genericStds.h"
+
+#define NORM_QMF_ENERGY 9.31322574615479E-10 /* 2^-30 */
+
+/* static FIXP_DBL ABS_THRES = fixMax( FL2FXCONST_DBL(1.28e5 *
+ * NORM_QMF_ENERGY), (FIXP_DBL)1)  Minimum threshold for detecting changes */
+#define ABS_THRES ((FIXP_DBL)16)
+
+/*******************************************************************************
+ Functionname:  spectralChange
+ *******************************************************************************
+ \brief   Calculates a measure for the spectral change within the frame
+
+ The function says how good it would be to split the frame at the given border
+ position into 2 envelopes.
+
+ The return value delta_sum is scaled with the factor 1/64
+
+ \return  calculated value
+*******************************************************************************/
+#define NRG_SHIFT 3 /* for energy summation */
+
+static FIXP_DBL spectralChange(
+    FIXP_DBL Energies[NUMBER_TIME_SLOTS_2304][MAX_FREQ_COEFFS],
+    INT *scaleEnergies, FIXP_DBL EnergyTotal, INT nSfb, INT start, INT border,
+    INT YBufferWriteOffset, INT stop, INT *result_e) {
+  INT i, j;
+  INT len1, len2;
+  SCHAR energies_e_diff[NUMBER_TIME_SLOTS_2304], energies_e, energyTotal_e = 19,
+                                                             energies_e_add;
+  SCHAR prevEnergies_e_diff, newEnergies_e_diff;
+  FIXP_DBL tmp0, tmp1;
+  FIXP_DBL delta, delta_sum;
+  INT accu_e, tmp_e;
+
+  delta_sum = FL2FXCONST_DBL(0.0f);
+  *result_e = 0;
+
+  len1 = border - start;
+  len2 = stop - border;
+
+  /* prefer borders near the middle of the frame */
+  FIXP_DBL pos_weight;
+  pos_weight = FL2FXCONST_DBL(0.5f) - (len1 * GetInvInt(len1 + len2));
+  pos_weight = /*FL2FXCONST_DBL(1.0)*/ (FIXP_DBL)MAXVAL_DBL -
+               (fMult(pos_weight, pos_weight) << 2);
+
+  /*** Calc scaling for energies ***/
+  FDK_ASSERT(scaleEnergies[0] >= 0);
+  FDK_ASSERT(scaleEnergies[1] >= 0);
+
+  energies_e = 19 - fMin(scaleEnergies[0], scaleEnergies[1]);
+
+  /* limit shift for energy accumulation, energies_e can be -10 min. */
+  if (energies_e < -10) {
+    energies_e_add = -10 - energies_e;
+    energies_e = -10;
+  } else if (energies_e > 17) {
+    energies_e_add = energies_e - 17;
+    energies_e = 17;
+  } else {
+    energies_e_add = 0;
+  }
+
+  /* compensate scaling differences between scaleEnergies[0] and
+   * scaleEnergies[1]  */
+  prevEnergies_e_diff = scaleEnergies[0] -
+                        fMin(scaleEnergies[0], scaleEnergies[1]) +
+                        energies_e_add + NRG_SHIFT;
+  newEnergies_e_diff = scaleEnergies[1] -
+                       fMin(scaleEnergies[0], scaleEnergies[1]) +
+                       energies_e_add + NRG_SHIFT;
+
+  prevEnergies_e_diff = fMin(prevEnergies_e_diff, DFRACT_BITS - 1);
+  newEnergies_e_diff = fMin(newEnergies_e_diff, DFRACT_BITS - 1);
+
+  for (i = start; i < YBufferWriteOffset; i++) {
+    energies_e_diff[i] = prevEnergies_e_diff;
+  }
+  for (i = YBufferWriteOffset; i < stop; i++) {
+    energies_e_diff[i] = newEnergies_e_diff;
+  }
+
+  /* Sum up energies of all QMF-timeslots for both halfs */
+  FDK_ASSERT(len1 <= 8); /* otherwise an overflow is possible */
+  FDK_ASSERT(len2 <= 8); /* otherwise an overflow is possible */
+
+  for (j = 0; j < nSfb; j++) {
+    FIXP_DBL accu1 = FL2FXCONST_DBL(0.f);
+    FIXP_DBL accu2 = FL2FXCONST_DBL(0.f);
+    accu_e = energies_e + 3;
+
+    /* Sum up energies in first half */
+    for (i = start; i < border; i++) {
+      accu1 += scaleValue(Energies[i][j], -energies_e_diff[i]);
+    }
+
+    /* Sum up energies in second half */
+    for (i = border; i < stop; i++) {
+      accu2 += scaleValue(Energies[i][j], -energies_e_diff[i]);
+    }
+
+    /* Ensure certain energy to prevent division by zero and to prevent
+     * splitting for very low levels */
+    accu1 = fMax(accu1, (FIXP_DBL)len1);
+    accu2 = fMax(accu2, (FIXP_DBL)len2);
+
+/* Energy change in current band */
+#define LN2 FL2FXCONST_DBL(0.6931471806f) /* ln(2) */
+    tmp0 = fLog2(accu2, accu_e) - fLog2(accu1, accu_e);
+    tmp1 = fLog2((FIXP_DBL)len1, 31) - fLog2((FIXP_DBL)len2, 31);
+    delta = fMult(LN2, (tmp0 + tmp1));
+    delta = (FIXP_DBL)fAbs(delta);
+
+    /* Weighting with amplitude ratio of this band */
+    accu_e++; /* scale at least one bit due to (accu1+accu2) */
+    accu1 >>= 1;
+    accu2 >>= 1;
+
+    if (accu_e & 1) {
+      accu_e++; /* for a defined square result exponent, the exponent has to be
+                   even */
+      accu1 >>= 1;
+      accu2 >>= 1;
+    }
+
+    delta_sum += fMult(sqrtFixp(accu1 + accu2), delta);
+    *result_e = ((accu_e >> 1) + LD_DATA_SHIFT);
+  }
+
+  if (energyTotal_e & 1) {
+    energyTotal_e += 1; /* for a defined square result exponent, the exponent
+                           has to be even */
+    EnergyTotal >>= 1;
+  }
+
+  delta_sum = fMult(delta_sum, invSqrtNorm2(EnergyTotal, &tmp_e));
+  *result_e = *result_e + (tmp_e - (energyTotal_e >> 1));
+
+  return fMult(delta_sum, pos_weight);
+}
+
+/*******************************************************************************
+ Functionname:  addLowbandEnergies
+ *******************************************************************************
+ \brief   Calculates total lowband energy
+
+ The input values Energies[0] (low-band) are scaled by the factor
+ 2^(14-*scaleEnergies[0])
+ The input values Energies[1] (high-band) are scaled by the factor
+ 2^(14-*scaleEnergies[1])
+
+ \return  total energy in the lowband, scaled by the factor 2^19
+*******************************************************************************/
+static FIXP_DBL addLowbandEnergies(FIXP_DBL **Energies, int *scaleEnergies,
+                                   int YBufferWriteOffset, int nrgSzShift,
+                                   int tran_off, UCHAR *freqBandTable,
+                                   int slots) {
+  INT nrgTotal_e;
+  FIXP_DBL nrgTotal_m;
+  FIXP_DBL accu1 = FL2FXCONST_DBL(0.0f);
+  FIXP_DBL accu2 = FL2FXCONST_DBL(0.0f);
+  int tran_offdiv2 = tran_off >> nrgSzShift;
+  const int sc1 =
+      DFRACT_BITS -
+      fNormz((FIXP_DBL)fMax(
+          1, (freqBandTable[0] * (YBufferWriteOffset - tran_offdiv2) - 1)));
+  const int sc2 =
+      DFRACT_BITS -
+      fNormz((FIXP_DBL)fMax(
+          1, (freqBandTable[0] *
+                  (tran_offdiv2 + (slots >> nrgSzShift) - YBufferWriteOffset) -
+              1)));
+  int ts, k;
+
+  /* Sum up lowband energy from one frame at offset tran_off */
+  /* freqBandTable[LORES] has MAX_FREQ_COEFFS/2 +1 coeefs max. */
+  for (ts = tran_offdiv2; ts < YBufferWriteOffset; ts++) {
+    for (k = 0; k < freqBandTable[0]; k++) {
+      accu1 += Energies[ts][k] >> sc1;
+    }
+  }
+  for (; ts < tran_offdiv2 + (slots >> nrgSzShift); ts++) {
+    for (k = 0; k < freqBandTable[0]; k++) {
+      accu2 += Energies[ts][k] >> sc2;
+    }
+  }
+
+  nrgTotal_m = fAddNorm(accu1, (sc1 - 5) - scaleEnergies[0], accu2,
+                        (sc2 - 5) - scaleEnergies[1], &nrgTotal_e);
+  nrgTotal_m = scaleValueSaturate(nrgTotal_m, nrgTotal_e);
+
+  return (nrgTotal_m);
+}
+
+/*******************************************************************************
+ Functionname:  addHighbandEnergies
+ *******************************************************************************
+ \brief   Add highband energies
+
+ Highband energies are mapped to an array with smaller dimension:
+ Its time resolution is only 1 SBR-timeslot and its frequency resolution
+ is 1 SBR-band. Therefore the data to be fed into the spectralChange
+ function is reduced.
+
+ The values EnergiesM are scaled by the factor (2^19-scaleEnergies[0]) for
+ slots<YBufferWriteOffset and by the factor (2^19-scaleEnergies[1]) for
+ slots>=YBufferWriteOffset.
+
+ \return  total energy in the highband, scaled by factor 2^19
+*******************************************************************************/
+
+static FIXP_DBL addHighbandEnergies(
+    FIXP_DBL **RESTRICT Energies, /*!< input */
+    INT *scaleEnergies, INT YBufferWriteOffset,
+    FIXP_DBL EnergiesM[NUMBER_TIME_SLOTS_2304]
+                      [MAX_FREQ_COEFFS], /*!< Combined output */
+    UCHAR *RESTRICT freqBandTable, INT nSfb, INT sbrSlots, INT timeStep) {
+  INT i, j, k, slotIn, slotOut, scale[2];
+  INT li, ui;
+  FIXP_DBL nrgTotal;
+  FIXP_DBL accu = FL2FXCONST_DBL(0.0f);
+
+  /* Combine QMF-timeslots to SBR-timeslots,
+     combine QMF-bands to SBR-bands,
+     combine Left and Right channel */
+  for (slotOut = 0; slotOut < sbrSlots; slotOut++) {
+    /* Note: Below slotIn = slotOut and not slotIn = timeStep*slotOut
+       because the Energies[] time resolution is always the SBR slot resolution
+       regardless of the timeStep. */
+    slotIn = slotOut;
+
+    for (j = 0; j < nSfb; j++) {
+      accu = FL2FXCONST_DBL(0.0f);
+
+      li = freqBandTable[j];
+      ui = freqBandTable[j + 1];
+
+      for (k = li; k < ui; k++) {
+        for (i = 0; i < timeStep; i++) {
+          accu += Energies[slotIn][k] >> 5;
+        }
+      }
+      EnergiesM[slotOut][j] = accu;
+    }
+  }
+
+  /* scale energies down before add up */
+  scale[0] = fixMin(8, scaleEnergies[0]);
+  scale[1] = fixMin(8, scaleEnergies[1]);
+
+  if ((scaleEnergies[0] - scale[0]) > (DFRACT_BITS - 1) ||
+      (scaleEnergies[1] - scale[1]) > (DFRACT_BITS - 1))
+    nrgTotal = FL2FXCONST_DBL(0.0f);
+  else {
+    /* Now add all energies */
+    accu = FL2FXCONST_DBL(0.0f);
+
+    for (slotOut = 0; slotOut < YBufferWriteOffset; slotOut++) {
+      for (j = 0; j < nSfb; j++) {
+        accu += (EnergiesM[slotOut][j] >> scale[0]);
+      }
+    }
+    nrgTotal = accu >> (scaleEnergies[0] - scale[0]);
+
+    for (slotOut = YBufferWriteOffset; slotOut < sbrSlots; slotOut++) {
+      for (j = 0; j < nSfb; j++) {
+        accu += (EnergiesM[slotOut][j] >> scale[0]);
+      }
+    }
+    nrgTotal = fAddSaturate(nrgTotal, accu >> (scaleEnergies[1] - scale[1]));
+  }
+
+  return (nrgTotal);
+}
+
+/*******************************************************************************
+ Functionname:  FDKsbrEnc_frameSplitter
+ *******************************************************************************
+ \brief   Decides if a FIXFIX-frame shall be splitted into 2 envelopes
+
+ If no transient has been detected before, the frame can still be splitted
+ into 2 envelopes.
+*******************************************************************************/
+void FDKsbrEnc_frameSplitter(
+    FIXP_DBL **Energies, INT *scaleEnergies,
+    HANDLE_SBR_TRANSIENT_DETECTOR h_sbrTransientDetector, UCHAR *freqBandTable,
+    UCHAR *tran_vector, int YBufferWriteOffset, int YBufferSzShift, int nSfb,
+    int timeStep, int no_cols, FIXP_DBL *tonality) {
+  if (tran_vector[1] == 0) /* no transient was detected */
+  {
+    FIXP_DBL delta;
+    INT delta_e;
+    FIXP_DBL(*EnergiesM)[MAX_FREQ_COEFFS];
+    FIXP_DBL EnergyTotal, newLowbandEnergy, newHighbandEnergy;
+    INT border;
+    INT sbrSlots = fMultI(GetInvInt(timeStep), no_cols);
+    C_ALLOC_SCRATCH_START(_EnergiesM, FIXP_DBL,
+                          NUMBER_TIME_SLOTS_2304 * MAX_FREQ_COEFFS)
+
+    FDK_ASSERT(sbrSlots * timeStep == no_cols);
+
+    EnergiesM = (FIXP_DBL(*)[MAX_FREQ_COEFFS])_EnergiesM;
+
+    /*
+      Get Lowband-energy over a range of 2 frames (Look half a frame back and
+      ahead).
+    */
+    newLowbandEnergy = addLowbandEnergies(
+        Energies, scaleEnergies, YBufferWriteOffset, YBufferSzShift,
+        h_sbrTransientDetector->tran_off, freqBandTable, no_cols);
+
+    newHighbandEnergy =
+        addHighbandEnergies(Energies, scaleEnergies, YBufferWriteOffset,
+                            EnergiesM, freqBandTable, nSfb, sbrSlots, timeStep);
+
+    {
+      /* prevLowBandEnergy: Corresponds to 1 frame, starting with half a frame
+         look-behind newLowbandEnergy:  Corresponds to 1 frame, starting in the
+         middle of the current frame */
+      EnergyTotal = (newLowbandEnergy >> 1) +
+                    (h_sbrTransientDetector->prevLowBandEnergy >>
+                     1); /* mean of new and prev LB NRG */
+      EnergyTotal =
+          fAddSaturate(EnergyTotal, newHighbandEnergy); /* Add HB NRG */
+      /* The below border should specify the same position as the middle border
+         of a FIXFIX-frame with 2 envelopes. */
+      border = (sbrSlots + 1) >> 1;
+
+      if ((INT)EnergyTotal & 0xffffffe0 &&
+          (scaleEnergies[0] < 32 || scaleEnergies[1] < 32)) /* i.e. > 31 */ {
+        delta = spectralChange(EnergiesM, scaleEnergies, EnergyTotal, nSfb, 0,
+                               border, YBufferWriteOffset, sbrSlots, &delta_e);
+      } else {
+        delta = FL2FXCONST_DBL(0.0f);
+        delta_e = 0;
+
+        /* set tonality to 0 when energy is very low, since the amplitude
+           resolution should then be low as well                          */
+        *tonality = FL2FXCONST_DBL(0.0f);
+      }
+
+      if (fIsLessThan(h_sbrTransientDetector->split_thr_m,
+                      h_sbrTransientDetector->split_thr_e, delta, delta_e)) {
+        tran_vector[0] = 1; /* Set flag for splitting */
+      } else {
+        tran_vector[0] = 0;
+      }
+    }
+
+    /* Update prevLowBandEnergy */
+    h_sbrTransientDetector->prevLowBandEnergy = newLowbandEnergy;
+    h_sbrTransientDetector->prevHighBandEnergy = newHighbandEnergy;
+    C_ALLOC_SCRATCH_END(_EnergiesM, FIXP_DBL,
+                        NUMBER_TIME_SLOTS_2304 * MAX_FREQ_COEFFS)
+  }
+}
+
+/*
+ * Calculate transient energy threshold for each QMF band
+ */
+static void calculateThresholds(FIXP_DBL **RESTRICT Energies,
+                                INT *RESTRICT scaleEnergies,
+                                FIXP_DBL *RESTRICT thresholds,
+                                int YBufferWriteOffset, int YBufferSzShift,
+                                int noCols, int noRows, int tran_off) {
+  FIXP_DBL mean_val, std_val, temp;
+  FIXP_DBL i_noCols;
+  FIXP_DBL i_noCols1;
+  FIXP_DBL accu, accu0, accu1;
+  int scaleFactor0, scaleFactor1, commonScale;
+  int i, j;
+
+  i_noCols = GetInvInt(noCols + tran_off) << YBufferSzShift;
+  i_noCols1 = GetInvInt(noCols + tran_off - 1) << YBufferSzShift;
+
+  /* calc minimum scale of energies of previous and current frame */
+  commonScale = fixMin(scaleEnergies[0], scaleEnergies[1]);
+
+  /* calc scalefactors to adapt energies to common scale */
+  scaleFactor0 = fixMin((scaleEnergies[0] - commonScale), (DFRACT_BITS - 1));
+  scaleFactor1 = fixMin((scaleEnergies[1] - commonScale), (DFRACT_BITS - 1));
+
+  FDK_ASSERT((scaleFactor0 >= 0) && (scaleFactor1 >= 0));
+
+  /* calculate standard deviation in every subband */
+  for (i = 0; i < noRows; i++) {
+    int startEnergy = (tran_off >> YBufferSzShift);
+    int endEnergy = ((noCols >> YBufferSzShift) + tran_off);
+    int shift;
+
+    /* calculate mean value over decimated energy values (downsampled by 2). */
+    accu0 = accu1 = FL2FXCONST_DBL(0.0f);
+
+    for (j = startEnergy; j < YBufferWriteOffset; j++)
+      accu0 = fMultAddDiv2(accu0, Energies[j][i], i_noCols);
+    for (; j < endEnergy; j++)
+      accu1 = fMultAddDiv2(accu1, Energies[j][i], i_noCols);
+
+    mean_val = ((accu0 << 1) >> scaleFactor0) +
+               ((accu1 << 1) >> scaleFactor1); /* average */
+    shift = fixMax(
+        0, CountLeadingBits(mean_val) -
+               6); /* -6 to keep room for accumulating upto N = 24 values */
+
+    /* calculate standard deviation */
+    accu = FL2FXCONST_DBL(0.0f);
+
+    /* summe { ((mean_val-nrg)^2) * i_noCols1 } */
+    for (j = startEnergy; j < YBufferWriteOffset; j++) {
+      temp = ((FIXP_DBL)mean_val - ((FIXP_DBL)Energies[j][i] >> scaleFactor0))
+             << shift;
+      temp = fPow2Div2(temp);
+      accu = fMultAddDiv2(accu, temp, i_noCols1);
+    }
+    for (; j < endEnergy; j++) {
+      temp = ((FIXP_DBL)mean_val - ((FIXP_DBL)Energies[j][i] >> scaleFactor1))
+             << shift;
+      temp = fPow2Div2(temp);
+      accu = fMultAddDiv2(accu, temp, i_noCols1);
+    }
+    accu <<= 2;
+    std_val = sqrtFixp(accu) >> shift; /* standard deviation */
+
+    /*
+    Take new threshold as average of calculated standard deviation ratio
+    and old threshold if greater than absolute threshold
+    */
+    temp = (commonScale <= (DFRACT_BITS - 1))
+               ? fMult(FL2FXCONST_DBL(0.66f), thresholds[i]) +
+                     (fMult(FL2FXCONST_DBL(0.34f), std_val) >> commonScale)
+               : (FIXP_DBL)0;
+
+    thresholds[i] = fixMax(ABS_THRES, temp);
+
+    FDK_ASSERT(commonScale >= 0);
+  }
+}
+
+/*
+ * Calculate transient levels for each QMF time slot.
+ */
+static void extractTransientCandidates(
+    FIXP_DBL **RESTRICT Energies, INT *RESTRICT scaleEnergies,
+    FIXP_DBL *RESTRICT thresholds, FIXP_DBL *RESTRICT transients,
+    int YBufferWriteOffset, int YBufferSzShift, int noCols, int start_band,
+    int stop_band, int tran_off, int addPrevSamples) {
+  FIXP_DBL i_thres;
+  C_ALLOC_SCRATCH_START(EnergiesTemp, FIXP_DBL, 2 * 32)
+  int tmpScaleEnergies0, tmpScaleEnergies1;
+  int endCond;
+  int startEnerg, endEnerg;
+  int i, j, jIndex, jpBM;
+
+  tmpScaleEnergies0 = scaleEnergies[0];
+  tmpScaleEnergies1 = scaleEnergies[1];
+
+  /* Scale value for first energies, upto YBufferWriteOffset */
+  tmpScaleEnergies0 = fixMin(tmpScaleEnergies0, MAX_SHIFT_DBL);
+  /* Scale value for first energies, from YBufferWriteOffset upwards */
+  tmpScaleEnergies1 = fixMin(tmpScaleEnergies1, MAX_SHIFT_DBL);
+
+  FDK_ASSERT((tmpScaleEnergies0 >= 0) && (tmpScaleEnergies1 >= 0));
+
+  /* Keep addPrevSamples extra previous transient candidates. */
+  FDKmemmove(transients, transients + noCols - addPrevSamples,
+             (tran_off + addPrevSamples) * sizeof(FIXP_DBL));
+  FDKmemclear(transients + tran_off + addPrevSamples,
+              noCols * sizeof(FIXP_DBL));
+
+  endCond = noCols; /* Amount of new transient values to be calculated. */
+  startEnerg = (tran_off - 3) >> YBufferSzShift; /* >>YBufferSzShift because of
+                                                    amount of energy values. -3
+                                                    because of neighbors being
+                                                    watched. */
+  endEnerg =
+      ((noCols + (YBufferWriteOffset << YBufferSzShift)) - 1) >>
+      YBufferSzShift; /* YBufferSzShift shifts because of half energy values. */
+
+  /* Compute differential values with two different weightings in every subband
+   */
+  for (i = start_band; i < stop_band; i++) {
+    FIXP_DBL thres = thresholds[i];
+
+    if ((LONG)thresholds[i] >= 256)
+      i_thres = (LONG)((LONG)MAXVAL_DBL / ((((LONG)thresholds[i])) + 1))
+                << (32 - 24);
+    else
+      i_thres = (LONG)MAXVAL_DBL;
+
+    /* Copy one timeslot and de-scale and de-squish */
+    if (YBufferSzShift == 1) {
+      for (j = startEnerg; j < YBufferWriteOffset; j++) {
+        FIXP_DBL tmp = Energies[j][i];
+        EnergiesTemp[(j << 1) + 1] = EnergiesTemp[j << 1] =
+            tmp >> tmpScaleEnergies0;
+      }
+      for (; j <= endEnerg; j++) {
+        FIXP_DBL tmp = Energies[j][i];
+        EnergiesTemp[(j << 1) + 1] = EnergiesTemp[j << 1] =
+            tmp >> tmpScaleEnergies1;
+      }
+    } else {
+      for (j = startEnerg; j < YBufferWriteOffset; j++) {
+        FIXP_DBL tmp = Energies[j][i];
+        EnergiesTemp[j] = tmp >> tmpScaleEnergies0;
+      }
+      for (; j <= endEnerg; j++) {
+        FIXP_DBL tmp = Energies[j][i];
+        EnergiesTemp[j] = tmp >> tmpScaleEnergies1;
+      }
+    }
+
+    /* Detect peaks in energy values. */
+
+    jIndex = tran_off;
+    jpBM = jIndex + addPrevSamples;
+
+    for (j = endCond; j--; jIndex++, jpBM++) {
+      FIXP_DBL delta, tran;
+      int d;
+
+      delta = (FIXP_DBL)0;
+      tran = (FIXP_DBL)0;
+
+      for (d = 1; d < 4; d++) {
+        delta += EnergiesTemp[jIndex + d]; /* R */
+        delta -= EnergiesTemp[jIndex - d]; /* L */
+        delta -= thres;
+
+        if (delta > (FIXP_DBL)0) {
+          tran = fMultAddDiv2(tran, i_thres, delta);
+        }
+      }
+      transients[jpBM] += (tran << 1);
+    }
+  }
+  C_ALLOC_SCRATCH_END(EnergiesTemp, FIXP_DBL, 2 * 32)
+}
+
+void FDKsbrEnc_transientDetect(HANDLE_SBR_TRANSIENT_DETECTOR h_sbrTran,
+                               FIXP_DBL **Energies, INT *scaleEnergies,
+                               UCHAR *transient_info, int YBufferWriteOffset,
+                               int YBufferSzShift, int timeStep,
+                               int frameMiddleBorder) {
+  int no_cols = h_sbrTran->no_cols;
+  int qmfStartSample;
+  int addPrevSamples;
+  int timeStepShift = 0;
+  int i, cond;
+
+  /* Where to start looking for transients in the transient candidate buffer */
+  qmfStartSample = timeStep * frameMiddleBorder;
+  /* We need to look one value backwards in the transients, so we might need one
+   * more previous value. */
+  addPrevSamples = (qmfStartSample > 0) ? 0 : 1;
+
+  switch (timeStep) {
+    case 1:
+      timeStepShift = 0;
+      break;
+    case 2:
+      timeStepShift = 1;
+      break;
+    case 4:
+      timeStepShift = 2;
+      break;
+  }
+
+  calculateThresholds(Energies, scaleEnergies, h_sbrTran->thresholds,
+                      YBufferWriteOffset, YBufferSzShift, h_sbrTran->no_cols,
+                      h_sbrTran->no_rows, h_sbrTran->tran_off);
+
+  extractTransientCandidates(
+      Energies, scaleEnergies, h_sbrTran->thresholds, h_sbrTran->transients,
+      YBufferWriteOffset, YBufferSzShift, h_sbrTran->no_cols, 0,
+      h_sbrTran->no_rows, h_sbrTran->tran_off, addPrevSamples);
+
+  transient_info[0] = 0;
+  transient_info[1] = 0;
+  transient_info[2] = 0;
+
+  /* Offset by the amount of additional previous transient candidates being
+   * kept. */
+  qmfStartSample += addPrevSamples;
+
+  /* Check for transients in second granule (pick the last value of subsequent
+   * values)  */
+  for (i = qmfStartSample; i < qmfStartSample + no_cols; i++) {
+    cond = (h_sbrTran->transients[i] <
+            fMult(FL2FXCONST_DBL(0.9f), h_sbrTran->transients[i - 1])) &&
+           (h_sbrTran->transients[i - 1] > h_sbrTran->tran_thr);
+
+    if (cond) {
+      transient_info[0] = (i - qmfStartSample) >> timeStepShift;
+      transient_info[1] = 1;
+      break;
+    }
+  }
+
+  if (h_sbrTran->frameShift != 0) {
+    /* transient prediction for LDSBR */
+    /* Check for transients in first <frameShift> qmf-slots of second frame */
+    for (i = qmfStartSample + no_cols;
+         i < qmfStartSample + no_cols + h_sbrTran->frameShift; i++) {
+      cond = (h_sbrTran->transients[i] <
+              fMult(FL2FXCONST_DBL(0.9f), h_sbrTran->transients[i - 1])) &&
+             (h_sbrTran->transients[i - 1] > h_sbrTran->tran_thr);
+
+      if (cond) {
+        int pos = (int)((i - qmfStartSample - no_cols) >> timeStepShift);
+        if ((pos < 3) && (transient_info[1] == 0)) {
+          transient_info[2] = 1;
+        }
+        break;
+      }
+    }
+  }
+}
+
+int FDKsbrEnc_InitSbrTransientDetector(
+    HANDLE_SBR_TRANSIENT_DETECTOR h_sbrTransientDetector,
+    UINT sbrSyntaxFlags, /* SBR syntax flags derived from AOT. */
+    INT frameSize, INT sampleFreq, sbrConfigurationPtr params, int tran_fc,
+    int no_cols, int no_rows, int YBufferWriteOffset, int YBufferSzShift,
+    int frameShift, int tran_off) {
+  INT totalBitrate =
+      params->codecSettings.standardBitrate * params->codecSettings.nChannels;
+  INT codecBitrate = params->codecSettings.bitRate;
+  FIXP_DBL bitrateFactor_m, framedur_fix;
+  INT bitrateFactor_e, tmp_e;
+
+  FDKmemclear(h_sbrTransientDetector, sizeof(SBR_TRANSIENT_DETECTOR));
+
+  h_sbrTransientDetector->frameShift = frameShift;
+  h_sbrTransientDetector->tran_off = tran_off;
+
+  if (codecBitrate) {
+    bitrateFactor_m = fDivNorm((FIXP_DBL)totalBitrate,
+                               (FIXP_DBL)(codecBitrate << 2), &bitrateFactor_e);
+    bitrateFactor_e += 2;
+  } else {
+    bitrateFactor_m = FL2FXCONST_DBL(1.0 / 4.0);
+    bitrateFactor_e = 2;
+  }
+
+  framedur_fix = fDivNorm(frameSize, sampleFreq);
+
+  /* The longer the frames, the more often should the FIXFIX-
+  case transmit 2 envelopes instead of 1.
+  Frame durations below 10 ms produce the highest threshold
+  so that practically always only 1 env is transmitted. */
+  FIXP_DBL tmp = framedur_fix - FL2FXCONST_DBL(0.010);
+
+  tmp = fixMax(tmp, FL2FXCONST_DBL(0.0001));
+  tmp = fDivNorm(FL2FXCONST_DBL(0.000075), fPow2(tmp), &tmp_e);
+
+  bitrateFactor_e = (tmp_e + bitrateFactor_e);
+
+  if (sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) {
+    bitrateFactor_e--; /* divide by 2 */
+  }
+
+  FDK_ASSERT(no_cols <= 32);
+  FDK_ASSERT(no_rows <= 64);
+
+  h_sbrTransientDetector->no_cols = no_cols;
+  h_sbrTransientDetector->tran_thr =
+      (FIXP_DBL)((params->tran_thr << (32 - 24 - 1)) / no_rows);
+  h_sbrTransientDetector->tran_fc = tran_fc;
+  h_sbrTransientDetector->split_thr_m = fMult(tmp, bitrateFactor_m);
+  h_sbrTransientDetector->split_thr_e = bitrateFactor_e;
+  h_sbrTransientDetector->no_rows = no_rows;
+  h_sbrTransientDetector->mode = params->tran_det_mode;
+  h_sbrTransientDetector->prevLowBandEnergy = FL2FXCONST_DBL(0.0f);
+
+  return (0);
+}
+
+#define ENERGY_SCALING_SIZE 32
+
+INT FDKsbrEnc_InitSbrFastTransientDetector(
+    HANDLE_FAST_TRAN_DET h_sbrFastTransientDetector,
+    const INT time_slots_per_frame, const INT bandwidth_qmf_slot,
+    const INT no_qmf_channels, const INT sbr_qmf_1st_band) {
+  int i;
+  int buff_size;
+  FIXP_DBL myExp;
+  FIXP_DBL myExpSlot;
+
+  h_sbrFastTransientDetector->lookahead = TRAN_DET_LOOKAHEAD;
+  h_sbrFastTransientDetector->nTimeSlots = time_slots_per_frame;
+
+  buff_size = h_sbrFastTransientDetector->nTimeSlots +
+              h_sbrFastTransientDetector->lookahead;
+
+  for (i = 0; i < buff_size; i++) {
+    h_sbrFastTransientDetector->delta_energy[i] = FL2FXCONST_DBL(0.0f);
+    h_sbrFastTransientDetector->energy_timeSlots[i] = FL2FXCONST_DBL(0.0f);
+    h_sbrFastTransientDetector->lowpass_energy[i] = FL2FXCONST_DBL(0.0f);
+    h_sbrFastTransientDetector->transientCandidates[i] = 0;
+  }
+
+  FDK_ASSERT(bandwidth_qmf_slot > 0.f);
+  h_sbrFastTransientDetector->stopBand =
+      fMin(TRAN_DET_STOP_FREQ / bandwidth_qmf_slot, no_qmf_channels);
+  h_sbrFastTransientDetector->startBand =
+      fMin(sbr_qmf_1st_band,
+           h_sbrFastTransientDetector->stopBand - TRAN_DET_MIN_QMFBANDS);
+
+  FDK_ASSERT(h_sbrFastTransientDetector->startBand < no_qmf_channels);
+  FDK_ASSERT(h_sbrFastTransientDetector->startBand <
+             h_sbrFastTransientDetector->stopBand);
+  FDK_ASSERT(h_sbrFastTransientDetector->startBand > 1);
+  FDK_ASSERT(h_sbrFastTransientDetector->stopBand > 1);
+
+  /* the energy weighting and adding up has a headroom of 6 Bits,
+     so up to 64 bands can be added without potential overflow. */
+  FDK_ASSERT(h_sbrFastTransientDetector->stopBand -
+                 h_sbrFastTransientDetector->startBand <=
+             64);
+
+/* QMF_HP_dB_SLOPE_FIX says that we want a 20 dB per 16 kHz HP filter.
+   The following lines map this to the QMF bandwidth. */
+#define EXP_E 7 /* 64 (=64) multiplications max, max. allowed sum is 0.5 */
+  myExp = fMultNorm(QMF_HP_dBd_SLOPE_FIX, 0, (FIXP_DBL)bandwidth_qmf_slot,
+                    DFRACT_BITS - 1, EXP_E);
+  myExpSlot = myExp;
+
+  for (i = 0; i < 64; i++) {
+    /* Calculate dBf over all qmf bands:
+       dBf = (10^(0.002266f/10*bw(slot)))^(band) =
+           = 2^(log2(10)*0.002266f/10*bw(slot)*band) =
+           = 2^(0.00075275f*bw(slot)*band)                                   */
+
+    FIXP_DBL dBf_m; /* dBf mantissa        */
+    INT dBf_e;      /* dBf exponent        */
+    INT tmp;
+
+    INT dBf_int;        /* dBf integer part    */
+    FIXP_DBL dBf_fract; /* dBf fractional part */
+
+    /* myExp*(i+1) = myExp_int - myExp_fract
+       myExp*(i+1) is split up here for better accuracy of CalcInvLdData(),
+       for its result can be split up into an integer and a fractional part */
+
+    /* Round up to next integer */
+    FIXP_DBL myExp_int =
+        (myExpSlot & (FIXP_DBL)0xfe000000) + (FIXP_DBL)0x02000000;
+
+    /* This is the fractional part that needs to be substracted */
+    FIXP_DBL myExp_fract = myExp_int - myExpSlot;
+
+    /* Calc integer part */
+    dBf_int = CalcInvLdData(myExp_int);
+    /* The result needs to be re-scaled. The ld(myExp_int) had been scaled by
+       EXP_E, the CalcInvLdData expects the operand to be scaled by
+       LD_DATA_SHIFT. Therefore, the correctly scaled result is
+       dBf_int^(2^(EXP_E-LD_DATA_SHIFT)), which is dBf_int^2 */
+
+    if (dBf_int <=
+        46340) { /* compare with maximum allowed value for signed integer
+                    multiplication, 46340 =
+                    (INT)floor(sqrt((double)(((UINT)1<<(DFRACT_BITS-1))-1))) */
+      dBf_int *= dBf_int;
+
+      /* Calc fractional part */
+      dBf_fract = CalcInvLdData(-myExp_fract);
+      /* The result needs to be re-scaled. The ld(myExp_fract) had been scaled
+         by EXP_E, the CalcInvLdData expects the operand to be scaled by
+         LD_DATA_SHIFT. Therefore, the correctly scaled result is
+         dBf_fract^(2^(EXP_E-LD_DATA_SHIFT)), which is dBf_fract^2 */
+      dBf_fract = fMultNorm(dBf_fract, dBf_fract, &tmp);
+
+      /* Get worst case scaling of multiplication result */
+      dBf_e = (DFRACT_BITS - 1 - tmp) - CountLeadingBits(dBf_int);
+
+      /* Now multiply integer with fractional part of the result, thus resulting
+         in the overall accurate fractional result */
+      dBf_m = fMultNorm(dBf_int, DFRACT_BITS - 1, dBf_fract, tmp, dBf_e);
+
+      myExpSlot += myExp;
+    } else {
+      dBf_m = (FIXP_DBL)0;
+      dBf_e = 0;
+    }
+
+    /* Keep the results */
+    h_sbrFastTransientDetector->dBf_m[i] = dBf_m;
+    h_sbrFastTransientDetector->dBf_e[i] = dBf_e;
+  }
+
+  /* Make sure that dBf is greater than 1.0 (because it should be a highpass) */
+  /* ... */
+
+  return 0;
+}
+
+void FDKsbrEnc_fastTransientDetect(
+    const HANDLE_FAST_TRAN_DET h_sbrFastTransientDetector,
+    const FIXP_DBL *const *Energies, const int *const scaleEnergies,
+    const INT YBufferWriteOffset, UCHAR *const tran_vector) {
+  int timeSlot, band;
+
+  FIXP_DBL max_delta_energy; /* helper to store maximum energy ratio          */
+  int max_delta_energy_scale; /* helper to store scale of maximum energy ratio
+                               */
+  int ind_max = 0; /* helper to store index of maximum energy ratio */
+  int isTransientInFrame = 0;
+
+  const int nTimeSlots = h_sbrFastTransientDetector->nTimeSlots;
+  const int lookahead = h_sbrFastTransientDetector->lookahead;
+  const int startBand = h_sbrFastTransientDetector->startBand;
+  const int stopBand = h_sbrFastTransientDetector->stopBand;
+
+  int *transientCandidates = h_sbrFastTransientDetector->transientCandidates;
+
+  FIXP_DBL *energy_timeSlots = h_sbrFastTransientDetector->energy_timeSlots;
+  int *energy_timeSlots_scale =
+      h_sbrFastTransientDetector->energy_timeSlots_scale;
+
+  FIXP_DBL *delta_energy = h_sbrFastTransientDetector->delta_energy;
+  int *delta_energy_scale = h_sbrFastTransientDetector->delta_energy_scale;
+
+  const FIXP_DBL thr = TRAN_DET_THRSHLD;
+  const INT thr_scale = TRAN_DET_THRSHLD_SCALE;
+
+  /*reset transient info*/
+  tran_vector[2] = 0;
+
+  /* reset transient candidates */
+  FDKmemclear(transientCandidates + lookahead, nTimeSlots * sizeof(int));
+
+  for (timeSlot = lookahead; timeSlot < nTimeSlots + lookahead; timeSlot++) {
+    int i, norm;
+    FIXP_DBL tmpE = FL2FXCONST_DBL(0.0f);
+    int headroomEnSlot = DFRACT_BITS - 1;
+
+    FIXP_DBL smallNRG = FL2FXCONST_DBL(1e-2f);
+    FIXP_DBL denominator;
+    INT denominator_scale;
+
+    /* determine minimum headroom of energy values for this timeslot */
+    for (band = startBand; band < stopBand; band++) {
+      int tmp_headroom = fNormz(Energies[timeSlot][band]) - 1;
+      if (tmp_headroom < headroomEnSlot) {
+        headroomEnSlot = tmp_headroom;
+      }
+    }
+
+    for (i = 0, band = startBand; band < stopBand; band++, i++) {
+      /* energy is weighted by weightingfactor stored in dBf_m array */
+      /* dBf_m index runs from 0 to stopBand-startband               */
+      /* energy shifted by calculated headroom for maximum precision */
+      FIXP_DBL weightedEnergy =
+          fMult(Energies[timeSlot][band] << headroomEnSlot,
+                h_sbrFastTransientDetector->dBf_m[i]);
+
+      /* energy is added up                                                */
+      /* shift by 6 to have a headroom for maximum 64 additions            */
+      /* shift by dBf_e to handle weighting factor dependent scale factors */
+      tmpE +=
+          weightedEnergy >> (6 + (10 - h_sbrFastTransientDetector->dBf_e[i]));
+    }
+
+    /* store calculated energy for timeslot */
+    energy_timeSlots[timeSlot] = tmpE;
+
+    /* calculate overall scale factor for energy of this timeslot */
+    /* =   original scale factor of energies
+     * (-scaleEnergies[0]+2*QMF_SCALE_OFFSET or
+     * -scaleEnergies[1]+2*QMF_SCALE_OFFSET    */
+    /*     depending on YBufferWriteOffset) */
+    /*   + weighting factor scale            (10) */
+    /*   + adding up scale factor            ( 6) */
+    /*   - headroom of energy value          (headroomEnSlot) */
+    if (timeSlot < YBufferWriteOffset) {
+      energy_timeSlots_scale[timeSlot] =
+          (-scaleEnergies[0] + 2 * QMF_SCALE_OFFSET) + (10 + 6) -
+          headroomEnSlot;
+    } else {
+      energy_timeSlots_scale[timeSlot] =
+          (-scaleEnergies[1] + 2 * QMF_SCALE_OFFSET) + (10 + 6) -
+          headroomEnSlot;
+    }
+
+    /* Add a small energy to the denominator, thus making the transient
+       detection energy-dependent. Loud transients are being detected,
+       silent ones not. */
+
+    /* make sure that smallNRG does not overflow */
+    if (-energy_timeSlots_scale[timeSlot - 1] + 1 > 5) {
+      denominator = smallNRG;
+      denominator_scale = 0;
+    } else {
+      /* Leave an additional headroom of 1 bit for this addition. */
+      smallNRG =
+          scaleValue(smallNRG, -(energy_timeSlots_scale[timeSlot - 1] + 1));
+      denominator = (energy_timeSlots[timeSlot - 1] >> 1) + smallNRG;
+      denominator_scale = energy_timeSlots_scale[timeSlot - 1] + 1;
+    }
+
+    delta_energy[timeSlot] =
+        fDivNorm(energy_timeSlots[timeSlot], denominator, &norm);
+    delta_energy_scale[timeSlot] =
+        energy_timeSlots_scale[timeSlot] - denominator_scale + norm;
+  }
+
+  /*get transient candidates*/
+  /* For every timeslot, check if delta(E) exceeds the threshold. If it did,
+     it could potentially be marked as a transient candidate. However, the 2
+     slots before the current one must not be transients with an energy higher
+     than 1.4*E(current). If both aren't transients or if the energy of the
+     current timesolot is more than 1.4 times higher than the energy in the
+     last or the one before the last slot, it is marked as a transient.*/
+
+  FDK_ASSERT(lookahead >= 2);
+  for (timeSlot = lookahead; timeSlot < nTimeSlots + lookahead; timeSlot++) {
+    FIXP_DBL energy_cur_slot_weighted =
+        fMult(energy_timeSlots[timeSlot], FL2FXCONST_DBL(1.0f / 1.4f));
+    if (!fIsLessThan(delta_energy[timeSlot], delta_energy_scale[timeSlot], thr,
+                     thr_scale) &&
+        (((transientCandidates[timeSlot - 2] == 0) &&
+          (transientCandidates[timeSlot - 1] == 0)) ||
+         !fIsLessThan(energy_cur_slot_weighted,
+                      energy_timeSlots_scale[timeSlot],
+                      energy_timeSlots[timeSlot - 1],
+                      energy_timeSlots_scale[timeSlot - 1]) ||
+         !fIsLessThan(energy_cur_slot_weighted,
+                      energy_timeSlots_scale[timeSlot],
+                      energy_timeSlots[timeSlot - 2],
+                      energy_timeSlots_scale[timeSlot - 2]))) {
+      /* in case of strong transients, subsequent
+       * qmf slots might be recognized as transients. */
+      transientCandidates[timeSlot] = 1;
+    }
+  }
+
+  /*get transient with max energy*/
+  max_delta_energy = FL2FXCONST_DBL(0.0f);
+  max_delta_energy_scale = 0;
+  ind_max = 0;
+  isTransientInFrame = 0;
+  for (timeSlot = 0; timeSlot < nTimeSlots; timeSlot++) {
+    int scale = fMax(delta_energy_scale[timeSlot], max_delta_energy_scale);
+    if (transientCandidates[timeSlot] &&
+        ((delta_energy[timeSlot] >> (scale - delta_energy_scale[timeSlot])) >
+         (max_delta_energy >> (scale - max_delta_energy_scale)))) {
+      max_delta_energy = delta_energy[timeSlot];
+      max_delta_energy_scale = scale;
+      ind_max = timeSlot;
+      isTransientInFrame = 1;
+    }
+  }
+
+  /*from all transient candidates take the one with the biggest energy*/
+  if (isTransientInFrame) {
+    tran_vector[0] = ind_max;
+    tran_vector[1] = 1;
+  } else {
+    /*reset transient info*/
+    tran_vector[0] = tran_vector[1] = 0;
+  }
+
+  /*check for transients in lookahead*/
+  for (timeSlot = nTimeSlots; timeSlot < nTimeSlots + lookahead; timeSlot++) {
+    if (transientCandidates[timeSlot]) {
+      tran_vector[2] = 1;
+    }
+  }
+
+  /*update buffers*/
+  for (timeSlot = 0; timeSlot < lookahead; timeSlot++) {
+    transientCandidates[timeSlot] = transientCandidates[nTimeSlots + timeSlot];
+
+    /* fixpoint stuff */
+    energy_timeSlots[timeSlot] = energy_timeSlots[nTimeSlots + timeSlot];
+    energy_timeSlots_scale[timeSlot] =
+        energy_timeSlots_scale[nTimeSlots + timeSlot];
+
+    delta_energy[timeSlot] = delta_energy[nTimeSlots + timeSlot];
+    delta_energy_scale[timeSlot] = delta_energy_scale[nTimeSlots + timeSlot];
+  }
+}
diff --git a/fdk-aac/libSBRenc/src/tran_det.h b/fdk-aac/libSBRenc/src/tran_det.h
new file mode 100644
index 0000000..d10a7db
--- /dev/null
+++ b/fdk-aac/libSBRenc/src/tran_det.h
@@ -0,0 +1,191 @@
+/* -----------------------------------------------------------------------------
+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 encoder library ******************************
+
+   Author(s):
+
+   Description:
+
+*******************************************************************************/
+
+/*!
+  \file
+  \brief  Transient detector prototypes $Revision: 95111 $
+*/
+#ifndef TRAN_DET_H
+#define TRAN_DET_H
+
+#include "sbr_encoder.h"
+#include "sbr_def.h"
+
+typedef struct {
+  FIXP_DBL transients[32 + (32 / 2)];
+  FIXP_DBL thresholds[64];
+  FIXP_DBL tran_thr;    /* Master threshold for transient signals */
+  FIXP_DBL split_thr_m; /* Threshold for splitting FIXFIX-frames into 2 env */
+  INT split_thr_e;      /* Scale for splitting threshold */
+  FIXP_DBL prevLowBandEnergy;  /* Energy of low band */
+  FIXP_DBL prevHighBandEnergy; /* Energy of high band */
+  INT tran_fc;                 /* Number of lowband subbands to discard  */
+  INT no_cols;
+  INT no_rows;
+  INT mode;
+
+  int frameShift;
+  int tran_off; /* Offset for reading energy values. */
+} SBR_TRANSIENT_DETECTOR;
+
+typedef SBR_TRANSIENT_DETECTOR *HANDLE_SBR_TRANSIENT_DETECTOR;
+
+#define TRAN_DET_LOOKAHEAD 2
+#define TRAN_DET_START_FREQ 4500 /*start frequency for transient detection*/
+#define TRAN_DET_STOP_FREQ 13500 /*stop frequency for transient detection*/
+#define TRAN_DET_MIN_QMFBANDS                    \
+  4 /* minimum qmf bands for transient detection \
+     */
+#define QMF_HP_dBd_SLOPE_FIX \
+  FL2FXCONST_DBL(0.00075275f) /* 0.002266f/10 * log2(10) */
+#define TRAN_DET_THRSHLD FL2FXCONST_DBL(5.0f / 8.0f)
+#define TRAN_DET_THRSHLD_SCALE (3)
+
+typedef struct {
+  INT transientCandidates[32 + TRAN_DET_LOOKAHEAD];
+  INT nTimeSlots;
+  INT lookahead;
+  INT startBand;
+  INT stopBand;
+
+  FIXP_DBL dBf_m[64];
+  INT dBf_e[64];
+
+  FIXP_DBL energy_timeSlots[32 + TRAN_DET_LOOKAHEAD];
+  INT energy_timeSlots_scale[32 + TRAN_DET_LOOKAHEAD];
+
+  FIXP_DBL delta_energy[32 + TRAN_DET_LOOKAHEAD];
+  INT delta_energy_scale[32 + TRAN_DET_LOOKAHEAD];
+
+  FIXP_DBL lowpass_energy[32 + TRAN_DET_LOOKAHEAD];
+  INT lowpass_energy_scale[32 + TRAN_DET_LOOKAHEAD];
+} FAST_TRAN_DETECTOR;
+typedef FAST_TRAN_DETECTOR *HANDLE_FAST_TRAN_DET;
+
+INT FDKsbrEnc_InitSbrFastTransientDetector(
+    HANDLE_FAST_TRAN_DET h_sbrFastTransientDetector,
+    const INT time_slots_per_frame, const INT bandwidth_qmf_slot,
+    const INT no_qmf_channels, const INT sbr_qmf_1st_band);
+
+void FDKsbrEnc_fastTransientDetect(
+    const HANDLE_FAST_TRAN_DET h_sbrFastTransientDetector,
+    const FIXP_DBL *const *Energies, const int *const scaleEnergies,
+    const INT YBufferWriteOffset, UCHAR *const tran_vector);
+
+void FDKsbrEnc_transientDetect(
+    HANDLE_SBR_TRANSIENT_DETECTOR h_sbrTransientDetector, FIXP_DBL **Energies,
+    INT *scaleEnergies, UCHAR *tran_vector, int YBufferWriteOffset,
+    int YBufferSzShift, int timeStep, int frameMiddleBorder);
+
+int FDKsbrEnc_InitSbrTransientDetector(
+    HANDLE_SBR_TRANSIENT_DETECTOR h_sbrTransientDetector,
+    UINT sbrSyntaxFlags, /* SBR syntax flags derived from AOT. */
+    INT frameSize, INT sampleFreq, sbrConfigurationPtr params, int tran_fc,
+    int no_cols, int no_rows, int YBufferWriteOffset, int YBufferSzShift,
+    int frameShift, int tran_off);
+
+void FDKsbrEnc_frameSplitter(
+    FIXP_DBL **Energies, INT *scaleEnergies,
+    HANDLE_SBR_TRANSIENT_DETECTOR h_sbrTransientDetector, UCHAR *freqBandTable,
+    UCHAR *tran_vector, int YBufferWriteOffset, int YBufferSzShift, int nSfb,
+    int timeStep, int no_cols, FIXP_DBL *tonality);
+#endif