Snapshot 2bda038c163298531d47394bc2c09e1409c5d0db

Change-Id: If584e579464f28b97d50e51fc76ba654a5536c54

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diff --git a/libSBRenc/include/sbr_encoder.h b/libSBRenc/include/sbr_encoder.h
new file mode 100644
index 0000000..992c20c
--- /dev/null
+++ b/libSBRenc/include/sbr_encoder.h
@@ -0,0 +1,394 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/***************************  Fraunhofer IIS ***********************
+
+   Author(s):
+   Description: SBR encoder top level processing prototype
+
+******************************************************************************/
+
+#ifndef __SBR_ENCODER_H
+#define __SBR_ENCODER_H
+
+#include "common_fix.h"
+#include "FDK_audio.h"
+
+#include "FDK_bitstream.h"
+
+/* core coder helpers */
+#define MAX_TRANS_FAC         8
+#define MAX_CODEC_FRAME_RATIO 2
+#define MAX_PAYLOAD_SIZE    256
+
+typedef struct
+{
+  INT bitRate;
+  INT nChannels;
+  INT sampleFreq;
+  INT transFac;
+  INT standardBitrate;
+} CODEC_PARAM;
+
+typedef enum
+{
+  SBR_MONO,
+  SBR_LEFT_RIGHT,
+  SBR_COUPLING,
+  SBR_SWITCH_LRC
+} SBR_STEREO_MODE;
+
+/* bitstream syntax flags */
+enum
+{
+  SBR_SYNTAX_LOW_DELAY = 0x0001,
+  SBR_SYNTAX_SCALABLE  = 0x0002,
+  SBR_SYNTAX_CRC       = 0x0004,
+  SBR_SYNTAX_DRM_CRC   = 0x0008
+};
+
+typedef struct
+{
+  UINT             bitrateFrom;      /*!< inclusive */
+  UINT             bitrateTo;        /*!< exclusive */
+
+  USHORT           sampleRate;       /*!<   */
+  UCHAR            numChannels;      /*!<   */
+
+  UCHAR            startFreq;        /*!< bs_start_freq */
+  UCHAR            startFreqSpeech;  /*!< bs_start_freq for speech config flag */
+  UCHAR            stopFreq;         /*!< bs_stop_freq */
+  UCHAR            stopFreqSpeech;   /*!< bs_stop_freq for speech config flag */
+
+  UCHAR            numNoiseBands;    /*!<   */
+  UCHAR            noiseFloorOffset; /*!<   */
+  SCHAR            noiseMaxLevel;    /*!<   */
+  SBR_STEREO_MODE  stereoMode;       /*!<   */
+  UCHAR            freqScale;        /*!<   */
+} sbrTuningTable_t;
+
+typedef struct sbrConfiguration
+{
+  /*
+     core coder dependent configurations
+  */
+  CODEC_PARAM codecSettings;  /*!< Core coder settings. To be set from core coder. */
+  INT SendHeaderDataTime;     /*!< SBR header send update frequency in ms. */
+  INT useWaveCoding;          /*!< Flag: usage of wavecoding tool. */
+  INT crcSbr;                 /*!< Flag: usage of SBR-CRC. */
+  INT dynBwSupported;         /*!< Flag: support for dynamic bandwidth in this combination. */
+  INT parametricCoding;       /*!< Flag: usage of parametric coding tool. */
+  int freq_res_fixfix[3];     /*!< Frequency resolution of envelopes in frame class FIXFIX
+                                 0=1 Env; 1=2 Env; 2=4 Env; */
+  /*
+     core coder dependent tuning parameters
+  */
+  INT tran_thr;             /*!< SBR transient detector threshold (* 100). */
+  INT noiseFloorOffset;     /*!< Noise floor offset.      */
+  UINT useSpeechConfig;     /*!< Flag: adapt tuning parameters according to speech. */
+
+
+
+  /*
+     core coder independent configurations
+  */
+  INT sbrFrameSize;           /*!< SBR frame size in samples. Will be calculated from core coder settings. */
+  INT sbr_data_extra;         /*!< Flag usage of data extra. */
+  INT amp_res;                /*!< Amplitude resolution. */
+  INT ana_max_level;          /*!< Noise insertion maximum level. */
+  INT tran_fc;                /*!< Transient detector start frequency. */
+  INT tran_det_mode;          /*!< Transient detector mode. */
+  INT spread;                 /*!< Flag: usage of SBR spread. */
+  INT stat;                   /*!< Flag: usage of static framing. */
+  INT e;                      /*!< Number of envelopes when static framing is chosen. */
+  SBR_STEREO_MODE stereoMode; /*!< SBR stereo mode. */
+  INT deltaTAcrossFrames;     /*!< Flag: allow time-delta coding. */
+  FIXP_DBL dF_edge_1stEnv;    /*!< Extra fraction delta-F coding is allowed to be more expensive. */
+  FIXP_DBL dF_edge_incr;      /*!< Increment dF_edge_1stEnv this much if dT-coding was used this frame. */
+  INT sbr_invf_mode;          /*!< Inverse filtering mode. */
+  INT sbr_xpos_mode;          /*!< Transposer mode. */
+  INT sbr_xpos_ctrl;          /*!< Transposer control. */
+  INT sbr_xpos_level;         /*!< Transposer 3rd order level. */
+  INT startFreq;              /*!< The start frequency table index. */
+  INT stopFreq;               /*!< The stop frequency table index. */
+  INT useSaPan;               /*!< Flag: usage of SAPAN stereo. */
+  INT dynBwEnabled;           /*!< Flag: usage of dynamic bandwidth. */
+  INT bParametricStereo;      /*!< Flag: usage of parametric stereo coding tool. */
+  INT bDownSampledSbr;        /*!< Signal downsampled SBR is used. */
+
+  /*
+     header_extra1 configuration
+  */
+  UCHAR freqScale;            /*!< Frequency grouping. */
+  INT alterScale;             /*!< Scale resolution. */
+  INT sbr_noise_bands;        /*!< Number of noise bands. */
+
+
+  /*
+     header_extra2 configuration
+  */
+  INT sbr_limiter_bands;      /*!< Number of limiter bands. */
+  INT sbr_limiter_gains;      /*!< Gain of limiter. */
+  INT sbr_interpol_freq;      /*!< Flag: use interpolation in freq. direction. */
+  INT sbr_smoothing_length;   /*!< Flag: choose length 4 or 0 (=on, off). */
+  UCHAR init_amp_res_FF;
+} sbrConfiguration, *sbrConfigurationPtr ;
+
+typedef struct
+{
+  UINT sbrSyntaxFlags;                  /**< SBR syntax flags derived from AOT. */
+  INT nChannels;                        /**< Number of channels.  */
+
+  INT nSfb[2];                          /**< Number of SBR scalefactor bands for LO_RES and HI_RES (?) */
+  INT num_Master;                       /**< Number of elements in v_k_master. */
+  INT sampleFreq;                       /**< SBR sampling frequency. */
+  INT frameSize;
+  INT xOverFreq;                        /**< The SBR start frequency. */
+  INT dynXOverFreq;                     /**< Used crossover frequency when dynamic bandwidth is enabled. */
+  INT noQmfBands;                       /**< Number of QMF frequency bands. */
+  INT noQmfSlots;                       /**< Number of QMF slots. */
+
+  UCHAR *freqBandTable[2];              /**< Frequency table for low and hires, only MAX_FREQ_COEFFS/2 +1 coeefs actually needed for lowres. */
+  UCHAR *v_k_master;                    /**< Master BandTable where freqBandTable is derived from. */
+
+
+  SBR_STEREO_MODE stereoMode;
+  INT noEnvChannels;                    /**< Number of envelope channels. */
+
+  INT useWaveCoding;                    /**< Flag indicates whether to use wave coding at all.      */
+  INT useParametricCoding;              /**< Flag indicates whether to use para coding at all.      */
+  INT xposCtrlSwitch;                   /**< Flag indicates whether to switch xpos ctrl on the fly. */
+  INT switchTransposers;                /**< Flag indicates whether to switch xpos on the fly .     */
+  UCHAR initAmpResFF;
+} SBR_CONFIG_DATA;
+
+typedef SBR_CONFIG_DATA *HANDLE_SBR_CONFIG_DATA;
+
+typedef struct {
+  MP4_ELEMENT_ID elType;
+  INT bitRate;
+  int instanceTag;
+  UCHAR fParametricStereo;
+  UCHAR nChannelsInEl;
+  UCHAR ChannelIndex[2];
+} SBR_ELEMENT_INFO;
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+typedef struct SBR_ENCODER *HANDLE_SBR_ENCODER;
+
+/**
+ * \brief  Get the max required input buffer size including delay balancing space
+ *         for N audio channels.
+ * \param noChannels  Number of audio channels.
+ * \return            Max required input buffer size in bytes.
+ */
+INT sbrEncoder_GetInBufferSize(int noChannels);
+
+INT sbrEncoder_Open(
+        HANDLE_SBR_ENCODER  *phSbrEncoder,
+        INT                  nElements,
+        INT                  nChannels,
+        INT                  supportPS
+        );
+
+/**
+ * \brief get closest working bit rate to specified desired bit rate for a single SBR element
+ * \param bitRate the desired target bit rate
+ * \param numChannels the amount of audio channels
+ * \param coreSampleRate the sample rate of the core coder
+ * \param the current Audio Object Type
+ * \return closest working bit rate to bitRate value
+ */
+UINT sbrEncoder_LimitBitRate(UINT bitRate, UINT numChannels, UINT coreSampleRate, AUDIO_OBJECT_TYPE aot);
+
+/**
+ * \brief                  Initialize SBR Encoder instance.
+ * \param phSbrEncoder     Pointer to a SBR Encoder instance.
+ * \param elInfo           Structure that describes the element/channel arrangement.
+ * \param noElements       Amount of elements described in elInfo.
+ * \param inputBuffer      Pointer to the encoder audio buffer
+ * \param bandwidth        Returns the core audio encoder bandwidth (output)
+ * \param bufferOffset     Returns the offset for the audio input data in order to do delay balancing.
+ * \param numChannels      Input: Encoder input channels. output: core encoder channels.
+ * \param sampleRate       Input: Encoder samplerate. output core encoder samplerate.
+ * \param frameLength      Input: Encoder frameLength. output core encoder frameLength.
+ * \param aot              Input: Desired AOT. output AOT to be used after parameter checking.
+ * \param delay            Input: core encoder delay. Output: total delay because of SBR.
+ * \param transformFactor  The core encoder transform factor (blockswitching).
+ * \return                 0 on success, and non-zero if failed.
+ */
+INT sbrEncoder_Init( HANDLE_SBR_ENCODER hSbrEncoder,
+                     SBR_ELEMENT_INFO elInfo[(6)],
+                     int              noElements,
+                     INT_PCM *inputBuffer,
+                     INT   *bandwidth,
+                     INT   *bufferOffset,
+                     INT   *numChannels,
+                     INT   *sampleRate,
+                     INT   *frameLength,
+                     AUDIO_OBJECT_TYPE *aot,
+                     int *delay,
+                     int  transformFactor,
+                     ULONG statesInitFlag
+                    );
+
+/**
+ * \brief             Do delay line buffers housekeeping. To be called after each encoded audio frame.
+ * \param hEnvEnc     SBR Encoder handle.
+ * \param timeBuffer  Pointer to the encoder audio buffer.
+ * \return            0 on success, and non-zero if failed.
+ */
+INT sbrEncoder_UpdateBuffers(HANDLE_SBR_ENCODER hEnvEnc,
+                             INT_PCM *timeBuffer
+                            );
+
+/**
+ * \brief               Close SBR encoder instance.
+ * \param phEbrEncoder  Handle of SBR encoder instance to be closed.
+ * \return              void
+ */
+void sbrEncoder_Close(HANDLE_SBR_ENCODER *phEbrEncoder);
+
+/**
+ * \brief               Encode SBR data of one complete audio frame.
+ * \param hEnvEncoder   Handle of SBR encoder instance.
+ * \param samples       Time samples, always interleaved.
+ * \param timeInStride  Channel stride factor of samples buffer.
+ * \param sbrDataBits   Size of SBR payload in bits.
+ * \param sbrData       SBR payload.
+ * \return              0 on success, and non-zero if failed.
+ */
+INT sbrEncoder_EncodeFrame(HANDLE_SBR_ENCODER  hEnvEncoder,
+                           INT_PCM            *samples,
+                           UINT                timeInStride,
+                           UINT                sbrDataBits[(6)],
+                           UCHAR               sbrData[(6)][MAX_PAYLOAD_SIZE]
+                          );
+
+/**
+ * \brief               Write SBR headers of one SBR element.
+ * \param sbrEncoder    Handle of the SBR encoder instance.
+ * \param hBs           Handle of bit stream handle to write SBR header to.
+ * \param element_index Index of the SBR element which header should be written.
+ * \param fSendHeaders  Flag indicating that the SBR encoder should send more headers in the SBR payload or not.
+ * \return              void
+ */
+void sbrEncoder_GetHeader(SBR_ENCODER   *sbrEncoder,
+                          HANDLE_FDK_BITSTREAM hBs,
+                          INT            element_index,
+                          int            fSendHeaders);
+
+/**
+ * \brief              SBR encoder bitrate estimation.
+ * \param hSbrEncoder  SBR encoder handle.
+ * \return             Estimated bitrate.
+ */
+INT sbrEncoder_GetEstimateBitrate(HANDLE_SBR_ENCODER hSbrEncoder);
+
+
+/**
+ * \brief              Delay between input data and downsampled output data.
+ * \param hSbrEncoder  SBR encoder handle.
+ * \return             Delay.
+ */
+INT sbrEncoder_GetInputDataDelay(HANDLE_SBR_ENCODER hSbrEncoder);
+
+/**
+ * \brief       Get decoder library version info.
+ * \param info  Pointer to an allocated LIB_INFO struct, where library info is written to.
+ * \return      0 on sucess.
+ */
+INT sbrEncoder_GetLibInfo(LIB_INFO *info);
+
+void sbrPrintRAM(void);
+
+void sbrPrintROM(void);
+
+#ifdef __cplusplus
+        }
+#endif
+
+#endif /* ifndef __SBR_MAIN_H */
diff --git a/libSBRenc/src/bit_sbr.cpp b/libSBRenc/src/bit_sbr.cpp
new file mode 100644
index 0000000..734a8aa
--- /dev/null
+++ b/libSBRenc/src/bit_sbr.cpp
@@ -0,0 +1,1059 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/*!
+  \file
+  \brief  SBR bit writing routines  
+*/
+
+
+#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
+                                          ,UINT  sbrSyntaxFlags
+                                          );
+
+
+
+static INT encodeSbrChannelPairElement (HANDLE_SBR_ENV_DATA      sbrEnvDataLeft,
+                                        HANDLE_SBR_ENV_DATA      sbrEnvDataRight,
+                                        HANDLE_PARAMETRIC_STEREO hParametricStereo,
+                                        HANDLE_FDK_BITSTREAM     hBitStream,
+                                        INT                      coupling);
+
+
+static INT encodeSbrGrid (HANDLE_SBR_ENV_DATA   sbrEnvData,
+                          HANDLE_FDK_BITSTREAM  hBitStream);
+
+static int encodeLowDelaySbrGrid ( HANDLE_SBR_ENV_DATA sbrEnvData,
+                                   HANDLE_FDK_BITSTREAM hBitStream,
+                                   int transmitFreqs);
+
+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)
+{
+  int bits;
+
+  bits = 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);
+    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
+                               ,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));
+      }
+  }
+  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,
+                             INT                   coupling)
+{
+  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));
+          }
+    } 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));
+      }
+
+      /* 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));
+      }
+    } 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,
+                         int transmitFreqs
+                       )
+{
+  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/libSBRenc/src/bit_sbr.h b/libSBRenc/src/bit_sbr.h
new file mode 100644
index 0000000..bf170c1
--- /dev/null
+++ b/libSBRenc/src/bit_sbr.h
@@ -0,0 +1,261 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/*!
+  \file
+  \brief  SBR bit writing  
+*/
+#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 NrSendHeaderData;            /**< input from commandline */
+  INT CountSendHeaderData;         /**< modulo count. If < 0 then no counting is done (no SBR headers) */
+};
+
+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 sbrdata
+  */
+  SR_MODE sampleRateMode;
+
+  /*
+    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;
+  INT freq_res_fixfix;
+
+
+  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;
+
+  /* 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/libSBRenc/src/cmondata.h b/libSBRenc/src/cmondata.h
new file mode 100644
index 0000000..c3be1d7
--- /dev/null
+++ b/libSBRenc/src/cmondata.h
@@ -0,0 +1,110 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/*!
+  \file
+  \brief  Core Coder's and SBR's shared data structure definition  
+*/
+#ifndef __SBR_CMONDATA_H
+#define __SBR_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/libSBRenc/src/code_env.cpp b/libSBRenc/src/code_env.cpp
new file mode 100644
index 0000000..7c169e6
--- /dev/null
+++ b/libSBRenc/src/code_env.cpp
@@ -0,0 +1,641 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+#include "code_env.h"
+#include "sbr_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)(((INT)(LONG)h_sbrCodeEnvelope->dF_edge_incr*h_sbrCodeEnvelope->dF_edge_incr_fac) >> (DFRACT_BITS-16));
+
+  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/libSBRenc/src/code_env.h b/libSBRenc/src/code_env.h
new file mode 100644
index 0000000..dd2b9ae
--- /dev/null
+++ b/libSBRenc/src/code_env.h
@@ -0,0 +1,153 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/*!
+  \file
+  \brief  DPCM Envelope coding  
+*/
+
+#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/libSBRenc/src/env_bit.cpp b/libSBRenc/src/env_bit.cpp
new file mode 100644
index 0000000..55c6967
--- /dev/null
+++ b/libSBRenc/src/env_bit.cpp
@@ -0,0 +1,250 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/*!
+  \file
+  \brief  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/libSBRenc/src/env_bit.h b/libSBRenc/src/env_bit.h
new file mode 100644
index 0000000..38578f8
--- /dev/null
+++ b/libSBRenc/src/env_bit.h
@@ -0,0 +1,126 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/*!
+  \file
+  \brief  Remaining SBR Bit Writing Routines
+*/
+
+#ifndef BIT_ENV_H
+#define BIT_ENV_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                  crcReg,
+                                UINT                 sbrSyntaxFlags);
+
+
+
+
+
+#endif /* #ifndef BIT_ENV_H */
diff --git a/libSBRenc/src/env_est.cpp b/libSBRenc/src/env_est.cpp
new file mode 100644
index 0000000..9423f27
--- /dev/null
+++ b/libSBRenc/src/env_est.cpp
@@ -0,0 +1,1866 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+#include "env_est.h"
+#include "tran_det.h"
+
+#include "qmf.h"
+
+#include "fram_gen.h"
+#include "bit_sbr.h"
+#include "cmondata.h"
+#include "sbr_ram.h"
+
+
+#include "genericStds.h"
+
+#define QUANT_ERROR_THRES 200
+#define Y_NRG_SCALE 5 /* noCols = 32 -> shift(5) */
+
+
+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};
+
+
+/***************************************************************************/
+/*!
+
+  \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, QMF_CHANNELS*QMF_MAX_TIME_SLOTS/2);
+
+  FDK_ASSERT(numberBands <= QMF_CHANNELS);
+  FDK_ASSERT(numberCols <= QMF_MAX_TIME_SLOTS);
+
+  /* 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+=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 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, QMF_CHANNELS*QMF_MAX_TIME_SLOTS/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,                 /*!< 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, QMF_MAX_TIME_SLOTS*QMF_CHANNELS/2);
+
+  FDK_ASSERT(numberBands <= QMF_CHANNELS);
+  FDK_ASSERT(numberCols <= QMF_MAX_TIME_SLOTS/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 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 2 consecutive timeslots */
+      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 2 timeslots */
+        tr0 = r0[j]; ti0 = i0[j];
+
+        /* Scale QMF Values and Calc Energy of both timeslots */
+        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, QMF_MAX_TIME_SLOTS*QMF_CHANNELS/2);
+}
+
+/***************************************************************************/
+/*!
+
+  \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  */
+                                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>>6 = 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-1-6); */ /* 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, FIXP_DBL nrgSum, 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));
+
+    /* 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 i, 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 (i = 0; i < nEnvelopes; i++) {
+
+    FIXP_DBL pNrgLeft[QMF_MAX_TIME_SLOTS];
+    FIXP_DBL pNrgRight[QMF_MAX_TIME_SLOTS];
+    FIXP_DBL envNrgLeft  = FL2FXCONST_DBL(0.0f);
+    FIXP_DBL envNrgRight = FL2FXCONST_DBL(0.0f);
+    int      missingHarmonic[QMF_MAX_TIME_SLOTS];
+    int      count[QMF_MAX_TIME_SLOTS];
+
+    start_pos = timeStep * frame_info->borders[i];
+    stop_pos = timeStep * frame_info->borders[i + 1];
+    freq_res = frame_info->freqRes[i];
+    no_of_bands = h_con->nSfb[freq_res];
+
+    if (i == short_env) {
+      stop_pos -= fixMax(2, timeStep);  /* consider at least 2 QMF slots less for short envelopes (envelopes just before transients) */
+    }
+
+    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;
+      envNrgRight += nrgRight;
+
+    } /* 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, no_of_bands);
+        if (stereoMode == SBR_COUPLING) {
+          nrgRight = nmhLoweringEnergy(nrgRight, envNrgRight, no_of_bands);
+        }
+      }
+
+      if (stereoMode == SBR_COUPLING) {
+        /* calc operation later with log */
+        nrgLeft2 = nrgLeft;
+        nrgLeft = (nrgRight + nrgLeft) >> 1;
+      }
+
+      /* nrgLeft = f20_log2(nrgLeft / (PFLOAT)(count * h_sbr->sbrQmf.no_channels))+(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]*h_con->noQmfBands)) << (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));
+      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;
+
+        if (nrgRight != FL2FXCONST_DBL(0.0f)) {
+          int sc0 = CountLeadingBits(nrgLeft2);
+          int 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;
+        }
+        else
+          nrgRight =  FL2FXCONST_DBL(0.5f);   /* ld64(4294967296.0f) */
+
+        /* 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 */
+
+  } /* i*/
+}
+
+/***************************************************************************/
+/*!
+
+  \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]);
+
+
+
+  /*
+    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);
+
+
+
+  /*
+    Transient detection COEFF Transform OK
+  */
+  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);
+
+
+}
+
+/***************************************************************************/
+/*!
+
+  \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;
+  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,
+                                                     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,
+                                                       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 (hEnvChan->encEnvData.ldGrid)
+       hEnvChan->encEnvData.currentAmpResFF = (AMP_RES)h_con->initAmpResFF;
+     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))
+    {
+      int i;
+
+      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)*QMF_CHANNELS);
+      }
+      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* YBuffer = GetRam_Sbr_envYBuffer(channel);
+
+  FDKmemclear(hSbrCut,sizeof(SBR_EXTRACT_ENVELOPE));
+  hSbrCut->p_YBuffer = YBuffer;
+
+
+  for (i = 0; i < (QMF_MAX_TIME_SLOTS>>1); i++) {
+    hSbrCut->YBuffer[i] = YBuffer + (i*QMF_CHANNELS);
+  }
+  FIXP_DBL *YBufferDyn = GetRam_Sbr_envYBuffer(chInEl, dynamic_RAM);
+  INT n=0;
+  for (; i < QMF_MAX_TIME_SLOTS; i++,n++) {
+    hSbrCut->YBuffer[i] = YBufferDyn + (n*QMF_CHANNELS);
+  }
+
+  FIXP_DBL* rBuffer = GetRam_Sbr_envRBuffer(0, dynamic_RAM);
+  FIXP_DBL* iBuffer = GetRam_Sbr_envIBuffer(0, dynamic_RAM);
+
+  for (i = 0; i < QMF_MAX_TIME_SLOTS; i++) {
+    hSbrCut->rBuffer[i] = rBuffer + (i*QMF_CHANNELS);
+    hSbrCut->iBuffer[i] = iBuffer + (i*QMF_CHANNELS);
+  }
+
+  return 0;
+}
+
+
+/***************************************************************************/
+/*!
+
+  \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;
+#ifndef FULL_DELAY
+    hSbrCut->YBufferWriteOffset = (no_cols>>1)+off*time_step;
+#else
+    hSbrCut->YBufferWriteOffset = no_cols+off*time_step;
+#endif
+  } 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        <=   QMF_CHANNELS);
+
+  /* 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<=QMF_MAX_TIME_SLOTS);
+
+  FIXP_DBL *YBufferDyn = GetRam_Sbr_envYBuffer(chInEl, dynamic_RAM);
+  INT n=0;
+  for (i=(QMF_MAX_TIME_SLOTS>>1); i < QMF_MAX_TIME_SLOTS; i++,n++) {
+    hSbrCut->YBuffer[i] = YBufferDyn + (n*QMF_CHANNELS);
+  }
+
+  if(statesInitFlag) {
+    for (i=0; i<YBufferLength; i++) {
+      FDKmemclear( hSbrCut->YBuffer[i],QMF_CHANNELS*sizeof(FIXP_DBL));
+    }
+  }
+
+  for (i = 0; i < rBufferLength; i++) {
+    FDKmemclear( hSbrCut->rBuffer[i],QMF_CHANNELS*sizeof(FIXP_DBL));
+    FDKmemclear( hSbrCut->iBuffer[i],QMF_CHANNELS*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/libSBRenc/src/env_est.h b/libSBRenc/src/env_est.h
new file mode 100644
index 0000000..4c30a50
--- /dev/null
+++ b/libSBRenc/src/env_est.h
@@ -0,0 +1,224 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/*!
+  \file
+  \brief  Envelope estimation structs and prototypes  
+*/
+#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[QMF_MAX_TIME_SLOTS];
+  FIXP_DBL  *iBuffer[QMF_MAX_TIME_SLOTS];
+
+  FIXP_DBL  *p_YBuffer;
+
+  FIXP_DBL  *YBuffer[QMF_MAX_TIME_SLOTS];
+  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
+{
+  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/libSBRenc/src/fram_gen.cpp b/libSBRenc/src/fram_gen.cpp
new file mode 100644
index 0000000..afef6e4
--- /dev/null
+++ b/libSBRenc/src/fram_gen.cpp
@@ -0,0 +1,2053 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+#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,
+                                  INT 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
+                               );
+
+
+/*!
+  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,
+                    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;
+  INT freq_res_fixfix = hSbrEnvFrame->freq_res_fixfix;
+
+  if (ldGrid) {
+    /* in case there was a transient at the very end of the previous frame, start with a transient envelope */
+    if(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
+    ---------------------------------------------------------------------------*/
+    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
+                           );
+
+      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,
+                        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
+                               )
+{
+  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];
+    hSbrFrameInfo->freqRes[i] = freqResTable[k];
+    hSbrGrid->v_f[i] = freqResTable[k];
+  }
+
+  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,
+                          INT freq_res_fixfix
+                          ,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 = freq_res_fixfix;
+
+  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;
+  } 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;
+      length_v_freq = 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: Calculates frame_info struct from control signal.
+
+ Arguments:   hSbrGrid - source
+              hSbrFrameInfo - destination
+
+ Return:      void; hSbrFrameInfo contains the updated FRAME_INFO struct
+
+*******************************************************************************/
+static void
+ctrlSignal2FrameInfo (HANDLE_SBR_GRID hSbrGrid,
+                      HANDLE_SBR_FRAME_INFO hSbrFrameInfo,
+                      INT freq_res_fixfix)
+{
+  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);
+
+    /* At this point all frequency resolutions are set to FREQ_RES_HIGH, so
+     * only if freq_res_fixfix is set to FREQ_RES_LOW, they all have to be
+     * changed.
+     * snd */
+    if (freq_res_fixfix == FREQ_RES_LOW) {
+      for (i = 0; i < hSbrFrameInfo->nEnvelopes; i++) {
+        hSbrFrameInfo->freqRes[i] = FREQ_RES_LOW;
+      }
+    }
+    /* ELD: store current frequency resolution */
+    hSbrGrid->v_f[0] = hSbrFrameInfo->freqRes[0];
+    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/libSBRenc/src/fram_gen.h b/libSBRenc/src/fram_gen.h
new file mode 100644
index 0000000..fe4b262
--- /dev/null
+++ b/libSBRenc/src/fram_gen.h
@@ -0,0 +1,305 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/*!
+  \file
+  \brief  Framing generator prototypes and structs  
+*/
+#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 */
+
+#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  0
+#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 */
+  INT freq_res_fixfix;      /*!< envelope frequency resolution to use for bs_frame_class = FIXFIX */
+
+  /* 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,
+                          INT freq_res_fixfix
+                          ,int ldGrid
+                          );
+
+HANDLE_SBR_FRAME_INFO
+FDKsbrEnc_frameInfoGenerator (HANDLE_SBR_ENVELOPE_FRAME hSbrEnvFrame,
+                    UCHAR *v_transient_info,
+                    UCHAR *v_transient_info_pre,
+                    int ldGrid,
+                    const int *v_tuning);
+
+#endif
diff --git a/libSBRenc/src/invf_est.cpp b/libSBRenc/src/invf_est.cpp
new file mode 100644
index 0000000..788ab7c
--- /dev/null
+++ b/libSBRenc/src/invf_est.cpp
@@ -0,0 +1,529 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+#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 FIXP_DBL *quantStepsSbr  = detectorParams->quantStepsSbr;
+  const FIXP_DBL *quantStepsOrig = detectorParams->quantStepsOrig;
+  const FIXP_DBL *nrgBorders     = detectorParams->nrgBorders;
+  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,quantStepsSbr,numRegionsSbr*sizeof(FIXP_DBL));
+  FDKmemcpy(quantStepsOrigTmp,quantStepsOrig,numRegionsOrig*sizeof(FIXP_DBL));
+
+  if(*prevRegionSbr < numRegionsSbr)
+    quantStepsSbrTmp[*prevRegionSbr] = quantStepsSbr[*prevRegionSbr] + hysteresis;
+  if(*prevRegionSbr > 0)
+    quantStepsSbrTmp[*prevRegionSbr - 1] = quantStepsSbr[*prevRegionSbr - 1] - hysteresis;
+
+  if(*prevRegionOrig < numRegionsOrig)
+    quantStepsOrigTmp[*prevRegionOrig] = quantStepsOrig[*prevRegionOrig] + hysteresis;
+  if(*prevRegionOrig > 0)
+    quantStepsOrigTmp[*prevRegionOrig - 1] = quantStepsOrig[*prevRegionOrig - 1] - hysteresis;
+
+  regionSbr  = findRegion(sbrQuotaMeanFilt, quantStepsSbrTmp, numRegionsSbr);
+  regionOrig = findRegion(origQuotaMeanFilt, quantStepsOrigTmp, numRegionsOrig);
+  regionNrg  = findRegion(nrg,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/libSBRenc/src/invf_est.h b/libSBRenc/src/invf_est.h
new file mode 100644
index 0000000..7c66cf6
--- /dev/null
+++ b/libSBRenc/src/invf_est.h
@@ -0,0 +1,175 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/*!
+  \file
+  \brief  Inverse Filtering detection prototypes  
+*/
+#ifndef _INV_FILT_DET_H
+#define _INV_FILT_DET_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/libSBRenc/src/mh_det.cpp b/libSBRenc/src/mh_det.cpp
new file mode 100644
index 0000000..a7fa208
--- /dev/null
+++ b/libSBRenc/src/mh_det.cpp
@@ -0,0 +1,1438 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+#include "mh_det.h"
+
+#include "sbr_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,li, ui,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++) {
+      li = pFreqBandTable[i];
+      ui = pFreqBandTable[i+1];
+
+      /* detection in adjacent channels.*/
+      if(pAddHarmSfb[i] && pAddHarmSfb[i+1]) {
+        FIXP_DBL maxVal1, maxVal2;
+        INT maxPos1, maxPos2, maxPosTime1, maxPosTime2;
+
+        li = pFreqBandTable[i];
+        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     Do detection for one tonality estimate.
+
+
+  \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))
+                  ? fixMax(fMult(mhThresh.decayGuideDiff,guideVectors.guideVectorDiff[i]), mhThresh.thresHoldDiffGuide)
+                  : mhThresh.thresHoldDiff;
+
+    thresTemp = fixMin(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){
+
+    if(totNoEst > 1){
+      start = detectionStart;
+
+      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_mem, FIXP_DBL, (3*MAX_NO_OF_ESTIMATES/2*MAX_FREQ_COEFFS));
+  FIXP_DBL *scratch = scratch_mem;
+  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);
+
+  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;
+}
+
+
+/**************************************************************************/
+/*!
+  \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);
+
+  switch(frameSize){
+  case 2048:
+      hs->transientPosOffset = FRAME_MIDDLE_SLOT_2048;
+      hs->timeSlots          = NUMBER_TIME_SLOTS_2048;
+      break;
+  case 1920:
+      hs->transientPosOffset = FRAME_MIDDLE_SLOT_1920;
+      hs->timeSlots          = NUMBER_TIME_SLOTS_1920;
+      break;
+  case 1024:
+      hs->transientPosOffset = FRAME_MIDDLE_SLOT_512LD;
+      hs->timeSlots          = 16;
+      break;
+  case 960:
+      hs->transientPosOffset = FRAME_MIDDLE_SLOT_512LD;
+      hs->timeSlots          = 15;
+      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/libSBRenc/src/mh_det.h b/libSBRenc/src/mh_det.h
new file mode 100644
index 0000000..ac62532
--- /dev/null
+++ b/libSBRenc/src/mh_det.h
@@ -0,0 +1,196 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/*!
+  \file
+  \brief  missing harmonics detection header file  
+*/
+
+#ifndef __MH_DETECT_H
+#define __MH_DETECT_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/libSBRenc/src/nf_est.cpp b/libSBRenc/src/nf_est.cpp
new file mode 100644
index 0000000..62bcc79
--- /dev/null
+++ b/libSBRenc/src/nf_est.cpp
@@ -0,0 +1,575 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+#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  (3)
+
+
+
+/**************************************************************************/
+/*!
+  \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 *smoothFilter, /*!< 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(smoothFilter[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 = fMult(*noiseLevel, noiseFloorOffset)<<(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. */
+                         int 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;
+
+  if (sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) {
+    nNoiseEnvelopes = 1;
+    startPos[0] = startIndex;
+    stopPos[0]  = startIndex + min(numberOfEstimatesPerFrame,2);
+  } else
+  if(nNoiseEnvelopes == 1){
+    startPos[0] = startIndex;
+    stopPos[0]  = startIndex + 2;
+  }
+  else{
+    startPos[0] = startIndex;
+    stopPos[0]  = startIndex + 1;
+    startPos[1] = startIndex + 1;
+    stopPos[1]  = startIndex + 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,      /*!< 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 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 {
+    FDK_ASSERT(noiseFloorOffset<=8); /* because of NOISE_FLOOR_OFFSET_SCALING */
+
+      /* Assumes the noise floor offset in tuning table are in q31    */
+      /* Currently the table contains only 0 for noise floor offset   */
+      /* 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,             /*!< Frequany 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;
+
+        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);
+
+        h_sbrNoiseFloorEstimate->noNoiseBands = (INT)((tmp + (FIXP_DBL)1) >> 1);
+
+        if (h_sbrNoiseFloorEstimate->noNoiseBands > MAX_NUM_NOISE_COEFFS)
+          h_sbrNoiseFloorEstimate->noNoiseBands = MAX_NUM_NOISE_COEFFS;
+
+        if( h_sbrNoiseFloorEstimate->noNoiseBands==0)
+            h_sbrNoiseFloorEstimate->noNoiseBands=1;
+    }
+
+
+    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/libSBRenc/src/nf_est.h b/libSBRenc/src/nf_est.h
new file mode 100644
index 0000000..084899a
--- /dev/null
+++ b/libSBRenc/src/nf_est.h
@@ -0,0 +1,147 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/*!
+  \file
+  \brief  Noise floor estimation structs and prototypes  
+*/
+
+#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. */
+                         int 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/libSBRenc/src/ps_bitenc.cpp b/libSBRenc/src/ps_bitenc.cpp
new file mode 100644
index 0000000..419d989
--- /dev/null
+++ b/libSBRenc/src/ps_bitenc.cpp
@@ -0,0 +1,696 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/*****************************  MPEG Audio Encoder  ***************************
+
+   Initial author:       N. Rettelbach
+   contents/description: Parametric Stereo bitstream encoder
+
+******************************************************************************/
+
+#include "ps_main.h"
+
+
+#include "ps_const.h"
+#include "ps_bitenc.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 const INT psBands[] =
+{
+  PS_BANDS_COARSE,
+  PS_BANDS_MID
+};
+
+static INT getNoBands(PS_RESOLUTION mode)
+{
+  if(mode>=6)
+    return 0;
+
+  if(mode>=3)
+    mode = (PS_RESOLUTION)(mode-3);
+
+  return psBands[mode];
+}
+
+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((PS_RESOLUTION)psOut->iidMode),
+                           psOut->deltaIPD[env],
+                           &error);
+
+      bitCnt += FDKsbrEnc_WriteBits_ps( hBitBuf, psOut->deltaOPD[env], 1);
+      bitCnt += FDKsbrEnc_EncodeOpd( hBitBuf,
+                           psOut->opd[env],
+                           opdLast,
+                           getNoBands((PS_RESOLUTION)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((PS_RESOLUTION)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((PS_RESOLUTION)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/libSBRenc/src/ps_bitenc.h b/libSBRenc/src/ps_bitenc.h
new file mode 100644
index 0000000..cfc5af7
--- /dev/null
+++ b/libSBRenc/src/ps_bitenc.h
@@ -0,0 +1,177 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/*****************************  MPEG Audio Encoder  ***************************
+
+   Initial author:       N. Rettelbach
+   contents/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  /* #ifndef PS_BITENC_H */
diff --git a/libSBRenc/src/ps_const.h b/libSBRenc/src/ps_const.h
new file mode 100644
index 0000000..08101e2
--- /dev/null
+++ b/libSBRenc/src/ps_const.h
@@ -0,0 +1,148 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/*****************************  MPEG Audio Encoder  ***************************
+
+   Initial author:       N. Rettelbach
+   contents/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        ( QMF_MAX_TIME_SLOTS )
+#define  HYBRID_READ_OFFSET      ( 10 )
+
+#define MAX_HYBRID_BANDS         ( (QMF_CHANNELS-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/libSBRenc/src/ps_encode.cpp b/libSBRenc/src/ps_encode.cpp
new file mode 100644
index 0000000..e60f83d
--- /dev/null
+++ b/libSBRenc/src/ps_encode.cpp
@@ -0,0 +1,1054 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/*****************************  MPEG Audio Encoder  ***************************
+
+   Initial Authors:      M. Neuendorf, N. Rettelbach, M. Multrus
+   Contents/Description: PS parameter extraction, encoding
+
+******************************************************************************/
+/*!
+  \file
+  \brief  PS parameter extraction, encoding functions  
+*/
+
+#include "ps_main.h"
+
+
+#include "sbr_ram.h"
+#include "ps_encode.h"
+
+#include "qmf.h"
+
+#include "ps_const.h"
+#include "sbr_misc.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] = {
+  0xce000000, 0xdc000000, 0xe4000000, 0xec000000, 0xf2000000, 0xf8000000, 0xfc000000, 0x00000000,
+  0x04000000, 0x08000000, 0x0e000000, 0x14000000, 0x1c000000, 0x24000000, 0x32000000
+};
+
+static const FIXP_DBL iidQuantFine_fx[31] = {
+  0x9c000001, 0xa6000001, 0xb0000001, 0xba000001, 0xc4000000, 0xce000000, 0xd4000000, 0xda000000,
+  0xe0000000, 0xe6000000, 0xec000000, 0xf0000000, 0xf4000000, 0xf8000000, 0xfc000000, 0x00000000,
+  0x04000000, 0x08000000, 0x0c000000, 0x10000000, 0x14000000, 0x1a000000, 0x20000000, 0x26000000,
+  0x2c000000, 0x32000000, 0x3c000000, 0x45ffffff, 0x4fffffff, 0x59ffffff, 0x63ffffff
+};
+
+
+
+static const FIXP_DBL iccQuant[8] = {
+  0x7fffffff, 0x77ef9d7f, 0x6babc97f, 0x4ceaf27f, 0x2f0ed3c0, 0x00000000, 0xb49ba601, 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;
+
+  bitsIidFreq = bitsIidTime = 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 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],
+                         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);
+      */
+      FIXP_DBL ICC, invNrg = CalcInvLdData ( -((ldPwrL[env][i]>>1) + (ldPwrR[env][i]>>1) + (FIXP_DBL)1) );
+      INT      scale, invScale = CountLeadingBits(invNrg);
+
+      scale = (DFRACT_BITS-1) - invScale;
+      ICC = fMult(pwrCr[env][i], invNrg<<invScale) ;
+      icc[env][i] = SATURATE_LEFT_SHIFT(ICC, scale, DFRACT_BITS);
+    }
+
+    for (; i<psBands; i++) {
+      INT sc1, sc2;
+      FIXP_DBL cNrgR, cNrgI, ICC;
+
+      sc1 = CountLeadingBits( fixMax(fixp_abs(pwrCr[env][i]),fixp_abs(pwrCi[env][i])) ) ;
+      cNrgR = fPow2Div2((pwrCr[env][i]<<sc1));       /* squared nrg's expect explicit scaling */
+      cNrgI = fPow2Div2((pwrCi[env][i]<<sc1));
+
+      ICC = CalcInvLdData( (CalcLdData((cNrgR + cNrgI)>>1)>>1) - (FIXP_DBL)((sc1-1)<<(DFRACT_BITS-1-LD_DATA_SHIFT)) );
+
+      FIXP_DBL invNrg = CalcInvLdData ( -((ldPwrL[env][i]>>1) + (ldPwrR[env][i]>>1) + (FIXP_DBL)1) );
+      sc1 = CountLeadingBits(invNrg);
+      invNrg <<= sc1;
+
+      sc2 = CountLeadingBits(ICC);
+      ICC = fMult(ICC<<sc2,invNrg);
+
+      sc1 = ( (DFRACT_BITS-1) - sc1 - sc2 );
+      if (sc1 < 0) {
+          ICC >>= -sc1;
+      }
+      else {
+          if (ICC >= ((FIXP_DBL)MAXVAL_DBL>>sc1) )
+              ICC = (FIXP_DBL)MAXVAL_DBL;
+          else
+              ICC <<= sc1;
+      }
+
+      icc[env][i] = ICC;
+    }
+  }
+}
+
+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_QMF l_real = (hybridData[col][0][0][subband]) << scale;
+          FIXP_QMF l_imag = (hybridData[col][0][1][subband]) << scale;
+          FIXP_QMF r_real = (hybridData[col][1][0][subband]) << scale;
+          FIXP_QMF 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->ldPwrL, pwrData->ldPwrR, 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->ldPwrL, pwrData->ldPwrR, 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];
+    }
+
+    for(env=0; env<hPsOut->nEnvelopes; env++) {
+      hPsOut->deltaIID[env] = (PS_DELTA)hPsData->iidDiffMode[env];
+
+      for(band=0; band<psBands; band++) {
+        hPsOut->iid[env][band] = hPsData->iidIdx[env][band];
+      }
+    }
+
+    for(env=0; env<hPsOut->nEnvelopes; env++) {
+      hPsOut->deltaICC[env] = (PS_DELTA)hPsData->iccDiffMode[env];
+      for(band=0; band<psBands; 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/libSBRenc/src/ps_encode.h b/libSBRenc/src/ps_encode.h
new file mode 100644
index 0000000..a12f7c6
--- /dev/null
+++ b/libSBRenc/src/ps_encode.h
@@ -0,0 +1,187 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/*****************************  MPEG Audio Encoder  ***************************
+
+   Initial author:       M. Neuendorf, N. Rettelbach, M. Multrus
+   contents/description: PS Parameter extraction, encoding
+
+******************************************************************************/
+/*!
+  \file
+  \brief  PS parameter extraction, encoding functions  
+*/
+
+#ifndef __INCLUDED_PS_ENCODE_H
+#define __INCLUDED_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/libSBRenc/src/ps_main.cpp b/libSBRenc/src/ps_main.cpp
new file mode 100644
index 0000000..bbab25a
--- /dev/null
+++ b/libSBRenc/src/ps_main.cpp
@@ -0,0 +1,618 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/*****************************  MPEG Audio Encoder  ***************************
+
+   Initial Authors:      M. Multrus
+   Contents/Description: PS Wrapper, Downmix
+
+******************************************************************************/
+
+#include "ps_main.h"
+
+
+/* Includes ******************************************************************/
+
+#include "ps_const.h"
+#include "ps_bitenc.h"
+
+#include "sbr_ram.h"
+
+/*--------------- function declarations --------------------*/
+static void psFindBestScaling(
+        HANDLE_PARAMETRIC_STEREO  hParametricStereo,
+        FIXP_DBL                 *hybridData[HYBRID_FRAMESIZE][MAX_PS_CHANNELS][2],
+        UCHAR                    *dynBandScale,
+        FIXP_QMF                 *maxBandValue,
+        SCHAR                    *dmxScale
+        );
+
+/*------------- function definitions ----------------*/
+FDK_PSENC_ERROR PSEnc_Create(
+        HANDLE_PARAMETRIC_STEREO *phParametricStereo
+        )
+{
+  FDK_PSENC_ERROR error = PSENC_OK;
+
+  if (phParametricStereo==NULL) {
+    error = PSENC_INVALID_HANDLE;
+  }
+  else {
+    int i;
+    HANDLE_PARAMETRIC_STEREO hParametricStereo = NULL;
+
+    if (NULL==(hParametricStereo = GetRam_ParamStereo())) {
+      error = PSENC_MEMORY_ERROR;
+      goto bail;
+    }
+    FDKmemclear(hParametricStereo, sizeof(PARAMETRIC_STEREO));
+
+    if (PSENC_OK != (error = FDKsbrEnc_CreatePSEncode(&hParametricStereo->hPsEncode))) {
+      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;
+      }
+    }
+
+    *phParametricStereo = hParametricStereo; /* return allocated handle */
+  }
+bail:
+  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,
+             QMF_CHANNELS,
+             QMF_CHANNELS,
+             1
+             );
+    } /* ch */
+
+    FDKhybridSynthesisInit(
+          &hParametricStereo->fdkHybSynFilter,
+           THREE_TO_TEN,
+           QMF_CHANNELS,
+           QMF_CHANNELS
+           );
+
+    /* 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));
+
+    /* clear scaling buffer */
+    FDKmemclear(hParametricStereo->dynBandScale, sizeof(UCHAR)*PS_MAX_BANDS);
+    FDKmemclear(hParametricStereo->maxBandValue, sizeof(FIXP_QMF)*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_QMF       **RESTRICT mixRealQmfData,
+       FIXP_QMF       **RESTRICT mixImagQmfData,
+       INT_PCM                  *downsampledOutSignal,
+       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_ALLOC_SCRATCH_START(pWorkBuffer, FIXP_QMF, QMF_CHANNELS*2);    
+
+    /* 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_QMF 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_QMF)(tmpLeftReal + tmpRightReal))>>dynScale;
+          tmpHybrid[1][k] = fMultDiv2(stereoScaleFactor, (FIXP_QMF)(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_ALLOC_SCRATCH_END(pWorkBuffer, FIXP_QMF, QMF_CHANNELS*2);
+
+
+  {
+    const INT noQmfSlots2 = hParametricStereo->noQmfSlots>>1;
+    const int noQmfBands  = hParametricStereo->noQmfBands;
+
+    INT scale, i, j, slotOffset;
+
+    FIXP_QMF tmp[2][QMF_CHANNELS];
+
+    for (i=0; i<noQmfSlots2; i++) {
+      FDKmemcpy(tmp[0], hParametricStereo->qmfDelayLines[0][i], noQmfBands*sizeof(FIXP_QMF));
+      FDKmemcpy(tmp[1], hParametricStereo->qmfDelayLines[1][i], noQmfBands*sizeof(FIXP_QMF));
+
+      FDKmemcpy(hParametricStereo->qmfDelayLines[0][i], mixRealQmfData[i+noQmfSlots2], noQmfBands*sizeof(FIXP_QMF));
+      FDKmemcpy(hParametricStereo->qmfDelayLines[1][i], mixImagQmfData[i+noQmfSlots2], noQmfBands*sizeof(FIXP_QMF));
+
+      FDKmemcpy(mixRealQmfData[i+noQmfSlots2], mixRealQmfData[i], noQmfBands*sizeof(FIXP_QMF));
+      FDKmemcpy(mixImagQmfData[i+noQmfSlots2], mixImagQmfData[i], noQmfBands*sizeof(FIXP_QMF));
+
+      FDKmemcpy(mixRealQmfData[i], tmp[0], noQmfBands*sizeof(FIXP_QMF));
+      FDKmemcpy(mixImagQmfData[i], tmp[1], noQmfBands*sizeof(FIXP_QMF));
+    }
+
+    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 = FDKmin(*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                      timeInStride,
+        QMF_FILTER_BANK         **hQmfAnalysis,
+        FIXP_QMF **RESTRICT       downmixedRealQmfData,
+        FIXP_QMF **RESTRICT       downmixedImagQmfData,
+        INT_PCM                  *downsampledOutSignal,
+        HANDLE_QMF_FILTER_BANK    sbrSynthQmf,
+        SCHAR                    *qmfScale,
+        const int                 sendHeader
+        )
+{
+  FDK_PSENC_ERROR error = PSENC_OK;
+  INT noQmfBands  = hParametricStereo->noQmfBands;
+  INT psQmfScale[MAX_PS_CHANNELS] = {0};
+  int psCh, i;
+  C_ALLOC_SCRATCH_START(pWorkBuffer, FIXP_DBL, QMF_CHANNELS*4);    
+
+  for (psCh = 0; psCh<MAX_PS_CHANNELS; psCh ++) {
+
+    for (i = 0; i < hQmfAnalysis[psCh]->no_col; i++) {
+
+      qmfAnalysisFilteringSlot(
+          hQmfAnalysis[psCh],
+         &pWorkBuffer[2*QMF_CHANNELS], /* qmfReal[QMF_CHANNELS] */
+         &pWorkBuffer[3*QMF_CHANNELS], /* qmfImag[QMF_CHANNELS] */
+          samples[psCh]+i*(hQmfAnalysis[psCh]->no_channels*timeInStride),
+          timeInStride,
+         &pWorkBuffer[0*QMF_CHANNELS]  /* qmf workbuffer 2*QMF_CHANNELS */
+          );
+
+      FDKhybridAnalysisApply(
+         &hParametricStereo->fdkHybAnaFilter[psCh],
+         &pWorkBuffer[2*QMF_CHANNELS],  /* qmfReal[QMF_CHANNELS] */
+         &pWorkBuffer[3*QMF_CHANNELS],  /* qmfImag[QMF_CHANNELS] */
+          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_ALLOC_SCRATCH_END(pWorkBuffer, FIXP_DBL, QMF_CHANNELS*4);
+
+  /* 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[HYBRID_FRAMESIZE+i][0][0], MAX_HYBRID_BANDS*sizeof(FIXP_DBL)); /* left, real */
+    FDKmemcpy(hParametricStereo->pHybridData[i][0][1], hParametricStereo->pHybridData[HYBRID_FRAMESIZE+i][0][1], MAX_HYBRID_BANDS*sizeof(FIXP_DBL)); /* left, imag */
+    FDKmemcpy(hParametricStereo->pHybridData[i][1][0], hParametricStereo->pHybridData[HYBRID_FRAMESIZE+i][1][0], MAX_HYBRID_BANDS*sizeof(FIXP_DBL)); /* right, real */
+    FDKmemcpy(hParametricStereo->pHybridData[i][1][1], hParametricStereo->pHybridData[HYBRID_FRAMESIZE+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, &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_QMF                 *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_QMF maxVal [2][PS_MAX_BANDS];
+  FIXP_QMF 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 */
+    {
+      FIXP_QMF tmp = maxVal[0][bin];
+      int i;
+      for (col=0; col<frameSize-HYBRID_READ_OFFSET; col++) {
+        for (i = hPsEncode->iidGroupBorders[group]; i < hPsEncode->iidGroupBorders[group+1]; i++) {
+          tmp = fixMax(tmp, (FIXP_QMF)fixp_abs(hybridData[col][0][0][i]));
+          tmp = fixMax(tmp, (FIXP_QMF)fixp_abs(hybridData[col][0][1][i]));
+          tmp = fixMax(tmp, (FIXP_QMF)fixp_abs(hybridData[col][1][0][i]));
+          tmp = fixMax(tmp, (FIXP_QMF)fixp_abs(hybridData[col][1][1][i]));
+        }
+      }
+      maxVal[0][bin] = tmp;
+
+      tmp = maxVal[1][bin];
+      for (col=frameSize-HYBRID_READ_OFFSET; col<frameSize; col++) {
+        for (i = hPsEncode->iidGroupBorders[group]; i < hPsEncode->iidGroupBorders[group+1]; i++) {
+          tmp = fixMax(tmp, (FIXP_QMF)fixp_abs(hybridData[col][0][0][i]));
+          tmp = fixMax(tmp, (FIXP_QMF)fixp_abs(hybridData[col][0][1][i]));
+          tmp = fixMax(tmp, (FIXP_QMF)fixp_abs(hybridData[col][1][0][i]));
+          tmp = fixMax(tmp, (FIXP_QMF)fixp_abs(hybridData[col][1][1][i]));
+        }
+      }
+      maxVal[1][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(FX_QMF2FX_DBL(maxValue))));
+#endif
+
+}
+
diff --git a/libSBRenc/src/ps_main.h b/libSBRenc/src/ps_main.h
new file mode 100644
index 0000000..6180299
--- /dev/null
+++ b/libSBRenc/src/ps_main.h
@@ -0,0 +1,271 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/*****************************  MPEG Audio Encoder  ***************************
+
+   Initial Authors:      Markus Multrus
+   Contents/Description: PS Wrapper, Downmix header file
+
+******************************************************************************/
+
+#ifndef __INCLUDED_PS_MAIN_H
+#define __INCLUDED_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_QMF                    qmfDelayLines[2][QMF_MAX_TIME_SLOTS>>1][QMF_CHANNELS];
+  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*(QMF_CHANNELS-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_QMF **RESTRICT       downmixedRealQmfData,
+        FIXP_QMF **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 /* __INCLUDED_PS_MAIN_H */
diff --git a/libSBRenc/src/resampler.cpp b/libSBRenc/src/resampler.cpp
new file mode 100644
index 0000000..e8ab263
--- /dev/null
+++ b/libSBRenc/src/resampler.cpp
@@ -0,0 +1,507 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/*!
+  \file
+  \brief  FDK resampler tool box: 
+  \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)
+};
+
+#ifdef RS_BIQUAD_SCATTERGAIN
+static const FIXP_DBL g48 = FL2FXCONST_DBL(0.67436532061161992682404480717671 - 0.001);
+#else
+static const FIXP_DBL g48 = FL2FXCONST_DBL(0.002712866530047) - (FIXP_DBL)0x8000;
+#endif
+
+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)
+};
+
+#ifdef RS_BIQUAD_SCATTERGAIN
+static const FIXP_DBL g45 = FL2FXCONST_DBL(0.60547428891341319051142629706723 - 0.001);
+#else
+static const FIXP_DBL g45 = FL2FXCONST_DBL(0.00242743980909524) - (FIXP_DBL)0x8000;
+#endif
+
+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)
+};
+
+#ifdef RS_BIQUAD_SCATTERGAIN
+static const FIXP_DBL g41 = FL2FXCONST_DBL(0.44578514476476679750811222123569);
+#else
+static const FIXP_DBL g41 = FL2FXCONST_DBL(0.00155956951169248);
+#endif
+
+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)
+};
+
+#ifdef RS_BIQUAD_SCATTERGAIN
+static const FIXP_DBL g35 = FL2FXCONST_DBL(0.34290853574973898694521267606792);
+#else
+static const FIXP_DBL g35 = FL2FXCONST_DBL(0.00162580994125131);
+#endif
+
+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)
+};
+
+#ifdef RS_BIQUAD_SCATTERGAIN
+static const FIXP_DBL g25 = FL2FXCONST_DBL(0.17533917408936346960080259950471);
+#else
+static const FIXP_DBL g25 = FL2FXCONST_DBL(0.000945182835294559);
+#endif
+
+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 (only 2 supported at the momment) */
+
+{
+  UINT i;
+  const struct FILTER_PARAM *currentSet=NULL;
+
+  FDK_ASSERT(ratio == 2);
+  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 inStride)
+{
+  INT_PCM output;
+  int i, n;
+
+
+#ifdef RS_BIQUAD_SCATTERGAIN
+#define BIQUAD_SCALE 3
+#else
+#define BIQUAD_SCALE 12
+#endif
+
+  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*inStride]) << (DFRACT_BITS-SAMPLE_BITS-BIQUAD_SCALE);
+#elif (SAMPLE_BITS == 32)
+    input =  pInput[n*inStride] >> BIQUAD_SCALE;
+#else
+#error NOT IMPLEMENTED
+#endif
+
+#ifndef RS_BIQUAD_SCATTERGAIN /* Merged Direct form I */
+
+    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);
+
+#else /* Direct form II */
+
+    /* Loop over sections */
+    for (i=0; i<downFilter->noCoeffs; i++)
+    {
+      FIXP_BQS state1, state2;
+      FIXP_DBL state0;
+
+      /* Load states */
+      state1 = states[i][s1];
+      state2 = states[i][s2];
+
+      state0 = input  - fMult(state1, coeff[A1]) - fMult(state2, coeff[A2]);
+      y = state0      + fMult(state1, coeff[B1]) + fMult(state2, coeff[B2]);
+      /* Apply scattered gain */
+      y = fMult(y, downFilter->gain);
+
+      /* Store new state in normalized form */
+#ifdef RS_BIQUAD_STATES16
+      /* Do not saturate any state value ! The result would be unacceptable. Rounding makes SNR around 10dB better. */
+      states[i][s2] = (FIXP_BQS)(LONG)((state0 + (FIXP_DBL)(1<<(DFRACT_BITS-FRACT_BITS-2))) >> (DFRACT_BITS-FRACT_BITS-1));
+#else
+      states[i][s2] = state0<<1;
+#endif
+
+      /* Feedback output to next section. */
+      input=y;
+
+      /* Step to next coef set */
+      coeff += BIQUAD_COEFSTEP;
+    }
+    downFilter->ptr ^= 1;
+  }
+
+#endif
+
+  /* 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 inStride,              /*!< increment of input samples */
+                         INT_PCM *outSamples,       /*!< pointer to output samples */
+                         INT *numOutSamples,        /*!< pointer tp number of output samples */
+                         INT outStride              /*!< increment of output samples */
+                         )
+{
+    INT i;
+    *numOutSamples=0;
+
+    for(i=0; i<numInSamples; i+=DownSampler->ratio)
+    {
+      *outSamples = AdvanceFilter(&(DownSampler->downFilter), &inSamples[i*inStride], DownSampler->ratio, inStride);
+      outSamples += outStride;
+    }
+    *numOutSamples = numInSamples/DownSampler->ratio;
+
+    return 0;
+}
+
diff --git a/libSBRenc/src/resampler.h b/libSBRenc/src/resampler.h
new file mode 100644
index 0000000..29e170c
--- /dev/null
+++ b/libSBRenc/src/resampler.h
@@ -0,0 +1,151 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+#ifndef __RESAMPLER_H
+#define __RESAMPLER_H
+/*!
+  \file
+  \brief  Fixed Point Resampler Tool Box  
+*/
+
+#include "common_fix.h"
+
+
+/**************************************************************************/
+/*                         BIQUAD Filter Structure                           */
+/**************************************************************************/
+
+#define MAXNR_SECTIONS  (15)
+
+#ifdef RS_BIQUAD_STATES16
+typedef FIXP_SGL FIXP_BQS;
+#else
+typedef FIXP_DBL FIXP_BQS;
+#endif
+
+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 inStride,              /*!< increment of input samples      */
+                         INT_PCM *outSamples,       /*!< pointer to output samples */
+                         INT *numOutSamples,        /*!< pointer tp number of output samples */
+                         INT outstride);            /*!< increment of output samples */
+
+
+
+#endif /* __RESAMPLER_H */
diff --git a/libSBRenc/src/sbr.h b/libSBRenc/src/sbr.h
new file mode 100644
index 0000000..13caadd
--- /dev/null
+++ b/libSBRenc/src/sbr.h
@@ -0,0 +1,164 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/*!
+  \file
+  \brief  Main SBR structs definitions  
+*/
+
+#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 DELAY_FRAMES 2
+
+typedef struct {
+    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 {
+    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+DELAY_FRAMES][MAX_PAYLOAD_SIZE];
+    UINT                      payloadDelayLineSize[1+DELAY_FRAMES];                 /* Sizes in bits */
+
+} SBR_ELEMENT;
+typedef SBR_ELEMENT* HANDLE_SBR_ELEMENT;
+
+struct SBR_ENCODER
+{
+  HANDLE_SBR_ELEMENT   sbrElement[(6)];
+  HANDLE_SBR_CHANNEL   pSbrChannel[(6)];
+  QMF_FILTER_BANK      QmfAnalysis[(6)];
+  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                  fTimeDomainDownsampling;  /* Flag signalling time domain downsampling instead of QMF downsampling. */
+  int                  nBitstrDelay;             /* Amount of SBR frames to be delayed in bitstream domain. */
+  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;
+
+} ;
+
+
+#endif /* __SBR_H */
diff --git a/libSBRenc/src/sbr_def.h b/libSBRenc/src/sbr_def.h
new file mode 100644
index 0000000..1d99f7f
--- /dev/null
+++ b/libSBRenc/src/sbr_def.h
@@ -0,0 +1,286 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/*!
+  \file
+  \brief  SBR main definitions  
+*/
+#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 handBack
+
+/* #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 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 QMF_CHANNELS                            64
+#define QMF_FILTER_LENGTH                      640
+#define QMF_MAX_TIME_SLOTS                      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
+
+
+/* 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;
+
+typedef enum
+{
+  SINGLE_RATE,
+  DUAL_RATE
+}
+SR_MODE;
+
+typedef enum
+{
+  FREQ_RES_LOW = 0,
+  FREQ_RES_HIGH
+}
+FREQ_RES;
+
+
+#endif
diff --git a/libSBRenc/src/sbr_encoder.cpp b/libSBRenc/src/sbr_encoder.cpp
new file mode 100644
index 0000000..0dcc1b2
--- /dev/null
+++ b/libSBRenc/src/sbr_encoder.cpp
@@ -0,0 +1,2237 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/***************************  Fraunhofer IIS FDK Tools  ***********************
+
+   Author(s): Andreas Ehret
+   Description: SBR encoder top level processing.
+
+******************************************************************************/
+
+#include "sbr_encoder.h"
+
+#include "sbr_ram.h"
+#include "sbr_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 3
+#define SBRENCODER_LIB_VL1 2
+#define SBRENCODER_LIB_VL2 1
+
+
+
+/***************************************************************************/
+/*
+ * SBR Delay balancing definitions.
+ */
+
+/*
+      input buffer (1ch)
+
+      |------------ 1537   -------------|-----|---------- 2048 -------------|
+           (core2sbr delay     )          ds     (read, core and ds area)
+*/
+
+#define DOWN_SMPL_FAC        (2)
+
+#define SFL(fl)            (fl*DOWN_SMPL_FAC)   /* SBR frame length (hardcoded to downsample factor of 2) */
+#define STS(fl)            (SFL(fl)/64)  /* SBR Time Slots */
+
+#define DELAY_QMF_ANA      (640 - 64)    /* Full bandwidth */
+#define DELAY_QMF_ANAELD   (32)
+#define DELAY_HYB_ANA      (10*64)       /* + 0.5 */
+#define DELAY_HYB_SYN      (6*64 - 32)
+#define DELAY_QMF_SYNELD   (32)
+#define DELAY_DEC_QMF      (6*64)        /* Decoder QMF overlap */
+#define DELAY_QMF_SYN      (2)           /* NO_POLY/2 */
+#define DELAY_QMF_DS       (32)          /* QMF synthesis for downsampled time signal */
+
+/* Delay in QMF paths */
+#define DELAY_SBR(fl)      (DELAY_QMF_ANA + (64*STS(fl)-1) + DELAY_QMF_SYN)
+#define DELAY_PS(fl)       (DELAY_QMF_ANA + DELAY_HYB_ANA + DELAY_DEC_QMF + (64*STS(fl)-1) + DELAY_HYB_SYN + DELAY_QMF_SYN)
+#define DELAY_ELDSBR(fl)   (DELAY_QMF_ANAELD + (((fl)+((fl)/2))*2 - 1) + DELAY_QMF_SYNELD)
+
+/* Delay differences for SBR and SBR+PS */
+    #define MAX_DS_FILTER_DELAY (34)                                      /* the additional max downsampler filter delay (source fs) */
+#define DELAY_AAC2SBR(fl)      ((/*RESAMPLER +*/ /*(DELAY_AAC(fl)*2) + */ DELAY_QMF_ANA + DELAY_DEC_QMF + DELAY_QMF_SYN) - DELAY_SBR(fl)) /* 1537 */
+#define DELAY_ELD2SBR(fl)      ((/*RESAMPLER +*/ /*(DELAY_ELD(fl)*2) + */ DELAY_QMF_ANAELD + DELAY_QMF_SYNELD) - DELAY_ELDSBR(fl))
+
+#define DELAY_AAC2PS(fl)       ((DELAY_QMF_ANA + DELAY_QMF_DS + /*(DELAY_AAC(fl)*2)*/ + DELAY_QMF_ANA + DELAY_DEC_QMF + DELAY_HYB_SYN + DELAY_QMF_SYN) - DELAY_PS(fl)) /* 2048 - 463*2 */
+
+/* Assumption: that the sample delay resulting of of DELAY_AAC2PS is always smaller than the sample delay implied by DELAY_AAC2SBR */
+#define MAX_SAMPLE_DELAY       (DELAY_AAC2SBR(1024) + MAX_DS_FILTER_DELAY)
+
+/***************************************************************************/
+
+
+
+#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, paramSetTop, bitRateClosestLowerIndex=-1, bitRateClosestUpperIndex=-1, found = 0;
+  UINT bitRateClosestUpper = 0, bitRateClosestLower=DISTANCE_CEIL_VALUE;
+
+  FDK_ASSERT(SBRENC_TUNING_SIZE == sizeof(sbrTuningTable)/sizeof(sbrTuningTable[0]));
+
+  if (core == AOT_ER_AAC_ELD) {
+    paramSetTop = SBRENC_TUNING_SIZE;
+    i = SBRENC_AACLC_TUNING_SIZE;
+  } else {
+    paramSetTop = SBRENC_AACLC_TUNING_SIZE;
+    i = 0;
+  }
+
+  for (; i < paramSetTop ; i++) {
+    if ( numChannels == sbrTuningTable [i].numChannels
+      && sampleRate == sbrTuningTable [i].sampleRate )
+    {
+      found = 1;
+      if ((bitrate >= sbrTuningTable [i].bitrateFrom) &&
+          (bitrate < sbrTuningTable [i].bitrateTo)) {
+            bitRateClosestLower = bitrate;
+            bitRateClosestUpper = bitrate;
+            //FDKprintf("entry %d\n", i);
+        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 (pBitRateClosest != NULL)
+  {
+    /* Is there was at least one matching tuning entry found then 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 (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  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] */
+                   AUDIO_OBJECT_TYPE core
+                   )
+{
+  INT idx = INVALID_TABLE_IDX;
+  UINT sampleRateCore;
+
+  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;
+  }
+
+  /* try DOWN_SMPL_FAC of the input sampling rate */
+  sampleRateCore = sampleRateInput/DOWN_SMPL_FAC;
+  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 fsCore,              /*! the core 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;
+  UINT sampleRate;
+
+  /* set the codec settings */
+  config->codecSettings.bitRate         = bitRate;
+  config->codecSettings.nChannels       = numChannels;
+  config->codecSettings.sampleFreq      = fsCore;
+  config->codecSettings.transFac        = transFac;
+  config->codecSettings.standardBitrate = standardBitrate;
+  sampleRate  = fsCore * DOWN_SMPL_FAC;
+
+  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 (sampleRate==44100 || sampleRate==48000) {
+        if (vbrMode<40) bitRate = 32000;
+      }
+    }
+  }
+
+  idx = getSbrTuningTableIndex(bitRate,numChannels,fsCore, 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;
+    }
+
+    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 ;
+
+    /* 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 confifuration
+ returns:      error status
+ input:        - core codec type,
+               - fac of SBR to core frame length,
+               - core frame length
+ output:       initialized SBR configuration
+
+*****************************************************************************/
+static UINT
+FDKsbrEnc_InitializeSbrDefaults (sbrConfigurationPtr config,
+                                 INT                 coreSbrFrameLenFac,
+                                 UINT                codecGranuleLen)
+{
+    if ( (coreSbrFrameLenFac != 2) ||
+         (codecGranuleLen*coreSbrFrameLenFac > QMF_CHANNELS*QMF_MAX_TIME_SLOTS) )
+       return(1);
+
+    config->SendHeaderDataTime     = 1000;
+    config->useWaveCoding          = 0;
+    config->crcSbr                 = 0;
+    config->dynBwSupported         = 1;
+    config->tran_thr               = 13000;
+    config->parametricCoding       = 1;
+
+    config->sbrFrameSize  = codecGranuleLen * coreSbrFrameLenFac;
+
+
+    /* 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;
+    config->bDownSampledSbr = 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 */
+
+
+    /* 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<(6); el++)
+    {
+      if (hSbrEncoder->sbrElement[el]!=NULL) {
+        sbrEncoder_ElementClose(&hSbrEncoder->sbrElement[el]);
+      }
+    }
+
+    /* Close sbr Channels */
+    for (ch=0; ch<(6); 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 */
+    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,
+                               INT noQmfChannels)
+{
+  INT k0, k2;
+
+  if(FDKsbrEnc_FindStartAndStopBand(sbrConfigData->sampleFreq,
+                                    noQmfChannels,
+                                    sbrHeaderData->sbr_start_frequency,
+                                    sbrHeaderData->sbr_stop_frequency,
+                                    sbrHeaderData->sampleRateMode,
+                                    &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,
+                           sbrHeaderData->sampleRateMode,
+                           noQmfChannels))
+    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 / noQmfChannels+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 interleaved */
+                         UINT                 timeInStride,            /*!< time buffer channel interleaving stride */
+                         UINT                *sbrDataBits,             /*!< Size of SBR payload  */
+                         UCHAR               *sbrData,                 /*!< SBR payload  */
+                         int                  clearOutput              /*!< Do not consider any input signal */
+                        )
+{
+  HANDLE_SBR_ELEMENT hSbrElement = hEnvEncoder->sbrElement[iElement];
+  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,
+                              hSbrElement->sbrConfigData.noQmfBands))
+        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);
+
+  INT error = noError;
+
+  /* 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)
+      {
+        C_ALLOC_SCRATCH_START(qmfWorkBuffer, FIXP_DBL, QMF_CHANNELS*2);
+        QMF_SCALE_FACTOR tmpScale;
+        FIXP_DBL **pQmfReal, **pQmfImag;
+
+
+        /* Obtain pointers to QMF buffers. */
+        pQmfReal = sbrExtrEnv->rBuffer;
+        pQmfImag = sbrExtrEnv->iBuffer;
+
+        qmfAnalysisFiltering( hSbrElement->hQmfAnalysis[ch],
+                               pQmfReal,
+                               pQmfImag,
+                              &tmpScale,
+                               samples + hSbrElement->elInfo.ChannelIndex[ch],
+                               timeInStride,
+                               qmfWorkBuffer );
+
+        C_ALLOC_SCRATCH_END(qmfWorkBuffer, FIXP_DBL, QMF_CHANNELS*2);
+
+        h_envChan->qmfScale = tmpScale.lb_scale + 7;
+
+      } /* fParametricStereo == 0 */
+
+
+      /*
+        Parametric Stereo processing
+      */
+      if (hSbrElement->elInfo.fParametricStereo)
+      {
+
+        /* 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],samples + hSbrElement->elInfo.ChannelIndex[1]};
+          error = FDKsbrEnc_PSEnc_ParametricStereoProcessing( hEnvEncoder->hParametricStereo,
+                                                              pSamples,
+                                                              timeInStride,
+                                                              hSbrElement->hQmfAnalysis,
+                                                              sbrExtrEnv->rBuffer,
+                                                              sbrExtrEnv->iBuffer,
+                                                              samples + hSbrElement->elInfo.ChannelIndex[ch],
+                                                             &hEnvEncoder->qmfSynthesisPS,
+                                                             &qmfScale,
+                                                              psHeaderActive );
+          if (noError != error)
+          {
+            error = handBack(error);
+          }
+          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->sbrChannel[1]->hEnvChannel,
+                                &hSbrElement->CmonData,
+                                 eData,
+                                 fData,
+                                 clearOutput
+                               );
+
+  /*
+    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);
+
+
+  }
+
+
+/*******************************/
+
+  if (hEnvEncoder->fTimeDomainDownsampling)
+  {
+    int ch;
+    int nChannels = hSbrElement->sbrConfigData.nChannels;
+
+    for (ch=0; ch < nChannels; ch++)
+    {
+      INT nOutSamples;
+
+      FDKaacEnc_Downsample(&hSbrElement->sbrChannel[ch]->downSampler,
+                            samples + hSbrElement->elInfo.ChannelIndex[ch] + hEnvEncoder->bufferOffset,
+                            hSbrElement->sbrConfigData.frameSize,
+                            timeInStride,
+                            samples + hSbrElement->elInfo.ChannelIndex[ch],
+                           &nOutSamples,
+                            hEnvEncoder->nChannels);
+    }
+  } /* downsample */
+
+
+  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 = 1;
+  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:
+    timeSlots = 16;
+    break;
+  case 1920:
+  case 960:
+    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,64,sbrConfigData->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 (params->sbrFrameSize) {
+      /* The factor of 2 is by definition. */
+      case 2048: tran_off = 8 + FRAME_MIDDLE_SLOT_2048 * timeStep; break;
+      case 1920: tran_off = 7 + FRAME_MIDDLE_SLOT_1920 * timeStep; break;
+      default: return 1; break;
+    }
+  }
+  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.ldGrid
+                            );
+
+  if(FDKsbrEnc_InitSbrTransientDetector (&hEnv->sbrTransientDetector,
+                                          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));
+
+      hSbrEncoder->pSBRdynamic_RAM = (UCHAR*)GetRam_SbrDynamic_RAM();
+      hSbrEncoder->dynamicRam = hSbrEncoder->pSBRdynamic_RAM;
+
+
+  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;
+    }
+  }
+
+  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;
+    }
+
+  }
+
+  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;
+    }
+  }
+
+  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[(6)],
+                    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;
+  } /* coreEl */
+
+  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       nBitstrDelay,
+                       int       nElement,
+                       ULONG     statesInitFlag
+                      ,UCHAR    *dynamic_RAM
+                      )
+{
+  UCHAR    *bitstreamBuffer;
+  int ch, i;
+
+  if ((params->codecSettings.nChannels < 1) || (params->codecSettings.nChannels > MAX_NUM_CHANNELS)){
+    return(1);
+  }
+
+  /* initialize the encoder handle  and structs*/
+  bitstreamBuffer = hSbrElement->payloadDelayLine[nBitstrDelay];
+
+  /* init and set syntax flags */
+  hSbrElement->sbrConfigData.sbrSyntaxFlags = 0;
+
+  switch (aot) {
+  case AOT_DRM_MPEG_PS:
+  case AOT_DRM_SBR:
+    hSbrElement->sbrConfigData.sbrSyntaxFlags |= SBR_SYNTAX_SCALABLE;
+    hSbrElement->sbrConfigData.sbrSyntaxFlags |= SBR_SYNTAX_DRM_CRC;
+    hSbrElement->sbrConfigData.sbrSyntaxFlags |= SBR_SYNTAX_CRC;
+    break;
+  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 = QMF_CHANNELS;
+  hSbrElement->sbrConfigData.noQmfSlots = params->sbrFrameSize/hSbrElement->sbrConfigData.noQmfBands;
+
+  FDKinitBitStream(&hSbrElement->CmonData.sbrBitbuf, bitstreamBuffer, MAX_PAYLOAD_SIZE*sizeof(UCHAR), 0, BS_WRITER);
+
+  /*
+    now initialize sbrConfigData, sbrHeaderData and sbrBitstreamData,
+  */
+  hSbrElement->sbrConfigData.nChannels = params->codecSettings.nChannels;
+
+  if(params->codecSettings.nChannels == 2)
+     hSbrElement->sbrConfigData.stereoMode = params->stereoMode;
+  else
+     hSbrElement->sbrConfigData.stereoMode = SBR_MONO;
+
+  hSbrElement->sbrConfigData.frameSize   = params->sbrFrameSize;
+
+  /* implicit rule for sampleRateMode */
+  /* run in "multirate" mode where sbr fs is 2 * codec fs */
+  hSbrElement->sbrHeaderData.sampleRateMode = DUAL_RATE;
+  hSbrElement->sbrConfigData.sampleFreq = 2 * params->codecSettings.sampleFreq;
+
+  hSbrElement->sbrBitstreamData.CountSendHeaderData = 0;
+  if (params->SendHeaderDataTime > 0 ) {
+
+    hSbrElement->sbrBitstreamData.NrSendHeaderData = (INT)(params->SendHeaderDataTime * hSbrElement->sbrConfigData.sampleFreq
+                                               / (1000 * hSbrElement->sbrConfigData.frameSize));
+    hSbrElement->sbrBitstreamData.NrSendHeaderData = fixMax(hSbrElement->sbrBitstreamData.NrSendHeaderData,1);
+  }
+  else {
+   hSbrElement->sbrBitstreamData.NrSendHeaderData = 0;
+  }
+
+  hSbrElement->sbrHeaderData.sbr_data_extra = params->sbr_data_extra;
+  hSbrElement->sbrBitstreamData.HeaderActive = 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;
+
+  /* init freq band table */
+  if(updateFreqBandTable(&hSbrElement->sbrConfigData,
+                         &hSbrElement->sbrHeaderData,
+                         hSbrElement->sbrConfigData.noQmfBands))
+  {
+    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 );
+  }
+
+  /*  */
+  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 = (1024*DOWN_SMPL_FAC);
+  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
+                                 )
+{
+    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->sbrElement[el]->sbrConfigData.nChannels,
+                                     NULL,
+                                     NULL,
+                                     1
+                                    ))
+          return -1;
+          sbrEncoder_UpdateBuffers(hEnvEnc, timeBuffer);
+      }
+    }
+	  return 0;
+}
+
+UINT sbrEncoder_LimitBitRate(UINT bitRate, UINT numChannels, UINT coreSampleRate, AUDIO_OBJECT_TYPE aot)
+{
+  UINT newBitRate;
+  INT index;
+
+  FDK_ASSERT(numChannels > 0 && numChannels <= 2);
+  if (aot == AOT_PS) {
+    if (numChannels == 2) {
+      index = getPsTuningTableIndex(bitRate, &newBitRate);
+      if (index == INVALID_TABLE_IDX) {
+        bitRate = newBitRate;
+      }
+      /* Set numChannels to 1 because for PS we need a SBR SCE (mono) element. */
+      numChannels = 1;
+    } else {
+      return 0;
+    }
+  }
+  index = getSbrTuningTableIndex(bitRate, numChannels, coreSampleRate, aot, &newBitRate);
+  if (index != INVALID_TABLE_IDX) {
+    newBitRate = bitRate;
+  }
+
+  return newBitRate;
+}
+
+
+INT sbrEncoder_Init(
+                     HANDLE_SBR_ENCODER hSbrEncoder,
+                     SBR_ELEMENT_INFO elInfo[(6)],
+                     int              noElements,
+                     INT_PCM *inputBuffer,
+                     INT   *coreBandwidth,
+                     INT   *inputBufferOffset,
+                     INT   *numChannels,
+                     INT   *sampleRate,
+                     INT   *frameLength,
+                     AUDIO_OBJECT_TYPE *aot,
+                     int *delay,
+                     int  transformFactor,
+                     ULONG  statesInitFlag
+                    )
+{
+    HANDLE_ERROR_INFO errorInfo = noError;
+    sbrConfiguration sbrConfig[(6)];
+    INT error = 0;
+    INT lowestBandwidth;
+    /* Save input parameters */
+    INT inputSampleRate = *sampleRate;
+    int coreFrameLength = *frameLength;
+    int inputBandWidth = *coreBandwidth;
+    int inputChannels = *numChannels;
+
+    int downsampledOffset = 0;
+    int sbrOffset = 0;
+    int downsamplerDelay = 0;
+    int downsample = 0;
+    int nBitstrDelay = 0;
+    int lowestSbrStartFreq, lowestSbrStopFreq;
+    int lowDelay = 0;
+    int usePs = 0;
+
+    /* check whether SBR setting is available for the current encoder configuration (bitrate, samplerate) */
+    if ( (*aot==AOT_PS) || (*aot==AOT_MP2_PS) || (*aot==AOT_DABPLUS_PS) || (*aot==AOT_DRM_MPEG_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 {
+        switch (*aot) {
+          case AOT_MP2_PS:
+            *aot = AOT_MP2_SBR;
+            break;
+          case AOT_DABPLUS_PS:
+            *aot = AOT_DABPLUS_SBR;
+            break;
+          case AOT_DRM_MPEG_PS:
+            *aot = AOT_DRM_SBR;
+            break;
+          case AOT_PS:
+          default:
+            *aot = AOT_SBR;
+        }
+        usePs = 0;
+      }
+    } /* usePs */
+
+    /* check whether SBR setting is available for the current encoder configuration (bitrate, samplerate) */
+    {
+      int delayDiff = 0;
+      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, *aot) )
+        {
+            /* otherwise - change to AAC-LC */
+            switch (*aot) {
+              case AOT_MP2_SBR:
+              case AOT_MP2_PS:
+                *aot = AOT_MP2_AAC_LC;
+                break;
+              case AOT_DABPLUS_SBR:
+              case AOT_DABPLUS_PS:
+                *aot = AOT_DABPLUS_AAC_LC;
+                break;
+              case AOT_DRM_SBR:
+              case AOT_DRM_MPEG_PS:
+                *aot = AOT_DRM_AAC;
+                break;
+              case AOT_ER_AAC_ELD:
+                break;
+              case AOT_SBR:
+              case AOT_PS:
+              default:
+                *aot = AOT_AAC_LC;
+            }
+            error = 1;
+            goto bail;
+        }
+      }
+
+      *sampleRate /= DOWN_SMPL_FAC;
+
+      /* Determine Delay balancing and new encoder delay */
+      if (lowDelay) {
+        downsample = 1;  /* activate downsampler */
+        delayDiff = (*delay*DOWN_SMPL_FAC) + DELAY_ELD2SBR(coreFrameLength);
+        *delay = DELAY_ELDSBR(coreFrameLength);
+      }
+      else if (usePs) {
+        delayDiff = (*delay*DOWN_SMPL_FAC) + DELAY_AAC2PS(coreFrameLength);
+        *delay = DELAY_PS(coreFrameLength);
+      }
+      else {
+        downsample = 1;  /* activate downsampler */
+        delayDiff = (*delay*DOWN_SMPL_FAC) + DELAY_AAC2SBR(coreFrameLength);
+        *delay = DELAY_SBR(coreFrameLength);
+      }
+
+      /* Take care about downsampled data bound to the SBR path */
+      if (!downsample && delayDiff > 0) {
+        /*
+         * We must tweak the balancing into a situation where the downsampled path
+         * is the one to be delayed, because delaying the QMF domain input, also delays
+         * the downsampled audio, counteracting to the purpose of delay balancing.
+         */
+        while ( delayDiff > 0 )
+        {
+          /* Encoder delay increases */
+          *delay += coreFrameLength*DOWN_SMPL_FAC;
+          /* Add one frame delay to SBR path */
+          delayDiff -= coreFrameLength*DOWN_SMPL_FAC;
+          nBitstrDelay += 1;
+        }
+      } else {
+        *delay += fixp_abs(delayDiff);
+      }
+
+      if (delayDiff < 0) {
+        /* Delay AAC data */
+        delayDiff = -delayDiff;
+        /* Multiply downsampled offset by AAC core channels. Divide by 2 because of half samplerate of downsampled data. */
+        downsampledOffset = (delayDiff*(*numChannels))/DOWN_SMPL_FAC;
+        sbrOffset = 0;
+      } else {
+        /* Delay SBR input */
+        if ( delayDiff > (int)coreFrameLength*DOWN_SMPL_FAC )
+        {
+          /* Do bitstream frame-wise delay balancing if we have more than SBR framelength samples delay difference */
+          delayDiff -= coreFrameLength*DOWN_SMPL_FAC;
+          nBitstrDelay = 1;
+        }
+        /* Multiply input offset by input channels */
+        sbrOffset = delayDiff*(*numChannels);
+        downsampledOffset = 0;
+      }
+
+      hSbrEncoder->nBitstrDelay = nBitstrDelay;
+      hSbrEncoder->nChannels = *numChannels;
+      hSbrEncoder->frameSize = *frameLength*DOWN_SMPL_FAC;
+      hSbrEncoder->fTimeDomainDownsampling = downsample;
+      hSbrEncoder->estimateBitrate = 0;
+      hSbrEncoder->inputDataDelay = 0;
+
+
+      /* Open SBR elements */
+      el = -1;
+      lowestSbrStartFreq = lowestSbrStopFreq = 9999;
+      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
+         */
+        FDKsbrEnc_InitializeSbrDefaults ( &sbrConfig[el],
+                                           DOWN_SMPL_FAC,
+                                           coreFrameLength);
+        /*
+         * Modify sbrConfig structure according to Element parameters
+         */
+        FDKsbrEnc_AdjustSbrSettings ( &sbrConfig[el],
+                                       elInfo[coreEl].bitRate,
+                                       elInfo[coreEl].nChannelsInEl,
+                                      *sampleRate,
+                                       transformFactor,
+                                       24000,
+                                       0,
+                                       0,                                            /* useSpeechConfig */
+                                       0,                                            /* lcsMode */
+                                       usePs, /* bParametricStereo */
+                                       *aot);
+
+        /* Find common frequency border for all SBR elements */
+        lowestSbrStartFreq = fixMin(lowestSbrStartFreq, sbrConfig[el].startFreq);
+        lowestSbrStopFreq = fixMin(lowestSbrStopFreq, 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 = lowestSbrStartFreq;
+        sbrConfig[el].stopFreq = lowestSbrStopFreq;
+
+        /* initialize SBR element, and get core bandwidth */
+        error = FDKsbrEnc_EnvInit(hSbrEncoder->sbrElement[el],
+                                  &sbrConfig[el],
+                                  &bandwidth,
+                                  *aot,
+                                   nBitstrDelay,
+                                   el,
+                                   statesInitFlag
+                                  ,hSbrEncoder->dynamicRam
+                                  );
+
+        if (error != 0) {
+          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->fTimeDomainDownsampling)
+      {
+        for (el=0; el<hSbrEncoder->noElements; el++)
+        {
+          HANDLE_SBR_ELEMENT hSbrEl = hSbrEncoder->sbrElement[el];
+          INT Wc, ch;
+
+          /* Calculated required normalized cutoff frequency (Wc = 1.0 -> lowestBandwidth = inputSampleRate/2) */
+          Wc = (2*lowestBandwidth)*1000 / inputSampleRate;
+
+          for (ch=0; ch<hSbrEl->elInfo.nChannelsInEl; ch++)
+          {
+            FDKaacEnc_InitDownsampler (&hSbrEl->sbrChannel[ch]->downSampler, Wc, DOWN_SMPL_FAC);
+          }
+
+          FDK_ASSERT (hSbrEl->sbrChannel[0]->downSampler.delay <=MAX_DS_FILTER_DELAY && hSbrEl->sbrChannel[0]->downSampler.delay <=MAX_DS_FILTER_DELAY);
+          downsamplerDelay = hSbrEl->sbrChannel[0]->downSampler.delay;
+        } /* third element loop */
+
+        /* lfe */
+        FDKaacEnc_InitDownsampler (&hSbrEncoder->lfeDownSampler, 0, DOWN_SMPL_FAC);
+
+        /* Add the resampler additional delay to get the final delay and buffer offset values. */
+        if (sbrOffset > 0 || downsampledOffset <= ((downsamplerDelay * (*numChannels))/DOWN_SMPL_FAC)) {
+          sbrOffset += (downsamplerDelay - downsampledOffset) * (*numChannels) ;
+          *delay += downsamplerDelay - downsampledOffset;
+          downsampledOffset = 0;
+        } else {
+          downsampledOffset -= (downsamplerDelay * (*numChannels))/DOWN_SMPL_FAC;
+          sbrOffset = 0;
+        }
+
+        hSbrEncoder->inputDataDelay = downsamplerDelay;
+      }
+
+
+      /* Assign core encoder Bandwidth */
+      *coreBandwidth = lowestBandwidth;
+
+      /* Estimate sbr bitrate, 2.5 kBit/s per sbr channel */
+      hSbrEncoder->estimateBitrate += 2500 * (*numChannels);
+
+      /* initialize parametric stereo */
+      if (usePs)
+      {
+        PSENC_CONFIG psEncConfig;
+        FDK_ASSERT(hSbrEncoder->noElements == 1);
+        INT psTuningTableIdx = getPsTuningTableIndex(elInfo[0].bitRate, NULL);
+
+        psEncConfig.frameSize           = *frameLength; //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 += ( (((*sampleRate) * 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);
+
+          if(noError != (errorInfo = PSEnc_Init( hSbrEncoder->hParametricStereo,
+                                                &psEncConfig,
+                                                 hSbrEncoder->sbrElement[0]->sbrConfigData.noQmfSlots,
+                                                 hSbrEncoder->sbrElement[0]->sbrConfigData.noQmfBands
+                                                ,hSbrEncoder->dynamicRam
+                                                 )))
+          {
+            errorInfo = handBack(errorInfo);
+          }
+        }
+      }
+
+      hSbrEncoder->downsampledOffset = downsampledOffset;
+      hSbrEncoder->downmixSize = coreFrameLength*(*numChannels);
+      hSbrEncoder->bufferOffset = sbrOffset;
+      /* Delay Compensation: fill bitstream delay buffer with zero input signal */
+      if ( hSbrEncoder->nBitstrDelay > 0 )
+      {
+        error = FDKsbrEnc_DelayCompensation (hSbrEncoder, inputBuffer);
+        if (error != 0)
+          goto bail;
+      }
+
+      /* Set Output frame length */
+      *frameLength = coreFrameLength*DOWN_SMPL_FAC;
+      /* Input buffer offset */
+      *inputBufferOffset = fixMax(sbrOffset, downsampledOffset);
+
+
+    }
+
+    return error;
+
+bail:
+    /* Restore input settings */
+    *sampleRate = inputSampleRate;
+    *frameLength = coreFrameLength;
+    *numChannels = inputChannels;
+    *coreBandwidth = inputBandWidth;
+
+    return error;
+ }
+
+
+INT
+sbrEncoder_EncodeFrame(  HANDLE_SBR_ENCODER   hSbrEncoder,
+                         INT_PCM             *samples,
+                         UINT                 timeInStride,
+                         UINT                 sbrDataBits[(6)],
+                         UCHAR                sbrData[(6)][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,
+                                        timeInStride,
+                                       &sbrDataBits[el],
+                                        sbrData[el],
+                                        0
+                                       );
+      if (error)
+        return error;
+    }
+  }
+
+  if ( (hSbrEncoder->lfeChIdx!=-1) && (hSbrEncoder->fTimeDomainDownsampling) )
+  {
+      INT nOutSamples;
+
+      FDKaacEnc_Downsample(&hSbrEncoder->lfeDownSampler,
+                            samples + hSbrEncoder->downsampledOffset + hSbrEncoder->bufferOffset + hSbrEncoder->lfeChIdx,
+                            hSbrEncoder->frameSize,
+                            timeInStride,
+                            samples + hSbrEncoder->downsampledOffset + hSbrEncoder->lfeChIdx,
+                           &nOutSamples,
+                            hSbrEncoder->nChannels);
+  } /* lfe downsampler */
+
+  return 0;
+}
+
+
+INT sbrEncoder_UpdateBuffers(
+                            HANDLE_SBR_ENCODER hSbrEncoder,
+                            INT_PCM *timeBuffer
+                            )
+ {
+    if ( hSbrEncoder->downsampledOffset > 0 ) {
+      /* Move delayed downsampled data */
+      FDKmemcpy ( timeBuffer,
+                  timeBuffer + hSbrEncoder->downmixSize,
+                  sizeof(INT_PCM) * (hSbrEncoder->downsampledOffset) );
+    } else {
+      /* Move delayed input data */
+      FDKmemcpy ( timeBuffer,
+        timeBuffer + hSbrEncoder->nChannels * hSbrEncoder->frameSize,
+                  sizeof(INT_PCM) * hSbrEncoder->bufferOffset );
+    }
+    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_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_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);
+  info->build_date = __DATE__;
+  info->build_time = __TIME__;
+  info->title = "SBR Encoder";
+
+  /* Set flags */
+  info->flags = 0
+    | CAPF_SBR_HQ
+    | CAPF_SBR_PS_MPEG
+    ;
+  /* End of flags */
+
+  return 0;
+}
diff --git a/libSBRenc/src/sbr_misc.cpp b/libSBRenc/src/sbr_misc.cpp
new file mode 100644
index 0000000..c44be22
--- /dev/null
+++ b/libSBRenc/src/sbr_misc.cpp
@@ -0,0 +1,272 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/*!
+  \file
+  \brief  Sbr miscellaneous helper functions  
+*/
+#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/libSBRenc/src/sbr_misc.h b/libSBRenc/src/sbr_misc.h
new file mode 100644
index 0000000..33b9cf9
--- /dev/null
+++ b/libSBRenc/src/sbr_misc.h
@@ -0,0 +1,106 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/*!
+  \file
+  \brief  Sbr miscellaneous helper functions prototypes  
+  \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/libSBRenc/src/sbr_ram.cpp b/libSBRenc/src/sbr_ram.cpp
new file mode 100644
index 0000000..e304c39
--- /dev/null
+++ b/libSBRenc/src/sbr_ram.cpp
@@ -0,0 +1,222 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/*!
+  \file
+  \brief Memory layout
+   
+
+  This module declares all static and dynamic memory spaces
+*/
+#include "sbr_ram.h"
+
+#include "sbr.h"
+#include "genericStds.h"
+
+C_ALLOC_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, (6))
+C_ALLOC_MEM2(Ram_SbrElement, SBR_ELEMENT, 1, (6))
+
+/*! Filter states for QMF-analysis. <br>
+  Dimension: #MAXNRSBRCHANNELS * #SBR_QMF_FILTER_LENGTH
+*/
+C_AALLOC_MEM2_L (Ram_Sbr_QmfStatesAnalysis, FIXP_QAS, QMF_FILTER_LENGTH, (6), 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*QMF_CHANNELS), (6), 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*QMF_CHANNELS), (6))
+
+/*! Frequency band table (low res) <br>
+  Dimension #MAX_FREQ_COEFFS/2+1
+*/
+C_ALLOC_MEM2 (Ram_Sbr_freqBandTableLO, UCHAR, (MAX_FREQ_COEFFS/2+1), (6))
+
+/*! Frequency band table (high res) <br>
+  Dimension #MAX_FREQ_COEFFS +1
+*/
+C_ALLOC_MEM2 (Ram_Sbr_freqBandTableHI, UCHAR, (MAX_FREQ_COEFFS+1), (6))
+
+/*! vk matser table <br>
+  Dimension #MAX_FREQ_COEFFS +1
+*/
+C_ALLOC_MEM2 (Ram_Sbr_v_k_master, UCHAR, (MAX_FREQ_COEFFS+1), (6))
+
+
+/*
+  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), (6))
+
+/*! sbr_prevCompVec[ <br>
+  Dimension #MAX_NUM_CHANNELS*#MAX_FREQ_COEFFS]
+*/
+C_ALLOC_MEM2 (Ram_Sbr_prevEnvelopeCompensation, UCHAR, MAX_FREQ_COEFFS, (6))
+/*! sbr_guideScfb[ <br>
+  Dimension #MAX_NUM_CHANNELS*#MAX_FREQ_COEFFS]
+*/
+C_ALLOC_MEM2 (Ram_Sbr_guideScfb, UCHAR, MAX_FREQ_COEFFS, (6))
+
+/*! 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), (6))
+C_ALLOC_MEM2 (Ram_Sbr_guideVectorDiff, FIXP_DBL, (MAX_NO_OF_ESTIMATES*MAX_FREQ_COEFFS), (6))
+C_ALLOC_MEM2 (Ram_Sbr_guideVectorOrig, FIXP_DBL, (MAX_NO_OF_ESTIMATES*MAX_FREQ_COEFFS), (6))
+
+/*
+  Static Parametric Stereo memory
+*/
+C_AALLOC_MEM_L(Ram_PsQmfStatesSynthesis, FIXP_DBL, QMF_FILTER_LENGTH/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, (QMF_MAX_TIME_SLOTS/2 * QMF_CHANNELS), (6))
+
+  FIXP_DBL* GetRam_Sbr_envYBuffer (int n, UCHAR* dynamic_RAM) {
+    FDK_ASSERT(dynamic_RAM!=0);
+    return ((FIXP_DBL*) (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);
+    return ((FIXP_DBL*) (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);
+    return ((FIXP_DBL*) (dynamic_RAM + OFFSET_QMF + (ENV_R_BUFF_BYTE) + (n*(ENV_R_BUFF_BYTE+ENV_I_BUFF_BYTE))));
+  }
+
+
+
+
+/* @} */
+
+
+
+
+
diff --git a/libSBRenc/src/sbr_ram.h b/libSBRenc/src/sbr_ram.h
new file mode 100644
index 0000000..0f9e9cc
--- /dev/null
+++ b/libSBRenc/src/sbr_ram.h
@@ -0,0 +1,187 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/*!
+\file
+\brief Memory layout
+ 
+*/
+#ifndef __SBR_RAM_H
+#define __SBR_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*QMF_MAX_TIME_SLOTS)) )
+
+  #define ENV_R_BUFF_BYTE      ( (sizeof(FIXP_DBL)*((QMF_MAX_TIME_SLOTS) * MAX_HYBRID_BANDS)) )
+  #define ENV_I_BUFF_BYTE      ( (sizeof(FIXP_DBL)*((QMF_MAX_TIME_SLOTS) * MAX_HYBRID_BANDS)) )
+  #define Y_BUF_CH_BYTE        ( (2*sizeof(FIXP_DBL)*((QMF_MAX_TIME_SLOTS) * MAX_HYBRID_BANDS)) )
+
+
+#define ENV_R_BUF_PS_BYTE    ( (sizeof(FIXP_DBL)*QMF_MAX_TIME_SLOTS * QMF_CHANNELS / 2) )
+#define ENV_I_BUF_PS_BYTE    ( (sizeof(FIXP_DBL)*QMF_MAX_TIME_SLOTS * QMF_CHANNELS / 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>>1 )
+
+
+/* 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/libSBRenc/src/sbr_rom.cpp b/libSBRenc/src/sbr_rom.cpp
new file mode 100644
index 0000000..6fd51a2
--- /dev/null
+++ b/libSBRenc/src/sbr_rom.cpp
@@ -0,0 +1,740 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/*!
+  \file
+  \brief  Definition of constant tables
+   
+
+  This module contains most of the constant data that can be stored in ROM.
+*/
+
+#include "sbr_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[SBRENC_TUNING_SIZE] =
+{
+
+  /*** AAC ***/
+  /*                        sf,sfsp,sf,sfsp,nnb,nfo,saml,SM,FS*/
+
+  /*** mono ***/
+
+  /* 8/16 kHz dual rate */
+  { 8000,  10000,   8000, 1,  7, 6, 11,10, 1, 0, 6, SBR_MONO, 3 },
+  { 10000, 12000,   8000, 1, 11, 7, 13,12, 1, 0, 6, SBR_MONO, 3 },
+  { 12000, 16001,   8000, 1, 14,10, 13,13, 1, 0, 6, SBR_MONO, 3 },
+  { 16000, 24000,   8000, 1, 14,10, 14,14, 2, 0, 3, SBR_MONO, 2 }, /* placebo */
+  { 24000, 32000,   8000, 1, 14,10, 14,14, 2, 0, 3, SBR_MONO, 2 }, /* placebo */
+  { 32000, 48001,   8000, 1, 14,11, 15,15, 2, 0, 3, SBR_MONO, 2 }, /* placebo */ /* bitrates higher than 48000 not supported by AAC core */
+
+  /* 11/22 kHz dual rate */
+  {  8000, 10000,  11025, 1,  5, 4,  6, 6, 1, 0, 6, SBR_MONO, 3 },
+  { 10000, 12000,  11025, 1,  8, 5, 12, 9, 1, 0, 6, SBR_MONO, 3 },
+  { 12000, 16000,  11025, 1, 12, 8, 13, 8, 1, 0, 6, SBR_MONO, 3 },
+  { 16000, 20000,  11025, 1, 12, 8, 13, 8, 1, 0, 6, SBR_MONO, 3 }, /* at such "high" bitrates it's better to upsample the input */
+  { 20000, 24001,  11025, 1, 13, 9, 13, 8, 1, 0, 6, SBR_MONO, 3 }, /* signal by a factor of 2 before sending it into the encoder */
+  { 24000, 32000,  11025, 1, 14,10, 14, 9, 2, 0, 3, SBR_MONO, 2 }, /* placebo */
+  { 32000, 48000,  11025, 1, 15,11, 15,10, 2, 0, 3, SBR_MONO, 2 }, /* placebo */
+  { 48000, 64001,  11025, 1, 15,11, 15,10, 2, 0, 3, SBR_MONO, 1 }, /* placebo */
+
+  /* 12/24 kHz dual rate */
+  {  8000, 10000,  12000, 1,  4, 3,  6, 6, 1, 0, 6, SBR_MONO, 3 }, /* nominal:  8 kbit/s */
+  { 10000, 12000,  12000, 1,  7, 4, 11, 8, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 10 kbit/s */
+  { 12000, 16000,  12000, 1, 11, 7, 12, 8, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 12 kbit/s */
+  { 16000, 20000,  12000, 1, 11, 7, 12, 8, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 16 kbit/s */ /* at such "high" bitrates it's better to upsample the input */
+  { 20000, 24001,  12000, 1, 12, 8, 12, 8, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 20 kbit/s */ /* signal by a factor of 2 before sending it into the encoder */
+  { 24000, 32000,  12000, 1, 13, 9, 13, 9, 2, 0, 3, SBR_MONO, 2 }, /* placebo */
+  { 32000, 48000,  12000, 1, 14,10, 14,10, 2, 0, 3, SBR_MONO, 2 }, /* placebo */
+  { 48000, 64001,  12000, 1, 15,11, 15,11, 2, 0, 3, SBR_MONO, 1 }, /* placebo */
+
+  /* 16/32 kHz dual rate */
+  {  8000, 10000,  16000, 1,  1, 1, 0, 0,  1, 0, 6, SBR_MONO, 3 }, /* nominal:  8 kbit/s */
+  { 10000, 12000,  16000, 1,  2, 1, 6, 0,  1, 0, 6, SBR_MONO, 3 }, /* nominal: 10 kbit/s */
+  { 12000, 16000,  16000, 1,  4, 2, 6, 0,  1, 0, 6, SBR_MONO, 3 }, /* nominal: 12 kbit/s */
+  { 16000, 18000,  16000, 1,  4, 2, 8, 3,  1, 0, 6, SBR_MONO, 3 }, /* nominal: 16 kbit/s */
+  { 18000, 22000,  16000, 1,  6, 5,11, 7,  2, 0, 6, SBR_MONO, 2 }, /* nominal: 20 kbit/s */
+  { 22000, 28000,  16000, 1, 10, 9,12, 8,  2, 0, 6, SBR_MONO, 2 }, /* nominal: 24 kbit/s */
+  { 28000, 36000,  16000, 1, 12,12,13,13,  2, 0, 3, SBR_MONO, 2 }, /* nominal: 32 kbit/s */
+  { 36000, 44000,  16000, 1, 14,14,13,13,  2, 0, 3, SBR_MONO, 1 }, /* nominal: 40 kbit/s */
+  { 44000, 64001,  16000, 1, 15,15,13,13,  2, 0, 3, SBR_MONO, 1 }, /* nominal: 48 kbit/s */
+
+  /* 22.05/44.1 kHz dual rate */
+  /* {  8000, 11369,  22050, 1,  1, 1, 1, 1,  1, 0, 6, SBR_MONO, 3 }, */ /* nominal:  8 kbit/s */   /* encoder can not work stable at this extremely low bitrate */
+  { 11369, 16000,  22050, 1,  3, 1, 4, 4,  1, 0, 6, SBR_MONO, 3 }, /* nominal: 12 kbit/s */
+  { 16000, 18000,  22050, 1,  3, 1, 5, 4,  1, 0, 6, SBR_MONO, 3 }, /* nominal: 16 kbit/s */
+  { 18000, 22000,  22050, 1,  4, 4, 8, 5,  2, 0, 6, SBR_MONO, 2 }, /* nominal: 20 kbit/s */
+  { 22000, 28000,  22050, 1,  7, 6, 8, 6,  2, 0, 6, SBR_MONO, 2 }, /* nominal: 24 kbit/s */
+  { 28000, 36000,  22050, 1, 10,10, 9, 9,  2, 0, 3, SBR_MONO, 2 }, /* nominal: 32 kbit/s */
+  { 36000, 44000,  22050, 1, 11,11,10,10,  2, 0, 3, SBR_MONO, 1 }, /* nominal: 40 kbit/s */
+  { 44000, 64001,  22050, 1, 13,13,12,12,  2, 0, 3, SBR_MONO, 1 }, /* nominal: 48 kbit/s */
+
+  /* 24/48 kHz dual rate */
+  /* {  8000, 12000,  24000, 1,  1, 1, 1, 1,  1, 0, 6, SBR_MONO, 3 }, */ /* nominal:  8 kbit/s */   /* encoder can not work stable at this extremely low bitrate */
+  { 12000, 16000,  24000, 1,  3, 1, 4, 4,  1, 0, 6, SBR_MONO, 3 }, /* nominal: 12 kbit/s */
+  { 16000, 18000,  24000, 1,  3, 1, 5, 4,  1, 0, 6, SBR_MONO, 3 }, /* nominal: 16 kbit/s */
+  { 18000, 22000,  24000, 1,  4, 3, 8, 5,  2, 0, 6, SBR_MONO, 2 }, /* nominal: 20 kbit/s */
+  { 22000, 28000,  24000, 1,  7, 6, 8, 6,  2, 0, 6, SBR_MONO, 2 }, /* nominal: 24 kbit/s */
+  { 28000, 36000,  24000, 1, 10,10, 9, 9,  2, 0, 3, SBR_MONO, 2 }, /* nominal: 32 kbit/s */
+  { 36000, 44000,  24000, 1, 11,11,10,10,  2, 0, 3, SBR_MONO, 1 }, /* nominal: 40 kbit/s */
+  { 44000, 64001,  24000, 1, 13,13,11,11,  2, 0, 3, SBR_MONO, 1 }, /* nominal: 48 kbit/s */
+
+  /* 32/64 kHz dual rate */  /* placebo settings */
+  {  24000, 36000, 32000, 1,  4, 4, 4, 4,  2, 0, 3, SBR_MONO, 3 }, /* lowest range  */
+  {  36000, 60000, 32000, 1,  7, 7, 6, 6,  2, 0, 3, SBR_MONO, 2 }, /* lowest range */
+  {  60000, 72000, 32000, 1,  9, 9, 8, 8,  2, 0, 3, SBR_MONO, 1 }, /* low range */
+  {  72000,100000, 32000, 1, 11,11,10,10,  2, 0, 3, SBR_MONO, 1 }, /* SBR sweet spot */
+  { 100000,160001, 32000, 1, 13,13,11,11,  2, 0, 3, SBR_MONO, 1 }, /* backwards compatible */
+
+  /* 44.1/88.2 kHz dual rate */  /* placebo settings */
+  {  24000, 36000, 44100, 1,  4, 4, 4, 4,  2, 0, 3, SBR_MONO, 3 }, /* lowest range (multichannel rear) */
+  {  36000, 60000, 44100, 1,  7, 7, 6, 6,  2, 0, 3, SBR_MONO, 2 }, /* lowest range (multichannel rear) */
+  {  60000, 72000, 44100, 1,  9, 9, 8, 8,  2, 0, 3, SBR_MONO, 1 }, /* low range */
+  {  72000,100000, 44100, 1, 11,11,10,10,  2, 0, 3, SBR_MONO, 1 }, /* SBR sweet spot */
+  { 100000,160001, 44100, 1, 13,13,11,11,  2, 0, 3, SBR_MONO, 1 }, /* backwards compatible */
+
+  /* 48/96 kHz dual rate */  /* not yet finally tuned */
+  {  24000, 36000, 48000, 1,  4, 4, 9, 9,  2, 0, 3, SBR_MONO, 3 }, /* lowest range (multichannel rear) */
+  {  36000, 60000, 48000, 1,  7, 7,10,10,  2, 0, 3, SBR_MONO, 2 }, /* nominal: 40 */
+  {  60000, 72000, 48000, 1,  9, 9,10,10,  2, 0, 3, SBR_MONO, 1 }, /* nominal: 64 */
+  {  72000,100000, 48000, 1, 11,11,11,11,  2, 0, 3, SBR_MONO, 1 }, /* nominal: 80 */
+  { 100000,160001, 48000, 1, 13,13,11,11,  2, 0, 3, SBR_MONO, 1 }, /* nominal: 128 */
+
+  /*** stereo ***/
+  /* 08/16 kHz dual rate */
+  { 16000, 24000,   8000, 2,  6, 6,  9, 7, 1, 0,-3, SBR_SWITCH_LRC, 3 }, /* nominal: 20 kbit/s */ /* placebo */
+  { 24000, 28000,   8000, 2,  9, 9, 11, 9, 1, 0,-3, SBR_SWITCH_LRC, 3 }, /* nominal: 24 kbit/s */
+  { 28000, 36000,   8000, 2, 11, 9, 11, 9, 2, 0,-3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */
+  { 36000, 44000,   8000, 2, 13,11, 13,11, 2, 0,-3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */
+  { 44000, 52000,   8000, 2, 14,12, 13,12, 2, 0,-3, SBR_SWITCH_LRC, 2 }, /* nominal: 48 kbit/s */
+  { 52000, 60000,   8000, 2, 15,15, 13,13, 3, 0,-3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */
+  { 60000, 76000,   8000, 2, 15,15, 13,13, 3, 0,-3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */
+  { 76000,128001,   8000, 2, 15,15, 13,13, 3, 0,-3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */
+
+  /* 11/22 kHz dual rate */
+  { 16000, 24000,  11025, 2,  7, 5, 9, 7,  1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 20 kbit/s */ /* placebo */
+  { 24000, 28000,  11025, 2, 10, 8,10, 8,  1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 24 kbit/s */
+  { 28000, 36000,  11025, 2, 12, 8,12, 8,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */
+  { 36000, 44000,  11025, 2, 13, 9,13, 9,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */
+  { 44000, 52000,  11025, 2, 14,11,13,11,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 48 kbit/s */
+  { 52000, 60000,  11025, 2, 15,15,13,13,  3, 0, -3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */
+  { 60000, 76000,  11025, 2, 15,15,13,13,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */
+  { 76000,128001,  11025, 2, 15,15,13,13,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */
+
+  /* 12/24 kHz dual rate */
+  { 16000, 24000,  12000, 2,  6, 4, 9, 7,  1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 20 kbit/s */ /* placebo */
+  { 24000, 28000,  12000, 2,  9, 7,10, 8,  1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 24 kbit/s */
+  { 28000, 36000,  12000, 2, 11, 7,12, 8,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */
+  { 36000, 44000,  12000, 2, 12, 9,12, 9,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */
+  { 44000, 52000,  12000, 2, 13,12,13,12,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 48 kbit/s */
+  { 52000, 60000,  12000, 2, 14,14,13,13,  3, 0, -3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */
+  { 60000, 76000,  12000, 2, 15,15,13,13,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */
+  { 76000,128001,  12000, 2, 15,15,13,13,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */
+
+  /* 16/32 kHz dual rate */
+  { 16000, 24000,  16000, 2,  4, 2, 1, 0,  1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 20 kbit/s */
+  { 24000, 28000,  16000, 2,  8, 7,10, 8,  1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 24 kbit/s */
+  { 28000, 36000,  16000, 2, 10, 9,12,11,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */
+  { 36000, 44000,  16000, 2, 13,13,13,13,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */
+  { 44000, 52000,  16000, 2, 15,15,13,13,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 48 kbit/s */
+  { 52000, 60000,  16000, 2, 15,15,13,13,  3, 0, -3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */
+  { 60000, 76000,  16000, 2, 15,15,13,13,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */
+  { 76000,128001,  16000, 2, 15,15,13,13,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */
+
+  /* 22.05/44.1 kHz dual rate */
+  { 16000, 24000,  22050, 2,  2, 1, 1, 0,  1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 20 kbit/s */
+  { 24000, 28000,  22050, 2,  5, 4, 6, 5,  1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 24 kbit/s */
+  { 28000, 32000,  22050, 2,  5, 4, 8, 7,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 28 kbit/s */
+  { 32000, 36000,  22050, 2,  7, 6, 8, 7,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */
+  { 36000, 44000,  22050, 2, 10,10, 9, 9,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */
+  { 44000, 52000,  22050, 2, 12,12, 9, 9,  3, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 48 kbit/s */
+  { 52000, 60000,  22050, 2, 13,13,10,10,  3, 0, -3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */
+  { 60000, 76000,  22050, 2, 14,14,12,12,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */
+  { 76000,128001,  22050, 2, 14,14,12,12,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */
+
+  /* 24/48 kHz dual rate */
+  { 16000, 24000,  24000, 2,  2, 1, 1, 0,  1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 20 kbit/s */
+  { 24000, 28000,  24000, 2,  5, 5, 6, 6,  1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 24 kbit/s */
+  { 28000, 36000,  24000, 2,  7, 6, 8, 7,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */
+  { 36000, 44000,  24000, 2, 10,10, 9, 9,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */
+  { 44000, 52000,  24000, 2, 12,12, 9, 9,  3, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 48 kbit/s */
+  { 52000, 60000,  24000, 2, 13,13,10,10,  3, 0, -3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */
+  { 60000, 76000,  24000, 2, 14,14,12,12,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */
+  { 76000,128001,  24000, 2, 15,15,12,12,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */
+
+  /* 32/64 kHz dual rate */  /* placebo settings */
+  {  32000, 60000, 32000, 2,  4, 4, 4, 4,  2, 0, -3, SBR_SWITCH_LRC, 3 }, /* lowest range (multichannel rear) */
+  {  60000, 80000, 32000, 2,  7, 7, 6, 6,  3, 0, -3, SBR_SWITCH_LRC, 2 }, /* lowest range (multichannel rear) */
+  {  80000,112000, 32000, 2,  9, 9, 8, 8,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* low range */
+  { 112000,144000, 32000, 2, 11,11,10,10,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* SBR sweet spot */
+  { 144000,256001, 32000, 2, 13,13,11,11,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* backwards compatible */
+
+  /* 44.1/88.2 kHz dual rate */  /* placebo settings */
+  {  32000, 60000, 44100, 2,  4, 4, 4, 4,  2, 0, -3, SBR_SWITCH_LRC, 3 }, /* lowest range (multichannel rear) */
+  {  60000, 80000, 44100, 2,  7, 7, 6, 6,  3, 0, -3, SBR_SWITCH_LRC, 2 }, /* lowest range (multichannel rear) */
+  {  80000,112000, 44100, 2,  9, 9, 8, 8,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* low range */
+  { 112000,144000, 44100, 2, 11,11,10,10,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* SBR sweet spot */
+  { 144000,256001, 44100, 2, 13,13,11,11,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* backwards compatible */
+
+  /* 48/96 kHz dual rate */  /* not yet finally tuned */
+  {  32000, 60000, 48000, 2,  4, 4, 9, 9,  2, 0, -3, SBR_SWITCH_LRC, 3 }, /* lowest range (multichannel rear) */
+  {  60000, 80000, 48000, 2,  7, 7, 9, 9,  3, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 64 */
+  {  80000,112000, 48000, 2,  9, 9,10,10,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 96 */
+  { 112000,144000, 48000, 2, 11,11,11,11,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 128 */
+  { 144000,256001, 48000, 2, 13,13,11,11,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 192 */
+
+  /** AAC LOW DELAY SECTION **/
+
+  /* 22.05/44.1 kHz dual rate */
+  {  8000, 11369,  22050, 1,  1, 1, 1, 1,  1, 0, 6, SBR_MONO, 3 }, /* nominal:  8 kbit/s */ /**changed (not changed !!)*/
+  { 11369, 16000,  22050, 1,  1, 0, 3, 3,  1, 0, 6, SBR_MONO, 3 }, /* nominal: 12 kbit/s */
+  { 16000, 18000,  22050, 1,  2, 4, 4, 3,  1, 0, 6, SBR_MONO, 3 }, /* nominal: 16 kbit/s */
+  { 18000, 22000,  22050, 1,  4, 4, 5, 5,  2, 0, 6, SBR_MONO, 2 }, /* nominal: 20 kbit/s */
+  { 22000, 28000,  22050, 1,  4, 4, 6, 5,  2, 0, 6, SBR_MONO, 2 }, /* nominal: 24 kbit/s */
+  { 28000, 36000,  22050, 1,  7, 8, 8, 8,  2, 0, 3, SBR_MONO, 2 }, /* nominal: 32 kbit/s */
+  { 36000, 44000,  22050, 1,  9, 9, 9, 9,  2, 0, 3, SBR_MONO, 1 }, /* nominal: 40 kbit/s */
+  { 44000, 52000,  22050, 1, 11,11,11,11,  2, 0, 3, SBR_MONO, 1 }, /* nominal: 48 kbit/s */
+  { 52000, 64001,  22050, 1, 12,11,11,11,  2, 0, 3, SBR_MONO, 1 }, /* nominal: 56 kbit/s */
+
+  /* 24/48 kHz dual rate */
+  {  8000, 12000,  24000, 1,  1, 1, 1, 1,  1, 0, 6, SBR_MONO, 3 }, /* nominal:  8 kbit/s */ /**changed (not changed !!)*/
+  { 12000, 16000,  24000, 1,  1, 0, 3, 3,  1, 0, 6, SBR_MONO, 3 }, /* nominal: 12 kbit/s */
+  { 16000, 18000,  24000, 1,  2, 4, 4, 3,  1, 0, 6, SBR_MONO, 3 }, /* nominal: 16 kbit/s */
+  { 18000, 22000,  24000, 1,  4, 4, 5, 5,  2, 0, 6, SBR_MONO, 2 }, /* nominal: 20 kbit/s */
+  { 22000, 28000,  24000, 1,  4, 4, 6, 5,  2, 0, 6, SBR_MONO, 2 }, /* nominal: 24 kbit/s */
+  { 28000, 36000,  24000, 1,  6, 8, 8, 8,  2, 0, 3, SBR_MONO, 2 }, /* nominal: 32 kbit/s */
+  { 36000, 44000,  24000, 1,  8, 9, 9, 9,  2, 0, 3, SBR_MONO, 1 }, /* nominal: 40 kbit/s */
+  { 44000, 52000,  24000, 1, 12,11,11,10,  2, 0, 3, SBR_MONO, 1 }, /* nominal: 48 kbit/s */
+  { 52000, 64001,  24000, 1, 13,11,11,10,  2, 0, 3, SBR_MONO, 1 }, /* nominal: 48 kbit/s */
+
+  /* 22.05/44.1 kHz dual rate */
+  { 24000, 28000,  22050, 2,  3, 2, 5, 4,  1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 24 kbit/s */
+  { 28000, 32000,  22050, 2,  3, 2, 7, 6,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 28 kbit/s */
+  { 32000, 36000,  22050, 2,  5, 4, 7, 6,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */
+  { 36000, 44000,  22050, 2,  5, 8, 8, 8,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */
+  { 44000, 52000,  22050, 2,  7,10, 8, 8,  3, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 48 kbit/s */
+  { 52000, 60000,  22050, 2,  9,11, 9, 9,  3, 0, -3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */
+  { 60000, 76000,  22050, 2, 10,12,10,11,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */
+  { 76000, 82000,  22050, 2, 12,12,11,11,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */
+  { 82000,128001,  22050, 2, 13,12,11,11,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */
+
+  /* 24/48 kHz dual rate */
+  { 24000, 28000,  24000, 2,  3, 3, 5, 5,  1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 24 kbit/s */
+  { 28000, 36000,  24000, 2,  5, 4, 7, 6,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */
+  { 36000, 44000,  24000, 2,  4, 8, 8, 8,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */
+  { 44000, 52000,  24000, 2,  6,10, 8, 8,  3, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 48 kbit/s */
+  { 52000, 60000,  24000, 2,  9,11, 9, 9,  3, 0, -3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */
+  { 60000, 76000,  24000, 2, 11,12,10,11,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */
+  { 76000, 88000,  24000, 2, 12,13,11,11,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */
+  { 88000,128001,  24000, 2, 13,13,11,11,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 92 kbit/s */
+
+
+};
+
+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/libSBRenc/src/sbr_rom.h b/libSBRenc/src/sbr_rom.h
new file mode 100644
index 0000000..4cd1ba5
--- /dev/null
+++ b/libSBRenc/src/sbr_rom.h
@@ -0,0 +1,132 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/*!
+\file
+\brief Declaration of constant tables
+ 
+*/
+#ifndef __SBR_ROM_H
+#define __SBR_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];
+
+#define SBRENC_AACLC_TUNING_SIZE   124
+#define SBRENC_AACELD_TUNING_SIZE   35
+#define SBRENC_AACELD2_TUNING_SIZE  31
+
+#define SBRENC_TUNING_SIZE (SBRENC_AACLC_TUNING_SIZE + SBRENC_AACELD_TUNING_SIZE)
+
+extern const sbrTuningTable_t sbrTuningTable[SBRENC_TUNING_SIZE];
+
+extern const psTuningTable_t psTuningTable[4];
+
+
+#endif
diff --git a/libSBRenc/src/sbrenc_freq_sca.cpp b/libSBRenc/src/sbrenc_freq_sca.cpp
new file mode 100644
index 0000000..bbcb29e
--- /dev/null
+++ b/libSBRenc/src/sbrenc_freq_sca.cpp
@@ -0,0 +1,671 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/*!
+  \file
+  \brief  frequency scale  
+*/
+
+#include "sbrenc_freq_sca.h"
+#include "sbr_misc.h"
+
+#include "genericStds.h"
+
+/*  StartFreq */
+static INT getStartFreq(INT fs, const INT start_freq);
+
+/* StopFreq */
+static INT getStopFreq(INT fs, const INT stop_freq, const INT noChannels);
+
+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 QMFbands, INT fs)
+{
+  INT result;
+
+  if ( startFreq < 0 || startFreq > 15) {
+    return -1;
+  }
+  /* Update startFreq struct */
+  result = getStartFreq(fs, startFreq);
+
+  result = (result*fs/QMFbands+1)>>1;
+
+  return (result);
+
+} /* End FDKsbrEnc_getSbrStartFreqRAW */
+
+
+/*******************************************************************************
+ Functionname:  getSbrStopFreq
+ *******************************************************************************
+ Description:
+
+ Arguments:
+
+ Return:
+ *******************************************************************************/
+INT FDKsbrEnc_getSbrStopFreqRAW  (INT stopFreq, INT QMFbands, INT fs)
+{
+  INT result;
+
+  if ( stopFreq < 0 || stopFreq > 13)
+    return -1;
+
+
+  /* Uppdate stopFreq struct */
+  result = getStopFreq( fs, stopFreq, QMFbands);
+  result =   (result*fs/QMFbands+1)>>1;
+
+  return (result);
+} /* End getSbrStopFreq */
+
+
+/*******************************************************************************
+ Functionname:  getStartFreq
+ *******************************************************************************
+ Description:
+
+ Arguments:
+
+ Return:
+ *******************************************************************************/
+static INT
+getStartFreq(INT fs, const INT start_freq)
+{
+  INT k0_min;
+
+  switch(fs){
+  case 16000: k0_min = 24;
+    break;
+  case 22050: k0_min = 17;
+    break;
+  case 24000: k0_min = 16;
+    break;
+  case 32000: k0_min = 16;
+    break;
+  case 44100: k0_min = 12;
+    break;
+  case 48000: k0_min = 11;
+    break;
+  case 64000: k0_min = 10;
+    break;
+  case 88200: k0_min = 7;
+    break;
+  case 96000: k0_min = 7;
+    break;
+  default:
+    k0_min=11; /* illegal fs */
+  }
+
+
+  switch (fs) {
+
+  case 16000:
+    {
+      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 22050:
+    {
+      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 24000:
+    {
+      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 32000:
+    {
+      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 44100:
+  case 48000:
+  case 64000:
+    {
+      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 88200:
+  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 fs, const INT stop_freq, const INT noChannels)
+{
+  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};
+
+  switch(fs){
+  case 16000: k1_min = 48;
+              v_stop_freq =v_stop_freq_16;
+    break;
+  case 22050: k1_min = 35;
+              v_stop_freq =v_stop_freq_22;
+    break;
+  case 24000: k1_min = 32;
+              v_stop_freq =v_stop_freq_24;
+    break;
+  case 32000: k1_min = 32;
+              v_stop_freq =v_stop_freq_32;
+    break;
+  case 44100: k1_min = 23;
+              v_stop_freq =v_stop_freq_44;
+    break;
+  case 48000: k1_min = 21;
+              v_stop_freq =v_stop_freq_48;
+    break;
+  case 64000: k1_min = 20;
+              v_stop_freq =v_stop_freq_64;
+    break;
+  case 88200: k1_min = 15;
+              v_stop_freq =v_stop_freq_88;
+    break;
+  case 96000: k1_min = 13;
+              v_stop_freq =v_stop_freq_96;
+    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:
+
+ Return:
+ *******************************************************************************/
+INT
+FDKsbrEnc_FindStartAndStopBand(const INT samplingFreq,
+                     const INT noChannels,
+                     const INT startFreq,
+                     const INT stopFreq,
+                     const SR_MODE sampleRateMode,
+                     INT *k0,
+                     INT *k2)
+{
+
+  /* Update startFreq struct */
+  *k0 = getStartFreq(samplingFreq, startFreq);
+
+  /* Test if start freq is outside corecoder range */
+  if( ( sampleRateMode == 1 ) &&
+      ( samplingFreq*noChannels  <
+        2**k0 * samplingFreq) ) {
+    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(samplingFreq, stopFreq, noChannels);
+  } 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 ( (samplingFreq == 44100) && ( (*k2 - *k0) > MAX_FREQ_COEFFS_FS44100 ) )
+    return (1); /* Number of bands exceeds valid range of MAX_FREQ_COEFFS for fs=44.1kHz */
+
+  if ( (samplingFreq >= 48000) && ( (*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*/
+        {
+          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, SR_MODE drOrSr,
+            INT noQMFChannels)
+{
+  INT i;
+  INT divider;
+  INT max1,max2;
+
+  /* Check if we use a Dual rate => diver=2 else 1 */
+  divider = (drOrSr == DUAL_RATE) ? 2 : 1;
+
+  if( (v_k_master[*xover_band] > (noQMFChannels/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] < (noQMFChannels/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/libSBRenc/src/sbrenc_freq_sca.h b/libSBRenc/src/sbrenc_freq_sca.h
new file mode 100644
index 0000000..613694a
--- /dev/null
+++ b/libSBRenc/src/sbrenc_freq_sca.h
@@ -0,0 +1,129 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/*!
+  \file
+  \brief  frequency scale prototypes  
+*/
+#ifndef __FREQ_SCA2_H
+#define __FREQ_SCA2_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,
+            SR_MODE drOrSr,
+            INT noQMFChannels);
+
+void  FDKsbrEnc_UpdateLoRes(UCHAR * v_lores,
+                  INT *num_lores,
+                  UCHAR * v_hires,
+                  INT num_hires);
+
+INT
+FDKsbrEnc_FindStartAndStopBand(const INT samplingFreq,
+                     const INT noChannels,
+                     const INT startFreq,
+                     const INT stop_freq,
+                     const SR_MODE sampleRateMode,
+                     INT *k0,
+                     INT *k2);
+
+INT FDKsbrEnc_getSbrStartFreqRAW (INT startFreq, INT QMFbands, INT fs );
+INT FDKsbrEnc_getSbrStopFreqRAW  (INT stopFreq, INT QMFbands, INT fs);
+#endif
diff --git a/libSBRenc/src/ton_corr.cpp b/libSBRenc/src/ton_corr.cpp
new file mode 100644
index 0000000..3142870
--- /dev/null
+++ b/libSBRenc/src/ton_corr.cpp
@@ -0,0 +1,883 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+#include "ton_corr.h"
+
+#include "sbr_ram.h"
+#include "sbr_misc.h"
+#include "genericStds.h"
+#include "autocorr2nd.h"
+
+
+
+/***************************************************************************
+
+  Send autoCorrSecondOrder to mlfile
+
+****************************************************************************/
+
+/**************************************************************************/
+/*!
+  \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;     /* Numer of Bands */
+  INT     buffLen           = hTonCorr->bufferLength;      /* Numer 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;
+
+#define BAND_V_SIZE QMF_MAX_TIME_SLOTS
+#define NUM_V_COMBINE 8 /* Must be a divisor of 64 and fulfill the ASSERTs below */
+
+  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 accross 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++;
+    }
+  }
+
+  FDK_ASSERT(noEstPerFrame == 2);
+
+
+  C_ALLOC_SCRATCH_END(realBuf, 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--;
+    targetStopBand = patchParam[patch].targetStartBand + patchParam[patch].numBandsInPatch;
+  }
+
+  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);
+
+  FDKmemclear(hTonCorr, sizeof(SBR_TON_CORR_EST));
+
+  for (i=0; i<MAX_NO_OF_ESTIMATES; i++) {
+    hTonCorr->quotaMatrix[i] = quotaMatrix + (i*QMF_CHANNELS);
+    hTonCorr->signMatrix[i]  = signMatrix  + (i*QMF_CHANNELS);
+  }
+
+  FDKsbrEnc_CreateSbrMissingHarmonicsDetector (&hTonCorr->sbrMissingHarmonicsDetector, chan);
+
+  return 0;
+}
+
+
+
+/**************************************************************************/
+/*!
+  \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.
+  Assing 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 = 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 = 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.*/
+  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/libSBRenc/src/ton_corr.h b/libSBRenc/src/ton_corr.h
new file mode 100644
index 0000000..a37eca5
--- /dev/null
+++ b/libSBRenc/src/ton_corr.h
@@ -0,0 +1,212 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/*!
+  \file
+  \brief  General tonality correction detector module.
+*/
+#ifndef _TON_CORR_EST_H
+#define _TON_CORR_EST_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.*/
+  INT 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[QMF_CHANNELS];      /*!< Vector holding the averaged energies for every QMF channel */
+
+  SCHAR indexVector[QMF_CHANNELS];           /*!< 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/libSBRenc/src/tran_det.cpp b/libSBRenc/src/tran_det.cpp
new file mode 100644
index 0000000..b6cde99
--- /dev/null
+++ b/libSBRenc/src/tran_det.cpp
@@ -0,0 +1,701 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+#include "tran_det.h"
+
+#include "fram_gen.h"
+#include "sbr_ram.h"
+#include "sbr_misc.h"
+
+#include "genericStds.h"
+
+#define NORM_QMF_ENERGY 5.684341886080801486968994140625e-14 /* 2^-44 */
+
+/* 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
+*******************************************************************************/
+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 stop)
+{
+  INT i,j;
+  INT len1,len2;
+  FIXP_DBL delta,tmp0,tmp1,tmp2;
+  FIXP_DBL accu1,accu2,delta_sum,result;
+
+  FDK_ASSERT(scaleEnergies[0] >= 0);
+
+  /* equal for aac (would be not equal for mp3) */
+  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);
+
+  delta_sum = FL2FXCONST_DBL(0.0f);
+
+  /* Sum up energies of all QMF-timeslots for both halfs */
+  for (j=0; j<nSfb; j++) {
+    #define NRG_SCALE  3
+    /* init with some energy to prevent division by zero
+       and to prevent splitting for very low levels */
+    accu1 = ((FL2FXCONST_DBL((1.0e6*NORM_QMF_ENERGY*8.0/32))) << fixMin(scaleEnergies[0],25))>>NRG_SCALE;  /* complex init for compare with original version */
+    accu2 = ((FL2FXCONST_DBL((1.0e6*NORM_QMF_ENERGY*8.0/32))) << fixMin(scaleEnergies[0],25))>>NRG_SCALE;  /* can be simplified in dsp implementation */
+
+    /* Sum up energies in first half */
+    for (i=start; i<border; i++) {
+      accu1 += (Energies[i][j]>>NRG_SCALE);
+    }
+
+    /* Sum up energies in second half */
+    for (i=border; i<stop; i++) {
+      accu2 += (Energies[i][j]>>NRG_SCALE);
+    }
+
+    /* Energy change in current band */
+    tmp0 = CalcLdData(accu2);
+    tmp1 = CalcLdData(accu1);
+    tmp2 = (tmp0 - tmp1 + CalcLdData(len1)-CalcLdData(len2));
+    delta = fixp_abs(fMult(tmp2, FL2FXCONST_DBL(0.6931471806f)));
+
+    /* Weighting with amplitude ratio of this band */
+    result = (EnergyTotal == FL2FXCONST_DBL(0.0f))
+            ? FL2FXCONST_DBL(0.f)
+            : FDKsbrEnc_LSI_divide_scale_fract( (accu1+accu2),
+                                      (EnergyTotal>>NRG_SCALE)+(FIXP_DBL)1,
+                                      (FIXP_DBL)MAXVAL_DBL >> fixMin(scaleEnergies[0],(DFRACT_BITS-1)) );
+
+    delta_sum += (FIXP_DBL)(fMult(sqrtFixp(result), delta));
+  }
+
+  return fMult(delta_sum, pos_weight);
+}
+
+
+/*******************************************************************************
+ Functionname:  addLowbandEnergies
+ *******************************************************************************
+ \brief   Calculates total lowband energy
+
+ The return value nrgTotal is scaled by the factor (1/32.0)
+
+ \return  total energy in the lowband
+*******************************************************************************/
+static FIXP_DBL addLowbandEnergies(FIXP_DBL **Energies,
+                                   int       *scaleEnergies,
+                                   int        YBufferWriteOffset,
+                                   int        nrgSzShift,
+                                   int        tran_off,
+                                   UCHAR     *freqBandTable,
+                                   int        slots)
+{
+  FIXP_DBL nrgTotal;
+  FIXP_DBL accu1 = FL2FXCONST_DBL(0.0f);
+  FIXP_DBL accu2 = FL2FXCONST_DBL(0.0f);
+  int tran_offdiv2 = tran_off>>nrgSzShift;
+  int ts,k;
+
+  /* Sum up lowband energy from one frame at offset tran_off */
+  for (ts=tran_offdiv2; ts<YBufferWriteOffset; ts++) {
+    for (k = 0; k < freqBandTable[0]; k++) {
+      accu1 += Energies[ts][k] >> 6;
+    }
+  }
+  for (; ts<tran_offdiv2+(slots>>nrgSzShift); ts++) {
+    for (k = 0; k < freqBandTable[0]; k++) {
+      accu2 += Energies[ts][k] >> 6;
+    }
+  }
+
+  nrgTotal = ( (accu1 >> fixMin(scaleEnergies[0],(DFRACT_BITS-1)))
+           +   (accu2 >> fixMin(scaleEnergies[1],(DFRACT_BITS-1))) ) << (2);
+
+  return(nrgTotal);
+}
+
+
+/*******************************************************************************
+ 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 (1/32.0) and scaleEnergies[0]
+ The return value nrgTotal is scaled by the factor (1/32.0)
+
+ \return  total energy in the highband
+*******************************************************************************/
+
+static FIXP_DBL addHighbandEnergies(FIXP_DBL **RESTRICT Energies, /*!< input */
+                                    INT       *scaleEnergies,
+                                    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;
+  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++) {
+    slotIn = 2*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+i)>>1][k] >> 5);
+        }
+      }
+      EnergiesM[slotOut][j] = accu;
+    }
+  }
+
+  scale = fixMin(8,scaleEnergies[0]);      /* scale energies down before add up */
+
+  if ((scaleEnergies[0]-1) > (DFRACT_BITS-1) )
+    nrgTotal = FL2FXCONST_DBL(0.0f);
+  else {
+    /* Now add all energies */
+    accu = FL2FXCONST_DBL(0.0f);
+    for (slotOut=0; slotOut<sbrSlots; slotOut++) {
+      for (j=0; j<nSfb; j++) {
+        accu += (EnergiesM[slotOut][j] >> scale);
+      }
+    }
+    nrgTotal = accu >> (scaleEnergies[0]-scale);
+  }
+
+  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)
+{
+  if (tran_vector[1]==0) /* no transient was detected */
+  {
+    FIXP_DBL delta;
+    FIXP_DBL EnergiesM[NUMBER_TIME_SLOTS_2304][MAX_FREQ_COEFFS];
+    FIXP_DBL EnergyTotal,newLowbandEnergy,newHighbandEnergy;
+    INT border;
+    INT sbrSlots = fMultI(GetInvInt(timeStep),no_cols);
+
+    FDK_ASSERT( sbrSlots * timeStep == no_cols );
+
+    /*
+      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,
+                                            EnergiesM,
+                                            freqBandTable,
+                                            nSfb,
+                                            sbrSlots,
+                                            timeStep);
+
+    if ( h_sbrTransientDetector->frameShift != 0 ) {
+      if (tran_vector[1]==0)
+        tran_vector[0] = 0;
+    } else
+    {
+      /* 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 + h_sbrTransientDetector->prevLowBandEnergy) >> 1;
+      EnergyTotal += newHighbandEnergy;
+      /* The below border should specify the same position as the middle border
+         of a FIXFIX-frame with 2 envelopes. */
+      border = (sbrSlots+1) >> 1;
+
+      delta = spectralChange(EnergiesM,
+                             scaleEnergies,
+                             EnergyTotal,
+                             nSfb,
+                             0,
+                             border,
+                             sbrSlots);
+
+      if (delta > (h_sbrTransientDetector->split_thr >> LD_DATA_SHIFT)) /* delta scaled by 1/64 */
+        tran_vector[0] = 1; /* Set flag for splitting */
+      else
+        tran_vector[0] = 0;
+    }
+
+    /* Update prevLowBandEnergy */
+    h_sbrTransientDetector->prevLowBandEnergy = newLowbandEnergy;
+    h_sbrTransientDetector->prevHighBandEnergy = newHighbandEnergy;
+  }
+}
+
+/*
+ * 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 += fMult(Energies[j][i], i_noCols);
+    for (; j<endEnergy; j++)
+      accu1 += fMult(Energies[j][i], i_noCols);
+
+    mean_val = (accu0 >> scaleFactor0) + (accu1 >> 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 = fPow2(temp);
+      temp = fMult(temp, i_noCols1);
+      accu += temp;
+    }
+    for (; j<endEnergy; j++) {
+      temp = ((FIXP_DBL)mean_val - ((FIXP_DBL)Energies[j][i] >> scaleFactor1))<<shift;
+      temp = fPow2(temp);
+      temp = fMult(temp, i_noCols1);
+      accu += temp;
+    }
+
+    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*QMF_MAX_TIME_SLOTS);
+  FIXP_DBL *RESTRICT pEnergiesTemp = EnergiesTemp;
+  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 += pEnergiesTemp[jIndex+d]; /* R */
+        delta -= pEnergiesTemp[jIndex-d]; /* L */
+        delta -= thres;
+
+        if ( delta > (FIXP_DBL)0 ) {
+          tran += fMult(i_thres, delta);
+        }
+      }
+      transients[jpBM] += tran;
+    }
+  }
+  C_ALLOC_SCRATCH_END(EnergiesTemp, FIXP_DBL, 2*QMF_MAX_TIME_SLOTS);
+}
+
+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,
+                                   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_fix, framedur_fix;
+    INT scale_0, scale_1;
+
+    FDKmemclear(h_sbrTransientDetector,sizeof(SBR_TRANSIENT_DETECTOR));
+
+    h_sbrTransientDetector->frameShift = frameShift;
+    h_sbrTransientDetector->tran_off = tran_off;
+
+    if(codecBitrate) {
+      bitrateFactor_fix = fDivNorm((FIXP_DBL)totalBitrate, (FIXP_DBL)(codecBitrate<<2),&scale_0);
+    }
+    else {
+      bitrateFactor_fix = FL2FXCONST_DBL(1.0/4.0);
+      scale_0 = 0;
+    }
+
+    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), &scale_1);
+
+    scale_1 = -(scale_1 + scale_0 + 2);
+
+    FDK_ASSERT(no_cols <= QMF_MAX_TIME_SLOTS);
+    FDK_ASSERT(no_rows <= QMF_CHANNELS);
+
+    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;
+
+    if (scale_1>=0) {
+      h_sbrTransientDetector->split_thr = fMult(tmp, bitrateFactor_fix) >> scale_1;
+    }
+    else {
+      h_sbrTransientDetector->split_thr = fMult(tmp, bitrateFactor_fix) << (-scale_1);
+    }
+
+    h_sbrTransientDetector->no_rows = no_rows;
+    h_sbrTransientDetector->mode = params->tran_det_mode;
+    h_sbrTransientDetector->prevLowBandEnergy = FL2FXCONST_DBL(0.0f);
+
+    return (0);
+}
+
diff --git a/libSBRenc/src/tran_det.h b/libSBRenc/src/tran_det.h
new file mode 100644
index 0000000..7e9a93c
--- /dev/null
+++ b/libSBRenc/src/tran_det.h
@@ -0,0 +1,150 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/*!
+  \file
+  \brief  Transient detector prototypes  
+*/
+#ifndef __TRAN_DET_H
+#define __TRAN_DET_H
+
+#include "sbr_encoder.h"
+#include "sbr_def.h"
+
+typedef struct
+{
+  FIXP_DBL  transients[QMF_MAX_TIME_SLOTS+(QMF_MAX_TIME_SLOTS/2)];
+  FIXP_DBL  thresholds[QMF_CHANNELS];
+  FIXP_DBL  tran_thr;              /* Master threshold for transient signals */
+  FIXP_DBL  split_thr;             /* Threshold for splitting FIXFIX-frames into 2 env */
+  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;
+
+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,
+                            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);
+
+#endif