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diff --git a/libSBRdec/include/sbrdecoder.h b/libSBRdec/include/sbrdecoder.h
new file mode 100644
index 0000000..300a6d9
--- /dev/null
+++ b/libSBRdec/include/sbrdecoder.h
@@ -0,0 +1,337 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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 SBR decoder library ******************
+
+   Author(s):
+   Description: SBR decoder front-end prototypes and definitions.
+
+******************************************************************************/
+
+#ifndef __SBRDECODER_H
+#define __SBRDECODER_H
+
+#include "common_fix.h"
+
+#include "FDK_bitstream.h"
+#include "FDK_audio.h"
+
+
+#define SBR_DEBUG_EXTHLP "\
+--- SBR ---\n\
+    0x00000010 Ancillary data and SBR-Header\n\
+    0x00000020 SBR-Side info\n\
+    0x00000040 Decoded SBR-bitstream data, e.g. envelope data\n\
+    0x00000080 SBR-Bitstream statistics\n\
+    0x00000100 Miscellaneous SBR-messages\n\
+    0x00000200 SBR-Energies and gains in the adjustor\n\
+    0x00000400 Fatal SBR errors\n\
+    0x00000800 Transposer coefficients for inverse filtering\n\
+"
+
+/* Capability flags */
+#define CAPF_SBR_LP          0x00000001   /*!< Flag indicating library's capability of Low Power mode.            */
+#define CAPF_SBR_HQ          0x00000002   /*!< Flag indicating library's capability of High Quality mode.         */
+#define CAPF_SBR_DRM_BS      0x00000004   /*!< Flag indicating library's capability to decode DRM SBR data.       */
+#define CAPF_SBR_CONCEALMENT 0x00000008   /*!< Flag indicating library's capability to conceal erroneous frames.  */
+#define CAPF_SBR_DRC         0x00000010   /*!< Flag indicating library's capability for Dynamic Range Control.    */
+#define CAPF_SBR_PS_MPEG     0x00000020   /*!< Flag indicating library's capability to do MPEG Parametric Stereo. */
+#define CAPF_SBR_PS_DRM      0x00000040   /*!< Flag indicating library's capability to do DRM Parametric Stereo.  */
+
+typedef enum
+{
+  SBRDEC_OK = 0,                       /*!< All fine.                                                       */
+  /* SBRDEC_CONCEAL,                                                                                        */
+  /* SBRDEC_NOSYNCH,                                                                                        */
+  /* SBRDEC_ILLEGAL_PROGRAM,                                                                                */
+  /* SBRDEC_ILLEGAL_TAG,                                                                                    */
+  /* SBRDEC_ILLEGAL_CHN_CONFIG,                                                                             */
+  /* SBRDEC_ILLEGAL_SECTION,                                                                                */
+  /* SBRDEC_ILLEGAL_SCFACTORS,                                                                              */
+  /* SBRDEC_ILLEGAL_PULSE_DATA,                                                                             */
+  /* SBRDEC_MAIN_PROFILE_NOT_IMPLEMENTED,                                                                   */
+  /* SBRDEC_GC_NOT_IMPLEMENTED,                                                                             */
+  /* SBRDEC_ILLEGAL_PLUS_ELE_ID,                                                                            */
+  SBRDEC_CREATE_ERROR,                 /*!<                                                                 */
+  SBRDEC_NOT_INITIALIZED,              /*!<                                                                 */
+  SBRDEC_MEM_ALLOC_FAILED,             /*!< Memory allocation failed. Probably not enough memory available. */
+  SBRDEC_PARSE_ERROR,                  /*!<                                                                 */
+  SBRDEC_UNSUPPORTED_CONFIG,           /*!<                                                                 */
+  SBRDEC_SET_PARAM_FAIL                /*!<                                                                 */
+} SBR_ERROR;
+
+typedef enum
+{
+  SBR_SYSTEM_BITSTREAM_DELAY,          /*!< System: Switch to enable an additional SBR bitstream delay of one frame. */
+  SBR_QMF_MODE,                        /*!< Set QMF mode, either complex or low power. */
+  SBR_LD_QMF_TIME_ALIGN,               /*!< Set QMF type, either LD-MPS or CLDFB. Relevant for ELD streams only. */
+  SBR_BS_INTERRUPTION                  /*!< Signal bit stream interruption. Value is ignored. */
+} SBRDEC_PARAM;
+
+typedef struct SBR_DECODER_INSTANCE *HANDLE_SBRDECODER;
+
+
+#ifdef __cplusplus
+extern "C"
+{
+#endif
+
+
+/**
+ * \brief        Allocates and initializes one SBR decoder instance.
+ * \param pSelf  Pointer to where a SBR decoder handle is copied into.
+ * \return       Error code.
+ */
+SBR_ERROR sbrDecoder_Open ( HANDLE_SBRDECODER *pSelf );
+
+/**
+ * \brief  Initialize a SBR decoder runtime instance. Must be called before decoding starts.
+ *
+ * \param self             Handle to a SBR decoder instance.
+ * \param sampleRateIn     Input samplerate of the SBR decoder instance.
+ * \param sampleRateOut    Output samplerate of the SBR decoder instance.
+ * \param samplesPerFrame  Number of samples per frames.
+ * \param coreCodec        Audio Object Type (AOT) of the core codec.
+ * \param elementID        Table with MPEG-4 element Ids in canonical order.
+ * \param forceReset       Flag that enforces a complete decoder reset.
+ *
+ * \return  Error code.
+ */
+SBR_ERROR sbrDecoder_InitElement (
+        HANDLE_SBRDECODER       self,
+        const int               sampleRateIn,
+        const int               sampleRateOut,
+        const int               samplesPerFrame,
+        const AUDIO_OBJECT_TYPE coreCodec,
+        const MP4_ELEMENT_ID    elementID,
+        const int               elementIndex
+        );
+
+/**
+ * \brief pass out of band SBR header to SBR decoder
+ *
+ * \param self         Handle to a SBR decoder instance.
+ * \param hBs          bit stream handle data source.
+ * \param elementID    SBR element ID.
+ * \param elementIndex SBR element index.
+ *
+ * \return  Error code.
+ */
+INT sbrDecoder_Header (
+        HANDLE_SBRDECODER       self,
+        HANDLE_FDK_BITSTREAM    hBs,
+        const INT sampleRateIn,
+        const INT sampleRateOut,
+        const INT samplesPerFrame,
+        const AUDIO_OBJECT_TYPE coreCodec,
+        const MP4_ELEMENT_ID    elementID,
+        const INT               elementIndex
+        );
+
+/**
+ * \brief        Set a parameter of the SBR decoder runtime instance.
+ * \param self   SBR decoder handle.
+ * \param param  Parameter which will be set if successfull.
+ * \param value  New parameter value.
+ * \return       Error code.
+ */
+SBR_ERROR sbrDecoder_SetParam ( HANDLE_SBRDECODER   self,
+                                const SBRDEC_PARAM  param,
+                                const INT           value );
+
+/**
+ * \brief  Feed DRC channel data into a SBR decoder runtime instance.
+ *
+ * \param self                    SBR decoder handle.
+ * \param ch                      Channel number to which the DRC data is associated to.
+ * \param numBands                Number of DRC bands.
+ * \param pNextFact_mag           Pointer to a table with the DRC factor magnitudes.
+ * \param nextFact_exp            Exponent for all DRC factors.
+ * \param drcInterpolationScheme  DRC interpolation scheme.
+ * \param winSequence             Window sequence from core coder (eight short or one long window).
+ * \param pBandTop                Pointer to a table with the top borders for all DRC bands.
+ *
+ * \return  Error code.
+ */
+SBR_ERROR sbrDecoder_drcFeedChannel ( HANDLE_SBRDECODER  self,
+                                      INT                ch,
+                                      UINT               numBands,
+                                      FIXP_DBL          *pNextFact_mag,
+                                      INT                nextFact_exp,
+                                      SHORT              drcInterpolationScheme,
+                                      UCHAR              winSequence,
+                                      USHORT            *pBandTop );
+
+/**
+ * \brief  Disable SBR DRC for a certain channel.
+ *
+ * \param hSbrDecoder  SBR decoder handle.
+ * \param ch           Number of the channel that has to be disabled.
+ *
+ * \return  None.
+ */
+void sbrDecoder_drcDisable ( HANDLE_SBRDECODER  self,
+                             INT                ch );
+
+
+/**
+ * \brief  Parse one SBR element data extension data block. The bit stream position will
+ *         be placed at the end of the SBR payload block. The remaining bits will be returned
+ *         into *count if a payload length is given (byPayLen > 0). If no SBR payload length is
+ *         given (bsPayLen < 0) then the bit stream position on return will be random after this
+ *         function call in case of errors, and any further decoding will be completely pointless.
+ *
+ * \param self           SBR decoder handle.
+ * \param hBs            Bit stream handle as data source.
+ * \param count          Pointer to an integer where the amount of parsed SBR payload bits is stored into.
+ * \param bsPayLen       If > 0 this value is the SBR payload length. If < 0, the SBR payload length is unknown.
+ * \param flags          CRC flag (0: EXT_SBR_DATA; 1: EXT_SBR_DATA_CRC)
+ * \param prev_element   Previous MPEG-4 element ID.
+ * \param element_index  Index of the current element.
+ *
+ * \return  Error code.
+ */
+SBR_ERROR sbrDecoder_Parse (
+        HANDLE_SBRDECODER  self,
+        HANDLE_FDK_BITSTREAM hBs,
+        int *count,
+        int  bsPayLen,
+        int  crcFlag,
+        MP4_ELEMENT_ID prev_element,
+        int element_index,
+        int fGlobalIndependencyFlag
+        );
+
+/**
+ * \brief  This function decodes the given SBR bitstreams and applies SBR to the given time data.
+ * 
+ * SBR-processing works InPlace. I.e. the calling function has to provide
+ * a time domain buffer timeData which can hold the completely decoded
+ * result.
+ *
+ * Left and right channel are read and stored according to the
+ * interleaving flag, frame length and number of channels.
+ *
+ * \param self            Handle of an open SBR decoder instance.
+ * \param hSbrBs          SBR Bitstream handle.
+ * \param timeData        Pointer to input and finally upsampled output data.
+ * \param numChannels     Pointer to a buffer holding the number of channels in time data buffer.
+ * \param sampleRate      Output samplerate.
+ * \param channelMapping  Channel mapping indices.
+ * \param interleaved     Flag indicating if time data is stored interleaved (1: Interleaved time data, 0: non-interleaved timedata).
+ * \param coreDecodedOk   Flag indicating if the core decoder did not find any error (0: core decoder found errors, 1: no errors).
+ * \param psDecoded       Pointer to a buffer holding a flag. Input: PS is possible, Output: PS has been rendered.
+ *
+ * \return  Error code.
+ */
+SBR_ERROR sbrDecoder_Apply ( HANDLE_SBRDECODER    self,
+                             INT_PCM             *timeData,
+                             int                 *numChannels,
+                             int                 *sampleRate,
+                             const UCHAR          channelMapping[(6)],
+                             const int            interleaved,
+                             const int            coreDecodedOk,
+                             UCHAR               *psDecoded );
+
+
+/**
+ * \brief       Close SBR decoder instance and free memory.
+ * \param self  SBR decoder handle.
+ * \return      Error Code.
+ */
+SBR_ERROR sbrDecoder_Close ( HANDLE_SBRDECODER *self );
+
+
+/**
+ * \brief       Get SBR decoder library information.
+ * \param info  Pointer to a LIB_INFO struct, where library information is written to.
+ * \return      0 on success, -1 if invalid handle or if no free element is available to write information to.
+ */
+INT sbrDecoder_GetLibInfo( LIB_INFO *info );    
+
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif
diff --git a/libSBRdec/src/arm/env_calc_arm.cpp b/libSBRdec/src/arm/env_calc_arm.cpp
new file mode 100644
index 0000000..4e3a6de
--- /dev/null
+++ b/libSBRdec/src/arm/env_calc_arm.cpp
@@ -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
+----------------------------------------------------------------------------------------------------------- */
+
+/********************************  Fraunhofer IIS  ***************************
+
+   Author(s):   Arthur Tritthart
+   Description: (ARM optimised) SBR domain coding
+
+******************************************************************************/
+#ifndef INCLUSION_GUARD_CALC_ENV_ARM
+#define INCLUSION_GUARD_CALC_ENV_ARM
+
+
+/*!
+  \brief   Compute maximal value of a complex array (re/im) of a given width
+           Negative values are temporarily logically or'ed with 0xFFFFFFFF
+           instead of negating the value, if the sign bit is set.
+  \param   maxVal  Preset maximal value
+  \param   reTmp   real input signal
+  \param   imTmp   imaginary input signal
+  \return  new maximal value
+*/
+
+#ifdef FUNCTION_FDK_get_maxval
+__asm FIXP_DBL FDK_get_maxval (FIXP_DBL maxVal, FIXP_DBL *reTmp, FIXP_DBL *imTmp, int width )
+{
+
+  /* Register map:
+     r0    maxVal
+     r1    reTmp
+     r2    imTmp
+     r3    width
+     r4    real
+     r5    imag
+  */
+    PUSH    {r4-r5}
+
+    MOVS    r3, r3, ASR #1
+    ADC     r3, r3, #0
+    BCS     FDK_get_maxval_loop_2nd_part
+    BEQ     FDK_get_maxval_loop_end
+
+FDK_get_maxval_loop
+    LDR     r4, [r1], #4
+    LDR     r5, [r2], #4
+    EOR     r4, r4, r4, ASR #31
+    EOR     r5, r5, r5, ASR #31
+    ORR     r0, r0, r4
+    ORR     r0, r0, r5
+
+FDK_get_maxval_loop_2nd_part
+    LDR     r4, [r1], #4
+    LDR     r5, [r2], #4
+    EOR     r4, r4, r4, ASR #31
+    EOR     r5, r5, r5, ASR #31
+    ORR     r0, r0, r4
+    ORR     r0, r0, r5
+
+    SUBS    r3, r3, #1
+    BNE     FDK_get_maxval_loop
+
+FDK_get_maxval_loop_end
+    POP     {r4-r5}
+    BX      lr
+}
+#endif /* FUNCTION_FDK_get_maxval */
+
+#endif /* INCLUSION_GUARD_CALC_ENV_ARM */
diff --git a/libSBRdec/src/arm/lpp_tran_arm.cpp b/libSBRdec/src/arm/lpp_tran_arm.cpp
new file mode 100644
index 0000000..541e7c7
--- /dev/null
+++ b/libSBRdec/src/arm/lpp_tran_arm.cpp
@@ -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
+----------------------------------------------------------------------------------------------------------- */
+
+/********************************  Fraunhofer IIS  ***************************
+
+   Author(s):   Arthur Tritthart
+   Description: (ARM optimised) LPP transposer subroutines
+
+******************************************************************************/
+
+
+#if defined(__arm__)
+
+
+#define FUNCTION_LPPTRANSPOSER_func1
+
+#ifdef FUNCTION_LPPTRANSPOSER_func1
+
+/* Note: This code requires only 43 cycles per iteration instead of 61 on ARM926EJ-S */
+__attribute__ ((noinline)) static void lppTransposer_func1(
+  FIXP_DBL *lowBandReal,
+  FIXP_DBL *lowBandImag,
+  FIXP_DBL **qmfBufferReal,
+  FIXP_DBL **qmfBufferImag,
+  int loops,
+  int hiBand,
+  int dynamicScale,
+  int descale,
+  FIXP_SGL a0r,
+  FIXP_SGL a0i,
+  FIXP_SGL a1r,
+  FIXP_SGL a1i)
+{
+
+  FIXP_DBL real1, real2, imag1, imag2, accu1, accu2;
+
+  real2 = lowBandReal[-2];
+  real1 = lowBandReal[-1];
+  imag2 = lowBandImag[-2];
+  imag1 = lowBandImag[-1];
+  for(int i=0; i < loops; i++)
+  {
+    accu1 = fMultDiv2(         a0r,real1);
+    accu2 = fMultDiv2(         a0i,imag1);
+    accu1 = fMultAddDiv2(accu1,a1r,real2);
+    accu2 = fMultAddDiv2(accu2,a1i,imag2);
+    real2 = fMultDiv2(         a1i,real2);
+    accu1 = accu1 - accu2;
+    accu1 = accu1 >> dynamicScale;
+
+    accu2 = fMultAddDiv2(real2,a1r,imag2);
+    real2 = real1;
+    imag2 = imag1;
+    accu2 = fMultAddDiv2(accu2,a0i,real1);
+    real1 = lowBandReal[i];
+    accu2 = fMultAddDiv2(accu2,a0r,imag1);
+    imag1 = lowBandImag[i];
+    accu2 = accu2 >> dynamicScale;
+
+    accu1 <<= 1;
+    accu2 <<= 1;
+
+    qmfBufferReal[i][hiBand] = accu1 + (real1>>descale);
+    qmfBufferImag[i][hiBand] = accu2 + (imag1>>descale);
+  }
+}
+#endif  /* #ifdef FUNCTION_LPPTRANSPOSER_func1 */
+#endif  /* __arm__ */
+
+
+
diff --git a/libSBRdec/src/env_calc.cpp b/libSBRdec/src/env_calc.cpp
new file mode 100644
index 0000000..11df761
--- /dev/null
+++ b/libSBRdec/src/env_calc.cpp
@@ -0,0 +1,2229 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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 calculation  
+
+  The envelope adjustor compares the energies present in the transposed
+  highband to the reference energies conveyed with the bitstream.
+  The highband is amplified (sometimes) or attenuated (mostly) to the
+  desired level.
+
+  The spectral shape of the reference energies can be changed several times per
+  frame if necessary. Each set of energy values corresponding to a certain range
+  in time will be called an <em>envelope</em> here.
+  The bitstream supports several frequency scales and two resolutions. Normally,
+  one or more QMF-subbands are grouped to one SBR-band. An envelope contains
+  reference energies for each SBR-band.
+  In addition to the energy envelopes, noise envelopes are transmitted that
+  define the ratio of energy which is generated by adding noise instead of
+  transposing the lowband. The noise envelopes are given in a coarser time
+  and frequency resolution.
+  If a signal contains strong tonal components, synthetic sines can be
+  generated in individual SBR bands.
+
+  An overlap buffer of 6 QMF-timeslots is used to allow a more
+  flexible alignment of the envelopes in time that is not restricted to the
+  core codec's frame borders.
+  Therefore the envelope adjustor has access to the spectral data of the
+  current frame as well as the last 6 QMF-timeslots of the previous frame.
+  However, in average only the data of 1 frame is being processed as
+  the adjustor is called once per frame.
+
+  Depending on the frequency range set in the bitstream, only QMF-subbands between
+  <em>lowSubband</em> and <em>highSubband</em> are adjusted.
+
+  Scaling of spectral data to maximize SNR (see #QMF_SCALE_FACTOR) as well as a special Mantissa-Exponent format
+  ( see  calculateSbrEnvelope() ) are being used. The main entry point for this modules is calculateSbrEnvelope().
+
+  \sa sbr_scale.h, #QMF_SCALE_FACTOR, calculateSbrEnvelope(), \ref documentationOverview
+*/
+
+
+#include "env_calc.h"
+
+#include "sbrdec_freq_sca.h"
+#include "env_extr.h"
+#include "transcendent.h"
+#include "sbr_ram.h"
+#include "sbr_rom.h"
+
+#include "genericStds.h"           /* need FDKpow() for debug outputs */
+
+#if defined(__arm__)
+#include "arm/env_calc_arm.cpp"
+#endif
+
+typedef struct
+{
+    FIXP_DBL nrgRef[MAX_FREQ_COEFFS];
+    FIXP_DBL nrgEst[MAX_FREQ_COEFFS];
+    FIXP_DBL nrgGain[MAX_FREQ_COEFFS];
+    FIXP_DBL noiseLevel[MAX_FREQ_COEFFS];
+    FIXP_DBL nrgSine[MAX_FREQ_COEFFS];
+
+    SCHAR   nrgRef_e[MAX_FREQ_COEFFS];
+    SCHAR   nrgEst_e[MAX_FREQ_COEFFS];
+    SCHAR   nrgGain_e[MAX_FREQ_COEFFS];
+    SCHAR   noiseLevel_e[MAX_FREQ_COEFFS];
+    SCHAR   nrgSine_e[MAX_FREQ_COEFFS];
+}
+ENV_CALC_NRGS;
+
+/*static*/ void equalizeFiltBufferExp(FIXP_DBL *filtBuffer,
+                                  SCHAR   *filtBuffer_e,
+                                  FIXP_DBL *NrgGain,
+                                  SCHAR   *NrgGain_e,
+                                  int    subbands);
+
+/*static*/ void calcNrgPerSubband(FIXP_DBL  **analysBufferReal,
+                              FIXP_DBL  **analysBufferImag,
+                              int       lowSubband, int highSubband,
+                              int       start_pos,  int next_pos,
+                              SCHAR     frameExp,
+                              FIXP_DBL *nrgEst,
+                              SCHAR    *nrgEst_e );
+
+/*static*/ void calcNrgPerSfb(FIXP_DBL  **analysBufferReal,
+                          FIXP_DBL  **analysBufferImag,
+                          int       nSfb,
+                          UCHAR    *freqBandTable,
+                          int       start_pos,  int next_pos,
+                          SCHAR     input_e,
+                          FIXP_DBL *nrg_est,
+                          SCHAR    *nrg_est_e );
+
+/*static*/ void calcSubbandGain(FIXP_DBL  nrgRef, SCHAR nrgRef_e, ENV_CALC_NRGS* nrgs, int c,
+                            FIXP_DBL  tmpNoise, SCHAR tmpNoise_e,
+                            UCHAR     sinePresentFlag,
+                            UCHAR     sineMapped,
+                            int       noNoiseFlag);
+
+/*static*/ void calcAvgGain(ENV_CALC_NRGS* nrgs,
+                        int        lowSubband,
+                        int        highSubband,
+                        FIXP_DBL  *sumRef_m,
+                        SCHAR     *sumRef_e,
+                        FIXP_DBL  *ptrAvgGain_m,
+                        SCHAR     *ptrAvgGain_e);
+
+/*static*/ void adjustTimeSlotLC(FIXP_DBL  *ptrReal,
+                           ENV_CALC_NRGS* nrgs,
+                           UCHAR *ptrHarmIndex,
+                           int    lowSubbands,
+                           int    noSubbands,
+                           int    scale_change,
+                           int    noNoiseFlag,
+                           int   *ptrPhaseIndex,
+                           int    fCldfb);
+/*static*/ void adjustTimeSlotHQ(FIXP_DBL  *ptrReal,
+                           FIXP_DBL  *ptrImag,
+                           HANDLE_SBR_CALCULATE_ENVELOPE h_sbr_cal_env,
+                           ENV_CALC_NRGS* nrgs,
+                           int    lowSubbands,
+                           int    noSubbands,
+                           int    scale_change,
+                           FIXP_SGL smooth_ratio,
+                           int    noNoiseFlag,
+                           int    filtBufferNoiseShift);
+
+
+/*!
+  \brief     Map sine flags from bitstream to QMF bands
+
+  The bitstream carries only 1 sine flag per band and frame.
+  This function maps every sine flag from the bitstream to a specific QMF subband
+  and to a specific envelope where the sine shall start.
+  The result is stored in the vector sineMapped which contains one entry per
+  QMF subband. The value of an entry specifies the envelope where a sine
+  shall start. A value of #MAX_ENVELOPES indicates that no sine is present
+  in the subband.
+  The missing harmonics flags from the previous frame (harmFlagsPrev) determine
+  if a sine starts at the beginning of the frame or at the transient position.
+  Additionally, the flags in harmFlagsPrev are being updated by this function
+  for the next frame.
+*/
+/*static*/ void mapSineFlags(UCHAR *freqBandTable, /*!< Band borders (there's only 1 flag per band) */
+                         int nSfb,                     /*!< Number of bands in the table */
+                         UCHAR *addHarmonics,           /*!< vector with 1 flag per sfb */
+                         int *harmFlagsPrev,           /*!< Packed 'addHarmonics' */
+                         int tranEnv,                  /*!< Transient position */
+                         SCHAR *sineMapped)            /*!< Resulting vector of sine start positions for each QMF band */
+
+{
+  int i;
+  int lowSubband2 = freqBandTable[0]<<1;
+  int bitcount = 0;
+  int oldflags = *harmFlagsPrev;
+  int newflags = 0;
+
+  /*
+    Format of harmFlagsPrev:
+
+    first word = flags for highest 16 sfb bands in use
+    second word = flags for next lower 16 sfb bands (if present)
+    third word = flags for lowest 16 sfb bands (if present)
+
+    Up to MAX_FREQ_COEFFS sfb bands can be flagged for a sign.
+    The lowest bit of the first word corresponds to the _highest_ sfb band in use.
+    This is ensures that each flag is  mapped to the same QMF band even after a
+    change of the crossover-frequency.
+  */
+
+
+  /* Reset the output vector first */
+  FDKmemset(sineMapped, MAX_ENVELOPES,MAX_FREQ_COEFFS); /* MAX_ENVELOPES means 'no sine' */
+
+  freqBandTable += nSfb;
+  addHarmonics  += nSfb-1;
+
+  for (i=nSfb; i!=0; i--) {
+    int ui = *freqBandTable--;                 /* Upper limit of the current scale factor band. */
+    int li = *freqBandTable;                   /* Lower limit of the current scale factor band. */
+
+    if ( *addHarmonics-- ) {                   /* There is a sine in this band */
+
+      unsigned int mask = 1 << bitcount;
+      newflags |= mask;                        /* Set flag */
+
+      /*
+        If there was a sine in the last frame, let it continue from the first envelope on
+        else start at the transient position.
+      */
+      sineMapped[(ui+li-lowSubband2) >> 1] = ( oldflags & mask ) ? 0 : tranEnv;
+    }
+
+    if ((++bitcount == 16) || i==1) {
+      bitcount = 0;
+      *harmFlagsPrev++ = newflags;
+      oldflags = *harmFlagsPrev;               /* Fetch 16 of the old flags */
+      newflags = 0;
+    }
+  }
+}
+
+
+/*!
+  \brief     Reduce gain-adjustment induced aliasing for real valued filterbank.
+*/
+/*static*/ void
+aliasingReduction(FIXP_DBL* degreeAlias,       /*!< estimated aliasing for each QMF channel */
+                  ENV_CALC_NRGS* nrgs,
+                  int*      useAliasReduction, /*!< synthetic sine engergy for each subband, used as flag */
+                  int       noSubbands)        /*!< number of QMF channels to process */
+{
+  FIXP_DBL* nrgGain   = nrgs->nrgGain;          /*!< subband gains to be modified */
+  SCHAR*    nrgGain_e = nrgs->nrgGain_e;        /*!< subband gains to be modified (exponents) */
+  FIXP_DBL* nrgEst    = nrgs->nrgEst;           /*!< subband energy before amplification */
+  SCHAR*    nrgEst_e  = nrgs->nrgEst_e;         /*!< subband energy before amplification (exponents) */
+  int grouping = 0, index = 0, noGroups, k;
+  int groupVector[MAX_FREQ_COEFFS];
+
+  /* Calculate grouping*/
+  for (k = 0; k < noSubbands-1; k++ ){
+    if ( (degreeAlias[k + 1] != FL2FXCONST_DBL(0.0f)) && useAliasReduction[k] ) {
+      if(grouping==0){
+        groupVector[index++] = k;
+        grouping = 1;
+      }
+      else{
+        if(groupVector[index-1] + 3 == k){
+          groupVector[index++] = k + 1;
+          grouping = 0;
+        }
+      }
+    }
+    else{
+      if(grouping){
+        if(useAliasReduction[k])
+          groupVector[index++] = k + 1;
+        else
+          groupVector[index++] = k;
+        grouping = 0;
+      }
+    }
+  }
+
+  if(grouping){
+    groupVector[index++] = noSubbands;
+  }
+  noGroups = index >> 1;
+
+
+  /*Calculate new gain*/
+  for (int group = 0; group < noGroups; group ++) {
+    FIXP_DBL nrgOrig = FL2FXCONST_DBL(0.0f);    /* Original signal energy in current group of bands */
+    SCHAR    nrgOrig_e = 0;
+    FIXP_DBL nrgAmp = FL2FXCONST_DBL(0.0f);     /* Amplified signal energy in group (using current gains) */
+    SCHAR    nrgAmp_e = 0;
+    FIXP_DBL nrgMod = FL2FXCONST_DBL(0.0f);   /* Signal energy in group when applying modified gains */
+    SCHAR    nrgMod_e = 0;
+    FIXP_DBL groupGain;         /* Total energy gain in group */
+    SCHAR    groupGain_e;
+    FIXP_DBL compensation;      /* Compensation factor for the energy change when applying modified gains */
+    SCHAR    compensation_e;
+
+    int startGroup = groupVector[2*group];
+    int stopGroup  = groupVector[2*group+1];
+
+    /* Calculate total energy in group before and after amplification with current gains: */
+    for(k = startGroup; k < stopGroup; k++){
+      /* Get original band energy */
+      FIXP_DBL tmp = nrgEst[k];
+      SCHAR    tmp_e = nrgEst_e[k];
+
+      FDK_add_MantExp(tmp, tmp_e, nrgOrig, nrgOrig_e, &nrgOrig, &nrgOrig_e);
+
+      /* Multiply band energy with current gain */
+      tmp = fMult(tmp,nrgGain[k]);
+      tmp_e = tmp_e + nrgGain_e[k];
+
+      FDK_add_MantExp(tmp, tmp_e, nrgAmp, nrgAmp_e, &nrgAmp, &nrgAmp_e);
+    }
+
+    /* Calculate total energy gain in group */
+    FDK_divide_MantExp(nrgAmp, nrgAmp_e,
+                       nrgOrig, nrgOrig_e,
+                       &groupGain, &groupGain_e);
+
+    for(k = startGroup; k < stopGroup; k++){
+      FIXP_DBL tmp;
+      SCHAR    tmp_e;
+
+      FIXP_DBL alpha = degreeAlias[k];
+      if (k < noSubbands - 1) {
+        if (degreeAlias[k + 1] > alpha)
+          alpha = degreeAlias[k + 1];
+      }
+
+      /* Modify gain depending on the degree of aliasing */
+      FDK_add_MantExp( fMult(alpha,groupGain), groupGain_e,
+                       fMult(/*FL2FXCONST_DBL(1.0f)*/ (FIXP_DBL)MAXVAL_DBL - alpha,nrgGain[k]), nrgGain_e[k],
+                       &nrgGain[k], &nrgGain_e[k] );
+
+      /* Apply modified gain to original energy */
+      tmp = fMult(nrgGain[k],nrgEst[k]);
+      tmp_e = nrgGain_e[k] + nrgEst_e[k];
+
+      /* Accumulate energy with modified gains applied */
+      FDK_add_MantExp( tmp, tmp_e,
+                       nrgMod, nrgMod_e,
+                       &nrgMod, &nrgMod_e );
+    }
+
+    /* Calculate compensation factor to retain the energy of the amplified signal */
+    FDK_divide_MantExp(nrgAmp, nrgAmp_e,
+                       nrgMod, nrgMod_e,
+                       &compensation, &compensation_e);
+
+    /* Apply compensation factor to all gains of the group */
+    for(k = startGroup; k < stopGroup; k++){
+      nrgGain[k] = fMult(nrgGain[k],compensation);
+      nrgGain_e[k] = nrgGain_e[k] + compensation_e;
+    }
+  }
+}
+
+
+ /* Convert headroom bits to exponent */
+#define SCALE2EXP(s) (15-(s))
+#define EXP2SCALE(e) (15-(e))
+
+/*!
+  \brief  Apply spectral envelope to subband samples
+
+  This function is called from sbr_dec.cpp in each frame.
+
+  To enhance accuracy and due to the usage of tables for squareroots and
+  inverse, some calculations are performed with the operands being split
+  into mantissa and exponent. The variable names in the source code carry
+  the suffixes <em>_m</em> and  <em>_e</em> respectively. The control data
+  in #hFrameData containts envelope data which is represented by this format but
+  stored in single words. (See requantizeEnvelopeData() for details). This data
+  is unpacked within calculateSbrEnvelope() to follow the described suffix convention.
+
+  The actual value (comparable to the corresponding float-variable in the
+  research-implementation) of a mantissa/exponent-pair can be calculated as
+
+  \f$ value = value\_m * 2^{value\_e} \f$
+
+  All energies and noise levels decoded from the bitstream suit for an
+  original signal magnitude of \f$\pm 32768 \f$ rather than \f$ \pm 1\f$. Therefore,
+  the scale factor <em>hb_scale</em> passed into this function will be converted
+  to an 'input exponent' (#input_e), which fits the internal representation.
+
+  Before the actual processing, an exponent #adj_e for resulting adjusted
+  samples is derived from the maximum reference energy.
+
+  Then, for each envelope, the following steps are performed:
+
+  \li Calculate energy in the signal to be adjusted. Depending on the the value of
+      #interpolFreq (interpolation mode), this is either done seperately
+      for each QMF-subband or for each SBR-band.
+      The resulting energies are stored in #nrgEst_m[#MAX_FREQ_COEFFS] (mantissas)
+      and #nrgEst_e[#MAX_FREQ_COEFFS] (exponents).
+  \li Calculate gain and noise level for each subband:<br>
+      \f$ gain  = \sqrt{ \frac{nrgRef}{nrgEst} \cdot (1 - noiseRatio) }
+          \hspace{2cm}
+          noise = \sqrt{ nrgRef \cdot noiseRatio }
+      \f$<br>
+      where <em>noiseRatio</em> and <em>nrgRef</em> are extracted from the
+      bitstream and <em>nrgEst</em> is the subband energy before adjustment.
+      The resulting gains are stored in #nrgGain_m[#MAX_FREQ_COEFFS]
+      (mantissas) and #nrgGain_e[#MAX_FREQ_COEFFS] (exponents), the noise levels
+      are stored in #noiseLevel_m[#MAX_FREQ_COEFFS] and #noiseLevel_e[#MAX_FREQ_COEFFS]
+      (exponents).
+      The sine levels are stored in #nrgSine_m[#MAX_FREQ_COEFFS]
+      and #nrgSine_e[#MAX_FREQ_COEFFS].
+  \li Noise limiting: The gain for each subband is limited both absolutely
+      and relatively compared to the total gain over all subbands.
+  \li Boost gain: Calculate and apply boost factor for each limiter band
+      in order to compensate for the energy loss imposed by the limiting.
+  \li Apply gains and add noise: The gains and noise levels are applied
+      to all timeslots of the current envelope. A short FIR-filter (length 4
+      QMF-timeslots) can be used to smooth the sudden change at the envelope borders.
+      Each complex subband sample of the current timeslot is multiplied by the
+      smoothed gain, then random noise with the calculated level is added.
+
+  \note
+  To reduce the stack size, some of the local arrays could be located within
+  the time output buffer. Of the 512 samples temporarily available there,
+  about half the size is already used by #SBR_FRAME_DATA. A pointer to the
+  remaining free memory could be supplied by an additional argument to calculateSbrEnvelope()
+  in sbr_dec:
+
+  \par
+  \code
+    calculateSbrEnvelope (&hSbrDec->sbrScaleFactor,
+                          &hSbrDec->SbrCalculateEnvelope,
+                          hHeaderData,
+                          hFrameData,
+                          QmfBufferReal,
+                          QmfBufferImag,
+                          timeOutPtr + sizeof(SBR_FRAME_DATA)/sizeof(Float) + 1);
+  \endcode
+
+  \par
+  Within calculateSbrEnvelope(), some pointers could be defined instead of the arrays
+  #nrgRef_m, #nrgRef_e, #nrgEst_m, #nrgEst_e, #noiseLevel_m:
+
+  \par
+  \code
+    fract*        nrgRef_m = timeOutPtr;
+    SCHAR*        nrgRef_e = nrgRef_m + MAX_FREQ_COEFFS;
+    fract*        nrgEst_m = nrgRef_e + MAX_FREQ_COEFFS;
+    SCHAR*        nrgEst_e = nrgEst_m + MAX_FREQ_COEFFS;
+    fract*        noiseLevel_m = nrgEst_e + MAX_FREQ_COEFFS;
+  \endcode
+
+  <br>
+*/
+void
+calculateSbrEnvelope (QMF_SCALE_FACTOR  *sbrScaleFactor,           /*!< Scaling factors */
+                      HANDLE_SBR_CALCULATE_ENVELOPE h_sbr_cal_env, /*!< Handle to struct filled by the create-function */
+                      HANDLE_SBR_HEADER_DATA hHeaderData,          /*!< Static control data */
+                      HANDLE_SBR_FRAME_DATA  hFrameData,           /*!< Control data of current frame */
+                      FIXP_DBL **analysBufferReal,                 /*!< Real part of subband samples to be processed */
+                      FIXP_DBL **analysBufferImag,                 /*!< Imag part of subband samples to be processed */
+                      const int useLP,
+                      FIXP_DBL *degreeAlias,                       /*!< Estimated aliasing for each QMF channel */
+                      const UINT flags,
+                      const int frameErrorFlag
+                      )
+{
+  int c, i, j, envNoise = 0;
+  UCHAR*   borders = hFrameData->frameInfo.borders;
+
+  FIXP_SGL *noiseLevels       = hFrameData->sbrNoiseFloorLevel;
+  HANDLE_FREQ_BAND_DATA hFreq = &hHeaderData->freqBandData;
+
+  int lowSubband  = hFreq->lowSubband;
+  int highSubband = hFreq->highSubband;
+  int noSubbands  = highSubband - lowSubband;
+
+  int    noNoiseBands = hFreq->nNfb;
+  int    no_cols      = hHeaderData->numberTimeSlots * hHeaderData->timeStep;
+  UCHAR  first_start  = borders[0] * hHeaderData->timeStep;
+
+  SCHAR  sineMapped[MAX_FREQ_COEFFS];
+  SCHAR  ov_adj_e = SCALE2EXP(sbrScaleFactor->ov_hb_scale);
+  SCHAR  adj_e = 0;
+  SCHAR  output_e;
+  SCHAR  final_e = 0;
+
+  SCHAR  maxGainLimit_e = (frameErrorFlag) ? MAX_GAIN_CONCEAL_EXP : MAX_GAIN_EXP;
+
+  int useAliasReduction[64];
+  UCHAR smooth_length = 0;
+
+  FIXP_SGL * pIenv = hFrameData->iEnvelope;
+
+  /*
+    Extract sine flags for all QMF bands
+  */
+  mapSineFlags(hFreq->freqBandTable[1],
+               hFreq->nSfb[1],
+               hFrameData->addHarmonics,
+               h_sbr_cal_env->harmFlagsPrev,
+               hFrameData->frameInfo.tranEnv,
+               sineMapped);
+
+
+  /*
+    Scan for maximum in bufferd noise levels.
+    This is needed in case that we had strong noise in the previous frame
+    which is smoothed into the current frame.
+    The resulting exponent is used as start value for the maximum search
+    in reference energies
+  */
+  if (!useLP)
+    adj_e = h_sbr_cal_env->filtBufferNoise_e - getScalefactor(h_sbr_cal_env->filtBufferNoise, noSubbands);
+
+  /*
+    Scan for maximum reference energy to be able
+    to select appropriate values for adj_e and final_e.
+  */
+
+  for (i = 0; i < hFrameData->frameInfo.nEnvelopes; i++) {
+    INT maxSfbNrg_e = -FRACT_BITS+NRG_EXP_OFFSET; /* start value for maximum search */
+
+    /* Fetch frequency resolution for current envelope: */
+    for (j=hFreq->nSfb[hFrameData->frameInfo.freqRes[i]]; j!=0; j--) {
+      maxSfbNrg_e = fixMax(maxSfbNrg_e,(INT)((LONG)(*pIenv++) & MASK_E));
+    }
+    maxSfbNrg_e -= NRG_EXP_OFFSET;
+
+    /* Energy -> magnitude (sqrt halfens exponent) */
+    maxSfbNrg_e = (maxSfbNrg_e+1) >> 1;  /* +1 to go safe (round to next higher int) */
+
+    /* Some safety margin is needed for 2 reasons:
+       - The signal energy is not equally spread over all subband samples in
+         a specific sfb of an envelope (Nrg could be too high by a factor of
+         envWidth * sfbWidth)
+       - Smoothing can smear high gains of the previous envelope into the current
+    */
+    maxSfbNrg_e += 6;
+
+    if (borders[i] < hHeaderData->numberTimeSlots)
+      /* This envelope affects timeslots that belong to the output frame */
+      adj_e = (maxSfbNrg_e > adj_e) ? maxSfbNrg_e : adj_e;
+
+    if (borders[i+1] > hHeaderData->numberTimeSlots)
+      /* This envelope affects timeslots after the output frame */
+      final_e =  (maxSfbNrg_e > final_e) ? maxSfbNrg_e : final_e;
+
+  }
+
+  /*
+    Calculate adjustment factors and apply them for every envelope.
+  */
+  pIenv = hFrameData->iEnvelope;
+
+  for (i = 0; i < hFrameData->frameInfo.nEnvelopes; i++) {
+
+    int k, noNoiseFlag;
+    SCHAR  noise_e, input_e = SCALE2EXP(sbrScaleFactor->hb_scale);
+    C_ALLOC_SCRATCH_START(pNrgs, ENV_CALC_NRGS, 1);
+
+    /*
+      Helper variables.
+    */
+    UCHAR start_pos = hHeaderData->timeStep * borders[i];  /* Start-position in time (subband sample) for current envelope. */
+    UCHAR stop_pos = hHeaderData->timeStep * borders[i+1]; /* Stop-position in time (subband sample) for current envelope. */
+    UCHAR freq_res = hFrameData->frameInfo.freqRes[i];     /* Frequency resolution for current envelope. */
+
+
+    /* Always do fully initialize the temporary energy table. This prevents negative energies and extreme gain factors in
+       cases where the number of limiter bands exceeds the number of subbands. The latter can be caused by undetected bit
+       errors and is tested by some streams from the certification set. */
+    FDKmemclear(pNrgs, sizeof(ENV_CALC_NRGS));
+
+    /* If the start-pos of the current envelope equals the stop pos of the current
+       noise envelope, increase the pointer (i.e. choose the next noise-floor).*/
+    if (borders[i] == hFrameData->frameInfo.bordersNoise[envNoise+1]){
+      noiseLevels += noNoiseBands;   /* The noise floor data is stored in a row [noiseFloor1 noiseFloor2...].*/
+      envNoise++;
+    }
+
+    if(i==hFrameData->frameInfo.tranEnv || i==h_sbr_cal_env->prevTranEnv) /* attack */
+    {
+      noNoiseFlag = 1;
+      if (!useLP)
+        smooth_length = 0;  /* No smoothing on attacks! */
+    }
+    else {
+      noNoiseFlag = 0;
+      if (!useLP)
+        smooth_length = (1 - hHeaderData->bs_data.smoothingLength) << 2;  /* can become either 0 or 4 */
+    }
+
+
+    /*
+      Energy estimation in transposed highband.
+    */
+    if (hHeaderData->bs_data.interpolFreq)
+      calcNrgPerSubband(analysBufferReal,
+                        (useLP) ? NULL : analysBufferImag,
+                        lowSubband, highSubband,
+                        start_pos, stop_pos,
+                        input_e,
+                        pNrgs->nrgEst,
+                        pNrgs->nrgEst_e);
+    else
+      calcNrgPerSfb(analysBufferReal,
+                    (useLP) ? NULL : analysBufferImag,
+                    hFreq->nSfb[freq_res],
+                    hFreq->freqBandTable[freq_res],
+                    start_pos, stop_pos,
+                    input_e,
+                    pNrgs->nrgEst,
+                    pNrgs->nrgEst_e);
+
+    /*
+      Calculate subband gains
+    */
+    {
+      UCHAR * table = hFreq->freqBandTable[freq_res];
+      UCHAR * pUiNoise = &hFreq->freqBandTableNoise[1]; /*! Upper limit of the current noise floor band. */
+
+      FIXP_SGL * pNoiseLevels = noiseLevels;
+
+      FIXP_DBL tmpNoise = FX_SGL2FX_DBL((FIXP_SGL)((LONG)(*pNoiseLevels) & MASK_M));
+      SCHAR    tmpNoise_e = (UCHAR)((LONG)(*pNoiseLevels++) & MASK_E) - NOISE_EXP_OFFSET;
+
+      int cc = 0;
+      c = 0;
+      for (j = 0; j < hFreq->nSfb[freq_res]; j++) {
+
+        FIXP_DBL refNrg   = FX_SGL2FX_DBL((FIXP_SGL)((LONG)(*pIenv) & MASK_M));
+        SCHAR    refNrg_e = (SCHAR)((LONG)(*pIenv) & MASK_E) - NRG_EXP_OFFSET;
+
+        UCHAR sinePresentFlag = 0;
+        int li = table[j];
+        int ui = table[j+1];
+
+        for (k=li; k<ui; k++) {
+          sinePresentFlag |= (i >= sineMapped[cc]);
+          cc++;
+        }
+
+        for (k=li; k<ui; k++) {
+          if (k >= *pUiNoise) {
+            tmpNoise = FX_SGL2FX_DBL((FIXP_SGL)((LONG)(*pNoiseLevels) & MASK_M));
+            tmpNoise_e = (SCHAR)((LONG)(*pNoiseLevels++) & MASK_E) - NOISE_EXP_OFFSET;
+
+            pUiNoise++;
+          }
+
+          FDK_ASSERT(k >= lowSubband);
+
+          if (useLP)
+            useAliasReduction[k-lowSubband] = !sinePresentFlag;
+
+          pNrgs->nrgSine[c] = FL2FXCONST_DBL(0.0f);
+          pNrgs->nrgSine_e[c] = 0;
+
+          calcSubbandGain(refNrg, refNrg_e, pNrgs, c,
+                          tmpNoise, tmpNoise_e,
+                          sinePresentFlag, i >= sineMapped[c],
+                          noNoiseFlag);
+
+          pNrgs->nrgRef[c]   = refNrg;
+          pNrgs->nrgRef_e[c] = refNrg_e;
+
+          c++;
+        }
+        pIenv++;
+      }
+    }
+
+    /*
+      Noise limiting
+    */
+
+    for (c = 0; c < hFreq->noLimiterBands; c++) {
+
+      FIXP_DBL sumRef, boostGain, maxGain;
+      FIXP_DBL accu = FL2FXCONST_DBL(0.0f);
+      SCHAR   sumRef_e, boostGain_e, maxGain_e, accu_e = 0;
+
+      calcAvgGain(pNrgs,
+                  hFreq->limiterBandTable[c], hFreq->limiterBandTable[c+1],
+                  &sumRef, &sumRef_e,
+                  &maxGain, &maxGain_e);
+
+      /* Multiply maxGain with limiterGain: */
+      maxGain = fMult(maxGain, FDK_sbrDecoder_sbr_limGains_m[hHeaderData->bs_data.limiterGains]);
+      maxGain_e += FDK_sbrDecoder_sbr_limGains_e[hHeaderData->bs_data.limiterGains];
+
+      /* Scale mantissa of MaxGain into range between 0.5 and 1: */
+      if (maxGain == FL2FXCONST_DBL(0.0f))
+        maxGain_e = -FRACT_BITS;
+      else {
+        SCHAR charTemp = CountLeadingBits(maxGain);
+        maxGain_e -= charTemp;
+        maxGain  <<= (int)charTemp;
+      }
+
+      if (maxGain_e >= maxGainLimit_e) { /* upper limit (e.g. 96 dB) */
+        maxGain = FL2FXCONST_DBL(0.5f);
+        maxGain_e = maxGainLimit_e;
+      }
+
+
+      /* Every subband gain is compared to the scaled "average gain"
+         and limited if necessary: */
+      for (k = hFreq->limiterBandTable[c]; k < hFreq->limiterBandTable[c+1]; k++) {
+        if ( (pNrgs->nrgGain_e[k] > maxGain_e) || (pNrgs->nrgGain_e[k] == maxGain_e && pNrgs->nrgGain[k]>maxGain) ) {
+
+          FIXP_DBL noiseAmp;
+          SCHAR    noiseAmp_e;
+
+          FDK_divide_MantExp(maxGain, maxGain_e, pNrgs->nrgGain[k], pNrgs->nrgGain_e[k], &noiseAmp, &noiseAmp_e);
+          pNrgs->noiseLevel[k]    = fMult(pNrgs->noiseLevel[k],noiseAmp);
+          pNrgs->noiseLevel_e[k] += noiseAmp_e;
+          pNrgs->nrgGain[k]       = maxGain;
+          pNrgs->nrgGain_e[k]     = maxGain_e;
+        }
+      }
+
+      /* -- Boost gain
+        Calculate and apply boost factor for each limiter band:
+        1. Check how much energy would be present when using the limited gain
+        2. Calculate boost factor by comparison with reference energy
+        3. Apply boost factor to compensate for the energy loss due to limiting
+      */
+      for (k = hFreq->limiterBandTable[c]; k < hFreq->limiterBandTable[c + 1]; k++) {
+
+        /* 1.a  Add energy of adjusted signal (using preliminary gain) */
+        FIXP_DBL  tmp   = fMult(pNrgs->nrgGain[k],pNrgs->nrgEst[k]);
+        SCHAR     tmp_e = pNrgs->nrgGain_e[k] + pNrgs->nrgEst_e[k];
+        FDK_add_MantExp(tmp, tmp_e, accu, accu_e, &accu, &accu_e);
+
+        /* 1.b  Add sine energy (if present) */
+        if(pNrgs->nrgSine[k] != FL2FXCONST_DBL(0.0f)) {
+          FDK_add_MantExp(pNrgs->nrgSine[k], pNrgs->nrgSine_e[k], accu, accu_e, &accu, &accu_e);
+        }
+        else {
+          /* 1.c  Add noise energy (if present) */
+          if(noNoiseFlag == 0) {
+            FDK_add_MantExp(pNrgs->noiseLevel[k], pNrgs->noiseLevel_e[k], accu, accu_e, &accu, &accu_e);
+          }
+        }
+      }
+
+      /* 2.a  Calculate ratio of wanted energy and accumulated energy */
+      if (accu == (FIXP_DBL)0) { /* If divisor is 0, limit quotient to +4 dB */
+        boostGain = FL2FXCONST_DBL(0.6279716f);
+        boostGain_e = 2;
+      } else {
+        INT div_e;
+        boostGain = fDivNorm(sumRef, accu, &div_e);
+        boostGain_e = sumRef_e - accu_e + div_e;
+      }
+
+
+      /* 2.b Result too high? --> Limit the boost factor to +4 dB */
+      if((boostGain_e  > 3) ||
+         (boostGain_e == 2 && boostGain > FL2FXCONST_DBL(0.6279716f)) ||
+         (boostGain_e == 3 && boostGain > FL2FXCONST_DBL(0.3139858f)) )
+      {
+        boostGain = FL2FXCONST_DBL(0.6279716f);
+        boostGain_e = 2;
+      }
+      /* 3.  Multiply all signal components with the boost factor */
+      for (k = hFreq->limiterBandTable[c]; k < hFreq->limiterBandTable[c + 1]; k++) {
+        pNrgs->nrgGain[k]   = fMultDiv2(pNrgs->nrgGain[k],boostGain);
+        pNrgs->nrgGain_e[k] = pNrgs->nrgGain_e[k] + boostGain_e + 1;
+
+        pNrgs->nrgSine[k]   = fMultDiv2(pNrgs->nrgSine[k],boostGain);
+        pNrgs->nrgSine_e[k] = pNrgs->nrgSine_e[k] + boostGain_e + 1;
+
+        pNrgs->noiseLevel[k]   = fMultDiv2(pNrgs->noiseLevel[k],boostGain);
+        pNrgs->noiseLevel_e[k] = pNrgs->noiseLevel_e[k] + boostGain_e + 1;
+      }
+    }
+    /* End of noise limiting */
+
+    if (useLP)
+      aliasingReduction(degreeAlias+lowSubband,
+                        pNrgs,
+                        useAliasReduction,
+                        noSubbands);
+
+    /* For the timeslots within the range for the output frame,
+       use the same scale for the noise levels.
+       Drawback: If the envelope exceeds the frame border, the noise levels
+                 will have to be rescaled later to fit final_e of
+                 the gain-values.
+    */
+    noise_e = (start_pos < no_cols) ? adj_e : final_e;
+
+    /*
+      Convert energies to amplitude levels
+    */
+    for (k=0; k<noSubbands; k++) {
+      FDK_sqrt_MantExp(&pNrgs->nrgSine[k],    &pNrgs->nrgSine_e[k],    &noise_e);
+      FDK_sqrt_MantExp(&pNrgs->nrgGain[k],    &pNrgs->nrgGain_e[k],    &pNrgs->nrgGain_e[k]);
+      FDK_sqrt_MantExp(&pNrgs->noiseLevel[k], &pNrgs->noiseLevel_e[k], &noise_e);
+    }
+
+
+
+    /*
+      Apply calculated gains and adaptive noise
+    */
+
+    /* assembleHfSignals() */
+    {
+      int scale_change, sc_change;
+      FIXP_SGL smooth_ratio;
+      int filtBufferNoiseShift=0;
+
+      /* Initialize smoothing buffers with the first valid values */
+      if (h_sbr_cal_env->startUp)
+      {
+        if (!useLP) {
+          h_sbr_cal_env->filtBufferNoise_e = noise_e;
+
+          FDKmemcpy(h_sbr_cal_env->filtBuffer_e,    pNrgs->nrgGain_e,  noSubbands*sizeof(SCHAR));
+          FDKmemcpy(h_sbr_cal_env->filtBufferNoise, pNrgs->noiseLevel, noSubbands*sizeof(FIXP_DBL));
+          FDKmemcpy(h_sbr_cal_env->filtBuffer,      pNrgs->nrgGain,    noSubbands*sizeof(FIXP_DBL));
+
+        }
+        h_sbr_cal_env->startUp = 0;
+      }
+
+      if (!useLP) {
+
+        equalizeFiltBufferExp(h_sbr_cal_env->filtBuffer,    /* buffered */
+                              h_sbr_cal_env->filtBuffer_e,  /* buffered */
+                              pNrgs->nrgGain,               /* current  */
+                              pNrgs->nrgGain_e,             /* current  */
+                              noSubbands);
+
+        /* Adapt exponent of buffered noise levels to the current exponent
+           so they can easily be smoothed */
+        if((h_sbr_cal_env->filtBufferNoise_e - noise_e)>=0) {
+          int shift = fixMin(DFRACT_BITS-1,(int)(h_sbr_cal_env->filtBufferNoise_e - noise_e));
+          for (k=0; k<noSubbands; k++)
+            h_sbr_cal_env->filtBufferNoise[k] <<= shift;
+        }
+        else {
+          int shift = fixMin(DFRACT_BITS-1,-(int)(h_sbr_cal_env->filtBufferNoise_e - noise_e));
+          for (k=0; k<noSubbands; k++)
+            h_sbr_cal_env->filtBufferNoise[k] >>= shift;
+        }
+
+        h_sbr_cal_env->filtBufferNoise_e = noise_e;
+      }
+
+      /* find best scaling! */
+      scale_change = -(DFRACT_BITS-1);
+      for(k=0;k<noSubbands;k++) {
+          scale_change = fixMax(scale_change,(int)pNrgs->nrgGain_e[k]);
+      }
+      sc_change = (start_pos<no_cols)? adj_e - input_e : final_e - input_e;
+
+      if ((scale_change-sc_change+1)<0)
+          scale_change-=(scale_change-sc_change+1);
+
+      scale_change = (scale_change-sc_change)+1;
+
+      for(k=0;k<noSubbands;k++) {
+          int sc = scale_change-pNrgs->nrgGain_e[k] + (sc_change-1);
+          pNrgs->nrgGain[k]  >>= sc;
+          pNrgs->nrgGain_e[k] += sc;
+      }
+
+      if (!useLP) {
+        for(k=0;k<noSubbands;k++) {
+          int sc = scale_change-h_sbr_cal_env->filtBuffer_e[k] + (sc_change-1);
+          h_sbr_cal_env->filtBuffer[k] >>= sc;
+        }
+      }
+
+      for (j = start_pos; j < stop_pos; j++)
+      {
+        /* This timeslot is located within the first part of the processing buffer
+           and will be fed into the QMF-synthesis for the current frame.
+               adj_e - input_e
+           This timeslot will not yet be fed into the QMF so we do not care
+           about the adj_e.
+               sc_change = final_e - input_e
+        */
+        if ( (j==no_cols) && (start_pos<no_cols) )
+        {
+          int shift = (int) (noise_e - final_e);
+          if (!useLP)
+            filtBufferNoiseShift = shift;               /* shifting of h_sbr_cal_env->filtBufferNoise[k] will be applied in function adjustTimeSlotHQ() */
+          if (shift>=0) {
+            shift = fixMin(DFRACT_BITS-1,shift);
+            for (k=0; k<noSubbands; k++) {
+              pNrgs->nrgSine[k] <<= shift;
+              pNrgs->noiseLevel[k]  <<= shift;
+              /*
+              if (!useLP)
+                h_sbr_cal_env->filtBufferNoise[k]  <<= shift;
+              */
+            }
+          }
+          else {
+            shift = fixMin(DFRACT_BITS-1,-shift);
+            for (k=0; k<noSubbands; k++) {
+              pNrgs->nrgSine[k] >>= shift;
+              pNrgs->noiseLevel[k]  >>= shift;
+              /*
+              if (!useLP)
+                h_sbr_cal_env->filtBufferNoise[k]  >>= shift;
+              */
+            }
+          }
+
+          /* update noise scaling */
+          noise_e = final_e;
+          if (!useLP)
+            h_sbr_cal_env->filtBufferNoise_e = noise_e;  /* scaling value unused! */
+
+          /* update gain buffer*/
+          sc_change -= (final_e - input_e);
+
+          if (sc_change<0) {
+            for(k=0;k<noSubbands;k++) {
+                pNrgs->nrgGain[k]  >>= -sc_change;
+                pNrgs->nrgGain_e[k] += -sc_change;
+            }
+            if (!useLP) {
+              for(k=0;k<noSubbands;k++) {
+                    h_sbr_cal_env->filtBuffer[k] >>= -sc_change;
+              }
+            }
+          } else {
+            scale_change+=sc_change;
+          }
+
+        } // if
+
+        if (!useLP) {
+
+          /* Prevent the smoothing filter from running on constant levels */
+          if (j-start_pos < smooth_length)
+            smooth_ratio = FDK_sbrDecoder_sbr_smoothFilter[j-start_pos];
+
+          else
+            smooth_ratio = FL2FXCONST_SGL(0.0f);
+
+          adjustTimeSlotHQ(&analysBufferReal[j][lowSubband],
+                           &analysBufferImag[j][lowSubband],
+                           h_sbr_cal_env,
+                           pNrgs,
+                           lowSubband,
+                           noSubbands,
+                           scale_change,
+                           smooth_ratio,
+                           noNoiseFlag,
+                           filtBufferNoiseShift);
+        }
+        else
+        {
+          adjustTimeSlotLC(&analysBufferReal[j][lowSubband],
+                           pNrgs,
+                          &h_sbr_cal_env->harmIndex,
+                           lowSubband,
+                           noSubbands,
+                           scale_change,
+                           noNoiseFlag,
+                          &h_sbr_cal_env->phaseIndex,
+                           (flags & SBRDEC_ELD_GRID));
+        }
+      } // for
+
+      if (!useLP) {
+        /* Update time-smoothing-buffers for gains and noise levels
+           The gains and the noise values of the current envelope are copied into the buffer.
+           This has to be done at the end of each envelope as the values are required for
+           a smooth transition to the next envelope. */
+        FDKmemcpy(h_sbr_cal_env->filtBuffer,      pNrgs->nrgGain,    noSubbands*sizeof(FIXP_DBL));
+        FDKmemcpy(h_sbr_cal_env->filtBuffer_e,    pNrgs->nrgGain_e,  noSubbands*sizeof(SCHAR));
+        FDKmemcpy(h_sbr_cal_env->filtBufferNoise, pNrgs->noiseLevel, noSubbands*sizeof(FIXP_DBL));
+      }
+
+    }
+    C_ALLOC_SCRATCH_END(pNrgs, ENV_CALC_NRGS, 1);
+  }
+
+  /* Rescale output samples */
+  {
+    FIXP_DBL maxVal;
+    int ov_reserve, reserve;
+
+    /* Determine headroom in old adjusted samples */
+    maxVal = maxSubbandSample( analysBufferReal,
+                              (useLP) ? NULL : analysBufferImag,
+                               lowSubband,
+                               highSubband,
+                               0,
+                               first_start);
+
+    ov_reserve = fNorm(maxVal);
+
+    /* Determine headroom in new adjusted samples */
+    maxVal = maxSubbandSample( analysBufferReal,
+                               (useLP) ? NULL : analysBufferImag,
+                               lowSubband,
+                               highSubband,
+                               first_start,
+                               no_cols);
+
+    reserve = fNorm(maxVal);
+
+    /* Determine common output exponent */
+    if (ov_adj_e - ov_reserve  >  adj_e - reserve ) /* set output_e to the maximum */
+      output_e = ov_adj_e - ov_reserve;
+    else
+      output_e = adj_e - reserve;
+
+    /* Rescale old samples */
+    rescaleSubbandSamples( analysBufferReal,
+                           (useLP) ? NULL : analysBufferImag,
+                           lowSubband, highSubband,
+                           0, first_start,
+                           ov_adj_e - output_e);
+
+    /* Rescale new samples */
+    rescaleSubbandSamples( analysBufferReal,
+                           (useLP) ? NULL : analysBufferImag,
+                           lowSubband, highSubband,
+                           first_start, no_cols,
+                           adj_e - output_e);
+  }
+
+  /* Update hb_scale */
+  sbrScaleFactor->hb_scale = EXP2SCALE(output_e);
+
+  /* Save the current final exponent for the next frame: */
+  sbrScaleFactor->ov_hb_scale = EXP2SCALE(final_e);
+
+
+  /* We need to remeber to the next frame that the transient
+     will occur in the first envelope (if tranEnv == nEnvelopes). */
+  if(hFrameData->frameInfo.tranEnv == hFrameData->frameInfo.nEnvelopes)
+    h_sbr_cal_env->prevTranEnv = 0;
+  else
+    h_sbr_cal_env->prevTranEnv = -1;
+
+}
+
+
+/*!
+  \brief   Create envelope instance
+
+  Must be called once for each channel before calculateSbrEnvelope() can be used.
+
+  \return  errorCode, 0 if successful
+*/
+SBR_ERROR
+createSbrEnvelopeCalc (HANDLE_SBR_CALCULATE_ENVELOPE hs,   /*!< pointer to envelope instance */
+                       HANDLE_SBR_HEADER_DATA hHeaderData, /*!< static SBR control data, initialized with defaults */
+                       const int chan,                     /*!< Channel for which to assign buffers */
+                       const UINT flags)
+{
+  SBR_ERROR err = SBRDEC_OK;
+  int i;
+
+  /* Clear previous missing harmonics flags */
+  for (i=0; i<(MAX_FREQ_COEFFS+15)>>4; i++) {
+    hs->harmFlagsPrev[i] = 0;
+  }
+  hs->harmIndex = 0;
+
+  /*
+    Setup pointers for time smoothing.
+    The buffer itself will be initialized later triggered by the startUp-flag.
+  */
+  hs->prevTranEnv = -1;
+
+
+  /* initialization */
+  resetSbrEnvelopeCalc(hs);
+
+  if (chan==0) { /* do this only once */
+    err = resetFreqBandTables(hHeaderData, flags);
+  }
+
+  return err;
+}
+
+/*!
+  \brief   Create envelope instance
+
+  Must be called once for each channel before calculateSbrEnvelope() can be used.
+
+  \return  errorCode, 0 if successful
+*/
+int
+deleteSbrEnvelopeCalc (HANDLE_SBR_CALCULATE_ENVELOPE hs)
+{
+  return 0;
+}
+
+
+/*!
+  \brief   Reset envelope instance
+
+  This function must be called for each channel on a change of configuration.
+  Note that resetFreqBandTables should also be called in this case.
+
+  \return  errorCode, 0 if successful
+*/
+void
+resetSbrEnvelopeCalc (HANDLE_SBR_CALCULATE_ENVELOPE hCalEnv) /*!< pointer to envelope instance */
+{
+  hCalEnv->phaseIndex = 0;
+
+  /* Noise exponent needs to be reset because the output exponent for the next frame depends on it */
+  hCalEnv->filtBufferNoise_e = 0;
+
+  hCalEnv->startUp = 1;
+}
+
+
+/*!
+  \brief  Equalize exponents of the buffered gain values and the new ones
+
+  After equalization of exponents, the FIR-filter addition for smoothing
+  can be performed.
+  This function is called once for each envelope before adjusting.
+*/
+/*static*/ void equalizeFiltBufferExp(FIXP_DBL *filtBuffer,     /*!< bufferd gains */
+                                  SCHAR    *filtBuffer_e,   /*!< exponents of bufferd gains */
+                                  FIXP_DBL *nrgGain,        /*!< gains for current envelope */
+                                  SCHAR    *nrgGain_e,      /*!< exponents of gains for current envelope */
+                                  int       subbands)       /*!< Number of QMF subbands */
+{
+  int   band;
+  int  diff;
+
+  for (band=0; band<subbands; band++){
+    diff = (int) (nrgGain_e[band] - filtBuffer_e[band]);
+    if (diff>0) {
+      filtBuffer[band] >>= diff;   /* Compensate for the scale change by shifting the mantissa. */
+      filtBuffer_e[band] += diff;  /* New gain is bigger, use its exponent */
+    }
+    else if (diff<0) {
+      /* The buffered gains seem to be larger, but maybe there
+         are some unused bits left in the mantissa */
+
+      int reserve = CntLeadingZeros(fixp_abs(filtBuffer[band]))-1;
+
+      if ((-diff) <= reserve) {
+        /* There is enough space in the buffered mantissa so
+           that we can take the new exponent as common.
+        */
+        filtBuffer[band] <<= (-diff);
+        filtBuffer_e[band] += diff;  /* becomes equal to *ptrNewExp */
+      }
+      else {
+        filtBuffer[band] <<= reserve;   /* Shift the mantissa as far as possible: */
+        filtBuffer_e[band] -= reserve;  /* Compensate in the exponent: */
+
+        /* For the remaining difference, change the new gain value */
+        diff = fixMin(-(reserve + diff),DFRACT_BITS-1);
+        nrgGain[band] >>= diff;
+        nrgGain_e[band] += diff;
+      }
+    }
+  }
+}
+
+/*!
+  \brief  Shift left the mantissas of all subband samples
+          in the giventime and frequency range by the specified number of bits.
+
+  This function is used to rescale the audio data in the overlap buffer
+  which has already been envelope adjusted with the last frame.
+*/
+void rescaleSubbandSamples(FIXP_DBL ** re,   /*!< Real part of input and output subband samples */
+                           FIXP_DBL ** im,   /*!< Imaginary part of input and output subband samples */
+                           int lowSubband,   /*!< Begin of frequency range to process */
+                           int highSubband,  /*!< End of frequency range to process */
+                           int start_pos,    /*!< Begin of time rage (QMF-timeslot) */
+                           int next_pos,     /*!< End of time rage (QMF-timeslot) */
+                           int shift)        /*!< number of bits to shift */
+{
+  int width = highSubband-lowSubband;
+
+  if ( (width > 0) && (shift!=0) ) {
+    if (im!=NULL) {
+      for (int l=start_pos; l<next_pos; l++) {
+          scaleValues(&re[l][lowSubband], width, shift);
+          scaleValues(&im[l][lowSubband], width, shift);
+      }
+    } else
+    {
+      for (int l=start_pos; l<next_pos; l++) {
+          scaleValues(&re[l][lowSubband], width, shift);
+      }
+    }
+  }
+}
+
+
+/*!
+  \brief   Determine headroom for shifting
+
+  Determine by how much the spectrum can be shifted left
+  for better accuracy in later processing.
+
+  \return  Number of free bits in the biggest spectral value
+*/
+
+FIXP_DBL maxSubbandSample( FIXP_DBL ** re,   /*!< Real part of input and output subband samples */
+                           FIXP_DBL ** im,   /*!< Real part of input and output subband samples */
+                           int lowSubband,   /*!< Begin of frequency range to process */
+                           int highSubband,  /*!< Number of QMF bands to process */
+                           int start_pos,    /*!< Begin of time rage (QMF-timeslot) */
+                           int next_pos      /*!< End of time rage (QMF-timeslot) */
+                          )
+{
+  FIXP_DBL maxVal = FL2FX_DBL(0.0f);
+  unsigned int width = highSubband - lowSubband;
+
+  FDK_ASSERT(width <= (64));
+
+  if ( width > 0 ) {
+    if (im!=NULL) 
+    {
+      for (int l=start_pos; l<next_pos; l++) 
+      {
+#ifdef FUNCTION_FDK_get_maxval
+        maxVal = FDK_get_maxval(maxVal, &re[l][lowSubband], &im[l][lowSubband], width);
+#else
+        int k=width;
+        FIXP_DBL *reTmp = &re[l][lowSubband];
+        FIXP_DBL *imTmp = &im[l][lowSubband];
+        do{
+          FIXP_DBL tmp1 = *(reTmp++);
+          FIXP_DBL tmp2 = *(imTmp++);
+          maxVal |= (FIXP_DBL)((LONG)(tmp1)^((LONG)tmp1>>(DFRACT_BITS-1)));
+          maxVal |= (FIXP_DBL)((LONG)(tmp2)^((LONG)tmp2>>(DFRACT_BITS-1)));
+        } while(--k!=0);
+#endif
+      }
+    } else
+    {
+      for (int l=start_pos; l<next_pos; l++) {
+        int k=width;
+        FIXP_DBL *reTmp = &re[l][lowSubband];
+        do{
+          FIXP_DBL tmp = *(reTmp++);
+          maxVal |= (FIXP_DBL)((LONG)(tmp)^((LONG)tmp>>(DFRACT_BITS-1)));
+        }while(--k!=0);
+      }
+    }
+  }
+
+  return(maxVal);
+}
+
+#define SHIFT_BEFORE_SQUARE (3) /* (7/2) */
+/*!<
+  If the accumulator does not provide enough overflow bits or
+  does not provide a high dynamic range, the below energy calculation
+  requires an additional shift operation for each sample.
+  On the other hand, doing the shift allows using a single-precision
+  multiplication for the square (at least 16bit x 16bit).
+  For even values of OVRFLW_BITS (0, 2, 4, 6), saturated arithmetic
+  is required for the energy accumulation.
+  Theoretically, the sample-squares can sum up to a value of 76,
+  requiring 7 overflow bits. However since such situations are *very*
+  rare, accu can be limited to 64.
+  In case native saturated arithmetic is not available, overflows
+  can be prevented by replacing the above #define by
+    #define SHIFT_BEFORE_SQUARE ((8 - OVRFLW_BITS) / 2)
+  which will result in slightly reduced accuracy.
+*/
+
+/*!
+  \brief  Estimates the mean energy of each filter-bank channel for the
+          duration of the current envelope
+
+  This function is used when interpolFreq is true.
+*/
+/*static*/ void calcNrgPerSubband(FIXP_DBL  **analysBufferReal, /*!< Real part of subband samples */
+                              FIXP_DBL  **analysBufferImag, /*!< Imaginary part of subband samples */
+                              int       lowSubband,           /*!< Begin of the SBR frequency range */
+                              int       highSubband,          /*!< High end of the SBR frequency range */
+                              int       start_pos,            /*!< First QMF-slot of current envelope */
+                              int       next_pos,             /*!< Last QMF-slot of current envelope + 1 */
+                              SCHAR     frameExp,             /*!< Common exponent for all input samples */
+                              FIXP_DBL *nrgEst,               /*!< resulting Energy (0..1) */
+                              SCHAR    *nrgEst_e )            /*!< Exponent of resulting Energy */
+{
+  FIXP_SGL invWidth;
+  SCHAR  preShift;
+  SCHAR  shift;
+  FIXP_DBL sum;
+  int k,l;
+
+  /* Divide by width of envelope later: */
+  invWidth = FX_DBL2FX_SGL(GetInvInt(next_pos - start_pos));
+  /* The common exponent needs to be doubled because all mantissas are squared: */
+  frameExp = frameExp << 1;
+
+  for (k=lowSubband; k<highSubband; k++) {
+    FIXP_DBL  bufferReal[(((1024)/(32))+(6))];
+    FIXP_DBL  bufferImag[(((1024)/(32))+(6))];
+    FIXP_DBL maxVal = FL2FX_DBL(0.0f);
+
+    if (analysBufferImag!=NULL)
+    {
+      for (l=start_pos;l<next_pos;l++) 
+      {
+        bufferImag[l] = analysBufferImag[l][k];
+        maxVal |= (FIXP_DBL)((LONG)(bufferImag[l])^((LONG)bufferImag[l]>>(DFRACT_BITS-1)));
+        bufferReal[l] = analysBufferReal[l][k];
+        maxVal |= (FIXP_DBL)((LONG)(bufferReal[l])^((LONG)bufferReal[l]>>(DFRACT_BITS-1)));
+      }
+    }
+    else
+    {
+      for (l=start_pos;l<next_pos;l++)
+      {
+        bufferReal[l] = analysBufferReal[l][k];
+        maxVal |= (FIXP_DBL)((LONG)(bufferReal[l])^((LONG)bufferReal[l]>>(DFRACT_BITS-1)));
+      }
+    }
+
+    if (maxVal!=FL2FXCONST_DBL(0.f)) {
+
+
+      /* If the accu does not provide enough overflow bits, we cannot
+         shift the samples up to the limit.
+         Instead, keep up to 3 free bits in each sample, i.e. up to
+         6 bits after calculation of square.
+         Please note the comment on saturated arithmetic above!
+      */
+      FIXP_DBL accu = FL2FXCONST_DBL(0.0f);
+      preShift = CntLeadingZeros(maxVal)-1;
+      preShift -= SHIFT_BEFORE_SQUARE;
+
+      if (preShift>=0) {
+        if (analysBufferImag!=NULL) {
+          for (l=start_pos; l<next_pos; l++) {
+            FIXP_DBL temp1 = bufferReal[l] << (int)preShift;
+            FIXP_DBL temp2 = bufferImag[l] << (int)preShift;
+            accu = fPow2AddDiv2(accu, temp1);
+            accu = fPow2AddDiv2(accu, temp2);
+          }
+        } else
+        {
+          for (l=start_pos; l<next_pos; l++) {
+            FIXP_DBL temp = bufferReal[l] << (int)preShift;
+            accu = fPow2AddDiv2(accu, temp);
+          }
+        }
+      }
+      else {    /* if negative shift value */
+        int negpreShift = -preShift;
+        if (analysBufferImag!=NULL) {
+          for (l=start_pos; l<next_pos; l++) {
+            FIXP_DBL temp1 = bufferReal[l] >> (int)negpreShift;
+            FIXP_DBL temp2 = bufferImag[l] >> (int)negpreShift;
+            accu = fPow2AddDiv2(accu, temp1);
+            accu = fPow2AddDiv2(accu, temp2);
+          }
+        } else
+        {
+          for (l=start_pos; l<next_pos; l++) {
+            FIXP_DBL temp = bufferReal[l] >> (int)negpreShift;
+            accu = fPow2AddDiv2(accu, temp);
+          }
+        }
+      }
+      accu <<= 1;
+
+      /* Convert double precision to Mantissa/Exponent: */
+      shift = fNorm(accu);
+      sum = accu << (int)shift;
+
+      /* Divide by width of envelope and apply frame scale: */
+      *nrgEst++ = fMult(sum, invWidth);
+      shift += 2 * preShift;
+      if (analysBufferImag!=NULL)
+        *nrgEst_e++ = frameExp - shift;
+      else
+        *nrgEst_e++ = frameExp - shift + 1;  /* +1 due to missing imag. part */
+    } /* maxVal!=0 */
+    else {
+
+      /* Prevent a zero-mantissa-number from being misinterpreted
+         due to its exponent. */
+      *nrgEst++ = FL2FXCONST_DBL(0.0f);
+      *nrgEst_e++ = 0;
+    }
+  }
+}
+
+/*!
+  \brief   Estimates the mean energy of each Scale factor band for the
+           duration of the current envelope.
+
+  This function is used when interpolFreq is false.
+*/
+/*static*/ void calcNrgPerSfb(FIXP_DBL  **analysBufferReal,  /*!< Real part of subband samples */
+                          FIXP_DBL  **analysBufferImag,  /*!< Imaginary part of subband samples */
+                          int       nSfb,                /*!< Number of scale factor bands */
+                          UCHAR    *freqBandTable,       /*!< First Subband for each Sfb */
+                          int       start_pos,           /*!< First QMF-slot of current envelope */
+                          int       next_pos,            /*!< Last QMF-slot of current envelope + 1 */
+                          SCHAR     input_e,             /*!< Common exponent for all input samples */
+                          FIXP_DBL *nrgEst,              /*!< resulting Energy (0..1) */
+                          SCHAR    *nrgEst_e )           /*!< Exponent of resulting Energy */
+{
+  FIXP_SGL  invWidth;
+  FIXP_DBL  temp;
+  SCHAR  preShift;
+  SCHAR   shift, sum_e;
+  FIXP_DBL  sum;
+
+  int j,k,l,li,ui;
+  FIXP_DBL sumAll, sumLine; /* Single precision would be sufficient,
+                             but overflow bits are required for accumulation */
+
+  /* Divide by width of envelope later: */
+  invWidth = FX_DBL2FX_SGL(GetInvInt(next_pos - start_pos));
+  /* The common exponent needs to be doubled because all mantissas are squared: */
+  input_e = input_e << 1;
+
+  for(j=0; j<nSfb; j++) {
+    li = freqBandTable[j];
+    ui = freqBandTable[j+1];
+
+    FIXP_DBL maxVal = maxSubbandSample( analysBufferReal,
+                                        analysBufferImag,
+                                        li,
+                                        ui,
+                                        start_pos,
+                                        next_pos );
+
+    if (maxVal!=FL2FXCONST_DBL(0.f)) {
+
+      preShift = CntLeadingZeros(maxVal)-1;
+
+      /* If the accu does not provide enough overflow bits, we cannot
+         shift the samples up to the limit.
+         Instead, keep up to 3 free bits in each sample, i.e. up to
+         6 bits after calculation of square.
+         Please note the comment on saturated arithmetic above!
+      */
+      preShift -= SHIFT_BEFORE_SQUARE;
+
+      sumAll = FL2FXCONST_DBL(0.0f);
+
+
+      for (k=li; k<ui; k++) {
+
+        sumLine = FL2FXCONST_DBL(0.0f);
+
+        if (analysBufferImag!=NULL) {
+          if (preShift>=0) {
+            for (l=start_pos; l<next_pos; l++) {
+              temp   = analysBufferReal[l][k] << (int)preShift;
+              sumLine += fPow2Div2(temp);
+              temp   = analysBufferImag[l][k] << (int)preShift;
+              sumLine += fPow2Div2(temp);
+
+            }
+          } else {
+            for (l=start_pos; l<next_pos; l++) {
+              temp   = analysBufferReal[l][k] >> -(int)preShift;
+              sumLine += fPow2Div2(temp);
+              temp   = analysBufferImag[l][k] >> -(int)preShift;
+              sumLine += fPow2Div2(temp);
+            }
+          }
+        } else
+        {
+          if (preShift>=0) {
+            for (l=start_pos; l<next_pos; l++) {
+              temp   = analysBufferReal[l][k] << (int)preShift;
+              sumLine += fPow2Div2(temp);
+            }
+          } else {
+            for (l=start_pos; l<next_pos; l++) {
+              temp   = analysBufferReal[l][k] >> -(int)preShift;
+              sumLine += fPow2Div2(temp);
+            }
+          }
+        }
+
+        /* The number of QMF-channels per SBR bands may be up to 15.
+           Shift right to avoid overflows in sum over all channels. */
+        sumLine = sumLine >> (4-1);
+        sumAll  += sumLine;
+      }
+
+      /* Convert double precision to Mantissa/Exponent: */
+      shift = fNorm(sumAll);
+      sum = sumAll << (int)shift;
+
+      /* Divide by width of envelope: */
+      sum = fMult(sum,invWidth);
+
+      /* Divide by width of Sfb: */
+      sum = fMult(sum, FX_DBL2FX_SGL(GetInvInt(ui-li)));
+
+      /* Set all Subband energies in the Sfb to the average energy: */
+      if (analysBufferImag!=NULL)
+        sum_e = input_e + 4 - shift;  /* -4 to compensate right-shift */
+      else
+        sum_e = input_e + 4 + 1 - shift;  /* -4 to compensate right-shift; +1 due to missing imag. part */
+
+      sum_e -= 2 * preShift;
+    } /* maxVal!=0 */
+    else {
+
+      /* Prevent a zero-mantissa-number from being misinterpreted
+         due to its exponent. */
+      sum = FL2FXCONST_DBL(0.0f);
+      sum_e = 0;
+    }
+
+    for (k=li; k<ui; k++)
+    {
+      *nrgEst++   = sum;
+      *nrgEst_e++ = sum_e;
+    }
+  }
+}
+
+
+/*!
+  \brief  Calculate gain, noise, and additional sine level for one subband.
+
+  The resulting energy gain is given by mantissa and exponent.
+*/
+/*static*/ void calcSubbandGain(FIXP_DBL  nrgRef,            /*!< Reference Energy according to envelope data */
+                            SCHAR     nrgRef_e,          /*!< Reference Energy according to envelope data (exponent) */
+                            ENV_CALC_NRGS* nrgs,
+                            int       i,
+                            FIXP_DBL  tmpNoise,          /*!< Relative noise level */
+                            SCHAR     tmpNoise_e,        /*!< Relative noise level (exponent) */
+                            UCHAR     sinePresentFlag,   /*!< Indicates if sine is present on band */
+                            UCHAR     sineMapped,        /*!< Indicates if sine must be added */
+                            int       noNoiseFlag)       /*!< Flag to suppress noise addition */
+{
+  FIXP_DBL  nrgEst          = nrgs->nrgEst[i];            /*!< Energy in transposed signal */
+  SCHAR     nrgEst_e        = nrgs->nrgEst_e[i];          /*!< Energy in transposed signal (exponent) */
+  FIXP_DBL *ptrNrgGain      = &nrgs->nrgGain[i];          /*!< Resulting energy gain */
+  SCHAR    *ptrNrgGain_e    = &nrgs->nrgGain_e[i];        /*!< Resulting energy gain (exponent) */
+  FIXP_DBL *ptrNoiseLevel   = &nrgs->noiseLevel[i];       /*!< Resulting absolute noise energy */
+  SCHAR    *ptrNoiseLevel_e = &nrgs->noiseLevel_e[i];     /*!< Resulting absolute noise energy (exponent) */
+  FIXP_DBL *ptrNrgSine      = &nrgs->nrgSine[i];          /*!< Additional sine energy */
+  SCHAR    *ptrNrgSine_e    = &nrgs->nrgSine_e[i];        /*!< Additional sine energy (exponent) */
+
+  FIXP_DBL a, b, c;
+  SCHAR    a_e, b_e, c_e;
+
+  /*
+     This addition of 1 prevents divisions by zero in the reference code.
+     For very small energies in nrgEst, it prevents the gains from becoming
+     very high which could cause some trouble due to the smoothing.
+  */
+  b_e = (int)(nrgEst_e - 1);
+  if (b_e>=0) {
+    nrgEst = (FL2FXCONST_DBL(0.5f) >> (INT)fixMin(b_e+1,DFRACT_BITS-1)) + (nrgEst >> 1);
+    nrgEst_e += 1;  /* shift by 1 bit to avoid overflow */
+
+  } else {
+    nrgEst = (nrgEst >> (INT)(fixMin(-b_e+1,DFRACT_BITS-1))) + (FL2FXCONST_DBL(0.5f) >> 1);
+    nrgEst_e = 2;  /* shift by 1 bit to avoid overflow */
+  }
+
+  /*  A = NrgRef * TmpNoise */
+  a = fMult(nrgRef,tmpNoise);
+  a_e = nrgRef_e + tmpNoise_e;
+
+  /*  B = 1 + TmpNoise */
+  b_e = (int)(tmpNoise_e - 1);
+  if (b_e>=0) {
+    b = (FL2FXCONST_DBL(0.5f) >> (INT)fixMin(b_e+1,DFRACT_BITS-1)) + (tmpNoise >> 1);
+    b_e = tmpNoise_e + 1;  /* shift by 1 bit to avoid overflow */
+  } else {
+    b = (tmpNoise >> (INT)(fixMin(-b_e+1,DFRACT_BITS-1))) + (FL2FXCONST_DBL(0.5f) >> 1);
+    b_e = 2;  /* shift by 1 bit to avoid overflow */
+  }
+
+  /*  noiseLevel = A / B = (NrgRef * TmpNoise) / (1 + TmpNoise) */
+  FDK_divide_MantExp( a,  a_e,
+                      b,  b_e,
+                      ptrNoiseLevel, ptrNoiseLevel_e);
+
+  if (sinePresentFlag) {
+
+    /*  C = (1 + TmpNoise) * NrgEst */
+    c = fMult(b,nrgEst);
+    c_e = b_e + nrgEst_e;
+
+    /*  gain = A / C = (NrgRef * TmpNoise) / (1 + TmpNoise) * NrgEst */
+    FDK_divide_MantExp( a,  a_e,
+                        c,  c_e,
+                        ptrNrgGain, ptrNrgGain_e);
+
+    if (sineMapped) {
+
+      /*  sineLevel = nrgRef/ (1 + TmpNoise) */
+      FDK_divide_MantExp( nrgRef,  nrgRef_e,
+                          b,  b_e,
+                          ptrNrgSine, ptrNrgSine_e);
+    }
+  }
+  else {
+    if (noNoiseFlag) {
+      /*  B = NrgEst */
+      b = nrgEst;
+      b_e = nrgEst_e;
+    }
+    else {
+      /*  B = NrgEst * (1 + TmpNoise) */
+      b = fMult(b,nrgEst);
+      b_e = b_e + nrgEst_e;
+    }
+
+
+    /*  gain = nrgRef / B */
+    FDK_divide_MantExp( nrgRef,  nrgRef_e,
+                        b,  b_e,
+                        ptrNrgGain, ptrNrgGain_e);
+  }
+}
+
+
+/*!
+  \brief  Calculate "average gain" for the specified subband range.
+
+  This is rather a gain of the average magnitude than the average
+  of gains!
+  The result is used as a relative limit for all gains within the
+  current "limiter band" (a certain frequency range).
+*/
+/*static*/ void calcAvgGain(ENV_CALC_NRGS* nrgs,
+                        int        lowSubband,    /*!< Begin of the limiter band */
+                        int        highSubband,   /*!< High end of the limiter band */
+                        FIXP_DBL  *ptrSumRef,
+                        SCHAR     *ptrSumRef_e,
+                        FIXP_DBL  *ptrAvgGain,  /*!< Resulting overall gain (mantissa) */
+                        SCHAR     *ptrAvgGain_e)  /*!< Resulting overall gain (exponent) */
+{
+  FIXP_DBL  *nrgRef   = nrgs->nrgRef;       /*!< Reference Energy according to envelope data */
+  SCHAR     *nrgRef_e = nrgs->nrgRef_e;     /*!< Reference Energy according to envelope data (exponent) */
+  FIXP_DBL  *nrgEst   = nrgs->nrgEst;       /*!< Energy in transposed signal */
+  SCHAR     *nrgEst_e = nrgs->nrgEst_e;     /*!< Energy in transposed signal (exponent) */
+
+  FIXP_DBL sumRef = 1;
+  FIXP_DBL sumEst = 1;
+  SCHAR    sumRef_e = -FRACT_BITS;
+  SCHAR    sumEst_e = -FRACT_BITS;
+  int      k;
+
+  for (k=lowSubband; k<highSubband; k++){
+    /* Add nrgRef[k] to sumRef: */
+    FDK_add_MantExp( sumRef, sumRef_e,
+                     nrgRef[k], nrgRef_e[k],
+                     &sumRef, &sumRef_e );
+
+    /* Add nrgEst[k] to sumEst: */
+    FDK_add_MantExp( sumEst, sumEst_e,
+                     nrgEst[k], nrgEst_e[k],
+                     &sumEst, &sumEst_e );
+  }
+
+  FDK_divide_MantExp(sumRef, sumRef_e,
+                     sumEst, sumEst_e,
+                     ptrAvgGain, ptrAvgGain_e);
+
+  *ptrSumRef = sumRef;
+  *ptrSumRef_e = sumRef_e;
+}
+
+
+/*!
+  \brief   Amplify one timeslot of the signal with the calculated gains
+           and add the noisefloor.
+*/
+
+/*static*/ void adjustTimeSlotLC(FIXP_DBL *ptrReal,       /*!< Subband samples to be adjusted, real part */
+                             ENV_CALC_NRGS* nrgs,
+                             UCHAR    *ptrHarmIndex,  /*!< Harmonic index */
+                             int       lowSubband,    /*!< Lowest QMF-channel in the currently used SBR range. */
+                             int       noSubbands,    /*!< Number of QMF subbands */
+                             int       scale_change,  /*!< Number of bits to shift adjusted samples */
+                             int       noNoiseFlag,   /*!< Flag to suppress noise addition */
+                             int      *ptrPhaseIndex, /*!< Start index to random number array */
+                             int       fCldfb)        /*!< CLDFB 80 flag */
+{
+  FIXP_DBL *pGain       = nrgs->nrgGain;     /*!< Gains of current envelope */
+  FIXP_DBL *pNoiseLevel = nrgs->noiseLevel;  /*!< Noise levels of current envelope */
+  FIXP_DBL *pSineLevel  = nrgs->nrgSine;     /*!< Sine levels */
+
+  int    k;
+  int    index = *ptrPhaseIndex;
+  UCHAR  harmIndex = *ptrHarmIndex;
+  UCHAR  freqInvFlag = (lowSubband & 1);
+  FIXP_DBL  signalReal, sineLevel, sineLevelNext, sineLevelPrev;
+  int    tone_count = 0;
+  int    sineSign = 1;
+
+  #define C1   ((FIXP_SGL)FL2FXCONST_SGL(2.f*0.00815f))
+  #define C1_CLDFB ((FIXP_SGL)FL2FXCONST_SGL(2.f*0.16773f))
+
+  /*
+    First pass for k=0 pulled out of the loop:
+  */
+
+  index = (index + 1) & (SBR_NF_NO_RANDOM_VAL - 1);
+
+  /*
+    The next multiplication constitutes the actual envelope adjustment
+    of the signal and should be carried out with full accuracy
+    (supplying #FRACT_BITS valid bits).
+  */
+  signalReal    = fMultDiv2(*ptrReal,*pGain++) << ((int)scale_change);
+  sineLevel     = *pSineLevel++;
+  sineLevelNext = (noSubbands > 1) ? pSineLevel[0] : FL2FXCONST_DBL(0.0f);
+
+  if (sineLevel!=FL2FXCONST_DBL(0.0f)) tone_count++;
+
+  else if (!noNoiseFlag)
+        /* Add noisefloor to the amplified signal */
+        signalReal += (fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][0], pNoiseLevel[0])<<4);
+
+  if (fCldfb) {
+
+    if (!(harmIndex&0x1)) {
+      /* harmIndex 0,2 */
+      signalReal += (harmIndex&0x2) ? -sineLevel : sineLevel;
+      *ptrReal++ = signalReal;
+    }
+    else {
+      /* harmIndex 1,3 in combination with freqInvFlag */
+      int shift = (int) (scale_change+1);
+      shift = (shift>=0) ? fixMin(DFRACT_BITS-1,shift) : fixMax(-(DFRACT_BITS-1),shift);
+
+      FIXP_DBL tmp1 = scaleValue( fMultDiv2(C1_CLDFB, sineLevel), -shift );
+
+      FIXP_DBL tmp2 = fMultDiv2(C1_CLDFB, sineLevelNext);
+
+
+      /* save switch and compare operations and reduce to XOR statement */
+      if ( ((harmIndex>>1)&0x1)^freqInvFlag) {
+          *(ptrReal-1) += tmp1;
+          signalReal   -= tmp2;
+      } else {
+          *(ptrReal-1) -= tmp1;
+          signalReal   += tmp2;
+      }
+      *ptrReal++ = signalReal;
+      freqInvFlag = !freqInvFlag;
+    }
+
+  } else
+  {
+    if (!(harmIndex&0x1)) {
+      /* harmIndex 0,2 */
+      signalReal += (harmIndex&0x2) ? -sineLevel : sineLevel;
+      *ptrReal++ = signalReal;
+    }
+    else {
+      /* harmIndex 1,3 in combination with freqInvFlag */
+      int shift = (int) (scale_change+1);
+      shift = (shift>=0) ? fixMin(DFRACT_BITS-1,shift) : fixMax(-(DFRACT_BITS-1),shift);
+
+      FIXP_DBL tmp1 = (shift>=0) ? ( fMultDiv2(C1, sineLevel) >> shift )
+                                 : ( fMultDiv2(C1, sineLevel) << (-shift) );
+      FIXP_DBL tmp2 = fMultDiv2(C1, sineLevelNext);
+
+
+      /* save switch and compare operations and reduce to XOR statement */
+      if ( ((harmIndex>>1)&0x1)^freqInvFlag) {
+          *(ptrReal-1) += tmp1;
+          signalReal   -= tmp2;
+      } else {
+          *(ptrReal-1) -= tmp1;
+          signalReal   += tmp2;
+      }
+      *ptrReal++ = signalReal;
+      freqInvFlag = !freqInvFlag;
+    }
+  }
+
+  pNoiseLevel++;
+
+  if ( noSubbands > 2 ) {
+    if (!(harmIndex&0x1)) {
+      /* harmIndex 0,2 */
+      if(!harmIndex) 
+      {
+        sineSign = 0;
+      }
+
+      for (k=noSubbands-2; k!=0; k--) {
+        FIXP_DBL sinelevel = *pSineLevel++;
+        index++;
+        if (((signalReal = (sineSign ? -sinelevel : sinelevel)) == FL2FXCONST_DBL(0.0f))  && !noNoiseFlag) 
+        {
+          /* Add noisefloor to the amplified signal */
+          index &= (SBR_NF_NO_RANDOM_VAL - 1);
+          signalReal += (fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][0], pNoiseLevel[0])<<4);
+        }
+        
+        /* The next multiplication constitutes the actual envelope adjustment of the signal. */
+        signalReal += fMultDiv2(*ptrReal,*pGain++) << ((int)scale_change);
+
+        pNoiseLevel++;
+        *ptrReal++ = signalReal;
+      } /* for ... */
+    }
+    else {
+      /* harmIndex 1,3 in combination with freqInvFlag */
+      if (harmIndex==1) freqInvFlag = !freqInvFlag;
+
+      for (k=noSubbands-2; k!=0; k--) {
+        index++;
+        /* The next multiplication constitutes the actual envelope adjustment of the signal. */
+        signalReal = fMultDiv2(*ptrReal,*pGain++) << ((int)scale_change);
+
+        if (*pSineLevel++!=FL2FXCONST_DBL(0.0f)) tone_count++;
+        else if (!noNoiseFlag) {
+          /* Add noisefloor to the amplified signal */
+          index &= (SBR_NF_NO_RANDOM_VAL - 1);
+          signalReal += (fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][0], pNoiseLevel[0])<<4);
+        }
+
+        pNoiseLevel++;
+
+        if (tone_count <= 16) {
+          FIXP_DBL addSine = fMultDiv2((pSineLevel[-2] - pSineLevel[0]), C1);
+          signalReal += (freqInvFlag) ? (-addSine) : (addSine);
+        }
+
+        *ptrReal++ = signalReal;
+        freqInvFlag = !freqInvFlag;
+      } /* for ... */
+    }
+  }
+
+  if (noSubbands > -1) {
+    index++;
+    /* The next multiplication constitutes the actual envelope adjustment of the signal. */
+    signalReal    = fMultDiv2(*ptrReal,*pGain) << ((int)scale_change);
+    sineLevelPrev = fMultDiv2(pSineLevel[-1],FL2FX_SGL(0.0163f));
+    sineLevel     = pSineLevel[0];
+
+    if (pSineLevel[0]!=FL2FXCONST_DBL(0.0f)) tone_count++;
+    else if (!noNoiseFlag) {
+        /* Add noisefloor to the amplified signal */
+        index &= (SBR_NF_NO_RANDOM_VAL - 1);
+        signalReal = signalReal + (fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][0], pNoiseLevel[0])<<4);
+    }
+
+    if (!(harmIndex&0x1)) {
+      /* harmIndex 0,2 */
+      *ptrReal = signalReal + ( (sineSign) ? -sineLevel : sineLevel);
+    }
+    else {
+      /* harmIndex 1,3 in combination with freqInvFlag */
+      if(tone_count <= 16){
+        if (freqInvFlag) {
+          *ptrReal++   = signalReal - sineLevelPrev;
+          if (noSubbands + lowSubband < 63)
+            *ptrReal = *ptrReal + fMultDiv2(C1, sineLevel);
+        }
+        else {
+          *ptrReal++ = signalReal + sineLevelPrev;
+          if (noSubbands + lowSubband < 63)
+            *ptrReal = *ptrReal - fMultDiv2(C1, sineLevel);
+        }
+      }
+      else *ptrReal = signalReal;
+    }
+  }
+  *ptrHarmIndex = (harmIndex + 1) & 3;
+  *ptrPhaseIndex = index & (SBR_NF_NO_RANDOM_VAL - 1);
+}
+void adjustTimeSlotHQ(FIXP_DBL *RESTRICT ptrReal,           /*!< Subband samples to be adjusted, real part */
+                      FIXP_DBL *RESTRICT ptrImag,               /*!< Subband samples to be adjusted, imag part */
+                      HANDLE_SBR_CALCULATE_ENVELOPE h_sbr_cal_env,
+                      ENV_CALC_NRGS* nrgs,
+                      int       lowSubband,            /*!< Lowest QMF-channel in the currently used SBR range. */
+                      int       noSubbands,            /*!< Number of QMF subbands */
+                      int       scale_change,          /*!< Number of bits to shift adjusted samples */
+                      FIXP_SGL  smooth_ratio,          /*!< Impact of last envelope */
+                      int       noNoiseFlag,           /*!< Start index to random number array */
+                      int       filtBufferNoiseShift)  /*!< Shift factor of filtBufferNoise */
+{
+
+  FIXP_DBL *RESTRICT gain       = nrgs->nrgGain;        /*!< Gains of current envelope */
+  FIXP_DBL *RESTRICT noiseLevel = nrgs->noiseLevel;     /*!< Noise levels of current envelope */
+  FIXP_DBL *RESTRICT pSineLevel = nrgs->nrgSine;        /*!< Sine levels */
+
+  FIXP_DBL *RESTRICT filtBuffer      = h_sbr_cal_env->filtBuffer;      /*!< Gains of last envelope */
+  FIXP_DBL *RESTRICT filtBufferNoise = h_sbr_cal_env->filtBufferNoise; /*!< Noise levels of last envelope */
+  UCHAR    *RESTRICT ptrHarmIndex    =&h_sbr_cal_env->harmIndex;       /*!< Harmonic index */
+  int      *RESTRICT ptrPhaseIndex   =&h_sbr_cal_env->phaseIndex;      /*!< Start index to random number array */
+
+  int    k;
+  FIXP_DBL signalReal, signalImag;
+  FIXP_DBL noiseReal,  noiseImag;
+  FIXP_DBL  smoothedGain, smoothedNoise;
+  FIXP_SGL direct_ratio = /*FL2FXCONST_SGL(1.0f) */ (FIXP_SGL)MAXVAL_SGL - smooth_ratio;
+  int    index = *ptrPhaseIndex;
+  UCHAR   harmIndex = *ptrHarmIndex;
+  register int freqInvFlag = (lowSubband & 1);
+  FIXP_DBL sineLevel;
+  int shift;
+
+  *ptrPhaseIndex = (index+noSubbands) & (SBR_NF_NO_RANDOM_VAL - 1);
+  *ptrHarmIndex = (harmIndex + 1) & 3;
+
+  /*
+    Possible optimization:
+    smooth_ratio and harmIndex stay constant during the loop.
+    It might be faster to include a separate loop in each path.
+
+    the check for smooth_ratio is now outside the loop and the workload
+    of the whole function decreased by about 20 %
+  */
+
+  filtBufferNoiseShift += 1;      /* due to later use of fMultDiv2 instead of fMult */
+  if (filtBufferNoiseShift<0)
+    shift = fixMin(DFRACT_BITS-1,-filtBufferNoiseShift);
+  else
+    shift = fixMin(DFRACT_BITS-1, filtBufferNoiseShift);
+
+  if (smooth_ratio > FL2FXCONST_SGL(0.0f)) {
+
+    for (k=0; k<noSubbands; k++) {
+      /*
+        Smoothing: The old envelope has been bufferd and a certain ratio
+        of the old gains and noise levels is used.
+      */
+
+      smoothedGain = fMult(smooth_ratio,filtBuffer[k]) +
+                     fMult(direct_ratio,gain[k]);
+
+      if (filtBufferNoiseShift<0) {
+        smoothedNoise = (fMultDiv2(smooth_ratio,filtBufferNoise[k])>>shift) +
+                         fMult(direct_ratio,noiseLevel[k]);
+      }
+      else {
+        smoothedNoise = (fMultDiv2(smooth_ratio,filtBufferNoise[k])<<shift) +
+                         fMult(direct_ratio,noiseLevel[k]);
+      }
+
+      /*
+        The next 2 multiplications constitute the actual envelope adjustment
+        of the signal and should be carried out with full accuracy
+        (supplying #DFRACT_BITS valid bits).
+      */
+      signalReal = fMultDiv2(*ptrReal,smoothedGain)<<((int)scale_change);
+      signalImag = fMultDiv2(*ptrImag,smoothedGain)<<((int)scale_change);
+
+      index++;
+
+      if (pSineLevel[k] != FL2FXCONST_DBL(0.0f)) {
+        sineLevel = pSineLevel[k];
+
+        switch(harmIndex) {
+        case 0:
+          *ptrReal++ = (signalReal + sineLevel);
+          *ptrImag++ = (signalImag);
+          break;
+        case 2:
+          *ptrReal++ = (signalReal - sineLevel);
+          *ptrImag++ = (signalImag);
+          break;
+        case 1:
+          *ptrReal++ = (signalReal);
+          if (freqInvFlag)
+            *ptrImag++ = (signalImag - sineLevel);
+          else
+            *ptrImag++ = (signalImag + sineLevel);
+          break;
+        case 3:
+          *ptrReal++ = signalReal;
+          if (freqInvFlag)
+            *ptrImag++ = (signalImag + sineLevel);
+          else
+            *ptrImag++ = (signalImag - sineLevel);
+          break;
+        }
+      }
+      else {
+        if (noNoiseFlag) {
+          /* Just the amplified signal is saved */
+          *ptrReal++ = (signalReal);
+          *ptrImag++ = (signalImag);
+        }
+        else {
+          /* Add noisefloor to the amplified signal */
+          index &= (SBR_NF_NO_RANDOM_VAL - 1);
+          noiseReal = fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][0], smoothedNoise)<<4;
+          noiseImag = fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][1], smoothedNoise)<<4;
+          *ptrReal++ = (signalReal + noiseReal);
+          *ptrImag++ = (signalImag + noiseImag);
+        }
+      }
+      freqInvFlag ^= 1;
+    }
+
+  }
+  else 
+  {
+    for (k=0; k<noSubbands; k++) 
+    {
+      smoothedGain  = gain[k];
+      signalReal = fMultDiv2(*ptrReal, smoothedGain) << scale_change;
+      signalImag = fMultDiv2(*ptrImag, smoothedGain) << scale_change;
+
+      index++;
+
+      if ((sineLevel = pSineLevel[k]) != FL2FXCONST_DBL(0.0f)) 
+      {
+        switch (harmIndex) 
+        {
+        case 0:
+          signalReal += sineLevel;
+          break;
+        case 1:
+          if (freqInvFlag)
+            signalImag -= sineLevel;
+          else
+            signalImag += sineLevel;
+          break;
+        case 2:
+          signalReal -= sineLevel;
+          break;
+        case 3:
+          if (freqInvFlag)
+            signalImag += sineLevel;
+          else
+            signalImag -= sineLevel;
+          break;
+        }
+      }
+      else 
+      {
+        if (noNoiseFlag == 0)
+        {
+          /* Add noisefloor to the amplified signal */
+          smoothedNoise = noiseLevel[k];
+          index &= (SBR_NF_NO_RANDOM_VAL - 1);
+          noiseReal = fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][0], smoothedNoise);
+          noiseImag = fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][1], smoothedNoise);
+          signalReal += noiseReal<<4;
+          signalImag += noiseImag<<4;
+        }
+      }
+      *ptrReal++ = signalReal;
+      *ptrImag++ = signalImag;
+
+      freqInvFlag ^= 1;
+    }
+  }
+}
+
+
+/*!
+  \brief   Reset limiter bands.
+
+  Build frequency band table for the gain limiter dependent on
+  the previously generated transposer patch areas.
+
+  \return  SBRDEC_OK if ok,  SBRDEC_UNSUPPORTED_CONFIG on error
+*/
+SBR_ERROR
+ResetLimiterBands ( UCHAR *limiterBandTable,   /*!< Resulting band borders in QMF channels */
+                    UCHAR *noLimiterBands,     /*!< Resulting number of limiter band */
+                    UCHAR *freqBandTable,      /*!< Table with possible band borders */
+                    int noFreqBands,                   /*!< Number of bands in freqBandTable */
+                    const PATCH_PARAM *patchParam,     /*!< Transposer patch parameters */
+                    int noPatches,                     /*!< Number of transposer patches */
+                    int limiterBands)                  /*!< Selected 'band density' from bitstream */
+{
+  int i, k, isPatchBorder[2], loLimIndex, hiLimIndex, tempNoLim, nBands;
+  UCHAR workLimiterBandTable[MAX_FREQ_COEFFS / 2 + MAX_NUM_PATCHES + 1];
+  int patchBorders[MAX_NUM_PATCHES + 1];
+  int kx, k2;
+  FIXP_DBL temp;
+
+  int lowSubband = freqBandTable[0];
+  int highSubband = freqBandTable[noFreqBands];
+
+  /* 1 limiter band. */
+  if(limiterBands == 0) {
+    limiterBandTable[0] = 0;
+    limiterBandTable[1] = highSubband - lowSubband;
+    nBands = 1;
+  } else {
+    for (i = 0; i < noPatches; i++) {
+      patchBorders[i] = patchParam[i].guardStartBand - lowSubband;
+    }
+    patchBorders[i] = highSubband - lowSubband;
+
+    /* 1.2, 2, or 3 limiter bands/octave plus bandborders at patchborders. */
+    for (k = 0; k <= noFreqBands; k++) {
+      workLimiterBandTable[k] = freqBandTable[k] - lowSubband;
+    }
+    for (k = 1; k < noPatches; k++) {
+      workLimiterBandTable[noFreqBands + k] = patchBorders[k];
+    }
+
+    tempNoLim = nBands = noFreqBands + noPatches - 1;
+    shellsort(workLimiterBandTable, tempNoLim + 1);
+
+    loLimIndex = 0;
+    hiLimIndex = 1;
+
+
+    while (hiLimIndex <= tempNoLim) {
+      k2 = workLimiterBandTable[hiLimIndex] + lowSubband;
+      kx = workLimiterBandTable[loLimIndex] + lowSubband;
+
+      temp = FX_SGL2FX_DBL(FDK_getNumOctavesDiv8(kx,k2)); /* Number of octaves */
+      temp = fMult(temp, FDK_sbrDecoder_sbr_limiterBandsPerOctaveDiv4[limiterBands]);
+
+      if (temp < FL2FXCONST_DBL (0.49f)>>5) {
+        if (workLimiterBandTable[hiLimIndex] == workLimiterBandTable[loLimIndex]) {
+          workLimiterBandTable[hiLimIndex] = highSubband;
+          nBands--;
+          hiLimIndex++;
+          continue;
+        }
+        isPatchBorder[0] = isPatchBorder[1] = 0;
+        for (k = 0; k <= noPatches; k++) {
+          if (workLimiterBandTable[hiLimIndex] == patchBorders[k]) {
+            isPatchBorder[1] = 1;
+            break;
+          }
+        }
+        if (!isPatchBorder[1]) {
+          workLimiterBandTable[hiLimIndex] = highSubband;
+          nBands--;
+          hiLimIndex++;
+          continue;
+        }
+        for (k = 0; k <= noPatches; k++) {
+          if (workLimiterBandTable[loLimIndex] == patchBorders[k]) {
+            isPatchBorder[0] = 1;
+            break;
+          }
+        }
+        if (!isPatchBorder[0]) {
+          workLimiterBandTable[loLimIndex] = highSubband;
+          nBands--;
+        }
+      }
+      loLimIndex = hiLimIndex;
+      hiLimIndex++;
+
+    }
+    shellsort(workLimiterBandTable, tempNoLim + 1);
+
+    /* Test if algorithm exceeded maximum allowed limiterbands */
+    if( nBands > MAX_NUM_LIMITERS || nBands <= 0) {
+      return SBRDEC_UNSUPPORTED_CONFIG;
+    }
+
+    /* Copy limiterbands from working buffer into final destination */
+    for (k = 0; k <= nBands; k++) {
+      limiterBandTable[k] = workLimiterBandTable[k];
+    }
+  }
+  *noLimiterBands = nBands;
+
+  return SBRDEC_OK;
+}
+
diff --git a/libSBRdec/src/env_calc.h b/libSBRdec/src/env_calc.h
new file mode 100644
index 0000000..d21e0b3
--- /dev/null
+++ b/libSBRdec/src/env_calc.h
@@ -0,0 +1,165 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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 calculation prototypes  
+*/
+#ifndef __ENV_CALC_H
+#define __ENV_CALC_H
+
+#include "sbrdecoder.h"
+#include "env_extr.h"  /* for HANDLE_SBR_HEADER_DATA */
+#include "sbr_scale.h"
+
+
+typedef struct
+{
+  FIXP_DBL  filtBuffer[MAX_FREQ_COEFFS];          /*!< previous gains (required for smoothing) */
+  FIXP_DBL  filtBufferNoise[MAX_FREQ_COEFFS];     /*!< previous noise levels (required for smoothing) */
+  SCHAR     filtBuffer_e[MAX_FREQ_COEFFS];        /*!< Exponents of previous gains */
+  SCHAR     filtBufferNoise_e;                    /*!< Common exponent of previous noise levels */
+
+  int startUp;               /*!< flag to signal initial conditions in buffers */
+  int phaseIndex;            /*!< Index for randomPase array */
+  int prevTranEnv;           /*!< The transient envelope of the previous frame. */
+
+  int harmFlagsPrev[(MAX_FREQ_COEFFS+15)/16];
+  /*!< Words with 16 flags each indicating where a sine was added in the previous frame.*/
+  UCHAR harmIndex;            /*!< Current phase of synthetic sine */
+
+}
+SBR_CALCULATE_ENVELOPE;
+
+typedef SBR_CALCULATE_ENVELOPE *HANDLE_SBR_CALCULATE_ENVELOPE;
+
+
+
+void
+calculateSbrEnvelope (QMF_SCALE_FACTOR  *sbrScaleFactor,
+                      HANDLE_SBR_CALCULATE_ENVELOPE h_sbr_cal_env,
+                      HANDLE_SBR_HEADER_DATA hHeaderData,
+                      HANDLE_SBR_FRAME_DATA hFrameData,
+                      FIXP_DBL **analysBufferReal,
+                      FIXP_DBL **analysBufferImag,                 /*!< Imag part of subband samples to be processed */
+                      const int useLP,
+                      FIXP_DBL *degreeAlias,                       /*!< Estimated aliasing for each QMF channel */
+                      const UINT flags,
+                      const int frameErrorFlag
+                      );
+
+SBR_ERROR
+createSbrEnvelopeCalc (HANDLE_SBR_CALCULATE_ENVELOPE hSbrCalculateEnvelope,
+                       HANDLE_SBR_HEADER_DATA hHeaderData,
+                       const int chan,
+                       const UINT flags);
+
+int
+deleteSbrEnvelopeCalc (HANDLE_SBR_CALCULATE_ENVELOPE hSbrCalculateEnvelope);
+
+void
+resetSbrEnvelopeCalc (HANDLE_SBR_CALCULATE_ENVELOPE hCalEnv);
+
+SBR_ERROR
+ResetLimiterBands ( UCHAR *limiterBandTable,
+                    UCHAR *noLimiterBands,
+                    UCHAR *freqBandTable,
+                    int noFreqBands,
+                    const PATCH_PARAM *patchParam,
+                    int noPatches,
+                    int limiterBands);
+
+void rescaleSubbandSamples( FIXP_DBL ** re,
+                            FIXP_DBL ** im,
+                            int lowSubband, int noSubbands,
+                            int start_pos,  int next_pos,
+                            int shift);
+
+FIXP_DBL maxSubbandSample( FIXP_DBL ** analysBufferReal_m,
+                           FIXP_DBL ** analysBufferImag_m,
+                           int lowSubband,
+                           int highSubband,
+                           int start_pos,
+                           int stop_pos);
+
+#endif // __ENV_CALC_H
diff --git a/libSBRdec/src/env_dec.cpp b/libSBRdec/src/env_dec.cpp
new file mode 100644
index 0000000..ac6c299
--- /dev/null
+++ b/libSBRdec/src/env_dec.cpp
@@ -0,0 +1,852 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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 decoding  
+  This module provides envelope decoding and error concealment algorithms. The main
+  entry point is decodeSbrData().
+
+  \sa decodeSbrData(),\ref documentationOverview
+*/
+
+#include "env_dec.h"
+
+#include "env_extr.h"
+#include "transcendent.h"
+
+#include "genericStds.h"
+
+
+static void decodeEnvelope (HANDLE_SBR_HEADER_DATA hHeaderData,
+                            HANDLE_SBR_FRAME_DATA h_sbr_data,
+                            HANDLE_SBR_PREV_FRAME_DATA h_prev_data,
+                            HANDLE_SBR_PREV_FRAME_DATA h_prev_data_otherChannel);
+static void sbr_envelope_unmapping (HANDLE_SBR_HEADER_DATA hHeaderData,
+                                    HANDLE_SBR_FRAME_DATA h_data_left,
+                                    HANDLE_SBR_FRAME_DATA h_data_right);
+static void requantizeEnvelopeData (HANDLE_SBR_FRAME_DATA h_sbr_data,
+                                    int ampResolution);
+static void deltaToLinearPcmEnvelopeDecoding (HANDLE_SBR_HEADER_DATA hHeaderData,
+                                              HANDLE_SBR_FRAME_DATA h_sbr_data,
+                                              HANDLE_SBR_PREV_FRAME_DATA h_prev_data);
+static void decodeNoiseFloorlevels (HANDLE_SBR_HEADER_DATA hHeaderData,
+                                    HANDLE_SBR_FRAME_DATA h_sbr_data,
+                                    HANDLE_SBR_PREV_FRAME_DATA h_prev_data);
+static void timeCompensateFirstEnvelope (HANDLE_SBR_HEADER_DATA hHeaderData,
+                                         HANDLE_SBR_FRAME_DATA h_sbr_data,
+                                         HANDLE_SBR_PREV_FRAME_DATA h_prev_data);
+static int checkEnvelopeData (HANDLE_SBR_HEADER_DATA hHeaderData,
+                              HANDLE_SBR_FRAME_DATA h_sbr_data,
+                              HANDLE_SBR_PREV_FRAME_DATA h_prev_data);
+
+
+
+#define SBR_ENERGY_PAN_OFFSET   (12 << ENV_EXP_FRACT)
+#define SBR_MAX_ENERGY          (35 << ENV_EXP_FRACT)
+
+#define DECAY                   ( 1 << ENV_EXP_FRACT)
+
+#if ENV_EXP_FRACT
+#define DECAY_COUPLING          ( 1 << (ENV_EXP_FRACT-1) ) /*!< corresponds to a value of 0.5 */
+#else
+#define DECAY_COUPLING            1  /*!< If the energy data is not shifted, use 1 instead of 0.5 */
+#endif
+
+
+/*!
+  \brief  Convert table index
+*/
+static int indexLow2High(int offset, /*!< mapping factor */
+                         int index,  /*!< index to scalefactor band */
+                         int res)    /*!< frequency resolution */
+{
+  if(res == 0)
+  {
+    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);
+}
+
+
+/*!
+  \brief  Update previous envelope value for delta-coding
+
+  The current envelope values needs to be stored for delta-coding
+  in the next frame.  The stored envelope is always represented with
+  the high frequency resolution.  If the current envelope uses the
+  low frequency resolution, the energy value will be mapped to the
+  corresponding high-res bands.
+*/
+static void mapLowResEnergyVal(FIXP_SGL currVal,  /*!< current energy value */
+                               FIXP_SGL* prevData,/*!< pointer to previous data vector */
+                               int offset,      /*!< mapping factor */
+                               int index,       /*!< index to scalefactor band */
+                               int res)         /*!< frequeny resolution */
+{
+  if(res == 0)
+  {
+    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;
+}
+
+
+
+/*!
+  \brief    Convert raw envelope and noisefloor data to energy levels
+
+  This function is being called by sbrDecoder_ParseElement() and provides two important algorithms:
+
+  First the function decodes envelopes and noise floor levels as described in requantizeEnvelopeData()
+  and sbr_envelope_unmapping(). The function also implements concealment algorithms in case there are errors
+  within the sbr data. For both operations fractional arithmetic is used.
+  Therefore you might encounter different output values on your target
+  system compared to the reference implementation.
+*/
+void
+decodeSbrData (HANDLE_SBR_HEADER_DATA hHeaderData,          /*!< Static control data */
+               HANDLE_SBR_FRAME_DATA h_data_left,           /*!< pointer to left channel frame data */
+               HANDLE_SBR_PREV_FRAME_DATA h_prev_data_left, /*!< pointer to left channel previous frame data */
+               HANDLE_SBR_FRAME_DATA h_data_right,          /*!< pointer to right channel frame data */
+               HANDLE_SBR_PREV_FRAME_DATA h_prev_data_right)/*!< pointer to right channel previous frame data */
+{
+  FIXP_SGL tempSfbNrgPrev[MAX_FREQ_COEFFS];
+  int errLeft;
+
+  /* Save previous energy values to be able to reuse them later for concealment. */
+  FDKmemcpy (tempSfbNrgPrev, h_prev_data_left->sfb_nrg_prev, MAX_FREQ_COEFFS * sizeof(FIXP_SGL));
+
+  decodeEnvelope (hHeaderData, h_data_left, h_prev_data_left, h_prev_data_right);
+  decodeNoiseFloorlevels (hHeaderData, h_data_left, h_prev_data_left);
+
+  if(h_data_right != NULL) {
+    errLeft = hHeaderData->frameErrorFlag;
+    decodeEnvelope (hHeaderData, h_data_right, h_prev_data_right, h_prev_data_left);
+    decodeNoiseFloorlevels (hHeaderData, h_data_right, h_prev_data_right);
+
+    if (!errLeft && hHeaderData->frameErrorFlag) {
+      /* If an error occurs in the right channel where the left channel seemed ok,
+         we apply concealment also on the left channel. This ensures that the coupling
+         modes of both channels match and that we have the same number of envelopes in
+         coupling mode.
+         However, as the left channel has already been processed before, the resulting
+         energy levels are not the same as if the left channel had been concealed
+         during the first call of decodeEnvelope().
+      */
+      /* Restore previous energy values for concealment, because the values have been
+         overwritten by the first call of decodeEnvelope(). */
+      FDKmemcpy (h_prev_data_left->sfb_nrg_prev, tempSfbNrgPrev, MAX_FREQ_COEFFS * sizeof(FIXP_SGL));
+      /* Do concealment */
+      decodeEnvelope (hHeaderData, h_data_left, h_prev_data_left, h_prev_data_right);
+    }
+
+    if (h_data_left->coupling) {
+      sbr_envelope_unmapping (hHeaderData, h_data_left, h_data_right);
+    }
+  }
+
+  /* Display the data for debugging: */
+}
+
+
+/*!
+  \brief   Convert from coupled channels to independent L/R data
+*/
+static void
+sbr_envelope_unmapping (HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */
+                        HANDLE_SBR_FRAME_DATA h_data_left,  /*!< pointer to left channel */
+                        HANDLE_SBR_FRAME_DATA h_data_right) /*!< pointer to right channel */
+{
+  int i;
+  FIXP_SGL tempL_m, tempR_m, tempRplus1_m, newL_m, newR_m;
+  SCHAR   tempL_e, tempR_e, tempRplus1_e, newL_e, newR_e;
+
+
+  /* 1. Unmap (already dequantized) coupled envelope energies */
+
+  for (i = 0; i < h_data_left->nScaleFactors; i++) {
+    tempR_m = (FIXP_SGL)((LONG)h_data_right->iEnvelope[i] & MASK_M);
+    tempR_e = (SCHAR)((LONG)h_data_right->iEnvelope[i] & MASK_E);
+
+    tempR_e -= (18 + NRG_EXP_OFFSET);  /* -18 = ld(UNMAPPING_SCALE / h_data_right->nChannels) */
+    tempL_m = (FIXP_SGL)((LONG)h_data_left->iEnvelope[i] & MASK_M);
+    tempL_e = (SCHAR)((LONG)h_data_left->iEnvelope[i] & MASK_E);
+
+    tempL_e -= NRG_EXP_OFFSET;
+
+    /* Calculate tempRight+1 */
+    FDK_add_MantExp( tempR_m, tempR_e,
+                     FL2FXCONST_SGL(0.5f), 1,  /* 1.0 */
+                     &tempRplus1_m, &tempRplus1_e);
+
+    FDK_divide_MantExp( tempL_m, tempL_e+1,  /*  2 * tempLeft */
+                       tempRplus1_m, tempRplus1_e,
+                       &newR_m, &newR_e );
+
+    if (newR_m >= ((FIXP_SGL)MAXVAL_SGL - ROUNDING)) {
+      newR_m >>= 1;
+      newR_e += 1;
+    }
+
+    newL_m = FX_DBL2FX_SGL(fMult(tempR_m,newR_m));
+    newL_e = tempR_e + newR_e;
+
+    h_data_right->iEnvelope[i] = ((FIXP_SGL)((SHORT)(FIXP_SGL)(newR_m + ROUNDING) & MASK_M)) +
+                                  (FIXP_SGL)((SHORT)(FIXP_SGL)(newR_e + NRG_EXP_OFFSET) & MASK_E);
+    h_data_left->iEnvelope[i] =  ((FIXP_SGL)((SHORT)(FIXP_SGL)(newL_m + ROUNDING) & MASK_M)) +
+                                  (FIXP_SGL)((SHORT)(FIXP_SGL)(newL_e + NRG_EXP_OFFSET) & MASK_E);
+  }
+
+  /* 2. Dequantize and unmap coupled noise floor levels */
+
+  for (i = 0; i < hHeaderData->freqBandData.nNfb * h_data_left->frameInfo.nNoiseEnvelopes; i++) {
+
+    tempL_e = (SCHAR)(6 - (LONG)h_data_left->sbrNoiseFloorLevel[i]);
+    tempR_e = (SCHAR)((LONG)h_data_right->sbrNoiseFloorLevel[i] - 12) /*SBR_ENERGY_PAN_OFFSET*/;
+
+    /* Calculate tempR+1 */
+    FDK_add_MantExp( FL2FXCONST_SGL(0.5f), 1+tempR_e, /* tempR */
+                     FL2FXCONST_SGL(0.5f), 1,         /*  1.0  */
+                     &tempRplus1_m, &tempRplus1_e);
+
+    /* Calculate 2*tempLeft/(tempR+1) */
+    FDK_divide_MantExp( FL2FXCONST_SGL(0.5f), tempL_e+2,  /*  2 * tempLeft */
+                       tempRplus1_m, tempRplus1_e,
+                       &newR_m, &newR_e );
+
+    /* if (newR_m >= ((FIXP_SGL)MAXVAL_SGL - ROUNDING)) {
+      newR_m >>= 1;
+      newR_e += 1;
+    } */
+
+    /* L = tempR * R */
+    newL_m = newR_m;
+    newL_e = newR_e + tempR_e;
+    h_data_right->sbrNoiseFloorLevel[i] = ((FIXP_SGL)((SHORT)(FIXP_SGL)(newR_m + ROUNDING) & MASK_M)) +
+                                           (FIXP_SGL)((SHORT)(FIXP_SGL)(newR_e + NOISE_EXP_OFFSET) & MASK_E);
+    h_data_left->sbrNoiseFloorLevel[i] =  ((FIXP_SGL)((SHORT)(FIXP_SGL)(newL_m + ROUNDING) & MASK_M)) +
+                                           (FIXP_SGL)((SHORT)(FIXP_SGL)(newL_e + NOISE_EXP_OFFSET) & MASK_E);
+  }
+}
+
+
+/*!
+  \brief    Simple alternative to the real SBR concealment
+
+  If the real frameInfo is not available due to a frame loss, a replacement will
+  be constructed with 1 envelope spanning the whole frame (FIX-FIX).
+  The delta-coded energies are set to negative values, resulting in a fade-down.
+  In case of coupling, the balance-channel will move towards the center.
+*/
+static void
+leanSbrConcealment(HANDLE_SBR_HEADER_DATA hHeaderData,     /*!< Static control data */
+                   HANDLE_SBR_FRAME_DATA  h_sbr_data,      /*!< pointer to current data */
+                   HANDLE_SBR_PREV_FRAME_DATA h_prev_data  /*!< pointer to data of last frame */
+                   )
+{
+  FIXP_SGL target;  /* targeted level for sfb_nrg_prev during fade-down */
+  FIXP_SGL step;    /* speed of fade */
+  int i;
+
+  int currentStartPos = h_prev_data->stopPos - hHeaderData->numberTimeSlots;
+  int currentStopPos = hHeaderData->numberTimeSlots;
+
+
+  /* Use some settings of the previous frame */
+  h_sbr_data->ampResolutionCurrentFrame = h_prev_data->ampRes;
+  h_sbr_data->coupling = h_prev_data->coupling;
+  for(i=0;i<MAX_INVF_BANDS;i++)
+    h_sbr_data->sbr_invf_mode[i] = h_prev_data->sbr_invf_mode[i];
+
+  /* Generate concealing control data */
+
+  h_sbr_data->frameInfo.nEnvelopes = 1;
+  h_sbr_data->frameInfo.borders[0] = currentStartPos;
+  h_sbr_data->frameInfo.borders[1] = currentStopPos;
+  h_sbr_data->frameInfo.freqRes[0] = 1;
+  h_sbr_data->frameInfo.tranEnv = -1;  /* no transient */
+  h_sbr_data->frameInfo.nNoiseEnvelopes = 1;
+  h_sbr_data->frameInfo.bordersNoise[0] = currentStartPos;
+  h_sbr_data->frameInfo.bordersNoise[1] = currentStopPos;
+
+  h_sbr_data->nScaleFactors = hHeaderData->freqBandData.nSfb[1];
+
+  /* Generate fake envelope data */
+
+  h_sbr_data->domain_vec[0] = 1;
+
+  if (h_sbr_data->coupling == COUPLING_BAL) {
+    target = (FIXP_SGL)SBR_ENERGY_PAN_OFFSET;
+    step = (FIXP_SGL)DECAY_COUPLING;
+  }
+  else {
+    target = FL2FXCONST_SGL(0.0f);
+    step   = (FIXP_SGL)DECAY;
+  }
+  if (hHeaderData->bs_info.ampResolution == 0) {
+    target <<= 1;
+    step   <<= 1;
+  }
+
+  for (i=0; i < h_sbr_data->nScaleFactors; i++) {
+    if (h_prev_data->sfb_nrg_prev[i] > target)
+      h_sbr_data->iEnvelope[i] = -step;
+    else
+      h_sbr_data->iEnvelope[i] = step;
+  }
+
+  /* Noisefloor levels are always cleared ... */
+
+  h_sbr_data->domain_vec_noise[0] = 1;
+  for (i=0; i < hHeaderData->freqBandData.nNfb; i++)
+    h_sbr_data->sbrNoiseFloorLevel[i] = FL2FXCONST_SGL(0.0f);
+
+  /* ... and so are the sines */
+  FDKmemclear(h_sbr_data->addHarmonics, MAX_FREQ_COEFFS);
+}
+
+
+/*!
+  \brief   Build reference energies and noise levels from bitstream elements
+*/
+static void
+decodeEnvelope (HANDLE_SBR_HEADER_DATA hHeaderData,     /*!< Static control data */
+                HANDLE_SBR_FRAME_DATA  h_sbr_data,      /*!< pointer to current data */
+                HANDLE_SBR_PREV_FRAME_DATA h_prev_data, /*!< pointer to data of last frame */
+                HANDLE_SBR_PREV_FRAME_DATA otherChannel /*!< other channel's last frame data */
+                )
+{
+  int i;
+  int fFrameError = hHeaderData->frameErrorFlag;
+  FIXP_SGL tempSfbNrgPrev[MAX_FREQ_COEFFS];
+
+  if (!fFrameError) {
+    /*
+      To avoid distortions after bad frames, set the error flag if delta coding in time occurs.
+      However, SBR can take a little longer to come up again.
+    */
+    if ( h_prev_data->frameErrorFlag ) {
+      if (h_sbr_data->domain_vec[0] != 0) {
+        fFrameError = 1;
+      }
+    } else {
+      /* Check that the previous stop position and the current start position match.
+         (Could be done in checkFrameInfo(), but the previous frame data is not available there) */
+      if ( h_sbr_data->frameInfo.borders[0] != h_prev_data->stopPos - hHeaderData->numberTimeSlots ) {
+        /* Both the previous as well as the current frame are flagged to be ok, but they do not match! */
+        if (h_sbr_data->domain_vec[0] == 1) {
+          /* Prefer concealment over delta-time coding between the mismatching frames */
+          fFrameError = 1;
+        }
+        else {
+          /* Close the gap in time by triggering timeCompensateFirstEnvelope() */
+          fFrameError = 1;
+        }
+      }
+    }
+  }
+
+
+  if (fFrameError)       /* Error is detected */
+    {
+      leanSbrConcealment(hHeaderData,
+                         h_sbr_data,
+                         h_prev_data);
+
+      /* decode the envelope data to linear PCM */
+      deltaToLinearPcmEnvelopeDecoding (hHeaderData, h_sbr_data, h_prev_data);
+    }
+  else                          /*Do a temporary dummy decoding and check that the envelope values are within limits */
+    {
+      if (h_prev_data->frameErrorFlag) {
+        timeCompensateFirstEnvelope (hHeaderData, h_sbr_data, h_prev_data);
+        if (h_sbr_data->coupling != h_prev_data->coupling) {
+          /*
+            Coupling mode has changed during concealment.
+             The stored energy levels need to be converted.
+           */
+          for (i = 0; i < hHeaderData->freqBandData.nSfb[1]; i++) {
+            /* Former Level-Channel will be used for both channels */
+            if (h_prev_data->coupling == COUPLING_BAL)
+              h_prev_data->sfb_nrg_prev[i] = otherChannel->sfb_nrg_prev[i];
+            /* Former L/R will be combined as the new Level-Channel */
+            else if (h_sbr_data->coupling == COUPLING_LEVEL)
+              h_prev_data->sfb_nrg_prev[i] = (h_prev_data->sfb_nrg_prev[i] + otherChannel->sfb_nrg_prev[i]) >> 1;
+            else if (h_sbr_data->coupling == COUPLING_BAL)
+              h_prev_data->sfb_nrg_prev[i] = (FIXP_SGL)SBR_ENERGY_PAN_OFFSET;
+          }
+        }
+      }
+      FDKmemcpy (tempSfbNrgPrev, h_prev_data->sfb_nrg_prev,
+              MAX_FREQ_COEFFS * sizeof (FIXP_SGL));
+
+      deltaToLinearPcmEnvelopeDecoding (hHeaderData, h_sbr_data, h_prev_data);
+
+      fFrameError = checkEnvelopeData (hHeaderData, h_sbr_data, h_prev_data);
+
+      if (fFrameError)
+        {
+          hHeaderData->frameErrorFlag = 1;
+          FDKmemcpy (h_prev_data->sfb_nrg_prev, tempSfbNrgPrev,
+                  MAX_FREQ_COEFFS * sizeof (FIXP_SGL));
+          decodeEnvelope (hHeaderData, h_sbr_data, h_prev_data, otherChannel);
+          return;
+        }
+    }
+
+  requantizeEnvelopeData (h_sbr_data, h_sbr_data->ampResolutionCurrentFrame);
+
+  hHeaderData->frameErrorFlag = fFrameError;
+}
+
+
+/*!
+  \brief   Verify that envelope energies are within the allowed range
+  \return  0 if all is fine, 1 if an envelope value was too high
+*/
+static int
+checkEnvelopeData (HANDLE_SBR_HEADER_DATA hHeaderData,     /*!< Static control data */
+                   HANDLE_SBR_FRAME_DATA h_sbr_data,       /*!< pointer to current data */
+                   HANDLE_SBR_PREV_FRAME_DATA h_prev_data  /*!< pointer to data of last frame */
+                   )
+{
+  FIXP_SGL *iEnvelope = h_sbr_data->iEnvelope;
+  FIXP_SGL *sfb_nrg_prev = h_prev_data->sfb_nrg_prev;
+  int    i = 0, errorFlag = 0;
+  FIXP_SGL sbr_max_energy =
+    (h_sbr_data->ampResolutionCurrentFrame == 1) ? SBR_MAX_ENERGY : (SBR_MAX_ENERGY << 1);
+
+  /*
+    Range check for current energies
+  */
+  for (i = 0; i < h_sbr_data->nScaleFactors; i++) {
+    if (iEnvelope[i] > sbr_max_energy) {
+      errorFlag = 1;
+    }
+    if (iEnvelope[i] < FL2FXCONST_SGL(0.0f)) {
+      errorFlag = 1;
+      /* iEnvelope[i] = FL2FXCONST_SGL(0.0f); */
+    }
+  }
+
+  /*
+    Range check for previous energies
+  */
+  for (i = 0; i < hHeaderData->freqBandData.nSfb[1]; i++) {
+    sfb_nrg_prev[i] = fixMax(sfb_nrg_prev[i], FL2FXCONST_SGL(0.0f));
+    sfb_nrg_prev[i] = fixMin(sfb_nrg_prev[i], sbr_max_energy);
+  }
+
+  return (errorFlag);
+}
+
+
+/*!
+  \brief   Verify that the noise levels are within the allowed range
+
+  The function is equivalent to checkEnvelopeData().
+  When the noise-levels are being decoded, it is already too late for
+  concealment. Therefore the noise levels are simply limited here.
+*/
+static void
+limitNoiseLevels(HANDLE_SBR_HEADER_DATA hHeaderData,     /*!< Static control data */
+                 HANDLE_SBR_FRAME_DATA h_sbr_data)       /*!< pointer to current data */
+{
+  int i;
+  int nNfb = hHeaderData->freqBandData.nNfb;
+
+  /*
+    Set range limits. The exact values depend on the coupling mode.
+    However this limitation is primarily intended to avoid unlimited
+    accumulation of the delta-coded noise levels.
+  */
+  #define lowerLimit   ((FIXP_SGL)0)     /* lowerLimit actually refers to the _highest_ noise energy */
+  #define upperLimit   ((FIXP_SGL)35)    /* upperLimit actually refers to the _lowest_ noise energy */
+
+  /*
+    Range check for current noise levels
+  */
+  for (i = 0; i < h_sbr_data->frameInfo.nNoiseEnvelopes * nNfb; i++) {
+    h_sbr_data->sbrNoiseFloorLevel[i] = fixMin(h_sbr_data->sbrNoiseFloorLevel[i], upperLimit);
+    h_sbr_data->sbrNoiseFloorLevel[i] = fixMax(h_sbr_data->sbrNoiseFloorLevel[i], lowerLimit);
+  }
+}
+
+
+/*!
+  \brief   Compensate for the wrong timing that might occur after a frame error.
+*/
+static void
+timeCompensateFirstEnvelope (HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */
+                             HANDLE_SBR_FRAME_DATA h_sbr_data,   /*!< pointer to actual data */
+                             HANDLE_SBR_PREV_FRAME_DATA h_prev_data)  /*!< pointer to data of last frame */
+{
+  int i, nScalefactors;
+  FRAME_INFO *pFrameInfo = &h_sbr_data->frameInfo;
+  UCHAR *nSfb = hHeaderData->freqBandData.nSfb;
+  int estimatedStartPos = h_prev_data->stopPos - hHeaderData->numberTimeSlots;
+  int refLen, newLen, shift;
+  FIXP_SGL  deltaExp;
+
+  /* Original length of first envelope according to bitstream */
+  refLen = pFrameInfo->borders[1] - pFrameInfo->borders[0];
+  /* Corrected length of first envelope (concealing can make the first envelope longer) */
+  newLen = pFrameInfo->borders[1] - estimatedStartPos;
+
+  if (newLen <= 0) {
+    /* An envelope length of <= 0 would not work, so we don't use it.
+       May occur if the previous frame was flagged bad due to a mismatch
+       of the old and new frame infos. */
+    newLen = refLen;
+    estimatedStartPos = pFrameInfo->borders[0];
+  }
+
+  deltaExp = FDK_getNumOctavesDiv8(newLen, refLen);
+
+  /* Shift by -3 to rescale ld-table, 1-ampRes to enable coarser steps */
+  shift = (FRACT_BITS - 1 - ENV_EXP_FRACT + 1 - h_sbr_data->ampResolutionCurrentFrame - 3);
+  deltaExp = deltaExp >> shift;
+  pFrameInfo->borders[0] = estimatedStartPos;
+  pFrameInfo->bordersNoise[0] = estimatedStartPos;
+
+  if (h_sbr_data->coupling != COUPLING_BAL) {
+    nScalefactors = (pFrameInfo->freqRes[0]) ? nSfb[1] : nSfb[0];
+
+    for (i = 0; i < nScalefactors; i++)
+      h_sbr_data->iEnvelope[i] = h_sbr_data->iEnvelope[i] + deltaExp;
+  }
+}
+
+
+
+/*!
+  \brief   Convert each envelope value from logarithmic to linear domain
+
+  Energy levels are transmitted in powers of 2, i.e. only the exponent
+  is extracted from the bitstream.
+  Therefore, normally only integer exponents can occur. However during
+  fading (in case of a corrupt bitstream), a fractional part can also
+  occur. The data in the array iEnvelope is shifted left by ENV_EXP_FRACT
+  compared to an integer representation so that numbers smaller than 1
+  can be represented.
+
+  This function calculates a mantissa corresponding to the fractional
+  part of the exponent for each reference energy. The array iEnvelope
+  is converted in place to save memory. Input and output data must
+  be interpreted differently, as shown in the below figure:
+
+  \image html  EnvelopeData.png
+
+  The data is then used in calculateSbrEnvelope().
+*/
+static void
+requantizeEnvelopeData (HANDLE_SBR_FRAME_DATA h_sbr_data, int ampResolution)
+{
+  int i;
+  FIXP_SGL mantissa;
+  int ampShift = 1 - ampResolution;
+  int exponent;
+
+  /* In case that ENV_EXP_FRACT is changed to something else but 0 or 8,
+     the initialization of this array has to be adapted!
+  */
+#if ENV_EXP_FRACT
+  static const FIXP_SGL pow2[ENV_EXP_FRACT] =
+  {
+    FL2FXCONST_SGL(0.5f * pow(2.0f, pow(0.5f, 1))), /* 0.7071 */
+    FL2FXCONST_SGL(0.5f * pow(2.0f, pow(0.5f, 2))), /* 0.5946 */
+    FL2FXCONST_SGL(0.5f * pow(2.0f, pow(0.5f, 3))),
+    FL2FXCONST_SGL(0.5f * pow(2.0f, pow(0.5f, 4))),
+    FL2FXCONST_SGL(0.5f * pow(2.0f, pow(0.5f, 5))),
+    FL2FXCONST_SGL(0.5f * pow(2.0f, pow(0.5f, 6))),
+    FL2FXCONST_SGL(0.5f * pow(2.0f, pow(0.5f, 7))),
+    FL2FXCONST_SGL(0.5f * pow(2.0f, pow(0.5f, 8)))  /* 0.5013 */
+  };
+
+  int bit, mask;
+#endif
+
+  for (i = 0; i < h_sbr_data->nScaleFactors; i++) {
+    exponent = (LONG)h_sbr_data->iEnvelope[i];
+
+#if ENV_EXP_FRACT
+
+    exponent = exponent >> ampShift;
+    mantissa = 0.5f;
+
+    /* Amplify mantissa according to the fractional part of the
+       exponent (result will be between 0.500000 and 0.999999)
+    */
+    mask = 1;  /* begin with lowest bit of exponent */
+
+    for ( bit=ENV_EXP_FRACT-1; bit>=0; bit-- ) {
+      if (exponent & mask) {
+        /* The current bit of the exponent is set,
+           multiply mantissa with the corresponding factor: */
+        mantissa = (FIXP_SGL)( (mantissa * pow2[bit]) << 1);
+      }
+      /* Advance to next bit */
+      mask = mask << 1;
+    }
+
+    /* Make integer part of exponent right aligned */
+    exponent = exponent >> ENV_EXP_FRACT;
+
+#else
+    /* In case of the high amplitude resolution, 1 bit of the exponent gets lost by the shift.
+       This will be compensated by a mantissa of 0.5*sqrt(2) instead of 0.5 if that bit is 1. */
+    mantissa = (exponent & ampShift) ? FL2FXCONST_SGL(0.707106781186548f) : FL2FXCONST_SGL(0.5f);
+    exponent = exponent >> ampShift;
+#endif
+
+    /*
+      Mantissa was set to 0.5 (instead of 1.0, therefore increase exponent by 1).
+      Multiply by L=nChannels=64 by increasing exponent by another 6.
+      => Increase exponent by 7
+    */
+    exponent += 7 + NRG_EXP_OFFSET;
+
+    /* Combine mantissa and exponent and write back the result */
+    h_sbr_data->iEnvelope[i] = (FIXP_SGL)(((LONG)mantissa & MASK_M) | (exponent & MASK_E));
+
+  }
+}
+
+
+/*!
+  \brief   Build new reference energies from old ones and delta coded data
+*/
+static void
+deltaToLinearPcmEnvelopeDecoding (HANDLE_SBR_HEADER_DATA hHeaderData,     /*!< Static control data */
+                                  HANDLE_SBR_FRAME_DATA h_sbr_data,       /*!< pointer to current data */
+                                  HANDLE_SBR_PREV_FRAME_DATA h_prev_data) /*!< pointer to previous data */
+{
+  int i, domain, no_of_bands, band, freqRes;
+
+  FIXP_SGL *sfb_nrg_prev = h_prev_data->sfb_nrg_prev;
+  FIXP_SGL *ptr_nrg = h_sbr_data->iEnvelope;
+
+  int offset = 2 * hHeaderData->freqBandData.nSfb[0] - hHeaderData->freqBandData.nSfb[1];
+
+  for (i = 0; i < h_sbr_data->frameInfo.nEnvelopes; i++) {
+    domain = h_sbr_data->domain_vec[i];
+    freqRes = h_sbr_data->frameInfo.freqRes[i];
+
+    FDK_ASSERT(freqRes >= 0 && freqRes <= 1);
+
+    no_of_bands = hHeaderData->freqBandData.nSfb[freqRes];
+
+    FDK_ASSERT(no_of_bands < (64));
+
+    if (domain == 0)
+    {
+      mapLowResEnergyVal(*ptr_nrg, sfb_nrg_prev, offset, 0, freqRes);
+      ptr_nrg++;
+      for (band = 1; band < no_of_bands; band++)
+      {
+        *ptr_nrg = *ptr_nrg + *(ptr_nrg-1);
+        mapLowResEnergyVal(*ptr_nrg, sfb_nrg_prev, offset, band, freqRes);
+        ptr_nrg++;
+      }
+    }
+    else
+    {
+      for (band = 0; band < no_of_bands; band++)
+      {
+        *ptr_nrg = *ptr_nrg + sfb_nrg_prev[indexLow2High(offset, band, freqRes)];
+        mapLowResEnergyVal(*ptr_nrg, sfb_nrg_prev, offset, band, freqRes);
+        ptr_nrg++;
+      }
+    }
+  }
+}
+
+
+/*!
+  \brief   Build new noise levels from old ones and delta coded data
+*/
+static void
+decodeNoiseFloorlevels (HANDLE_SBR_HEADER_DATA hHeaderData,     /*!< Static control data */
+                        HANDLE_SBR_FRAME_DATA h_sbr_data,       /*!< pointer to current data */
+                        HANDLE_SBR_PREV_FRAME_DATA h_prev_data) /*!< pointer to previous data */
+{
+  int i;
+  int nNfb = hHeaderData->freqBandData.nNfb;
+  int nNoiseFloorEnvelopes = h_sbr_data->frameInfo.nNoiseEnvelopes;
+
+  /* Decode first noise envelope */
+
+  if (h_sbr_data->domain_vec_noise[0] == 0) {
+    FIXP_SGL noiseLevel = h_sbr_data->sbrNoiseFloorLevel[0];
+    for (i = 1; i < nNfb; i++) {
+      noiseLevel += h_sbr_data->sbrNoiseFloorLevel[i];
+      h_sbr_data->sbrNoiseFloorLevel[i] = noiseLevel;
+    }
+  }
+  else {
+    for (i = 0; i < nNfb; i++) {
+      h_sbr_data->sbrNoiseFloorLevel[i] += h_prev_data->prevNoiseLevel[i];
+    }
+  }
+
+  /* If present, decode the second noise envelope
+     Note:  nNoiseFloorEnvelopes can only be 1 or 2 */
+
+  if (nNoiseFloorEnvelopes > 1) {
+    if (h_sbr_data->domain_vec_noise[1] == 0) {
+      FIXP_SGL noiseLevel = h_sbr_data->sbrNoiseFloorLevel[nNfb];
+      for (i = nNfb + 1; i < 2*nNfb; i++) {
+        noiseLevel += h_sbr_data->sbrNoiseFloorLevel[i];
+        h_sbr_data->sbrNoiseFloorLevel[i] = noiseLevel;
+      }
+    }
+    else {
+      for (i = 0; i < nNfb; i++) {
+        h_sbr_data->sbrNoiseFloorLevel[i + nNfb] += h_sbr_data->sbrNoiseFloorLevel[i];
+      }
+    }
+  }
+
+  limitNoiseLevels(hHeaderData, h_sbr_data);
+
+  /* Update prevNoiseLevel with the last noise envelope */
+  for (i = 0; i < nNfb; i++)
+    h_prev_data->prevNoiseLevel[i] = h_sbr_data->sbrNoiseFloorLevel[i + nNfb*(nNoiseFloorEnvelopes-1)];
+
+
+  /* Requantize the noise floor levels in COUPLING_OFF-mode */
+  if (!h_sbr_data->coupling) {
+    int nf_e;
+
+    for (i = 0; i < nNoiseFloorEnvelopes*nNfb; i++) {
+      nf_e = 6 - (LONG)h_sbr_data->sbrNoiseFloorLevel[i] + 1 + NOISE_EXP_OFFSET;
+      /* +1 to compensate for a mantissa of 0.5 instead of 1.0 */
+
+      h_sbr_data->sbrNoiseFloorLevel[i] =
+        (FIXP_SGL)( ((LONG)FL2FXCONST_SGL(0.5f)) +  /* mantissa */
+                  (nf_e & MASK_E) ); /* exponent */
+
+    }
+  }
+}
diff --git a/libSBRdec/src/env_dec.h b/libSBRdec/src/env_dec.h
new file mode 100644
index 0000000..3e656ed
--- /dev/null
+++ b/libSBRdec/src/env_dec.h
@@ -0,0 +1,101 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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 decoding  
+*/
+#ifndef __ENV_DEC_H
+#define __ENV_DEC_H
+
+#include "sbrdecoder.h"
+#include "env_extr.h"
+
+void decodeSbrData (HANDLE_SBR_HEADER_DATA hHeaderData,
+                    HANDLE_SBR_FRAME_DATA h_data_left,
+                    HANDLE_SBR_PREV_FRAME_DATA h_prev_data_left,
+                    HANDLE_SBR_FRAME_DATA h_data_right,
+                    HANDLE_SBR_PREV_FRAME_DATA h_prev_data_right);
+
+
+#endif
diff --git a/libSBRdec/src/env_extr.cpp b/libSBRdec/src/env_extr.cpp
new file mode 100644
index 0000000..716fb91
--- /dev/null
+++ b/libSBRdec/src/env_extr.cpp
@@ -0,0 +1,1395 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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 extraction  
+  The functions provided by this module are mostly called by applySBR(). After it is
+  determined that there is valid SBR data, sbrGetHeaderData() might be called if the current
+  SBR data contains an \ref SBR_HEADER_ELEMENT as opposed to a \ref SBR_STANDARD_ELEMENT. This function
+  may return various error codes as defined in #SBR_HEADER_STATUS . Most importantly it returns HEADER_RESET when decoder
+  settings need to be recalculated according to the SBR specifications. In that case applySBR()
+  will initiatite the required re-configuration.
+
+  The header data is stored in a #SBR_HEADER_DATA structure.
+
+  The actual SBR data for the current frame is decoded into SBR_FRAME_DATA stuctures by sbrGetChannelPairElement()
+  [for stereo streams] and sbrGetSingleChannelElement() [for mono streams]. There is no fractional arithmetic involved.
+
+  Once the information is extracted, the data needs to be further prepared before the actual decoding process.
+  This is done in decodeSbrData().
+
+  \sa Description of buffer management in applySBR(). \ref documentationOverview
+
+  <h1>About the SBR data format:</h1>
+
+  Each frame includes SBR data (side chain information), and can be either the \ref SBR_HEADER_ELEMENT or the \ref SBR_STANDARD_ELEMENT.
+  Parts of the data can be protected by a CRC checksum.
+
+  \anchor SBR_HEADER_ELEMENT <h2>The SBR_HEADER_ELEMENT</h2>
+
+  The SBR_HEADER_ELEMENT can be transmitted with every frame, however, it typically is send every second or so. It contains fundamental
+  information such as SBR sampling frequency and frequency range as well as control signals that do not require frequent changes. It also
+  includes the \ref SBR_STANDARD_ELEMENT.
+
+  Depending on the changes between the information in a current SBR_HEADER_ELEMENT and the previous SBR_HEADER_ELEMENT, the SBR decoder might need
+  to be reset and reconfigured (e.g. new tables need to be calculated).
+
+  \anchor SBR_STANDARD_ELEMENT <h2>The SBR_STANDARD_ELEMENT</h2>
+
+  This data can be subdivided into "side info" and "raw data", where side info is defined as signals needed to decode the raw data
+  and some decoder tuning signals. Raw data is referred to as PCM and Huffman coded envelope and noise floor estimates. The side info also
+  includes information about the time-frequency grid for the current frame.
+
+  \sa \ref documentationOverview
+*/
+
+#include "env_extr.h"
+
+#include "sbr_ram.h"
+#include "sbr_rom.h"
+#include "huff_dec.h"
+
+
+#include "psbitdec.h"
+
+#define DRM_PARAMETRIC_STEREO   0
+#define EXTENSION_ID_PS_CODING  2
+
+
+static int extractFrameInfo (HANDLE_FDK_BITSTREAM   hBs,
+                             HANDLE_SBR_HEADER_DATA hHeaderData,
+                             HANDLE_SBR_FRAME_DATA  h_frame_data,
+                             const UINT             nrOfChannels,
+                             const UINT             flags
+                            );
+
+
+static int sbrGetEnvelope (HANDLE_SBR_HEADER_DATA hHeaderData,
+                           HANDLE_SBR_FRAME_DATA  h_frame_data,
+                           HANDLE_FDK_BITSTREAM   hBs,
+                           const UINT flags);
+
+static void sbrGetDirectionControlData (HANDLE_SBR_FRAME_DATA hFrameData,
+                                        HANDLE_FDK_BITSTREAM hBs);
+
+static void sbrGetNoiseFloorData (HANDLE_SBR_HEADER_DATA hHeaderData,
+                                  HANDLE_SBR_FRAME_DATA  h_frame_data,
+                                  HANDLE_FDK_BITSTREAM   hBs);
+
+static int checkFrameInfo (FRAME_INFO *pFrameInfo, int numberOfTimeSlots, int overlap, int timeStep);
+
+SBR_ERROR
+initHeaderData (
+        HANDLE_SBR_HEADER_DATA  hHeaderData,
+        const int               sampleRateIn,
+        const int               sampleRateOut,
+        const int               samplesPerFrame,
+        const UINT              flags
+        )
+{
+  HANDLE_FREQ_BAND_DATA hFreq = &hHeaderData->freqBandData;
+  SBR_ERROR sbrError = SBRDEC_OK;
+  int numAnalysisBands;
+
+  if ( sampleRateIn == sampleRateOut ) {
+    hHeaderData->sbrProcSmplRate = sampleRateOut<<1;
+    numAnalysisBands = 32;
+  } else {
+    hHeaderData->sbrProcSmplRate = sampleRateOut;
+    if ( (sampleRateOut>>1) == sampleRateIn) {
+      /* 1:2 */
+      numAnalysisBands = 32;
+    } else if ( (sampleRateOut>>2) == sampleRateIn ) {
+      /* 1:4 */
+      numAnalysisBands = 32;
+    } else if ( (sampleRateOut*3)>>3 == (sampleRateIn*8)>>3 ) {
+      /* 3:8, 3/4 core frame length */
+      numAnalysisBands = 24;
+    } else {
+      sbrError = SBRDEC_UNSUPPORTED_CONFIG;
+      goto bail;
+    }
+  }
+
+  /* Fill in default values first */
+  hHeaderData->syncState          = SBR_NOT_INITIALIZED;
+  hHeaderData->status             = 0;
+  hHeaderData->frameErrorFlag     = 0;
+
+  hHeaderData->bs_info.ampResolution     = 1;
+  hHeaderData->bs_info.xover_band        = 0;
+  hHeaderData->bs_info.sbr_preprocessing = 0;
+
+  hHeaderData->bs_data.startFreq       = 5;
+  hHeaderData->bs_data.stopFreq        = 0;
+  hHeaderData->bs_data.freqScale       = 2;
+  hHeaderData->bs_data.alterScale      = 1;
+  hHeaderData->bs_data.noise_bands     = 2;
+  hHeaderData->bs_data.limiterBands    = 2;
+  hHeaderData->bs_data.limiterGains    = 2;
+  hHeaderData->bs_data.interpolFreq    = 1;
+  hHeaderData->bs_data.smoothingLength = 1;
+
+  hHeaderData->timeStep = (flags & SBRDEC_ELD_GRID) ? 1 : 2;
+
+  /* Setup pointers to frequency band tables */
+  hFreq->freqBandTable[0]  = hFreq->freqBandTableLo;
+  hFreq->freqBandTable[1] = hFreq->freqBandTableHi;
+
+  /* Patch some entries */
+  if (sampleRateOut > 24000) {    /* Trigger an error if SBR is going to be processed without     */
+    hHeaderData->bs_data.startFreq = 7;   /*   having read these frequency values from bit stream before. */
+    hHeaderData->bs_data.stopFreq  = 3;
+  }
+
+  /* One SBR timeslot corresponds to the amount of samples equal to the amount of analysis bands, divided by the timestep. */
+  hHeaderData->numberTimeSlots = (samplesPerFrame/numAnalysisBands) >> (hHeaderData->timeStep - 1);
+  if (hHeaderData->numberTimeSlots > (16)) {
+    sbrError = SBRDEC_UNSUPPORTED_CONFIG;
+  }
+
+  hHeaderData->numberOfAnalysisBands = numAnalysisBands;
+
+bail:
+  return sbrError;
+}
+
+
+/*!
+  \brief   Initialize the SBR_PREV_FRAME_DATA struct
+*/
+void
+initSbrPrevFrameData (HANDLE_SBR_PREV_FRAME_DATA h_prev_data, /*!< handle to struct SBR_PREV_FRAME_DATA */
+                      int timeSlots)                          /*!< Framelength in SBR-timeslots */
+{
+  int i;
+
+  /* Set previous energy and noise levels to 0 for the case
+     that decoding starts in the middle of a bitstream */
+  for (i=0; i < MAX_FREQ_COEFFS; i++)
+    h_prev_data->sfb_nrg_prev[i] = (FIXP_DBL)0;
+  for (i=0; i < MAX_NOISE_COEFFS; i++)
+    h_prev_data->prevNoiseLevel[i] = (FIXP_DBL)0;
+  for (i=0; i < MAX_INVF_BANDS; i++)
+    h_prev_data->sbr_invf_mode[i] = INVF_OFF;
+
+  h_prev_data->stopPos = timeSlots;
+  h_prev_data->coupling = COUPLING_OFF;
+  h_prev_data->ampRes = 0;
+}
+
+
+/*!
+  \brief   Read header data from bitstream
+
+  \return  error status - 0 if ok
+*/
+SBR_HEADER_STATUS
+sbrGetHeaderData (HANDLE_SBR_HEADER_DATA hHeaderData,
+                  HANDLE_FDK_BITSTREAM   hBs,
+                  const UINT             flags,
+                  const int              fIsSbrData)
+{
+  SBR_HEADER_DATA_BS *pBsData;
+  SBR_HEADER_DATA_BS lastHeader;
+  SBR_HEADER_DATA_BS_INFO lastInfo;
+  int headerExtra1=0, headerExtra2=0;
+
+  /* Copy SBR bit stream header to temporary header */
+  lastHeader = hHeaderData->bs_data;
+  lastInfo   = hHeaderData->bs_info;
+
+  /* Read new header from bitstream */
+  {
+    pBsData = &hHeaderData->bs_data;
+  }
+
+  {
+    hHeaderData->bs_info.ampResolution = FDKreadBits (hBs, 1);
+  }
+
+  pBsData->startFreq = FDKreadBits (hBs, 4);
+  pBsData->stopFreq = FDKreadBits (hBs, 4);
+
+  {
+    hHeaderData->bs_info.xover_band = FDKreadBits (hBs, 3);
+    FDKreadBits (hBs, 2);
+  }
+
+  headerExtra1 = FDKreadBits (hBs, 1);
+  headerExtra2 = FDKreadBits (hBs, 1);
+
+  /* Handle extra header information */
+  if( headerExtra1) 
+  {
+    pBsData->freqScale = FDKreadBits (hBs, 2);
+    pBsData->alterScale = FDKreadBits (hBs, 1);
+    pBsData->noise_bands = FDKreadBits (hBs, 2);
+  }
+  else {
+    pBsData->freqScale   = 2;
+    pBsData->alterScale  = 1;
+    pBsData->noise_bands = 2;
+  }
+
+  if (headerExtra2) {
+    pBsData->limiterBands = FDKreadBits (hBs, 2);
+    pBsData->limiterGains = FDKreadBits (hBs, 2);
+    pBsData->interpolFreq = FDKreadBits (hBs, 1);
+    pBsData->smoothingLength = FDKreadBits (hBs, 1);
+  }
+  else {
+    pBsData->limiterBands    = 2;
+    pBsData->limiterGains    = 2;
+    pBsData->interpolFreq    = 1;
+    pBsData->smoothingLength = 1;
+  }
+
+  /* Look for new settings. IEC 14496-3, 4.6.18.3.1 */
+  if(hHeaderData->syncState != SBR_ACTIVE ||
+     lastHeader.startFreq   != pBsData->startFreq   ||
+     lastHeader.stopFreq    != pBsData->stopFreq    ||
+     lastHeader.freqScale   != pBsData->freqScale   ||
+     lastHeader.alterScale  != pBsData->alterScale  ||
+     lastHeader.noise_bands != pBsData->noise_bands ||
+     lastInfo.xover_band    != hHeaderData->bs_info.xover_band) {
+    return HEADER_RESET; /* New settings */
+  }
+
+  return HEADER_OK;
+}
+
+/*!
+  \brief   Get missing harmonics parameters (only used for AAC+SBR)
+
+  \return  error status - 0 if ok
+*/
+int
+sbrGetSyntheticCodedData(HANDLE_SBR_HEADER_DATA hHeaderData,
+                         HANDLE_SBR_FRAME_DATA  hFrameData,
+                         HANDLE_FDK_BITSTREAM   hBs)
+{
+  int i, bitsRead = 0;
+
+  int flag = FDKreadBits(hBs,1);
+  bitsRead++;
+
+  if(flag){
+    for(i=0;i<hHeaderData->freqBandData.nSfb[1];i++){
+      hFrameData->addHarmonics[i]  = FDKreadBits (hBs, 1 );
+      bitsRead++;
+    }
+  }
+  else {
+    for(i=0; i<MAX_FREQ_COEFFS; i++)
+      hFrameData->addHarmonics[i]  = 0;
+  }
+  return(bitsRead);
+}
+
+/*!
+  \brief      Reads extension data from the bitstream
+
+  The bitstream format allows up to 4 kinds of extended data element.
+  Extended data may contain several elements, each identified by a 2-bit-ID.
+  So far, no extended data elements are defined hence the first 2 parameters
+  are unused. The data should be skipped in order to update the number
+  of read bits for the consistency check in applySBR().
+*/
+static int  extractExtendedData(
+                                HANDLE_SBR_HEADER_DATA hHeaderData,    /*!< handle to SBR header */
+                                HANDLE_FDK_BITSTREAM   hBs             /*!< Handle to the bit buffer */
+                               ,HANDLE_PS_DEC hParametricStereoDec     /*!< Parametric Stereo Decoder */
+                                ) {
+  INT nBitsLeft;
+  int extended_data;
+  int i, frameOk = 1;
+
+
+  extended_data = FDKreadBits(hBs, 1);
+
+  if (extended_data) {
+    int cnt;
+    int bPsRead = 0;
+
+    cnt = FDKreadBits(hBs, 4);
+    if (cnt == (1<<4)-1)
+      cnt += FDKreadBits(hBs, 8);
+
+
+    nBitsLeft = 8 * cnt;
+
+    /* sanity check for cnt */
+    if (nBitsLeft > (INT)FDKgetValidBits(hBs)) {
+      /* limit nBitsLeft */
+      nBitsLeft = (INT)FDKgetValidBits(hBs);
+      /* set frame error */
+      frameOk = 0;
+    }
+
+    while (nBitsLeft > 7) {
+      int extension_id = FDKreadBits(hBs, 2);
+      nBitsLeft -= 2;
+
+      switch(extension_id) {
+
+
+
+        case EXTENSION_ID_PS_CODING:
+
+        /* Read PS data from bitstream */
+
+        if (hParametricStereoDec != NULL) {
+          if(bPsRead && !hParametricStereoDec->bsData[hParametricStereoDec->bsReadSlot].mpeg.bPsHeaderValid) {
+            cnt = nBitsLeft >> 3; /* number of remaining bytes */
+            for (i=0; i<cnt; i++)
+              FDKreadBits(hBs, 8);
+            nBitsLeft -= cnt * 8;
+          } else {
+            nBitsLeft -= ReadPsData(hParametricStereoDec, hBs, nBitsLeft);
+            bPsRead = 1;
+          }
+        }
+
+          /* parametric stereo detected, could set channelMode accordingly here  */
+          /*                                                                     */
+          /* "The usage of this parametric stereo extension to HE-AAC is         */
+          /* signalled implicitly in the bitstream. Hence, if an sbr_extension() */
+          /* with bs_extension_id==EXTENSION_ID_PS is found in the SBR part of   */
+          /* the bitstream, a decoder supporting the combination of SBR and PS   */
+          /* shall operate the PS tool to generate a stereo output signal."      */
+          /* source: ISO/IEC 14496-3:2001/FDAM 2:2004(E)                         */
+
+        break;
+
+
+      default:
+        cnt = nBitsLeft >> 3; /* number of remaining bytes */
+        for (i=0; i<cnt; i++)
+          FDKreadBits(hBs, 8);
+        nBitsLeft -= cnt * 8;
+        break;
+      }
+    }
+
+    if (nBitsLeft < 0) {
+      frameOk = 0;
+      goto bail;
+    }
+    else {
+      /* Read fill bits for byte alignment */
+      FDKreadBits(hBs, nBitsLeft);
+    }
+  }
+
+bail:
+  return (frameOk);
+}
+
+
+/*!
+  \brief   Read bitstream elements of one channel
+
+  \return  SbrFrameOK:  1=ok, 0=error
+*/
+int
+sbrGetSingleChannelElement (HANDLE_SBR_HEADER_DATA hHeaderData,          /*!< Static control data */
+                            HANDLE_SBR_FRAME_DATA  hFrameData,           /*!< Control data of current frame */
+                            HANDLE_FDK_BITSTREAM   hBs,                  /*!< Handle to struct BIT_BUF */
+                            HANDLE_PS_DEC          hParametricStereoDec, /*!< Handle to PS decoder */
+                            const UINT             flags,
+                            const int              overlap
+                           )
+{
+  int i;
+
+
+  hFrameData->coupling = COUPLING_OFF;
+
+  {
+    /* Reserved bits */
+    if (FDKreadBits(hBs, 1)) {  /* bs_data_extra */
+      FDKreadBits(hBs, 4);
+      if (flags & SBRDEC_SYNTAX_SCAL) {
+        FDKreadBits(hBs, 4);
+      }
+    }
+  }
+
+  if (flags & SBRDEC_SYNTAX_SCAL) {
+    FDKreadBits (hBs, 1);     /* bs_coupling */
+  }
+
+  /*
+    Grid control
+  */
+  if ( !extractFrameInfo ( hBs, hHeaderData, hFrameData, 1, flags) )
+   return 0;
+
+  if ( !checkFrameInfo (&hFrameData->frameInfo, hHeaderData->numberTimeSlots, overlap, hHeaderData->timeStep) )
+    return 0;
+
+
+  /*
+    Fetch domain vectors (time or frequency direction for delta-coding)
+  */
+  sbrGetDirectionControlData (hFrameData, hBs);
+
+  for (i=0; i<hHeaderData->freqBandData.nInvfBands; i++) {
+    hFrameData->sbr_invf_mode[i] =
+      (INVF_MODE) FDKreadBits (hBs, 2);
+  }
+
+
+
+  /* raw data */
+  if ( !sbrGetEnvelope (hHeaderData, hFrameData, hBs, flags) )
+    return 0;
+
+
+  sbrGetNoiseFloorData (hHeaderData, hFrameData, hBs);
+
+  sbrGetSyntheticCodedData(hHeaderData, hFrameData, hBs);
+
+  {
+    /* sbr extended data */
+    if (! extractExtendedData(
+                               hHeaderData,
+                               hBs
+                              ,hParametricStereoDec
+                               )) {
+      return 0;
+    }
+  }
+
+  return 1;
+}
+
+
+
+/*!
+  \brief      Read bitstream elements of a channel pair
+  \return     SbrFrameOK
+*/
+int
+sbrGetChannelPairElement (HANDLE_SBR_HEADER_DATA hHeaderData,    /*!< Static control data */
+                          HANDLE_SBR_FRAME_DATA  hFrameDataLeft, /*!< Dynamic control data for first channel */
+                          HANDLE_SBR_FRAME_DATA  hFrameDataRight,/*!< Dynamic control data for second channel */
+                          HANDLE_FDK_BITSTREAM   hBs,            /*!< handle to struct BIT_BUF */
+                          const UINT flags,
+                          const int overlap )
+{
+  int i, bit;
+
+
+  /* Reserved bits */
+  if (FDKreadBits(hBs, 1)) {  /* bs_data_extra */
+    FDKreadBits(hBs, 4);
+    FDKreadBits(hBs, 4);
+  }
+
+  /* Read coupling flag */
+  bit = FDKreadBits (hBs, 1);
+
+  if (bit) {
+    hFrameDataLeft->coupling = COUPLING_LEVEL;
+    hFrameDataRight->coupling = COUPLING_BAL;
+  }
+  else {
+    hFrameDataLeft->coupling = COUPLING_OFF;
+    hFrameDataRight->coupling = COUPLING_OFF;
+  }
+
+
+  /*
+    Grid control
+  */
+  if ( !extractFrameInfo (hBs, hHeaderData, hFrameDataLeft, 2, flags) )
+    return 0;
+
+  if ( !checkFrameInfo (&hFrameDataLeft->frameInfo, hHeaderData->numberTimeSlots, overlap, hHeaderData->timeStep) )
+    return 0;
+
+  if (hFrameDataLeft->coupling) {
+    FDKmemcpy (&hFrameDataRight->frameInfo, &hFrameDataLeft->frameInfo, sizeof(FRAME_INFO));
+    hFrameDataRight->ampResolutionCurrentFrame = hFrameDataLeft->ampResolutionCurrentFrame;
+  }
+  else {
+    if ( !extractFrameInfo (hBs, hHeaderData, hFrameDataRight, 2, flags) )
+      return 0;
+
+    if ( !checkFrameInfo (&hFrameDataRight->frameInfo, hHeaderData->numberTimeSlots, overlap, hHeaderData->timeStep) )
+      return 0;
+  }
+
+  /*
+    Fetch domain vectors (time or frequency direction for delta-coding)
+  */
+  sbrGetDirectionControlData (hFrameDataLeft, hBs);
+  sbrGetDirectionControlData (hFrameDataRight, hBs);
+
+  for (i=0; i<hHeaderData->freqBandData.nInvfBands; i++) {
+    hFrameDataLeft->sbr_invf_mode[i] = (INVF_MODE) FDKreadBits (hBs, 2);
+  }
+
+  if (hFrameDataLeft->coupling) {
+    for (i=0; i<hHeaderData->freqBandData.nInvfBands; i++) {
+      hFrameDataRight->sbr_invf_mode[i] = hFrameDataLeft->sbr_invf_mode[i];
+    }
+
+
+    if ( !sbrGetEnvelope (hHeaderData, hFrameDataLeft, hBs, flags) ) {
+      return 0;
+    }
+
+    sbrGetNoiseFloorData (hHeaderData, hFrameDataLeft, hBs);
+
+    if ( !sbrGetEnvelope (hHeaderData, hFrameDataRight, hBs, flags) ) {
+      return 0;
+    }
+  }
+  else {
+
+    for (i=0; i<hHeaderData->freqBandData.nInvfBands; i++) {
+      hFrameDataRight->sbr_invf_mode[i] = (INVF_MODE) FDKreadBits (hBs, 2);
+    }
+
+
+
+    if ( !sbrGetEnvelope (hHeaderData, hFrameDataLeft, hBs, flags) )
+      return 0;
+
+    if ( !sbrGetEnvelope (hHeaderData, hFrameDataRight, hBs, flags) )
+      return 0;
+
+    sbrGetNoiseFloorData (hHeaderData, hFrameDataLeft, hBs);
+
+  }
+  sbrGetNoiseFloorData (hHeaderData, hFrameDataRight, hBs);
+
+  sbrGetSyntheticCodedData(hHeaderData, hFrameDataLeft, hBs);
+  sbrGetSyntheticCodedData(hHeaderData, hFrameDataRight, hBs);
+
+  {
+    if (! extractExtendedData(
+                               hHeaderData,
+                               hBs
+                              ,NULL
+                             ) ) {
+      return 0;
+    }
+  }
+
+  return 1;
+}
+
+
+
+
+/*!
+  \brief   Read direction control data from bitstream
+*/
+void
+sbrGetDirectionControlData (HANDLE_SBR_FRAME_DATA h_frame_data, /*!< handle to struct SBR_FRAME_DATA */
+                            HANDLE_FDK_BITSTREAM  hBs)          /*!< handle to struct BIT_BUF */
+{
+  int i;
+
+  for (i = 0; i < h_frame_data->frameInfo.nEnvelopes; i++) {
+    h_frame_data->domain_vec[i] = FDKreadBits (hBs, 1);
+  }
+
+  for (i = 0; i < h_frame_data->frameInfo.nNoiseEnvelopes; i++) {
+    h_frame_data->domain_vec_noise[i] = FDKreadBits (hBs, 1);
+  }
+}
+
+
+
+/*!
+  \brief   Read noise-floor-level data from bitstream
+*/
+void
+sbrGetNoiseFloorData (HANDLE_SBR_HEADER_DATA hHeaderData,  /*!< Static control data */
+                      HANDLE_SBR_FRAME_DATA  h_frame_data, /*!< handle to struct SBR_FRAME_DATA */
+                      HANDLE_FDK_BITSTREAM   hBs)          /*!< handle to struct BIT_BUF */
+{
+  int i,j;
+  int delta;
+  COUPLING_MODE coupling;
+  int noNoiseBands = hHeaderData->freqBandData.nNfb;
+
+  Huffman hcb_noiseF;
+  Huffman hcb_noise;
+  int envDataTableCompFactor;
+
+  coupling = h_frame_data->coupling;
+
+
+  /*
+    Select huffman codebook depending on coupling mode
+  */
+  if (coupling == COUPLING_BAL) {
+    hcb_noise = (Huffman)&FDK_sbrDecoder_sbr_huffBook_NoiseBalance11T;
+    hcb_noiseF = (Huffman)&FDK_sbrDecoder_sbr_huffBook_EnvBalance11F;  /* "sbr_huffBook_NoiseBalance11F" */
+    envDataTableCompFactor = 1;
+  }
+  else {
+    hcb_noise = (Huffman)&FDK_sbrDecoder_sbr_huffBook_NoiseLevel11T;
+    hcb_noiseF = (Huffman)&FDK_sbrDecoder_sbr_huffBook_EnvLevel11F;    /* "sbr_huffBook_NoiseLevel11F" */
+    envDataTableCompFactor = 0;
+  }
+
+  /*
+    Read raw noise-envelope data
+  */
+  for (i=0; i<h_frame_data->frameInfo.nNoiseEnvelopes; i++) {
+
+
+    if (h_frame_data->domain_vec_noise[i] == 0) {
+      if (coupling == COUPLING_BAL) {
+        h_frame_data->sbrNoiseFloorLevel[i*noNoiseBands] =
+          (FIXP_SGL) (((int)FDKreadBits (hBs, 5)) << envDataTableCompFactor);
+      }
+      else {
+        h_frame_data->sbrNoiseFloorLevel[i*noNoiseBands] =
+          (FIXP_SGL) (int)FDKreadBits (hBs, 5);
+      }
+
+      for (j = 1; j < noNoiseBands; j++) {
+        delta = DecodeHuffmanCW(hcb_noiseF, hBs);
+        h_frame_data->sbrNoiseFloorLevel[i*noNoiseBands+j] = (FIXP_SGL) (delta << envDataTableCompFactor);
+      }
+    }
+    else {
+      for (j = 0; j < noNoiseBands; j++) {
+        delta = DecodeHuffmanCW(hcb_noise, hBs);
+        h_frame_data->sbrNoiseFloorLevel[i*noNoiseBands+j] = (FIXP_SGL) (delta << envDataTableCompFactor);
+      }
+    }
+  }
+}
+
+
+/*!
+  \brief   Read envelope data from bitstream
+*/
+static int
+sbrGetEnvelope (HANDLE_SBR_HEADER_DATA hHeaderData,  /*!< Static control data */
+                HANDLE_SBR_FRAME_DATA  h_frame_data, /*!< handle to struct SBR_FRAME_DATA */
+                HANDLE_FDK_BITSTREAM   hBs,          /*!< handle to struct BIT_BUF */
+                const UINT             flags)
+{
+  int i, j;
+  UCHAR no_band[MAX_ENVELOPES];
+  int delta = 0;
+  int offset = 0;
+  COUPLING_MODE coupling = h_frame_data->coupling;
+  int ampRes = hHeaderData->bs_info.ampResolution;
+  int nEnvelopes = h_frame_data->frameInfo.nEnvelopes;
+  int envDataTableCompFactor;
+  int start_bits, start_bits_balance;
+  Huffman hcb_t, hcb_f;
+
+  h_frame_data->nScaleFactors = 0;
+
+  if ( (h_frame_data->frameInfo.frameClass == 0) && (nEnvelopes == 1) ) {
+    if (flags & SBRDEC_ELD_GRID)
+      ampRes = h_frame_data->ampResolutionCurrentFrame;
+    else
+      ampRes = 0;
+  }
+  h_frame_data->ampResolutionCurrentFrame = ampRes;
+
+  /*
+    Set number of bits for first value depending on amplitude resolution
+  */
+  if(ampRes == 1)
+  {
+    start_bits = 6;
+    start_bits_balance = 5;
+  }
+  else
+  {
+    start_bits = 7;
+    start_bits_balance = 6;
+  }
+
+  /*
+    Calculate number of values for each envelope and alltogether
+  */
+  for (i = 0; i < nEnvelopes; i++) {
+    no_band[i] = hHeaderData->freqBandData.nSfb[h_frame_data->frameInfo.freqRes[i]];
+    h_frame_data->nScaleFactors += no_band[i];
+  }
+  if (h_frame_data->nScaleFactors > MAX_NUM_ENVELOPE_VALUES)
+    return 0;
+
+  /*
+    Select Huffman codebook depending on coupling mode and amplitude resolution
+  */
+  if (coupling == COUPLING_BAL) {
+    envDataTableCompFactor = 1;
+    if (ampRes == 0) {
+      hcb_t = (Huffman)&FDK_sbrDecoder_sbr_huffBook_EnvBalance10T;
+      hcb_f = (Huffman)&FDK_sbrDecoder_sbr_huffBook_EnvBalance10F;
+    }
+    else {
+      hcb_t = (Huffman)&FDK_sbrDecoder_sbr_huffBook_EnvBalance11T;
+      hcb_f = (Huffman)&FDK_sbrDecoder_sbr_huffBook_EnvBalance11F;
+    }
+  }
+  else {
+    envDataTableCompFactor = 0;
+    if (ampRes == 0) {
+      hcb_t = (Huffman)&FDK_sbrDecoder_sbr_huffBook_EnvLevel10T;
+      hcb_f = (Huffman)&FDK_sbrDecoder_sbr_huffBook_EnvLevel10F;
+    }
+    else {
+      hcb_t = (Huffman)&FDK_sbrDecoder_sbr_huffBook_EnvLevel11T;
+      hcb_f = (Huffman)&FDK_sbrDecoder_sbr_huffBook_EnvLevel11F;
+    }
+  }
+
+  /*
+    Now read raw envelope data
+  */
+  for (j = 0, offset = 0; j < nEnvelopes; j++) {
+
+
+    if (h_frame_data->domain_vec[j] == 0) {
+      if (coupling == COUPLING_BAL) {
+        h_frame_data->iEnvelope[offset] =
+          (FIXP_SGL) (( (int)FDKreadBits(hBs, start_bits_balance)) << envDataTableCompFactor);
+      }
+      else {
+        h_frame_data->iEnvelope[offset] =
+          (FIXP_SGL) (int)FDKreadBits (hBs, start_bits);
+      }
+    }
+
+    for (i = (1 - h_frame_data->domain_vec[j]); i < no_band[j]; i++) {
+
+      if (h_frame_data->domain_vec[j] == 0) {
+        delta = DecodeHuffmanCW(hcb_f, hBs);
+      }
+      else {
+        delta = DecodeHuffmanCW(hcb_t, hBs);
+      }
+
+      h_frame_data->iEnvelope[offset + i] = (FIXP_SGL) (delta << envDataTableCompFactor);
+    }
+    offset += no_band[j];
+  }
+
+#if ENV_EXP_FRACT
+  /* Convert from int to scaled fract (ENV_EXP_FRACT bits for the fractional part) */
+  for (i = 0; i < h_frame_data->nScaleFactors; i++) {
+    h_frame_data->iEnvelope[i] <<= ENV_EXP_FRACT;
+  }
+#endif
+
+  return 1;
+}
+
+
+//static const FRAME_INFO v_frame_info1_8 = { 0, 1, {0, 8}, {1}, -1, 1, {0, 8} };
+static const FRAME_INFO v_frame_info2_8 = { 0, 2, {0, 4, 8}, {1, 1}, -1, 2, {0, 4, 8} };
+static const FRAME_INFO v_frame_info4_8 = { 0, 4, {0, 2, 4, 6, 8}, {1, 1, 1, 1}, -1, 2, {0, 4, 8} };
+
+/***************************************************************************/
+/*!
+  \brief    Generates frame info for FIXFIXonly frame class used for low delay version
+
+  \return   nothing
+ ****************************************************************************/
+ static void generateFixFixOnly ( FRAME_INFO *hSbrFrameInfo,
+                                  int tranPosInternal,
+                                  int numberTimeSlots
+                                )
+{
+    int nEnv, i, tranIdx;
+    const int *pTable;
+
+    switch (numberTimeSlots) {
+        case 8:
+            pTable = FDK_sbrDecoder_envelopeTable_8[tranPosInternal];
+            break;
+        case 15:
+            pTable = FDK_sbrDecoder_envelopeTable_15[tranPosInternal];
+            break;
+        case 16:
+            pTable = FDK_sbrDecoder_envelopeTable_16[tranPosInternal];
+            break;
+        default:
+            FDK_ASSERT(0);
+    }
+
+    /* look number of envelopes in table */
+    nEnv = pTable[0];
+    /* look up envelope 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;
+    hSbrFrameInfo->nEnvelopes = nEnv;
+
+   /* transient idx */
+    tranIdx = hSbrFrameInfo->tranEnv = pTable[1];
+
+    /* add noise floors */
+    hSbrFrameInfo->bordersNoise[0] = 0;
+    hSbrFrameInfo->bordersNoise[1] = hSbrFrameInfo->borders[tranIdx?tranIdx:1];
+    hSbrFrameInfo->bordersNoise[2] = numberTimeSlots;
+    /* nEnv is always > 1, so nNoiseEnvelopes is always 2 (IEC 14496-3 4.6.19.3.2) */
+    hSbrFrameInfo->nNoiseEnvelopes = 2;
+}
+
+/*!
+  \brief  Extracts LowDelaySBR control data from the bitstream.
+
+  \return zero for bitstream error, one for correct.
+*/
+static int
+extractLowDelayGrid (HANDLE_FDK_BITSTREAM hBitBuf,          /*!< bitbuffer handle */
+                     HANDLE_SBR_HEADER_DATA hHeaderData,
+                     HANDLE_SBR_FRAME_DATA h_frame_data, /*!< contains the FRAME_INFO struct to be filled */
+                     int timeSlots
+                    )
+{
+  FRAME_INFO * pFrameInfo = &h_frame_data->frameInfo;
+  INT numberTimeSlots = hHeaderData->numberTimeSlots;
+  INT temp = 0, k;
+
+      /* FIXFIXonly framing case */
+      h_frame_data->frameInfo.frameClass = 0;
+
+      /* get the transient position from the bitstream */
+      switch (timeSlots){
+        case 8:
+          /* 3bit transient position (temp={0;..;7}) */
+          temp = FDKreadBits( hBitBuf, 3);
+          break;
+
+        case 16:
+        case 15:
+          /* 4bit transient position (temp={0;..;15}) */
+          temp = FDKreadBits( hBitBuf, 4);
+          break;
+
+        default:
+          return 0;
+      }
+
+      /* calculate borders according to the transient position */
+      generateFixFixOnly ( pFrameInfo,
+                           temp,
+                           numberTimeSlots
+                         );
+
+      /* decode freq res: */
+      for (k = 0; k < pFrameInfo->nEnvelopes; k++) {
+          pFrameInfo->freqRes[k] = (UCHAR) FDKreadBits (hBitBuf, 1); /* f = F [1 bits] */          
+      }
+
+
+  return 1;
+}
+
+/*!
+  \brief   Extract the frame information (structure FRAME_INFO) from the bitstream
+  \return  Zero for bitstream error, one for correct.
+*/
+int
+extractFrameInfo ( HANDLE_FDK_BITSTREAM   hBs,          /*!< bitbuffer handle */
+                   HANDLE_SBR_HEADER_DATA hHeaderData,  /*!< Static control data */
+                   HANDLE_SBR_FRAME_DATA  h_frame_data, /*!< pointer to memory where the frame-info will be stored */
+                   const UINT             nrOfChannels,
+                   const UINT             flags
+                 )
+{
+  FRAME_INFO * pFrameInfo = &h_frame_data->frameInfo;
+  int numberTimeSlots = hHeaderData->numberTimeSlots;
+  int pointer_bits = 0, nEnv = 0, b = 0, border, i, n = 0,
+    k, p, aL, aR, nL, nR,
+    temp = 0, staticFreqRes;
+  UCHAR frameClass;
+
+  if (flags & SBRDEC_ELD_GRID) {
+      /* CODEC_AACLD (LD+SBR) only uses the normal 0 Grid for non-transient Frames and the LowDelayGrid for transient Frames */
+      frameClass = FDKreadBits (hBs, 1); /* frameClass = [1 bit] */
+      if ( frameClass == 1 ) {
+        /* if frameClass == 1, extract LowDelaySbrGrid, otherwise extract normal SBR-Grid for FIXIFX */
+        /* extract the AACLD-Sbr-Grid */
+        pFrameInfo->frameClass = frameClass;
+        extractLowDelayGrid (hBs, hHeaderData, h_frame_data, numberTimeSlots);
+        return 1;
+      }
+  } else
+  {
+    frameClass = FDKreadBits (hBs, 2); /* frameClass = C [2 bits] */
+  }
+
+
+  switch (frameClass) {
+  case 0:
+    temp = FDKreadBits (hBs, 2);     /* E [2 bits ] */
+    nEnv = (int) (1 << temp);    /* E -> e */
+
+    if ((flags & SBRDEC_ELD_GRID) && (nEnv == 1))
+      h_frame_data->ampResolutionCurrentFrame = FDKreadBits( hBs, 1); /* new ELD Syntax 07-11-09 */
+
+    staticFreqRes = FDKreadBits (hBs, 1);
+
+    {
+      if (nEnv > MAX_ENVELOPES_HEAAC)
+        return 0;
+    }
+
+    b = nEnv + 1;
+    switch (nEnv) {
+    case 1:
+      switch (numberTimeSlots) {
+        case 15:
+          FDKmemcpy (pFrameInfo, &FDK_sbrDecoder_sbr_frame_info1_15, sizeof(FRAME_INFO));
+          break;
+        case 16:
+          FDKmemcpy (pFrameInfo, &FDK_sbrDecoder_sbr_frame_info1_16, sizeof(FRAME_INFO));
+          break;
+        default:
+          FDK_ASSERT(0);
+      }
+      break;
+    case 2:
+      switch (numberTimeSlots) {
+        case 15:
+          FDKmemcpy (pFrameInfo, &FDK_sbrDecoder_sbr_frame_info2_15, sizeof(FRAME_INFO));
+          break;
+        case 16:
+          FDKmemcpy (pFrameInfo, &FDK_sbrDecoder_sbr_frame_info2_16, sizeof(FRAME_INFO));
+          break;
+        default:
+          FDK_ASSERT(0);
+      }
+      break;
+    case 4:
+      switch (numberTimeSlots) {
+        case 15:
+          FDKmemcpy (pFrameInfo, &FDK_sbrDecoder_sbr_frame_info4_15, sizeof(FRAME_INFO));
+          break;
+        case 16:
+          FDKmemcpy (pFrameInfo, &FDK_sbrDecoder_sbr_frame_info4_16, sizeof(FRAME_INFO));
+          break;
+        default:
+          FDK_ASSERT(0);
+      }
+      break;
+    case 8:
+#if (MAX_ENVELOPES >= 8)
+      switch (numberTimeSlots) {
+        case 15:
+          FDKmemcpy (pFrameInfo, &FDK_sbrDecoder_sbr_frame_info8_15, sizeof(FRAME_INFO));
+          break;
+        case 16:
+          FDKmemcpy (pFrameInfo, &FDK_sbrDecoder_sbr_frame_info8_16, sizeof(FRAME_INFO));
+          break;
+        default:
+          FDK_ASSERT(0);
+      }
+      break;
+#else
+      return 0;
+#endif
+    }
+    /* Apply correct freqRes (High is default) */
+    if (!staticFreqRes) {
+      for (i = 0; i < nEnv ; i++)
+        pFrameInfo->freqRes[i] = 0;
+    }
+
+    break;
+  case 1:
+  case 2:
+    temp = FDKreadBits (hBs, 2);  /* A [2 bits] */
+
+    n    = FDKreadBits (hBs, 2);  /* n = N [2 bits] */
+
+    nEnv = n + 1;                             /* # envelopes */
+    b = nEnv + 1;                             /* # borders   */
+
+    break;
+  }
+
+  switch (frameClass) {
+  case 1:
+    /* Decode borders: */
+    pFrameInfo->borders[0] = 0;               /* first border          */
+    border = temp + numberTimeSlots;          /* A -> aR               */
+    i = b-1;                                  /* frame info index for last border */
+    pFrameInfo->borders[i] = border;          /* last border                      */
+
+    for (k = 0; k < n; k++) {
+      temp = FDKreadBits (hBs, 2);/* R [2 bits] */
+      border -= (2 * temp + 2);               /* R -> r                */
+      pFrameInfo->borders[--i] = border;
+    }
+
+
+    /* Decode pointer: */
+    pointer_bits = DFRACT_BITS - 1 - CountLeadingBits((FIXP_DBL)(n+1));
+    p = FDKreadBits (hBs, pointer_bits);     /* p = P [pointer_bits bits] */
+
+    if (p > n+1)
+      return 0;
+
+    pFrameInfo->tranEnv = p ? n + 2 - p : -1;
+
+
+    /* Decode freq res: */
+    for (k = n; k >= 0; k--) {
+      pFrameInfo->freqRes[k] = FDKreadBits (hBs, 1); /* f = F [1 bits] */
+    }
+
+
+    /* Calculate noise floor middle border: */
+    if (p == 0 || p == 1)
+      pFrameInfo->bordersNoise[1] = pFrameInfo->borders[n];
+    else
+      pFrameInfo->bordersNoise[1] = pFrameInfo->borders[pFrameInfo->tranEnv];
+
+    break;
+
+  case 2:
+    /* Decode borders: */
+    border = temp;                            /* A -> aL */
+    pFrameInfo->borders[0] = border;          /* first border */
+
+    for (k = 1; k <= n; k++) {
+      temp = FDKreadBits (hBs, 2);/* R [2 bits] */
+      border += (2 * temp + 2);               /* R -> r                */
+      pFrameInfo->borders[k] = border;
+    }
+    pFrameInfo->borders[k] = numberTimeSlots; /* last border */
+
+
+    /* Decode pointer: */
+    pointer_bits = DFRACT_BITS - 1 - CountLeadingBits((FIXP_DBL)(n+1));
+    p = FDKreadBits (hBs, pointer_bits);     /* p = P [pointer_bits bits] */
+    if (p > n+1)
+      return 0;
+
+    if (p == 0 || p == 1)
+      pFrameInfo->tranEnv = -1;
+    else
+      pFrameInfo->tranEnv = p - 1;
+
+
+
+    /* Decode freq res: */
+    for (k = 0; k <= n; k++) {
+      pFrameInfo->freqRes[k] = FDKreadBits(hBs, 1); /* f = F [1 bits] */
+    }
+
+
+
+    /* Calculate noise floor middle border: */
+    switch (p) {
+    case 0:
+      pFrameInfo->bordersNoise[1] = pFrameInfo->borders[1];
+      break;
+    case 1:
+      pFrameInfo->bordersNoise[1] = pFrameInfo->borders[n];
+      break;
+    default:
+      pFrameInfo->bordersNoise[1] = pFrameInfo->borders[pFrameInfo->tranEnv];
+      break;
+    }
+
+    break;
+
+  case 3:
+    /* v_ctrlSignal = [frameClass,aL,aR,nL,nR,v_rL,v_rR,p,v_fLR]; */
+
+    aL = FDKreadBits (hBs, 2);       /* AL [2 bits], AL -> aL */
+
+    aR = FDKreadBits (hBs, 2) + numberTimeSlots;     /* AR [2 bits], AR -> aR */
+
+    nL = FDKreadBits (hBs, 2);       /* nL = NL [2 bits] */
+
+    nR = FDKreadBits (hBs, 2);       /* nR = NR [2 bits] */
+
+
+
+    /*-------------------------------------------------------------------------
+      Calculate help variables
+      --------------------------------------------------------------------------*/
+
+    /* general: */
+    nEnv = nL + nR + 1;            /* # envelopes */
+    if (nEnv > MAX_ENVELOPES)
+      return 0;
+    b = nEnv + 1;                  /* # borders   */
+
+
+
+    /*-------------------------------------------------------------------------
+      Decode envelopes
+      --------------------------------------------------------------------------*/
+
+
+    /* L-borders:   */
+    border            = aL;                   /* first border */
+    pFrameInfo->borders[0] = border;
+
+    for (k = 1; k <= nL; k++) {
+      temp = FDKreadBits (hBs, 2);/* R [2 bits] */
+      border += (2 * temp + 2);               /* R -> r                */
+      pFrameInfo->borders[k] = border;
+    }
+
+
+    /* R-borders:  */
+    border = aR;                              /* last border */
+    i      = nEnv;
+
+    pFrameInfo->borders[i] = border;
+
+    for (k = 0; k < nR; k++) {
+      temp = FDKreadBits (hBs, 2);/* R [2 bits] */
+      border -= (2 * temp + 2);               /* R -> r                */
+      pFrameInfo->borders[--i] = border;
+    }
+
+
+    /* decode pointer: */
+    pointer_bits = DFRACT_BITS - 1 - CountLeadingBits((FIXP_DBL)(nL+nR+1));
+    p = FDKreadBits (hBs, pointer_bits);     /* p = P [pointer_bits bits] */
+
+    if (p > nL+nR+1)
+      return 0;
+
+    pFrameInfo->tranEnv = p ? b - p : -1;
+
+
+
+    /* decode freq res: */
+    for (k = 0; k < nEnv; k++) {
+      pFrameInfo->freqRes[k] = FDKreadBits(hBs, 1); /* f = F [1 bits] */
+    }
+
+
+
+    /*-------------------------------------------------------------------------
+      Decode noise floors
+      --------------------------------------------------------------------------*/
+    pFrameInfo->bordersNoise[0] = aL;
+
+    if (nEnv == 1) {
+      /* 1 noise floor envelope: */
+      pFrameInfo->bordersNoise[1] = aR;
+    }
+    else {
+      /* 2 noise floor envelopes */
+      if (p == 0 || p == 1)
+        pFrameInfo->bordersNoise[1] = pFrameInfo->borders[nEnv - 1];
+      else
+        pFrameInfo->bordersNoise[1] = pFrameInfo->borders[pFrameInfo->tranEnv];
+      pFrameInfo->bordersNoise[2] = aR;
+    }
+    break;
+  }
+
+
+  /*
+    Store number of envelopes, noise floor envelopes and frame class
+  */
+  pFrameInfo->nEnvelopes = nEnv;
+
+  if (nEnv == 1)
+    pFrameInfo->nNoiseEnvelopes = 1;
+  else
+    pFrameInfo->nNoiseEnvelopes = 2;
+
+  pFrameInfo->frameClass = frameClass;
+
+  if (pFrameInfo->frameClass == 2 || pFrameInfo->frameClass == 1) {
+    /* calculate noise floor first and last borders: */
+    pFrameInfo->bordersNoise[0] = pFrameInfo->borders[0];
+    pFrameInfo->bordersNoise[pFrameInfo->nNoiseEnvelopes] = pFrameInfo->borders[nEnv];
+  }
+
+
+  return 1;
+}
+
+
+/*!
+  \brief   Check if the frameInfo vector has reasonable values.
+  \return  Zero for error, one for correct
+*/
+static int
+checkFrameInfo (FRAME_INFO * pFrameInfo, /*!< pointer to frameInfo */
+                int numberOfTimeSlots,   /*!< QMF time slots per frame */
+                int overlap,             /*!< Amount of overlap QMF time slots */
+                int timeStep)            /*!< QMF slots to SBR slots step factor */
+{
+  int maxPos,i,j;
+  int startPos;
+  int stopPos;
+  int tranEnv;
+  int startPosNoise;
+  int stopPosNoise;
+  int nEnvelopes = pFrameInfo->nEnvelopes;
+  int nNoiseEnvelopes = pFrameInfo->nNoiseEnvelopes;
+
+  if(nEnvelopes < 1 || nEnvelopes > MAX_ENVELOPES)
+    return 0;
+
+  if(nNoiseEnvelopes > MAX_NOISE_ENVELOPES)
+    return 0;
+
+  startPos        = pFrameInfo->borders[0];
+  stopPos         = pFrameInfo->borders[nEnvelopes];
+  tranEnv         = pFrameInfo->tranEnv;
+  startPosNoise   = pFrameInfo->bordersNoise[0];
+  stopPosNoise    = pFrameInfo->bordersNoise[nNoiseEnvelopes];
+
+  if (overlap < 0 || overlap > (6)) {
+    return 0;
+  }
+  if (timeStep < 1 || timeStep > 2) {
+    return 0;
+  }
+  maxPos = numberOfTimeSlots + (overlap/timeStep);
+
+  /* Check that the start and stop positions of the frame are reasonable values. */
+  if( (startPos < 0) || (startPos >= stopPos) )
+    return 0;
+  if( startPos > maxPos-numberOfTimeSlots ) /* First env. must start in or directly after the overlap buffer */
+    return 0;
+  if( stopPos < numberOfTimeSlots ) /* One complete frame must be ready for output after processing */
+    return 0;
+  if(stopPos > maxPos)
+    return 0;
+
+  /* Check that the  start border for every envelope is strictly later in time */
+  for(i=0;i<nEnvelopes;i++) {
+    if(pFrameInfo->borders[i] >= pFrameInfo->borders[i+1])
+      return 0;
+  }
+
+  /* Check that the envelope to be shortened is actually among the envelopes */
+  if(tranEnv>nEnvelopes)
+    return 0;
+
+
+  /* Check the noise borders */
+  if(nEnvelopes==1 && nNoiseEnvelopes>1)
+    return 0;
+
+  if(startPos != startPosNoise || stopPos != stopPosNoise)
+    return 0;
+
+
+  /* Check that the  start border for every noise-envelope is strictly later in time*/
+  for(i=0; i<nNoiseEnvelopes; i++) {
+    if(pFrameInfo->bordersNoise[i] >= pFrameInfo->bordersNoise[i+1])
+      return 0;
+  }
+
+  /* Check that every noise border is the same as an envelope border*/
+  for(i=0; i<nNoiseEnvelopes; i++) {
+    startPosNoise = pFrameInfo->bordersNoise[i];
+
+    for(j=0; j<nEnvelopes; j++) {
+      if(pFrameInfo->borders[j] == startPosNoise)
+        break;
+    }
+    if(j==nEnvelopes)
+      return 0;
+  }
+
+  return 1;
+}
diff --git a/libSBRdec/src/env_extr.h b/libSBRdec/src/env_extr.h
new file mode 100644
index 0000000..be46246
--- /dev/null
+++ b/libSBRdec/src/env_extr.h
@@ -0,0 +1,319 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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 extraction prototypes  
+*/
+
+#ifndef __ENVELOPE_EXTRACTION_H
+#define __ENVELOPE_EXTRACTION_H
+
+#include "sbrdecoder.h"
+
+#include "FDK_bitstream.h"
+#include "lpp_tran.h"
+
+#include "psdec.h"
+
+#define ENV_EXP_FRACT 0
+/*!< Shift raw envelope data to support fractional numbers.
+  Can be set to 8 instead of 0 to enhance accuracy during concealment.
+  This is not required for conformance and #requantizeEnvelopeData() will
+  become more expensive.
+*/
+
+#define EXP_BITS 6
+/*!< Size of exponent-part of a pseudo float envelope value (should be at least 6).
+  The remaining bits in each word are used for the mantissa (should be at least 10).
+  This format is used in the arrays iEnvelope[] and sbrNoiseFloorLevel[]
+  in the FRAME_DATA struct which must fit in a certain part of the output buffer
+  (See buffer management in sbr_dec.cpp).
+  Exponents and mantissas could also be stored in separate arrays.
+  Accessing the exponent or the mantissa would be simplified and the masks #MASK_E
+  resp. #MASK_M would   no longer be required.
+*/
+
+#define MASK_M (((1 << (FRACT_BITS - EXP_BITS)) - 1) << EXP_BITS)  /*!< Mask for extracting the mantissa of a pseudo float envelope value */
+#define MASK_E ((1 << EXP_BITS) - 1)           /*!< Mask for extracting the exponent of a pseudo float envelope value */
+
+#define SIGN_EXT ( ((SCHAR)-1) ^ MASK_E)        /*!< a CHAR-constant with all bits above our sign-bit set */
+#define ROUNDING ( (FIXP_SGL)(1<<(EXP_BITS-1)) ) /*!< 0.5-offset for rounding the mantissa of a pseudo-float envelope value */
+#define NRG_EXP_OFFSET  16                     /*!< Will be added to the reference energy's exponent to prevent negative numbers */
+#define NOISE_EXP_OFFSET  38                   /*!< Will be added to the noise level exponent to prevent negative numbers */
+
+typedef enum
+{
+  HEADER_NOT_PRESENT,
+  HEADER_OK,
+  HEADER_RESET
+}
+SBR_HEADER_STATUS;
+
+typedef enum
+{
+  SBR_NOT_INITIALIZED,
+  UPSAMPLING,
+  SBR_HEADER,
+  SBR_ACTIVE
+}
+SBR_SYNC_STATE;
+
+
+typedef enum
+{
+  COUPLING_OFF = 0,
+  COUPLING_LEVEL,
+  COUPLING_BAL
+}
+COUPLING_MODE;
+
+typedef struct
+{
+  UCHAR nSfb[2];           /*!< Number of SBR-bands for low and high freq-resolution */
+  UCHAR nNfb;              /*!< Actual number of noise bands to read from the bitstream*/
+  UCHAR numMaster;         /*!< Number of SBR-bands in v_k_master */
+  UCHAR lowSubband;        /*!< QMF-band where SBR frequency range starts */
+  UCHAR highSubband;       /*!< QMF-band where SBR frequency range ends */
+  UCHAR limiterBandTable[MAX_NUM_LIMITERS+1]; /*!< Limiter band table. */
+  UCHAR noLimiterBands;    /*!< Number of limiter bands. */
+  UCHAR nInvfBands;        /*!< Number of bands for inverse filtering */
+  UCHAR *freqBandTable[2]; /*!< Pointers to freqBandTableLo and freqBandTableHi */
+  UCHAR freqBandTableLo[MAX_FREQ_COEFFS/2+1];
+                                   /*!< Mapping of SBR bands to QMF bands for low frequency resolution */
+  UCHAR freqBandTableHi[MAX_FREQ_COEFFS+1];
+                                   /*!< Mapping of SBR bands to QMF bands for high frequency resolution */
+  UCHAR freqBandTableNoise[MAX_NOISE_COEFFS+1];
+                                   /*!< Mapping of SBR noise bands to QMF bands */
+  UCHAR v_k_master[MAX_FREQ_COEFFS+1];
+                                   /*!< Master BandTable which freqBandTable is derived from */
+}
+FREQ_BAND_DATA;
+
+typedef FREQ_BAND_DATA *HANDLE_FREQ_BAND_DATA;
+
+#define SBRDEC_ELD_GRID        1
+#define SBRDEC_SYNTAX_SCAL     2
+#define SBRDEC_SYNTAX_USAC     4
+#define SBRDEC_SYNTAX_RSVD50   8
+#define SBRDEC_LOW_POWER      16  /* Flag indicating that Low Power QMF mode shall be used. */
+#define SBRDEC_PS_DECODED     32  /* Flag indicating that PS was decoded and rendered. */
+#define SBRDEC_LD_MPS_QMF    512  /* Flag indicating that the LD-MPS QMF shall be used. */
+
+#define SBRDEC_HDR_STAT_RESET  1
+#define SBRDEC_HDR_STAT_UPDATE 2
+
+typedef struct {
+  UCHAR ampResolution;       /*!< Amplitude resolution of envelope values (0: 1.5dB, 1: 3dB) */
+  UCHAR xover_band;          /*!< Start index in #v_k_master[] used for dynamic crossover frequency */
+  UCHAR sbr_preprocessing;   /*!< SBR prewhitening flag. */
+} SBR_HEADER_DATA_BS_INFO;
+
+typedef struct {
+  /* Changes in these variables causes a reset of the decoder */
+  UCHAR startFreq;           /*!< Index for SBR start frequency */
+  UCHAR stopFreq;            /*!< Index for SBR highest frequency */
+  UCHAR freqScale;           /*!< 0: linear scale,  1-3 logarithmic scales */
+  UCHAR alterScale;          /*!< Flag for coarser frequency resolution */
+  UCHAR noise_bands;         /*!< Noise bands per octave, read from bitstream*/
+
+  /* don't require reset */
+  UCHAR limiterBands;        /*!< Index for number of limiter bands per octave */
+  UCHAR limiterGains;        /*!< Index to select gain limit */
+  UCHAR interpolFreq;        /*!< Select gain calculation method (1: per QMF channel, 0: per SBR band) */
+  UCHAR smoothingLength;     /*!< Smoothing of gains over time (0: on  1: off) */
+
+} SBR_HEADER_DATA_BS;
+
+typedef struct
+{
+  SBR_SYNC_STATE syncState;    /*!< The current initialization status of the header */
+
+  UCHAR status;                /*!< Flags field used for signaling a reset right before the processing starts and an update from config (e.g. ASC). */
+  UCHAR frameErrorFlag;        /*!< Frame data valid flag. CAUTION: This variable will be overwritten by the flag stored in the element structure.
+                                    This is necessary because of the frame delay. There it might happen that different slots use the same header. */
+  UCHAR numberTimeSlots;       /*!< AAC: 16,15 */
+  UCHAR numberOfAnalysisBands; /*!< Number of QMF analysis bands */
+  UCHAR timeStep;              /*!< Time resolution of SBR in QMF-slots */
+  UINT  sbrProcSmplRate;       /*!< SBR processing sampling frequency (!= OutputSamplingRate)
+                                     (always: CoreSamplingRate * UpSamplingFactor; even in single rate mode) */
+
+  SBR_HEADER_DATA_BS      bs_data;  /*!< current SBR header. */
+  SBR_HEADER_DATA_BS_INFO bs_info;  /*!< SBR info. */
+
+  FREQ_BAND_DATA freqBandData;  /*!< Pointer to struct #FREQ_BAND_DATA */
+}
+SBR_HEADER_DATA;
+
+typedef SBR_HEADER_DATA *HANDLE_SBR_HEADER_DATA;
+
+
+typedef struct
+{
+  UCHAR frameClass;               /*!< Select grid type */
+  UCHAR nEnvelopes;               /*!< Number of envelopes */
+  UCHAR borders[MAX_ENVELOPES+1]; /*!< Envelope borders (in SBR-timeslots, e.g. mp3PRO: 0..11) */
+  UCHAR freqRes[MAX_ENVELOPES];   /*!< Frequency resolution for each envelope (0=low, 1=high) */
+  SCHAR  tranEnv;                 /*!< Transient envelope, -1 if none */
+  UCHAR nNoiseEnvelopes;          /*!< Number of noise envelopes */
+  UCHAR bordersNoise[MAX_NOISE_ENVELOPES+1];/*!< borders of noise envelopes */
+}
+FRAME_INFO;
+
+
+typedef struct
+{
+  FIXP_SGL sfb_nrg_prev[MAX_FREQ_COEFFS];    /*!< Previous envelope (required for differential-coded values) */
+  FIXP_SGL prevNoiseLevel[MAX_NOISE_COEFFS]; /*!< Previous noise envelope (required for differential-coded values) */
+  COUPLING_MODE coupling;                    /*!< Stereo-mode of previous frame */
+  INVF_MODE sbr_invf_mode[MAX_INVF_BANDS];   /*!< Previous strength of filtering in transposer */
+  UCHAR ampRes;                              /*!< Previous amplitude resolution (0: 1.5dB, 1: 3dB) */
+  UCHAR stopPos;                             /*!< Position in time where last envelope ended */
+  UCHAR frameErrorFlag;                      /*!< Previous frame status */
+}
+SBR_PREV_FRAME_DATA;
+
+typedef SBR_PREV_FRAME_DATA *HANDLE_SBR_PREV_FRAME_DATA;
+
+
+typedef struct
+{
+  int nScaleFactors;                    /*!< total number of scalefactors in frame */
+
+  FRAME_INFO frameInfo;                 /*!< time grid for current frame */
+  UCHAR domain_vec[MAX_ENVELOPES];      /*!< Bitfield containing direction of delta-coding for each envelope (0:frequency, 1:time) */
+  UCHAR domain_vec_noise[MAX_NOISE_ENVELOPES]; /*!< Same as above, but for noise envelopes */
+
+  INVF_MODE sbr_invf_mode[MAX_INVF_BANDS]; /*!< Strength of filtering in transposer */
+  COUPLING_MODE coupling;                  /*!< Stereo-mode */
+  int ampResolutionCurrentFrame;           /*!< Amplitude resolution of envelope values (0: 1.5dB, 1: 3dB) */
+
+  UCHAR addHarmonics[MAX_FREQ_COEFFS];     /*!< Flags for synthetic sine addition */
+
+  FIXP_SGL iEnvelope[MAX_NUM_ENVELOPE_VALUES];       /*!< Envelope data */
+  FIXP_SGL sbrNoiseFloorLevel[MAX_NUM_NOISE_VALUES]; /*!< Noise envelope data */
+}
+SBR_FRAME_DATA;
+
+typedef SBR_FRAME_DATA *HANDLE_SBR_FRAME_DATA;
+
+void initSbrPrevFrameData (HANDLE_SBR_PREV_FRAME_DATA h_prev_data,
+                           int timeSlots);
+
+
+int sbrGetSingleChannelElement (HANDLE_SBR_HEADER_DATA hHeaderData,
+                                HANDLE_SBR_FRAME_DATA  hFrameData,
+                                HANDLE_FDK_BITSTREAM   hBitBuf,
+                                HANDLE_PS_DEC hParametricStereoDec,
+                                const UINT             flags,
+                                const int              overlap
+                               );
+
+int sbrGetChannelPairElement (HANDLE_SBR_HEADER_DATA hHeaderData,
+                              HANDLE_SBR_FRAME_DATA  hFrameDataLeft,
+                              HANDLE_SBR_FRAME_DATA  hFrameDataRight,
+                              HANDLE_FDK_BITSTREAM   hBitBuf,
+                              const UINT             flags,
+                              const int              overlap);
+
+SBR_HEADER_STATUS
+sbrGetHeaderData (HANDLE_SBR_HEADER_DATA headerData,
+                  HANDLE_FDK_BITSTREAM   hBitBuf,
+                  const UINT             flags,
+                  const int              fIsSbrData);
+
+/*!
+  \brief     Initialize SBR header data
+
+  Copy default values to the header data struct and patch some entries
+  depending on the core codec.
+*/
+SBR_ERROR
+initHeaderData (
+        HANDLE_SBR_HEADER_DATA  hHeaderData,
+        const int               sampleRateIn,
+        const int               sampleRateOut,
+        const int               samplesPerFrame,
+        const UINT              flags
+        );
+#endif
diff --git a/libSBRdec/src/huff_dec.cpp b/libSBRdec/src/huff_dec.cpp
new file mode 100644
index 0000000..6e00b23
--- /dev/null
+++ b/libSBRdec/src/huff_dec.cpp
@@ -0,0 +1,120 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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  Huffman Decoder  
+*/
+
+#include "huff_dec.h"
+
+/***************************************************************************/
+/*!
+  \brief     Decodes one huffman code word
+
+  Reads bits from the bitstream until a valid codeword is found.
+  The table entries are interpreted either as index to the next entry
+  or - if negative - as the codeword.
+
+  \return    decoded value
+
+  \author
+
+****************************************************************************/
+int
+DecodeHuffmanCW (Huffman h,                /*!< pointer to huffman codebook table */
+                 HANDLE_FDK_BITSTREAM hBs) /*!< Handle to Bitbuffer */
+{
+  SCHAR index = 0;
+  int value, bit;
+
+  while (index >= 0) {
+    bit = FDKreadBits (hBs, 1);
+    index = h[index][bit];
+  }
+
+  value = index+64; /* Add offset */
+
+
+  return value;
+}
diff --git a/libSBRdec/src/huff_dec.h b/libSBRdec/src/huff_dec.h
new file mode 100644
index 0000000..7b2b50b
--- /dev/null
+++ b/libSBRdec/src/huff_dec.h
@@ -0,0 +1,100 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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  Huffman Decoder  
+*/
+#ifndef __HUFF_DEC_H
+#define __HUFF_DEC_H
+
+#include "sbrdecoder.h"
+#include "FDK_bitstream.h"
+
+typedef const SCHAR (*Huffman)[2];
+
+int
+DecodeHuffmanCW (Huffman h,
+                 HANDLE_FDK_BITSTREAM hBitBuf);
+
+#endif
diff --git a/libSBRdec/src/lpp_tran.cpp b/libSBRdec/src/lpp_tran.cpp
new file mode 100644
index 0000000..2d51831
--- /dev/null
+++ b/libSBRdec/src/lpp_tran.cpp
@@ -0,0 +1,1000 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/*!
+  \file
+  \brief  Low Power Profile Transposer,  
+  This module provides the transposer. The main entry point is lppTransposer(). The function generates
+  high frequency content by copying data from the low band (provided by core codec) into the high band.
+  This process is also referred to as "patching". The function also implements spectral whitening by means of
+  inverse filtering based on LPC coefficients.
+
+  Together with the QMF filterbank the transposer can be tested using a supplied test program. See main_audio.cpp for details.
+  This module does use fractional arithmetic and the accuracy of the computations has an impact on the overall sound quality.
+  The module also needs to take into account the different scaling of spectral data.
+
+  \sa lppTransposer(), main_audio.cpp, sbr_scale.h, \ref documentationOverview
+*/
+
+#include "lpp_tran.h"
+
+#include "sbr_ram.h"
+#include "sbr_rom.h"
+
+#include "genericStds.h"
+#include "autocorr2nd.h"
+
+
+
+#if defined(__arm__)
+#include "arm/lpp_tran_arm.cpp"
+#endif
+
+
+
+#define LPC_SCALE_FACTOR  2
+
+
+/*!
+ *
+ * \brief Get bandwidth expansion factor from filtering level
+ *
+ * Returns a filter parameter (bandwidth expansion factor) depending on
+ * the desired filtering level signalled in the bitstream.
+ * When switching the filtering level from LOW to OFF, an additional
+ * level is being inserted to achieve a smooth transition.
+ */
+
+#ifndef FUNCTION_mapInvfMode
+static FIXP_DBL
+mapInvfMode (INVF_MODE mode,
+             INVF_MODE prevMode,
+             WHITENING_FACTORS whFactors)
+{
+  switch (mode) {
+  case INVF_LOW_LEVEL:
+    if(prevMode == INVF_OFF)
+      return whFactors.transitionLevel;
+    else
+      return whFactors.lowLevel;
+
+  case INVF_MID_LEVEL:
+    return whFactors.midLevel;
+
+  case INVF_HIGH_LEVEL:
+    return whFactors.highLevel;
+
+  default:
+    if(prevMode == INVF_LOW_LEVEL)
+      return whFactors.transitionLevel;
+    else
+      return whFactors.off;
+  }
+}
+#endif /* #ifndef FUNCTION_mapInvfMode */
+
+/*!
+ *
+ * \brief Perform inverse filtering level emphasis
+ *
+ * Retrieve bandwidth expansion factor and apply smoothing for each filter band
+ *
+ */
+
+#ifndef FUNCTION_inverseFilteringLevelEmphasis
+static void
+inverseFilteringLevelEmphasis(HANDLE_SBR_LPP_TRANS hLppTrans,/*!< Handle of lpp transposer  */
+                              UCHAR nInvfBands,              /*!< Number of bands for inverse filtering */
+                              INVF_MODE *sbr_invf_mode,      /*!< Current inverse filtering modes */
+                              INVF_MODE *sbr_invf_mode_prev, /*!< Previous inverse filtering modes */
+                              FIXP_DBL * bwVector            /*!< Resulting filtering levels */
+                              )
+{
+  for(int i = 0; i < nInvfBands; i++) {
+    FIXP_DBL accu;
+    FIXP_DBL bwTmp = mapInvfMode (sbr_invf_mode[i],
+                                  sbr_invf_mode_prev[i],
+                                  hLppTrans->pSettings->whFactors);
+
+    if(bwTmp < hLppTrans->bwVectorOld[i]) {
+      accu = fMultDiv2(FL2FXCONST_DBL(0.75f),bwTmp) +
+             fMultDiv2(FL2FXCONST_DBL(0.25f),hLppTrans->bwVectorOld[i]);
+    }
+    else {
+      accu = fMultDiv2(FL2FXCONST_DBL(0.90625f),bwTmp) +
+             fMultDiv2(FL2FXCONST_DBL(0.09375f),hLppTrans->bwVectorOld[i]);
+    }
+
+    if (accu <  FL2FXCONST_DBL(0.015625f)>>1)
+      bwVector[i] = FL2FXCONST_DBL(0.0f);
+    else
+      bwVector[i] = fixMin(accu<<1,FL2FXCONST_DBL(0.99609375f));
+  }
+}
+#endif /* #ifndef FUNCTION_inverseFilteringLevelEmphasis */
+
+/* Resulting autocorrelation determinant exponent */
+#define ACDET_EXP (2*(DFRACT_BITS+sbrScaleFactor->lb_scale+10-ac.det_scale))
+#define AC_EXP (-sbrScaleFactor->lb_scale+LPC_SCALE_FACTOR)
+#define ALPHA_EXP (-sbrScaleFactor->lb_scale+LPC_SCALE_FACTOR+1)
+/* Resulting transposed QMF values exponent 16 bit normalized samplebits assumed. */
+#define QMFOUT_EXP ((SAMPLE_BITS-15)-sbrScaleFactor->lb_scale)
+
+/*!
+ *
+ * \brief Perform transposition by patching of subband samples.
+ * This function serves as the main entry point into the module. The function determines the areas for the
+ * patching process (these are the source range as well as the target range) and implements spectral whitening
+ * by means of inverse filtering. The function autoCorrelation2nd() is an auxiliary function for calculating the
+ * LPC coefficients for the filtering.  The actual calculation of the LPC coefficients and the implementation
+ * of the filtering are done as part of lppTransposer().
+ *
+ * Note that the filtering is done on all available QMF subsamples, whereas the patching is only done on those QMF
+ * subsamples that will be used in the next QMF synthesis. The filtering is also implemented before the patching
+ * includes further dependencies on parameters from the SBR data.
+ *
+ */
+
+void lppTransposer (HANDLE_SBR_LPP_TRANS hLppTrans,    /*!< Handle of lpp transposer  */
+                    QMF_SCALE_FACTOR  *sbrScaleFactor, /*!< Scaling factors */
+                    FIXP_DBL **qmfBufferReal,          /*!< Pointer to pointer to real part of subband samples (source) */
+
+                    FIXP_DBL *degreeAlias,             /*!< Vector for results of aliasing estimation */
+                    FIXP_DBL **qmfBufferImag,          /*!< Pointer to pointer to imaginary part of subband samples (source) */
+                    const int useLP,
+                    const int timeStep,                /*!< Time step of envelope */
+                    const int firstSlotOffs,           /*!< Start position in time */
+                    const int lastSlotOffs,            /*!< Number of overlap-slots into next frame */
+                    const int nInvfBands,              /*!< Number of bands for inverse filtering */
+                    INVF_MODE *sbr_invf_mode,          /*!< Current inverse filtering modes */
+                    INVF_MODE *sbr_invf_mode_prev      /*!< Previous inverse filtering modes */
+                    )
+{
+  INT    bwIndex[MAX_NUM_PATCHES];
+  FIXP_DBL  bwVector[MAX_NUM_PATCHES];       /*!< pole moving factors */
+
+  int    i;
+  int    loBand, start, stop;
+  TRANSPOSER_SETTINGS *pSettings = hLppTrans->pSettings;
+  PATCH_PARAM *patchParam = pSettings->patchParam;
+  int    patch;
+
+  FIXP_SGL  alphar[LPC_ORDER], a0r, a1r;
+  FIXP_SGL  alphai[LPC_ORDER], a0i=0, a1i=0;
+  FIXP_SGL  bw = FL2FXCONST_SGL(0.0f);
+
+  int    autoCorrLength;
+
+  FIXP_DBL k1, k1_below=0, k1_below2=0;
+
+  ACORR_COEFS ac;
+  int    startSample;
+  int    stopSample;
+  int    stopSampleClear;
+
+  int comLowBandScale;
+  int ovLowBandShift;
+  int lowBandShift;
+/*  int ovHighBandShift;*/
+  int targetStopBand;
+
+
+  alphai[0] = FL2FXCONST_SGL(0.0f);
+  alphai[1] = FL2FXCONST_SGL(0.0f);
+
+
+  startSample = firstSlotOffs * timeStep;
+  stopSample  = pSettings->nCols + lastSlotOffs * timeStep;
+
+
+  inverseFilteringLevelEmphasis(hLppTrans, nInvfBands, sbr_invf_mode, sbr_invf_mode_prev, bwVector);
+
+  stopSampleClear = stopSample;
+
+  autoCorrLength = pSettings->nCols + pSettings->overlap;
+
+  /* Set upper subbands to zero:
+     This is required in case that the patches do not cover the complete highband
+     (because the last patch would be too short).
+     Possible optimization: Clearing bands up to usb would be sufficient here. */
+  targetStopBand = patchParam[pSettings->noOfPatches-1].targetStartBand
+                 + patchParam[pSettings->noOfPatches-1].numBandsInPatch;
+
+  int memSize = ((64) - targetStopBand) * sizeof(FIXP_DBL);
+
+  if (!useLP) {
+    for (i = startSample; i < stopSampleClear; i++) {
+      FDKmemclear(&qmfBufferReal[i][targetStopBand], memSize);
+      FDKmemclear(&qmfBufferImag[i][targetStopBand], memSize);
+    }
+  } else
+  for (i = startSample; i < stopSampleClear; i++) {
+    FDKmemclear(&qmfBufferReal[i][targetStopBand], memSize);
+  }
+
+  /* init bwIndex for each patch */
+  FDKmemclear(bwIndex, pSettings->noOfPatches*sizeof(INT));
+
+  /*
+    Calc common low band scale factor
+  */
+  comLowBandScale = fixMin(sbrScaleFactor->ov_lb_scale,sbrScaleFactor->lb_scale);
+
+  ovLowBandShift =  sbrScaleFactor->ov_lb_scale - comLowBandScale;
+  lowBandShift   =  sbrScaleFactor->lb_scale - comLowBandScale;
+  /*  ovHighBandShift = firstSlotOffs == 0 ? ovLowBandShift:0;*/
+
+  /* outer loop over bands to do analysis only once for each band */
+
+  if (!useLP) {
+    start = pSettings->lbStartPatching;
+    stop = pSettings->lbStopPatching;
+  } else
+  {
+    start = fixMax(1, pSettings->lbStartPatching - 2);
+    stop = patchParam[0].targetStartBand;
+  }
+
+
+  for ( loBand = start; loBand <  stop; loBand++ ) {
+
+    FIXP_DBL  lowBandReal[(((1024)/(32))+(6))+LPC_ORDER];
+    FIXP_DBL *plowBandReal = lowBandReal;
+    FIXP_DBL **pqmfBufferReal = qmfBufferReal;
+    FIXP_DBL  lowBandImag[(((1024)/(32))+(6))+LPC_ORDER];
+    FIXP_DBL *plowBandImag = lowBandImag;
+    FIXP_DBL **pqmfBufferImag = qmfBufferImag;
+    int resetLPCCoeffs=0;
+    int dynamicScale = DFRACT_BITS-1-LPC_SCALE_FACTOR;
+    int acDetScale = 0; /* scaling of autocorrelation determinant */
+
+    for(i=0;i<LPC_ORDER;i++){
+      *plowBandReal++ = hLppTrans->lpcFilterStatesReal[i][loBand];
+      if (!useLP)
+        *plowBandImag++ = hLppTrans->lpcFilterStatesImag[i][loBand];
+    }
+
+    /*
+      Take old slope length qmf slot source values out of (overlap)qmf buffer
+    */
+    if (!useLP) {
+      for(i=0;i<pSettings->nCols+pSettings->overlap;i++){
+        *plowBandReal++ = (*pqmfBufferReal++)[loBand];
+        *plowBandImag++ = (*pqmfBufferImag++)[loBand];
+      }
+    } else
+    {
+      /* pSettings->overlap is always even */
+      FDK_ASSERT((pSettings->overlap & 1) == 0);
+
+      for(i=0;i<((pSettings->overlap+pSettings->nCols)>>1);i++) {
+        *plowBandReal++ = (*pqmfBufferReal++)[loBand];
+        *plowBandReal++ = (*pqmfBufferReal++)[loBand];
+      }
+      if (pSettings->nCols & 1) {
+        *plowBandReal++ = (*pqmfBufferReal++)[loBand];
+      }
+    }
+
+    /*
+      Determine dynamic scaling value.
+     */
+    dynamicScale = fixMin(dynamicScale, getScalefactor(lowBandReal, LPC_ORDER+pSettings->overlap) + ovLowBandShift);
+    dynamicScale = fixMin(dynamicScale, getScalefactor(&lowBandReal[LPC_ORDER+pSettings->overlap], pSettings->nCols) + lowBandShift);
+    if (!useLP) {
+      dynamicScale = fixMin(dynamicScale, getScalefactor(lowBandImag, LPC_ORDER+pSettings->overlap) + ovLowBandShift);
+      dynamicScale = fixMin(dynamicScale, getScalefactor(&lowBandImag[LPC_ORDER+pSettings->overlap], pSettings->nCols) + lowBandShift);
+    }
+    dynamicScale = fixMax(0, dynamicScale-1); /* one additional bit headroom to prevent -1.0 */
+
+    /*
+      Scale temporal QMF buffer.
+     */
+    scaleValues(&lowBandReal[0], LPC_ORDER+pSettings->overlap, dynamicScale-ovLowBandShift);
+    scaleValues(&lowBandReal[LPC_ORDER+pSettings->overlap], pSettings->nCols, dynamicScale-lowBandShift);
+
+    if (!useLP) {
+      scaleValues(&lowBandImag[0], LPC_ORDER+pSettings->overlap, dynamicScale-ovLowBandShift);
+      scaleValues(&lowBandImag[LPC_ORDER+pSettings->overlap], pSettings->nCols, dynamicScale-lowBandShift);
+    }
+
+
+      if (!useLP) {
+        acDetScale += autoCorr2nd_cplx(&ac, lowBandReal+LPC_ORDER, lowBandImag+LPC_ORDER, autoCorrLength);
+      }
+      else
+      {
+        acDetScale += autoCorr2nd_real(&ac, lowBandReal+LPC_ORDER, autoCorrLength);
+      }
+
+      /* Examine dynamic of determinant in autocorrelation. */
+      acDetScale += 2*(comLowBandScale + dynamicScale);
+      acDetScale *= 2;              /* two times reflection coefficent scaling */
+      acDetScale += ac.det_scale;   /* ac scaling of determinant */
+
+      /* In case of determinant < 10^-38, resetLPCCoeffs=1 has to be enforced. */
+      if (acDetScale>126 ) {
+        resetLPCCoeffs = 1;
+      }
+
+
+    alphar[1] = FL2FXCONST_SGL(0.0f);
+    if (!useLP)
+      alphai[1] = FL2FXCONST_SGL(0.0f);
+
+    if (ac.det != FL2FXCONST_DBL(0.0f)) {
+      FIXP_DBL tmp,absTmp,absDet;
+
+      absDet = fixp_abs(ac.det);
+
+      if (!useLP) {
+        tmp = ( fMultDiv2(ac.r01r,ac.r12r) >> (LPC_SCALE_FACTOR-1) ) -
+              ( (fMultDiv2(ac.r01i,ac.r12i) + fMultDiv2(ac.r02r,ac.r11r)) >> (LPC_SCALE_FACTOR-1) );
+      } else
+      {
+        tmp = ( fMultDiv2(ac.r01r,ac.r12r) >> (LPC_SCALE_FACTOR-1) ) -
+              ( fMultDiv2(ac.r02r,ac.r11r) >> (LPC_SCALE_FACTOR-1) );
+      }
+      absTmp = fixp_abs(tmp);
+
+      /*
+        Quick check: is first filter coeff >= 1(4)
+       */
+      {
+        INT scale;
+        FIXP_DBL result = fDivNorm(absTmp, absDet, &scale);
+        scale = scale+ac.det_scale;
+
+        if ( (scale > 0) && (result >= (FIXP_DBL)MAXVAL_DBL>>scale) ) {
+          resetLPCCoeffs = 1;
+        }
+        else {
+          alphar[1] = FX_DBL2FX_SGL(scaleValue(result,scale));
+          if((tmp<FL2FX_DBL(0.0f)) ^ (ac.det<FL2FX_DBL(0.0f))) {
+            alphar[1] = -alphar[1];
+          }
+        }
+      }
+
+      if (!useLP)
+      {
+        tmp =  ( fMultDiv2(ac.r01i,ac.r12r) >> (LPC_SCALE_FACTOR-1) ) +
+               ( (fMultDiv2(ac.r01r,ac.r12i) - (FIXP_DBL)fMultDiv2(ac.r02i,ac.r11r)) >> (LPC_SCALE_FACTOR-1) ) ;
+
+        absTmp = fixp_abs(tmp);
+
+        /*
+        Quick check: is second filter coeff >= 1(4)
+        */
+        {
+          INT scale;
+          FIXP_DBL result = fDivNorm(absTmp, absDet, &scale);
+          scale = scale+ac.det_scale;
+
+          if ( (scale > 0) && (result >= /*FL2FXCONST_DBL(1.f)*/ (FIXP_DBL)MAXVAL_DBL>>scale) ) {
+            resetLPCCoeffs = 1;
+          }
+          else {
+            alphai[1] = FX_DBL2FX_SGL(scaleValue(result,scale));
+            if((tmp<FL2FX_DBL(0.0f)) ^ (ac.det<FL2FX_DBL(0.0f))) {
+              alphai[1] = -alphai[1];
+            }
+          }
+        }
+      }
+    }
+
+    alphar[0] =  FL2FXCONST_SGL(0.0f);
+    if (!useLP)
+      alphai[0] = FL2FXCONST_SGL(0.0f);
+
+    if ( ac.r11r != FL2FXCONST_DBL(0.0f) ) {
+
+      /* ac.r11r is always >=0 */
+      FIXP_DBL tmp,absTmp;
+
+      if (!useLP) {
+        tmp = (ac.r01r>>(LPC_SCALE_FACTOR+1)) +
+              (fMultDiv2(alphar[1],ac.r12r) + fMultDiv2(alphai[1],ac.r12i));
+      } else
+      {
+        if(ac.r01r>=FL2FXCONST_DBL(0.0f))
+          tmp = (ac.r01r>>(LPC_SCALE_FACTOR+1)) + fMultDiv2(alphar[1],ac.r12r);
+        else
+          tmp = -((-ac.r01r)>>(LPC_SCALE_FACTOR+1)) + fMultDiv2(alphar[1],ac.r12r);
+      }
+
+      absTmp = fixp_abs(tmp);
+
+      /*
+        Quick check: is first filter coeff >= 1(4)
+      */
+
+      if (absTmp >= (ac.r11r>>1)) {
+        resetLPCCoeffs=1;
+      }
+      else {
+        INT scale;
+        FIXP_DBL result = fDivNorm(absTmp, fixp_abs(ac.r11r), &scale);
+        alphar[0] =  FX_DBL2FX_SGL(scaleValue(result,scale+1));
+
+        if((tmp>FL2FX_DBL(0.0f)) ^ (ac.r11r<FL2FX_DBL(0.0f)))
+          alphar[0] = -alphar[0];
+      }
+
+      if (!useLP)
+      {
+        tmp = (ac.r01i>>(LPC_SCALE_FACTOR+1)) +
+              (fMultDiv2(alphai[1],ac.r12r) - fMultDiv2(alphar[1],ac.r12i));
+
+        absTmp = fixp_abs(tmp);
+
+        /*
+        Quick check: is second filter coeff >= 1(4)
+        */
+        if (absTmp >= (ac.r11r>>1)) {
+          resetLPCCoeffs=1;
+        }
+        else {
+          INT scale;
+          FIXP_DBL result = fDivNorm(absTmp, fixp_abs(ac.r11r), &scale);
+          alphai[0] = FX_DBL2FX_SGL(scaleValue(result,scale+1));
+          if((tmp>FL2FX_DBL(0.0f)) ^ (ac.r11r<FL2FX_DBL(0.0f)))
+            alphai[0] = -alphai[0];
+        }
+      }
+    }
+
+
+    if (!useLP)
+    {
+      /* Now check the quadratic criteria */
+      if( (fMultDiv2(alphar[0],alphar[0]) + fMultDiv2(alphai[0],alphai[0])) >= FL2FXCONST_DBL(0.5f) )
+        resetLPCCoeffs=1;
+      if( (fMultDiv2(alphar[1],alphar[1]) + fMultDiv2(alphai[1],alphai[1])) >= FL2FXCONST_DBL(0.5f) )
+        resetLPCCoeffs=1;
+    }
+
+    if(resetLPCCoeffs){
+      alphar[0] = FL2FXCONST_SGL(0.0f);
+      alphar[1] = FL2FXCONST_SGL(0.0f);
+      if (!useLP)
+      {
+        alphai[0] = FL2FXCONST_SGL(0.0f);
+        alphai[1] = FL2FXCONST_SGL(0.0f);
+      }
+    }
+
+    if (useLP)
+    {
+
+      /* Aliasing detection */
+      if(ac.r11r==FL2FXCONST_DBL(0.0f)) {
+        k1 = FL2FXCONST_DBL(0.0f);
+      }
+      else {
+        if ( fixp_abs(ac.r01r) >= fixp_abs(ac.r11r) ) {
+          if ( fMultDiv2(ac.r01r,ac.r11r) < FL2FX_DBL(0.0f)) {
+            k1 = (FIXP_DBL)MAXVAL_DBL /*FL2FXCONST_SGL(1.0f)*/;
+          }else {
+            /* Since this value is squared later, it must not ever become -1.0f. */
+            k1 = (FIXP_DBL)(MINVAL_DBL+1) /*FL2FXCONST_SGL(-1.0f)*/;
+          }
+        }
+        else {
+          INT scale;
+          FIXP_DBL result = fDivNorm(fixp_abs(ac.r01r), fixp_abs(ac.r11r), &scale);
+          k1 = scaleValue(result,scale);
+
+          if(!((ac.r01r<FL2FX_DBL(0.0f)) ^ (ac.r11r<FL2FX_DBL(0.0f)))) {
+            k1 = -k1;
+          }
+        }
+      }
+      if(loBand > 1){
+        /* Check if the gain should be locked */
+        FIXP_DBL deg = /*FL2FXCONST_DBL(1.0f)*/ (FIXP_DBL)MAXVAL_DBL - fPow2(k1_below);
+        degreeAlias[loBand] = FL2FXCONST_DBL(0.0f);
+        if (((loBand & 1) == 0) && (k1 < FL2FXCONST_DBL(0.0f))){
+          if (k1_below < FL2FXCONST_DBL(0.0f)) {         /* 2-Ch Aliasing Detection */
+            degreeAlias[loBand] = (FIXP_DBL)MAXVAL_DBL /*FL2FXCONST_DBL(1.0f)*/;
+            if ( k1_below2 > FL2FXCONST_DBL(0.0f) ) {    /* 3-Ch Aliasing Detection */
+              degreeAlias[loBand-1] = deg;
+            }
+          }
+          else if ( k1_below2 > FL2FXCONST_DBL(0.0f) ) { /* 3-Ch Aliasing Detection */
+            degreeAlias[loBand]   = deg;
+          }
+        }
+        if (((loBand & 1) == 1) && (k1 > FL2FXCONST_DBL(0.0f))){
+          if (k1_below > FL2FXCONST_DBL(0.0f)) {         /* 2-CH Aliasing Detection */
+            degreeAlias[loBand] = (FIXP_DBL)MAXVAL_DBL /*FL2FXCONST_DBL(1.0f)*/;
+            if ( k1_below2 < FL2FXCONST_DBL(0.0f) ) {    /* 3-CH Aliasing Detection */
+              degreeAlias[loBand-1] = deg;
+            }
+          }
+          else if ( k1_below2 < FL2FXCONST_DBL(0.0f) ) { /* 3-CH Aliasing Detection */
+            degreeAlias[loBand]   = deg;
+          }
+        }
+      }
+      /* remember k1 values of the 2 QMF channels below the current channel */
+      k1_below2 = k1_below;
+      k1_below = k1;
+    }
+
+    patch = 0;
+
+    while ( patch < pSettings->noOfPatches ) { /* inner loop over every patch */
+
+      int hiBand = loBand + patchParam[patch].targetBandOffs;
+
+      if ( loBand < patchParam[patch].sourceStartBand
+           || loBand >= patchParam[patch].sourceStopBand
+           //|| hiBand >= hLppTrans->pSettings->noChannels
+           ) {
+        /* Lowband not in current patch - proceed */
+        patch++;
+        continue;
+      }
+
+      FDK_ASSERT( hiBand < (64) );
+
+      /* bwIndex[patch] is already initialized with value from previous band inside this patch */
+      while (hiBand >= pSettings->bwBorders[bwIndex[patch]])
+        bwIndex[patch]++;
+
+
+      /*
+        Filter Step 2: add the left slope with the current filter to the buffer
+                       pure source values are already in there
+      */
+      bw = FX_DBL2FX_SGL(bwVector[bwIndex[patch]]);
+
+      a0r = FX_DBL2FX_SGL(fMult(bw,alphar[0])); /* Apply current bandwidth expansion factor */
+
+
+      if (!useLP)
+        a0i = FX_DBL2FX_SGL(fMult(bw,alphai[0]));
+      bw =  FX_DBL2FX_SGL(fPow2(bw));
+      a1r = FX_DBL2FX_SGL(fMult(bw,alphar[1]));
+      if (!useLP)
+        a1i = FX_DBL2FX_SGL(fMult(bw,alphai[1]));
+
+
+
+      /*
+        Filter Step 3: insert the middle part which won't be windowed
+      */
+
+      if ( bw <= FL2FXCONST_SGL(0.0f) ) {
+        if (!useLP) {
+          int descale = fixMin(DFRACT_BITS-1, (LPC_SCALE_FACTOR+dynamicScale));
+          for(i = startSample; i < stopSample; i++ ) {
+            qmfBufferReal[i][hiBand] = lowBandReal[LPC_ORDER+i]>>descale;
+            qmfBufferImag[i][hiBand] = lowBandImag[LPC_ORDER+i]>>descale;
+          }
+        } else
+        {
+          int descale = fixMin(DFRACT_BITS-1, (LPC_SCALE_FACTOR+dynamicScale));
+          for(i = startSample; i < stopSample; i++ ) {
+            qmfBufferReal[i][hiBand] = lowBandReal[LPC_ORDER+i]>>descale;
+          }
+        }
+      }
+      else {  /* bw <= 0 */
+
+        if (!useLP) {
+          int descale = fixMin(DFRACT_BITS-1, (LPC_SCALE_FACTOR+dynamicScale));
+#ifdef FUNCTION_LPPTRANSPOSER_func1
+          lppTransposer_func1(lowBandReal+LPC_ORDER+startSample,lowBandImag+LPC_ORDER+startSample,
+                              qmfBufferReal+startSample,qmfBufferImag+startSample,
+                              stopSample-startSample, (int) hiBand,
+                              dynamicScale,descale,
+                              a0r, a0i, a1r, a1i);
+#else
+          for(i = startSample; i < stopSample; i++ ) {
+            FIXP_DBL accu1, accu2;
+
+            accu1 = (fMultDiv2(a0r,lowBandReal[LPC_ORDER+i-1]) - fMultDiv2(a0i,lowBandImag[LPC_ORDER+i-1]) +
+                     fMultDiv2(a1r,lowBandReal[LPC_ORDER+i-2]) - fMultDiv2(a1i,lowBandImag[LPC_ORDER+i-2]))>>dynamicScale;
+            accu2 = (fMultDiv2(a0i,lowBandReal[LPC_ORDER+i-1]) + fMultDiv2(a0r,lowBandImag[LPC_ORDER+i-1]) +
+                     fMultDiv2(a1i,lowBandReal[LPC_ORDER+i-2]) + fMultDiv2(a1r,lowBandImag[LPC_ORDER+i-2]))>>dynamicScale;
+
+            qmfBufferReal[i][hiBand] = (lowBandReal[LPC_ORDER+i]>>descale) + (accu1<<1);
+            qmfBufferImag[i][hiBand] = (lowBandImag[LPC_ORDER+i]>>descale) + (accu2<<1);
+          }
+#endif
+        } else
+        {
+          int descale = fixMin(DFRACT_BITS-1, (LPC_SCALE_FACTOR+dynamicScale));
+
+          FDK_ASSERT(dynamicScale >= 0);
+          for(i = startSample; i < stopSample; i++ ) {
+            FIXP_DBL accu1;
+
+            accu1 = (fMultDiv2(a0r,lowBandReal[LPC_ORDER+i-1]) + fMultDiv2(a1r,lowBandReal[LPC_ORDER+i-2]))>>dynamicScale;
+
+            qmfBufferReal[i][hiBand] = (lowBandReal[LPC_ORDER+i]>>descale) + (accu1<<1);
+          }
+        }
+      } /* bw <= 0 */
+
+      patch++;
+
+    }  /* inner loop over patches */
+
+     /*
+     * store the unmodified filter coefficients if there is
+     * an overlapping envelope
+     *****************************************************************/
+
+
+  }  /* outer loop over bands (loBand) */
+
+  if (useLP)
+  {
+    for ( loBand = pSettings->lbStartPatching; loBand <  pSettings->lbStopPatching; loBand++ ) {
+      patch = 0;
+      while ( patch < pSettings->noOfPatches ) {
+
+        UCHAR hiBand = loBand + patchParam[patch].targetBandOffs;
+
+        if ( loBand < patchParam[patch].sourceStartBand
+          || loBand >= patchParam[patch].sourceStopBand
+          || hiBand >= (64)              /* Highband out of range (biterror) */
+          ) {
+          /* Lowband not in current patch or highband out of range (might be caused by biterrors)- proceed */
+          patch++;
+          continue;
+        }
+
+        if(hiBand != patchParam[patch].targetStartBand)
+          degreeAlias[hiBand] = degreeAlias[loBand];
+
+        patch++;
+      }
+    }/* end  for loop */
+  }
+
+ for (i = 0; i < nInvfBands; i++ ) {
+   hLppTrans->bwVectorOld[i] = bwVector[i];
+ }
+
+  /*
+    set high band scale factor
+  */
+  sbrScaleFactor->hb_scale = comLowBandScale-(LPC_SCALE_FACTOR);
+
+}
+
+/*!
+ *
+ * \brief Initialize one low power transposer instance
+ *
+ *
+ */
+SBR_ERROR
+createLppTransposer (HANDLE_SBR_LPP_TRANS hs, /*!< Handle of low power transposer  */
+                     TRANSPOSER_SETTINGS *pSettings, /*!< Pointer to settings */
+                     const int  highBandStartSb, /*!< ? */
+                     UCHAR *v_k_master,         /*!< Master table */
+                     const int numMaster,       /*!< Valid entries in master table */
+                     const int usb,             /*!< Highband area stop subband */
+                     const int timeSlots,       /*!< Number of time slots */
+                     const int nCols,           /*!< Number of colums (codec qmf bank) */
+                     UCHAR *noiseBandTable,     /*!< Mapping of SBR noise bands to QMF bands */
+                     const int  noNoiseBands,   /*!< Number of noise bands */
+                     UINT   fs,                 /*!< Sample Frequency */
+                     const int chan,            /*!< Channel number */
+                     const int overlap
+                     )
+{
+  /* FB inverse filtering settings */
+  hs->pSettings = pSettings;
+
+  pSettings->nCols = nCols;
+  pSettings->overlap = overlap;
+
+  switch (timeSlots) {
+
+  case 15:
+  case 16:
+    break;
+
+  default:
+    return SBRDEC_UNSUPPORTED_CONFIG; /* Unimplemented */
+  }
+
+  if (chan==0) {
+    /* Init common data only once */
+    hs->pSettings->nCols = nCols;
+
+    return resetLppTransposer (hs,
+                               highBandStartSb,
+                               v_k_master,
+                               numMaster,
+                               noiseBandTable,
+                               noNoiseBands,
+                               usb,
+                               fs);
+  }
+  return SBRDEC_OK;
+}
+
+
+static int findClosestEntry(UCHAR goalSb, UCHAR *v_k_master, UCHAR numMaster, UCHAR direction)
+{
+  int index;
+
+  if( goalSb <= v_k_master[0] )
+    return v_k_master[0];
+
+  if( goalSb >= v_k_master[numMaster] )
+    return v_k_master[numMaster];
+
+  if(direction) {
+    index = 0;
+    while( v_k_master[index] < goalSb ) {
+      index++;
+    }
+  } else {
+    index = numMaster;
+    while( v_k_master[index] > goalSb ) {
+      index--;
+    }
+  }
+
+  return v_k_master[index];
+}
+
+
+/*!
+ *
+ * \brief Reset memory for one lpp transposer instance
+ *
+ * \return SBRDEC_OK on success, SBRDEC_UNSUPPORTED_CONFIG on error
+ */
+SBR_ERROR
+resetLppTransposer (HANDLE_SBR_LPP_TRANS hLppTrans,  /*!< Handle of lpp transposer  */
+                    UCHAR  highBandStartSb,          /*!< High band area: start subband */
+                    UCHAR *v_k_master,               /*!< Master table */
+                    UCHAR  numMaster,                /*!< Valid entries in master table */
+                    UCHAR *noiseBandTable,           /*!< Mapping of SBR noise bands to QMF bands */
+                    UCHAR  noNoiseBands,             /*!< Number of noise bands */
+                    UCHAR  usb,                      /*!< High band area: stop subband */
+                    UINT   fs                        /*!< SBR output sampling frequency */
+                    )
+{
+  TRANSPOSER_SETTINGS *pSettings = hLppTrans->pSettings;
+  PATCH_PARAM  *patchParam = pSettings->patchParam;
+
+  int i, patch;
+  int targetStopBand;
+  int sourceStartBand;
+  int patchDistance;
+  int numBandsInPatch;
+
+  int lsb = v_k_master[0];                 /* Start subband expressed in "non-critical" sampling terms*/
+  int xoverOffset = highBandStartSb - lsb; /* Calculate distance in QMF bands between k0 and kx */
+  int startFreqHz;
+
+  int desiredBorder;
+
+  usb = fixMin(usb, v_k_master[numMaster]); /* Avoid endless loops (compare with float code). */
+
+  /*
+   * Plausibility check
+   */
+
+  if ( lsb - SHIFT_START_SB < 4 ) {
+    return SBRDEC_UNSUPPORTED_CONFIG;
+  }
+
+
+  /*
+   * Initialize the patching parameter
+   */
+ desiredBorder = 21;
+ if (fs < 92017) {
+    desiredBorder = 23;
+  }
+  if (fs < 75132) {
+    desiredBorder = 32;
+  }
+  if (fs < 55426) {
+    desiredBorder = 43;
+  }
+  if (fs < 46009) {
+    desiredBorder = 46;
+  }
+  if (fs < 35777) {
+    desiredBorder = 64;
+  }
+
+  desiredBorder = findClosestEntry(desiredBorder, v_k_master, numMaster, 1); /* Adapt region to master-table */
+
+  /* First patch */
+  sourceStartBand = SHIFT_START_SB + xoverOffset;
+  targetStopBand = lsb + xoverOffset; /* upperBand */
+
+  /* Even (odd) numbered channel must be patched to even (odd) numbered channel */
+  patch = 0;
+  while(targetStopBand < usb) {
+
+    /* Too many patches?
+       Allow MAX_NUM_PATCHES+1 patches here.
+       we need to check later again, since patch might be the highest patch
+       AND contain less than 3 bands => actual number of patches will be reduced by 1.
+    */
+    if (patch > MAX_NUM_PATCHES) {
+      return SBRDEC_UNSUPPORTED_CONFIG;
+    }
+
+    patchParam[patch].guardStartBand = targetStopBand;
+    patchParam[patch].targetStartBand = targetStopBand;
+
+    numBandsInPatch = desiredBorder - targetStopBand;                   /* Get the desired range of the patch */
+
+    if ( numBandsInPatch >= lsb - sourceStartBand ) {
+      /* Desired number bands are not available -> patch whole source range */
+      patchDistance   = targetStopBand - sourceStartBand;        /* Get the targetOffset */
+      patchDistance   = patchDistance & ~1;                      /* Rounding off odd numbers and make all even */
+      numBandsInPatch = lsb - (targetStopBand - patchDistance);  /* Update number of bands to be patched */
+      numBandsInPatch = findClosestEntry(targetStopBand + numBandsInPatch, v_k_master, numMaster, 0) -
+                        targetStopBand;  /* Adapt region to master-table */
+    }
+
+    /* Desired number bands are available -> get the minimal even patching distance */
+    patchDistance   = numBandsInPatch + targetStopBand - lsb;  /* Get minimal distance */
+    patchDistance   = (patchDistance + 1) & ~1;                /* Rounding up odd numbers and make all even */
+
+    if (numBandsInPatch > 0) {
+      patchParam[patch].sourceStartBand = targetStopBand - patchDistance;
+      patchParam[patch].targetBandOffs  = patchDistance;
+      patchParam[patch].numBandsInPatch = numBandsInPatch;
+      patchParam[patch].sourceStopBand  = patchParam[patch].sourceStartBand + numBandsInPatch;
+
+      targetStopBand += patchParam[patch].numBandsInPatch;
+      patch++;
+    }
+
+    /* All patches but first */
+    sourceStartBand = SHIFT_START_SB;
+
+    /* Check if we are close to desiredBorder */
+    if( desiredBorder - targetStopBand < 3)  /* MPEG doc */
+    {
+      desiredBorder = usb;
+    }
+
+  }
+
+  patch--;
+
+  /* If highest patch contains less than three subband: skip it */
+  if ( (patch>0) && (patchParam[patch].numBandsInPatch < 3) ) {
+    patch--;
+    targetStopBand = patchParam[patch].targetStartBand + patchParam[patch].numBandsInPatch;
+  }
+
+  /* now check if we don't have one too many */
+  if (patch >= MAX_NUM_PATCHES) {
+    return SBRDEC_UNSUPPORTED_CONFIG;
+  }
+
+  pSettings->noOfPatches = patch + 1;
+
+  /* Check lowest and highest source subband */
+  pSettings->lbStartPatching = targetStopBand;
+  pSettings->lbStopPatching  = 0;
+  for ( patch = 0; patch < pSettings->noOfPatches; patch++ ) {
+    pSettings->lbStartPatching = fixMin( pSettings->lbStartPatching, patchParam[patch].sourceStartBand );
+    pSettings->lbStopPatching  = fixMax( pSettings->lbStopPatching, patchParam[patch].sourceStopBand );
+  }
+
+  for(i = 0 ; i < noNoiseBands; i++){
+    pSettings->bwBorders[i] = noiseBandTable[i+1];
+  }
+
+  /*
+   * Choose whitening factors
+   */
+
+  startFreqHz = ( (lsb + xoverOffset)*fs ) >> 7;  /* Shift does a division by 2*(64) */
+
+  for( i = 1; i < NUM_WHFACTOR_TABLE_ENTRIES; i++ )
+  {
+    if( startFreqHz < FDK_sbrDecoder_sbr_whFactorsIndex[i])
+      break;
+  }
+  i--;
+
+  pSettings->whFactors.off = FDK_sbrDecoder_sbr_whFactorsTable[i][0];
+  pSettings->whFactors.transitionLevel = FDK_sbrDecoder_sbr_whFactorsTable[i][1];
+  pSettings->whFactors.lowLevel = FDK_sbrDecoder_sbr_whFactorsTable[i][2];
+  pSettings->whFactors.midLevel = FDK_sbrDecoder_sbr_whFactorsTable[i][3];
+  pSettings->whFactors.highLevel = FDK_sbrDecoder_sbr_whFactorsTable[i][4];
+
+  return SBRDEC_OK;
+}
diff --git a/libSBRdec/src/lpp_tran.h b/libSBRdec/src/lpp_tran.h
new file mode 100644
index 0000000..1838c06
--- /dev/null
+++ b/libSBRdec/src/lpp_tran.h
@@ -0,0 +1,242 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/*!
+  \file
+  \brief  Low Power Profile Transposer,  
+*/
+
+#ifndef _LPP_TRANS_H
+#define _LPP_TRANS_H
+
+#include "sbrdecoder.h"
+#include "qmf.h"
+
+/*
+  Common
+*/
+#define QMF_OUT_SCALE            8
+
+/*
+  Env-Adjust
+*/
+#define MAX_NOISE_ENVELOPES      2
+#define MAX_NOISE_COEFFS         5
+#define MAX_NUM_NOISE_VALUES     (MAX_NOISE_ENVELOPES * MAX_NOISE_COEFFS)
+#define MAX_NUM_LIMITERS         12
+
+/* Set MAX_ENVELOPES to the largest value of all supported BSFORMATs
+   by overriding MAX_ENVELOPES in the correct order: */
+#define MAX_ENVELOPES_HEAAC      5
+#define MAX_ENVELOPES            MAX_ENVELOPES_HEAAC
+
+#define MAX_FREQ_COEFFS          48
+#define MAX_FREQ_COEFFS_FS44100  35
+#define MAX_FREQ_COEFFS_FS48000  32
+
+
+#define MAX_NUM_ENVELOPE_VALUES  (MAX_ENVELOPES * MAX_FREQ_COEFFS)
+
+#define MAX_GAIN_EXP             34
+/* Maximum gain will be sqrt(0.5 * 2^MAX_GAIN_EXP)
+   example: 34=99dB   */
+#define MAX_GAIN_CONCEAL_EXP     1
+/* Maximum gain will be sqrt(0.5 * 2^MAX_GAIN_CONCEAL_EXP) in concealment case (0dB) */
+
+/*
+  LPP Transposer
+*/
+#define LPC_ORDER                2
+
+#define MAX_INVF_BANDS           MAX_NOISE_COEFFS
+
+#define MAX_NUM_PATCHES          6
+#define SHIFT_START_SB           1              /*!< lowest subband of source range */
+
+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;
+
+
+/** parameter set for one single patch */
+typedef struct {
+  UCHAR  sourceStartBand;   /*!< first band in lowbands where to take the samples from */
+  UCHAR  sourceStopBand;    /*!< first band in lowbands which is not included in the patch anymore */
+  UCHAR  guardStartBand;    /*!< first band in highbands to be filled with zeros in order to
+                                         reduce interferences between patches */
+  UCHAR  targetStartBand;   /*!< first band in highbands to be filled with whitened lowband signal */
+  UCHAR  targetBandOffs;    /*!< difference between 'startTargetBand' and 'startSourceBand' */
+  UCHAR  numBandsInPatch;   /*!< number of consecutive bands in this one patch */
+} PATCH_PARAM;
+
+
+/** whitening factors for different levels of whitening
+    need to be initialized corresponding to crossover frequency */
+typedef struct {
+  FIXP_DBL  off;                       /*!< bw factor for signal OFF */
+  FIXP_DBL  transitionLevel;
+  FIXP_DBL  lowLevel;                  /*!< bw factor for signal LOW_LEVEL */
+  FIXP_DBL  midLevel;                  /*!< bw factor for signal MID_LEVEL */
+  FIXP_DBL  highLevel;                 /*!< bw factor for signal HIGH_LEVEL */
+} WHITENING_FACTORS;
+
+
+/*! The transposer settings are calculated on a header reset and are shared by both channels. */
+typedef struct {
+  UCHAR  nCols;                       /*!< number subsamples of a codec frame */
+  UCHAR  noOfPatches;                 /*!< number of patches */
+  UCHAR  lbStartPatching;             /*!< first band of lowbands that will be patched */
+  UCHAR  lbStopPatching;              /*!< first band that won't be patched anymore*/
+  UCHAR  bwBorders[MAX_NUM_NOISE_VALUES]; /*!< spectral bands with different inverse filtering levels */
+
+  PATCH_PARAM    patchParam[MAX_NUM_PATCHES]; /*!< new parameter set for patching */
+  WHITENING_FACTORS whFactors;                /*!< the pole moving factors for certain whitening levels as indicated
+                                                   in the bitstream depending on the crossover frequency */
+  UCHAR  overlap;                             /*!< Overlap size */
+} TRANSPOSER_SETTINGS;
+
+
+typedef struct
+{
+  TRANSPOSER_SETTINGS *pSettings;                /*!< Common settings for both channels */
+  FIXP_DBL  bwVectorOld[MAX_NUM_PATCHES];        /*!< pole moving factors of past frame */
+  FIXP_DBL  lpcFilterStatesReal[LPC_ORDER][(32)];     /*!< pointer array to save filter states */
+  FIXP_DBL  lpcFilterStatesImag[LPC_ORDER][(32)];     /*!< pointer array to save filter states */
+}
+SBR_LPP_TRANS;
+
+typedef SBR_LPP_TRANS *HANDLE_SBR_LPP_TRANS;
+
+
+void lppTransposer (HANDLE_SBR_LPP_TRANS hLppTrans,
+                    QMF_SCALE_FACTOR  *sbrScaleFactor,
+                    FIXP_DBL **qmfBufferReal,
+
+                    FIXP_DBL *degreeAlias,
+                    FIXP_DBL **qmfBufferImag,
+                    const int useLP,
+                    const int timeStep,
+                    const int firstSlotOffset,
+                    const int lastSlotOffset,
+                    const int nInvfBands,
+                    INVF_MODE *sbr_invf_mode,
+                    INVF_MODE *sbr_invf_mode_prev
+                    );
+
+
+SBR_ERROR
+createLppTransposer (HANDLE_SBR_LPP_TRANS hLppTrans,
+                     TRANSPOSER_SETTINGS *pSettings,
+                     const int  highBandStartSb,
+                     UCHAR *v_k_master,
+                     const int  numMaster,
+                     const int  usb,
+                     const int  timeSlots,
+                     const int  nCols,
+                     UCHAR *noiseBandTable,
+                     const int  noNoiseBands,
+                     UINT   fs,
+                     const int  chan,
+                     const int overlap);
+
+
+SBR_ERROR
+resetLppTransposer (HANDLE_SBR_LPP_TRANS hLppTrans,
+                    UCHAR  highBandStartSb,
+                    UCHAR *v_k_master,
+                    UCHAR  numMaster,
+                    UCHAR *noiseBandTable,
+                    UCHAR  noNoiseBands,
+                    UCHAR  usb,
+                    UINT   fs);
+
+
+
+#endif /* _LPP_TRANS_H */
+
diff --git a/libSBRdec/src/psbitdec.cpp b/libSBRdec/src/psbitdec.cpp
new file mode 100644
index 0000000..dfd532f
--- /dev/null
+++ b/libSBRdec/src/psbitdec.cpp
@@ -0,0 +1,593 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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 "psbitdec.h"
+
+
+#include "sbr_rom.h"
+#include "huff_dec.h"
+
+/* PS dec privat functions */
+SBR_ERROR ResetPsDec(HANDLE_PS_DEC h_ps_d);
+void ResetPsDeCor   (HANDLE_PS_DEC h_ps_d);
+
+/***************************************************************************/
+/*!
+  \brief  huffman decoding by codebook table
+
+  \return index of huffman codebook table
+
+****************************************************************************/
+static SCHAR
+decode_huff_cw (Huffman h,                      /*!< pointer to huffman codebook table */
+                HANDLE_FDK_BITSTREAM hBitBuf,   /*!< Handle to Bitbuffer */
+                int *length)                    /*!< length of huffman codeword (or NULL) */
+{
+  UCHAR bit = 0;
+  SCHAR index = 0;
+  UCHAR bitCount = 0;
+
+  while (index >= 0) {
+    bit = FDKreadBits (hBitBuf, 1);
+    bitCount++;
+    index = h[index][bit];
+  }
+  if (length) {
+    *length = bitCount;
+  }
+  return( index+64 ); /* Add offset */
+}
+
+/***************************************************************************/
+/*!
+  \brief  helper function - limiting of value to min/max values
+
+  \return limited value
+
+****************************************************************************/
+
+static SCHAR
+limitMinMax(SCHAR i,
+            SCHAR min,
+            SCHAR max)
+{
+  if (i<min)
+    return min;
+  else if (i>max)
+    return max;
+  else
+    return i;
+}
+
+/***************************************************************************/
+/*!
+  \brief  Decodes delta values in-place and updates
+          data buffers according to quantization classes.
+
+  When delta coded in frequency the first element is deltacode from zero.
+  aIndex buffer is decoded from delta values to actual values.
+
+  \return none
+
+****************************************************************************/
+static void
+deltaDecodeArray(SCHAR enable,
+                 SCHAR *aIndex,          /*!< ICC/IID parameters */
+                 SCHAR *aPrevFrameIndex, /*!< ICC/IID parameters  of previous frame */
+                 SCHAR DtDf,
+                 UCHAR nrElements,       /*!< as conveyed in bitstream */
+                                         /*!< output array size: nrElements*stride */
+                 UCHAR stride,           /*!< 1=dflt, 2=half freq. resolution */
+                 SCHAR minIdx,
+                 SCHAR maxIdx)
+{
+  int i;
+
+  /* Delta decode */
+  if ( enable==1 ) {
+    if (DtDf == 0)  {   /* Delta coded in freq */
+      aIndex[0] = 0 + aIndex[0];
+      aIndex[0] = limitMinMax(aIndex[0],minIdx,maxIdx);
+      for (i = 1; i < nrElements; i++) {
+        aIndex[i] = aIndex[i-1] + aIndex[i];
+        aIndex[i] = limitMinMax(aIndex[i],minIdx,maxIdx);
+      }
+    }
+    else { /* Delta time */
+      for (i = 0; i < nrElements; i++) {
+        aIndex[i] = aPrevFrameIndex[i*stride] + aIndex[i];
+        aIndex[i] = limitMinMax(aIndex[i],minIdx,maxIdx);
+      }
+    }
+  }
+  else { /* No data is sent, set index to zero */
+    for (i = 0; i < nrElements; i++) {
+      aIndex[i] = 0;
+    }
+  }
+  if (stride==2) {
+    for (i=nrElements*stride-1; i>0; i--) {
+      aIndex[i] = aIndex[i>>1];
+    }
+  }
+}
+
+/***************************************************************************/
+/*!
+  \brief Mapping of ICC/IID parameters to 20 stereo bands
+
+  \return none
+
+****************************************************************************/
+static void map34IndexTo20 (SCHAR *aIndex, /*!< decoded ICC/IID parameters */
+                            UCHAR noBins)  /*!< number of stereo bands     */
+{
+  aIndex[0]  = (2*aIndex[0]+aIndex[1])/3;
+  aIndex[1]  = (aIndex[1]+2*aIndex[2])/3;
+  aIndex[2]  = (2*aIndex[3]+aIndex[4])/3;
+  aIndex[3]  = (aIndex[4]+2*aIndex[5])/3;
+  aIndex[4]  = (aIndex[6]+aIndex[7])/2;
+  aIndex[5]  = (aIndex[8]+aIndex[9])/2;
+  aIndex[6]  = aIndex[10];
+  aIndex[7]  = aIndex[11];
+  aIndex[8]  = (aIndex[12]+aIndex[13])/2;
+  aIndex[9]  = (aIndex[14]+aIndex[15])/2;
+  aIndex[10] = aIndex[16];
+  /* For IPD/OPD it stops here */
+
+  if (noBins == NO_HI_RES_BINS)
+  {
+    aIndex[11] = aIndex[17];
+    aIndex[12] = aIndex[18];
+    aIndex[13] = aIndex[19];
+    aIndex[14] = (aIndex[20]+aIndex[21])/2;
+    aIndex[15] = (aIndex[22]+aIndex[23])/2;
+    aIndex[16] = (aIndex[24]+aIndex[25])/2;
+    aIndex[17] = (aIndex[26]+aIndex[27])/2;
+    aIndex[18] = (aIndex[28]+aIndex[29]+aIndex[30]+aIndex[31])/4;
+    aIndex[19] = (aIndex[32]+aIndex[33])/2;
+  }
+}
+
+/***************************************************************************/
+/*!
+  \brief  Decodes delta coded IID, ICC, IPD and OPD indices
+
+  \return PS processing flag. If set to 1
+
+****************************************************************************/
+int
+DecodePs( struct PS_DEC *h_ps_d,      /*!< PS handle */
+          const UCHAR    frameError ) /*!< Flag telling that frame had errors */
+{
+  MPEG_PS_BS_DATA *pBsData;
+  UCHAR gr, env;
+  int   bPsHeaderValid, bPsDataAvail;
+
+  /* Shortcuts to avoid deferencing and keep the code readable */
+  pBsData = &h_ps_d->bsData[h_ps_d->processSlot].mpeg;
+  bPsHeaderValid = pBsData->bPsHeaderValid;
+  bPsDataAvail = (h_ps_d->bPsDataAvail[h_ps_d->processSlot] == ppt_mpeg) ? 1 : 0;
+
+ /***************************************************************************************
+  * Decide whether to process or to conceal PS data or not.                             */
+
+  if ( ( h_ps_d->psDecodedPrv && !frameError && !bPsDataAvail)
+    || (!h_ps_d->psDecodedPrv && (frameError || !bPsDataAvail || !bPsHeaderValid)) ) {
+    /* Don't apply PS processing.
+     * Declare current PS header and bitstream data invalid. */
+    pBsData->bPsHeaderValid = 0;
+    h_ps_d->bPsDataAvail[h_ps_d->processSlot] = ppt_none;
+    return (0);
+  }
+
+  if (frameError || !bPsHeaderValid)
+  { /* no new PS data available (e.g. frame loss) */
+    /* => keep latest data constant (i.e. FIX with noEnv=0) */
+    pBsData->noEnv = 0;
+  }
+
+ /***************************************************************************************
+  * Decode bitstream payload or prepare parameter for concealment:
+  */
+  for (env=0; env<pBsData->noEnv; env++) {
+    SCHAR *aPrevIidIndex;
+    SCHAR *aPrevIccIndex;
+
+    UCHAR noIidSteps = pBsData->bFineIidQ?NO_IID_STEPS_FINE:NO_IID_STEPS;
+
+    if (env==0) {
+      aPrevIidIndex = h_ps_d->specificTo.mpeg.aIidPrevFrameIndex;
+      aPrevIccIndex = h_ps_d->specificTo.mpeg.aIccPrevFrameIndex;
+    }
+    else {
+      aPrevIidIndex = pBsData->aaIidIndex[env-1];
+      aPrevIccIndex = pBsData->aaIccIndex[env-1];
+    }
+
+    deltaDecodeArray(pBsData->bEnableIid,
+                     pBsData->aaIidIndex[env],
+                     aPrevIidIndex,
+                     pBsData->abIidDtFlag[env],
+                     FDK_sbrDecoder_aNoIidBins[pBsData->freqResIid],
+                     (pBsData->freqResIid)?1:2,
+                     -noIidSteps,
+                     noIidSteps);
+
+    deltaDecodeArray(pBsData->bEnableIcc,
+                     pBsData->aaIccIndex[env],
+                     aPrevIccIndex,
+                     pBsData->abIccDtFlag[env],
+                     FDK_sbrDecoder_aNoIccBins[pBsData->freqResIcc],
+                     (pBsData->freqResIcc)?1:2,
+                     0,
+                     NO_ICC_STEPS-1);
+  }   /* for (env=0; env<pBsData->noEnv; env++) */
+
+  /* handling of FIX noEnv=0 */
+  if (pBsData->noEnv==0) {
+    /* set noEnv=1, keep last parameters or force 0 if not enabled */
+    pBsData->noEnv = 1;
+
+    if (pBsData->bEnableIid) {
+      for (gr = 0; gr < NO_HI_RES_IID_BINS; gr++) {
+        pBsData->aaIidIndex[pBsData->noEnv-1][gr] =
+          h_ps_d->specificTo.mpeg.aIidPrevFrameIndex[gr];
+      }
+    }
+    else {
+      for (gr = 0; gr < NO_HI_RES_IID_BINS; gr++) {
+        pBsData->aaIidIndex[pBsData->noEnv-1][gr] = 0;
+      }
+    }
+
+    if (pBsData->bEnableIcc) {
+      for (gr = 0; gr < NO_HI_RES_ICC_BINS; gr++) {
+        pBsData->aaIccIndex[pBsData->noEnv-1][gr] =
+          h_ps_d->specificTo.mpeg.aIccPrevFrameIndex[gr];
+      }
+    }
+    else {
+      for (gr = 0; gr < NO_HI_RES_ICC_BINS; gr++) {
+        pBsData->aaIccIndex[pBsData->noEnv-1][gr] = 0;
+      }
+    }
+  }
+
+  /* Update previous frame index buffers */
+  for (gr = 0; gr < NO_HI_RES_IID_BINS; gr++) {
+    h_ps_d->specificTo.mpeg.aIidPrevFrameIndex[gr] =
+      pBsData->aaIidIndex[pBsData->noEnv-1][gr];
+  }
+  for (gr = 0; gr < NO_HI_RES_ICC_BINS; gr++) {
+    h_ps_d->specificTo.mpeg.aIccPrevFrameIndex[gr] =
+      pBsData->aaIccIndex[pBsData->noEnv-1][gr];
+  }
+
+  /* PS data from bitstream (if avail) was decoded now */
+  h_ps_d->bPsDataAvail[h_ps_d->processSlot] = ppt_none;
+
+  /* handling of env borders for FIX & VAR */
+  if (pBsData->bFrameClass == 0) {
+    /* FIX_BORDERS NoEnv=0,1,2,4 */
+    pBsData->aEnvStartStop[0] = 0;
+    for (env=1; env<pBsData->noEnv; env++) {
+      pBsData->aEnvStartStop[env] =
+        (env * h_ps_d->noSubSamples) / pBsData->noEnv;
+    }
+    pBsData->aEnvStartStop[pBsData->noEnv] = h_ps_d->noSubSamples;
+    /* 1024 (32 slots) env borders:  0, 8, 16, 24, 32 */
+    /*  960 (30 slots) env borders:  0, 7, 15, 22, 30 */
+  }
+  else {   /* if (h_ps_d->bFrameClass == 0) */
+    /* VAR_BORDERS NoEnv=1,2,3,4 */
+    pBsData->aEnvStartStop[0] = 0;
+
+    /* handle case aEnvStartStop[noEnv]<noSubSample for VAR_BORDERS by
+       duplicating last PS parameters and incrementing noEnv */
+    if (pBsData->aEnvStartStop[pBsData->noEnv] < h_ps_d->noSubSamples) {
+      for (gr = 0; gr < NO_HI_RES_IID_BINS; gr++) {
+        pBsData->aaIidIndex[pBsData->noEnv][gr] =
+          pBsData->aaIidIndex[pBsData->noEnv-1][gr];
+      }
+      for (gr = 0; gr < NO_HI_RES_ICC_BINS; gr++) {
+        pBsData->aaIccIndex[pBsData->noEnv][gr] =
+          pBsData->aaIccIndex[pBsData->noEnv-1][gr];
+      }
+      pBsData->noEnv++;
+      pBsData->aEnvStartStop[pBsData->noEnv] = h_ps_d->noSubSamples;
+    }
+
+    /* enforce strictly monotonic increasing borders */
+    for (env=1; env<pBsData->noEnv; env++) {
+      UCHAR thr;
+      thr = (UCHAR)h_ps_d->noSubSamples - (pBsData->noEnv - env);
+      if (pBsData->aEnvStartStop[env] > thr) {
+        pBsData->aEnvStartStop[env] = thr;
+      }
+      else {
+        thr = pBsData->aEnvStartStop[env-1]+1;
+        if (pBsData->aEnvStartStop[env] < thr) {
+          pBsData->aEnvStartStop[env] = thr;
+        }
+      }
+    }
+  }   /* if (h_ps_d->bFrameClass == 0) ... else */
+
+  /* copy data prior to possible 20<->34 in-place mapping */
+  for (env=0; env<pBsData->noEnv; env++) {
+    UCHAR i;
+    for (i=0; i<NO_HI_RES_IID_BINS; i++) {
+      h_ps_d->specificTo.mpeg.coef.aaIidIndexMapped[env][i] = pBsData->aaIidIndex[env][i];
+    }
+    for (i=0; i<NO_HI_RES_ICC_BINS; i++) {
+      h_ps_d->specificTo.mpeg.coef.aaIccIndexMapped[env][i] = pBsData->aaIccIndex[env][i];
+    }
+  }
+
+
+  /* MPEG baseline PS */
+  /* Baseline version of PS always uses the hybrid filter structure with 20 stereo bands. */
+  /* If ICC/IID parameters for 34 stereo bands are decoded they have to be mapped to 20   */
+  /* stereo bands.                                                                        */
+  /* Additionaly the IPD/OPD parameters won't be used.                                    */
+
+  for (env=0; env<pBsData->noEnv; env++) {
+    if (pBsData->freqResIid == 2)
+      map34IndexTo20 (h_ps_d->specificTo.mpeg.coef.aaIidIndexMapped[env], NO_HI_RES_IID_BINS);
+    if (pBsData->freqResIcc == 2)
+      map34IndexTo20 (h_ps_d->specificTo.mpeg.coef.aaIccIndexMapped[env], NO_HI_RES_ICC_BINS);
+
+    /* IPD/OPD is disabled in baseline version and thus was removed here */
+  }
+
+  return (1);
+}
+
+
+/***************************************************************************/
+/*!
+
+  \brief  Reads parametric stereo data from bitstream
+
+  \return
+
+****************************************************************************/
+unsigned int
+ReadPsData (HANDLE_PS_DEC h_ps_d,          /*!< handle to struct PS_DEC */
+            HANDLE_FDK_BITSTREAM hBitBuf,  /*!< handle to struct BIT_BUF */
+            int nBitsLeft                  /*!< max number of bits available */
+           )
+{
+  MPEG_PS_BS_DATA *pBsData;
+
+  UCHAR     gr, env;
+  SCHAR     dtFlag;
+  INT       startbits;
+  Huffman   CurrentTable;
+  SCHAR     bEnableHeader;
+
+  if (!h_ps_d)
+    return 0;
+
+  pBsData = &h_ps_d->bsData[h_ps_d->bsReadSlot].mpeg;
+
+  if (h_ps_d->bsReadSlot != h_ps_d->bsLastSlot) {
+    /* Copy last header data */
+    FDKmemcpy(pBsData, &h_ps_d->bsData[h_ps_d->bsLastSlot].mpeg, sizeof(MPEG_PS_BS_DATA));
+  }
+
+
+  startbits = (INT) FDKgetValidBits(hBitBuf);
+
+  bEnableHeader = (SCHAR) FDKreadBits (hBitBuf, 1);
+
+  /* Read header */
+  if (bEnableHeader) {
+    pBsData->bPsHeaderValid = 1;
+    pBsData->bEnableIid = (UCHAR) FDKreadBits (hBitBuf, 1);
+    if (pBsData->bEnableIid) {
+      pBsData->modeIid = (UCHAR) FDKreadBits (hBitBuf, 3);
+    }
+
+    pBsData->bEnableIcc = (UCHAR) FDKreadBits (hBitBuf, 1);
+    if (pBsData->bEnableIcc) {
+      pBsData->modeIcc = (UCHAR) FDKreadBits (hBitBuf, 3);
+    }
+
+    pBsData->bEnableExt = (UCHAR) FDKreadBits (hBitBuf, 1);
+  }
+
+  pBsData->bFrameClass = (UCHAR) FDKreadBits (hBitBuf, 1);
+  if (pBsData->bFrameClass == 0) {
+    /* FIX_BORDERS NoEnv=0,1,2,4 */
+    pBsData->noEnv = FDK_sbrDecoder_aFixNoEnvDecode[(UCHAR) FDKreadBits (hBitBuf, 2)];
+    /* all additional handling of env borders is now in DecodePs() */
+  }
+  else {
+    /* VAR_BORDERS NoEnv=1,2,3,4 */
+    pBsData->noEnv = 1+(UCHAR) FDKreadBits (hBitBuf, 2);
+    for (env=1; env<pBsData->noEnv+1; env++)
+      pBsData->aEnvStartStop[env] = ((UCHAR) FDKreadBits (hBitBuf, 5)) + 1;
+    /* all additional handling of env borders is now in DecodePs() */
+  }
+
+  /* verify that IID & ICC modes (quant grid, freq res) are supported */
+  if ((pBsData->modeIid > 5) || (pBsData->modeIcc > 5)) {
+    /* no useful PS data could be read from bitstream */
+    h_ps_d->bPsDataAvail[h_ps_d->bsReadSlot] = ppt_none;
+    /* discard all remaining bits */
+    nBitsLeft -= startbits - FDKgetValidBits(hBitBuf);
+    while (nBitsLeft) {
+      int i = nBitsLeft;
+      if (i>8) {
+        i = 8;
+      }
+      FDKreadBits (hBitBuf, i);
+      nBitsLeft -= i;
+    }
+    return (startbits - FDKgetValidBits(hBitBuf));
+  }
+
+  if (pBsData->modeIid > 2){
+    pBsData->freqResIid = pBsData->modeIid-3;
+    pBsData->bFineIidQ = 1;
+  }
+  else{
+    pBsData->freqResIid = pBsData->modeIid;
+    pBsData->bFineIidQ = 0;
+  }
+
+  if (pBsData->modeIcc > 2){
+    pBsData->freqResIcc = pBsData->modeIcc-3;
+  }
+  else{
+    pBsData->freqResIcc = pBsData->modeIcc;
+  }
+
+
+  /* Extract IID data */
+  if (pBsData->bEnableIid) {
+    for (env=0; env<pBsData->noEnv; env++) {
+      dtFlag = (SCHAR)FDKreadBits (hBitBuf, 1);
+      if (!dtFlag)
+      {
+        if (pBsData->bFineIidQ)
+          CurrentTable = (Huffman)&aBookPsIidFineFreqDecode;
+        else
+          CurrentTable = (Huffman)&aBookPsIidFreqDecode;
+      }
+      else
+      {
+        if (pBsData->bFineIidQ)
+         CurrentTable = (Huffman)&aBookPsIidFineTimeDecode;
+        else
+          CurrentTable = (Huffman)&aBookPsIidTimeDecode;
+      }
+
+      for (gr = 0; gr < FDK_sbrDecoder_aNoIidBins[pBsData->freqResIid]; gr++)
+        pBsData->aaIidIndex[env][gr] = decode_huff_cw(CurrentTable,hBitBuf,NULL);
+      pBsData->abIidDtFlag[env] = dtFlag;
+    }
+  }
+
+  /* Extract ICC data */
+  if (pBsData->bEnableIcc) {
+    for (env=0; env<pBsData->noEnv; env++) {
+      dtFlag = (SCHAR)FDKreadBits (hBitBuf, 1);
+      if (!dtFlag)
+        CurrentTable = (Huffman)&aBookPsIccFreqDecode;
+      else
+        CurrentTable = (Huffman)&aBookPsIccTimeDecode;
+
+      for (gr = 0; gr < FDK_sbrDecoder_aNoIccBins[pBsData->freqResIcc]; gr++)
+        pBsData->aaIccIndex[env][gr] = decode_huff_cw(CurrentTable,hBitBuf,NULL);
+      pBsData->abIccDtFlag[env] = dtFlag;
+    }
+  }
+
+  if (pBsData->bEnableExt) {
+
+    /*!
+    Decoders that support only the baseline version of the PS tool are allowed
+    to ignore the IPD/OPD data, but according header data has to be parsed.
+    ISO/IEC 14496-3 Subpart 8 Annex 4
+    */
+
+    int cnt = FDKreadBits(hBitBuf, PS_EXTENSION_SIZE_BITS);
+    if (cnt == (1<<PS_EXTENSION_SIZE_BITS)-1) {
+      cnt += FDKreadBits(hBitBuf, PS_EXTENSION_ESC_COUNT_BITS);
+    }
+    while (cnt--)
+      FDKreadBits(hBitBuf, 8);
+  }
+
+
+  /* new PS data was read from bitstream */
+  h_ps_d->bPsDataAvail[h_ps_d->bsReadSlot] = ppt_mpeg;
+
+
+
+  return (startbits - FDKgetValidBits(hBitBuf));
+}
+
diff --git a/libSBRdec/src/psbitdec.h b/libSBRdec/src/psbitdec.h
new file mode 100644
index 0000000..3b65468
--- /dev/null
+++ b/libSBRdec/src/psbitdec.h
@@ -0,0 +1,103 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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 __PSBITDEC_H
+#define __PSBITDEC_H
+
+#include "sbrdecoder.h"
+
+
+#include "psdec.h"
+
+
+unsigned int
+ReadPsData (struct PS_DEC  *h_ps_d,
+            HANDLE_FDK_BITSTREAM  hBs,
+            int nBitsLeft);
+
+int
+DecodePs(struct PS_DEC *h_ps_d,
+         const UCHAR    frameError);
+
+
+#endif /* __PSBITDEC_H */
diff --git a/libSBRdec/src/psdec.cpp b/libSBRdec/src/psdec.cpp
new file mode 100644
index 0000000..d494c65
--- /dev/null
+++ b/libSBRdec/src/psdec.cpp
@@ -0,0 +1,1414 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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  parametric stereo decoder  
+*/
+
+#include "psdec.h"
+
+
+
+#include "FDK_bitbuffer.h"
+#include "psdec_hybrid.h"
+
+#include "sbr_rom.h"
+#include "sbr_ram.h"
+
+#include "FDK_tools_rom.h"
+
+#include "genericStds.h"
+
+#include "FDK_trigFcts.h"
+
+
+/********************************************************************/
+/*                       MLQUAL DEFINES                             */
+/********************************************************************/
+
+  #define FRACT_ZERO FRACT_BITS-1
+/********************************************************************/
+
+SBR_ERROR ResetPsDec( HANDLE_PS_DEC h_ps_d );
+
+void ResetPsDeCor( HANDLE_PS_DEC h_ps_d );
+
+
+/***** HELPERS *****/
+
+static void assignTimeSlotsPS (FIXP_DBL *bufAdr, FIXP_DBL **bufPtr, const int numSlots, const int numChan);
+
+
+
+/*******************/
+
+#define DIV3 FL2FXCONST_DBL(1.f/3.f)     /* division 3.0 */
+#define DIV1_5 FL2FXCONST_DBL(2.f/3.f)   /* division 1.5 */
+
+/***************************************************************************/
+/*!
+  \brief  Creates one instance of the PS_DEC struct
+
+  \return Error info
+
+****************************************************************************/
+int
+CreatePsDec( HANDLE_PS_DEC *h_PS_DEC,   /*!< pointer to the module state */
+             int aacSamplesPerFrame
+           )
+{
+  SBR_ERROR errorInfo = SBRDEC_OK;
+  HANDLE_PS_DEC  h_ps_d;
+  int i;
+
+  if (*h_PS_DEC == NULL) {
+    /* Get ps dec ram */
+    h_ps_d = GetRam_ps_dec();
+    if (h_ps_d == NULL) {
+      errorInfo = SBRDEC_MEM_ALLOC_FAILED;
+      goto bail;
+    }
+  } else {
+    /* Reset an open instance */
+    h_ps_d = *h_PS_DEC;
+  }
+
+   /* initialisation */
+  switch (aacSamplesPerFrame) {
+  case 960:
+    h_ps_d->noSubSamples = 30;              /* col */
+    break;
+  case 1024:
+    h_ps_d->noSubSamples = 32;              /* col */
+    break;
+  default:
+    h_ps_d->noSubSamples = -1;
+    break;
+  }
+
+  if (h_ps_d->noSubSamples >  MAX_NUM_COL
+   || h_ps_d->noSubSamples <= 0)
+  {
+    goto bail;
+  }
+  h_ps_d->noChannels   = NO_QMF_CHANNELS;   /* row */
+
+  h_ps_d->psDecodedPrv   =  0;
+  h_ps_d->procFrameBased = -1;
+  for (i = 0; i < (1)+1; i++) {
+    h_ps_d->bPsDataAvail[i]  =  ppt_none;
+  }
+
+
+  for (i = 0; i < (1)+1; i++) {
+    FDKmemclear(&h_ps_d->bsData[i].mpeg, sizeof(MPEG_PS_BS_DATA));
+  }
+
+  errorInfo = ResetPsDec( h_ps_d );
+
+  if ( errorInfo != SBRDEC_OK )
+    goto bail;
+
+  ResetPsDeCor( h_ps_d );
+
+  *h_PS_DEC = h_ps_d;
+
+
+
+  return 0;
+
+bail:
+  DeletePsDec(&h_ps_d);
+
+  return -1;
+} /*END CreatePsDec */
+
+/***************************************************************************/
+/*!
+  \brief  Delete one instance of the PS_DEC struct
+
+  \return Error info
+
+****************************************************************************/
+int
+DeletePsDec( HANDLE_PS_DEC *h_PS_DEC)  /*!< pointer to the module state */
+{
+  if (*h_PS_DEC == NULL) {
+    return -1;
+  }
+
+
+  FreeRam_ps_dec(h_PS_DEC);
+
+
+  return 0;
+} /*END DeletePsDec */
+
+/***************************************************************************/
+/*!
+  \brief resets some values of the PS handle to default states
+
+  \return
+
+****************************************************************************/
+SBR_ERROR ResetPsDec( HANDLE_PS_DEC h_ps_d )  /*!< pointer to the module state */
+{
+  SBR_ERROR errorInfo = SBRDEC_OK;
+  INT i;
+
+  const UCHAR noQmfBandsInHybrid20 = 3;
+  /* const UCHAR noQmfBandsInHybrid34 = 5; */
+
+  const UCHAR aHybridResolution20[] = { HYBRID_8_CPLX,
+                                        HYBRID_2_REAL,
+                                        HYBRID_2_REAL };
+
+  h_ps_d->specificTo.mpeg.delayBufIndex   = 0;
+
+  /* explicitly init state variables to safe values (until first ps header arrives) */
+
+  h_ps_d->specificTo.mpeg.lastUsb        =  0;
+
+  h_ps_d->specificTo.mpeg.scaleFactorPsDelayBuffer = -(DFRACT_BITS-1);
+
+  FDKmemclear(h_ps_d->specificTo.mpeg.aDelayBufIndexDelayQmf, (NO_QMF_CHANNELS-FIRST_DELAY_SB)*sizeof(UCHAR));
+  h_ps_d->specificTo.mpeg.noSampleDelay = delayIndexQmf[0];
+
+  for (i=0 ; i < NO_SERIAL_ALLPASS_LINKS; i++) {
+    h_ps_d->specificTo.mpeg.aDelayRBufIndexSer[i] = 0;
+  }
+
+  h_ps_d->specificTo.mpeg.pAaRealDelayBufferQmf[0] = h_ps_d->specificTo.mpeg.aaQmfDelayBufReal;
+
+  assignTimeSlotsPS ( h_ps_d->specificTo.mpeg.pAaRealDelayBufferQmf[0] + (NO_QMF_CHANNELS-FIRST_DELAY_SB),
+                     &h_ps_d->specificTo.mpeg.pAaRealDelayBufferQmf[1],
+                      h_ps_d->specificTo.mpeg.noSampleDelay-1,
+                      (NO_DELAY_BUFFER_BANDS-FIRST_DELAY_SB));
+
+  h_ps_d->specificTo.mpeg.pAaImagDelayBufferQmf[0] = h_ps_d->specificTo.mpeg.aaQmfDelayBufImag;
+
+  assignTimeSlotsPS ( h_ps_d->specificTo.mpeg.pAaImagDelayBufferQmf[0] + (NO_QMF_CHANNELS-FIRST_DELAY_SB),
+                     &h_ps_d->specificTo.mpeg.pAaImagDelayBufferQmf[1],
+                      h_ps_d->specificTo.mpeg.noSampleDelay-1,
+                      (NO_DELAY_BUFFER_BANDS-FIRST_DELAY_SB));
+
+  /* Hybrid Filter Bank 1 creation. */
+  errorInfo = InitHybridFilterBank ( &h_ps_d->specificTo.mpeg.hybrid,
+                                      h_ps_d->noSubSamples,
+                                      noQmfBandsInHybrid20,
+                                      aHybridResolution20 );
+
+  for ( i = 0; i < NO_IID_GROUPS; i++ )
+  {
+    h_ps_d->specificTo.mpeg.h11rPrev[i] = FL2FXCONST_DBL(0.5f);
+    h_ps_d->specificTo.mpeg.h12rPrev[i] = FL2FXCONST_DBL(0.5f);
+  }
+
+  FDKmemclear( h_ps_d->specificTo.mpeg.h21rPrev, sizeof( h_ps_d->specificTo.mpeg.h21rPrev ) );
+  FDKmemclear( h_ps_d->specificTo.mpeg.h22rPrev, sizeof( h_ps_d->specificTo.mpeg.h22rPrev ) );
+
+  return errorInfo;
+}
+
+/***************************************************************************/
+/*!
+  \brief  clear some buffers used in decorrelation process
+
+  \return
+
+****************************************************************************/
+void ResetPsDeCor( HANDLE_PS_DEC h_ps_d )  /*!< pointer to the module state */
+{
+  INT i;
+
+  FDKmemclear(h_ps_d->specificTo.mpeg.aPeakDecayFastBin, NO_MID_RES_BINS*sizeof(FIXP_DBL));
+  FDKmemclear(h_ps_d->specificTo.mpeg.aPrevNrgBin, NO_MID_RES_BINS*sizeof(FIXP_DBL));
+  FDKmemclear(h_ps_d->specificTo.mpeg.aPrevPeakDiffBin, NO_MID_RES_BINS*sizeof(FIXP_DBL));
+  FDKmemclear(h_ps_d->specificTo.mpeg.aPowerPrevScal, NO_MID_RES_BINS*sizeof(SCHAR));
+
+  for (i=0 ; i < FIRST_DELAY_SB ; i++) {
+    FDKmemclear(h_ps_d->specificTo.mpeg.aaaRealDelayRBufferSerQmf[i], NO_DELAY_LENGTH_VECTORS*sizeof(FIXP_DBL));
+    FDKmemclear(h_ps_d->specificTo.mpeg.aaaImagDelayRBufferSerQmf[i], NO_DELAY_LENGTH_VECTORS*sizeof(FIXP_DBL));
+  }
+  for (i=0 ; i < NO_SUB_QMF_CHANNELS ; i++) {
+    FDKmemclear(h_ps_d->specificTo.mpeg.aaaRealDelayRBufferSerSubQmf[i], NO_DELAY_LENGTH_VECTORS*sizeof(FIXP_DBL));
+    FDKmemclear(h_ps_d->specificTo.mpeg.aaaImagDelayRBufferSerSubQmf[i], NO_DELAY_LENGTH_VECTORS*sizeof(FIXP_DBL));
+  }
+
+}
+
+/*******************************************************************************/
+
+/* slot based funcion prototypes */
+
+static void deCorrelateSlotBased( HANDLE_PS_DEC h_ps_d,
+
+                                  FIXP_DBL    *mHybridRealLeft,
+                                  FIXP_DBL    *mHybridImagLeft,
+                                  SCHAR        sf_mHybridLeft,
+
+                                  FIXP_DBL    *rIntBufferLeft,
+                                  FIXP_DBL    *iIntBufferLeft,
+                                  SCHAR        sf_IntBuffer,
+
+                                  FIXP_DBL    *mHybridRealRight,
+                                  FIXP_DBL    *mHybridImagRight,
+
+                                  FIXP_DBL    *rIntBufferRight,
+                                  FIXP_DBL    *iIntBufferRight );
+
+static void applySlotBasedRotation( HANDLE_PS_DEC h_ps_d,
+
+                                    FIXP_DBL  *mHybridRealLeft,
+                                    FIXP_DBL  *mHybridImagLeft,
+
+                                    FIXP_DBL  *QmfLeftReal,
+                                    FIXP_DBL  *QmfLeftImag,
+
+                                    FIXP_DBL  *mHybridRealRight,
+                                    FIXP_DBL  *mHybridImagRight,
+
+                                    FIXP_DBL  *QmfRightReal,
+                                    FIXP_DBL  *QmfRightImag
+                                  );
+
+
+/***************************************************************************/
+/*!
+  \brief  Get scale factor for all ps delay buffer.
+
+  \return
+
+****************************************************************************/
+static
+int getScaleFactorPsStatesBuffer(HANDLE_PS_DEC   h_ps_d)
+{
+  INT i;
+  int scale = DFRACT_BITS-1;
+
+  for (i=0; i<NO_QMF_BANDS_HYBRID20; i++) {
+    scale = fMin(scale, getScalefactor(h_ps_d->specificTo.mpeg.hybrid.mQmfBufferRealSlot[i], NO_SUB_QMF_CHANNELS));
+    scale = fMin(scale, getScalefactor(h_ps_d->specificTo.mpeg.hybrid.mQmfBufferImagSlot[i], NO_SUB_QMF_CHANNELS));
+  }
+
+  for (i=0; i<NO_SAMPLE_DELAY_ALLPASS; i++) {
+    scale = fMin(scale, getScalefactor(h_ps_d->specificTo.mpeg.aaRealDelayBufferQmf[i], FIRST_DELAY_SB));
+    scale = fMin(scale, getScalefactor(h_ps_d->specificTo.mpeg.aaImagDelayBufferQmf[i], FIRST_DELAY_SB));
+  }
+
+  for (i=0; i<NO_SAMPLE_DELAY_ALLPASS; i++) {
+    scale = fMin(scale, getScalefactor(h_ps_d->specificTo.mpeg.aaRealDelayBufferSubQmf[i], NO_SUB_QMF_CHANNELS));
+    scale = fMin(scale, getScalefactor(h_ps_d->specificTo.mpeg.aaImagDelayBufferSubQmf[i], NO_SUB_QMF_CHANNELS));
+  }
+
+  for (i=0; i<FIRST_DELAY_SB; i++) {
+    scale = fMin(scale, getScalefactor(h_ps_d->specificTo.mpeg.aaaRealDelayRBufferSerQmf[i], NO_DELAY_LENGTH_VECTORS));
+    scale = fMin(scale, getScalefactor(h_ps_d->specificTo.mpeg.aaaImagDelayRBufferSerQmf[i], NO_DELAY_LENGTH_VECTORS));
+  }
+
+  for (i=0; i<NO_SUB_QMF_CHANNELS; i++) {
+    scale = fMin(scale, getScalefactor(h_ps_d->specificTo.mpeg.aaaRealDelayRBufferSerSubQmf[i], NO_DELAY_LENGTH_VECTORS));
+    scale = fMin(scale, getScalefactor(h_ps_d->specificTo.mpeg.aaaImagDelayRBufferSerSubQmf[i], NO_DELAY_LENGTH_VECTORS));
+  }
+
+  for (i=0; i<MAX_DELAY_BUFFER_SIZE; i++)
+  {
+    INT len;
+    if (i==0)
+      len = NO_QMF_CHANNELS-FIRST_DELAY_SB;
+    else
+      len = NO_DELAY_BUFFER_BANDS-FIRST_DELAY_SB;
+
+    scale = fMin(scale, getScalefactor(h_ps_d->specificTo.mpeg.pAaRealDelayBufferQmf[i], len));
+    scale = fMin(scale, getScalefactor(h_ps_d->specificTo.mpeg.pAaImagDelayBufferQmf[i], len));
+  }
+
+  return (scale);
+}
+
+/***************************************************************************/
+/*!
+  \brief  Rescale all ps delay buffer.
+
+  \return
+
+****************************************************************************/
+static
+void scalePsStatesBuffer(HANDLE_PS_DEC h_ps_d,
+                         int           scale)
+{
+  INT i;
+
+  if (scale < 0)
+    scale = fixMax((INT)scale,(INT)-(DFRACT_BITS-1));
+  else
+    scale = fixMin((INT)scale,(INT)DFRACT_BITS-1);
+
+  for (i=0; i<NO_QMF_BANDS_HYBRID20; i++) {
+    scaleValues( h_ps_d->specificTo.mpeg.hybrid.mQmfBufferRealSlot[i], NO_SUB_QMF_CHANNELS, scale );
+    scaleValues( h_ps_d->specificTo.mpeg.hybrid.mQmfBufferImagSlot[i], NO_SUB_QMF_CHANNELS, scale );
+  }
+
+  for (i=0; i<NO_SAMPLE_DELAY_ALLPASS; i++) {
+    scaleValues( h_ps_d->specificTo.mpeg.aaRealDelayBufferQmf[i], FIRST_DELAY_SB, scale );
+    scaleValues( h_ps_d->specificTo.mpeg.aaImagDelayBufferQmf[i], FIRST_DELAY_SB, scale );
+  }
+
+  for (i=0; i<NO_SAMPLE_DELAY_ALLPASS; i++) {
+    scaleValues( h_ps_d->specificTo.mpeg.aaRealDelayBufferSubQmf[i], NO_SUB_QMF_CHANNELS, scale );
+    scaleValues( h_ps_d->specificTo.mpeg.aaImagDelayBufferSubQmf[i], NO_SUB_QMF_CHANNELS, scale );
+  }
+
+  for (i=0; i<FIRST_DELAY_SB; i++) {
+    scaleValues( h_ps_d->specificTo.mpeg.aaaRealDelayRBufferSerQmf[i], NO_DELAY_LENGTH_VECTORS, scale );
+    scaleValues( h_ps_d->specificTo.mpeg.aaaImagDelayRBufferSerQmf[i], NO_DELAY_LENGTH_VECTORS, scale );
+  }
+
+  for (i=0; i<NO_SUB_QMF_CHANNELS; i++) {
+    scaleValues( h_ps_d->specificTo.mpeg.aaaRealDelayRBufferSerSubQmf[i], NO_DELAY_LENGTH_VECTORS, scale );
+    scaleValues( h_ps_d->specificTo.mpeg.aaaImagDelayRBufferSerSubQmf[i], NO_DELAY_LENGTH_VECTORS, scale );
+  }
+
+  for (i=0; i<MAX_DELAY_BUFFER_SIZE; i++) {
+    INT len;
+    if (i==0)
+      len = NO_QMF_CHANNELS-FIRST_DELAY_SB;
+    else
+      len = NO_DELAY_BUFFER_BANDS-FIRST_DELAY_SB;
+
+    scaleValues( h_ps_d->specificTo.mpeg.pAaRealDelayBufferQmf[i], len, scale );
+    scaleValues( h_ps_d->specificTo.mpeg.pAaImagDelayBufferQmf[i], len, scale );
+  }
+
+  scale <<= 1;
+
+  scaleValues( h_ps_d->specificTo.mpeg.aPeakDecayFastBin, NO_MID_RES_BINS, scale );
+  scaleValues( h_ps_d->specificTo.mpeg.aPrevPeakDiffBin, NO_MID_RES_BINS, scale );
+  scaleValues( h_ps_d->specificTo.mpeg.aPrevNrgBin, NO_MID_RES_BINS, scale );
+}
+
+/***************************************************************************/
+/*!
+  \brief  Scale input channel to the same scalefactor and rescale hybrid
+          filterbank values
+
+  \return
+
+****************************************************************************/
+
+void scalFilterBankValues( HANDLE_PS_DEC   h_ps_d,
+                           FIXP_DBL      **fixpQmfReal,
+                           FIXP_DBL      **fixpQmfImag,
+                           int             lsb,
+                           int             scaleFactorLowBandSplitLow,
+                           int             scaleFactorLowBandSplitHigh,
+                           SCHAR          *scaleFactorLowBand_lb,
+                           SCHAR          *scaleFactorLowBand_hb,
+                           int             scaleFactorHighBands,
+                           INT            *scaleFactorHighBand,
+                           INT             noCols
+                         )
+{
+  INT maxScal;
+
+  INT i;
+
+  scaleFactorHighBands        =  -scaleFactorHighBands;
+  scaleFactorLowBandSplitLow  =  -scaleFactorLowBandSplitLow;
+  scaleFactorLowBandSplitHigh =  -scaleFactorLowBandSplitHigh;
+
+  /* get max scale factor */
+  maxScal = fixMax(scaleFactorHighBands,fixMax(scaleFactorLowBandSplitLow, scaleFactorLowBandSplitHigh ));
+
+  {
+    int headroom  = getScaleFactorPsStatesBuffer(h_ps_d);
+    maxScal   = fixMax(maxScal,(INT)(h_ps_d->specificTo.mpeg.scaleFactorPsDelayBuffer-headroom));
+    maxScal  += 1;
+  }
+
+  /* scale whole left channel to the same scale factor */
+
+  /* low band ( overlap buffer ) */
+  if ( maxScal != scaleFactorLowBandSplitLow ) {
+    INT scale = scaleFactorLowBandSplitLow - maxScal;
+    for ( i=0; i<(6); i++ ) {
+      scaleValues( fixpQmfReal[i], lsb, scale );
+      scaleValues( fixpQmfImag[i], lsb, scale );
+    }
+  }
+  /* low band ( current frame ) */
+  if ( maxScal != scaleFactorLowBandSplitHigh ) {
+    INT scale = scaleFactorLowBandSplitHigh - maxScal;
+    /* for ( i=(6); i<(6)+MAX_NUM_COL; i++ ) { */
+    for ( i=(6); i<(6)+noCols; i++ ) {
+      scaleValues( fixpQmfReal[i], lsb, scale );
+      scaleValues( fixpQmfImag[i], lsb, scale );
+    }
+  }
+  /* high band */
+  if ( maxScal != scaleFactorHighBands ) {
+    INT scale = scaleFactorHighBands - maxScal;
+    /* for ( i=0; i<MAX_NUM_COL; i++ ) { */
+    for ( i=0; i<noCols; i++ ) {
+      scaleValues( &fixpQmfReal[i][lsb], (64)-lsb, scale );
+      scaleValues( &fixpQmfImag[i][lsb], (64)-lsb, scale );
+    }
+  }
+
+  if ( maxScal != h_ps_d->specificTo.mpeg.scaleFactorPsDelayBuffer )
+    scalePsStatesBuffer(h_ps_d,(h_ps_d->specificTo.mpeg.scaleFactorPsDelayBuffer-maxScal));
+
+  h_ps_d->specificTo.mpeg.hybrid.sf_mQmfBuffer = maxScal;
+  h_ps_d->specificTo.mpeg.scaleFactorPsDelayBuffer = maxScal;
+
+  *scaleFactorHighBand += maxScal - scaleFactorHighBands;
+
+  h_ps_d->rescal = maxScal - scaleFactorLowBandSplitHigh;
+  h_ps_d->sf_IntBuffer = maxScal;
+
+  *scaleFactorLowBand_lb += maxScal - scaleFactorLowBandSplitLow;
+  *scaleFactorLowBand_hb += maxScal - scaleFactorLowBandSplitHigh;
+}
+
+void rescalFilterBankValues( HANDLE_PS_DEC   h_ps_d,                      /* parametric stereo decoder handle     */
+                             FIXP_DBL      **QmfBufferReal,               /* qmf filterbank values                */
+                             FIXP_DBL      **QmfBufferImag,               /* qmf filterbank values                */
+                             int             lsb,                         /* sbr start subband                    */
+                             INT             noCols)
+{
+  int i;
+  /* scale back 6 timeslots look ahead for hybrid filterbank to original value */
+  for ( i=noCols; i<noCols + (6); i++ ) {
+    scaleValues( QmfBufferReal[i], lsb, h_ps_d->rescal );
+    scaleValues( QmfBufferImag[i], lsb, h_ps_d->rescal );
+  }
+}
+
+/***************************************************************************/
+/*!
+  \brief  Generate decorrelated side channel using allpass/delay
+
+  \return
+
+****************************************************************************/
+static void
+deCorrelateSlotBased( HANDLE_PS_DEC h_ps_d,            /*!< pointer to the module state */
+
+                      FIXP_DBL    *mHybridRealLeft,    /*!< left (mono) hybrid values real */
+                      FIXP_DBL    *mHybridImagLeft,    /*!< left (mono) hybrid values imag */
+                      SCHAR        sf_mHybridLeft,     /*!< scalefactor for left (mono) hybrid bands */
+
+                      FIXP_DBL    *rIntBufferLeft,     /*!< real qmf bands left (mono) (38x64) */
+                      FIXP_DBL    *iIntBufferLeft,     /*!< real qmf bands left (mono) (38x64) */
+                      SCHAR        sf_IntBuffer,       /*!< scalefactor for all left and right qmf bands   */
+
+                      FIXP_DBL    *mHybridRealRight,   /*!< right (decorrelated) hybrid values real */
+                      FIXP_DBL    *mHybridImagRight,   /*!< right (decorrelated) hybrid values imag */
+
+                      FIXP_DBL    *rIntBufferRight,    /*!< real qmf bands right (decorrelated) (38x64) */
+                      FIXP_DBL    *iIntBufferRight )   /*!< real qmf bands right (decorrelated) (38x64) */
+{
+
+  INT  i, m, sb, gr, bin;
+
+  FIXP_DBL peakDiff, nrg, transRatio;
+
+  FIXP_DBL *RESTRICT aaLeftReal;
+  FIXP_DBL *RESTRICT aaLeftImag;
+
+  FIXP_DBL *RESTRICT aaRightReal;
+  FIXP_DBL *RESTRICT aaRightImag;
+
+  FIXP_DBL *RESTRICT pRealDelayBuffer;
+  FIXP_DBL *RESTRICT pImagDelayBuffer;
+
+  C_ALLOC_SCRATCH_START(aaPowerSlot, FIXP_DBL, NO_MID_RES_BINS);
+  C_ALLOC_SCRATCH_START(aaTransRatioSlot, FIXP_DBL, NO_MID_RES_BINS);
+
+/*!
+<pre>
+   parameter index       qmf bands             hybrid bands
+  ----------------------------------------------------------------------------
+         0                   0                      0,7
+         1                   0                      1,6
+         2                   0                      2
+         3                   0                      3           HYBRID BANDS
+         4                   1                      9
+         5                   1                      8
+         6                   2                     10
+         7                   2                     11
+  ----------------------------------------------------------------------------
+         8                   3
+         9                   4
+        10                   5
+        11                   6
+        12                   7
+        13                   8
+        14                   9,10      (2 )                      QMF BANDS
+        15                   11 - 13   (3 )
+        16                   14 - 17   (4 )
+        17                   18 - 22   (5 )
+        18                   23 - 34   (12)
+        19                   35 - 63   (29)
+  ----------------------------------------------------------------------------
+</pre>
+*/
+
+  #define FLTR_SCALE 3
+
+  /* hybrid bands (parameter index 0 - 7) */
+  aaLeftReal  = mHybridRealLeft;
+  aaLeftImag  = mHybridImagLeft;
+
+  aaPowerSlot[0] = ( fMultAddDiv2( fMultDiv2(aaLeftReal[0],  aaLeftReal[0]),  aaLeftImag[0],  aaLeftImag[0] ) >> FLTR_SCALE ) +
+                   ( fMultAddDiv2( fMultDiv2(aaLeftReal[7],  aaLeftReal[7]),  aaLeftImag[7],  aaLeftImag[7] ) >> FLTR_SCALE );
+
+  aaPowerSlot[1] = ( fMultAddDiv2( fMultDiv2(aaLeftReal[1],  aaLeftReal[1]),  aaLeftImag[1],  aaLeftImag[1] ) >> FLTR_SCALE ) +
+                   ( fMultAddDiv2( fMultDiv2(aaLeftReal[6],  aaLeftReal[6]),  aaLeftImag[6],  aaLeftImag[6] ) >> FLTR_SCALE );
+
+  aaPowerSlot[2] =   fMultAddDiv2( fMultDiv2(aaLeftReal[2],  aaLeftReal[2]),  aaLeftImag[2],  aaLeftImag[2] ) >> FLTR_SCALE;
+  aaPowerSlot[3] =   fMultAddDiv2( fMultDiv2(aaLeftReal[3],  aaLeftReal[3]),  aaLeftImag[3],  aaLeftImag[3] ) >> FLTR_SCALE;
+
+  aaPowerSlot[4] =   fMultAddDiv2( fMultDiv2(aaLeftReal[9],  aaLeftReal[9]),  aaLeftImag[9],  aaLeftImag[9] ) >> FLTR_SCALE;
+  aaPowerSlot[5] =   fMultAddDiv2( fMultDiv2(aaLeftReal[8],  aaLeftReal[8]),  aaLeftImag[8],  aaLeftImag[8] ) >> FLTR_SCALE;
+
+  aaPowerSlot[6] =   fMultAddDiv2( fMultDiv2(aaLeftReal[10], aaLeftReal[10]), aaLeftImag[10], aaLeftImag[10] ) >> FLTR_SCALE;
+  aaPowerSlot[7] =   fMultAddDiv2( fMultDiv2(aaLeftReal[11], aaLeftReal[11]), aaLeftImag[11], aaLeftImag[11] ) >> FLTR_SCALE;
+
+  /* qmf bands (parameter index 8 - 19) */
+  for ( bin = 8; bin < NO_MID_RES_BINS; bin++ ) {
+    FIXP_DBL slotNrg = FL2FXCONST_DBL(0.f);
+
+    for ( i = groupBorders20[bin+2]; i < groupBorders20[bin+3]; i++ ) {  /* max loops: 29 */
+      slotNrg += fMultAddDiv2 ( fMultDiv2(rIntBufferLeft[i], rIntBufferLeft[i]), iIntBufferLeft[i], iIntBufferLeft[i]) >> FLTR_SCALE;
+    }
+    aaPowerSlot[bin] = slotNrg;
+
+  }
+
+
+  /* calculation of transient ratio */
+  for (bin=0; bin < NO_MID_RES_BINS; bin++) {   /* noBins = 20 ( BASELINE_PS ) */
+
+    h_ps_d->specificTo.mpeg.aPeakDecayFastBin[bin] = fMult( h_ps_d->specificTo.mpeg.aPeakDecayFastBin[bin], PEAK_DECAY_FACTOR );
+
+    if (h_ps_d->specificTo.mpeg.aPeakDecayFastBin[bin] < aaPowerSlot[bin]) {
+      h_ps_d->specificTo.mpeg.aPeakDecayFastBin[bin] = aaPowerSlot[bin];
+    }
+
+    /* calculate PSmoothPeakDecayDiffNrg */
+    peakDiff = fMultAdd ( (h_ps_d->specificTo.mpeg.aPrevPeakDiffBin[bin]>>1),
+                 INT_FILTER_COEFF, h_ps_d->specificTo.mpeg.aPeakDecayFastBin[bin] - aaPowerSlot[bin] - h_ps_d->specificTo.mpeg.aPrevPeakDiffBin[bin]);
+
+    /* save peakDiff for the next frame */
+    h_ps_d->specificTo.mpeg.aPrevPeakDiffBin[bin] = peakDiff;
+
+    nrg = h_ps_d->specificTo.mpeg.aPrevNrgBin[bin] + fMult( INT_FILTER_COEFF, aaPowerSlot[bin] - h_ps_d->specificTo.mpeg.aPrevNrgBin[bin] );
+
+    /* Negative energies don't exist. But sometimes they appear due to rounding. */
+
+    nrg = fixMax(nrg,FL2FXCONST_DBL(0.f));
+
+    /* save nrg for the next frame */
+    h_ps_d->specificTo.mpeg.aPrevNrgBin[bin] = nrg;
+
+    nrg = fMult( nrg, TRANSIENT_IMPACT_FACTOR );
+
+    /* save transient impact factor */
+    if ( peakDiff <= nrg || peakDiff == FL2FXCONST_DBL(0.0) ) {
+      aaTransRatioSlot[bin] = (FIXP_DBL)MAXVAL_DBL /* FL2FXCONST_DBL(1.0f)*/;
+    }
+    else if ( nrg <= FL2FXCONST_DBL(0.0f) ) {
+        aaTransRatioSlot[bin] = FL2FXCONST_DBL(0.f);
+      }
+    else {
+      /* scale to denominator */
+      INT scale_left = fixMax(0, CntLeadingZeros(peakDiff) - 1);
+      aaTransRatioSlot[bin] = schur_div( nrg<<scale_left, peakDiff<<scale_left, 16);
+    }
+  } /* bin */
+
+
+
+
+  #define DELAY_GROUP_OFFSET    20
+  #define NR_OF_DELAY_GROUPS     2
+
+  FIXP_DBL rTmp, iTmp, rTmp0, iTmp0, rR0, iR0;
+
+  INT TempDelay     = h_ps_d->specificTo.mpeg.delayBufIndex;  /* set delay indices */
+
+  pRealDelayBuffer = h_ps_d->specificTo.mpeg.aaRealDelayBufferSubQmf[TempDelay];
+  pImagDelayBuffer = h_ps_d->specificTo.mpeg.aaImagDelayBufferSubQmf[TempDelay];
+
+  aaLeftReal  = mHybridRealLeft;
+  aaLeftImag  = mHybridImagLeft;
+  aaRightReal = mHybridRealRight;
+  aaRightImag = mHybridImagRight;
+
+  /************************/
+  /* ICC groups :  0 -  9 */
+  /************************/
+
+  /* gr = ICC groups */
+  for (gr=0; gr < SUBQMF_GROUPS; gr++) {
+
+    transRatio = aaTransRatioSlot[bins2groupMap20[gr]];
+
+    /* sb = subQMF/QMF subband */
+    sb = groupBorders20[gr];
+
+    /* Update delay buffers, sample delay allpass = 2 */
+    rTmp0 = pRealDelayBuffer[sb];
+    iTmp0 = pImagDelayBuffer[sb];
+
+    pRealDelayBuffer[sb] = aaLeftReal[sb];
+    pImagDelayBuffer[sb] = aaLeftImag[sb];
+
+    /* delay by fraction */
+    cplxMultDiv2(&rR0, &iR0, rTmp0, iTmp0, aaFractDelayPhaseFactorReSubQmf20[sb], aaFractDelayPhaseFactorImSubQmf20[sb]);
+    rR0<<=1;
+    iR0<<=1;
+
+    FIXP_DBL *pAaaRealDelayRBufferSerSubQmf = h_ps_d->specificTo.mpeg.aaaRealDelayRBufferSerSubQmf[sb];
+    FIXP_DBL *pAaaImagDelayRBufferSerSubQmf = h_ps_d->specificTo.mpeg.aaaImagDelayRBufferSerSubQmf[sb];
+
+    for (m=0; m<NO_SERIAL_ALLPASS_LINKS ; m++) {
+
+      INT tmpDelayRSer = h_ps_d->specificTo.mpeg.aDelayRBufIndexSer[m];
+
+      /* get delayed values from according buffer : m(0)=3; m(1)=4; m(2)=5; */
+      rTmp0 = pAaaRealDelayRBufferSerSubQmf[tmpDelayRSer];
+      iTmp0 = pAaaImagDelayRBufferSerSubQmf[tmpDelayRSer];
+
+      /* delay by fraction */
+      cplxMultDiv2(&rTmp, &iTmp, rTmp0, iTmp0, aaFractDelayPhaseFactorSerReSubQmf20[sb][m], aaFractDelayPhaseFactorSerImSubQmf20[sb][m]);
+
+      rTmp = (rTmp - fMultDiv2(aAllpassLinkDecaySer[m], rR0)) << 1;
+      iTmp = (iTmp - fMultDiv2(aAllpassLinkDecaySer[m], iR0)) << 1;
+
+      pAaaRealDelayRBufferSerSubQmf[tmpDelayRSer] = rR0 + fMult(aAllpassLinkDecaySer[m], rTmp);
+      pAaaImagDelayRBufferSerSubQmf[tmpDelayRSer] = iR0 + fMult(aAllpassLinkDecaySer[m], iTmp);
+
+      rR0 = rTmp;
+      iR0 = iTmp;
+
+      pAaaRealDelayRBufferSerSubQmf += aAllpassLinkDelaySer[m];
+      pAaaImagDelayRBufferSerSubQmf += aAllpassLinkDelaySer[m];
+
+    } /* m */
+
+    /* duck if a past transient is found */
+    aaRightReal[sb] = fMult(transRatio, rR0);
+    aaRightImag[sb] = fMult(transRatio, iR0);
+
+  } /* gr */
+
+
+  scaleValues( mHybridRealLeft,  NO_SUB_QMF_CHANNELS, -SCAL_HEADROOM );
+  scaleValues( mHybridImagLeft,  NO_SUB_QMF_CHANNELS, -SCAL_HEADROOM );
+  scaleValues( mHybridRealRight, NO_SUB_QMF_CHANNELS, -SCAL_HEADROOM );
+  scaleValues( mHybridImagRight, NO_SUB_QMF_CHANNELS, -SCAL_HEADROOM );
+
+
+  /************************/
+
+  aaLeftReal  = rIntBufferLeft;
+  aaLeftImag  = iIntBufferLeft;
+  aaRightReal = rIntBufferRight;
+  aaRightImag = iIntBufferRight;
+
+  pRealDelayBuffer = h_ps_d->specificTo.mpeg.aaRealDelayBufferQmf[TempDelay];
+  pImagDelayBuffer = h_ps_d->specificTo.mpeg.aaImagDelayBufferQmf[TempDelay];
+
+  /************************/
+  /* ICC groups : 10 - 19 */
+  /************************/
+
+
+  /* gr = ICC groups */
+  for (gr=SUBQMF_GROUPS; gr < NO_IID_GROUPS - NR_OF_DELAY_GROUPS; gr++) {
+
+    transRatio = aaTransRatioSlot[bins2groupMap20[gr]];
+
+    /* sb = subQMF/QMF subband */
+    for (sb = groupBorders20[gr]; sb < groupBorders20[gr+1]; sb++) {
+      FIXP_DBL resR, resI;
+
+      /* decayScaleFactor = 1.0f + decay_cutoff * DECAY_SLOPE - DECAY_SLOPE * sb; DECAY_SLOPE = 0.05 */
+      FIXP_DBL decayScaleFactor = decayScaleFactTable[sb];
+
+      /* Update delay buffers, sample delay allpass = 2 */
+      rTmp0 = pRealDelayBuffer[sb];
+      iTmp0 = pImagDelayBuffer[sb];
+
+      pRealDelayBuffer[sb] = aaLeftReal[sb];
+      pImagDelayBuffer[sb] = aaLeftImag[sb];
+
+      /* delay by fraction */
+      cplxMultDiv2(&rR0, &iR0, rTmp0, iTmp0, aaFractDelayPhaseFactorReQmf[sb], aaFractDelayPhaseFactorImQmf[sb]);
+      rR0<<=1;
+      iR0<<=1;
+
+      resR = fMult(decayScaleFactor, rR0);
+      resI = fMult(decayScaleFactor, iR0);
+
+      FIXP_DBL *pAaaRealDelayRBufferSerQmf = h_ps_d->specificTo.mpeg.aaaRealDelayRBufferSerQmf[sb];
+      FIXP_DBL *pAaaImagDelayRBufferSerQmf = h_ps_d->specificTo.mpeg.aaaImagDelayRBufferSerQmf[sb];
+
+      for (m=0; m<NO_SERIAL_ALLPASS_LINKS ; m++) {
+
+        INT tmpDelayRSer = h_ps_d->specificTo.mpeg.aDelayRBufIndexSer[m];
+
+        /* get delayed values from according buffer : m(0)=3; m(1)=4; m(2)=5; */
+        rTmp0 = pAaaRealDelayRBufferSerQmf[tmpDelayRSer];
+        iTmp0 = pAaaImagDelayRBufferSerQmf[tmpDelayRSer];
+
+        /* delay by fraction */
+        cplxMultDiv2(&rTmp, &iTmp, rTmp0, iTmp0, aaFractDelayPhaseFactorSerReQmf[sb][m], aaFractDelayPhaseFactorSerImQmf[sb][m]);
+
+        rTmp = (rTmp - fMultDiv2(aAllpassLinkDecaySer[m], resR))<<1;
+        iTmp = (iTmp - fMultDiv2(aAllpassLinkDecaySer[m], resI))<<1;
+
+        resR = fMult(decayScaleFactor, rTmp);
+        resI = fMult(decayScaleFactor, iTmp);
+
+        pAaaRealDelayRBufferSerQmf[tmpDelayRSer] = rR0 + fMult(aAllpassLinkDecaySer[m], resR);
+        pAaaImagDelayRBufferSerQmf[tmpDelayRSer] = iR0 + fMult(aAllpassLinkDecaySer[m], resI);
+
+        rR0 = rTmp;
+        iR0 = iTmp;
+
+        pAaaRealDelayRBufferSerQmf += aAllpassLinkDelaySer[m];
+        pAaaImagDelayRBufferSerQmf += aAllpassLinkDelaySer[m];
+
+      } /* m */
+
+      /* duck if a past transient is found */
+      aaRightReal[sb] = fMult(transRatio, rR0);
+      aaRightImag[sb] = fMult(transRatio, iR0);
+
+    } /* sb */
+  } /* gr */
+
+  /************************/
+  /* ICC groups : 20,  21 */
+  /************************/
+
+
+  /* gr = ICC groups */
+  for (gr=DELAY_GROUP_OFFSET; gr < NO_IID_GROUPS; gr++) {
+
+    INT sbStart = groupBorders20[gr];
+    INT sbStop  = groupBorders20[gr+1];
+
+    UCHAR *pDelayBufIdx = &h_ps_d->specificTo.mpeg.aDelayBufIndexDelayQmf[sbStart-FIRST_DELAY_SB];
+
+    transRatio = aaTransRatioSlot[bins2groupMap20[gr]];
+
+    /* sb = subQMF/QMF subband */
+    for (sb = sbStart; sb < sbStop; sb++) {
+
+      /* Update delay buffers */
+      rR0 = h_ps_d->specificTo.mpeg.pAaRealDelayBufferQmf[*pDelayBufIdx][sb-FIRST_DELAY_SB];
+      iR0 = h_ps_d->specificTo.mpeg.pAaImagDelayBufferQmf[*pDelayBufIdx][sb-FIRST_DELAY_SB];
+
+      h_ps_d->specificTo.mpeg.pAaRealDelayBufferQmf[*pDelayBufIdx][sb-FIRST_DELAY_SB] = aaLeftReal[sb];
+      h_ps_d->specificTo.mpeg.pAaImagDelayBufferQmf[*pDelayBufIdx][sb-FIRST_DELAY_SB] = aaLeftImag[sb];
+
+      /* duck if a past transient is found */
+      aaRightReal[sb] = fMult(transRatio, rR0);
+      aaRightImag[sb] = fMult(transRatio, iR0);
+
+      if (++(*pDelayBufIdx) >= delayIndexQmf[sb]) {
+        *pDelayBufIdx = 0;
+      }
+      pDelayBufIdx++;
+
+    } /* sb */
+  } /* gr */
+
+
+  /* Update delay buffer index */
+  if (++h_ps_d->specificTo.mpeg.delayBufIndex >= NO_SAMPLE_DELAY_ALLPASS)
+    h_ps_d->specificTo.mpeg.delayBufIndex = 0;
+
+  for (m=0; m<NO_SERIAL_ALLPASS_LINKS ; m++) {
+    if (++h_ps_d->specificTo.mpeg.aDelayRBufIndexSer[m] >= aAllpassLinkDelaySer[m])
+      h_ps_d->specificTo.mpeg.aDelayRBufIndexSer[m] = 0;
+  }
+
+
+  scaleValues( &rIntBufferLeft[NO_QMF_BANDS_HYBRID20],  NO_QMF_CHANNELS-NO_QMF_BANDS_HYBRID20, -SCAL_HEADROOM );
+  scaleValues( &iIntBufferLeft[NO_QMF_BANDS_HYBRID20],  NO_QMF_CHANNELS-NO_QMF_BANDS_HYBRID20, -SCAL_HEADROOM );
+  scaleValues( &rIntBufferRight[NO_QMF_BANDS_HYBRID20], NO_QMF_CHANNELS-NO_QMF_BANDS_HYBRID20, -SCAL_HEADROOM );
+  scaleValues( &iIntBufferRight[NO_QMF_BANDS_HYBRID20], NO_QMF_CHANNELS-NO_QMF_BANDS_HYBRID20, -SCAL_HEADROOM );
+
+  /* free memory on scratch */
+  C_ALLOC_SCRATCH_END(aaTransRatioSlot, FIXP_DBL, NO_MID_RES_BINS);
+  C_ALLOC_SCRATCH_END(aaPowerSlot, FIXP_DBL, NO_MID_RES_BINS);
+}
+
+
+void initSlotBasedRotation( HANDLE_PS_DEC h_ps_d, /*!< pointer to the module state */
+                            int env,
+                            int usb
+                            ) {
+
+  INT     group = 0;
+  INT     bin =  0;
+  INT     noIidSteps;
+
+/*  const UCHAR *pQuantizedIIDs;*/
+
+  FIXP_SGL  invL;
+  FIXP_DBL  ScaleL, ScaleR;
+  FIXP_DBL  Alpha, Beta;
+  FIXP_DBL  h11r, h12r, h21r, h22r;
+
+  const FIXP_DBL  *PScaleFactors;
+
+  /* Overwrite old values in delay buffers when upper subband is higher than in last frame */
+  if (env == 0) {
+
+    if ((usb > h_ps_d->specificTo.mpeg.lastUsb) && h_ps_d->specificTo.mpeg.lastUsb) {
+
+      INT i,k,length;
+
+      for (i=h_ps_d->specificTo.mpeg.lastUsb ; i < FIRST_DELAY_SB; i++) {
+        FDKmemclear(h_ps_d->specificTo.mpeg.aaaRealDelayRBufferSerQmf[i], NO_DELAY_LENGTH_VECTORS*sizeof(FIXP_DBL));
+        FDKmemclear(h_ps_d->specificTo.mpeg.aaaImagDelayRBufferSerQmf[i], NO_DELAY_LENGTH_VECTORS*sizeof(FIXP_DBL));
+      }
+
+      for (k=0 ; k<NO_SAMPLE_DELAY_ALLPASS; k++) {
+        FDKmemclear(h_ps_d->specificTo.mpeg.pAaRealDelayBufferQmf[k], FIRST_DELAY_SB*sizeof(FIXP_DBL));
+      }
+      length = (usb-FIRST_DELAY_SB)*sizeof(FIXP_DBL);
+      if(length>0) {
+        FDKmemclear(h_ps_d->specificTo.mpeg.pAaRealDelayBufferQmf[0], length);
+        FDKmemclear(h_ps_d->specificTo.mpeg.pAaImagDelayBufferQmf[0], length);
+      }
+      length = (fixMin(NO_DELAY_BUFFER_BANDS,(INT)usb)-FIRST_DELAY_SB)*sizeof(FIXP_DBL);
+      if(length>0) {
+        for (k=1 ; k < h_ps_d->specificTo.mpeg.noSampleDelay; k++) {
+          FDKmemclear(h_ps_d->specificTo.mpeg.pAaRealDelayBufferQmf[k], length);
+          FDKmemclear(h_ps_d->specificTo.mpeg.pAaImagDelayBufferQmf[k], length);
+        }
+      }
+    }
+    h_ps_d->specificTo.mpeg.lastUsb = usb;
+  } /* env == 0 */
+
+  if (h_ps_d->bsData[h_ps_d->processSlot].mpeg.bFineIidQ)
+  {
+    PScaleFactors = ScaleFactorsFine; /* values are shiftet right by one */
+    noIidSteps = NO_IID_STEPS_FINE;
+    /*pQuantizedIIDs = quantizedIIDsFine;*/
+  }
+
+  else
+  {
+    PScaleFactors = ScaleFactors; /* values are shiftet right by one */
+    noIidSteps = NO_IID_STEPS;
+    /*pQuantizedIIDs = quantizedIIDs;*/
+  }
+
+
+  /* dequantize and decode */
+  for ( group = 0; group < NO_IID_GROUPS; group++ ) {
+
+    bin = bins2groupMap20[group];
+
+    /*!
+    <h3> type 'A' rotation </h3>
+    mixing procedure R_a, used in baseline version<br>
+
+     Scale-factor vectors c1 and c2 are precalculated in initPsTables () and stored in
+     scaleFactors[] and scaleFactorsFine[] = pScaleFactors [].
+     From the linearized IID parameters (intensity differences), two scale factors are
+     calculated. They are used to obtain the coefficients h11... h22.
+    */
+
+    /* ScaleR and ScaleL are scaled by 1 shift right */
+
+    ScaleR = PScaleFactors[noIidSteps + h_ps_d->specificTo.mpeg.coef.aaIidIndexMapped[env][bin]];
+    ScaleL = PScaleFactors[noIidSteps - h_ps_d->specificTo.mpeg.coef.aaIidIndexMapped[env][bin]];
+
+    Beta   = fMult (fMult( Alphas[h_ps_d->specificTo.mpeg.coef.aaIccIndexMapped[env][bin]], ( ScaleR - ScaleL )), FIXP_SQRT05);
+    Alpha  = Alphas[h_ps_d->specificTo.mpeg.coef.aaIccIndexMapped[env][bin]]>>1;
+
+    /* Alpha and Beta are now both scaled by 2 shifts right */
+
+    /* calculate the coefficients h11... h22 from scale-factors and ICC parameters */
+
+    /* h values are scaled by 1 shift right */
+    {
+      FIXP_DBL trigData[4];
+
+      inline_fixp_cos_sin(Beta + Alpha, Beta - Alpha, 2, trigData);
+      h11r = fMult( ScaleL, trigData[0]);
+      h12r = fMult( ScaleR, trigData[2]);
+      h21r = fMult( ScaleL, trigData[1]);
+      h22r = fMult( ScaleR, trigData[3]);
+    }
+    /*****************************************************************************************/
+    /* Interpolation of the matrices H11... H22:                                             */
+    /*                                                                                       */
+    /* H11(k,n) = H11(k,n[e]) + (n-n[e]) * (H11(k,n[e+1] - H11(k,n[e])) / (n[e+1] - n[e])    */
+    /* ...                                                                                   */
+    /*****************************************************************************************/
+
+    /* invL = 1/(length of envelope) */
+    invL = FX_DBL2FX_SGL(GetInvInt(h_ps_d->bsData[h_ps_d->processSlot].mpeg.aEnvStartStop[env + 1] - h_ps_d->bsData[h_ps_d->processSlot].mpeg.aEnvStartStop[env]));
+
+    h_ps_d->specificTo.mpeg.coef.H11r[group]  = h_ps_d->specificTo.mpeg.h11rPrev[group];
+    h_ps_d->specificTo.mpeg.coef.H12r[group]  = h_ps_d->specificTo.mpeg.h12rPrev[group];
+    h_ps_d->specificTo.mpeg.coef.H21r[group]  = h_ps_d->specificTo.mpeg.h21rPrev[group];
+    h_ps_d->specificTo.mpeg.coef.H22r[group]  = h_ps_d->specificTo.mpeg.h22rPrev[group];
+
+    h_ps_d->specificTo.mpeg.coef.DeltaH11r[group]  = fMult ( h11r - h_ps_d->specificTo.mpeg.coef.H11r[group], invL );
+    h_ps_d->specificTo.mpeg.coef.DeltaH12r[group]  = fMult ( h12r - h_ps_d->specificTo.mpeg.coef.H12r[group], invL );
+    h_ps_d->specificTo.mpeg.coef.DeltaH21r[group]  = fMult ( h21r - h_ps_d->specificTo.mpeg.coef.H21r[group], invL );
+    h_ps_d->specificTo.mpeg.coef.DeltaH22r[group]  = fMult ( h22r - h_ps_d->specificTo.mpeg.coef.H22r[group], invL );
+
+    /* update prev coefficients for interpolation in next envelope */
+
+    h_ps_d->specificTo.mpeg.h11rPrev[group] = h11r;
+    h_ps_d->specificTo.mpeg.h12rPrev[group] = h12r;
+    h_ps_d->specificTo.mpeg.h21rPrev[group] = h21r;
+    h_ps_d->specificTo.mpeg.h22rPrev[group] = h22r;
+
+  } /* group loop */
+}
+
+
+static void applySlotBasedRotation( HANDLE_PS_DEC h_ps_d,        /*!< pointer to the module state */
+
+                                    FIXP_DBL  *mHybridRealLeft,  /*!< hybrid values real left  */
+                                    FIXP_DBL  *mHybridImagLeft,  /*!< hybrid values imag left  */
+
+                                    FIXP_DBL  *QmfLeftReal,      /*!< real bands left qmf channel */
+                                    FIXP_DBL  *QmfLeftImag,      /*!< imag bands left qmf channel */
+
+                                    FIXP_DBL  *mHybridRealRight, /*!< hybrid values real right  */
+                                    FIXP_DBL  *mHybridImagRight, /*!< hybrid values imag right  */
+
+                                    FIXP_DBL  *QmfRightReal,     /*!< real bands right qmf channel */
+                                    FIXP_DBL  *QmfRightImag      /*!< imag bands right qmf channel */
+                                   )
+{
+  INT     group;
+  INT     subband;
+
+  FIXP_DBL *RESTRICT HybrLeftReal;
+  FIXP_DBL *RESTRICT HybrLeftImag;
+  FIXP_DBL *RESTRICT HybrRightReal;
+  FIXP_DBL *RESTRICT HybrRightImag;
+
+  FIXP_DBL tmpLeft, tmpRight;
+
+
+  /**********************************************************************************************/
+  /*!
+  <h2> Mapping </h2>
+
+  The number of stereo bands that is actually used depends on the number of availble
+  parameters for IID and ICC:
+  <pre>
+   nr. of IID para.| nr. of ICC para. | nr. of Stereo bands
+   ----------------|------------------|-------------------
+     10,20         |     10,20        |        20
+     10,20         |     34           |        34
+     34            |     10,20        |        34
+     34            |     34           |        34
+  </pre>
+  In the case the number of parameters for IIS and ICC differs from the number of stereo
+  bands, a mapping from the lower number to the higher number of parameters is applied.
+  Index mapping of IID and ICC parameters is already done in psbitdec.cpp. Further mapping is
+  not needed here in baseline version.
+  **********************************************************************************************/
+
+  /************************************************************************************************/
+  /*!
+  <h2> Mixing </h2>
+
+  To generate the QMF subband signals for the subband samples n = n[e]+1 ,,, n_[e+1] the
+  parameters at position n[e] and n[e+1] are required as well as the subband domain signals
+  s_k(n) and d_k(n) for n = n[e]+1... n_[e+1]. n[e] represents the start position for
+  envelope e. The border positions n[e] are handled in DecodePS().
+
+  The stereo sub subband signals are constructed as:
+  <pre>
+  l_k(n) = H11(k,n) s_k(n) + H21(k,n) d_k(n)
+  r_k(n) = H21(k,n) s_k(n) + H22(k,n) d_k(n)
+  </pre>
+  In order to obtain the matrices H11(k,n)... H22 (k,n), the vectors h11(b)... h22(b) need to
+  be calculated first (b: parameter index). Depending on ICC mode either mixing procedure R_a
+  or R_b is used for that. For both procedures, the parameters for parameter position n[e+1]
+  is used.
+  ************************************************************************************************/
+
+
+  /************************************************************************************************/
+  /*!
+  <h2>Phase parameters </h2>
+  With disabled phase parameters (which is the case in baseline version), the H-matrices are
+  just calculated by:
+
+  <pre>
+  H11(k,n[e+1] = h11(b(k))
+  (...)
+  b(k): parameter index according to mapping table
+  </pre>
+
+  <h2>Processing of the samples in the sub subbands </h2>
+  this loop includes the interpolation of the coefficients Hxx
+  ************************************************************************************************/
+
+
+  /* loop thru all groups ... */
+  HybrLeftReal  = mHybridRealLeft;
+  HybrLeftImag  = mHybridImagLeft;
+  HybrRightReal = mHybridRealRight;
+  HybrRightImag = mHybridImagRight;
+
+  /******************************************************/
+  /* construct stereo sub subband signals according to: */
+  /*                                                    */
+  /* l_k(n) = H11(k,n) s_k(n) + H21(k,n) d_k(n)         */
+  /* r_k(n) = H12(k,n) s_k(n) + H22(k,n) d_k(n)         */
+  /******************************************************/
+  for ( group = 0; group < SUBQMF_GROUPS; group++ ) {
+
+    h_ps_d->specificTo.mpeg.coef.H11r[group] += h_ps_d->specificTo.mpeg.coef.DeltaH11r[group];
+    h_ps_d->specificTo.mpeg.coef.H12r[group] += h_ps_d->specificTo.mpeg.coef.DeltaH12r[group];
+    h_ps_d->specificTo.mpeg.coef.H21r[group] += h_ps_d->specificTo.mpeg.coef.DeltaH21r[group];
+    h_ps_d->specificTo.mpeg.coef.H22r[group] += h_ps_d->specificTo.mpeg.coef.DeltaH22r[group];
+
+    subband = groupBorders20[group];
+
+    tmpLeft  = fMultAddDiv2( fMultDiv2(h_ps_d->specificTo.mpeg.coef.H11r[group], HybrLeftReal[subband]), h_ps_d->specificTo.mpeg.coef.H21r[group], HybrRightReal[subband]);
+    tmpRight = fMultAddDiv2( fMultDiv2(h_ps_d->specificTo.mpeg.coef.H12r[group], HybrLeftReal[subband]), h_ps_d->specificTo.mpeg.coef.H22r[group], HybrRightReal[subband]);
+    HybrLeftReal [subband] = tmpLeft<<1;
+    HybrRightReal[subband] = tmpRight<<1;
+
+    tmpLeft  = fMultAdd( fMultDiv2(h_ps_d->specificTo.mpeg.coef.H11r[group], HybrLeftImag[subband]), h_ps_d->specificTo.mpeg.coef.H21r[group], HybrRightImag[subband]);
+    tmpRight = fMultAdd( fMultDiv2(h_ps_d->specificTo.mpeg.coef.H12r[group], HybrLeftImag[subband]), h_ps_d->specificTo.mpeg.coef.H22r[group], HybrRightImag[subband]);
+    HybrLeftImag [subband] = tmpLeft;
+    HybrRightImag[subband] = tmpRight;
+  }
+
+  /* continue in the qmf buffers */
+  HybrLeftReal  = QmfLeftReal;
+  HybrLeftImag  = QmfLeftImag;
+  HybrRightReal = QmfRightReal;
+  HybrRightImag = QmfRightImag;
+
+  for (; group < NO_IID_GROUPS; group++ ) {
+
+    h_ps_d->specificTo.mpeg.coef.H11r[group] += h_ps_d->specificTo.mpeg.coef.DeltaH11r[group];
+    h_ps_d->specificTo.mpeg.coef.H12r[group] += h_ps_d->specificTo.mpeg.coef.DeltaH12r[group];
+    h_ps_d->specificTo.mpeg.coef.H21r[group] += h_ps_d->specificTo.mpeg.coef.DeltaH21r[group];
+    h_ps_d->specificTo.mpeg.coef.H22r[group] += h_ps_d->specificTo.mpeg.coef.DeltaH22r[group];
+
+    for ( subband = groupBorders20[group]; subband < groupBorders20[group + 1]; subband++ )
+    {
+      tmpLeft  = fMultAdd( fMultDiv2(h_ps_d->specificTo.mpeg.coef.H11r[group], HybrLeftReal[subband]), h_ps_d->specificTo.mpeg.coef.H21r[group], HybrRightReal[subband]);
+      tmpRight = fMultAdd( fMultDiv2(h_ps_d->specificTo.mpeg.coef.H12r[group], HybrLeftReal[subband]), h_ps_d->specificTo.mpeg.coef.H22r[group], HybrRightReal[subband]);
+      HybrLeftReal [subband] = tmpLeft;
+      HybrRightReal[subband] = tmpRight;
+
+      tmpLeft  = fMultAdd( fMultDiv2(h_ps_d->specificTo.mpeg.coef.H11r[group], HybrLeftImag[subband]), h_ps_d->specificTo.mpeg.coef.H21r[group], HybrRightImag[subband]);
+      tmpRight = fMultAdd( fMultDiv2(h_ps_d->specificTo.mpeg.coef.H12r[group], HybrLeftImag[subband]), h_ps_d->specificTo.mpeg.coef.H22r[group], HybrRightImag[subband]);
+      HybrLeftImag [subband] = tmpLeft;
+      HybrRightImag[subband] = tmpRight;
+
+    } /* subband */
+  }
+}
+
+
+/***************************************************************************/
+/*!
+  \brief  Applies IID, ICC, IPD and OPD parameters to the current frame.
+
+  \return none
+
+****************************************************************************/
+void
+ApplyPsSlot( HANDLE_PS_DEC h_ps_d,         /*!< handle PS_DEC*/
+             FIXP_DBL  **rIntBufferLeft,   /*!< real bands left qmf channel (38x64)  */
+             FIXP_DBL  **iIntBufferLeft,   /*!< imag bands left qmf channel (38x64)  */
+             FIXP_DBL  *rIntBufferRight,   /*!< real bands right qmf channel (38x64) */
+             FIXP_DBL  *iIntBufferRight    /*!< imag bands right qmf channel (38x64) */
+           )
+{
+
+  /*!
+  The 64-band QMF representation of the monaural signal generated by the SBR tool
+  is used as input of the PS tool. After the PS processing, the outputs of the left
+  and right hybrid synthesis filterbanks are used to generate the stereo output
+  signal.
+
+  <pre>
+
+             -------------            ----------            -------------
+            | Hybrid      | M_n[k,m] |          | L_n[k,m] | Hybrid      | l[n]
+   m[n] --->| analysis    |--------->|          |--------->| synthesis   |----->
+            | filter bank |          |          |          | filter bank |
+             -------------           | Stereo   |           -------------
+                   |                 | recon-   |
+                   |                 | stuction |
+                  \|/                |          |
+             -------------           |          |
+            | De-         | D_n[k,m] |          |
+            | correlation |--------->|          |
+             -------------           |          |           -------------
+                                     |          | R_n[k,m] | Hybrid      | r[n]
+                                     |          |--------->| synthesis   |----->
+   IID, ICC ------------------------>|          |          | filter bank |
+  (IPD, OPD)                          ----------            -------------
+
+  m[n]:      QMF represantation of the mono input
+  M_n[k,m]:  (sub-)sub-band domain signals of the mono input
+  D_n[k,m]:  decorrelated (sub-)sub-band domain signals
+  L_n[k,m]:  (sub-)sub-band domain signals of the left output
+  R_n[k,m]:  (sub-)sub-band domain signals of the right output
+  l[n],r[n]: left/right output signals
+
+  </pre>
+  */
+
+  /* get temporary hybrid qmf values of one timeslot */
+  C_ALLOC_SCRATCH_START(hybridRealLeft, FIXP_DBL, NO_SUB_QMF_CHANNELS);
+  C_ALLOC_SCRATCH_START(hybridImagLeft, FIXP_DBL, NO_SUB_QMF_CHANNELS);
+  C_ALLOC_SCRATCH_START(hybridRealRight, FIXP_DBL, NO_SUB_QMF_CHANNELS);
+  C_ALLOC_SCRATCH_START(hybridImagRight, FIXP_DBL, NO_SUB_QMF_CHANNELS);
+
+  SCHAR sf_IntBuffer     = h_ps_d->sf_IntBuffer;
+
+  /* clear workbuffer */
+  FDKmemclear(hybridRealLeft,  NO_SUB_QMF_CHANNELS*sizeof(FIXP_DBL));
+  FDKmemclear(hybridImagLeft,  NO_SUB_QMF_CHANNELS*sizeof(FIXP_DBL));
+  FDKmemclear(hybridRealRight, NO_SUB_QMF_CHANNELS*sizeof(FIXP_DBL));
+  FDKmemclear(hybridImagRight, NO_SUB_QMF_CHANNELS*sizeof(FIXP_DBL));
+
+
+  /*!
+  Hybrid analysis filterbank:
+  The lower 3 (5) of the 64 QMF subbands are further split to provide better frequency resolution.
+  for PS processing.
+  For the 10 and 20 stereo bands configuration, the QMF band H_0(w) is split
+  up into 8 (sub-) sub-bands and the QMF bands H_1(w) and H_2(w) are spit into 2 (sub-)
+  4th. (See figures 8.20 and 8.22 of ISO/IEC 14496-3:2001/FDAM 2:2004(E) )
+  */
+
+
+  if (h_ps_d->procFrameBased == 1)    /* If we have switched from frame to slot based processing  */
+  {                                   /* fill hybrid delay buffer.                                */
+    h_ps_d->procFrameBased = 0;
+
+    fillHybridDelayLine( rIntBufferLeft,
+                         iIntBufferLeft,
+                         hybridRealLeft,
+                         hybridImagLeft,
+                         hybridRealRight,
+                         hybridImagRight,
+                        &h_ps_d->specificTo.mpeg.hybrid );
+  }
+
+  slotBasedHybridAnalysis ( rIntBufferLeft[HYBRID_FILTER_DELAY], /* qmf filterbank values                         */
+                            iIntBufferLeft[HYBRID_FILTER_DELAY], /* qmf filterbank values                         */
+                            hybridRealLeft,                      /* hybrid filterbank values                      */
+                            hybridImagLeft,                      /* hybrid filterbank values                      */
+                           &h_ps_d->specificTo.mpeg.hybrid);          /* hybrid filterbank handle                      */
+
+
+  SCHAR hybridScal = h_ps_d->specificTo.mpeg.hybrid.sf_mQmfBuffer;
+
+
+  /*!
+  Decorrelation:
+  By means of all-pass filtering and delaying, the (sub-)sub-band samples s_k(n) are
+  converted into de-correlated (sub-)sub-band samples d_k(n).
+  - k: frequency in hybrid spectrum
+  - n: time index
+  */
+
+  deCorrelateSlotBased( h_ps_d,              /* parametric stereo decoder handle       */
+                        hybridRealLeft,      /* left hybrid time slot                  */
+                        hybridImagLeft,
+                        hybridScal,      /* scale factor of left hybrid time slot  */
+                        rIntBufferLeft[0],   /* left qmf time slot                     */
+                        iIntBufferLeft[0],
+                        sf_IntBuffer,        /* scale factor of left and right qmf time slot */
+                        hybridRealRight,     /* right hybrid time slot                 */
+                        hybridImagRight,
+                        rIntBufferRight,     /* right qmf time slot                    */
+                        iIntBufferRight );
+
+
+
+  /*!
+  Stereo Processing:
+  The sets of (sub-)sub-band samples s_k(n) and d_k(n) are processed according to
+  the stereo cues which are defined per stereo band.
+  */
+
+
+  applySlotBasedRotation( h_ps_d,            /* parametric stereo decoder handle       */
+                          hybridRealLeft,    /* left hybrid time slot                  */
+                          hybridImagLeft,
+                          rIntBufferLeft[0], /* left qmf time slot                     */
+                          iIntBufferLeft[0],
+                          hybridRealRight,   /* right hybrid time slot                 */
+                          hybridImagRight,
+                          rIntBufferRight,   /* right qmf time slot                    */
+                          iIntBufferRight );
+
+
+
+
+  /*!
+  Hybrid synthesis filterbank:
+  The stereo processed hybrid subband signals l_k(n) and r_k(n) are fed into the hybrid synthesis
+  filterbanks which are identical to the 64 complex synthesis filterbank of the SBR tool. The
+  input to the filterbank are slots of 64 QMF samples. For each slot the filterbank outputs one
+  block of 64 samples of one reconstructed stereo channel. The hybrid synthesis filterbank is
+  computed seperatly for the left and right channel.
+  */
+
+
+  /* left channel */
+  slotBasedHybridSynthesis ( hybridRealLeft,         /* one timeslot of hybrid filterbank values */
+                             hybridImagLeft,
+                             rIntBufferLeft[0],      /* one timeslot of qmf filterbank values    */
+                             iIntBufferLeft[0],
+                            &h_ps_d->specificTo.mpeg.hybrid );      /* hybrid filterbank handle                 */
+
+  /* right channel */
+  slotBasedHybridSynthesis ( hybridRealRight,        /* one timeslot of hybrid filterbank values */
+                             hybridImagRight,
+                             rIntBufferRight,        /* one timeslot of qmf filterbank values    */
+                             iIntBufferRight,
+                            &h_ps_d->specificTo.mpeg.hybrid );      /* hybrid filterbank handle                 */
+
+
+
+
+
+
+
+  /* free temporary hybrid qmf values of one timeslot */
+  C_ALLOC_SCRATCH_END(hybridImagRight, FIXP_DBL, NO_SUB_QMF_CHANNELS);
+  C_ALLOC_SCRATCH_END(hybridRealRight, FIXP_DBL, NO_SUB_QMF_CHANNELS);
+  C_ALLOC_SCRATCH_END(hybridImagLeft, FIXP_DBL, NO_SUB_QMF_CHANNELS);
+  C_ALLOC_SCRATCH_END(hybridRealLeft, FIXP_DBL, NO_SUB_QMF_CHANNELS);
+
+}/* END ApplyPsSlot */
+
+
+/***************************************************************************/
+/*!
+
+  \brief  assigns timeslots to an array
+
+  \return
+
+****************************************************************************/
+
+static void assignTimeSlotsPS (FIXP_DBL *bufAdr,
+                               FIXP_DBL **bufPtr,
+                               const int numSlots,
+                               const int numChan)
+{
+  FIXP_DBL  *ptr;
+  int slot;
+  ptr = bufAdr;
+  for(slot=0; slot < numSlots; slot++) {
+   bufPtr [slot] = ptr;
+    ptr += numChan;
+  }
+}
+
diff --git a/libSBRdec/src/psdec.h b/libSBRdec/src/psdec.h
new file mode 100644
index 0000000..e3a0424
--- /dev/null
+++ b/libSBRdec/src/psdec.h
@@ -0,0 +1,352 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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 decoder  
+*/
+#ifndef __PSDEC_H
+#define __PSDEC_H
+
+#include "sbrdecoder.h"
+
+
+
+/* This PS decoder implements the baseline version. So it always uses the     */
+/* hybrid filter structure for 20 stereo bands and does not implemet IPD/OPD  */
+/* synthesis. The baseline version has to support the complete PS bitstream   */
+/* syntax. But IPD/OPD data is ignored and set to 0. If 34 stereo band config */
+/* is used in the bitstream for IIS/ICC the decoded parameters are mapped to  */
+/* 20 stereo bands.                                                           */
+
+
+#include "FDK_bitstream.h"
+
+#include "psdec_hybrid.h"
+
+#define SCAL_HEADROOM                     ( 2 )
+
+#define PS_EXTENSION_SIZE_BITS            ( 4 )
+#define PS_EXTENSION_ESC_COUNT_BITS       ( 8 )
+
+#define NO_QMF_CHANNELS                   ( 64 )
+#define MAX_NUM_COL                       ( 32 )
+
+
+  #define NO_QMF_BANDS_HYBRID20           ( 3  )
+  #define NO_SUB_QMF_CHANNELS             ( 12 )
+
+  #define NRG_INT_COEFF                   ( 0.75f )
+  #define INT_FILTER_COEFF                (FL2FXCONST_DBL( 1.0f - NRG_INT_COEFF ))
+  #define PEAK_DECAY_FACTOR               (FL2FXCONST_DBL( 0.765928338364649f ))
+  #define TRANSIENT_IMPACT_FACTOR         (FL2FXCONST_DBL( 2.0 / 3.0 ))
+
+  #define NO_SERIAL_ALLPASS_LINKS         ( 3 )
+  #define MAX_NO_PS_ENV                   ( 4 + 1 )   /* +1 needed for VAR_BORDER */
+
+  #define MAX_DELAY_BUFFER_SIZE           ( 14 )
+  #define NO_DELAY_BUFFER_BANDS           ( 35 )
+
+  #define NO_HI_RES_BINS                  ( 34 )
+  #define NO_MID_RES_BINS                 ( 20 )
+  #define NO_LOW_RES_BINS                 ( 10 )
+
+  #define FIRST_DELAY_SB                  ( 23 )
+  #define NO_SAMPLE_DELAY_ALLPASS         ( 2 )
+  #define NO_DELAY_LENGTH_VECTORS         ( 12 )     /* d(m): d(0)=3 + d(1)=4 + d(2)=5 */
+
+  #define NO_HI_RES_IID_BINS              ( NO_HI_RES_BINS )
+  #define NO_HI_RES_ICC_BINS              ( NO_HI_RES_BINS )
+
+  #define NO_MID_RES_IID_BINS             ( NO_MID_RES_BINS )
+  #define NO_MID_RES_ICC_BINS             ( NO_MID_RES_BINS )
+
+  #define NO_LOW_RES_IID_BINS             ( NO_LOW_RES_BINS )
+  #define NO_LOW_RES_ICC_BINS             ( NO_LOW_RES_BINS )
+
+  #define SUBQMF_GROUPS                   ( 10 )
+  #define QMF_GROUPS                      ( 12 )
+
+  #define SUBQMF_GROUPS_HI_RES            ( 32 )
+  #define QMF_GROUPS_HI_RES               ( 18 )
+
+  #define NO_IID_GROUPS                   ( SUBQMF_GROUPS + QMF_GROUPS )
+  #define NO_IID_GROUPS_HI_RES            ( SUBQMF_GROUPS_HI_RES + QMF_GROUPS_HI_RES )
+
+  #define NO_IID_STEPS                    ( 7 )  /* 1 .. + 7 */
+  #define NO_IID_STEPS_FINE               ( 15 ) /* 1 .. +15 */
+  #define NO_ICC_STEPS                    ( 8 )  /* 0 .. + 7 */
+
+  #define NO_IID_LEVELS                   ( 2 * NO_IID_STEPS + 1 )      /* - 7 ..  + 7 */
+  #define NO_IID_LEVELS_FINE              ( 2 * NO_IID_STEPS_FINE + 1 ) /* -15 ..  +15 */
+  #define NO_ICC_LEVELS                   ( NO_ICC_STEPS )              /*   0 ..  + 7 */
+
+  #define FIXP_SQRT05                     ((FIXP_DBL)0x5a827980) /* 1/SQRT2 */
+
+  struct PS_DEC_COEFFICIENTS {
+
+    FIXP_DBL H11r[NO_IID_GROUPS];                       /*!< coefficients of the sub-subband groups                               */
+    FIXP_DBL H12r[NO_IID_GROUPS];                       /*!< coefficients of the sub-subband groups                               */
+    FIXP_DBL H21r[NO_IID_GROUPS];                       /*!< coefficients of the sub-subband groups                               */
+    FIXP_DBL H22r[NO_IID_GROUPS];                       /*!< coefficients of the sub-subband groups                               */
+
+    FIXP_DBL DeltaH11r[NO_IID_GROUPS];                  /*!< coefficients of the sub-subband groups                               */
+    FIXP_DBL DeltaH12r[NO_IID_GROUPS];                  /*!< coefficients of the sub-subband groups                               */
+    FIXP_DBL DeltaH21r[NO_IID_GROUPS];                  /*!< coefficients of the sub-subband groups                               */
+    FIXP_DBL DeltaH22r[NO_IID_GROUPS];                  /*!< coefficients of the sub-subband groups                               */
+
+    SCHAR aaIidIndexMapped[MAX_NO_PS_ENV][NO_HI_RES_IID_BINS];       /*!< The mapped IID index for all envelopes and all IID bins */
+    SCHAR aaIccIndexMapped[MAX_NO_PS_ENV][NO_HI_RES_ICC_BINS];       /*!< The mapped ICC index for all envelopes and all ICC bins */
+
+  };
+
+
+
+
+typedef enum {
+  ppt_none = 0,
+  ppt_mpeg = 1,
+  ppt_drm  = 2
+} PS_PAYLOAD_TYPE;
+
+
+typedef struct {
+  UCHAR bPsHeaderValid;                                      /*!< set if new header is available from bitstream                 */
+
+  UCHAR bEnableIid;                                          /*!< One bit denoting the presence of IID parameters               */
+  UCHAR bEnableIcc;                                          /*!< One bit denoting the presence of ICC parameters               */
+  UCHAR bEnableExt;                                          /*!< The PS extension layer is enabled using the enable_ext bit.
+                                                                  If it is set to %1 the IPD and OPD parameters are sent.
+                                                                  If it is disabled, i.e. %0, the extension layer is skipped.   */
+
+  UCHAR modeIid;                                             /*!< The configuration of IID parameters (number of bands and
+                                                                  quantisation grid, iid_quant) is determined by iid_mode.      */
+  UCHAR modeIcc;                                             /*!< The configuration of Inter-channel Coherence parameters
+                                                                  (number of bands and quantisation grid) is determined by
+                                                                  icc_mode.                                                     */
+
+  UCHAR freqResIid;                                          /*!< 0=low, 1=mid or 2=high frequency resolution for iid           */
+  UCHAR freqResIcc;                                          /*!< 0=low, 1=mid or 2=high frequency resolution for icc           */
+
+  UCHAR bFineIidQ;                                           /*!< Use fine Iid quantisation.                                    */
+
+  UCHAR bFrameClass;                                         /*!< The frame_class bit determines whether the parameter
+                                                                  positions of the current frame are uniformly spaced
+                                                                  accross the frame or they are defined using the positions
+                                                                  described by border_position.                                 */
+
+  UCHAR noEnv;                                               /*!< The number of envelopes per frame                             */
+  UCHAR aEnvStartStop[MAX_NO_PS_ENV+1];                      /*!< In case of variable parameter spacing the parameter
+                                                                  positions are determined by border_position                   */
+
+  SCHAR abIidDtFlag[MAX_NO_PS_ENV];                                /*!< Deltacoding time/freq flag for IID, 0 => freq           */
+  SCHAR abIccDtFlag[MAX_NO_PS_ENV];                                /*!< Deltacoding time/freq flag for ICC, 0 => freq           */
+
+  SCHAR aaIidIndex[MAX_NO_PS_ENV][NO_HI_RES_IID_BINS];             /*!< The IID index for all envelopes and all IID bins        */
+  SCHAR aaIccIndex[MAX_NO_PS_ENV][NO_HI_RES_ICC_BINS];             /*!< The ICC index for all envelopes and all ICC bins        */
+
+} MPEG_PS_BS_DATA;
+
+
+
+struct PS_DEC {
+
+  SCHAR noSubSamples;
+  SCHAR noChannels;
+
+  SCHAR procFrameBased;                                      /*!< Helper to detected switching from frame based to slot based
+                                                                  processing                                                    */
+
+  PS_PAYLOAD_TYPE bPsDataAvail[(1)+1];   /*!< set if new data available from bitstream                      */
+  UCHAR psDecodedPrv;                                        /*!< set if PS has been processed in the last frame                */
+
+  /* helpers for frame delay line */
+  UCHAR bsLastSlot;                                          /*!< Index of last read slot.                                      */
+  UCHAR bsReadSlot;                                          /*!< Index of current read slot for additional delay.              */
+  UCHAR processSlot;                                         /*!< Index of current slot for processing (need for add. delay).   */
+
+
+  INT  rescal;
+  INT  sf_IntBuffer;
+
+  union {  /* Bitstream data */
+    MPEG_PS_BS_DATA  mpeg;                                   /*!< Struct containing all MPEG specific PS data from bitstream.   */
+  } bsData[(1)+1];
+
+  shouldBeUnion {  /* Static data */
+    struct {
+      SCHAR aIidPrevFrameIndex[NO_HI_RES_IID_BINS];          /*!< The IID index for previous frame                              */
+      SCHAR aIccPrevFrameIndex[NO_HI_RES_ICC_BINS];          /*!< The ICC index for previous frame                              */
+
+      UCHAR delayBufIndex;                                         /*!< Pointer to where the latest sample is in buffer         */
+      UCHAR noSampleDelay;                                         /*!< How many QMF samples delay is used.                     */
+      UCHAR lastUsb;                                               /*!< uppermost WMF delay band of last frame                  */
+
+      UCHAR aDelayRBufIndexSer[NO_SERIAL_ALLPASS_LINKS];             /*!< Delay buffer for reverb filter                        */
+      UCHAR aDelayBufIndexDelayQmf[NO_QMF_CHANNELS-FIRST_DELAY_SB];  /*!< Delay buffer for ICC group 20 & 21                    */
+
+      SCHAR scaleFactorPsDelayBuffer;                                /*!< Scale factor for ps delay buffer                      */
+
+      /* hybrid filter bank delay lines */
+      FIXP_DBL aaQmfDelayBufReal[(NO_QMF_CHANNELS-FIRST_DELAY_SB) + (MAX_DELAY_BUFFER_SIZE-1)*(NO_DELAY_BUFFER_BANDS-FIRST_DELAY_SB)];
+      FIXP_DBL aaQmfDelayBufImag[(NO_QMF_CHANNELS-FIRST_DELAY_SB) + (MAX_DELAY_BUFFER_SIZE-1)*(NO_DELAY_BUFFER_BANDS-FIRST_DELAY_SB)];
+
+      FIXP_DBL *pAaRealDelayBufferQmf[MAX_DELAY_BUFFER_SIZE];          /*!< Real part delay buffer                                  */
+      FIXP_DBL *pAaImagDelayBufferQmf[MAX_DELAY_BUFFER_SIZE];          /*!< Imaginary part delay buffer                             */
+
+      FIXP_DBL aaRealDelayBufferQmf[NO_SAMPLE_DELAY_ALLPASS][FIRST_DELAY_SB]; /*!< Real part delay buffer     */
+      FIXP_DBL aaImagDelayBufferQmf[NO_SAMPLE_DELAY_ALLPASS][FIRST_DELAY_SB]; /*!< Imaginary part delay buffer*/
+
+      FIXP_DBL aaRealDelayBufferSubQmf[NO_SAMPLE_DELAY_ALLPASS][NO_SUB_QMF_CHANNELS];          /*!< Real part delay buffer          */
+      FIXP_DBL aaImagDelayBufferSubQmf[NO_SAMPLE_DELAY_ALLPASS][NO_SUB_QMF_CHANNELS];          /*!< Imaginary part delay buffer     */
+
+      FIXP_DBL aaaRealDelayRBufferSerQmf[FIRST_DELAY_SB][NO_DELAY_LENGTH_VECTORS];             /*!< Real part delay buffer          */
+      FIXP_DBL aaaImagDelayRBufferSerQmf[FIRST_DELAY_SB][NO_DELAY_LENGTH_VECTORS];             /*!< Imaginary part delay buffer     */
+
+      FIXP_DBL aaaRealDelayRBufferSerSubQmf[NO_SUB_QMF_CHANNELS][NO_DELAY_LENGTH_VECTORS];     /*!< Real part delay buffer          */
+      FIXP_DBL aaaImagDelayRBufferSerSubQmf[NO_SUB_QMF_CHANNELS][NO_DELAY_LENGTH_VECTORS];     /*!< Imaginary part delay buffer     */
+
+      HYBRID hybrid;                                      /*!< hybrid filter bank struct 1 or 2.                          */
+
+      FIXP_DBL aPrevNrgBin[NO_MID_RES_BINS];              /*!< energy of previous frame                                             */
+      FIXP_DBL aPrevPeakDiffBin[NO_MID_RES_BINS];         /*!< peak difference of previous frame                                    */
+      FIXP_DBL aPeakDecayFastBin[NO_MID_RES_BINS];        /*!< Saved max. peak decay value per bin                                  */
+      SCHAR aPowerPrevScal[NO_MID_RES_BINS];              /*!< Last power value (each bin) of previous frame                        */
+
+      FIXP_DBL h11rPrev[NO_IID_GROUPS];                   /*!< previous calculated h(xy) coefficients                               */
+      FIXP_DBL h12rPrev[NO_IID_GROUPS];                   /*!< previous calculated h(xy) coefficients                               */
+      FIXP_DBL h21rPrev[NO_IID_GROUPS];                   /*!< previous calculated h(xy) coefficients                               */
+      FIXP_DBL h22rPrev[NO_IID_GROUPS];                   /*!< previous calculated h(xy) coefficients                               */
+
+      PS_DEC_COEFFICIENTS  coef;                         /*!< temporal coefficients (reusable scratch memory)                 */
+
+    } mpeg;
+
+  } specificTo;
+
+
+};
+
+typedef struct PS_DEC *HANDLE_PS_DEC;
+
+
+int CreatePsDec(HANDLE_PS_DEC *h_PS_DEC, int aacSamplesPerFrame);
+
+int DeletePsDec(HANDLE_PS_DEC *h_PS_DEC);
+
+void
+scalFilterBankValues( HANDLE_PS_DEC   h_ps_d,                        /* parametric stereo decoder handle     */
+                      FIXP_DBL      **fixpQmfReal,                   /* qmf filterbank values                */
+                      FIXP_DBL      **fixpQmfImag,                   /* qmf filterbank values                */
+                      int             lsb,                           /* sbr start subband                    */
+                      int             scaleFactorLowBandSplitLow,
+                      int             scaleFactorLowBandSplitHigh,
+                      SCHAR          *scaleFactorLowBand_lb,
+                      SCHAR          *scaleFactorLowBand_hb,
+                      int             scaleFactorHighBands,
+                      INT            *scaleFactorHighBand,
+                      INT             noCols);
+
+void
+rescalFilterBankValues( HANDLE_PS_DEC   h_ps_d,                      /* parametric stereo decoder handle     */
+                        FIXP_DBL      **QmfBufferReal,               /* qmf filterbank values                */
+                        FIXP_DBL      **QmfBufferImag,               /* qmf filterbank values                */
+                        int             lsb,                         /* sbr start subband                    */
+                        INT             noCols);
+
+
+void
+initSlotBasedRotation( HANDLE_PS_DEC h_ps_d,
+                       int env,
+                       int usb);
+
+void
+ApplyPsSlot( HANDLE_PS_DEC h_ps_d,            /* parametric stereo decoder handle    */
+             FIXP_DBL **rIntBufferLeft,       /* real values of left qmf timeslot    */
+             FIXP_DBL **iIntBufferLeft,       /* imag values of left qmf timeslot    */
+             FIXP_DBL *rIntBufferRight,       /* real values of right qmf timeslot   */
+             FIXP_DBL *iIntBufferRight);      /* imag values of right qmf timeslot   */
+
+
+
+#endif  /* __PSDEC_H */
diff --git a/libSBRdec/src/psdec_hybrid.cpp b/libSBRdec/src/psdec_hybrid.cpp
new file mode 100644
index 0000000..7fc2c0a
--- /dev/null
+++ b/libSBRdec/src/psdec_hybrid.cpp
@@ -0,0 +1,652 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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 "psdec_hybrid.h"
+
+
+#include "fft.h"
+#include "sbr_ram.h"
+
+#include "FDK_tools_rom.h"
+#include "sbr_rom.h"
+
+/*******************************************************************************
+ Functionname:  InitHybridFilterBank
+ *******************************************************************************
+
+ Description:   Init one instance of HANDLE_HYBRID stuct
+
+ Arguments:
+
+ Return:        none
+
+*******************************************************************************/
+
+
+SBR_ERROR
+InitHybridFilterBank ( HANDLE_HYBRID hs,          /*!< Handle to HYBRID struct. */
+                       SCHAR frameSize,           /*!< Framesize (in Qmf súbband samples). */
+                       SCHAR noBands,             /*!< Number of Qmf bands for hybrid filtering. */
+                       const UCHAR *pResolution ) /*!< Resolution in Qmf bands (length noBands). */
+{
+  SCHAR i;
+  UCHAR maxNoChannels = 0;
+
+  for (i = 0; i < noBands; i++) {
+    hs->pResolution[i] = pResolution[i];
+    if(pResolution[i] > maxNoChannels)
+      maxNoChannels = pResolution[i];
+  }
+
+  hs->nQmfBands     = noBands;
+  hs->frameSize     = frameSize;
+  hs->qmfBufferMove = HYBRID_FILTER_LENGTH - 1;
+
+  hs->sf_mQmfBuffer = 0;
+
+  return SBRDEC_OK;
+}
+
+/*******************************************************************************
+ Functionname:  dualChannelFiltering
+ *******************************************************************************
+
+ Description:   fast 2-channel real-valued filtering with 6-tap delay.
+
+ Arguments:
+
+ Return:        none
+
+*******************************************************************************/
+
+/*!
+2 channel filter
+<pre>
+   Filter Coefs:
+   0.0,
+   0.01899487526049,
+   0.0,
+   -0.07293139167538,
+   0.0,
+   0.30596630545168,
+   0.5,
+   0.30596630545168,
+   0.0,
+   -0.07293139167538,
+   0.0,
+   0.01899487526049,
+   0.0
+
+
+   Filter design:
+   h[q,n] = g[n] * cos(2pi/2 * q * (n-6) );  n = 0..12,  q = 0,1;
+
+   ->  h[0,n] = g[n] * 1;
+   ->  h[1,n] = g[n] * pow(-1,n);
+</pre>
+*/
+
+static void slotBasedDualChannelFiltering( const FIXP_DBL *pQmfReal,
+                                           const FIXP_DBL *pQmfImag,
+
+                                           FIXP_DBL       *mHybridReal,
+                                           FIXP_DBL       *mHybridImag)
+{
+
+  FIXP_DBL  t1, t3, t5, t6;
+
+  /* symmetric filter coefficients */
+
+  /*  you don't have to shift the result after fMult because of p2_13_20 <= 0.5 */
+  t1 = fMultDiv2(p2_13_20[1] , ( (pQmfReal[1] >> 1) + (pQmfReal[11] >> 1)));
+  t3 = fMultDiv2(p2_13_20[3] , ( (pQmfReal[3] >> 1) + (pQmfReal[ 9] >> 1)));
+  t5 = fMultDiv2(p2_13_20[5] , ( (pQmfReal[5] >> 1) + (pQmfReal[ 7] >> 1)));
+  t6 = fMultDiv2(p2_13_20[6] ,   (pQmfReal[6] >> 1) );
+
+  mHybridReal[0] = (t1 + t3 + t5 + t6) << 2;
+  mHybridReal[1] = (- t1 - t3 - t5 + t6) << 2;
+
+  t1 = fMultDiv2(p2_13_20[1] , ( (pQmfImag[1] >> 1) + (pQmfImag[11] >> 1)));
+  t3 = fMultDiv2(p2_13_20[3] , ( (pQmfImag[3] >> 1) + (pQmfImag[ 9] >> 1)));
+  t5 = fMultDiv2(p2_13_20[5] , ( (pQmfImag[5] >> 1) + (pQmfImag[ 7] >> 1)));
+  t6 = fMultDiv2(p2_13_20[6] ,    pQmfImag[6] >> 1 );
+
+  mHybridImag[0] = (t1 + t3 + t5 + t6) << 2;
+  mHybridImag[1] = (- t1 - t3 - t5 + t6) << 2;
+}
+
+
+/*******************************************************************************
+ Functionname:  eightChannelFiltering
+ *******************************************************************************
+
+ Description:   fast 8-channel complex-valued filtering with 6-tap delay.
+
+ Arguments:
+
+ Return:        none
+
+*******************************************************************************/
+/*!
+   8 channel filter
+
+   Implementation using a FFT of length 8
+<pre>
+   prototype filter coefficients:
+   0.00746082949812   0.02270420949825   0.04546865930473   0.07266113929591   0.09885108575264   0.11793710567217
+   0.125
+   0.11793710567217   0.09885108575264   0.07266113929591   0.04546865930473   0.02270420949825   0.00746082949812
+
+   Filter design:
+   N = 13; Q = 8;
+   h[q,n]       = g[n] * exp(j * 2 * pi / Q * (q + .5) * (n - 6));  n = 0..(N-1),  q = 0..(Q-1);
+
+   Time Signal:   x[t];
+   Filter Bank Output
+   y[q,t] = conv(x[t],h[q,t]) = conv(h[q,t],x[t]) = sum(x[k] * h[q, t - k] ) = sum(h[q, k] * x[t - k] ); k = 0..(N-1);
+
+   y[q,t] =   x[t - 12]*h[q, 12]  +  x[t - 11]*h[q, 11]  +  x[t - 10]*h[q, 10]  +  x[t -  9]*h[q,  9]
+           +  x[t -  8]*h[q,  8]  +  x[t -  7]*h[q,  7]
+           +  x[t -  6]*h[q,  6]
+           +  x[t -  5]*h[q,  5]  +  x[t -  4]*h[q,  4]
+           +  x[t -  3]*h[q,  3]  +  x[t -  2]*h[q,  2]  +  x[t -  1]*h[q,  1]  +  x[t -  0]*h[q,  0];
+
+   h'[q, n] = h[q,(N-1)-n] = g[n] * exp(j * 2 * pi / Q * (q + .5) * (6 - n));  n = 0..(N-1),  q = 0..(Q-1);
+
+   y[q,t] =   x[t - 12]*h'[q,  0]  +  x[t - 11]*h'[q,  1]  +  x[t - 10]*h'[q,  2]  +  x[t -  9]*h'[q,  3]
+           +  x[t -  8]*h'[q,  4]  +  x[t -  7]*h'[q,  5]
+           +  x[t -  6]*h'[q,  6]
+           +  x[t -  5]*h'[q,  7]  +  x[t -  4]*h'[q,  8]
+           +  x[t -  3]*h'[q,  9]  +  x[t -  2]*h'[q, 10]  +  x[t -  1]*h'[q, 11]  +  x[t -  0]*h'[q, 12];
+
+   Try to split off FFT Modulation Term:
+   FFT(x[t], q) = sum(x[t+k]*exp(-j*2*pi/N *q * k))
+                                           c                                           m
+   Step 1:  h'[q,n] = g[n] * ( exp(j * 2 * pi / 8 * .5 * (6 - n)) ) * ( exp (j * 2 * pi / 8 * q * (6 - n)) );
+
+    h'[q,n] = g[n] *c[n] * m[q,n]; (see above)
+    c[n]    = exp( j * 2 * pi / 8 * .5 * (6 - n) );
+    m[q,n]  = exp( j * 2 * pi / 8 *  q * (6 - n) );
+
+    y[q,t] = x[t -  0]*g[0]*c[0]*m[q,0]  +  x[t -  1]*g[1]*c[ 1]*m[q, 1]  + ...
+             ...                         +  x[t - 12]*g[2]*c[12]*m[q,12];
+
+                                                                              |
+    n                   m                            *exp(-j*2*pi)            |   n'                   fft
+-------------------------------------------------------------------------------------------------------------------------
+    0       exp( j * 2 * pi / 8 * q * 6) ->  exp(-j * 2 * pi / 8 * q * 2)     |   2         exp(-j * 2 * pi / 8 * q * 0)
+    1       exp( j * 2 * pi / 8 * q * 5) ->  exp(-j * 2 * pi / 8 * q * 3)     |   3         exp(-j * 2 * pi / 8 * q * 1)
+    2       exp( j * 2 * pi / 8 * q * 4) ->  exp(-j * 2 * pi / 8 * q * 4)     |   4         exp(-j * 2 * pi / 8 * q * 2)
+    3       exp( j * 2 * pi / 8 * q * 3) ->  exp(-j * 2 * pi / 8 * q * 5)     |   5         exp(-j * 2 * pi / 8 * q * 3)
+    4       exp( j * 2 * pi / 8 * q * 2) ->  exp(-j * 2 * pi / 8 * q * 6)     |   6         exp(-j * 2 * pi / 8 * q * 4)
+    5       exp( j * 2 * pi / 8 * q * 1) ->  exp(-j * 2 * pi / 8 * q * 7)     |   7         exp(-j * 2 * pi / 8 * q * 5)
+    6       exp( j * 2 * pi / 8 * q * 0)                                      |   0         exp(-j * 2 * pi / 8 * q * 6)
+    7       exp(-j * 2 * pi / 8 * q * 1)                                      |   1         exp(-j * 2 * pi / 8 * q * 7)
+    8       exp(-j * 2 * pi / 8 * q * 2)                                      |   2
+    9       exp(-j * 2 * pi / 8 * q * 3)                                      |   3
+    10      exp(-j * 2 * pi / 8 * q * 4)                                      |   4
+    11      exp(-j * 2 * pi / 8 * q * 5)                                      |   5
+    12      exp(-j * 2 * pi / 8 * q * 6)                                      |   6
+
+
+    now use fft modulation coefficients
+    m[6]  =       = fft[0]
+    m[7]  =       = fft[1]
+    m[8]  = m[ 0] = fft[2]
+    m[9]  = m[ 1] = fft[3]
+    m[10] = m[ 2] = fft[4]
+    m[11] = m[ 3] = fft[5]
+    m[12] = m[ 4] = fft[6]
+            m[ 5] = fft[7]
+
+    y[q,t] = (                       x[t- 6]*g[ 6]*c[ 6] ) * fft[q,0]  +
+             (                       x[t- 7]*g[ 7]*c[ 7] ) * fft[q,1]  +
+             ( x[t- 0]*g[ 0]*c[ 0] + x[t- 8]*g[ 8]*c[ 8] ) * fft[q,2]  +
+             ( x[t- 1]*g[ 1]*c[ 1] + x[t- 9]*g[ 9]*c[ 9] ) * fft[q,3]  +
+             ( x[t- 2]*g[ 2]*c[ 2] + x[t-10]*g[10]*c[10] ) * fft[q,4]  +
+             ( x[t- 3]*g[ 3]*c[ 3] + x[t-11]*g[11]*c[11] ) * fft[q,5]  +
+             ( x[t- 4]*g[ 4]*c[ 4] + x[t-12]*g[12]*c[12] ) * fft[q,6]  +
+             ( x[t- 5]*g[ 5]*c[ 5]                       ) * fft[q,7];
+
+    pre twiddle factors c[n] = exp(j * 2 * pi / 8 * .5 * (6 - n));
+    n                c]           |  n                c[n]         |  n                c[n]
+---------------------------------------------------------------------------------------------------
+    0       exp( j * 6 * pi / 8)  |  1       exp( j * 5 * pi / 8)  |  2       exp( j * 4 * pi / 8)
+    3       exp( j * 3 * pi / 8)  |  4       exp( j * 2 * pi / 8)  |  5       exp( j * 1 * pi / 8)
+    6       exp( j * 0 * pi / 8)  |  7       exp(-j * 1 * pi / 8)  |  8       exp(-j * 2 * pi / 8)
+    9       exp(-j * 3 * pi / 8)  | 10       exp(-j * 4 * pi / 8)  | 11       exp(-j * 5 * pi / 8)
+   12       exp(-j * 6 * pi / 8)  |                                |
+</pre>
+*/
+
+/* defining rotation factors for *ChannelFiltering */
+
+#define cos0Pi    FL2FXCONST_DBL( 1.f)
+#define sin0Pi    FL2FXCONST_DBL( 0.f)
+
+#define cos1Pi    FL2FXCONST_DBL(-1.f)
+#define sin1Pi    FL2FXCONST_DBL( 0.f)
+
+#define cos1Pi_2  FL2FXCONST_DBL( 0.f)
+#define sin1Pi_2  FL2FXCONST_DBL( 1.f)
+
+#define cos1Pi_3  FL2FXCONST_DBL( 0.5f)
+#define sin1Pi_3  FL2FXCONST_DBL( 0.86602540378444f)
+
+#define cos0Pi_4  cos0Pi
+#define cos1Pi_4  FL2FXCONST_DBL(0.70710678118655f)
+#define cos2Pi_4  cos1Pi_2
+#define cos3Pi_4  (-cos1Pi_4)
+#define cos4Pi_4  (-cos0Pi_4)
+#define cos5Pi_4  cos3Pi_4
+#define cos6Pi_4  cos2Pi_4
+
+#define sin0Pi_4  sin0Pi
+#define sin1Pi_4  FL2FXCONST_DBL(0.70710678118655f)
+#define sin2Pi_4  sin1Pi_2
+#define sin3Pi_4  sin1Pi_4
+#define sin4Pi_4  sin0Pi_4
+#define sin5Pi_4  (-sin3Pi_4)
+#define sin6Pi_4  (-sin2Pi_4)
+
+#define cos0Pi_8  cos0Pi
+#define cos1Pi_8  FL2FXCONST_DBL(0.92387953251129f)
+#define cos2Pi_8  cos1Pi_4
+#define cos3Pi_8  FL2FXCONST_DBL(0.38268343236509f)
+#define cos4Pi_8  cos2Pi_4
+#define cos5Pi_8  (-cos3Pi_8)
+#define cos6Pi_8  (-cos2Pi_8)
+
+#define sin0Pi_8  sin0Pi
+#define sin1Pi_8  cos3Pi_8
+#define sin2Pi_8  sin1Pi_4
+#define sin3Pi_8  cos1Pi_8
+#define sin4Pi_8  sin2Pi_4
+#define sin5Pi_8  sin3Pi_8
+#define sin6Pi_8  sin1Pi_4
+
+#if defined(ARCH_PREFER_MULT_32x16)
+  #define FIXP_HYB FIXP_SGL
+  #define FIXP_CAST FX_DBL2FX_SGL
+#else
+  #define FIXP_HYB FIXP_DBL
+  #define FIXP_CAST
+#endif
+
+static const FIXP_HYB  cr[13] =
+{
+   FIXP_CAST(cos6Pi_8), FIXP_CAST(cos5Pi_8), FIXP_CAST(cos4Pi_8),
+   FIXP_CAST(cos3Pi_8), FIXP_CAST(cos2Pi_8), FIXP_CAST(cos1Pi_8),
+   FIXP_CAST(cos0Pi_8),
+   FIXP_CAST(cos1Pi_8), FIXP_CAST(cos2Pi_8), FIXP_CAST(cos3Pi_8),
+   FIXP_CAST(cos4Pi_8), FIXP_CAST(cos5Pi_8), FIXP_CAST(cos6Pi_8)
+};
+
+static const FIXP_HYB  ci[13] =
+{
+   FIXP_CAST( sin6Pi_8), FIXP_CAST( sin5Pi_8), FIXP_CAST( sin4Pi_8),
+   FIXP_CAST( sin3Pi_8), FIXP_CAST( sin2Pi_8), FIXP_CAST( sin1Pi_8),
+   FIXP_CAST( sin0Pi_8) ,
+   FIXP_CAST(-sin1Pi_8), FIXP_CAST(-sin2Pi_8), FIXP_CAST(-sin3Pi_8),
+   FIXP_CAST(-sin4Pi_8), FIXP_CAST(-sin5Pi_8), FIXP_CAST(-sin6Pi_8)
+};
+
+static void slotBasedEightChannelFiltering( const FIXP_DBL *pQmfReal,
+                                            const FIXP_DBL *pQmfImag,
+
+                                            FIXP_DBL  *mHybridReal,
+                                            FIXP_DBL  *mHybridImag)
+{
+
+  int bin;
+  FIXP_DBL _fft[128 + ALIGNMENT_DEFAULT - 1];
+  FIXP_DBL *fft = (FIXP_DBL *)ALIGN_PTR(_fft);
+
+#if defined(ARCH_PREFER_MULT_32x16)
+  const FIXP_SGL *p = p8_13_20; /* BASELINE_PS */
+#else
+  const FIXP_DBL *p = p8_13_20; /* BASELINE_PS */
+#endif
+
+  /* pre twiddeling */
+
+  /*   x*(a*b + c*d) = fMultDiv2(x, fMultAddDiv2(fMultDiv2(a, b), c, d)) */
+  /*   x*(a*b - c*d) = fMultDiv2(x, fMultSubDiv2(fMultDiv2(a, b), c, d)) */
+  FIXP_DBL accu1, accu2, accu3, accu4;
+
+  #define TWIDDLE_1(n_0,n_1,n_2)                                                        \
+         cplxMultDiv2(&accu1, &accu2, pQmfReal[n_0], pQmfImag[n_0], cr[n_0], ci[n_0]);  \
+         accu1 = fMultDiv2(p[n_0], accu1);                                              \
+         accu2 = fMultDiv2(p[n_0], accu2);                                              \
+         cplxMultDiv2(&accu3, &accu4, pQmfReal[n_1], pQmfImag[n_1], cr[n_1], ci[n_1]);  \
+         accu3 = fMultDiv2(p[n_1], accu3);                                              \
+         accu4 = fMultDiv2(p[n_1], accu4);                                              \
+         fft[FIXP_FFT_IDX_R(n_2)] = accu1 + accu3;                                      \
+         fft[FIXP_FFT_IDX_I(n_2)] = accu2 + accu4;
+
+  #define TWIDDLE_0(n_0,n_1)                                                            \
+         cplxMultDiv2(&accu1, &accu2, pQmfReal[n_0], pQmfImag[n_0], cr[n_0], ci[n_0]);  \
+         fft[FIXP_FFT_IDX_R(n_1)] = fMultDiv2(p[n_0], accu1);                           \
+         fft[FIXP_FFT_IDX_I(n_1)] = fMultDiv2(p[n_0], accu2);
+
+  TWIDDLE_0( 6, 0)
+  TWIDDLE_0( 7, 1)
+
+  TWIDDLE_1( 0, 8, 2)
+  TWIDDLE_1( 1, 9, 3)
+  TWIDDLE_1( 2,10, 4)
+  TWIDDLE_1( 3,11, 5)
+  TWIDDLE_1( 4,12, 6)
+
+  TWIDDLE_0( 5, 7)
+
+  fft_8 (fft);
+
+  /* resort fft data into output array*/
+  for(bin=0; bin<8;bin++ ) {
+    mHybridReal[bin] = fft[FIXP_FFT_IDX_R(bin)] << 4;
+    mHybridImag[bin] = fft[FIXP_FFT_IDX_I(bin)] << 4;
+  }
+}
+
+
+/*******************************************************************************
+ Functionname:  fillHybridDelayLine
+ *******************************************************************************
+
+ Description:   The delay line of the hybrid filter is filled and copied from
+                left to right.
+
+ Return:        none
+
+*******************************************************************************/
+
+void
+fillHybridDelayLine( FIXP_DBL **fixpQmfReal,          /*!< Qmf real Values    */
+                     FIXP_DBL **fixpQmfImag,          /*!< Qmf imag Values    */
+                     FIXP_DBL   fixpHybridLeftR[12],  /*!< Hybrid real Values left channel  */
+                     FIXP_DBL   fixpHybridLeftI[12],  /*!< Hybrid imag Values left channel  */
+                     FIXP_DBL   fixpHybridRightR[12], /*!< Hybrid real Values right channel */
+                     FIXP_DBL   fixpHybridRightI[12], /*!< Hybrid imag Values right channel */
+                     HANDLE_HYBRID hHybrid )
+{
+  int i;
+
+  for (i = 0; i < HYBRID_FILTER_DELAY; i++) {
+    slotBasedHybridAnalysis ( fixpQmfReal[i],
+                              fixpQmfReal[i],
+                              fixpHybridLeftR,
+                              fixpHybridLeftI,
+                              hHybrid );
+  }
+
+  FDKmemcpy(fixpHybridRightR, fixpHybridLeftR, sizeof(FIXP_DBL)*NO_SUB_QMF_CHANNELS);
+  FDKmemcpy(fixpHybridRightI, fixpHybridLeftI, sizeof(FIXP_DBL)*NO_SUB_QMF_CHANNELS);
+}
+
+
+/*******************************************************************************
+ Functionname:  slotBasedHybridAnalysis
+ *******************************************************************************
+
+ Description:   The lower QMF subbands are further split to provide better
+                frequency resolution for PS processing.
+
+  Return:        none
+
+*******************************************************************************/
+
+
+void
+slotBasedHybridAnalysis ( FIXP_DBL *fixpQmfReal,      /*!< Qmf real Values */
+                          FIXP_DBL *fixpQmfImag,      /*!< Qmf imag Values */
+
+                          FIXP_DBL  fixpHybridReal[12],   /*!< Hybrid real Values */
+                          FIXP_DBL  fixpHybridImag[12],   /*!< Hybrid imag Values */
+
+                          HANDLE_HYBRID hHybrid)
+{
+  int  k, band;
+  HYBRID_RES hybridRes;
+  int  chOffset = 0;
+
+  C_ALLOC_SCRATCH_START(pTempRealSlot, FIXP_DBL, 4*HYBRID_FILTER_LENGTH);
+
+  FIXP_DBL *pTempImagSlot = pTempRealSlot + HYBRID_FILTER_LENGTH;
+  FIXP_DBL *pWorkRealSlot = pTempImagSlot + HYBRID_FILTER_LENGTH;
+  FIXP_DBL *pWorkImagSlot = pWorkRealSlot + HYBRID_FILTER_LENGTH;
+
+  /*!
+  Hybrid filtering is applied to the first hHybrid->nQmfBands QMF bands (3 when 10 or 20 stereo bands
+  are used, 5 when 34 stereo bands are used). For the remaining QMF bands a delay would be necessary.
+  But there is no need to implement a delay because there is a look-ahead of HYBRID_FILTER_DELAY = 6
+  QMF samples in the low-band buffer.
+  */
+
+  for(band = 0; band < hHybrid->nQmfBands; band++) {
+
+    /*  get hybrid resolution per qmf band                */
+    /*  in case of baseline ps 10/20 band stereo mode :   */
+    /*                                                    */
+    /*             qmfBand[0] : 8 ( HYBRID_8_CPLX )       */
+    /*             qmfBand[1] : 2 ( HYBRID_2_REAL )       */
+    /*             qmfBand[2] : 2 ( HYBRID_2_REAL )       */
+    /*                                                    */
+    /*  (split the 3 lower qmf band to 12 hybrid bands)   */
+
+    hybridRes = (HYBRID_RES)hHybrid->pResolution[band];
+
+    FDKmemcpy(pWorkRealSlot, hHybrid->mQmfBufferRealSlot[band], hHybrid->qmfBufferMove * sizeof(FIXP_DBL));
+    FDKmemcpy(pWorkImagSlot, hHybrid->mQmfBufferImagSlot[band], hHybrid->qmfBufferMove * sizeof(FIXP_DBL));
+
+    pWorkRealSlot[hHybrid->qmfBufferMove] = fixpQmfReal[band];
+    pWorkImagSlot[hHybrid->qmfBufferMove] = fixpQmfImag[band];
+
+    FDKmemcpy(hHybrid->mQmfBufferRealSlot[band], pWorkRealSlot + 1, hHybrid->qmfBufferMove * sizeof(FIXP_DBL));
+    FDKmemcpy(hHybrid->mQmfBufferImagSlot[band], pWorkImagSlot + 1, hHybrid->qmfBufferMove * sizeof(FIXP_DBL));
+
+    if (fixpQmfReal) {
+
+      /* actual filtering only if output signal requested */
+      switch( hybridRes ) {
+
+      /* HYBRID_2_REAL & HYBRID_8_CPLX are only needful for baseline ps */
+      case HYBRID_2_REAL:
+
+        slotBasedDualChannelFiltering( pWorkRealSlot,
+                                       pWorkImagSlot,
+                                       pTempRealSlot,
+                                       pTempImagSlot);
+        break;
+
+      case HYBRID_8_CPLX:
+
+        slotBasedEightChannelFiltering( pWorkRealSlot,
+                                        pWorkImagSlot,
+                                        pTempRealSlot,
+                                        pTempImagSlot);
+        break;
+
+      default:
+        FDK_ASSERT(0);
+      }
+
+      for(k = 0; k < (SCHAR)hybridRes; k++) {
+        fixpHybridReal [chOffset + k] = pTempRealSlot[k];
+        fixpHybridImag [chOffset + k] = pTempImagSlot[k];
+      }
+      chOffset += hybridRes;
+    } /* if (mHybridReal) */
+  }
+
+  /* group hybrid channels 3+4 -> 3 and 2+5 -> 2 */
+  fixpHybridReal[3] += fixpHybridReal[4];
+  fixpHybridImag[3] += fixpHybridImag[4];
+  fixpHybridReal[4] = (FIXP_DBL)0;
+  fixpHybridImag[4] = (FIXP_DBL)0;
+
+  fixpHybridReal[2] += fixpHybridReal[5];
+  fixpHybridImag[2] += fixpHybridImag[5];
+  fixpHybridReal[5] = (FIXP_DBL)0;
+  fixpHybridImag[5] = (FIXP_DBL)0;
+
+  /* free memory on scratch */
+  C_ALLOC_SCRATCH_END(pTempRealSlot, FIXP_DBL, 4*HYBRID_FILTER_LENGTH);
+
+}
+
+
+/*******************************************************************************
+ Functionname:  slotBasedHybridSynthesis
+ *******************************************************************************
+
+ Description:  The coefficients offering higher resolution for the lower QMF
+               channel are simply added prior to the synthesis with the 54
+               subbands QMF.
+
+ Arguments:
+
+ Return:        none
+
+*******************************************************************************/
+
+/*! <pre>
+      l,r0(n) ---\
+      l,r1(n) ---- + --\
+      l,r2(n) ---/      \
+                         + --> F0(w)
+      l,r3(n) ---\      /
+      l,r4(n) ---- + --/
+      l,r5(n) ---/
+
+
+      l,r6(n) ---\
+                  + ---------> F1(w)
+      l,r7(n) ---/
+
+
+      l,r8(n) ---\
+                  + ---------> F2(w)
+      l,r9(n) ---/
+
+    </pre>
+      Hybrid QMF synthesis filterbank for the 10 and 20 stereo-bands configurations. The
+      coefficients offering higher resolution for the lower QMF channel are simply added
+      prior to the synthesis with the 54 subbands QMF.
+
+      [see ISO/IEC 14496-3:2001/FDAM 2:2004(E) - Page 52]
+*/
+
+
+void
+slotBasedHybridSynthesis ( FIXP_DBL  *fixpHybridReal,  /*!< Hybrid real Values */
+                           FIXP_DBL  *fixpHybridImag,  /*!< Hybrid imag Values */
+                           FIXP_DBL  *fixpQmfReal,     /*!< Qmf real Values */
+                           FIXP_DBL  *fixpQmfImag,     /*!< Qmf imag Values */
+                           HANDLE_HYBRID hHybrid )     /*!< Handle to HYBRID struct. */
+{
+  int  k, band;
+
+  HYBRID_RES hybridRes;
+  int  chOffset = 0;
+
+  for(band = 0; band < hHybrid->nQmfBands; band++) {
+
+    FIXP_DBL qmfReal = FL2FXCONST_DBL(0.f);
+    FIXP_DBL qmfImag = FL2FXCONST_DBL(0.f);
+    hybridRes = (HYBRID_RES)hHybrid->pResolution[band];
+
+    for(k = 0; k < (SCHAR)hybridRes; k++) {
+      qmfReal += fixpHybridReal[chOffset + k];
+      qmfImag += fixpHybridImag[chOffset + k];
+    }
+
+    fixpQmfReal[band] = qmfReal;
+    fixpQmfImag[band] = qmfImag;
+
+    chOffset += hybridRes;
+  }
+}
+
+
+
diff --git a/libSBRdec/src/psdec_hybrid.h b/libSBRdec/src/psdec_hybrid.h
new file mode 100644
index 0000000..6503df9
--- /dev/null
+++ b/libSBRdec/src/psdec_hybrid.h
@@ -0,0 +1,165 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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 __HYBRID_H
+#define __HYBRID_H
+
+#include "sbrdecoder.h"
+
+
+#define HYBRID_FILTER_LENGTH  13
+#define HYBRID_FILTER_DELAY    6
+
+
+#define FAST_FILTER2
+#define FAST_FILTER4
+#define FAST_FILTER8
+#define FAST_FILTER12
+
+#define FFT_IDX_R(a)  (2*a)
+#define FFT_IDX_I(a)  (2*a+1)
+
+#define FIXP_FFT_IDX_R(a)   (a<<1)
+#define FIXP_FFT_IDX_I(a)  ((a<<1) + 1)
+
+
+typedef enum {
+
+  HYBRID_2_REAL     = 2,
+  HYBRID_4_CPLX     = 4,
+  HYBRID_8_CPLX     = 8,
+  HYBRID_12_CPLX    = 12
+
+} HYBRID_RES;
+
+typedef struct
+{
+  SCHAR   nQmfBands;
+  SCHAR   frameSize;
+  SCHAR   qmfBufferMove;
+
+  UCHAR   pResolution[3];
+
+  FIXP_DBL mQmfBufferRealSlot[3][HYBRID_FILTER_LENGTH];   /**< Stores old Qmf samples. */
+  FIXP_DBL mQmfBufferImagSlot[3][HYBRID_FILTER_LENGTH];
+  SCHAR sf_mQmfBuffer;
+
+} HYBRID;
+
+typedef HYBRID *HANDLE_HYBRID;
+
+void
+fillHybridDelayLine( FIXP_DBL **fixpQmfReal,
+                     FIXP_DBL **fixpQmfImag,
+                     FIXP_DBL   fixpHybridLeftR[12],
+                     FIXP_DBL   fixpHybridLeftI[12],
+                     FIXP_DBL   fixpHybridRightR[12],
+                     FIXP_DBL   fixpHybridRightI[12],
+                     HANDLE_HYBRID hHybrid );
+
+void
+slotBasedHybridAnalysis ( FIXP_DBL *fixpQmfReal,
+                          FIXP_DBL *fixpQmfImag,
+
+                          FIXP_DBL  *fixpHybridReal,
+                          FIXP_DBL  *fixpHybridImag,
+
+                          HANDLE_HYBRID hHybrid);
+
+
+void
+slotBasedHybridSynthesis ( FIXP_DBL  *fixpHybridReal,
+                           FIXP_DBL  *fixpHybridImag,
+
+                           FIXP_DBL  *fixpQmfReal,
+                           FIXP_DBL  *fixpQmfImag,
+
+                           HANDLE_HYBRID hHybrid );
+
+SBR_ERROR InitHybridFilterBank ( HANDLE_HYBRID  hHybrid,
+                                 SCHAR frameSize,
+                                 SCHAR noBands,
+                                 const UCHAR *pResolution );
+
+
+#endif /* __HYBRID_H */
diff --git a/libSBRdec/src/sbr_crc.cpp b/libSBRdec/src/sbr_crc.cpp
new file mode 100644
index 0000000..760bd1f
--- /dev/null
+++ b/libSBRdec/src/sbr_crc.cpp
@@ -0,0 +1,183 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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  CRC check coutines  
+*/
+
+#include "sbr_crc.h"
+
+#include "FDK_bitstream.h"
+#include "transcendent.h"
+
+#define MAXCRCSTEP    16
+#define MAXCRCSTEP_LD  4
+
+/*!
+  \brief     crc calculation
+*/
+static ULONG
+calcCRC (HANDLE_CRC hCrcBuf, ULONG bValue, int nBits)
+{
+  int i;
+  ULONG bMask = (1UL << (nBits - 1));
+
+  for (i = 0; i < nBits; i++, bMask >>= 1) {
+    USHORT flag = (hCrcBuf->crcState & hCrcBuf->crcMask) ? 1 : 0;
+    USHORT flag1 = (bMask & bValue) ? 1 : 0;
+
+    flag ^= flag1;
+    hCrcBuf->crcState <<= 1;
+    if (flag)
+      hCrcBuf->crcState ^= hCrcBuf->crcPoly;
+  }
+
+  return (hCrcBuf->crcState);
+}
+
+
+/*!
+  \brief     crc
+*/
+static int
+getCrc (HANDLE_FDK_BITSTREAM hBs, ULONG NrBits)
+{
+  int i;
+  CRC_BUFFER CrcBuf;
+
+  CrcBuf.crcState = SBR_CRC_START;
+  CrcBuf.crcPoly  = SBR_CRC_POLY;
+  CrcBuf.crcMask  = SBR_CRC_MASK;
+
+  int CrcStep = NrBits>>MAXCRCSTEP_LD;
+
+  int CrcNrBitsRest = (NrBits - CrcStep * MAXCRCSTEP);
+  ULONG bValue;
+
+  for (i = 0; i < CrcStep; i++) {
+    bValue = FDKreadBits (hBs, MAXCRCSTEP);
+    calcCRC (&CrcBuf, bValue, MAXCRCSTEP);
+  }
+
+  bValue = FDKreadBits (hBs, CrcNrBitsRest);
+  calcCRC (&CrcBuf, bValue, CrcNrBitsRest);
+
+  return (CrcBuf.crcState & SBR_CRC_RANGE);
+
+}
+
+
+/*!
+  \brief   crc interface
+  \return  1: CRC OK, 0: CRC check failure
+*/
+int
+SbrCrcCheck (HANDLE_FDK_BITSTREAM hBs, /*!< handle to bit-buffer  */
+             LONG NrBits)              /*!< max. CRC length       */
+{
+  int crcResult = 1;
+  ULONG NrCrcBits;
+  ULONG crcCheckResult;
+  LONG NrBitsAvailable;
+  ULONG crcCheckSum;
+
+  crcCheckSum = FDKreadBits (hBs, 10);
+
+  NrBitsAvailable = FDKgetValidBits(hBs);
+  if (NrBitsAvailable <= 0){
+    return 0;
+  }
+
+  NrCrcBits = fixMin ((INT)NrBits, (INT)NrBitsAvailable);
+
+  crcCheckResult = getCrc (hBs, NrCrcBits);
+  FDKpushBack(hBs, (NrBitsAvailable - FDKgetValidBits(hBs)) );
+
+
+  if (crcCheckResult != crcCheckSum) {
+    crcResult = 0;
+  }
+
+  return (crcResult);
+}
diff --git a/libSBRdec/src/sbr_crc.h b/libSBRdec/src/sbr_crc.h
new file mode 100644
index 0000000..542843d
--- /dev/null
+++ b/libSBRdec/src/sbr_crc.h
@@ -0,0 +1,123 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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  CRC checking routines  
+*/
+#ifndef __SBR_CRC_H
+#define __SBR_CRC_H
+
+#include "sbrdecoder.h"
+
+#include "FDK_bitstream.h"
+
+/* some useful crc polynoms:
+
+crc5: x^5+x^4+x^2+x^1+1
+crc6: x^6+x^5+x^3+x^2+x+1
+crc7: x^7+x^6+x^2+1
+crc8: x^8+x^2+x+x+1
+*/
+
+/* default SBR CRC */  /* 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_START                 0x0000
+#define SBR_CRC_RANGE                 0x03FF
+
+typedef struct
+{
+  USHORT crcState;
+  USHORT crcMask;
+  USHORT crcPoly;
+}
+CRC_BUFFER;
+
+typedef CRC_BUFFER *HANDLE_CRC;
+
+int SbrCrcCheck (HANDLE_FDK_BITSTREAM hBitBuf,
+                 LONG                 NrCrcBits);
+
+
+#endif
diff --git a/libSBRdec/src/sbr_deb.cpp b/libSBRdec/src/sbr_deb.cpp
new file mode 100644
index 0000000..aa37ffe
--- /dev/null
+++ b/libSBRdec/src/sbr_deb.cpp
@@ -0,0 +1,90 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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  Print selected debug messages   
+*/
+
+#include "sbr_deb.h"
+
diff --git a/libSBRdec/src/sbr_deb.h b/libSBRdec/src/sbr_deb.h
new file mode 100644
index 0000000..324dea9
--- /dev/null
+++ b/libSBRdec/src/sbr_deb.h
@@ -0,0 +1,94 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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  Debugging aids   
+*/
+
+#ifndef __SBR_DEB_H
+#define __SBR_DEB_H
+
+#include "sbrdecoder.h"
+
+#endif
diff --git a/libSBRdec/src/sbr_dec.cpp b/libSBRdec/src/sbr_dec.cpp
new file mode 100644
index 0000000..208120c
--- /dev/null
+++ b/libSBRdec/src/sbr_dec.cpp
@@ -0,0 +1,1046 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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 decoder  
+  This module provides the actual decoder implementation. The SBR data (side information) is already
+  decoded. Only three functions are provided:
+
+  \li 1.) createSbrDec(): One time initialization
+  \li 2.) resetSbrDec(): Called by sbr_Apply() when the information contained in an SBR_HEADER_ELEMENT requires a reset
+  and recalculation of important SBR structures.
+  \li 3.) sbr_dec(): The actual decoder. Calls the different tools such as filterbanks, lppTransposer(), and calculateSbrEnvelope()
+  [the envelope adjuster].
+
+  \sa sbr_dec(), \ref documentationOverview
+*/
+
+#include "sbr_dec.h"
+
+#include "sbr_ram.h"
+#include "env_extr.h"
+#include "env_calc.h"
+#include "scale.h"
+
+#include "genericStds.h"
+
+#include "sbrdec_drc.h"
+
+
+
+static void assignLcTimeSlots( HANDLE_SBR_DEC hSbrDec,                     /*!< handle to Decoder channel */
+                               FIXP_DBL  **QmfBufferReal,
+                               int noCols )
+{
+  int slot, i;
+  FIXP_DBL  *ptr;
+
+  /* Number of QMF timeslots in the overlap buffer: */
+  ptr = hSbrDec->pSbrOverlapBuffer;
+  for(slot=0; slot<hSbrDec->LppTrans.pSettings->overlap; slot++) {
+    QmfBufferReal[slot] = ptr; ptr += (64);
+  }
+
+  /* Assign timeslots to Workbuffer1 */
+  ptr = hSbrDec->WorkBuffer1;
+  for(i=0; i<noCols; i++) {
+    QmfBufferReal[slot] = ptr; ptr += (64);
+    slot++;
+  }
+}
+
+
+static void assignHqTimeSlots( HANDLE_SBR_DEC hSbrDec,                     /*!< handle to Decoder channel */
+                               FIXP_DBL  **QmfBufferReal,
+                               FIXP_DBL  **QmfBufferImag,
+                               int noCols )
+{
+  FIXP_DBL  *ptr;
+  int slot;
+
+  /* Number of QMF timeslots in one half of a frame (size of Workbuffer1 or 2): */
+  int halflen = (noCols >> 1) + hSbrDec->LppTrans.pSettings->overlap;
+  int totCols = noCols + hSbrDec->LppTrans.pSettings->overlap;
+
+  /* Number of QMF timeslots in the overlap buffer: */
+  ptr = hSbrDec->pSbrOverlapBuffer;
+  for(slot=0; slot<hSbrDec->LppTrans.pSettings->overlap; slot++) {
+    QmfBufferReal[slot] = ptr; ptr += (64);
+    QmfBufferImag[slot] = ptr; ptr += (64);
+  }
+
+  /* Assign first half of timeslots to Workbuffer1 */
+  ptr = hSbrDec->WorkBuffer1;
+  for(; slot<halflen; slot++) {
+    QmfBufferReal[slot] = ptr; ptr += (64);
+    QmfBufferImag[slot] = ptr; ptr += (64);
+  }
+
+  /* Assign second half of timeslots to Workbuffer2 */
+  ptr = hSbrDec->WorkBuffer2;
+  for(; slot<totCols; slot++) {
+    QmfBufferReal[slot] = ptr; ptr += (64);
+    QmfBufferImag[slot] = ptr; ptr += (64);
+  }
+}
+
+
+static void assignTimeSlots( HANDLE_SBR_DEC hSbrDec,                     /*!< handle to Decoder channel */
+                             int noCols,
+                             int useLP )
+{
+ /* assign qmf time slots */
+  hSbrDec->useLP = useLP;
+  if (useLP) {
+    hSbrDec->SynthesisQMF.flags |= QMF_FLAG_LP;
+    hSbrDec->AnalysiscQMF.flags |= QMF_FLAG_LP;
+  } else {
+    hSbrDec->SynthesisQMF.flags &= ~QMF_FLAG_LP;
+    hSbrDec->AnalysiscQMF.flags &= ~QMF_FLAG_LP;
+  }
+  if (!useLP)
+    assignHqTimeSlots( hSbrDec, hSbrDec->QmfBufferReal, hSbrDec->QmfBufferImag, noCols );
+  else
+  {
+    assignLcTimeSlots( hSbrDec, hSbrDec->QmfBufferReal, noCols );
+  }
+}
+
+static void changeQmfType( HANDLE_SBR_DEC hSbrDec,                     /*!< handle to Decoder channel */
+                           int useLdTimeAlign )
+{
+  UINT synQmfFlags = hSbrDec->SynthesisQMF.flags;
+  UINT anaQmfFlags = hSbrDec->AnalysiscQMF.flags;
+  int  resetSynQmf = 0;
+  int  resetAnaQmf = 0;
+
+  /* assign qmf type */
+  if (useLdTimeAlign) {
+    if (synQmfFlags & QMF_FLAG_CLDFB) {
+      /* change the type to MPSLD */
+      synQmfFlags &= ~QMF_FLAG_CLDFB;
+      synQmfFlags |=  QMF_FLAG_MPSLDFB;
+      resetSynQmf = 1;
+    }
+    if (anaQmfFlags & QMF_FLAG_CLDFB) {
+      /* change the type to MPSLD */
+      anaQmfFlags &= ~QMF_FLAG_CLDFB;
+      anaQmfFlags |=  QMF_FLAG_MPSLDFB;
+      resetAnaQmf = 1;
+    }
+  } else {
+    if (synQmfFlags & QMF_FLAG_MPSLDFB) {
+      /* change the type to CLDFB */
+      synQmfFlags &= ~QMF_FLAG_MPSLDFB;
+      synQmfFlags |=  QMF_FLAG_CLDFB;
+      resetSynQmf = 1;
+    }
+    if (anaQmfFlags & QMF_FLAG_MPSLDFB) {
+      /* change the type to CLDFB */
+      anaQmfFlags &= ~QMF_FLAG_MPSLDFB;
+      anaQmfFlags |=  QMF_FLAG_CLDFB;
+      resetAnaQmf = 1;
+    }
+  }
+
+  if (resetAnaQmf) {
+    int qmfErr = qmfInitAnalysisFilterBank (
+           &hSbrDec->AnalysiscQMF,
+            hSbrDec->anaQmfStates,
+            hSbrDec->AnalysiscQMF.no_col,
+            hSbrDec->AnalysiscQMF.lsb,
+            hSbrDec->AnalysiscQMF.usb,
+            hSbrDec->AnalysiscQMF.no_channels,
+            anaQmfFlags | QMF_FLAG_KEEP_STATES
+            );
+    if (qmfErr != 0) {
+      FDK_ASSERT(0);
+    }
+  }
+
+  if (resetSynQmf) {
+    int qmfErr = qmfInitSynthesisFilterBank (
+           &hSbrDec->SynthesisQMF,
+            hSbrDec->pSynQmfStates,
+            hSbrDec->SynthesisQMF.no_col,
+            hSbrDec->SynthesisQMF.lsb,
+            hSbrDec->SynthesisQMF.usb,
+            hSbrDec->SynthesisQMF.no_channels,
+            synQmfFlags | QMF_FLAG_KEEP_STATES
+            );
+
+    if (qmfErr != 0) {
+      FDK_ASSERT(0);
+    }
+  }
+}
+
+
+/*!
+  \brief      SBR decoder core function for one channel
+
+  \image html  BufferMgmtDetailed-1632.png
+
+  Besides the filter states of the QMF filter bank and the LPC-states of
+  the LPP-Transposer, processing is mainly based on four buffers:
+  #timeIn, #timeOut, #WorkBuffer2 and #OverlapBuffer. The #WorkBuffer2
+  is reused for all channels and might be used by the core decoder, a
+  static overlap buffer is required for each channel. Du to in-place
+  processing, #timeIn and #timeOut point to identical locations.
+
+  The spectral data is organized in so-called slots, each slot
+  containing 64 bands of complex data. The number of slots per frame is
+  dependend on the frame size. For mp3PRO, there are 18 slots per frame
+  and 6 slots per #OverlapBuffer. It is not necessary to have the slots
+  in located consecutive address ranges.
+
+  To optimize memory usage and to minimize the number of memory
+  accesses, the memory management is organized as follows (Slot numbers
+  based on mp3PRO):
+
+  1.) Input time domain signal is located in #timeIn, the last slots
+  (0..5) of the spectral data of the previous frame are located in the
+  #OverlapBuffer. In addition, #frameData of the current frame resides
+  in the upper part of #timeIn.
+
+  2.) During the cplxAnalysisQmfFiltering(), 32 samples from #timeIn are transformed
+  into a slot of up to 32 complex spectral low band values at a
+  time. The first spectral slot -- nr. 6 -- is written at slot number
+  zero of #WorkBuffer2. #WorkBuffer2 will be completely filled with
+  spectral data.
+
+  3.) LPP-Transposition in lppTransposer() is processed on 24 slots. During the
+  transposition, the high band part of the spectral data is replicated
+  based on the low band data.
+
+  Envelope Adjustment is processed on the high band part of the spectral
+  data only by calculateSbrEnvelope().
+
+  4.) The cplxSynthesisQmfFiltering() creates 64 time domain samples out
+  of a slot of 64 complex spectral values at a time. The first 6 slots
+  in #timeOut are filled from the results of spectral slots 0..5 in the
+  #OverlapBuffer. The consecutive slots in timeOut are now filled with
+  the results of spectral slots 6..17.
+
+  5.) The preprocessed slots 18..23 have to be stored in the
+  #OverlapBuffer.
+
+*/
+
+void
+sbr_dec ( HANDLE_SBR_DEC hSbrDec,            /*!< handle to Decoder channel */
+          INT_PCM *timeIn,                   /*!< pointer to input time signal */
+          INT_PCM *timeOut,                  /*!< pointer to output time signal */
+          HANDLE_SBR_DEC hSbrDecRight,       /*!< handle to Decoder channel right */
+          INT_PCM *timeOutRight,             /*!< pointer to output time signal */
+          const int strideIn,                /*!< Time data traversal strideIn */
+          const int strideOut,               /*!< Time data traversal strideOut */
+          HANDLE_SBR_HEADER_DATA hHeaderData,/*!< Static control data */
+          HANDLE_SBR_FRAME_DATA hFrameData,  /*!< Control data of current frame */
+          HANDLE_SBR_PREV_FRAME_DATA hPrevFrameData,  /*!< Some control data of last frame */
+          const int applyProcessing,         /*!< Flag for SBR operation */
+          HANDLE_PS_DEC h_ps_d,
+          const UINT flags
+         )
+{
+  int i, slot, reserve;
+  int saveLbScale;
+  int ov_len;
+  int lastSlotOffs;
+  FIXP_DBL maxVal;
+
+  /* 1+1/3 frames of spectral data: */
+  FIXP_DBL **QmfBufferReal = hSbrDec->QmfBufferReal;
+  FIXP_DBL **QmfBufferImag = hSbrDec->QmfBufferImag;
+
+ /* Number of QMF timeslots in the overlap buffer: */
+ ov_len = hSbrDec->LppTrans.pSettings->overlap;
+
+ /* Number of QMF slots per frame */
+  int noCols = hHeaderData->numberTimeSlots * hHeaderData->timeStep;
+
+ /* assign qmf time slots */
+  if ( ((flags & SBRDEC_LOW_POWER ) ? 1 : 0) != ((hSbrDec->SynthesisQMF.flags & QMF_FLAG_LP) ? 1 : 0) ) {
+    assignTimeSlots( hSbrDec, hHeaderData->numberTimeSlots * hHeaderData->timeStep, flags & SBRDEC_LOW_POWER);
+  }
+
+  if (flags & SBRDEC_ELD_GRID) {
+    /* Choose the right low delay filter bank */
+    changeQmfType( hSbrDec, (flags & SBRDEC_LD_MPS_QMF) ? 1 : 0 );
+  }
+
+  /*
+    low band codec signal subband filtering
+   */
+
+  {
+    C_AALLOC_SCRATCH_START(qmfTemp, FIXP_DBL, 2*(64));
+
+    qmfAnalysisFiltering( &hSbrDec->AnalysiscQMF,
+                           QmfBufferReal + ov_len,
+                           QmfBufferImag + ov_len,
+                          &hSbrDec->sbrScaleFactor,
+                           timeIn,
+                           strideIn,
+                           qmfTemp
+                         );
+
+    C_AALLOC_SCRATCH_END(qmfTemp, FIXP_DBL, 2*(64));
+  }
+
+  /*
+    Clear upper half of spectrum
+  */
+  {
+    int nAnalysisBands = hHeaderData->numberOfAnalysisBands;
+
+    if (! (flags & SBRDEC_LOW_POWER)) {
+      for (slot = ov_len; slot < noCols+ov_len; slot++) {
+        FDKmemclear(&QmfBufferReal[slot][nAnalysisBands],((64)-nAnalysisBands)*sizeof(FIXP_DBL));
+        FDKmemclear(&QmfBufferImag[slot][nAnalysisBands],((64)-nAnalysisBands)*sizeof(FIXP_DBL));
+      }
+    } else
+    for (slot = ov_len; slot < noCols+ov_len; slot++) {
+      FDKmemclear(&QmfBufferReal[slot][nAnalysisBands],((64)-nAnalysisBands)*sizeof(FIXP_DBL));
+    }
+  }
+
+
+
+  /*
+    Shift spectral data left to gain accuracy in transposer and adjustor
+  */
+  maxVal = maxSubbandSample( QmfBufferReal,
+                            (flags & SBRDEC_LOW_POWER) ? NULL : QmfBufferImag,
+                             0,
+                             hSbrDec->AnalysiscQMF.lsb,
+                             ov_len,
+                             noCols+ov_len );
+
+  reserve = fixMax(0,CntLeadingZeros(maxVal)-1) ;
+  reserve = fixMin(reserve,DFRACT_BITS-1-hSbrDec->sbrScaleFactor.lb_scale);
+
+  /* If all data is zero, lb_scale could become too large */
+  rescaleSubbandSamples( QmfBufferReal,
+                         (flags & SBRDEC_LOW_POWER) ? NULL : QmfBufferImag,
+                         0,
+                         hSbrDec->AnalysiscQMF.lsb,
+                         ov_len,
+                         noCols+ov_len,
+                         reserve);
+
+  hSbrDec->sbrScaleFactor.lb_scale += reserve;
+
+  /*
+    save low band scale, wavecoding or parametric stereo may modify it
+  */
+  saveLbScale = hSbrDec->sbrScaleFactor.lb_scale;
+
+
+  if (applyProcessing)
+  {
+    UCHAR * borders = hFrameData->frameInfo.borders;
+    lastSlotOffs =  borders[hFrameData->frameInfo.nEnvelopes] - hHeaderData->numberTimeSlots;
+
+    FIXP_DBL degreeAlias[(64)];
+
+    /* The transposer will override most values in degreeAlias[].
+       The array needs to be cleared at least from lowSubband to highSubband before. */
+    if (flags & SBRDEC_LOW_POWER)
+      FDKmemclear(&degreeAlias[hHeaderData->freqBandData.lowSubband], (hHeaderData->freqBandData.highSubband-hHeaderData->freqBandData.lowSubband)*sizeof(FIXP_DBL));
+
+    /*
+      Inverse filtering of lowband and transposition into the SBR-frequency range
+    */
+
+    lppTransposer ( &hSbrDec->LppTrans,
+                    &hSbrDec->sbrScaleFactor,
+                    QmfBufferReal,
+                    degreeAlias,                  // only used if useLP = 1
+                    QmfBufferImag,
+                    flags & SBRDEC_LOW_POWER,
+                    hHeaderData->timeStep,
+                    borders[0],
+                    lastSlotOffs,
+                    hHeaderData->freqBandData.nInvfBands,
+                    hFrameData->sbr_invf_mode,
+                    hPrevFrameData->sbr_invf_mode );
+
+
+
+
+
+    /*
+      Adjust envelope of current frame.
+    */
+
+    calculateSbrEnvelope (&hSbrDec->sbrScaleFactor,
+                          &hSbrDec->SbrCalculateEnvelope,
+                          hHeaderData,
+                          hFrameData,
+                          QmfBufferReal,
+                          QmfBufferImag,
+                          flags & SBRDEC_LOW_POWER,
+
+                          degreeAlias,
+                          flags,
+                          (hHeaderData->frameErrorFlag || hPrevFrameData->frameErrorFlag));
+
+
+    /*
+      Update hPrevFrameData (to be used in the next frame)
+    */
+    for (i=0; i<hHeaderData->freqBandData.nInvfBands; i++) {
+      hPrevFrameData->sbr_invf_mode[i] = hFrameData->sbr_invf_mode[i];
+    }
+    hPrevFrameData->coupling = hFrameData->coupling;
+    hPrevFrameData->stopPos = borders[hFrameData->frameInfo.nEnvelopes];
+    hPrevFrameData->ampRes = hFrameData->ampResolutionCurrentFrame;
+  }
+  else {
+    /* Reset hb_scale if no highband is present, because hb_scale is considered in the QMF-synthesis */
+    hSbrDec->sbrScaleFactor.hb_scale = saveLbScale;
+  }
+
+
+  for (i=0; i<LPC_ORDER; i++){
+    /*
+      Store the unmodified qmf Slots values (required for LPC filtering)
+    */
+    if (! (flags & SBRDEC_LOW_POWER)) {
+      FDKmemcpy(hSbrDec->LppTrans.lpcFilterStatesReal[i], QmfBufferReal[noCols-LPC_ORDER+i], hSbrDec->AnalysiscQMF.lsb*sizeof(FIXP_DBL));
+      FDKmemcpy(hSbrDec->LppTrans.lpcFilterStatesImag[i], QmfBufferImag[noCols-LPC_ORDER+i], hSbrDec->AnalysiscQMF.lsb*sizeof(FIXP_DBL));
+    } else
+    FDKmemcpy(hSbrDec->LppTrans.lpcFilterStatesReal[i], QmfBufferReal[noCols-LPC_ORDER+i], hSbrDec->AnalysiscQMF.lsb*sizeof(FIXP_DBL));
+  }
+
+  /*
+    Synthesis subband filtering.
+  */
+
+  if ( ! (flags & SBRDEC_PS_DECODED) ) {
+
+    {
+      int outScalefactor = 0;
+
+      if (h_ps_d != NULL) {
+        h_ps_d->procFrameBased = 1;  /* we here do frame based processing */
+      }
+
+
+      sbrDecoder_drcApply(&hSbrDec->sbrDrcChannel,
+                           QmfBufferReal,
+                           (flags & SBRDEC_LOW_POWER) ? NULL : QmfBufferImag,
+                           hSbrDec->SynthesisQMF.no_col,
+                          &outScalefactor
+                          );
+
+
+
+      qmfChangeOutScalefactor(&hSbrDec->SynthesisQMF, outScalefactor );
+
+      {
+        C_AALLOC_SCRATCH_START(qmfTemp, FIXP_DBL, 2*(64));
+
+        qmfSynthesisFiltering( &hSbrDec->SynthesisQMF,
+                                QmfBufferReal,
+                                (flags & SBRDEC_LOW_POWER) ? NULL : QmfBufferImag,
+                               &hSbrDec->sbrScaleFactor,
+                                hSbrDec->LppTrans.pSettings->overlap,
+                                timeOut,
+                                strideOut,
+                                qmfTemp);
+
+        C_AALLOC_SCRATCH_END(qmfTemp, FIXP_DBL, 2*(64));
+      }
+
+    }
+
+  } else { /* (flags & SBRDEC_PS_DECODED) */
+    INT i, sdiff, outScalefactor, scaleFactorLowBand, scaleFactorHighBand;
+    SCHAR scaleFactorLowBand_ov, scaleFactorLowBand_no_ov;
+
+    HANDLE_QMF_FILTER_BANK synQmf      = &hSbrDec->SynthesisQMF;
+    HANDLE_QMF_FILTER_BANK synQmfRight = &hSbrDecRight->SynthesisQMF;
+
+    /* adapt scaling */
+    sdiff = hSbrDec->sbrScaleFactor.lb_scale - reserve;                  /* Scaling difference         */
+    scaleFactorHighBand   = sdiff - hSbrDec->sbrScaleFactor.hb_scale;    /* Scale of current high band */
+    scaleFactorLowBand_ov = sdiff - hSbrDec->sbrScaleFactor.ov_lb_scale; /* Scale of low band overlapping QMF data */
+    scaleFactorLowBand_no_ov = sdiff - hSbrDec->sbrScaleFactor.lb_scale; /* Scale of low band current QMF data     */
+    outScalefactor  = 0;                                                 /* Initial output scale */
+
+    if (h_ps_d->procFrameBased == 1)    /* If we have switched from frame to slot based processing copy filter states */
+    { /* procFrameBased will be unset later */
+      /* copy filter states from left to right */
+      FDKmemcpy(synQmfRight->FilterStates, synQmf->FilterStates, ((640)-(64))*sizeof(FIXP_QSS));
+    }
+
+    /* scale ALL qmf vales ( real and imag ) of mono / left channel to the
+       same scale factor ( ov_lb_sf, lb_sf and hq_sf )                      */
+    scalFilterBankValues( h_ps_d,                             /* parametric stereo decoder handle     */
+                          QmfBufferReal,                      /* qmf filterbank values                */
+                          QmfBufferImag,                      /* qmf filterbank values                */
+                          synQmf->lsb,                        /* sbr start subband                    */
+                          hSbrDec->sbrScaleFactor.ov_lb_scale,
+                          hSbrDec->sbrScaleFactor.lb_scale,
+                         &scaleFactorLowBand_ov,              /* adapt scaling values */
+                         &scaleFactorLowBand_no_ov,           /* adapt scaling values */
+                          hSbrDec->sbrScaleFactor.hb_scale,   /* current frame ( highband ) */
+                         &scaleFactorHighBand,
+                          synQmf->no_col);
+
+    /* use the same synthese qmf values for left and right channel */
+    synQmfRight->no_col = synQmf->no_col;
+    synQmfRight->lsb    = synQmf->lsb;
+    synQmfRight->usb    = synQmf->usb;
+
+    int env=0;
+
+      outScalefactor += (SCAL_HEADROOM+1); /* psDiffScale! */
+
+    {
+      C_ALLOC_SCRATCH_START(pWorkBuffer, FIXP_DBL, 2*(64));
+
+      int maxShift = 0;
+
+      if (hSbrDec->sbrDrcChannel.prevFact_exp > maxShift) {
+        maxShift = hSbrDec->sbrDrcChannel.prevFact_exp;
+      }
+      if (hSbrDec->sbrDrcChannel.currFact_exp > maxShift) {
+        maxShift = hSbrDec->sbrDrcChannel.currFact_exp;
+      }
+      if (hSbrDec->sbrDrcChannel.nextFact_exp > maxShift) {
+        maxShift = hSbrDec->sbrDrcChannel.nextFact_exp;
+      }
+
+      /* copy DRC data to right channel (with PS both channels use the same DRC gains) */
+      FDKmemcpy(&hSbrDecRight->sbrDrcChannel, &hSbrDec->sbrDrcChannel, sizeof(SBRDEC_DRC_CHANNEL));
+
+      for (i = 0; i < synQmf->no_col; i++) {  /* ----- no_col loop ----- */
+
+        INT outScalefactorR, outScalefactorL;
+        outScalefactorR = outScalefactorL = outScalefactor;
+
+        /* qmf timeslot of right channel */
+        FIXP_DBL* rQmfReal = pWorkBuffer;
+        FIXP_DBL* rQmfImag = pWorkBuffer + 64;
+
+
+        {
+          if ( i == h_ps_d->bsData[h_ps_d->processSlot].mpeg.aEnvStartStop[env] ) {
+            initSlotBasedRotation( h_ps_d, env, hHeaderData->freqBandData.highSubband );
+            env++;
+          }
+
+          ApplyPsSlot( h_ps_d,                   /* parametric stereo decoder handle  */
+                      (QmfBufferReal + i),       /* one timeslot of left/mono channel */
+                      (QmfBufferImag + i),       /* one timeslot of left/mono channel */
+                       rQmfReal,                 /* one timeslot or right channel     */
+                       rQmfImag);                /* one timeslot or right channel     */
+        }
+
+
+        scaleFactorLowBand = (i<(6)) ? scaleFactorLowBand_ov : scaleFactorLowBand_no_ov;
+
+
+        sbrDecoder_drcApplySlot ( /* right channel */
+                                 &hSbrDecRight->sbrDrcChannel,
+                                  rQmfReal,
+                                  rQmfImag,
+                                  i,
+                                  synQmfRight->no_col,
+                                  maxShift
+                                );
+
+        outScalefactorR += maxShift;
+
+        sbrDecoder_drcApplySlot ( /* left channel */
+                                 &hSbrDec->sbrDrcChannel,
+                                 *(QmfBufferReal + i),
+                                 *(QmfBufferImag + i),
+                                  i,
+                                  synQmf->no_col,
+                                  maxShift
+                                );
+
+        outScalefactorL += maxShift;
+
+
+        /* scale filter states for left and right channel */
+        qmfChangeOutScalefactor( synQmf, outScalefactorL );
+        qmfChangeOutScalefactor( synQmfRight, outScalefactorR );
+
+        {
+
+          qmfSynthesisFilteringSlot( synQmfRight,
+                                     rQmfReal,                /* QMF real buffer */
+                                     rQmfImag,                /* QMF imag buffer */
+                                     scaleFactorLowBand,
+                                     scaleFactorHighBand,
+                                     timeOutRight+(i*synQmf->no_channels*strideOut),
+                                     strideOut,
+                                     pWorkBuffer);
+
+          qmfSynthesisFilteringSlot( synQmf,
+                                   *(QmfBufferReal + i),      /* QMF real buffer */
+                                   *(QmfBufferImag + i),      /* QMF imag buffer */
+                                     scaleFactorLowBand,
+                                     scaleFactorHighBand,
+                                     timeOut+(i*synQmf->no_channels*strideOut),
+                                     strideOut,
+                                     pWorkBuffer);
+
+        }
+      } /* no_col loop  i  */
+
+      /* scale back (6) timeslots look ahead for hybrid filterbank to original value */
+      rescalFilterBankValues( h_ps_d,
+                              QmfBufferReal,
+                              QmfBufferImag,
+                              synQmf->lsb,
+                              synQmf->no_col );
+
+      C_ALLOC_SCRATCH_END(pWorkBuffer, FIXP_DBL, 2*(64));
+    }
+  }
+
+  sbrDecoder_drcUpdateChannel( &hSbrDec->sbrDrcChannel );
+
+
+  /*
+    Update overlap buffer
+    Even bands above usb are copied to avoid outdated spectral data in case
+    the stop frequency raises.
+  */
+
+  if (hSbrDec->LppTrans.pSettings->overlap > 0)
+  {
+    if (! (flags & SBRDEC_LOW_POWER)) {
+      for ( i=0; i<hSbrDec->LppTrans.pSettings->overlap; i++ ) {
+        FDKmemcpy(QmfBufferReal[i], QmfBufferReal[i+noCols], (64)*sizeof(FIXP_DBL));
+        FDKmemcpy(QmfBufferImag[i], QmfBufferImag[i+noCols], (64)*sizeof(FIXP_DBL));
+      }
+    } else
+      for ( i=0; i<hSbrDec->LppTrans.pSettings->overlap; i++ ) {
+        FDKmemcpy(QmfBufferReal[i], QmfBufferReal[i+noCols], (64)*sizeof(FIXP_DBL));
+      }
+  }
+
+  hSbrDec->sbrScaleFactor.ov_lb_scale = saveLbScale;
+
+  /* Save current frame status */
+  hPrevFrameData->frameErrorFlag = hHeaderData->frameErrorFlag;
+
+} // sbr_dec()
+
+
+/*!
+  \brief     Creates sbr decoder structure
+  \return    errorCode, 0 if successful
+*/
+SBR_ERROR
+createSbrDec (SBR_CHANNEL * hSbrChannel,
+              HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */
+              TRANSPOSER_SETTINGS *pSettings,
+              const int     downsampleFac,        /*!< Downsampling factor */
+              const UINT    qmfFlags,             /*!< flags -> 1: HQ/LP selector, 2: CLDFB */
+              const UINT    flags,
+              const int     overlap, 
+              int           chan)                 /*!< Channel for which to assign buffers etc. */
+
+{
+  SBR_ERROR err = SBRDEC_OK;
+  int timeSlots = hHeaderData->numberTimeSlots;   /* Number of SBR slots per frame */
+  int noCols = timeSlots * hHeaderData->timeStep; /* Number of QMF slots per frame */
+  HANDLE_SBR_DEC hs = &(hSbrChannel->SbrDec);
+
+  /* Initialize scale factors */
+  hs->sbrScaleFactor.ov_lb_scale  = 0;
+  hs->sbrScaleFactor.ov_hb_scale  = 0;
+  hs->sbrScaleFactor.hb_scale     = 0;
+
+
+  /*
+    create envelope calculator
+  */
+  err = createSbrEnvelopeCalc (&hs->SbrCalculateEnvelope,
+                               hHeaderData,
+                               chan,
+                               flags);
+  if (err != SBRDEC_OK) {
+    return err;
+  }
+
+  /*
+    create QMF filter banks
+  */
+  {
+    int qmfErr;
+
+    qmfErr = qmfInitAnalysisFilterBank (
+                    &hs->AnalysiscQMF,
+                     hs->anaQmfStates,
+                     noCols,
+                     hHeaderData->freqBandData.lowSubband,
+                     hHeaderData->freqBandData.highSubband,
+                     hHeaderData->numberOfAnalysisBands,
+                     qmfFlags & (~QMF_FLAG_KEEP_STATES)
+                     );
+    if (qmfErr != 0) {
+      return SBRDEC_UNSUPPORTED_CONFIG;
+    }
+  }
+  if (hs->pSynQmfStates == NULL) {
+    hs->pSynQmfStates = GetRam_sbr_QmfStatesSynthesis(chan);
+    if (hs->pSynQmfStates == NULL)
+      return SBRDEC_MEM_ALLOC_FAILED;
+  }
+
+  {
+    int qmfErr;
+
+    qmfErr = qmfInitSynthesisFilterBank (
+           &hs->SynthesisQMF,
+            hs->pSynQmfStates,
+            noCols,
+            hHeaderData->freqBandData.lowSubband,
+            hHeaderData->freqBandData.highSubband,
+            (64) / downsampleFac,
+            qmfFlags & (~QMF_FLAG_KEEP_STATES)
+            );
+
+    if (qmfErr != 0) {
+      return SBRDEC_UNSUPPORTED_CONFIG;
+    }
+  }
+  initSbrPrevFrameData (&hSbrChannel->prevFrameData, timeSlots);
+
+  /*
+    create transposer
+  */
+  err = createLppTransposer (&hs->LppTrans,
+                             pSettings,
+                             hHeaderData->freqBandData.lowSubband,
+                             hHeaderData->freqBandData.v_k_master,
+                             hHeaderData->freqBandData.numMaster,
+                             hs->SynthesisQMF.usb,
+                             timeSlots,
+                             hs->AnalysiscQMF.no_col,
+                             hHeaderData->freqBandData.freqBandTableNoise,
+                             hHeaderData->freqBandData.nNfb,
+                             hHeaderData->sbrProcSmplRate,
+                             chan,
+                             overlap );
+  if (err != SBRDEC_OK) {
+    return err;
+  }
+
+  /* The CLDFB does not have overlap */
+  if ((qmfFlags & QMF_FLAG_CLDFB) == 0) {
+    if (hs->pSbrOverlapBuffer == NULL) {
+      hs->pSbrOverlapBuffer = GetRam_sbr_OverlapBuffer(chan);
+      if (hs->pSbrOverlapBuffer == NULL)  {
+        return SBRDEC_MEM_ALLOC_FAILED;
+      }
+    } else {
+      /* Clear overlap buffer */
+      FDKmemclear( hs->pSbrOverlapBuffer,
+                   sizeof(FIXP_DBL) * 2 * (6) * (64)
+                 );
+    }
+  }
+
+  /* assign qmf time slots */
+  assignTimeSlots( &hSbrChannel->SbrDec, hHeaderData->numberTimeSlots * hHeaderData->timeStep, qmfFlags & QMF_FLAG_LP);
+
+  return err;
+}
+
+/*!
+  \brief     Delete sbr decoder structure
+  \return    errorCode, 0 if successful
+*/
+int
+deleteSbrDec (SBR_CHANNEL * hSbrChannel)
+{
+  HANDLE_SBR_DEC hs = &hSbrChannel->SbrDec;
+
+  deleteSbrEnvelopeCalc (&hs->SbrCalculateEnvelope);
+
+  /* delete QMF filter states */
+  if (hs->pSynQmfStates != NULL) {
+    FreeRam_sbr_QmfStatesSynthesis(&hs->pSynQmfStates);
+  }
+
+
+  if (hs->pSbrOverlapBuffer != NULL) {
+    FreeRam_sbr_OverlapBuffer(&hs->pSbrOverlapBuffer);
+  }
+
+  return 0;
+}
+
+
+/*!
+  \brief     resets sbr decoder structure
+  \return    errorCode, 0 if successful
+*/
+SBR_ERROR
+resetSbrDec (HANDLE_SBR_DEC hSbrDec,
+             HANDLE_SBR_HEADER_DATA hHeaderData,
+             HANDLE_SBR_PREV_FRAME_DATA hPrevFrameData,
+             const int useLP,
+             const int downsampleFac
+             )
+{
+  SBR_ERROR sbrError = SBRDEC_OK;
+  
+  int old_lsb = hSbrDec->SynthesisQMF.lsb;
+  int new_lsb = hHeaderData->freqBandData.lowSubband;
+  int l, startBand, stopBand, startSlot, size;
+
+  int source_scale, target_scale, delta_scale, target_lsb, target_usb, reserve;
+  FIXP_DBL maxVal;
+
+  /* overlapBuffer point to first (6) slots */
+  FIXP_DBL  **OverlapBufferReal = hSbrDec->QmfBufferReal;
+  FIXP_DBL  **OverlapBufferImag = hSbrDec->QmfBufferImag;
+
+  /* assign qmf time slots */
+  assignTimeSlots( hSbrDec, hHeaderData->numberTimeSlots * hHeaderData->timeStep, useLP);
+
+
+
+  resetSbrEnvelopeCalc (&hSbrDec->SbrCalculateEnvelope);
+
+  hSbrDec->SynthesisQMF.lsb = hHeaderData->freqBandData.lowSubband;
+  hSbrDec->SynthesisQMF.usb = fixMin((INT)hSbrDec->SynthesisQMF.no_channels, (INT)hHeaderData->freqBandData.highSubband);
+
+  hSbrDec->AnalysiscQMF.lsb = hSbrDec->SynthesisQMF.lsb;
+  hSbrDec->AnalysiscQMF.usb = hSbrDec->SynthesisQMF.usb;
+
+
+  /*
+    The following initialization of spectral data in the overlap buffer
+    is required for dynamic x-over or a change of the start-freq for 2 reasons:
+
+    1. If the lowband gets _wider_, unadjusted data would remain
+
+    2. If the lowband becomes _smaller_, the highest bands of the old lowband
+       must be cleared because the whitening would be affected
+  */
+  startBand = old_lsb;
+  stopBand  = new_lsb;
+  startSlot = hHeaderData->timeStep * (hPrevFrameData->stopPos - hHeaderData->numberTimeSlots);
+  size      = fixMax(0,stopBand-startBand);
+
+  /* keep already adjusted data in the x-over-area */
+  if (!useLP) {
+    for (l=startSlot; l<hSbrDec->LppTrans.pSettings->overlap; l++) {
+      FDKmemclear(&OverlapBufferReal[l][startBand], size*sizeof(FIXP_DBL));
+      FDKmemclear(&OverlapBufferImag[l][startBand], size*sizeof(FIXP_DBL));
+    }
+  } else
+  for (l=startSlot; l<hSbrDec->LppTrans.pSettings->overlap ; l++) {
+    FDKmemclear(&OverlapBufferReal[l][startBand], size*sizeof(FIXP_DBL));
+  }
+
+
+  /*
+    reset LPC filter states
+  */
+  startBand = fixMin(old_lsb,new_lsb);
+  stopBand  = fixMax(old_lsb,new_lsb);
+  size      = fixMax(0,stopBand-startBand);
+
+  FDKmemclear(&hSbrDec->LppTrans.lpcFilterStatesReal[0][startBand], size*sizeof(FIXP_DBL));
+  FDKmemclear(&hSbrDec->LppTrans.lpcFilterStatesReal[1][startBand], size*sizeof(FIXP_DBL));
+  if (!useLP) {
+    FDKmemclear(&hSbrDec->LppTrans.lpcFilterStatesImag[0][startBand], size*sizeof(FIXP_DBL));
+    FDKmemclear(&hSbrDec->LppTrans.lpcFilterStatesImag[1][startBand], size*sizeof(FIXP_DBL));
+  }
+
+
+  /*
+    Rescale already processed spectral data between old and new x-over frequency.
+    This must be done because of the separate scalefactors for lowband and highband.
+  */
+  startBand = fixMin(old_lsb,new_lsb);
+  stopBand =  fixMax(old_lsb,new_lsb);
+
+  if (new_lsb > old_lsb) {
+    /* The x-over-area was part of the highband before and will now belong to the lowband */
+    source_scale = hSbrDec->sbrScaleFactor.ov_hb_scale;
+    target_scale = hSbrDec->sbrScaleFactor.ov_lb_scale;
+    target_lsb   = 0;
+    target_usb   = old_lsb;
+  }
+  else {
+    /* The x-over-area was part of the lowband before and will now belong to the highband */
+    source_scale = hSbrDec->sbrScaleFactor.ov_lb_scale;
+    target_scale = hSbrDec->sbrScaleFactor.ov_hb_scale;
+    /* jdr: The values old_lsb and old_usb might be wrong because the previous frame might have been "upsamling". */
+    target_lsb   = hSbrDec->SynthesisQMF.lsb;
+    target_usb   = hSbrDec->SynthesisQMF.usb;
+  }
+
+  /* Shift left all samples of the x-over-area as much as possible
+     An unnecessary coarse scale could cause ov_lb_scale or ov_hb_scale to be
+     adapted and the accuracy in the next frame would seriously suffer! */
+
+  maxVal = maxSubbandSample( OverlapBufferReal,
+                             (useLP) ? NULL : OverlapBufferImag,
+                             startBand,
+                             stopBand,
+                             0,
+                             startSlot);
+
+  reserve = CntLeadingZeros(maxVal)-1;
+  reserve = fixMin(reserve,DFRACT_BITS-1-source_scale);
+
+  rescaleSubbandSamples( OverlapBufferReal,
+                         (useLP) ? NULL : OverlapBufferImag,
+                         startBand,
+                         stopBand,
+                         0,
+                         startSlot,
+                         reserve);
+  source_scale += reserve;
+
+  delta_scale = target_scale - source_scale;
+
+  if (delta_scale > 0) { /* x-over-area is dominant */
+    delta_scale = -delta_scale;
+    startBand = target_lsb;
+    stopBand = target_usb;
+
+    if (new_lsb > old_lsb) {
+      /* The lowband has to be rescaled */
+      hSbrDec->sbrScaleFactor.ov_lb_scale = source_scale;
+    }
+    else {
+      /* The highband has be be rescaled */
+      hSbrDec->sbrScaleFactor.ov_hb_scale = source_scale;
+    }
+  }
+
+  FDK_ASSERT(startBand <= stopBand);
+
+  if (!useLP) {
+    for (l=0; l<startSlot; l++) {
+      scaleValues( OverlapBufferReal[l] + startBand, stopBand-startBand, delta_scale );
+      scaleValues( OverlapBufferImag[l] + startBand, stopBand-startBand, delta_scale );
+    }
+  } else
+  for (l=0; l<startSlot; l++) {
+    scaleValues( OverlapBufferReal[l] + startBand, stopBand-startBand, delta_scale );
+  }
+
+
+  /*
+    Initialize transposer and limiter
+  */
+  sbrError = resetLppTransposer (&hSbrDec->LppTrans,
+                                 hHeaderData->freqBandData.lowSubband,
+                                 hHeaderData->freqBandData.v_k_master,
+                                 hHeaderData->freqBandData.numMaster,
+                                 hHeaderData->freqBandData.freqBandTableNoise,
+                                 hHeaderData->freqBandData.nNfb,
+                                 hHeaderData->freqBandData.highSubband,
+                                 hHeaderData->sbrProcSmplRate);
+  if (sbrError != SBRDEC_OK)
+    return sbrError;
+
+  sbrError = ResetLimiterBands ( hHeaderData->freqBandData.limiterBandTable,
+                                 &hHeaderData->freqBandData.noLimiterBands,
+                                 hHeaderData->freqBandData.freqBandTable[0],
+                                 hHeaderData->freqBandData.nSfb[0],
+                                 hSbrDec->LppTrans.pSettings->patchParam,
+                                 hSbrDec->LppTrans.pSettings->noOfPatches,
+                                 hHeaderData->bs_data.limiterBands);
+
+
+  return sbrError;
+}
diff --git a/libSBRdec/src/sbr_dec.h b/libSBRdec/src/sbr_dec.h
new file mode 100644
index 0000000..309327f
--- /dev/null
+++ b/libSBRdec/src/sbr_dec.h
@@ -0,0 +1,210 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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 decoder  
+*/
+#ifndef __SBR_DEC_H
+#define __SBR_DEC_H
+
+#include "sbrdecoder.h"
+
+#include "lpp_tran.h"
+#include "qmf.h"
+#include "env_calc.h"
+#include "FDK_audio.h"
+
+
+#include "sbrdec_drc.h"
+
+#define SACDEC_ALIGNMENT_FIX
+
+typedef struct
+{
+  QMF_FILTER_BANK     AnalysiscQMF;
+  QMF_FILTER_BANK     SynthesisQMF;
+
+  SBR_CALCULATE_ENVELOPE  SbrCalculateEnvelope;
+  SBR_LPP_TRANS           LppTrans;
+
+  QMF_SCALE_FACTOR sbrScaleFactor;
+  QMF_SCALE_FACTOR sbrScaleFactorRight;
+
+  /*! Delayed spectral data needed for the dynamic framing of SBR. Not required in case of CLDFB */
+  FIXP_DBL * pSbrOverlapBuffer;
+
+  /* References to workbuffers */
+  FIXP_DBL * WorkBuffer1;
+  FIXP_DBL * WorkBuffer2;
+
+  /* QMF filter states */
+  FIXP_QAS   anaQmfStates[(320)];
+  FIXP_QSS * pSynQmfStates;
+
+  /* Reference pointer arrays for QMF time slots,
+     mixed among overlap and current slots. */
+  FIXP_DBL * QmfBufferReal[(((1024)/(32))+(6))];
+  FIXP_DBL * QmfBufferImag[(((1024)/(32))+(6))];
+  int useLP;
+
+  /* QMF domain extension time slot reference pointer array */
+
+  SBRDEC_DRC_CHANNEL  sbrDrcChannel;
+
+} SBR_DEC;
+
+typedef SBR_DEC *HANDLE_SBR_DEC;
+
+
+typedef struct
+{
+  SBR_FRAME_DATA      frameData[(1)+1];
+  SBR_PREV_FRAME_DATA prevFrameData;
+  SBR_DEC SbrDec;
+}
+SBR_CHANNEL;
+
+typedef SBR_CHANNEL *HANDLE_SBR_CHANNEL;
+
+void
+SbrDecodeAndProcess (HANDLE_SBR_DEC hSbrDec,
+                     INT_PCM *timeIn,
+                     HANDLE_SBR_HEADER_DATA hHeaderData,
+                     HANDLE_SBR_FRAME_DATA hFrameData,
+                     HANDLE_SBR_PREV_FRAME_DATA hPrevFrameData,
+                     int applyProcessing,
+                     int channelNr
+                     , UCHAR useLP
+                     );
+
+
+void
+SbrConstructTimeOutput (HANDLE_SBR_DEC hSbrDec,            /*!< handle to Decoder channel */
+                        INT_PCM *timeOut,                    /*!< pointer to output time signal */
+                        HANDLE_SBR_HEADER_DATA hHeaderData,/*!< Static control data */
+                        HANDLE_SBR_PREV_FRAME_DATA hPrevFrameData,  /*!< Some control data of last frame */
+                        int channelNr
+                       ,UCHAR useLP
+                        );
+
+
+void
+sbr_dec (HANDLE_SBR_DEC hSbrDec,            /*!< handle to Decoder channel */
+         INT_PCM *timeIn,                   /*!< pointer to input time signal */
+         INT_PCM *timeOut,                  /*!< pointer to output time signal */
+         HANDLE_SBR_DEC hSbrDecRight,       /*!< handle to Decoder channel right */
+         INT_PCM *timeOutRight,             /*!< pointer to output time signal */
+         const int strideIn,                /*!< Time data traversal strideIn */
+         const int strideOut,               /*!< Time data traversal strideOut */
+         HANDLE_SBR_HEADER_DATA hHeaderData,/*!< Static control data */
+         HANDLE_SBR_FRAME_DATA hFrameData,  /*!< Control data of current frame */
+         HANDLE_SBR_PREV_FRAME_DATA hPrevFrameData,  /*!< Some control data of last frame */
+         const int applyProcessing,         /*!< Flag for SBR operation */
+         HANDLE_PS_DEC h_ps_d,
+         const UINT flags
+        );
+
+
+
+SBR_ERROR
+createSbrDec (SBR_CHANNEL * hSbrChannel,
+              HANDLE_SBR_HEADER_DATA hHeaderData,
+              TRANSPOSER_SETTINGS *pSettings,
+              const int downsampleFac,
+              const UINT qmfFlags,
+              const UINT flags,
+              const int overlap,
+              int chan);
+
+int
+deleteSbrDec (SBR_CHANNEL * hSbrChannel);
+
+SBR_ERROR
+resetSbrDec (HANDLE_SBR_DEC hSbrDec,
+             HANDLE_SBR_HEADER_DATA hHeaderData,
+             HANDLE_SBR_PREV_FRAME_DATA hPrevFrameData,
+             const int useLP,
+             const int downsampleFac);
+
+#endif
diff --git a/libSBRdec/src/sbr_ram.cpp b/libSBRdec/src/sbr_ram.cpp
new file mode 100644
index 0000000..ee95e01
--- /dev/null
+++ b/libSBRdec/src/sbr_ram.cpp
@@ -0,0 +1,194 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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"
+
+
+
+
+#define WORKBUFFER1_TAG 0
+#define WORKBUFFER2_TAG 1
+
+/*!
+  \name StaticSbrData
+
+  Static memory areas, must not be overwritten in other sections of the decoder
+*/
+/* @{ */
+
+/*! SBR Decoder main structure */
+C_ALLOC_MEM(Ram_SbrDecoder, struct SBR_DECODER_INSTANCE, 1)
+/*! SBR Decoder element data  <br>
+  Dimension: (4) */
+C_ALLOC_MEM2(Ram_SbrDecElement, SBR_DECODER_ELEMENT, 1, (4))
+/*! SBR Decoder individual channel data  <br>
+  Dimension: (6) */
+C_ALLOC_MEM2(Ram_SbrDecChannel, SBR_CHANNEL, 1, (6)+1)
+
+/*! Filter states for QMF-synthesis.  <br>
+  Dimension: #(6) * (#QMF_FILTER_STATE_SYN_SIZE-#(64)) */
+C_AALLOC_MEM2_L(Ram_sbr_QmfStatesSynthesis, FIXP_QSS, (640)-(64), (6)+1, SECT_DATA_L1)
+
+/*! Delayed spectral data needed for the dynamic framing of SBR.
+  For mp3PRO, 1/3 of a frame is buffered (#(6) 6) */
+C_AALLOC_MEM2(Ram_sbr_OverlapBuffer, FIXP_DBL, 2 * (6) * (64), (6)+1)
+
+/*! Static Data of PS */
+
+C_ALLOC_MEM(Ram_ps_dec, PS_DEC, 1)
+
+
+/* @} */
+
+
+/*!
+  \name DynamicSbrData
+
+  Dynamic memory areas, might be reused in other algorithm sections,
+  e.g. the core decoder
+  <br>
+  Depending on the mode set by DONT_USE_CORE_WORKBUFFER, workbuffers are
+  defined additionally to the CoreWorkbuffer.
+  <br>
+  The size of WorkBuffers is ((1024)/(32))*(64) = 2048.
+  <br>
+  WorkBuffer2 is a pointer to the CoreWorkBuffer wich is reused here in the SBR part. In case of
+  DONT_USE_CORE_WORKBUFFER, the CoreWorkbuffer is not used and the according
+  Workbuffer2 is defined locally in this file.
+  <br>
+  WorkBuffer1 is reused in the AAC core (-> aacdecoder.cpp, aac_ram.cpp)
+  <br>
+
+  Use of WorkBuffers:
+  <pre>
+
+    -------------------------------------------------------------
+    AAC core:
+
+      CoreWorkbuffer: spectral coefficients
+      WorkBuffer1:    CAacDecoderChannelInfo, CAacDecoderDynamicData
+
+    -------------------------------------------------------------
+    SBR part:
+      ----------------------------------------------
+      Low Power Mode (useLP=1 or LOW_POWER_SBR_ONLY), see assignLcTimeSlots()
+
+        SLOT_BASED_PROTOTYPE_SYN_FILTER
+
+        WorkBuffer1                                WorkBuffer2(=CoreWorkbuffer)
+         ________________                           ________________
+        | RealLeft       |                         | RealRight      |
+        |________________|                         |________________|
+
+      ----------------------------------------------
+      High Quality Mode (!LOW_POWER_SBR_ONLY and useLP=0), see assignHqTimeSlots()
+
+         SLOTBASED_PS
+
+         WorkBuffer1                                WorkBuffer2(=CoreWorkbuffer)
+         ________________                           ________________
+        | Real/Imag      |  interleaved            | Real/Imag      |  interleaved
+        |________________|  first half actual ch   |________________|  second half actual ch
+
+    -------------------------------------------------------------
+
+  </pre>
+
+*/
+/* @{ */
+C_ALLOC_MEM_OVERLAY(Ram_SbrDecWorkBuffer1, FIXP_DBL, ((1024)/(32))*(64), SECT_DATA_L1, WORKBUFFER1_TAG)
+C_ALLOC_MEM_OVERLAY(Ram_SbrDecWorkBuffer2, FIXP_DBL, ((1024)/(32))*(64), SECT_DATA_L2, WORKBUFFER2_TAG)
+
+/* @} */
+
+
+
+
diff --git a/libSBRdec/src/sbr_ram.h b/libSBRdec/src/sbr_ram.h
new file mode 100644
index 0000000..5469e51
--- /dev/null
+++ b/libSBRdec/src/sbr_ram.h
@@ -0,0 +1,158 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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 "sbrdecoder.h"
+
+#include "env_extr.h"
+#include "sbr_dec.h"
+
+
+
+#define SBRDEC_MAX_CH_PER_ELEMENT  (2)
+
+typedef struct
+{
+  SBR_CHANNEL          *pSbrChannel[SBRDEC_MAX_CH_PER_ELEMENT];
+  TRANSPOSER_SETTINGS   transposerSettings; /* Common transport settings for each individual channel of an element */
+  HANDLE_FDK_BITSTREAM  hBs;
+
+  MP4_ELEMENT_ID        elementID;          /* Element ID set during initialization. Can be used for concealment */
+  int                   nChannels;          /* Number of elements output channels (=2 in case of PS) */
+
+  UCHAR frameErrorFlag[(1)+1];  /* Frame error status (for every slot in the delay line).
+                                                       Will be copied into header at the very beginning of decodeElement() routine. */
+
+  UCHAR useFrameSlot;          /* Index which defines which slot will be decoded/filled next (used with additional delay) */
+  UCHAR useHeaderSlot[(1)+1];   /* Index array that provides the link between header and frame data
+                                                       (important when processing with additional delay). */
+} SBR_DECODER_ELEMENT;
+
+
+struct SBR_DECODER_INSTANCE
+{
+  SBR_DECODER_ELEMENT  *pSbrElement[(4)];
+  SBR_HEADER_DATA       sbrHeader[(4)][(1)+1];      /* Sbr header for each individual channel of an element */
+
+  FIXP_DBL *workBuffer1;
+  FIXP_DBL *workBuffer2;
+
+  HANDLE_PS_DEC  hParametricStereoDec;
+
+  /* Global parameters */
+  AUDIO_OBJECT_TYPE coreCodec; /* AOT of core codec */
+  int numSbrElements;
+  int numSbrChannels;
+  INT sampleRateIn;            /* SBR decoder input sampling rate; might be different than the transposer input sampling rate. */
+  INT sampleRateOut;           /* Sampling rate of the SBR decoder output audio samples. */
+  USHORT codecFrameSize;
+  UCHAR synDownsampleFac;
+  UCHAR numDelayFrames;        /* The current number of additional delay frames used for processing. */
+
+  UINT flags;
+
+};
+
+H_ALLOC_MEM(Ram_SbrDecElement, SBR_DECODER_ELEMENT)
+H_ALLOC_MEM(Ram_SbrDecChannel, SBR_CHANNEL)
+H_ALLOC_MEM(Ram_SbrDecoder, struct SBR_DECODER_INSTANCE)
+
+H_ALLOC_MEM(Ram_sbr_QmfStatesSynthesis, FIXP_QSS)
+H_ALLOC_MEM(Ram_sbr_OverlapBuffer, FIXP_DBL)
+
+
+H_ALLOC_MEM(Ram_ps_dec, PS_DEC)
+
+
+H_ALLOC_MEM_OVERLAY(Ram_SbrDecWorkBuffer1, FIXP_DBL)
+H_ALLOC_MEM_OVERLAY(Ram_SbrDecWorkBuffer2, FIXP_DBL)
+
+
+#endif /* _SBR_RAM_H_ */
diff --git a/libSBRdec/src/sbr_rom.cpp b/libSBRdec/src/sbr_rom.cpp
new file mode 100644
index 0000000..8520b14
--- /dev/null
+++ b/libSBRdec/src/sbr_rom.cpp
@@ -0,0 +1,1412 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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"
+
+
+
+
+/*!
+  \name   StartStopBands
+  \brief  Start and stop subbands of the highband.
+
+  k_o = startMin + offset[bs_start_freq];
+  startMin = {3000,4000,5000} * (128/FS_sbr) / FS_sbr < 32Khz, 32Khz <= FS_sbr < 64KHz, 64KHz <= FS_sbr
+  The stop subband can also be calculated to save memory by defining #CALC_STOP_BAND.
+*/
+//@{
+const UCHAR FDK_sbrDecoder_sbr_start_freq_16[16] = {16, 17, 18, 19, 20, 21, 22, 23, 24, 25, 26, 27, 28, 29, 30, 31};
+const UCHAR FDK_sbrDecoder_sbr_start_freq_22[16] = {12, 13, 14, 15, 16, 17, 18, 19, 20, 21, 22, 23, 24, 26, 28, 30};
+const UCHAR FDK_sbrDecoder_sbr_start_freq_24[16] = {11, 13, 14, 15, 16, 17, 18, 19, 20, 21, 22, 23, 25, 27, 29, 32};
+const UCHAR FDK_sbrDecoder_sbr_start_freq_32[16] = {10, 12, 14, 15, 16, 17, 18, 19, 20, 21, 22, 23, 25, 27, 29, 32};
+const UCHAR FDK_sbrDecoder_sbr_start_freq_40[16] = {12, 13, 14, 15, 16, 17, 18, 19, 20, 21, 22, 24, 26, 28, 30, 32};
+const UCHAR FDK_sbrDecoder_sbr_start_freq_44[16] = { 8, 10, 11, 12, 13, 14, 15, 16, 17, 18, 19, 21, 23, 25, 28, 32};
+const UCHAR FDK_sbrDecoder_sbr_start_freq_48[16] = { 7,  9, 10, 11, 12, 13, 14, 15, 16, 17, 18, 20, 22, 24, 27, 31};
+//@}
+
+
+/*!
+  \name   Whitening
+  \brief  Coefficients for spectral whitening in the transposer
+*/
+//@{
+/*! Assignment of whitening tuning depending on the crossover frequency */
+const USHORT FDK_sbrDecoder_sbr_whFactorsIndex[NUM_WHFACTOR_TABLE_ENTRIES] = {
+  0,
+  5000,
+  6000,
+  6500,
+  7000,
+  7500,
+  8000,
+  9000,
+  10000
+};
+
+/*!
+  \brief Whithening levels tuning table
+
+  With the current tuning, there are some redundant entries:
+
+  \li  NUM_WHFACTOR_TABLE_ENTRIES can be reduced by 3,
+  \li  the first coloumn can be eliminated.
+
+*/
+const FIXP_DBL FDK_sbrDecoder_sbr_whFactorsTable[NUM_WHFACTOR_TABLE_ENTRIES][6] = {
+  /* OFF_LEVEL, TRANSITION_LEVEL, LOW_LEVEL, MID_LEVEL, HIGH_LEVEL */
+  {  FL2FXCONST_DBL(0.00f), FL2FXCONST_DBL(0.6f), FL2FXCONST_DBL(0.75f), FL2FXCONST_DBL(0.90f), FL2FXCONST_DBL(0.98f)},  /* < 5000 */
+  {  FL2FXCONST_DBL(0.00f), FL2FXCONST_DBL(0.6f), FL2FXCONST_DBL(0.75f), FL2FXCONST_DBL(0.90f), FL2FXCONST_DBL(0.98f)},  /* 5000 < 6000 */
+  {  FL2FXCONST_DBL(0.00f), FL2FXCONST_DBL(0.6f), FL2FXCONST_DBL(0.75f), FL2FXCONST_DBL(0.90f), FL2FXCONST_DBL(0.98f)},  /* 6000 < 6500 */
+  {  FL2FXCONST_DBL(0.00f), FL2FXCONST_DBL(0.6f), FL2FXCONST_DBL(0.75f), FL2FXCONST_DBL(0.90f), FL2FXCONST_DBL(0.98f)},  /* 6500 < 7000 */
+  {  FL2FXCONST_DBL(0.00f), FL2FXCONST_DBL(0.6f), FL2FXCONST_DBL(0.75f), FL2FXCONST_DBL(0.90f), FL2FXCONST_DBL(0.98f)},  /* 7000 < 7500 */
+  {  FL2FXCONST_DBL(0.00f), FL2FXCONST_DBL(0.6f), FL2FXCONST_DBL(0.75f), FL2FXCONST_DBL(0.90f), FL2FXCONST_DBL(0.98f)},  /* 7500 < 8000 */
+  {  FL2FXCONST_DBL(0.00f), FL2FXCONST_DBL(0.6f), FL2FXCONST_DBL(0.75f), FL2FXCONST_DBL(0.90f), FL2FXCONST_DBL(0.98f)},  /* 8000 < 9000 */
+  {  FL2FXCONST_DBL(0.00f), FL2FXCONST_DBL(0.6f), FL2FXCONST_DBL(0.75f), FL2FXCONST_DBL(0.90f), FL2FXCONST_DBL(0.98f)},  /* 9000 < 10000 */
+  {  FL2FXCONST_DBL(0.00f), FL2FXCONST_DBL(0.6f), FL2FXCONST_DBL(0.75f), FL2FXCONST_DBL(0.90f), FL2FXCONST_DBL(0.98f)},  /* > 10000 */
+};
+
+
+//@}
+
+
+/*!
+  \name   EnvAdj
+  \brief  Constants and tables used for envelope adjustment
+*/
+//@{
+
+/*! Mantissas of gain limits */
+const FIXP_SGL FDK_sbrDecoder_sbr_limGains_m[4] =
+{
+  FL2FXCONST_SGL(0.5011932025f),   /*!< -3 dB. Gain limit when limiterGains in frameData is 0 */
+  FL2FXCONST_SGL(0.5f),            /*!< 0 dB.  Gain limit when limiterGains in frameData is 1 */
+  FL2FXCONST_SGL(0.9976346258f),   /*!< +3 dB. Gain limit when limiterGains in frameData is 2 */
+  FL2FXCONST_SGL(0.6776263578f)    /*!< Inf.   Gain limit when limiterGains in frameData is 3 */
+};
+
+/*! Exponents of gain limits */
+const UCHAR FDK_sbrDecoder_sbr_limGains_e[4] =
+{
+  0, 1, 1, 67
+};
+
+/*! Constants for calculating the number of limiter bands */
+const FIXP_SGL FDK_sbrDecoder_sbr_limiterBandsPerOctaveDiv4[4] =
+{
+  FL2FXCONST_SGL(1.0f / 4.0f),
+  FL2FXCONST_SGL(1.2f / 4.0f),
+  FL2FXCONST_SGL(2.0f / 4.0f),
+  FL2FXCONST_SGL(3.0f / 4.0f)
+};
+
+/*! Ratio of old gains and noise levels for the first 4 timeslots of an envelope */
+const FIXP_SGL FDK_sbrDecoder_sbr_smoothFilter[4] = {
+  FL2FXCONST_SGL(0.66666666666666f),
+  FL2FXCONST_SGL(0.36516383427084f),
+  FL2FXCONST_SGL(0.14699433520835f),
+  FL2FXCONST_SGL(0.03183050093751f)
+};
+
+
+/*! Real and imaginary part of random noise which will be modulated
+  to the desired level. An accuracy of 13 bits is sufficient for these
+  random numbers.
+*/
+const FIXP_SGL FDK_sbrDecoder_sbr_randomPhase[SBR_NF_NO_RANDOM_VAL][2] = {
+  { FL2FXCONST_SGL(-0.99948153278296f / 8.0), FL2FXCONST_SGL(-0.59483417516607f / 8.0) },
+  { FL2FXCONST_SGL( 0.97113454393991f / 8.0), FL2FXCONST_SGL(-0.67528515225647f / 8.0) },
+  { FL2FXCONST_SGL( 0.14130051758487f / 8.0), FL2FXCONST_SGL(-0.95090983575689f / 8.0) },
+  { FL2FXCONST_SGL(-0.47005496701697f / 8.0), FL2FXCONST_SGL(-0.37340549728647f / 8.0) },
+  { FL2FXCONST_SGL( 0.80705063769351f / 8.0), FL2FXCONST_SGL( 0.29653668284408f / 8.0) },
+  { FL2FXCONST_SGL(-0.38981478896926f / 8.0), FL2FXCONST_SGL( 0.89572605717087f / 8.0) },
+  { FL2FXCONST_SGL(-0.01053049862020f / 8.0), FL2FXCONST_SGL(-0.66959058036166f / 8.0) },
+  { FL2FXCONST_SGL(-0.91266367957293f / 8.0), FL2FXCONST_SGL(-0.11522938140034f / 8.0) },
+  { FL2FXCONST_SGL( 0.54840422910309f / 8.0), FL2FXCONST_SGL( 0.75221367176302f / 8.0) },
+  { FL2FXCONST_SGL( 0.40009252867955f / 8.0), FL2FXCONST_SGL(-0.98929400334421f / 8.0) },
+  { FL2FXCONST_SGL(-0.99867974711855f / 8.0), FL2FXCONST_SGL(-0.88147068645358f / 8.0) },
+  { FL2FXCONST_SGL(-0.95531076805040f / 8.0), FL2FXCONST_SGL( 0.90908757154593f / 8.0) },
+  { FL2FXCONST_SGL(-0.45725933317144f / 8.0), FL2FXCONST_SGL(-0.56716323646760f / 8.0) },
+  { FL2FXCONST_SGL(-0.72929675029275f / 8.0), FL2FXCONST_SGL(-0.98008272727324f / 8.0) },
+  { FL2FXCONST_SGL( 0.75622801399036f / 8.0), FL2FXCONST_SGL( 0.20950329995549f / 8.0) },
+  { FL2FXCONST_SGL( 0.07069442601050f / 8.0), FL2FXCONST_SGL(-0.78247898470706f / 8.0) },
+  { FL2FXCONST_SGL( 0.74496252926055f / 8.0), FL2FXCONST_SGL(-0.91169004445807f / 8.0) },
+  { FL2FXCONST_SGL(-0.96440182703856f / 8.0), FL2FXCONST_SGL(-0.94739918296622f / 8.0) },
+  { FL2FXCONST_SGL( 0.30424629369539f / 8.0), FL2FXCONST_SGL(-0.49438267012479f / 8.0) },
+  { FL2FXCONST_SGL( 0.66565033746925f / 8.0), FL2FXCONST_SGL( 0.64652935542491f / 8.0) },
+  { FL2FXCONST_SGL( 0.91697008020594f / 8.0), FL2FXCONST_SGL( 0.17514097332009f / 8.0) },
+  { FL2FXCONST_SGL(-0.70774918760427f / 8.0), FL2FXCONST_SGL( 0.52548653416543f / 8.0) },
+  { FL2FXCONST_SGL(-0.70051415345560f / 8.0), FL2FXCONST_SGL(-0.45340028808763f / 8.0) },
+  { FL2FXCONST_SGL(-0.99496513054797f / 8.0), FL2FXCONST_SGL(-0.90071908066973f / 8.0) },
+  { FL2FXCONST_SGL( 0.98164490790123f / 8.0), FL2FXCONST_SGL(-0.77463155528697f / 8.0) },
+  { FL2FXCONST_SGL(-0.54671580548181f / 8.0), FL2FXCONST_SGL(-0.02570928536004f / 8.0) },
+  { FL2FXCONST_SGL(-0.01689629065389f / 8.0), FL2FXCONST_SGL( 0.00287506445732f / 8.0) },
+  { FL2FXCONST_SGL(-0.86110349531986f / 8.0), FL2FXCONST_SGL( 0.42548583726477f / 8.0) },
+  { FL2FXCONST_SGL(-0.98892980586032f / 8.0), FL2FXCONST_SGL(-0.87881132267556f / 8.0) },
+  { FL2FXCONST_SGL( 0.51756627678691f / 8.0), FL2FXCONST_SGL( 0.66926784710139f / 8.0) },
+  { FL2FXCONST_SGL(-0.99635026409640f / 8.0), FL2FXCONST_SGL(-0.58107730574765f / 8.0) },
+  { FL2FXCONST_SGL(-0.99969370862163f / 8.0), FL2FXCONST_SGL( 0.98369989360250f / 8.0) },
+  { FL2FXCONST_SGL( 0.55266258627194f / 8.0), FL2FXCONST_SGL( 0.59449057465591f / 8.0) },
+  { FL2FXCONST_SGL( 0.34581177741673f / 8.0), FL2FXCONST_SGL( 0.94879421061866f / 8.0) },
+  { FL2FXCONST_SGL( 0.62664209577999f / 8.0), FL2FXCONST_SGL(-0.74402970906471f / 8.0) },
+  { FL2FXCONST_SGL(-0.77149701404973f / 8.0), FL2FXCONST_SGL(-0.33883658042801f / 8.0) },
+  { FL2FXCONST_SGL(-0.91592244254432f / 8.0), FL2FXCONST_SGL( 0.03687901376713f / 8.0) },
+  { FL2FXCONST_SGL(-0.76285492357887f / 8.0), FL2FXCONST_SGL(-0.91371867919124f / 8.0) },
+  { FL2FXCONST_SGL( 0.79788337195331f / 8.0), FL2FXCONST_SGL(-0.93180971199849f / 8.0) },
+  { FL2FXCONST_SGL( 0.54473080610200f / 8.0), FL2FXCONST_SGL(-0.11919206037186f / 8.0) },
+  { FL2FXCONST_SGL(-0.85639281671058f / 8.0), FL2FXCONST_SGL( 0.42429854760451f / 8.0) },
+  { FL2FXCONST_SGL(-0.92882402971423f / 8.0), FL2FXCONST_SGL( 0.27871809078609f / 8.0) },
+  { FL2FXCONST_SGL(-0.11708371046774f / 8.0), FL2FXCONST_SGL(-0.99800843444966f / 8.0) },
+  { FL2FXCONST_SGL( 0.21356749817493f / 8.0), FL2FXCONST_SGL(-0.90716295627033f / 8.0) },
+  { FL2FXCONST_SGL(-0.76191692573909f / 8.0), FL2FXCONST_SGL( 0.99768118356265f / 8.0) },
+  { FL2FXCONST_SGL( 0.98111043100884f / 8.0), FL2FXCONST_SGL(-0.95854459734407f / 8.0) },
+  { FL2FXCONST_SGL(-0.85913269895572f / 8.0), FL2FXCONST_SGL( 0.95766566168880f / 8.0) },
+  { FL2FXCONST_SGL(-0.93307242253692f / 8.0), FL2FXCONST_SGL( 0.49431757696466f / 8.0) },
+  { FL2FXCONST_SGL( 0.30485754879632f / 8.0), FL2FXCONST_SGL(-0.70540034357529f / 8.0) },
+  { FL2FXCONST_SGL( 0.85289650925190f / 8.0), FL2FXCONST_SGL( 0.46766131791044f / 8.0) },
+  { FL2FXCONST_SGL( 0.91328082618125f / 8.0), FL2FXCONST_SGL(-0.99839597361769f / 8.0) },
+  { FL2FXCONST_SGL(-0.05890199924154f / 8.0), FL2FXCONST_SGL( 0.70741827819497f / 8.0) },
+  { FL2FXCONST_SGL( 0.28398686150148f / 8.0), FL2FXCONST_SGL( 0.34633555702188f / 8.0) },
+  { FL2FXCONST_SGL( 0.95258164539612f / 8.0), FL2FXCONST_SGL(-0.54893416026939f / 8.0) },
+  { FL2FXCONST_SGL(-0.78566324168507f / 8.0), FL2FXCONST_SGL(-0.75568541079691f / 8.0) },
+  { FL2FXCONST_SGL(-0.95789495447877f / 8.0), FL2FXCONST_SGL(-0.20423194696966f / 8.0) },
+  { FL2FXCONST_SGL( 0.82411158711197f / 8.0), FL2FXCONST_SGL( 0.96654618432562f / 8.0) },
+  { FL2FXCONST_SGL(-0.65185446735885f / 8.0), FL2FXCONST_SGL(-0.88734990773289f / 8.0) },
+  { FL2FXCONST_SGL(-0.93643603134666f / 8.0), FL2FXCONST_SGL( 0.99870790442385f / 8.0) },
+  { FL2FXCONST_SGL( 0.91427159529618f / 8.0), FL2FXCONST_SGL(-0.98290505544444f / 8.0) },
+  { FL2FXCONST_SGL(-0.70395684036886f / 8.0), FL2FXCONST_SGL( 0.58796798221039f / 8.0) },
+  { FL2FXCONST_SGL( 0.00563771969365f / 8.0), FL2FXCONST_SGL( 0.61768196727244f / 8.0) },
+  { FL2FXCONST_SGL( 0.89065051931895f / 8.0), FL2FXCONST_SGL( 0.52783352697585f / 8.0) },
+  { FL2FXCONST_SGL(-0.68683707712762f / 8.0), FL2FXCONST_SGL( 0.80806944710339f / 8.0) },
+  { FL2FXCONST_SGL( 0.72165342518718f / 8.0), FL2FXCONST_SGL(-0.69259857349564f / 8.0) },
+  { FL2FXCONST_SGL(-0.62928247730667f / 8.0), FL2FXCONST_SGL( 0.13627037407335f / 8.0) },
+  { FL2FXCONST_SGL( 0.29938434065514f / 8.0), FL2FXCONST_SGL(-0.46051329682246f / 8.0) },
+  { FL2FXCONST_SGL(-0.91781958879280f / 8.0), FL2FXCONST_SGL(-0.74012716684186f / 8.0) },
+  { FL2FXCONST_SGL( 0.99298717043688f / 8.0), FL2FXCONST_SGL( 0.40816610075661f / 8.0) },
+  { FL2FXCONST_SGL( 0.82368298622748f / 8.0), FL2FXCONST_SGL(-0.74036047190173f / 8.0) },
+  { FL2FXCONST_SGL(-0.98512833386833f / 8.0), FL2FXCONST_SGL(-0.99972330709594f / 8.0) },
+  { FL2FXCONST_SGL(-0.95915368242257f / 8.0), FL2FXCONST_SGL(-0.99237800466040f / 8.0) },
+  { FL2FXCONST_SGL(-0.21411126572790f / 8.0), FL2FXCONST_SGL(-0.93424819052545f / 8.0) },
+  { FL2FXCONST_SGL(-0.68821476106884f / 8.0), FL2FXCONST_SGL(-0.26892306315457f / 8.0) },
+  { FL2FXCONST_SGL( 0.91851997982317f / 8.0), FL2FXCONST_SGL( 0.09358228901785f / 8.0) },
+  { FL2FXCONST_SGL(-0.96062769559127f / 8.0), FL2FXCONST_SGL( 0.36099095133739f / 8.0) },
+  { FL2FXCONST_SGL( 0.51646184922287f / 8.0), FL2FXCONST_SGL(-0.71373332873917f / 8.0) },
+  { FL2FXCONST_SGL( 0.61130721139669f / 8.0), FL2FXCONST_SGL( 0.46950141175917f / 8.0) },
+  { FL2FXCONST_SGL( 0.47336129371299f / 8.0), FL2FXCONST_SGL(-0.27333178296162f / 8.0) },
+  { FL2FXCONST_SGL( 0.90998308703519f / 8.0), FL2FXCONST_SGL( 0.96715662938132f / 8.0) },
+  { FL2FXCONST_SGL( 0.44844799194357f / 8.0), FL2FXCONST_SGL( 0.99211574628306f / 8.0) },
+  { FL2FXCONST_SGL( 0.66614891079092f / 8.0), FL2FXCONST_SGL( 0.96590176169121f / 8.0) },
+  { FL2FXCONST_SGL( 0.74922239129237f / 8.0), FL2FXCONST_SGL(-0.89879858826087f / 8.0) },
+  { FL2FXCONST_SGL(-0.99571588506485f / 8.0), FL2FXCONST_SGL( 0.52785521494349f / 8.0) },
+  { FL2FXCONST_SGL( 0.97401082477563f / 8.0), FL2FXCONST_SGL(-0.16855870075190f / 8.0) },
+  { FL2FXCONST_SGL( 0.72683747733879f / 8.0), FL2FXCONST_SGL(-0.48060774432251f / 8.0) },
+  { FL2FXCONST_SGL( 0.95432193457128f / 8.0), FL2FXCONST_SGL( 0.68849603408441f / 8.0) },
+  { FL2FXCONST_SGL(-0.72962208425191f / 8.0), FL2FXCONST_SGL(-0.76608443420917f / 8.0) },
+  { FL2FXCONST_SGL(-0.85359479233537f / 8.0), FL2FXCONST_SGL( 0.88738125901579f / 8.0) },
+  { FL2FXCONST_SGL(-0.81412430338535f / 8.0), FL2FXCONST_SGL(-0.97480768049637f / 8.0) },
+  { FL2FXCONST_SGL(-0.87930772356786f / 8.0), FL2FXCONST_SGL( 0.74748307690436f / 8.0) },
+  { FL2FXCONST_SGL(-0.71573331064977f / 8.0), FL2FXCONST_SGL(-0.98570608178923f / 8.0) },
+  { FL2FXCONST_SGL( 0.83524300028228f / 8.0), FL2FXCONST_SGL( 0.83702537075163f / 8.0) },
+  { FL2FXCONST_SGL(-0.48086065601423f / 8.0), FL2FXCONST_SGL(-0.98848504923531f / 8.0) },
+  { FL2FXCONST_SGL( 0.97139128574778f / 8.0), FL2FXCONST_SGL( 0.80093621198236f / 8.0) },
+  { FL2FXCONST_SGL( 0.51992825347895f / 8.0), FL2FXCONST_SGL( 0.80247631400510f / 8.0) },
+  { FL2FXCONST_SGL(-0.00848591195325f / 8.0), FL2FXCONST_SGL(-0.76670128000486f / 8.0) },
+  { FL2FXCONST_SGL(-0.70294374303036f / 8.0), FL2FXCONST_SGL( 0.55359910445577f / 8.0) },
+  { FL2FXCONST_SGL(-0.95894428168140f / 8.0), FL2FXCONST_SGL(-0.43265504344783f / 8.0) },
+  { FL2FXCONST_SGL( 0.97079252950321f / 8.0), FL2FXCONST_SGL( 0.09325857238682f / 8.0) },
+  { FL2FXCONST_SGL(-0.92404293670797f / 8.0), FL2FXCONST_SGL( 0.85507704027855f / 8.0) },
+  { FL2FXCONST_SGL(-0.69506469500450f / 8.0), FL2FXCONST_SGL( 0.98633412625459f / 8.0) },
+  { FL2FXCONST_SGL( 0.26559203620024f / 8.0), FL2FXCONST_SGL( 0.73314307966524f / 8.0) },
+  { FL2FXCONST_SGL( 0.28038443336943f / 8.0), FL2FXCONST_SGL( 0.14537913654427f / 8.0) },
+  { FL2FXCONST_SGL(-0.74138124825523f / 8.0), FL2FXCONST_SGL( 0.99310339807762f / 8.0) },
+  { FL2FXCONST_SGL(-0.01752795995444f / 8.0), FL2FXCONST_SGL(-0.82616635284178f / 8.0) },
+  { FL2FXCONST_SGL(-0.55126773094930f / 8.0), FL2FXCONST_SGL(-0.98898543862153f / 8.0) },
+  { FL2FXCONST_SGL( 0.97960898850996f / 8.0), FL2FXCONST_SGL(-0.94021446752851f / 8.0) },
+  { FL2FXCONST_SGL(-0.99196309146936f / 8.0), FL2FXCONST_SGL( 0.67019017358456f / 8.0) },
+  { FL2FXCONST_SGL(-0.67684928085260f / 8.0), FL2FXCONST_SGL( 0.12631491649378f / 8.0) },
+  { FL2FXCONST_SGL( 0.09140039465500f / 8.0), FL2FXCONST_SGL(-0.20537731453108f / 8.0) },
+  { FL2FXCONST_SGL(-0.71658965751996f / 8.0), FL2FXCONST_SGL(-0.97788200391224f / 8.0) },
+  { FL2FXCONST_SGL( 0.81014640078925f / 8.0), FL2FXCONST_SGL( 0.53722648362443f / 8.0) },
+  { FL2FXCONST_SGL( 0.40616991671205f / 8.0), FL2FXCONST_SGL(-0.26469008598449f / 8.0) },
+  { FL2FXCONST_SGL(-0.67680188682972f / 8.0), FL2FXCONST_SGL( 0.94502052337695f / 8.0) },
+  { FL2FXCONST_SGL( 0.86849774348749f / 8.0), FL2FXCONST_SGL(-0.18333598647899f / 8.0) },
+  { FL2FXCONST_SGL(-0.99500381284851f / 8.0), FL2FXCONST_SGL(-0.02634122068550f / 8.0) },
+  { FL2FXCONST_SGL( 0.84329189340667f / 8.0), FL2FXCONST_SGL( 0.10406957462213f / 8.0) },
+  { FL2FXCONST_SGL(-0.09215968531446f / 8.0), FL2FXCONST_SGL( 0.69540012101253f / 8.0) },
+  { FL2FXCONST_SGL( 0.99956173327206f / 8.0), FL2FXCONST_SGL(-0.12358542001404f / 8.0) },
+  { FL2FXCONST_SGL(-0.79732779473535f / 8.0), FL2FXCONST_SGL(-0.91582524736159f / 8.0) },
+  { FL2FXCONST_SGL( 0.96349973642406f / 8.0), FL2FXCONST_SGL( 0.96640458041000f / 8.0) },
+  { FL2FXCONST_SGL(-0.79942778496547f / 8.0), FL2FXCONST_SGL( 0.64323902822857f / 8.0) },
+  { FL2FXCONST_SGL(-0.11566039853896f / 8.0), FL2FXCONST_SGL( 0.28587846253726f / 8.0) },
+  { FL2FXCONST_SGL(-0.39922954514662f / 8.0), FL2FXCONST_SGL( 0.94129601616966f / 8.0) },
+  { FL2FXCONST_SGL( 0.99089197565987f / 8.0), FL2FXCONST_SGL(-0.92062625581587f / 8.0) },
+  { FL2FXCONST_SGL( 0.28631285179909f / 8.0), FL2FXCONST_SGL(-0.91035047143603f / 8.0) },
+  { FL2FXCONST_SGL(-0.83302725605608f / 8.0), FL2FXCONST_SGL(-0.67330410892084f / 8.0) },
+  { FL2FXCONST_SGL( 0.95404443402072f / 8.0), FL2FXCONST_SGL( 0.49162765398743f / 8.0) },
+  { FL2FXCONST_SGL(-0.06449863579434f / 8.0), FL2FXCONST_SGL( 0.03250560813135f / 8.0) },
+  { FL2FXCONST_SGL(-0.99575054486311f / 8.0), FL2FXCONST_SGL( 0.42389784469507f / 8.0) },
+  { FL2FXCONST_SGL(-0.65501142790847f / 8.0), FL2FXCONST_SGL( 0.82546114655624f / 8.0) },
+  { FL2FXCONST_SGL(-0.81254441908887f / 8.0), FL2FXCONST_SGL(-0.51627234660629f / 8.0) },
+  { FL2FXCONST_SGL(-0.99646369485481f / 8.0), FL2FXCONST_SGL( 0.84490533520752f / 8.0) },
+  { FL2FXCONST_SGL( 0.00287840603348f / 8.0), FL2FXCONST_SGL( 0.64768261158166f / 8.0) },
+  { FL2FXCONST_SGL( 0.70176989408455f / 8.0), FL2FXCONST_SGL(-0.20453028573322f / 8.0) },
+  { FL2FXCONST_SGL( 0.96361882270190f / 8.0), FL2FXCONST_SGL( 0.40706967140989f / 8.0) },
+  { FL2FXCONST_SGL(-0.68883758192426f / 8.0), FL2FXCONST_SGL( 0.91338958840772f / 8.0) },
+  { FL2FXCONST_SGL(-0.34875585502238f / 8.0), FL2FXCONST_SGL( 0.71472290693300f / 8.0) },
+  { FL2FXCONST_SGL( 0.91980081243087f / 8.0), FL2FXCONST_SGL( 0.66507455644919f / 8.0) },
+  { FL2FXCONST_SGL(-0.99009048343881f / 8.0), FL2FXCONST_SGL( 0.85868021604848f / 8.0) },
+  { FL2FXCONST_SGL( 0.68865791458395f / 8.0), FL2FXCONST_SGL( 0.55660316809678f / 8.0) },
+  { FL2FXCONST_SGL(-0.99484402129368f / 8.0), FL2FXCONST_SGL(-0.20052559254934f / 8.0) },
+  { FL2FXCONST_SGL( 0.94214511408023f / 8.0), FL2FXCONST_SGL(-0.99696425367461f / 8.0) },
+  { FL2FXCONST_SGL(-0.67414626793544f / 8.0), FL2FXCONST_SGL( 0.49548221180078f / 8.0) },
+  { FL2FXCONST_SGL(-0.47339353684664f / 8.0), FL2FXCONST_SGL(-0.85904328834047f / 8.0) },
+  { FL2FXCONST_SGL( 0.14323651387360f / 8.0), FL2FXCONST_SGL(-0.94145598222488f / 8.0) },
+  { FL2FXCONST_SGL(-0.29268293575672f / 8.0), FL2FXCONST_SGL( 0.05759224927952f / 8.0) },
+  { FL2FXCONST_SGL( 0.43793861458754f / 8.0), FL2FXCONST_SGL(-0.78904969892724f / 8.0) },
+  { FL2FXCONST_SGL(-0.36345126374441f / 8.0), FL2FXCONST_SGL( 0.64874435357162f / 8.0) },
+  { FL2FXCONST_SGL(-0.08750604656825f / 8.0), FL2FXCONST_SGL( 0.97686944362527f / 8.0) },
+  { FL2FXCONST_SGL(-0.96495267812511f / 8.0), FL2FXCONST_SGL(-0.53960305946511f / 8.0) },
+  { FL2FXCONST_SGL( 0.55526940659947f / 8.0), FL2FXCONST_SGL( 0.78891523734774f / 8.0) },
+  { FL2FXCONST_SGL( 0.73538215752630f / 8.0), FL2FXCONST_SGL( 0.96452072373404f / 8.0) },
+  { FL2FXCONST_SGL(-0.30889773919437f / 8.0), FL2FXCONST_SGL(-0.80664389776860f / 8.0) },
+  { FL2FXCONST_SGL( 0.03574995626194f / 8.0), FL2FXCONST_SGL(-0.97325616900959f / 8.0) },
+  { FL2FXCONST_SGL( 0.98720684660488f / 8.0), FL2FXCONST_SGL( 0.48409133691962f / 8.0) },
+  { FL2FXCONST_SGL(-0.81689296271203f / 8.0), FL2FXCONST_SGL(-0.90827703628298f / 8.0) },
+  { FL2FXCONST_SGL( 0.67866860118215f / 8.0), FL2FXCONST_SGL( 0.81284503870856f / 8.0) },
+  { FL2FXCONST_SGL(-0.15808569732583f / 8.0), FL2FXCONST_SGL( 0.85279555024382f / 8.0) },
+  { FL2FXCONST_SGL( 0.80723395114371f / 8.0), FL2FXCONST_SGL(-0.24717418514605f / 8.0) },
+  { FL2FXCONST_SGL( 0.47788757329038f / 8.0), FL2FXCONST_SGL(-0.46333147839295f / 8.0) },
+  { FL2FXCONST_SGL( 0.96367554763201f / 8.0), FL2FXCONST_SGL( 0.38486749303242f / 8.0) },
+  { FL2FXCONST_SGL(-0.99143875716818f / 8.0), FL2FXCONST_SGL(-0.24945277239809f / 8.0) },
+  { FL2FXCONST_SGL( 0.83081876925833f / 8.0), FL2FXCONST_SGL(-0.94780851414763f / 8.0) },
+  { FL2FXCONST_SGL(-0.58753191905341f / 8.0), FL2FXCONST_SGL( 0.01290772389163f / 8.0) },
+  { FL2FXCONST_SGL( 0.95538108220960f / 8.0), FL2FXCONST_SGL(-0.85557052096538f / 8.0) },
+  { FL2FXCONST_SGL(-0.96490920476211f / 8.0), FL2FXCONST_SGL(-0.64020970923102f / 8.0) },
+  { FL2FXCONST_SGL(-0.97327101028521f / 8.0), FL2FXCONST_SGL( 0.12378128133110f / 8.0) },
+  { FL2FXCONST_SGL( 0.91400366022124f / 8.0), FL2FXCONST_SGL( 0.57972471346930f / 8.0) },
+  { FL2FXCONST_SGL(-0.99925837363824f / 8.0), FL2FXCONST_SGL( 0.71084847864067f / 8.0) },
+  { FL2FXCONST_SGL(-0.86875903507313f / 8.0), FL2FXCONST_SGL(-0.20291699203564f / 8.0) },
+  { FL2FXCONST_SGL(-0.26240034795124f / 8.0), FL2FXCONST_SGL(-0.68264554369108f / 8.0) },
+  { FL2FXCONST_SGL(-0.24664412953388f / 8.0), FL2FXCONST_SGL(-0.87642273115183f / 8.0) },
+  { FL2FXCONST_SGL( 0.02416275806869f / 8.0), FL2FXCONST_SGL( 0.27192914288905f / 8.0) },
+  { FL2FXCONST_SGL( 0.82068619590515f / 8.0), FL2FXCONST_SGL(-0.85087787994476f / 8.0) },
+  { FL2FXCONST_SGL( 0.88547373760759f / 8.0), FL2FXCONST_SGL(-0.89636802901469f / 8.0) },
+  { FL2FXCONST_SGL(-0.18173078152226f / 8.0), FL2FXCONST_SGL(-0.26152145156800f / 8.0) },
+  { FL2FXCONST_SGL( 0.09355476558534f / 8.0), FL2FXCONST_SGL( 0.54845123045604f / 8.0) },
+  { FL2FXCONST_SGL(-0.54668414224090f / 8.0), FL2FXCONST_SGL( 0.95980774020221f / 8.0) },
+  { FL2FXCONST_SGL( 0.37050990604091f / 8.0), FL2FXCONST_SGL(-0.59910140383171f / 8.0) },
+  { FL2FXCONST_SGL(-0.70373594262891f / 8.0), FL2FXCONST_SGL( 0.91227665827081f / 8.0) },
+  { FL2FXCONST_SGL(-0.34600785879594f / 8.0), FL2FXCONST_SGL(-0.99441426144200f / 8.0) },
+  { FL2FXCONST_SGL(-0.68774481731008f / 8.0), FL2FXCONST_SGL(-0.30238837956299f / 8.0) },
+  { FL2FXCONST_SGL(-0.26843291251234f / 8.0), FL2FXCONST_SGL( 0.83115668004362f / 8.0) },
+  { FL2FXCONST_SGL( 0.49072334613242f / 8.0), FL2FXCONST_SGL(-0.45359708737775f / 8.0) },
+  { FL2FXCONST_SGL( 0.38975993093975f / 8.0), FL2FXCONST_SGL( 0.95515358099121f / 8.0) },
+  { FL2FXCONST_SGL(-0.97757125224150f / 8.0), FL2FXCONST_SGL( 0.05305894580606f / 8.0) },
+  { FL2FXCONST_SGL(-0.17325552859616f / 8.0), FL2FXCONST_SGL(-0.92770672250494f / 8.0) },
+  { FL2FXCONST_SGL( 0.99948035025744f / 8.0), FL2FXCONST_SGL( 0.58285545563426f / 8.0) },
+  { FL2FXCONST_SGL(-0.64946246527458f / 8.0), FL2FXCONST_SGL( 0.68645507104960f / 8.0) },
+  { FL2FXCONST_SGL(-0.12016920576437f / 8.0), FL2FXCONST_SGL(-0.57147322153312f / 8.0) },
+  { FL2FXCONST_SGL(-0.58947456517751f / 8.0), FL2FXCONST_SGL(-0.34847132454388f / 8.0) },
+  { FL2FXCONST_SGL(-0.41815140454465f / 8.0), FL2FXCONST_SGL( 0.16276422358861f / 8.0) },
+  { FL2FXCONST_SGL( 0.99885650204884f / 8.0), FL2FXCONST_SGL( 0.11136095490444f / 8.0) },
+  { FL2FXCONST_SGL(-0.56649614128386f / 8.0), FL2FXCONST_SGL(-0.90494866361587f / 8.0) },
+  { FL2FXCONST_SGL( 0.94138021032330f / 8.0), FL2FXCONST_SGL( 0.35281916733018f / 8.0) },
+  { FL2FXCONST_SGL(-0.75725076534641f / 8.0), FL2FXCONST_SGL( 0.53650549640587f / 8.0) },
+  { FL2FXCONST_SGL( 0.20541973692630f / 8.0), FL2FXCONST_SGL(-0.94435144369918f / 8.0) },
+  { FL2FXCONST_SGL( 0.99980371023351f / 8.0), FL2FXCONST_SGL( 0.79835913565599f / 8.0) },
+  { FL2FXCONST_SGL( 0.29078277605775f / 8.0), FL2FXCONST_SGL( 0.35393777921520f / 8.0) },
+  { FL2FXCONST_SGL(-0.62858772103030f / 8.0), FL2FXCONST_SGL( 0.38765693387102f / 8.0) },
+  { FL2FXCONST_SGL( 0.43440904467688f / 8.0), FL2FXCONST_SGL(-0.98546330463232f / 8.0) },
+  { FL2FXCONST_SGL(-0.98298583762390f / 8.0), FL2FXCONST_SGL( 0.21021524625209f / 8.0) },
+  { FL2FXCONST_SGL( 0.19513029146934f / 8.0), FL2FXCONST_SGL(-0.94239832251867f / 8.0) },
+  { FL2FXCONST_SGL(-0.95476662400101f / 8.0), FL2FXCONST_SGL( 0.98364554179143f / 8.0) },
+  { FL2FXCONST_SGL( 0.93379635304810f / 8.0), FL2FXCONST_SGL(-0.70881994583682f / 8.0) },
+  { FL2FXCONST_SGL(-0.85235410573336f / 8.0), FL2FXCONST_SGL(-0.08342347966410f / 8.0) },
+  { FL2FXCONST_SGL(-0.86425093011245f / 8.0), FL2FXCONST_SGL(-0.45795025029466f / 8.0) },
+  { FL2FXCONST_SGL( 0.38879779059045f / 8.0), FL2FXCONST_SGL( 0.97274429344593f / 8.0) },
+  { FL2FXCONST_SGL( 0.92045124735495f / 8.0), FL2FXCONST_SGL(-0.62433652524220f / 8.0) },
+  { FL2FXCONST_SGL( 0.89162532251878f / 8.0), FL2FXCONST_SGL( 0.54950955570563f / 8.0) },
+  { FL2FXCONST_SGL(-0.36834336949252f / 8.0), FL2FXCONST_SGL( 0.96458298020975f / 8.0) },
+  { FL2FXCONST_SGL( 0.93891760988045f / 8.0), FL2FXCONST_SGL(-0.89968353740388f / 8.0) },
+  { FL2FXCONST_SGL( 0.99267657565094f / 8.0), FL2FXCONST_SGL(-0.03757034316958f / 8.0) },
+  { FL2FXCONST_SGL(-0.94063471614176f / 8.0), FL2FXCONST_SGL( 0.41332338538963f / 8.0) },
+  { FL2FXCONST_SGL( 0.99740224117019f / 8.0), FL2FXCONST_SGL(-0.16830494996370f / 8.0) },
+  { FL2FXCONST_SGL(-0.35899413170555f / 8.0), FL2FXCONST_SGL(-0.46633226649613f / 8.0) },
+  { FL2FXCONST_SGL( 0.05237237274947f / 8.0), FL2FXCONST_SGL(-0.25640361602661f / 8.0) },
+  { FL2FXCONST_SGL( 0.36703583957424f / 8.0), FL2FXCONST_SGL(-0.38653265641875f / 8.0) },
+  { FL2FXCONST_SGL( 0.91653180367913f / 8.0), FL2FXCONST_SGL(-0.30587628726597f / 8.0) },
+  { FL2FXCONST_SGL( 0.69000803499316f / 8.0), FL2FXCONST_SGL( 0.90952171386132f / 8.0) },
+  { FL2FXCONST_SGL(-0.38658751133527f / 8.0), FL2FXCONST_SGL( 0.99501571208985f / 8.0) },
+  { FL2FXCONST_SGL(-0.29250814029851f / 8.0), FL2FXCONST_SGL( 0.37444994344615f / 8.0) },
+  { FL2FXCONST_SGL(-0.60182204677608f / 8.0), FL2FXCONST_SGL( 0.86779651036123f / 8.0) },
+  { FL2FXCONST_SGL(-0.97418588163217f / 8.0), FL2FXCONST_SGL( 0.96468523666475f / 8.0) },
+  { FL2FXCONST_SGL( 0.88461574003963f / 8.0), FL2FXCONST_SGL( 0.57508405276414f / 8.0) },
+  { FL2FXCONST_SGL( 0.05198933055162f / 8.0), FL2FXCONST_SGL( 0.21269661669964f / 8.0) },
+  { FL2FXCONST_SGL(-0.53499621979720f / 8.0), FL2FXCONST_SGL( 0.97241553731237f / 8.0) },
+  { FL2FXCONST_SGL(-0.49429560226497f / 8.0), FL2FXCONST_SGL( 0.98183865291903f / 8.0) },
+  { FL2FXCONST_SGL(-0.98935142339139f / 8.0), FL2FXCONST_SGL(-0.40249159006933f / 8.0) },
+  { FL2FXCONST_SGL(-0.98081380091130f / 8.0), FL2FXCONST_SGL(-0.72856895534041f / 8.0) },
+  { FL2FXCONST_SGL(-0.27338148835532f / 8.0), FL2FXCONST_SGL( 0.99950922447209f / 8.0) },
+  { FL2FXCONST_SGL( 0.06310802338302f / 8.0), FL2FXCONST_SGL(-0.54539587529618f / 8.0) },
+  { FL2FXCONST_SGL(-0.20461677199539f / 8.0), FL2FXCONST_SGL(-0.14209977628489f / 8.0) },
+  { FL2FXCONST_SGL( 0.66223843141647f / 8.0), FL2FXCONST_SGL( 0.72528579940326f / 8.0) },
+  { FL2FXCONST_SGL(-0.84764345483665f / 8.0), FL2FXCONST_SGL( 0.02372316801261f / 8.0) },
+  { FL2FXCONST_SGL(-0.89039863483811f / 8.0), FL2FXCONST_SGL( 0.88866581484602f / 8.0) },
+  { FL2FXCONST_SGL( 0.95903308477986f / 8.0), FL2FXCONST_SGL( 0.76744927173873f / 8.0) },
+  { FL2FXCONST_SGL( 0.73504123909879f / 8.0), FL2FXCONST_SGL(-0.03747203173192f / 8.0) },
+  { FL2FXCONST_SGL(-0.31744434966056f / 8.0), FL2FXCONST_SGL(-0.36834111883652f / 8.0) },
+  { FL2FXCONST_SGL(-0.34110827591623f / 8.0), FL2FXCONST_SGL( 0.40211222807691f / 8.0) },
+  { FL2FXCONST_SGL( 0.47803883714199f / 8.0), FL2FXCONST_SGL(-0.39423219786288f / 8.0) },
+  { FL2FXCONST_SGL( 0.98299195879514f / 8.0), FL2FXCONST_SGL( 0.01989791390047f / 8.0) },
+  { FL2FXCONST_SGL(-0.30963073129751f / 8.0), FL2FXCONST_SGL(-0.18076720599336f / 8.0) },
+  { FL2FXCONST_SGL( 0.99992588229018f / 8.0), FL2FXCONST_SGL(-0.26281872094289f / 8.0) },
+  { FL2FXCONST_SGL(-0.93149731080767f / 8.0), FL2FXCONST_SGL(-0.98313162570490f / 8.0) },
+  { FL2FXCONST_SGL( 0.99923472302773f / 8.0), FL2FXCONST_SGL(-0.80142993767554f / 8.0) },
+  { FL2FXCONST_SGL(-0.26024169633417f / 8.0), FL2FXCONST_SGL(-0.75999759855752f / 8.0) },
+  { FL2FXCONST_SGL(-0.35712514743563f / 8.0), FL2FXCONST_SGL( 0.19298963768574f / 8.0) },
+  { FL2FXCONST_SGL(-0.99899084509530f / 8.0), FL2FXCONST_SGL( 0.74645156992493f / 8.0) },
+  { FL2FXCONST_SGL( 0.86557171579452f / 8.0), FL2FXCONST_SGL( 0.55593866696299f / 8.0) },
+  { FL2FXCONST_SGL( 0.33408042438752f / 8.0), FL2FXCONST_SGL( 0.86185953874709f / 8.0) },
+  { FL2FXCONST_SGL( 0.99010736374716f / 8.0), FL2FXCONST_SGL( 0.04602397576623f / 8.0) },
+  { FL2FXCONST_SGL(-0.66694269691195f / 8.0), FL2FXCONST_SGL(-0.91643611810148f / 8.0) },
+  { FL2FXCONST_SGL( 0.64016792079480f / 8.0), FL2FXCONST_SGL( 0.15649530836856f / 8.0) },
+  { FL2FXCONST_SGL( 0.99570534804836f / 8.0), FL2FXCONST_SGL( 0.45844586038111f / 8.0) },
+  { FL2FXCONST_SGL(-0.63431466947340f / 8.0), FL2FXCONST_SGL( 0.21079116459234f / 8.0) },
+  { FL2FXCONST_SGL(-0.07706847005931f / 8.0), FL2FXCONST_SGL(-0.89581437101329f / 8.0) },
+  { FL2FXCONST_SGL( 0.98590090577724f / 8.0), FL2FXCONST_SGL( 0.88241721133981f / 8.0) },
+  { FL2FXCONST_SGL( 0.80099335254678f / 8.0), FL2FXCONST_SGL(-0.36851896710853f / 8.0) },
+  { FL2FXCONST_SGL( 0.78368131392666f / 8.0), FL2FXCONST_SGL( 0.45506999802597f / 8.0) },
+  { FL2FXCONST_SGL( 0.08707806671691f / 8.0), FL2FXCONST_SGL( 0.80938994918745f / 8.0) },
+  { FL2FXCONST_SGL(-0.86811883080712f / 8.0), FL2FXCONST_SGL( 0.39347308654705f / 8.0) },
+  { FL2FXCONST_SGL(-0.39466529740375f / 8.0), FL2FXCONST_SGL(-0.66809432114456f / 8.0) },
+  { FL2FXCONST_SGL( 0.97875325649683f / 8.0), FL2FXCONST_SGL(-0.72467840967746f / 8.0) },
+  { FL2FXCONST_SGL(-0.95038560288864f / 8.0), FL2FXCONST_SGL( 0.89563219587625f / 8.0) },
+  { FL2FXCONST_SGL( 0.17005239424212f / 8.0), FL2FXCONST_SGL( 0.54683053962658f / 8.0) },
+  { FL2FXCONST_SGL(-0.76910792026848f / 8.0), FL2FXCONST_SGL(-0.96226617549298f / 8.0) },
+  { FL2FXCONST_SGL( 0.99743281016846f / 8.0), FL2FXCONST_SGL( 0.42697157037567f / 8.0) },
+  { FL2FXCONST_SGL( 0.95437383549973f / 8.0), FL2FXCONST_SGL( 0.97002324109952f / 8.0) },
+  { FL2FXCONST_SGL( 0.99578905365569f / 8.0), FL2FXCONST_SGL(-0.54106826257356f / 8.0) },
+  { FL2FXCONST_SGL( 0.28058259829990f / 8.0), FL2FXCONST_SGL(-0.85361420634036f / 8.0) },
+  { FL2FXCONST_SGL( 0.85256524470573f / 8.0), FL2FXCONST_SGL(-0.64567607735589f / 8.0) },
+  { FL2FXCONST_SGL(-0.50608540105128f / 8.0), FL2FXCONST_SGL(-0.65846015480300f / 8.0) },
+  { FL2FXCONST_SGL(-0.97210735183243f / 8.0), FL2FXCONST_SGL(-0.23095213067791f / 8.0) },
+  { FL2FXCONST_SGL( 0.95424048234441f / 8.0), FL2FXCONST_SGL(-0.99240147091219f / 8.0) },
+  { FL2FXCONST_SGL(-0.96926570524023f / 8.0), FL2FXCONST_SGL( 0.73775654896574f / 8.0) },
+  { FL2FXCONST_SGL( 0.30872163214726f / 8.0), FL2FXCONST_SGL( 0.41514960556126f / 8.0) },
+  { FL2FXCONST_SGL(-0.24523839572639f / 8.0), FL2FXCONST_SGL( 0.63206633394807f / 8.0) },
+  { FL2FXCONST_SGL(-0.33813265086024f / 8.0), FL2FXCONST_SGL(-0.38661779441897f / 8.0) },
+  { FL2FXCONST_SGL(-0.05826828420146f / 8.0), FL2FXCONST_SGL(-0.06940774188029f / 8.0) },
+  { FL2FXCONST_SGL(-0.22898461455054f / 8.0), FL2FXCONST_SGL( 0.97054853316316f / 8.0) },
+  { FL2FXCONST_SGL(-0.18509915019881f / 8.0), FL2FXCONST_SGL( 0.47565762892084f / 8.0) },
+  { FL2FXCONST_SGL(-0.10488238045009f / 8.0), FL2FXCONST_SGL(-0.87769947402394f / 8.0) },
+  { FL2FXCONST_SGL(-0.71886586182037f / 8.0), FL2FXCONST_SGL( 0.78030982480538f / 8.0) },
+  { FL2FXCONST_SGL( 0.99793873738654f / 8.0), FL2FXCONST_SGL( 0.90041310491497f / 8.0) },
+  { FL2FXCONST_SGL( 0.57563307626120f / 8.0), FL2FXCONST_SGL(-0.91034337352097f / 8.0) },
+  { FL2FXCONST_SGL( 0.28909646383717f / 8.0), FL2FXCONST_SGL( 0.96307783970534f / 8.0) },
+  { FL2FXCONST_SGL( 0.42188998312520f / 8.0), FL2FXCONST_SGL( 0.48148651230437f / 8.0) },
+  { FL2FXCONST_SGL( 0.93335049681047f / 8.0), FL2FXCONST_SGL(-0.43537023883588f / 8.0) },
+  { FL2FXCONST_SGL(-0.97087374418267f / 8.0), FL2FXCONST_SGL( 0.86636445711364f / 8.0) },
+  { FL2FXCONST_SGL( 0.36722871286923f / 8.0), FL2FXCONST_SGL( 0.65291654172961f / 8.0) },
+  { FL2FXCONST_SGL(-0.81093025665696f / 8.0), FL2FXCONST_SGL( 0.08778370229363f / 8.0) },
+  { FL2FXCONST_SGL(-0.26240603062237f / 8.0), FL2FXCONST_SGL(-0.92774095379098f / 8.0) },
+  { FL2FXCONST_SGL( 0.83996497984604f / 8.0), FL2FXCONST_SGL( 0.55839849139647f / 8.0) },
+  { FL2FXCONST_SGL(-0.99909615720225f / 8.0), FL2FXCONST_SGL(-0.96024605713970f / 8.0) },
+  { FL2FXCONST_SGL( 0.74649464155061f / 8.0), FL2FXCONST_SGL( 0.12144893606462f / 8.0) },
+  { FL2FXCONST_SGL(-0.74774595569805f / 8.0), FL2FXCONST_SGL(-0.26898062008959f / 8.0) },
+  { FL2FXCONST_SGL( 0.95781667469567f / 8.0), FL2FXCONST_SGL(-0.79047927052628f / 8.0) },
+  { FL2FXCONST_SGL( 0.95472308713099f / 8.0), FL2FXCONST_SGL(-0.08588776019550f / 8.0) },
+  { FL2FXCONST_SGL( 0.48708332746299f / 8.0), FL2FXCONST_SGL( 0.99999041579432f / 8.0) },
+  { FL2FXCONST_SGL( 0.46332038247497f / 8.0), FL2FXCONST_SGL( 0.10964126185063f / 8.0) },
+  { FL2FXCONST_SGL(-0.76497004940162f / 8.0), FL2FXCONST_SGL( 0.89210929242238f / 8.0) },
+  { FL2FXCONST_SGL( 0.57397389364339f / 8.0), FL2FXCONST_SGL( 0.35289703373760f / 8.0) },
+  { FL2FXCONST_SGL( 0.75374316974495f / 8.0), FL2FXCONST_SGL( 0.96705214651335f / 8.0) },
+  { FL2FXCONST_SGL(-0.59174397685714f / 8.0), FL2FXCONST_SGL(-0.89405370422752f / 8.0) },
+  { FL2FXCONST_SGL( 0.75087906691890f / 8.0), FL2FXCONST_SGL(-0.29612672982396f / 8.0) },
+  { FL2FXCONST_SGL(-0.98607857336230f / 8.0), FL2FXCONST_SGL( 0.25034911730023f / 8.0) },
+  { FL2FXCONST_SGL(-0.40761056640505f / 8.0), FL2FXCONST_SGL(-0.90045573444695f / 8.0) },
+  { FL2FXCONST_SGL( 0.66929266740477f / 8.0), FL2FXCONST_SGL( 0.98629493401748f / 8.0) },
+  { FL2FXCONST_SGL(-0.97463695257310f / 8.0), FL2FXCONST_SGL(-0.00190223301301f / 8.0) },
+  { FL2FXCONST_SGL( 0.90145509409859f / 8.0), FL2FXCONST_SGL( 0.99781390365446f / 8.0) },
+  { FL2FXCONST_SGL(-0.87259289048043f / 8.0), FL2FXCONST_SGL( 0.99233587353666f / 8.0) },
+  { FL2FXCONST_SGL(-0.91529461447692f / 8.0), FL2FXCONST_SGL(-0.15698707534206f / 8.0) },
+  { FL2FXCONST_SGL(-0.03305738840705f / 8.0), FL2FXCONST_SGL(-0.37205262859764f / 8.0) },
+  { FL2FXCONST_SGL( 0.07223051368337f / 8.0), FL2FXCONST_SGL(-0.88805001733626f / 8.0) },
+  { FL2FXCONST_SGL( 0.99498012188353f / 8.0), FL2FXCONST_SGL( 0.97094358113387f / 8.0) },
+  { FL2FXCONST_SGL(-0.74904939500519f / 8.0), FL2FXCONST_SGL( 0.99985483641521f / 8.0) },
+  { FL2FXCONST_SGL( 0.04585228574211f / 8.0), FL2FXCONST_SGL( 0.99812337444082f / 8.0) },
+  { FL2FXCONST_SGL(-0.89054954257993f / 8.0), FL2FXCONST_SGL(-0.31791913188064f / 8.0) },
+  { FL2FXCONST_SGL(-0.83782144651251f / 8.0), FL2FXCONST_SGL( 0.97637632547466f / 8.0) },
+  { FL2FXCONST_SGL( 0.33454804933804f / 8.0), FL2FXCONST_SGL(-0.86231516800408f / 8.0) },
+  { FL2FXCONST_SGL(-0.99707579362824f / 8.0), FL2FXCONST_SGL( 0.93237990079441f / 8.0) },
+  { FL2FXCONST_SGL(-0.22827527843994f / 8.0), FL2FXCONST_SGL( 0.18874759397997f / 8.0) },
+  { FL2FXCONST_SGL( 0.67248046289143f / 8.0), FL2FXCONST_SGL(-0.03646211390569f / 8.0) },
+  { FL2FXCONST_SGL(-0.05146538187944f / 8.0), FL2FXCONST_SGL(-0.92599700120679f / 8.0) },
+  { FL2FXCONST_SGL( 0.99947295749905f / 8.0), FL2FXCONST_SGL( 0.93625229707912f / 8.0) },
+  { FL2FXCONST_SGL( 0.66951124390363f / 8.0), FL2FXCONST_SGL( 0.98905825623893f / 8.0) },
+  { FL2FXCONST_SGL(-0.99602956559179f / 8.0), FL2FXCONST_SGL(-0.44654715757688f / 8.0) },
+  { FL2FXCONST_SGL( 0.82104905483590f / 8.0), FL2FXCONST_SGL( 0.99540741724928f / 8.0) },
+  { FL2FXCONST_SGL( 0.99186510988782f / 8.0), FL2FXCONST_SGL( 0.72023001312947f / 8.0) },
+  { FL2FXCONST_SGL(-0.65284592392918f / 8.0), FL2FXCONST_SGL( 0.52186723253637f / 8.0) },
+  { FL2FXCONST_SGL( 0.93885443798188f / 8.0), FL2FXCONST_SGL(-0.74895312615259f / 8.0) },
+  { FL2FXCONST_SGL( 0.96735248738388f / 8.0), FL2FXCONST_SGL( 0.90891816978629f / 8.0) },
+  { FL2FXCONST_SGL(-0.22225968841114f / 8.0), FL2FXCONST_SGL( 0.57124029781228f / 8.0) },
+  { FL2FXCONST_SGL(-0.44132783753414f / 8.0), FL2FXCONST_SGL(-0.92688840659280f / 8.0) },
+  { FL2FXCONST_SGL(-0.85694974219574f / 8.0), FL2FXCONST_SGL( 0.88844532719844f / 8.0) },
+  { FL2FXCONST_SGL( 0.91783042091762f / 8.0), FL2FXCONST_SGL(-0.46356892383970f / 8.0) },
+  { FL2FXCONST_SGL( 0.72556974415690f / 8.0), FL2FXCONST_SGL(-0.99899555770747f / 8.0) },
+  { FL2FXCONST_SGL(-0.99711581834508f / 8.0), FL2FXCONST_SGL( 0.58211560180426f / 8.0) },
+  { FL2FXCONST_SGL( 0.77638976371966f / 8.0), FL2FXCONST_SGL( 0.94321834873819f / 8.0) },
+  { FL2FXCONST_SGL( 0.07717324253925f / 8.0), FL2FXCONST_SGL( 0.58638399856595f / 8.0) },
+  { FL2FXCONST_SGL(-0.56049829194163f / 8.0), FL2FXCONST_SGL( 0.82522301569036f / 8.0) },
+  { FL2FXCONST_SGL( 0.98398893639988f / 8.0), FL2FXCONST_SGL( 0.39467440420569f / 8.0) },
+  { FL2FXCONST_SGL( 0.47546946844938f / 8.0), FL2FXCONST_SGL( 0.68613044836811f / 8.0) },
+  { FL2FXCONST_SGL( 0.65675089314631f / 8.0), FL2FXCONST_SGL( 0.18331637134880f / 8.0) },
+  { FL2FXCONST_SGL( 0.03273375457980f / 8.0), FL2FXCONST_SGL(-0.74933109564108f / 8.0) },
+  { FL2FXCONST_SGL(-0.38684144784738f / 8.0), FL2FXCONST_SGL( 0.51337349030406f / 8.0) },
+  { FL2FXCONST_SGL(-0.97346267944545f / 8.0), FL2FXCONST_SGL(-0.96549364384098f / 8.0) },
+  { FL2FXCONST_SGL(-0.53282156061942f / 8.0), FL2FXCONST_SGL(-0.91423265091354f / 8.0) },
+  { FL2FXCONST_SGL( 0.99817310731176f / 8.0), FL2FXCONST_SGL( 0.61133572482148f / 8.0) },
+  { FL2FXCONST_SGL(-0.50254500772635f / 8.0), FL2FXCONST_SGL(-0.88829338134294f / 8.0) },
+  { FL2FXCONST_SGL( 0.01995873238855f / 8.0), FL2FXCONST_SGL( 0.85223515096765f / 8.0) },
+  { FL2FXCONST_SGL( 0.99930381973804f / 8.0), FL2FXCONST_SGL( 0.94578896296649f / 8.0) },
+  { FL2FXCONST_SGL( 0.82907767600783f / 8.0), FL2FXCONST_SGL(-0.06323442598128f / 8.0) },
+  { FL2FXCONST_SGL(-0.58660709669728f / 8.0), FL2FXCONST_SGL( 0.96840773806582f / 8.0) },
+  { FL2FXCONST_SGL(-0.17573736667267f / 8.0), FL2FXCONST_SGL(-0.48166920859485f / 8.0) },
+  { FL2FXCONST_SGL( 0.83434292401346f / 8.0), FL2FXCONST_SGL(-0.13023450646997f / 8.0) },
+  { FL2FXCONST_SGL( 0.05946491307025f / 8.0), FL2FXCONST_SGL( 0.20511047074866f / 8.0) },
+  { FL2FXCONST_SGL( 0.81505484574602f / 8.0), FL2FXCONST_SGL(-0.94685947861369f / 8.0) },
+  { FL2FXCONST_SGL(-0.44976380954860f / 8.0), FL2FXCONST_SGL( 0.40894572671545f / 8.0) },
+  { FL2FXCONST_SGL(-0.89746474625671f / 8.0), FL2FXCONST_SGL( 0.99846578838537f / 8.0) },
+  { FL2FXCONST_SGL( 0.39677256130792f / 8.0), FL2FXCONST_SGL(-0.74854668609359f / 8.0) },
+  { FL2FXCONST_SGL(-0.07588948563079f / 8.0), FL2FXCONST_SGL( 0.74096214084170f / 8.0) },
+  { FL2FXCONST_SGL( 0.76343198951445f / 8.0), FL2FXCONST_SGL( 0.41746629422634f / 8.0) },
+  { FL2FXCONST_SGL(-0.74490104699626f / 8.0), FL2FXCONST_SGL( 0.94725911744610f / 8.0) },
+  { FL2FXCONST_SGL( 0.64880119792759f / 8.0), FL2FXCONST_SGL( 0.41336660830571f / 8.0) },
+  { FL2FXCONST_SGL( 0.62319537462542f / 8.0), FL2FXCONST_SGL(-0.93098313552599f / 8.0) },
+  { FL2FXCONST_SGL( 0.42215817594807f / 8.0), FL2FXCONST_SGL(-0.07712787385208f / 8.0) },
+  { FL2FXCONST_SGL( 0.02704554141885f / 8.0), FL2FXCONST_SGL(-0.05417518053666f / 8.0) },
+  { FL2FXCONST_SGL( 0.80001773566818f / 8.0), FL2FXCONST_SGL( 0.91542195141039f / 8.0) },
+  { FL2FXCONST_SGL(-0.79351832348816f / 8.0), FL2FXCONST_SGL(-0.36208897989136f / 8.0) },
+  { FL2FXCONST_SGL( 0.63872359151636f / 8.0), FL2FXCONST_SGL( 0.08128252493444f / 8.0) },
+  { FL2FXCONST_SGL( 0.52890520960295f / 8.0), FL2FXCONST_SGL( 0.60048872455592f / 8.0) },
+  { FL2FXCONST_SGL( 0.74238552914587f / 8.0), FL2FXCONST_SGL( 0.04491915291044f / 8.0) },
+  { FL2FXCONST_SGL( 0.99096131449250f / 8.0), FL2FXCONST_SGL(-0.19451182854402f / 8.0) },
+  { FL2FXCONST_SGL(-0.80412329643109f / 8.0), FL2FXCONST_SGL(-0.88513818199457f / 8.0) },
+  { FL2FXCONST_SGL(-0.64612616129736f / 8.0), FL2FXCONST_SGL( 0.72198674804544f / 8.0) },
+  { FL2FXCONST_SGL( 0.11657770663191f / 8.0), FL2FXCONST_SGL(-0.83662833815041f / 8.0) },
+  { FL2FXCONST_SGL(-0.95053182488101f / 8.0), FL2FXCONST_SGL(-0.96939905138082f / 8.0) },
+  { FL2FXCONST_SGL(-0.62228872928622f / 8.0), FL2FXCONST_SGL( 0.82767262846661f / 8.0) },
+  { FL2FXCONST_SGL( 0.03004475787316f / 8.0), FL2FXCONST_SGL(-0.99738896333384f / 8.0) },
+  { FL2FXCONST_SGL(-0.97987214341034f / 8.0), FL2FXCONST_SGL( 0.36526129686425f / 8.0) },
+  { FL2FXCONST_SGL(-0.99986980746200f / 8.0), FL2FXCONST_SGL(-0.36021610299715f / 8.0) },
+  { FL2FXCONST_SGL( 0.89110648599879f / 8.0), FL2FXCONST_SGL(-0.97894250343044f / 8.0) },
+  { FL2FXCONST_SGL( 0.10407960510582f / 8.0), FL2FXCONST_SGL( 0.77357793811619f / 8.0) },
+  { FL2FXCONST_SGL( 0.95964737821728f / 8.0), FL2FXCONST_SGL(-0.35435818285502f / 8.0) },
+  { FL2FXCONST_SGL( 0.50843233159162f / 8.0), FL2FXCONST_SGL( 0.96107691266205f / 8.0) },
+  { FL2FXCONST_SGL( 0.17006334670615f / 8.0), FL2FXCONST_SGL(-0.76854025314829f / 8.0) },
+  { FL2FXCONST_SGL( 0.25872675063360f / 8.0), FL2FXCONST_SGL( 0.99893303933816f / 8.0) },
+  { FL2FXCONST_SGL(-0.01115998681937f / 8.0), FL2FXCONST_SGL( 0.98496019742444f / 8.0) },
+  { FL2FXCONST_SGL(-0.79598702973261f / 8.0), FL2FXCONST_SGL( 0.97138411318894f / 8.0) },
+  { FL2FXCONST_SGL(-0.99264708948101f / 8.0), FL2FXCONST_SGL(-0.99542822402536f / 8.0) },
+  { FL2FXCONST_SGL(-0.99829663752818f / 8.0), FL2FXCONST_SGL( 0.01877138824311f / 8.0) },
+  { FL2FXCONST_SGL(-0.70801016548184f / 8.0), FL2FXCONST_SGL( 0.33680685948117f / 8.0) },
+  { FL2FXCONST_SGL(-0.70467057786826f / 8.0), FL2FXCONST_SGL( 0.93272777501857f / 8.0) },
+  { FL2FXCONST_SGL( 0.99846021905254f / 8.0), FL2FXCONST_SGL(-0.98725746254433f / 8.0) },
+  { FL2FXCONST_SGL(-0.63364968534650f / 8.0), FL2FXCONST_SGL(-0.16473594423746f / 8.0) },
+  { FL2FXCONST_SGL(-0.16258217500792f / 8.0), FL2FXCONST_SGL(-0.95939125400802f / 8.0) },
+  { FL2FXCONST_SGL(-0.43645594360633f / 8.0), FL2FXCONST_SGL(-0.94805030113284f / 8.0) },
+  { FL2FXCONST_SGL(-0.99848471702976f / 8.0), FL2FXCONST_SGL( 0.96245166923809f / 8.0) },
+  { FL2FXCONST_SGL(-0.16796458968998f / 8.0), FL2FXCONST_SGL(-0.98987511890470f / 8.0) },
+  { FL2FXCONST_SGL(-0.87979225745213f / 8.0), FL2FXCONST_SGL(-0.71725725041680f / 8.0) },
+  { FL2FXCONST_SGL( 0.44183099021786f / 8.0), FL2FXCONST_SGL(-0.93568974498761f / 8.0) },
+  { FL2FXCONST_SGL( 0.93310180125532f / 8.0), FL2FXCONST_SGL(-0.99913308068246f / 8.0) },
+  { FL2FXCONST_SGL(-0.93941931782002f / 8.0), FL2FXCONST_SGL(-0.56409379640356f / 8.0) },
+  { FL2FXCONST_SGL(-0.88590003188677f / 8.0), FL2FXCONST_SGL( 0.47624600491382f / 8.0) },
+  { FL2FXCONST_SGL( 0.99971463703691f / 8.0), FL2FXCONST_SGL(-0.83889954253462f / 8.0) },
+  { FL2FXCONST_SGL(-0.75376385639978f / 8.0), FL2FXCONST_SGL( 0.00814643438625f / 8.0) },
+  { FL2FXCONST_SGL( 0.93887685615875f / 8.0), FL2FXCONST_SGL(-0.11284528204636f / 8.0) },
+  { FL2FXCONST_SGL( 0.85126435782309f / 8.0), FL2FXCONST_SGL( 0.52349251543547f / 8.0) },
+  { FL2FXCONST_SGL( 0.39701421446381f / 8.0), FL2FXCONST_SGL( 0.81779634174316f / 8.0) },
+  { FL2FXCONST_SGL(-0.37024464187437f / 8.0), FL2FXCONST_SGL(-0.87071656222959f / 8.0) },
+  { FL2FXCONST_SGL(-0.36024828242896f / 8.0), FL2FXCONST_SGL( 0.34655735648287f / 8.0) },
+  { FL2FXCONST_SGL(-0.93388812549209f / 8.0), FL2FXCONST_SGL(-0.84476541096429f / 8.0) },
+  { FL2FXCONST_SGL(-0.65298804552119f / 8.0), FL2FXCONST_SGL(-0.18439575450921f / 8.0) },
+  { FL2FXCONST_SGL( 0.11960319006843f / 8.0), FL2FXCONST_SGL( 0.99899346780168f / 8.0) },
+  { FL2FXCONST_SGL( 0.94292565553160f / 8.0), FL2FXCONST_SGL( 0.83163906518293f / 8.0) },
+  { FL2FXCONST_SGL( 0.75081145286948f / 8.0), FL2FXCONST_SGL(-0.35533223142265f / 8.0) },
+  { FL2FXCONST_SGL( 0.56721979748394f / 8.0), FL2FXCONST_SGL(-0.24076836414499f / 8.0) },
+  { FL2FXCONST_SGL( 0.46857766746029f / 8.0), FL2FXCONST_SGL(-0.30140233457198f / 8.0) },
+  { FL2FXCONST_SGL( 0.97312313923635f / 8.0), FL2FXCONST_SGL(-0.99548191630031f / 8.0) },
+  { FL2FXCONST_SGL(-0.38299976567017f / 8.0), FL2FXCONST_SGL( 0.98516909715427f / 8.0) },
+  { FL2FXCONST_SGL( 0.41025800019463f / 8.0), FL2FXCONST_SGL( 0.02116736935734f / 8.0) },
+  { FL2FXCONST_SGL( 0.09638062008048f / 8.0), FL2FXCONST_SGL( 0.04411984381457f / 8.0) },
+  { FL2FXCONST_SGL(-0.85283249275397f / 8.0), FL2FXCONST_SGL( 0.91475563922421f / 8.0) },
+  { FL2FXCONST_SGL( 0.88866808958124f / 8.0), FL2FXCONST_SGL(-0.99735267083226f / 8.0) },
+  { FL2FXCONST_SGL(-0.48202429536989f / 8.0), FL2FXCONST_SGL(-0.96805608884164f / 8.0) },
+  { FL2FXCONST_SGL( 0.27572582416567f / 8.0), FL2FXCONST_SGL( 0.58634753335832f / 8.0) },
+  { FL2FXCONST_SGL(-0.65889129659168f / 8.0), FL2FXCONST_SGL( 0.58835634138583f / 8.0) },
+  { FL2FXCONST_SGL( 0.98838086953732f / 8.0), FL2FXCONST_SGL( 0.99994349600236f / 8.0) },
+  { FL2FXCONST_SGL(-0.20651349620689f / 8.0), FL2FXCONST_SGL( 0.54593044066355f / 8.0) },
+  { FL2FXCONST_SGL(-0.62126416356920f / 8.0), FL2FXCONST_SGL(-0.59893681700392f / 8.0) },
+  { FL2FXCONST_SGL( 0.20320105410437f / 8.0), FL2FXCONST_SGL(-0.86879180355289f / 8.0) },
+  { FL2FXCONST_SGL(-0.97790548600584f / 8.0), FL2FXCONST_SGL( 0.96290806999242f / 8.0) },
+  { FL2FXCONST_SGL( 0.11112534735126f / 8.0), FL2FXCONST_SGL( 0.21484763313301f / 8.0) },
+  { FL2FXCONST_SGL(-0.41368337314182f / 8.0), FL2FXCONST_SGL( 0.28216837680365f / 8.0) },
+  { FL2FXCONST_SGL( 0.24133038992960f / 8.0), FL2FXCONST_SGL( 0.51294362630238f / 8.0) },
+  { FL2FXCONST_SGL(-0.66393410674885f / 8.0), FL2FXCONST_SGL(-0.08249679629081f / 8.0) },
+  { FL2FXCONST_SGL(-0.53697829178752f / 8.0), FL2FXCONST_SGL(-0.97649903936228f / 8.0) },
+  { FL2FXCONST_SGL(-0.97224737889348f / 8.0), FL2FXCONST_SGL( 0.22081333579837f / 8.0) },
+  { FL2FXCONST_SGL( 0.87392477144549f / 8.0), FL2FXCONST_SGL(-0.12796173740361f / 8.0) },
+  { FL2FXCONST_SGL( 0.19050361015753f / 8.0), FL2FXCONST_SGL( 0.01602615387195f / 8.0) },
+  { FL2FXCONST_SGL(-0.46353441212724f / 8.0), FL2FXCONST_SGL(-0.95249041539006f / 8.0) },
+  { FL2FXCONST_SGL(-0.07064096339021f / 8.0), FL2FXCONST_SGL(-0.94479803205886f / 8.0) },
+  { FL2FXCONST_SGL(-0.92444085484466f / 8.0), FL2FXCONST_SGL(-0.10457590187436f / 8.0) },
+  { FL2FXCONST_SGL(-0.83822593578728f / 8.0), FL2FXCONST_SGL(-0.01695043208885f / 8.0) },
+  { FL2FXCONST_SGL( 0.75214681811150f / 8.0), FL2FXCONST_SGL(-0.99955681042665f / 8.0) },
+  { FL2FXCONST_SGL(-0.42102998829339f / 8.0), FL2FXCONST_SGL( 0.99720941999394f / 8.0) },
+  { FL2FXCONST_SGL(-0.72094786237696f / 8.0), FL2FXCONST_SGL(-0.35008961934255f / 8.0) },
+  { FL2FXCONST_SGL( 0.78843311019251f / 8.0), FL2FXCONST_SGL( 0.52851398958271f / 8.0) },
+  { FL2FXCONST_SGL( 0.97394027897442f / 8.0), FL2FXCONST_SGL(-0.26695944086561f / 8.0) },
+  { FL2FXCONST_SGL( 0.99206463477946f / 8.0), FL2FXCONST_SGL(-0.57010120849429f / 8.0) },
+  { FL2FXCONST_SGL( 0.76789609461795f / 8.0), FL2FXCONST_SGL(-0.76519356730966f / 8.0) },
+  { FL2FXCONST_SGL(-0.82002421836409f / 8.0), FL2FXCONST_SGL(-0.73530179553767f / 8.0) },
+  { FL2FXCONST_SGL( 0.81924990025724f / 8.0), FL2FXCONST_SGL( 0.99698425250579f / 8.0) },
+  { FL2FXCONST_SGL(-0.26719850873357f / 8.0), FL2FXCONST_SGL( 0.68903369776193f / 8.0) },
+  { FL2FXCONST_SGL(-0.43311260380975f / 8.0), FL2FXCONST_SGL( 0.85321815947490f / 8.0) },
+  { FL2FXCONST_SGL( 0.99194979673836f / 8.0), FL2FXCONST_SGL( 0.91876249766422f / 8.0) },
+  { FL2FXCONST_SGL(-0.80692001248487f / 8.0), FL2FXCONST_SGL(-0.32627540663214f / 8.0) },
+  { FL2FXCONST_SGL( 0.43080003649976f / 8.0), FL2FXCONST_SGL(-0.21919095636638f / 8.0) },
+  { FL2FXCONST_SGL( 0.67709491937357f / 8.0), FL2FXCONST_SGL(-0.95478075822906f / 8.0) },
+  { FL2FXCONST_SGL( 0.56151770568316f / 8.0), FL2FXCONST_SGL(-0.70693811747778f / 8.0) },
+  { FL2FXCONST_SGL( 0.10831862810749f / 8.0), FL2FXCONST_SGL(-0.08628837174592f / 8.0) },
+  { FL2FXCONST_SGL( 0.91229417540436f / 8.0), FL2FXCONST_SGL(-0.65987351408410f / 8.0) },
+  { FL2FXCONST_SGL(-0.48972893932274f / 8.0), FL2FXCONST_SGL( 0.56289246362686f / 8.0) },
+  { FL2FXCONST_SGL(-0.89033658689697f / 8.0), FL2FXCONST_SGL(-0.71656563987082f / 8.0) },
+  { FL2FXCONST_SGL( 0.65269447475094f / 8.0), FL2FXCONST_SGL( 0.65916004833932f / 8.0) },
+  { FL2FXCONST_SGL( 0.67439478141121f / 8.0), FL2FXCONST_SGL(-0.81684380846796f / 8.0) },
+  { FL2FXCONST_SGL(-0.47770832416973f / 8.0), FL2FXCONST_SGL(-0.16789556203025f / 8.0) },
+  { FL2FXCONST_SGL(-0.99715979260878f / 8.0), FL2FXCONST_SGL(-0.93565784007648f / 8.0) },
+  { FL2FXCONST_SGL(-0.90889593602546f / 8.0), FL2FXCONST_SGL( 0.62034397054380f / 8.0) },
+  { FL2FXCONST_SGL(-0.06618622548177f / 8.0), FL2FXCONST_SGL(-0.23812217221359f / 8.0) },
+  { FL2FXCONST_SGL( 0.99430266919728f / 8.0), FL2FXCONST_SGL( 0.18812555317553f / 8.0) },
+  { FL2FXCONST_SGL( 0.97686402381843f / 8.0), FL2FXCONST_SGL(-0.28664534366620f / 8.0) },
+  { FL2FXCONST_SGL( 0.94813650221268f / 8.0), FL2FXCONST_SGL(-0.97506640027128f / 8.0) },
+  { FL2FXCONST_SGL(-0.95434497492853f / 8.0), FL2FXCONST_SGL(-0.79607978501983f / 8.0) },
+  { FL2FXCONST_SGL(-0.49104783137150f / 8.0), FL2FXCONST_SGL( 0.32895214359663f / 8.0) },
+  { FL2FXCONST_SGL( 0.99881175120751f / 8.0), FL2FXCONST_SGL( 0.88993983831354f / 8.0) },
+  { FL2FXCONST_SGL( 0.50449166760303f / 8.0), FL2FXCONST_SGL(-0.85995072408434f / 8.0) },
+  { FL2FXCONST_SGL( 0.47162891065108f / 8.0), FL2FXCONST_SGL(-0.18680204049569f / 8.0) },
+  { FL2FXCONST_SGL(-0.62081581361840f / 8.0), FL2FXCONST_SGL( 0.75000676218956f / 8.0) },
+  { FL2FXCONST_SGL(-0.43867015250812f / 8.0), FL2FXCONST_SGL( 0.99998069244322f / 8.0) },
+  { FL2FXCONST_SGL( 0.98630563232075f / 8.0), FL2FXCONST_SGL(-0.53578899600662f / 8.0) },
+  { FL2FXCONST_SGL(-0.61510362277374f / 8.0), FL2FXCONST_SGL(-0.89515019899997f / 8.0) },
+  { FL2FXCONST_SGL(-0.03841517601843f / 8.0), FL2FXCONST_SGL(-0.69888815681179f / 8.0) },
+  { FL2FXCONST_SGL(-0.30102157304644f / 8.0), FL2FXCONST_SGL(-0.07667808922205f / 8.0) },
+  { FL2FXCONST_SGL( 0.41881284182683f / 8.0), FL2FXCONST_SGL( 0.02188098922282f / 8.0) },
+  { FL2FXCONST_SGL(-0.86135454941237f / 8.0), FL2FXCONST_SGL( 0.98947480909359f / 8.0) },
+  { FL2FXCONST_SGL( 0.67226861393788f / 8.0), FL2FXCONST_SGL(-0.13494389011014f / 8.0) },
+  { FL2FXCONST_SGL(-0.70737398842068f / 8.0), FL2FXCONST_SGL(-0.76547349325992f / 8.0) },
+  { FL2FXCONST_SGL( 0.94044946687963f / 8.0), FL2FXCONST_SGL( 0.09026201157416f / 8.0) },
+  { FL2FXCONST_SGL(-0.82386352534327f / 8.0), FL2FXCONST_SGL( 0.08924768823676f / 8.0) },
+  { FL2FXCONST_SGL(-0.32070666698656f / 8.0), FL2FXCONST_SGL( 0.50143421908753f / 8.0) },
+  { FL2FXCONST_SGL( 0.57593163224487f / 8.0), FL2FXCONST_SGL(-0.98966422921509f / 8.0) },
+  { FL2FXCONST_SGL(-0.36326018419965f / 8.0), FL2FXCONST_SGL( 0.07440243123228f / 8.0) },
+  { FL2FXCONST_SGL( 0.99979044674350f / 8.0), FL2FXCONST_SGL(-0.14130287347405f / 8.0) },
+  { FL2FXCONST_SGL(-0.92366023326932f / 8.0), FL2FXCONST_SGL(-0.97979298068180f / 8.0) },
+  { FL2FXCONST_SGL(-0.44607178518598f / 8.0), FL2FXCONST_SGL(-0.54233252016394f / 8.0) },
+  { FL2FXCONST_SGL( 0.44226800932956f / 8.0), FL2FXCONST_SGL( 0.71326756742752f / 8.0) },
+  { FL2FXCONST_SGL( 0.03671907158312f / 8.0), FL2FXCONST_SGL( 0.63606389366675f / 8.0) },
+  { FL2FXCONST_SGL( 0.52175424682195f / 8.0), FL2FXCONST_SGL(-0.85396826735705f / 8.0) },
+  { FL2FXCONST_SGL(-0.94701139690956f / 8.0), FL2FXCONST_SGL(-0.01826348194255f / 8.0) },
+  { FL2FXCONST_SGL(-0.98759606946049f / 8.0), FL2FXCONST_SGL( 0.82288714303073f / 8.0) },
+  { FL2FXCONST_SGL( 0.87434794743625f / 8.0), FL2FXCONST_SGL( 0.89399495655433f / 8.0) },
+  { FL2FXCONST_SGL(-0.93412041758744f / 8.0), FL2FXCONST_SGL( 0.41374052024363f / 8.0) },
+  { FL2FXCONST_SGL( 0.96063943315511f / 8.0), FL2FXCONST_SGL( 0.93116709541280f / 8.0) },
+  { FL2FXCONST_SGL( 0.97534253457837f / 8.0), FL2FXCONST_SGL( 0.86150930812689f / 8.0) },
+  { FL2FXCONST_SGL( 0.99642466504163f / 8.0), FL2FXCONST_SGL( 0.70190043427512f / 8.0) },
+  { FL2FXCONST_SGL(-0.94705089665984f / 8.0), FL2FXCONST_SGL(-0.29580042814306f / 8.0) },
+  { FL2FXCONST_SGL( 0.91599807087376f / 8.0), FL2FXCONST_SGL(-0.98147830385781f / 8.0) }
+};
+//@}
+
+/*
+static const FIXP_SGL harmonicPhase [2][4] = {
+  { 1.0, 0.0, -1.0,  0.0},
+  { 0.0, 1.0,  0.0, -1.0}
+};
+*/
+
+
+/* The CLDFB-80 is not linear phase (unsymmetric), but the exact
+   phase difference between adjacent bands, at exact positions
+   (in this case exactly in the frequency band centre), can of
+   course be determined anyway. While the standard symmetric QMF
+   bank has a phase difference of 0.5*pi, the CLDFB-80
+   bank has the difference 0.2337*pi. */
+const FIXP_SGL harmonicPhaseX [2][4] = {
+  { FL2FXCONST_SGL( 7.423735494778151e-001), FL2FXCONST_SGL(-6.699862036159475e-001),
+    FL2FXCONST_SGL(-7.423735494778152e-001), FL2FXCONST_SGL( 6.699862036159474e-001) },
+  { FL2FXCONST_SGL( 7.423735494778151e-001), FL2FXCONST_SGL( 6.699862036159476e-001),
+    FL2FXCONST_SGL(-7.423735494778151e-001), FL2FXCONST_SGL(-6.699862036159476e-001) }
+};
+
+/* tables for SBR and AAC LD */
+/* table for 8 time slot index */
+const int FDK_sbrDecoder_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 15 time slot index */
+const int FDK_sbrDecoder_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},
+};
+
+/* table for 16 time slot index */
+const int FDK_sbrDecoder_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},
+};
+
+/*!
+  \name FrameInfoDefaults
+
+  Predefined envelope positions for the FIX-FIX case (static framing)
+*/
+//@{
+const FRAME_INFO FDK_sbrDecoder_sbr_frame_info1_15 = { 0, 1, {0, 15,  0,  0,  0, 0}, {1, 0, 0, 0, 0}, -1, 1, {0, 15,  0} };
+const FRAME_INFO FDK_sbrDecoder_sbr_frame_info2_15 = { 0, 2, {0,  8, 15,  0,  0, 0}, {1, 1, 0, 0, 0}, -1, 2, {0,  8, 15} };
+const FRAME_INFO FDK_sbrDecoder_sbr_frame_info4_15 = { 0, 4, {0,  4,  8, 12, 15, 0}, {1, 1, 1, 1, 0}, -1, 2, {0,  8, 15} };
+#if (MAX_ENVELOPES >= 8)
+const FRAME_INFO FDK_sbrDecoder_sbr_frame_info8_15 = { 0, 8, {0, 2, 4, 6, 8, 10, 12, 14, 15}, {1, 1, 1, 1, 1, 1, 1, 1}, -1, 2, {0, 8, 15} };
+#endif
+
+const FRAME_INFO FDK_sbrDecoder_sbr_frame_info1_16 = { 0, 1, {0, 16,  0,  0,  0, 0}, {1, 0, 0, 0, 0}, -1, 1, {0, 16, 0} };
+const FRAME_INFO FDK_sbrDecoder_sbr_frame_info2_16 = { 0, 2, {0,  8, 16,  0,  0, 0}, {1, 1, 0, 0, 0}, -1, 2, {0, 8, 16} };
+const FRAME_INFO FDK_sbrDecoder_sbr_frame_info4_16 = { 0, 4, {0,  4,  8, 12, 16, 0}, {1, 1, 1, 1, 0}, -1, 2, {0, 8, 16} };
+
+#if (MAX_ENVELOPES >= 8)
+const FRAME_INFO FDK_sbrDecoder_sbr_frame_info8_16 = { 0, 8, {0, 2, 4, 6, 8, 10, 12, 14, 16}, {1, 1, 1, 1, 1, 1, 1, 1}, -1, 2, {0, 8, 16} };
+#endif
+
+
+//@}
+
+/*!
+  \name SBR_HuffmanTables
+
+  SBR Huffman Table Overview:        \n
+                                     \n
+ o envelope level,   1.5 dB:         \n
+    1)  sbr_huffBook_EnvLevel10T[120][2]   \n
+    2)  sbr_huffBook_EnvLevel10F[120][2]   \n
+                                     \n
+ o envelope balance, 1.5 dB:         \n
+    3)  sbr_huffBook_EnvBalance10T[48][2]  \n
+    4)  sbr_huffBook_EnvBalance10F[48][2]  \n
+                                     \n
+ o envelope level,   3.0 dB:         \n
+    5)  sbr_huffBook_EnvLevel11T[62][2]    \n
+    6)  sbr_huffBook_EnvLevel11F[62][2]    \n
+                                     \n
+ o envelope balance, 3.0 dB:         \n
+    7)  sbr_huffBook_EnvBalance11T[24][2]  \n
+    8)  sbr_huffBook_EnvBalance11F[24][2]  \n
+                                     \n
+ o noise level,      3.0 dB:         \n
+    9)  sbr_huffBook_NoiseLevel11T[62][2]  \n
+    -) (sbr_huffBook_EnvLevel11F[62][2] is used for freq dir)\n
+                                     \n
+ o noise balance,    3.0 dB:         \n
+   10)  sbr_huffBook_NoiseBalance11T[24][2]\n
+    -) (sbr_huffBook_EnvBalance11F[24][2] is used for freq dir)\n
+                                     \n
+  (1.5 dB is never used for noise)
+
+*/
+//@{
+const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvLevel10T[120][2] = {
+  {   1,   2 },    { -64, -65 },    {   3,   4 },    { -63, -66 },
+  {   5,   6 },    { -62, -67 },    {   7,   8 },    { -61, -68 },
+  {   9,  10 },    { -60, -69 },    {  11,  12 },    { -59, -70 },
+  {  13,  14 },    { -58, -71 },    {  15,  16 },    { -57, -72 },
+  {  17,  18 },    { -73, -56 },    {  19,  21 },    { -74,  20 },
+  { -55, -75 },    {  22,  26 },    {  23,  24 },    { -54, -76 },
+  { -77,  25 },    { -53, -78 },    {  27,  34 },    {  28,  29 },
+  { -52, -79 },    {  30,  31 },    { -80, -51 },    {  32,  33 },
+  { -83, -82 },    { -81, -50 },    {  35,  57 },    {  36,  40 },
+  {  37,  38 },    { -88, -84 },    { -48,  39 },    { -90, -85 },
+  {  41,  46 },    {  42,  43 },    { -49, -87 },    {  44,  45 },
+  { -89, -86 },    {-124,-123 },    {  47,  50 },    {  48,  49 },
+  {-122,-121 },    {-120,-119 },    {  51,  54 },    {  52,  53 },
+  {-118,-117 },    {-116,-115 },    {  55,  56 },    {-114,-113 },
+  {-112,-111 },    {  58,  89 },    {  59,  74 },    {  60,  67 },
+  {  61,  64 },    {  62,  63 },    {-110,-109 },    {-108,-107 },
+  {  65,  66 },    {-106,-105 },    {-104,-103 },    {  68,  71 },
+  {  69,  70 },    {-102,-101 },    {-100, -99 },    {  72,  73 },
+  { -98, -97 },    { -96, -95 },    {  75,  82 },    {  76,  79 },
+  {  77,  78 },    { -94, -93 },    { -92, -91 },    {  80,  81 },
+  { -47, -46 },    { -45, -44 },    {  83,  86 },    {  84,  85 },
+  { -43, -42 },    { -41, -40 },    {  87,  88 },    { -39, -38 },
+  { -37, -36 },    {  90, 105 },    {  91,  98 },    {  92,  95 },
+  {  93,  94 },    { -35, -34 },    { -33, -32 },    {  96,  97 },
+  { -31, -30 },    { -29, -28 },    {  99, 102 },    { 100, 101 },
+  { -27, -26 },    { -25, -24 },    { 103, 104 },    { -23, -22 },
+  { -21, -20 },    { 106, 113 },    { 107, 110 },    { 108, 109 },
+  { -19, -18 },    { -17, -16 },    { 111, 112 },    { -15, -14 },
+  { -13, -12 },    { 114, 117 },    { 115, 116 },    { -11, -10 },
+  {  -9,  -8 },    { 118, 119 },    {  -7,  -6 },    {  -5,  -4 }
+};
+
+const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvLevel10F[120][2] = {
+  {   1,   2 },    { -64, -65 },    {   3,   4 },    { -63, -66 },
+  {   5,   6 },    { -67, -62 },    {   7,   8 },    { -68, -61 },
+  {   9,  10 },    { -69, -60 },    {  11,  13 },    { -70,  12 },
+  { -59, -71 },    {  14,  16 },    { -58,  15 },    { -72, -57 },
+  {  17,  19 },    { -73,  18 },    { -56, -74 },    {  20,  23 },
+  {  21,  22 },    { -55, -75 },    { -54, -53 },    {  24,  27 },
+  {  25,  26 },    { -76, -52 },    { -77, -51 },    {  28,  31 },
+  {  29,  30 },    { -50, -78 },    { -79, -49 },    {  32,  36 },
+  {  33,  34 },    { -48, -47 },    { -80,  35 },    { -81, -82 },
+  {  37,  47 },    {  38,  41 },    {  39,  40 },    { -83, -46 },
+  { -45, -84 },    {  42,  44 },    { -85,  43 },    { -44, -43 },
+  {  45,  46 },    { -88, -87 },    { -86, -90 },    {  48,  66 },
+  {  49,  56 },    {  50,  53 },    {  51,  52 },    { -92, -42 },
+  { -41, -39 },    {  54,  55 },    {-105, -89 },    { -38, -37 },
+  {  57,  60 },    {  58,  59 },    { -94, -91 },    { -40, -36 },
+  {  61,  63 },    { -20,  62 },    {-115,-110 },    {  64,  65 },
+  {-108,-107 },    {-101, -97 },    {  67,  89 },    {  68,  75 },
+  {  69,  72 },    {  70,  71 },    { -95, -93 },    { -34, -27 },
+  {  73,  74 },    { -22, -17 },    { -16,-124 },    {  76,  82 },
+  {  77,  79 },    {-123,  78 },    {-122,-121 },    {  80,  81 },
+  {-120,-119 },    {-118,-117 },    {  83,  86 },    {  84,  85 },
+  {-116,-114 },    {-113,-112 },    {  87,  88 },    {-111,-109 },
+  {-106,-104 },    {  90, 105 },    {  91,  98 },    {  92,  95 },
+  {  93,  94 },    {-103,-102 },    {-100, -99 },    {  96,  97 },
+  { -98, -96 },    { -35, -33 },    {  99, 102 },    { 100, 101 },
+  { -32, -31 },    { -30, -29 },    { 103, 104 },    { -28, -26 },
+  { -25, -24 },    { 106, 113 },    { 107, 110 },    { 108, 109 },
+  { -23, -21 },    { -19, -18 },    { 111, 112 },    { -15, -14 },
+  { -13, -12 },    { 114, 117 },    { 115, 116 },    { -11, -10 },
+  {  -9,  -8 },    { 118, 119 },    {  -7,  -6 },    {  -5,  -4 }
+};
+
+const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvBalance10T[48][2] = {
+  { -64,   1 },    { -63,   2 },    { -65,   3 },    { -62,   4 },
+  { -66,   5 },    { -61,   6 },    { -67,   7 },    { -60,   8 },
+  { -68,   9 },    {  10,  11 },    { -69, -59 },    {  12,  13 },
+  { -70, -58 },    {  14,  28 },    {  15,  21 },    {  16,  18 },
+  { -57,  17 },    { -71, -56 },    {  19,  20 },    { -88, -87 },
+  { -86, -85 },    {  22,  25 },    {  23,  24 },    { -84, -83 },
+  { -82, -81 },    {  26,  27 },    { -80, -79 },    { -78, -77 },
+  {  29,  36 },    {  30,  33 },    {  31,  32 },    { -76, -75 },
+  { -74, -73 },    {  34,  35 },    { -72, -55 },    { -54, -53 },
+  {  37,  41 },    {  38,  39 },    { -52, -51 },    { -50,  40 },
+  { -49, -48 },    {  42,  45 },    {  43,  44 },    { -47, -46 },
+  { -45, -44 },    {  46,  47 },    { -43, -42 },    { -41, -40 }
+};
+
+const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvBalance10F[48][2] = {
+  { -64,   1 },    { -65,   2 },    { -63,   3 },    { -66,   4 },
+  { -62,   5 },    { -61,   6 },    { -67,   7 },    { -68,   8 },
+  { -60,   9 },    {  10,  11 },    { -69, -59 },    { -70,  12 },
+  { -58,  13 },    {  14,  17 },    { -71,  15 },    { -57,  16 },
+  { -56, -73 },    {  18,  32 },    {  19,  25 },    {  20,  22 },
+  { -72,  21 },    { -88, -87 },    {  23,  24 },    { -86, -85 },
+  { -84, -83 },    {  26,  29 },    {  27,  28 },    { -82, -81 },
+  { -80, -79 },    {  30,  31 },    { -78, -77 },    { -76, -75 },
+  {  33,  40 },    {  34,  37 },    {  35,  36 },    { -74, -55 },
+  { -54, -53 },    {  38,  39 },    { -52, -51 },    { -50, -49 },
+  {  41,  44 },    {  42,  43 },    { -48, -47 },    { -46, -45 },
+  {  45,  46 },    { -44, -43 },    { -42,  47 },    { -41, -40 }
+};
+
+const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvLevel11T[62][2] = {
+  { -64,   1 },    { -65,   2 },    { -63,   3 },    { -66,   4 },
+  { -62,   5 },    { -67,   6 },    { -61,   7 },    { -68,   8 },
+  { -60,   9 },    {  10,  11 },    { -69, -59 },    {  12,  14 },
+  { -70,  13 },    { -71, -58 },    {  15,  18 },    {  16,  17 },
+  { -72, -57 },    { -73, -74 },    {  19,  22 },    { -56,  20 },
+  { -55,  21 },    { -54, -77 },    {  23,  31 },    {  24,  25 },
+  { -75, -76 },    {  26,  27 },    { -78, -53 },    {  28,  29 },
+  { -52, -95 },    { -94,  30 },    { -93, -92 },    {  32,  47 },
+  {  33,  40 },    {  34,  37 },    {  35,  36 },    { -91, -90 },
+  { -89, -88 },    {  38,  39 },    { -87, -86 },    { -85, -84 },
+  {  41,  44 },    {  42,  43 },    { -83, -82 },    { -81, -80 },
+  {  45,  46 },    { -79, -51 },    { -50, -49 },    {  48,  55 },
+  {  49,  52 },    {  50,  51 },    { -48, -47 },    { -46, -45 },
+  {  53,  54 },    { -44, -43 },    { -42, -41 },    {  56,  59 },
+  {  57,  58 },    { -40, -39 },    { -38, -37 },    {  60,  61 },
+  { -36, -35 },    { -34, -33 }
+};
+
+const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvLevel11F[62][2] = {
+  { -64,   1 },    { -65,   2 },    { -63,   3 },    { -66,   4 },
+  { -62,   5 },    { -67,   6 },    {   7,   8 },    { -61, -68 },
+  {   9,  10 },    { -60, -69 },    {  11,  12 },    { -59, -70 },
+  {  13,  14 },    { -58, -71 },    {  15,  16 },    { -57, -72 },
+  {  17,  19 },    { -56,  18 },    { -55, -73 },    {  20,  24 },
+  {  21,  22 },    { -74, -54 },    { -53,  23 },    { -75, -76 },
+  {  25,  30 },    {  26,  27 },    { -52, -51 },    {  28,  29 },
+  { -77, -79 },    { -50, -49 },    {  31,  39 },    {  32,  35 },
+  {  33,  34 },    { -78, -46 },    { -82, -88 },    {  36,  37 },
+  { -83, -48 },    { -47,  38 },    { -86, -85 },    {  40,  47 },
+  {  41,  44 },    {  42,  43 },    { -80, -44 },    { -43, -42 },
+  {  45,  46 },    { -39, -87 },    { -84, -40 },    {  48,  55 },
+  {  49,  52 },    {  50,  51 },    { -95, -94 },    { -93, -92 },
+  {  53,  54 },    { -91, -90 },    { -89, -81 },    {  56,  59 },
+  {  57,  58 },    { -45, -41 },    { -38, -37 },    {  60,  61 },
+  { -36, -35 },    { -34, -33 }
+};
+
+const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvBalance11T[24][2] = {
+  { -64,   1 },    { -63,   2 },    { -65,   3 },    { -66,   4 },
+  { -62,   5 },    { -61,   6 },    { -67,   7 },    { -68,   8 },
+  { -60,   9 },    {  10,  16 },    {  11,  13 },    { -69,  12 },
+  { -76, -75 },    {  14,  15 },    { -74, -73 },    { -72, -71 },
+  {  17,  20 },    {  18,  19 },    { -70, -59 },    { -58, -57 },
+  {  21,  22 },    { -56, -55 },    { -54,  23 },    { -53, -52 }
+};
+
+const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvBalance11F[24][2] = {
+  { -64,   1 },    { -65,   2 },    { -63,   3 },    { -66,   4 },
+  { -62,   5 },    { -61,   6 },    { -67,   7 },    { -68,   8 },
+  { -60,   9 },    {  10,  13 },    { -69,  11 },    { -59,  12 },
+  { -58, -76 },    {  14,  17 },    {  15,  16 },    { -75, -74 },
+  { -73, -72 },    {  18,  21 },    {  19,  20 },    { -71, -70 },
+  { -57, -56 },    {  22,  23 },    { -55, -54 },    { -53, -52 }
+};
+
+const SCHAR FDK_sbrDecoder_sbr_huffBook_NoiseLevel11T[62][2] = {
+  { -64,   1 },    { -63,   2 },    { -65,   3 },    { -66,   4 },
+  { -62,   5 },    { -67,   6 },    {   7,   8 },    { -61, -68 },
+  {   9,  30 },    {  10,  15 },    { -60,  11 },    { -69,  12 },
+  {  13,  14 },    { -59, -53 },    { -95, -94 },    {  16,  23 },
+  {  17,  20 },    {  18,  19 },    { -93, -92 },    { -91, -90 },
+  {  21,  22 },    { -89, -88 },    { -87, -86 },    {  24,  27 },
+  {  25,  26 },    { -85, -84 },    { -83, -82 },    {  28,  29 },
+  { -81, -80 },    { -79, -78 },    {  31,  46 },    {  32,  39 },
+  {  33,  36 },    {  34,  35 },    { -77, -76 },    { -75, -74 },
+  {  37,  38 },    { -73, -72 },    { -71, -70 },    {  40,  43 },
+  {  41,  42 },    { -58, -57 },    { -56, -55 },    {  44,  45 },
+  { -54, -52 },    { -51, -50 },    {  47,  54 },    {  48,  51 },
+  {  49,  50 },    { -49, -48 },    { -47, -46 },    {  52,  53 },
+  { -45, -44 },    { -43, -42 },    {  55,  58 },    {  56,  57 },
+  { -41, -40 },    { -39, -38 },    {  59,  60 },    { -37, -36 },
+  { -35,  61 },    { -34, -33 }
+};
+
+const SCHAR FDK_sbrDecoder_sbr_huffBook_NoiseBalance11T[24][2] = {
+  { -64,   1 },    { -65,   2 },    { -63,   3 },    {   4,   9 },
+  { -66,   5 },    { -62,   6 },    {   7,   8 },    { -76, -75 },
+  { -74, -73 },    {  10,  17 },    {  11,  14 },    {  12,  13 },
+  { -72, -71 },    { -70, -69 },    {  15,  16 },    { -68, -67 },
+  { -61, -60 },    {  18,  21 },    {  19,  20 },    { -59, -58 },
+  { -57, -56 },    {  22,  23 },    { -55, -54 },    { -53, -52 }
+};
+//@}
+
+
+
+
+/*!
+  \name  parametric stereo
+  \brief constants used by the parametric stereo part of the decoder
+
+*/
+
+
+/* constants used in psbitdec.cpp */
+
+/* FIX_BORDER can have 0, 1, 2, 4 envelopes */
+const UCHAR FDK_sbrDecoder_aFixNoEnvDecode[4] = {0, 1, 2, 4};
+
+
+/* IID & ICC Huffman codebooks */
+const SCHAR aBookPsIidTimeDecode[28][2] = {
+  { -64,   1 },    { -65,   2 },    { -63,   3 },    { -66,   4 },
+  { -62,   5 },    { -67,   6 },    { -61,   7 },    { -68,   8 },
+  { -60,   9 },    { -69,  10 },    { -59,  11 },    { -70,  12 },
+  { -58,  13 },    { -57,  14 },    { -71,  15 },    {  16,  17 },
+  { -56, -72 },    {  18,  21 },    {  19,  20 },    { -55, -78 },
+  { -77, -76 },    {  22,  25 },    {  23,  24 },    { -75, -74 },
+  { -73, -54 },    {  26,  27 },    { -53, -52 },    { -51, -50 }
+};
+
+const SCHAR aBookPsIidFreqDecode[28][2] = {
+  { -64,   1 },    {   2,   3 },    { -63, -65 },    {   4,   5 },
+  { -62, -66 },    {   6,   7 },    { -61, -67 },    {   8,   9 },
+  { -68, -60 },    { -59,  10 },    { -69,  11 },    { -58,  12 },
+  { -70,  13 },    { -71,  14 },    { -57,  15 },    {  16,  17 },
+  { -56, -72 },    {  18,  19 },    { -55, -54 },    {  20,  21 },
+  { -73, -53 },    {  22,  24 },    { -74,  23 },    { -75, -78 },
+  {  25,  26 },    { -77, -76 },    { -52,  27 },    { -51, -50 }
+};
+
+const SCHAR aBookPsIccTimeDecode[14][2] = {
+  { -64,   1 },    { -63,   2 },    { -65,   3 },    { -62,   4 },
+  { -66,   5 },    { -61,   6 },    { -67,   7 },    { -60,   8 },
+  { -68,   9 },    { -59,  10 },    { -69,  11 },    { -58,  12 },
+  { -70,  13 },    { -71, -57 }
+};
+
+const SCHAR aBookPsIccFreqDecode[14][2] = {
+  { -64,   1 },    { -63,   2 },    { -65,   3 },    { -62,   4 },
+  { -66,   5 },    { -61,   6 },    { -67,   7 },    { -60,   8 },
+  { -59,   9 },    { -68,  10 },    { -58,  11 },    { -69,  12 },
+  { -57,  13 },    { -70, -71 }
+};
+
+/* IID-fine Huffman codebooks */
+
+const SCHAR aBookPsIidFineTimeDecode[60][2] = {
+  {   1, -64 },    { -63,   2 },    {   3, -65 },    {   4,  59 },
+  {   5,   7 },    {   6, -67 },    { -68, -60 },    { -61,   8 },
+  {   9,  11 },    { -59,  10 },    { -70, -58 },    {  12,  41 },
+  {  13,  20 },    {  14, -71 },    { -55,  15 },    { -53,  16 },
+  {  17, -77 },    {  18,  19 },    { -85, -84 },    { -46, -45 },
+  { -57,  21 },    {  22,  40 },    {  23,  29 },    { -51,  24 },
+  {  25,  26 },    { -83, -82 },    {  27,  28 },    { -90, -38 },
+  { -92, -91 },    {  30,  37 },    {  31,  34 },    {  32,  33 },
+  { -35, -34 },    { -37, -36 },    {  35,  36 },    { -94, -93 },
+  { -89, -39 },    {  38, -79 },    {  39, -81 },    { -88, -40 },
+  { -74, -54 },    {  42, -69 },    {  43,  44 },    { -72, -56 },
+  {  45,  52 },    {  46,  50 },    {  47, -76 },    { -49,  48 },
+  { -47,  49 },    { -87, -41 },    { -52,  51 },    { -78, -50 },
+  {  53, -73 },    {  54, -75 },    {  55,  57 },    {  56, -80 },
+  { -86, -42 },    { -48,  58 },    { -44, -43 },    { -66, -62 }
+};
+
+
+const SCHAR aBookPsIidFineFreqDecode[60][2] = {
+  {   1, -64 },    {   2,   4 },    {   3, -65 },    { -66, -62 },
+  { -63,   5 },    {   6,   7 },    { -67, -61 },    {   8,   9 },
+  { -68, -60 },    {  10,  11 },    { -69, -59 },    {  12,  13 },
+  { -70, -58 },    {  14,  18 },    { -57,  15 },    {  16, -72 },
+  { -54,  17 },    { -75, -53 },    {  19,  37 },    { -56,  20 },
+  {  21, -73 },    {  22,  29 },    {  23, -76 },    {  24, -78 },
+  {  25,  28 },    {  26,  27 },    { -85, -43 },    { -83, -45 },
+  { -81, -47 },    { -52,  30 },    { -50,  31 },    {  32, -79 },
+  {  33,  34 },    { -82, -46 },    {  35,  36 },    { -90, -89 },
+  { -92, -91 },    {  38, -71 },    { -55,  39 },    {  40, -74 },
+  {  41,  50 },    {  42, -77 },    { -49,  43 },    {  44,  47 },
+  {  45,  46 },    { -86, -42 },    { -88, -87 },    {  48,  49 },
+  { -39, -38 },    { -41, -40 },    { -51,  51 },    {  52,  59 },
+  {  53,  56 },    {  54,  55 },    { -35, -34 },    { -37, -36 },
+  {  57,  58 },    { -94, -93 },    { -84, -44 },    { -80, -48 }
+};
+
+/* constants used in psdec.cpp */
+
+const FIXP_DBL decayScaleFactTable[64] = {
+
+ FL2FXCONST_DBL(1.000000), FL2FXCONST_DBL(1.000000), FL2FXCONST_DBL(1.000000), FL2FXCONST_DBL(1.000000),
+ FL2FXCONST_DBL(0.950000), FL2FXCONST_DBL(0.900000), FL2FXCONST_DBL(0.850000), FL2FXCONST_DBL(0.800000),
+ FL2FXCONST_DBL(0.750000), FL2FXCONST_DBL(0.700000), FL2FXCONST_DBL(0.650000), FL2FXCONST_DBL(0.600000),
+ FL2FXCONST_DBL(0.550000), FL2FXCONST_DBL(0.500000), FL2FXCONST_DBL(0.450000), FL2FXCONST_DBL(0.400000),
+ FL2FXCONST_DBL(0.350000), FL2FXCONST_DBL(0.300000), FL2FXCONST_DBL(0.250000), FL2FXCONST_DBL(0.200000),
+ FL2FXCONST_DBL(0.150000), FL2FXCONST_DBL(0.100000), FL2FXCONST_DBL(0.050000), FL2FXCONST_DBL(0.000000),
+ FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000),
+ FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000),
+ FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000),
+ FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000),
+ FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000),
+ FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000),
+ FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000),
+ FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000),
+ FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000),
+ FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000) };
+
+/* the values of the following 3 tables are shiftet right by 1 ! */
+const FIXP_DBL ScaleFactors[NO_IID_LEVELS] = {
+
+    0x5a5ded00, 0x59cd0400, 0x58c29680, 0x564c2e80, 0x52a3d480,
+    0x4c8be080, 0x46df3080, 0x40000000, 0x384ba5c0, 0x304c2980,
+    0x24e9f640, 0x1b4a2940, 0x11b5c0a0, 0x0b4e2540, 0x0514ea90
+};
+
+const FIXP_DBL ScaleFactorsFine[NO_IID_LEVELS_FINE] = {
+
+    0x5a825c00, 0x5a821c00, 0x5a815100, 0x5a7ed000, 0x5a76e600,
+    0x5a5ded00, 0x5a39b880, 0x59f1fd00, 0x5964d680, 0x5852ca00,
+    0x564c2e80, 0x54174480, 0x50ea7500, 0x4c8be080, 0x46df3080,
+    0x40000000, 0x384ba5c0, 0x304c2980, 0x288dd240, 0x217a2900,
+    0x1b4a2940, 0x13c5ece0, 0x0e2b0090, 0x0a178ef0, 0x072ab798,
+    0x0514ea90, 0x02dc5944, 0x019bf87c, 0x00e7b173, 0x00824b8b,
+    0x00494568
+};
+const FIXP_DBL Alphas[NO_ICC_LEVELS] = {
+
+    0x00000000, 0x0b6b5be0, 0x12485f80, 0x1da2fa40,
+    0x2637ebc0, 0x3243f6c0, 0x466b7480, 0x6487ed80
+};
+
+#if defined(ARCH_PREFER_MULT_32x16)
+#define  FIXP_PS FIXP_SGL
+#define  FXP_CAST(a) FX_DBL2FX_SGL((FIXP_DBL)a)
+#define  FL2FXCONST_PS FL2FXCONST_SGL
+#else
+#define  FIXP_PS FIXP_DBL
+#define  FXP_CAST
+#define  FL2FXCONST_PS FL2FXCONST_DBL
+#endif
+
+const FIXP_PS aAllpassLinkDecaySer[NO_SERIAL_ALLPASS_LINKS] = {
+FXP_CAST(0x53625b00), FXP_CAST(0x4848af00), FXP_CAST(0x3ea94d00) };
+
+const FIXP_PS aaFractDelayPhaseFactorReQmf[NO_QMF_CHANNELS] = {
+FXP_CAST(0x68b92180), FXP_CAST(0xde396900), FXP_CAST(0x80650380), FXP_CAST(0xcb537e40), FXP_CAST(0x5beb8f00), FXP_CAST(0x72f29200), FXP_CAST(0xf1f43c50), FXP_CAST(0x83896280),
+FXP_CAST(0xb9b99c00), FXP_CAST(0x4cda8f00), FXP_CAST(0x7a576e00), FXP_CAST(0x060799e0), FXP_CAST(0x89be5280), FXP_CAST(0xa9dab600), FXP_CAST(0x3be51b00), FXP_CAST(0x7eb91900),
+FXP_CAST(0x19f4f540), FXP_CAST(0x92dcb380), FXP_CAST(0x9c1ad700), FXP_CAST(0x29761940), FXP_CAST(0x7ffbf500), FXP_CAST(0x2d3eb180), FXP_CAST(0x9eab0a00), FXP_CAST(0x90d0aa80),
+FXP_CAST(0x1601bcc0), FXP_CAST(0x7e180e80), FXP_CAST(0x3f6b3940), FXP_CAST(0xacdeeb00), FXP_CAST(0x88435b00), FXP_CAST(0x0202a768), FXP_CAST(0x79194f80), FXP_CAST(0x5007fd00),
+FXP_CAST(0xbd1ecf00), FXP_CAST(0x82a8d100), FXP_CAST(0xedf6e5e0), FXP_CAST(0x711f3500), FXP_CAST(0x5eac4480), FXP_CAST(0xcf0447c0), FXP_CAST(0x80245f80), FXP_CAST(0xda5cd4c0),
+FXP_CAST(0x665c0800), FXP_CAST(0x6afbc500), FXP_CAST(0xe21e85e0), FXP_CAST(0x80c5e500), FXP_CAST(0xc7b003c0), FXP_CAST(0x59139f80), FXP_CAST(0x74a8e400), FXP_CAST(0xf5f51f40),
+FXP_CAST(0x84896680), FXP_CAST(0xb6662b00), FXP_CAST(0x4999b600), FXP_CAST(0x7b76a300), FXP_CAST(0x0a0b0650), FXP_CAST(0x8b572b80), FXP_CAST(0xa6ec4580), FXP_CAST(0x384fda80),
+FXP_CAST(0x7f3a1f00), FXP_CAST(0x1de19ec0), FXP_CAST(0x95045000), FXP_CAST(0x99a3e180), FXP_CAST(0x25a30740), FXP_CAST(0x7fdb9e80), FXP_CAST(0x30fbdb00), FXP_CAST(0xa153d500) };
+
+const FIXP_PS aaFractDelayPhaseFactorImQmf[NO_QMF_CHANNELS] = {
+FXP_CAST(0xb6663a80), FXP_CAST(0x84896200), FXP_CAST(0xf5f50c70), FXP_CAST(0x74a8dc80), FXP_CAST(0x5913ad00), FXP_CAST(0xc7b01480), FXP_CAST(0x80c5e300), FXP_CAST(0xe21e73a0),
+FXP_CAST(0x6afbba80), FXP_CAST(0x665c1380), FXP_CAST(0xda5ce6c0), FXP_CAST(0x80246080), FXP_CAST(0xcf043640), FXP_CAST(0x5eac3800), FXP_CAST(0x711f3e00), FXP_CAST(0xedf6f8a0),
+FXP_CAST(0x82a8d500), FXP_CAST(0xbd1ebe80), FXP_CAST(0x5007ee00), FXP_CAST(0x79195580), FXP_CAST(0x0202ba40), FXP_CAST(0x88436180), FXP_CAST(0xacdedc80), FXP_CAST(0x3f6b28c0),
+FXP_CAST(0x7e181180), FXP_CAST(0x1601cf40), FXP_CAST(0x90d0b380), FXP_CAST(0x9eaafd80), FXP_CAST(0x2d3e9fc0), FXP_CAST(0x7ffbf580), FXP_CAST(0x29762b00), FXP_CAST(0x9c1ae280),
+FXP_CAST(0x92dca980), FXP_CAST(0x19f4e2c0), FXP_CAST(0x7eb91680), FXP_CAST(0x3be52b80), FXP_CAST(0xa9dac400), FXP_CAST(0x89be4b80), FXP_CAST(0x06078710), FXP_CAST(0x7a576880),
+FXP_CAST(0x4cda9e00), FXP_CAST(0xb9b9ac00), FXP_CAST(0x83895e00), FXP_CAST(0xf1f42990), FXP_CAST(0x72f28a00), FXP_CAST(0x5beb9c00), FXP_CAST(0xcb538f40), FXP_CAST(0x80650200),
+FXP_CAST(0xde3956c0), FXP_CAST(0x68b91680), FXP_CAST(0x68b92c00), FXP_CAST(0xde397b40), FXP_CAST(0x80650500), FXP_CAST(0xcb536d00), FXP_CAST(0x5beb8180), FXP_CAST(0x72f29a80),
+FXP_CAST(0xf1f44f10), FXP_CAST(0x83896700), FXP_CAST(0xb9b98c80), FXP_CAST(0x4cda8000), FXP_CAST(0x7a577380), FXP_CAST(0x0607acb8), FXP_CAST(0x89be5a00), FXP_CAST(0xa9daa800) };
+
+const FIXP_PS aaFractDelayPhaseFactorReSubQmf20[NO_SUB_QMF_CHANNELS] = {
+FXP_CAST(0x7e807380), FXP_CAST(0x72b9bb00), FXP_CAST(0x5c44ee80), FXP_CAST(0x3d3938c0), FXP_CAST(0x80000000), FXP_CAST(0x80000000),
+FXP_CAST(0x72b9bb00), FXP_CAST(0x7e807380), FXP_CAST(0xba914700), FXP_CAST(0x050677b0), FXP_CAST(0x895cc380), FXP_CAST(0x834e4900) };
+
+const FIXP_PS aaFractDelayPhaseFactorImSubQmf20[NO_SUB_QMF_CHANNELS] = {
+FXP_CAST(0xec791720), FXP_CAST(0xc73ca080), FXP_CAST(0xa748ea00), FXP_CAST(0x8f976980), FXP_CAST(0x00000000), FXP_CAST(0x00000000),
+FXP_CAST(0x38c35f80), FXP_CAST(0x1386e8e0), FXP_CAST(0x9477d000), FXP_CAST(0x80194380), FXP_CAST(0xcff26140), FXP_CAST(0x1ce70d40) };
+
+const FIXP_PS aaFractDelayPhaseFactorSerReQmf[NO_QMF_CHANNELS][NO_SERIAL_ALLPASS_LINKS] = {
+{FXP_CAST(0x63e52480), FXP_CAST(0x30fbc540), FXP_CAST(0x6d73af00)}, {FXP_CAST(0xc7b01280), FXP_CAST(0x89be5100), FXP_CAST(0xf7c31cb0)}, {FXP_CAST(0x83896200), FXP_CAST(0x7641af00), FXP_CAST(0x8aee2700)},
+{FXP_CAST(0x0202b330), FXP_CAST(0xcf043ac0), FXP_CAST(0x9bfab500)}, {FXP_CAST(0x7d572c80), FXP_CAST(0xcf043ac0), FXP_CAST(0x1893b960)}, {FXP_CAST(0x34ac7fc0), FXP_CAST(0x7641af00), FXP_CAST(0x7abf7980)},
+{FXP_CAST(0x99a3ee00), FXP_CAST(0x89be5100), FXP_CAST(0x58eead80)}, {FXP_CAST(0x9eab0580), FXP_CAST(0x30fbc540), FXP_CAST(0xd77dae40)}, {FXP_CAST(0x3be52140), FXP_CAST(0x30fbc540), FXP_CAST(0x819b8500)},
+{FXP_CAST(0x7b769d80), FXP_CAST(0x89be5100), FXP_CAST(0xb3a12280)}, {FXP_CAST(0xf9f86878), FXP_CAST(0x7641af00), FXP_CAST(0x37c519c0)}, {FXP_CAST(0x81e7ef80), FXP_CAST(0xcf043ac0), FXP_CAST(0x7ff16880)},
+{FXP_CAST(0xcf043cc0), FXP_CAST(0xcf043ac0), FXP_CAST(0x3e8b2340)}, {FXP_CAST(0x68b92280), FXP_CAST(0x7641af00), FXP_CAST(0xb9e4a900)}, {FXP_CAST(0x5eac3980), FXP_CAST(0x89be5100), FXP_CAST(0x80a05200)},
+{FXP_CAST(0xc094cd00), FXP_CAST(0x30fbc540), FXP_CAST(0xd051dc80)}, {FXP_CAST(0x85a89400), FXP_CAST(0x30fbc540), FXP_CAST(0x53483b00)}, {FXP_CAST(0x0a0af5e0), FXP_CAST(0x89be5100), FXP_CAST(0x7cb1b680)},
+{FXP_CAST(0x7eb91900), FXP_CAST(0x7641af00), FXP_CAST(0x2006e8c0)}, {FXP_CAST(0x2d3ea680), FXP_CAST(0xcf043ac0), FXP_CAST(0xa0ec1c00)}, {FXP_CAST(0x95044180), FXP_CAST(0xcf043ac0), FXP_CAST(0x880d2180)},
+{FXP_CAST(0xa4147300), FXP_CAST(0x7641af00), FXP_CAST(0xf0282870)}, {FXP_CAST(0x42e13f80), FXP_CAST(0x89be5100), FXP_CAST(0x694c4a00)}, {FXP_CAST(0x79195200), FXP_CAST(0x30fbc540), FXP_CAST(0x71374780)},
+{FXP_CAST(0xf1f43550), FXP_CAST(0x30fbc540), FXP_CAST(0xff6593ea)}, {FXP_CAST(0x80c5e280), FXP_CAST(0x89be5100), FXP_CAST(0x8e39ec00)}, {FXP_CAST(0xd689e480), FXP_CAST(0x7641af00), FXP_CAST(0x97648100)},
+{FXP_CAST(0x6d235300), FXP_CAST(0xcf043ac0), FXP_CAST(0x110a20c0)}, {FXP_CAST(0x5913a800), FXP_CAST(0xcf043ac0), FXP_CAST(0x785d4f80)}, {FXP_CAST(0xb9b99a00), FXP_CAST(0x7641af00), FXP_CAST(0x5e440880)},
+{FXP_CAST(0x88436100), FXP_CAST(0x89be5100), FXP_CAST(0xdece7000)}, {FXP_CAST(0x12091320), FXP_CAST(0x30fbc540), FXP_CAST(0x8309f800)}, {FXP_CAST(0x7f9afd00), FXP_CAST(0x30fbc540), FXP_CAST(0xada33f00)},
+{FXP_CAST(0x25a31700), FXP_CAST(0x89be5100), FXP_CAST(0x30cc3600)}, {FXP_CAST(0x90d0ab80), FXP_CAST(0x7641af00), FXP_CAST(0x7f7cbe80)}, {FXP_CAST(0xa9dabf00), FXP_CAST(0xcf043ac0), FXP_CAST(0x45182580)},
+{FXP_CAST(0x4999cb80), FXP_CAST(0xcf043ac0), FXP_CAST(0xc0681c80)}, {FXP_CAST(0x7641ac80), FXP_CAST(0x7641af00), FXP_CAST(0x80194380)}, {FXP_CAST(0xe9fe3300), FXP_CAST(0x89be5100), FXP_CAST(0xc95184c0)},
+{FXP_CAST(0x80246000), FXP_CAST(0x30fbc540), FXP_CAST(0x4d55d800)}, {FXP_CAST(0xde396fc0), FXP_CAST(0x30fbc540), FXP_CAST(0x7e324000)}, {FXP_CAST(0x711f3f00), FXP_CAST(0x89be5100), FXP_CAST(0x275ce480)},
+{FXP_CAST(0x53211700), FXP_CAST(0x7641af00), FXP_CAST(0xa6343580)}, {FXP_CAST(0xb3256780), FXP_CAST(0xcf043ac0), FXP_CAST(0x85997b80)}, {FXP_CAST(0x8b572680), FXP_CAST(0xcf043ac0), FXP_CAST(0xe89ba660)},
+{FXP_CAST(0x19f4f780), FXP_CAST(0x7641af00), FXP_CAST(0x64c4e100)}, {FXP_CAST(0x7ffbf580), FXP_CAST(0x89be5100), FXP_CAST(0x7493a380)}, {FXP_CAST(0x1de18100), FXP_CAST(0x30fbc540), FXP_CAST(0x070897f0)},
+{FXP_CAST(0x8d0d6a80), FXP_CAST(0x30fbc540), FXP_CAST(0x91ed6f00)}, {FXP_CAST(0xaff81380), FXP_CAST(0x89be5100), FXP_CAST(0x932db000)}, {FXP_CAST(0x5007fb00), FXP_CAST(0x7641af00), FXP_CAST(0x0970feb0)},
+{FXP_CAST(0x72f28d00), FXP_CAST(0xcf043ac0), FXP_CAST(0x758d6500)}, {FXP_CAST(0xe21e6cc0), FXP_CAST(0xcf043ac0), FXP_CAST(0x63436f80)}, {FXP_CAST(0x80040b00), FXP_CAST(0x7641af00), FXP_CAST(0xe63d7600)},
+{FXP_CAST(0xe60b1ae0), FXP_CAST(0x89be5100), FXP_CAST(0x84ea5c80)}, {FXP_CAST(0x74a8e100), FXP_CAST(0x30fbc540), FXP_CAST(0xa7f07500)}, {FXP_CAST(0x4cda8980), FXP_CAST(0x30fbc540), FXP_CAST(0x29a6d340)},
+{FXP_CAST(0xacdeda80), FXP_CAST(0x89be5100), FXP_CAST(0x7e93d600)}, {FXP_CAST(0x8ee0c980), FXP_CAST(0x7641af00), FXP_CAST(0x4b662680)}, {FXP_CAST(0x21c6a280), FXP_CAST(0xcf043ac0), FXP_CAST(0xc7258c80)},
+{FXP_CAST(0x7fdb9f00), FXP_CAST(0xcf043ac0), FXP_CAST(0x8006d500)}, {FXP_CAST(0x1601ba60), FXP_CAST(0x7641af00), FXP_CAST(0xc2830940)}, {FXP_CAST(0x89be4c80), FXP_CAST(0x89be5100), FXP_CAST(0x471cf100)},
+{FXP_CAST(0xb6664400), FXP_CAST(0x30fbc540), FXP_CAST(0x7f3fb800)}};
+
+const FIXP_PS aaFractDelayPhaseFactorSerImQmf[NO_QMF_CHANNELS][NO_SERIAL_ALLPASS_LINKS] = {
+{FXP_CAST(0xaff80c80), FXP_CAST(0x89be5100), FXP_CAST(0xbda29e00)}, {FXP_CAST(0x8d0d6f00), FXP_CAST(0x30fbc540), FXP_CAST(0x8043ee80)}, {FXP_CAST(0x1de18a20), FXP_CAST(0x30fbc540), FXP_CAST(0xcc3e7840)},
+{FXP_CAST(0x7ffbf500), FXP_CAST(0x89be5100), FXP_CAST(0x4fdfc180)}, {FXP_CAST(0x19f4ee40), FXP_CAST(0x7641af00), FXP_CAST(0x7d9e4c00)}, {FXP_CAST(0x8b572300), FXP_CAST(0xcf043ac0), FXP_CAST(0x244a2940)},
+{FXP_CAST(0xb3256f00), FXP_CAST(0xcf043ac0), FXP_CAST(0xa3f0a500)}, {FXP_CAST(0x53211e00), FXP_CAST(0x7641af00), FXP_CAST(0x86944500)}, {FXP_CAST(0x711f3a80), FXP_CAST(0x89be5100), FXP_CAST(0xebc72040)},
+{FXP_CAST(0xde3966c0), FXP_CAST(0x30fbc540), FXP_CAST(0x66b87e00)}, {FXP_CAST(0x80246080), FXP_CAST(0x30fbc540), FXP_CAST(0x73362c00)}, {FXP_CAST(0xe9fe3c40), FXP_CAST(0x89be5100), FXP_CAST(0x03d1d110)},
+{FXP_CAST(0x7641b000), FXP_CAST(0x7641af00), FXP_CAST(0x90520c80)}, {FXP_CAST(0x4999c380), FXP_CAST(0xcf043ac0), FXP_CAST(0x94e80a80)}, {FXP_CAST(0xa9dab800), FXP_CAST(0xcf043ac0), FXP_CAST(0x0ca570e0)},
+{FXP_CAST(0x90d0b000), FXP_CAST(0x7641af00), FXP_CAST(0x76c9bc80)}, {FXP_CAST(0x25a32000), FXP_CAST(0x89be5100), FXP_CAST(0x61338500)}, {FXP_CAST(0x7f9afc80), FXP_CAST(0x30fbc540), FXP_CAST(0xe318f060)},
+{FXP_CAST(0x120909c0), FXP_CAST(0x30fbc540), FXP_CAST(0x84124e00)}, {FXP_CAST(0x88435d80), FXP_CAST(0x89be5100), FXP_CAST(0xaa4d2f80)}, {FXP_CAST(0xb9b9a200), FXP_CAST(0x7641af00), FXP_CAST(0x2cae1800)},
+{FXP_CAST(0x5913ae80), FXP_CAST(0xcf043ac0), FXP_CAST(0x7f040680)}, {FXP_CAST(0x6d234e00), FXP_CAST(0xcf043ac0), FXP_CAST(0x48c6a100)}, {FXP_CAST(0xd689db80), FXP_CAST(0x7641af00), FXP_CAST(0xc44860c0)},
+{FXP_CAST(0x80c5e380), FXP_CAST(0x89be5100), FXP_CAST(0x80005d00)}, {FXP_CAST(0xf1f43eb0), FXP_CAST(0x30fbc540), FXP_CAST(0xc55a3a00)}, {FXP_CAST(0x79195500), FXP_CAST(0x30fbc540), FXP_CAST(0x49c3de00)},
+{FXP_CAST(0x42e13700), FXP_CAST(0x89be5100), FXP_CAST(0x7edc5b00)}, {FXP_CAST(0xa4146c80), FXP_CAST(0x7641af00), FXP_CAST(0x2b8c2c00)}, {FXP_CAST(0x95044680), FXP_CAST(0xcf043ac0), FXP_CAST(0xa968c100)},
+{FXP_CAST(0x2d3eaf40), FXP_CAST(0xcf043ac0), FXP_CAST(0x8460fd80)}, {FXP_CAST(0x7eb91780), FXP_CAST(0x7641af00), FXP_CAST(0xe44621e0)}, {FXP_CAST(0x0a0aec80), FXP_CAST(0x89be5100), FXP_CAST(0x61fb5c00)},
+{FXP_CAST(0x85a89100), FXP_CAST(0x30fbc540), FXP_CAST(0x76555780)}, {FXP_CAST(0xc094d500), FXP_CAST(0x30fbc540), FXP_CAST(0x0b71f790)}, {FXP_CAST(0x5eac4000), FXP_CAST(0x89be5100), FXP_CAST(0x94401a80)},
+{FXP_CAST(0x68b91d80), FXP_CAST(0x7641af00), FXP_CAST(0x90ea3980)}, {FXP_CAST(0xcf043440), FXP_CAST(0xcf043ac0), FXP_CAST(0x05067a08)}, {FXP_CAST(0x81e7f180), FXP_CAST(0xcf043ac0), FXP_CAST(0x73bb6d00)},
+{FXP_CAST(0xf9f871e0), FXP_CAST(0x7641af00), FXP_CAST(0x65ff0e00)}, {FXP_CAST(0x7b76a000), FXP_CAST(0x89be5100), FXP_CAST(0xea9664c0)}, {FXP_CAST(0x3be518c0), FXP_CAST(0x30fbc540), FXP_CAST(0x8633e880)},
+{FXP_CAST(0x9eaaff00), FXP_CAST(0x30fbc540), FXP_CAST(0xa4c84500)}, {FXP_CAST(0x99a3f400), FXP_CAST(0x89be5100), FXP_CAST(0x2571eac0)}, {FXP_CAST(0x34ac8840), FXP_CAST(0x7641af00), FXP_CAST(0x7dd82b00)},
+{FXP_CAST(0x7d572a80), FXP_CAST(0xcf043ac0), FXP_CAST(0x4eed8400)}, {FXP_CAST(0x0202a9c4), FXP_CAST(0xcf043ac0), FXP_CAST(0xcb249700)}, {FXP_CAST(0x83896000), FXP_CAST(0x7641af00), FXP_CAST(0x80318200)},
+{FXP_CAST(0xc7b01b00), FXP_CAST(0x89be5100), FXP_CAST(0xbeab7580)}, {FXP_CAST(0x63e52a80), FXP_CAST(0x30fbc540), FXP_CAST(0x4364b700)}, {FXP_CAST(0x63e51f00), FXP_CAST(0x30fbc540), FXP_CAST(0x7fa6bd00)},
+{FXP_CAST(0xc7b00a00), FXP_CAST(0x89be5100), FXP_CAST(0x32a67940)}, {FXP_CAST(0x83896400), FXP_CAST(0x7641af00), FXP_CAST(0xaf2fd200)}, {FXP_CAST(0x0202bc9c), FXP_CAST(0xcf043ac0), FXP_CAST(0x829e6e80)},
+{FXP_CAST(0x7d572e80), FXP_CAST(0xcf043ac0), FXP_CAST(0xdcde6b80)}, {FXP_CAST(0x34ac7700), FXP_CAST(0x7641af00), FXP_CAST(0x5ce4e280)}, {FXP_CAST(0x99a3e880), FXP_CAST(0x89be5100), FXP_CAST(0x79089c00)},
+{FXP_CAST(0x9eab0b80), FXP_CAST(0x30fbc540), FXP_CAST(0x1307ae80)}, {FXP_CAST(0x3be52980), FXP_CAST(0x30fbc540), FXP_CAST(0x98906880)}, {FXP_CAST(0x7b769b00), FXP_CAST(0x89be5100), FXP_CAST(0x8d51b300)},
+{FXP_CAST(0xf9f85f10), FXP_CAST(0x7641af00), FXP_CAST(0xfd62ee24)}, {FXP_CAST(0x81e7ee00), FXP_CAST(0xcf043ac0), FXP_CAST(0x70439680)}, {FXP_CAST(0xcf044580), FXP_CAST(0xcf043ac0), FXP_CAST(0x6a6d9600)},
+{FXP_CAST(0x68b92800), FXP_CAST(0x7641af00), FXP_CAST(0xf2275f80)}};
+
+const FIXP_PS aaFractDelayPhaseFactorSerReSubQmf20[NO_SUB_QMF_CHANNELS][NO_SERIAL_ALLPASS_LINKS] = {
+{FXP_CAST(0x7e2df000), FXP_CAST(0x7a7d0580), FXP_CAST(0x7ed03e00)}, {FXP_CAST(0x6fec9a80), FXP_CAST(0x5133cc80), FXP_CAST(0x7573df00)}, {FXP_CAST(0x55063900), FXP_CAST(0x0c8bd360), FXP_CAST(0x636c0400)},
+{FXP_CAST(0x3084ca00), FXP_CAST(0xc3a94580), FXP_CAST(0x4a0d6700)}, {FXP_CAST(0x80000000), FXP_CAST(0x80000000), FXP_CAST(0x80000000)}, {FXP_CAST(0x80000000), FXP_CAST(0x80000000), FXP_CAST(0x80000000)},
+{FXP_CAST(0x6fec9a80), FXP_CAST(0x5133cc80), FXP_CAST(0x7573df00)}, {FXP_CAST(0x7e2df000), FXP_CAST(0x7a7d0580), FXP_CAST(0x7ed03e00)}, {FXP_CAST(0xa4c84280), FXP_CAST(0xb8e31300), FXP_CAST(0xd5af0140)},
+{FXP_CAST(0xf0f488a0), FXP_CAST(0x8275a100), FXP_CAST(0x1a72e360)}, {FXP_CAST(0x80aaa680), FXP_CAST(0x471ced00), FXP_CAST(0x9d2ead80)}, {FXP_CAST(0x9477d100), FXP_CAST(0x7d8a5f00), FXP_CAST(0x8151df80)}};
+
+const FIXP_PS aaFractDelayPhaseFactorSerImSubQmf20[NO_SUB_QMF_CHANNELS][NO_SERIAL_ALLPASS_LINKS] = {
+{FXP_CAST(0xea7d08a0), FXP_CAST(0xdad7f3c0), FXP_CAST(0xee9c9f60)}, {FXP_CAST(0xc1e54140), FXP_CAST(0x9d0dfe80), FXP_CAST(0xcd1e7300)}, {FXP_CAST(0xa051a580), FXP_CAST(0x809dc980), FXP_CAST(0xaf61c400)},
+{FXP_CAST(0x898d4e00), FXP_CAST(0x8f1d3400), FXP_CAST(0x97988280)}, {FXP_CAST(0x00000000), FXP_CAST(0x00000000), FXP_CAST(0x00000000)}, {FXP_CAST(0x00000000), FXP_CAST(0x00000000), FXP_CAST(0x00000000)},
+{FXP_CAST(0x3e1abec0), FXP_CAST(0x62f20180), FXP_CAST(0x32e18d00)}, {FXP_CAST(0x1582f760), FXP_CAST(0x25280c40), FXP_CAST(0x116360a0)}, {FXP_CAST(0xa6343800), FXP_CAST(0x6a6d9880), FXP_CAST(0x87327a00)},
+{FXP_CAST(0x80e32200), FXP_CAST(0xe70747c0), FXP_CAST(0x82c32b00)}, {FXP_CAST(0xf2f42420), FXP_CAST(0x6a6d9880), FXP_CAST(0xaea47080)}, {FXP_CAST(0x456eba00), FXP_CAST(0xe70747c0), FXP_CAST(0xedaa8640)}};
+
+const FIXP_PS p8_13_20[13] =
+{
+  FL2FXCONST_PS(0.00746082949812f),  FL2FXCONST_PS(0.02270420949825f),  FL2FXCONST_PS(0.04546865930473f),  FL2FXCONST_PS(0.07266113929591f),
+  FL2FXCONST_PS(0.09885108575264f),  FL2FXCONST_PS(0.11793710567217f),  FL2FXCONST_PS(0.125f           ),  FL2FXCONST_PS(0.11793710567217f),
+  FL2FXCONST_PS(0.09885108575264f),  FL2FXCONST_PS(0.07266113929591f),  FL2FXCONST_PS(0.04546865930473f),  FL2FXCONST_PS(0.02270420949825f),
+  FL2FXCONST_PS(0.00746082949812f)
+};
+
+const FIXP_PS p2_13_20[13] =
+{
+  FL2FXCONST_PS(0.0f), FL2FXCONST_PS( 0.01899487526049f),  FL2FXCONST_PS(0.0f), FL2FXCONST_PS(-0.07293139167538f),
+  FL2FXCONST_PS(0.0f), FL2FXCONST_PS( 0.30596630545168f),  FL2FXCONST_PS(0.5f), FL2FXCONST_PS( 0.30596630545168f),
+  FL2FXCONST_PS(0.0f), FL2FXCONST_PS(-0.07293139167538f),  FL2FXCONST_PS(0.0f), FL2FXCONST_PS( 0.01899487526049f),
+  FL2FXCONST_PS(0.0f)
+};
+
+
+
+const UCHAR aAllpassLinkDelaySer[] = {    3,     4,     5};
+
+const UCHAR delayIndexQmf[NO_QMF_CHANNELS] = {
+  14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14,
+  14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14,
+  14, 14, 14,  1,  1,  1,  1,  1,  1,  1,  1,  1,  1,  1,  1,  1,
+   1,  1,  1,  1,  1,  1,  1,  1,  1,  1,  1,  1,  1,  1,  1,  1
+};
+
+const UCHAR groupBorders20[NO_IID_GROUPS + 1] =
+{
+   6,  7,  0,  1,  2,  3,                            /* 6 subqmf subbands - 0th qmf subband */
+   9,  8,                                            /* 2 subqmf subbands - 1st qmf subband */
+  10, 11,                                            /* 2 subqmf subbands - 2nd qmf subband */
+   3,  4,  5,  6,  7,  8,
+   9,  11, 14, 18, 23, 35, 64
+};
+
+const UCHAR groupBorders34[NO_IID_GROUPS_HI_RES + 1] =
+{
+   0,  1,  2,  3,  4,  5,  6,  7,  8,  9,  10, 11,   /* 12 subqmf subbands - 0th qmf subband */
+  12, 13, 14, 15, 16, 17, 18, 19,                    /*  8 subqmf subbands - 1st qmf subband */
+  20, 21, 22, 23,                                    /*  4 subqmf subbands - 2nd qmf subband */
+  24, 25, 26, 27,                                    /*  4 subqmf subbands - 3nd qmf subband */
+  28, 29, 30, 31,                                    /*  4 subqmf subbands - 4nd qmf subband */
+  32-27, 33-27, 34-27, 35-27, 36-27, 37-27, 38-27,
+  40-27, 42-27, 44-27, 46-27, 48-27, 51-27, 54-27,
+  57-27, 60-27, 64-27, 68-27, 91-27
+};
+
+const UCHAR bins2groupMap20[NO_IID_GROUPS] =
+{
+  1, 0,
+  0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 17, 18, 19
+};
+
+const UCHAR quantizedIIDs[NO_IID_STEPS] =
+{
+    2, 4, 7, 10, 14, 18, 25
+};
+const UCHAR quantizedIIDsFine[NO_IID_STEPS_FINE] =
+{
+    2, 4, 6, 8, 10, 13, 16, 19, 22, 25, 30, 35, 40, 45, 50
+};
+
+const UCHAR FDK_sbrDecoder_aNoIidBins[3] = {NO_LOW_RES_IID_BINS,
+                           NO_MID_RES_IID_BINS,
+                           NO_HI_RES_IID_BINS};
+
+const UCHAR FDK_sbrDecoder_aNoIccBins[3] = {NO_LOW_RES_ICC_BINS,
+                           NO_MID_RES_ICC_BINS,
+                           NO_HI_RES_ICC_BINS};
+
+
+
+/************************************************************************/
+/*!
+   \brief   Create lookup tables for some arithmetic functions
+
+   The tables would normally be defined as const arrays,
+   but initialization at run time allows to specify their accuracy.
+*/
+/************************************************************************/
+
+/*   1/x-table:  (example for INV_TABLE_BITS 8)
+
+     The table covers an input range from 0.5 to 1.0 with a step size of 1/512,
+     starting at 0.5 + 1/512.
+     Each table entry corresponds to an input interval starting 1/1024 below the
+     exact value and ending 1/1024 above it.
+
+     The table is actually a 0.5/x-table, so that the output range is again
+     0.5...1.0 and the exponent of the result must be increased by 1.
+
+     Input range           Index in table      result
+     -------------------------------------------------------------------
+     0.500000...0.500976          -            0.5 / 0.500000 = 1.000000
+     0.500976...0.502930          0            0.5 / 0.501953 = 0.996109
+     0.502930...0.500488          1            0.5 / 0.503906 = 0.992248
+             ...
+     0.999023...1.000000         255           0.5 / 1.000000 = 0.500000
+
+       for (i=0; i<INV_TABLE_SIZE; i++) {
+         d = 0.5f / ( 0.5f+(double)(i+1)/(INV_TABLE_SIZE*2) ) ;
+         invTable[i] = FL2FX_SGL(d);
+       }
+*/
+const FIXP_SGL FDK_sbrDecoder_invTable[INV_TABLE_SIZE] =
+{
+  0x7f80, 0x7f01, 0x7e83, 0x7e07, 0x7d8b, 0x7d11, 0x7c97, 0x7c1e,
+  0x7ba6, 0x7b2f, 0x7ab9, 0x7a44, 0x79cf, 0x795c, 0x78e9, 0x7878,
+  0x7807, 0x7796, 0x7727, 0x76b9, 0x764b, 0x75de, 0x7572, 0x7506,
+  0x749c, 0x7432, 0x73c9, 0x7360, 0x72f9, 0x7292, 0x722c, 0x71c6,
+  0x7161, 0x70fd, 0x709a, 0x7037, 0x6fd5, 0x6f74, 0x6f13, 0x6eb3,
+  0x6e54, 0x6df5, 0x6d97, 0x6d39, 0x6cdc, 0x6c80, 0x6c24, 0x6bc9,
+  0x6b6f, 0x6b15, 0x6abc, 0x6a63, 0x6a0b, 0x69b3, 0x695c, 0x6906,
+  0x68b0, 0x685a, 0x6806, 0x67b1, 0x675e, 0x670a, 0x66b8, 0x6666,
+  0x6614, 0x65c3, 0x6572, 0x6522, 0x64d2, 0x6483, 0x6434, 0x63e6,
+  0x6399, 0x634b, 0x62fe, 0x62b2, 0x6266, 0x621b, 0x61d0, 0x6185,
+  0x613b, 0x60f2, 0x60a8, 0x6060, 0x6017, 0x5fcf, 0x5f88, 0x5f41,
+  0x5efa, 0x5eb4, 0x5e6e, 0x5e28, 0x5de3, 0x5d9f, 0x5d5a, 0x5d17,
+  0x5cd3, 0x5c90, 0x5c4d, 0x5c0b, 0x5bc9, 0x5b87, 0x5b46, 0x5b05,
+  0x5ac4, 0x5a84, 0x5a44, 0x5a05, 0x59c6, 0x5987, 0x5949, 0x590a,
+  0x58cd, 0x588f, 0x5852, 0x5815, 0x57d9, 0x579d, 0x5761, 0x5725,
+  0x56ea, 0x56af, 0x5675, 0x563b, 0x5601, 0x55c7, 0x558e, 0x5555,
+  0x551c, 0x54e3, 0x54ab, 0x5473, 0x543c, 0x5405, 0x53ce, 0x5397,
+  0x5360, 0x532a, 0x52f4, 0x52bf, 0x5289, 0x5254, 0x521f, 0x51eb,
+  0x51b7, 0x5183, 0x514f, 0x511b, 0x50e8, 0x50b5, 0x5082, 0x5050,
+  0x501d, 0x4feb, 0x4fba, 0x4f88, 0x4f57, 0x4f26, 0x4ef5, 0x4ec4,
+  0x4e94, 0x4e64, 0x4e34, 0x4e04, 0x4dd5, 0x4da6, 0x4d77, 0x4d48,
+  0x4d19, 0x4ceb, 0x4cbd, 0x4c8f, 0x4c61, 0x4c34, 0x4c07, 0x4bd9,
+  0x4bad, 0x4b80, 0x4b54, 0x4b27, 0x4afb, 0x4acf, 0x4aa4, 0x4a78,
+  0x4a4d, 0x4a22, 0x49f7, 0x49cd, 0x49a2, 0x4978, 0x494e, 0x4924,
+  0x48fa, 0x48d1, 0x48a7, 0x487e, 0x4855, 0x482d, 0x4804, 0x47dc,
+  0x47b3, 0x478b, 0x4763, 0x473c, 0x4714, 0x46ed, 0x46c5, 0x469e,
+  0x4677, 0x4651, 0x462a, 0x4604, 0x45de, 0x45b8, 0x4592, 0x456c,
+  0x4546, 0x4521, 0x44fc, 0x44d7, 0x44b2, 0x448d, 0x4468, 0x4444,
+  0x441f, 0x43fb, 0x43d7, 0x43b3, 0x4390, 0x436c, 0x4349, 0x4325,
+  0x4302, 0x42df, 0x42bc, 0x4299, 0x4277, 0x4254, 0x4232, 0x4210,
+  0x41ee, 0x41cc, 0x41aa, 0x4189, 0x4167, 0x4146, 0x4125, 0x4104,
+  0x40e3, 0x40c2, 0x40a1, 0x4081, 0x4060, 0x4040, 0x4020, 0x4000
+};
+
diff --git a/libSBRdec/src/sbr_rom.h b/libSBRdec/src/sbr_rom.h
new file mode 100644
index 0000000..912f7e4
--- /dev/null
+++ b/libSBRdec/src/sbr_rom.h
@@ -0,0 +1,232 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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 __rom_H
+#define __rom_H
+
+#include "sbrdecoder.h"
+#include "env_extr.h"
+#include "qmf.h"
+
+#define INV_INT_TABLE_SIZE      49
+#define SBR_NF_NO_RANDOM_VAL  512   /*!< Size of random number array for noise floor */
+
+/*
+  Frequency scales
+*/
+extern const UCHAR FDK_sbrDecoder_sbr_start_freq_16[16];
+extern const UCHAR FDK_sbrDecoder_sbr_start_freq_22[16];
+extern const UCHAR FDK_sbrDecoder_sbr_start_freq_24[16];
+extern const UCHAR FDK_sbrDecoder_sbr_start_freq_32[16];
+extern const UCHAR FDK_sbrDecoder_sbr_start_freq_40[16];
+extern const UCHAR FDK_sbrDecoder_sbr_start_freq_44[16];
+extern const UCHAR FDK_sbrDecoder_sbr_start_freq_48[16];
+
+/*
+  Low-Power-Profile Transposer
+*/
+#define NUM_WHFACTOR_TABLE_ENTRIES  9
+extern const USHORT FDK_sbrDecoder_sbr_whFactorsIndex[NUM_WHFACTOR_TABLE_ENTRIES];
+extern const FIXP_DBL FDK_sbrDecoder_sbr_whFactorsTable[NUM_WHFACTOR_TABLE_ENTRIES][6];
+
+
+
+/*
+  Envelope Adjustor
+*/
+extern const FIXP_SGL FDK_sbrDecoder_sbr_limGains_m[4];
+extern const UCHAR    FDK_sbrDecoder_sbr_limGains_e[4];
+extern const FIXP_SGL FDK_sbrDecoder_sbr_limiterBandsPerOctaveDiv4[4];
+extern const FIXP_SGL FDK_sbrDecoder_sbr_smoothFilter[4];
+extern const FIXP_SGL FDK_sbrDecoder_sbr_randomPhase[SBR_NF_NO_RANDOM_VAL][2];
+extern const FIXP_SGL harmonicPhaseX [2][4];
+
+/*
+  Envelope Extractor
+*/
+extern const int FDK_sbrDecoder_envelopeTable_8  [8][5];
+extern const int FDK_sbrDecoder_envelopeTable_15 [15][6];
+extern const int FDK_sbrDecoder_envelopeTable_16 [16][6];
+
+extern const FRAME_INFO FDK_sbrDecoder_sbr_frame_info1_15;
+extern const FRAME_INFO FDK_sbrDecoder_sbr_frame_info2_15;
+extern const FRAME_INFO FDK_sbrDecoder_sbr_frame_info4_15;
+extern const FRAME_INFO FDK_sbrDecoder_sbr_frame_info8_15;
+
+extern const FRAME_INFO FDK_sbrDecoder_sbr_frame_info1_16;
+extern const FRAME_INFO FDK_sbrDecoder_sbr_frame_info2_16;
+extern const FRAME_INFO FDK_sbrDecoder_sbr_frame_info4_16;
+extern const FRAME_INFO FDK_sbrDecoder_sbr_frame_info8_16;
+
+extern const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvLevel10T[120][2];
+extern const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvLevel10F[120][2];
+extern const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvBalance10T[48][2];
+extern const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvBalance10F[48][2];
+extern const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvLevel11T[62][2];
+extern const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvLevel11F[62][2];
+extern const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvBalance11T[24][2];
+extern const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvBalance11F[24][2];
+extern const SCHAR FDK_sbrDecoder_sbr_huffBook_NoiseLevel11T[62][2];
+extern const SCHAR FDK_sbrDecoder_sbr_huffBook_NoiseBalance11T[24][2];
+
+
+/*
+ Parametric stereo
+*/
+
+
+extern const FIXP_DBL decayScaleFactTable[NO_QMF_CHANNELS];
+
+/* FIX_BORDER can have 0, 1, 2, 4 envelops */
+extern const UCHAR FDK_sbrDecoder_aFixNoEnvDecode[4];
+
+/* IID & ICC Huffman codebooks */
+extern const SCHAR aBookPsIidTimeDecode[28][2];
+extern const SCHAR aBookPsIidFreqDecode[28][2];
+extern const SCHAR aBookPsIccTimeDecode[14][2];
+extern const SCHAR aBookPsIccFreqDecode[14][2];
+
+/* IID-fine Huffman codebooks */
+
+extern const SCHAR aBookPsIidFineTimeDecode[60][2];
+extern const SCHAR aBookPsIidFineFreqDecode[60][2];
+
+/* the values of the following 3 tables are shiftet right by 1 ! */
+extern const FIXP_DBL ScaleFactors[NO_IID_LEVELS];
+extern const FIXP_DBL ScaleFactorsFine[NO_IID_LEVELS_FINE];
+extern const FIXP_DBL Alphas[NO_ICC_LEVELS];
+
+#if defined(ARCH_PREFER_MULT_32x16)
+extern const FIXP_SGL aAllpassLinkDecaySer[NO_SERIAL_ALLPASS_LINKS];
+extern const FIXP_SGL aaFractDelayPhaseFactorReQmf[NO_QMF_CHANNELS];
+extern const FIXP_SGL aaFractDelayPhaseFactorImQmf[NO_QMF_CHANNELS];
+extern const FIXP_SGL aaFractDelayPhaseFactorReSubQmf20[NO_SUB_QMF_CHANNELS];
+extern const FIXP_SGL aaFractDelayPhaseFactorImSubQmf20[NO_SUB_QMF_CHANNELS];
+
+extern const FIXP_SGL aaFractDelayPhaseFactorSerReQmf[NO_QMF_CHANNELS][NO_SERIAL_ALLPASS_LINKS];
+extern const FIXP_SGL aaFractDelayPhaseFactorSerImQmf[NO_QMF_CHANNELS][NO_SERIAL_ALLPASS_LINKS];
+extern const FIXP_SGL aaFractDelayPhaseFactorSerReSubQmf20[NO_SUB_QMF_CHANNELS][NO_SERIAL_ALLPASS_LINKS];
+extern const FIXP_SGL aaFractDelayPhaseFactorSerImSubQmf20[NO_SUB_QMF_CHANNELS][NO_SERIAL_ALLPASS_LINKS];
+
+extern const FIXP_SGL p8_13_20[13];
+extern const FIXP_SGL p2_13_20[13];
+
+#else
+extern const FIXP_DBL aAllpassLinkDecaySer[NO_SERIAL_ALLPASS_LINKS];
+extern const FIXP_DBL aaFractDelayPhaseFactorReQmf[NO_QMF_CHANNELS];
+extern const FIXP_DBL aaFractDelayPhaseFactorImQmf[NO_QMF_CHANNELS];
+extern const FIXP_DBL aaFractDelayPhaseFactorReSubQmf20[NO_SUB_QMF_CHANNELS];
+extern const FIXP_DBL aaFractDelayPhaseFactorImSubQmf20[NO_SUB_QMF_CHANNELS];
+
+extern const FIXP_DBL aaFractDelayPhaseFactorSerReQmf[NO_QMF_CHANNELS][NO_SERIAL_ALLPASS_LINKS];
+extern const FIXP_DBL aaFractDelayPhaseFactorSerImQmf[NO_QMF_CHANNELS][NO_SERIAL_ALLPASS_LINKS];
+extern const FIXP_DBL aaFractDelayPhaseFactorSerReSubQmf20[NO_SUB_QMF_CHANNELS][NO_SERIAL_ALLPASS_LINKS];
+extern const FIXP_DBL aaFractDelayPhaseFactorSerImSubQmf20[NO_SUB_QMF_CHANNELS][NO_SERIAL_ALLPASS_LINKS];
+
+extern const FIXP_DBL p8_13_20[13];
+extern const FIXP_DBL p2_13_20[13];
+#endif
+
+extern const UCHAR aAllpassLinkDelaySer[3];
+extern const UCHAR delayIndexQmf[NO_QMF_CHANNELS];
+extern const UCHAR groupBorders20[NO_IID_GROUPS + 1];
+extern const UCHAR groupBorders34[NO_IID_GROUPS_HI_RES + 1];
+extern const UCHAR bins2groupMap20[NO_IID_GROUPS];
+extern const UCHAR quantizedIIDs[NO_IID_STEPS];
+extern const UCHAR quantizedIIDsFine[NO_IID_STEPS_FINE];
+extern const UCHAR FDK_sbrDecoder_aNoIidBins[3];
+extern const UCHAR FDK_sbrDecoder_aNoIccBins[3];
+
+
+/* Lookup tables for some arithmetic functions */
+
+#define INV_TABLE_BITS 8
+#define INV_TABLE_SIZE (1<<INV_TABLE_BITS)
+extern const FIXP_SGL FDK_sbrDecoder_invTable[INV_TABLE_SIZE];
+
+#endif // __rom_H
diff --git a/libSBRdec/src/sbr_scale.h b/libSBRdec/src/sbr_scale.h
new file mode 100644
index 0000000..a98fc05
--- /dev/null
+++ b/libSBRdec/src/sbr_scale.h
@@ -0,0 +1,123 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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 scaling factors,  
+To deal with the dynamic range in the different processing stages, a
+fixed point specific code has to rely on scaling factors. A floating
+point code carries a scaling factor -- the exponent -- for each value,
+so scaling is not necessary there.
+
+The output of the core decoder (low band) is scaled up to cover as much
+as possible bits for each value. As high band and low band are processed
+in different algorithm sections, they require their own scaling
+factors. In addition, any static buffers, e.g. filter states, require a
+separate scaling factor as well. The code takes care to do the proper
+adjustment, if scaling factors of a filter state and the time signal differ.
+
+\sa #QMF_SCALE_FACTOR, \ref documentationOverview
+*/
+
+#ifndef __SBR_SCALE_H
+#define __SBR_SCALE_H
+
+/*!
+\verbatim
+  scale:
+    0             left aligned e.g. |max| >=0.5
+    FRACT_BITS-1  zero e.g |max| = 0
+\endverbatim
+
+  Dynamic scaling is used to achieve sufficient accuracy even when the signal
+  energy is low.  The dynamic framing of SBR produces a variable overlap area
+  where samples from the previous QMF-Analysis are stored. Depending on the
+  start position and stop position of the current SBR envelopes, the processing
+  buffer consists of differently scaled regions like illustrated in the below
+  figure.
+
+  \image html scales.png Scale
+*/
+
+
+#endif
diff --git a/libSBRdec/src/sbrdec_drc.cpp b/libSBRdec/src/sbrdec_drc.cpp
new file mode 100644
index 0000000..ce5247a
--- /dev/null
+++ b/libSBRdec/src/sbrdec_drc.cpp
@@ -0,0 +1,512 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/*****************************  MPEG-4 AAC Decoder  **************************
+
+   Author(s):   Christian Griebel
+   Description: Dynamic range control (DRC) decoder tool for SBR
+
+******************************************************************************/
+
+#include "sbrdec_drc.h"
+
+
+/* DRC - Offset table for QMF interpolation. */
+static const int offsetTab[2][16] =
+{
+  { 0, 4, 8, 12, 16, 20, 24, 28, 0, 0, 0, 0, 0, 0, 0, 0 },  /* 1024 framing */
+  { 0, 4, 8, 12, 16, 19, 22, 26, 0, 0, 0, 0, 0, 0, 0, 0 }   /*  960 framing */
+};
+
+/*!
+  \brief Initialize DRC QMF factors
+
+  \hDrcData Handle to DRC channel data.
+
+  \return none
+*/
+void sbrDecoder_drcInitChannel (
+    HANDLE_SBR_DRC_CHANNEL  hDrcData )
+{
+  int band;
+
+  if (hDrcData == NULL) {
+    return;
+  }
+
+  for (band = 0; band < (64); band++) {
+    hDrcData->prevFact_mag[band] = (FIXP_DBL)MAXVAL_DBL /*FL2FXCONST_DBL(1.0f)*/;
+  }
+
+  for (band = 0; band < SBRDEC_MAX_DRC_BANDS; band++) {
+    hDrcData->currFact_mag[band] = (FIXP_DBL)MAXVAL_DBL /*FL2FXCONST_DBL(1.0f)*/;
+    hDrcData->nextFact_mag[band] = (FIXP_DBL)MAXVAL_DBL /*FL2FXCONST_DBL(1.0f)*/;
+  }
+
+  hDrcData->prevFact_exp = 0;
+  hDrcData->currFact_exp = 0;
+  hDrcData->nextFact_exp = 0;
+
+  hDrcData->numBandsCurr = 0;
+  hDrcData->numBandsNext = 0;
+
+  hDrcData->winSequenceCurr = 0;
+  hDrcData->winSequenceNext = 0;
+
+  hDrcData->drcInterpolationSchemeCurr = 0;
+  hDrcData->drcInterpolationSchemeNext = 0;
+
+  hDrcData->enable = 0;
+}
+
+
+/*!
+  \brief Swap DRC QMF scaling factors after they have been applied.
+
+  \hDrcData Handle to DRC channel data.
+
+  \return none
+*/
+void sbrDecoder_drcUpdateChannel (
+    HANDLE_SBR_DRC_CHANNEL  hDrcData )
+{
+  if (hDrcData == NULL) {
+    return;
+  }
+  if (hDrcData->enable != 1) {
+    return;
+  }
+
+  /* swap previous data */
+  FDKmemcpy( hDrcData->currFact_mag,
+             hDrcData->nextFact_mag,
+             SBRDEC_MAX_DRC_BANDS * sizeof(FIXP_DBL) );
+
+  hDrcData->currFact_exp = hDrcData->nextFact_exp;
+
+  hDrcData->numBandsCurr = hDrcData->numBandsNext;
+
+  FDKmemcpy( hDrcData->bandTopCurr,
+             hDrcData->bandTopNext,
+             SBRDEC_MAX_DRC_BANDS * sizeof(USHORT) );
+
+  hDrcData->drcInterpolationSchemeCurr = hDrcData->drcInterpolationSchemeNext;
+
+  hDrcData->winSequenceCurr = hDrcData->winSequenceNext;
+}
+
+
+/*!
+  \brief Apply DRC factors slot based.
+
+  \hDrcData Handle to DRC channel data.
+  \qmfRealSlot Pointer to real valued QMF data of one time slot.
+  \qmfImagSlot Pointer to the imaginary QMF data of one time slot.
+  \col Number of the time slot.
+  \numQmfSubSamples Total number of time slots for one frame.
+  \scaleFactor Pointer to the out scale factor of the time slot.
+
+  \return None.
+*/
+void sbrDecoder_drcApplySlot (
+    HANDLE_SBR_DRC_CHANNEL  hDrcData,
+    FIXP_DBL   *qmfRealSlot,
+    FIXP_DBL   *qmfImagSlot,
+    int  col,
+    int  numQmfSubSamples,
+    int  maxShift
+  )
+{
+  const int *offset;
+
+  int band, bottomMdct, topMdct, bin, useLP;
+  int indx = numQmfSubSamples - (numQmfSubSamples >> 1) - 10;   /* l_border */
+  int frameLenFlag = (numQmfSubSamples == 30) ? 1 : 0;
+
+  const FIXP_DBL *fact_mag = NULL;
+  INT fact_exp = 0;
+  UINT numBands = 0;
+  USHORT *bandTop = NULL;
+  int shortDrc = 0;
+
+  FIXP_DBL alphaValue = FL2FXCONST_DBL(0.0f);
+
+  if (hDrcData == NULL) {
+    return;
+  }
+  if (hDrcData->enable != 1) {
+    return;
+  }
+
+  offset = offsetTab[frameLenFlag];
+
+  useLP = (qmfImagSlot == NULL) ? 1 : 0;
+
+  col += indx;
+  bottomMdct = 0;
+  bin = 0;
+
+  /* get respective data and calc interpolation factor */
+  if (col < (numQmfSubSamples>>1)) {  /* first half of current frame */
+    if (hDrcData->winSequenceCurr != 2) { /* long window */
+      int j = col + (numQmfSubSamples>>1);
+
+      if (hDrcData->drcInterpolationSchemeCurr == 0) {
+        INT k = (frameLenFlag) ? 0x4444444 : 0x4000000;
+
+        alphaValue = (FIXP_DBL)(j * k);
+      }
+      else {
+        if (j >= offset[hDrcData->drcInterpolationSchemeCurr - 1]) {
+          alphaValue = FL2FXCONST_DBL(1.0f);
+        }
+      }
+    }
+    else {  /* short windows */
+      shortDrc = 1;
+    }
+
+    fact_mag = hDrcData->currFact_mag;
+    fact_exp = hDrcData->currFact_exp;
+    numBands = hDrcData->numBandsCurr;
+    bandTop = hDrcData->bandTopCurr;
+  }
+  else if (col < numQmfSubSamples) {  /* second half of current frame */
+    if (hDrcData->winSequenceNext != 2) { /* next: long window */
+      int j = col - (numQmfSubSamples>>1);
+
+      if (hDrcData->drcInterpolationSchemeNext == 0) {
+        INT k = (frameLenFlag) ? 0x4444444 : 0x4000000;
+
+        alphaValue = (FIXP_DBL)(j * k);
+      }
+      else {
+        if (j >= offset[hDrcData->drcInterpolationSchemeNext - 1]) {
+          alphaValue = FL2FXCONST_DBL(1.0f);
+        }
+      }
+
+      fact_mag = hDrcData->nextFact_mag;
+      fact_exp = hDrcData->nextFact_exp;
+      numBands = hDrcData->numBandsNext;
+      bandTop = hDrcData->bandTopNext;
+    }
+    else {  /* next: short windows */
+      if (hDrcData->winSequenceCurr != 2) {  /* current: long window */
+        alphaValue = (FIXP_DBL)0;
+
+        fact_mag = hDrcData->nextFact_mag;
+        fact_exp = hDrcData->nextFact_exp;
+        numBands = hDrcData->numBandsNext;
+        bandTop = hDrcData->bandTopNext;
+      }
+      else {  /* current: short windows */
+        shortDrc = 1;
+
+        fact_mag = hDrcData->currFact_mag;
+        fact_exp = hDrcData->currFact_exp;
+        numBands = hDrcData->numBandsCurr;
+        bandTop = hDrcData->bandTopCurr;
+      }
+    }
+  }
+  else {  /* first half of next frame */
+    if (hDrcData->winSequenceNext != 2) { /* long window */
+      int j = col - (numQmfSubSamples>>1);
+
+      if (hDrcData->drcInterpolationSchemeNext == 0) {
+        INT k = (frameLenFlag) ? 0x4444444 : 0x4000000;
+
+        alphaValue = (FIXP_DBL)(j * k);
+      }
+      else {
+        if (j >= offset[hDrcData->drcInterpolationSchemeNext - 1]) {
+          alphaValue = FL2FXCONST_DBL(1.0f);
+        }
+      }
+    }
+    else {  /* short windows */
+      shortDrc = 1;
+    }
+
+    fact_mag = hDrcData->nextFact_mag;
+    fact_exp = hDrcData->nextFact_exp;
+    numBands = hDrcData->numBandsNext;
+    bandTop = hDrcData->bandTopNext;
+
+    col -= numQmfSubSamples;
+  }
+
+
+  /* process bands */
+  for (band = 0; band < (int)numBands; band++) {
+    int bottomQmf, topQmf;
+
+    FIXP_DBL drcFact_mag = FL2FXCONST_DBL(1.0f);
+
+    topMdct = (bandTop[band]+1) << 2;
+
+    if (!shortDrc) {  /* long window */
+      if (frameLenFlag) {
+        /* 960 framing */
+        bottomMdct = 30 * (bottomMdct / 30);
+        topMdct    = 30 * (topMdct / 30);
+
+        bottomQmf = fMultIfloor((FIXP_DBL)0x4444444, bottomMdct);
+        topQmf    = fMultIfloor((FIXP_DBL)0x4444444, topMdct);
+      }
+      else {
+        /* 1024 framing */
+        bottomMdct &= ~0x1f;
+        topMdct    &= ~0x1f;
+
+        bottomQmf = bottomMdct >> 5;
+        topQmf    = topMdct >> 5;
+      }
+
+      if (band == ((int)numBands-1)) {
+        topQmf = (64);
+      }
+    
+      for (bin = bottomQmf; bin < topQmf; bin++) {
+        FIXP_DBL drcFact1_mag = hDrcData->prevFact_mag[bin];
+        FIXP_DBL drcFact2_mag = fact_mag[band];
+
+        /* normalize scale factors */
+        if (hDrcData->prevFact_exp < maxShift) {
+          drcFact1_mag >>= maxShift - hDrcData->prevFact_exp;
+        }
+        if (fact_exp < maxShift) {
+          drcFact2_mag >>= maxShift - fact_exp;
+        }
+
+        /* interpolate */
+        drcFact_mag = fMult(alphaValue, drcFact2_mag) + fMult((FL2FXCONST_DBL(1.0f) - alphaValue), drcFact1_mag);
+
+        /* apply scaling */
+        qmfRealSlot[bin] = fMult(qmfRealSlot[bin], drcFact_mag);
+        if (!useLP) {
+          qmfImagSlot[bin] = fMult(qmfImagSlot[bin], drcFact_mag);
+        }
+
+        /* save previous factors */
+        if (col == (numQmfSubSamples>>1)-1) {
+          hDrcData->prevFact_mag[bin] = fact_mag[band];
+        }
+      }
+    }
+    else {  /* short windows */
+      int startSample, stopSample;
+      FIXP_DBL invFrameSizeDiv8 = (frameLenFlag) ? (FIXP_DBL)0x1111111 : (FIXP_DBL)0x1000000;
+
+      if (frameLenFlag) {
+        /*  960 framing */
+        bottomMdct = 30/8 * (bottomMdct*8/30);
+        topMdct    = 30/8 * (topMdct*8/30);
+      }
+      else {
+        /* 1024 framing */
+        bottomMdct &= ~0x03;
+        topMdct    &= ~0x03;
+      }
+
+      /* startSample is truncated to the nearest corresponding start subsample in
+         the QMF of the short window bottom is present in:*/
+      startSample  = ((fMultIfloor( invFrameSizeDiv8, bottomMdct ) & 0x7) * numQmfSubSamples) >> 3;
+
+      /* stopSample is rounded upwards to the nearest corresponding stop subsample
+         in the QMF of the short window top is present in. */
+      stopSample  = ((fMultIceil( invFrameSizeDiv8, topMdct ) & 0xf) * numQmfSubSamples) >> 3;
+
+      bottomQmf = fMultIfloor( invFrameSizeDiv8, ((bottomMdct%(numQmfSubSamples<<2)) << 5) );
+      topQmf    = fMultIfloor( invFrameSizeDiv8, ((topMdct%(numQmfSubSamples<<2)) << 5) );
+
+      /* extend last band */
+      if (band == ((int)numBands-1)) {
+        topQmf = (64);
+        stopSample = numQmfSubSamples;
+      }
+
+      if (topQmf == 0) {
+        topQmf = (64);
+      }
+
+      /* save previous factors */
+      if (stopSample == numQmfSubSamples) {
+        int tmpBottom = bottomQmf;
+
+        if (((numQmfSubSamples-1) & ~0x03) > startSample) {
+            tmpBottom = 0;    /* band starts in previous short window */
+        }
+
+        for (bin = tmpBottom; bin < topQmf; bin++) {
+          hDrcData->prevFact_mag[bin] = fact_mag[band];
+        }
+      }
+
+      /* apply */
+      if ((col >= startSample) && (col < stopSample)) {
+        if ((col & ~0x03) > startSample) {
+            bottomQmf = 0;    /* band starts in previous short window */
+        }
+        if (col < ((stopSample-1) & ~0x03)) {
+            topQmf = (64);   /* band ends in next short window */
+        }
+
+        drcFact_mag = fact_mag[band];
+
+        /* normalize scale factor */
+        if (fact_exp < maxShift) {
+          drcFact_mag >>= maxShift - fact_exp;
+        }
+
+        /* apply scaling */
+        for (bin = bottomQmf; bin < topQmf; bin++) {
+          qmfRealSlot[bin] = fMult(qmfRealSlot[bin], drcFact_mag);
+          if (!useLP) {
+            qmfImagSlot[bin] = fMult(qmfImagSlot[bin], drcFact_mag);
+          }
+        }
+      }
+    }
+
+    bottomMdct = topMdct;
+  }   /* end of bands loop */
+
+  if (col == (numQmfSubSamples>>1)-1) {
+    hDrcData->prevFact_exp = fact_exp;
+  }
+}
+
+
+/*!
+  \brief Apply DRC factors frame based.
+
+  \hDrcData Handle to DRC channel data.
+  \qmfRealSlot Pointer to real valued QMF data of the whole frame.
+  \qmfImagSlot Pointer to the imaginary QMF data of the whole frame.
+  \numQmfSubSamples Total number of time slots for one frame.
+  \scaleFactor Pointer to the out scale factor of the frame.
+
+  \return None.
+*/
+void sbrDecoder_drcApply (
+    HANDLE_SBR_DRC_CHANNEL  hDrcData,
+    FIXP_DBL **QmfBufferReal,
+    FIXP_DBL **QmfBufferImag,
+    int  numQmfSubSamples,
+    int *scaleFactor
+  )
+{
+  int col;
+  int maxShift = 0;
+
+  /* get max scale factor */
+  if (hDrcData->prevFact_exp > maxShift) {
+    maxShift = hDrcData->prevFact_exp;
+  }
+  if (hDrcData->currFact_exp > maxShift) {
+    maxShift = hDrcData->currFact_exp;
+  }
+  if (hDrcData->nextFact_exp > maxShift) {
+    maxShift = hDrcData->nextFact_exp;
+  }
+
+  for (col = 0; col < numQmfSubSamples; col++)
+  {
+    FIXP_DBL *qmfSlotReal = QmfBufferReal[col];
+    FIXP_DBL *qmfSlotImag = (QmfBufferImag == NULL) ? NULL : QmfBufferImag[col];
+
+    sbrDecoder_drcApplySlot (
+      hDrcData,
+      qmfSlotReal,
+      qmfSlotImag,
+      col,
+      numQmfSubSamples,
+      maxShift
+    );
+  }
+
+  *scaleFactor += maxShift;
+}
+
diff --git a/libSBRdec/src/sbrdec_drc.h b/libSBRdec/src/sbrdec_drc.h
new file mode 100644
index 0000000..2577e89
--- /dev/null
+++ b/libSBRdec/src/sbrdec_drc.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
+----------------------------------------------------------------------------------------------------------- */
+
+/*****************************  MPEG-4 AAC Decoder  **************************
+
+   Author(s):   Christian Griebel
+   Description: Dynamic range control (DRC) decoder tool for SBR
+
+******************************************************************************/
+
+#ifndef _SBRDEC_DRC_H_
+#define _SBRDEC_DRC_H_
+
+#include "sbrdecoder.h"
+
+
+#define SBRDEC_MAX_DRC_CHANNELS  (6)
+#define SBRDEC_MAX_DRC_BANDS     ( 16 )
+
+typedef struct
+{
+  FIXP_DBL prevFact_mag[(64)];
+  INT      prevFact_exp;
+
+  FIXP_DBL currFact_mag[SBRDEC_MAX_DRC_BANDS];
+  FIXP_DBL nextFact_mag[SBRDEC_MAX_DRC_BANDS];
+  INT      currFact_exp;
+  INT      nextFact_exp;
+
+  UINT     numBandsCurr;
+  UINT     numBandsNext;
+  USHORT   bandTopCurr[SBRDEC_MAX_DRC_BANDS];
+  USHORT   bandTopNext[SBRDEC_MAX_DRC_BANDS];
+
+  SHORT    drcInterpolationSchemeCurr;
+  SHORT    drcInterpolationSchemeNext;
+
+  SHORT    enable;
+
+  UCHAR    winSequenceCurr;
+  UCHAR    winSequenceNext;
+
+} SBRDEC_DRC_CHANNEL;
+
+typedef SBRDEC_DRC_CHANNEL * HANDLE_SBR_DRC_CHANNEL;
+
+
+void sbrDecoder_drcInitChannel (
+    HANDLE_SBR_DRC_CHANNEL  hDrcData );
+
+void sbrDecoder_drcUpdateChannel (
+    HANDLE_SBR_DRC_CHANNEL  hDrcData );
+
+void sbrDecoder_drcApplySlot (
+    HANDLE_SBR_DRC_CHANNEL  hDrcData,
+    FIXP_DBL *qmfRealSlot,
+    FIXP_DBL *qmfImagSlot,
+    int  col,
+    int  numQmfSubSamples,
+    int  maxShift );
+
+void sbrDecoder_drcApply (
+    HANDLE_SBR_DRC_CHANNEL  hDrcData,
+    FIXP_DBL **QmfBufferReal,
+    FIXP_DBL **QmfBufferImag,
+    int  numQmfSubSamples,
+    int *scaleFactor );
+
+
+#endif  /* _SBRDEC_DRC_H_ */
diff --git a/libSBRdec/src/sbrdec_freq_sca.cpp b/libSBRdec/src/sbrdec_freq_sca.cpp
new file mode 100644
index 0000000..b877545
--- /dev/null
+++ b/libSBRdec/src/sbrdec_freq_sca.cpp
@@ -0,0 +1,805 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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 calculation  
+*/
+
+#include "sbrdec_freq_sca.h"
+
+#include "transcendent.h"
+#include "sbr_rom.h"
+#include "env_extr.h"
+
+#include "genericStds.h"      /* need log() for debug-code only */
+
+#define MAX_OCTAVE         29
+#define MAX_SECOND_REGION  50
+
+
+static int  numberOfBands(FIXP_SGL bpo_div16, int start, int stop, int warpFlag);
+static void CalcBands(UCHAR * diff, UCHAR start, UCHAR stop, UCHAR num_bands);
+static SBR_ERROR modifyBands(UCHAR max_band, UCHAR * diff, UCHAR length);
+static void cumSum(UCHAR start_value, UCHAR* diff, UCHAR length, UCHAR *start_adress);
+
+
+
+/*!
+  \brief     Retrieve QMF-band where the SBR range starts
+
+  Convert startFreq which was read from the bitstream into a
+  QMF-channel number.
+
+  \return  Number of start band
+*/
+static UCHAR
+getStartBand(UINT   fs,               /*!< Output sampling frequency */
+             UCHAR  startFreq,        /*!< Index to table of possible start bands */
+             UINT   headerDataFlags)  /*!< Info to SBR mode */
+{
+  INT  band;
+  UINT fsMapped;
+
+    fsMapped = fs;
+
+  switch (fsMapped) {
+    case 48000:
+      band = FDK_sbrDecoder_sbr_start_freq_48[startFreq];
+      break;
+    case 44100:
+      band = FDK_sbrDecoder_sbr_start_freq_44[startFreq];
+      break;
+    case 32000:
+      band = FDK_sbrDecoder_sbr_start_freq_32[startFreq];
+      break;
+    case 24000:
+      band = FDK_sbrDecoder_sbr_start_freq_24[startFreq];
+      break;
+    case 22050:
+      band = FDK_sbrDecoder_sbr_start_freq_22[startFreq];
+      break;
+    case 16000:
+      band = FDK_sbrDecoder_sbr_start_freq_16[startFreq];
+      break;
+    default:
+      band = 255;
+  }
+
+  return band;
+}
+
+
+/*!
+  \brief     Retrieve QMF-band where the SBR range starts
+
+  Convert startFreq which was read from the bitstream into a
+  QMF-channel number.
+
+  \return  Number of start band
+*/
+static UCHAR
+getStopBand(UINT   fs,               /*!< Output sampling frequency */
+            UCHAR  stopFreq,         /*!< Index to table of possible start bands */
+            UINT   headerDataFlags,  /*!< Info to SBR mode */
+            UCHAR  k0)               /*!< Start freq index */
+{
+  UCHAR k2;
+
+  if (stopFreq < 14) {
+    INT    stopMin;
+    UCHAR  diff_tot[MAX_OCTAVE + MAX_SECOND_REGION];
+    UCHAR *diff0 = diff_tot;
+    UCHAR *diff1 = diff_tot+MAX_OCTAVE;
+
+    if (fs < 32000) {
+      stopMin = (((2*6000*2*(64)) / fs) + 1) >> 1;
+    }
+    else {
+      if (fs < 64000) {
+        stopMin = (((2*8000*2*(64)) / fs) + 1) >> 1;
+      }
+      else {
+        stopMin = (((2*10000*2*(64)) / fs) + 1) >> 1;
+      }
+    }
+
+    /*
+      Choose a stop band between k1 and 64 depending on stopFreq (0..13),
+      based on a logarithmic scale.
+      The vectors diff0 and diff1 are used temporarily here.
+    */
+    CalcBands( diff0, stopMin, 64, 13);
+    shellsort( diff0, 13);
+    cumSum(stopMin, diff0, 13, diff1);
+    k2 = diff1[stopFreq];
+  }
+  else if (stopFreq==14)
+    k2 = 2*k0;
+  else
+    k2 = 3*k0;
+
+  /* Limit to Nyquist */
+  if (k2 > (64))
+    k2 = (64);
+
+
+  /* Range checks */
+  /* 1 <= difference <= 48; 1 <= fs <= 96000 */
+  if ( ((k2 - k0) > MAX_FREQ_COEFFS) || (k2 <= k0) ) {
+    return 255;
+  }
+
+  if (headerDataFlags & (SBRDEC_SYNTAX_USAC|SBRDEC_SYNTAX_RSVD50)) {
+    /* 1 <= difference <= 35; 42000 <= fs <= 96000 */
+    if ( (fs >= 42000) && ( (k2 - k0) > MAX_FREQ_COEFFS_FS44100 ) ) {
+      return 255;
+    }
+    /* 1 <= difference <= 32; 46009 <= fs <= 96000 */
+    if ( (fs >= 46009) && ( (k2 - k0) > MAX_FREQ_COEFFS_FS48000 ) ) {
+      return 255;
+    }
+  }
+  else {
+    /* 1 <= difference <= 35; fs == 44100 */
+    if ( (fs == 44100) && ( (k2 - k0) > MAX_FREQ_COEFFS_FS44100 ) ) {
+      return 255;
+    }
+    /* 1 <= difference <= 32; 48000 <= fs <= 96000 */
+    if ( (fs >= 48000) && ( (k2 - k0) > MAX_FREQ_COEFFS_FS48000 ) ) {
+      return 255;
+    }
+  }
+
+  return k2;
+}
+
+
+/*!
+  \brief     Generates master frequency tables
+
+  Frequency tables are calculated according to the selected domain
+  (linear/logarithmic) and granularity.
+  IEC 14496-3 4.6.18.3.2.1
+
+  \return  errorCode, 0 if successful
+*/
+SBR_ERROR
+sbrdecUpdateFreqScale(UCHAR * v_k_master,    /*!< Master table to be created */
+                      UCHAR *numMaster,      /*!< Number of entries in master table */
+                      UINT   fs,             /*!< SBR working sampling rate */
+                      HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Control data from bitstream */
+                      UINT flags)
+{
+  FIXP_SGL bpo_div16;        /* bands_per_octave divided by 16 */
+  INT      dk=0;
+
+  /* Internal variables */
+  UCHAR  k0, k2, i;
+  UCHAR  num_bands0 = 0;
+  UCHAR  num_bands1 = 0;
+  UCHAR  diff_tot[MAX_OCTAVE + MAX_SECOND_REGION];
+  UCHAR *diff0 = diff_tot;
+  UCHAR *diff1 = diff_tot+MAX_OCTAVE;
+  INT    k2_achived;
+  INT    k2_diff;
+  INT    incr=0;
+
+  /*
+    Determine start band
+  */
+  k0 = getStartBand(fs, hHeaderData->bs_data.startFreq, flags);
+  if (k0 == 255) {
+    return SBRDEC_UNSUPPORTED_CONFIG;
+  }
+
+  /*
+    Determine stop band
+  */
+  k2 = getStopBand(fs, hHeaderData->bs_data.stopFreq, flags, k0);
+  if (k2 == 255) {
+    return SBRDEC_UNSUPPORTED_CONFIG;
+  }
+
+  if(hHeaderData->bs_data.freqScale>0) { /* Bark */
+    INT k1;
+
+    if(hHeaderData->bs_data.freqScale==1) {
+      bpo_div16 = FL2FXCONST_SGL(12.0f/16.0f);
+    }
+    else if(hHeaderData->bs_data.freqScale==2) {
+      bpo_div16 = FL2FXCONST_SGL(10.0f/16.0f);
+    }
+    else {
+      bpo_div16 =  FL2FXCONST_SGL(8.0f/16.0f);
+    }
+
+
+    if( 1000 * k2 > 2245 * k0 ) { /* Two or more regions */
+      k1 = 2*k0;
+
+      num_bands0 = numberOfBands(bpo_div16, k0, k1, 0);
+      num_bands1 = numberOfBands(bpo_div16, k1, k2, hHeaderData->bs_data.alterScale );
+      if ( num_bands0 < 1) {
+        return SBRDEC_UNSUPPORTED_CONFIG;
+      }
+      if ( num_bands1 < 1 ) {
+        return SBRDEC_UNSUPPORTED_CONFIG;
+      }
+
+      CalcBands(diff0, k0, k1, num_bands0);
+      shellsort( diff0, num_bands0);
+      if (diff0[0] == 0) {
+#ifdef DEBUG_TOOLS
+#endif
+        return SBRDEC_UNSUPPORTED_CONFIG;
+      }
+
+      cumSum(k0, diff0, num_bands0, v_k_master);
+
+      CalcBands(diff1, k1, k2, num_bands1);
+      shellsort( diff1, num_bands1);
+      if(diff0[num_bands0-1] > diff1[0]) {
+        SBR_ERROR err;
+
+        err = modifyBands(diff0[num_bands0-1],diff1, num_bands1);
+        if (err)
+          return SBRDEC_UNSUPPORTED_CONFIG;
+      }
+
+      /* Add 2nd region */
+      cumSum(k1, diff1, num_bands1, &v_k_master[num_bands0]);
+      *numMaster = num_bands0 + num_bands1;     /* Output nr of bands */
+
+    }
+    else { /* Only one region */
+      k1=k2;
+
+      num_bands0 = numberOfBands(bpo_div16, k0, k1, 0);
+      if ( num_bands0 < 1) {
+        return SBRDEC_UNSUPPORTED_CONFIG;
+      }
+      CalcBands(diff0, k0, k1, num_bands0);
+      shellsort(diff0, num_bands0);
+      if (diff0[0] == 0) {
+#ifdef DEBUG_TOOLS
+#endif
+        return SBRDEC_UNSUPPORTED_CONFIG;
+      }
+
+      cumSum(k0, diff0, num_bands0, v_k_master);
+      *numMaster = num_bands0;        /* Output nr of bands */
+
+    }
+  }
+  else { /* Linear mode */
+     if (hHeaderData->bs_data.alterScale==0) {
+        dk = 1;
+        /* FLOOR to get to few number of bands (next lower even number) */
+        num_bands0 = (k2 - k0) & 254;
+      } else {
+        dk = 2;
+        num_bands0 = ( ((k2 - k0) >> 1) + 1 ) & 254; /* ROUND to the closest fit */
+      }
+
+      if (num_bands0 < 1) {
+        return SBRDEC_UNSUPPORTED_CONFIG;
+        /* We must return already here because 'i' can become negative below. */
+      }
+
+      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 achieved */
+      /* and we got too wide SBR area */
+      if (k2_diff < 0) {
+          incr = 1;
+          i = 0;
+      }
+
+      /* If linear scale wasn't achieved */
+      /* and we got too small SBR area */
+      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 */
+    *numMaster = num_bands0;  /* Output nr of bands */
+  }
+
+  if (*numMaster < 1) {
+    return SBRDEC_UNSUPPORTED_CONFIG;
+  }
+
+
+  /*
+    Print out the calculated table
+  */
+
+  return SBRDEC_OK;
+}
+
+
+/*!
+  \brief     Calculate frequency ratio of one SBR band
+
+  All SBR bands should span a constant frequency range in the logarithmic
+  domain. This function calculates the ratio of any SBR band's upper and lower
+  frequency.
+
+ \return    num_band-th root of k_start/k_stop
+*/
+static FIXP_SGL calcFactorPerBand(int k_start, int k_stop, int num_bands)
+{
+/* Scaled bandfactor and step 1 bit right to avoid overflow
+ * use double data type */
+  FIXP_DBL bandfactor = FL2FXCONST_DBL(0.25f); /* Start value */
+  FIXP_DBL step = FL2FXCONST_DBL(0.125f);      /* Initial increment for factor */
+
+  int    direction = 1;
+
+/* Because saturation can't be done in INT IIS,
+ * changed start and stop data type from FIXP_SGL to FIXP_DBL */
+  FIXP_DBL start = k_start << (DFRACT_BITS-8);
+  FIXP_DBL stop = k_stop << (DFRACT_BITS-8);
+
+  FIXP_DBL temp;
+
+  int   j, i=0;
+
+  while ( step > FL2FXCONST_DBL(0.0f)) {
+    i++;
+    temp = stop;
+
+    /* Calculate temp^num_bands: */
+    for (j=0; j<num_bands; j++)
+      //temp = fMult(temp,bandfactor);
+      temp = fMultDiv2(temp,bandfactor)<<2;
+
+    if (temp<start) { /* Factor too strong, make it weaker */
+      if (direction == 0)
+        /* Halfen step. Right shift is not done as fract because otherwise the
+           lowest bit cannot be cleared due to rounding */
+        step = (FIXP_DBL)((LONG)step >> 1);
+      direction = 1;
+      bandfactor = bandfactor + step;
+    }
+    else {  /* Factor is too weak: make it stronger */
+      if (direction == 1)
+        step = (FIXP_DBL)((LONG)step >> 1);
+      direction = 0;
+      bandfactor = bandfactor - step;
+    }
+
+    if (i>100) {
+      step = FL2FXCONST_DBL(0.0f);
+    }
+  }
+  return FX_DBL2FX_SGL(bandfactor<<1);
+}
+
+
+/*!
+  \brief     Calculate number of SBR bands between start and stop band
+
+  Given the number of bands per octave, this function calculates how many
+  bands fit in the given frequency range.
+  When the warpFlag is set, the 'band density' is decreased by a factor
+  of 1/1.3
+
+  \return    number of bands
+*/
+static int
+numberOfBands(FIXP_SGL bpo_div16, /*!< Input: number of bands per octave divided by 16 */
+              int    start,     /*!< First QMF band of SBR frequency range */
+              int    stop,      /*!< Last QMF band of SBR frequency range + 1 */
+              int    warpFlag)  /*!< Stretching flag */
+{
+  FIXP_SGL num_bands_div128;
+  int    num_bands;
+
+  num_bands_div128 = FX_DBL2FX_SGL(fMult(FDK_getNumOctavesDiv8(start,stop),bpo_div16));
+
+  if (warpFlag) {
+    /* Apply the warp factor of 1.3 to get wider bands.  We use a value
+       of 32768/25200 instead of the exact value to avoid critical cases
+       of rounding.
+    */
+    num_bands_div128 = FX_DBL2FX_SGL(fMult(num_bands_div128, FL2FXCONST_SGL(25200.0/32768.0)));
+  }
+
+  /* add scaled 1 for rounding to even numbers: */
+  num_bands_div128 = num_bands_div128 + FL2FXCONST_SGL( 1.0f/128.0f );
+  /* scale back to right aligned integer and double the value: */
+  num_bands = 2 * ((LONG)num_bands_div128 >> (FRACT_BITS - 7));
+
+  return(num_bands);
+}
+
+
+/*!
+  \brief     Calculate width of SBR bands
+
+  Given the desired number of bands within the SBR frequency range,
+  this function calculates the width of each SBR band in QMF channels.
+  The bands get wider from start to stop (bark scale).
+*/
+static void
+CalcBands(UCHAR * diff,    /*!< Vector of widths to be calculated */
+          UCHAR start,     /*!< Lower end of subband range */
+          UCHAR stop,      /*!< Upper end of subband range */
+          UCHAR num_bands) /*!< Desired number of bands */
+{
+  int i;
+  int previous;
+  int current;
+  FIXP_SGL exact, temp;
+  FIXP_SGL bandfactor = calcFactorPerBand(start, stop, num_bands);
+
+  previous = stop; /* Start with highest QMF channel */
+  exact = (FIXP_SGL)(stop << (FRACT_BITS-8)); /* Shift left to gain some accuracy */
+
+  for(i=num_bands-1; i>=0; i--) {
+    /* Calculate border of next lower sbr band */
+    exact = FX_DBL2FX_SGL(fMult(exact,bandfactor));
+
+    /* Add scaled 0.5 for rounding:
+       We use a value 128/256 instead of 0.5 to avoid some critical cases of rounding. */
+    temp = exact +  FL2FXCONST_SGL(128.0/32768.0);
+
+    /* scale back to right alinged integer: */
+    current = (LONG)temp >> (FRACT_BITS-8);
+
+    /* Save width of band i */
+    diff[i] = previous - current;
+    previous = current;
+  }
+}
+
+
+/*!
+  \brief     Calculate cumulated sum vector from delta vector
+*/
+static void
+cumSum(UCHAR start_value, UCHAR* diff, UCHAR 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];
+}
+
+
+/*!
+  \brief     Adapt width of frequency bands in the second region
+
+  If SBR spans more than 2 octaves, the upper part of a bark-frequency-scale
+  is calculated separately. This function tries to avoid that the second region
+  starts with a band smaller than the highest band of the first region.
+*/
+static SBR_ERROR
+modifyBands(UCHAR max_band_previous, UCHAR * diff, UCHAR 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])>>1 )
+    change = (diff[length-1]-diff[0])>>1;
+
+  diff[0] += change;
+  diff[length-1] -= change;
+  shellsort(diff, length);
+
+  return SBRDEC_OK;
+}
+
+
+/*!
+  \brief   Update high resolution frequency band table
+*/
+static void
+sbrdecUpdateHiRes(UCHAR * h_hires,
+                  UCHAR * num_hires,
+                  UCHAR * v_k_master,
+                  UCHAR num_bands,
+                  UCHAR xover_band)
+{
+  UCHAR i;
+
+  *num_hires = num_bands-xover_band;
+
+  for(i=xover_band; i<=num_bands; i++) {
+    h_hires[i-xover_band] = v_k_master[i];
+  }
+}
+
+
+/*!
+  \brief  Build low resolution table out of high resolution table
+*/
+static void
+sbrdecUpdateLoRes(UCHAR * h_lores,
+                  UCHAR * num_lores,
+                  UCHAR * h_hires,
+                  UCHAR num_hires)
+{
+  UCHAR i;
+
+  if( (num_hires & 1) == 0) {
+    /* If even number of hires bands */
+    *num_lores = num_hires >> 1;
+    /* 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) >> 1;
+    /* 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];
+    }
+  }
+}
+
+
+/*!
+  \brief   Derive a low-resolution frequency-table from the master frequency table
+*/
+void
+sbrdecDownSampleLoRes(UCHAR *v_result,
+                      UCHAR num_result,
+                      UCHAR *freqBandTableRef,
+                      UCHAR num_Ref)
+{
+  int step;
+  int i,j;
+  int org_length,result_length;
+  int v_index[MAX_FREQ_COEFFS>>1];
+
+  /* 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;
+    org_length = org_length - step;
+    result_length--;
+    v_index[i] = v_index[i-1] + step;
+  }
+
+  for(j=0;j<=i;j++) {
+    /* Use downsample vector to index LoResolution vector */
+    v_result[j]=freqBandTableRef[v_index[j]];
+  }
+
+}
+
+
+/*!
+  \brief   Sorting routine
+*/
+void shellsort(UCHAR *in, UCHAR n)
+{
+
+  int i, j, v, w;
+  int inc = 1;
+
+  do
+    inc = 3 * inc + 1;
+  while (inc <= n);
+
+  do {
+    inc = inc / 3;
+    for (i = inc; i < n; i++) {
+      v = in[i];
+      j = i;
+      while ((w=in[j-inc]) > v) {
+        in[j] = w;
+        j -= inc;
+        if (j < inc)
+          break;
+      }
+      in[j] = v;
+    }
+  } while (inc > 1);
+
+}
+
+
+
+/*!
+  \brief   Reset frequency band tables
+  \return  errorCode, 0 if successful
+*/
+SBR_ERROR
+resetFreqBandTables(HANDLE_SBR_HEADER_DATA hHeaderData, const UINT flags)
+{
+  SBR_ERROR err = SBRDEC_OK;
+  int k2,kx, lsb, usb;
+  int     intTemp;
+  UCHAR    nBandsLo, nBandsHi;
+  HANDLE_FREQ_BAND_DATA hFreq = &hHeaderData->freqBandData;
+
+  /* Calculate master frequency function */
+  err = sbrdecUpdateFreqScale(hFreq->v_k_master,
+                              &hFreq->numMaster,
+                              hHeaderData->sbrProcSmplRate,
+                              hHeaderData,
+                              flags);
+
+  if ( err || (hHeaderData->bs_info.xover_band > hFreq->numMaster) ) {
+    return SBRDEC_UNSUPPORTED_CONFIG;
+  }
+
+  /* Derive Hiresolution from master frequency function */
+  sbrdecUpdateHiRes(hFreq->freqBandTable[1], &nBandsHi, hFreq->v_k_master, hFreq->numMaster, hHeaderData->bs_info.xover_band );
+  /* Derive Loresolution from Hiresolution */
+  sbrdecUpdateLoRes(hFreq->freqBandTable[0], &nBandsLo, hFreq->freqBandTable[1], nBandsHi);
+
+
+  hFreq->nSfb[0] = nBandsLo;
+  hFreq->nSfb[1] = nBandsHi;
+
+  /* Check index to freqBandTable[0] */
+  if ( !(nBandsLo > 0) || (nBandsLo > (MAX_FREQ_COEFFS>>1)) ) {
+    return SBRDEC_UNSUPPORTED_CONFIG;
+  }
+
+  lsb = hFreq->freqBandTable[0][0];
+  usb = hFreq->freqBandTable[0][nBandsLo];
+
+  /* Additional check for lsb */
+  if ( (lsb > (32)) || (lsb >= usb) ) {
+    return SBRDEC_UNSUPPORTED_CONFIG;
+  }
+
+
+  /* Calculate number of noise bands */
+
+  k2 = hFreq->freqBandTable[1][nBandsHi];
+  kx = hFreq->freqBandTable[1][0];
+
+  if (hHeaderData->bs_data.noise_bands == 0)
+  {
+    hFreq->nNfb = 1;
+  }
+  else /* Calculate no of noise bands 1,2 or 3 bands/octave */
+  {
+    /* Fetch number of octaves divided by 32 */
+    intTemp = (LONG)FDK_getNumOctavesDiv8(kx,k2) >> 2;
+
+    /* Integer-Multiplication with number of bands: */
+    intTemp = intTemp * hHeaderData->bs_data.noise_bands;
+
+    /* Add scaled 0.5 for rounding: */
+    intTemp = intTemp + (LONG)FL2FXCONST_SGL(0.5f/32.0f);
+
+    /* Convert to right-aligned integer: */
+    intTemp = intTemp >> (FRACT_BITS - 1 /*sign*/ - 5 /* rescale */);
+
+    /* Compare with float calculation */
+    FDK_ASSERT( intTemp ==  (int)((hHeaderData->bs_data.noise_bands * FDKlog( (float)k2/kx) / (float)(FDKlog(2.0)))+0.5) );
+
+    if( intTemp==0)
+      intTemp=1;
+
+    hFreq->nNfb = intTemp;
+  }
+
+  hFreq->nInvfBands = hFreq->nNfb;
+
+  if( hFreq->nNfb > MAX_NOISE_COEFFS ) {
+    return SBRDEC_UNSUPPORTED_CONFIG;
+  }
+
+  /* Get noise bands */
+  sbrdecDownSampleLoRes(hFreq->freqBandTableNoise,
+                        hFreq->nNfb,
+                        hFreq->freqBandTable[0],
+                        nBandsLo);
+
+
+
+
+  hFreq->lowSubband  = lsb;
+  hFreq->highSubband = usb;
+
+  return SBRDEC_OK;
+}
diff --git a/libSBRdec/src/sbrdec_freq_sca.h b/libSBRdec/src/sbrdec_freq_sca.h
new file mode 100644
index 0000000..eebdd52
--- /dev/null
+++ b/libSBRdec/src/sbrdec_freq_sca.h
@@ -0,0 +1,107 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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_SCA_H
+#define __FREQ_SCA_H
+
+#include "sbrdecoder.h"
+#include "env_extr.h"
+
+int
+sbrdecUpdateFreqScale(UCHAR * v_k_master,
+                      UCHAR *numMaster,
+                      HANDLE_SBR_HEADER_DATA headerData);
+
+void sbrdecDownSampleLoRes(UCHAR *v_result, UCHAR num_result,
+                           UCHAR *freqBandTableRef, UCHAR num_Ref);
+
+void shellsort(UCHAR *in, UCHAR n);
+
+SBR_ERROR
+resetFreqBandTables(HANDLE_SBR_HEADER_DATA hHeaderData, const UINT flags);
+
+#endif
diff --git a/libSBRdec/src/sbrdecoder.cpp b/libSBRdec/src/sbrdecoder.cpp
new file mode 100644
index 0000000..a40e5ba
--- /dev/null
+++ b/libSBRdec/src/sbrdecoder.cpp
@@ -0,0 +1,1527 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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 decoder frontend  
+  This module provides a frontend to the SBR decoder. The function openSBR() is called for
+  initialization. The function sbrDecoder_Apply() is called for each frame. sbr_Apply() will call the
+  required functions to decode the raw SBR data (provided by env_extr.cpp), to decode the envelope data and noise floor levels [decodeSbrData()],
+  and to finally apply SBR to the current frame [sbr_dec()].
+
+  \sa sbrDecoder_Apply(), \ref documentationOverview
+*/
+
+/*!
+  \page documentationOverview Overview of important information resources and source code documentation
+
+  The primary source code documentation is based on generated and cross-referenced HTML files using
+  <a HREF="http://www.doxygen.org">doxygen</a>. As part of this documentation
+  you can find more extensive descriptions about key concepts and algorithms at the following locations:
+
+  <h2>Programming</h2>
+
+  \li Buffer management: sbrDecoder_Apply() and sbr_dec()
+  \li Internal scale factors to maximize SNR on fixed point processors: #QMF_SCALE_FACTOR
+  \li Special mantissa-exponent format: Created in requantizeEnvelopeData() and used in calculateSbrEnvelope()
+
+  <h2>Algorithmic details</h2>
+  \li About the SBR data format: \ref SBR_HEADER_ELEMENT and \ref SBR_STANDARD_ELEMENT
+  \li Details about the bitstream decoder: env_extr.cpp
+  \li Details about the QMF filterbank and the provided polyphase implementation: qmf_dec.cpp
+  \li Details about the transposer: lpp_tran.cpp
+  \li Details about the envelope adjuster: env_calc.cpp
+
+*/
+
+#include "sbrdecoder.h"
+
+#include "FDK_bitstream.h"
+
+#include "sbrdec_freq_sca.h"
+#include "env_extr.h"
+#include "sbr_dec.h"
+#include "env_dec.h"
+#include "sbr_crc.h"
+#include "sbr_ram.h"
+#include "sbr_rom.h"
+#include "lpp_tran.h"
+#include "transcendent.h"
+
+
+#include "sbrdec_drc.h"
+
+#include "psbitdec.h"
+
+
+/* Decoder library info */
+#define SBRDECODER_LIB_VL0 2
+#define SBRDECODER_LIB_VL1 1
+#define SBRDECODER_LIB_VL2 2
+#define SBRDECODER_LIB_TITLE "SBR Decoder"
+#define SBRDECODER_LIB_BUILD_DATE __DATE__
+#define SBRDECODER_LIB_BUILD_TIME __TIME__
+
+
+
+
+static UCHAR getHeaderSlot( UCHAR currentSlot, UCHAR hdrSlotUsage[(1)+1] )
+{
+  UINT  occupied = 0;
+  int   s;
+  UCHAR slot = hdrSlotUsage[currentSlot];
+
+  FDK_ASSERT((1)+1 < 32);
+
+  for (s = 0; s < (1)+1; s++) {
+    if ( (hdrSlotUsage[s] == slot)
+      && (s != slot) ) {
+      occupied = 1;
+      break;
+    }
+  }
+
+  if (occupied) {
+    occupied = 0;
+
+    for (s = 0; s < (1)+1; s++) {
+      occupied |= 1 << hdrSlotUsage[s];
+    }
+    for (s = 0; s < (1)+1; s++) {
+      if ( !(occupied & 0x1) ) {
+        slot = s;
+        break;
+      }
+      occupied >>= 1;
+    }
+  }
+
+  return slot;
+}
+
+static void copySbrHeader( HANDLE_SBR_HEADER_DATA hDst, const HANDLE_SBR_HEADER_DATA hSrc )
+{
+  /* copy the whole header memory (including pointers) */
+  FDKmemcpy( hDst, hSrc, sizeof(SBR_HEADER_DATA) );
+
+  /* update pointers */
+  hDst->freqBandData.freqBandTable[0]  = hDst->freqBandData.freqBandTableLo;
+  hDst->freqBandData.freqBandTable[1] = hDst->freqBandData.freqBandTableHi;
+}
+
+
+/*!
+  \brief Reset SBR decoder.
+
+  Reset should only be called if SBR has been sucessfully detected by
+  an appropriate checkForPayload() function.
+
+  \return Error code.
+*/
+static
+SBR_ERROR sbrDecoder_ResetElement (
+        HANDLE_SBRDECODER    self,
+        int                  sampleRateIn,
+        int                  sampleRateOut,
+        int                  samplesPerFrame,
+        const MP4_ELEMENT_ID elementID,
+        const int            elementIndex,
+        const int            overlap
+        )
+{
+  SBR_ERROR sbrError = SBRDEC_OK;
+  HANDLE_SBR_HEADER_DATA hSbrHeader;
+  UINT qmfFlags = 0;
+
+  int i, synDownsampleFac;
+
+  /* Check in/out samplerates */
+  if ( sampleRateIn < 6400
+    || sampleRateIn > 24000
+     )
+  {
+    sbrError = SBRDEC_UNSUPPORTED_CONFIG;
+    goto bail;
+  }
+
+  if ( sampleRateOut > 48000 )
+  {
+    sbrError = SBRDEC_UNSUPPORTED_CONFIG;
+    goto bail;
+  }
+
+  /* Set QMF mode flags */
+  if (self->flags & SBRDEC_LOW_POWER)
+    qmfFlags |= QMF_FLAG_LP;
+
+  if (self->coreCodec == AOT_ER_AAC_ELD) {
+    if (self->flags & SBRDEC_LD_MPS_QMF) {
+      qmfFlags |=  QMF_FLAG_MPSLDFB;
+    } else {
+      qmfFlags |=  QMF_FLAG_CLDFB;
+    }
+  }
+
+  /* Set downsampling factor for synthesis filter bank */
+  if (sampleRateOut == 0)
+  {
+    /* no single rate mode */
+      sampleRateOut = sampleRateIn<<1; /* In case of implicit signalling, assume dual rate SBR */
+  }
+
+  if ( sampleRateIn == sampleRateOut ) {
+    synDownsampleFac = 2;
+  } else {
+    synDownsampleFac = 1;
+  }
+
+  self->synDownsampleFac = synDownsampleFac;
+  self->sampleRateOut = sampleRateOut;
+
+  {
+    int i;
+
+    for (i = 0; i < (1)+1; i++)
+    {
+      hSbrHeader = &(self->sbrHeader[elementIndex][i]);
+
+      /* init a default header such that we can at least do upsampling later */
+      sbrError = initHeaderData(
+              hSbrHeader,
+              sampleRateIn,
+              sampleRateOut,
+              samplesPerFrame,
+              self->flags
+              );
+    }
+  }
+
+  if (sbrError != SBRDEC_OK) {
+    goto bail;
+  }
+
+  /* Init SBR channels going to be assigned to a SBR element */
+  {
+    int ch;
+
+    for (ch=0; ch<self->pSbrElement[elementIndex]->nChannels; ch++)
+    {
+      /* and create sbrDec */
+      sbrError = createSbrDec (self->pSbrElement[elementIndex]->pSbrChannel[ch],
+                          hSbrHeader,
+                         &self->pSbrElement[elementIndex]->transposerSettings,
+                          synDownsampleFac,
+                          qmfFlags,
+                          self->flags,
+                          overlap,
+                          ch );
+
+      if (sbrError != SBRDEC_OK) {
+        goto bail;
+      }
+    }
+  }
+
+  //FDKmemclear(sbr_OverlapBuffer, sizeof(sbr_OverlapBuffer));
+
+  if (self->numSbrElements == 1) {
+    switch ( self->coreCodec ) {
+    case AOT_AAC_LC:
+    case AOT_SBR:
+    case AOT_PS:
+    case AOT_ER_AAC_SCAL:
+    case AOT_DRM_AAC:
+    case AOT_DRM_SURROUND:
+      if (CreatePsDec ( &self->hParametricStereoDec, samplesPerFrame )) {
+        sbrError = SBRDEC_CREATE_ERROR;
+        goto bail;
+      }
+      break;
+    default:
+      break;
+    }
+  }
+
+  /* Init frame delay slot handling */
+  self->pSbrElement[elementIndex]->useFrameSlot = 0;
+  for (i = 0; i < ((1)+1); i++) {
+    self->pSbrElement[elementIndex]->useHeaderSlot[i] = i;
+  }
+
+bail:
+
+  return sbrError;
+}
+
+
+SBR_ERROR sbrDecoder_Open ( HANDLE_SBRDECODER  * pSelf )
+{
+  HANDLE_SBRDECODER    self = NULL;
+  SBR_ERROR sbrError = SBRDEC_OK;
+
+  /* Get memory for this instance */
+  self = GetRam_SbrDecoder();
+  if (self == NULL) {
+    sbrError = SBRDEC_MEM_ALLOC_FAILED;
+    goto bail;
+  }
+
+  self->workBuffer1 = GetRam_SbrDecWorkBuffer1();
+  self->workBuffer2 = GetRam_SbrDecWorkBuffer2();
+
+  if (  self->workBuffer1 == NULL
+     || self->workBuffer2 == NULL )
+  {
+    sbrError = SBRDEC_MEM_ALLOC_FAILED;
+    goto bail;
+  }
+
+  /*
+  Already zero because of calloc
+  self->numSbrElements = 0;
+  self->numSbrChannels = 0;
+  self->codecFrameSize = 0;
+  */
+
+  self->numDelayFrames = (1);  /* set to the max value by default */
+
+  *pSelf = self;
+
+bail:
+  return sbrError;
+}
+
+/**
+ * \brief determine if the given core codec AOT can be processed or not.
+ * \param coreCodec core codec audio object type.
+ * \return 1 if SBR can be processed, 0 if SBR cannot be processed/applied.
+ */
+static
+int sbrDecoder_isCoreCodecValid(AUDIO_OBJECT_TYPE coreCodec)
+{
+  switch (coreCodec) {
+    case AOT_AAC_LC:
+    case AOT_SBR:
+    case AOT_PS:
+    case AOT_ER_AAC_SCAL:
+    case AOT_ER_AAC_ELD:
+      return 1;
+    default:
+      return 0;
+  }
+}
+
+static
+void sbrDecoder_DestroyElement (
+        HANDLE_SBRDECODER       self,
+        const int               elementIndex
+        )
+{
+  if (self->pSbrElement[elementIndex] != NULL) {
+    int ch;
+
+    for (ch=0; ch<SBRDEC_MAX_CH_PER_ELEMENT; ch++) {
+      if (self->pSbrElement[elementIndex]->pSbrChannel[ch] != NULL) {
+        deleteSbrDec( self->pSbrElement[elementIndex]->pSbrChannel[ch] );
+        FreeRam_SbrDecChannel( &self->pSbrElement[elementIndex]->pSbrChannel[ch] );
+        self->numSbrChannels -= 1;
+      }
+    }
+    FreeRam_SbrDecElement( &self->pSbrElement[elementIndex] );
+    self->numSbrElements -= 1;
+  }
+}
+
+
+SBR_ERROR sbrDecoder_InitElement (
+        HANDLE_SBRDECODER       self,
+        const int               sampleRateIn,
+        const int               sampleRateOut,
+        const int               samplesPerFrame,
+        const AUDIO_OBJECT_TYPE coreCodec,
+        const MP4_ELEMENT_ID    elementID,
+        const int               elementIndex
+        )
+{
+  SBR_ERROR sbrError = SBRDEC_OK;
+  int chCnt=0;
+  int nSbrElementsStart = self->numSbrElements;
+
+  /* Check core codec AOT */
+  if (! sbrDecoder_isCoreCodecValid(coreCodec) || elementIndex >= (4)) {
+    sbrError = SBRDEC_UNSUPPORTED_CONFIG;
+    goto bail;
+  }
+
+  if ( elementID != ID_SCE && elementID != ID_CPE && elementID != ID_LFE )
+  {
+    sbrError = SBRDEC_UNSUPPORTED_CONFIG;
+    goto bail;
+  }
+
+  if (  self->sampleRateIn == sampleRateIn
+     && self->codecFrameSize == samplesPerFrame
+     && self->coreCodec == coreCodec
+     && self->pSbrElement[elementIndex] != NULL
+     && self->pSbrElement[elementIndex]->elementID == elementID
+     )
+  {
+     /* Nothing to do */
+     return SBRDEC_OK;
+  }
+
+  self->sampleRateIn = sampleRateIn;
+  self->codecFrameSize = samplesPerFrame;
+  self->coreCodec = coreCodec;
+
+  self->flags = 0;
+  self->flags |= (coreCodec == AOT_ER_AAC_ELD) ? SBRDEC_ELD_GRID : 0;
+
+  /* Init SBR elements */
+  {
+    int elChannels, ch;
+
+    if (self->pSbrElement[elementIndex] == NULL) {
+      self->pSbrElement[elementIndex] = GetRam_SbrDecElement(elementIndex);
+      if (self->pSbrElement[elementIndex] == NULL) {
+        sbrError = SBRDEC_MEM_ALLOC_FAILED;
+        goto bail;
+      }
+      self->numSbrElements ++;
+    } else {
+      self->numSbrChannels -= self->pSbrElement[elementIndex]->nChannels;
+    }
+
+    /* Save element ID for sanity checks and to have a fallback for concealment. */
+    self->pSbrElement[elementIndex]->elementID = elementID;
+
+    /* Determine amount of channels for this element */
+    switch (elementID) {
+      case ID_NONE:
+      case ID_CPE: elChannels=2;
+        break;
+      case ID_LFE:
+      case ID_SCE: elChannels=1;
+        break;
+      default: elChannels=0;
+        break;
+    }
+
+    /* Handle case of Parametric Stereo */
+    if ( elementIndex == 0 && elementID == ID_SCE ) {
+      switch (coreCodec) {
+        case AOT_AAC_LC:
+        case AOT_SBR:
+        case AOT_PS:
+        case AOT_ER_AAC_SCAL:
+        case AOT_DRM_AAC:
+        case AOT_DRM_SURROUND:
+          elChannels = 2;
+          break;
+        default:
+          break;
+      }
+    }
+
+    self->pSbrElement[elementIndex]->nChannels = elChannels;
+
+    for (ch=0; ch<elChannels; ch++)
+    {
+      if (self->pSbrElement[elementIndex]->pSbrChannel[ch] == NULL) {
+        self->pSbrElement[elementIndex]->pSbrChannel[ch] = GetRam_SbrDecChannel(chCnt);
+        if (self->pSbrElement[elementIndex]->pSbrChannel[ch] == NULL) {
+          sbrError = SBRDEC_MEM_ALLOC_FAILED;
+          goto bail;
+        }
+      }
+      self->numSbrChannels ++;
+
+      sbrDecoder_drcInitChannel( &self->pSbrElement[elementIndex]->pSbrChannel[ch]->SbrDec.sbrDrcChannel );
+
+      /* Add reference pointer to workbuffers. */
+      self->pSbrElement[elementIndex]->pSbrChannel[ch]->SbrDec.WorkBuffer1 = self->workBuffer1;
+      self->pSbrElement[elementIndex]->pSbrChannel[ch]->SbrDec.WorkBuffer2 = self->workBuffer2;
+      chCnt++;
+    }
+    if (elChannels == 1 && self->pSbrElement[elementIndex]->pSbrChannel[ch] != NULL) {
+      deleteSbrDec( self->pSbrElement[elementIndex]->pSbrChannel[ch] );
+      FreeRam_SbrDecChannel( &self->pSbrElement[elementIndex]->pSbrChannel[ch] );
+    }
+  }
+
+  /* clear error flags for all delay slots */
+  FDKmemclear(self->pSbrElement[elementIndex]->frameErrorFlag, ((1)+1)*sizeof(UCHAR));
+
+  /* Initialize this instance */
+  sbrError = sbrDecoder_ResetElement( 
+          self,
+          sampleRateIn,
+          sampleRateOut,
+          samplesPerFrame,
+          elementID,
+          elementIndex,
+          (coreCodec == AOT_ER_AAC_ELD) ? 0 : (6)
+          );
+
+
+
+bail:
+  if (sbrError != SBRDEC_OK) {
+    if (nSbrElementsStart < self->numSbrElements) {
+      /* Free the memory allocated for this element */
+      sbrDecoder_DestroyElement( self, elementIndex );
+    } else if (self->pSbrElement[elementIndex] != NULL) {
+      /* Set error flag to trigger concealment */
+      self->pSbrElement[elementIndex]->frameErrorFlag[self->pSbrElement[elementIndex]->useFrameSlot] = 1;;
+    }
+  }
+
+  return sbrError;
+}
+
+/**
+ * \brief Apply decoded SBR header for one element.
+ * \param self SBR decoder instance handle
+ * \param hSbrHeader SBR header handle to be processed.
+ * \param hSbrChannel pointer array to the SBR element channels corresponding to the SBR header.
+ * \param headerStatus header status value returned from SBR header parser.
+ * \param numElementChannels amount of channels for the SBR element whos header is to be processed.
+ */
+static
+SBR_ERROR sbrDecoder_HeaderUpdate(
+        HANDLE_SBRDECODER self,
+        HANDLE_SBR_HEADER_DATA hSbrHeader,
+        SBR_HEADER_STATUS headerStatus,
+        HANDLE_SBR_CHANNEL hSbrChannel[],
+        const int numElementChannels
+        )
+{
+  SBR_ERROR errorStatus = SBRDEC_OK;
+ 
+  /*
+    change of control data, reset decoder
+  */
+  errorStatus = resetFreqBandTables(hSbrHeader, self->flags);
+
+  if (errorStatus == SBRDEC_OK) {
+    if (hSbrHeader->syncState == UPSAMPLING && headerStatus != HEADER_RESET)
+    {
+      /* As the default header would limit the frequency range,
+         lowSubband and highSubband must be patched. */
+      hSbrHeader->freqBandData.lowSubband = hSbrHeader->numberOfAnalysisBands;
+      hSbrHeader->freqBandData.highSubband = hSbrHeader->numberOfAnalysisBands;
+    }
+
+    /* Trigger a reset before processing this slot */
+    hSbrHeader->status |= SBRDEC_HDR_STAT_RESET;
+  }
+
+  return errorStatus;
+}
+
+INT sbrDecoder_Header (
+        HANDLE_SBRDECODER       self,
+        HANDLE_FDK_BITSTREAM    hBs,
+        const INT sampleRateIn,
+        const INT sampleRateOut,
+        const INT samplesPerFrame,
+        const AUDIO_OBJECT_TYPE coreCodec,
+        const MP4_ELEMENT_ID    elementID,
+        const INT               elementIndex
+        )
+{
+  SBR_HEADER_STATUS headerStatus;
+  HANDLE_SBR_HEADER_DATA hSbrHeader;
+  SBR_ERROR sbrError = SBRDEC_OK;
+  int headerIndex;
+
+  if ( self == NULL || elementIndex > (4) )
+  {
+    return SBRDEC_UNSUPPORTED_CONFIG;
+  }
+
+  if (! sbrDecoder_isCoreCodecValid(coreCodec)) {
+    return SBRDEC_UNSUPPORTED_CONFIG;
+  }
+
+  sbrError = sbrDecoder_InitElement(
+          self,
+          sampleRateIn,
+          sampleRateOut,
+          samplesPerFrame,
+          coreCodec,
+          elementID,
+          elementIndex
+          );
+
+  if (sbrError != SBRDEC_OK) {
+    goto bail;
+  }
+
+  headerIndex = getHeaderSlot(self->pSbrElement[elementIndex]->useFrameSlot,
+                              self->pSbrElement[elementIndex]->useHeaderSlot);
+  hSbrHeader = &(self->sbrHeader[elementIndex][headerIndex]);
+
+  headerStatus = sbrGetHeaderData ( hSbrHeader,
+                                    hBs,
+                                    self->flags,
+                                    0);
+
+
+  {
+    SBR_DECODER_ELEMENT *pSbrElement;
+
+    pSbrElement = self->pSbrElement[elementIndex];
+
+    /* Sanity check */
+    if (pSbrElement != NULL) {
+      if ( (elementID == ID_CPE && pSbrElement->nChannels != 2)
+        || (elementID != ID_CPE && pSbrElement->nChannels != 1) )
+      {
+        return SBRDEC_UNSUPPORTED_CONFIG;
+      }
+      if ( headerStatus == HEADER_RESET ) {
+
+        sbrError = sbrDecoder_HeaderUpdate(
+              self,
+              hSbrHeader,
+              headerStatus,
+              pSbrElement->pSbrChannel,
+              pSbrElement->nChannels
+              );
+
+        if (sbrError == SBRDEC_OK) {
+          hSbrHeader->syncState = SBR_HEADER;
+          hSbrHeader->status   |= SBRDEC_HDR_STAT_UPDATE;
+        }
+        /* else {
+          Since we already have overwritten the old SBR header the only way out is UPSAMPLING!
+          This will be prepared in the next step.
+        } */
+      }
+    }
+  }
+bail:
+  return sbrError;
+}
+
+
+SBR_ERROR sbrDecoder_SetParam (HANDLE_SBRDECODER   self,
+                               const SBRDEC_PARAM  param,
+                               const INT           value )
+{
+  SBR_ERROR errorStatus = SBRDEC_OK;
+
+  /* configure the subsystems */
+  switch (param)
+  {
+  case SBR_SYSTEM_BITSTREAM_DELAY:
+    if (value < 0 || value > (1)) {
+      errorStatus = SBRDEC_SET_PARAM_FAIL;
+      break;
+    }
+    if (self == NULL) {
+      errorStatus = SBRDEC_NOT_INITIALIZED;
+    } else {
+      self->numDelayFrames = (UCHAR)value;
+    }
+    break;
+  case SBR_QMF_MODE:
+    if (self == NULL) {
+      errorStatus = SBRDEC_NOT_INITIALIZED;
+    } else {
+      if (value == 1) {
+        self->flags |= SBRDEC_LOW_POWER;
+      } else {
+        self->flags &= ~SBRDEC_LOW_POWER;
+      }
+    }
+    break;
+  case SBR_LD_QMF_TIME_ALIGN:
+    if (self == NULL) {
+      errorStatus = SBRDEC_NOT_INITIALIZED;
+    } else {
+      if (value == 1) {
+        self->flags |= SBRDEC_LD_MPS_QMF;
+      } else {
+        self->flags &= ~SBRDEC_LD_MPS_QMF;
+      }
+    }
+    break;
+  case SBR_BS_INTERRUPTION:
+    {
+      int elementIndex;
+      /* Loop over SBR elements */
+      for (elementIndex = 0; elementIndex < self->numSbrElements; elementIndex++)
+      {
+        HANDLE_SBR_HEADER_DATA hSbrHeader;
+        int headerIndex = getHeaderSlot(self->pSbrElement[elementIndex]->useFrameSlot,
+                                        self->pSbrElement[elementIndex]->useHeaderSlot);
+
+        hSbrHeader = &(self->sbrHeader[elementIndex][headerIndex]);
+
+        /* Set sync state UPSAMPLING for the corresponding slot.
+           This switches off bitstream parsing until a new header arrives. */
+        hSbrHeader->syncState = UPSAMPLING;
+        hSbrHeader->status   |= SBRDEC_HDR_STAT_UPDATE;
+      }
+    }
+    break;
+  default:
+    errorStatus = SBRDEC_SET_PARAM_FAIL;
+    break;
+  }  /* switch(param) */
+
+  return (errorStatus);
+}
+
+static
+SBRDEC_DRC_CHANNEL * sbrDecoder_drcGetChannel( const HANDLE_SBRDECODER self, const INT channel )
+{
+  SBRDEC_DRC_CHANNEL *pSbrDrcChannelData = NULL;
+  int elementIndex, elChanIdx=0, numCh=0;
+
+  for (elementIndex = 0; (elementIndex < (4)) && (numCh <= channel); elementIndex++)
+  {
+    SBR_DECODER_ELEMENT *pSbrElement = self->pSbrElement[elementIndex];
+    int c, elChannels;
+
+    elChanIdx = 0;
+    if (pSbrElement == NULL) break;
+
+    /* Determine amount of channels for this element */
+    switch (pSbrElement->elementID) {
+      case ID_CPE: elChannels = 2;
+        break;
+      case ID_LFE:
+      case ID_SCE: elChannels = 1;
+        break;
+      case ID_NONE:
+      default: elChannels = 0;
+        break;
+    }
+
+    /* Limit with actual allocated element channels */
+    elChannels = FDKmin(elChannels, pSbrElement->nChannels);
+
+    for (c = 0; (c < elChannels) && (numCh <= channel); c++) {
+      if (pSbrElement->pSbrChannel[elChanIdx] != NULL) {
+        numCh++;
+        elChanIdx++;
+      }
+    }
+  }
+  elementIndex -= 1;
+  elChanIdx -= 1;
+
+  if (elChanIdx < 0 || elementIndex < 0) {
+    return NULL;
+  }
+
+  if ( self->pSbrElement[elementIndex] != NULL ) {
+    if ( self->pSbrElement[elementIndex]->pSbrChannel[elChanIdx] != NULL )
+    {
+      pSbrDrcChannelData = &self->pSbrElement[elementIndex]->pSbrChannel[elChanIdx]->SbrDec.sbrDrcChannel;
+    }
+  }
+
+  return (pSbrDrcChannelData);
+}
+
+SBR_ERROR sbrDecoder_drcFeedChannel ( HANDLE_SBRDECODER  self,
+                                      INT                ch,
+                                      UINT               numBands,
+                                      FIXP_DBL          *pNextFact_mag,
+                                      INT                nextFact_exp,
+                                      SHORT              drcInterpolationScheme,
+                                      UCHAR              winSequence,
+                                      USHORT            *pBandTop )
+{
+  SBRDEC_DRC_CHANNEL *pSbrDrcChannelData = NULL;
+
+  if (self == NULL) {
+    return SBRDEC_NOT_INITIALIZED;
+  }
+  if (ch > (6) || pNextFact_mag == NULL) {
+    return SBRDEC_SET_PARAM_FAIL;
+  }
+
+  /* Find the right SBR channel */
+  pSbrDrcChannelData = sbrDecoder_drcGetChannel( self, ch );
+
+  if ( pSbrDrcChannelData != NULL ) {
+    int i;
+
+    pSbrDrcChannelData->enable   = 1;
+    pSbrDrcChannelData->numBandsNext = numBands;
+
+    pSbrDrcChannelData->winSequenceNext            = winSequence;
+    pSbrDrcChannelData->drcInterpolationSchemeNext = drcInterpolationScheme;
+    pSbrDrcChannelData->nextFact_exp               = nextFact_exp;
+
+    for (i = 0; i < (int)numBands; i++) {
+      pSbrDrcChannelData->bandTopNext[i]  = pBandTop[i];
+      pSbrDrcChannelData->nextFact_mag[i] = pNextFact_mag[i];
+    }
+  }
+
+  return SBRDEC_OK;
+}
+
+
+void sbrDecoder_drcDisable ( HANDLE_SBRDECODER  self,
+                             INT                ch )
+{
+  SBRDEC_DRC_CHANNEL *pSbrDrcChannelData = NULL;
+
+  if ( (self == NULL)
+    || (ch > (6))
+    || (self->numSbrElements == 0)
+    || (self->numSbrChannels == 0) ) {
+    return;
+  }
+
+  /* Find the right SBR channel */
+  pSbrDrcChannelData = sbrDecoder_drcGetChannel( self, ch );
+
+  if ( pSbrDrcChannelData != NULL ) {
+    pSbrDrcChannelData->enable = 0;
+  }
+}
+
+
+
+SBR_ERROR sbrDecoder_Parse(
+        HANDLE_SBRDECODER self,
+        HANDLE_FDK_BITSTREAM  hBs,
+        int *count,
+        int  bsPayLen,
+        int  crcFlag,
+        MP4_ELEMENT_ID prevElement,
+        int elementIndex,
+        int fGlobalIndependencyFlag
+        )
+{
+  SBR_DECODER_ELEMENT   *hSbrElement;
+  HANDLE_SBR_HEADER_DATA hSbrHeader;
+  HANDLE_SBR_CHANNEL    *pSbrChannel;
+
+  SBR_FRAME_DATA *hFrameDataLeft;
+  SBR_FRAME_DATA *hFrameDataRight;
+
+  SBR_ERROR errorStatus = SBRDEC_OK;
+  SBR_SYNC_STATE initialSyncState;
+  SBR_HEADER_STATUS headerStatus = HEADER_NOT_PRESENT;
+
+  INT  startPos;
+  INT  CRCLen = 0;
+
+  int  stereo;
+  int  fDoDecodeSbrData = 1;
+
+  int lastSlot, lastHdrSlot = 0, thisHdrSlot;
+
+  /* Remember start position of  SBR element */
+  startPos = FDKgetValidBits(hBs);
+
+  /* SBR sanity checks */
+  if ( self == NULL || self->pSbrElement[elementIndex] == NULL ) {
+    errorStatus = SBRDEC_NOT_INITIALIZED;
+    goto bail;
+  } 
+
+  hSbrElement = self->pSbrElement[elementIndex];
+
+  lastSlot    = (hSbrElement->useFrameSlot > 0) ? hSbrElement->useFrameSlot-1 : self->numDelayFrames;
+  lastHdrSlot =  hSbrElement->useHeaderSlot[lastSlot];
+  thisHdrSlot =  getHeaderSlot( hSbrElement->useFrameSlot, hSbrElement->useHeaderSlot );  /* Get a free header slot not used by frames not processed yet. */
+
+  /* Assign the free slot to store a new header if there is one. */
+  hSbrHeader = &self->sbrHeader[elementIndex][thisHdrSlot];
+
+  pSbrChannel = hSbrElement->pSbrChannel;
+  stereo = (hSbrElement->elementID == ID_CPE) ? 1 : 0;
+
+  hFrameDataLeft  = &self->pSbrElement[elementIndex]->pSbrChannel[0]->frameData[hSbrElement->useFrameSlot];
+  hFrameDataRight = &self->pSbrElement[elementIndex]->pSbrChannel[1]->frameData[hSbrElement->useFrameSlot];
+
+  initialSyncState = hSbrHeader->syncState;
+
+  /* reset PS flag; will be set after PS was found */
+  self->flags &= ~SBRDEC_PS_DECODED;
+
+  if (hSbrHeader->status & SBRDEC_HDR_STAT_UPDATE) {
+    /* Got a new header from extern (e.g. from an ASC) */
+    headerStatus = HEADER_OK;
+    hSbrHeader->status &= ~SBRDEC_HDR_STAT_UPDATE;
+  }
+  else if (thisHdrSlot != lastHdrSlot) {
+    /* Copy the last header into this slot otherwise the
+       header compare will trigger more HEADER_RESETs than needed. */
+    copySbrHeader( hSbrHeader, &self->sbrHeader[elementIndex][lastHdrSlot] );
+  }
+
+  /*
+     Check if bit stream data is valid and matches the element context
+  */
+  if ( ((prevElement != ID_SCE) && (prevElement != ID_CPE)) || prevElement != hSbrElement->elementID) {
+    /* In case of LFE we also land here, since there is no LFE SBR element (do upsampling only) */
+    fDoDecodeSbrData = 0;
+  }
+
+  if (fDoDecodeSbrData)
+  {
+    if ((INT)FDKgetValidBits(hBs) <= 0) {
+      fDoDecodeSbrData = 0;
+    }
+  }
+
+  /*
+     SBR CRC-check
+  */
+  if (fDoDecodeSbrData)
+  {
+    if (crcFlag == 1) {
+      switch (self->coreCodec) {
+      case AOT_ER_AAC_ELD:
+        FDKpushFor (hBs, 10);
+        /* check sbrcrc later: we don't know the payload length now */
+        break;
+      default:
+        CRCLen = bsPayLen - 10;                     /* change: 0 => i */
+        if (CRCLen < 0) {
+          fDoDecodeSbrData = 0;
+        } else {
+          fDoDecodeSbrData = SbrCrcCheck (hBs, CRCLen);
+        }
+        break;
+      }
+    }
+  } /* if (fDoDecodeSbrData) */
+
+  /*
+     Read in the header data and issue a reset if change occured
+  */
+  if (fDoDecodeSbrData)
+  {
+    int sbrHeaderPresent;
+
+    {
+      sbrHeaderPresent = FDKreadBit(hBs);
+    }
+
+    if ( sbrHeaderPresent ) {
+      headerStatus = sbrGetHeaderData (hSbrHeader,
+                                       hBs,
+                                       self->flags,
+                                       1);
+    }
+
+    if (headerStatus == HEADER_RESET)
+    {
+      errorStatus = sbrDecoder_HeaderUpdate(
+            self,
+            hSbrHeader,
+            headerStatus,
+            pSbrChannel,
+            hSbrElement->nChannels 
+            );
+
+      if (errorStatus == SBRDEC_OK) {
+        hSbrHeader->syncState = SBR_HEADER;
+      } else {
+        hSbrHeader->syncState = SBR_NOT_INITIALIZED;
+      }
+    }
+
+    if (errorStatus != SBRDEC_OK) {
+      fDoDecodeSbrData = 0;
+    }
+  } /* if (fDoDecodeSbrData) */
+
+  /*
+    Print debugging output only if state has changed
+  */
+
+  /* read frame data */
+  if ((hSbrHeader->syncState >= SBR_HEADER) && fDoDecodeSbrData) {
+    int sbrFrameOk;
+    /* read the SBR element data */
+    if (stereo) {
+      sbrFrameOk = sbrGetChannelPairElement(hSbrHeader,
+                                            hFrameDataLeft,
+                                            hFrameDataRight,
+                                            hBs,
+                                            self->flags,
+                                            self->pSbrElement[elementIndex]->transposerSettings.overlap);
+    }
+    else {
+      if (self->hParametricStereoDec != NULL) {
+        /* update slot index for PS bitstream parsing */
+        self->hParametricStereoDec->bsLastSlot = self->hParametricStereoDec->bsReadSlot;
+        self->hParametricStereoDec->bsReadSlot = hSbrElement->useFrameSlot;
+      }
+      sbrFrameOk = sbrGetSingleChannelElement(hSbrHeader,
+                                              hFrameDataLeft,
+                                              hBs,
+                                              self->hParametricStereoDec,
+                                              self->flags,
+                                              self->pSbrElement[elementIndex]->transposerSettings.overlap);
+    }
+    if (!sbrFrameOk) {
+      fDoDecodeSbrData = 0;
+    }
+    else {
+      INT valBits;
+
+      if (bsPayLen > 0) {
+        valBits = bsPayLen - ((INT)startPos - (INT)FDKgetValidBits(hBs));
+      } else {
+        valBits = (INT)FDKgetValidBits(hBs);
+      }
+
+      if ( crcFlag == 1 ) {
+        switch (self->coreCodec) {
+        case AOT_ER_AAC_ELD:
+          {
+            /* late crc check for eld */
+            INT payloadbits = (INT)startPos - (INT)FDKgetValidBits(hBs) - startPos;
+            INT crcLen      = payloadbits - 10;
+            FDKpushBack(hBs, payloadbits);
+            fDoDecodeSbrData      = SbrCrcCheck (hBs, crcLen);
+            FDKpushFor(hBs, crcLen);
+          }
+          break;
+        default:
+          break;
+        }
+      }
+
+      /* sanity check of remaining bits */
+      if (valBits < 0) {
+        fDoDecodeSbrData = 0;
+      } else {
+        switch (self->coreCodec) {
+        case AOT_SBR:
+        case AOT_PS:
+        case AOT_AAC_LC:
+          {
+            /* This sanity check is only meaningful with General Audio bitstreams */
+            int alignBits = valBits & 0x7;
+
+            if (valBits > alignBits) {
+              fDoDecodeSbrData = 0;
+            }
+          }
+          break;
+        default:
+          /* No sanity check available */
+          break;
+        }
+      }
+    }
+  }
+
+  if (!fDoDecodeSbrData) {
+    /* Set error flag for this slot to trigger concealment */
+    self->pSbrElement[elementIndex]->frameErrorFlag[hSbrElement->useFrameSlot] = 1;
+    errorStatus = SBRDEC_PARSE_ERROR;
+  } else {
+    /* Everything seems to be ok so clear the error flag */
+    self->pSbrElement[elementIndex]->frameErrorFlag[hSbrElement->useFrameSlot] = 0;
+  }
+
+  if (!stereo) {
+    /* Turn coupling off explicitely to avoid access to absent right frame data
+       that might occur with corrupt bitstreams. */
+    hFrameDataLeft->coupling = COUPLING_OFF;
+  }
+
+bail:
+  if (errorStatus == SBRDEC_OK) {
+    if (headerStatus == HEADER_NOT_PRESENT) {
+      /* Use the old header for this frame */
+      hSbrElement->useHeaderSlot[hSbrElement->useFrameSlot] = lastHdrSlot;
+    } else {
+      /* Use the new header for this frame */
+      hSbrElement->useHeaderSlot[hSbrElement->useFrameSlot] = thisHdrSlot;
+    }
+
+    /* Move frame pointer to the next slot which is up to be decoded/applied next */
+    hSbrElement->useFrameSlot = (hSbrElement->useFrameSlot+1) % (self->numDelayFrames+1);
+  }
+
+  *count -= startPos - FDKgetValidBits(hBs);
+
+  return errorStatus;
+}
+
+
+/**
+ * \brief Render one SBR element into time domain signal.
+ * \param self SBR decoder handle
+ * \param timeData pointer to output buffer
+ * \param interleaved flag indicating interleaved channel output
+ * \param channelMapping pointer to UCHAR array where next 2 channel offsets are stored. 
+ * \param elementIndex enumerating index of the SBR element to render.
+ * \param numInChannels number of channels from core coder (reading stride).
+ * \param numOutChannels pointer to a location to return number of output channels.
+ * \param psPossible flag indicating if PS is possible or not.
+ * \return SBRDEC_OK if successfull, else error code
+ */
+static SBR_ERROR
+sbrDecoder_DecodeElement (
+        HANDLE_SBRDECODER    self,
+        INT_PCM             *timeData,
+        const int            interleaved,
+        const UCHAR         *channelMapping,
+        const int            elementIndex,
+        const int            numInChannels,
+        int                 *numOutChannels,
+        const int            psPossible
+        )
+{
+  SBR_DECODER_ELEMENT *hSbrElement = self->pSbrElement[elementIndex];
+  HANDLE_SBR_CHANNEL    *pSbrChannel = self->pSbrElement[elementIndex]->pSbrChannel;
+  HANDLE_SBR_HEADER_DATA hSbrHeader = &self->sbrHeader[elementIndex][hSbrElement->useHeaderSlot[hSbrElement->useFrameSlot]];
+  HANDLE_PS_DEC h_ps_d = self->hParametricStereoDec;
+
+  /* get memory for frame data from scratch */
+  SBR_FRAME_DATA *hFrameDataLeft  = &hSbrElement->pSbrChannel[0]->frameData[hSbrElement->useFrameSlot];
+  SBR_FRAME_DATA *hFrameDataRight = &hSbrElement->pSbrChannel[1]->frameData[hSbrElement->useFrameSlot];
+
+  SBR_ERROR errorStatus = SBRDEC_OK;
+
+
+  INT  strideIn, strideOut, offset0, offset1;
+  INT  codecFrameSize = self->codecFrameSize;
+
+  int  stereo = (hSbrElement->elementID == ID_CPE) ? 1 : 0;
+  int  numElementChannels = hSbrElement->nChannels; /* Number of channels of the current SBR element */
+
+  /* Update the header error flag */
+  hSbrHeader->frameErrorFlag = hSbrElement->frameErrorFlag[hSbrElement->useFrameSlot];
+
+  /*
+     Prepare filterbank for upsampling if no valid bit stream data is available.
+   */
+  if ( hSbrHeader->syncState == SBR_NOT_INITIALIZED )
+  {
+    errorStatus = initHeaderData(
+            hSbrHeader,
+            self->sampleRateIn,
+            self->sampleRateOut,
+            codecFrameSize,
+            self->flags
+            );
+
+    if (errorStatus != SBRDEC_OK) {
+      return errorStatus;
+    }
+
+    hSbrHeader->syncState = UPSAMPLING;
+
+    errorStatus = sbrDecoder_HeaderUpdate(
+            self,
+            hSbrHeader,
+            HEADER_NOT_PRESENT,
+            pSbrChannel,
+            hSbrElement->nChannels
+            );
+
+    if (errorStatus != SBRDEC_OK) {
+      hSbrHeader->syncState = SBR_NOT_INITIALIZED;
+      return errorStatus;
+    }
+  }
+
+  /* reset */
+  if (hSbrHeader->status & SBRDEC_HDR_STAT_RESET) {
+    int ch;
+    for (ch = 0 ; ch < numElementChannels; ch++) {
+      SBR_ERROR errorStatusTmp = SBRDEC_OK;
+
+      errorStatusTmp = resetSbrDec (
+             &pSbrChannel[ch]->SbrDec,
+              hSbrHeader,
+             &pSbrChannel[ch]->prevFrameData,
+              self->flags & SBRDEC_LOW_POWER,
+              self->synDownsampleFac
+              );
+
+      if (errorStatusTmp != SBRDEC_OK) {
+        errorStatus = errorStatusTmp;
+      }
+    }
+    hSbrHeader->status &= ~SBRDEC_HDR_STAT_RESET;
+  }
+
+  /* decoding */
+  if ( (hSbrHeader->syncState == SBR_ACTIVE)
+    || ((hSbrHeader->syncState == SBR_HEADER) && (hSbrHeader->frameErrorFlag == 0)) )
+  {
+    errorStatus = SBRDEC_OK;
+
+    decodeSbrData (hSbrHeader,
+                   hFrameDataLeft,
+                  &pSbrChannel[0]->prevFrameData,
+                   (stereo) ? hFrameDataRight : NULL,
+                   (stereo) ? &pSbrChannel[1]->prevFrameData : NULL);
+
+
+    /* Now we have a full parameter set and can do parameter
+       based concealment instead of plain upsampling. */
+    hSbrHeader->syncState = SBR_ACTIVE;
+  }
+
+  /* decode PS data if available */
+  if (h_ps_d != NULL && psPossible) {
+    int applyPs = 1;
+
+    /* define which frame delay line slot to process */
+    h_ps_d->processSlot = hSbrElement->useFrameSlot;
+
+    applyPs = DecodePs(h_ps_d, hSbrHeader->frameErrorFlag);
+    self->flags |= (applyPs) ? SBRDEC_PS_DECODED : 0;
+  }
+
+  /* Set strides for reading and writing */
+  if (interleaved) {
+    strideIn = numInChannels;
+    if ( psPossible )
+      strideOut = (numInChannels < 2) ? 2 : numInChannels;
+    else
+      strideOut = numInChannels;
+    offset0 = channelMapping[0];
+    offset1 = channelMapping[1];
+  } else {
+    strideIn  = 1;
+    strideOut = 1;
+    offset0 = channelMapping[0]*2*codecFrameSize;
+    offset1 = channelMapping[1]*2*codecFrameSize;
+  }
+
+  /* use same buffers for left and right channel and apply PS per timeslot */
+  /* Process left channel */
+//FDKprintf("self->codecFrameSize %d\t%d\n",self->codecFrameSize,self->sampleRateIn);
+  sbr_dec (&pSbrChannel[0]->SbrDec,
+            timeData + offset0,
+            timeData + offset0,
+           &pSbrChannel[1]->SbrDec,
+            timeData + offset1,
+            strideIn,
+            strideOut,
+            hSbrHeader,
+            hFrameDataLeft,
+           &pSbrChannel[0]->prevFrameData,
+            (hSbrHeader->syncState == SBR_ACTIVE),
+            h_ps_d,
+            self->flags
+          );
+
+  if (stereo) {
+    /* Process right channel */
+    sbr_dec (&pSbrChannel[1]->SbrDec,
+              timeData + offset1,
+              timeData + offset1,
+              NULL,
+              NULL,
+              strideIn,
+              strideOut,
+              hSbrHeader,
+              hFrameDataRight,
+             &pSbrChannel[1]->prevFrameData,
+              (hSbrHeader->syncState == SBR_ACTIVE),
+              NULL,
+              self->flags
+            );
+  }
+
+  if (h_ps_d != NULL) {
+    /* save PS status for next run */
+    h_ps_d->psDecodedPrv = (self->flags & SBRDEC_PS_DECODED) ? 1 : 0 ;
+  }
+
+  if ( psPossible 
+    )
+  {
+    FDK_ASSERT(strideOut > 1);
+    if ( !(self->flags & SBRDEC_PS_DECODED) ) {
+      /* A decoder which is able to decode PS has to produce a stereo output even if no PS data is availble. */
+      /* So copy left channel to right channel.                                                              */
+      if (interleaved) {
+        INT_PCM *ptr;
+        INT i;
+        FDK_ASSERT(strideOut == 2);
+
+        ptr = timeData;
+        for (i = codecFrameSize; i--; ) 
+        {
+          INT_PCM tmp; /* This temporal variable is required because some compilers can't do *ptr++ = *ptr++ correctly. */
+          tmp = *ptr++; *ptr++ = tmp;
+          tmp = *ptr++; *ptr++ = tmp;
+        }
+      } else {
+        FDKmemcpy( timeData+2*codecFrameSize, timeData, 2*codecFrameSize*sizeof(INT_PCM) );
+      }
+    }
+    *numOutChannels = 2;  /* Output minimum two channels when PS is enabled. */
+  }
+
+  return errorStatus;
+}
+
+
+SBR_ERROR sbrDecoder_Apply ( HANDLE_SBRDECODER   self,
+                             INT_PCM            *timeData,
+                             int                *numChannels,
+                             int                *sampleRate,
+                             const UCHAR         channelMapping[(6)],
+                             const int           interleaved,
+                             const int           coreDecodedOk,
+                             UCHAR              *psDecoded )
+{
+  SBR_ERROR errorStatus = SBRDEC_OK;
+
+  int   psPossible = 0;
+  int   sbrElementNum;
+  int   numCoreChannels = *numChannels;
+  int   numSbrChannels  = 0;
+
+  psPossible = *psDecoded;
+
+  if (self->numSbrElements < 1) {
+    /* exit immediately to avoid access violations */
+    return SBRDEC_CREATE_ERROR;
+  }
+
+  /* Sanity check of allocated SBR elements. */
+  for (sbrElementNum=0; sbrElementNum<self->numSbrElements; sbrElementNum++) {
+    if (self->pSbrElement[sbrElementNum] == NULL) {
+      return SBRDEC_CREATE_ERROR;
+    }
+  }
+
+  if (self->numSbrElements != 1 || self->pSbrElement[0]->elementID != ID_SCE) {
+    psPossible = 0;
+  }
+
+
+  /* In case of non-interleaved time domain data and upsampling, make room for bigger SBR output. */
+  if (self->synDownsampleFac == 1 && interleaved == 0) {
+    int c, outputFrameSize;
+
+    outputFrameSize =
+            self->pSbrElement[0]->pSbrChannel[0]->SbrDec.SynthesisQMF.no_channels
+            * self->pSbrElement[0]->pSbrChannel[0]->SbrDec.SynthesisQMF.no_col;
+
+    for (c=numCoreChannels-1; c>0; c--) {
+      FDKmemmove(timeData + c*outputFrameSize, timeData + c*self->codecFrameSize , self->codecFrameSize*sizeof(INT_PCM));
+    }
+  }
+
+
+  /* Make sure that even if no SBR data was found/parsed *psDecoded is returned 1 if psPossible was 0. */
+  if (psPossible == 0) {
+    self->flags &= ~SBRDEC_PS_DECODED;
+  }
+
+  /* Loop over SBR elements */
+  for (sbrElementNum = 0; sbrElementNum<self->numSbrElements; sbrElementNum++)
+  {
+    int numElementChan;
+
+    if (psPossible && self->pSbrElement[sbrElementNum]->pSbrChannel[1] == NULL) {
+      errorStatus = SBRDEC_UNSUPPORTED_CONFIG;
+      goto bail;
+    }
+
+    numElementChan = (self->pSbrElement[sbrElementNum]->elementID == ID_CPE) ? 2 : 1;
+
+    /* If core signal is bad then force upsampling */
+    if ( ! coreDecodedOk ) {
+      self->pSbrElement[sbrElementNum]->frameErrorFlag[self->pSbrElement[sbrElementNum]->useFrameSlot] = 1;
+    }
+
+    errorStatus = sbrDecoder_DecodeElement (
+                                 self,
+                                 timeData,
+                                 interleaved,
+                                 channelMapping,
+                                 sbrElementNum,
+                                 numCoreChannels,
+                                &numElementChan,
+                                 psPossible
+                               );
+
+    if (errorStatus != SBRDEC_OK) {
+      goto bail;
+    }
+
+    numSbrChannels += numElementChan;
+    channelMapping += numElementChan;
+
+    if (numSbrChannels >= numCoreChannels) {
+      break;
+    }
+  }
+
+  /* Update numChannels and samplerate */
+  *numChannels = numSbrChannels;
+  *sampleRate = self->sampleRateOut;
+  *psDecoded = (self->flags & SBRDEC_PS_DECODED) ? 1 : 0;
+
+
+
+bail:
+
+  return errorStatus;
+}
+
+
+SBR_ERROR sbrDecoder_Close ( HANDLE_SBRDECODER *pSelf )
+{
+  HANDLE_SBRDECODER self = *pSelf;
+  int i;
+
+  if (self != NULL)
+  {
+    if (self->hParametricStereoDec != NULL) {
+      DeletePsDec ( &self->hParametricStereoDec );
+    }
+
+    if (self->workBuffer1 != NULL) {
+      FreeRam_SbrDecWorkBuffer1(&self->workBuffer1);
+    }
+    if (self->workBuffer2 != NULL) {
+      FreeRam_SbrDecWorkBuffer2(&self->workBuffer2);
+    }
+
+    for (i = 0; i < (4); i++) {
+      sbrDecoder_DestroyElement( self, i );
+    }
+
+    FreeRam_SbrDecoder(pSelf);
+  }
+
+  return SBRDEC_OK;
+}
+
+
+INT sbrDecoder_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_SBRDEC;
+  info->version = LIB_VERSION(SBRDECODER_LIB_VL0, SBRDECODER_LIB_VL1, SBRDECODER_LIB_VL2);
+  LIB_VERSION_STRING(info);
+  info->build_date = (char *)SBRDECODER_LIB_BUILD_DATE;
+  info->build_time = (char *)SBRDECODER_LIB_BUILD_TIME;
+  info->title      = (char *)SBRDECODER_LIB_TITLE;
+
+  /* Set flags */
+  info->flags = 0
+    | CAPF_SBR_HQ
+    | CAPF_SBR_LP
+    | CAPF_SBR_PS_MPEG
+    | CAPF_SBR_CONCEALMENT
+    | CAPF_SBR_DRC
+      ;
+  /* End of flags */
+
+  return 0;
+}
+
diff --git a/libSBRdec/src/transcendent.h b/libSBRdec/src/transcendent.h
new file mode 100644
index 0000000..f0ee21e
--- /dev/null
+++ b/libSBRdec/src/transcendent.h
@@ -0,0 +1,355 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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 Fixed Point Arithmetic Library Interface  
+*/
+
+#ifndef __TRANSCENDENT_H
+#define __TRANSCENDENT_H
+
+#include "sbrdecoder.h"
+#include "sbr_rom.h"
+
+/************************************************************************/
+/*!
+  \brief   Get number of octaves between frequencies a and b
+
+  The Result is scaled with 1/8.
+  The valid range for a and b is 1 to LOG_DUALIS_TABLE_SIZE.
+
+  \return   ld(a/b) / 8
+*/
+/************************************************************************/
+static inline FIXP_SGL FDK_getNumOctavesDiv8(INT a, /*!< lower band */
+                                             INT b) /*!< upper band */
+{
+  return ( (SHORT)((LONG)(CalcLdInt(b) - CalcLdInt(a))>>(FRACT_BITS-3)) );
+}
+
+
+/************************************************************************/
+/*!
+  \brief   Add two values given by mantissa and exponent.
+
+  Mantissas are in fract format with values between 0 and 1. <br>
+  The base for exponents is 2.  Example:  \f$  a = a\_m * 2^{a\_e}  \f$<br>
+*/
+/************************************************************************/
+inline void FDK_add_MantExp(FIXP_SGL a_m, /*!< Mantissa of 1st operand a */
+                     SCHAR     a_e,       /*!< Exponent of 1st operand a */
+                     FIXP_SGL  b_m,       /*!< Mantissa of 2nd operand b */
+                     SCHAR     b_e,       /*!< Exponent of 2nd operand b */
+                     FIXP_SGL *ptrSum_m,  /*!< Mantissa of result */
+                     SCHAR    *ptrSum_e)  /*!< Exponent of result */
+{
+  FIXP_DBL accu;
+  int   shift;
+  int   shiftAbs;
+
+  FIXP_DBL shiftedMantissa;
+  FIXP_DBL otherMantissa;
+
+  /* Equalize exponents of the summands.
+     For the smaller summand, the exponent is adapted and
+     for compensation, the mantissa is shifted right. */
+
+  shift = (int)(a_e - b_e);
+
+  shiftAbs = (shift>0)? shift : -shift;
+  shiftAbs = (shiftAbs < DFRACT_BITS-1)? shiftAbs : DFRACT_BITS-1;
+  shiftedMantissa = (shift>0)? (FX_SGL2FX_DBL(b_m) >> shiftAbs) : (FX_SGL2FX_DBL(a_m) >> shiftAbs);
+  otherMantissa = (shift>0)? FX_SGL2FX_DBL(a_m) : FX_SGL2FX_DBL(b_m);
+  *ptrSum_e = (shift>0)? a_e : b_e;
+
+  accu = (shiftedMantissa >> 1) + (otherMantissa >> 1);
+  /* shift by 1 bit to avoid overflow */
+
+  if ( (accu >= (FL2FXCONST_DBL(0.5f) - (FIXP_DBL)1)) || (accu <= FL2FXCONST_DBL(-0.5f)) )
+    *ptrSum_e += 1;
+  else
+    accu = (shiftedMantissa + otherMantissa);
+
+  *ptrSum_m = FX_DBL2FX_SGL(accu);
+
+}
+
+inline void FDK_add_MantExp(FIXP_DBL a,   /*!< Mantissa of 1st operand a */
+                     SCHAR     a_e,       /*!< Exponent of 1st operand a */
+                     FIXP_DBL  b,         /*!< Mantissa of 2nd operand b */
+                     SCHAR     b_e,       /*!< Exponent of 2nd operand b */
+                     FIXP_DBL *ptrSum,    /*!< Mantissa of result */
+                     SCHAR    *ptrSum_e)  /*!< Exponent of result */
+{
+  FIXP_DBL accu;
+  int   shift;
+  int   shiftAbs;
+
+  FIXP_DBL shiftedMantissa;
+  FIXP_DBL otherMantissa;
+
+  /* Equalize exponents of the summands.
+     For the smaller summand, the exponent is adapted and
+     for compensation, the mantissa is shifted right. */
+
+  shift = (int)(a_e - b_e);
+
+  shiftAbs = (shift>0)? shift : -shift;
+  shiftAbs = (shiftAbs < DFRACT_BITS-1)? shiftAbs : DFRACT_BITS-1;
+  shiftedMantissa = (shift>0)? (b >> shiftAbs) : (a >> shiftAbs);
+  otherMantissa = (shift>0)? a : b;
+  *ptrSum_e = (shift>0)? a_e : b_e;
+
+  accu = (shiftedMantissa >> 1) + (otherMantissa >> 1);
+  /* shift by 1 bit to avoid overflow */
+
+  if ( (accu >= (FL2FXCONST_DBL(0.5f) - (FIXP_DBL)1)) || (accu <= FL2FXCONST_DBL(-0.5f)) )
+    *ptrSum_e += 1;
+  else
+    accu = (shiftedMantissa + otherMantissa);
+
+  *ptrSum = accu;
+
+}
+
+/************************************************************************/
+/*!
+  \brief   Divide two values given by mantissa and exponent.
+
+  Mantissas are in fract format with values between 0 and 1. <br>
+  The base for exponents is 2.  Example:  \f$  a = a\_m * 2^{a\_e}  \f$<br>
+
+  For performance reasons, the division is based on a table lookup
+  which limits accuracy.
+*/
+/************************************************************************/
+static inline void FDK_divide_MantExp(FIXP_SGL a_m,           /*!< Mantissa of dividend a */
+                                      SCHAR     a_e,          /*!< Exponent of dividend a */
+                                      FIXP_SGL  b_m,          /*!< Mantissa of divisor b */
+                                      SCHAR     b_e,          /*!< Exponent of divisor b */
+                                      FIXP_SGL *ptrResult_m,  /*!< Mantissa of quotient a/b */
+                                      SCHAR    *ptrResult_e)  /*!< Exponent of quotient a/b */
+
+{
+  int preShift, postShift, index, shift;
+  FIXP_DBL ratio_m;
+  FIXP_SGL  bInv_m = FL2FXCONST_SGL(0.0f);
+
+  preShift = CntLeadingZeros(FX_SGL2FX_DBL(b_m));
+
+  /*
+    Shift b into the range from 0..INV_TABLE_SIZE-1,
+
+    E.g. 10 bits must be skipped for INV_TABLE_BITS 8:
+    - leave 8 bits as index for table
+    - skip sign bit,
+    - skip first bit of mantissa, because this is always the same (>0.5)
+
+    We are dealing with energies, so we need not care
+    about negative numbers
+  */
+
+  /*
+    The first interval has half width so the lowest bit of the index is
+    needed for a doubled resolution.
+  */
+  shift = (FRACT_BITS - 2 - INV_TABLE_BITS - preShift);
+
+  index = (shift<0)? (LONG)b_m << (-shift) : (LONG)b_m >> shift;
+
+
+  /* The index has INV_TABLE_BITS +1 valid bits here. Clear the other bits. */
+  index &= (1 << (INV_TABLE_BITS+1)) - 1;
+
+    /* Remove offset of half an interval */
+  index--;
+
+    /* Now the lowest bit is shifted out */
+  index = index >> 1;
+
+    /* Fetch inversed mantissa from table: */
+  bInv_m = (index<0)? bInv_m : FDK_sbrDecoder_invTable[index];
+
+    /* Multiply a with the inverse of b: */
+  ratio_m = (index<0)? FX_SGL2FX_DBL(a_m >> 1) : fMultDiv2(bInv_m,a_m);
+
+  postShift = CntLeadingZeros(ratio_m)-1;
+
+  *ptrResult_m = FX_DBL2FX_SGL(ratio_m << postShift);
+  *ptrResult_e = a_e - b_e + 1 + preShift - postShift;
+}
+
+static inline void FDK_divide_MantExp(FIXP_DBL a_m,           /*!< Mantissa of dividend a */
+                                      SCHAR     a_e,          /*!< Exponent of dividend a */
+                                      FIXP_DBL  b_m,          /*!< Mantissa of divisor b */
+                                      SCHAR     b_e,          /*!< Exponent of divisor b */
+                                      FIXP_DBL *ptrResult_m,  /*!< Mantissa of quotient a/b */
+                                      SCHAR    *ptrResult_e)  /*!< Exponent of quotient a/b */
+
+{
+  int preShift, postShift, index, shift;
+  FIXP_DBL ratio_m;
+  FIXP_SGL  bInv_m = FL2FXCONST_SGL(0.0f);
+
+  preShift = CntLeadingZeros(b_m);
+
+  /*
+    Shift b into the range from 0..INV_TABLE_SIZE-1,
+
+    E.g. 10 bits must be skipped for INV_TABLE_BITS 8:
+    - leave 8 bits as index for table
+    - skip sign bit,
+    - skip first bit of mantissa, because this is always the same (>0.5)
+
+    We are dealing with energies, so we need not care
+    about negative numbers
+  */
+
+  /*
+    The first interval has half width so the lowest bit of the index is
+    needed for a doubled resolution.
+  */
+  shift = (DFRACT_BITS - 2 - INV_TABLE_BITS - preShift);
+
+  index = (shift<0)? (LONG)b_m << (-shift) : (LONG)b_m >> shift;
+
+
+  /* The index has INV_TABLE_BITS +1 valid bits here. Clear the other bits. */
+  index &= (1 << (INV_TABLE_BITS+1)) - 1;
+
+    /* Remove offset of half an interval */
+  index--;
+
+    /* Now the lowest bit is shifted out */
+  index = index >> 1;
+
+    /* Fetch inversed mantissa from table: */
+  bInv_m = (index<0)? bInv_m : FDK_sbrDecoder_invTable[index];
+
+    /* Multiply a with the inverse of b: */
+  ratio_m = (index<0)? (a_m >> 1) : fMultDiv2(bInv_m,a_m);
+
+  postShift = CntLeadingZeros(ratio_m)-1;
+
+  *ptrResult_m = ratio_m << postShift;
+  *ptrResult_e = a_e - b_e + 1 + preShift - postShift;
+}
+
+/*!
+  \brief   Calculate the squareroot of a number given by mantissa and exponent
+
+  Mantissa is in fract format with values between 0 and 1. <br>
+  The base for the exponent is 2.  Example:  \f$  a = a\_m * 2^{a\_e}  \f$<br>
+  The operand is addressed via pointers and will be overwritten with the result.
+
+  For performance reasons, the square root is based on a table lookup
+  which limits accuracy.
+*/
+static inline void FDK_sqrt_MantExp(FIXP_DBL *mantissa,    /*!< Pointer to mantissa */
+                                    SCHAR    *exponent,
+                                    const SCHAR *destScale)
+{
+  FIXP_DBL input_m = *mantissa;
+  int   input_e = (int) *exponent;
+  FIXP_DBL result = FL2FXCONST_DBL(0.0f);
+  int    result_e = -FRACT_BITS;
+
+  /* Call lookup square root, which does internally normalization. */
+  result   = sqrtFixp_lookup(input_m, &input_e);
+  result_e = input_e;
+
+  /* Write result */
+  if (exponent==destScale) {
+    *mantissa = result;
+    *exponent = result_e;
+  } else {
+    int shift = result_e - *destScale;
+    *mantissa = (shift>=0) ? result << (INT)fixMin(DFRACT_BITS-1,shift)
+                           : result >> (INT)fixMin(DFRACT_BITS-1,-shift);
+    *exponent = *destScale;
+  }
+}
+
+
+#endif