Snapshot 2bda038c163298531d47394bc2c09e1409c5d0dbandroid-cts-4.1_r4android-cts-4.1_r2android-cts-4.1_r1android-4.1_pre2_android-4.1.2_r2.1android-4.1.2_r2android-4.1.2_r1android-4.1.1_r6.1android-4.1.1_r6android-4.1.1_r5android-4.1.1_r4android-4.1.1_r3android-4.1.1_r2android-4.1.1_r1_android-4.1.1_r1.1android-4.1-sdk_pre1_jb-releasejb-mr0-releasejb-dev

Change-Id: If584e579464f28b97d50e51fc76ba654a5536c54

13 files changed, 4688 insertions, 0 deletions

diff --git a/libMpegTPDec/include/mpegFileRead.h b/libMpegTPDec/include/mpegFileRead.h
new file mode 100644
index 0000000..d45bee2
--- /dev/null
+++ b/libMpegTPDec/include/mpegFileRead.h
@@ -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
+----------------------------------------------------------------------------------------------------------- */
+
+/*****************************  MPEG-4 AAC Decoder  **************************
+
+   Author(s):   Manuel Jander
+   Description: Bitstream data provider for MP4 decoders
+
+******************************************************************************/
+
+#include "machine_type.h"
+#include "FDK_audio.h"
+
+#define MPFREAD_MP4FF_DISABLE
+
+#ifndef MPFREAD_MP4FF_DISABLE
+  /*!< If MPFREAD_MP4FF_ENABLE is set, include support for MPEG ISO fileformat.
+       If not set, no .mp4, .m4a and .3gp files can be used for input. */
+  #define MPFREAD_MP4FF_ENABLE
+#endif
+
+/* maximum number of layers which can be read        */
+/* shall equal max number of layers read by iisisoff */
+#define FILEREAD_MAX_LAYERS (2)
+
+typedef struct STRUCT_FILEREAD *HANDLE_FILEREAD;
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/**
+ * \brief           Open an MPEG audio file and try to detect its format.
+ * \param filename  String of the filename to be opened.
+ * \param fileFormat Skip file format detection and use given format if fileFormat != FF_UNKNOWN.
+                    Else store detected format into *fileFmt.
+ * \param transportType Skip transport type detection and use given format if transportType != TT_UNKNOWN.
+                    Else store detected format into *fileFmt.
+ * \param conf      Pointer to unsigned char to hold the AudioSpecificConfig of the input file, if
+                    any (MPEG 4 file format). In case of RAW LATM it holds the StreamMuxConfig.
+ * \param confSize  Pointer to an integer, where the length of the ASC or SMC (in case of RAW LATM)
+                    is stored to.
+ * \return          MPEG file read handle.
+ */
+HANDLE_FILEREAD mpegFileRead_Open( const char     *filename,
+                                   FILE_FORMAT     fileFormat,
+                                   TRANSPORT_TYPE  transportType,
+                                   UCHAR          *conf[],
+                                   UINT            confSize[],
+                                   INT            *noOfLayers
+                                 );
+
+/**
+ * \brief           Get the file format of the input file.
+ * \param hDataSrc  MPEG file read handle.
+ * \return          File format of the input file.
+ */
+FILE_FORMAT mpegFileRead_GetFileFormat(HANDLE_FILEREAD hDataSrc);
+
+/**
+ * \brief           Get the transport type of the input file.
+ * \param hDataSrc  MPEG file read handle.
+ * \return          Transport type of the input file.
+ */
+TRANSPORT_TYPE mpegFileRead_GetTransportType(HANDLE_FILEREAD hDataSrc);
+
+/**
+ * \brief Read data from MPEG file. In case of packet file, read one packet, in case
+ *        of streaming file with embedded synchronisation layer (LOAS/ADTS...), just
+ *        fill the buffer.
+ *
+ * \param hMpegFile   MPEG file read handle.
+ * \param inBuffer    Pointer to input buffer.
+ * \param bufferSize  Size of input buffer.
+ * \param bytesValid  Number of bytes that were read.
+ * \return            0 on success, -1 if unsupported file format or file read error.
+ */
+int mpegFileRead_Read( HANDLE_FILEREAD   hMpegFile,
+                       UCHAR            *inBuffer[],
+                       UINT              bufferSize,
+                       UINT             *bytesValid
+                     );
+
+/**
+ * \brief            Seek in file from origin by given offset in frames.
+ * \param hMpegFile  MPEG file read handle.
+ * \param origin     If 0, the origin is the file beginning (absolute seek).
+ *                   If 1, the origin is the current position (relative seek).
+ * \param offset     The amount of frames to seek from the given origin.
+ * \return           0 on sucess, -1 if offset < 0 or file read error.
+ */
+int mpegFileRead_seek( HANDLE_FILEREAD   hMpegFile,
+                       INT               origin,
+                       INT               offset
+                      );
+
+/**
+ * \brief            Get file position in percent.
+ * \param hMpegFile  MPEG file read handle.
+ * \return           File position in percent.
+ */
+int mpegFileRead_getPercent(HANDLE_FILEREAD hMpegFile);
+
+
+/**
+ * \brief           Close MPEG audio file.
+ * \param hMpegFile Mpeg file read handle.
+ * \return          0 on sucess.
+ */
+int mpegFileRead_Close(HANDLE_FILEREAD *hMpegFile);
+
+#ifdef __cplusplus
+}
+#endif
diff --git a/libMpegTPDec/include/tp_data.h b/libMpegTPDec/include/tp_data.h
new file mode 100644
index 0000000..6e8218a
--- /dev/null
+++ b/libMpegTPDec/include/tp_data.h
@@ -0,0 +1,340 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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): Manuel Jander
+   Description: MPEG Transport data tables
+
+******************************************************************************/
+
+#ifndef __TP_DATA_H__
+#define __TP_DATA_H__
+
+#include "machine_type.h"
+#include "FDK_audio.h"
+#include "FDK_bitstream.h"
+
+/*
+ * Configuration
+ */
+#define TP_GA_ENABLE
+/* #define TP_CELP_ENABLE */
+/* #define TP_HVXC_ENABLE */
+/* #define TP_SLS_ENABLE */
+#define TP_ELD_ENABLE
+/* #define TP_USAC_ENABLE */
+/* #define TP_RSVD50_ENABLE */
+
+#if defined(TP_GA_ENABLE) || defined(TP_SLS_ENABLE)
+#define TP_PCE_ENABLE       /**< Enable full PCE support */
+#endif
+
+/**
+ * ProgramConfig struct.
+ */
+/* ISO/IEC 14496-3 4.4.1.1 Table 4.2 Program config element */
+#define PC_FSB_CHANNELS_MAX 16 /* Front/Side/Back channels */
+#define PC_LFE_CHANNELS_MAX 4
+#define PC_ASSOCDATA_MAX    8
+#define PC_CCEL_MAX         16 /* CC elements */
+#define PC_COMMENTLENGTH    256
+
+typedef struct
+{
+#ifdef TP_PCE_ENABLE
+  /* PCE bitstream elements: */
+  UCHAR ElementInstanceTag;
+  UCHAR Profile;
+  UCHAR SamplingFrequencyIndex;
+  UCHAR NumFrontChannelElements;
+  UCHAR NumSideChannelElements;
+  UCHAR NumBackChannelElements;
+  UCHAR NumLfeChannelElements;
+  UCHAR NumAssocDataElements;
+  UCHAR NumValidCcElements;
+
+  UCHAR MonoMixdownPresent;
+  UCHAR MonoMixdownElementNumber;
+
+  UCHAR StereoMixdownPresent;
+  UCHAR StereoMixdownElementNumber;
+
+  UCHAR MatrixMixdownIndexPresent;
+  UCHAR MatrixMixdownIndex;
+  UCHAR PseudoSurroundEnable;
+
+  UCHAR FrontElementIsCpe[PC_FSB_CHANNELS_MAX];
+  UCHAR FrontElementTagSelect[PC_FSB_CHANNELS_MAX];
+
+  UCHAR SideElementIsCpe[PC_FSB_CHANNELS_MAX];
+  UCHAR SideElementTagSelect[PC_FSB_CHANNELS_MAX];
+
+  UCHAR BackElementIsCpe[PC_FSB_CHANNELS_MAX];
+  UCHAR BackElementTagSelect[PC_FSB_CHANNELS_MAX];
+
+  UCHAR LfeElementTagSelect[PC_LFE_CHANNELS_MAX];
+
+  UCHAR AssocDataElementTagSelect[PC_ASSOCDATA_MAX];
+
+  UCHAR CcElementIsIndSw[PC_CCEL_MAX];
+  UCHAR ValidCcElementTagSelect[PC_CCEL_MAX];
+
+  UCHAR CommentFieldBytes;
+  UCHAR Comment[PC_COMMENTLENGTH];
+#endif /* TP_PCE_ENABLE */
+
+  /* Helper variables for administration: */
+  UCHAR isValid;               /*!< Flag showing if PCE has been read successfully. */
+  UCHAR NumChannels;           /*!< Amount of audio channels summing all channel elements including LFEs */
+  UCHAR NumEffectiveChannels;  /*!< Amount of audio channels summing only SCEs and CPEs */
+  UCHAR elCounter;
+
+} CProgramConfig;
+
+typedef enum {
+  ASCEXT_UNKOWN = -1,
+  ASCEXT_SBR    = 0x2b7,
+  ASCEXT_PS     = 0x548,
+  ASCEXT_MPS    = 0x76a,
+  ASCEXT_SAOC   = 0x7cb,
+  ASCEXT_LDMPS  = 0x7cc
+
+} TP_ASC_EXTENSION_ID;
+
+#ifdef TP_GA_ENABLE
+/**
+ * GaSpecificConfig struct
+ */
+typedef struct {
+  UINT m_frameLengthFlag ;
+  UINT m_dependsOnCoreCoder ;
+  UINT m_coreCoderDelay ;
+
+  UINT m_extensionFlag ;
+  UINT m_extensionFlag3 ;
+
+  UINT m_layer;
+  UINT m_numOfSubFrame;
+  UINT m_layerLength;
+
+} CSGaSpecificConfig;
+#endif /* TP_GA_ENABLE */
+
+
+
+
+#ifdef TP_ELD_ENABLE
+
+typedef enum {
+  ELDEXT_TERM  = 0x0,   /* Termination tag */
+  ELDEXT_SAOC  = 0x1,   /* SAOC config */
+  ELDEXT_LDSAC = 0x2    /* LD MPEG Surround config */
+  /* reserved */
+} ASC_ELD_EXT_TYPE;
+
+typedef struct {
+  UCHAR m_frameLengthFlag;
+
+  UCHAR m_sbrPresentFlag;
+  UCHAR m_useLdQmfTimeAlign;  /* Use LD-MPS QMF in SBR to achive time alignment */
+  UCHAR m_sbrSamplingRate;
+  UCHAR m_sbrCrcFlag;
+
+} CSEldSpecificConfig;
+#endif /* TP_ELD_ENABLE */
+
+
+
+
+/**
+ * Audio configuration struct, suitable for encoder and decoder configuration.
+ */
+typedef struct {
+
+  /* XYZ Specific Data */
+  union {
+#ifdef TP_GA_ENABLE
+    CSGaSpecificConfig    m_gaSpecificConfig;       /**< General audio specific configuration.         */
+#endif /* TP_GA_ENABLE */
+#ifdef TP_ELD_ENABLE
+    CSEldSpecificConfig   m_eldSpecificConfig;      /**< ELD specific configuration.                   */
+#endif /* TP_ELD_ENABLE */
+  } m_sc;
+  
+  /* Common ASC parameters */
+#ifdef TP_PCE_ENABLE
+  CProgramConfig        m_progrConfigElement;     /**< Program configuration.                          */
+#endif /* TP_PCE_ENABLE */
+
+  AUDIO_OBJECT_TYPE     m_aot;                    /**< Audio Object Type.                              */
+  UINT                  m_samplingFrequency;      /**< Samplerate.                                     */
+  UINT                  m_samplesPerFrame;        /**< Amount of samples per frame.                    */
+  UINT                  m_directMapping;          /**< Document this please !!                         */
+
+  AUDIO_OBJECT_TYPE     m_extensionAudioObjectType;        /**< Audio object type                      */
+  UINT                  m_extensionSamplingFrequency;      /**< Samplerate                             */  
+
+  SCHAR                 m_channelConfiguration;   /**< Channel configuration index                     */
+
+  SCHAR                 m_epConfig;               /**< Error protection index                           */
+  SCHAR                 m_vcb11Flag;              /**< aacSectionDataResilienceFlag                     */
+  SCHAR                 m_rvlcFlag;               /**< aacScalefactorDataResilienceFlag                 */
+  SCHAR                 m_hcrFlag;                /**< aacSpectralDataResilienceFlag                    */
+
+  SCHAR                 m_sbrPresentFlag;         /**< Flag indicating the presence of SBR data in the bitstream               */
+  SCHAR                 m_psPresentFlag;          /**< Flag indicating the presence of parametric stereo data in the bitstream */
+  UCHAR                 m_samplingFrequencyIndex; /**< Samplerate index                                 */
+  UCHAR                 m_extensionSamplingFrequencyIndex; /**< Samplerate index                        */
+  SCHAR                 m_extensionChannelConfiguration;   /**< Channel configuration index             */
+
+} CSAudioSpecificConfig;
+
+typedef INT (*cbUpdateConfig_t)(void*, const CSAudioSpecificConfig*);
+typedef INT (*cbSsc_t)(
+        void*, HANDLE_FDK_BITSTREAM,
+        const AUDIO_OBJECT_TYPE coreCodec,
+        const INT samplingFrequency,
+        const INT muxMode,
+        const INT configBytes
+        );
+typedef INT (*cbSbr_t)(
+        void *                  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
+        );
+
+typedef struct  {
+  cbUpdateConfig_t cbUpdateConfig; /*!< Function pointer for Config change notify callback.  */
+  void *cbUpdateConfigData;        /*!< User data pointer for Config change notify callback. */
+  cbSsc_t cbSsc;                   /*!< Function pointer for SSC parser callback. */
+  void *cbSscData;                 /*!< User data pointer for SSC parser callback. */
+  cbSbr_t cbSbr;                   /*!< Function pointer for SBR header parser callback. */
+  void *cbSbrData;                 /*!< User data pointer for SBR header parser callback. */
+} CSTpCallBacks;
+
+static const UINT SamplingRateTable[] = 
+{ 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050, 16000, 12000, 11025, 8000, 7350, 0, 0,
+  0
+};
+
+static inline
+int getSamplingRateIndex( UINT samplingRate )
+{
+  UINT sf_index, tableSize=sizeof(SamplingRateTable)/sizeof(UINT);
+
+  for (sf_index=0; sf_index<tableSize; sf_index++) {
+    if( SamplingRateTable[sf_index] == samplingRate ) break;
+  }
+
+  if (sf_index>tableSize-1) {
+    return tableSize-1;
+  }
+
+  return sf_index;
+}
+
+/*
+ * Get Channel count from channel configuration
+ */
+static inline int getNumberOfTotalChannels(int channelConfig)
+{
+  if (channelConfig > 0 && channelConfig < 8)
+    return (channelConfig == 7)?8:channelConfig;
+  else
+    return 0;
+}
+
+static inline
+int getNumberOfEffectiveChannels(const int channelConfig)
+{
+  const int n[] = {0,1,2,3,4,5,5,7};
+  return n[channelConfig];
+}
+
+#endif /* __TP_DATA_H__ */
diff --git a/libMpegTPDec/include/tpdec_lib.h b/libMpegTPDec/include/tpdec_lib.h
new file mode 100644
index 0000000..7317204
--- /dev/null
+++ b/libMpegTPDec/include/tpdec_lib.h
@@ -0,0 +1,493 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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 Transport Decoder  ***********************
+
+   Author(s): Manuel Jander
+   Description: MPEG Transport decoder
+
+******************************************************************************/
+
+#ifndef __TPDEC_LIB_H__
+#define __TPDEC_LIB_H__
+
+#include "tp_data.h"
+
+#include "FDK_bitstream.h"
+
+#define TRANSPORTDEC_INBUF_SIZE  ( 8192 )   /*!< Size is in bytes.
+                                                 Set the transport input buffer size carefully and
+                                                 assure that it fulfills the requirements of the
+                                                 supported transport format(s).                    */
+
+typedef enum {
+  TRANSPORTDEC_OK = 0,             /*!< All fine.                                                  */
+
+  /* Synchronization errors. Wait for new input data and try again. */
+  tpdec_sync_error_start   = 0x100,
+  TRANSPORTDEC_NOT_ENOUGH_BITS,    /*!< Out of bits. Provide more bits and try again.              */
+  TRANSPORTDEC_SYNC_ERROR,         /*!< No sync was found or sync got lost. Keep trying.           */
+  tpdec_sync_error_end,
+
+  /* Decode errors. Mostly caused due to bit errors. */
+  tpdec_decode_error_start = 0x400,
+  TRANSPORTDEC_PARSE_ERROR,        /*!< Bitstream data showed inconsistencies (wrong syntax).      */
+  TRANSPORTDEC_UNSUPPORTED_FORMAT, /*!< Unsupported format or feature found in the bitstream data. */
+  TRANSPORTDEC_CRC_ERROR,          /*!< CRC error encountered in bitstream data.                   */
+  tpdec_decode_error_end,
+
+  /* Fatal errors. Stop immediately on one of these errors! */
+  tpdec_fatal_error_start  = 0x200,
+  TRANSPORTDEC_UNKOWN_ERROR,       /*!< An unknown error occured.                                  */
+  TRANSPORTDEC_INVALID_PARAMETER,  /*!< An invalid parameter was passed to a function.             */
+  TRANSPORTDEC_NEED_TO_RESTART,    /*!< The decoder needs to be restarted, since the requiered
+                                        configuration change cannot be performed.                  */
+  tpdec_fatal_error_end
+
+} TRANSPORTDEC_ERROR;
+
+
+/** Macro to identify decode errors. */
+#define TPDEC_IS_DECODE_ERROR(err) ( ((err>=tpdec_decode_error_start) && (err<=tpdec_decode_error_end)) ? 1 : 0)
+/** Macro to identify fatal errors. */
+#define TPDEC_IS_FATAL_ERROR(err)  ( ((err>=tpdec_fatal_error_start)  && (err<=tpdec_fatal_error_end))  ? 1 : 0)
+
+
+/**
+ * \brief Parameter identifiers for transportDec_SetParam()
+ */
+typedef enum {
+  TPDEC_PARAM_MINIMIZE_DELAY = 1,        /** Delay minimization strategy. 0: none, 1: discard as many frames as possible. */
+  TPDEC_PARAM_EARLY_CONFIG,              /** Enable early config discovery. */
+  TPDEC_PARAM_IGNORE_BUFFERFULLNESS,     /** Ignore buffer fullness. */
+  TPDEC_PARAM_SET_BITRATE,               /** Set average bit rate for bit stream interruption frame misses estimation. */
+  TPDEC_PARAM_RESET,                     /** Reset transport decoder instance status. */
+  TPDEC_PARAM_BURST_PERIOD               /** Set data reception burst period in mili seconds. */
+} TPDEC_PARAM;
+
+/* ISO/IEC 14496-3 4.4.1.1 Table 4.2 Program config element */
+#define PC_FSB_CHANNELS_MAX 16 /* Front/Side/Back channels */
+#define PC_LFE_CHANNELS_MAX 4
+#define PC_ASSOCDATA_MAX    8
+#define PC_CCEL_MAX         16 /* CC elements */
+#define PC_COMMENTLENGTH    256
+
+
+/*!
+  \brief               Reset Program Config Element.
+  \param pPce          Program Config Element structure.
+  \return              void
+*/
+void  CProgramConfig_Reset ( CProgramConfig *pPce );
+
+/*!
+  \brief               Initialize Program Config Element.
+  \param pPce          Program Config Element structure.
+  \return              void
+*/
+void  CProgramConfig_Init ( CProgramConfig *pPce );
+
+/*!
+  \brief               Inquire state of present Program Config Element structure.
+  \param pPce          Program Config Element structure.
+  \return              1 if the PCE structure is filled correct,
+                       0 if no valid PCE present.
+*/
+int  CProgramConfig_IsValid ( const CProgramConfig *pPce );
+
+#ifdef TP_PCE_ENABLE
+/*!
+  \brief               Read Program Config Element.
+  \param pPce          Program Config Element structure.
+  \param bs            Bitstream buffer to read from.
+  \param alignAnchor   Align bitstream to alignAnchor bits after all read operations.
+  \return              void
+*/
+void  CProgramConfig_Read ( CProgramConfig       *pPce,
+                            HANDLE_FDK_BITSTREAM  bs,
+                            UINT                  alignAnchor );
+#endif /* TP_PCE_ENABLE */
+
+/**
+ * \brief Lookup and verify a given element. The decoder calls this
+ *        method with every new element ID found in the bitstream.
+ *
+ * \param pPce        A valid Program config structure.
+ * \param tag         Tag of the current element to be looked up.
+ * \param channelIdx  The current channel count of the decoder parser.
+ * \param chMapping   Array to store the canonical channel mapping indexes.
+ * \param chType      Array to store the audio channel type.
+ * \param chIndex     Array to store the individual audio channel type index.
+ * \param elMapping   Pointer where the canonical element index is stored.
+ * \param elType      The element id of the current element to be looked up.
+ *
+ * \return            Non-zero if the element belongs to the current program, zero
+ *                    if it does not.
+ */
+int CProgramConfig_LookupElement(
+        CProgramConfig *pPce,
+        const UINT      channelConfig,
+        const UINT      tag,
+        const UINT      channelIdx,
+        UCHAR           chMapping[],
+        AUDIO_CHANNEL_TYPE chType[],
+        UCHAR           chIndex[],
+        UCHAR          *elMapping,
+        MP4_ELEMENT_ID  elList[],
+        MP4_ELEMENT_ID  elType
+        );
+
+/**
+ * \brief        Get table of elements in canonical order.
+ * \param pPce   A valid program config structure.
+ * \param table  An array where the element IDs are stored.
+ * \return       Total element count including all SCE, CPE and LFE.
+ */
+int CProgramConfig_GetElementTable( const CProgramConfig *pPce,
+                                    MP4_ELEMENT_ID  table[],
+                                    const INT elListSize );
+
+/**
+ * \brief       Initialize a given AudioSpecificConfig structure.
+ * \param pAsc  A pointer to an allocated CSAudioSpecificConfig struct.
+ * \return      void
+ */
+void AudioSpecificConfig_Init(CSAudioSpecificConfig *pAsc);
+
+/**
+ * \brief   Parse a AudioSpecificConfig from a given bitstream handle.
+ *
+ * \param pAsc                         A pointer to an allocated CSAudioSpecificConfig struct.
+ * \param hBs                          Bitstream handle.
+ * \param fExplicitBackwardCompatible  Do explicit backward compatibility parsing if set (flag).
+ * \param cb pointer to structure holding callback information
+ *
+ * \return  Total element count including all SCE, CPE and LFE.
+ */
+TRANSPORTDEC_ERROR AudioSpecificConfig_Parse(
+        CSAudioSpecificConfig *pAsc,
+        HANDLE_FDK_BITSTREAM hBs,
+        int fExplicitBackwardCompatible,
+        CSTpCallBacks *cb
+        );
+
+/* CELP stuff */
+enum {
+  MPE     = 0,
+  RPE     = 1,
+  fs8KHz  = 0,
+  fs16KHz = 1
+};
+
+/* Defintion of flags that can be passed to transportDecOpen() */
+#define TP_FLAG_MPEG4 1
+
+/* Capability flags */
+#define CAPF_TPDEC_ADIF        0x00001000  /**< Flag indicating support for ADIF transport format.        */
+#define CAPF_TPDEC_ADTS        0x00002000  /**< Flag indicating support for ADTS transport format.        */
+#define CAPF_TPDEC_LOAS        0x00004000  /**< Flag indicating support for LOAS transport format.        */
+#define CAPF_TPDEC_LATM        0x00008000  /**< Flag indicating support for LATM transport format.        */
+#define CAPF_TPDEC_RAWPACKETS  0x00010000  /**< Flag indicating support for raw packets transport format. */
+
+typedef struct TRANSPORTDEC *HANDLE_TRANSPORTDEC;
+
+
+/**
+ * \brief Configure Transport Decoder via a binary coded AudioSpecificConfig or StreamMuxConfig.
+ *        The previously requested configuration callback will be called as well. The buffer conf
+ *        must containt a SMC in case of LOAS/LATM transport format, and an ASC elseways.
+ *
+ * \param hTp     Handle of a transport decoder.
+ * \param conf    UCHAR buffer of the binary coded config (ASC or SMC).
+ * \param length  The length in bytes of the conf buffer.
+ *
+ * \return        Error code.
+ */
+TRANSPORTDEC_ERROR transportDec_OutOfBandConfig( const HANDLE_TRANSPORTDEC hTp,
+                                                 UCHAR              *conf,
+                                                 const UINT          length,
+                                                 const UINT          layer );
+
+/**
+ * \brief Open Transport medium for reading.
+ *
+ * \param transportDecFmt Format of the transport decoder medium to be accessed.
+ * \param flags           Transport decoder flags. Currently only TP_FLAG_MPEG4, which signals a
+ *                        MPEG4 capable decoder (relevant for ADTS only).
+ *
+ * \return   A pointer to a valid and allocated HANDLE_TRANSPORTDEC or a null pointer on failure.
+ */
+HANDLE_TRANSPORTDEC transportDec_Open( TRANSPORT_TYPE transportDecFmt,
+                                       const UINT flags );
+
+/**
+ * \brief                Register configuration change callback.
+ * \param hTp            Handle of transport decoder.
+ * \param cbUpdateConfig Pointer to a callback function to handle audio config changes.
+ * \param user_data      void pointer for user data passed to the callback as first parameter.
+ * \return               0 on success.
+ */
+int transportDec_RegisterAscCallback (
+        HANDLE_TRANSPORTDEC hTp,
+        const cbUpdateConfig_t cbUpdateConfig,
+        void* user_data );
+
+/**
+ * \brief                Register SSC parser callback.
+ * \param hTp            Handle of transport decoder.
+ * \param cbUpdateConfig Pointer to a callback function to handle SSC parsing.
+ * \param user_data      void pointer for user data passed to the callback as first parameter.
+ * \return               0 on success.
+ */
+int transportDec_RegisterSscCallback (
+        HANDLE_TRANSPORTDEC hTp,
+        const cbSsc_t cbSscParse,
+        void* user_data );
+
+/**
+ * \brief                Register SBR header parser callback.
+ * \param hTp            Handle of transport decoder.
+ * \param cbUpdateConfig Pointer to a callback function to handle SBR header parsing.
+ * \param user_data      void pointer for user data passed to the callback as first parameter.
+ * \return               0 on success. 
+ */
+int transportDec_RegisterSbrCallback( HANDLE_TRANSPORTDEC hTpDec, const cbSbr_t cbSbr, void* user_data);
+
+/**
+ * \brief Fill internal input buffer with bitstream data from the external input buffer.
+ *  The function only copies such data as long as the decoder-internal input buffer is not full.
+ *  So it grabs whatever it can from pBuffer and returns information (bytesValid) so that at a
+ *  subsequent call of %transportDec_FillData(), the right position in pBuffer can be determined to
+ *  grab the next data.
+ *
+ * \param hTp         Handle of transportDec.
+ * \param pBuffer     Pointer to external input buffer.
+ * \param bufferSize  Size of external input buffer. This argument is required because decoder-internally
+ *                    we need the information to calculate the offset to pBuffer, where the next
+ *                    available data is, which is then fed into the decoder-internal buffer (as much
+ *                    as possible). Our example framework implementation fills the buffer at pBuffer
+ *                    again, once it contains no available valid bytes anymore (meaning bytesValid equal 0).
+ * \param bytesValid  Number of bitstream bytes in the external bitstream buffer that have not yet been
+ *                    copied into the decoder's internal bitstream buffer by calling this function.
+ *                    The value is updated according to the amount of newly copied bytes.
+ * \param layer       The layer the bitstream belongs to.
+ * \return            Error code.
+ */
+TRANSPORTDEC_ERROR transportDec_FillData(
+        const HANDLE_TRANSPORTDEC  hTp,
+        UCHAR                     *pBuffer,
+        const UINT                 bufferSize,
+        UINT                      *pBytesValid,
+        const INT                  layer );
+
+/**
+ * \brief      Get transportDec bitstream handle.
+ * \param hTp  Pointer to a transport decoder handle.
+ * \return     HANDLE_FDK_BITSTREAM bitstream handle.
+ */
+HANDLE_FDK_BITSTREAM transportDec_GetBitstream ( const HANDLE_TRANSPORTDEC hTp, const UINT layer );
+
+/**
+ * \brief      Get transport format.
+ * \param hTp  Pointer to a transport decoder handle.
+ * \return     The transport format.
+ */
+TRANSPORT_TYPE transportDec_GetFormat ( const HANDLE_TRANSPORTDEC hTp );
+
+/**
+ * \brief Get the current buffer fullness value.
+ *
+ * \param hTp     Handle of a transport decoder.
+ *
+ * \return        Buffer fullness
+ */
+INT transportDec_GetBufferFullness( const HANDLE_TRANSPORTDEC hTp );
+
+/**
+ * \brief       Close and deallocate transportDec.
+ * \param phTp  Pointer to a previously allocated transport decoder handle.
+ * \return      void
+ */
+void transportDec_Close ( HANDLE_TRANSPORTDEC *phTp );
+
+/**
+ * \brief         Read one access unit from the transportDec medium.
+ * \param hTp     Handle of transportDec.
+ * \param length  On return, this value is overwritten with the actual access unit length in bits.
+ *                Set to -1 if length is unknown.
+ * \return        Error code.
+ */
+TRANSPORTDEC_ERROR transportDec_ReadAccessUnit ( const HANDLE_TRANSPORTDEC hTp, const UINT layer );
+
+/**
+ * \brief Get the remaining amount of bits of the current access unit. The result
+ *        can be below zero, meaning that too many bits have been read.
+ * \param hTp     Handle of transportDec.
+ * \return amount of remaining bits.
+ */
+INT transportDec_GetAuBitsRemaining( const HANDLE_TRANSPORTDEC hTp, const UINT layer );
+
+/**
+ * \brief Get the total amount of bits of the current access unit.
+ * \param hTp     Handle of transportDec.
+ * \return amount of total bits.
+ */
+INT transportDec_GetAuBitsTotal( const HANDLE_TRANSPORTDEC hTp, const UINT layer );
+
+/**
+ * \brief      This function is required to be called when the decoder has finished parsing
+ *             one Access Unit for bitstream housekeeping.
+ * \param hTp  Transport Handle.
+ * \return     Error code.
+ */
+TRANSPORTDEC_ERROR transportDec_EndAccessUnit ( const HANDLE_TRANSPORTDEC hTp );
+
+/**
+ * \brief      Obtain the amount of missing access units if applicable in case of 
+ *             a bit stream synchronization error. Each time transportDec_ReadAccessUnit()
+ *             returns TRANSPORTDEC_SYNC_ERROR this function can be called to retrieve an estimate
+ *             of the amount of missing access units. This works only in case of constant average
+ *             bit rate (has to be known) and if the parameter TPDEC_PARAM_SET_BITRATE has been set
+ *             accordingly.
+ * \param hTp  Transport Handle.
+ * \param pNAccessUnits pointer to a memory location where the estimated lost frame count will be stored into.      
+ * \return     Error code.
+ */
+TRANSPORTDEC_ERROR transportDec_GetMissingAccessUnitCount ( INT *pNAccessUnits, HANDLE_TRANSPORTDEC hTp );
+
+
+/**
+ * \brief        Set a given setting.
+ * \param hTp    Transport Handle.
+ * \param param  Identifier of the parameter to be changed.
+ * \param value  Value for the parameter to be changed.
+ * \return       Error code.
+ */
+TRANSPORTDEC_ERROR transportDec_SetParam ( const HANDLE_TRANSPORTDEC hTp,
+                                           const TPDEC_PARAM         param,
+                                           const INT                 value );
+
+/**
+ * \brief        Get number of subframes (for LATM or ADTS)
+ * \param hTp    Transport Handle.
+ * \return       Number of ADTS/LATM subframes (return 1 for all other transport types).
+ */
+UINT transportDec_GetNrOfSubFrames(HANDLE_TRANSPORTDEC hTp);
+
+
+/**
+ * \brief       Get info structure of transport decoder library.
+ * \param info  A pointer to an allocated LIB_INFO struct.
+ * \return      Error code.
+ */
+TRANSPORTDEC_ERROR transportDec_GetLibInfo( LIB_INFO *info );
+
+/* ADTS CRC support */
+
+/**
+ * \brief        Set current bitstream position as start of a new data region.
+ * \param hTp    Transport handle.
+ * \param mBits  Size in bits of the data region. Set to 0 if it should not be of a fixed size.
+ * \return       Data region ID, which should be used when calling transportDec_CrcEndReg().
+ */
+int transportDec_CrcStartReg ( const HANDLE_TRANSPORTDEC hTp,
+                               const INT mBits );
+
+/**
+ * \brief        Set end of data region.
+ * \param hTp    Transport handle.
+ * \param reg    Data region ID, opbtained from transportDec_CrcStartReg().
+ * \return       void
+ */
+void transportDec_CrcEndReg ( const HANDLE_TRANSPORTDEC hTp,
+                              const INT reg );
+
+/**
+ * \brief      Calculate ADTS crc and check if it is correct. The ADTS checksum is held internally.
+ * \param hTp  Transport handle.
+ * \return     Return TRANSPORTDEC_OK if the CRC is ok, or error if CRC is not correct.
+ */
+TRANSPORTDEC_ERROR transportDec_CrcCheck ( const HANDLE_TRANSPORTDEC hTp );
+
+
+#endif /* #ifndef __TPDEC_LIB_H__ */
diff --git a/libMpegTPDec/src/mpegFileFormat.h b/libMpegTPDec/src/mpegFileFormat.h
new file mode 100644
index 0000000..7ddaa0d
--- /dev/null
+++ b/libMpegTPDec/src/mpegFileFormat.h
@@ -0,0 +1,114 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/***************************  Fraunhofer IIS FDK Tools  ***********************
+
+   Author(s):   Oliver Moser
+   Description: bitstream format detection routines
+
+******************************************************************************/
+
+#if !defined(__BITSTREAM_FORMAT_H__)
+#define __BITSTREAM_FORMAT_H__
+
+#include "machine_type.h"
+#include "FDK_audio.h"
+
+/**
+ * \brief Try to find out the format of a file, given the few first bytes.
+ * \param fileData pointer to a buffer holding the first bytes of a file.
+ * \param pAu pointer to UCHAR*, returns the address of the first AU found or NULL.
+ * \param length pointer to the length of the buffer fileData. Return length of first AU.
+ * \return the detected file format, or FF_UNKNOWN in case of failure.
+ */
+FILE_FORMAT GetFileFormat(UCHAR *fileData, UCHAR **pAu, UINT *length);
+
+/**
+ * \brief Try to find out the transport type contained in a given file.
+ * \param filename name of the file to be analysed.
+ * \param fileFormat pointer to a variable where the detected file format is stored into.
+ * \return the detected transport type or TT_UNKNOWN in case of failure.
+ */
+TRANSPORT_TYPE GetTransportType(const char* filename, FILE_FORMAT *fileFormat);
+
+#endif
diff --git a/libMpegTPDec/src/tpdec_adif.cpp b/libMpegTPDec/src/tpdec_adif.cpp
new file mode 100644
index 0000000..b6d6087
--- /dev/null
+++ b/libMpegTPDec/src/tpdec_adif.cpp
@@ -0,0 +1,155 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/*****************************  MPEG-4 AAC Decoder  **************************
+
+   Author(s):   Josef Hoepfl
+   Description: ADIF reader
+
+******************************************************************************/
+
+#include "tpdec_adif.h"
+
+
+#include "FDK_bitstream.h"
+#include "genericStds.h"
+
+TRANSPORTDEC_ERROR adifRead_DecodeHeader(
+        CAdifHeader          *pAdifHeader,
+        CProgramConfig       *pPce,
+        HANDLE_FDK_BITSTREAM  bs
+        )
+{
+  int i;
+  TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK;
+  UINT startAnchor = FDKgetValidBits(bs);
+
+  if ((INT)startAnchor < MIN_ADIF_HEADERLENGTH) {
+    return (TRANSPORTDEC_NOT_ENOUGH_BITS);
+  }
+
+  if (FDKreadBits(bs,8) != 'A') {
+    return (TRANSPORTDEC_SYNC_ERROR);
+  }
+  if (FDKreadBits(bs,8) != 'D') {
+    return (TRANSPORTDEC_SYNC_ERROR);
+  }
+  if (FDKreadBits(bs,8) != 'I') {
+    return (TRANSPORTDEC_SYNC_ERROR);
+  }
+  if (FDKreadBits(bs,8) != 'F') {
+    return (TRANSPORTDEC_SYNC_ERROR);
+  }
+
+  if ( (pAdifHeader->CopyrightIdPresent = FDKreadBits(bs,1)) != 0 )
+    FDKpushBiDirectional(bs,72);  /* CopyrightId */
+
+
+  pAdifHeader->OriginalCopy = FDKreadBits(bs,1);
+  pAdifHeader->Home = FDKreadBits(bs,1);
+  pAdifHeader->BitstreamType = FDKreadBits(bs,1);
+
+  /* pAdifHeader->BitRate = FDKreadBits(bs, 23); */
+  pAdifHeader->BitRate = FDKreadBits(bs,16);
+  pAdifHeader->BitRate <<= 7;
+  pAdifHeader->BitRate |= FDKreadBits(bs,7);
+
+  pAdifHeader->NumProgramConfigElements = FDKreadBits(bs,4) + 1;
+
+  if (pAdifHeader->BitstreamType == 0) {
+    FDKpushBiDirectional(bs,20);  /* adif_buffer_fullness */
+  }
+
+  /* Parse all PCEs but keep only one */
+  for (i=0; i < pAdifHeader->NumProgramConfigElements; i++)
+  {
+    CProgramConfig_Read(pPce, bs, startAnchor);
+  }
+
+  FDKbyteAlign(bs, startAnchor);
+
+  return (ErrorStatus);
+}
+
+
+
diff --git a/libMpegTPDec/src/tpdec_adif.h b/libMpegTPDec/src/tpdec_adif.h
new file mode 100644
index 0000000..7b9b071
--- /dev/null
+++ b/libMpegTPDec/src/tpdec_adif.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
+----------------------------------------------------------------------------------------------------------- */
+
+/*****************************  MPEG-4 AAC Decoder  **************************
+
+   Author(s):   Josef Hoepfl
+   Description: ADIF reader
+
+******************************************************************************/
+
+#ifndef TPDEC_ADIF_H
+#define TPDEC_ADIF_H
+
+#include "tpdec_lib.h"
+
+#define MIN_ADIF_HEADERLENGTH  63  /* in bits */
+
+typedef struct
+{
+  INT NumProgramConfigElements;
+  UINT BitRate;
+  UCHAR CopyrightIdPresent;
+  UCHAR OriginalCopy;
+  UCHAR Home;
+  UCHAR BitstreamType;
+} CAdifHeader;
+
+/**
+ * \brief Parse a ADIF header at the given bitstream and store the parsed data into a given CAdifHeader
+ *        and CProgramConfig struct
+ *
+ * \param pAdifHeader pointer to a CAdifHeader structure to hold the parsed ADIF header data.
+ * \param pPce pointer to a CProgramConfig structure where the last PCE will remain.
+ *
+ * \return TRANSPORTDEC_ERROR error code
+ */
+TRANSPORTDEC_ERROR adifRead_DecodeHeader(
+        CAdifHeader         *pAdifHeader,
+        CProgramConfig      *pPce,
+        HANDLE_FDK_BITSTREAM bs
+        );
+
+#endif /* TPDEC_ADIF_H */
diff --git a/libMpegTPDec/src/tpdec_adts.cpp b/libMpegTPDec/src/tpdec_adts.cpp
new file mode 100644
index 0000000..92e2e59
--- /dev/null
+++ b/libMpegTPDec/src/tpdec_adts.cpp
@@ -0,0 +1,388 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/*****************************  MPEG-4 AAC Decoder  **************************
+
+   Author(s):   Josef Hoepfl
+   Description: ADTS interface
+
+******************************************************************************/
+
+#include "tpdec_adts.h"
+
+
+#include "FDK_bitstream.h"
+
+
+
+void adtsRead_CrcInit(HANDLE_ADTS pAdts) /*!< pointer to adts crc info stucture */
+{
+  FDKcrcInit(&pAdts->crcInfo, 0x8005, 0xFFFF, 16);
+}
+
+int adtsRead_CrcStartReg(
+                     HANDLE_ADTS pAdts,          /*!< pointer to adts stucture */
+                     HANDLE_FDK_BITSTREAM hBs,   /*!< handle to current bit buffer structure */
+                     int mBits                   /*!< number of bits in crc region */
+                   )
+{
+  if (pAdts->bs.protection_absent) {
+    return 0;
+  }
+
+  return ( FDKcrcStartReg(&pAdts->crcInfo, hBs, mBits) );
+
+}
+
+void adtsRead_CrcEndReg(
+                    HANDLE_ADTS pAdts, /*!< pointer to adts crc info stucture */
+                    HANDLE_FDK_BITSTREAM hBs,   /*!< handle to current bit buffer structure */
+                    int reg                    /*!< crc region */
+                  )
+{
+  if (pAdts->bs.protection_absent == 0)
+  {
+    FDKcrcEndReg(&pAdts->crcInfo, hBs, reg);
+  }
+}
+
+TRANSPORTDEC_ERROR adtsRead_CrcCheck( HANDLE_ADTS pAdts )
+{
+  TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK;
+  USHORT crc;
+
+  if (pAdts->bs.protection_absent)
+    return TRANSPORTDEC_OK;
+
+  crc = FDKcrcGetCRC(&pAdts->crcInfo);
+  if (crc != pAdts->crcReadValue)
+  {
+    return (TRANSPORTDEC_CRC_ERROR);
+  }
+
+  return (ErrorStatus);
+}
+
+
+
+#define Adts_Length_SyncWord                     12
+#define Adts_Length_Id                            1
+#define Adts_Length_Layer                         2
+#define Adts_Length_ProtectionAbsent              1
+#define Adts_Length_Profile                       2
+#define Adts_Length_SamplingFrequencyIndex        4
+#define Adts_Length_PrivateBit                    1
+#define Adts_Length_ChannelConfiguration          3
+#define Adts_Length_OriginalCopy                  1
+#define Adts_Length_Home                          1
+#define Adts_Length_CopyrightIdentificationBit    1
+#define Adts_Length_CopyrightIdentificationStart  1
+#define Adts_Length_FrameLength                  13
+#define Adts_Length_BufferFullness               11
+#define Adts_Length_NumberOfRawDataBlocksInFrame  2
+#define Adts_Length_CrcCheck                     16
+
+TRANSPORTDEC_ERROR adtsRead_DecodeHeader(
+        HANDLE_ADTS           pAdts,
+        CSAudioSpecificConfig *pAsc,
+        HANDLE_FDK_BITSTREAM  hBs,
+        const INT             ignoreBufferFullness
+        )
+{
+  INT crcReg;
+
+  INT valBits;
+  INT cmp_buffer_fullness;
+  int i, adtsHeaderLength;
+
+  STRUCT_ADTS_BS bs;
+
+#ifdef TP_PCE_ENABLE
+  CProgramConfig oldPce;
+  /* Store the old PCE temporarily. Maybe we'll need it later if we
+     have channelConfig=0 and no PCE in this frame. */
+  FDKmemcpy(&oldPce, &pAsc->m_progrConfigElement, sizeof(CProgramConfig));
+#endif
+
+  valBits = FDKgetValidBits(hBs);
+
+  /* adts_fixed_header */
+  bs.mpeg_id           = FDKreadBits(hBs, Adts_Length_Id);
+  bs.layer             = FDKreadBits(hBs, Adts_Length_Layer);
+  bs.protection_absent = FDKreadBits(hBs, Adts_Length_ProtectionAbsent);
+  bs.profile           = FDKreadBits(hBs, Adts_Length_Profile);
+  bs.sample_freq_index = FDKreadBits(hBs, Adts_Length_SamplingFrequencyIndex);
+  bs.private_bit       = FDKreadBits(hBs, Adts_Length_PrivateBit);
+  bs.channel_config    = FDKreadBits(hBs, Adts_Length_ChannelConfiguration);
+  bs.original          = FDKreadBits(hBs, Adts_Length_OriginalCopy);
+  bs.home              = FDKreadBits(hBs, Adts_Length_Home);
+
+  /* adts_variable_header */
+  bs.copyright_id    = FDKreadBits(hBs, Adts_Length_CopyrightIdentificationBit);
+  bs.copyright_start = FDKreadBits(hBs, Adts_Length_CopyrightIdentificationStart);
+  bs.frame_length    = FDKreadBits(hBs, Adts_Length_FrameLength);
+  bs.adts_fullness   = FDKreadBits(hBs, Adts_Length_BufferFullness);
+  bs.num_raw_blocks  = FDKreadBits(hBs, Adts_Length_NumberOfRawDataBlocksInFrame);
+  bs.num_pce_bits    = 0;
+
+  adtsHeaderLength = ADTS_HEADERLENGTH;
+
+  if (!bs.protection_absent) {
+    FDKcrcReset(&pAdts->crcInfo);
+    FDKpushBack(hBs, 56);   /* complete fixed and variable header! */
+    crcReg = FDKcrcStartReg(&pAdts->crcInfo, hBs, 0);
+    FDKpushFor(hBs, 56);
+  }
+
+  if (! bs.protection_absent && bs.num_raw_blocks>0) {
+    for (i=0; i<bs.num_raw_blocks; i++) {
+      pAdts->rawDataBlockDist[i] = (USHORT)FDKreadBits(hBs, 16);
+      adtsHeaderLength += 16;
+    }
+    /* Change raw data blocks to delta values */
+    pAdts->rawDataBlockDist[bs.num_raw_blocks] = bs.frame_length - 7 - bs.num_raw_blocks*2 - 2 ;
+    for (i=bs.num_raw_blocks; i>0; i--) {
+      pAdts->rawDataBlockDist[i] -= pAdts->rawDataBlockDist[i-1];
+    }
+  }
+
+  /* adts_error_check */
+  if (!bs.protection_absent)
+  {
+    USHORT crc_check;
+
+    FDKcrcEndReg(&pAdts->crcInfo, hBs, crcReg);
+    crc_check = FDKreadBits(hBs, Adts_Length_CrcCheck);
+    adtsHeaderLength += Adts_Length_CrcCheck;
+
+    pAdts->crcReadValue = crc_check;
+    /* Check header CRC in case of multiple raw data blocks */
+    if (bs.num_raw_blocks > 0) {
+      if (pAdts->crcReadValue != FDKcrcGetCRC(&pAdts->crcInfo)) {
+        return TRANSPORTDEC_CRC_ERROR;
+      }
+      /* Reset CRC for the upcoming raw_data_block() */
+      FDKcrcReset(&pAdts->crcInfo);
+    }
+  }
+
+
+  /* check if valid header */
+  if (
+      (bs.layer != 0) ||                          // we only support MPEG ADTS
+      (bs.sample_freq_index >= 13)                // we only support 96kHz - 7350kHz
+      ) {
+    FDKpushFor(hBs, bs.frame_length * 8);   // try again one frame later
+    return TRANSPORTDEC_UNSUPPORTED_FORMAT;
+  }
+
+  /* special treatment of id-bit */
+  if ( (bs.mpeg_id == 0) && (pAdts->decoderCanDoMpeg4 == 0) )
+  {
+    /* MPEG-2 decoder cannot play MPEG-4 bitstreams */
+
+
+    FDKpushFor(hBs, bs.frame_length * 8);  // try again one frame later
+    return TRANSPORTDEC_UNSUPPORTED_FORMAT;
+  }
+
+  if (!ignoreBufferFullness)
+  {
+    cmp_buffer_fullness = bs.frame_length*8 + bs.adts_fullness*32*getNumberOfEffectiveChannels(bs.channel_config);
+
+
+    /* Evaluate buffer fullness */
+    if (bs.adts_fullness != 0x7FF)
+    {
+      if (pAdts->BufferFullnesStartFlag)
+      {
+        if ( valBits < cmp_buffer_fullness )
+        {
+          /* Condition for start of decoding is not fulfilled */
+
+          /* The current frame will not be decoded */
+          FDKpushBack(hBs, adtsHeaderLength);
+
+          if ( (cmp_buffer_fullness+adtsHeaderLength) > ((TRANSPORTDEC_INBUF_SIZE<<3)-7) ) {
+            return TRANSPORTDEC_SYNC_ERROR;
+          } else {
+            return TRANSPORTDEC_NOT_ENOUGH_BITS;
+          }
+        }
+        else
+        {
+          pAdts->BufferFullnesStartFlag = 0;
+        }
+      }
+    }
+  }
+
+
+  /* Get info from ADTS header */
+  AudioSpecificConfig_Init(pAsc);
+  pAsc->m_aot                    = (AUDIO_OBJECT_TYPE)(bs.profile + 1);
+  pAsc->m_samplingFrequencyIndex = bs.sample_freq_index;
+  pAsc->m_samplingFrequency      = SamplingRateTable[bs.sample_freq_index];
+  pAsc->m_channelConfiguration   = bs.channel_config;
+  pAsc->m_samplesPerFrame        = 1024;
+
+#ifdef TP_PCE_ENABLE
+  if (bs.channel_config == 0)
+  {
+    int pceBits = 0;
+    UINT alignAnchor = FDKgetValidBits(hBs);
+    
+    if (FDKreadBits(hBs,3) == ID_PCE) {
+      /* Got luck! Parse the PCE */
+      int crcReg;
+      crcReg = adtsRead_CrcStartReg(pAdts, hBs, 0);
+
+      CProgramConfig_Read(&pAsc->m_progrConfigElement, hBs, alignAnchor);
+
+      adtsRead_CrcEndReg(pAdts, hBs, crcReg);
+      pceBits = alignAnchor - FDKgetValidBits(hBs);
+      /* store the number of PCE bits */
+      bs.num_pce_bits = pceBits;
+    }
+    else {
+      /* No PCE in this frame! Push back the ID tag bits. */
+      FDKpushBack(hBs,3);
+
+      /* Encoders do not have to write a PCE in each frame.
+         So if we already have a valid PCE we have to use it. */
+      if (  oldPce.isValid
+        && (bs.sample_freq_index == pAdts->bs.sample_freq_index)  /* we could compare the complete fixed header (bytes) here! */
+        && (bs.channel_config    == pAdts->bs.channel_config) /* == 0 */
+        && (bs.mpeg_id           == pAdts->bs.mpeg_id) )
+      { /* Restore previous PCE which is still valid */
+        FDKmemcpy(&pAsc->m_progrConfigElement, &oldPce, sizeof(CProgramConfig));
+      }
+      else if (bs.mpeg_id == 0) {
+        /* If not it seems that we have a implicit channel configuration.
+           This mode is not allowed in the context of ISO/IEC 14496-3.
+           Skip this frame and try the next one. */
+        FDKpushFor(hBs, (bs.frame_length<<3) - adtsHeaderLength - 3);
+        return TRANSPORTDEC_UNSUPPORTED_FORMAT;
+      }
+      /* else {
+         ISO/IEC 13818-7 implicit channel mapping is allowed.
+         So just open the box of chocolates to see what we got.
+      } */
+    }
+  }
+#endif /* TP_PCE_ENABLE */
+
+  /* Copy bit stream data struct to persistent memory now, once we passed all sanity checks above. */
+  FDKmemcpy(&pAdts->bs, &bs, sizeof(STRUCT_ADTS_BS));
+
+  return TRANSPORTDEC_OK;
+}
+
+int adtsRead_GetRawDataBlockLength(
+        HANDLE_ADTS pAdts,
+        INT         blockNum
+        )
+{
+  int length;
+
+  if (pAdts->bs.num_raw_blocks == 0) {
+    length = (pAdts->bs.frame_length - 7) << 3;    /* aac_frame_length subtracted by the header size (7 bytes). */
+    if (pAdts->bs.protection_absent == 0)
+      length -= 16;                                                                   /* substract 16 bit CRC */
+  } else {
+    if (pAdts->bs.protection_absent) {
+      length = -1; /* raw data block length is unknown */
+    } else {
+      if (blockNum < 0 || blockNum > 3) {
+        return TRANSPORTDEC_INVALID_PARAMETER;
+      }
+      length = (pAdts->rawDataBlockDist[blockNum] << 3) - 16;
+    }
+  }
+  if (blockNum == 0) {
+    length -= pAdts->bs.num_pce_bits;
+  }
+  return length;
+}
+
+
diff --git a/libMpegTPDec/src/tpdec_adts.h b/libMpegTPDec/src/tpdec_adts.h
new file mode 100644
index 0000000..def6aa0
--- /dev/null
+++ b/libMpegTPDec/src/tpdec_adts.h
@@ -0,0 +1,236 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/*****************************  MPEG-4 AAC Decoder  **************************
+
+   Author(s):   Josef Hoepfl
+   Description: ADTS interface
+
+******************************************************************************/
+
+#ifndef TPDEC_ADTS_H
+#define TPDEC_ADTS_H
+
+#include "tpdec_lib.h"
+
+
+#define ADTS_SYNCWORD               ( 0xfff )
+#define ADTS_SYNCLENGTH             ( 12 )     /* in bits */
+#define ADTS_HEADERLENGTH           ( 56 )     /* minimum header size in bits */
+#define ADTS_FIXED_HEADERLENGTH     ( 28 )     /* in bits */
+#define ADTS_VARIABLE_HEADERLENGTH  ( ADTS_HEADERLENGTH - ADTS_FIXED_HEADERLENGTH )
+
+#ifdef CHECK_TWO_SYNCS
+ #define ADTS_MIN_TP_BUF_SIZE       ( 8191 + 2 )
+#else
+ #define ADTS_MIN_TP_BUF_SIZE       ( 8191 )
+#endif
+
+#include "FDK_crc.h"
+
+typedef struct {
+  /* ADTS header fields */
+  UCHAR mpeg_id;
+  UCHAR layer;
+  UCHAR protection_absent;
+  UCHAR profile;
+  UCHAR sample_freq_index;
+  UCHAR private_bit;
+  UCHAR channel_config;
+  UCHAR original;
+  UCHAR home;
+  UCHAR copyright_id;
+  UCHAR copyright_start;
+  USHORT frame_length;
+  USHORT adts_fullness;
+  UCHAR num_raw_blocks;
+  UCHAR num_pce_bits;
+} STRUCT_ADTS_BS;
+
+struct STRUCT_ADTS {
+
+  STRUCT_ADTS_BS bs;
+
+  UCHAR decoderCanDoMpeg4;
+  UCHAR BufferFullnesStartFlag;
+
+  FDK_CRCINFO crcInfo;             /* CRC state info */
+  USHORT      crcReadValue;        /* CRC value read from bitstream data */
+  USHORT      rawDataBlockDist[4]; /* distance between each raw data block. Not the same as found in the bitstream */
+} ;
+
+typedef struct STRUCT_ADTS *HANDLE_ADTS;
+
+/*!
+  \brief Initialize ADTS CRC
+
+  The function initialzes the crc buffer and the crc lookup table.
+
+  \return  none
+*/
+void adtsRead_CrcInit( HANDLE_ADTS pAdts );
+
+/**
+ * \brief Starts CRC region with a maximum number of bits
+ *        If mBits is positive zero padding will be used for CRC calculation, if there
+ *        are less than mBits bits available.
+ *        If mBits is negative no zero padding is done.
+ *        If mBits is zero the memory for the buffer is allocated dynamically, the
+ *        number of bits is not limited.
+ *
+ * \param pAdts ADTS data handle
+ * \param hBs bitstream handle, on which the CRC region referes to
+ * \param mBits max number of bits in crc region to be considered
+ *
+ * \return  ID for the created region, -1 in case of an error
+ */
+int adtsRead_CrcStartReg(
+        HANDLE_ADTS pAdts,
+        HANDLE_FDK_BITSTREAM hBs,
+        int mBits
+        );
+
+/**
+ * \brief Ends CRC region identified by reg
+ *
+ * \param pAdts ADTS data handle
+ * \param hBs bitstream handle, on which the CRC region referes to
+ * \param reg CRC regions ID returned by adtsRead_CrcStartReg()
+ *
+ * \return  none
+ */
+void adtsRead_CrcEndReg(
+        HANDLE_ADTS pAdts,
+        HANDLE_FDK_BITSTREAM hBs,
+        int reg
+        );
+
+/**
+ * \brief Check CRC
+ *
+ * Checks if the currently calculated CRC matches the CRC field read from the bitstream
+ * Deletes all CRC regions.
+ *
+ * \param pAdts ADTS data handle
+ *
+ * \return Returns 0 if they are identical otherwise 1
+ */
+TRANSPORTDEC_ERROR adtsRead_CrcCheck( HANDLE_ADTS pAdts );
+
+
+/**
+ * \brief Check if we have a valid ADTS frame at the current bitbuffer position
+ *
+ * This function assumes enough bits in buffer for the current frame.
+ * It reads out the header bits to prepare the bitbuffer for the decode loop.
+ * In case the header bits show an invalid bitstream/frame, the whole frame is skipped.
+ *
+ * \param pAdts ADTS data handle which is filled with parsed ADTS header data
+ * \param bs handle of bitstream from whom the ADTS header is read
+ *
+ * \return  error status
+ */
+TRANSPORTDEC_ERROR adtsRead_DecodeHeader(
+        HANDLE_ADTS           pAdts,
+        CSAudioSpecificConfig *pAsc,
+        HANDLE_FDK_BITSTREAM  bs,
+        const INT             ignoreBufferFullness
+        );
+
+/**
+ * \brief Get the raw data block length of the given block number.
+ *
+ * \param pAdts ADTS data handle
+ * \param blockNum current raw data block index
+ * \param pLength pointer to an INT where the length of the given raw data block is stored into
+ *                the returned value might be -1, in which case the raw data block length is unknown.
+ *
+ * \return  error status
+ */
+int adtsRead_GetRawDataBlockLength(
+        HANDLE_ADTS pAdts,
+        INT         blockNum
+        );
+
+
+#endif /* TPDEC_ADTS_H */
diff --git a/libMpegTPDec/src/tpdec_asc.cpp b/libMpegTPDec/src/tpdec_asc.cpp
new file mode 100644
index 0000000..d6e1a0b
--- /dev/null
+++ b/libMpegTPDec/src/tpdec_asc.cpp
@@ -0,0 +1,846 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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):   Daniel Homm
+   Description:
+
+******************************************************************************/
+
+#include "tpdec_lib.h"
+#include "tp_data.h"
+
+
+void CProgramConfig_Reset(CProgramConfig *pPce)
+{
+  pPce->elCounter = 0;
+}
+
+void CProgramConfig_Init(CProgramConfig *pPce)
+{
+  FDKmemclear(pPce, sizeof(CProgramConfig));
+#ifdef TP_PCE_ENABLE
+  pPce->SamplingFrequencyIndex = 0xf;
+#endif
+}
+
+int  CProgramConfig_IsValid ( const CProgramConfig *pPce )
+{
+  return ( (pPce->isValid) ? 1 : 0);
+}
+
+#ifdef TP_PCE_ENABLE
+void CProgramConfig_Read(
+                          CProgramConfig *pPce,
+                          HANDLE_FDK_BITSTREAM bs,
+                          UINT alignmentAnchor
+                        )
+{
+  int i;
+
+  pPce->NumEffectiveChannels = 0;
+  pPce->NumChannels = 0;
+  pPce->ElementInstanceTag = (UCHAR) FDKreadBits(bs,4);
+  pPce->Profile = (UCHAR) FDKreadBits(bs,2);
+  pPce->SamplingFrequencyIndex = (UCHAR) FDKreadBits(bs,4);
+  pPce->NumFrontChannelElements = (UCHAR) FDKreadBits(bs,4);
+  pPce->NumSideChannelElements = (UCHAR) FDKreadBits(bs,4);
+  pPce->NumBackChannelElements = (UCHAR) FDKreadBits(bs,4);
+  pPce->NumLfeChannelElements = (UCHAR) FDKreadBits(bs,2);
+  pPce->NumAssocDataElements = (UCHAR) FDKreadBits(bs,3);
+  pPce->NumValidCcElements = (UCHAR) FDKreadBits(bs,4);
+
+  if ((pPce->MonoMixdownPresent = (UCHAR) FDKreadBits(bs,1)) != 0)
+  {
+    pPce->MonoMixdownElementNumber = (UCHAR) FDKreadBits(bs,4);
+  }
+
+  if ((pPce->StereoMixdownPresent = (UCHAR) FDKreadBits(bs,1)) != 0)
+  {
+    pPce->StereoMixdownElementNumber = (UCHAR) FDKreadBits(bs,4);
+  }
+
+  if ((pPce->MatrixMixdownIndexPresent = (UCHAR) FDKreadBits(bs,1)) != 0)
+  {
+    pPce->MatrixMixdownIndex = (UCHAR) FDKreadBits(bs,2);
+    pPce->PseudoSurroundEnable = (UCHAR) FDKreadBits(bs,1);
+  }
+
+  for (i=0; i < pPce->NumFrontChannelElements; i++)
+  {
+    pPce->FrontElementIsCpe[i] = (UCHAR) FDKreadBits(bs,1);
+    pPce->FrontElementTagSelect[i] = (UCHAR) FDKreadBits(bs,4);
+    pPce->NumChannels += pPce->FrontElementIsCpe[i] ? 2 : 1;
+  }
+
+  for (i=0; i < pPce->NumSideChannelElements; i++)
+  {
+    pPce->SideElementIsCpe[i] = (UCHAR) FDKreadBits(bs,1);
+    pPce->SideElementTagSelect[i] = (UCHAR) FDKreadBits(bs,4);
+    pPce->NumChannels += pPce->SideElementIsCpe[i] ? 2 : 1;
+  }
+
+  for (i=0; i < pPce->NumBackChannelElements; i++)
+  {
+    pPce->BackElementIsCpe[i] = (UCHAR) FDKreadBits(bs,1);
+    pPce->BackElementTagSelect[i] = (UCHAR) FDKreadBits(bs,4);
+    pPce->NumChannels += pPce->BackElementIsCpe[i] ? 2 : 1;
+  }
+
+  pPce->NumEffectiveChannels = pPce->NumChannels;
+
+  for (i=0; i < pPce->NumLfeChannelElements; i++)
+  {
+    pPce->LfeElementTagSelect[i] = (UCHAR) FDKreadBits(bs,4);
+    pPce->NumChannels += 1;
+  }
+
+  for (i=0; i < pPce->NumAssocDataElements; i++)
+  {
+    pPce->AssocDataElementTagSelect[i] = (UCHAR) FDKreadBits(bs,4);
+  }
+
+  for (i=0; i < pPce->NumValidCcElements; i++)
+  {
+    pPce->CcElementIsIndSw[i] = (UCHAR) FDKreadBits(bs,1);
+    pPce->ValidCcElementTagSelect[i] = (UCHAR) FDKreadBits(bs,4);
+  }
+
+  FDKbyteAlign(bs, alignmentAnchor);
+
+  pPce->CommentFieldBytes = (UCHAR) FDKreadBits(bs,8);
+
+  for (i=0; i < pPce->CommentFieldBytes; i++)
+  {
+    UCHAR text;
+
+    text = (UCHAR)FDKreadBits(bs,8);
+
+    if (i < PC_COMMENTLENGTH)
+    {
+      pPce->Comment[i] = text;
+    }
+  }
+
+  pPce->isValid = 1;
+}
+#endif /* TP_PCE_ENABLE */
+
+/**
+ * \brief get implicit audio channel type for given channelConfig and MPEG ordered channel index
+ * \param channelConfig MPEG channelConfiguration from 1 upto 7
+ * \param index MPEG channel order index
+ * \return audio channel type.
+ */
+void getImplicitAudioChannelTypeAndIndex(
+        AUDIO_CHANNEL_TYPE *chType,
+        UCHAR *chIndex,
+        UINT channelConfig,
+        UINT index
+        )
+{
+  if (index < 3) {
+    *chType = ACT_FRONT;
+    *chIndex = index;
+  } else {
+    switch (channelConfig) {
+      case MODE_1_2_1:
+      case MODE_1_2_2:
+      case MODE_1_2_2_1:
+        switch (index) {
+          case 3:
+          case 4:
+            *chType = ACT_BACK;
+            *chIndex = index - 3;
+            break;
+          case 5:
+            *chType = ACT_LFE;
+            *chIndex = 0;
+            break;
+        }
+        break;
+      case MODE_1_2_2_2_1:
+        switch (index) {
+          case 3:
+          case 4:
+            *chType = ACT_SIDE;
+            *chIndex = index - 3;
+            break;
+          case 5:
+          case 6:
+            *chType = ACT_BACK;
+            *chIndex = index - 5;
+            break;
+          case 7:
+            *chType = ACT_LFE;
+            *chIndex = 0;
+            break;
+        }
+        break;
+      default:
+        *chType = ACT_NONE;
+        break;
+    }
+  }
+}
+
+int CProgramConfig_LookupElement(
+        CProgramConfig *pPce,
+        const UINT      channelConfig,
+        const UINT      tag,
+        const UINT      channelIdx,
+        UCHAR           chMapping[],
+        AUDIO_CHANNEL_TYPE chType[],
+        UCHAR           chIndex[],
+        UCHAR          *elMapping,
+        MP4_ELEMENT_ID  elList[],
+        MP4_ELEMENT_ID  elType
+       )
+{
+  if (channelConfig > 0)
+  {
+    /* Constant channel mapping must have
+       been set during initialization. */
+    if ( elType == ID_SCE
+      || elType == ID_CPE
+      || elType == ID_LFE )
+    {
+      *elMapping = pPce->elCounter;
+      if (elList[pPce->elCounter] != elType) {
+        /* Not in the list */
+        return 0;
+      }
+      /* Assume all front channels */
+      getImplicitAudioChannelTypeAndIndex(&chType[channelIdx], &chIndex[channelIdx], channelConfig, channelIdx);
+      if (elType == ID_CPE) {
+        chType[channelIdx+1] = chType[channelIdx];
+        chIndex[channelIdx+1] = chIndex[channelIdx]+1;
+      }
+      pPce->elCounter++;
+    }
+    /* Accept all non-channel elements, too. */
+    return 1;
+  }
+  else
+  {
+#ifdef TP_PCE_ENABLE
+    if (!pPce->isValid)
+#endif /* TP_PCE_ENABLE */
+    {
+      /* Implicit channel mapping. */
+      if ( elType == ID_SCE
+        || elType == ID_CPE
+        || elType == ID_LFE )
+      {
+        /* Store all channel element IDs */
+        elList[pPce->elCounter] = elType;
+        *elMapping = pPce->elCounter++;
+      }
+    }
+#ifdef  TP_PCE_ENABLE
+    else {
+      /* Accept the additional channel(s), only if the tag is in the lists */
+      int isCpe = 0, i;
+      int cc = 0, fc = 0, sc = 0, bc = 0, lc = 0, ec = 0; /* Channel and element counters */
+
+      switch (elType)
+      {
+      case ID_CPE:
+        isCpe = 1;
+      case ID_SCE:
+        /* search in front channels */
+        for (i = 0; i < pPce->NumFrontChannelElements; i++) {
+          if (isCpe == pPce->FrontElementIsCpe[i] && pPce->FrontElementTagSelect[i] == tag) {
+            chMapping[cc] = channelIdx;
+            chType[cc] = ACT_FRONT;
+            chIndex[cc] = fc;
+            if (isCpe) {
+              chMapping[cc+1] = channelIdx+1;
+              chType[cc+1] = ACT_FRONT;
+              chIndex[cc+1] = fc+1;
+            }
+            *elMapping = ec;
+            return 1;
+          }
+          ec++;
+          if (pPce->FrontElementIsCpe[i]) {
+            cc+=2; fc+=2;
+          } else {
+            cc++; fc++;
+          }
+        }
+        /* search in side channels */
+        for (i = 0; i < pPce->NumSideChannelElements; i++) {
+          if (isCpe == pPce->SideElementIsCpe[i] && pPce->SideElementTagSelect[i] == tag) {
+            chMapping[cc] = channelIdx;
+            chType[cc] = ACT_SIDE;
+            chIndex[cc] = sc;
+            if (isCpe) {
+              chMapping[cc+1] = channelIdx+1;
+              chType[cc+1] = ACT_SIDE;
+              chIndex[cc+1] = sc+1;
+            }
+            *elMapping = ec;
+            return 1;
+          }
+          ec++;
+          if (pPce->SideElementIsCpe[i]) {
+            cc+=2; sc+=2;
+          } else {
+            cc++; sc++;
+          }
+        }
+        /* search in back channels */
+        for (i = 0; i < pPce->NumBackChannelElements; i++) {
+          if (isCpe == pPce->BackElementIsCpe[i] && pPce->BackElementTagSelect[i] == tag) {
+            chMapping[cc] = channelIdx;
+            chType[cc] = ACT_BACK;
+            chIndex[cc] = bc;
+            if (isCpe) {
+              chMapping[cc+1] = channelIdx+1;
+              chType[cc+1] = ACT_BACK;
+              chIndex[cc+1] = bc+1;
+            }
+            *elMapping = ec;
+            return 1;
+          }
+          ec++;
+          if (pPce->BackElementIsCpe[i]) {
+            cc+=2; bc+=2;
+          } else {
+            cc++; bc++;
+          }
+        }
+        break;
+
+      case ID_LFE:
+        /* Initialize channel counter and element counter */
+        cc = pPce->NumEffectiveChannels;
+        ec = pPce->NumFrontChannelElements+ pPce->NumSideChannelElements + pPce->NumBackChannelElements;
+        /* search in lfe channels */
+        for (i = 0; i < pPce->NumLfeChannelElements; i++) {
+          if ( pPce->LfeElementTagSelect[i] == tag ) {
+            chMapping[cc] = channelIdx;
+            *elMapping = ec;
+            chType[cc] = ACT_LFE;
+            chIndex[cc] = lc;
+            return 1;
+          }
+          ec++;
+          cc++;
+          lc++;
+        }
+        break;
+
+      /* Non audio elements */
+      case ID_CCE:
+        /* search in cce channels */
+        for (i = 0; i < pPce->NumValidCcElements; i++) {
+          if (pPce->ValidCcElementTagSelect[i] == tag) {
+            return 1;
+          }
+        }
+        break;
+      case ID_DSE:
+        /* search associated data elements */
+        for (i = 0; i < pPce->NumAssocDataElements; i++) {
+          if (pPce->AssocDataElementTagSelect[i] == tag) {
+            return 1;
+          }
+        }
+        break;
+      default:
+        return 0;
+      }
+      return 0;  /* not found in any list */
+    }
+#endif /* TP_PCE_ENABLE */
+  }
+
+  return 1;
+}
+
+#ifdef  TP_PCE_ENABLE
+int CProgramConfig_GetElementTable(
+        const CProgramConfig *pPce,
+        MP4_ELEMENT_ID  elList[],
+        const INT elListSize
+       )
+{
+  int i, el = 0;
+
+  if ( elListSize 
+    < pPce->NumFrontChannelElements + pPce->NumSideChannelElements + pPce->NumBackChannelElements + pPce->NumLfeChannelElements 
+    )
+  {
+    return 0;
+  }
+
+  for (i=0; i < pPce->NumFrontChannelElements; i++)
+  {
+    elList[el++] = (pPce->FrontElementIsCpe[i]) ?  ID_CPE : ID_SCE;
+  }
+
+  for (i=0; i < pPce->NumSideChannelElements; i++)
+  {
+    elList[el++] = (pPce->SideElementIsCpe[i]) ?  ID_CPE : ID_SCE;
+  }
+
+  for (i=0; i < pPce->NumBackChannelElements; i++)
+  {
+    elList[el++] = (pPce->BackElementIsCpe[i]) ?  ID_CPE : ID_SCE;
+  }
+
+  for (i=0; i < pPce->NumLfeChannelElements; i++)
+  {
+    elList[el++] = ID_LFE;
+  }
+
+
+  return el;
+}
+#endif
+
+static AUDIO_OBJECT_TYPE getAOT(HANDLE_FDK_BITSTREAM bs)
+{
+  int tmp = 0;
+
+  tmp = FDKreadBits(bs,5);
+  if (tmp == AOT_ESCAPE) {
+    int tmp2 = FDKreadBits(bs,6);
+    tmp = 32 + tmp2;
+  }
+
+  return (AUDIO_OBJECT_TYPE)tmp;
+}
+
+static INT getSampleRate(HANDLE_FDK_BITSTREAM bs, UCHAR *index, int nBits)
+{
+  INT sampleRate;
+  int idx;
+
+  idx = FDKreadBits(bs, nBits);
+  if( idx == (1<<nBits)-1 ) {
+    if(FDKgetValidBits(bs) < 24) {
+      return 0;
+    }
+    sampleRate = FDKreadBits(bs,24);
+  } else {
+    sampleRate = SamplingRateTable[idx];
+  }
+
+  *index = idx;
+
+  return sampleRate;
+}
+
+#ifdef TP_GA_ENABLE
+static
+TRANSPORTDEC_ERROR GaSpecificConfig_Parse( CSGaSpecificConfig    *self,
+                                           CSAudioSpecificConfig *asc,
+                                           HANDLE_FDK_BITSTREAM   bs,
+                                           UINT                   ascStartAnchor )
+{
+  TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK;
+
+  self->m_frameLengthFlag = FDKreadBits(bs,1);
+
+  self->m_dependsOnCoreCoder = FDKreadBits(bs,1);
+
+  if( self->m_dependsOnCoreCoder )
+    self->m_coreCoderDelay = FDKreadBits(bs,14);
+
+  self->m_extensionFlag = FDKreadBits(bs,1);
+
+  if( asc->m_channelConfiguration == 0 ) {
+    CProgramConfig_Read(&asc->m_progrConfigElement, bs, ascStartAnchor);
+  }
+
+  if ((asc->m_aot == AOT_AAC_SCAL) || (asc->m_aot == AOT_ER_AAC_SCAL)) {
+    self->m_layer = FDKreadBits(bs,3);
+  }
+
+  if (self->m_extensionFlag) {
+    if (asc->m_aot == AOT_ER_BSAC) {
+      self->m_numOfSubFrame = FDKreadBits(bs,5);
+      self->m_layerLength   = FDKreadBits(bs,11);
+    }
+
+    if ((asc->m_aot == AOT_ER_AAC_LC)   || (asc->m_aot == AOT_ER_AAC_LTP)  ||
+        (asc->m_aot == AOT_ER_AAC_SCAL) || (asc->m_aot == AOT_ER_AAC_LD))
+    {
+      asc->m_vcb11Flag = FDKreadBits(bs,1); /* aacSectionDataResilienceFlag */
+      asc->m_rvlcFlag  = FDKreadBits(bs,1); /* aacScalefactorDataResilienceFlag */
+      asc->m_hcrFlag   = FDKreadBits(bs,1); /* aacSpectralDataResilienceFlag */
+    }
+
+    self->m_extensionFlag3 = FDKreadBits(bs,1);
+
+  }
+  return (ErrorStatus);
+}
+#endif /* TP_GA_ENABLE */
+
+
+
+
+
+#ifdef TP_ELD_ENABLE
+
+static INT ld_sbr_header( const CSAudioSpecificConfig *asc,
+                           HANDLE_FDK_BITSTREAM hBs, 
+                           CSTpCallBacks *cb )
+{
+  const int channelConfiguration = asc->m_channelConfiguration;
+  int i = 0;
+  INT error = 0;
+
+  if (channelConfiguration == 2) {
+    error = cb->cbSbr(cb->cbSbrData, hBs, asc->m_samplingFrequency, asc->m_extensionSamplingFrequency, asc->m_samplesPerFrame, AOT_ER_AAC_ELD, ID_CPE, i++);
+  } else {
+    error = cb->cbSbr(cb->cbSbrData, hBs, asc->m_samplingFrequency, asc->m_extensionSamplingFrequency, asc->m_samplesPerFrame, AOT_ER_AAC_ELD, ID_SCE, i++);
+  }
+
+  switch ( channelConfiguration ) {
+    case 5:
+      error |= cb->cbSbr(cb->cbSbrData, hBs, asc->m_samplingFrequency, asc->m_extensionSamplingFrequency, asc->m_samplesPerFrame, AOT_ER_AAC_ELD, ID_CPE, i++);
+    case 3:
+      error |= cb->cbSbr(cb->cbSbrData, hBs, asc->m_samplingFrequency, asc->m_extensionSamplingFrequency, asc->m_samplesPerFrame, AOT_ER_AAC_ELD, ID_CPE, i++);
+      break;
+
+    case 7:
+      error |= cb->cbSbr(cb->cbSbrData, hBs, asc->m_samplingFrequency, asc->m_extensionSamplingFrequency, asc->m_samplesPerFrame, AOT_ER_AAC_ELD, ID_SCE, i++);
+    case 6:
+      error |= cb->cbSbr(cb->cbSbrData, hBs, asc->m_samplingFrequency, asc->m_extensionSamplingFrequency, asc->m_samplesPerFrame, AOT_ER_AAC_ELD, ID_CPE, i++);
+    case 4:
+      error |= cb->cbSbr(cb->cbSbrData, hBs, asc->m_samplingFrequency, asc->m_extensionSamplingFrequency, asc->m_samplesPerFrame, AOT_ER_AAC_ELD, ID_CPE, i++);
+      break;
+  }
+
+  return error;
+}
+
+static
+TRANSPORTDEC_ERROR EldSpecificConfig_Parse(
+        CSAudioSpecificConfig *asc,
+        HANDLE_FDK_BITSTREAM hBs,
+        CSTpCallBacks *cb
+        )
+{
+  TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK;
+  CSEldSpecificConfig *esc = &asc->m_sc.m_eldSpecificConfig;
+  ASC_ELD_EXT_TYPE eldExtType;
+  int eldExtLen, len, cnt;
+
+  FDKmemclear(esc, sizeof(CSEldSpecificConfig));
+
+  esc->m_frameLengthFlag = FDKreadBits(hBs, 1 );
+  if (esc->m_frameLengthFlag) {
+    asc->m_samplesPerFrame = 480;
+  } else {
+    asc->m_samplesPerFrame = 512;
+  }
+
+  asc->m_vcb11Flag = FDKreadBits(hBs, 1 );
+  asc->m_rvlcFlag  = FDKreadBits(hBs, 1 );
+  asc->m_hcrFlag   = FDKreadBits(hBs, 1 );
+
+  esc->m_sbrPresentFlag     = FDKreadBits(hBs, 1 );
+
+  if (esc->m_sbrPresentFlag == 1) {
+    esc->m_sbrSamplingRate    = FDKreadBits(hBs, 1 ); /* 0: single rate, 1: dual rate */
+    esc->m_sbrCrcFlag         = FDKreadBits(hBs, 1 );
+
+    asc->m_extensionSamplingFrequency = asc->m_samplingFrequency << esc->m_sbrSamplingRate;
+
+    if (cb->cbSbr != NULL){
+      if ( 0 != ld_sbr_header(asc, hBs, cb) ) {
+        return TRANSPORTDEC_PARSE_ERROR;
+      }
+    }
+  }
+  esc->m_useLdQmfTimeAlign = 0;
+
+  /* new ELD syntax */
+  /* parse ExtTypeConfigData */
+  while ((eldExtType = (ASC_ELD_EXT_TYPE)FDKreadBits(hBs, 4 )) != ELDEXT_TERM) {
+    eldExtLen = len = FDKreadBits(hBs, 4 );
+    if ( len == 0xf ) {
+      len = FDKreadBits(hBs, 8 );
+      eldExtLen += len;
+
+      if ( len == 0xff ) {
+        len = FDKreadBits(hBs, 16 );
+        eldExtLen += len;
+      }
+    }
+
+    switch (eldExtType) {
+      case ELDEXT_LDSAC:
+        esc->m_useLdQmfTimeAlign = 1;
+        if (cb->cbSsc != NULL) {
+          ErrorStatus = (TRANSPORTDEC_ERROR)cb->cbSsc(
+                  cb->cbSscData,
+                  hBs,
+                  asc->m_aot,
+                  asc->m_samplingFrequency,
+                  1,  /* muxMode */
+                  len
+                  );
+        } else {
+          ErrorStatus = TRANSPORTDEC_UNSUPPORTED_FORMAT;
+        }
+        if (ErrorStatus != TRANSPORTDEC_OK) {
+          goto bail;
+        }
+        break;
+      default:
+        for(cnt=0; cnt<len; cnt++) {
+          FDKreadBits(hBs, 8 );
+        }
+        break;
+      /* add future eld extension configs here */
+    }
+  }
+bail:
+  return (ErrorStatus);
+}
+#endif /* TP_ELD_ENABLE */
+
+
+
+/*
+ * API Functions
+ */
+
+void AudioSpecificConfig_Init(CSAudioSpecificConfig *asc)
+{
+  FDKmemclear(asc, sizeof(CSAudioSpecificConfig));
+
+  /* Init all values that should not be zero. */
+  asc->m_aot                    = AOT_NONE;
+  asc->m_samplingFrequencyIndex = 0xf;
+  asc->m_epConfig               = -1;
+  asc->m_extensionAudioObjectType        = AOT_NULL_OBJECT;
+#ifdef TP_PCE_ENABLE
+  CProgramConfig_Init(&asc->m_progrConfigElement);
+#endif
+}
+
+TRANSPORTDEC_ERROR AudioSpecificConfig_Parse(
+        CSAudioSpecificConfig *self,
+        HANDLE_FDK_BITSTREAM   bs,
+        int                    fExplicitBackwardCompatible,
+        CSTpCallBacks      *cb
+        )
+{
+  TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK;
+  UINT ascStartAnchor = FDKgetValidBits(bs);
+  int frameLengthFlag = -1;
+
+  AudioSpecificConfig_Init(self);
+
+  self->m_aot = getAOT(bs);
+  self->m_samplingFrequency = getSampleRate(bs, &self->m_samplingFrequencyIndex, 4);
+  if (self->m_samplingFrequency <= 0) {
+    return TRANSPORTDEC_PARSE_ERROR;
+  }
+
+  self->m_channelConfiguration = FDKreadBits(bs,4);
+
+  /* SBR extension ( explicit non-backwards compatible mode ) */
+  self->m_sbrPresentFlag = 0;
+  self->m_psPresentFlag  = 0;
+
+  if ( self->m_aot == AOT_SBR || self->m_aot == AOT_PS ) {
+    self->m_extensionAudioObjectType = AOT_SBR;
+
+    self->m_sbrPresentFlag = 1;
+    if ( self->m_aot == AOT_PS ) {
+      self->m_psPresentFlag = 1;
+    }
+
+    self->m_extensionSamplingFrequency = getSampleRate(bs, &self->m_extensionSamplingFrequencyIndex, 4);
+    self->m_aot = getAOT(bs);
+
+  } else {
+    self->m_extensionAudioObjectType = AOT_NULL_OBJECT;
+  }
+
+  /* Parse whatever specific configs */
+  switch (self->m_aot)
+  {
+#ifdef TP_GA_ENABLE
+    case AOT_AAC_LC:
+    case AOT_ER_AAC_LC:
+    case AOT_ER_AAC_LD:
+    case AOT_ER_AAC_SCAL:
+    case AOT_ER_BSAC:
+      if ((ErrorStatus = GaSpecificConfig_Parse(&self->m_sc.m_gaSpecificConfig, self, bs, ascStartAnchor)) != TRANSPORTDEC_OK ) {
+        return (ErrorStatus);
+      }
+      frameLengthFlag = self->m_sc.m_gaSpecificConfig.m_frameLengthFlag;
+      break;
+#endif /* TP_GA_ENABLE */
+    case AOT_MPEGS:
+      if (cb->cbSsc != NULL) {
+        cb->cbSsc(
+                cb->cbSscData,
+                bs,
+                self->m_aot,
+                self->m_samplingFrequency,
+                1,
+                0  /* don't know the length */
+                );
+      } else {
+        return TRANSPORTDEC_UNSUPPORTED_FORMAT;
+      }
+      break;
+#ifdef TP_ELD_ENABLE
+    case AOT_ER_AAC_ELD:
+      if ((ErrorStatus = EldSpecificConfig_Parse(self, bs, cb)) != TRANSPORTDEC_OK ) {
+        return (ErrorStatus);
+      }
+      frameLengthFlag = self->m_sc.m_eldSpecificConfig.m_frameLengthFlag;
+      self->m_sbrPresentFlag = self->m_sc.m_eldSpecificConfig.m_sbrPresentFlag;
+      self->m_extensionSamplingFrequency = (self->m_sc.m_eldSpecificConfig.m_sbrSamplingRate+1) * self->m_samplingFrequency;
+      break;
+#endif /* TP_ELD_ENABLE */
+
+    default:
+      return TRANSPORTDEC_UNSUPPORTED_FORMAT;
+      break;
+  }
+
+  /* Frame length */
+  switch (self->m_aot)
+  {
+#if defined(TP_GA_ENABLE) || defined(TP_USAC_ENABLE)
+    case AOT_AAC_LC:
+    case AOT_ER_AAC_LC:
+    case AOT_ER_AAC_SCAL:
+    case AOT_ER_BSAC:
+    /*case AOT_USAC:*/
+      if (!frameLengthFlag)
+        self->m_samplesPerFrame = 1024;
+      else
+        self->m_samplesPerFrame = 960;
+      break;
+#endif /* TP_GA_ENABLE */
+#if defined(TP_GA_ENABLE)
+    case AOT_ER_AAC_LD:
+      if (!frameLengthFlag)
+        self->m_samplesPerFrame = 512;
+      else
+        self->m_samplesPerFrame = 480;
+      break;
+#endif /* defined(TP_GA_ENABLE) */
+    default:
+      break;
+  }
+
+  switch (self->m_aot)
+  {
+    case AOT_ER_AAC_LC:
+    case AOT_ER_AAC_LD:
+    case AOT_ER_AAC_ELD:
+    case AOT_ER_AAC_SCAL:
+    case AOT_ER_CELP:
+    case AOT_ER_HVXC:
+    case AOT_ER_BSAC:
+      self->m_epConfig = FDKreadBits(bs,2);
+
+      if (self->m_epConfig > 1) {
+        return TRANSPORTDEC_UNSUPPORTED_FORMAT; // EPCONFIG;
+      }
+      break;
+    default:
+      break;
+  }
+
+
+  return (ErrorStatus);
+}
+
+
diff --git a/libMpegTPDec/src/tpdec_latm.cpp b/libMpegTPDec/src/tpdec_latm.cpp
new file mode 100644
index 0000000..99b905c
--- /dev/null
+++ b/libMpegTPDec/src/tpdec_latm.cpp
@@ -0,0 +1,423 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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):   Daniel Homm
+   Description:
+
+******************************************************************************/
+
+#include "tpdec_latm.h"
+
+
+#include "FDK_bitstream.h"
+
+
+#define TPDEC_TRACKINDEX(p,l) (2*(p) + (l))
+
+static
+UINT CLatmDemux_GetValue(HANDLE_FDK_BITSTREAM bs)
+{
+  UCHAR bytesForValue = 0, tmp = 0;
+  int value = 0;
+
+  bytesForValue = (UCHAR) FDKreadBits(bs,2);
+
+  for (UINT i=0; i<=bytesForValue; i++) {
+    value <<= 8;
+    tmp = (UCHAR) FDKreadBits(bs,8);
+    value += tmp;
+  }
+
+  return value;
+}
+
+
+static
+TRANSPORTDEC_ERROR CLatmDemux_ReadAudioMuxElement(
+        HANDLE_FDK_BITSTREAM bs,
+        CLatmDemux *pLatmDemux,
+        int m_muxConfigPresent,
+        CSTpCallBacks *pTpDecCallbacks,
+        CSAudioSpecificConfig *pAsc
+        )
+{
+  TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK;
+
+  if (m_muxConfigPresent) {
+    pLatmDemux->m_useSameStreamMux = FDKreadBits(bs,1);
+
+    if (!pLatmDemux->m_useSameStreamMux) {
+      if ((ErrorStatus = CLatmDemux_ReadStreamMuxConfig(bs, pLatmDemux, pTpDecCallbacks, pAsc))) {
+        return (ErrorStatus);
+      }
+    }
+  }
+
+  if (pLatmDemux->m_AudioMuxVersionA == 0) {
+    /* Do only once per call, because parsing and decoding is done in-line. */
+    if ((ErrorStatus = CLatmDemux_ReadPayloadLengthInfo(bs,pLatmDemux))) {
+      return (ErrorStatus);
+    }
+  } else {
+    /* audioMuxVersionA > 0 is reserved for future extensions */
+    ErrorStatus = TRANSPORTDEC_UNSUPPORTED_FORMAT;
+  }
+
+  return (ErrorStatus);
+}
+
+TRANSPORTDEC_ERROR CLatmDemux_Read(
+        HANDLE_FDK_BITSTREAM bs,
+        CLatmDemux *pLatmDemux,
+        TRANSPORT_TYPE tt,
+        CSTpCallBacks *pTpDecCallbacks,
+        CSAudioSpecificConfig *pAsc,
+        const INT ignoreBufferFullness
+        )
+{
+  UINT cntBits;
+  UINT cmpBufferFullness;
+  UINT audioMuxLengthBytesLast = 0;
+  TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK;
+
+  cntBits = FDKgetValidBits(bs);
+
+  if ((INT)cntBits < MIN_LATM_HEADERLENGTH) {
+    return TRANSPORTDEC_NOT_ENOUGH_BITS;
+  }
+
+  if ((ErrorStatus = CLatmDemux_ReadAudioMuxElement(bs, pLatmDemux, (tt != TT_MP4_LATM_MCP0), pTpDecCallbacks, pAsc)))
+    return (ErrorStatus);
+
+  if (!ignoreBufferFullness)
+  {
+    cmpBufferFullness =   24+audioMuxLengthBytesLast*8
+                        + pLatmDemux->m_linfo[0][0].m_bufferFullness* pAsc[TPDEC_TRACKINDEX(0,0)].m_channelConfiguration*32;
+
+    /* evaluate buffer fullness */
+
+    if (pLatmDemux->m_linfo[0][0].m_bufferFullness != 0xFF)
+    {
+      if (!pLatmDemux->BufferFullnessAchieved)
+      {
+        if (cntBits < cmpBufferFullness)
+        {
+          /* condition for start of decoding is not fulfilled */
+
+          /* the current frame will not be decoded */
+          return TRANSPORTDEC_NOT_ENOUGH_BITS;
+        }
+        else
+        {
+          pLatmDemux->BufferFullnessAchieved = 1;
+        }
+      }
+    }
+  }
+
+  return (ErrorStatus);
+}
+
+
+TRANSPORTDEC_ERROR CLatmDemux_ReadStreamMuxConfig(
+        HANDLE_FDK_BITSTREAM bs,
+        CLatmDemux *pLatmDemux,
+        CSTpCallBacks *pTpDecCallbacks,
+        CSAudioSpecificConfig *pAsc
+        )
+{
+  LATM_LAYER_INFO *p_linfo = NULL;
+  TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK;
+
+  pLatmDemux->m_AudioMuxVersion = FDKreadBits(bs,1);
+
+  if (pLatmDemux->m_AudioMuxVersion == 0) {
+    pLatmDemux->m_AudioMuxVersionA = 0;
+  } else {
+    pLatmDemux->m_AudioMuxVersionA = FDKreadBits(bs,1);
+  }
+
+  if (pLatmDemux->m_AudioMuxVersionA == 0) {
+    if (pLatmDemux->m_AudioMuxVersion == 1) {
+      pLatmDemux->m_taraBufferFullness = CLatmDemux_GetValue(bs);
+    }
+    pLatmDemux->m_allStreamsSameTimeFraming = FDKreadBits(bs,1);
+    pLatmDemux->m_noSubFrames = FDKreadBits(bs,6) + 1;
+    pLatmDemux->m_numProgram  = FDKreadBits(bs,4) + 1;
+
+    if (pLatmDemux->m_numProgram > 1) {
+      return TRANSPORTDEC_UNSUPPORTED_FORMAT;
+    }
+
+    int idCnt = 0;
+    for (UINT prog = 0; prog < pLatmDemux->m_numProgram; prog++) {
+      pLatmDemux->m_numLayer = FDKreadBits(bs,3) + 1;
+      if (pLatmDemux->m_numLayer > 2) {
+        return TRANSPORTDEC_UNSUPPORTED_FORMAT;
+      }
+
+      for (UINT lay = 0; lay < pLatmDemux->m_numLayer; lay++) {
+        p_linfo = &pLatmDemux->m_linfo[prog][lay];
+
+        p_linfo->m_streamID = idCnt++;
+        p_linfo->m_frameLengthInBits = 0;
+
+        if( (prog == 0) && (lay == 0) ) {
+          pLatmDemux->m_useSameConfig = 0;
+        } else {
+          pLatmDemux->m_useSameConfig = FDKreadBits(bs,1);
+        }
+
+        if (pLatmDemux->m_useSameConfig) {
+          if (lay > 1) {
+            FDKmemcpy(&pAsc[TPDEC_TRACKINDEX(prog,lay)], &pAsc[TPDEC_TRACKINDEX(prog,lay-1)], sizeof(CSAudioSpecificConfig));
+          } else {
+            return TRANSPORTDEC_PARSE_ERROR;
+          }
+        } else {
+          if (pLatmDemux->m_AudioMuxVersion == 1)
+          {
+            FDK_BITSTREAM tmpBs;
+            UINT ascStartPos, ascLen=0;
+
+            ascLen = CLatmDemux_GetValue(bs);
+            ascStartPos = FDKgetValidBits(bs);
+            tmpBs = *bs;
+            FDKsyncCache(&tmpBs);
+            tmpBs.hBitBuf.ValidBits = ascLen;
+            
+            /* Read ASC */
+            if ((ErrorStatus = AudioSpecificConfig_Parse(&pAsc[TPDEC_TRACKINDEX(prog,lay)], &tmpBs, 1, pTpDecCallbacks))) {
+              return (ErrorStatus);
+            }
+
+            /* The field p_linfo->m_ascLen could be wrong, so check if */
+            if ( 0 > (INT)FDKgetValidBits(&tmpBs)) {
+              return TRANSPORTDEC_PARSE_ERROR;
+            }
+            FDKpushFor(bs, ascLen); /* position bitstream after ASC */
+          }
+          else {
+            /* Read ASC */
+            if ((ErrorStatus = AudioSpecificConfig_Parse(&pAsc[TPDEC_TRACKINDEX(prog,lay)], bs, 0, pTpDecCallbacks))) {
+              return (ErrorStatus);
+            }
+          }
+          {
+            int cbError;
+
+            cbError = pTpDecCallbacks->cbUpdateConfig(pTpDecCallbacks->cbUpdateConfigData, &pAsc[TPDEC_TRACKINDEX(prog,lay)]);
+            if (cbError != 0) {
+              return TRANSPORTDEC_UNKOWN_ERROR;
+            }
+          }
+        }
+
+        p_linfo->m_frameLengthType = FDKreadBits(bs,3);
+        switch( p_linfo->m_frameLengthType ) {
+        case 0:
+          p_linfo->m_bufferFullness = FDKreadBits(bs,8);
+
+          if (!pLatmDemux->m_allStreamsSameTimeFraming) {
+            if ((lay > 0) && (pAsc[TPDEC_TRACKINDEX(prog,lay)].m_aot == AOT_AAC_SCAL || pAsc[TPDEC_TRACKINDEX(prog,lay)].m_aot == AOT_ER_AAC_SCAL)) {
+              return TRANSPORTDEC_UNSUPPORTED_FORMAT;
+            }
+          }
+          break;
+        case 1:
+          /* frameLength = FDKreadBits(bs,9); */
+        case 3:
+        case 4:
+        case 5:
+          /* CELP */
+        case 6:
+        case 7:
+          /* HVXC */
+        default:
+          return TRANSPORTDEC_PARSE_ERROR; //_LATM_INVALIDFRAMELENGTHTYPE;
+
+        }  /* switch framelengthtype*/
+
+      }  /* layer loop */
+    }  /* prog loop */
+
+    pLatmDemux->m_otherDataPresent = FDKreadBits(bs,1);
+    pLatmDemux->m_otherDataLength  = 0;
+
+    if (pLatmDemux->m_otherDataPresent) {
+      int otherDataLenEsc  = 0;
+      do {
+        pLatmDemux->m_otherDataLength <<= 8;   // *= 256
+        otherDataLenEsc = FDKreadBits(bs,1);
+        pLatmDemux->m_otherDataLength += FDKreadBits(bs,8);
+      } while (otherDataLenEsc);
+    }
+
+    pLatmDemux->m_crcCheckPresent = FDKreadBits(bs,1);
+    pLatmDemux->m_crcCheckSum     = 0;
+
+    if (pLatmDemux->m_crcCheckPresent) {
+      pLatmDemux->m_crcCheckSum = FDKreadBits(bs,8);
+    }
+
+  }
+  else {
+    /* audioMuxVersionA > 0 is reserved for future extensions */
+    ErrorStatus = TRANSPORTDEC_UNSUPPORTED_FORMAT;
+  }
+  return (ErrorStatus);
+}
+
+TRANSPORTDEC_ERROR CLatmDemux_ReadPayloadLengthInfo(HANDLE_FDK_BITSTREAM bs, CLatmDemux *pLatmDemux)
+{
+  TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK;
+  int totalPayloadBits = 0;
+
+  if( pLatmDemux->m_allStreamsSameTimeFraming == 1 ) {
+    for (UINT prog=0; prog<pLatmDemux->m_numProgram; prog++ ) {
+      for (UINT lay=0; lay<pLatmDemux->m_numLayer; lay++ ) {
+        LATM_LAYER_INFO *p_linfo = &pLatmDemux->m_linfo[prog][lay];
+
+        switch (p_linfo->m_frameLengthType ) {
+        case 0:
+          p_linfo->m_frameLengthInBits = CLatmDemux_ReadAuChunkLengthInfo(bs);
+          totalPayloadBits += p_linfo->m_frameLengthInBits;
+          break;
+        case 3:
+        case 5:
+        case 7:
+        default:
+          return TRANSPORTDEC_PARSE_ERROR; //AAC_DEC_LATM_INVALIDFRAMELENGTHTYPE;
+        }
+      }
+    }
+  }
+  else {
+    ErrorStatus = TRANSPORTDEC_PARSE_ERROR; //AAC_DEC_LATM_TIMEFRAMING;
+  }
+  if (pLatmDemux->m_audioMuxLengthBytes > 0 && totalPayloadBits > pLatmDemux->m_audioMuxLengthBytes*8) {
+    return TRANSPORTDEC_PARSE_ERROR;
+  }
+  return (ErrorStatus);
+}
+
+int CLatmDemux_ReadAuChunkLengthInfo(HANDLE_FDK_BITSTREAM bs)
+{
+  UCHAR endFlag;
+  int len = 0;
+
+  do {
+    UCHAR tmp = (UCHAR) FDKreadBits(bs,8);
+    endFlag = (tmp < 255);
+
+    len += tmp;
+
+  } while( endFlag == 0 );
+
+  len <<= 3;  /* convert from bytes to bits */
+
+  return len;
+}
+
+int CLatmDemux_GetFrameLengthInBits(CLatmDemux *pLatmDemux)
+{
+  return pLatmDemux->m_linfo[0][0].m_frameLengthInBits;
+}
+
+int CLatmDemux_GetOtherDataPresentFlag(CLatmDemux *pLatmDemux)
+{
+  return pLatmDemux->m_otherDataPresent ? 1 : 0;
+}
+
+int CLatmDemux_GetOtherDataLength(CLatmDemux *pLatmDemux)
+{
+  return pLatmDemux->m_otherDataLength;
+}
+
+UINT CLatmDemux_GetNrOfSubFrames(CLatmDemux *pLatmDemux)
+{
+  return pLatmDemux->m_noSubFrames;
+}
+
diff --git a/libMpegTPDec/src/tpdec_latm.h b/libMpegTPDec/src/tpdec_latm.h
new file mode 100644
index 0000000..f1fcc19
--- /dev/null
+++ b/libMpegTPDec/src/tpdec_latm.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
+----------------------------------------------------------------------------------------------------------- */
+
+/*****************************  MPEG-4 AAC Decoder  **************************
+
+   Author(s):   Daniel Homm
+   Description:
+
+******************************************************************************/
+
+#ifndef LATM_DEMUX_H
+#define LATM_DEMUX_H
+
+#include "tpdec_lib.h"
+
+
+#include "FDK_bitstream.h"
+
+#define MIN_LATM_HEADERLENGTH  9
+#define MIN_LOAS_HEADERLENGTH  MIN_LATM_HEADERLENGTH + 24   /* both in bits */
+ #define MIN_TP_BUF_SIZE_LOAS   ( 8194 )
+
+enum {
+  LATM_MAX_PROG = 1,
+  LATM_MAX_LAYER = 2,
+  LATM_MAX_VAR_CHUNKS=16,
+  LATM_MAX_ID=16
+};
+
+typedef struct {
+  UINT m_frameLengthType;
+  UINT m_bufferFullness;
+  UINT m_streamID;
+  UINT m_frameLengthInBits;
+} LATM_LAYER_INFO;
+
+typedef struct {
+  LATM_LAYER_INFO m_linfo[LATM_MAX_PROG][LATM_MAX_LAYER];
+  UINT m_taraBufferFullness;
+  UINT m_otherDataLength;
+  UINT m_audioMuxLengthBytes;          /* Length of LOAS payload */
+
+  UCHAR m_useSameStreamMux;
+  UCHAR m_AudioMuxVersion;
+  UCHAR m_AudioMuxVersionA;
+  UCHAR m_allStreamsSameTimeFraming;
+  UCHAR m_noSubFrames;
+  UCHAR m_numProgram;
+  UCHAR m_numLayer;
+  UCHAR m_useSameConfig;
+
+  UCHAR m_otherDataPresent;
+  UCHAR m_crcCheckPresent;
+  UCHAR m_crcCheckSum;
+
+  SCHAR BufferFullnessAchieved;
+} CLatmDemux;
+
+int CLatmDemux_ReadAuChunkLengthInfo(HANDLE_FDK_BITSTREAM bs);
+
+TRANSPORTDEC_ERROR CLatmDemux_Read(
+        HANDLE_FDK_BITSTREAM bs,
+        CLatmDemux *pLatmDemux,
+        TRANSPORT_TYPE tt,
+        CSTpCallBacks *pTpDecCallbacks,
+        CSAudioSpecificConfig *pAsc,
+        const INT ignoreBufferFullness
+        );
+
+TRANSPORTDEC_ERROR CLatmDemux_ReadStreamMuxConfig(
+        HANDLE_FDK_BITSTREAM bs,
+        CLatmDemux *pLatmDemux,
+        CSTpCallBacks *pTpDecCallbacks,
+        CSAudioSpecificConfig *pAsc
+        );
+
+TRANSPORTDEC_ERROR CLatmDemux_ReadPayloadLengthInfo(HANDLE_FDK_BITSTREAM bs, CLatmDemux *pLatmDemux);
+
+int  CLatmDemux_GetFrameLengthInBits(CLatmDemux *pLatmDemux);
+int  CLatmDemux_GetOtherDataPresentFlag(CLatmDemux *pLatmDemux);
+int  CLatmDemux_GetOtherDataLength(CLatmDemux *pLatmDemux);
+UINT CLatmDemux_GetNrOfSubFrames(CLatmDemux *pLatmDemux);
+
+
+#endif /* LATM_DEMUX_H */
diff --git a/libMpegTPDec/src/tpdec_lib.cpp b/libMpegTPDec/src/tpdec_lib.cpp
new file mode 100644
index 0000000..2dca5d5
--- /dev/null
+++ b/libMpegTPDec/src/tpdec_lib.cpp
@@ -0,0 +1,1203 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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 Transport Decoder  ************************
+
+   Author(s): Manuel Jander
+   Description: MPEG Transport decoder
+
+******************************************************************************/
+
+#include "tpdec_lib.h"
+
+/* library version */
+#include "version"
+
+
+#include "tp_data.h"
+
+#include "tpdec_adts.h"
+
+#include "tpdec_adif.h"
+
+#include "tpdec_latm.h"
+
+
+
+#define MODULE_NAME "transportDec"
+
+typedef union {
+  STRUCT_ADTS adts;
+
+  CAdifHeader adif;
+
+  CLatmDemux latm;
+
+
+} transportdec_parser_t;
+
+struct TRANSPORTDEC
+{
+  TRANSPORT_TYPE transportFmt;     /*!< MPEG4 transportDec type. */
+
+  CSTpCallBacks callbacks;         /*!< Struct holding callback and its data */
+
+  FDK_BITSTREAM bitStream[2]; /* Bitstream reader */
+  UCHAR *bsBuffer;                 /* Internal bitstreamd data buffer (unallocated in case of TT_MP4_RAWPACKETS) */
+
+  transportdec_parser_t parser;    /* Format specific parser structs. */
+
+  CSAudioSpecificConfig asc[(1*2)]; /* Audio specific config from the last config found. */
+  UINT  globalFramePos;            /* Global transport frame reference bit position. */
+  UINT  accessUnitAnchor[2];    /* Current access unit start bit position. */
+  INT   auLength[2];            /* Length of current access unit. */
+  INT   numberOfRawDataBlocks;     /* Current number of raw data blocks contained remaining from the current transport frame. */
+  UINT  avgBitRate;                /* Average bit rate used for frame loss estimation. */
+  UINT  lastValidBufferFullness;   /* Last valid buffer fullness value for frame loss estimation */
+  INT   remainder;                 /* Reminder in division during lost access unit estimation. */
+  INT   missingAccessUnits;        /* Estimated missing access units. */
+  UINT  burstPeriod;               /* Data burst period in mili seconds. */
+  UINT  holdOffFrames;             /* Amount of frames that were already hold off due to buffer fullness condition not being met. */
+  UINT  flags;                     /* Flags. */
+};
+
+/* Flag bitmasks for "flags" member of struct TRANSPORTDEC */
+#define TPDEC_SYNCOK                1
+#define TPDEC_MINIMIZE_DELAY        2
+#define TPDEC_IGNORE_BUFFERFULLNESS 4
+#define TPDEC_EARLY_CONFIG          8
+#define TPDEC_LOST_FRAMES_PENDING  16
+#define TPDEC_CONFIG_FOUND         32
+
+C_ALLOC_MEM(Ram_TransportDecoder, TRANSPORTDEC, 1)
+C_ALLOC_MEM(Ram_TransportDecoderBuffer, UCHAR, TRANSPORTDEC_INBUF_SIZE)
+
+
+
+
+HANDLE_TRANSPORTDEC transportDec_Open( const TRANSPORT_TYPE transportFmt, const UINT flags)
+{
+  HANDLE_TRANSPORTDEC hInput;
+
+  hInput = GetRam_TransportDecoder(0);
+  if ( hInput == NULL ) {
+    return NULL;
+  }
+
+  /* Init transportDec struct. */
+  hInput->transportFmt = transportFmt;
+
+  switch (transportFmt) {
+
+  case TT_MP4_ADIF:
+    break;
+
+  case TT_MP4_ADTS:
+    if (flags & TP_FLAG_MPEG4)
+      hInput->parser.adts.decoderCanDoMpeg4 = 1;
+    else
+      hInput->parser.adts.decoderCanDoMpeg4 = 0;
+    adtsRead_CrcInit(&hInput->parser.adts);
+    hInput->parser.adts.BufferFullnesStartFlag = 1;
+    hInput->numberOfRawDataBlocks = 0;
+    break;
+
+
+  case TT_MP4_LATM_MCP0:
+  case TT_MP4_LATM_MCP1:
+  case TT_MP4_LOAS:
+  case TT_MP4_RAW:
+    break;
+
+  default:
+    FreeRam_TransportDecoder(&hInput);
+    hInput = NULL;
+    break;
+  }
+
+  if (hInput != NULL) {
+    /* Create bitstream */
+    if ( (transportFmt == TT_MP4_RAW)
+      || (transportFmt == TT_DRM) ){
+      hInput->bsBuffer = NULL;
+    } else {
+      hInput->bsBuffer = GetRam_TransportDecoderBuffer(0);
+      if (hInput->bsBuffer == NULL) {
+          transportDec_Close( &hInput );
+          return NULL;
+      }
+      FDKinitBitStream(&hInput->bitStream[0], hInput->bsBuffer, TRANSPORTDEC_INBUF_SIZE, 0, BS_READER);
+    }
+
+    hInput->burstPeriod = 0;
+  }
+
+  return hInput;
+}
+
+TRANSPORTDEC_ERROR transportDec_OutOfBandConfig(HANDLE_TRANSPORTDEC hTp, UCHAR *conf, const UINT length, UINT layer )
+{
+  TRANSPORTDEC_ERROR     err        = TRANSPORTDEC_OK;
+
+  FDK_BITSTREAM bs;
+  HANDLE_FDK_BITSTREAM  hBs        = &bs;
+
+  FDKinitBitStream(hBs, conf, 0x80000000, length<<3, BS_READER);
+
+  /* config transport decoder */
+  switch (hTp->transportFmt) {
+    case TT_MP4_LATM_MCP0:
+    case TT_MP4_LATM_MCP1:
+    case TT_MP4_LOAS:
+      {
+        if (layer != 0) {
+          return TRANSPORTDEC_INVALID_PARAMETER;
+        }
+        CLatmDemux *pLatmDemux = &hTp->parser.latm;
+        err = CLatmDemux_ReadStreamMuxConfig(hBs, pLatmDemux, &hTp->callbacks, hTp->asc);
+        if (err != TRANSPORTDEC_OK) {
+          return err;
+        }
+      }
+      break;
+    default:
+    case TT_MP4_RAW:
+      err = AudioSpecificConfig_Parse(&hTp->asc[layer], hBs, 1, &hTp->callbacks);
+      break;
+  }
+  if (err == TRANSPORTDEC_OK) {
+    int errC;
+
+    errC = hTp->callbacks.cbUpdateConfig(hTp->callbacks.cbUpdateConfigData, &hTp->asc[layer]);
+    if (errC != 0) {
+      err = TRANSPORTDEC_PARSE_ERROR;
+    }
+  }
+
+  if (err == TRANSPORTDEC_OK) {
+    hTp->flags |= TPDEC_CONFIG_FOUND;
+  }
+
+  return err;
+}
+
+int transportDec_RegisterAscCallback( HANDLE_TRANSPORTDEC hTpDec, const cbUpdateConfig_t cbUpdateConfig, void* user_data)
+{
+  if (hTpDec == NULL) {
+    return -1;
+  }
+  hTpDec->callbacks.cbUpdateConfig = cbUpdateConfig;
+  hTpDec->callbacks.cbUpdateConfigData = user_data;
+  return 0;
+}
+
+int transportDec_RegisterSscCallback( HANDLE_TRANSPORTDEC hTpDec, const cbSsc_t cbSsc, void* user_data)
+{
+  if (hTpDec == NULL) {
+    return -1;
+  }
+  hTpDec->callbacks.cbSsc = cbSsc;
+  hTpDec->callbacks.cbSscData = user_data;
+  return 0;
+}
+
+int transportDec_RegisterSbrCallback( HANDLE_TRANSPORTDEC hTpDec, const cbSbr_t cbSbr, void* user_data)
+{
+  if (hTpDec == NULL) {
+    return -1;
+  }
+  hTpDec->callbacks.cbSbr = cbSbr;
+  hTpDec->callbacks.cbSbrData = user_data;
+  return 0;
+}
+
+TRANSPORTDEC_ERROR transportDec_FillData(
+        const HANDLE_TRANSPORTDEC  hTp,
+        UCHAR                     *pBuffer,
+        const UINT                 bufferSize,
+        UINT                      *pBytesValid,
+        const INT                  layer )
+{
+  HANDLE_FDK_BITSTREAM hBs;
+
+  if ( (hTp == NULL)
+    || (layer >= 2) ) {
+    return TRANSPORTDEC_INVALID_PARAMETER;
+  }
+
+  if (*pBytesValid == 0) {
+    /* nothing to do */
+    return TRANSPORTDEC_OK;
+  }
+
+  /* set bitbuffer shortcut */
+  hBs = &hTp->bitStream[layer];
+
+  switch (hTp->transportFmt) {
+  case TT_MP4_RAW:
+  case TT_DRM:
+    /* For packet based transport, pass input buffer to bitbuffer without copying the data.
+       Unfortunately we do not know the actual buffer size. And the FDK bit buffer implementation
+       needs a number 2^x. So we assume the maximum of 48 channels with 6144 bits per channel
+       and round it up to the next power of 2 => 65536 bytes */
+    FDKinitBitStream(hBs, pBuffer, 0x10000, (*pBytesValid)<<3, BS_READER);
+    *pBytesValid = 0;
+    break;
+
+  default:
+    /* ... else feed bitbuffer with new stream data (append). */
+    if (hTp->numberOfRawDataBlocks <= 0) {
+      FDKfeedBuffer (hBs, pBuffer, bufferSize, pBytesValid) ;
+    }
+  }
+
+  return TRANSPORTDEC_OK;
+}
+
+HANDLE_FDK_BITSTREAM transportDec_GetBitstream( const HANDLE_TRANSPORTDEC hTp, const UINT layer )
+{
+  return &hTp->bitStream[layer];
+}
+
+TRANSPORT_TYPE transportDec_GetFormat( const HANDLE_TRANSPORTDEC hTp )
+{
+  return hTp->transportFmt;
+}
+
+INT transportDec_GetBufferFullness( const HANDLE_TRANSPORTDEC hTp )
+{
+  INT bufferFullness = -1;
+
+  switch (hTp->transportFmt) {
+    case TT_MP4_ADTS:
+      if (hTp->parser.adts.bs.adts_fullness != 0x7ff) {
+        bufferFullness = hTp->parser.adts.bs.frame_length*8 + hTp->parser.adts.bs.adts_fullness * 32 * getNumberOfEffectiveChannels(hTp->parser.adts.bs.channel_config);
+      }
+      break;
+    case TT_MP4_LOAS:
+    case TT_MP4_LATM_MCP0:
+    case TT_MP4_LATM_MCP1:
+      if (hTp->parser.latm.m_linfo[0][0].m_bufferFullness != 0xff) {
+        bufferFullness = hTp->parser.latm.m_linfo[0][0].m_bufferFullness;
+      }
+      break;
+    default:
+      break;
+  }
+
+  return bufferFullness;
+}
+
+/**
+ * \brief Determine additional buffer fullness contraint due to burst data reception.
+ *        The parameter TPDEC_PARAM_BURSTPERIOD must have been set as a precondition.
+ * \param hTp transport decoder handle.
+ * \param bufferFullness the buffer fullness value of the first frame to be decoded.
+ * \param bitsAvail the amount of available bits at the end of the first frame to be decoded.
+ * \return error code
+ */
+static 
+TRANSPORTDEC_ERROR additionalHoldOffNeeded(
+        HANDLE_TRANSPORTDEC hTp,
+        INT                 bufferFullness,
+        INT                 bitsAvail
+        )
+{
+  INT checkLengthBits, avgBitsPerFrame;
+  INT maxAU; /* maximum number of frames per Master Frame */
+  INT samplesPerFrame = hTp->asc->m_samplesPerFrame;
+  INT samplingFrequency = (INT)hTp->asc->m_samplingFrequency;
+
+  if ( (hTp->avgBitRate == 0) || (hTp->burstPeriod == 0) ) {
+    return TRANSPORTDEC_OK;
+  }
+  if ( (samplesPerFrame == 0 ) || (samplingFrequency == 0) ) {
+    return TRANSPORTDEC_NOT_ENOUGH_BITS;
+  }
+
+  /* One Master Frame is sent every hTp->burstPeriod ms */
+  maxAU = hTp->burstPeriod * samplingFrequency + (samplesPerFrame*1000 - 1);
+  maxAU = maxAU / (samplesPerFrame*1000);
+  /* Subtract number of frames which were already held off. */
+  maxAU -= hTp->holdOffFrames;
+
+  avgBitsPerFrame = hTp->avgBitRate * samplesPerFrame + (samplingFrequency-1);
+  avgBitsPerFrame = avgBitsPerFrame / samplingFrequency;
+
+  /* Consider worst case of bufferFullness quantization. */
+  switch (hTp->transportFmt) {
+    case TT_MP4_ADIF:
+    case TT_MP4_ADTS:
+    case TT_MP4_LOAS:
+    case TT_MP4_LATM_MCP0:
+    case TT_MP4_LATM_MCP1:
+      bufferFullness += 31;
+      break;
+    default:
+      break;
+  }
+
+  checkLengthBits = bufferFullness + (maxAU-1)*avgBitsPerFrame;
+
+  /* Check if buffer is big enough to fullfill buffer fullness condition */
+  if ( (checkLengthBits /*+headerBits*/) > ((TRANSPORTDEC_INBUF_SIZE<<3)-7) ) {
+    return TRANSPORTDEC_SYNC_ERROR;
+  }
+
+  if ( bitsAvail < checkLengthBits ) {
+    return TRANSPORTDEC_NOT_ENOUGH_BITS;
+  }
+  else {
+    return TRANSPORTDEC_OK;
+  }
+}
+
+/**
+ * \brief adjust bit stream position and the end of an access unit.
+ * \param hTp transport decoder handle.
+ * \return error code.
+ */
+static
+TRANSPORTDEC_ERROR transportDec_AdjustEndOfAccessUnit(HANDLE_TRANSPORTDEC hTp)
+{
+  HANDLE_FDK_BITSTREAM hBs = &hTp->bitStream[0];
+  TRANSPORTDEC_ERROR err = TRANSPORTDEC_OK;
+
+  switch (hTp->transportFmt) {
+    case TT_MP4_LOAS:
+    case TT_MP4_LATM_MCP0:
+    case TT_MP4_LATM_MCP1:
+      if ( hTp->numberOfRawDataBlocks == 0 )
+      {
+        /* Check global frame length */
+        if (hTp->transportFmt == TT_MP4_LOAS && hTp->parser.latm.m_audioMuxLengthBytes > 0)
+        {
+          int loasOffset;
+
+          loasOffset = (hTp->parser.latm.m_audioMuxLengthBytes*8 + FDKgetValidBits(hBs)) - hTp->globalFramePos;
+          if (loasOffset != 0) {
+            FDKpushBiDirectional(hBs, loasOffset);
+            /* For ELD and other payloads there is an unknown amount of padding, so ignore unread bits, but
+               throw an error only if too many bits where read. */
+            if (loasOffset < 0) {
+              err = TRANSPORTDEC_PARSE_ERROR;
+            }
+          }
+        }
+
+        /* Do global LOAS/LATM audioMuxElement byte alignment */
+        FDKbyteAlign(hBs, hTp->globalFramePos);
+      }
+      break;
+    default:
+      break;
+  }
+
+  return err;
+}
+
+
+/* How many bits to advance for synchronization search. */
+#define TPDEC_SYNCSKIP 8
+
+static
+TRANSPORTDEC_ERROR synchronization(
+        HANDLE_TRANSPORTDEC hTp,
+        INT                *pHeaderBits
+        )
+{
+  TRANSPORTDEC_ERROR err = TRANSPORTDEC_OK, errFirstFrame = TRANSPORTDEC_OK;
+  HANDLE_FDK_BITSTREAM hBs = &hTp->bitStream[0];
+
+  INT syncLayerFrameBits = 0; /* Length of sync layer frame (i.e. LOAS) */
+  INT rawDataBlockLength = 0, rawDataBlockLengthPrevious;
+  INT totalBits;
+  INT headerBits = 0, headerBitsFirstFrame = 0, headerBitsPrevious;
+  INT numFramesTraversed = 0, fTraverseMoreFrames, fConfigFound = 0, startPos, startPosFirstFrame = -1;
+  INT numRawDataBlocksFirstFrame = 0, numRawDataBlocksPrevious, globalFramePosFirstFrame = 0, rawDataBlockLengthFirstFrame = 0;
+  INT ignoreBufferFullness = hTp->flags & (TPDEC_IGNORE_BUFFERFULLNESS|TPDEC_SYNCOK);
+
+  /* Synch parameters */
+  INT syncLength;      /* Length of sync word in bits */
+  UINT syncWord;       /* Sync word to be found */
+  UINT syncMask;       /* Mask for sync word (for adding one bit, so comprising one bit less) */
+  C_ALLOC_SCRATCH_START(contextFirstFrame, transportdec_parser_t, 1);
+
+  totalBits = (INT)FDKgetValidBits(hBs);
+
+  if (totalBits <= 0) {
+    /* Return sync error, because this happens only in case of severly damaged bit streams.
+       Returning TRANSPORTDEC_NOT_ENOUGH_BITS here is very dangerous. */
+    /* numberOfRawDataBlocks must be always reset in case of sync errors. */
+    hTp->numberOfRawDataBlocks = 0;
+    goto bail;
+  }
+
+  fTraverseMoreFrames = (hTp->flags & (TPDEC_MINIMIZE_DELAY|TPDEC_EARLY_CONFIG)) && ! (hTp->flags & TPDEC_SYNCOK);
+
+  /* Set transport specific sync parameters */
+  switch (hTp->transportFmt) {
+    case TT_MP4_ADTS:
+      syncWord = ADTS_SYNCWORD;
+      syncLength = ADTS_SYNCLENGTH;
+      break;
+    case TT_MP4_LOAS:
+      syncWord = 0x2B7;
+      syncLength = 11;
+      break;
+    default:
+      syncWord = 0;
+      syncLength = 0;
+      break;
+  }
+
+  syncMask = (1<<syncLength)-1;
+
+  do {
+    INT bitsAvail = 0;     /* Bits available in bitstream buffer    */
+    INT checkLengthBits;   /* Helper to check remaining bits and buffer boundaries */
+    UINT synch;            /* Current sync word read from bitstream */
+
+    headerBitsPrevious = headerBits;
+
+    bitsAvail = (INT)FDKgetValidBits(hBs);
+
+    if (hTp->numberOfRawDataBlocks == 0) {
+      /* search synchword */
+
+      FDK_ASSERT( (bitsAvail % TPDEC_SYNCSKIP) == 0);
+
+      if ((bitsAvail-syncLength) < TPDEC_SYNCSKIP) {
+        err = TRANSPORTDEC_NOT_ENOUGH_BITS;
+        headerBits = 0;
+      } else {
+
+        synch = FDKreadBits(hBs, syncLength);
+
+        if ( !(hTp->flags & TPDEC_SYNCOK) ) {
+          for (; (bitsAvail-syncLength) >= TPDEC_SYNCSKIP; bitsAvail-=TPDEC_SYNCSKIP) {
+            if (synch == syncWord) {
+              break;
+            }
+            synch = ((synch << TPDEC_SYNCSKIP) & syncMask) | FDKreadBits(hBs, TPDEC_SYNCSKIP);
+          }
+        }
+        if (synch != syncWord) {
+          /* No correct syncword found. */
+          err = TRANSPORTDEC_SYNC_ERROR;
+        } else {
+          err = TRANSPORTDEC_OK;
+        }
+        headerBits = syncLength;
+      }
+    } else {
+      headerBits = 0;
+    }
+
+    /* Save previous raw data block data */
+    rawDataBlockLengthPrevious = rawDataBlockLength;
+    numRawDataBlocksPrevious = hTp->numberOfRawDataBlocks;
+
+    /* Parse transport header (raw data block granularity) */
+    startPos = FDKgetValidBits(hBs);
+
+    if (err == TRANSPORTDEC_OK )
+    {
+      switch (hTp->transportFmt) {
+        case TT_MP4_ADTS:
+          if (hTp->numberOfRawDataBlocks <= 0)
+          {
+            int errC;
+
+            /* Parse ADTS header */
+            err = adtsRead_DecodeHeader( &hTp->parser.adts, &hTp->asc[0], hBs, ignoreBufferFullness );
+            if (err != TRANSPORTDEC_OK) {
+              if (err != TRANSPORTDEC_NOT_ENOUGH_BITS) {
+                err = TRANSPORTDEC_SYNC_ERROR;
+              }
+            } else {
+              errC = hTp->callbacks.cbUpdateConfig(hTp->callbacks.cbUpdateConfigData, &hTp->asc[0]);
+              if (errC != 0) {
+                err = TRANSPORTDEC_SYNC_ERROR;
+              } else {
+                hTp->numberOfRawDataBlocks = hTp->parser.adts.bs.num_raw_blocks+1;
+                /* CAUTION: The PCE (if available) is declared to be a part of the header! */
+                hTp->globalFramePos = FDKgetValidBits(hBs) + hTp->parser.adts.bs.num_pce_bits;
+              }
+            }
+          }
+          else {
+            /* Reset CRC because the next bits are the beginning of a raw_data_block() */
+            FDKcrcReset(&hTp->parser.adts.crcInfo);
+            hTp->globalFramePos = FDKgetValidBits(hBs);
+          }
+          if (err == TRANSPORTDEC_OK) {
+            hTp->numberOfRawDataBlocks--;
+            rawDataBlockLength = adtsRead_GetRawDataBlockLength(&hTp->parser.adts, (hTp->parser.adts.bs.num_raw_blocks-hTp->numberOfRawDataBlocks));
+            syncLayerFrameBits = (hTp->parser.adts.bs.frame_length<<3) - (startPos - FDKgetValidBits(hBs)) - syncLength;
+            if (syncLayerFrameBits <= 0) {
+              err = TRANSPORTDEC_SYNC_ERROR;
+            }
+          } else {
+            hTp->numberOfRawDataBlocks = 0;
+          }
+          break;
+        case TT_MP4_LOAS:
+          if (hTp->numberOfRawDataBlocks <= 0)
+          {
+            syncLayerFrameBits = FDKreadBits(hBs, 13);
+            hTp->parser.latm.m_audioMuxLengthBytes = syncLayerFrameBits;
+            syncLayerFrameBits <<= 3;
+          }
+        case TT_MP4_LATM_MCP1:
+        case TT_MP4_LATM_MCP0:
+          if (hTp->numberOfRawDataBlocks <= 0)
+          {
+            hTp->globalFramePos = FDKgetValidBits(hBs);
+
+            err = CLatmDemux_Read(
+                    hBs,
+                   &hTp->parser.latm,
+                    hTp->transportFmt,
+                   &hTp->callbacks,
+                    hTp->asc,
+                    ignoreBufferFullness);
+
+            if (err != TRANSPORTDEC_OK) {
+              if (err != TRANSPORTDEC_NOT_ENOUGH_BITS) {
+                err = TRANSPORTDEC_SYNC_ERROR;
+              }
+            } else {
+              hTp->numberOfRawDataBlocks = CLatmDemux_GetNrOfSubFrames(&hTp->parser.latm);
+              syncLayerFrameBits -= startPos - FDKgetValidBits(hBs) - (13);            
+            }
+          } else {
+            err = CLatmDemux_ReadPayloadLengthInfo(hBs, &hTp->parser.latm);
+            if (err != TRANSPORTDEC_OK) {
+              err = TRANSPORTDEC_SYNC_ERROR;
+            }
+          }
+          if (err == TRANSPORTDEC_OK) {
+            rawDataBlockLength = CLatmDemux_GetFrameLengthInBits(&hTp->parser.latm);
+            hTp->numberOfRawDataBlocks--;
+          } else {
+            hTp->numberOfRawDataBlocks = 0;
+          }
+          break;
+        default:
+          {
+            syncLayerFrameBits = 0;
+          }
+          break;
+      }
+    }
+
+    headerBits += startPos - (INT)FDKgetValidBits(hBs);
+    bitsAvail -= headerBits;
+
+    checkLengthBits  = syncLayerFrameBits;
+
+    /* Check if the whole frame would fit the bitstream buffer */
+    if (err == TRANSPORTDEC_OK) {
+      if ( (checkLengthBits+headerBits) > ((TRANSPORTDEC_INBUF_SIZE<<3)-7) ) {
+        /* We assume that the size of the transport bit buffer has been
+           chosen to meet all system requirements, thus this condition
+           is considered a synchronisation error. */
+        err = TRANSPORTDEC_SYNC_ERROR;
+      } else {
+        if ( bitsAvail < checkLengthBits ) {
+          err = TRANSPORTDEC_NOT_ENOUGH_BITS;
+        }
+      }
+    }
+
+    if (err == TRANSPORTDEC_NOT_ENOUGH_BITS) {
+      break;
+    }
+
+
+    if (err == TRANSPORTDEC_SYNC_ERROR) {
+      int bits;
+
+      /* Enforce re-sync of transport headers. */
+      hTp->numberOfRawDataBlocks = 0;
+
+      /* Ensure that the bit amount lands and a multiple of TPDEC_SYNCSKIP */
+      bits = (bitsAvail + headerBits) % TPDEC_SYNCSKIP;
+      /* Rewind - TPDEC_SYNCSKIP, in order to look for a synch one bit ahead next time. */
+      FDKpushBiDirectional(hBs, -(headerBits - TPDEC_SYNCSKIP) + bits);
+      bitsAvail += headerBits - TPDEC_SYNCSKIP - bits;
+      headerBits = 0;      
+    }
+
+    /* Frame traversal */
+    if ( fTraverseMoreFrames )
+    {
+      /* Save parser context for early config discovery "rewind all frames" */
+      if ( (hTp->flags & TPDEC_EARLY_CONFIG) && !(hTp->flags & TPDEC_MINIMIZE_DELAY))
+      {
+        /* ignore buffer fullness if just traversing additional frames for ECD */
+        ignoreBufferFullness = 1;
+
+        /* Save context in order to return later */
+        if ( err == TRANSPORTDEC_OK && startPosFirstFrame == -1 ) {
+          startPosFirstFrame = FDKgetValidBits(hBs);
+          numRawDataBlocksFirstFrame = hTp->numberOfRawDataBlocks;
+          globalFramePosFirstFrame = hTp->globalFramePos;
+          rawDataBlockLengthFirstFrame = rawDataBlockLength;
+          headerBitsFirstFrame = headerBits;
+          errFirstFrame = err;
+          FDKmemcpy(contextFirstFrame, &hTp->parser, sizeof(transportdec_parser_t));
+        }
+
+        /* Break when config was found or it is not possible anymore to find a config */
+        if (startPosFirstFrame != -1 && (fConfigFound || err != TRANSPORTDEC_OK)) {
+          break;
+        }
+      }
+
+      if (err == TRANSPORTDEC_OK) {
+        FDKpushFor(hBs, rawDataBlockLength);
+        bitsAvail -= rawDataBlockLength;
+        numFramesTraversed++;
+        /* Ignore error here itentionally. */
+        transportDec_AdjustEndOfAccessUnit(hTp);
+      }
+    }
+  } while ( fTraverseMoreFrames || (err == TRANSPORTDEC_SYNC_ERROR && !(hTp->flags & TPDEC_SYNCOK)));
+
+  /* Restore context in case of ECD frame traversal */
+  if ( startPosFirstFrame != -1 && (fConfigFound || err != TRANSPORTDEC_OK) ) {
+    FDKpushBiDirectional(hBs, FDKgetValidBits(hBs) - startPosFirstFrame);
+    FDKmemcpy(&hTp->parser, contextFirstFrame, sizeof(transportdec_parser_t));
+    hTp->numberOfRawDataBlocks = numRawDataBlocksFirstFrame;
+    hTp->globalFramePos = globalFramePosFirstFrame;
+    rawDataBlockLength = rawDataBlockLengthFirstFrame;
+    headerBits = headerBitsFirstFrame;
+    err = errFirstFrame;
+    numFramesTraversed = 0;
+  } 
+
+  /* Additional burst data mode buffer fullness check. */
+  if ( !(hTp->flags & (TPDEC_IGNORE_BUFFERFULLNESS|TPDEC_SYNCOK)) && err == TRANSPORTDEC_OK) {
+    err = additionalHoldOffNeeded(hTp, transportDec_GetBufferFullness(hTp), FDKgetValidBits(hBs) - syncLayerFrameBits);
+    if (err == TRANSPORTDEC_NOT_ENOUGH_BITS) {
+      hTp->holdOffFrames++;
+    }
+  }
+  
+  /* Rewind for retry because of not enough bits */
+  if (err == TRANSPORTDEC_NOT_ENOUGH_BITS) {
+    FDKpushBack(hBs, headerBits);
+    headerBits = 0;
+  }
+  else {
+    /* reset hold off frame counter */
+    hTp->holdOffFrames = 0;
+  }
+
+  /* Return to last good frame in case of frame traversal but not ECD. */
+  if (numFramesTraversed > 0) {
+    FDKpushBack(hBs, rawDataBlockLengthPrevious);
+    if (err != TRANSPORTDEC_OK) {
+      hTp->numberOfRawDataBlocks = numRawDataBlocksPrevious;
+      headerBits = headerBitsPrevious;
+    }
+    err = TRANSPORTDEC_OK;
+  }
+
+bail:
+  hTp->auLength[0] = rawDataBlockLength;
+
+  if (err == TRANSPORTDEC_OK) {
+    hTp->flags |= TPDEC_SYNCOK;
+  }
+
+  if (pHeaderBits != NULL) {
+    *pHeaderBits = headerBits;
+  }
+
+  if (err == TRANSPORTDEC_SYNC_ERROR) {
+    hTp->flags &= ~TPDEC_SYNCOK;
+  }
+
+  C_ALLOC_SCRATCH_END(contextFirstFrame, transportdec_parser_t, 1);
+
+  return err;
+}
+
+/**
+ * \brief Synchronize to stream and estimate the amount of missing access units due
+ *        to a current synchronization error in case of constant average bit rate.
+ */
+static
+TRANSPORTDEC_ERROR transportDec_readStream ( HANDLE_TRANSPORTDEC hTp, const UINT layer )
+{
+
+  TRANSPORTDEC_ERROR error = TRANSPORTDEC_OK;
+  HANDLE_FDK_BITSTREAM hBs = &hTp->bitStream[layer];
+  INT nAU = -1;
+  INT headerBits;
+  INT bitDistance, bfDelta;
+
+  /* Obtain distance to next synch word */
+  bitDistance = FDKgetValidBits(hBs);
+  error = synchronization(hTp, &headerBits);
+  bitDistance -= FDKgetValidBits(hBs);
+
+
+  FDK_ASSERT(bitDistance >= 0);
+
+  if (error == TRANSPORTDEC_SYNC_ERROR || (hTp->flags & TPDEC_LOST_FRAMES_PENDING))
+  {
+    /* Check if estimating lost access units is feasible. */
+    if (hTp->avgBitRate > 0 && hTp->asc[0].m_samplesPerFrame > 0 && hTp->asc[0].m_samplingFrequency > 0)
+    {
+      if (error == TRANSPORTDEC_OK)
+      {
+        int aj;
+
+        aj = transportDec_GetBufferFullness(hTp);
+        if (aj > 0) {
+          bfDelta = aj;
+        } else {
+          bfDelta = 0;
+        }
+        /* sync was ok: last of a series of bad access units. */
+        hTp->flags &= ~TPDEC_LOST_FRAMES_PENDING;
+        /* Add up bitDistance until end of the current frame. Later we substract
+           this frame from the grand total, since this current successfully synchronized
+           frame should not be skipped of course; but it must be accounted into the
+           bufferfulness math. */
+        bitDistance += hTp->auLength[0];
+      } else {
+        if ( !(hTp->flags & TPDEC_LOST_FRAMES_PENDING) ) {
+          /* sync not ok: one of many bad access units. */
+          hTp->flags |= TPDEC_LOST_FRAMES_PENDING;
+          bfDelta = - (INT)hTp->lastValidBufferFullness;
+        } else {
+          bfDelta = 0;
+        }
+      }
+
+      {
+        int num, denom;
+
+        /* Obtain estimate of number of lost frames */
+        num = hTp->asc[0].m_samplingFrequency * (bfDelta + bitDistance) + hTp->remainder;
+        denom = hTp->avgBitRate * hTp->asc[0].m_samplesPerFrame;
+        if (num > 0) {
+          nAU = num / denom;
+          hTp->remainder = num % denom;
+        } else {
+          hTp->remainder = num;
+        }
+
+        if (error == TRANSPORTDEC_OK)
+        {
+          /* Final adjustment of remainder, taken -1 into account because current
+             frame should not be skipped, thus substract -1 or do nothing instead
+             of +1-1 accordingly. */
+          if ( (denom - hTp->remainder) >= hTp->remainder ) {
+            nAU--;
+          }
+            
+          if (nAU < 0) {
+            /* There was one frame too much concealed, so unfortunately we will have to skip one good frame. */
+            transportDec_EndAccessUnit(hTp);
+            error = synchronization(hTp, &headerBits);             
+            nAU = -1;
+#ifdef DEBUG
+            FDKprintf("ERROR: Bufferfullness accounting failed. remainder=%d, nAU=%d\n", hTp->remainder, nAU);
+#endif
+          }
+          hTp->remainder = 0;
+          /* Enforce last missed frames to be concealed. */
+          if (nAU > 0) {
+            FDKpushBack(hBs, headerBits);
+          }
+        }
+      }
+    }
+  }
+
+  /* Be sure that lost frames are handled correctly. This is necessary due to some
+     sync error sequences where later it turns out that there is not enough data, but
+     the bits upto the sync word are discarded, thus causing a value of nAU > 0 */
+  if (nAU > 0) {
+    error = TRANSPORTDEC_SYNC_ERROR;
+  }
+
+  hTp->missingAccessUnits = nAU;
+
+  return error;
+}
+
+/* returns error code */
+TRANSPORTDEC_ERROR transportDec_ReadAccessUnit( const HANDLE_TRANSPORTDEC hTp, const UINT layer )
+{
+  TRANSPORTDEC_ERROR err = TRANSPORTDEC_OK;
+  HANDLE_FDK_BITSTREAM hBs;
+
+  if (!hTp) {
+    return TRANSPORTDEC_INVALID_PARAMETER;
+  }
+
+  hBs = &hTp->bitStream[layer];
+
+  switch (hTp->transportFmt) {
+
+    case TT_MP4_ADIF:
+      /* Read header if not already done */
+      if (!(hTp->flags & TPDEC_CONFIG_FOUND))
+      {
+        CProgramConfig *pce;
+
+        AudioSpecificConfig_Init(&hTp->asc[0]);
+        pce = &hTp->asc[0].m_progrConfigElement;
+        err = adifRead_DecodeHeader(&hTp->parser.adif, pce, hBs);
+        if (err)
+          goto bail;
+
+        /* Map adif header to ASC */
+        hTp->asc[0].m_aot                    = (AUDIO_OBJECT_TYPE)(pce->Profile + 1);
+        hTp->asc[0].m_samplingFrequencyIndex = pce->SamplingFrequencyIndex;
+        hTp->asc[0].m_samplingFrequency      = SamplingRateTable[pce->SamplingFrequencyIndex];
+        hTp->asc[0].m_channelConfiguration   = 0;
+        hTp->asc[0].m_samplesPerFrame        = 1024;
+        hTp->avgBitRate                      = hTp->parser.adif.BitRate;
+
+        /* Call callback to decoder. */
+        {
+          int errC;
+
+          errC = hTp->callbacks.cbUpdateConfig(hTp->callbacks.cbUpdateConfigData, &hTp->asc[0]);
+          if (errC == 0) {
+            hTp->flags |= TPDEC_CONFIG_FOUND;
+          } else {
+            err = TRANSPORTDEC_PARSE_ERROR;
+            goto bail;
+          }
+        }
+      }
+      hTp->auLength[layer] = -1; /* Access Unit data length is unknown. */
+      break;
+
+    case TT_MP4_RAW:
+      if ((INT)FDKgetValidBits(hBs) <= 0 && layer == 0) {
+        err = TRANSPORTDEC_NOT_ENOUGH_BITS;
+      }
+      /* One Access Unit was filled into buffer.
+         So get the length out of the buffer. */
+      hTp->auLength[layer] = FDKgetValidBits(hBs);
+      hTp->flags |= TPDEC_SYNCOK;
+      break;
+
+    case TT_RSVD50:
+    case TT_MP4_ADTS:
+    case TT_MP4_LOAS:
+    case TT_MP4_LATM_MCP0:
+    case TT_MP4_LATM_MCP1:
+      err = transportDec_readStream(hTp, layer);
+      break;
+
+    default:
+      err = TRANSPORTDEC_UNSUPPORTED_FORMAT;
+      break;
+  }
+
+  if (err == TRANSPORTDEC_OK) {
+    hTp->accessUnitAnchor[layer] = FDKgetValidBits(hBs);
+  } else {
+    hTp->accessUnitAnchor[layer] = 0;
+  }
+
+bail:
+  return err;
+}
+
+INT transportDec_GetAuBitsRemaining( const HANDLE_TRANSPORTDEC hTp, const UINT layer )
+{
+  INT bits;
+
+  if (hTp->accessUnitAnchor[layer] > 0 && hTp->auLength[layer] > 0) {
+    bits = hTp->auLength[layer] - (hTp->accessUnitAnchor[layer] - FDKgetValidBits(&hTp->bitStream[layer]));
+  } else {
+    bits = FDKgetValidBits(&hTp->bitStream[layer]);
+  }
+
+  return bits;
+}
+
+INT transportDec_GetAuBitsTotal( const HANDLE_TRANSPORTDEC hTp, const UINT layer )
+{
+  return hTp->auLength[layer];
+}
+
+TRANSPORTDEC_ERROR transportDec_GetMissingAccessUnitCount ( INT *pNAccessUnits, HANDLE_TRANSPORTDEC hTp )
+{
+  *pNAccessUnits = hTp->missingAccessUnits;
+
+  return TRANSPORTDEC_OK;
+}
+
+/* Inform the transportDec layer that reading of access unit has finished. */
+TRANSPORTDEC_ERROR transportDec_EndAccessUnit(HANDLE_TRANSPORTDEC hTp)
+{
+  TRANSPORTDEC_ERROR err = TRANSPORTDEC_OK;
+
+  err = transportDec_AdjustEndOfAccessUnit(hTp);
+
+  switch (hTp->transportFmt) {
+    case TT_MP4_LOAS:
+    case TT_MP4_LATM_MCP0:
+    case TT_MP4_LATM_MCP1:
+      break;
+    default:
+      break;
+  }
+
+  return err;
+}
+
+TRANSPORTDEC_ERROR transportDec_SetParam ( const HANDLE_TRANSPORTDEC hTp,
+                                           const TPDEC_PARAM        param,
+                                           const INT                value)
+{
+  TRANSPORTDEC_ERROR error = TRANSPORTDEC_OK;
+
+  switch (param) {
+    case TPDEC_PARAM_MINIMIZE_DELAY:
+      if (value) {
+        hTp->flags |= TPDEC_MINIMIZE_DELAY;
+      } else {
+        hTp->flags &= ~TPDEC_MINIMIZE_DELAY;
+      }
+      break;
+    case TPDEC_PARAM_EARLY_CONFIG:
+      if (value) {
+        hTp->flags |= TPDEC_EARLY_CONFIG;
+      } else {
+        hTp->flags &= ~TPDEC_EARLY_CONFIG;
+      }
+      break;
+    case TPDEC_PARAM_IGNORE_BUFFERFULLNESS:
+      if (value) {
+        hTp->flags |= TPDEC_IGNORE_BUFFERFULLNESS;
+      } else {
+        hTp->flags &= ~TPDEC_IGNORE_BUFFERFULLNESS;
+      }
+      break;
+    case TPDEC_PARAM_SET_BITRATE:
+      hTp->avgBitRate = value;
+      break;
+    case TPDEC_PARAM_BURST_PERIOD:
+      hTp->burstPeriod = value;
+      break;
+    case TPDEC_PARAM_RESET:
+      {
+        int i;
+
+        for (i=0; i<(1*2); i++) {
+          FDKresetBitbuffer(&hTp->bitStream[i]);
+          hTp->auLength[i] = 0;
+          hTp->accessUnitAnchor[i] = 0;
+        }        
+        hTp->flags &= ~(TPDEC_SYNCOK|TPDEC_LOST_FRAMES_PENDING);
+        hTp->remainder = 0;
+        hTp->avgBitRate = 0;
+        hTp->missingAccessUnits = 0;
+        hTp->numberOfRawDataBlocks = 0;
+        hTp->globalFramePos = 0;
+        hTp->holdOffFrames = 0;
+      }
+      break;
+  }
+
+  return error;
+}
+
+UINT transportDec_GetNrOfSubFrames(HANDLE_TRANSPORTDEC hTp)
+{
+  UINT nSubFrames = 0;
+
+  if (hTp == NULL)
+    return 0;
+
+  if (hTp->transportFmt==TT_MP4_LATM_MCP1 || hTp->transportFmt==TT_MP4_LATM_MCP0 || hTp->transportFmt==TT_MP4_LOAS)
+    nSubFrames = CLatmDemux_GetNrOfSubFrames(&hTp->parser.latm);
+  else if (hTp->transportFmt==TT_MP4_ADTS)
+    nSubFrames = hTp->parser.adts.bs.num_raw_blocks;
+
+  return nSubFrames;
+}
+
+void transportDec_Close(HANDLE_TRANSPORTDEC *phTp)
+{
+  if (phTp != NULL)
+  {
+    if (*phTp != NULL) {
+      if ((*phTp)->transportFmt != TT_MP4_RAW && (*phTp)->transportFmt != TT_DRM) {
+        FreeRam_TransportDecoderBuffer(&(*phTp)->bsBuffer);
+      }
+      if (*phTp != NULL) {
+        FreeRam_TransportDecoder(phTp);
+      }
+    }
+  }
+}
+
+TRANSPORTDEC_ERROR transportDec_GetLibInfo( LIB_INFO *info )
+{
+  int i;
+
+  if (info == NULL) {
+    return TRANSPORTDEC_UNKOWN_ERROR;
+  }
+
+  /* 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 TRANSPORTDEC_UNKOWN_ERROR;
+  info += i;
+
+  info->module_id  = FDK_TPDEC;
+  info->build_date = __DATE__;
+  info->build_time = __TIME__;
+  info->title      = TP_LIB_TITLE;
+  info->version    = LIB_VERSION(TP_LIB_VL0, TP_LIB_VL1, TP_LIB_VL2);
+  LIB_VERSION_STRING(info);
+  info->flags = 0
+    | CAPF_ADIF
+    | CAPF_ADTS
+    | CAPF_LATM
+    | CAPF_LOAS
+    | CAPF_RAWPACKETS
+    ;
+
+  return TRANSPORTDEC_OK; /* FDKERR_NOERROR; */
+}
+
+
+int  transportDec_CrcStartReg(HANDLE_TRANSPORTDEC pTp, INT mBits)
+{
+  switch (pTp->transportFmt) {
+  case TT_MP4_ADTS:
+    return adtsRead_CrcStartReg(&pTp->parser.adts, &pTp->bitStream[0], mBits);
+  default:
+    return 0;
+  }
+}
+
+void transportDec_CrcEndReg(HANDLE_TRANSPORTDEC pTp, INT reg)
+{
+  switch (pTp->transportFmt) {
+  case TT_MP4_ADTS:
+    adtsRead_CrcEndReg(&pTp->parser.adts, &pTp->bitStream[0], reg);
+    break;
+  default:
+    break;
+  }
+}
+
+TRANSPORTDEC_ERROR transportDec_CrcCheck(HANDLE_TRANSPORTDEC pTp)
+{
+  switch (pTp->transportFmt) {
+  case TT_MP4_ADTS:
+    if ( (pTp->parser.adts.bs.num_raw_blocks > 0) && (pTp->parser.adts.bs.protection_absent == 0) )
+    {
+      HANDLE_FDK_BITSTREAM hBs = &pTp->bitStream[0];
+      int bitDiff;
+      
+      /* Calculate possible offset to CRC value. */
+      bitDiff  = pTp->parser.adts.rawDataBlockDist[pTp->parser.adts.bs.num_raw_blocks-pTp->numberOfRawDataBlocks]<<3;
+      bitDiff -= pTp->globalFramePos - FDKgetValidBits(hBs) + 16;
+      FDKpushBiDirectional(hBs, bitDiff);
+      pTp->parser.adts.crcReadValue = FDKreadBits(hBs, 16);
+    }
+    return adtsRead_CrcCheck(&pTp->parser.adts);
+  default:
+    return TRANSPORTDEC_OK;
+  }
+}
diff --git a/libMpegTPDec/src/version b/libMpegTPDec/src/version
new file mode 100644
index 0000000..5b9a425
--- /dev/null
+++ b/libMpegTPDec/src/version
@@ -0,0 +1,8 @@
+
+/* library info */
+#define TP_LIB_VL0 2
+#define TP_LIB_VL1 2
+#define TP_LIB_VL2 1
+#define TP_LIB_TITLE "MPEG Transport"
+#define TP_LIB_BUILD_DATE __DATE__
+#define TP_LIB_BUILD_TIME __TIME__