Remove FDK-AAC

12 files changed, 0 insertions, 4091 deletions

diff --git a/libMpegTPEnc/src/tpenc_adif.cpp b/libMpegTPEnc/src/tpenc_adif.cpp
deleted file mode 100644
index b48a32e..0000000
--- a/libMpegTPEnc/src/tpenc_adif.cpp
+++ /dev/null
@@ -1,182 +0,0 @@
-
-/* -----------------------------------------------------------------------------------------------------------
-Software License for The Fraunhofer FDK AAC Codec Library for Android
-
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
-  All rights reserved.
-
- 1.    INTRODUCTION
-The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
-the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
-This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
-
-AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
-audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
-independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
-of the MPEG specifications.
-
-Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
-may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
-individually for the purpose of encoding or decoding bit streams in products that are compliant with
-the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
-these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
-software may already be covered under those patent licenses when it is used for those licensed purposes only.
-
-Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
-are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
-applications information and documentation.
-
-2.    COPYRIGHT LICENSE
-
-Redistribution and use in source and binary forms, with or without modification, are permitted without
-payment of copyright license fees provided that you satisfy the following conditions:
-
-You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
-your modifications thereto in source code form.
-
-You must retain the complete text of this software license in the documentation and/or other materials
-provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
-You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
-modifications thereto to recipients of copies in binary form.
-
-The name of Fraunhofer may not be used to endorse or promote products derived from this library without
-prior written permission.
-
-You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
-software or your modifications thereto.
-
-Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
-and the date of any change. For modified versions of the FDK AAC Codec, the term
-"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
-"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
-
-3.    NO PATENT LICENSE
-
-NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
-ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
-respect to this software.
-
-You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
-by appropriate patent licenses.
-
-4.    DISCLAIMER
-
-This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
-"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
-of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
-CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
-including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
-or business interruption, however caused and on any theory of liability, whether in contract, strict
-liability, or tort (including negligence), arising in any way out of the use of this software, even if
-advised of the possibility of such damage.
-
-5.    CONTACT INFORMATION
-
-Fraunhofer Institute for Integrated Circuits IIS
-Attention: Audio and Multimedia Departments - FDK AAC LL
-Am Wolfsmantel 33
-91058 Erlangen, Germany
-
-www.iis.fraunhofer.de/amm
-amm-info@iis.fraunhofer.de
------------------------------------------------------------------------------------------------------------ */
-
-/******************************** MPEG Audio Encoder **************************
-
-   contents/description: ADIF Transport Headers writing
-
-******************************************************************************/
-
-#include "tpenc_adif.h"
-
-#include "tpenc_lib.h"
-#include "tpenc_asc.h"
-
-
-
-int adifWrite_EncodeHeader(ADIF_INFO *adif,
-                                 HANDLE_FDK_BITSTREAM hBs,
-                                 INT adif_buffer_fullness)
-{
-  /* ADIF/PCE/ADTS definitions */
-  const char adifId[5]="ADIF";
-  const int  copyRightIdPresent=0;
-  const int  originalCopy=0;
-  const int  home=0;
-
-  int i;
-
-  INT sampleRate = adif->samplingRate;
-  INT totalBitRate = adif->bitRate;
-
-  if (adif->headerWritten)
-    return 0;
-
-  /* Align inside PCE with respect to the first bit of the header */
-  UINT alignAnchor = FDKgetValidBits(hBs);
-
-  /* Signal variable bitrate if buffer fullnes exceeds 20 bit */
-  adif->bVariableRate = ( adif_buffer_fullness >= (INT)(0x1<<20) ) ? 1 : 0;
-
-  FDKwriteBits(hBs, adifId[0],8);
-  FDKwriteBits(hBs, adifId[1],8);
-  FDKwriteBits(hBs, adifId[2],8);
-  FDKwriteBits(hBs, adifId[3],8);
-
-
-  FDKwriteBits(hBs, copyRightIdPresent ? 1:0,1);
-
-  if(copyRightIdPresent) {
-    for(i=0;i<72;i++) {
-      FDKwriteBits(hBs,0,1);
-    }
-  }
-  FDKwriteBits(hBs, originalCopy ? 1:0,1);
-  FDKwriteBits(hBs, home ? 1:0,1);
-  FDKwriteBits(hBs, adif->bVariableRate?1:0, 1);
-  FDKwriteBits(hBs, totalBitRate,23);
-
-  /* we write only one PCE at the moment */
-  FDKwriteBits(hBs, 0, 4);
-
-  if(!adif->bVariableRate) {
-    FDKwriteBits(hBs, adif_buffer_fullness, 20);
-  }
-
-  /* Write PCE */
-  transportEnc_writePCE(hBs, adif->cm, sampleRate, adif->instanceTag, adif->profile, 0, 0, alignAnchor);
-
-  return 0;
-}
-
-int adifWrite_GetHeaderBits(ADIF_INFO *adif)
-{
-  /* ADIF definitions */
-  const int  copyRightIdPresent=0;
-
-  if (adif->headerWritten)
-    return 0;
-
-  int bits = 0;
-
-  bits += 8*4; /* ADIF ID */
-
-  bits += 1; /* Copyright present */
-
-  if (copyRightIdPresent)
-    bits += 72;          /* Copyright ID */
-
-  bits += 26;
-
-  bits += 4; /* Number of PCE's */
-
-  if(!adif->bVariableRate) {
-    bits += 20;
-  }
-
-  /* write PCE */
-  bits = transportEnc_GetPCEBits(adif->cm, 0, bits);
-
-  return bits;
-}
-
diff --git a/libMpegTPEnc/src/tpenc_adif.h b/libMpegTPEnc/src/tpenc_adif.h
deleted file mode 100644
index d590354..0000000
--- a/libMpegTPEnc/src/tpenc_adif.h
+++ /dev/null
@@ -1,135 +0,0 @@
-
-/* -----------------------------------------------------------------------------------------------------------
-Software License for The Fraunhofer FDK AAC Codec Library for Android
-
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
-  All rights reserved.
-
- 1.    INTRODUCTION
-The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
-the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
-This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
-
-AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
-audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
-independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
-of the MPEG specifications.
-
-Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
-may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
-individually for the purpose of encoding or decoding bit streams in products that are compliant with
-the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
-these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
-software may already be covered under those patent licenses when it is used for those licensed purposes only.
-
-Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
-are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
-applications information and documentation.
-
-2.    COPYRIGHT LICENSE
-
-Redistribution and use in source and binary forms, with or without modification, are permitted without
-payment of copyright license fees provided that you satisfy the following conditions:
-
-You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
-your modifications thereto in source code form.
-
-You must retain the complete text of this software license in the documentation and/or other materials
-provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
-You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
-modifications thereto to recipients of copies in binary form.
-
-The name of Fraunhofer may not be used to endorse or promote products derived from this library without
-prior written permission.
-
-You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
-software or your modifications thereto.
-
-Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
-and the date of any change. For modified versions of the FDK AAC Codec, the term
-"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
-"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
-
-3.    NO PATENT LICENSE
-
-NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
-ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
-respect to this software.
-
-You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
-by appropriate patent licenses.
-
-4.    DISCLAIMER
-
-This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
-"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
-of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
-CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
-including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
-or business interruption, however caused and on any theory of liability, whether in contract, strict
-liability, or tort (including negligence), arising in any way out of the use of this software, even if
-advised of the possibility of such damage.
-
-5.    CONTACT INFORMATION
-
-Fraunhofer Institute for Integrated Circuits IIS
-Attention: Audio and Multimedia Departments - FDK AAC LL
-Am Wolfsmantel 33
-91058 Erlangen, Germany
-
-www.iis.fraunhofer.de/amm
-amm-info@iis.fraunhofer.de
------------------------------------------------------------------------------------------------------------ */
-
-/******************************** MPEG Audio Encoder **************************
-
-   Initial author:       Alex Goeschel
-   contents/description: Transport Headers support
-
-******************************************************************************/
-
-#ifndef TPENC_ADIF_H
-#define TPENC_ADIF_H
-
-#include "machine_type.h"
-#include "FDK_bitstream.h"
-
-#include "tp_data.h"
-
-typedef struct {
-  CHANNEL_MODE cm;
-  INT samplingRate;
-  INT bitRate;
-  int profile;
-  int bVariableRate;
-  int instanceTag;
-  int headerWritten;
-} ADIF_INFO;
-
-/**
- * \brief encodes ADIF Header
- *
- * \param adif pointer to ADIF_INFO structure
- * \param hBitStream handle of bitstream, where the ADIF header is written into
- * \param adif_buffer_fullness buffer fullness value for the ADIF header
- *
- * \return 0 on success
- */
-int adifWrite_EncodeHeader(
-        ADIF_INFO           *adif,
-        HANDLE_FDK_BITSTREAM hBitStream,
-        INT                  adif_buffer_fullness
-        );
-
-/**
- * \brief Get bit demand of a ADIF header
- *
- * \param adif pointer to ADIF_INFO structure
- *
- * \return amount of bits required to write the ADIF header according to the data
- *         contained in the adif parameter
- */
-int adifWrite_GetHeaderBits( ADIF_INFO *adif );
-
-#endif /* TPENC_ADIF_H */
-
diff --git a/libMpegTPEnc/src/tpenc_adts.cpp b/libMpegTPEnc/src/tpenc_adts.cpp
deleted file mode 100644
index f4f3178..0000000
--- a/libMpegTPEnc/src/tpenc_adts.cpp
+++ /dev/null
@@ -1,315 +0,0 @@
-
-/* -----------------------------------------------------------------------------------------------------------
-Software License for The Fraunhofer FDK AAC Codec Library for Android
-
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
-  All rights reserved.
-
- 1.    INTRODUCTION
-The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
-the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
-This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
-
-AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
-audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
-independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
-of the MPEG specifications.
-
-Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
-may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
-individually for the purpose of encoding or decoding bit streams in products that are compliant with
-the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
-these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
-software may already be covered under those patent licenses when it is used for those licensed purposes only.
-
-Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
-are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
-applications information and documentation.
-
-2.    COPYRIGHT LICENSE
-
-Redistribution and use in source and binary forms, with or without modification, are permitted without
-payment of copyright license fees provided that you satisfy the following conditions:
-
-You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
-your modifications thereto in source code form.
-
-You must retain the complete text of this software license in the documentation and/or other materials
-provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
-You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
-modifications thereto to recipients of copies in binary form.
-
-The name of Fraunhofer may not be used to endorse or promote products derived from this library without
-prior written permission.
-
-You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
-software or your modifications thereto.
-
-Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
-and the date of any change. For modified versions of the FDK AAC Codec, the term
-"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
-"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
-
-3.    NO PATENT LICENSE
-
-NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
-ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
-respect to this software.
-
-You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
-by appropriate patent licenses.
-
-4.    DISCLAIMER
-
-This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
-"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
-of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
-CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
-including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
-or business interruption, however caused and on any theory of liability, whether in contract, strict
-liability, or tort (including negligence), arising in any way out of the use of this software, even if
-advised of the possibility of such damage.
-
-5.    CONTACT INFORMATION
-
-Fraunhofer Institute for Integrated Circuits IIS
-Attention: Audio and Multimedia Departments - FDK AAC LL
-Am Wolfsmantel 33
-91058 Erlangen, Germany
-
-www.iis.fraunhofer.de/amm
-amm-info@iis.fraunhofer.de
------------------------------------------------------------------------------------------------------------ */
-
-/******************************** MPEG Audio Encoder **************************
-
-   Initial author:       Alex Groeschel
-   contents/description: ADTS Transport Headers support
-
-******************************************************************************/
-
-#include "tpenc_adts.h"
-
-
-#include "tpenc_lib.h"
-#include "tpenc_asc.h"
-
-
-int adtsWrite_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->protection_absent) {
-    return 0;
-  }
-  return ( FDKcrcStartReg(&pAdts->crcInfo, hBs, mBits) );
-}
-
-void adtsWrite_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->protection_absent == 0)
-  {
-    FDKcrcEndReg(&pAdts->crcInfo, hBs, reg);
-  }
-}
-
-int adtsWrite_GetHeaderBits( HANDLE_ADTS hAdts )
-{
-  int bits = 0;
-
-  if (hAdts->currentBlock == 0) {
-    /* Static and variable header bits */
-    bits = 56;
-    if (!hAdts->protection_absent) {
-      /* Add header/ single raw data block CRC bits */
-      bits += 16;
-      if (hAdts->num_raw_blocks>0) {
-        /* Add bits of raw data block position markers */
-        bits += (hAdts->num_raw_blocks)*16;
-      }
-    }
-  }
-  if (!hAdts->protection_absent && hAdts->num_raw_blocks>0) {
-    /* Add raw data block CRC bits. Not really part of the header, put they cause bit overhead to be accounted. */
-    bits += 16;
-  }
-
-  hAdts->headerBits = bits;
-
-  return bits;
-}
-
-INT adtsWrite_Init(HANDLE_ADTS hAdts, CODER_CONFIG *config)
-{
-  /* Sanity checks */
-  if ( config->nSubFrames < 1
-    || config->nSubFrames > 4
-    || (int)config->aot > 4
-    || (int)config->aot < 1 ) {
-    return -1;
-  }
-
-  /* fixed header */
-  if (config->flags & CC_MPEG_ID) {
-    hAdts->mpeg_id = 0; /* MPEG 4 */
-  } else {
-    hAdts->mpeg_id = 1; /* MPEG 2 */
-  }
-  hAdts->layer=0;
-  hAdts->protection_absent = ! (config->flags & CC_PROTECTION);
-  hAdts->profile = ((int)config->aot) - 1;
-  hAdts->sample_freq_index = getSamplingRateIndex(config->samplingRate);
-  hAdts->sample_freq = config->samplingRate;
-  hAdts->private_bit=0;
-  hAdts->channel_mode = config->channelMode;
-  hAdts->original=0;
-  hAdts->home=0;
-  /* variable header */
-  hAdts->copyright_id=0;
-  hAdts->copyright_start=0;
-
-  hAdts->num_raw_blocks=config->nSubFrames-1; /* 0 means 1 raw data block */
-
-  FDKcrcInit(&hAdts->crcInfo, 0x8005, 0xFFFF, 16);
-
-  hAdts->currentBlock = 0;
-
-
-  return 0;
-}
-
-int adtsWrite_EncodeHeader(HANDLE_ADTS hAdts,
-                                 HANDLE_FDK_BITSTREAM hBitStream,
-                                 int buffer_fullness,
-                                 int frame_length)
-{
-  INT crcIndex = 0;
-
-
-  hAdts->headerBits = adtsWrite_GetHeaderBits(hAdts);
-
-  FDK_ASSERT(((frame_length+hAdts->headerBits)/8)<0x2000);      /*13 bit*/
-  FDK_ASSERT(buffer_fullness<0x800);    /* 11 bit   */
-
-  if (!hAdts->protection_absent) {
-    FDKcrcReset(&hAdts->crcInfo);
-  }
-
-  if (hAdts->currentBlock == 0) {
-    FDKresetBitbuffer(hBitStream, BS_WRITER);
-  }
-
-  hAdts->subFrameStartBit = FDKgetValidBits(hBitStream);
-
-  /* Skip new header if this is raw data block 1..n */
-  if (hAdts->currentBlock == 0)
-  {
-    FDKresetBitbuffer(hBitStream, BS_WRITER);
-
-    if (hAdts->num_raw_blocks == 0) {
-      crcIndex = adtsWrite_CrcStartReg(hAdts, hBitStream, 0);
-    }
-
-    /* fixed header */
-    FDKwriteBits(hBitStream, 0xFFF, 12);
-    FDKwriteBits(hBitStream, hAdts->mpeg_id, 1);
-    FDKwriteBits(hBitStream, hAdts->layer, 2);
-    FDKwriteBits(hBitStream, hAdts->protection_absent, 1);
-    FDKwriteBits(hBitStream, hAdts->profile, 2);
-    FDKwriteBits(hBitStream, hAdts->sample_freq_index, 4);
-    FDKwriteBits(hBitStream, hAdts->private_bit, 1);
-    FDKwriteBits(hBitStream, getChannelConfig(hAdts->channel_mode), 3);
-    FDKwriteBits(hBitStream, hAdts->original, 1);
-    FDKwriteBits(hBitStream, hAdts->home, 1);
-    /* variable header */
-    FDKwriteBits(hBitStream, hAdts->copyright_id, 1);
-    FDKwriteBits(hBitStream, hAdts->copyright_start, 1);
-    FDKwriteBits(hBitStream, (frame_length + hAdts->headerBits)>>3, 13);
-    FDKwriteBits(hBitStream, buffer_fullness, 11);
-    FDKwriteBits(hBitStream, hAdts->num_raw_blocks, 2);
-
-    if (!hAdts->protection_absent) {
-      int i;
-
-      /* End header CRC portion for single raw data block and write dummy zero values for unknown fields. */
-      if (hAdts->num_raw_blocks == 0) {
-        adtsWrite_CrcEndReg(hAdts, hBitStream, crcIndex);
-      } else {
-        for (i=0; i<hAdts->num_raw_blocks; i++) {
-          FDKwriteBits(hBitStream, 0, 16);
-        }
-      }
-      FDKwriteBits(hBitStream, 0, 16);
-    }
-  } /* End of ADTS header */
-
-  return 0;
-}
-
-void adtsWrite_EndRawDataBlock(HANDLE_ADTS hAdts,
-                          HANDLE_FDK_BITSTREAM hBs,
-                          int *pBits)
-{
-  if (!hAdts->protection_absent) {
-    FDK_BITSTREAM bsWriter;
-
-    FDKinitBitStream(&bsWriter, hBs->hBitBuf.Buffer, hBs->hBitBuf.bufSize, 0, BS_WRITER);
-    FDKpushFor(&bsWriter, 56);
-
-    if (hAdts->num_raw_blocks == 0) {
-      FDKwriteBits(&bsWriter, FDKcrcGetCRC(&hAdts->crcInfo), 16);
-    } else {
-      int distance;
-
-      /* Write CRC of current raw data block */
-      FDKwriteBits(hBs, FDKcrcGetCRC(&hAdts->crcInfo), 16);
-
-      /* Write distance to current data block */
-      if (hAdts->currentBlock < hAdts->num_raw_blocks) {
-        FDKpushFor(&bsWriter, hAdts->currentBlock*16);
-        distance = FDKgetValidBits(hBs) - (56 + (hAdts->num_raw_blocks)*16 + 16);
-        FDKwriteBits(&bsWriter, distance>>3, 16);
-      }
-    }
-    FDKsyncCache(&bsWriter);
-  }
-
-  /* Write total frame lenth for multiple raw data blocks and header CRC */
-  if (hAdts->num_raw_blocks > 0 && hAdts->currentBlock == hAdts->num_raw_blocks) {
-    FDK_BITSTREAM bsWriter;
-    int crcIndex = 0;
-
-    FDKinitBitStream(&bsWriter, hBs->hBitBuf.Buffer, hBs->hBitBuf.bufSize, 0, BS_WRITER);
-
-    if (!hAdts->protection_absent) {
-      FDKcrcReset(&hAdts->crcInfo);
-      crcIndex = FDKcrcStartReg(&hAdts->crcInfo, &bsWriter, 0);
-    }
-    /* Write total frame length */
-    FDKpushFor(&bsWriter, 56-28+2);
-    FDKwriteBits(&bsWriter, FDKgetValidBits(hBs)>>3, 13);
-
-    /* Write header CRC */
-    if (!hAdts->protection_absent) {
-      FDKpushFor(&bsWriter, 11+2 + (hAdts->num_raw_blocks)*16);
-      FDKcrcEndReg(&hAdts->crcInfo, &bsWriter, crcIndex);
-      FDKwriteBits(&bsWriter, FDKcrcGetCRC(&hAdts->crcInfo), 16);
-    }
-    FDKsyncCache(&bsWriter);
-  }
-
-  /* Correct *pBits to reflect the amount of bits of the current subframe */
-  *pBits -= hAdts->subFrameStartBit;
-  if (!hAdts->protection_absent && hAdts->num_raw_blocks > 0) {
-    /* Fixup CRC bits, since they come after each raw data block */
-    *pBits += 16;
-  }
-  hAdts->currentBlock++;
-}
-
diff --git a/libMpegTPEnc/src/tpenc_adts.h b/libMpegTPEnc/src/tpenc_adts.h
deleted file mode 100644
index c12c7c7..0000000
--- a/libMpegTPEnc/src/tpenc_adts.h
+++ /dev/null
@@ -1,218 +0,0 @@
-
-/* -----------------------------------------------------------------------------------------------------------
-Software License for The Fraunhofer FDK AAC Codec Library for Android
-
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
-  All rights reserved.
-
- 1.    INTRODUCTION
-The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
-the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
-This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
-
-AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
-audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
-independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
-of the MPEG specifications.
-
-Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
-may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
-individually for the purpose of encoding or decoding bit streams in products that are compliant with
-the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
-these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
-software may already be covered under those patent licenses when it is used for those licensed purposes only.
-
-Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
-are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
-applications information and documentation.
-
-2.    COPYRIGHT LICENSE
-
-Redistribution and use in source and binary forms, with or without modification, are permitted without
-payment of copyright license fees provided that you satisfy the following conditions:
-
-You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
-your modifications thereto in source code form.
-
-You must retain the complete text of this software license in the documentation and/or other materials
-provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
-You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
-modifications thereto to recipients of copies in binary form.
-
-The name of Fraunhofer may not be used to endorse or promote products derived from this library without
-prior written permission.
-
-You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
-software or your modifications thereto.
-
-Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
-and the date of any change. For modified versions of the FDK AAC Codec, the term
-"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
-"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
-
-3.    NO PATENT LICENSE
-
-NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
-ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
-respect to this software.
-
-You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
-by appropriate patent licenses.
-
-4.    DISCLAIMER
-
-This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
-"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
-of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
-CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
-including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
-or business interruption, however caused and on any theory of liability, whether in contract, strict
-liability, or tort (including negligence), arising in any way out of the use of this software, even if
-advised of the possibility of such damage.
-
-5.    CONTACT INFORMATION
-
-Fraunhofer Institute for Integrated Circuits IIS
-Attention: Audio and Multimedia Departments - FDK AAC LL
-Am Wolfsmantel 33
-91058 Erlangen, Germany
-
-www.iis.fraunhofer.de/amm
-amm-info@iis.fraunhofer.de
------------------------------------------------------------------------------------------------------------ */
-
-/******************************** MPEG Audio Encoder **************************
-
-   Initial author:       Alex Groeschel
-   contents/description: ADTS Transport writer
-
-******************************************************************************/
-
-#ifndef TPENC_ADTS_H
-#define TPENC_ADTS_H
-
-
-
-#include "tp_data.h"
-
-#include "FDK_crc.h"
-
-typedef struct {
-  INT sample_freq;
-  CHANNEL_MODE channel_mode;
-  UCHAR decoderCanDoMpeg4;
-  UCHAR mpeg_id;
-  UCHAR layer;
-  UCHAR protection_absent;
-  UCHAR profile;
-  UCHAR sample_freq_index;
-  UCHAR private_bit;
-  UCHAR original;
-  UCHAR home;
-  UCHAR copyright_id;
-  UCHAR copyright_start;
-  USHORT frame_length;
-  UCHAR num_raw_blocks;
-  UCHAR BufferFullnesStartFlag;
-  int headerBits;                /*!< Header bit demand for the current raw data block */
-  int currentBlock;              /*!< Index of current raw data block */
-  int subFrameStartBit;          /*!< Bit position where the current raw data block begins */
-  FDK_CRCINFO crcInfo;
-} STRUCT_ADTS;
-
-typedef STRUCT_ADTS *HANDLE_ADTS;
-
-/**
- * \brief Initialize ADTS data structure
- *
- * \param hAdts ADTS data handle
- * \param config a valid CODER_CONFIG struct from where the required
- *        information for the ADTS header is extrated from
- *
- * \return 0 in case of success.
- */
-INT adtsWrite_Init(
-        HANDLE_ADTS   hAdts,
-        CODER_CONFIG *config
-        );
-
-/**
- * \brief Get the total bit overhead caused by ADTS
- *
- * \hAdts handle to ADTS data
- *
- * \return Amount of additional bits required for the current raw data block
- */
-int adtsWrite_GetHeaderBits( HANDLE_ADTS hAdts );
-
-/**
- * \brief Write an ADTS header into the given bitstream. May not write a header
- *        in case of multiple raw data blocks.
- *
- * \param hAdts ADTS data handle
- * \param hBitStream bitstream handle into which the ADTS may be written into
- * \param buffer_fullness the buffer fullness value for the ADTS header
- * \param the current raw data block length
- *
- * \return 0 in case of success.
- */
-INT adtsWrite_EncodeHeader(
-        HANDLE_ADTS          hAdts,
-        HANDLE_FDK_BITSTREAM hBitStream,
-        int                  bufferFullness,
-        int                  frame_length
-        );
-/**
- * \brief Finish a ADTS raw data block
- *
- * \param hAdts ADTS data handle
- * \param hBs bitstream handle into which the ADTS may be written into
- * \param pBits a pointer to a integer holding the current bitstream buffer bit count,
- *              which is corrected to the current raw data block boundary.
- *
- */
-void adtsWrite_EndRawDataBlock(
-        HANDLE_ADTS          hAdts,
-        HANDLE_FDK_BITSTREAM hBs,
-        int                 *bits
-        );
-
-
-/**
- * \brief Start 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 of which the CRC region ends
- * \param mBits limit of number of bits to be considered for the requested CRC region
- *
- * \return ID for the created region, -1 in case of an error
- */
-int adtsWrite_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 of which the CRC region ends
- * \param reg a CRC region ID returned previously by adtsWrite_CrcStartReg()
- */
-void adtsWrite_CrcEndReg(
-        HANDLE_ADTS          pAdts,
-        HANDLE_FDK_BITSTREAM hBs,
-        int                  reg
-        );
-
-
-
-
-#endif /* TPENC_ADTS_H */
-
diff --git a/libMpegTPEnc/src/tpenc_asc.cpp b/libMpegTPEnc/src/tpenc_asc.cpp
deleted file mode 100644
index bc4302e..0000000
--- a/libMpegTPEnc/src/tpenc_asc.cpp
+++ /dev/null
@@ -1,576 +0,0 @@
-
-/* -----------------------------------------------------------------------------------------------------------
-Software License for The Fraunhofer FDK AAC Codec Library for Android
-
-© Copyright  1995 - 2013 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 Encoder  **************************
-
-   Author(s):
-   Description:
-
-******************************************************************************/
-
-#include "tp_data.h"
-
-#include "tpenc_lib.h"
-#include "tpenc_asc.h"
-#include "FDK_bitstream.h"
-#include "genericStds.h"
-
-#define PCE_MAX_ELEMENTS 8
-
-/**
- *  Describe a PCE based on placed channel elements and element type sequence.
- */
-typedef struct {
-
-    UCHAR    num_front_channel_elements;     /*!< Number of front channel elements. */
-    UCHAR    num_side_channel_elements;      /*!< Number of side channel elements. */
-    UCHAR    num_back_channel_elements;      /*!< Number of back channel elements. */
-    UCHAR    num_lfe_channel_elements;       /*!< Number of lfe channel elements. */
-    MP4_ELEMENT_ID el_list[PCE_MAX_ELEMENTS];/*!< List contains sequence describing the elements
-                                                  in present channel mode. (MPEG order) */
-} PCE_CONFIGURATION;
-
-
-/**
- *  Map an incoming channel mode to a existing PCE configuration entry.
- */
-typedef struct {
-
-    CHANNEL_MODE        channel_mode;        /*!< Present channel mode. */
-    PCE_CONFIGURATION   pce_configuration;   /*!< Program config element description. */
-
-} CHANNEL_CONFIGURATION;
-
-
-/**
- * \brief Table contains all supported channel modes and according PCE configuration description.
- *
- * The number of channel element parameter describes the kind of consecutively elements.
- * E.g. MODE_1_2_2_2_1 means:
- *                          - First 3 elements (SCE,CPE,CPE) are front channel elements.
- *                          - Next element (CPE) is a back channel element.
- *                          - Last element (LFE) is a lfe channel element.
- */
-static const CHANNEL_CONFIGURATION pceConfigTab[] =
-{
-  { MODE_1,                        {  1, 0, 0, 0, { ID_SCE,  ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE } } },
-  { MODE_2,                        {  1, 0, 0, 0, { ID_CPE,  ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE } } },
-  { MODE_1_2,                      {  2, 0, 0, 0, { ID_SCE,  ID_CPE,  ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE } } },
-  { MODE_1_2_1,                    {  2, 0, 1, 0, { ID_SCE,  ID_CPE,  ID_SCE,  ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE } } },
-  { MODE_1_2_2,                    {  2, 0, 1, 0, { ID_SCE,  ID_CPE,  ID_CPE,  ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE } } },
-  { MODE_1_2_2_1,                  {  2, 0, 1, 1, { ID_SCE,  ID_CPE,  ID_CPE,  ID_LFE, ID_NONE, ID_NONE, ID_NONE, ID_NONE } } },
-  { MODE_1_2_2_2_1,                {  3, 0, 1, 1, { ID_SCE,  ID_CPE,  ID_CPE,  ID_CPE,  ID_LFE,  ID_NONE, ID_NONE, ID_NONE } } },
-
-
-  { MODE_1_1,                      {  2, 0, 0, 0, { ID_SCE,  ID_SCE,  ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE } } },
-  { MODE_1_1_1_1,                  {  2, 2, 0, 0, { ID_SCE,  ID_SCE,  ID_SCE,  ID_SCE,  ID_NONE, ID_NONE, ID_NONE, ID_NONE } } },
-  { MODE_1_1_1_1_1_1,              {  2, 2, 2, 0, { ID_SCE,  ID_SCE,  ID_SCE,  ID_SCE,  ID_SCE,  ID_SCE,  ID_NONE, ID_NONE } } },
-  { MODE_1_1_1_1_1_1_1_1,          {  3, 2, 3, 0, { ID_SCE,  ID_SCE,  ID_SCE,  ID_SCE,  ID_SCE,  ID_SCE,  ID_SCE,  ID_SCE } } },
-
-  { MODE_2_2,                      {  1, 0, 1, 0, { ID_CPE,  ID_CPE,  ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE } } },
-  { MODE_2_2_2,                    {  1, 1, 1, 0, { ID_CPE,  ID_CPE,  ID_CPE,  ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE } } },
-  { MODE_2_2_2_2,                  {  4, 0, 0, 0, { ID_CPE,  ID_CPE,  ID_CPE,  ID_CPE,  ID_NONE, ID_NONE, ID_NONE, ID_NONE } } },
-
-  { MODE_2_1,                      {  1, 0, 1, 0, { ID_CPE,  ID_SCE,  ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE } } },
-
-  { MODE_7_1_REAR_SURROUND,        {  2, 0, 2, 1, { ID_SCE,  ID_CPE,  ID_CPE,  ID_CPE,  ID_LFE,  ID_NONE, ID_NONE, ID_NONE } } },
-  { MODE_7_1_FRONT_CENTER,         {  3, 0, 1, 1, { ID_SCE,  ID_CPE,  ID_CPE,  ID_CPE,  ID_LFE,  ID_NONE, ID_NONE, ID_NONE } } },
-
-};
-
-
-/**
- * \brief  Get program config element description for existing channel mode.
- *
- * \param channel_mode          Current channel mode.
- *
- * \return
- *          - Pointer to PCE_CONFIGURATION entry, on success.
- *          - NULL, on failure.
- */
-static const PCE_CONFIGURATION* getPceEntry(
-        const CHANNEL_MODE channel_mode
-        )
-{
-  UINT i;
-  const PCE_CONFIGURATION *pce_config = NULL;
-
-  for (i=0; i < (sizeof(pceConfigTab)/sizeof(CHANNEL_CONFIGURATION)); i++) {
-    if (pceConfigTab[i].channel_mode == channel_mode) {
-      pce_config = &pceConfigTab[i].pce_configuration;
-    }
-  }
-
-  return pce_config;
-}
-
-int getChannelConfig( CHANNEL_MODE channel_mode )
-{
-  INT chan_config = 0;
-
-  switch(channel_mode) {
-    case MODE_1:         chan_config = 1; break;
-    case MODE_2:         chan_config = 2; break;
-    case MODE_1_2:       chan_config = 3; break;
-    case MODE_1_2_1:     chan_config = 4; break;
-    case MODE_1_2_2:     chan_config = 5; break;
-    case MODE_1_2_2_1:   chan_config = 6; break;
-    case MODE_1_2_2_2_1: chan_config = 7; break;
-
-    default:             chan_config = 0;
-  }
-
-  return chan_config;
-}
-
-CHANNEL_MODE transportEnc_GetChannelMode( int noChannels )
-{
-  CHANNEL_MODE chMode;
-
-  if (noChannels <= 8 && noChannels > 0)
-    chMode = (CHANNEL_MODE)((noChannels == 8) ? 7 : noChannels); /* see : iso/mpeg4 v1 audio subpart1*/
-  else
-    chMode = MODE_UNKNOWN;
-
-  return chMode;
-}
-
-#ifdef TP_PCE_ENABLE
-int transportEnc_writePCE(HANDLE_FDK_BITSTREAM hBs,
-                          CHANNEL_MODE channelMode,
-                          INT sampleRate,
-                          int instanceTagPCE,
-                          int profile,
-                          int matrixMixdownA,
-                          int pseudoSurroundEnable,
-                          UINT alignAnchor)
-{
-  int sampleRateIndex, i;
-  const PCE_CONFIGURATION* config = NULL;
-  const MP4_ELEMENT_ID* pEl_list = NULL;
-  UCHAR cpeCnt=0, sceCnt=0, lfeCnt=0;
-
-  sampleRateIndex = getSamplingRateIndex(sampleRate);
-  if (sampleRateIndex == 15) {
-    return -1;
-  }
-
-  if ((config=getPceEntry(channelMode))==NULL) {
-    return -1;
-  }
-
-  /* Pointer to first element in element list. */
-  pEl_list = &config->el_list[0];
-
-  FDKwriteBits(hBs, instanceTagPCE,  4);                        /* Element instance tag */
-  FDKwriteBits(hBs, profile,         2);                        /* Object type */
-  FDKwriteBits(hBs, sampleRateIndex, 4);                        /* Sample rate index*/
-
-  FDKwriteBits(hBs, config->num_front_channel_elements, 4);     /* Front channel Elements */
-  FDKwriteBits(hBs, config->num_side_channel_elements , 4);     /* No Side Channel Elements */
-  FDKwriteBits(hBs, config->num_back_channel_elements , 4);     /* No Back channel Elements */
-  FDKwriteBits(hBs, config->num_lfe_channel_elements  , 2);     /* No Lfe channel elements */
-
-  FDKwriteBits(hBs, 0, 3);                                      /* No assoc data elements */
-  FDKwriteBits(hBs, 0, 4);                                      /* No valid cc elements */
-  FDKwriteBits(hBs, 0, 1);                                      /* Mono mixdown present */
-  FDKwriteBits(hBs, 0, 1);                                      /* Stereo mixdown present */
-
-  if ( matrixMixdownA!=0 && ((channelMode==MODE_1_2_2)||(channelMode==MODE_1_2_2_1)) ) {
-      FDKwriteBits(hBs, 1, 1);                                  /* Matrix mixdown present */
-      FDKwriteBits(hBs, (matrixMixdownA-1)&0x3, 2);             /* matrix_mixdown_idx */
-      FDKwriteBits(hBs, (pseudoSurroundEnable)?1:0, 1);         /* pseudo_surround_enable */
-  }
-  else {
-      FDKwriteBits(hBs, 0, 1);                                  /* Matrix mixdown not present */
-  }
-
-  for(i=0; i<config->num_front_channel_elements; i++) {
-      UCHAR isCpe = (*pEl_list++==ID_CPE) ? 1 : 0;
-      UCHAR tag   = (isCpe) ? cpeCnt++ : sceCnt++;
-      FDKwriteBits(hBs, isCpe, 1);                              /* Front channel Elements is CPE? */
-      FDKwriteBits(hBs, tag, 4);                                /* Front channel Instance Tag.*/
-  }
-  for(i=0; i<config->num_side_channel_elements; i++) {
-      UCHAR isCpe = (*pEl_list++==ID_CPE) ? 1 : 0;
-      UCHAR tag   = (isCpe) ? cpeCnt++ : sceCnt++;
-      FDKwriteBits(hBs, isCpe, 1);                              /* Front channel Elements is CPE? */
-      FDKwriteBits(hBs, tag, 4);                                /* Front channel Instance Tag.*/
-  }
-  for(i=0; i<config->num_back_channel_elements; i++) {
-      UCHAR isCpe = (*pEl_list++==ID_CPE) ? 1 : 0;
-      UCHAR tag   = (isCpe) ? cpeCnt++ : sceCnt++;
-      FDKwriteBits(hBs, isCpe, 1);                              /* Front channel Elements is CPE? */
-      FDKwriteBits(hBs, tag, 4);                                /* Front channel Instance Tag.*/
-  }
-  for(i=0; i<config->num_lfe_channel_elements; i++) {
-      FDKwriteBits(hBs, lfeCnt++, 4);                           /* LFE channel Instance Tag. */
-  }
-
-  /* - num_valid_cc_elements always 0.
-     - num_assoc_data_elements always 0. */
-
-  /* Byte alignment: relative to alignAnchor
-       ADTS: align with respect to the first bit of the raw_data_block()
-       ADIF: align with respect to the first bit of the header
-       LATM: align with respect to the first bit of the ASC */
-  FDKbyteAlign(hBs, alignAnchor);                               /* Alignment */
-
-  FDKwriteBits(hBs, 0 ,8);                                      /* Do no write any comment. */
-
-  /* - comment_field_bytes always 0. */
-
-  return 0;
-}
-
-int transportEnc_GetPCEBits(CHANNEL_MODE channelMode,
-                            int matrixMixdownA,
-                            int bits)
-{
-  const PCE_CONFIGURATION* config = NULL;
-
-  if ((config=getPceEntry(channelMode))==NULL) {
-    return -1;  /* unsupported channelmapping */
-  }
-
-  bits += 4 + 2 + 4;        /* Element instance tag  + Object type + Sample rate index */
-  bits += 4 + 4 + 4 + 2;    /* No (front + side + back + lfe channel) elements */
-  bits += 3 + 4;            /* No (assoc data + valid cc) elements */
-  bits += 1 + 1 + 1 ;       /* Mono + Stereo + Matrix mixdown present */
-
-  if ( matrixMixdownA!=0 && ((channelMode==MODE_1_2_2)||(channelMode==MODE_1_2_2_1)) ) {
-    bits +=3;               /* matrix_mixdown_idx + pseudo_surround_enable */
-  }
-
-  bits += (1+4) * (INT)config->num_front_channel_elements;
-  bits += (1+4) * (INT)config->num_side_channel_elements;
-  bits += (1+4) * (INT)config->num_back_channel_elements;
-  bits +=   (4) * (INT)config->num_lfe_channel_elements;
-
-  /* - num_valid_cc_elements always 0.
-     - num_assoc_data_elements always 0. */
-
-  if ((bits%8) != 0) {
-    bits += (8 - (bits%8)); /* Alignment */
-  }
-
-  bits += 8;                /* Comment field  bytes */
-
-  /* - comment_field_bytes alwys 0. */
-
-  return bits;
-}
-#endif /* TP_PCE_ENABLE */
-
-static void writeAot(HANDLE_FDK_BITSTREAM hBitstreamBuffer, AUDIO_OBJECT_TYPE aot)
-{
-    int tmp = (int) aot;
-
-    if (tmp > 31) {
-        FDKwriteBits( hBitstreamBuffer, AOT_ESCAPE, 5 );
-        FDKwriteBits( hBitstreamBuffer, tmp-32, 6 );   /* AudioObjectType */
-    } else {
-        FDKwriteBits( hBitstreamBuffer, tmp, 5 );
-    }
-}
-
-static void writeSampleRate(HANDLE_FDK_BITSTREAM hBitstreamBuffer, int sampleRate)
-{
-  int sampleRateIndex = getSamplingRateIndex(sampleRate);
-
-  FDKwriteBits( hBitstreamBuffer, sampleRateIndex, 4 );
-  if( sampleRateIndex == 15 ) {
-    FDKwriteBits( hBitstreamBuffer, sampleRate, 24 );
-  }
-}
-
-#ifdef TP_GA_ENABLE
-static
-int transportEnc_writeGASpecificConfig(
-                                        HANDLE_FDK_BITSTREAM asc,
-                                        CODER_CONFIG *config,
-                                        int          extFlg,
-                                        UINT         alignAnchor
-                                       )
-{
-  int aot = config->aot;
-  int samplesPerFrame = config->samplesPerFrame;
-
-  /* start of GASpecificConfig according to ISO/IEC 14496-3 Subpart 4, 4.4.1 */
-  FDKwriteBits( asc, ((samplesPerFrame==960 || samplesPerFrame==480)?1:0), 1);  /* frameLengthFlag: 1 for a 960/480 (I)MDCT, 0 for a 1024/512 (I)MDCT*/
-  FDKwriteBits( asc, 0, 1);  /* dependsOnCoreCoder: Sampling Rate Coder Specific, see in ISO/IEC 14496-3 Subpart 4, 4.4.1 */
-  FDKwriteBits( asc, extFlg, 1 ); /* Extension Flag: Shall be 1 for aot = 17,19,20,21,22,23 */
-
-  /* Write PCE if channel config is not 1-7 */
-  if (getChannelConfig(config->channelMode) == 0) {
-      transportEnc_writePCE(asc, config->channelMode, config->samplingRate, 0, 1, config->matrixMixdownA, (config->flags&CC_PSEUDO_SURROUND)?1:0, alignAnchor);
-  }
-  if (extFlg) {
-    if (aot == AOT_ER_BSAC) {
-      FDKwriteBits( asc, config->BSACnumOfSubFrame, 5 ); /* numOfSubFrame */
-      FDKwriteBits( asc, config->BSAClayerLength, 11 );  /* layer_length */
-    }
-    if ((aot == AOT_ER_AAC_LC)   || (aot == AOT_ER_AAC_LTP)  ||
-        (aot == AOT_ER_AAC_SCAL) || (aot == AOT_ER_AAC_LD))
-    {
-      FDKwriteBits( asc, (config->flags & CC_VCB11) ? 1 : 0, 1 ); /* aacSectionDataResillienceFlag */
-      FDKwriteBits( asc, (config->flags & CC_RVLC) ? 1 : 0,  1 ); /* aacScaleFactorDataResillienceFlag */
-      FDKwriteBits( asc, (config->flags & CC_HCR) ? 1 : 0,   1 ); /* aacSpectralDataResillienceFlag */
-    }
-    FDKwriteBits( asc, 0, 1 ); /* extensionFlag3: reserved. Shall be '0' */
-  }
-  return 0;
-}
-#endif /* TP_GA_ENABLE */
-
-#ifdef TP_ELD_ENABLE
-
-static
-int transportEnc_writeELDSpecificConfig(
-                                         HANDLE_FDK_BITSTREAM hBs,
-                                         CODER_CONFIG *config,
-                                         int        epConfig,
-                                         CSTpCallBacks *cb
-                                        )
-{
-  /* ELD specific config */
-  if (config->channelMode == MODE_1_1) {
-    return -1;
-  }
-  FDKwriteBits(hBs, (config->samplesPerFrame == 480) ? 1 : 0, 1);
-
-  FDKwriteBits(hBs, (config->flags & CC_VCB11 ) ? 1:0, 1);
-  FDKwriteBits(hBs, (config->flags & CC_RVLC ) ? 1:0, 1);
-  FDKwriteBits(hBs, (config->flags & CC_HCR  ) ? 1:0, 1);
-
-  FDKwriteBits(hBs, (config->flags & CC_SBR) ? 1:0, 1); /* SBR header flag */
-  if ( (config->flags & CC_SBR) ) {
-    FDKwriteBits(hBs, (config->samplingRate == config->extSamplingRate) ? 0:1, 1); /* Samplerate Flag */
-    FDKwriteBits(hBs, (config->flags & CC_SBRCRC) ? 1:0, 1); /* SBR CRC flag*/
-
-    if (cb->cbSbr != NULL) {
-      const PCE_CONFIGURATION *pPce;
-      int e;
-
-      pPce = getPceEntry(config->channelMode);
-
-      for (e=0; e<PCE_MAX_ELEMENTS && pPce->el_list[e] != ID_NONE; e++  ) {
-        if ( (pPce->el_list[e] == ID_SCE) || (pPce->el_list[e] == ID_CPE) ) {
-          cb->cbSbr(cb->cbSbrData, hBs, 0, 0, 0, config->aot, pPce->el_list[e], e);
-        }
-      }
-    }
-  }
-
-  FDKwriteBits(hBs, 0, 4); /* ELDEXT_TERM */
-
-  return 0;
-}
-#endif /* TP_ELD_ENABLE */
-
-
-int transportEnc_writeASC (
-                            HANDLE_FDK_BITSTREAM asc,
-                            CODER_CONFIG *config,
-                            CSTpCallBacks *cb
-                           )
-{
-  UINT extFlag = 0;
-  int err;
-  int epConfig = 0;
-
-  /* Required for the PCE. */
-  UINT alignAnchor = FDKgetValidBits(asc);
-
-  /* Extension Flag: Shall be 1 for aot = 17,19,20,21,22,23,39 */
-  switch (config->aot) {
-    case AOT_ER_AAC_LC:
-    case AOT_ER_AAC_LTP:
-    case AOT_ER_AAC_SCAL:
-    case AOT_ER_TWIN_VQ:
-    case AOT_ER_BSAC:
-    case AOT_ER_AAC_LD:
-    case AOT_ER_AAC_ELD:
-    case AOT_USAC:
-        extFlag = 1;
-        break;
-    default:
-        break;
-  }
-
-  if (config->sbrSignaling==SIG_EXPLICIT_HIERARCHICAL && config->sbrPresent)
-    writeAot(asc, config->extAOT);
-  else
-    writeAot(asc, config->aot);
-
-  {
-    writeSampleRate(asc, config->samplingRate);
-  }
-
-  /* Try to guess a reasonable channel mode if not given */
-  if (config->channelMode == MODE_INVALID) {
-    config->channelMode = transportEnc_GetChannelMode(config->noChannels);
-    if (config->channelMode == MODE_INVALID)
-      return -1;
-  }
-
-  FDKwriteBits( asc, getChannelConfig(config->channelMode), 4 );
-
-  if (config->sbrSignaling==SIG_EXPLICIT_HIERARCHICAL && config->sbrPresent) {
-    writeSampleRate(asc, config->extSamplingRate);
-    writeAot(asc, config->aot);
-  }
-
-  switch (config->aot) {
-#ifdef TP_GA_ENABLE
-    case AOT_AAC_MAIN:
-    case AOT_AAC_LC:
-    case AOT_AAC_SSR:
-    case AOT_AAC_LTP:
-    case AOT_AAC_SCAL:
-    case AOT_TWIN_VQ:
-    case AOT_ER_AAC_LC:
-    case AOT_ER_AAC_LTP:
-    case AOT_ER_AAC_SCAL:
-    case AOT_ER_TWIN_VQ:
-    case AOT_ER_BSAC:
-    case AOT_ER_AAC_LD:
-      err = transportEnc_writeGASpecificConfig(asc, config, extFlag, alignAnchor);
-      if (err)
-        return err;
-      break;
-#endif /* TP_GA_ENABLE */
-#ifdef TP_ELD_ENABLE
-    case AOT_ER_AAC_ELD:
-      err = transportEnc_writeELDSpecificConfig(asc, config, epConfig, cb);
-      if (err)
-        return err;
-      break;
-#endif /* TP_ELD_ENABLE */
-    default:
-      return -1;
-  }
-
-  switch (config->aot) {
-    case AOT_ER_AAC_LC:
-    case AOT_ER_AAC_LTP:
-    case AOT_ER_AAC_SCAL:
-    case AOT_ER_TWIN_VQ:
-    case AOT_ER_BSAC:
-    case AOT_ER_AAC_LD:
-    case AOT_ER_CELP:
-    case AOT_ER_HVXC:
-    case AOT_ER_HILN:
-    case AOT_ER_PARA:
-    case AOT_ER_AAC_ELD:
-      FDKwriteBits( asc, 0, 2 ); /* epconfig 0 */
-      break;
-    default:
-      break;
-  }
-
-  /* backward compatible explicit signaling of extension AOT */
-  if (config->sbrSignaling==SIG_EXPLICIT_BW_COMPATIBLE)
-  {
-    TP_ASC_EXTENSION_ID ascExtId = ASCEXT_UNKOWN;
-
-    if (config->sbrPresent) {
-      ascExtId=ASCEXT_SBR;
-      FDKwriteBits( asc, ascExtId, 11 );
-      writeAot(asc, config->extAOT);
-      FDKwriteBits( asc, 1, 1 ); /* sbrPresentFlag=1 */
-      writeSampleRate(asc, config->extSamplingRate);
-      if (config->psPresent) {
-        ascExtId=ASCEXT_PS;
-        FDKwriteBits( asc, ascExtId, 11 );
-        FDKwriteBits( asc, 1, 1 ); /* psPresentFlag=1 */
-      }
-    }
-
-  }
-
-  /* Make sure all bits are sync'ed */
-  FDKsyncCache( asc );
-
-  return 0;
-}
diff --git a/libMpegTPEnc/src/tpenc_asc.h b/libMpegTPEnc/src/tpenc_asc.h
deleted file mode 100644
index 47fe7a1..0000000
--- a/libMpegTPEnc/src/tpenc_asc.h
+++ /dev/null
@@ -1,142 +0,0 @@
-
-/* -----------------------------------------------------------------------------------------------------------
-Software License for The Fraunhofer FDK AAC Codec Library for Android
-
-© Copyright  1995 - 2013 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 Encoder  **************************
-
-   Author(s): Manuel Jander
-   Description: Audio Specific Config writer
-
-******************************************************************************/
-
-#ifndef TPENC_ASC_H
-#define TPENC_ASC_H
-
-/**
- * \brief Get channel config from channel mode.
- *
- * \param channel_mode channel mode
- *
- * \return chanel config
- */
-int getChannelConfig( CHANNEL_MODE channel_mode );
-
-/**
- * \brief Write a Program Config Element.
- *
- * \param hBs bitstream handle into which the PCE is appended
- * \param channelMode the channel mode to be used
- * \param sampleRate the sample rate
- * \param instanceTagPCE the instance tag of the Program Config Element
- * \param profile the MPEG Audio profile to be used
- * \param matrix mixdown gain
- * \param pseudo surround indication
- * \param reference bitstream position for alignment
- * \return zero on success, non-zero on failure.
- */
-int transportEnc_writePCE(
-        HANDLE_FDK_BITSTREAM hBs,
-        CHANNEL_MODE         channelMode,
-        INT                  sampleRate,
-        int                  instanceTagPCE,
-        int                  profile,
-        int                  matrixMixdownA,
-        int                  pseudoSurroundEnable,
-        UINT                 alignAnchor
-        );
-
-/**
- * \brief Get the bit count required by a Program Config Element
- *
- * \param channelMode the channel mode to be used
- * \param matrix mixdown gain
- * \param bit offset at which the PCE would start
- * \return the amount of bits required for the PCE including the given bit offset.
- */
-int transportEnc_GetPCEBits(
-        CHANNEL_MODE channelMode,
-        int          matrixMixdownA,
-        int          bits
-        );
-
-#endif /* TPENC_ASC_H */
-
diff --git a/libMpegTPEnc/src/tpenc_dab.cpp b/libMpegTPEnc/src/tpenc_dab.cpp
deleted file mode 100644
index 202fecf..0000000
--- a/libMpegTPEnc/src/tpenc_dab.cpp
+++ /dev/null
@@ -1,467 +0,0 @@
-
-/* -----------------------------------------------------------------------------------------------------------
-Software License for The Fraunhofer FDK AAC Codec Library for Android
-
-� Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur F�rderung der angewandten Forschung e.V.
-  All rights reserved.
-
- 1.    INTRODUCTION
-The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
-the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
-This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
-
-AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
-audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
-independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
-of the MPEG specifications.
-
-Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
-may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
-individually for the purpose of encoding or decoding bit streams in products that are compliant with
-the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
-these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
-software may already be covered under those patent licenses when it is used for those licensed purposes only.
-
-Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
-are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
-applications information and documentation.
-
-2.    COPYRIGHT LICENSE
-
-Redistribution and use in source and binary forms, with or without modification, are permitted without
-payment of copyright license fees provided that you satisfy the following conditions:
-
-You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
-your modifications thereto in source code form.
-
-You must retain the complete text of this software license in the documentation and/or other materials
-provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
-You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
-modifications thereto to recipients of copies in binary form.
-
-The name of Fraunhofer may not be used to endorse or promote products derived from this library without
-prior written permission.
-
-You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
-software or your modifications thereto.
-
-Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
-and the date of any change. For modified versions of the FDK AAC Codec, the term
-"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
-"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
-
-3.    NO PATENT LICENSE
-
-NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
-ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
-respect to this software.
-
-You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
-by appropriate patent licenses.
-
-4.    DISCLAIMER
-
-This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
-"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
-of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
-CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
-including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
-or business interruption, however caused and on any theory of liability, whether in contract, strict
-liability, or tort (including negligence), arising in any way out of the use of this software, even if
-advised of the possibility of such damage.
-
-5.    CONTACT INFORMATION
-
-Fraunhofer Institute for Integrated Circuits IIS
-Attention: Audio and Multimedia Departments - FDK AAC LL
-Am Wolfsmantel 33
-91058 Erlangen, Germany
-
-www.iis.fraunhofer.de/amm
-amm-info@iis.fraunhofer.de
------------------------------------------------------------------------------------------------------------ */
-
-/******************************** MPEG Audio Encoder **************************
-
-   Initial author:       serge
-   contents/description: DAB Transport Headers support
-
-******************************************************************************/
-#include <stdio.h>
-#include "FDK_audio.h"
-#include "tpenc_dab.h"
-
-
-#include "tpenc_lib.h"
-#include "tpenc_asc.h"
-
-#include "common_fix.h"
-
-int dabWrite_CrcStartReg(
-                                 HANDLE_DAB pDab,          /*!< pointer to dab stucture */
-                                 HANDLE_FDK_BITSTREAM hBs,   /*!< handle to current bit buffer structure */
-                                 int mBits                   /*!< number of bits in crc region */
-                                )
-{
-  //fprintf(stderr, "dabWrite_CrcStartReg(%p): bits in crc region=%d\n", hBs, mBits);
-  return ( FDKcrcStartReg(&pDab->crcInfo2, hBs, mBits) );
-}
-
-void dabWrite_CrcEndReg(
-                                HANDLE_DAB pDab, /*!< pointer to dab crc info stucture */
-                                HANDLE_FDK_BITSTREAM hBs,   /*!< handle to current bit buffer structure */
-                                int reg                    /*!< crc region */
-                               )
-{
-    //fprintf(stderr, "dabWrite_CrcEndReg(%p): crc region=%d\n", hBs, reg);
-    FDKcrcEndReg(&pDab->crcInfo2, hBs, reg);
-}
-
-int dabWrite_GetHeaderBits( HANDLE_DAB hDab )
-{
-  int bits = 0;
-
-  if (hDab->currentBlock == 0) {
-    /* Static and variable header bits */
-    bits += 16; //header_firecode 16
-    bits +=  8; //rfa=1, dac_rate=1, sbr_flag=1, aac_channel_mode=1, ps_flag=1, mpeg_surround_config=3
-    bits += 12 * hDab->num_raw_blocks; //au_start[1...num_aus] 12 bit AU start position markers
-
-    //4 byte alignment
-    if (hDab->dac_rate == 0 || hDab->sbr_flag == 0)
-    	bits+=4;
-    //16sbr => 16 + 5 + 3 + 12*(2-1) => 36 => 40 bits 5
-    //24sbr => 16 + 5 + 3 + 12*(3-1) => 48 ok         6
-    //32sbr => 16 + 5 + 3 + 12*(4-1) => 60 => 64 bits 8
-    //48sbr => 16 + 5 + 3 + 12*(6-1) => 84 => 88 bits 11
-  }
-
-  /* Add raw data block CRC bits. Not really part of the header, put they cause bit overhead to be accounted. */
-  bits += 16;
-
-
-  return bits;
-}
-
-
-int dabWrite_CountTotalBitDemandHeader( HANDLE_DAB hDab, unsigned int streamDataLength )
-{
-	//fprintf(stderr, "streamDataLength=%d (%d bytes)\n", streamDataLength, streamDataLength >> 3);
-	return dabWrite_GetHeaderBits(hDab);
-}
-
-
-INT dabWrite_Init(HANDLE_DAB hDab, CODER_CONFIG *config)
-{
-  /* Sanity checks */
-	if((int)config->aot > 4
-    || (int)config->aot < 1 ) {
-    return -1;
-  }
-
-  /* Sanity checks DAB-specific */
-  if ( !(config->nSubFrames == 2 && config->samplingRate == 16000 && (config->flags & CC_SBR)) &&
-	   !(config->nSubFrames == 3 && config->samplingRate == 24000 && (config->flags & CC_SBR)) &&
-	   !(config->nSubFrames == 4 && config->samplingRate == 32000) &&
-	   !(config->nSubFrames == 6 && config->samplingRate == 48000)) {
-	  return -1;
-  }
-
-  hDab->dac_rate = 0;
-  hDab->aac_channel_mode=0;
-  hDab->sbr_flag = 0;
-  hDab->ps_flag = 0;
-  hDab->mpeg_surround_config=0;
-  hDab->subchannels_num=config->bitRate/8000;
-
-
-  if(config->samplingRate == 24000 || config->samplingRate == 48000)
-	  hDab->dac_rate = 1;
-
-  if (config->extAOT==AOT_SBR || config->extAOT == AOT_PS)
-	  hDab->sbr_flag = 1;
-
-  if(config->extAOT == AOT_PS)
-	  hDab->ps_flag = 1;
-
-
-  if(config->channelMode == MODE_2)
-	  hDab->aac_channel_mode =  1;
-
-  //fprintf(stderr, "hDab->dac_rate=%d\n", hDab->dac_rate);
-  //fprintf(stderr, "hDab->sbr_flag=%d\n", hDab->sbr_flag);
-  //fprintf(stderr, "hDab->ps_flag=%d\n", hDab->ps_flag);
-  //fprintf(stderr, "hDab->aac_channel_mode=%d\n", hDab->aac_channel_mode);
-  //fprintf(stderr, "hDab->subchannels_num=%d\n", hDab->subchannels_num);
-  //fprintf(stderr, "cc->nSubFrames=%d\n", config->nSubFrames);
-
-  hDab->num_raw_blocks=config->nSubFrames-1; /* 0 means 1 raw data block */
-
-  FDKcrcInit(&hDab->crcInfo, 0x1021, 0xFFFF, 16);
-  FDKcrcInit(&hDab->crcFire, 0x782d, 0, 16);
-  FDKcrcInit(&hDab->crcInfo2, 0x8005, 0xFFFF, 16);
-
-  hDab->currentBlock = 0;
-  hDab->headerBits = dabWrite_GetHeaderBits(hDab);
-
-  return 0;
-}
-
-int dabWrite_EncodeHeader(HANDLE_DAB hDab,
-                                 HANDLE_FDK_BITSTREAM hBitStream,
-                                 int buffer_fullness,
-                                 int frame_length)
-{
-  INT crcIndex = 0;
-
-
-  FDK_ASSERT(((frame_length+hDab->headerBits)/8)<0x2000);      /*13 bit*/
-  FDK_ASSERT(buffer_fullness<0x800);    /* 11 bit   */
-
-  FDKcrcReset(&hDab->crcInfo);
-
-
-//  fprintf(stderr, "dabWrite_EncodeHeader() hDab->currentBlock=%d, frame_length=%d, buffer_fullness=%d\n",
-//		  hDab->currentBlock, frame_length, buffer_fullness);
-
-//  if (hDab->currentBlock == 0) {
-//	//hDab->subFrameStartPrev=dabWrite_GetHeaderBits(hDab);
-//	fprintf(stderr, "header bits[%d] [%d]\n", hDab->subFrameStartPrev, hDab->subFrameStartPrev >> 3);
-//    FDKresetBitbuffer(hBitStream, BS_WRITER);
-//  }
-
-  //hDab->subFrameStartBit = FDKgetValidBits(hBitStream);
-//  fprintf(stderr, "dabWrite_EncodeHeader() hDab->subFrameStartBit=%d [%d]\n", hDab->subFrameStartBit, hDab->subFrameStartBit >> 3);
-
-  //hDab->subFrameStartBit = FDKgetValidBits(hBitStream);
-  /* Skip new header if this is raw data block 1..n */
-  if (hDab->currentBlock == 0)
-  {
-    FDKresetBitbuffer(hBitStream, BS_WRITER);
-//    fprintf(stderr, "dabWrite_EncodeHeader() after FDKresetBitbuffer=%d [%d]\n", FDKgetValidBits(hBitStream), FDKgetValidBits(hBitStream) >> 3);
-
-    /* fixed header */
-    FDKwriteBits(hBitStream, 0, 16); //header_firecode
-    FDKwriteBits(hBitStream, 0, 1); //rfa
-    FDKwriteBits(hBitStream, hDab->dac_rate, 1);
-    FDKwriteBits(hBitStream, hDab->sbr_flag, 1);
-    FDKwriteBits(hBitStream, hDab->aac_channel_mode, 1);
-    FDKwriteBits(hBitStream, hDab->ps_flag, 1);
-    FDKwriteBits(hBitStream, hDab->mpeg_surround_config, 3);
-    /* variable header */
-    int i;
-    for(i=0; i<hDab->num_raw_blocks; i++)
-    	FDKwriteBits(hBitStream, 0, 12);
-    /* padding */
-    if (hDab->dac_rate == 0 || hDab->sbr_flag == 0) {
-    	FDKwriteBits(hBitStream, 0, 4);
-    }
-  } /* End of DAB header */
-
-  hDab->subFrameStartBit = FDKgetValidBits(hBitStream);
-  FDK_ASSERT(FDKgetValidBits(hBitStream) % 8 == 0); //only aligned header
-
-//  fprintf(stderr, "dabWrite_EncodeHeader() FDKgetValidBits(hBitStream)=%d [%d]\n", FDKgetValidBits(hBitStream), FDKgetValidBits(hBitStream) >> 3);
-  return 0;
-}
-
-int dabWrite_writeExtensionFillPayload(HANDLE_FDK_BITSTREAM  hBitStream, int extPayloadBits)
-{
-#define EXT_TYPE_BITS         ( 4 )
-#define DATA_EL_VERSION_BITS  ( 4 )
-#define FILL_NIBBLE_BITS      ( 4 )
-
-#define EXT_TYPE_BITS         ( 4 )
-#define DATA_EL_VERSION_BITS  ( 4 )
-#define FILL_NIBBLE_BITS      ( 4 )
-
-	INT  extBitsUsed = 0;
-	INT extPayloadType = EXT_FIL;
-	//fprintf(stderr, "FDKaacEnc_writeExtensionPayload() extPayloadType=%d\n", extPayloadType);
-	if (extPayloadBits >= EXT_TYPE_BITS)
-	{
-	  UCHAR  fillByte = 0x00;  /* for EXT_FIL and EXT_FILL_DATA */
-
-	  if (hBitStream != NULL) {
-		FDKwriteBits(hBitStream, extPayloadType, EXT_TYPE_BITS);
-	  }
-	  extBitsUsed += EXT_TYPE_BITS;
-
-	  switch (extPayloadType) {
-		case EXT_FILL_DATA:
-		  fillByte = 0xA5;
-		case EXT_FIL:
-		default:
-		  if (hBitStream != NULL) {
-			int writeBits = extPayloadBits;
-			FDKwriteBits(hBitStream, 0x00, FILL_NIBBLE_BITS);
-			writeBits -= 8;  /* acount for the extension type and the fill nibble */
-			while (writeBits >= 8) {
-			  FDKwriteBits(hBitStream, fillByte, 8);
-			  writeBits -= 8;
-			}
-		  }
-		  extBitsUsed += FILL_NIBBLE_BITS + (extPayloadBits & ~0x7) - 8;
-		  break;
-	  }
-	}
-
-	return (extBitsUsed);
-}
-
-void dabWrite_FillRawDataBlock(HANDLE_FDK_BITSTREAM  hBitStream, int payloadBits)
-{
-	INT extBitsUsed = 0;
-#define EL_ID_BITS              ( 3 )
-#define FILL_EL_COUNT_BITS      ( 4 )
-#define FILL_EL_ESC_COUNT_BITS  ( 8 )
-#define MAX_FILL_DATA_BYTES     ( 269 )
-    while (payloadBits >= (EL_ID_BITS + FILL_EL_COUNT_BITS)) {
-      INT cnt, esc_count=-1, alignBits=7;
-
-      payloadBits -= EL_ID_BITS + FILL_EL_COUNT_BITS;
-      if (payloadBits >= 15*8) {
-        payloadBits -= FILL_EL_ESC_COUNT_BITS;
-        esc_count = 0;  /* write esc_count even if cnt becomes smaller 15 */
-      }
-      alignBits = 0;
-
-      cnt = fixMin(MAX_FILL_DATA_BYTES, (payloadBits+alignBits)>>3);
-
-      if (cnt >= 15) {
-        esc_count = cnt - 15 + 1;
-      }
-
-      if (hBitStream != NULL) {
-        /* write bitstream */
-        FDKwriteBits(hBitStream, ID_FIL, EL_ID_BITS);
-        if (esc_count >= 0) {
-          FDKwriteBits(hBitStream, 15, FILL_EL_COUNT_BITS);
-          FDKwriteBits(hBitStream, esc_count, FILL_EL_ESC_COUNT_BITS);
-        } else {
-          FDKwriteBits(hBitStream, cnt, FILL_EL_COUNT_BITS);
-        }
-      }
-
-      extBitsUsed += EL_ID_BITS + FILL_EL_COUNT_BITS + ((esc_count>=0) ? FILL_EL_ESC_COUNT_BITS : 0);
-
-      cnt = fixMin(cnt*8, payloadBits);  /* convert back to bits */
-#if 0
-      extBitsUsed += FDKaacEnc_writeExtensionPayload( hBitStream,
-                                                      pExtension->type,
-                                                      pExtension->pPayload,
-                                                      cnt );
-#else
-      extBitsUsed += dabWrite_writeExtensionFillPayload(hBitStream, cnt);
-#endif
-      payloadBits -= cnt;
-    }
-}
-
-void dabWrite_EndRawDataBlock(HANDLE_DAB hDab,
-                          HANDLE_FDK_BITSTREAM hBs,
-                          int *pBits)
-{
-    FDK_BITSTREAM bsWriter;
-	INT crcIndex = 0;
-	USHORT crcData;
-	INT writeBits=0;
-	INT writeBitsNonLastBlock=0;
-	INT writeBitsLastBlock=0;
-#if 1
-    if (hDab->currentBlock == hDab->num_raw_blocks) {
-    	//calculate byte-alignment before writing ID_FIL
-    	if((FDKgetValidBits(hBs)+3) % 8){
-    		writeBits = 8 - ((FDKgetValidBits(hBs)+3) % 8);
-    	}
-
-    	INT offset_end = hDab->subchannels_num*110*8 - 2*8 - 3;
-    	writeBitsLastBlock = offset_end - FDKgetValidBits(hBs);
-        dabWrite_FillRawDataBlock(hBs, writeBitsLastBlock);
-		FDKsyncCache(hBs);
-		//fprintf(stderr, "FIL-element written=%d\n", writeBitsLastBlock);
-		writeBitsLastBlock=writeBits;
-    }
-#endif
-	FDKwriteBits(hBs, 7, 3); //finalize AU: ID_END
-	FDKsyncCache(hBs);
-	//byte-align (if ID_FIL doesn't align it).
-	if(FDKgetValidBits(hBs) % 8){
-		writeBits = 8 - (FDKgetValidBits(hBs) % 8);
-        FDKwriteBits(hBs, 0x00, writeBits);
-		FDKsyncCache(hBs);
-	}
-
-	//fake-written bits alignment for last AU
-	if (hDab->currentBlock == hDab->num_raw_blocks)
-		writeBits=writeBitsLastBlock;
-
-	INT frameLen = (FDKgetValidBits(hBs) - hDab->subFrameStartBit) >> 3;
-	//fprintf(stderr, "frame=%d, offset writeBits=%d\n", frameLen, writeBits);
-
-	FDK_ASSERT(FDKgetValidBits(hBs) % 8 == 0); //only aligned au's
-    FDK_ASSERT(hDab->subchannels_num*110*8 >= FDKgetValidBits(hBs)+2*8); //don't overlap superframe
-
-    FDKinitBitStream(&bsWriter, hBs->hBitBuf.Buffer, hBs->hBitBuf.bufSize, 0, BS_WRITER);
-    FDKpushFor(&bsWriter, hDab->subFrameStartBit);
-    FDKcrcReset(&hDab->crcInfo);
-    hDab->crcIndex = FDKcrcStartReg(&hDab->crcInfo, &bsWriter, 0);
-#if 0
-    if (hDab->currentBlock == hDab->num_raw_blocks) {
-		INT offset_size = hDab->subchannels_num*110*8 - 2*8 - FDKgetValidBits(hBs);
-		//fprintf(stderr, "offset_size=%d\n", offset_size >> 3);
-		FDKpushFor(hBs, offset_size);
-    }
-#endif
-
-    FDKpushFor(&bsWriter, FDKgetValidBits(hBs) - hDab->subFrameStartBit);
-	FDKcrcEndReg(&hDab->crcInfo,  &bsWriter, hDab->crcIndex);
-	crcData = FDKcrcGetCRC(&hDab->crcInfo);
-	//fprintf(stderr, "crcData = %04x\n", crcData);
-    /* Write inverted CRC of current raw data block */
-    FDKwriteBits(hBs, crcData ^ 0xffff, 16);
-	FDKsyncCache(hBs);
-
-
-    /* Write distance to current data block */
-	if(hDab->currentBlock) {
-		FDKinitBitStream(&bsWriter, hBs->hBitBuf.Buffer, hBs->hBitBuf.bufSize, 0, BS_WRITER);
-	    FDKpushFor(&bsWriter, 24 + (hDab->currentBlock-1)*12);
-		//fprintf(stderr, "FDKwriteBits() = %d\n", hDab->subFrameStartBit>>3);
-		FDKwriteBits(&bsWriter, (hDab->subFrameStartBit>>3), 12);
-		FDKsyncCache(&bsWriter);
-	}
-
-	/* Write FireCode */
-	if (hDab->currentBlock == hDab->num_raw_blocks) {
-		FDKinitBitStream(&bsWriter, hBs->hBitBuf.Buffer, hBs->hBitBuf.bufSize, 0, BS_WRITER);
-		FDKpushFor(&bsWriter, 16);
-
-		FDKcrcReset(&hDab->crcFire);
-		crcIndex = FDKcrcStartReg(&hDab->crcFire, &bsWriter, 72);
-		FDKpushFor(&bsWriter, 9*8); //9bytes
-		FDKcrcEndReg(&hDab->crcFire, &bsWriter, crcIndex);
-
-		crcData = FDKcrcGetCRC(&hDab->crcFire);
-		//fprintf(stderr, "Firecode: %04x\n", crcData);
-
-		FDKinitBitStream(&bsWriter, hBs->hBitBuf.Buffer, hBs->hBitBuf.bufSize, 0, BS_WRITER);
-		FDKwriteBits(&bsWriter, crcData, 16);
-		FDKsyncCache(&bsWriter);
-	}
-
-    if (hDab->currentBlock == 0)
-    	*pBits += hDab->headerBits;
-    else
-        *pBits += 16;
-
-    *pBits += writeBits + 3; //size: ID_END + alignment
-
-    /* Correct *pBits to reflect the amount of bits of the current subframe */
-    *pBits -= hDab->subFrameStartBit;
-    /* Fixup CRC bits, since they come after each raw data block */
-
-    hDab->currentBlock++;
-    //fprintf(stderr, "dabWrite_EndRawDataBlock() *pBits=%d (%d)\n", *pBits, *pBits >> 3);
-}
-
diff --git a/libMpegTPEnc/src/tpenc_dab.h b/libMpegTPEnc/src/tpenc_dab.h
deleted file mode 100644
index 17b83c6..0000000
--- a/libMpegTPEnc/src/tpenc_dab.h
+++ /dev/null
@@ -1,217 +0,0 @@
-
-/* -----------------------------------------------------------------------------------------------------------
-Software License for The Fraunhofer FDK AAC Codec Library for Android
-
-� Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur F�rderung der angewandten Forschung e.V.
-  All rights reserved.
-
- 1.    INTRODUCTION
-The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
-the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
-This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
-
-AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
-audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
-independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
-of the MPEG specifications.
-
-Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
-may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
-individually for the purpose of encoding or decoding bit streams in products that are compliant with
-the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
-these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
-software may already be covered under those patent licenses when it is used for those licensed purposes only.
-
-Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
-are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
-applications information and documentation.
-
-2.    COPYRIGHT LICENSE
-
-Redistribution and use in source and binary forms, with or without modification, are permitted without
-payment of copyright license fees provided that you satisfy the following conditions:
-
-You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
-your modifications thereto in source code form.
-
-You must retain the complete text of this software license in the documentation and/or other materials
-provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
-You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
-modifications thereto to recipients of copies in binary form.
-
-The name of Fraunhofer may not be used to endorse or promote products derived from this library without
-prior written permission.
-
-You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
-software or your modifications thereto.
-
-Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
-and the date of any change. For modified versions of the FDK AAC Codec, the term
-"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
-"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
-
-3.    NO PATENT LICENSE
-
-NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
-ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
-respect to this software.
-
-You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
-by appropriate patent licenses.
-
-4.    DISCLAIMER
-
-This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
-"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
-of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
-CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
-including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
-or business interruption, however caused and on any theory of liability, whether in contract, strict
-liability, or tort (including negligence), arising in any way out of the use of this software, even if
-advised of the possibility of such damage.
-
-5.    CONTACT INFORMATION
-
-Fraunhofer Institute for Integrated Circuits IIS
-Attention: Audio and Multimedia Departments - FDK AAC LL
-Am Wolfsmantel 33
-91058 Erlangen, Germany
-
-www.iis.fraunhofer.de/amm
-amm-info@iis.fraunhofer.de
------------------------------------------------------------------------------------------------------------ */
-
-/******************************** MPEG Audio Encoder **************************
-
-   Initial author:       serge
-   contents/description: DAB Transport writer
-
-******************************************************************************/
-
-#ifndef TPENC_DAB_H
-#define TPENC_DAB_H
-
-
-
-#include "tp_data.h"
-
-#include "FDK_crc.h"
-
-typedef struct {
-  USHORT frame_length;
-  UCHAR dac_rate;
-  UCHAR aac_channel_mode;
-  UCHAR sbr_flag;
-  UCHAR ps_flag;
-  UCHAR mpeg_surround_config;
-  UCHAR num_raw_blocks;
-  UCHAR BufferFullnesStartFlag;
-  int subchannels_num;
-  int headerBits;                /*!< Header bit demand for the current raw data block */
-  int currentBlock;              /*!< Index of current raw data block */
-  int subFrameStartBit;          /*!< Bit position where the current raw data block begins */
-  //int subFrameStartPrev;         /*!< Bit position where the previous raw data block begins */
-  int crcIndex;
-  FDK_CRCINFO crcInfo;
-  FDK_CRCINFO crcFire;
-  FDK_CRCINFO crcInfo2;
-  USHORT tab[256];
-} STRUCT_DAB;
-
-typedef STRUCT_DAB *HANDLE_DAB;
-
-/**
- * \brief Initialize DAB data structure
- *
- * \param hDab DAB data handle
- * \param config a valid CODER_CONFIG struct from where the required
- *        information for the DAB header is extrated from
- *
- * \return 0 in case of success.
- */
-INT dabWrite_Init(
-        HANDLE_DAB   hDab,
-        CODER_CONFIG *config
-        );
-
-/**
- * \brief Get the total bit overhead caused by DAB
- *
- * \hDab handle to DAB data
- *
- * \return Amount of additional bits required for the current raw data block
- */
-int dabWrite_GetHeaderBits( HANDLE_DAB hDab );
-int dabWrite_CountTotalBitDemandHeader( HANDLE_DAB hDab, unsigned int streamDataLength );
-
-/**
- * \brief Write an DAB header into the given bitstream. May not write a header
- *        in case of multiple raw data blocks.
- *
- * \param hDab DAB data handle
- * \param hBitStream bitstream handle into which the DAB may be written into
- * \param buffer_fullness the buffer fullness value for the DAB header
- * \param the current raw data block length
- *
- * \return 0 in case of success.
- */
-INT dabWrite_EncodeHeader(
-        HANDLE_DAB          hDab,
-        HANDLE_FDK_BITSTREAM hBitStream,
-        int                  bufferFullness,
-        int                  frame_length
-        );
-/**
- * \brief Finish a DAB raw data block
- *
- * \param hDab DAB data handle
- * \param hBs bitstream handle into which the DAB may be written into
- * \param pBits a pointer to a integer holding the current bitstream buffer bit count,
- *              which is corrected to the current raw data block boundary.
- *
- */
-void dabWrite_EndRawDataBlock(
-        HANDLE_DAB          hDab,
-        HANDLE_FDK_BITSTREAM hBs,
-        int                 *bits
-        );
-
-
-/**
- * \brief Start 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 pDab DAB data handle
- * \param hBs bitstream handle of which the CRC region ends
- * \param mBits limit of number of bits to be considered for the requested CRC region
- *
- * \return ID for the created region, -1 in case of an error
- */
-int dabWrite_CrcStartReg(
-        HANDLE_DAB          pDab,
-        HANDLE_FDK_BITSTREAM hBs,
-        int                  mBits
-        );
-
-/**
- * \brief Ends CRC region identified by reg
- *
- * \param pDab DAB data handle
- * \param hBs bitstream handle of which the CRC region ends
- * \param reg a CRC region ID returned previously by dabWrite_CrcStartReg()
- */
-void dabWrite_CrcEndReg(
-        HANDLE_DAB          pDab,
-        HANDLE_FDK_BITSTREAM hBs,
-        int                  reg
-        );
-
-
-
-
-#endif /* TPENC_DAB_H */
-
diff --git a/libMpegTPEnc/src/tpenc_latm.cpp b/libMpegTPEnc/src/tpenc_latm.cpp
deleted file mode 100644
index 58e51ef..0000000
--- a/libMpegTPEnc/src/tpenc_latm.cpp
+++ /dev/null
@@ -1,882 +0,0 @@
-
-/* -----------------------------------------------------------------------------------------------------------
-Software License for The Fraunhofer FDK AAC Codec Library for Android
-
-© Copyright  1995 - 2013 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 Encoder  **************************
-
-   Author(s):
-   Description:
-
-******************************************************************************/
-
-#include "tpenc_latm.h"
-
-
-#include "genericStds.h"
-
-static const short celpFrameLengthTable[64] = {
- 154, 170, 186, 147, 156, 165, 114, 120,
- 186, 126, 132, 138, 142, 146, 154, 166,
- 174, 182, 190, 198, 206, 210, 214, 110,
- 114, 118, 120, 122, 218, 230, 242, 254,
- 266, 278, 286, 294, 318, 342, 358, 374,
- 390, 406, 422, 136, 142, 148, 154, 160,
- 166, 170, 174, 186, 198, 206, 214, 222,
- 230, 238, 216, 160, 280, 338, 0,   0
-};
-
-/*******
- write value to transport stream
- first two bits define the size of the value itself
- then the value itself, with a size of 0-3 bytes
-*******/
-static
-UINT transportEnc_LatmWriteValue(HANDLE_FDK_BITSTREAM hBs, int value)
-{
-  UCHAR valueBytes = 4;
-  unsigned int bitsWritten = 0;
-  int i;
-
-  if ( value < (1<<8) ) {
-    valueBytes = 1;
-  } else if ( value < (1<<16) ) {
-    valueBytes = 2;
-  } else if ( value < (1<<24) ) {
-    valueBytes = 3;
-  } else {
-    valueBytes = 4;
-  }
-
-  FDKwriteBits(hBs, valueBytes-1, 2 ); /* size of value in Bytes */
-  for (i=0; i<valueBytes; i++) {
-    /* write most significant Byte first */
-    FDKwriteBits(hBs, (UCHAR)(value>>((valueBytes-1-i)<<3)), 8);
-  }
-
-  bitsWritten = (valueBytes<<3)+2;
-
-  return bitsWritten;
-}
-
-static
-UINT transportEnc_LatmCountFixBitDemandHeader ( HANDLE_LATM_STREAM hAss )
-{
-  int bitDemand = 0;
-  int insertSetupData = 0 ;
-
-  /* only if start of new latm frame */
-  if (hAss->subFrameCnt==0)
-  {
-    /* AudioSyncStream */
-
-    if (hAss->tt == TT_MP4_LOAS) {
-      bitDemand += 11 ;             /* syncword */
-      bitDemand += 13 ;             /* audioMuxLengthBytes */
-    }
-
-    /* AudioMuxElement*/
-
-    /* AudioMuxElement::Stream Mux Config */
-    if (hAss->muxConfigPeriod > 0) {
-      insertSetupData = (hAss->latmFrameCounter == 0);
-    } else {
-      insertSetupData = 0;
-    }
-
-    if (hAss->tt != TT_MP4_LATM_MCP0) {
-      /* AudioMuxElement::useSameStreamMux Flag */
-      bitDemand+=1;
-
-      if( insertSetupData ) {
-        bitDemand += hAss->streamMuxConfigBits;
-      }
-    }
-
-    /* AudioMuxElement::otherDataBits */
-    bitDemand += 8*hAss->otherDataLenBytes;
-
-    /* AudioMuxElement::ByteAlign */
-    if ( bitDemand % 8 ) {
-       hAss->fillBits = 8 - (bitDemand % 8);
-       bitDemand += hAss->fillBits ;
-    } else {
-      hAss->fillBits = 0;
-    }
-  }
-
-  return bitDemand ;
-}
-
-static
-UINT transportEnc_LatmCountVarBitDemandHeader ( HANDLE_LATM_STREAM hAss , unsigned int streamDataLength )
-{
-  int bitDemand = 0;
-  int  prog, layer;
-
-  /* Payload Length Info*/
-  if( hAss->allStreamsSameTimeFraming ) {
-    for( prog=0; prog<hAss->noProgram; prog++ ) {
-      for( layer=0; layer<LATM_MAX_LAYERS; layer++ ) {
-        LATM_LAYER_INFO *p_linfo = &(hAss->m_linfo[prog][layer]);
-
-        if( p_linfo->streamID >= 0 ) {
-          switch( p_linfo->frameLengthType ) {
-          case 0:
-            if ( streamDataLength > 0 ) {
-              streamDataLength -= bitDemand ;
-              while( streamDataLength >= (255<<3) ) {
-                bitDemand+=8;
-                streamDataLength -= (255<<3);
-              }
-              bitDemand += 8;
-            }
-            break;
-
-          case 1:
-          case 4:
-          case 6:
-            bitDemand += 2;
-            break;
-
-          default:
-            return 0;
-          }
-        }
-      }
-    }
-  } else {
-    /* there are many possibilities to use this mechanism.  */
-    switch( hAss->varMode ) {
-    case LATMVAR_SIMPLE_SEQUENCE: {
-      /* Use the sequence generated by the encoder */
-      //int streamCntPosition = transportEnc_SetWritePointer( hAss->hAssemble, 0 );
-      //int streamCntPosition = FDKgetValidBits( hAss->hAssemble );
-      bitDemand+=4;
-
-      hAss->varStreamCnt = 0;
-      for( prog=0; prog<hAss->noProgram; prog++ ) {
-        for( layer=0; layer<LATM_MAX_LAYERS; layer++ ) {
-          LATM_LAYER_INFO *p_linfo = &(hAss->m_linfo[prog][layer]);
-
-          if( p_linfo->streamID >= 0 ) {
-
-            bitDemand+=4; /* streamID */
-            switch( p_linfo->frameLengthType ) {
-            case 0:
-              streamDataLength -= bitDemand ;
-              while( streamDataLength >= (255<<3) ) {
-                bitDemand+=8;
-                streamDataLength -= (255<<3);
-              }
-
-              bitDemand += 8;
-              break;
-              /*bitDemand += 1; endFlag
-              break;*/
-
-            case 1:
-            case 4:
-            case 6:
-
-              break;
-
-            default:
-              return  0;
-            }
-            hAss->varStreamCnt++;
-          }
-        }
-      }
-      bitDemand+=4;
-      //transportEnc_UpdateBitstreamField( hAss->hAssemble, streamCntPosition, hAss->varStreamCnt-1, 4 );
-      //UINT pos = streamCntPosition-FDKgetValidBits(hAss->hAssemble);
-      //FDKpushBack( hAss->hAssemble,  pos);
-      //FDKwriteBits( hAss->hAssemble, hAss->varStreamCnt-1, 4);
-      //FDKpushFor( hAss->hAssemble, pos-4);
-    }
-    break;
-
-    default:
-      return  0;
-    }
-  }
-
-  return bitDemand ;
-}
-
-TRANSPORTENC_ERROR
-CreateStreamMuxConfig(
-                      HANDLE_LATM_STREAM hAss,
-                      HANDLE_FDK_BITSTREAM hBs,
-                      int bufferFullness,
-                      CSTpCallBacks *cb
-                     )
-{
-  INT streamIDcnt, tmp;
-  int layer, prog;
-
-  USHORT coreFrameOffset=0;
-
-  hAss->audioMuxVersionA    = 0; /* for future extensions */
-  hAss->streamMuxConfigBits = 0;
-
-  FDKwriteBits( hBs, hAss->audioMuxVersion, 1 );                   /* audioMuxVersion */
-  hAss->streamMuxConfigBits += 1;
-
-  if ( hAss->audioMuxVersion == 1 ) {
-    FDKwriteBits( hBs, hAss->audioMuxVersionA, 1 );                /* audioMuxVersionA */
-    hAss->streamMuxConfigBits+=1;
-  }
-
-  if ( hAss->audioMuxVersionA == 0 )
-  {
-    if ( hAss->audioMuxVersion == 1 ) {
-      hAss->streamMuxConfigBits+= transportEnc_LatmWriteValue( hBs, hAss->taraBufferFullness );/* taraBufferFullness */
-    }
-    FDKwriteBits( hBs, hAss->allStreamsSameTimeFraming ? 1:0, 1 ); /* allStreamsSameTimeFraming */
-    FDKwriteBits( hBs, hAss->noSubframes-1, 6 );                   /* Number of Subframes */
-    FDKwriteBits( hBs, hAss->noProgram-1, 4 );                     /* Number of Programs */
-
-    hAss->streamMuxConfigBits+=11;
-
-    streamIDcnt = 0;
-    for( prog=0; prog<hAss->noProgram; prog++ ) {
-      int transLayer = 0;
-
-      FDKwriteBits( hBs, hAss->noLayer[prog]-1, 3 );
-      hAss->streamMuxConfigBits+=3;
-
-      for( layer=0; layer<LATM_MAX_LAYERS; layer++ ) {
-        LATM_LAYER_INFO   *p_linfo = &(hAss->m_linfo[prog][layer]);
-        CODER_CONFIG *p_lci   = hAss->config[prog][layer];
-
-        p_linfo->streamID = -1;
-
-        if( hAss->config[prog][layer] != NULL ) {
-          int useSameConfig = 0;
-
-          if( transLayer > 0 ) {
-            FDKwriteBits( hBs, useSameConfig ? 1 : 0, 1 );
-            hAss->streamMuxConfigBits+=1;
-          }
-          if( (useSameConfig == 0) || (transLayer==0) ) {
-            UINT bits;
-
-            if ( hAss->audioMuxVersion == 1 ) {
-              FDKpushFor(hBs, 2); /* align to ASC, even if we do not know the length of the "ascLen" field yet */
-            }
-
-            bits = FDKgetValidBits( hBs );
-
-            transportEnc_writeASC(
-                    hBs,
-                    hAss->config[prog][layer],
-                    cb
-                    );
-
-            bits = FDKgetValidBits( hBs ) - bits;
-
-            if ( hAss->audioMuxVersion == 1 ) {
-              FDKpushBack(hBs, bits+2);
-              hAss->streamMuxConfigBits += transportEnc_LatmWriteValue( hBs, bits );
-              transportEnc_writeASC(
-                      hBs,
-                      hAss->config[prog][layer],
-                      cb
-                      );
-            }
-
-            hAss->streamMuxConfigBits += bits; /* add asc length to smc summary */
-          }
-          transLayer++;
-
-          if( !hAss->allStreamsSameTimeFraming ) {
-            if( streamIDcnt >= LATM_MAX_STREAM_ID )
-              return TRANSPORTENC_INVALID_CONFIG;
-          }
-          p_linfo->streamID = streamIDcnt++;
-
-          switch( p_lci->aot ) {
-          case AOT_AAC_MAIN      :
-          case AOT_AAC_LC        :
-          case AOT_AAC_SSR       :
-          case AOT_AAC_LTP       :
-          case AOT_AAC_SCAL      :
-          case AOT_ER_AAC_LD     :
-          case AOT_ER_AAC_ELD    :
-          case AOT_USAC:
-          case AOT_RSVD50:
-            p_linfo->frameLengthType = 0;
-
-            FDKwriteBits( hBs, p_linfo->frameLengthType, 3 );                        /* frameLengthType */
-            FDKwriteBits( hBs, bufferFullness, 8 );                           /* bufferFullness */
-            hAss->streamMuxConfigBits+=11;
-
-            if ( !hAss->allStreamsSameTimeFraming ) {
-              CODER_CONFIG *p_lci_prev = hAss->config[prog][layer-1];
-              if ( ((p_lci->aot == AOT_AAC_SCAL) || (p_lci->aot == AOT_ER_AAC_SCAL)) &&
-                   ((p_lci_prev->aot == AOT_CELP) || (p_lci_prev->aot == AOT_ER_CELP)) ) {
-                FDKwriteBits( hBs, coreFrameOffset, 6 );                      /* coreFrameOffset */
-                hAss->streamMuxConfigBits+=6;
-              }
-            }
-            break;
-
-          case AOT_TWIN_VQ:
-            p_linfo->frameLengthType = 1;
-            tmp = ( (p_lci->bitsFrame+7) >> 3 ) - 20;                            /* transmission frame length in bytes */
-            if( (tmp < 0) ) {
-              return TRANSPORTENC_INVALID_TRANSMISSION_FRAME_LENGTH;
-            }
-            FDKwriteBits( hBs, p_linfo->frameLengthType, 3 );          /* frameLengthType */
-            FDKwriteBits( hBs, tmp, 9 );
-            hAss->streamMuxConfigBits+=12;
-
-            p_linfo->frameLengthBits = (tmp+20) << 3;
-            break;
-
-          case AOT_CELP:
-            p_linfo->frameLengthType = 4;
-            FDKwriteBits( hBs, p_linfo->frameLengthType, 3 );          /* frameLengthType */
-            hAss->streamMuxConfigBits+=3;
-            {
-              int i;
-              for( i=0; i<62; i++ ) {
-                if( celpFrameLengthTable[i] == p_lci->bitsFrame )
-                  break;
-              }
-              if( i>=62 ) {
-                return TRANSPORTENC_INVALID_CELP_FRAME_LENGTH;
-              }
-
-              FDKwriteBits( hBs, i, 6 );                                /* CELPframeLengthTabelIndex */
-              hAss->streamMuxConfigBits+=6;
-            }
-            p_linfo->frameLengthBits = p_lci->bitsFrame;
-            break;
-
-          case AOT_HVXC:
-            p_linfo->frameLengthType = 6;
-            FDKwriteBits( hBs, p_linfo->frameLengthType, 3 );          /* frameLengthType */
-            hAss->streamMuxConfigBits+=3;
-            {
-              int i;
-
-              if( p_lci->bitsFrame == 40 ) {
-                i = 0;
-              } else if( p_lci->bitsFrame == 80 ) {
-                i = 1;
-              } else {
-                return TRANSPORTENC_INVALID_FRAME_BITS;
-              }
-              FDKwriteBits( hBs, i, 1 );                                /* HVXCframeLengthTableIndex */
-              hAss->streamMuxConfigBits+=1;
-            }
-            p_linfo->frameLengthBits = p_lci->bitsFrame;
-            break;
-
-          case AOT_NULL_OBJECT:
-          default:
-            return TRANSPORTENC_INVALID_AOT;
-          }
-        }
-      }
-    }
-
-    FDKwriteBits( hBs, (hAss->otherDataLenBytes>0) ? 1:0, 1 );      /* otherDataPresent */
-    hAss->streamMuxConfigBits+=1;
-
-    if( hAss->otherDataLenBytes > 0 ) {
-
-      INT otherDataLenTmp = hAss->otherDataLenBytes;
-      INT escCnt = 0;
-      INT otherDataLenEsc = 1;
-
-      while(otherDataLenTmp) {
-        otherDataLenTmp >>= 8;
-        escCnt ++;
-      }
-
-      do {
-        otherDataLenTmp = (hAss->otherDataLenBytes>>(escCnt*8)) & 0xFF;
-        escCnt--;
-        otherDataLenEsc = escCnt>0;
-
-        FDKwriteBits( hBs, otherDataLenEsc, 1 );
-        FDKwriteBits( hBs, otherDataLenTmp, 8 );
-        hAss->streamMuxConfigBits+=9;
-      } while(otherDataLenEsc);
-    }
-
-    {
-      USHORT crcCheckPresent=0;
-      USHORT crcCheckSum=0;
-
-      FDKwriteBits( hBs, crcCheckPresent, 1 );               /* crcCheckPresent */
-      hAss->streamMuxConfigBits+=1;
-      if ( crcCheckPresent ){
-        FDKwriteBits( hBs, crcCheckSum, 8 );                 /* crcCheckSum */
-        hAss->streamMuxConfigBits+=8;
-      }
-    }
-
-  } else {  /* if ( audioMuxVersionA == 0 ) */
-
-    /* for future extensions */
-
-  }
-
-  return TRANSPORTENC_OK;
-}
-
-
-static TRANSPORTENC_ERROR
-WriteAuPayloadLengthInfo( HANDLE_FDK_BITSTREAM hBitStream, int AuLengthBits )
-{
-  int restBytes;
-
-  if( AuLengthBits % 8 )
-    return TRANSPORTENC_INVALID_AU_LENGTH;
-
-  while( AuLengthBits >= 255*8 ) {
-    FDKwriteBits( hBitStream, 255, 8 );  /* 255 shows incomplete AU */
-    AuLengthBits -= (255*8);
-  }
-
-  restBytes = (AuLengthBits) >> 3;
-  FDKwriteBits( hBitStream, restBytes, 8 );
-
-  return TRANSPORTENC_OK;
-}
-
-static
-TRANSPORTENC_ERROR transportEnc_LatmSetNrOfSubframes( HANDLE_LATM_STREAM hAss,
-                                                      INT noSubframes_next)    /* nr of access units / payloads within a latm frame */
-{
-  /* sanity chk */
-  if (noSubframes_next < 1 || noSubframes_next > MAX_NR_OF_SUBFRAMES) {
-    return TRANSPORTENC_LATM_INVALID_NR_OF_SUBFRAMES;
-  }
-
-  hAss->noSubframes_next = noSubframes_next;
-
-  /* if at start then we can take over the value immediately, otherwise we have to wait for the next SMC */
-  if ( (hAss->subFrameCnt == 0) && (hAss->latmFrameCounter == 0) ) {
-    hAss->noSubframes = noSubframes_next;
-  }
-
-  return TRANSPORTENC_OK;
-}
-
-static
-int allStreamsSameTimeFraming( HANDLE_LATM_STREAM hAss, UCHAR noProgram, UCHAR noLayer[] /* return */ )
-{
-  int prog, layer;
-
-  signed int lastNoSamples   = -1;
-  signed int minFrameSamples = FDK_INT_MAX;
-  signed int maxFrameSamples = 0;
-
-  signed int highestSamplingRate = -1;
-
-  for( prog=0; prog<noProgram; prog++ ) {
-    noLayer[prog] = 0;
-
-    for( layer=0; layer<LATM_MAX_LAYERS; layer++ )
-    {
-      if( hAss->config[prog][layer] != NULL )
-      {
-        INT hsfSamplesFrame;
-
-        noLayer[prog]++;
-
-        if( highestSamplingRate < 0 )
-          highestSamplingRate = hAss->config[prog][layer]->samplingRate;
-
-        hsfSamplesFrame = hAss->config[prog][layer]->samplesPerFrame  * highestSamplingRate / hAss->config[prog][layer]->samplingRate;
-
-        if( hsfSamplesFrame <= minFrameSamples ) minFrameSamples = hsfSamplesFrame;
-        if( hsfSamplesFrame >= maxFrameSamples ) maxFrameSamples = hsfSamplesFrame;
-
-        if( lastNoSamples == -1 ) {
-          lastNoSamples                             = hsfSamplesFrame;
-        } else {
-          if( hsfSamplesFrame != lastNoSamples ) {
-            return 0;
-          }
-        }
-      }
-    }
-  }
-
-  return 1;
-}
-
-/**
- * Initialize LATM/LOAS Stream and add layer 0 at program 0.
- */
-static
-TRANSPORTENC_ERROR transportEnc_InitLatmStream( HANDLE_LATM_STREAM hAss,
-                                                int                fractDelayPresent,
-                                                signed int         muxConfigPeriod, /* insert setup data every muxConfigPeriod frames */
-                                                UINT               audioMuxVersion,
-                                                TRANSPORT_TYPE     tt
-                                              )
-{
-  TRANSPORTENC_ERROR ErrorStatus = TRANSPORTENC_OK;
-
-  if (hAss == NULL)
-    return TRANSPORTENC_INVALID_PARAMETER;
-
-  hAss->tt = tt;
-
-  hAss->noProgram = 1;
-
-  hAss->audioMuxVersion = audioMuxVersion;
-
-  /* Fill noLayer array using hAss->config */
-  hAss->allStreamsSameTimeFraming = allStreamsSameTimeFraming( hAss, hAss->noProgram, hAss->noLayer );
-  /* Only allStreamsSameTimeFraming==1 is supported */
-  FDK_ASSERT(hAss->allStreamsSameTimeFraming);
-
-  hAss->fractDelayPresent = fractDelayPresent;
-  hAss->otherDataLenBytes = 0;
-
-  hAss->varMode = LATMVAR_SIMPLE_SEQUENCE;
-
-  /* initialize counters */
-  hAss->subFrameCnt                  = 0;
-  hAss->noSubframes                  = DEFAULT_LATM_NR_OF_SUBFRAMES;
-  hAss->noSubframes_next             = DEFAULT_LATM_NR_OF_SUBFRAMES;
-
-  /* sync layer related */
-  hAss->audioMuxLengthBytes     = 0;
-
-  hAss->latmFrameCounter        = 0;
-  hAss->muxConfigPeriod = muxConfigPeriod;
-
-  return ErrorStatus;
-}
-
-
-/**
- *
- */
-UINT transportEnc_LatmCountTotalBitDemandHeader ( HANDLE_LATM_STREAM hAss , unsigned int streamDataLength )
-{
-  UINT bitDemand = 0;
-
-  switch (hAss->tt) {
-  case TT_MP4_LOAS:
-  case TT_MP4_LATM_MCP0:
-  case TT_MP4_LATM_MCP1:
-    if (hAss->subFrameCnt == 0) {
-      bitDemand  = transportEnc_LatmCountFixBitDemandHeader ( hAss );
-    }
-    bitDemand += transportEnc_LatmCountVarBitDemandHeader ( hAss , streamDataLength /*- bitDemand*/);
-    break;
-  default:
-    break;
-  }
-
-  return bitDemand;
-}
-
-static TRANSPORTENC_ERROR
-AdvanceAudioMuxElement (
-        HANDLE_LATM_STREAM   hAss,
-        HANDLE_FDK_BITSTREAM hBs,
-        int                  auBits,
-        int                  bufferFullness,
-        CSTpCallBacks    *cb
-        )
-{
-  TRANSPORTENC_ERROR ErrorStatus = TRANSPORTENC_OK;
-  int insertMuxSetup;
-
-  /* Insert setup data to assemble Buffer */
-  if (hAss->subFrameCnt == 0)
-  {
-    if (hAss->muxConfigPeriod > 0) {
-      insertMuxSetup = (hAss->latmFrameCounter == 0);
-    } else  {
-      insertMuxSetup = 0;
-    }
-
-    if (hAss->tt != TT_MP4_LATM_MCP0) {
-      if( insertMuxSetup ) {
-        FDKwriteBits( hBs, 0, 1 );  /* useSameStreamMux useNewStreamMux */
-        CreateStreamMuxConfig(hAss, hBs, bufferFullness, cb);
-        if (ErrorStatus != TRANSPORTENC_OK)
-          return ErrorStatus;
-      } else {
-        FDKwriteBits( hBs, 1, 1 );   /* useSameStreamMux */
-      }
-    }
-  }
-
-  /* PayloadLengthInfo */
-  {
-    int prog, layer;
-
-    for (prog = 0; prog < hAss->noProgram; prog++) {
-      for (layer = 0; layer < hAss->noLayer[prog]; layer++) {
-        ErrorStatus = WriteAuPayloadLengthInfo( hBs, auBits );
-        if (ErrorStatus != TRANSPORTENC_OK)
-          return ErrorStatus;
-      }
-    }
-  }
-  /* At this point comes the access unit. */
-
-  return TRANSPORTENC_OK;
-}
-
-TRANSPORTENC_ERROR
-transportEnc_LatmWrite (
-        HANDLE_LATM_STREAM    hAss,
-        HANDLE_FDK_BITSTREAM  hBs,
-        int                   auBits,
-        int                   bufferFullness,
-        CSTpCallBacks     *cb
-        )
-{
-  TRANSPORTENC_ERROR ErrorStatus;
-
-  if (hAss->subFrameCnt == 0) {
-    /* Start new frame */
-    FDKresetBitbuffer(hBs, BS_WRITER);
-  }
-
-  hAss->latmSubframeStart = FDKgetValidBits(hBs);
-
-  /* Insert syncword and syncword distance
-     - only if loas
-     - we must update the syncword distance (=audiomuxlengthbytes) later
-   */
-  if( hAss->tt == TT_MP4_LOAS && hAss->subFrameCnt == 0)
-  {
-    /* Start new LOAS frame */
-    FDKwriteBits( hBs, 0x2B7, 11 );
-    hAss->audioMuxLengthBytes = 0;
-    hAss->audioMuxLengthBytesPos = FDKgetValidBits( hBs );  /* store read pointer position */
-    FDKwriteBits( hBs, hAss->audioMuxLengthBytes, 13 );
-  }
-
-  ErrorStatus = AdvanceAudioMuxElement(
-          hAss,
-          hBs,
-          auBits,
-          bufferFullness,
-          cb
-          );
-
-  if (ErrorStatus != TRANSPORTENC_OK)
-    return ErrorStatus;
-
-  return ErrorStatus;
-}
-
-void transportEnc_LatmAdjustSubframeBits(HANDLE_LATM_STREAM    hAss,
-                                         int                  *bits)
-{
-  /* Substract bits from possible previous subframe */
-  *bits -= hAss->latmSubframeStart;
-  /* Add fill bits */
-  if (hAss->subFrameCnt == 0)
-    *bits += hAss->fillBits;
-}
-
-
-void transportEnc_LatmGetFrame(HANDLE_LATM_STREAM    hAss,
-                               HANDLE_FDK_BITSTREAM  hBs,
-                               int                  *bytes)
-{
-
-  hAss->subFrameCnt++;
-  if (hAss->subFrameCnt >= hAss->noSubframes)
-  {
-
-    /* Add LOAS frame length if required. */
-    if (hAss->tt == TT_MP4_LOAS)
-    {
-      int latmBytes;
-
-      latmBytes = (FDKgetValidBits(hBs)+7) >> 3;
-
-      /* write length info into assembler buffer */
-      hAss->audioMuxLengthBytes = latmBytes - 3; /* 3=Syncword + length */
-      {
-        FDK_BITSTREAM tmpBuf;
-
-        FDKinitBitStream( &tmpBuf, hBs->hBitBuf.Buffer, hBs->hBitBuf.bufSize, 0, BS_WRITER ) ;
-        FDKpushFor( &tmpBuf, hAss->audioMuxLengthBytesPos );
-        FDKwriteBits( &tmpBuf, hAss->audioMuxLengthBytes, 13 );
-        FDKsyncCache( &tmpBuf );
-      }
-    }
-
-    /* Write AudioMuxElement byte alignment fill bits */
-    FDKwriteBits(hBs, 0, hAss->fillBits);
-
-    FDK_ASSERT( (FDKgetValidBits(hBs) % 8) == 0);
-
-    hAss->subFrameCnt = 0;
-
-    FDKsyncCache(hBs);
-    *bytes = (FDKgetValidBits(hBs) + 7)>>3;
-    //FDKfetchBuffer(hBs, buffer, (UINT*)bytes);
-
-    if (hAss->muxConfigPeriod > 0)
-    {
-      hAss->latmFrameCounter++;
-
-      if (hAss->latmFrameCounter >= hAss->muxConfigPeriod) {
-        hAss->latmFrameCounter = 0;
-        hAss->noSubframes = hAss->noSubframes_next;
-      }
-    }
-  } else {
-    /* No data this time */
-    *bytes = 0;
-  }
-}
-
-/**
- * Init LATM/LOAS
- */
-TRANSPORTENC_ERROR transportEnc_Latm_Init(
-        HANDLE_LATM_STREAM  hAss,
-        HANDLE_FDK_BITSTREAM hBs,
-        CODER_CONFIG  *layerConfig,
-        UINT audioMuxVersion,
-        TRANSPORT_TYPE tt,
-        CSTpCallBacks *cb
-        )
-{
-  TRANSPORTENC_ERROR ErrorStatus;
-  int fractDelayPresent = 0;
-  int prog, layer;
-
-  int setupDataDistanceFrames = layerConfig->headerPeriod;
-
-  FDK_ASSERT(setupDataDistanceFrames>=0);
-
-  for (prog=0; prog<LATM_MAX_PROGRAMS; prog++) {
-    for (layer=0; layer<LATM_MAX_LAYERS; layer++) {
-      hAss->config[prog][layer] = NULL;
-      hAss->m_linfo[prog][layer].streamID = -1;
-    }
-  }
-
-  hAss->config[0][0] = layerConfig;
-  hAss->m_linfo[0][0].streamID = 0;
-
-  ErrorStatus = transportEnc_InitLatmStream( hAss,
-                                             fractDelayPresent,
-                                             setupDataDistanceFrames,
-                                             (audioMuxVersion)?1:0,
-                                             tt
-                                             );
-  if (ErrorStatus != TRANSPORTENC_OK)
-    goto bail;
-
-  ErrorStatus = transportEnc_LatmSetNrOfSubframes(
-                                                   hAss,
-                                                   layerConfig->nSubFrames
-                                                  );
-  if (ErrorStatus != TRANSPORTENC_OK)
-    goto bail;
-
-  /* Get the size of the StreamMuxConfig somehow */
-  AdvanceAudioMuxElement(hAss, hBs, 0, 0, cb);
-  //CreateStreamMuxConfig(hAss, hBs, 0);
-
-bail:
-  return ErrorStatus;
-}
-
-
-
-
-
-
diff --git a/libMpegTPEnc/src/tpenc_latm.h b/libMpegTPEnc/src/tpenc_latm.h
deleted file mode 100644
index 34eea58..0000000
--- a/libMpegTPEnc/src/tpenc_latm.h
+++ /dev/null
@@ -1,264 +0,0 @@
-
-/* -----------------------------------------------------------------------------------------------------------
-Software License for The Fraunhofer FDK AAC Codec Library for Android
-
-© Copyright  1995 - 2013 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 Encoder  **************************
-
-   Author(s):
-   Description:
-
-******************************************************************************/
-
-#ifndef TPENC_LATM_H
-#define TPENC_LATM_H
-
-
-
-#include "tpenc_lib.h"
-#include "FDK_bitstream.h"
-
-
-#define DEFAULT_LATM_NR_OF_SUBFRAMES 1
-#define DEFAULT_LATM_SMC_REPEAT      8
-
-#define MAX_AAC_LAYERS        9
-
-#define LATM_MAX_PROGRAMS     1
-#define LATM_MAX_STREAM_ID   16
-
-#define LATM_MAX_LAYERS       1 /*MAX_AAC_LAYERS*/
-
-#define MAX_NR_OF_SUBFRAMES   2           /* set this carefully to avoid buffer overflows */
-
-typedef enum { LATMVAR_SIMPLE_SEQUENCE } LATM_VAR_MODE;
-
-typedef struct {
-  signed int frameLengthType;
-  signed int frameLengthBits;
-  signed int varFrameLengthTable[4];
-  signed int streamID;
-} LATM_LAYER_INFO;
-
-
-typedef struct {
-  LATM_LAYER_INFO      m_linfo[LATM_MAX_PROGRAMS][LATM_MAX_LAYERS];
-  CODER_CONFIG        *config[LATM_MAX_PROGRAMS][LATM_MAX_LAYERS];
-
-  LATM_VAR_MODE        varMode;
-  TRANSPORT_TYPE       tt;
-
-  int                  audioMuxLengthBytes;
-
-  int                  audioMuxLengthBytesPos;
-  int                  taraBufferFullness; /* state of the bit reservoir */
-  int                  varStreamCnt;
-  unsigned int         otherDataLenBytes;
-
-  UCHAR                latmFrameCounter;   /* Current frame number. Counts modulo muxConfigPeriod */
-  UCHAR                muxConfigPeriod;    /* Distance in frames between MuxConfig */
-
-  UCHAR                audioMuxVersion;    /* AMV1 supports transmission of taraBufferFullness and ASC lengths */
-  UCHAR                audioMuxVersionA;   /* for future extensions */
-
-  UCHAR                noProgram;
-  UCHAR                noLayer[LATM_MAX_PROGRAMS];
-  UCHAR                fractDelayPresent;
-
-  UCHAR                allStreamsSameTimeFraming;
-  UCHAR                subFrameCnt;        /* Current Subframe frame */
-  UCHAR                noSubframes;        /* Number of subframes    */
-  UINT                 latmSubframeStart;  /* Position of current subframe start */
-  UCHAR                noSubframes_next;
-
-  UCHAR                fillBits;           /* AudioMuxElement fill bits */
-  UCHAR                streamMuxConfigBits;
-
-} LATM_STREAM;
-
-typedef LATM_STREAM *HANDLE_LATM_STREAM;
-
-/**
- * \brief Initialize LATM_STREAM Handle. Creates automatically one program with one layer with
- *        the given layerConfig. The layerConfig must be persisten because references to this pointer
- *        are made at any time again.
- *        Use transportEnc_Latm_AddLayer() to add more programs/layers.
- *
- * \param hLatmStreamInfo HANDLE_LATM_STREAM handle
- * \param hBs Bitstream handle
- * \param layerConfig a valid CODER_CONFIG struct containing the current audio configuration parameters
- * \param audioMuxVersion the LATM audioMuxVersion to be used
- * \param tt the specific TRANSPORT_TYPE to be used, either TT_MP4_LOAS, TT_MP4_LATM_MCP1 or TT_MP4_LATM_MCP0 LOAS
- * \param cb callback information structure.
- *
- * \return an TRANSPORTENC_ERROR error code
- */
-TRANSPORTENC_ERROR transportEnc_Latm_Init(
-        HANDLE_LATM_STREAM hLatmStreamInfo,
-        HANDLE_FDK_BITSTREAM hBs,
-        CODER_CONFIG *layerConfig,
-        UINT audioMuxVersion,
-        TRANSPORT_TYPE tt,
-        CSTpCallBacks *cb
-        );
-
-/**
- * \brief Get bit demand of next LATM/LOAS header
- *
- * \param hAss HANDLE_LATM_STREAM handle
- * \param streamDataLength the length of the payload
- *
- * \return the number of bits required by the LATM/LOAS headers
- */
-unsigned int transportEnc_LatmCountTotalBitDemandHeader (
-                                                          HANDLE_LATM_STREAM hAss,
-                                                          unsigned int streamDataLength
-                                                         );
-
-/**
- * \brief Write LATM/LOAS header into given bitstream handle
- *
- * \param hLatmStreamInfo HANDLE_LATM_STREAM handle
- * \param hBitstream Bitstream handle
- * \param auBits amount of current payload bits
- * \param bufferFullness LATM buffer fullness value
- * \param cb callback information structure.
- *
- * \return an TRANSPORTENC_ERROR error code
- */
-TRANSPORTENC_ERROR
-transportEnc_LatmWrite (
-        HANDLE_LATM_STREAM    hAss,
-        HANDLE_FDK_BITSTREAM  hBitstream,
-        int                   auBits,
-        int                   bufferFullness,
-        CSTpCallBacks     *cb
-        );
-
-/**
- * \brief Adjust bit count relative to current subframe
- *
- * \param hAss HANDLE_LATM_STREAM handle
- * \param pBits pointer to an int, where the current frame bit count is contained,
- *        and where the subframe relative bit count will be returned into
- *
- * \return void
- */
-void transportEnc_LatmAdjustSubframeBits(HANDLE_LATM_STREAM    hAss,
-                                         int                  *pBits);
-
-/**
- * \brief Request an LATM frame, which may, or may not be available
- *
- * \param hAss HANDLE_LATM_STREAM handle
- * \param hBs Bitstream handle
- * \param pBytes pointer to an int, where the current frame byte count stored into.
- *        A return value of zero means that currently no LATM/LOAS frame can be returned.
- *        The latter is expected in case of multiple subframes being used.
- *
- * \return void
- */
-void transportEnc_LatmGetFrame(
-                               HANDLE_LATM_STREAM    hAss,
-                               HANDLE_FDK_BITSTREAM  hBs,
-                               int                  *pBytes
-                               );
-
-/**
- * \brief Write a StreamMuxConfig into the given bitstream handle
- *
- * \param hAss HANDLE_LATM_STREAM handle
- * \param hBs Bitstream handle
- * \param bufferFullness LATM buffer fullness value
- * \param cb callback information structure.
- *
- * \return void
- */
-TRANSPORTENC_ERROR
-CreateStreamMuxConfig(
-        HANDLE_LATM_STREAM hAss,
-        HANDLE_FDK_BITSTREAM hBs,
-        int bufferFullness,
-        CSTpCallBacks *cb
-        );
-
-
-#endif /* TPENC_LATM_H */
diff --git a/libMpegTPEnc/src/tpenc_lib.cpp b/libMpegTPEnc/src/tpenc_lib.cpp
deleted file mode 100644
index 058b8d4..0000000
--- a/libMpegTPEnc/src/tpenc_lib.cpp
+++ /dev/null
@@ -1,685 +0,0 @@
-
-/* -----------------------------------------------------------------------------------------------------------
-Software License for The Fraunhofer FDK AAC Codec Library for Android
-
-© Copyright  1995 - 2013 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 Encoder  ************************
-
-   Author(s): Manuel Jander
-   Description: MPEG Transport encode
-
-******************************************************************************/
-
-#include "tpenc_lib.h"
-
-/* library info */
-#include "version"
-
-#define MODULE_NAME "transportEnc"
-
-#include "tpenc_asc.h"
-#include "conv_string.h"
-
-#include "tpenc_adts.h"
-
-#include "tpenc_adif.h"
-
-#include "tpenc_dab.h"
-
-#include "tpenc_latm.h"
-
-typedef struct {
-  int curSubFrame;
-  int nSubFrames;
-  int prevBits;
-} RAWPACKETS_INFO;
-
-struct TRANSPORTENC
-{
-  CODER_CONFIG config;
-  TRANSPORT_TYPE transportFmt;          /*!< MPEG4 transport type. */
-
-  FDK_BITSTREAM bitStream;
-  UCHAR *bsBuffer;
-  INT bsBufferSize;
-
-  INT pceFrameCounter;                  /*!< Indicates frame period when PCE must be written in raw_data_block.
-                                             -1 means not to write a PCE in raw_dat_block. */
-  union {
-    STRUCT_ADTS adts;
-
-    ADIF_INFO adif;
-
-    STRUCT_DAB dab;
-
-    LATM_STREAM latm;
-
-    RAWPACKETS_INFO raw;
-
-
-
-  } writer;
-
-  CSTpCallBacks callbacks;
-};
-
-typedef struct _TRANSPORTENC_STRUCT TRANSPORTENC_STRUCT;
-
-
-/*
- * MEMORY Declaration
- */
-
-C_ALLOC_MEM(Ram_TransportEncoder, TRANSPORTENC, 1)
-
-TRANSPORTENC_ERROR transportEnc_Open( HANDLE_TRANSPORTENC *phTpEnc )
-{
-  HANDLE_TRANSPORTENC hTpEnc;
-
-  if ( phTpEnc == NULL ){
-    return TRANSPORTENC_INVALID_PARAMETER;
-  }
-
-  hTpEnc = GetRam_TransportEncoder(0);
-
-  if ( hTpEnc == NULL ) {
-    return TRANSPORTENC_NO_MEM;
-  }
-
-  *phTpEnc = hTpEnc;
-  return TRANSPORTENC_OK;
-}
-
-/**
- * \brief  Get frame period of PCE in raw_data_block.
- *
- *  - Write PCE only if necessary. PCE can be part of the ASC if chConfig==0 whererfore
- *    no additonal PCE will be written in raw_data_block.
- * - A matrixMixdown coefficient can only be written if chConfig is 5.0 or 5.1.
- * - The PCE repetition rate in raw_data_block can be controlled via headerPeriod parameter.
- *
- * \param channelConfig         Channel Configuration derived from Channel Mode
- * \param transportFmt          Format of the transport to be written.
- * \param headerPeriod          Chosen PCE frame repetition rate.
- * \param matrixMixdownA        Indicates if a valid Matrix Mixdown coefficient is available.
- *
- * \return  PCE frame repetition rate. -1 means no PCE present in raw_data_block.
- */
-static INT getPceRepetitionRate(
-        const int            channelConfig,
-        const TRANSPORT_TYPE transportFmt,
-        const int            headerPeriod,
-        const int            matrixMixdownA
-        )
-{
-  INT pceFrameCounter = -1; /* variable to be returned */
-
-  if (headerPeriod>0) {
-    switch ( channelConfig ) {
-      case 0:
-        switch (transportFmt) {
-          case TT_MP4_ADTS:
-          case TT_MP4_LATM_MCP0:
-          case TT_MP4_RAW:
-            pceFrameCounter = headerPeriod;
-            break;
-          case TT_MP4_ADIF:                  /* ADIF header comprises PCE */
-          case TT_MP4_LOAS:                  /* PCE in ASC if chChonfig==0 */
-          case TT_MP4_LATM_MCP1:             /* PCE in ASC if chChonfig==0 */
-          case TT_DRM:                       /* PCE not allowed in DRM */
-          case TT_DABPLUS:
-          default:
-            pceFrameCounter = -1;            /* no PCE in raw_data_block */
-        }
-        break;
-      case 5: /* MODE_1_2_2 */
-      case 6: /* MODE_1_2_2_1 */
-        /* matrixMixdownCoefficient can only be written if 5.0 and 5.1 config present. */
-        if (matrixMixdownA!=0) {
-          switch (transportFmt) {
-            case TT_MP4_ADIF:                /* ADIF header comprises PCE */
-            case TT_MP4_ADTS:
-            case TT_MP4_LOAS:                /* no PCE in ASC because chConfig!=0 */
-            case TT_MP4_LATM_MCP1:           /* no PCE in ASC because chConfig!=0 */
-            case TT_MP4_LATM_MCP0:
-            case TT_MP4_RAW:
-              pceFrameCounter = headerPeriod;
-              break;
-            case TT_DRM:                     /* PCE not allowed in DRM */
-            case TT_DABPLUS:
-            default:
-              pceFrameCounter = -1;          /* no PCE in raw_data_block */
-          } /* switch transportFmt */
-        } /* if matrixMixdownA!=0 */
-        break;
-      default:
-        pceFrameCounter = -1;                /* no PCE in raw_data_block */
-    } /* switch getChannelConfig() */
-  } /* if headerPeriod>0  */
-  else {
-    pceFrameCounter = -1;                    /* no PCE in raw_data_block */
-  }
-
-  return pceFrameCounter;
-}
-
-TRANSPORTENC_ERROR transportEnc_Init(
-        HANDLE_TRANSPORTENC hTpEnc,
-        UCHAR             *bsBuffer,
-        INT                bsBufferSize,
-        TRANSPORT_TYPE     transportFmt,
-        CODER_CONFIG      *cconfig,
-        UINT               flags
-        )
-{
-  /* Copy configuration structure */
-  FDKmemcpy(&hTpEnc->config, cconfig, sizeof(CODER_CONFIG));
-
-  /* Init transportEnc struct. */
-  hTpEnc->transportFmt = transportFmt;
-
-  hTpEnc->bsBuffer = bsBuffer;
-  hTpEnc->bsBufferSize = bsBufferSize;
-
-  FDKinitBitStream(&hTpEnc->bitStream, hTpEnc->bsBuffer, hTpEnc->bsBufferSize, 0, BS_WRITER);
-
-  switch (transportFmt) {
-
-  case TT_MP4_ADIF:
-    /* Sanity checks */
-    if ( (hTpEnc->config.aot != AOT_AAC_LC)
-       ||(hTpEnc->config.samplesPerFrame != 1024))
-    {
-      return TRANSPORTENC_INVALID_PARAMETER;
-    }
-    hTpEnc->writer.adif.headerWritten = 0;
-    hTpEnc->writer.adif.samplingRate = hTpEnc->config.samplingRate;
-    hTpEnc->writer.adif.bitRate = hTpEnc->config.bitRate;
-    hTpEnc->writer.adif.profile = ((int)hTpEnc->config.aot) - 1;
-    hTpEnc->writer.adif.cm = hTpEnc->config.channelMode;
-    hTpEnc->writer.adif.bVariableRate = 0;
-    hTpEnc->writer.adif.instanceTag = 0;
-    break;
-
-  case TT_MP4_ADTS:
-    /* Sanity checks */
-    if ( ( hTpEnc->config.aot != AOT_AAC_LC)
-       ||(hTpEnc->config.samplesPerFrame != 1024) )
-    {
-      return TRANSPORTENC_INVALID_PARAMETER;
-    }
-    if ( adtsWrite_Init(&hTpEnc->writer.adts, &hTpEnc->config) != 0) {
-      return TRANSPORTENC_INVALID_PARAMETER;
-    }
-    break;
-
-  case TT_DABPLUS:
-    /* Sanity checks */
-    if ( ( hTpEnc->config.aot != AOT_AAC_LC)
-       ||(hTpEnc->config.samplesPerFrame != 960) )
-    {
-      return TRANSPORTENC_INVALID_PARAMETER;
-    }
-    if ( dabWrite_Init(&hTpEnc->writer.dab, &hTpEnc->config) != 0) {
-      return TRANSPORTENC_INVALID_PARAMETER;
-    }
-    break;
-
-  case TT_MP4_LOAS:
-  case TT_MP4_LATM_MCP0:
-  case TT_MP4_LATM_MCP1:
-    {
-      TRANSPORTENC_ERROR error;
-
-      error = transportEnc_Latm_Init(
-                &hTpEnc->writer.latm,
-                &hTpEnc->bitStream,
-                &hTpEnc->config,
-                 flags & TP_FLAG_LATM_AMV,
-                 transportFmt,
-                &hTpEnc->callbacks
-                 );
-      if (error != TRANSPORTENC_OK) {
-        return error;
-      }
-    }
-    break;
-
-  case TT_MP4_RAW:
-    hTpEnc->writer.raw.curSubFrame = 0;
-    hTpEnc->writer.raw.nSubFrames = hTpEnc->config.nSubFrames;
-    break;
-
-
-
-  default:
-    return TRANSPORTENC_INVALID_PARAMETER;
-  }
-
-  /* pceFrameCounter indicates if PCE must be written in raw_data_block. */
-  hTpEnc->pceFrameCounter = getPceRepetitionRate(
-                    getChannelConfig(hTpEnc->config.channelMode),
-                    transportFmt,
-                    hTpEnc->config.headerPeriod,
-                    hTpEnc->config.matrixMixdownA);
-
-  return TRANSPORTENC_OK;
-}
-
-HANDLE_FDK_BITSTREAM transportEnc_GetBitstream( HANDLE_TRANSPORTENC hTp )
-{
-  return &hTp->bitStream;
-}
-
-int transportEnc_RegisterSbrCallback( HANDLE_TRANSPORTENC hTpEnc, const cbSbr_t cbSbr, void* user_data)
-{
-  if (hTpEnc == NULL) {
-    return -1;
-  }
-  hTpEnc->callbacks.cbSbr = cbSbr;
-  hTpEnc->callbacks.cbSbrData = user_data;
-  return 0;
-}
-
-
-TRANSPORTENC_ERROR transportEnc_WriteAccessUnit(
-                                               HANDLE_TRANSPORTENC hTp,
-                                               INT frameUsedBits,
-                                               int bufferFullness,
-                                               int ncc
-                                              )
-{
-  TRANSPORTENC_ERROR err = TRANSPORTENC_OK;
-
-  if (!hTp) {
-    return TRANSPORTENC_INVALID_PARAMETER;
-  }
-  HANDLE_FDK_BITSTREAM hBs = &hTp->bitStream;
-
-  /* In case of writing PCE in raw_data_block frameUsedBits must be adapted. */
-  if (hTp->pceFrameCounter>=hTp->config.headerPeriod) {
-    frameUsedBits += transportEnc_GetPCEBits(hTp->config.channelMode, hTp->config.matrixMixdownA, 3); /* Consider 3 bits ID signalling in alignment */
-  }
-
-  switch (hTp->transportFmt) {
-    case TT_MP4_ADIF:
-      FDKinitBitStream(&hTp->bitStream, hTp->bsBuffer, hTp->bsBufferSize, 0, BS_WRITER);
-      adifWrite_EncodeHeader(
-             &hTp->writer.adif,
-              hBs,
-              bufferFullness
-              );
-      break;
-    case TT_MP4_ADTS:
-      bufferFullness /= ncc;                          /* Number of Considered Channels */
-      bufferFullness /= 32;
-      bufferFullness = FDKmin(0x7FF, bufferFullness); /* Signal variable rate */
-      adtsWrite_EncodeHeader(
-             &hTp->writer.adts,
-             &hTp->bitStream,
-              bufferFullness,
-              frameUsedBits
-              );
-      break;
-    case TT_DABPLUS:
-      bufferFullness /= ncc;                          /* Number of Considered Channels */
-      bufferFullness /= 32;
-      bufferFullness = FDKmin(0x7FF, bufferFullness); /* Signal variable rate */
-      dabWrite_EncodeHeader(
-             &hTp->writer.dab,
-             &hTp->bitStream,
-              bufferFullness,
-              frameUsedBits
-              );
-      break;
-    case TT_MP4_LOAS:
-    case TT_MP4_LATM_MCP0:
-    case TT_MP4_LATM_MCP1:
-      bufferFullness /= ncc;                         /* Number of Considered Channels */
-      bufferFullness /= 32;
-      bufferFullness = FDKmin(0xFF, bufferFullness); /* Signal variable rate */
-      transportEnc_LatmWrite(
-             &hTp->writer.latm,
-              hBs,
-              frameUsedBits,
-              bufferFullness,
-             &hTp->callbacks
-              );
-    break;
-    case TT_MP4_RAW:
-      if (hTp->writer.raw.curSubFrame >= hTp->writer.raw.nSubFrames) {
-        hTp->writer.raw.curSubFrame = 0;
-        FDKinitBitStream(&hTp->bitStream, hTp->bsBuffer, hTp->bsBufferSize, 0, BS_WRITER);
-      }
-      hTp->writer.raw.prevBits = FDKgetValidBits(hBs);
-      break;
-    default:
-      err = TRANSPORTENC_UNSUPPORTED_FORMAT;
-      break;
-  }
-
-  /* Write PCE in raw_data_block if required */
-  if (hTp->pceFrameCounter>=hTp->config.headerPeriod) {
-    INT crcIndex = 0;
-    /* Align inside PCE with repsect to the first bit of the raw_data_block() */
-    UINT alignAnchor = FDKgetValidBits(&hTp->bitStream);
-
-    /* Write PCE element ID bits */
-    FDKwriteBits(&hTp->bitStream, ID_PCE, 3);
-
-    if ( (hTp->transportFmt==TT_MP4_ADTS) && !hTp->writer.adts.protection_absent) {
-      crcIndex = adtsWrite_CrcStartReg(&hTp->writer.adts, &hTp->bitStream, 0);
-    }
-
-    /* Write PCE as first raw_data_block element */
-    transportEnc_writePCE(&hTp->bitStream, hTp->config.channelMode, hTp->config.samplingRate, 0, 1, hTp->config.matrixMixdownA, (hTp->config.flags&CC_PSEUDO_SURROUND)?1:0, alignAnchor);
-
-    if ( (hTp->transportFmt==TT_MP4_ADTS) && !hTp->writer.adts.protection_absent) {
-      adtsWrite_CrcEndReg(&hTp->writer.adts, &hTp->bitStream, crcIndex);
-    }
-    hTp->pceFrameCounter = 0; /* reset pce frame counter */
-  }
-
-  if (hTp->pceFrameCounter!=-1) {
-    hTp->pceFrameCounter++; /* Update pceFrameCounter only if PCE writing is active. */
-  }
-
-  return err;
-}
-
-
-TRANSPORTENC_ERROR transportEnc_EndAccessUnit(HANDLE_TRANSPORTENC hTp, int *bits)
-{
-  switch (hTp->transportFmt) {
-    case TT_MP4_LATM_MCP0:
-    case TT_MP4_LATM_MCP1:
-    case TT_MP4_LOAS:
-      transportEnc_LatmAdjustSubframeBits(&hTp->writer.latm, bits);
-      break;
-    case TT_MP4_ADTS:
-      adtsWrite_EndRawDataBlock(&hTp->writer.adts, &hTp->bitStream, bits);
-      break;
-    case TT_DABPLUS:
-      dabWrite_EndRawDataBlock(&hTp->writer.dab, &hTp->bitStream, bits);
-      break;
-    case TT_MP4_ADIF:
-      /* Substract ADIF header from AU bits, not to be considered. */
-      *bits -= adifWrite_GetHeaderBits(&hTp->writer.adif);
-      hTp->writer.adif.headerWritten = 1;
-      break;
-    case TT_MP4_RAW:
-      *bits -= hTp->writer.raw.prevBits;
-      break;
-    default:
-      break;
-  }
-
-  return TRANSPORTENC_OK;
-}
-
-TRANSPORTENC_ERROR transportEnc_GetFrame(HANDLE_TRANSPORTENC hTpEnc, int *nbytes)
-{
-  HANDLE_FDK_BITSTREAM hBs = &hTpEnc->bitStream;
-
-  switch (hTpEnc->transportFmt) {
-    case TT_MP4_LATM_MCP0:
-    case TT_MP4_LATM_MCP1:
-    case TT_MP4_LOAS:
-      *nbytes = hTpEnc->bsBufferSize;
-      transportEnc_LatmGetFrame(&hTpEnc->writer.latm, hBs, nbytes);
-      break;
-    case TT_MP4_ADTS:
-      if (hTpEnc->writer.adts.currentBlock >= hTpEnc->writer.adts.num_raw_blocks+1) {
-        *nbytes = (FDKgetValidBits(hBs) + 7)>>3;
-        hTpEnc->writer.adts.currentBlock = 0;
-      } else {
-        *nbytes = 0;
-      }
-      break;
-    case TT_DABPLUS:
-      if (hTpEnc->writer.dab.currentBlock >= hTpEnc->writer.dab.num_raw_blocks+1) {
-        *nbytes = (FDKgetValidBits(hBs) + 7)>>3;
-        hTpEnc->writer.dab.currentBlock = 0;
-      } else {
-        *nbytes = 0;
-      }
-      break;
-    case TT_MP4_ADIF:
-      FDK_ASSERT((INT)FDKgetValidBits(hBs) >= 0);
-      *nbytes = (FDKgetValidBits(hBs) + 7)>>3;
-      break;
-    case TT_MP4_RAW:
-      FDKsyncCache(hBs);
-      hTpEnc->writer.raw.curSubFrame++;
-      *nbytes = ((FDKgetValidBits(hBs)-hTpEnc->writer.raw.prevBits) + 7)>>3;
-      break;
-    default:
-      break;
-  }
-
-  return TRANSPORTENC_OK;
-}
-
-INT transportEnc_GetStaticBits( HANDLE_TRANSPORTENC hTp, int auBits )
-{
-  INT nbits = 0, nPceBits = 0;
-
-  /* Write PCE within raw_data_block in transport lib. */
-  if (hTp->pceFrameCounter>=hTp->config.headerPeriod) {
-    nPceBits = transportEnc_GetPCEBits(hTp->config.channelMode, hTp->config.matrixMixdownA, 3); /* Consider 3 bits ID signalling in alignment */
-    auBits += nPceBits; /* Adapt required raw_data_block bit consumtpion for AU length information e.g. in LATM/LOAS configuration. */
-  }
-
-  switch (hTp->transportFmt) {
-    case TT_MP4_ADIF:
-    case TT_MP4_RAW:
-      nbits = 0; /* Do not consider the ADIF header into the total bitrate */
-      break;
-    case TT_MP4_ADTS:
-      nbits = adtsWrite_GetHeaderBits(&hTp->writer.adts);
-      break;
-    case TT_DABPLUS:
-      nbits = dabWrite_CountTotalBitDemandHeader(&hTp->writer.dab, auBits);
-      break;
-    case TT_MP4_LOAS:
-    case TT_MP4_LATM_MCP0:
-    case TT_MP4_LATM_MCP1:
-      nbits = transportEnc_LatmCountTotalBitDemandHeader( &hTp->writer.latm, auBits );
-      break;
-    default:
-      nbits = 0;
-      break;
-  }
-
-  /* PCE is written in the transport library therefore the bit consumption is part of the transport static bits. */
-  nbits += nPceBits;
-
-  return nbits;
-}
-
-void transportEnc_Close(HANDLE_TRANSPORTENC *phTp)
-{
-  if (phTp != NULL)
-  {
-    if (*phTp != NULL) {
-      FreeRam_TransportEncoder(phTp);
-    }
-  }
-}
-
-int transportEnc_CrcStartReg(HANDLE_TRANSPORTENC hTpEnc, int mBits)
-{
-  int crcReg = 0;
-
-  switch (hTpEnc->transportFmt) {
-  case TT_MP4_ADTS:
-    crcReg = adtsWrite_CrcStartReg(&hTpEnc->writer.adts, &hTpEnc->bitStream, mBits);
-    break;
-  case TT_DABPLUS:
-    crcReg = dabWrite_CrcStartReg(&hTpEnc->writer.dab, &hTpEnc->bitStream, mBits);
-    break;
-  default:
-    break;
-  }
-
-  return crcReg;
-}
-
-void transportEnc_CrcEndReg(HANDLE_TRANSPORTENC hTpEnc, int reg)
-{
-  switch (hTpEnc->transportFmt) {
-  case TT_MP4_ADTS:
-    adtsWrite_CrcEndReg(&hTpEnc->writer.adts, &hTpEnc->bitStream, reg);
-    break;
-  case TT_DABPLUS:
-    dabWrite_CrcEndReg(&hTpEnc->writer.dab, &hTpEnc->bitStream, reg);
-    break;
-  default:
-    break;
-  }
-}
-
-
-TRANSPORTENC_ERROR transportEnc_GetConf(HANDLE_TRANSPORTENC  hTpEnc,
-                                        CODER_CONFIG        *cc,
-                                        FDK_BITSTREAM       *dataBuffer,
-                                        UINT                *confType)
-{
-  TRANSPORTENC_ERROR tpErr = TRANSPORTENC_OK;
-  HANDLE_LATM_STREAM hLatmConfig = &hTpEnc->writer.latm;
-
-  *confType = 0; /* set confType variable to default */
-
-  /* write StreamMuxConfig or AudioSpecificConfig depending on format used */
-  switch (hTpEnc->transportFmt)
-  {
-    case TT_MP4_LATM_MCP0:
-    case TT_MP4_LATM_MCP1:
-    case TT_MP4_LOAS:
-      tpErr = CreateStreamMuxConfig(hLatmConfig, dataBuffer, 0, &hTpEnc->callbacks);
-      *confType = 1; /* config is SMC */
-      break;
-    default:
-      if (transportEnc_writeASC(dataBuffer, cc, &hTpEnc->callbacks) != 0) {
-        tpErr = TRANSPORTENC_UNKOWN_ERROR;
-      }
-  }
-
-  return tpErr;
-
-}
-
-TRANSPORTENC_ERROR transportEnc_GetLibInfo( LIB_INFO *info )
-{
-  int i;
-
-  if (info == NULL) {
-    return TRANSPORTENC_INVALID_PARAMETER;
-  }
-  /* 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 TRANSPORTENC_UNKOWN_ERROR;
-  }
-  info += i;
-
-  info->module_id = FDK_TPENC;
-  info->version = LIB_VERSION(TP_LIB_VL0, TP_LIB_VL1, TP_LIB_VL2);
-  LIB_VERSION_STRING(info);
-  info->build_date = __DATE__;
-  info->build_time = __TIME__;
-  info->title = TP_LIB_TITLE;
-
-  /* Set flags */
-  info->flags = 0
-    | CAPF_ADIF
-    | CAPF_ADTS
-    | CAPF_LATM
-    | CAPF_LOAS
-    | CAPF_RAWPACKETS
-    | CAPF_DAB_AAC
-    ;
-
-  return TRANSPORTENC_OK;
-}
-
diff --git a/libMpegTPEnc/src/version b/libMpegTPEnc/src/version
deleted file mode 100644
index 2803347..0000000
--- a/libMpegTPEnc/src/version
+++ /dev/null
@@ -1,8 +0,0 @@
-
-/* library info */
-#define TP_LIB_VL0 2
-#define TP_LIB_VL1 3
-#define TP_LIB_VL2 3
-#define TP_LIB_TITLE "MPEG Transport"
-#define TP_LIB_BUILD_DATE __DATE__
-#define TP_LIB_BUILD_TIME __TIME__