Snapshot 2bda038c163298531d47394bc2c09e1409c5d0db

Change-Id: If584e579464f28b97d50e51fc76ba654a5536c54

10 files changed, 3329 insertions, 0 deletions

diff --git a/libMpegTPEnc/src/tpenc_adif.cpp b/libMpegTPEnc/src/tpenc_adif.cpp
new file mode 100644
index 0000000..06bd30b
--- /dev/null
+++ b/libMpegTPEnc/src/tpenc_adif.cpp
@@ -0,0 +1,182 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/******************************** MPEG Audio Encoder **************************
+
+   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
new file mode 100644
index 0000000..0921712
--- /dev/null
+++ b/libMpegTPEnc/src/tpenc_adif.h
@@ -0,0 +1,135 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/******************************** MPEG Audio Encoder **************************
+
+   Initial author:       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
new file mode 100644
index 0000000..39460f0
--- /dev/null
+++ b/libMpegTPEnc/src/tpenc_adts.cpp
@@ -0,0 +1,315 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/******************************** MPEG Audio Encoder **************************
+
+   Initial author:       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
new file mode 100644
index 0000000..9123f1e
--- /dev/null
+++ b/libMpegTPEnc/src/tpenc_adts.h
@@ -0,0 +1,218 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/******************************** MPEG Audio Encoder **************************
+
+   Initial author:       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
new file mode 100644
index 0000000..6200c14
--- /dev/null
+++ b/libMpegTPEnc/src/tpenc_asc.cpp
@@ -0,0 +1,552 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/*****************************  MPEG-4 AAC 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 2 elements (SCE,CPE) are front channel elements.
+ *                          - Following element (CPE) is a side channel element.
+ *                          - 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,                {  2, 1, 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 } } }
+};
+
+
+/**
+ * \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&0x1, 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, 0, 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->extAOT == AOT_SBR || config->extAOT == AOT_PS)
+    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->extAOT == AOT_SBR || config->extAOT == AOT_PS) {
+    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;
+  }
+
+  /* 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
new file mode 100644
index 0000000..990f74f
--- /dev/null
+++ b/libMpegTPEnc/src/tpenc_asc.h
@@ -0,0 +1,142 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/*****************************  MPEG-4 AAC 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_latm.cpp b/libMpegTPEnc/src/tpenc_latm.cpp
new file mode 100644
index 0000000..54fd717
--- /dev/null
+++ b/libMpegTPEnc/src/tpenc_latm.cpp
@@ -0,0 +1,882 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/*****************************  MPEG-4 AAC 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
new file mode 100644
index 0000000..fd95a60
--- /dev/null
+++ b/libMpegTPEnc/src/tpenc_latm.h
@@ -0,0 +1,264 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/*****************************  MPEG-4 AAC 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
new file mode 100644
index 0000000..4f3177d
--- /dev/null
+++ b/libMpegTPEnc/src/tpenc_lib.cpp
@@ -0,0 +1,631 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/**************************  MPEG-4 Transport 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_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;
+
+    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 = GetRam_TransportEncoder(0);
+
+  if ( hTpEnc == NULL ) {
+    return TRANSPORTENC_INVALID_PARAMETER;
+  }
+
+  *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 */
+          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 */
+            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_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_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, 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_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_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_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;
+  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;
+  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
+    ;
+
+  return TRANSPORTENC_OK;
+}
+
diff --git a/libMpegTPEnc/src/version b/libMpegTPEnc/src/version
new file mode 100644
index 0000000..5b9a425
--- /dev/null
+++ b/libMpegTPEnc/src/version
@@ -0,0 +1,8 @@
+
+/* library info */
+#define TP_LIB_VL0 2
+#define TP_LIB_VL1 2
+#define TP_LIB_VL2 1
+#define TP_LIB_TITLE "MPEG Transport"
+#define TP_LIB_BUILD_DATE __DATE__
+#define TP_LIB_BUILD_TIME __TIME__