Include patched FDK-AAC in the repository

The initial idea was to get the DAB+ patch into upstream, but since
that follows the android source releases, there is no place for a custom
DAB+ patch there.

So instead of having to maintain a patched fdk-aac that has to have the
same .so version as the distribution package on which it is installed,
we prefer having a separate fdk-aac-dab library to avoid collision.

At that point, there's no reason to keep fdk-aac in a separate
repository, as odr-audioenc is the only tool that needs DAB+ encoding
support. Including it here simplifies installation, and makes it
consistent with toolame-dab, also shipped in this repository.

DAB+ decoding support (needed by ODR-SourceCompanion, dablin, etisnoop,
welle.io and others) can be done using upstream FDK-AAC.

14 files changed, 5446 insertions, 0 deletions

diff --git a/fdk-aac/libMpegTPEnc/include/tp_data.h b/fdk-aac/libMpegTPEnc/include/tp_data.h
new file mode 100644
index 0000000..00de356
--- /dev/null
+++ b/fdk-aac/libMpegTPEnc/include/tp_data.h
@@ -0,0 +1,466 @@
+/* -----------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2018 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 transport format encoder library *********************
+
+   Author(s):   Manuel Jander
+
+   Description: MPEG Transport data tables
+
+*******************************************************************************/
+
+#ifndef TP_DATA_H
+#define TP_DATA_H
+
+#include "machine_type.h"
+#include "FDK_audio.h"
+#include "FDK_bitstream.h"
+
+/*
+ * Configuration
+ */
+
+#define TP_USAC_MAX_SPEAKERS (24)
+
+#define TP_USAC_MAX_EXT_ELEMENTS ((24))
+
+#define TP_USAC_MAX_ELEMENTS ((24) + TP_USAC_MAX_EXT_ELEMENTS)
+
+#define TP_USAC_MAX_CONFIG_LEN                                         \
+  512 /* next power of two of maximum of escapedValue(hBs, 4, 4, 8) in \
+         AudioPreRoll() (285) */
+
+#define TPDEC_USAC_NUM_CONFIG_CHANGE_FRAMES \
+  (1) /* Number of frames for config change in USAC */
+
+enum {
+  TPDEC_FLUSH_OFF = 0,
+  TPDEC_RSV60_CFG_CHANGE_ATSC_FLUSH_ON = 1,
+  TPDEC_RSV60_DASH_IPF_ATSC_FLUSH_ON = 2,
+  TPDEC_USAC_DASH_IPF_FLUSH_ON = 3
+};
+
+enum {
+  TPDEC_BUILD_UP_OFF = 0,
+  TPDEC_RSV60_BUILD_UP_ON = 1,
+  TPDEC_RSV60_BUILD_UP_ON_IN_BAND = 2,
+  TPDEC_USAC_BUILD_UP_ON = 3,
+  TPDEC_RSV60_BUILD_UP_IDLE = 4,
+  TPDEC_RSV60_BUILD_UP_IDLE_IN_BAND = 5
+};
+
+/**
+ * ProgramConfig struct.
+ */
+/* ISO/IEC 14496-3 4.4.1.1 Table 4.2 Program config element */
+#define PC_FSB_CHANNELS_MAX 16 /* Front/Side/Back channels */
+#define PC_LFE_CHANNELS_MAX 4
+#define PC_ASSOCDATA_MAX 8
+#define PC_CCEL_MAX 16 /* CC elements */
+#define PC_COMMENTLENGTH 256
+#define PC_NUM_HEIGHT_LAYER 3
+
+typedef struct {
+  /* PCE bitstream elements: */
+  UCHAR ElementInstanceTag;
+  UCHAR Profile;
+  UCHAR SamplingFrequencyIndex;
+  UCHAR NumFrontChannelElements;
+  UCHAR NumSideChannelElements;
+  UCHAR NumBackChannelElements;
+  UCHAR NumLfeChannelElements;
+  UCHAR NumAssocDataElements;
+  UCHAR NumValidCcElements;
+
+  UCHAR MonoMixdownPresent;
+  UCHAR MonoMixdownElementNumber;
+
+  UCHAR StereoMixdownPresent;
+  UCHAR StereoMixdownElementNumber;
+
+  UCHAR MatrixMixdownIndexPresent;
+  UCHAR MatrixMixdownIndex;
+  UCHAR PseudoSurroundEnable;
+
+  UCHAR FrontElementIsCpe[PC_FSB_CHANNELS_MAX];
+  UCHAR FrontElementTagSelect[PC_FSB_CHANNELS_MAX];
+  UCHAR FrontElementHeightInfo[PC_FSB_CHANNELS_MAX];
+
+  UCHAR SideElementIsCpe[PC_FSB_CHANNELS_MAX];
+  UCHAR SideElementTagSelect[PC_FSB_CHANNELS_MAX];
+  UCHAR SideElementHeightInfo[PC_FSB_CHANNELS_MAX];
+
+  UCHAR BackElementIsCpe[PC_FSB_CHANNELS_MAX];
+  UCHAR BackElementTagSelect[PC_FSB_CHANNELS_MAX];
+  UCHAR BackElementHeightInfo[PC_FSB_CHANNELS_MAX];
+
+  UCHAR LfeElementTagSelect[PC_LFE_CHANNELS_MAX];
+
+  UCHAR AssocDataElementTagSelect[PC_ASSOCDATA_MAX];
+
+  UCHAR CcElementIsIndSw[PC_CCEL_MAX];
+  UCHAR ValidCcElementTagSelect[PC_CCEL_MAX];
+
+  UCHAR CommentFieldBytes;
+  UCHAR Comment[PC_COMMENTLENGTH];
+
+  /* Helper variables for administration: */
+  UCHAR isValid; /*!< Flag showing if PCE has been read successfully. */
+  UCHAR
+  NumChannels; /*!< Amount of audio channels summing all channel elements
+                  including LFEs */
+  UCHAR NumEffectiveChannels; /*!< Amount of audio channels summing only SCEs
+                                 and CPEs */
+  UCHAR elCounter;
+
+} CProgramConfig;
+
+typedef enum {
+  ASCEXT_UNKOWN = -1,
+  ASCEXT_SBR = 0x2b7,
+  ASCEXT_PS = 0x548,
+  ASCEXT_MPS = 0x76a,
+  ASCEXT_SAOC = 0x7cb,
+  ASCEXT_LDMPS = 0x7cc
+
+} TP_ASC_EXTENSION_ID;
+
+/**
+ * GaSpecificConfig struct
+ */
+typedef struct {
+  UINT m_frameLengthFlag;
+  UINT m_dependsOnCoreCoder;
+  UINT m_coreCoderDelay;
+
+  UINT m_extensionFlag;
+  UINT m_extensionFlag3;
+
+  UINT m_layer;
+  UINT m_numOfSubFrame;
+  UINT m_layerLength;
+
+} CSGaSpecificConfig;
+
+typedef enum {
+  ELDEXT_TERM = 0x0,         /* Termination tag */
+  ELDEXT_SAOC = 0x1,         /* SAOC config */
+  ELDEXT_LDSAC = 0x2,        /* LD MPEG Surround config */
+  ELDEXT_DOWNSCALEINFO = 0x3 /* ELD sample rate adaptation */
+  /* reserved */
+} ASC_ELD_EXT_TYPE;
+
+typedef struct {
+  UCHAR m_frameLengthFlag;
+
+  UCHAR m_sbrPresentFlag;
+  UCHAR
+  m_useLdQmfTimeAlign; /* Use LD-MPS QMF in SBR to achive time alignment */
+  UCHAR m_sbrSamplingRate;
+  UCHAR m_sbrCrcFlag;
+  UINT m_downscaledSamplingFrequency;
+
+} CSEldSpecificConfig;
+
+typedef struct {
+  USAC_EXT_ELEMENT_TYPE usacExtElementType;
+  USHORT usacExtElementConfigLength;
+  USHORT usacExtElementDefaultLength;
+  UCHAR usacExtElementPayloadFrag;
+  UCHAR usacExtElementHasAudioPreRoll;
+} CSUsacExtElementConfig;
+
+typedef struct {
+  MP4_ELEMENT_ID usacElementType;
+  UCHAR m_noiseFilling;
+  UCHAR m_harmonicSBR;
+  UCHAR m_interTes;
+  UCHAR m_pvc;
+  UCHAR m_stereoConfigIndex;
+  CSUsacExtElementConfig extElement;
+} CSUsacElementConfig;
+
+typedef struct {
+  UCHAR m_frameLengthFlag;
+  UCHAR m_coreSbrFrameLengthIndex;
+  UCHAR m_sbrRatioIndex;
+  UCHAR m_nUsacChannels; /* number of audio channels signaled in
+                            UsacDecoderConfig() / rsv603daDecoderConfig() via
+                            numElements and usacElementType */
+  UCHAR m_channelConfigurationIndex;
+  UINT m_usacNumElements;
+  CSUsacElementConfig element[TP_USAC_MAX_ELEMENTS];
+
+  UCHAR numAudioChannels;
+  UCHAR m_usacConfigExtensionPresent;
+  UCHAR elementLengthPresent;
+  UCHAR UsacConfig[TP_USAC_MAX_CONFIG_LEN];
+  USHORT UsacConfigBits;
+} CSUsacConfig;
+
+/**
+ * Audio configuration struct, suitable for encoder and decoder configuration.
+ */
+typedef struct {
+  /* XYZ Specific Data */
+  union {
+    CSGaSpecificConfig
+        m_gaSpecificConfig; /**< General audio specific configuration. */
+    CSEldSpecificConfig m_eldSpecificConfig; /**< ELD specific configuration. */
+    CSUsacConfig m_usacConfig; /**< USAC specific configuration               */
+  } m_sc;
+
+  /* Common ASC parameters */
+  CProgramConfig m_progrConfigElement; /**< Program configuration. */
+
+  AUDIO_OBJECT_TYPE m_aot;  /**< Audio Object Type.  */
+  UINT m_samplingFrequency; /**< Samplerate. */
+  UINT m_samplesPerFrame;   /**< Amount of samples per frame.   */
+  UINT m_directMapping; /**< Document this please !!                         */
+
+  AUDIO_OBJECT_TYPE m_extensionAudioObjectType; /**< Audio object type */
+  UINT m_extensionSamplingFrequency;            /**< Samplerate            */
+
+  SCHAR m_channelConfiguration; /**< Channel configuration index */
+
+  SCHAR m_epConfig;  /**< Error protection index                           */
+  SCHAR m_vcb11Flag; /**< aacSectionDataResilienceFlag                     */
+  SCHAR m_rvlcFlag;  /**< aacScalefactorDataResilienceFlag                 */
+  SCHAR m_hcrFlag;   /**< aacSpectralDataResilienceFlag                    */
+
+  SCHAR m_sbrPresentFlag; /**< Flag indicating the presence of SBR data in the
+                             bitstream               */
+  SCHAR
+  m_psPresentFlag; /**< Flag indicating the presence of parametric stereo
+                      data in the bitstream */
+  UCHAR m_samplingFrequencyIndex;          /**< Samplerate index          */
+  UCHAR m_extensionSamplingFrequencyIndex; /**< Samplerate index */
+  SCHAR m_extensionChannelConfiguration;   /**< Channel configuration index   */
+
+  UCHAR
+  configMode; /**< The flag indicates if the callback shall work in memory
+                 allocation mode or in config change detection mode */
+  UCHAR AacConfigChanged; /**< The flag will be set if at least one aac config
+                             parameter has changed that requires a memory
+                             reconfiguration, otherwise it will be cleared */
+  UCHAR SbrConfigChanged; /**< The flag will be set if at least one sbr config
+                             parameter has changed that requires a memory
+                             reconfiguration, otherwise it will be cleared */
+  UCHAR SacConfigChanged; /**< The flag will be set if at least one sac config
+                             parameter has changed that requires a memory
+                             reconfiguration, otherwise it will be cleared */
+
+  UCHAR
+  config[TP_USAC_MAX_CONFIG_LEN]; /**< Configuration stored as bitstream */
+  UINT configBits;                /**< Configuration length in bits */
+
+} CSAudioSpecificConfig;
+
+typedef struct {
+  SCHAR flushCnt;      /**< Flush frame counter */
+  UCHAR flushStatus;   /**< Flag indicates flush mode: on|off */
+  SCHAR buildUpCnt;    /**< Build up frame counter */
+  UCHAR buildUpStatus; /**< Flag indicates build up mode: on|off */
+  UCHAR cfgChanged; /**< Flag indicates that the config changed and the decoder
+                       needs to be initialized again via callback. Make sure
+                       that memory is freed before initialization. */
+  UCHAR contentChanged; /**< Flag indicates that the content changed i.e. a
+                           right truncation occured before */
+  UCHAR forceCfgChange; /**< Flag indicates if config change has to be forced
+                           even if new config is the same */
+} CCtrlCFGChange;
+
+typedef INT (*cbUpdateConfig_t)(void *, const CSAudioSpecificConfig *,
+                                const UCHAR configMode, UCHAR *configChanged);
+typedef INT (*cbFreeMem_t)(void *, const CSAudioSpecificConfig *);
+typedef INT (*cbCtrlCFGChange_t)(void *, const CCtrlCFGChange *);
+typedef INT (*cbSsc_t)(void *, HANDLE_FDK_BITSTREAM,
+                       const AUDIO_OBJECT_TYPE coreCodec,
+                       const INT samplingRate, const INT frameSize,
+                       const INT stereoConfigIndex,
+                       const INT coreSbrFrameLengthIndex, const INT configBytes,
+                       const UCHAR configMode, UCHAR *configChanged);
+
+typedef INT (*cbSbr_t)(void *self, HANDLE_FDK_BITSTREAM hBs,
+                       const INT sampleRateIn, const INT sampleRateOut,
+                       const INT samplesPerFrame,
+                       const AUDIO_OBJECT_TYPE coreCodec,
+                       const MP4_ELEMENT_ID elementID, const INT elementIndex,
+                       const UCHAR harmonicSbr, const UCHAR stereoConfigIndex,
+                       const UCHAR configMode, UCHAR *configChanged,
+                       const INT downscaleFactor);
+
+typedef INT (*cbUsac_t)(void *self, HANDLE_FDK_BITSTREAM hBs);
+
+typedef INT (*cbUniDrc_t)(void *self, HANDLE_FDK_BITSTREAM hBs,
+                          const INT fullPayloadLength, const INT payloadType,
+                          const INT subStreamIndex, const INT payloadStart,
+                          const AUDIO_OBJECT_TYPE);
+
+typedef struct {
+  cbUpdateConfig_t cbUpdateConfig; /*!< Function pointer for Config change
+                                      notify callback.  */
+  void *cbUpdateConfigData; /*!< User data pointer for Config change notify
+                               callback. */
+  cbFreeMem_t cbFreeMem;    /*!< Function pointer for free memory callback.  */
+  void *cbFreeMemData;      /*!< User data pointer for free memory callback. */
+  cbCtrlCFGChange_t cbCtrlCFGChange; /*!< Function pointer for config change
+                                        control callback. */
+  void *cbCtrlCFGChangeData; /*!< User data pointer for config change control
+                                callback. */
+  cbSsc_t cbSsc;             /*!< Function pointer for SSC parser callback. */
+  void *cbSscData;           /*!< User data pointer for SSC parser callback. */
+  cbSbr_t cbSbr;   /*!< Function pointer for SBR header parser callback. */
+  void *cbSbrData; /*!< User data pointer for SBR header parser callback. */
+  cbUsac_t cbUsac;
+  void *cbUsacData;
+  cbUniDrc_t cbUniDrc; /*!< Function pointer for uniDrcConfig and
+                          loudnessInfoSet parser callback. */
+  void *cbUniDrcData;  /*!< User data pointer for uniDrcConfig and
+                          loudnessInfoSet parser callback. */
+} CSTpCallBacks;
+
+static const UINT SamplingRateTable[] = {
+    96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050, 16000, 12000, 11025,
+    8000,  7350,  0,     0,     57600, 51200, 40000, 38400, 34150, 28800, 25600,
+    20000, 19200, 17075, 14400, 12800, 9600,  0,     0,     0,     0};
+
+static inline int getSamplingRateIndex(UINT samplingRate, UINT nBits) {
+  UINT sf_index;
+  UINT tableSize = (1 << nBits) - 1;
+
+  for (sf_index = 0; sf_index < tableSize; sf_index++) {
+    if (SamplingRateTable[sf_index] == samplingRate) break;
+  }
+
+  if (sf_index > tableSize) {
+    return tableSize - 1;
+  }
+
+  return sf_index;
+}
+
+/*
+ * Get Channel count from channel configuration
+ */
+static inline int getNumberOfTotalChannels(int channelConfig) {
+  switch (channelConfig) {
+    case 1:
+    case 2:
+    case 3:
+    case 4:
+    case 5:
+    case 6:
+      return channelConfig;
+    case 7:
+    case 12:
+    case 14:
+      return 8;
+    case 11:
+      return 7;
+    case 13:
+      return 24;
+    default:
+      return 0;
+  }
+}
+
+static inline int getNumberOfEffectiveChannels(
+    const int
+        channelConfig) { /* index: 0,1,2,3,4,5,6,7,8,9,10,11,12,13,14,15 */
+  const int n[] = {0, 1, 2, 3, 4, 5, 5, 7, 0, 0, 0, 6, 7, 22, 7, 0};
+  return n[channelConfig];
+}
+
+#endif /* TP_DATA_H */
diff --git a/fdk-aac/libMpegTPEnc/include/tpenc_lib.h b/fdk-aac/libMpegTPEnc/include/tpenc_lib.h
new file mode 100644
index 0000000..4eb89a7
--- /dev/null
+++ b/fdk-aac/libMpegTPEnc/include/tpenc_lib.h
@@ -0,0 +1,339 @@
+/* -----------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2018 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 transport format encoder library *********************
+
+   Author(s):   Manuel Jander
+
+   Description: MPEG Transport encode
+
+*******************************************************************************/
+
+#ifndef TPENC_LIB_H
+#define TPENC_LIB_H
+
+#include "tp_data.h"
+#include "FDK_bitstream.h"
+
+#define TRANSPORTENC_INBUF_SIZE 8192
+
+typedef enum {
+  TRANSPORTENC_OK = 0,             /*!< All fine.             */
+  TRANSPORTENC_NO_MEM,             /*!< Out of memory.             */
+  TRANSPORTENC_UNKOWN_ERROR = 1,   /*!< Unknown error (embarrasing).   */
+  TRANSPORTENC_INVALID_PARAMETER,  /*!< An invalid parameter was passed to a
+                                      function .          */
+  TRANSPORTENC_UNSUPPORTED_FORMAT, /*!< Unsupported transport format. */
+  TRANSPORTENC_NOT_ENOUGH_BITS,    /*!< Out of bits. Provide more bits and try
+                                      again.            */
+
+  TRANSPORTENC_INVALID_CONFIG, /*!< Error in configuration.               */
+  TRANSPORTENC_LATM_INVALID_NR_OF_SUBFRAMES, /*!< LATM: number of subframes out
+                                                of range.                  */
+  TRANSPORTENC_LOAS_NOT_AVAILABLE, /*!< LOAS format not supported.           */
+  TRANSPORTENC_INVALID_LATM_ALIGNMENT, /*!< AudioMuxElement length not aligned
+                                          to 1 byte.            */
+
+  TRANSPORTENC_INVALID_TRANSMISSION_FRAME_LENGTH, /*!< Invalid transmission
+                                                     frame length (< 0). */
+  TRANSPORTENC_INVALID_CELP_FRAME_LENGTH, /*!< Invalid CELP frame length found
+                                             (>= 62).                 */
+  TRANSPORTENC_INVALID_FRAME_BITS, /*!< Frame bits is not 40 and not 80. */
+  TRANSPORTENC_INVALID_AOT,        /*!< Unknown AOT found.        */
+  TRANSPORTENC_INVALID_AU_LENGTH   /*!< Invalid Access Unit length (not
+                                      byte-aligned).           */
+
+} TRANSPORTENC_ERROR;
+
+typedef struct TRANSPORTENC *HANDLE_TRANSPORTENC;
+
+/**
+ * \brief             Determine a reasonable channel configuration on the basis
+ * of channel_mode.
+ * \param noChannels  Number of audio channels.
+ * \return            CHANNEL_MODE value that matches the given amount of audio
+ * channels.
+ */
+CHANNEL_MODE transportEnc_GetChannelMode(int noChannels);
+
+/**
+ * \brief                Register SBR heaqder writer callback.
+ * \param hTp            Handle of transport decoder.
+ * \param cbUpdateConfig Pointer to a callback function to handle SBR header
+ * writing.
+ * \param user_data      void pointer for user data passed to the callback as
+ * first parameter.
+ * \return               0 on success.
+ */
+int transportEnc_RegisterSbrCallback(HANDLE_TRANSPORTENC hTpEnc,
+                                     const cbSbr_t cbSbr, void *user_data);
+
+/**
+ * \brief                Register USAC SC writer callback.
+ * \param hTp            Handle of transport decoder.
+ * \param cbUpdateConfig Pointer to a callback function to handle USAC
+ * SCwriting.
+ * \param user_data      void pointer for user data passed to the callback as
+ * first parameter.
+ * \return               0 on success.
+ */
+int transportEnc_RegisterUsacCallback(HANDLE_TRANSPORTENC hTpEnc,
+                                      const cbUsac_t cbUsac, void *user_data);
+
+/**
+ * \brief                Register SSC writer callback.
+ * \param hTp            Handle of transport decoder.
+ * \param cbUpdateConfig Pointer to a callback function to handle SSC writing.
+ * \param user_data      void pointer for user data passed to the callback as
+ * first parameter.
+ * \return               0 on success.
+ */
+int transportEnc_RegisterSscCallback(HANDLE_TRANSPORTENC hTpEnc,
+                                     const cbSsc_t cbSsc, void *user_data);
+
+/**
+ * \brief         Write ASC from given parameters.
+ * \param asc     A HANDLE_FDK_BITSTREAM where the ASC is written to.
+ * \param config  Structure containing the codec configuration settings.
+ * \param cb callback information structure.
+ * \return        0 on success.
+ */
+int transportEnc_writeASC(HANDLE_FDK_BITSTREAM asc, CODER_CONFIG *config,
+                          CSTpCallBacks *cb);
+
+/* Defintion of flags that can be passed to transportEnc_Open() */
+#define TP_FLAG_MPEG4 1     /** MPEG4 (instead of MPEG2) */
+#define TP_FLAG_LATM_AMV 2  /** LATM AudioMuxVersion  */
+#define TP_FLAG_LATM_AMVA 4 /** LATM AudioMuxVersionA */
+
+/**
+ * \brief               Allocate transport encoder.
+ * \param phTpEnc       Pointer to transport encoder handle.
+ * \return              Error code.
+ */
+TRANSPORTENC_ERROR transportEnc_Open(HANDLE_TRANSPORTENC *phTpEnc);
+
+/**
+ * \brief               Init transport encoder.
+ * \param bsBuffer      Pointer to transport encoder.
+ * \param bsBuffer      Pointer to bitstream buffer.
+ * \param bsBufferSize  Size in bytes of bsBuffer.
+ * \param transportFmt  Format of the transport to be written.
+ * \param config        Pointer to a valid CODER_CONFIG struct.
+ * \param flags         Transport encoder flags.
+ * \return              Error code.
+ */
+TRANSPORTENC_ERROR transportEnc_Init(HANDLE_TRANSPORTENC hTpEnc,
+                                     UCHAR *bsBuffer, INT bsBufferSize,
+                                     TRANSPORT_TYPE transportFmt,
+                                     CODER_CONFIG *config, UINT flags);
+
+/**
+ * \brief               Write additional bits in transport encoder.
+ * \param config        Pointer to a valid CODER_CONFIG struct.
+ * \param nBits         Number of additional bits.
+ * \return              Error code.
+ */
+TRANSPORTENC_ERROR transportEnc_AddOtherDataBits(HANDLE_TRANSPORTENC hTpEnc,
+                                                 const int nBits);
+
+/**
+ * \brief      Get transport encoder bitstream.
+ * \param hTp  Pointer to a transport encoder handle.
+ * \return     The handle to the requested FDK bitstream.
+ */
+HANDLE_FDK_BITSTREAM transportEnc_GetBitstream(HANDLE_TRANSPORTENC hTp);
+
+/**
+ * \brief         Get amount of bits required by the transport headers.
+ * \param hTp     Handle of transport encoder.
+ * \param auBits  Amount of payload bits required for the current subframe.
+ * \return        Error code.
+ */
+INT transportEnc_GetStaticBits(HANDLE_TRANSPORTENC hTp, int auBits);
+
+/**
+ * \brief       Close transport encoder. This function assures that all
+ * allocated memory is freed.
+ * \param phTp  Pointer to a previously allocated transport encoder handle.
+ */
+void transportEnc_Close(HANDLE_TRANSPORTENC *phTp);
+
+/**
+ * \brief                       Write one access unit.
+ * \param hTp                   Handle of transport encoder.
+ * \param total_bits            Amount of total access unit bits.
+ * \param bufferFullness        Value of current buffer fullness in bits.
+ * \param noConsideredChannels  Number of bitrate wise considered channels (all
+ * minus LFE channels).
+ * \return                      Error code.
+ */
+TRANSPORTENC_ERROR transportEnc_WriteAccessUnit(HANDLE_TRANSPORTENC hTp,
+                                                INT total_bits,
+                                                int bufferFullness,
+                                                int noConsideredChannels);
+
+/**
+ * \brief        Inform the transportEnc layer that writing of access unit has
+ * finished. This function is required to be called when the encoder has
+ * finished writing one Access one Access Unit for bitstream
+ * housekeeping.
+ * \param hTp    Transport handle.
+ * \param pBits  Pointer to an int, where the current amount of frame bits is
+ * passed and where the current amount of subframe bits is returned.
+ *
+ * OR:  This integer is modified by the amount of extra bit alignment that may
+ * occurr.
+ *
+ * \return       Error code.
+ */
+TRANSPORTENC_ERROR transportEnc_EndAccessUnit(HANDLE_TRANSPORTENC hTp,
+                                              int *pBits);
+
+/*
+ * \brief         Get a payload frame.
+ * \param hTpEnc  Transport encoder handle.
+ * \param nBytes  Pointer to an int to hold the frame size in bytes. Returns
+ * zero if currently there is no complete frame for output (number of sub frames
+ * > 1).
+ * \return        Error code.
+ */
+TRANSPORTENC_ERROR transportEnc_GetFrame(HANDLE_TRANSPORTENC hTpEnc,
+                                         int *nbytes);
+
+/* ADTS CRC support */
+
+/**
+ * \brief         Set current bitstream position as start of a new data region.
+ * \param hTpEnc  Transport encoder handle.
+ * \param mBits   Size in bits of the data region. Set to 0 if it should not be
+ * of a fixed size.
+ * \return        Data region ID, which should be used when calling
+ * transportEnc_CrcEndReg().
+ */
+int transportEnc_CrcStartReg(HANDLE_TRANSPORTENC hTpEnc, int mBits);
+
+/**
+ * \brief         Set end of data region.
+ * \param hTpEnc  Transport encoder handle.
+ * \param reg     Data region ID, opbtained from transportEnc_CrcStartReg().
+ * \return        void
+ */
+void transportEnc_CrcEndReg(HANDLE_TRANSPORTENC hTpEnc, int reg);
+
+/**
+ * \brief             Get AudioSpecificConfig or StreamMuxConfig from transport
+ * encoder handle and write it to dataBuffer.
+ * \param hTpEnc      Transport encoder handle.
+ * \param cc          Pointer to the current and valid configuration contained
+ * in a CODER_CONFIG struct.
+ * \param dataBuffer  Bitbuffer holding binary configuration.
+ * \param confType    Pointer to an UINT where the configuration type is
+ * returned (0:ASC, 1:SMC).
+ * \return            Error code.
+ */
+TRANSPORTENC_ERROR transportEnc_GetConf(HANDLE_TRANSPORTENC hTpEnc,
+                                        CODER_CONFIG *cc,
+                                        FDK_BITSTREAM *dataBuffer,
+                                        UINT *confType);
+
+/**
+ * \brief       Get information (version among other things) of the transport
+ * encoder library.
+ * \param info  Pointer to an allocated LIB_INFO struct.
+ * \return      Error code.
+ */
+TRANSPORTENC_ERROR transportEnc_GetLibInfo(LIB_INFO *info);
+
+#endif /* #ifndef TPENC_LIB_H */
diff --git a/fdk-aac/libMpegTPEnc/src/tp_version.h b/fdk-aac/libMpegTPEnc/src/tp_version.h
new file mode 100644
index 0000000..9f1aa22
--- /dev/null
+++ b/fdk-aac/libMpegTPEnc/src/tp_version.h
@@ -0,0 +1,118 @@
+/* -----------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2018 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 transport format encoder library *********************
+
+   Author(s):
+
+   Description:
+
+*******************************************************************************/
+
+#if !defined(TP_VERSION_H)
+#define TP_VERSION_H
+
+/* library info */
+#define TP_LIB_VL0 3
+#define TP_LIB_VL1 0
+#define TP_LIB_VL2 0
+#define TP_LIB_TITLE "MPEG Transport"
+#ifdef __ANDROID__
+#define TP_LIB_BUILD_DATE ""
+#define TP_LIB_BUILD_TIME ""
+#else
+#define TP_LIB_BUILD_DATE __DATE__
+#define TP_LIB_BUILD_TIME __TIME__
+#endif
+#endif /* !defined(TP_VERSION_H) */
diff --git a/fdk-aac/libMpegTPEnc/src/tpenc_adif.cpp b/fdk-aac/libMpegTPEnc/src/tpenc_adif.cpp
new file mode 100644
index 0000000..b281eff
--- /dev/null
+++ b/fdk-aac/libMpegTPEnc/src/tpenc_adif.cpp
@@ -0,0 +1,186 @@
+/* -----------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2018 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 transport format encoder library *********************
+
+   Author(s):
+
+   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 err = 0;
+
+  int i;
+
+  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, adif->samplingRate, adif->instanceTag,
+                        adif->profile, adif->matrixMixdownA,
+                        (adif->pseudoSurroundEnable) ? 1 : 0, alignAnchor);
+
+  return err;
+}
+
+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, adif->matrixMixdownA, bits);
+
+  return bits;
+}
diff --git a/fdk-aac/libMpegTPEnc/src/tpenc_adif.h b/fdk-aac/libMpegTPEnc/src/tpenc_adif.h
new file mode 100644
index 0000000..e001afc
--- /dev/null
+++ b/fdk-aac/libMpegTPEnc/src/tpenc_adif.h
@@ -0,0 +1,146 @@
+/* -----------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2018 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 transport format encoder library *********************
+
+   Author(s):   Alex Goeschel
+
+   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;
+  int matrixMixdownA;
+  int pseudoSurroundEnable;
+
+} 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/fdk-aac/libMpegTPEnc/src/tpenc_adts.cpp b/fdk-aac/libMpegTPEnc/src/tpenc_adts.cpp
new file mode 100644
index 0000000..3f7e62c
--- /dev/null
+++ b/fdk-aac/libMpegTPEnc/src/tpenc_adts.cpp
@@ -0,0 +1,319 @@
+/* -----------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2018 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 transport format encoder library *********************
+
+   Author(s):   Alex Groeschel
+
+   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, 4);
+  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 */
+
+  hAdts->channel_config_zero = config->channelConfigZero;
+
+  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, hAdts->channel_config_zero), 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/fdk-aac/libMpegTPEnc/src/tpenc_adts.h b/fdk-aac/libMpegTPEnc/src/tpenc_adts.h
new file mode 100644
index 0000000..fe86306
--- /dev/null
+++ b/fdk-aac/libMpegTPEnc/src/tpenc_adts.h
@@ -0,0 +1,208 @@
+/* -----------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2018 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 transport format encoder library *********************
+
+   Author(s):   Alex Groeschel
+
+   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;
+  UCHAR channel_config_zero;
+  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/fdk-aac/libMpegTPEnc/src/tpenc_asc.cpp b/fdk-aac/libMpegTPEnc/src/tpenc_asc.cpp
new file mode 100644
index 0000000..0b484a0
--- /dev/null
+++ b/fdk-aac/libMpegTPEnc/src/tpenc_asc.cpp
@@ -0,0 +1,996 @@
+/* -----------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2018 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 transport format encoder library *********************
+
+   Author(s):
+
+   Description:
+
+*******************************************************************************/
+
+#include "tp_data.h"
+
+#include "tpenc_lib.h"
+#include "tpenc_asc.h"
+#include "FDK_bitstream.h"
+#include "genericStds.h"
+
+#include "FDK_crc.h"
+
+#define PCE_HEIGHT_EXT_SYNC (0xAC)
+#define HEIGHT_NORMAL 0
+#define HEIGHT_TOP 1
+#define HEIGHT_BOTTOM 2
+#define MAX_FRONT_ELEMENTS 8
+#define MAX_SIDE_ELEMENTS 3
+#define MAX_BACK_ELEMENTS 4
+
+/**
+ *  Describe additional PCE height information for front, side and back channel
+ * elements.
+ */
+typedef struct {
+  UCHAR
+  num_front_height_channel_elements[2];      /*!< Number of front channel
+                                                elements in top [0] and bottom
+                                                [1] plane. */
+  UCHAR num_side_height_channel_elements[2]; /*!< Number of side channel
+                                                elements in top [0] and bottom
+                                                [1] plane. */
+  UCHAR num_back_height_channel_elements[2]; /*!< Number of back channel
+                                                elements in top [0] and bottom
+                                                [1] plane. */
+} PCE_HEIGHT_NUM;
+
+/**
+ *  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. */
+  const MP4_ELEMENT_ID
+      *pEl_type; /*!< List contains sequence describing the elements
+                      in present channel mode. (MPEG order) */
+  const PCE_HEIGHT_NUM *pHeight_num;
+} 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;
+
+/**
+ * The following arrays provide the IDs of the consecutive elements for each
+ * mode.
+ */
+static const MP4_ELEMENT_ID elType_1[] = {ID_SCE};
+static const MP4_ELEMENT_ID elType_2[] = {ID_CPE};
+static const MP4_ELEMENT_ID elType_1_2[] = {ID_SCE, ID_CPE};
+static const MP4_ELEMENT_ID elType_1_2_1[] = {ID_SCE, ID_CPE, ID_SCE};
+static const MP4_ELEMENT_ID elType_1_2_2[] = {ID_SCE, ID_CPE, ID_CPE};
+static const MP4_ELEMENT_ID elType_1_2_2_1[] = {ID_SCE, ID_CPE, ID_CPE, ID_LFE};
+static const MP4_ELEMENT_ID elType_1_2_2_2_1[] = {ID_SCE, ID_CPE, ID_CPE,
+                                                  ID_CPE, ID_LFE};
+static const MP4_ELEMENT_ID elType_6_1[] = {ID_SCE, ID_CPE, ID_CPE, ID_SCE,
+                                            ID_LFE};
+static const MP4_ELEMENT_ID elType_7_1_back[] = {ID_SCE, ID_CPE, ID_CPE, ID_CPE,
+                                                 ID_LFE};
+static const MP4_ELEMENT_ID elType_7_1_top_front[] = {ID_SCE, ID_CPE, ID_CPE,
+                                                      ID_LFE, ID_CPE};
+static const MP4_ELEMENT_ID elType_7_1_rear_surround[] = {
+    ID_SCE, ID_CPE, ID_CPE, ID_CPE, ID_LFE};
+static const MP4_ELEMENT_ID elType_7_1_front_center[] = {ID_SCE, ID_CPE, ID_CPE,
+                                                         ID_CPE, ID_LFE};
+
+/**
+ * The following arrays provide information on how many front, side and back
+ * elements are assigned to the top or bottom plane for each mode that comprises
+ * height information.
+ */
+static const PCE_HEIGHT_NUM heightNum_7_1_top_front = {{1, 0}, {0, 0}, {0, 0}};
+
+/**
+ * \brief Table contains all supported channel modes and according PCE
+ configuration description.
+ *
+ * The mode identifier is followed by the number of front, side, back, and LFE
+ elements.
+ * These are followed by a pointer to the IDs of the consecutive elements
+ (ID_SCE, ID_CPE, ID_LFE).
+ *
+ * For some modes (MODE_7_1_TOP_FRONT and MODE_22_2) additional height
+ information is transmitted.
+ * In this case the additional pointer provides information on how many front,
+ side and back elements
+ * are assigned to the top or bottom plane.The elements are arranged in the
+ following order: normal height (front, side, back, LFE), top height (front,
+ side, back), bottom height (front, side, back).
+ *
+ *
+ * E.g. MODE_7_1_TOP_FRONT means:
+ *                          - 3 elements are front channel elements.
+ *                          - 0 elements are side channel elements.
+ *                          - 1 element is back channel element.
+ *                          - 1 element is an LFE channel element.
+ *                          - the element order is ID_SCE, ID_CPE, ID_CPE,
+ ID_LFE, ID_CPE.
+ *                          - 1 of the front elements is in the top plane.
+ *
+ * This leads to the following mapping for the cconsecutive elements in the
+ MODE_7_1_TOP_FRONT bitstream:
+ *                          - ID_SCE -> normal height front,
+                            - ID_CPE -> normal height front,
+                            - ID_CPE -> normal height back,
+                            - ID_LFE -> normal height LFE,
+                            - ID_CPE -> top height front.
+ */
+static const CHANNEL_CONFIGURATION pceConfigTab[] = {
+    {MODE_1,
+     {1, 0, 0, 0, elType_1,
+      NULL}}, /* don't transmit height information in this mode */
+    {MODE_2,
+     {1, 0, 0, 0, elType_2,
+      NULL}}, /* don't transmit height information in this mode */
+    {MODE_1_2,
+     {2, 0, 0, 0, elType_1_2,
+      NULL}}, /* don't transmit height information in this mode */
+    {MODE_1_2_1,
+     {2, 0, 1, 0, elType_1_2_1,
+      NULL}}, /* don't transmit height information in this mode */
+    {MODE_1_2_2,
+     {2, 0, 1, 0, elType_1_2_2,
+      NULL}}, /* don't transmit height information in this mode */
+    {MODE_1_2_2_1,
+     {2, 0, 1, 1, elType_1_2_2_1,
+      NULL}}, /* don't transmit height information in this mode */
+    {MODE_1_2_2_2_1,
+     {3, 0, 1, 1, elType_1_2_2_2_1,
+      NULL}}, /* don't transmit height information in this mode */
+
+    {MODE_6_1,
+     {2, 0, 2, 1, elType_6_1,
+      NULL}}, /* don't transmit height information in this mode */
+    {MODE_7_1_BACK,
+     {2, 0, 2, 1, elType_7_1_back,
+      NULL}}, /* don't transmit height information in this mode */
+    {MODE_7_1_TOP_FRONT,
+     {3, 0, 1, 1, elType_7_1_top_front, &heightNum_7_1_top_front}},
+
+    {MODE_7_1_REAR_SURROUND,
+     {2, 0, 2, 1, elType_7_1_rear_surround,
+      NULL}}, /* don't transmit height information in this mode */
+    {MODE_7_1_FRONT_CENTER,
+     {3, 0, 1, 1, elType_7_1_front_center,
+      NULL}} /* don't transmit height information in this mode */
+};
+
+/**
+ * \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;
+      break;
+    }
+  }
+
+  return pce_config;
+}
+
+int getChannelConfig(const CHANNEL_MODE channel_mode,
+                     const UCHAR channel_config_zero) {
+  INT chan_config = 0;
+
+  if (channel_config_zero != 0) {
+    chan_config = 0;
+  } else {
+    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;
+      case MODE_6_1:
+        chan_config = 11;
+        break;
+      case MODE_7_1_BACK:
+        chan_config = 12;
+        break;
+      case MODE_7_1_TOP_FRONT:
+        chan_config = 14;
+        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;
+}
+
+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, frntCnt = 0, sdCnt = 0, bckCnt = 0,
+        isCpe = 0, tag = 0, normalFrontEnd = 0, normalSideEnd = 0,
+        normalBackEnd = 0, topFrontEnd = 0, topSideEnd = 0, topBackEnd = 0,
+        bottomFrontEnd = 0, bottomSideEnd = 0;
+#ifdef FDK_ASSERT_ENABLE
+  UCHAR bottomBackEnd = 0;
+#endif
+  enum elementDepth { FRONT, SIDE, BACK } elDepth;
+
+  sampleRateIndex = getSamplingRateIndex(sampleRate, 4);
+  if (sampleRateIndex == 15) {
+    return -1;
+  }
+
+  if ((config = getPceEntry(channelMode)) == NULL) {
+    return -1;
+  }
+
+  FDK_ASSERT(config->num_front_channel_elements <= MAX_FRONT_ELEMENTS);
+  FDK_ASSERT(config->num_side_channel_elements <= MAX_SIDE_ELEMENTS);
+  FDK_ASSERT(config->num_back_channel_elements <= MAX_BACK_ELEMENTS);
+
+  UCHAR frontIsCpe[MAX_FRONT_ELEMENTS] = {0},
+        frontTag[MAX_FRONT_ELEMENTS] = {0}, sideIsCpe[MAX_SIDE_ELEMENTS] = {0},
+        sideTag[MAX_SIDE_ELEMENTS] = {0}, backIsCpe[MAX_BACK_ELEMENTS] = {0},
+        backTag[MAX_BACK_ELEMENTS] = {0};
+
+  /* Write general information */
+
+  FDKwriteBits(hBs, instanceTagPCE, 4);  /* Element instance tag */
+  FDKwriteBits(hBs, profile, 2);         /* Object type */
+  FDKwriteBits(hBs, sampleRateIndex, 4); /* Sample rate index*/
+
+  FDKwriteBits(hBs, config->num_front_channel_elements,
+               4); /* Front channel Elements */
+  FDKwriteBits(hBs, config->num_side_channel_elements,
+               4); /* No Side Channel Elements */
+  FDKwriteBits(hBs, config->num_back_channel_elements,
+               4); /* No Back channel Elements */
+  FDKwriteBits(hBs, config->num_lfe_channel_elements,
+               2); /* No Lfe channel elements */
+
+  FDKwriteBits(hBs, 0, 3); /* No assoc data elements */
+  FDKwriteBits(hBs, 0, 4); /* No valid cc elements */
+  FDKwriteBits(hBs, 0, 1); /* Mono mixdown present */
+  FDKwriteBits(hBs, 0, 1); /* Stereo mixdown present */
+
+  if (matrixMixdownA != 0 &&
+      ((channelMode == MODE_1_2_2) || (channelMode == MODE_1_2_2_1))) {
+    FDKwriteBits(hBs, 1, 1); /* Matrix mixdown present */
+    FDKwriteBits(hBs, (matrixMixdownA - 1) & 0x3, 2); /* matrix_mixdown_idx */
+    FDKwriteBits(hBs, (pseudoSurroundEnable) ? 1 : 0,
+                 1); /* pseudo_surround_enable */
+  } else {
+    FDKwriteBits(hBs, 0, 1); /* Matrix mixdown not present */
+  }
+
+  if (config->pHeight_num != NULL) {
+    /* we have up to three different height levels, and in each height level we
+     * may have front, side and back channels. We need to know where each
+     * section ends to correctly count the tags */
+    normalFrontEnd = config->num_front_channel_elements -
+                     config->pHeight_num->num_front_height_channel_elements[0] -
+                     config->pHeight_num->num_front_height_channel_elements[1];
+    normalSideEnd = normalFrontEnd + config->num_side_channel_elements -
+                    config->pHeight_num->num_side_height_channel_elements[0] -
+                    config->pHeight_num->num_side_height_channel_elements[1];
+    normalBackEnd = normalSideEnd + config->num_back_channel_elements -
+                    config->pHeight_num->num_back_height_channel_elements[0] -
+                    config->pHeight_num->num_back_height_channel_elements[1];
+
+    topFrontEnd =
+        normalBackEnd + config->num_lfe_channel_elements +
+        config->pHeight_num->num_front_height_channel_elements[0]; /* only
+                                                                      normal
+                                                                      height
+                                                                      LFEs
+                                                                      assumed */
+    topSideEnd =
+        topFrontEnd + config->pHeight_num->num_side_height_channel_elements[0];
+    topBackEnd =
+        topSideEnd + config->pHeight_num->num_back_height_channel_elements[0];
+
+    bottomFrontEnd =
+        topBackEnd + config->pHeight_num->num_front_height_channel_elements[1];
+    bottomSideEnd = bottomFrontEnd +
+                    config->pHeight_num->num_side_height_channel_elements[1];
+#ifdef FDK_ASSERT_ENABLE
+    bottomBackEnd = bottomSideEnd +
+                    config->pHeight_num->num_back_height_channel_elements[1];
+#endif
+
+  } else {
+    /* we have only one height level, so we don't care about top or bottom */
+    normalFrontEnd = config->num_front_channel_elements;
+    normalSideEnd = normalFrontEnd + config->num_side_channel_elements;
+    normalBackEnd = normalSideEnd + config->num_back_channel_elements;
+  }
+
+  /* assign cpe and tag information to either front, side or back channels */
+
+  pEl_list = config->pEl_type;
+
+  for (i = 0; i < config->num_front_channel_elements +
+                      config->num_side_channel_elements +
+                      config->num_back_channel_elements +
+                      config->num_lfe_channel_elements;
+       i++) {
+    if (*pEl_list == ID_LFE) {
+      pEl_list++;
+      continue;
+    }
+    isCpe = (*pEl_list++ == ID_CPE) ? 1 : 0;
+    tag = (isCpe) ? cpeCnt++ : sceCnt++;
+
+    if (i < normalFrontEnd)
+      elDepth = FRONT;
+    else if (i < normalSideEnd)
+      elDepth = SIDE;
+    else if (i < normalBackEnd)
+      elDepth = BACK;
+    else if (i < topFrontEnd)
+      elDepth = FRONT;
+    else if (i < topSideEnd)
+      elDepth = SIDE;
+    else if (i < topBackEnd)
+      elDepth = BACK;
+    else if (i < bottomFrontEnd)
+      elDepth = FRONT;
+    else if (i < bottomSideEnd)
+      elDepth = SIDE;
+    else {
+      elDepth = BACK;
+      FDK_ASSERT(i < bottomBackEnd); /* won't fail if implementation of pce
+                                        configuration table is correct */
+    }
+
+    switch (elDepth) {
+      case FRONT:
+        FDK_ASSERT(frntCnt < config->num_front_channel_elements);
+        frontIsCpe[frntCnt] = isCpe;
+        frontTag[frntCnt++] = tag;
+        break;
+      case SIDE:
+        FDK_ASSERT(sdCnt < config->num_side_channel_elements);
+        sideIsCpe[sdCnt] = isCpe;
+        sideTag[sdCnt++] = tag;
+        break;
+      case BACK:
+        FDK_ASSERT(bckCnt < config->num_back_channel_elements);
+        backIsCpe[bckCnt] = isCpe;
+        backTag[bckCnt++] = tag;
+        break;
+    }
+  }
+
+  /* Write front channel isCpe and tags */
+  for (i = 0; i < config->num_front_channel_elements; i++) {
+    FDKwriteBits(hBs, frontIsCpe[i], 1);
+    FDKwriteBits(hBs, frontTag[i], 4);
+  }
+  /* Write side channel isCpe and tags */
+  for (i = 0; i < config->num_side_channel_elements; i++) {
+    FDKwriteBits(hBs, sideIsCpe[i], 1);
+    FDKwriteBits(hBs, sideTag[i], 4);
+  }
+  /* Write back channel isCpe and tags */
+  for (i = 0; i < config->num_back_channel_elements; i++) {
+    FDKwriteBits(hBs, backIsCpe[i], 1);
+    FDKwriteBits(hBs, backTag[i], 4);
+  }
+  /* Write LFE information */
+  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 */
+
+  /* Write comment information */
+
+  if (config->pHeight_num != NULL) {
+    /* embed height information in comment field */
+
+    INT commentBytes =
+        1 /* PCE_HEIGHT_EXT_SYNC */
+        + ((((config->num_front_channel_elements +
+              config->num_side_channel_elements +
+              config->num_back_channel_elements)
+             << 1) +
+            7) >>
+           3) /* 2 bit height info per element, round up to full bytes */
+        + 1;  /* CRC */
+
+    FDKwriteBits(hBs, commentBytes, 8); /* comment size. */
+
+    FDK_CRCINFO crcInfo; /* CRC state info */
+    INT crcReg;
+
+    FDKcrcInit(&crcInfo, 0x07, 0xFF, 8);
+    crcReg = FDKcrcStartReg(&crcInfo, hBs, 0);
+
+    FDKwriteBits(hBs, PCE_HEIGHT_EXT_SYNC, 8); /* indicate height extension */
+
+    /* front channel height information */
+    for (i = 0;
+         i < config->num_front_channel_elements -
+                 config->pHeight_num->num_front_height_channel_elements[0] -
+                 config->pHeight_num->num_front_height_channel_elements[1];
+         i++)
+      FDKwriteBits(hBs, HEIGHT_NORMAL, 2);
+    for (i = 0; i < config->pHeight_num->num_front_height_channel_elements[0];
+         i++)
+      FDKwriteBits(hBs, HEIGHT_TOP, 2);
+    for (i = 0; i < config->pHeight_num->num_front_height_channel_elements[1];
+         i++)
+      FDKwriteBits(hBs, HEIGHT_BOTTOM, 2);
+
+    /* side channel height information */
+    for (i = 0;
+         i < config->num_side_channel_elements -
+                 config->pHeight_num->num_side_height_channel_elements[0] -
+                 config->pHeight_num->num_side_height_channel_elements[1];
+         i++)
+      FDKwriteBits(hBs, HEIGHT_NORMAL, 2);
+    for (i = 0; i < config->pHeight_num->num_side_height_channel_elements[0];
+         i++)
+      FDKwriteBits(hBs, HEIGHT_TOP, 2);
+    for (i = 0; i < config->pHeight_num->num_side_height_channel_elements[1];
+         i++)
+      FDKwriteBits(hBs, HEIGHT_BOTTOM, 2);
+
+    /* back channel height information */
+    for (i = 0;
+         i < config->num_back_channel_elements -
+                 config->pHeight_num->num_back_height_channel_elements[0] -
+                 config->pHeight_num->num_back_height_channel_elements[1];
+         i++)
+      FDKwriteBits(hBs, HEIGHT_NORMAL, 2);
+    for (i = 0; i < config->pHeight_num->num_back_height_channel_elements[0];
+         i++)
+      FDKwriteBits(hBs, HEIGHT_TOP, 2);
+    for (i = 0; i < config->pHeight_num->num_back_height_channel_elements[1];
+         i++)
+      FDKwriteBits(hBs, HEIGHT_BOTTOM, 2);
+
+    FDKbyteAlign(hBs, alignAnchor); /* Alignment */
+
+    FDKcrcEndReg(&crcInfo, hBs, crcReg);
+    FDKwriteBits(hBs, FDKcrcGetCRC(&crcInfo), 8);
+
+  } else {
+    FDKwriteBits(hBs, 0,
+                 8); /* Do no write any comment or height information. */
+  }
+
+  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 */
+
+  if (config->pHeight_num != NULL) {
+    /* Comment field (height extension) */
+
+    bits +=
+        8 /* PCE_HEIGHT_EXT_SYNC */
+        +
+        ((config->num_front_channel_elements +
+          config->num_side_channel_elements + config->num_back_channel_elements)
+         << 1) /* 2 bit height info per element */
+        + 8;   /* CRC */
+
+    if ((bits % 8) != 0) {
+      bits += (8 - (bits % 8)); /* Alignment */
+    }
+  }
+
+  return bits;
+}
+
+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 hBs, int sampleRate,
+                            int nBits) {
+  int srIdx = getSamplingRateIndex(sampleRate, nBits);
+
+  FDKwriteBits(hBs, srIdx, nBits);
+  if (srIdx == (1 << nBits) - 1) {
+    FDKwriteBits(hBs, sampleRate, 24);
+  }
+}
+
+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, config->channelConfigZero) == 0) {
+    transportEnc_writePCE(asc, config->channelMode, config->samplingRate, 0, 1,
+                          config->matrixMixdownA,
+                          (config->flags & CC_PSEUDO_SURROUND) ? 1 : 0,
+                          alignAnchor);
+  }
+  if ((aot == AOT_AAC_SCAL) || (aot == AOT_ER_AAC_SCAL)) {
+    FDKwriteBits(asc, 0, 3); /* layerNr */
+  }
+  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;
+}
+
+static int transportEnc_writeELDSpecificConfig(HANDLE_FDK_BITSTREAM hBs,
+                                               CODER_CONFIG *config,
+                                               int epConfig,
+                                               CSTpCallBacks *cb) {
+  UINT frameLengthFlag = 0;
+  switch (config->samplesPerFrame) {
+    case 512:
+    case 256:
+    case 128:
+    case 64:
+      frameLengthFlag = 0;
+      break;
+    case 480:
+    case 240:
+    case 160:
+    case 120:
+    case 60:
+      frameLengthFlag = 1;
+      break;
+  }
+
+  FDKwriteBits(hBs, frameLengthFlag, 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, sbrElementIndex = 0;
+
+      pPce = getPceEntry(config->channelMode);
+
+      for (e = 0; e < pPce->num_front_channel_elements +
+                          pPce->num_side_channel_elements +
+                          pPce->num_back_channel_elements +
+                          pPce->num_lfe_channel_elements;
+           e++) {
+        if ((pPce->pEl_type[e] == ID_SCE) || (pPce->pEl_type[e] == ID_CPE)) {
+          cb->cbSbr(cb->cbSbrData, hBs, 0, 0, 0, config->aot, pPce->pEl_type[e],
+                    sbrElementIndex, 0, 0, 0, NULL, 1);
+          sbrElementIndex++;
+        }
+      }
+    }
+  }
+
+  if ((config->flags & CC_SAC) && (cb->cbSsc != NULL)) {
+    FDKwriteBits(hBs, ELDEXT_LDSAC, 4);
+
+    const INT eldExtLen =
+        (cb->cbSsc(cb->cbSscData, NULL, config->aot, config->extSamplingRate, 0,
+                   0, 0, 0, 0, NULL) +
+         7) >>
+        3;
+    INT cnt = eldExtLen;
+
+    if (cnt < 0xF) {
+      FDKwriteBits(hBs, cnt, 4);
+    } else {
+      FDKwriteBits(hBs, 0xF, 4);
+      cnt -= 0xF;
+
+      if (cnt < 0xFF) {
+        FDKwriteBits(hBs, cnt, 8);
+      } else {
+        FDKwriteBits(hBs, 0xFF, 8);
+        cnt -= 0xFF;
+
+        FDK_ASSERT(cnt <= 0xFFFF);
+        FDKwriteBits(hBs, cnt, 16);
+      }
+    }
+
+    cb->cbSsc(cb->cbSscData, hBs, config->aot, config->extSamplingRate, 0, 0, 0,
+              0, 0, NULL);
+  }
+
+  if (config->downscaleSamplingRate != 0 &&
+      config->downscaleSamplingRate != config->extSamplingRate) {
+    /* downscale active */
+
+    /* eldExtLenDsc: Number of bytes for the ELD downscale extension (srIdx
+       needs 1 byte
+       + downscaleSamplingRate needs additional 3 bytes) */
+    int eldExtLenDsc = 1;
+    int downscaleSamplingRate = config->downscaleSamplingRate;
+    FDKwriteBits(hBs, ELDEXT_DOWNSCALEINFO, 4); /* ELDEXT_DOWNSCALEINFO */
+
+    if ((downscaleSamplingRate != 96000) && (downscaleSamplingRate != 88200) &&
+        (downscaleSamplingRate != 64000) && (downscaleSamplingRate != 48000) &&
+        (downscaleSamplingRate != 44100) && (downscaleSamplingRate != 32000) &&
+        (downscaleSamplingRate != 24000) && (downscaleSamplingRate != 22050) &&
+        (downscaleSamplingRate != 16000) && (downscaleSamplingRate != 12000) &&
+        (downscaleSamplingRate != 11025) && (downscaleSamplingRate != 8000) &&
+        (downscaleSamplingRate != 7350)) {
+      eldExtLenDsc = 4; /* length extends to 4 if downscaleSamplingRate's value
+                           is not one of the listed values */
+    }
+
+    FDKwriteBits(hBs, eldExtLenDsc, 4);
+    writeSampleRate(hBs, downscaleSamplingRate, 4);
+    FDKwriteBits(hBs, 0x0, 4); /* fill_nibble */
+  }
+
+  FDKwriteBits(hBs, ELDEXT_TERM, 4); /* ELDEXT_TERM */
+
+  return 0;
+}
+
+static int transportEnc_writeUsacSpecificConfig(HANDLE_FDK_BITSTREAM hBs,
+                                                int extFlag, CODER_CONFIG *cc,
+                                                CSTpCallBacks *cb) {
+  FDK_BITSTREAM usacConf;
+  int usacConfigBits = cc->rawConfigBits;
+
+  if ((usacConfigBits <= 0) ||
+      ((usacConfigBits + 7) / 8 > (int)sizeof(cc->rawConfig))) {
+    return TRANSPORTENC_UNSUPPORTED_FORMAT;
+  }
+  FDKinitBitStream(&usacConf, cc->rawConfig, BUFSIZE_DUMMY_VALUE,
+                   usacConfigBits, BS_READER);
+
+  for (; usacConfigBits > 0; usacConfigBits--) {
+    UINT tmp = FDKreadBit(&usacConf);
+    FDKwriteBits(hBs, tmp, 1);
+  }
+  FDKsyncCache(hBs);
+
+  return TRANSPORTENC_OK;
+}
+
+int transportEnc_writeASC(HANDLE_FDK_BITSTREAM asc, CODER_CONFIG *config,
+                          CSTpCallBacks *cb) {
+  UINT extFlag = 0;
+  int err;
+  int epConfig = 0;
+
+  /* Required for the PCE. */
+  UINT alignAnchor = FDKgetValidBits(asc);
+
+  /* Extension Flag: Shall be 1 for aot = 17,19,20,21,22,23,39 */
+  switch (config->aot) {
+    case AOT_ER_AAC_LC:
+    case AOT_ER_AAC_LTP:
+    case AOT_ER_AAC_SCAL:
+    case AOT_ER_TWIN_VQ:
+    case AOT_ER_BSAC:
+    case AOT_ER_AAC_LD:
+    case AOT_ER_AAC_ELD:
+    case AOT_USAC:
+      extFlag = 1;
+      break;
+    default:
+      break;
+  }
+
+  if (config->sbrSignaling == SIG_EXPLICIT_HIERARCHICAL && config->sbrPresent)
+    writeAot(asc, config->extAOT);
+  else
+    writeAot(asc, config->aot);
+
+  /* In case of USAC it is the output not the core sampling rate */
+  writeSampleRate(asc, config->samplingRate, 4);
+
+  /* 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, config->channelConfigZero), 4);
+
+  if (config->sbrSignaling == SIG_EXPLICIT_HIERARCHICAL && config->sbrPresent) {
+    writeSampleRate(asc, config->extSamplingRate, 4);
+    writeAot(asc, config->aot);
+  }
+
+  switch (config->aot) {
+    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;
+    case AOT_ER_AAC_ELD:
+      err = transportEnc_writeELDSpecificConfig(asc, config, epConfig, cb);
+      if (err) return err;
+      break;
+    case AOT_USAC:
+      err = transportEnc_writeUsacSpecificConfig(asc, extFlag, config, cb);
+      if (err) {
+        return err;
+      }
+      break;
+    default:
+      return -1;
+  }
+
+  switch (config->aot) {
+    case AOT_ER_AAC_LC:
+    case AOT_ER_AAC_LTP:
+    case AOT_ER_AAC_SCAL:
+    case AOT_ER_TWIN_VQ:
+    case AOT_ER_BSAC:
+    case AOT_ER_AAC_LD:
+    case AOT_ER_CELP:
+    case AOT_ER_HVXC:
+    case AOT_ER_HILN:
+    case AOT_ER_PARA:
+    case AOT_ER_AAC_ELD:
+      FDKwriteBits(asc, 0, 2); /* epconfig 0 */
+      break;
+    default:
+      break;
+  }
+
+  /* backward compatible explicit signaling of extension AOT */
+  if (config->sbrSignaling == SIG_EXPLICIT_BW_COMPATIBLE) {
+    TP_ASC_EXTENSION_ID ascExtId = ASCEXT_UNKOWN;
+
+    if (config->sbrPresent) {
+      ascExtId = ASCEXT_SBR;
+      FDKwriteBits(asc, ascExtId, 11);
+      writeAot(asc, config->extAOT);
+      FDKwriteBits(asc, 1, 1); /* sbrPresentFlag=1 */
+      writeSampleRate(asc, config->extSamplingRate, 4);
+      if (config->psPresent) {
+        ascExtId = ASCEXT_PS;
+        FDKwriteBits(asc, ascExtId, 11);
+        FDKwriteBits(asc, 1, 1); /* psPresentFlag=1 */
+      }
+    }
+  }
+
+  /* Make sure all bits are sync'ed */
+  FDKsyncCache(asc);
+
+  return 0;
+}
diff --git a/fdk-aac/libMpegTPEnc/src/tpenc_asc.h b/fdk-aac/libMpegTPEnc/src/tpenc_asc.h
new file mode 100644
index 0000000..5f5621e
--- /dev/null
+++ b/fdk-aac/libMpegTPEnc/src/tpenc_asc.h
@@ -0,0 +1,147 @@
+/* -----------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2018 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 transport format encoder library *********************
+
+   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
+ * \param channel_config_zero no standard channel configuration
+ *
+ * \return chanel config
+ */
+int getChannelConfig(const CHANNEL_MODE channel_mode,
+                     const UCHAR channel_config_zero);
+
+/**
+ * \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/fdk-aac/libMpegTPEnc/src/tpenc_dab.cpp b/fdk-aac/libMpegTPEnc/src/tpenc_dab.cpp
new file mode 100644
index 0000000..202fecf
--- /dev/null
+++ b/fdk-aac/libMpegTPEnc/src/tpenc_dab.cpp
@@ -0,0 +1,467 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+� Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur F�rderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/******************************** MPEG Audio Encoder **************************
+
+   Initial author:       serge
+   contents/description: DAB Transport Headers support
+
+******************************************************************************/
+#include <stdio.h>
+#include "FDK_audio.h"
+#include "tpenc_dab.h"
+
+
+#include "tpenc_lib.h"
+#include "tpenc_asc.h"
+
+#include "common_fix.h"
+
+int dabWrite_CrcStartReg(
+                                 HANDLE_DAB pDab,          /*!< pointer to dab stucture */
+                                 HANDLE_FDK_BITSTREAM hBs,   /*!< handle to current bit buffer structure */
+                                 int mBits                   /*!< number of bits in crc region */
+                                )
+{
+  //fprintf(stderr, "dabWrite_CrcStartReg(%p): bits in crc region=%d\n", hBs, mBits);
+  return ( FDKcrcStartReg(&pDab->crcInfo2, hBs, mBits) );
+}
+
+void dabWrite_CrcEndReg(
+                                HANDLE_DAB pDab, /*!< pointer to dab crc info stucture */
+                                HANDLE_FDK_BITSTREAM hBs,   /*!< handle to current bit buffer structure */
+                                int reg                    /*!< crc region */
+                               )
+{
+    //fprintf(stderr, "dabWrite_CrcEndReg(%p): crc region=%d\n", hBs, reg);
+    FDKcrcEndReg(&pDab->crcInfo2, hBs, reg);
+}
+
+int dabWrite_GetHeaderBits( HANDLE_DAB hDab )
+{
+  int bits = 0;
+
+  if (hDab->currentBlock == 0) {
+    /* Static and variable header bits */
+    bits += 16; //header_firecode 16
+    bits +=  8; //rfa=1, dac_rate=1, sbr_flag=1, aac_channel_mode=1, ps_flag=1, mpeg_surround_config=3
+    bits += 12 * hDab->num_raw_blocks; //au_start[1...num_aus] 12 bit AU start position markers
+
+    //4 byte alignment
+    if (hDab->dac_rate == 0 || hDab->sbr_flag == 0)
+    	bits+=4;
+    //16sbr => 16 + 5 + 3 + 12*(2-1) => 36 => 40 bits 5
+    //24sbr => 16 + 5 + 3 + 12*(3-1) => 48 ok         6
+    //32sbr => 16 + 5 + 3 + 12*(4-1) => 60 => 64 bits 8
+    //48sbr => 16 + 5 + 3 + 12*(6-1) => 84 => 88 bits 11
+  }
+
+  /* Add raw data block CRC bits. Not really part of the header, put they cause bit overhead to be accounted. */
+  bits += 16;
+
+
+  return bits;
+}
+
+
+int dabWrite_CountTotalBitDemandHeader( HANDLE_DAB hDab, unsigned int streamDataLength )
+{
+	//fprintf(stderr, "streamDataLength=%d (%d bytes)\n", streamDataLength, streamDataLength >> 3);
+	return dabWrite_GetHeaderBits(hDab);
+}
+
+
+INT dabWrite_Init(HANDLE_DAB hDab, CODER_CONFIG *config)
+{
+  /* Sanity checks */
+	if((int)config->aot > 4
+    || (int)config->aot < 1 ) {
+    return -1;
+  }
+
+  /* Sanity checks DAB-specific */
+  if ( !(config->nSubFrames == 2 && config->samplingRate == 16000 && (config->flags & CC_SBR)) &&
+	   !(config->nSubFrames == 3 && config->samplingRate == 24000 && (config->flags & CC_SBR)) &&
+	   !(config->nSubFrames == 4 && config->samplingRate == 32000) &&
+	   !(config->nSubFrames == 6 && config->samplingRate == 48000)) {
+	  return -1;
+  }
+
+  hDab->dac_rate = 0;
+  hDab->aac_channel_mode=0;
+  hDab->sbr_flag = 0;
+  hDab->ps_flag = 0;
+  hDab->mpeg_surround_config=0;
+  hDab->subchannels_num=config->bitRate/8000;
+
+
+  if(config->samplingRate == 24000 || config->samplingRate == 48000)
+	  hDab->dac_rate = 1;
+
+  if (config->extAOT==AOT_SBR || config->extAOT == AOT_PS)
+	  hDab->sbr_flag = 1;
+
+  if(config->extAOT == AOT_PS)
+	  hDab->ps_flag = 1;
+
+
+  if(config->channelMode == MODE_2)
+	  hDab->aac_channel_mode =  1;
+
+  //fprintf(stderr, "hDab->dac_rate=%d\n", hDab->dac_rate);
+  //fprintf(stderr, "hDab->sbr_flag=%d\n", hDab->sbr_flag);
+  //fprintf(stderr, "hDab->ps_flag=%d\n", hDab->ps_flag);
+  //fprintf(stderr, "hDab->aac_channel_mode=%d\n", hDab->aac_channel_mode);
+  //fprintf(stderr, "hDab->subchannels_num=%d\n", hDab->subchannels_num);
+  //fprintf(stderr, "cc->nSubFrames=%d\n", config->nSubFrames);
+
+  hDab->num_raw_blocks=config->nSubFrames-1; /* 0 means 1 raw data block */
+
+  FDKcrcInit(&hDab->crcInfo, 0x1021, 0xFFFF, 16);
+  FDKcrcInit(&hDab->crcFire, 0x782d, 0, 16);
+  FDKcrcInit(&hDab->crcInfo2, 0x8005, 0xFFFF, 16);
+
+  hDab->currentBlock = 0;
+  hDab->headerBits = dabWrite_GetHeaderBits(hDab);
+
+  return 0;
+}
+
+int dabWrite_EncodeHeader(HANDLE_DAB hDab,
+                                 HANDLE_FDK_BITSTREAM hBitStream,
+                                 int buffer_fullness,
+                                 int frame_length)
+{
+  INT crcIndex = 0;
+
+
+  FDK_ASSERT(((frame_length+hDab->headerBits)/8)<0x2000);      /*13 bit*/
+  FDK_ASSERT(buffer_fullness<0x800);    /* 11 bit   */
+
+  FDKcrcReset(&hDab->crcInfo);
+
+
+//  fprintf(stderr, "dabWrite_EncodeHeader() hDab->currentBlock=%d, frame_length=%d, buffer_fullness=%d\n",
+//		  hDab->currentBlock, frame_length, buffer_fullness);
+
+//  if (hDab->currentBlock == 0) {
+//	//hDab->subFrameStartPrev=dabWrite_GetHeaderBits(hDab);
+//	fprintf(stderr, "header bits[%d] [%d]\n", hDab->subFrameStartPrev, hDab->subFrameStartPrev >> 3);
+//    FDKresetBitbuffer(hBitStream, BS_WRITER);
+//  }
+
+  //hDab->subFrameStartBit = FDKgetValidBits(hBitStream);
+//  fprintf(stderr, "dabWrite_EncodeHeader() hDab->subFrameStartBit=%d [%d]\n", hDab->subFrameStartBit, hDab->subFrameStartBit >> 3);
+
+  //hDab->subFrameStartBit = FDKgetValidBits(hBitStream);
+  /* Skip new header if this is raw data block 1..n */
+  if (hDab->currentBlock == 0)
+  {
+    FDKresetBitbuffer(hBitStream, BS_WRITER);
+//    fprintf(stderr, "dabWrite_EncodeHeader() after FDKresetBitbuffer=%d [%d]\n", FDKgetValidBits(hBitStream), FDKgetValidBits(hBitStream) >> 3);
+
+    /* fixed header */
+    FDKwriteBits(hBitStream, 0, 16); //header_firecode
+    FDKwriteBits(hBitStream, 0, 1); //rfa
+    FDKwriteBits(hBitStream, hDab->dac_rate, 1);
+    FDKwriteBits(hBitStream, hDab->sbr_flag, 1);
+    FDKwriteBits(hBitStream, hDab->aac_channel_mode, 1);
+    FDKwriteBits(hBitStream, hDab->ps_flag, 1);
+    FDKwriteBits(hBitStream, hDab->mpeg_surround_config, 3);
+    /* variable header */
+    int i;
+    for(i=0; i<hDab->num_raw_blocks; i++)
+    	FDKwriteBits(hBitStream, 0, 12);
+    /* padding */
+    if (hDab->dac_rate == 0 || hDab->sbr_flag == 0) {
+    	FDKwriteBits(hBitStream, 0, 4);
+    }
+  } /* End of DAB header */
+
+  hDab->subFrameStartBit = FDKgetValidBits(hBitStream);
+  FDK_ASSERT(FDKgetValidBits(hBitStream) % 8 == 0); //only aligned header
+
+//  fprintf(stderr, "dabWrite_EncodeHeader() FDKgetValidBits(hBitStream)=%d [%d]\n", FDKgetValidBits(hBitStream), FDKgetValidBits(hBitStream) >> 3);
+  return 0;
+}
+
+int dabWrite_writeExtensionFillPayload(HANDLE_FDK_BITSTREAM  hBitStream, int extPayloadBits)
+{
+#define EXT_TYPE_BITS         ( 4 )
+#define DATA_EL_VERSION_BITS  ( 4 )
+#define FILL_NIBBLE_BITS      ( 4 )
+
+#define EXT_TYPE_BITS         ( 4 )
+#define DATA_EL_VERSION_BITS  ( 4 )
+#define FILL_NIBBLE_BITS      ( 4 )
+
+	INT  extBitsUsed = 0;
+	INT extPayloadType = EXT_FIL;
+	//fprintf(stderr, "FDKaacEnc_writeExtensionPayload() extPayloadType=%d\n", extPayloadType);
+	if (extPayloadBits >= EXT_TYPE_BITS)
+	{
+	  UCHAR  fillByte = 0x00;  /* for EXT_FIL and EXT_FILL_DATA */
+
+	  if (hBitStream != NULL) {
+		FDKwriteBits(hBitStream, extPayloadType, EXT_TYPE_BITS);
+	  }
+	  extBitsUsed += EXT_TYPE_BITS;
+
+	  switch (extPayloadType) {
+		case EXT_FILL_DATA:
+		  fillByte = 0xA5;
+		case EXT_FIL:
+		default:
+		  if (hBitStream != NULL) {
+			int writeBits = extPayloadBits;
+			FDKwriteBits(hBitStream, 0x00, FILL_NIBBLE_BITS);
+			writeBits -= 8;  /* acount for the extension type and the fill nibble */
+			while (writeBits >= 8) {
+			  FDKwriteBits(hBitStream, fillByte, 8);
+			  writeBits -= 8;
+			}
+		  }
+		  extBitsUsed += FILL_NIBBLE_BITS + (extPayloadBits & ~0x7) - 8;
+		  break;
+	  }
+	}
+
+	return (extBitsUsed);
+}
+
+void dabWrite_FillRawDataBlock(HANDLE_FDK_BITSTREAM  hBitStream, int payloadBits)
+{
+	INT extBitsUsed = 0;
+#define EL_ID_BITS              ( 3 )
+#define FILL_EL_COUNT_BITS      ( 4 )
+#define FILL_EL_ESC_COUNT_BITS  ( 8 )
+#define MAX_FILL_DATA_BYTES     ( 269 )
+    while (payloadBits >= (EL_ID_BITS + FILL_EL_COUNT_BITS)) {
+      INT cnt, esc_count=-1, alignBits=7;
+
+      payloadBits -= EL_ID_BITS + FILL_EL_COUNT_BITS;
+      if (payloadBits >= 15*8) {
+        payloadBits -= FILL_EL_ESC_COUNT_BITS;
+        esc_count = 0;  /* write esc_count even if cnt becomes smaller 15 */
+      }
+      alignBits = 0;
+
+      cnt = fixMin(MAX_FILL_DATA_BYTES, (payloadBits+alignBits)>>3);
+
+      if (cnt >= 15) {
+        esc_count = cnt - 15 + 1;
+      }
+
+      if (hBitStream != NULL) {
+        /* write bitstream */
+        FDKwriteBits(hBitStream, ID_FIL, EL_ID_BITS);
+        if (esc_count >= 0) {
+          FDKwriteBits(hBitStream, 15, FILL_EL_COUNT_BITS);
+          FDKwriteBits(hBitStream, esc_count, FILL_EL_ESC_COUNT_BITS);
+        } else {
+          FDKwriteBits(hBitStream, cnt, FILL_EL_COUNT_BITS);
+        }
+      }
+
+      extBitsUsed += EL_ID_BITS + FILL_EL_COUNT_BITS + ((esc_count>=0) ? FILL_EL_ESC_COUNT_BITS : 0);
+
+      cnt = fixMin(cnt*8, payloadBits);  /* convert back to bits */
+#if 0
+      extBitsUsed += FDKaacEnc_writeExtensionPayload( hBitStream,
+                                                      pExtension->type,
+                                                      pExtension->pPayload,
+                                                      cnt );
+#else
+      extBitsUsed += dabWrite_writeExtensionFillPayload(hBitStream, cnt);
+#endif
+      payloadBits -= cnt;
+    }
+}
+
+void dabWrite_EndRawDataBlock(HANDLE_DAB hDab,
+                          HANDLE_FDK_BITSTREAM hBs,
+                          int *pBits)
+{
+    FDK_BITSTREAM bsWriter;
+	INT crcIndex = 0;
+	USHORT crcData;
+	INT writeBits=0;
+	INT writeBitsNonLastBlock=0;
+	INT writeBitsLastBlock=0;
+#if 1
+    if (hDab->currentBlock == hDab->num_raw_blocks) {
+    	//calculate byte-alignment before writing ID_FIL
+    	if((FDKgetValidBits(hBs)+3) % 8){
+    		writeBits = 8 - ((FDKgetValidBits(hBs)+3) % 8);
+    	}
+
+    	INT offset_end = hDab->subchannels_num*110*8 - 2*8 - 3;
+    	writeBitsLastBlock = offset_end - FDKgetValidBits(hBs);
+        dabWrite_FillRawDataBlock(hBs, writeBitsLastBlock);
+		FDKsyncCache(hBs);
+		//fprintf(stderr, "FIL-element written=%d\n", writeBitsLastBlock);
+		writeBitsLastBlock=writeBits;
+    }
+#endif
+	FDKwriteBits(hBs, 7, 3); //finalize AU: ID_END
+	FDKsyncCache(hBs);
+	//byte-align (if ID_FIL doesn't align it).
+	if(FDKgetValidBits(hBs) % 8){
+		writeBits = 8 - (FDKgetValidBits(hBs) % 8);
+        FDKwriteBits(hBs, 0x00, writeBits);
+		FDKsyncCache(hBs);
+	}
+
+	//fake-written bits alignment for last AU
+	if (hDab->currentBlock == hDab->num_raw_blocks)
+		writeBits=writeBitsLastBlock;
+
+	INT frameLen = (FDKgetValidBits(hBs) - hDab->subFrameStartBit) >> 3;
+	//fprintf(stderr, "frame=%d, offset writeBits=%d\n", frameLen, writeBits);
+
+	FDK_ASSERT(FDKgetValidBits(hBs) % 8 == 0); //only aligned au's
+    FDK_ASSERT(hDab->subchannels_num*110*8 >= FDKgetValidBits(hBs)+2*8); //don't overlap superframe
+
+    FDKinitBitStream(&bsWriter, hBs->hBitBuf.Buffer, hBs->hBitBuf.bufSize, 0, BS_WRITER);
+    FDKpushFor(&bsWriter, hDab->subFrameStartBit);
+    FDKcrcReset(&hDab->crcInfo);
+    hDab->crcIndex = FDKcrcStartReg(&hDab->crcInfo, &bsWriter, 0);
+#if 0
+    if (hDab->currentBlock == hDab->num_raw_blocks) {
+		INT offset_size = hDab->subchannels_num*110*8 - 2*8 - FDKgetValidBits(hBs);
+		//fprintf(stderr, "offset_size=%d\n", offset_size >> 3);
+		FDKpushFor(hBs, offset_size);
+    }
+#endif
+
+    FDKpushFor(&bsWriter, FDKgetValidBits(hBs) - hDab->subFrameStartBit);
+	FDKcrcEndReg(&hDab->crcInfo,  &bsWriter, hDab->crcIndex);
+	crcData = FDKcrcGetCRC(&hDab->crcInfo);
+	//fprintf(stderr, "crcData = %04x\n", crcData);
+    /* Write inverted CRC of current raw data block */
+    FDKwriteBits(hBs, crcData ^ 0xffff, 16);
+	FDKsyncCache(hBs);
+
+
+    /* Write distance to current data block */
+	if(hDab->currentBlock) {
+		FDKinitBitStream(&bsWriter, hBs->hBitBuf.Buffer, hBs->hBitBuf.bufSize, 0, BS_WRITER);
+	    FDKpushFor(&bsWriter, 24 + (hDab->currentBlock-1)*12);
+		//fprintf(stderr, "FDKwriteBits() = %d\n", hDab->subFrameStartBit>>3);
+		FDKwriteBits(&bsWriter, (hDab->subFrameStartBit>>3), 12);
+		FDKsyncCache(&bsWriter);
+	}
+
+	/* Write FireCode */
+	if (hDab->currentBlock == hDab->num_raw_blocks) {
+		FDKinitBitStream(&bsWriter, hBs->hBitBuf.Buffer, hBs->hBitBuf.bufSize, 0, BS_WRITER);
+		FDKpushFor(&bsWriter, 16);
+
+		FDKcrcReset(&hDab->crcFire);
+		crcIndex = FDKcrcStartReg(&hDab->crcFire, &bsWriter, 72);
+		FDKpushFor(&bsWriter, 9*8); //9bytes
+		FDKcrcEndReg(&hDab->crcFire, &bsWriter, crcIndex);
+
+		crcData = FDKcrcGetCRC(&hDab->crcFire);
+		//fprintf(stderr, "Firecode: %04x\n", crcData);
+
+		FDKinitBitStream(&bsWriter, hBs->hBitBuf.Buffer, hBs->hBitBuf.bufSize, 0, BS_WRITER);
+		FDKwriteBits(&bsWriter, crcData, 16);
+		FDKsyncCache(&bsWriter);
+	}
+
+    if (hDab->currentBlock == 0)
+    	*pBits += hDab->headerBits;
+    else
+        *pBits += 16;
+
+    *pBits += writeBits + 3; //size: ID_END + alignment
+
+    /* Correct *pBits to reflect the amount of bits of the current subframe */
+    *pBits -= hDab->subFrameStartBit;
+    /* Fixup CRC bits, since they come after each raw data block */
+
+    hDab->currentBlock++;
+    //fprintf(stderr, "dabWrite_EndRawDataBlock() *pBits=%d (%d)\n", *pBits, *pBits >> 3);
+}
+
diff --git a/fdk-aac/libMpegTPEnc/src/tpenc_dab.h b/fdk-aac/libMpegTPEnc/src/tpenc_dab.h
new file mode 100644
index 0000000..17b83c6
--- /dev/null
+++ b/fdk-aac/libMpegTPEnc/src/tpenc_dab.h
@@ -0,0 +1,217 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+� Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur F�rderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/******************************** MPEG Audio Encoder **************************
+
+   Initial author:       serge
+   contents/description: DAB Transport writer
+
+******************************************************************************/
+
+#ifndef TPENC_DAB_H
+#define TPENC_DAB_H
+
+
+
+#include "tp_data.h"
+
+#include "FDK_crc.h"
+
+typedef struct {
+  USHORT frame_length;
+  UCHAR dac_rate;
+  UCHAR aac_channel_mode;
+  UCHAR sbr_flag;
+  UCHAR ps_flag;
+  UCHAR mpeg_surround_config;
+  UCHAR num_raw_blocks;
+  UCHAR BufferFullnesStartFlag;
+  int subchannels_num;
+  int headerBits;                /*!< Header bit demand for the current raw data block */
+  int currentBlock;              /*!< Index of current raw data block */
+  int subFrameStartBit;          /*!< Bit position where the current raw data block begins */
+  //int subFrameStartPrev;         /*!< Bit position where the previous raw data block begins */
+  int crcIndex;
+  FDK_CRCINFO crcInfo;
+  FDK_CRCINFO crcFire;
+  FDK_CRCINFO crcInfo2;
+  USHORT tab[256];
+} STRUCT_DAB;
+
+typedef STRUCT_DAB *HANDLE_DAB;
+
+/**
+ * \brief Initialize DAB data structure
+ *
+ * \param hDab DAB data handle
+ * \param config a valid CODER_CONFIG struct from where the required
+ *        information for the DAB header is extrated from
+ *
+ * \return 0 in case of success.
+ */
+INT dabWrite_Init(
+        HANDLE_DAB   hDab,
+        CODER_CONFIG *config
+        );
+
+/**
+ * \brief Get the total bit overhead caused by DAB
+ *
+ * \hDab handle to DAB data
+ *
+ * \return Amount of additional bits required for the current raw data block
+ */
+int dabWrite_GetHeaderBits( HANDLE_DAB hDab );
+int dabWrite_CountTotalBitDemandHeader( HANDLE_DAB hDab, unsigned int streamDataLength );
+
+/**
+ * \brief Write an DAB header into the given bitstream. May not write a header
+ *        in case of multiple raw data blocks.
+ *
+ * \param hDab DAB data handle
+ * \param hBitStream bitstream handle into which the DAB may be written into
+ * \param buffer_fullness the buffer fullness value for the DAB header
+ * \param the current raw data block length
+ *
+ * \return 0 in case of success.
+ */
+INT dabWrite_EncodeHeader(
+        HANDLE_DAB          hDab,
+        HANDLE_FDK_BITSTREAM hBitStream,
+        int                  bufferFullness,
+        int                  frame_length
+        );
+/**
+ * \brief Finish a DAB raw data block
+ *
+ * \param hDab DAB data handle
+ * \param hBs bitstream handle into which the DAB may be written into
+ * \param pBits a pointer to a integer holding the current bitstream buffer bit count,
+ *              which is corrected to the current raw data block boundary.
+ *
+ */
+void dabWrite_EndRawDataBlock(
+        HANDLE_DAB          hDab,
+        HANDLE_FDK_BITSTREAM hBs,
+        int                 *bits
+        );
+
+
+/**
+ * \brief Start CRC region with a maximum number of bits
+ *        If mBits is positive zero padding will be used for CRC calculation, if there
+ *        are less than mBits bits available.
+ *        If mBits is negative no zero padding is done.
+ *        If mBits is zero the memory for the buffer is allocated dynamically, the
+ *        number of bits is not limited.
+ *
+ * \param pDab DAB data handle
+ * \param hBs bitstream handle of which the CRC region ends
+ * \param mBits limit of number of bits to be considered for the requested CRC region
+ *
+ * \return ID for the created region, -1 in case of an error
+ */
+int dabWrite_CrcStartReg(
+        HANDLE_DAB          pDab,
+        HANDLE_FDK_BITSTREAM hBs,
+        int                  mBits
+        );
+
+/**
+ * \brief Ends CRC region identified by reg
+ *
+ * \param pDab DAB data handle
+ * \param hBs bitstream handle of which the CRC region ends
+ * \param reg a CRC region ID returned previously by dabWrite_CrcStartReg()
+ */
+void dabWrite_CrcEndReg(
+        HANDLE_DAB          pDab,
+        HANDLE_FDK_BITSTREAM hBs,
+        int                  reg
+        );
+
+
+
+
+#endif /* TPENC_DAB_H */
+
diff --git a/fdk-aac/libMpegTPEnc/src/tpenc_latm.cpp b/fdk-aac/libMpegTPEnc/src/tpenc_latm.cpp
new file mode 100644
index 0000000..2d35d48
--- /dev/null
+++ b/fdk-aac/libMpegTPEnc/src/tpenc_latm.cpp
@@ -0,0 +1,850 @@
+/* -----------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2018 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 transport format encoder library *********************
+
+   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 += hAss->otherDataLenBits;
+
+    /* 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->taraBufferFullness = 0xFF;
+  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)) {
+            const UINT alignAnchor = FDKgetValidBits(hBs);
+
+            if (0 !=
+                (transportEnc_writeASC(hBs, hAss->config[prog][layer], cb))) {
+              return TRANSPORTENC_UNKOWN_ERROR;
+            }
+
+            if (hAss->audioMuxVersion == 1) {
+              UINT ascLen = transportEnc_LatmWriteValue(hBs, 0);
+              FDKbyteAlign(hBs, alignAnchor);
+              ascLen = FDKgetValidBits(hBs) - alignAnchor - ascLen;
+              FDKpushBack(hBs, FDKgetValidBits(hBs) - alignAnchor);
+
+              transportEnc_LatmWriteValue(hBs, ascLen);
+
+              if (0 !=
+                  (transportEnc_writeASC(hBs, hAss->config[prog][layer], cb))) {
+                return TRANSPORTENC_UNKOWN_ERROR;
+              }
+
+              FDKbyteAlign(hBs, alignAnchor); /* asc length fillbits */
+            }
+
+            hAss->streamMuxConfigBits +=
+                FDKgetValidBits(hBs) -
+                alignAnchor; /* 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:
+              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->otherDataLenBits > 0) ? 1 : 0,
+                 1); /* otherDataPresent */
+    hAss->streamMuxConfigBits += 1;
+
+    if (hAss->otherDataLenBits > 0) {
+      FDKwriteBits(hBs, 0, 1);
+      FDKwriteBits(hBs, hAss->otherDataLenBits, 8);
+      hAss->streamMuxConfigBits += 9;
+    }
+
+    FDKwriteBits(hBs, 0, 1); /* crcCheckPresent=0 */
+    hAss->streamMuxConfigBits += 1;
+
+  } 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->otherDataLenBits = 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 */
+        if (TRANSPORTENC_OK != (ErrorStatus = CreateStreamMuxConfig(
+                                    hAss, hBs, bufferFullness, cb))) {
+          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->otherDataLenBits;
+    *bits += hAss->fillBits;
+  }
+}
+
+TRANSPORTENC_ERROR transportEnc_LatmGetFrame(HANDLE_LATM_STREAM hAss,
+                                             HANDLE_FDK_BITSTREAM hBs,
+                                             int *pBytes) {
+  TRANSPORTENC_ERROR ErrorStatus = TRANSPORTENC_OK;
+
+  hAss->subFrameCnt++;
+  if (hAss->subFrameCnt >= hAss->noSubframes) {
+    /* Add LOAS frame length if required. */
+    if (hAss->tt == TT_MP4_LOAS) {
+      FDK_BITSTREAM tmpBuf;
+
+      /* Determine frame length info */
+      hAss->audioMuxLengthBytes =
+          ((FDKgetValidBits(hBs) + hAss->otherDataLenBits + 7) >> 3) -
+          3; /* 3=Syncword + length */
+
+      /* Check frame length info */
+      if (hAss->audioMuxLengthBytes >= (1 << 13)) {
+        ErrorStatus = TRANSPORTENC_INVALID_AU_LENGTH;
+        goto bail;
+      }
+
+      /* Write length info into assembler buffer */
+      FDKinitBitStream(&tmpBuf, hBs->hBitBuf.Buffer, hBs->hBitBuf.bufSize, 0,
+                       BS_WRITER);
+      FDKpushFor(&tmpBuf, hAss->audioMuxLengthBytesPos);
+      FDKwriteBits(&tmpBuf, hAss->audioMuxLengthBytes, 13);
+      FDKsyncCache(&tmpBuf);
+    }
+
+    /* Write AudioMuxElement other data bits */
+    FDKwriteBits(hBs, 0, hAss->otherDataLenBits);
+
+    /* Write AudioMuxElement byte alignment fill bits */
+    FDKwriteBits(hBs, 0, hAss->fillBits);
+
+    FDK_ASSERT((FDKgetValidBits(hBs) % 8) == 0);
+
+    hAss->subFrameCnt = 0;
+
+    FDKsyncCache(hBs);
+    *pBytes = (FDKgetValidBits(hBs) + 7) >> 3;
+
+    if (hAss->muxConfigPeriod > 0) {
+      hAss->latmFrameCounter++;
+
+      if (hAss->latmFrameCounter >= hAss->muxConfigPeriod) {
+        hAss->latmFrameCounter = 0;
+        hAss->noSubframes = hAss->noSubframes_next;
+      }
+    }
+  } else {
+    /* No data this time */
+    *pBytes = 0;
+  }
+
+bail:
+  return ErrorStatus;
+}
+
+/**
+ * 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 */
+  if (TRANSPORTENC_OK !=
+      (ErrorStatus = AdvanceAudioMuxElement(hAss, hBs, 0, 0, cb))) {
+    goto bail;
+  }
+
+  // CreateStreamMuxConfig(hAss, hBs, 0);
+
+bail:
+  return ErrorStatus;
+}
+
+TRANSPORTENC_ERROR transportEnc_LatmAddOtherDataBits(HANDLE_LATM_STREAM hAss,
+                                                     const int otherDataBits) {
+  TRANSPORTENC_ERROR ErrorStatus = TRANSPORTENC_OK;
+
+  if ((hAss->otherDataLenBits != 0) || (otherDataBits % 8 != 0)) {
+    /* This implementation allows to add other data bits only once.
+       To keep existing alignment only whole bytes are allowed. */
+    ErrorStatus = TRANSPORTENC_UNKOWN_ERROR;
+  } else {
+    /* Ensure correct addional bits in payload. */
+    if (hAss->tt == TT_MP4_LATM_MCP0) {
+      hAss->otherDataLenBits = otherDataBits;
+    } else {
+      hAss->otherDataLenBits = otherDataBits - 9;
+      hAss->streamMuxConfigBits += 9;
+    }
+  }
+
+  return ErrorStatus;
+}
diff --git a/fdk-aac/libMpegTPEnc/src/tpenc_latm.h b/fdk-aac/libMpegTPEnc/src/tpenc_latm.h
new file mode 100644
index 0000000..d650357
--- /dev/null
+++ b/fdk-aac/libMpegTPEnc/src/tpenc_latm.h
@@ -0,0 +1,274 @@
+/* -----------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2018 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 transport format encoder library *********************
+
+   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;
+
+  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 otherDataLenBits; /* AudioMuxElement other data bits */
+  UCHAR fillBits;         /* AudioMuxElement fill bits */
+  UINT 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 Write addional other data bits in AudioMuxElement
+ *
+ * \param hAss HANDLE_LATM_STREAM handle
+ * \param otherDataBits number of other data bits to be written
+ *
+ * \return an TRANSPORTENC_ERROR error code
+ */
+TRANSPORTENC_ERROR transportEnc_LatmAddOtherDataBits(HANDLE_LATM_STREAM hAss,
+                                                     const int otherDataBits);
+
+/**
+ * \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 an TRANSPORTENC_ERROR error code
+ */
+TRANSPORTENC_ERROR 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/fdk-aac/libMpegTPEnc/src/tpenc_lib.cpp b/fdk-aac/libMpegTPEnc/src/tpenc_lib.cpp
new file mode 100644
index 0000000..316c6e0
--- /dev/null
+++ b/fdk-aac/libMpegTPEnc/src/tpenc_lib.cpp
@@ -0,0 +1,713 @@
+/* -----------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2018 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 transport format encoder library *********************
+
+   Author(s):   Manuel Jander
+
+   Description: MPEG Transport encode
+
+*******************************************************************************/
+
+#include "tpenc_lib.h"
+
+/* library info */
+#include "tp_version.h"
+
+#define MODULE_NAME "transportEnc"
+
+#include "tpenc_asc.h"
+
+#include "tpenc_adts.h"
+
+#include "tpenc_adif.h"
+
+#include "tpenc_dab.h"
+
+#include "tpenc_latm.h"
+
+typedef struct {
+  int curSubFrame;
+  int nSubFrames;
+  int prevBits;
+} RAWPACKETS_INFO;
+
+struct TRANSPORTENC {
+  CODER_CONFIG config;
+  TRANSPORT_TYPE transportFmt; /*!< MPEG4 transport type. */
+
+  FDK_BITSTREAM bitStream;
+  UCHAR *bsBuffer;
+  INT bsBufferSize;
+
+  INT pceFrameCounter; /*!< Indicates frame period when PCE must be written in
+                          raw_data_block. -1 means not to write a PCE in
+                          raw_dat_block. */
+  union {
+    STRUCT_ADTS adts;
+
+    ADIF_INFO adif;
+
+    STRUCT_DAB dab;
+
+    LATM_STREAM latm;
+
+    RAWPACKETS_INFO raw;
+
+  } writer;
+
+  CSTpCallBacks callbacks;
+};
+
+typedef struct _TRANSPORTENC_STRUCT TRANSPORTENC_STRUCT;
+
+/*
+ * MEMORY Declaration
+ */
+
+C_ALLOC_MEM(Ram_TransportEncoder, struct TRANSPORTENC, 1)
+
+TRANSPORTENC_ERROR transportEnc_Open(HANDLE_TRANSPORTENC *phTpEnc) {
+  HANDLE_TRANSPORTENC hTpEnc;
+
+  if (phTpEnc == NULL) {
+    return TRANSPORTENC_INVALID_PARAMETER;
+  }
+
+  hTpEnc = GetRam_TransportEncoder(0);
+
+  if (hTpEnc == NULL) {
+    return TRANSPORTENC_NO_MEM;
+  }
+
+  *phTpEnc = hTpEnc;
+  return TRANSPORTENC_OK;
+}
+
+/**
+ * \brief  Get frame period of PCE in raw_data_block.
+ *
+ * - Write PCE only if necessary. PCE can be part of the ASC if chConfig==0
+ * whererfore no additonal PCE will be written in raw_data_block.
+ * - A matrixMixdown coefficient can only be written if chConfig is 5.0 or 5.1.
+ * - The PCE repetition rate in raw_data_block can be controlled via
+ * headerPeriod parameter.
+ *
+ * \param channelMode           Encoder Channel Mode.
+ * \param channelConfigZero     No standard channel configuration.
+ * \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 CHANNEL_MODE channelMode,
+                                const int channelConfigZero,
+                                const TRANSPORT_TYPE transportFmt,
+                                const int headerPeriod,
+                                const int matrixMixdownA) {
+  INT pceFrameCounter = -1; /* variable to be returned */
+
+  if (headerPeriod > 0) {
+    switch (getChannelConfig(channelMode, channelConfigZero)) {
+      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 */
+            if ((channelMode == MODE_1_2_2) || (channelMode == MODE_1_2_2_1)) {
+              pceFrameCounter = headerPeriod; /* repeating pce only meaningful
+                                                 for potential matrix mixdown */
+              break;
+            }
+            FDK_FALLTHROUGH;
+          case TT_MP4_LOAS:      /* PCE in ASC if chChonfig==0 */
+          case TT_MP4_LATM_MCP1: /* PCE in ASC if chChonfig==0 */
+          case TT_DABPLUS:
+          default:
+            pceFrameCounter = -1; /* no PCE in raw_data_block */
+        }
+        break;
+      case 5: /* MODE_1_2_2 */
+      case 6: /* MODE_1_2_2_1 */
+        /* matrixMixdownCoefficient can only be written if 5.0 and 5.1 config
+         * present. */
+        if (matrixMixdownA != 0) {
+          switch (transportFmt) {
+            case TT_MP4_ADIF: /* ADIF header comprises PCE */
+            case TT_MP4_ADTS:
+            case TT_MP4_LOAS:      /* no PCE in ASC because chConfig!=0 */
+            case TT_MP4_LATM_MCP1: /* no PCE in ASC because chConfig!=0 */
+            case TT_MP4_LATM_MCP0:
+            case TT_MP4_RAW:
+              pceFrameCounter = headerPeriod;
+              break;
+            case TT_DABPLUS:
+            default:
+              pceFrameCounter = -1; /* no PCE in raw_data_block */
+          }                         /* switch transportFmt */
+        }                           /* if matrixMixdownA!=0 */
+        break;
+      default:
+        pceFrameCounter = -1; /* no PCE in raw_data_block */
+    }                         /* switch getChannelConfig() */
+  }                           /* if headerPeriod>0  */
+  else {
+    pceFrameCounter = -1; /* no PCE in raw_data_block */
+  }
+
+  return pceFrameCounter;
+}
+
+TRANSPORTENC_ERROR transportEnc_Init(HANDLE_TRANSPORTENC hTpEnc,
+                                     UCHAR *bsBuffer, INT bsBufferSize,
+                                     TRANSPORT_TYPE transportFmt,
+                                     CODER_CONFIG *cconfig, UINT flags) {
+  /* Copy configuration structure */
+  FDKmemcpy(&hTpEnc->config, cconfig, sizeof(CODER_CONFIG));
+
+  /* Init transportEnc struct. */
+  hTpEnc->transportFmt = transportFmt;
+
+  hTpEnc->bsBuffer = bsBuffer;
+  hTpEnc->bsBufferSize = bsBufferSize;
+
+  FDKinitBitStream(&hTpEnc->bitStream, hTpEnc->bsBuffer, hTpEnc->bsBufferSize,
+                   0, BS_WRITER);
+
+  switch (transportFmt) {
+    case TT_MP4_ADIF:
+      /* Sanity checks */
+      if ((hTpEnc->config.aot != AOT_AAC_LC) ||
+          (hTpEnc->config.samplesPerFrame != 1024)) {
+        return TRANSPORTENC_INVALID_PARAMETER;
+      }
+      hTpEnc->writer.adif.headerWritten = 0;
+      hTpEnc->writer.adif.samplingRate = hTpEnc->config.samplingRate;
+      hTpEnc->writer.adif.bitRate = hTpEnc->config.bitRate;
+      hTpEnc->writer.adif.profile = ((int)hTpEnc->config.aot) - 1;
+      hTpEnc->writer.adif.cm = hTpEnc->config.channelMode;
+      hTpEnc->writer.adif.bVariableRate = 0;
+      hTpEnc->writer.adif.instanceTag = 0;
+      hTpEnc->writer.adif.matrixMixdownA = hTpEnc->config.matrixMixdownA;
+      hTpEnc->writer.adif.pseudoSurroundEnable =
+          (hTpEnc->config.flags & CC_PSEUDO_SURROUND) ? 1 : 0;
+      break;
+
+    case TT_MP4_ADTS:
+      /* Sanity checks */
+      if ((hTpEnc->config.aot != AOT_AAC_LC) ||
+          (hTpEnc->config.samplesPerFrame != 1024)) {
+        return TRANSPORTENC_INVALID_PARAMETER;
+      }
+      if (adtsWrite_Init(&hTpEnc->writer.adts, &hTpEnc->config) != 0) {
+        return TRANSPORTENC_INVALID_PARAMETER;
+      }
+      break;
+
+  case TT_DABPLUS:
+    /* Sanity checks */
+    if ( ( hTpEnc->config.aot != AOT_AAC_LC)
+       ||(hTpEnc->config.samplesPerFrame != 960) )
+    {
+      return TRANSPORTENC_INVALID_PARAMETER;
+    }
+    if ( dabWrite_Init(&hTpEnc->writer.dab, &hTpEnc->config) != 0) {
+      return TRANSPORTENC_INVALID_PARAMETER;
+    }
+    break;
+
+    case TT_MP4_LOAS:
+    case TT_MP4_LATM_MCP0:
+    case TT_MP4_LATM_MCP1: {
+      TRANSPORTENC_ERROR error;
+
+      error = transportEnc_Latm_Init(&hTpEnc->writer.latm, &hTpEnc->bitStream,
+                                     &hTpEnc->config, flags & TP_FLAG_LATM_AMV,
+                                     transportFmt, &hTpEnc->callbacks);
+      if (error != TRANSPORTENC_OK) {
+        return error;
+      }
+    } break;
+
+    case TT_MP4_RAW:
+      hTpEnc->writer.raw.curSubFrame = 0;
+      hTpEnc->writer.raw.nSubFrames = hTpEnc->config.nSubFrames;
+      break;
+
+    default:
+      return TRANSPORTENC_INVALID_PARAMETER;
+  }
+
+  /* pceFrameCounter indicates if PCE must be written in raw_data_block. */
+  hTpEnc->pceFrameCounter = getPceRepetitionRate(
+      hTpEnc->config.channelMode, hTpEnc->config.channelConfigZero,
+      transportFmt, hTpEnc->config.headerPeriod, hTpEnc->config.matrixMixdownA);
+
+  return TRANSPORTENC_OK;
+}
+
+TRANSPORTENC_ERROR transportEnc_AddOtherDataBits(HANDLE_TRANSPORTENC hTpEnc,
+                                                 const int nBits) {
+  TRANSPORTENC_ERROR tpErr = TRANSPORTENC_OK;
+
+  switch (hTpEnc->transportFmt) {
+    case TT_MP4_LATM_MCP0:
+    case TT_MP4_LATM_MCP1:
+    case TT_MP4_LOAS:
+      tpErr = transportEnc_LatmAddOtherDataBits(&hTpEnc->writer.latm, nBits);
+      break;
+    case TT_MP4_ADTS:
+    case TT_MP4_ADIF:
+    case TT_MP4_RAW:
+    default:
+      tpErr = TRANSPORTENC_UNKOWN_ERROR;
+  }
+
+  return tpErr;
+}
+
+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;
+}
+int transportEnc_RegisterUsacCallback(HANDLE_TRANSPORTENC hTpEnc,
+                                      const cbUsac_t cbUsac, void *user_data) {
+  if (hTpEnc == NULL) {
+    return -1;
+  }
+  hTpEnc->callbacks.cbUsac = cbUsac;
+  hTpEnc->callbacks.cbUsacData = user_data;
+  return 0;
+}
+
+int transportEnc_RegisterSscCallback(HANDLE_TRANSPORTENC hTpEnc,
+                                     const cbSsc_t cbSsc, void *user_data) {
+  if (hTpEnc == NULL) {
+    return -1;
+  }
+  hTpEnc->callbacks.cbSsc = cbSsc;
+  hTpEnc->callbacks.cbSscData = 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);
+      if (0 != adifWrite_EncodeHeader(&hTp->writer.adif, hBs, bufferFullness)) {
+        err = TRANSPORTENC_INVALID_CONFIG;
+      }
+      break;
+    case TT_MP4_ADTS:
+      bufferFullness /= ncc; /* Number of Considered Channels */
+      bufferFullness /= 32;
+      bufferFullness = FDKmin(0x7FF, bufferFullness); /* Signal variable rate */
+      adtsWrite_EncodeHeader(&hTp->writer.adts, &hTp->bitStream, bufferFullness,
+                             frameUsedBits);
+      break;
+    case TT_DABPLUS:
+      bufferFullness /= ncc;                          /* Number of Considered Channels */
+      bufferFullness /= 32;
+      bufferFullness = FDKmin(0x7FF, bufferFullness); /* Signal variable rate */
+      dabWrite_EncodeHeader(
+             &hTp->writer.dab,
+             &hTp->bitStream,
+              bufferFullness,
+              frameUsedBits
+              );
+      break;
+    case TT_MP4_LOAS:
+    case TT_MP4_LATM_MCP0:
+    case TT_MP4_LATM_MCP1:
+      bufferFullness /= ncc; /* Number of Considered Channels */
+      bufferFullness /= 32;
+      bufferFullness = FDKmin(0xFF, bufferFullness); /* Signal variable rate */
+      transportEnc_LatmWrite(&hTp->writer.latm, hBs, frameUsedBits,
+                             bufferFullness, &hTp->callbacks);
+      break;
+    case TT_MP4_RAW:
+      if (hTp->writer.raw.curSubFrame >= hTp->writer.raw.nSubFrames) {
+        hTp->writer.raw.curSubFrame = 0;
+        FDKinitBitStream(&hTp->bitStream, hTp->bsBuffer, hTp->bsBufferSize, 0,
+                         BS_WRITER);
+      }
+      hTp->writer.raw.prevBits = FDKgetValidBits(hBs);
+      break;
+    default:
+      err = TRANSPORTENC_UNSUPPORTED_FORMAT;
+      break;
+  }
+
+  /* Write PCE in raw_data_block if required */
+  if (hTp->pceFrameCounter >= hTp->config.headerPeriod) {
+    INT crcIndex = 0;
+    /* Align inside PCE with repsect to the first bit of the raw_data_block() */
+    UINT alignAnchor = FDKgetValidBits(&hTp->bitStream);
+
+    /* Write PCE element ID bits */
+    FDKwriteBits(&hTp->bitStream, ID_PCE, 3);
+
+    if ((hTp->transportFmt == TT_MP4_ADTS) &&
+        !hTp->writer.adts.protection_absent) {
+      crcIndex = adtsWrite_CrcStartReg(&hTp->writer.adts, &hTp->bitStream, 0);
+    }
+
+    /* Write PCE as first raw_data_block element */
+    transportEnc_writePCE(
+        &hTp->bitStream, hTp->config.channelMode, hTp->config.samplingRate, 0,
+        1, hTp->config.matrixMixdownA,
+        (hTp->config.flags & CC_PSEUDO_SURROUND) ? 1 : 0, alignAnchor);
+
+    if ((hTp->transportFmt == TT_MP4_ADTS) &&
+        !hTp->writer.adts.protection_absent) {
+      adtsWrite_CrcEndReg(&hTp->writer.adts, &hTp->bitStream, crcIndex);
+    }
+    hTp->pceFrameCounter = 0; /* reset pce frame counter */
+  }
+
+  if (hTp->pceFrameCounter != -1) {
+    hTp->pceFrameCounter++; /* Update pceFrameCounter only if PCE writing is
+                               active. */
+  }
+
+  return err;
+}
+
+TRANSPORTENC_ERROR transportEnc_EndAccessUnit(HANDLE_TRANSPORTENC hTp,
+                                              int *bits) {
+  switch (hTp->transportFmt) {
+    case TT_MP4_LATM_MCP0:
+    case TT_MP4_LATM_MCP1:
+    case TT_MP4_LOAS:
+      transportEnc_LatmAdjustSubframeBits(&hTp->writer.latm, bits);
+      break;
+    case TT_MP4_ADTS:
+      adtsWrite_EndRawDataBlock(&hTp->writer.adts, &hTp->bitStream, bits);
+      break;
+    case TT_DABPLUS:
+      dabWrite_EndRawDataBlock(&hTp->writer.dab, &hTp->bitStream, bits);
+      break;
+    case TT_MP4_ADIF:
+      /* Substract ADIF header from AU bits, not to be considered. */
+      *bits -= adifWrite_GetHeaderBits(&hTp->writer.adif);
+      hTp->writer.adif.headerWritten = 1;
+      break;
+    case TT_MP4_RAW:
+      *bits -= hTp->writer.raw.prevBits;
+      break;
+    default:
+      break;
+  }
+
+  return TRANSPORTENC_OK;
+}
+
+TRANSPORTENC_ERROR transportEnc_GetFrame(HANDLE_TRANSPORTENC hTpEnc,
+                                         int *nbytes) {
+  TRANSPORTENC_ERROR tpErr = TRANSPORTENC_OK;
+  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;
+      tpErr = transportEnc_LatmGetFrame(&hTpEnc->writer.latm, hBs, nbytes);
+      break;
+    case TT_MP4_ADTS:
+      if (hTpEnc->writer.adts.currentBlock >=
+          hTpEnc->writer.adts.num_raw_blocks + 1) {
+        *nbytes = (FDKgetValidBits(hBs) + 7) >> 3;
+        hTpEnc->writer.adts.currentBlock = 0;
+      } else {
+        *nbytes = 0;
+      }
+      break;
+    case TT_DABPLUS:
+      if (hTpEnc->writer.dab.currentBlock >= hTpEnc->writer.dab.num_raw_blocks+1) {
+        *nbytes = (FDKgetValidBits(hBs) + 7)>>3;
+        hTpEnc->writer.dab.currentBlock = 0;
+      } else {
+        *nbytes = 0;
+      }
+      break;
+    case TT_MP4_ADIF:
+      FDK_ASSERT((INT)FDKgetValidBits(hBs) >= 0);
+      *nbytes = (FDKgetValidBits(hBs) + 7) >> 3;
+      break;
+    case TT_MP4_RAW:
+      FDKsyncCache(hBs);
+      hTpEnc->writer.raw.curSubFrame++;
+      *nbytes = ((FDKgetValidBits(hBs) - hTpEnc->writer.raw.prevBits) + 7) >> 3;
+      break;
+    default:
+      break;
+  }
+
+  return tpErr;
+}
+
+INT transportEnc_GetStaticBits(HANDLE_TRANSPORTENC hTp, int auBits) {
+  INT nbits = 0, nPceBits = 0;
+
+  /* Write PCE within raw_data_block in transport lib. */
+  if (hTp->pceFrameCounter >= hTp->config.headerPeriod) {
+    nPceBits = transportEnc_GetPCEBits(
+        hTp->config.channelMode, hTp->config.matrixMixdownA,
+        3);             /* Consider 3 bits ID signalling in alignment */
+    auBits += nPceBits; /* Adapt required raw_data_block bit consumtpion for AU
+                           length information e.g. in LATM/LOAS configuration.
+                         */
+  }
+
+  switch (hTp->transportFmt) {
+    case TT_MP4_ADIF:
+    case TT_MP4_RAW:
+      nbits = 0; /* Do not consider the ADIF header into the total bitrate */
+      break;
+    case TT_MP4_ADTS:
+      nbits = adtsWrite_GetHeaderBits(&hTp->writer.adts);
+      break;
+    case TT_DABPLUS:
+      nbits = dabWrite_CountTotalBitDemandHeader(&hTp->writer.dab, auBits);
+      break;
+    case TT_MP4_LOAS:
+    case TT_MP4_LATM_MCP0:
+    case TT_MP4_LATM_MCP1:
+      nbits =
+          transportEnc_LatmCountTotalBitDemandHeader(&hTp->writer.latm, auBits);
+      break;
+    default:
+      nbits = 0;
+      break;
+  }
+
+  /* PCE is written in the transport library therefore the bit consumption is
+   * part of the transport static bits. */
+  nbits += nPceBits;
+
+  return nbits;
+}
+
+void transportEnc_Close(HANDLE_TRANSPORTENC *phTp) {
+  if (phTp != NULL) {
+    if (*phTp != NULL) {
+      FreeRam_TransportEncoder(phTp);
+    }
+  }
+}
+
+int transportEnc_CrcStartReg(HANDLE_TRANSPORTENC hTpEnc, int mBits) {
+  int crcReg = 0;
+
+  switch (hTpEnc->transportFmt) {
+    case TT_MP4_ADTS:
+      crcReg = adtsWrite_CrcStartReg(&hTpEnc->writer.adts, &hTpEnc->bitStream,
+                                     mBits);
+      break;
+    case TT_DABPLUS:
+      crcReg = dabWrite_CrcStartReg(&hTpEnc->writer.dab, &hTpEnc->bitStream, mBits);
+      break;
+    default:
+      break;
+  }
+
+  return crcReg;
+}
+
+void transportEnc_CrcEndReg(HANDLE_TRANSPORTENC hTpEnc, int reg) {
+  switch (hTpEnc->transportFmt) {
+    case TT_MP4_ADTS:
+      adtsWrite_CrcEndReg(&hTpEnc->writer.adts, &hTpEnc->bitStream, reg);
+      break;
+    case TT_DABPLUS:
+      dabWrite_CrcEndReg(&hTpEnc->writer.dab, &hTpEnc->bitStream, reg);
+      break;
+    default:
+      break;
+  }
+}
+
+TRANSPORTENC_ERROR transportEnc_GetConf(HANDLE_TRANSPORTENC hTpEnc,
+                                        CODER_CONFIG *cc,
+                                        FDK_BITSTREAM *dataBuffer,
+                                        UINT *confType) {
+  TRANSPORTENC_ERROR tpErr = TRANSPORTENC_OK;
+  HANDLE_LATM_STREAM hLatmConfig = &hTpEnc->writer.latm;
+
+  *confType = 0; /* set confType variable to default */
+
+  /* write StreamMuxConfig or AudioSpecificConfig depending on format used */
+  switch (hTpEnc->transportFmt) {
+    case TT_MP4_LATM_MCP0:
+    case TT_MP4_LATM_MCP1:
+    case TT_MP4_LOAS:
+      tpErr =
+          CreateStreamMuxConfig(hLatmConfig, dataBuffer, 0, &hTpEnc->callbacks);
+      *confType = 1; /* config is SMC */
+      break;
+    default:
+      if (transportEnc_writeASC(dataBuffer, cc, &hTpEnc->callbacks) != 0) {
+        tpErr = TRANSPORTENC_UNKOWN_ERROR;
+      }
+  }
+
+  return tpErr;
+}
+
+TRANSPORTENC_ERROR transportEnc_GetLibInfo(LIB_INFO *info) {
+  int i;
+
+  if (info == NULL) {
+    return TRANSPORTENC_INVALID_PARAMETER;
+  }
+  /* search for next free tab */
+  for (i = 0; i < FDK_MODULE_LAST; i++) {
+    if (info[i].module_id == FDK_NONE) break;
+  }
+  if (i == FDK_MODULE_LAST) {
+    return TRANSPORTENC_UNKOWN_ERROR;
+  }
+  info += i;
+
+  info->module_id = FDK_TPENC;
+  info->version = LIB_VERSION(TP_LIB_VL0, TP_LIB_VL1, TP_LIB_VL2);
+  LIB_VERSION_STRING(info);
+#ifdef __ANDROID__
+  info->build_date = "";
+  info->build_time = "";
+#else
+  info->build_date = __DATE__;
+  info->build_time = __TIME__;
+#endif
+  info->title = TP_LIB_TITLE;
+
+  /* Set flags */
+  info->flags =
+      0 | CAPF_ADIF | CAPF_ADTS | CAPF_LATM | CAPF_LOAS | CAPF_RAWPACKETS | CAPF_DAB_AAC;
+
+  return TRANSPORTENC_OK;
+}