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authorMatthias P. Braendli <matthias.braendli@mpb.li>2019-11-11 11:38:02 +0100
committerMatthias P. Braendli <matthias.braendli@mpb.li>2019-11-11 11:38:02 +0100
commit0e5af65c467b2423a0b857ae3ad98c91acc1e190 (patch)
treed07f69550d8886271e44fe79c4dcfb299cafbd38 /fdk-aac/libMpegTPEnc/src
parentefe406d9724f959c8bc2a31802559ca6d41fd897 (diff)
downloadODR-AudioEnc-0e5af65c467b2423a0b857ae3ad98c91acc1e190.tar.gz
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ODR-AudioEnc-0e5af65c467b2423a0b857ae3ad98c91acc1e190.zip
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.
Diffstat (limited to 'fdk-aac/libMpegTPEnc/src')
-rw-r--r--fdk-aac/libMpegTPEnc/src/tp_version.h118
-rw-r--r--fdk-aac/libMpegTPEnc/src/tpenc_adif.cpp186
-rw-r--r--fdk-aac/libMpegTPEnc/src/tpenc_adif.h146
-rw-r--r--fdk-aac/libMpegTPEnc/src/tpenc_adts.cpp319
-rw-r--r--fdk-aac/libMpegTPEnc/src/tpenc_adts.h208
-rw-r--r--fdk-aac/libMpegTPEnc/src/tpenc_asc.cpp996
-rw-r--r--fdk-aac/libMpegTPEnc/src/tpenc_asc.h147
-rw-r--r--fdk-aac/libMpegTPEnc/src/tpenc_dab.cpp467
-rw-r--r--fdk-aac/libMpegTPEnc/src/tpenc_dab.h217
-rw-r--r--fdk-aac/libMpegTPEnc/src/tpenc_latm.cpp850
-rw-r--r--fdk-aac/libMpegTPEnc/src/tpenc_latm.h274
-rw-r--r--fdk-aac/libMpegTPEnc/src/tpenc_lib.cpp713
12 files changed, 4641 insertions, 0 deletions
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;
+}