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authorMatthias P. Braendli <matthias.braendli@mpb.li>2016-09-10 20:15:44 +0200
committerMatthias P. Braendli <matthias.braendli@mpb.li>2016-09-10 20:15:44 +0200
commit14c7b800eaa23e9da7c92c7c4df397d0c191f097 (patch)
treed840b6ec41ff74d1184ca1dcd7731d08f1e9ebbb /libMpegTPEnc/src
parent78a801e4d716c6f2403cc56cf6c5b6f138f24b2f (diff)
downloadODR-AudioEnc-14c7b800eaa23e9da7c92c7c4df397d0c191f097.tar.gz
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Remove FDK-AAC
Diffstat (limited to 'libMpegTPEnc/src')
-rw-r--r--libMpegTPEnc/src/tpenc_adif.cpp182
-rw-r--r--libMpegTPEnc/src/tpenc_adif.h135
-rw-r--r--libMpegTPEnc/src/tpenc_adts.cpp315
-rw-r--r--libMpegTPEnc/src/tpenc_adts.h218
-rw-r--r--libMpegTPEnc/src/tpenc_asc.cpp576
-rw-r--r--libMpegTPEnc/src/tpenc_asc.h142
-rw-r--r--libMpegTPEnc/src/tpenc_dab.cpp467
-rw-r--r--libMpegTPEnc/src/tpenc_dab.h217
-rw-r--r--libMpegTPEnc/src/tpenc_latm.cpp882
-rw-r--r--libMpegTPEnc/src/tpenc_latm.h264
-rw-r--r--libMpegTPEnc/src/tpenc_lib.cpp685
-rw-r--r--libMpegTPEnc/src/version8
12 files changed, 0 insertions, 4091 deletions
diff --git a/libMpegTPEnc/src/tpenc_adif.cpp b/libMpegTPEnc/src/tpenc_adif.cpp
deleted file mode 100644
index b48a32e..0000000
--- a/libMpegTPEnc/src/tpenc_adif.cpp
+++ /dev/null
@@ -1,182 +0,0 @@
-
-/* -----------------------------------------------------------------------------------------------------------
-Software License for The Fraunhofer FDK AAC Codec Library for Android
-
-© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
- All rights reserved.
-
- 1. INTRODUCTION
-The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
-the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
-This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
-
-AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
-audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
-independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
-of the MPEG specifications.
-
-Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
-may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
-individually for the purpose of encoding or decoding bit streams in products that are compliant with
-the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
-these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
-software may already be covered under those patent licenses when it is used for those licensed purposes only.
-
-Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
-are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
-applications information and documentation.
-
-2. COPYRIGHT LICENSE
-
-Redistribution and use in source and binary forms, with or without modification, are permitted without
-payment of copyright license fees provided that you satisfy the following conditions:
-
-You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
-your modifications thereto in source code form.
-
-You must retain the complete text of this software license in the documentation and/or other materials
-provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
-You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
-modifications thereto to recipients of copies in binary form.
-
-The name of Fraunhofer may not be used to endorse or promote products derived from this library without
-prior written permission.
-
-You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
-software or your modifications thereto.
-
-Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
-and the date of any change. For modified versions of the FDK AAC Codec, the term
-"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
-"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
-
-3. NO PATENT LICENSE
-
-NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
-ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
-respect to this software.
-
-You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
-by appropriate patent licenses.
-
-4. DISCLAIMER
-
-This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
-"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
-of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
-CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
-including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
-or business interruption, however caused and on any theory of liability, whether in contract, strict
-liability, or tort (including negligence), arising in any way out of the use of this software, even if
-advised of the possibility of such damage.
-
-5. CONTACT INFORMATION
-
-Fraunhofer Institute for Integrated Circuits IIS
-Attention: Audio and Multimedia Departments - FDK AAC LL
-Am Wolfsmantel 33
-91058 Erlangen, Germany
-
-www.iis.fraunhofer.de/amm
-amm-info@iis.fraunhofer.de
------------------------------------------------------------------------------------------------------------ */
-
-/******************************** MPEG Audio Encoder **************************
-
- contents/description: ADIF Transport Headers writing
-
-******************************************************************************/
-
-#include "tpenc_adif.h"
-
-#include "tpenc_lib.h"
-#include "tpenc_asc.h"
-
-
-
-int adifWrite_EncodeHeader(ADIF_INFO *adif,
- HANDLE_FDK_BITSTREAM hBs,
- INT adif_buffer_fullness)
-{
- /* ADIF/PCE/ADTS definitions */
- const char adifId[5]="ADIF";
- const int copyRightIdPresent=0;
- const int originalCopy=0;
- const int home=0;
-
- int i;
-
- INT sampleRate = adif->samplingRate;
- INT totalBitRate = adif->bitRate;
-
- if (adif->headerWritten)
- return 0;
-
- /* Align inside PCE with respect to the first bit of the header */
- UINT alignAnchor = FDKgetValidBits(hBs);
-
- /* Signal variable bitrate if buffer fullnes exceeds 20 bit */
- adif->bVariableRate = ( adif_buffer_fullness >= (INT)(0x1<<20) ) ? 1 : 0;
-
- FDKwriteBits(hBs, adifId[0],8);
- FDKwriteBits(hBs, adifId[1],8);
- FDKwriteBits(hBs, adifId[2],8);
- FDKwriteBits(hBs, adifId[3],8);
-
-
- FDKwriteBits(hBs, copyRightIdPresent ? 1:0,1);
-
- if(copyRightIdPresent) {
- for(i=0;i<72;i++) {
- FDKwriteBits(hBs,0,1);
- }
- }
- FDKwriteBits(hBs, originalCopy ? 1:0,1);
- FDKwriteBits(hBs, home ? 1:0,1);
- FDKwriteBits(hBs, adif->bVariableRate?1:0, 1);
- FDKwriteBits(hBs, totalBitRate,23);
-
- /* we write only one PCE at the moment */
- FDKwriteBits(hBs, 0, 4);
-
- if(!adif->bVariableRate) {
- FDKwriteBits(hBs, adif_buffer_fullness, 20);
- }
-
- /* Write PCE */
- transportEnc_writePCE(hBs, adif->cm, sampleRate, adif->instanceTag, adif->profile, 0, 0, alignAnchor);
-
- return 0;
-}
-
-int adifWrite_GetHeaderBits(ADIF_INFO *adif)
-{
- /* ADIF definitions */
- const int copyRightIdPresent=0;
-
- if (adif->headerWritten)
- return 0;
-
- int bits = 0;
-
- bits += 8*4; /* ADIF ID */
-
- bits += 1; /* Copyright present */
-
- if (copyRightIdPresent)
- bits += 72; /* Copyright ID */
-
- bits += 26;
-
- bits += 4; /* Number of PCE's */
-
- if(!adif->bVariableRate) {
- bits += 20;
- }
-
- /* write PCE */
- bits = transportEnc_GetPCEBits(adif->cm, 0, bits);
-
- return bits;
-}
-
diff --git a/libMpegTPEnc/src/tpenc_adif.h b/libMpegTPEnc/src/tpenc_adif.h
deleted file mode 100644
index d590354..0000000
--- a/libMpegTPEnc/src/tpenc_adif.h
+++ /dev/null
@@ -1,135 +0,0 @@
-
-/* -----------------------------------------------------------------------------------------------------------
-Software License for The Fraunhofer FDK AAC Codec Library for Android
-
-© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
- All rights reserved.
-
- 1. INTRODUCTION
-The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
-the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
-This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
-
-AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
-audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
-independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
-of the MPEG specifications.
-
-Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
-may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
-individually for the purpose of encoding or decoding bit streams in products that are compliant with
-the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
-these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
-software may already be covered under those patent licenses when it is used for those licensed purposes only.
-
-Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
-are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
-applications information and documentation.
-
-2. COPYRIGHT LICENSE
-
-Redistribution and use in source and binary forms, with or without modification, are permitted without
-payment of copyright license fees provided that you satisfy the following conditions:
-
-You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
-your modifications thereto in source code form.
-
-You must retain the complete text of this software license in the documentation and/or other materials
-provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
-You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
-modifications thereto to recipients of copies in binary form.
-
-The name of Fraunhofer may not be used to endorse or promote products derived from this library without
-prior written permission.
-
-You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
-software or your modifications thereto.
-
-Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
-and the date of any change. For modified versions of the FDK AAC Codec, the term
-"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
-"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
-
-3. NO PATENT LICENSE
-
-NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
-ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
-respect to this software.
-
-You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
-by appropriate patent licenses.
-
-4. DISCLAIMER
-
-This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
-"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
-of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
-CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
-including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
-or business interruption, however caused and on any theory of liability, whether in contract, strict
-liability, or tort (including negligence), arising in any way out of the use of this software, even if
-advised of the possibility of such damage.
-
-5. CONTACT INFORMATION
-
-Fraunhofer Institute for Integrated Circuits IIS
-Attention: Audio and Multimedia Departments - FDK AAC LL
-Am Wolfsmantel 33
-91058 Erlangen, Germany
-
-www.iis.fraunhofer.de/amm
-amm-info@iis.fraunhofer.de
------------------------------------------------------------------------------------------------------------ */
-
-/******************************** MPEG Audio Encoder **************************
-
- Initial author: Alex Goeschel
- contents/description: Transport Headers support
-
-******************************************************************************/
-
-#ifndef TPENC_ADIF_H
-#define TPENC_ADIF_H
-
-#include "machine_type.h"
-#include "FDK_bitstream.h"
-
-#include "tp_data.h"
-
-typedef struct {
- CHANNEL_MODE cm;
- INT samplingRate;
- INT bitRate;
- int profile;
- int bVariableRate;
- int instanceTag;
- int headerWritten;
-} ADIF_INFO;
-
-/**
- * \brief encodes ADIF Header
- *
- * \param adif pointer to ADIF_INFO structure
- * \param hBitStream handle of bitstream, where the ADIF header is written into
- * \param adif_buffer_fullness buffer fullness value for the ADIF header
- *
- * \return 0 on success
- */
-int adifWrite_EncodeHeader(
- ADIF_INFO *adif,
- HANDLE_FDK_BITSTREAM hBitStream,
- INT adif_buffer_fullness
- );
-
-/**
- * \brief Get bit demand of a ADIF header
- *
- * \param adif pointer to ADIF_INFO structure
- *
- * \return amount of bits required to write the ADIF header according to the data
- * contained in the adif parameter
- */
-int adifWrite_GetHeaderBits( ADIF_INFO *adif );
-
-#endif /* TPENC_ADIF_H */
-
diff --git a/libMpegTPEnc/src/tpenc_adts.cpp b/libMpegTPEnc/src/tpenc_adts.cpp
deleted file mode 100644
index f4f3178..0000000
--- a/libMpegTPEnc/src/tpenc_adts.cpp
+++ /dev/null
@@ -1,315 +0,0 @@
-
-/* -----------------------------------------------------------------------------------------------------------
-Software License for The Fraunhofer FDK AAC Codec Library for Android
-
-© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
- All rights reserved.
-
- 1. INTRODUCTION
-The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
-the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
-This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
-
-AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
-audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
-independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
-of the MPEG specifications.
-
-Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
-may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
-individually for the purpose of encoding or decoding bit streams in products that are compliant with
-the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
-these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
-software may already be covered under those patent licenses when it is used for those licensed purposes only.
-
-Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
-are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
-applications information and documentation.
-
-2. COPYRIGHT LICENSE
-
-Redistribution and use in source and binary forms, with or without modification, are permitted without
-payment of copyright license fees provided that you satisfy the following conditions:
-
-You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
-your modifications thereto in source code form.
-
-You must retain the complete text of this software license in the documentation and/or other materials
-provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
-You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
-modifications thereto to recipients of copies in binary form.
-
-The name of Fraunhofer may not be used to endorse or promote products derived from this library without
-prior written permission.
-
-You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
-software or your modifications thereto.
-
-Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
-and the date of any change. For modified versions of the FDK AAC Codec, the term
-"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
-"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
-
-3. NO PATENT LICENSE
-
-NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
-ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
-respect to this software.
-
-You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
-by appropriate patent licenses.
-
-4. DISCLAIMER
-
-This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
-"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
-of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
-CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
-including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
-or business interruption, however caused and on any theory of liability, whether in contract, strict
-liability, or tort (including negligence), arising in any way out of the use of this software, even if
-advised of the possibility of such damage.
-
-5. CONTACT INFORMATION
-
-Fraunhofer Institute for Integrated Circuits IIS
-Attention: Audio and Multimedia Departments - FDK AAC LL
-Am Wolfsmantel 33
-91058 Erlangen, Germany
-
-www.iis.fraunhofer.de/amm
-amm-info@iis.fraunhofer.de
------------------------------------------------------------------------------------------------------------ */
-
-/******************************** MPEG Audio Encoder **************************
-
- Initial author: Alex Groeschel
- contents/description: ADTS Transport Headers support
-
-******************************************************************************/
-
-#include "tpenc_adts.h"
-
-
-#include "tpenc_lib.h"
-#include "tpenc_asc.h"
-
-
-int adtsWrite_CrcStartReg(
- HANDLE_ADTS pAdts, /*!< pointer to adts stucture */
- HANDLE_FDK_BITSTREAM hBs, /*!< handle to current bit buffer structure */
- int mBits /*!< number of bits in crc region */
- )
-{
- if (pAdts->protection_absent) {
- return 0;
- }
- return ( FDKcrcStartReg(&pAdts->crcInfo, hBs, mBits) );
-}
-
-void adtsWrite_CrcEndReg(
- HANDLE_ADTS pAdts, /*!< pointer to adts crc info stucture */
- HANDLE_FDK_BITSTREAM hBs, /*!< handle to current bit buffer structure */
- int reg /*!< crc region */
- )
-{
- if (pAdts->protection_absent == 0)
- {
- FDKcrcEndReg(&pAdts->crcInfo, hBs, reg);
- }
-}
-
-int adtsWrite_GetHeaderBits( HANDLE_ADTS hAdts )
-{
- int bits = 0;
-
- if (hAdts->currentBlock == 0) {
- /* Static and variable header bits */
- bits = 56;
- if (!hAdts->protection_absent) {
- /* Add header/ single raw data block CRC bits */
- bits += 16;
- if (hAdts->num_raw_blocks>0) {
- /* Add bits of raw data block position markers */
- bits += (hAdts->num_raw_blocks)*16;
- }
- }
- }
- if (!hAdts->protection_absent && hAdts->num_raw_blocks>0) {
- /* Add raw data block CRC bits. Not really part of the header, put they cause bit overhead to be accounted. */
- bits += 16;
- }
-
- hAdts->headerBits = bits;
-
- return bits;
-}
-
-INT adtsWrite_Init(HANDLE_ADTS hAdts, CODER_CONFIG *config)
-{
- /* Sanity checks */
- if ( config->nSubFrames < 1
- || config->nSubFrames > 4
- || (int)config->aot > 4
- || (int)config->aot < 1 ) {
- return -1;
- }
-
- /* fixed header */
- if (config->flags & CC_MPEG_ID) {
- hAdts->mpeg_id = 0; /* MPEG 4 */
- } else {
- hAdts->mpeg_id = 1; /* MPEG 2 */
- }
- hAdts->layer=0;
- hAdts->protection_absent = ! (config->flags & CC_PROTECTION);
- hAdts->profile = ((int)config->aot) - 1;
- hAdts->sample_freq_index = getSamplingRateIndex(config->samplingRate);
- hAdts->sample_freq = config->samplingRate;
- hAdts->private_bit=0;
- hAdts->channel_mode = config->channelMode;
- hAdts->original=0;
- hAdts->home=0;
- /* variable header */
- hAdts->copyright_id=0;
- hAdts->copyright_start=0;
-
- hAdts->num_raw_blocks=config->nSubFrames-1; /* 0 means 1 raw data block */
-
- FDKcrcInit(&hAdts->crcInfo, 0x8005, 0xFFFF, 16);
-
- hAdts->currentBlock = 0;
-
-
- return 0;
-}
-
-int adtsWrite_EncodeHeader(HANDLE_ADTS hAdts,
- HANDLE_FDK_BITSTREAM hBitStream,
- int buffer_fullness,
- int frame_length)
-{
- INT crcIndex = 0;
-
-
- hAdts->headerBits = adtsWrite_GetHeaderBits(hAdts);
-
- FDK_ASSERT(((frame_length+hAdts->headerBits)/8)<0x2000); /*13 bit*/
- FDK_ASSERT(buffer_fullness<0x800); /* 11 bit */
-
- if (!hAdts->protection_absent) {
- FDKcrcReset(&hAdts->crcInfo);
- }
-
- if (hAdts->currentBlock == 0) {
- FDKresetBitbuffer(hBitStream, BS_WRITER);
- }
-
- hAdts->subFrameStartBit = FDKgetValidBits(hBitStream);
-
- /* Skip new header if this is raw data block 1..n */
- if (hAdts->currentBlock == 0)
- {
- FDKresetBitbuffer(hBitStream, BS_WRITER);
-
- if (hAdts->num_raw_blocks == 0) {
- crcIndex = adtsWrite_CrcStartReg(hAdts, hBitStream, 0);
- }
-
- /* fixed header */
- FDKwriteBits(hBitStream, 0xFFF, 12);
- FDKwriteBits(hBitStream, hAdts->mpeg_id, 1);
- FDKwriteBits(hBitStream, hAdts->layer, 2);
- FDKwriteBits(hBitStream, hAdts->protection_absent, 1);
- FDKwriteBits(hBitStream, hAdts->profile, 2);
- FDKwriteBits(hBitStream, hAdts->sample_freq_index, 4);
- FDKwriteBits(hBitStream, hAdts->private_bit, 1);
- FDKwriteBits(hBitStream, getChannelConfig(hAdts->channel_mode), 3);
- FDKwriteBits(hBitStream, hAdts->original, 1);
- FDKwriteBits(hBitStream, hAdts->home, 1);
- /* variable header */
- FDKwriteBits(hBitStream, hAdts->copyright_id, 1);
- FDKwriteBits(hBitStream, hAdts->copyright_start, 1);
- FDKwriteBits(hBitStream, (frame_length + hAdts->headerBits)>>3, 13);
- FDKwriteBits(hBitStream, buffer_fullness, 11);
- FDKwriteBits(hBitStream, hAdts->num_raw_blocks, 2);
-
- if (!hAdts->protection_absent) {
- int i;
-
- /* End header CRC portion for single raw data block and write dummy zero values for unknown fields. */
- if (hAdts->num_raw_blocks == 0) {
- adtsWrite_CrcEndReg(hAdts, hBitStream, crcIndex);
- } else {
- for (i=0; i<hAdts->num_raw_blocks; i++) {
- FDKwriteBits(hBitStream, 0, 16);
- }
- }
- FDKwriteBits(hBitStream, 0, 16);
- }
- } /* End of ADTS header */
-
- return 0;
-}
-
-void adtsWrite_EndRawDataBlock(HANDLE_ADTS hAdts,
- HANDLE_FDK_BITSTREAM hBs,
- int *pBits)
-{
- if (!hAdts->protection_absent) {
- FDK_BITSTREAM bsWriter;
-
- FDKinitBitStream(&bsWriter, hBs->hBitBuf.Buffer, hBs->hBitBuf.bufSize, 0, BS_WRITER);
- FDKpushFor(&bsWriter, 56);
-
- if (hAdts->num_raw_blocks == 0) {
- FDKwriteBits(&bsWriter, FDKcrcGetCRC(&hAdts->crcInfo), 16);
- } else {
- int distance;
-
- /* Write CRC of current raw data block */
- FDKwriteBits(hBs, FDKcrcGetCRC(&hAdts->crcInfo), 16);
-
- /* Write distance to current data block */
- if (hAdts->currentBlock < hAdts->num_raw_blocks) {
- FDKpushFor(&bsWriter, hAdts->currentBlock*16);
- distance = FDKgetValidBits(hBs) - (56 + (hAdts->num_raw_blocks)*16 + 16);
- FDKwriteBits(&bsWriter, distance>>3, 16);
- }
- }
- FDKsyncCache(&bsWriter);
- }
-
- /* Write total frame lenth for multiple raw data blocks and header CRC */
- if (hAdts->num_raw_blocks > 0 && hAdts->currentBlock == hAdts->num_raw_blocks) {
- FDK_BITSTREAM bsWriter;
- int crcIndex = 0;
-
- FDKinitBitStream(&bsWriter, hBs->hBitBuf.Buffer, hBs->hBitBuf.bufSize, 0, BS_WRITER);
-
- if (!hAdts->protection_absent) {
- FDKcrcReset(&hAdts->crcInfo);
- crcIndex = FDKcrcStartReg(&hAdts->crcInfo, &bsWriter, 0);
- }
- /* Write total frame length */
- FDKpushFor(&bsWriter, 56-28+2);
- FDKwriteBits(&bsWriter, FDKgetValidBits(hBs)>>3, 13);
-
- /* Write header CRC */
- if (!hAdts->protection_absent) {
- FDKpushFor(&bsWriter, 11+2 + (hAdts->num_raw_blocks)*16);
- FDKcrcEndReg(&hAdts->crcInfo, &bsWriter, crcIndex);
- FDKwriteBits(&bsWriter, FDKcrcGetCRC(&hAdts->crcInfo), 16);
- }
- FDKsyncCache(&bsWriter);
- }
-
- /* Correct *pBits to reflect the amount of bits of the current subframe */
- *pBits -= hAdts->subFrameStartBit;
- if (!hAdts->protection_absent && hAdts->num_raw_blocks > 0) {
- /* Fixup CRC bits, since they come after each raw data block */
- *pBits += 16;
- }
- hAdts->currentBlock++;
-}
-
diff --git a/libMpegTPEnc/src/tpenc_adts.h b/libMpegTPEnc/src/tpenc_adts.h
deleted file mode 100644
index c12c7c7..0000000
--- a/libMpegTPEnc/src/tpenc_adts.h
+++ /dev/null
@@ -1,218 +0,0 @@
-
-/* -----------------------------------------------------------------------------------------------------------
-Software License for The Fraunhofer FDK AAC Codec Library for Android
-
-© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
- All rights reserved.
-
- 1. INTRODUCTION
-The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
-the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
-This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
-
-AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
-audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
-independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
-of the MPEG specifications.
-
-Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
-may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
-individually for the purpose of encoding or decoding bit streams in products that are compliant with
-the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
-these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
-software may already be covered under those patent licenses when it is used for those licensed purposes only.
-
-Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
-are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
-applications information and documentation.
-
-2. COPYRIGHT LICENSE
-
-Redistribution and use in source and binary forms, with or without modification, are permitted without
-payment of copyright license fees provided that you satisfy the following conditions:
-
-You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
-your modifications thereto in source code form.
-
-You must retain the complete text of this software license in the documentation and/or other materials
-provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
-You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
-modifications thereto to recipients of copies in binary form.
-
-The name of Fraunhofer may not be used to endorse or promote products derived from this library without
-prior written permission.
-
-You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
-software or your modifications thereto.
-
-Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
-and the date of any change. For modified versions of the FDK AAC Codec, the term
-"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
-"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
-
-3. NO PATENT LICENSE
-
-NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
-ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
-respect to this software.
-
-You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
-by appropriate patent licenses.
-
-4. DISCLAIMER
-
-This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
-"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
-of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
-CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
-including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
-or business interruption, however caused and on any theory of liability, whether in contract, strict
-liability, or tort (including negligence), arising in any way out of the use of this software, even if
-advised of the possibility of such damage.
-
-5. CONTACT INFORMATION
-
-Fraunhofer Institute for Integrated Circuits IIS
-Attention: Audio and Multimedia Departments - FDK AAC LL
-Am Wolfsmantel 33
-91058 Erlangen, Germany
-
-www.iis.fraunhofer.de/amm
-amm-info@iis.fraunhofer.de
------------------------------------------------------------------------------------------------------------ */
-
-/******************************** MPEG Audio Encoder **************************
-
- Initial author: Alex Groeschel
- contents/description: ADTS Transport writer
-
-******************************************************************************/
-
-#ifndef TPENC_ADTS_H
-#define TPENC_ADTS_H
-
-
-
-#include "tp_data.h"
-
-#include "FDK_crc.h"
-
-typedef struct {
- INT sample_freq;
- CHANNEL_MODE channel_mode;
- UCHAR decoderCanDoMpeg4;
- UCHAR mpeg_id;
- UCHAR layer;
- UCHAR protection_absent;
- UCHAR profile;
- UCHAR sample_freq_index;
- UCHAR private_bit;
- UCHAR original;
- UCHAR home;
- UCHAR copyright_id;
- UCHAR copyright_start;
- USHORT frame_length;
- UCHAR num_raw_blocks;
- UCHAR BufferFullnesStartFlag;
- int headerBits; /*!< Header bit demand for the current raw data block */
- int currentBlock; /*!< Index of current raw data block */
- int subFrameStartBit; /*!< Bit position where the current raw data block begins */
- FDK_CRCINFO crcInfo;
-} STRUCT_ADTS;
-
-typedef STRUCT_ADTS *HANDLE_ADTS;
-
-/**
- * \brief Initialize ADTS data structure
- *
- * \param hAdts ADTS data handle
- * \param config a valid CODER_CONFIG struct from where the required
- * information for the ADTS header is extrated from
- *
- * \return 0 in case of success.
- */
-INT adtsWrite_Init(
- HANDLE_ADTS hAdts,
- CODER_CONFIG *config
- );
-
-/**
- * \brief Get the total bit overhead caused by ADTS
- *
- * \hAdts handle to ADTS data
- *
- * \return Amount of additional bits required for the current raw data block
- */
-int adtsWrite_GetHeaderBits( HANDLE_ADTS hAdts );
-
-/**
- * \brief Write an ADTS header into the given bitstream. May not write a header
- * in case of multiple raw data blocks.
- *
- * \param hAdts ADTS data handle
- * \param hBitStream bitstream handle into which the ADTS may be written into
- * \param buffer_fullness the buffer fullness value for the ADTS header
- * \param the current raw data block length
- *
- * \return 0 in case of success.
- */
-INT adtsWrite_EncodeHeader(
- HANDLE_ADTS hAdts,
- HANDLE_FDK_BITSTREAM hBitStream,
- int bufferFullness,
- int frame_length
- );
-/**
- * \brief Finish a ADTS raw data block
- *
- * \param hAdts ADTS data handle
- * \param hBs bitstream handle into which the ADTS may be written into
- * \param pBits a pointer to a integer holding the current bitstream buffer bit count,
- * which is corrected to the current raw data block boundary.
- *
- */
-void adtsWrite_EndRawDataBlock(
- HANDLE_ADTS hAdts,
- HANDLE_FDK_BITSTREAM hBs,
- int *bits
- );
-
-
-/**
- * \brief Start CRC region with a maximum number of bits
- * If mBits is positive zero padding will be used for CRC calculation, if there
- * are less than mBits bits available.
- * If mBits is negative no zero padding is done.
- * If mBits is zero the memory for the buffer is allocated dynamically, the
- * number of bits is not limited.
- *
- * \param pAdts ADTS data handle
- * \param hBs bitstream handle of which the CRC region ends
- * \param mBits limit of number of bits to be considered for the requested CRC region
- *
- * \return ID for the created region, -1 in case of an error
- */
-int adtsWrite_CrcStartReg(
- HANDLE_ADTS pAdts,
- HANDLE_FDK_BITSTREAM hBs,
- int mBits
- );
-
-/**
- * \brief Ends CRC region identified by reg
- *
- * \param pAdts ADTS data handle
- * \param hBs bitstream handle of which the CRC region ends
- * \param reg a CRC region ID returned previously by adtsWrite_CrcStartReg()
- */
-void adtsWrite_CrcEndReg(
- HANDLE_ADTS pAdts,
- HANDLE_FDK_BITSTREAM hBs,
- int reg
- );
-
-
-
-
-#endif /* TPENC_ADTS_H */
-
diff --git a/libMpegTPEnc/src/tpenc_asc.cpp b/libMpegTPEnc/src/tpenc_asc.cpp
deleted file mode 100644
index bc4302e..0000000
--- a/libMpegTPEnc/src/tpenc_asc.cpp
+++ /dev/null
@@ -1,576 +0,0 @@
-
-/* -----------------------------------------------------------------------------------------------------------
-Software License for The Fraunhofer FDK AAC Codec Library for Android
-
-© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
- All rights reserved.
-
- 1. INTRODUCTION
-The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
-the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
-This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
-
-AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
-audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
-independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
-of the MPEG specifications.
-
-Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
-may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
-individually for the purpose of encoding or decoding bit streams in products that are compliant with
-the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
-these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
-software may already be covered under those patent licenses when it is used for those licensed purposes only.
-
-Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
-are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
-applications information and documentation.
-
-2. COPYRIGHT LICENSE
-
-Redistribution and use in source and binary forms, with or without modification, are permitted without
-payment of copyright license fees provided that you satisfy the following conditions:
-
-You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
-your modifications thereto in source code form.
-
-You must retain the complete text of this software license in the documentation and/or other materials
-provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
-You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
-modifications thereto to recipients of copies in binary form.
-
-The name of Fraunhofer may not be used to endorse or promote products derived from this library without
-prior written permission.
-
-You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
-software or your modifications thereto.
-
-Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
-and the date of any change. For modified versions of the FDK AAC Codec, the term
-"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
-"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
-
-3. NO PATENT LICENSE
-
-NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
-ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
-respect to this software.
-
-You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
-by appropriate patent licenses.
-
-4. DISCLAIMER
-
-This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
-"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
-of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
-CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
-including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
-or business interruption, however caused and on any theory of liability, whether in contract, strict
-liability, or tort (including negligence), arising in any way out of the use of this software, even if
-advised of the possibility of such damage.
-
-5. CONTACT INFORMATION
-
-Fraunhofer Institute for Integrated Circuits IIS
-Attention: Audio and Multimedia Departments - FDK AAC LL
-Am Wolfsmantel 33
-91058 Erlangen, Germany
-
-www.iis.fraunhofer.de/amm
-amm-info@iis.fraunhofer.de
------------------------------------------------------------------------------------------------------------ */
-
-/***************************** MPEG-4 AAC Encoder **************************
-
- Author(s):
- Description:
-
-******************************************************************************/
-
-#include "tp_data.h"
-
-#include "tpenc_lib.h"
-#include "tpenc_asc.h"
-#include "FDK_bitstream.h"
-#include "genericStds.h"
-
-#define PCE_MAX_ELEMENTS 8
-
-/**
- * Describe a PCE based on placed channel elements and element type sequence.
- */
-typedef struct {
-
- UCHAR num_front_channel_elements; /*!< Number of front channel elements. */
- UCHAR num_side_channel_elements; /*!< Number of side channel elements. */
- UCHAR num_back_channel_elements; /*!< Number of back channel elements. */
- UCHAR num_lfe_channel_elements; /*!< Number of lfe channel elements. */
- MP4_ELEMENT_ID el_list[PCE_MAX_ELEMENTS];/*!< List contains sequence describing the elements
- in present channel mode. (MPEG order) */
-} PCE_CONFIGURATION;
-
-
-/**
- * Map an incoming channel mode to a existing PCE configuration entry.
- */
-typedef struct {
-
- CHANNEL_MODE channel_mode; /*!< Present channel mode. */
- PCE_CONFIGURATION pce_configuration; /*!< Program config element description. */
-
-} CHANNEL_CONFIGURATION;
-
-
-/**
- * \brief Table contains all supported channel modes and according PCE configuration description.
- *
- * The number of channel element parameter describes the kind of consecutively elements.
- * E.g. MODE_1_2_2_2_1 means:
- * - First 3 elements (SCE,CPE,CPE) are front channel elements.
- * - Next element (CPE) is a back channel element.
- * - Last element (LFE) is a lfe channel element.
- */
-static const CHANNEL_CONFIGURATION pceConfigTab[] =
-{
- { MODE_1, { 1, 0, 0, 0, { ID_SCE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE } } },
- { MODE_2, { 1, 0, 0, 0, { ID_CPE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE } } },
- { MODE_1_2, { 2, 0, 0, 0, { ID_SCE, ID_CPE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE } } },
- { MODE_1_2_1, { 2, 0, 1, 0, { ID_SCE, ID_CPE, ID_SCE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE } } },
- { MODE_1_2_2, { 2, 0, 1, 0, { ID_SCE, ID_CPE, ID_CPE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE } } },
- { MODE_1_2_2_1, { 2, 0, 1, 1, { ID_SCE, ID_CPE, ID_CPE, ID_LFE, ID_NONE, ID_NONE, ID_NONE, ID_NONE } } },
- { MODE_1_2_2_2_1, { 3, 0, 1, 1, { ID_SCE, ID_CPE, ID_CPE, ID_CPE, ID_LFE, ID_NONE, ID_NONE, ID_NONE } } },
-
-
- { MODE_1_1, { 2, 0, 0, 0, { ID_SCE, ID_SCE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE } } },
- { MODE_1_1_1_1, { 2, 2, 0, 0, { ID_SCE, ID_SCE, ID_SCE, ID_SCE, ID_NONE, ID_NONE, ID_NONE, ID_NONE } } },
- { MODE_1_1_1_1_1_1, { 2, 2, 2, 0, { ID_SCE, ID_SCE, ID_SCE, ID_SCE, ID_SCE, ID_SCE, ID_NONE, ID_NONE } } },
- { MODE_1_1_1_1_1_1_1_1, { 3, 2, 3, 0, { ID_SCE, ID_SCE, ID_SCE, ID_SCE, ID_SCE, ID_SCE, ID_SCE, ID_SCE } } },
-
- { MODE_2_2, { 1, 0, 1, 0, { ID_CPE, ID_CPE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE } } },
- { MODE_2_2_2, { 1, 1, 1, 0, { ID_CPE, ID_CPE, ID_CPE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE } } },
- { MODE_2_2_2_2, { 4, 0, 0, 0, { ID_CPE, ID_CPE, ID_CPE, ID_CPE, ID_NONE, ID_NONE, ID_NONE, ID_NONE } } },
-
- { MODE_2_1, { 1, 0, 1, 0, { ID_CPE, ID_SCE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE } } },
-
- { MODE_7_1_REAR_SURROUND, { 2, 0, 2, 1, { ID_SCE, ID_CPE, ID_CPE, ID_CPE, ID_LFE, ID_NONE, ID_NONE, ID_NONE } } },
- { MODE_7_1_FRONT_CENTER, { 3, 0, 1, 1, { ID_SCE, ID_CPE, ID_CPE, ID_CPE, ID_LFE, ID_NONE, ID_NONE, ID_NONE } } },
-
-};
-
-
-/**
- * \brief Get program config element description for existing channel mode.
- *
- * \param channel_mode Current channel mode.
- *
- * \return
- * - Pointer to PCE_CONFIGURATION entry, on success.
- * - NULL, on failure.
- */
-static const PCE_CONFIGURATION* getPceEntry(
- const CHANNEL_MODE channel_mode
- )
-{
- UINT i;
- const PCE_CONFIGURATION *pce_config = NULL;
-
- for (i=0; i < (sizeof(pceConfigTab)/sizeof(CHANNEL_CONFIGURATION)); i++) {
- if (pceConfigTab[i].channel_mode == channel_mode) {
- pce_config = &pceConfigTab[i].pce_configuration;
- }
- }
-
- return pce_config;
-}
-
-int getChannelConfig( CHANNEL_MODE channel_mode )
-{
- INT chan_config = 0;
-
- switch(channel_mode) {
- case MODE_1: chan_config = 1; break;
- case MODE_2: chan_config = 2; break;
- case MODE_1_2: chan_config = 3; break;
- case MODE_1_2_1: chan_config = 4; break;
- case MODE_1_2_2: chan_config = 5; break;
- case MODE_1_2_2_1: chan_config = 6; break;
- case MODE_1_2_2_2_1: chan_config = 7; break;
-
- default: chan_config = 0;
- }
-
- return chan_config;
-}
-
-CHANNEL_MODE transportEnc_GetChannelMode( int noChannels )
-{
- CHANNEL_MODE chMode;
-
- if (noChannels <= 8 && noChannels > 0)
- chMode = (CHANNEL_MODE)((noChannels == 8) ? 7 : noChannels); /* see : iso/mpeg4 v1 audio subpart1*/
- else
- chMode = MODE_UNKNOWN;
-
- return chMode;
-}
-
-#ifdef TP_PCE_ENABLE
-int transportEnc_writePCE(HANDLE_FDK_BITSTREAM hBs,
- CHANNEL_MODE channelMode,
- INT sampleRate,
- int instanceTagPCE,
- int profile,
- int matrixMixdownA,
- int pseudoSurroundEnable,
- UINT alignAnchor)
-{
- int sampleRateIndex, i;
- const PCE_CONFIGURATION* config = NULL;
- const MP4_ELEMENT_ID* pEl_list = NULL;
- UCHAR cpeCnt=0, sceCnt=0, lfeCnt=0;
-
- sampleRateIndex = getSamplingRateIndex(sampleRate);
- if (sampleRateIndex == 15) {
- return -1;
- }
-
- if ((config=getPceEntry(channelMode))==NULL) {
- return -1;
- }
-
- /* Pointer to first element in element list. */
- pEl_list = &config->el_list[0];
-
- FDKwriteBits(hBs, instanceTagPCE, 4); /* Element instance tag */
- FDKwriteBits(hBs, profile, 2); /* Object type */
- FDKwriteBits(hBs, sampleRateIndex, 4); /* Sample rate index*/
-
- FDKwriteBits(hBs, config->num_front_channel_elements, 4); /* Front channel Elements */
- FDKwriteBits(hBs, config->num_side_channel_elements , 4); /* No Side Channel Elements */
- FDKwriteBits(hBs, config->num_back_channel_elements , 4); /* No Back channel Elements */
- FDKwriteBits(hBs, config->num_lfe_channel_elements , 2); /* No Lfe channel elements */
-
- FDKwriteBits(hBs, 0, 3); /* No assoc data elements */
- FDKwriteBits(hBs, 0, 4); /* No valid cc elements */
- FDKwriteBits(hBs, 0, 1); /* Mono mixdown present */
- FDKwriteBits(hBs, 0, 1); /* Stereo mixdown present */
-
- if ( matrixMixdownA!=0 && ((channelMode==MODE_1_2_2)||(channelMode==MODE_1_2_2_1)) ) {
- FDKwriteBits(hBs, 1, 1); /* Matrix mixdown present */
- FDKwriteBits(hBs, (matrixMixdownA-1)&0x3, 2); /* matrix_mixdown_idx */
- FDKwriteBits(hBs, (pseudoSurroundEnable)?1:0, 1); /* pseudo_surround_enable */
- }
- else {
- FDKwriteBits(hBs, 0, 1); /* Matrix mixdown not present */
- }
-
- for(i=0; i<config->num_front_channel_elements; i++) {
- UCHAR isCpe = (*pEl_list++==ID_CPE) ? 1 : 0;
- UCHAR tag = (isCpe) ? cpeCnt++ : sceCnt++;
- FDKwriteBits(hBs, isCpe, 1); /* Front channel Elements is CPE? */
- FDKwriteBits(hBs, tag, 4); /* Front channel Instance Tag.*/
- }
- for(i=0; i<config->num_side_channel_elements; i++) {
- UCHAR isCpe = (*pEl_list++==ID_CPE) ? 1 : 0;
- UCHAR tag = (isCpe) ? cpeCnt++ : sceCnt++;
- FDKwriteBits(hBs, isCpe, 1); /* Front channel Elements is CPE? */
- FDKwriteBits(hBs, tag, 4); /* Front channel Instance Tag.*/
- }
- for(i=0; i<config->num_back_channel_elements; i++) {
- UCHAR isCpe = (*pEl_list++==ID_CPE) ? 1 : 0;
- UCHAR tag = (isCpe) ? cpeCnt++ : sceCnt++;
- FDKwriteBits(hBs, isCpe, 1); /* Front channel Elements is CPE? */
- FDKwriteBits(hBs, tag, 4); /* Front channel Instance Tag.*/
- }
- for(i=0; i<config->num_lfe_channel_elements; i++) {
- FDKwriteBits(hBs, lfeCnt++, 4); /* LFE channel Instance Tag. */
- }
-
- /* - num_valid_cc_elements always 0.
- - num_assoc_data_elements always 0. */
-
- /* Byte alignment: relative to alignAnchor
- ADTS: align with respect to the first bit of the raw_data_block()
- ADIF: align with respect to the first bit of the header
- LATM: align with respect to the first bit of the ASC */
- FDKbyteAlign(hBs, alignAnchor); /* Alignment */
-
- FDKwriteBits(hBs, 0 ,8); /* Do no write any comment. */
-
- /* - comment_field_bytes always 0. */
-
- return 0;
-}
-
-int transportEnc_GetPCEBits(CHANNEL_MODE channelMode,
- int matrixMixdownA,
- int bits)
-{
- const PCE_CONFIGURATION* config = NULL;
-
- if ((config=getPceEntry(channelMode))==NULL) {
- return -1; /* unsupported channelmapping */
- }
-
- bits += 4 + 2 + 4; /* Element instance tag + Object type + Sample rate index */
- bits += 4 + 4 + 4 + 2; /* No (front + side + back + lfe channel) elements */
- bits += 3 + 4; /* No (assoc data + valid cc) elements */
- bits += 1 + 1 + 1 ; /* Mono + Stereo + Matrix mixdown present */
-
- if ( matrixMixdownA!=0 && ((channelMode==MODE_1_2_2)||(channelMode==MODE_1_2_2_1)) ) {
- bits +=3; /* matrix_mixdown_idx + pseudo_surround_enable */
- }
-
- bits += (1+4) * (INT)config->num_front_channel_elements;
- bits += (1+4) * (INT)config->num_side_channel_elements;
- bits += (1+4) * (INT)config->num_back_channel_elements;
- bits += (4) * (INT)config->num_lfe_channel_elements;
-
- /* - num_valid_cc_elements always 0.
- - num_assoc_data_elements always 0. */
-
- if ((bits%8) != 0) {
- bits += (8 - (bits%8)); /* Alignment */
- }
-
- bits += 8; /* Comment field bytes */
-
- /* - comment_field_bytes alwys 0. */
-
- return bits;
-}
-#endif /* TP_PCE_ENABLE */
-
-static void writeAot(HANDLE_FDK_BITSTREAM hBitstreamBuffer, AUDIO_OBJECT_TYPE aot)
-{
- int tmp = (int) aot;
-
- if (tmp > 31) {
- FDKwriteBits( hBitstreamBuffer, AOT_ESCAPE, 5 );
- FDKwriteBits( hBitstreamBuffer, tmp-32, 6 ); /* AudioObjectType */
- } else {
- FDKwriteBits( hBitstreamBuffer, tmp, 5 );
- }
-}
-
-static void writeSampleRate(HANDLE_FDK_BITSTREAM hBitstreamBuffer, int sampleRate)
-{
- int sampleRateIndex = getSamplingRateIndex(sampleRate);
-
- FDKwriteBits( hBitstreamBuffer, sampleRateIndex, 4 );
- if( sampleRateIndex == 15 ) {
- FDKwriteBits( hBitstreamBuffer, sampleRate, 24 );
- }
-}
-
-#ifdef TP_GA_ENABLE
-static
-int transportEnc_writeGASpecificConfig(
- HANDLE_FDK_BITSTREAM asc,
- CODER_CONFIG *config,
- int extFlg,
- UINT alignAnchor
- )
-{
- int aot = config->aot;
- int samplesPerFrame = config->samplesPerFrame;
-
- /* start of GASpecificConfig according to ISO/IEC 14496-3 Subpart 4, 4.4.1 */
- FDKwriteBits( asc, ((samplesPerFrame==960 || samplesPerFrame==480)?1:0), 1); /* frameLengthFlag: 1 for a 960/480 (I)MDCT, 0 for a 1024/512 (I)MDCT*/
- FDKwriteBits( asc, 0, 1); /* dependsOnCoreCoder: Sampling Rate Coder Specific, see in ISO/IEC 14496-3 Subpart 4, 4.4.1 */
- FDKwriteBits( asc, extFlg, 1 ); /* Extension Flag: Shall be 1 for aot = 17,19,20,21,22,23 */
-
- /* Write PCE if channel config is not 1-7 */
- if (getChannelConfig(config->channelMode) == 0) {
- transportEnc_writePCE(asc, config->channelMode, config->samplingRate, 0, 1, config->matrixMixdownA, (config->flags&CC_PSEUDO_SURROUND)?1:0, alignAnchor);
- }
- if (extFlg) {
- if (aot == AOT_ER_BSAC) {
- FDKwriteBits( asc, config->BSACnumOfSubFrame, 5 ); /* numOfSubFrame */
- FDKwriteBits( asc, config->BSAClayerLength, 11 ); /* layer_length */
- }
- if ((aot == AOT_ER_AAC_LC) || (aot == AOT_ER_AAC_LTP) ||
- (aot == AOT_ER_AAC_SCAL) || (aot == AOT_ER_AAC_LD))
- {
- FDKwriteBits( asc, (config->flags & CC_VCB11) ? 1 : 0, 1 ); /* aacSectionDataResillienceFlag */
- FDKwriteBits( asc, (config->flags & CC_RVLC) ? 1 : 0, 1 ); /* aacScaleFactorDataResillienceFlag */
- FDKwriteBits( asc, (config->flags & CC_HCR) ? 1 : 0, 1 ); /* aacSpectralDataResillienceFlag */
- }
- FDKwriteBits( asc, 0, 1 ); /* extensionFlag3: reserved. Shall be '0' */
- }
- return 0;
-}
-#endif /* TP_GA_ENABLE */
-
-#ifdef TP_ELD_ENABLE
-
-static
-int transportEnc_writeELDSpecificConfig(
- HANDLE_FDK_BITSTREAM hBs,
- CODER_CONFIG *config,
- int epConfig,
- CSTpCallBacks *cb
- )
-{
- /* ELD specific config */
- if (config->channelMode == MODE_1_1) {
- return -1;
- }
- FDKwriteBits(hBs, (config->samplesPerFrame == 480) ? 1 : 0, 1);
-
- FDKwriteBits(hBs, (config->flags & CC_VCB11 ) ? 1:0, 1);
- FDKwriteBits(hBs, (config->flags & CC_RVLC ) ? 1:0, 1);
- FDKwriteBits(hBs, (config->flags & CC_HCR ) ? 1:0, 1);
-
- FDKwriteBits(hBs, (config->flags & CC_SBR) ? 1:0, 1); /* SBR header flag */
- if ( (config->flags & CC_SBR) ) {
- FDKwriteBits(hBs, (config->samplingRate == config->extSamplingRate) ? 0:1, 1); /* Samplerate Flag */
- FDKwriteBits(hBs, (config->flags & CC_SBRCRC) ? 1:0, 1); /* SBR CRC flag*/
-
- if (cb->cbSbr != NULL) {
- const PCE_CONFIGURATION *pPce;
- int e;
-
- pPce = getPceEntry(config->channelMode);
-
- for (e=0; e<PCE_MAX_ELEMENTS && pPce->el_list[e] != ID_NONE; e++ ) {
- if ( (pPce->el_list[e] == ID_SCE) || (pPce->el_list[e] == ID_CPE) ) {
- cb->cbSbr(cb->cbSbrData, hBs, 0, 0, 0, config->aot, pPce->el_list[e], e);
- }
- }
- }
- }
-
- FDKwriteBits(hBs, 0, 4); /* ELDEXT_TERM */
-
- return 0;
-}
-#endif /* TP_ELD_ENABLE */
-
-
-int transportEnc_writeASC (
- HANDLE_FDK_BITSTREAM asc,
- CODER_CONFIG *config,
- CSTpCallBacks *cb
- )
-{
- UINT extFlag = 0;
- int err;
- int epConfig = 0;
-
- /* Required for the PCE. */
- UINT alignAnchor = FDKgetValidBits(asc);
-
- /* Extension Flag: Shall be 1 for aot = 17,19,20,21,22,23,39 */
- switch (config->aot) {
- case AOT_ER_AAC_LC:
- case AOT_ER_AAC_LTP:
- case AOT_ER_AAC_SCAL:
- case AOT_ER_TWIN_VQ:
- case AOT_ER_BSAC:
- case AOT_ER_AAC_LD:
- case AOT_ER_AAC_ELD:
- case AOT_USAC:
- extFlag = 1;
- break;
- default:
- break;
- }
-
- if (config->sbrSignaling==SIG_EXPLICIT_HIERARCHICAL && config->sbrPresent)
- writeAot(asc, config->extAOT);
- else
- writeAot(asc, config->aot);
-
- {
- writeSampleRate(asc, config->samplingRate);
- }
-
- /* Try to guess a reasonable channel mode if not given */
- if (config->channelMode == MODE_INVALID) {
- config->channelMode = transportEnc_GetChannelMode(config->noChannels);
- if (config->channelMode == MODE_INVALID)
- return -1;
- }
-
- FDKwriteBits( asc, getChannelConfig(config->channelMode), 4 );
-
- if (config->sbrSignaling==SIG_EXPLICIT_HIERARCHICAL && config->sbrPresent) {
- writeSampleRate(asc, config->extSamplingRate);
- writeAot(asc, config->aot);
- }
-
- switch (config->aot) {
-#ifdef TP_GA_ENABLE
- case AOT_AAC_MAIN:
- case AOT_AAC_LC:
- case AOT_AAC_SSR:
- case AOT_AAC_LTP:
- case AOT_AAC_SCAL:
- case AOT_TWIN_VQ:
- case AOT_ER_AAC_LC:
- case AOT_ER_AAC_LTP:
- case AOT_ER_AAC_SCAL:
- case AOT_ER_TWIN_VQ:
- case AOT_ER_BSAC:
- case AOT_ER_AAC_LD:
- err = transportEnc_writeGASpecificConfig(asc, config, extFlag, alignAnchor);
- if (err)
- return err;
- break;
-#endif /* TP_GA_ENABLE */
-#ifdef TP_ELD_ENABLE
- case AOT_ER_AAC_ELD:
- err = transportEnc_writeELDSpecificConfig(asc, config, epConfig, cb);
- if (err)
- return err;
- break;
-#endif /* TP_ELD_ENABLE */
- default:
- return -1;
- }
-
- switch (config->aot) {
- case AOT_ER_AAC_LC:
- case AOT_ER_AAC_LTP:
- case AOT_ER_AAC_SCAL:
- case AOT_ER_TWIN_VQ:
- case AOT_ER_BSAC:
- case AOT_ER_AAC_LD:
- case AOT_ER_CELP:
- case AOT_ER_HVXC:
- case AOT_ER_HILN:
- case AOT_ER_PARA:
- case AOT_ER_AAC_ELD:
- FDKwriteBits( asc, 0, 2 ); /* epconfig 0 */
- break;
- default:
- break;
- }
-
- /* backward compatible explicit signaling of extension AOT */
- if (config->sbrSignaling==SIG_EXPLICIT_BW_COMPATIBLE)
- {
- TP_ASC_EXTENSION_ID ascExtId = ASCEXT_UNKOWN;
-
- if (config->sbrPresent) {
- ascExtId=ASCEXT_SBR;
- FDKwriteBits( asc, ascExtId, 11 );
- writeAot(asc, config->extAOT);
- FDKwriteBits( asc, 1, 1 ); /* sbrPresentFlag=1 */
- writeSampleRate(asc, config->extSamplingRate);
- if (config->psPresent) {
- ascExtId=ASCEXT_PS;
- FDKwriteBits( asc, ascExtId, 11 );
- FDKwriteBits( asc, 1, 1 ); /* psPresentFlag=1 */
- }
- }
-
- }
-
- /* Make sure all bits are sync'ed */
- FDKsyncCache( asc );
-
- return 0;
-}
diff --git a/libMpegTPEnc/src/tpenc_asc.h b/libMpegTPEnc/src/tpenc_asc.h
deleted file mode 100644
index 47fe7a1..0000000
--- a/libMpegTPEnc/src/tpenc_asc.h
+++ /dev/null
@@ -1,142 +0,0 @@
-
-/* -----------------------------------------------------------------------------------------------------------
-Software License for The Fraunhofer FDK AAC Codec Library for Android
-
-© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
- All rights reserved.
-
- 1. INTRODUCTION
-The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
-the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
-This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
-
-AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
-audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
-independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
-of the MPEG specifications.
-
-Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
-may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
-individually for the purpose of encoding or decoding bit streams in products that are compliant with
-the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
-these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
-software may already be covered under those patent licenses when it is used for those licensed purposes only.
-
-Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
-are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
-applications information and documentation.
-
-2. COPYRIGHT LICENSE
-
-Redistribution and use in source and binary forms, with or without modification, are permitted without
-payment of copyright license fees provided that you satisfy the following conditions:
-
-You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
-your modifications thereto in source code form.
-
-You must retain the complete text of this software license in the documentation and/or other materials
-provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
-You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
-modifications thereto to recipients of copies in binary form.
-
-The name of Fraunhofer may not be used to endorse or promote products derived from this library without
-prior written permission.
-
-You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
-software or your modifications thereto.
-
-Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
-and the date of any change. For modified versions of the FDK AAC Codec, the term
-"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
-"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
-
-3. NO PATENT LICENSE
-
-NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
-ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
-respect to this software.
-
-You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
-by appropriate patent licenses.
-
-4. DISCLAIMER
-
-This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
-"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
-of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
-CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
-including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
-or business interruption, however caused and on any theory of liability, whether in contract, strict
-liability, or tort (including negligence), arising in any way out of the use of this software, even if
-advised of the possibility of such damage.
-
-5. CONTACT INFORMATION
-
-Fraunhofer Institute for Integrated Circuits IIS
-Attention: Audio and Multimedia Departments - FDK AAC LL
-Am Wolfsmantel 33
-91058 Erlangen, Germany
-
-www.iis.fraunhofer.de/amm
-amm-info@iis.fraunhofer.de
------------------------------------------------------------------------------------------------------------ */
-
-/***************************** MPEG-4 AAC Encoder **************************
-
- Author(s): Manuel Jander
- Description: Audio Specific Config writer
-
-******************************************************************************/
-
-#ifndef TPENC_ASC_H
-#define TPENC_ASC_H
-
-/**
- * \brief Get channel config from channel mode.
- *
- * \param channel_mode channel mode
- *
- * \return chanel config
- */
-int getChannelConfig( CHANNEL_MODE channel_mode );
-
-/**
- * \brief Write a Program Config Element.
- *
- * \param hBs bitstream handle into which the PCE is appended
- * \param channelMode the channel mode to be used
- * \param sampleRate the sample rate
- * \param instanceTagPCE the instance tag of the Program Config Element
- * \param profile the MPEG Audio profile to be used
- * \param matrix mixdown gain
- * \param pseudo surround indication
- * \param reference bitstream position for alignment
- * \return zero on success, non-zero on failure.
- */
-int transportEnc_writePCE(
- HANDLE_FDK_BITSTREAM hBs,
- CHANNEL_MODE channelMode,
- INT sampleRate,
- int instanceTagPCE,
- int profile,
- int matrixMixdownA,
- int pseudoSurroundEnable,
- UINT alignAnchor
- );
-
-/**
- * \brief Get the bit count required by a Program Config Element
- *
- * \param channelMode the channel mode to be used
- * \param matrix mixdown gain
- * \param bit offset at which the PCE would start
- * \return the amount of bits required for the PCE including the given bit offset.
- */
-int transportEnc_GetPCEBits(
- CHANNEL_MODE channelMode,
- int matrixMixdownA,
- int bits
- );
-
-#endif /* TPENC_ASC_H */
-
diff --git a/libMpegTPEnc/src/tpenc_dab.cpp b/libMpegTPEnc/src/tpenc_dab.cpp
deleted file mode 100644
index 202fecf..0000000
--- a/libMpegTPEnc/src/tpenc_dab.cpp
+++ /dev/null
@@ -1,467 +0,0 @@
-
-/* -----------------------------------------------------------------------------------------------------------
-Software License for The Fraunhofer FDK AAC Codec Library for Android
-
-� Copyright 1995 - 2012 Fraunhofer-Gesellschaft zur F�rderung der angewandten Forschung e.V.
- All rights reserved.
-
- 1. INTRODUCTION
-The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
-the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
-This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
-
-AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
-audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
-independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
-of the MPEG specifications.
-
-Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
-may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
-individually for the purpose of encoding or decoding bit streams in products that are compliant with
-the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
-these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
-software may already be covered under those patent licenses when it is used for those licensed purposes only.
-
-Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
-are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
-applications information and documentation.
-
-2. COPYRIGHT LICENSE
-
-Redistribution and use in source and binary forms, with or without modification, are permitted without
-payment of copyright license fees provided that you satisfy the following conditions:
-
-You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
-your modifications thereto in source code form.
-
-You must retain the complete text of this software license in the documentation and/or other materials
-provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
-You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
-modifications thereto to recipients of copies in binary form.
-
-The name of Fraunhofer may not be used to endorse or promote products derived from this library without
-prior written permission.
-
-You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
-software or your modifications thereto.
-
-Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
-and the date of any change. For modified versions of the FDK AAC Codec, the term
-"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
-"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
-
-3. NO PATENT LICENSE
-
-NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
-ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
-respect to this software.
-
-You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
-by appropriate patent licenses.
-
-4. DISCLAIMER
-
-This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
-"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
-of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
-CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
-including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
-or business interruption, however caused and on any theory of liability, whether in contract, strict
-liability, or tort (including negligence), arising in any way out of the use of this software, even if
-advised of the possibility of such damage.
-
-5. CONTACT INFORMATION
-
-Fraunhofer Institute for Integrated Circuits IIS
-Attention: Audio and Multimedia Departments - FDK AAC LL
-Am Wolfsmantel 33
-91058 Erlangen, Germany
-
-www.iis.fraunhofer.de/amm
-amm-info@iis.fraunhofer.de
------------------------------------------------------------------------------------------------------------ */
-
-/******************************** MPEG Audio Encoder **************************
-
- Initial author: serge
- contents/description: DAB Transport Headers support
-
-******************************************************************************/
-#include <stdio.h>
-#include "FDK_audio.h"
-#include "tpenc_dab.h"
-
-
-#include "tpenc_lib.h"
-#include "tpenc_asc.h"
-
-#include "common_fix.h"
-
-int dabWrite_CrcStartReg(
- HANDLE_DAB pDab, /*!< pointer to dab stucture */
- HANDLE_FDK_BITSTREAM hBs, /*!< handle to current bit buffer structure */
- int mBits /*!< number of bits in crc region */
- )
-{
- //fprintf(stderr, "dabWrite_CrcStartReg(%p): bits in crc region=%d\n", hBs, mBits);
- return ( FDKcrcStartReg(&pDab->crcInfo2, hBs, mBits) );
-}
-
-void dabWrite_CrcEndReg(
- HANDLE_DAB pDab, /*!< pointer to dab crc info stucture */
- HANDLE_FDK_BITSTREAM hBs, /*!< handle to current bit buffer structure */
- int reg /*!< crc region */
- )
-{
- //fprintf(stderr, "dabWrite_CrcEndReg(%p): crc region=%d\n", hBs, reg);
- FDKcrcEndReg(&pDab->crcInfo2, hBs, reg);
-}
-
-int dabWrite_GetHeaderBits( HANDLE_DAB hDab )
-{
- int bits = 0;
-
- if (hDab->currentBlock == 0) {
- /* Static and variable header bits */
- bits += 16; //header_firecode 16
- bits += 8; //rfa=1, dac_rate=1, sbr_flag=1, aac_channel_mode=1, ps_flag=1, mpeg_surround_config=3
- bits += 12 * hDab->num_raw_blocks; //au_start[1...num_aus] 12 bit AU start position markers
-
- //4 byte alignment
- if (hDab->dac_rate == 0 || hDab->sbr_flag == 0)
- bits+=4;
- //16sbr => 16 + 5 + 3 + 12*(2-1) => 36 => 40 bits 5
- //24sbr => 16 + 5 + 3 + 12*(3-1) => 48 ok 6
- //32sbr => 16 + 5 + 3 + 12*(4-1) => 60 => 64 bits 8
- //48sbr => 16 + 5 + 3 + 12*(6-1) => 84 => 88 bits 11
- }
-
- /* Add raw data block CRC bits. Not really part of the header, put they cause bit overhead to be accounted. */
- bits += 16;
-
-
- return bits;
-}
-
-
-int dabWrite_CountTotalBitDemandHeader( HANDLE_DAB hDab, unsigned int streamDataLength )
-{
- //fprintf(stderr, "streamDataLength=%d (%d bytes)\n", streamDataLength, streamDataLength >> 3);
- return dabWrite_GetHeaderBits(hDab);
-}
-
-
-INT dabWrite_Init(HANDLE_DAB hDab, CODER_CONFIG *config)
-{
- /* Sanity checks */
- if((int)config->aot > 4
- || (int)config->aot < 1 ) {
- return -1;
- }
-
- /* Sanity checks DAB-specific */
- if ( !(config->nSubFrames == 2 && config->samplingRate == 16000 && (config->flags & CC_SBR)) &&
- !(config->nSubFrames == 3 && config->samplingRate == 24000 && (config->flags & CC_SBR)) &&
- !(config->nSubFrames == 4 && config->samplingRate == 32000) &&
- !(config->nSubFrames == 6 && config->samplingRate == 48000)) {
- return -1;
- }
-
- hDab->dac_rate = 0;
- hDab->aac_channel_mode=0;
- hDab->sbr_flag = 0;
- hDab->ps_flag = 0;
- hDab->mpeg_surround_config=0;
- hDab->subchannels_num=config->bitRate/8000;
-
-
- if(config->samplingRate == 24000 || config->samplingRate == 48000)
- hDab->dac_rate = 1;
-
- if (config->extAOT==AOT_SBR || config->extAOT == AOT_PS)
- hDab->sbr_flag = 1;
-
- if(config->extAOT == AOT_PS)
- hDab->ps_flag = 1;
-
-
- if(config->channelMode == MODE_2)
- hDab->aac_channel_mode = 1;
-
- //fprintf(stderr, "hDab->dac_rate=%d\n", hDab->dac_rate);
- //fprintf(stderr, "hDab->sbr_flag=%d\n", hDab->sbr_flag);
- //fprintf(stderr, "hDab->ps_flag=%d\n", hDab->ps_flag);
- //fprintf(stderr, "hDab->aac_channel_mode=%d\n", hDab->aac_channel_mode);
- //fprintf(stderr, "hDab->subchannels_num=%d\n", hDab->subchannels_num);
- //fprintf(stderr, "cc->nSubFrames=%d\n", config->nSubFrames);
-
- hDab->num_raw_blocks=config->nSubFrames-1; /* 0 means 1 raw data block */
-
- FDKcrcInit(&hDab->crcInfo, 0x1021, 0xFFFF, 16);
- FDKcrcInit(&hDab->crcFire, 0x782d, 0, 16);
- FDKcrcInit(&hDab->crcInfo2, 0x8005, 0xFFFF, 16);
-
- hDab->currentBlock = 0;
- hDab->headerBits = dabWrite_GetHeaderBits(hDab);
-
- return 0;
-}
-
-int dabWrite_EncodeHeader(HANDLE_DAB hDab,
- HANDLE_FDK_BITSTREAM hBitStream,
- int buffer_fullness,
- int frame_length)
-{
- INT crcIndex = 0;
-
-
- FDK_ASSERT(((frame_length+hDab->headerBits)/8)<0x2000); /*13 bit*/
- FDK_ASSERT(buffer_fullness<0x800); /* 11 bit */
-
- FDKcrcReset(&hDab->crcInfo);
-
-
-// fprintf(stderr, "dabWrite_EncodeHeader() hDab->currentBlock=%d, frame_length=%d, buffer_fullness=%d\n",
-// hDab->currentBlock, frame_length, buffer_fullness);
-
-// if (hDab->currentBlock == 0) {
-// //hDab->subFrameStartPrev=dabWrite_GetHeaderBits(hDab);
-// fprintf(stderr, "header bits[%d] [%d]\n", hDab->subFrameStartPrev, hDab->subFrameStartPrev >> 3);
-// FDKresetBitbuffer(hBitStream, BS_WRITER);
-// }
-
- //hDab->subFrameStartBit = FDKgetValidBits(hBitStream);
-// fprintf(stderr, "dabWrite_EncodeHeader() hDab->subFrameStartBit=%d [%d]\n", hDab->subFrameStartBit, hDab->subFrameStartBit >> 3);
-
- //hDab->subFrameStartBit = FDKgetValidBits(hBitStream);
- /* Skip new header if this is raw data block 1..n */
- if (hDab->currentBlock == 0)
- {
- FDKresetBitbuffer(hBitStream, BS_WRITER);
-// fprintf(stderr, "dabWrite_EncodeHeader() after FDKresetBitbuffer=%d [%d]\n", FDKgetValidBits(hBitStream), FDKgetValidBits(hBitStream) >> 3);
-
- /* fixed header */
- FDKwriteBits(hBitStream, 0, 16); //header_firecode
- FDKwriteBits(hBitStream, 0, 1); //rfa
- FDKwriteBits(hBitStream, hDab->dac_rate, 1);
- FDKwriteBits(hBitStream, hDab->sbr_flag, 1);
- FDKwriteBits(hBitStream, hDab->aac_channel_mode, 1);
- FDKwriteBits(hBitStream, hDab->ps_flag, 1);
- FDKwriteBits(hBitStream, hDab->mpeg_surround_config, 3);
- /* variable header */
- int i;
- for(i=0; i<hDab->num_raw_blocks; i++)
- FDKwriteBits(hBitStream, 0, 12);
- /* padding */
- if (hDab->dac_rate == 0 || hDab->sbr_flag == 0) {
- FDKwriteBits(hBitStream, 0, 4);
- }
- } /* End of DAB header */
-
- hDab->subFrameStartBit = FDKgetValidBits(hBitStream);
- FDK_ASSERT(FDKgetValidBits(hBitStream) % 8 == 0); //only aligned header
-
-// fprintf(stderr, "dabWrite_EncodeHeader() FDKgetValidBits(hBitStream)=%d [%d]\n", FDKgetValidBits(hBitStream), FDKgetValidBits(hBitStream) >> 3);
- return 0;
-}
-
-int dabWrite_writeExtensionFillPayload(HANDLE_FDK_BITSTREAM hBitStream, int extPayloadBits)
-{
-#define EXT_TYPE_BITS ( 4 )
-#define DATA_EL_VERSION_BITS ( 4 )
-#define FILL_NIBBLE_BITS ( 4 )
-
-#define EXT_TYPE_BITS ( 4 )
-#define DATA_EL_VERSION_BITS ( 4 )
-#define FILL_NIBBLE_BITS ( 4 )
-
- INT extBitsUsed = 0;
- INT extPayloadType = EXT_FIL;
- //fprintf(stderr, "FDKaacEnc_writeExtensionPayload() extPayloadType=%d\n", extPayloadType);
- if (extPayloadBits >= EXT_TYPE_BITS)
- {
- UCHAR fillByte = 0x00; /* for EXT_FIL and EXT_FILL_DATA */
-
- if (hBitStream != NULL) {
- FDKwriteBits(hBitStream, extPayloadType, EXT_TYPE_BITS);
- }
- extBitsUsed += EXT_TYPE_BITS;
-
- switch (extPayloadType) {
- case EXT_FILL_DATA:
- fillByte = 0xA5;
- case EXT_FIL:
- default:
- if (hBitStream != NULL) {
- int writeBits = extPayloadBits;
- FDKwriteBits(hBitStream, 0x00, FILL_NIBBLE_BITS);
- writeBits -= 8; /* acount for the extension type and the fill nibble */
- while (writeBits >= 8) {
- FDKwriteBits(hBitStream, fillByte, 8);
- writeBits -= 8;
- }
- }
- extBitsUsed += FILL_NIBBLE_BITS + (extPayloadBits & ~0x7) - 8;
- break;
- }
- }
-
- return (extBitsUsed);
-}
-
-void dabWrite_FillRawDataBlock(HANDLE_FDK_BITSTREAM hBitStream, int payloadBits)
-{
- INT extBitsUsed = 0;
-#define EL_ID_BITS ( 3 )
-#define FILL_EL_COUNT_BITS ( 4 )
-#define FILL_EL_ESC_COUNT_BITS ( 8 )
-#define MAX_FILL_DATA_BYTES ( 269 )
- while (payloadBits >= (EL_ID_BITS + FILL_EL_COUNT_BITS)) {
- INT cnt, esc_count=-1, alignBits=7;
-
- payloadBits -= EL_ID_BITS + FILL_EL_COUNT_BITS;
- if (payloadBits >= 15*8) {
- payloadBits -= FILL_EL_ESC_COUNT_BITS;
- esc_count = 0; /* write esc_count even if cnt becomes smaller 15 */
- }
- alignBits = 0;
-
- cnt = fixMin(MAX_FILL_DATA_BYTES, (payloadBits+alignBits)>>3);
-
- if (cnt >= 15) {
- esc_count = cnt - 15 + 1;
- }
-
- if (hBitStream != NULL) {
- /* write bitstream */
- FDKwriteBits(hBitStream, ID_FIL, EL_ID_BITS);
- if (esc_count >= 0) {
- FDKwriteBits(hBitStream, 15, FILL_EL_COUNT_BITS);
- FDKwriteBits(hBitStream, esc_count, FILL_EL_ESC_COUNT_BITS);
- } else {
- FDKwriteBits(hBitStream, cnt, FILL_EL_COUNT_BITS);
- }
- }
-
- extBitsUsed += EL_ID_BITS + FILL_EL_COUNT_BITS + ((esc_count>=0) ? FILL_EL_ESC_COUNT_BITS : 0);
-
- cnt = fixMin(cnt*8, payloadBits); /* convert back to bits */
-#if 0
- extBitsUsed += FDKaacEnc_writeExtensionPayload( hBitStream,
- pExtension->type,
- pExtension->pPayload,
- cnt );
-#else
- extBitsUsed += dabWrite_writeExtensionFillPayload(hBitStream, cnt);
-#endif
- payloadBits -= cnt;
- }
-}
-
-void dabWrite_EndRawDataBlock(HANDLE_DAB hDab,
- HANDLE_FDK_BITSTREAM hBs,
- int *pBits)
-{
- FDK_BITSTREAM bsWriter;
- INT crcIndex = 0;
- USHORT crcData;
- INT writeBits=0;
- INT writeBitsNonLastBlock=0;
- INT writeBitsLastBlock=0;
-#if 1
- if (hDab->currentBlock == hDab->num_raw_blocks) {
- //calculate byte-alignment before writing ID_FIL
- if((FDKgetValidBits(hBs)+3) % 8){
- writeBits = 8 - ((FDKgetValidBits(hBs)+3) % 8);
- }
-
- INT offset_end = hDab->subchannels_num*110*8 - 2*8 - 3;
- writeBitsLastBlock = offset_end - FDKgetValidBits(hBs);
- dabWrite_FillRawDataBlock(hBs, writeBitsLastBlock);
- FDKsyncCache(hBs);
- //fprintf(stderr, "FIL-element written=%d\n", writeBitsLastBlock);
- writeBitsLastBlock=writeBits;
- }
-#endif
- FDKwriteBits(hBs, 7, 3); //finalize AU: ID_END
- FDKsyncCache(hBs);
- //byte-align (if ID_FIL doesn't align it).
- if(FDKgetValidBits(hBs) % 8){
- writeBits = 8 - (FDKgetValidBits(hBs) % 8);
- FDKwriteBits(hBs, 0x00, writeBits);
- FDKsyncCache(hBs);
- }
-
- //fake-written bits alignment for last AU
- if (hDab->currentBlock == hDab->num_raw_blocks)
- writeBits=writeBitsLastBlock;
-
- INT frameLen = (FDKgetValidBits(hBs) - hDab->subFrameStartBit) >> 3;
- //fprintf(stderr, "frame=%d, offset writeBits=%d\n", frameLen, writeBits);
-
- FDK_ASSERT(FDKgetValidBits(hBs) % 8 == 0); //only aligned au's
- FDK_ASSERT(hDab->subchannels_num*110*8 >= FDKgetValidBits(hBs)+2*8); //don't overlap superframe
-
- FDKinitBitStream(&bsWriter, hBs->hBitBuf.Buffer, hBs->hBitBuf.bufSize, 0, BS_WRITER);
- FDKpushFor(&bsWriter, hDab->subFrameStartBit);
- FDKcrcReset(&hDab->crcInfo);
- hDab->crcIndex = FDKcrcStartReg(&hDab->crcInfo, &bsWriter, 0);
-#if 0
- if (hDab->currentBlock == hDab->num_raw_blocks) {
- INT offset_size = hDab->subchannels_num*110*8 - 2*8 - FDKgetValidBits(hBs);
- //fprintf(stderr, "offset_size=%d\n", offset_size >> 3);
- FDKpushFor(hBs, offset_size);
- }
-#endif
-
- FDKpushFor(&bsWriter, FDKgetValidBits(hBs) - hDab->subFrameStartBit);
- FDKcrcEndReg(&hDab->crcInfo, &bsWriter, hDab->crcIndex);
- crcData = FDKcrcGetCRC(&hDab->crcInfo);
- //fprintf(stderr, "crcData = %04x\n", crcData);
- /* Write inverted CRC of current raw data block */
- FDKwriteBits(hBs, crcData ^ 0xffff, 16);
- FDKsyncCache(hBs);
-
-
- /* Write distance to current data block */
- if(hDab->currentBlock) {
- FDKinitBitStream(&bsWriter, hBs->hBitBuf.Buffer, hBs->hBitBuf.bufSize, 0, BS_WRITER);
- FDKpushFor(&bsWriter, 24 + (hDab->currentBlock-1)*12);
- //fprintf(stderr, "FDKwriteBits() = %d\n", hDab->subFrameStartBit>>3);
- FDKwriteBits(&bsWriter, (hDab->subFrameStartBit>>3), 12);
- FDKsyncCache(&bsWriter);
- }
-
- /* Write FireCode */
- if (hDab->currentBlock == hDab->num_raw_blocks) {
- FDKinitBitStream(&bsWriter, hBs->hBitBuf.Buffer, hBs->hBitBuf.bufSize, 0, BS_WRITER);
- FDKpushFor(&bsWriter, 16);
-
- FDKcrcReset(&hDab->crcFire);
- crcIndex = FDKcrcStartReg(&hDab->crcFire, &bsWriter, 72);
- FDKpushFor(&bsWriter, 9*8); //9bytes
- FDKcrcEndReg(&hDab->crcFire, &bsWriter, crcIndex);
-
- crcData = FDKcrcGetCRC(&hDab->crcFire);
- //fprintf(stderr, "Firecode: %04x\n", crcData);
-
- FDKinitBitStream(&bsWriter, hBs->hBitBuf.Buffer, hBs->hBitBuf.bufSize, 0, BS_WRITER);
- FDKwriteBits(&bsWriter, crcData, 16);
- FDKsyncCache(&bsWriter);
- }
-
- if (hDab->currentBlock == 0)
- *pBits += hDab->headerBits;
- else
- *pBits += 16;
-
- *pBits += writeBits + 3; //size: ID_END + alignment
-
- /* Correct *pBits to reflect the amount of bits of the current subframe */
- *pBits -= hDab->subFrameStartBit;
- /* Fixup CRC bits, since they come after each raw data block */
-
- hDab->currentBlock++;
- //fprintf(stderr, "dabWrite_EndRawDataBlock() *pBits=%d (%d)\n", *pBits, *pBits >> 3);
-}
-
diff --git a/libMpegTPEnc/src/tpenc_dab.h b/libMpegTPEnc/src/tpenc_dab.h
deleted file mode 100644
index 17b83c6..0000000
--- a/libMpegTPEnc/src/tpenc_dab.h
+++ /dev/null
@@ -1,217 +0,0 @@
-
-/* -----------------------------------------------------------------------------------------------------------
-Software License for The Fraunhofer FDK AAC Codec Library for Android
-
-� Copyright 1995 - 2012 Fraunhofer-Gesellschaft zur F�rderung der angewandten Forschung e.V.
- All rights reserved.
-
- 1. INTRODUCTION
-The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
-the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
-This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
-
-AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
-audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
-independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
-of the MPEG specifications.
-
-Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
-may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
-individually for the purpose of encoding or decoding bit streams in products that are compliant with
-the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
-these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
-software may already be covered under those patent licenses when it is used for those licensed purposes only.
-
-Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
-are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
-applications information and documentation.
-
-2. COPYRIGHT LICENSE
-
-Redistribution and use in source and binary forms, with or without modification, are permitted without
-payment of copyright license fees provided that you satisfy the following conditions:
-
-You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
-your modifications thereto in source code form.
-
-You must retain the complete text of this software license in the documentation and/or other materials
-provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
-You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
-modifications thereto to recipients of copies in binary form.
-
-The name of Fraunhofer may not be used to endorse or promote products derived from this library without
-prior written permission.
-
-You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
-software or your modifications thereto.
-
-Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
-and the date of any change. For modified versions of the FDK AAC Codec, the term
-"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
-"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
-
-3. NO PATENT LICENSE
-
-NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
-ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
-respect to this software.
-
-You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
-by appropriate patent licenses.
-
-4. DISCLAIMER
-
-This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
-"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
-of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
-CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
-including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
-or business interruption, however caused and on any theory of liability, whether in contract, strict
-liability, or tort (including negligence), arising in any way out of the use of this software, even if
-advised of the possibility of such damage.
-
-5. CONTACT INFORMATION
-
-Fraunhofer Institute for Integrated Circuits IIS
-Attention: Audio and Multimedia Departments - FDK AAC LL
-Am Wolfsmantel 33
-91058 Erlangen, Germany
-
-www.iis.fraunhofer.de/amm
-amm-info@iis.fraunhofer.de
------------------------------------------------------------------------------------------------------------ */
-
-/******************************** MPEG Audio Encoder **************************
-
- Initial author: serge
- contents/description: DAB Transport writer
-
-******************************************************************************/
-
-#ifndef TPENC_DAB_H
-#define TPENC_DAB_H
-
-
-
-#include "tp_data.h"
-
-#include "FDK_crc.h"
-
-typedef struct {
- USHORT frame_length;
- UCHAR dac_rate;
- UCHAR aac_channel_mode;
- UCHAR sbr_flag;
- UCHAR ps_flag;
- UCHAR mpeg_surround_config;
- UCHAR num_raw_blocks;
- UCHAR BufferFullnesStartFlag;
- int subchannels_num;
- int headerBits; /*!< Header bit demand for the current raw data block */
- int currentBlock; /*!< Index of current raw data block */
- int subFrameStartBit; /*!< Bit position where the current raw data block begins */
- //int subFrameStartPrev; /*!< Bit position where the previous raw data block begins */
- int crcIndex;
- FDK_CRCINFO crcInfo;
- FDK_CRCINFO crcFire;
- FDK_CRCINFO crcInfo2;
- USHORT tab[256];
-} STRUCT_DAB;
-
-typedef STRUCT_DAB *HANDLE_DAB;
-
-/**
- * \brief Initialize DAB data structure
- *
- * \param hDab DAB data handle
- * \param config a valid CODER_CONFIG struct from where the required
- * information for the DAB header is extrated from
- *
- * \return 0 in case of success.
- */
-INT dabWrite_Init(
- HANDLE_DAB hDab,
- CODER_CONFIG *config
- );
-
-/**
- * \brief Get the total bit overhead caused by DAB
- *
- * \hDab handle to DAB data
- *
- * \return Amount of additional bits required for the current raw data block
- */
-int dabWrite_GetHeaderBits( HANDLE_DAB hDab );
-int dabWrite_CountTotalBitDemandHeader( HANDLE_DAB hDab, unsigned int streamDataLength );
-
-/**
- * \brief Write an DAB header into the given bitstream. May not write a header
- * in case of multiple raw data blocks.
- *
- * \param hDab DAB data handle
- * \param hBitStream bitstream handle into which the DAB may be written into
- * \param buffer_fullness the buffer fullness value for the DAB header
- * \param the current raw data block length
- *
- * \return 0 in case of success.
- */
-INT dabWrite_EncodeHeader(
- HANDLE_DAB hDab,
- HANDLE_FDK_BITSTREAM hBitStream,
- int bufferFullness,
- int frame_length
- );
-/**
- * \brief Finish a DAB raw data block
- *
- * \param hDab DAB data handle
- * \param hBs bitstream handle into which the DAB may be written into
- * \param pBits a pointer to a integer holding the current bitstream buffer bit count,
- * which is corrected to the current raw data block boundary.
- *
- */
-void dabWrite_EndRawDataBlock(
- HANDLE_DAB hDab,
- HANDLE_FDK_BITSTREAM hBs,
- int *bits
- );
-
-
-/**
- * \brief Start CRC region with a maximum number of bits
- * If mBits is positive zero padding will be used for CRC calculation, if there
- * are less than mBits bits available.
- * If mBits is negative no zero padding is done.
- * If mBits is zero the memory for the buffer is allocated dynamically, the
- * number of bits is not limited.
- *
- * \param pDab DAB data handle
- * \param hBs bitstream handle of which the CRC region ends
- * \param mBits limit of number of bits to be considered for the requested CRC region
- *
- * \return ID for the created region, -1 in case of an error
- */
-int dabWrite_CrcStartReg(
- HANDLE_DAB pDab,
- HANDLE_FDK_BITSTREAM hBs,
- int mBits
- );
-
-/**
- * \brief Ends CRC region identified by reg
- *
- * \param pDab DAB data handle
- * \param hBs bitstream handle of which the CRC region ends
- * \param reg a CRC region ID returned previously by dabWrite_CrcStartReg()
- */
-void dabWrite_CrcEndReg(
- HANDLE_DAB pDab,
- HANDLE_FDK_BITSTREAM hBs,
- int reg
- );
-
-
-
-
-#endif /* TPENC_DAB_H */
-
diff --git a/libMpegTPEnc/src/tpenc_latm.cpp b/libMpegTPEnc/src/tpenc_latm.cpp
deleted file mode 100644
index 58e51ef..0000000
--- a/libMpegTPEnc/src/tpenc_latm.cpp
+++ /dev/null
@@ -1,882 +0,0 @@
-
-/* -----------------------------------------------------------------------------------------------------------
-Software License for The Fraunhofer FDK AAC Codec Library for Android
-
-© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
- All rights reserved.
-
- 1. INTRODUCTION
-The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
-the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
-This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
-
-AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
-audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
-independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
-of the MPEG specifications.
-
-Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
-may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
-individually for the purpose of encoding or decoding bit streams in products that are compliant with
-the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
-these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
-software may already be covered under those patent licenses when it is used for those licensed purposes only.
-
-Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
-are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
-applications information and documentation.
-
-2. COPYRIGHT LICENSE
-
-Redistribution and use in source and binary forms, with or without modification, are permitted without
-payment of copyright license fees provided that you satisfy the following conditions:
-
-You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
-your modifications thereto in source code form.
-
-You must retain the complete text of this software license in the documentation and/or other materials
-provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
-You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
-modifications thereto to recipients of copies in binary form.
-
-The name of Fraunhofer may not be used to endorse or promote products derived from this library without
-prior written permission.
-
-You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
-software or your modifications thereto.
-
-Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
-and the date of any change. For modified versions of the FDK AAC Codec, the term
-"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
-"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
-
-3. NO PATENT LICENSE
-
-NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
-ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
-respect to this software.
-
-You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
-by appropriate patent licenses.
-
-4. DISCLAIMER
-
-This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
-"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
-of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
-CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
-including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
-or business interruption, however caused and on any theory of liability, whether in contract, strict
-liability, or tort (including negligence), arising in any way out of the use of this software, even if
-advised of the possibility of such damage.
-
-5. CONTACT INFORMATION
-
-Fraunhofer Institute for Integrated Circuits IIS
-Attention: Audio and Multimedia Departments - FDK AAC LL
-Am Wolfsmantel 33
-91058 Erlangen, Germany
-
-www.iis.fraunhofer.de/amm
-amm-info@iis.fraunhofer.de
------------------------------------------------------------------------------------------------------------ */
-
-/***************************** MPEG-4 AAC Encoder **************************
-
- Author(s):
- Description:
-
-******************************************************************************/
-
-#include "tpenc_latm.h"
-
-
-#include "genericStds.h"
-
-static const short celpFrameLengthTable[64] = {
- 154, 170, 186, 147, 156, 165, 114, 120,
- 186, 126, 132, 138, 142, 146, 154, 166,
- 174, 182, 190, 198, 206, 210, 214, 110,
- 114, 118, 120, 122, 218, 230, 242, 254,
- 266, 278, 286, 294, 318, 342, 358, 374,
- 390, 406, 422, 136, 142, 148, 154, 160,
- 166, 170, 174, 186, 198, 206, 214, 222,
- 230, 238, 216, 160, 280, 338, 0, 0
-};
-
-/*******
- write value to transport stream
- first two bits define the size of the value itself
- then the value itself, with a size of 0-3 bytes
-*******/
-static
-UINT transportEnc_LatmWriteValue(HANDLE_FDK_BITSTREAM hBs, int value)
-{
- UCHAR valueBytes = 4;
- unsigned int bitsWritten = 0;
- int i;
-
- if ( value < (1<<8) ) {
- valueBytes = 1;
- } else if ( value < (1<<16) ) {
- valueBytes = 2;
- } else if ( value < (1<<24) ) {
- valueBytes = 3;
- } else {
- valueBytes = 4;
- }
-
- FDKwriteBits(hBs, valueBytes-1, 2 ); /* size of value in Bytes */
- for (i=0; i<valueBytes; i++) {
- /* write most significant Byte first */
- FDKwriteBits(hBs, (UCHAR)(value>>((valueBytes-1-i)<<3)), 8);
- }
-
- bitsWritten = (valueBytes<<3)+2;
-
- return bitsWritten;
-}
-
-static
-UINT transportEnc_LatmCountFixBitDemandHeader ( HANDLE_LATM_STREAM hAss )
-{
- int bitDemand = 0;
- int insertSetupData = 0 ;
-
- /* only if start of new latm frame */
- if (hAss->subFrameCnt==0)
- {
- /* AudioSyncStream */
-
- if (hAss->tt == TT_MP4_LOAS) {
- bitDemand += 11 ; /* syncword */
- bitDemand += 13 ; /* audioMuxLengthBytes */
- }
-
- /* AudioMuxElement*/
-
- /* AudioMuxElement::Stream Mux Config */
- if (hAss->muxConfigPeriod > 0) {
- insertSetupData = (hAss->latmFrameCounter == 0);
- } else {
- insertSetupData = 0;
- }
-
- if (hAss->tt != TT_MP4_LATM_MCP0) {
- /* AudioMuxElement::useSameStreamMux Flag */
- bitDemand+=1;
-
- if( insertSetupData ) {
- bitDemand += hAss->streamMuxConfigBits;
- }
- }
-
- /* AudioMuxElement::otherDataBits */
- bitDemand += 8*hAss->otherDataLenBytes;
-
- /* AudioMuxElement::ByteAlign */
- if ( bitDemand % 8 ) {
- hAss->fillBits = 8 - (bitDemand % 8);
- bitDemand += hAss->fillBits ;
- } else {
- hAss->fillBits = 0;
- }
- }
-
- return bitDemand ;
-}
-
-static
-UINT transportEnc_LatmCountVarBitDemandHeader ( HANDLE_LATM_STREAM hAss , unsigned int streamDataLength )
-{
- int bitDemand = 0;
- int prog, layer;
-
- /* Payload Length Info*/
- if( hAss->allStreamsSameTimeFraming ) {
- for( prog=0; prog<hAss->noProgram; prog++ ) {
- for( layer=0; layer<LATM_MAX_LAYERS; layer++ ) {
- LATM_LAYER_INFO *p_linfo = &(hAss->m_linfo[prog][layer]);
-
- if( p_linfo->streamID >= 0 ) {
- switch( p_linfo->frameLengthType ) {
- case 0:
- if ( streamDataLength > 0 ) {
- streamDataLength -= bitDemand ;
- while( streamDataLength >= (255<<3) ) {
- bitDemand+=8;
- streamDataLength -= (255<<3);
- }
- bitDemand += 8;
- }
- break;
-
- case 1:
- case 4:
- case 6:
- bitDemand += 2;
- break;
-
- default:
- return 0;
- }
- }
- }
- }
- } else {
- /* there are many possibilities to use this mechanism. */
- switch( hAss->varMode ) {
- case LATMVAR_SIMPLE_SEQUENCE: {
- /* Use the sequence generated by the encoder */
- //int streamCntPosition = transportEnc_SetWritePointer( hAss->hAssemble, 0 );
- //int streamCntPosition = FDKgetValidBits( hAss->hAssemble );
- bitDemand+=4;
-
- hAss->varStreamCnt = 0;
- for( prog=0; prog<hAss->noProgram; prog++ ) {
- for( layer=0; layer<LATM_MAX_LAYERS; layer++ ) {
- LATM_LAYER_INFO *p_linfo = &(hAss->m_linfo[prog][layer]);
-
- if( p_linfo->streamID >= 0 ) {
-
- bitDemand+=4; /* streamID */
- switch( p_linfo->frameLengthType ) {
- case 0:
- streamDataLength -= bitDemand ;
- while( streamDataLength >= (255<<3) ) {
- bitDemand+=8;
- streamDataLength -= (255<<3);
- }
-
- bitDemand += 8;
- break;
- /*bitDemand += 1; endFlag
- break;*/
-
- case 1:
- case 4:
- case 6:
-
- break;
-
- default:
- return 0;
- }
- hAss->varStreamCnt++;
- }
- }
- }
- bitDemand+=4;
- //transportEnc_UpdateBitstreamField( hAss->hAssemble, streamCntPosition, hAss->varStreamCnt-1, 4 );
- //UINT pos = streamCntPosition-FDKgetValidBits(hAss->hAssemble);
- //FDKpushBack( hAss->hAssemble, pos);
- //FDKwriteBits( hAss->hAssemble, hAss->varStreamCnt-1, 4);
- //FDKpushFor( hAss->hAssemble, pos-4);
- }
- break;
-
- default:
- return 0;
- }
- }
-
- return bitDemand ;
-}
-
-TRANSPORTENC_ERROR
-CreateStreamMuxConfig(
- HANDLE_LATM_STREAM hAss,
- HANDLE_FDK_BITSTREAM hBs,
- int bufferFullness,
- CSTpCallBacks *cb
- )
-{
- INT streamIDcnt, tmp;
- int layer, prog;
-
- USHORT coreFrameOffset=0;
-
- hAss->audioMuxVersionA = 0; /* for future extensions */
- hAss->streamMuxConfigBits = 0;
-
- FDKwriteBits( hBs, hAss->audioMuxVersion, 1 ); /* audioMuxVersion */
- hAss->streamMuxConfigBits += 1;
-
- if ( hAss->audioMuxVersion == 1 ) {
- FDKwriteBits( hBs, hAss->audioMuxVersionA, 1 ); /* audioMuxVersionA */
- hAss->streamMuxConfigBits+=1;
- }
-
- if ( hAss->audioMuxVersionA == 0 )
- {
- if ( hAss->audioMuxVersion == 1 ) {
- hAss->streamMuxConfigBits+= transportEnc_LatmWriteValue( hBs, hAss->taraBufferFullness );/* taraBufferFullness */
- }
- FDKwriteBits( hBs, hAss->allStreamsSameTimeFraming ? 1:0, 1 ); /* allStreamsSameTimeFraming */
- FDKwriteBits( hBs, hAss->noSubframes-1, 6 ); /* Number of Subframes */
- FDKwriteBits( hBs, hAss->noProgram-1, 4 ); /* Number of Programs */
-
- hAss->streamMuxConfigBits+=11;
-
- streamIDcnt = 0;
- for( prog=0; prog<hAss->noProgram; prog++ ) {
- int transLayer = 0;
-
- FDKwriteBits( hBs, hAss->noLayer[prog]-1, 3 );
- hAss->streamMuxConfigBits+=3;
-
- for( layer=0; layer<LATM_MAX_LAYERS; layer++ ) {
- LATM_LAYER_INFO *p_linfo = &(hAss->m_linfo[prog][layer]);
- CODER_CONFIG *p_lci = hAss->config[prog][layer];
-
- p_linfo->streamID = -1;
-
- if( hAss->config[prog][layer] != NULL ) {
- int useSameConfig = 0;
-
- if( transLayer > 0 ) {
- FDKwriteBits( hBs, useSameConfig ? 1 : 0, 1 );
- hAss->streamMuxConfigBits+=1;
- }
- if( (useSameConfig == 0) || (transLayer==0) ) {
- UINT bits;
-
- if ( hAss->audioMuxVersion == 1 ) {
- FDKpushFor(hBs, 2); /* align to ASC, even if we do not know the length of the "ascLen" field yet */
- }
-
- bits = FDKgetValidBits( hBs );
-
- transportEnc_writeASC(
- hBs,
- hAss->config[prog][layer],
- cb
- );
-
- bits = FDKgetValidBits( hBs ) - bits;
-
- if ( hAss->audioMuxVersion == 1 ) {
- FDKpushBack(hBs, bits+2);
- hAss->streamMuxConfigBits += transportEnc_LatmWriteValue( hBs, bits );
- transportEnc_writeASC(
- hBs,
- hAss->config[prog][layer],
- cb
- );
- }
-
- hAss->streamMuxConfigBits += bits; /* add asc length to smc summary */
- }
- transLayer++;
-
- if( !hAss->allStreamsSameTimeFraming ) {
- if( streamIDcnt >= LATM_MAX_STREAM_ID )
- return TRANSPORTENC_INVALID_CONFIG;
- }
- p_linfo->streamID = streamIDcnt++;
-
- switch( p_lci->aot ) {
- case AOT_AAC_MAIN :
- case AOT_AAC_LC :
- case AOT_AAC_SSR :
- case AOT_AAC_LTP :
- case AOT_AAC_SCAL :
- case AOT_ER_AAC_LD :
- case AOT_ER_AAC_ELD :
- case AOT_USAC:
- case AOT_RSVD50:
- p_linfo->frameLengthType = 0;
-
- FDKwriteBits( hBs, p_linfo->frameLengthType, 3 ); /* frameLengthType */
- FDKwriteBits( hBs, bufferFullness, 8 ); /* bufferFullness */
- hAss->streamMuxConfigBits+=11;
-
- if ( !hAss->allStreamsSameTimeFraming ) {
- CODER_CONFIG *p_lci_prev = hAss->config[prog][layer-1];
- if ( ((p_lci->aot == AOT_AAC_SCAL) || (p_lci->aot == AOT_ER_AAC_SCAL)) &&
- ((p_lci_prev->aot == AOT_CELP) || (p_lci_prev->aot == AOT_ER_CELP)) ) {
- FDKwriteBits( hBs, coreFrameOffset, 6 ); /* coreFrameOffset */
- hAss->streamMuxConfigBits+=6;
- }
- }
- break;
-
- case AOT_TWIN_VQ:
- p_linfo->frameLengthType = 1;
- tmp = ( (p_lci->bitsFrame+7) >> 3 ) - 20; /* transmission frame length in bytes */
- if( (tmp < 0) ) {
- return TRANSPORTENC_INVALID_TRANSMISSION_FRAME_LENGTH;
- }
- FDKwriteBits( hBs, p_linfo->frameLengthType, 3 ); /* frameLengthType */
- FDKwriteBits( hBs, tmp, 9 );
- hAss->streamMuxConfigBits+=12;
-
- p_linfo->frameLengthBits = (tmp+20) << 3;
- break;
-
- case AOT_CELP:
- p_linfo->frameLengthType = 4;
- FDKwriteBits( hBs, p_linfo->frameLengthType, 3 ); /* frameLengthType */
- hAss->streamMuxConfigBits+=3;
- {
- int i;
- for( i=0; i<62; i++ ) {
- if( celpFrameLengthTable[i] == p_lci->bitsFrame )
- break;
- }
- if( i>=62 ) {
- return TRANSPORTENC_INVALID_CELP_FRAME_LENGTH;
- }
-
- FDKwriteBits( hBs, i, 6 ); /* CELPframeLengthTabelIndex */
- hAss->streamMuxConfigBits+=6;
- }
- p_linfo->frameLengthBits = p_lci->bitsFrame;
- break;
-
- case AOT_HVXC:
- p_linfo->frameLengthType = 6;
- FDKwriteBits( hBs, p_linfo->frameLengthType, 3 ); /* frameLengthType */
- hAss->streamMuxConfigBits+=3;
- {
- int i;
-
- if( p_lci->bitsFrame == 40 ) {
- i = 0;
- } else if( p_lci->bitsFrame == 80 ) {
- i = 1;
- } else {
- return TRANSPORTENC_INVALID_FRAME_BITS;
- }
- FDKwriteBits( hBs, i, 1 ); /* HVXCframeLengthTableIndex */
- hAss->streamMuxConfigBits+=1;
- }
- p_linfo->frameLengthBits = p_lci->bitsFrame;
- break;
-
- case AOT_NULL_OBJECT:
- default:
- return TRANSPORTENC_INVALID_AOT;
- }
- }
- }
- }
-
- FDKwriteBits( hBs, (hAss->otherDataLenBytes>0) ? 1:0, 1 ); /* otherDataPresent */
- hAss->streamMuxConfigBits+=1;
-
- if( hAss->otherDataLenBytes > 0 ) {
-
- INT otherDataLenTmp = hAss->otherDataLenBytes;
- INT escCnt = 0;
- INT otherDataLenEsc = 1;
-
- while(otherDataLenTmp) {
- otherDataLenTmp >>= 8;
- escCnt ++;
- }
-
- do {
- otherDataLenTmp = (hAss->otherDataLenBytes>>(escCnt*8)) & 0xFF;
- escCnt--;
- otherDataLenEsc = escCnt>0;
-
- FDKwriteBits( hBs, otherDataLenEsc, 1 );
- FDKwriteBits( hBs, otherDataLenTmp, 8 );
- hAss->streamMuxConfigBits+=9;
- } while(otherDataLenEsc);
- }
-
- {
- USHORT crcCheckPresent=0;
- USHORT crcCheckSum=0;
-
- FDKwriteBits( hBs, crcCheckPresent, 1 ); /* crcCheckPresent */
- hAss->streamMuxConfigBits+=1;
- if ( crcCheckPresent ){
- FDKwriteBits( hBs, crcCheckSum, 8 ); /* crcCheckSum */
- hAss->streamMuxConfigBits+=8;
- }
- }
-
- } else { /* if ( audioMuxVersionA == 0 ) */
-
- /* for future extensions */
-
- }
-
- return TRANSPORTENC_OK;
-}
-
-
-static TRANSPORTENC_ERROR
-WriteAuPayloadLengthInfo( HANDLE_FDK_BITSTREAM hBitStream, int AuLengthBits )
-{
- int restBytes;
-
- if( AuLengthBits % 8 )
- return TRANSPORTENC_INVALID_AU_LENGTH;
-
- while( AuLengthBits >= 255*8 ) {
- FDKwriteBits( hBitStream, 255, 8 ); /* 255 shows incomplete AU */
- AuLengthBits -= (255*8);
- }
-
- restBytes = (AuLengthBits) >> 3;
- FDKwriteBits( hBitStream, restBytes, 8 );
-
- return TRANSPORTENC_OK;
-}
-
-static
-TRANSPORTENC_ERROR transportEnc_LatmSetNrOfSubframes( HANDLE_LATM_STREAM hAss,
- INT noSubframes_next) /* nr of access units / payloads within a latm frame */
-{
- /* sanity chk */
- if (noSubframes_next < 1 || noSubframes_next > MAX_NR_OF_SUBFRAMES) {
- return TRANSPORTENC_LATM_INVALID_NR_OF_SUBFRAMES;
- }
-
- hAss->noSubframes_next = noSubframes_next;
-
- /* if at start then we can take over the value immediately, otherwise we have to wait for the next SMC */
- if ( (hAss->subFrameCnt == 0) && (hAss->latmFrameCounter == 0) ) {
- hAss->noSubframes = noSubframes_next;
- }
-
- return TRANSPORTENC_OK;
-}
-
-static
-int allStreamsSameTimeFraming( HANDLE_LATM_STREAM hAss, UCHAR noProgram, UCHAR noLayer[] /* return */ )
-{
- int prog, layer;
-
- signed int lastNoSamples = -1;
- signed int minFrameSamples = FDK_INT_MAX;
- signed int maxFrameSamples = 0;
-
- signed int highestSamplingRate = -1;
-
- for( prog=0; prog<noProgram; prog++ ) {
- noLayer[prog] = 0;
-
- for( layer=0; layer<LATM_MAX_LAYERS; layer++ )
- {
- if( hAss->config[prog][layer] != NULL )
- {
- INT hsfSamplesFrame;
-
- noLayer[prog]++;
-
- if( highestSamplingRate < 0 )
- highestSamplingRate = hAss->config[prog][layer]->samplingRate;
-
- hsfSamplesFrame = hAss->config[prog][layer]->samplesPerFrame * highestSamplingRate / hAss->config[prog][layer]->samplingRate;
-
- if( hsfSamplesFrame <= minFrameSamples ) minFrameSamples = hsfSamplesFrame;
- if( hsfSamplesFrame >= maxFrameSamples ) maxFrameSamples = hsfSamplesFrame;
-
- if( lastNoSamples == -1 ) {
- lastNoSamples = hsfSamplesFrame;
- } else {
- if( hsfSamplesFrame != lastNoSamples ) {
- return 0;
- }
- }
- }
- }
- }
-
- return 1;
-}
-
-/**
- * Initialize LATM/LOAS Stream and add layer 0 at program 0.
- */
-static
-TRANSPORTENC_ERROR transportEnc_InitLatmStream( HANDLE_LATM_STREAM hAss,
- int fractDelayPresent,
- signed int muxConfigPeriod, /* insert setup data every muxConfigPeriod frames */
- UINT audioMuxVersion,
- TRANSPORT_TYPE tt
- )
-{
- TRANSPORTENC_ERROR ErrorStatus = TRANSPORTENC_OK;
-
- if (hAss == NULL)
- return TRANSPORTENC_INVALID_PARAMETER;
-
- hAss->tt = tt;
-
- hAss->noProgram = 1;
-
- hAss->audioMuxVersion = audioMuxVersion;
-
- /* Fill noLayer array using hAss->config */
- hAss->allStreamsSameTimeFraming = allStreamsSameTimeFraming( hAss, hAss->noProgram, hAss->noLayer );
- /* Only allStreamsSameTimeFraming==1 is supported */
- FDK_ASSERT(hAss->allStreamsSameTimeFraming);
-
- hAss->fractDelayPresent = fractDelayPresent;
- hAss->otherDataLenBytes = 0;
-
- hAss->varMode = LATMVAR_SIMPLE_SEQUENCE;
-
- /* initialize counters */
- hAss->subFrameCnt = 0;
- hAss->noSubframes = DEFAULT_LATM_NR_OF_SUBFRAMES;
- hAss->noSubframes_next = DEFAULT_LATM_NR_OF_SUBFRAMES;
-
- /* sync layer related */
- hAss->audioMuxLengthBytes = 0;
-
- hAss->latmFrameCounter = 0;
- hAss->muxConfigPeriod = muxConfigPeriod;
-
- return ErrorStatus;
-}
-
-
-/**
- *
- */
-UINT transportEnc_LatmCountTotalBitDemandHeader ( HANDLE_LATM_STREAM hAss , unsigned int streamDataLength )
-{
- UINT bitDemand = 0;
-
- switch (hAss->tt) {
- case TT_MP4_LOAS:
- case TT_MP4_LATM_MCP0:
- case TT_MP4_LATM_MCP1:
- if (hAss->subFrameCnt == 0) {
- bitDemand = transportEnc_LatmCountFixBitDemandHeader ( hAss );
- }
- bitDemand += transportEnc_LatmCountVarBitDemandHeader ( hAss , streamDataLength /*- bitDemand*/);
- break;
- default:
- break;
- }
-
- return bitDemand;
-}
-
-static TRANSPORTENC_ERROR
-AdvanceAudioMuxElement (
- HANDLE_LATM_STREAM hAss,
- HANDLE_FDK_BITSTREAM hBs,
- int auBits,
- int bufferFullness,
- CSTpCallBacks *cb
- )
-{
- TRANSPORTENC_ERROR ErrorStatus = TRANSPORTENC_OK;
- int insertMuxSetup;
-
- /* Insert setup data to assemble Buffer */
- if (hAss->subFrameCnt == 0)
- {
- if (hAss->muxConfigPeriod > 0) {
- insertMuxSetup = (hAss->latmFrameCounter == 0);
- } else {
- insertMuxSetup = 0;
- }
-
- if (hAss->tt != TT_MP4_LATM_MCP0) {
- if( insertMuxSetup ) {
- FDKwriteBits( hBs, 0, 1 ); /* useSameStreamMux useNewStreamMux */
- CreateStreamMuxConfig(hAss, hBs, bufferFullness, cb);
- if (ErrorStatus != TRANSPORTENC_OK)
- return ErrorStatus;
- } else {
- FDKwriteBits( hBs, 1, 1 ); /* useSameStreamMux */
- }
- }
- }
-
- /* PayloadLengthInfo */
- {
- int prog, layer;
-
- for (prog = 0; prog < hAss->noProgram; prog++) {
- for (layer = 0; layer < hAss->noLayer[prog]; layer++) {
- ErrorStatus = WriteAuPayloadLengthInfo( hBs, auBits );
- if (ErrorStatus != TRANSPORTENC_OK)
- return ErrorStatus;
- }
- }
- }
- /* At this point comes the access unit. */
-
- return TRANSPORTENC_OK;
-}
-
-TRANSPORTENC_ERROR
-transportEnc_LatmWrite (
- HANDLE_LATM_STREAM hAss,
- HANDLE_FDK_BITSTREAM hBs,
- int auBits,
- int bufferFullness,
- CSTpCallBacks *cb
- )
-{
- TRANSPORTENC_ERROR ErrorStatus;
-
- if (hAss->subFrameCnt == 0) {
- /* Start new frame */
- FDKresetBitbuffer(hBs, BS_WRITER);
- }
-
- hAss->latmSubframeStart = FDKgetValidBits(hBs);
-
- /* Insert syncword and syncword distance
- - only if loas
- - we must update the syncword distance (=audiomuxlengthbytes) later
- */
- if( hAss->tt == TT_MP4_LOAS && hAss->subFrameCnt == 0)
- {
- /* Start new LOAS frame */
- FDKwriteBits( hBs, 0x2B7, 11 );
- hAss->audioMuxLengthBytes = 0;
- hAss->audioMuxLengthBytesPos = FDKgetValidBits( hBs ); /* store read pointer position */
- FDKwriteBits( hBs, hAss->audioMuxLengthBytes, 13 );
- }
-
- ErrorStatus = AdvanceAudioMuxElement(
- hAss,
- hBs,
- auBits,
- bufferFullness,
- cb
- );
-
- if (ErrorStatus != TRANSPORTENC_OK)
- return ErrorStatus;
-
- return ErrorStatus;
-}
-
-void transportEnc_LatmAdjustSubframeBits(HANDLE_LATM_STREAM hAss,
- int *bits)
-{
- /* Substract bits from possible previous subframe */
- *bits -= hAss->latmSubframeStart;
- /* Add fill bits */
- if (hAss->subFrameCnt == 0)
- *bits += hAss->fillBits;
-}
-
-
-void transportEnc_LatmGetFrame(HANDLE_LATM_STREAM hAss,
- HANDLE_FDK_BITSTREAM hBs,
- int *bytes)
-{
-
- hAss->subFrameCnt++;
- if (hAss->subFrameCnt >= hAss->noSubframes)
- {
-
- /* Add LOAS frame length if required. */
- if (hAss->tt == TT_MP4_LOAS)
- {
- int latmBytes;
-
- latmBytes = (FDKgetValidBits(hBs)+7) >> 3;
-
- /* write length info into assembler buffer */
- hAss->audioMuxLengthBytes = latmBytes - 3; /* 3=Syncword + length */
- {
- FDK_BITSTREAM tmpBuf;
-
- FDKinitBitStream( &tmpBuf, hBs->hBitBuf.Buffer, hBs->hBitBuf.bufSize, 0, BS_WRITER ) ;
- FDKpushFor( &tmpBuf, hAss->audioMuxLengthBytesPos );
- FDKwriteBits( &tmpBuf, hAss->audioMuxLengthBytes, 13 );
- FDKsyncCache( &tmpBuf );
- }
- }
-
- /* Write AudioMuxElement byte alignment fill bits */
- FDKwriteBits(hBs, 0, hAss->fillBits);
-
- FDK_ASSERT( (FDKgetValidBits(hBs) % 8) == 0);
-
- hAss->subFrameCnt = 0;
-
- FDKsyncCache(hBs);
- *bytes = (FDKgetValidBits(hBs) + 7)>>3;
- //FDKfetchBuffer(hBs, buffer, (UINT*)bytes);
-
- if (hAss->muxConfigPeriod > 0)
- {
- hAss->latmFrameCounter++;
-
- if (hAss->latmFrameCounter >= hAss->muxConfigPeriod) {
- hAss->latmFrameCounter = 0;
- hAss->noSubframes = hAss->noSubframes_next;
- }
- }
- } else {
- /* No data this time */
- *bytes = 0;
- }
-}
-
-/**
- * Init LATM/LOAS
- */
-TRANSPORTENC_ERROR transportEnc_Latm_Init(
- HANDLE_LATM_STREAM hAss,
- HANDLE_FDK_BITSTREAM hBs,
- CODER_CONFIG *layerConfig,
- UINT audioMuxVersion,
- TRANSPORT_TYPE tt,
- CSTpCallBacks *cb
- )
-{
- TRANSPORTENC_ERROR ErrorStatus;
- int fractDelayPresent = 0;
- int prog, layer;
-
- int setupDataDistanceFrames = layerConfig->headerPeriod;
-
- FDK_ASSERT(setupDataDistanceFrames>=0);
-
- for (prog=0; prog<LATM_MAX_PROGRAMS; prog++) {
- for (layer=0; layer<LATM_MAX_LAYERS; layer++) {
- hAss->config[prog][layer] = NULL;
- hAss->m_linfo[prog][layer].streamID = -1;
- }
- }
-
- hAss->config[0][0] = layerConfig;
- hAss->m_linfo[0][0].streamID = 0;
-
- ErrorStatus = transportEnc_InitLatmStream( hAss,
- fractDelayPresent,
- setupDataDistanceFrames,
- (audioMuxVersion)?1:0,
- tt
- );
- if (ErrorStatus != TRANSPORTENC_OK)
- goto bail;
-
- ErrorStatus = transportEnc_LatmSetNrOfSubframes(
- hAss,
- layerConfig->nSubFrames
- );
- if (ErrorStatus != TRANSPORTENC_OK)
- goto bail;
-
- /* Get the size of the StreamMuxConfig somehow */
- AdvanceAudioMuxElement(hAss, hBs, 0, 0, cb);
- //CreateStreamMuxConfig(hAss, hBs, 0);
-
-bail:
- return ErrorStatus;
-}
-
-
-
-
-
-
diff --git a/libMpegTPEnc/src/tpenc_latm.h b/libMpegTPEnc/src/tpenc_latm.h
deleted file mode 100644
index 34eea58..0000000
--- a/libMpegTPEnc/src/tpenc_latm.h
+++ /dev/null
@@ -1,264 +0,0 @@
-
-/* -----------------------------------------------------------------------------------------------------------
-Software License for The Fraunhofer FDK AAC Codec Library for Android
-
-© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
- All rights reserved.
-
- 1. INTRODUCTION
-The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
-the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
-This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
-
-AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
-audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
-independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
-of the MPEG specifications.
-
-Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
-may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
-individually for the purpose of encoding or decoding bit streams in products that are compliant with
-the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
-these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
-software may already be covered under those patent licenses when it is used for those licensed purposes only.
-
-Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
-are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
-applications information and documentation.
-
-2. COPYRIGHT LICENSE
-
-Redistribution and use in source and binary forms, with or without modification, are permitted without
-payment of copyright license fees provided that you satisfy the following conditions:
-
-You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
-your modifications thereto in source code form.
-
-You must retain the complete text of this software license in the documentation and/or other materials
-provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
-You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
-modifications thereto to recipients of copies in binary form.
-
-The name of Fraunhofer may not be used to endorse or promote products derived from this library without
-prior written permission.
-
-You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
-software or your modifications thereto.
-
-Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
-and the date of any change. For modified versions of the FDK AAC Codec, the term
-"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
-"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
-
-3. NO PATENT LICENSE
-
-NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
-ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
-respect to this software.
-
-You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
-by appropriate patent licenses.
-
-4. DISCLAIMER
-
-This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
-"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
-of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
-CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
-including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
-or business interruption, however caused and on any theory of liability, whether in contract, strict
-liability, or tort (including negligence), arising in any way out of the use of this software, even if
-advised of the possibility of such damage.
-
-5. CONTACT INFORMATION
-
-Fraunhofer Institute for Integrated Circuits IIS
-Attention: Audio and Multimedia Departments - FDK AAC LL
-Am Wolfsmantel 33
-91058 Erlangen, Germany
-
-www.iis.fraunhofer.de/amm
-amm-info@iis.fraunhofer.de
------------------------------------------------------------------------------------------------------------ */
-
-/***************************** MPEG-4 AAC Encoder **************************
-
- Author(s):
- Description:
-
-******************************************************************************/
-
-#ifndef TPENC_LATM_H
-#define TPENC_LATM_H
-
-
-
-#include "tpenc_lib.h"
-#include "FDK_bitstream.h"
-
-
-#define DEFAULT_LATM_NR_OF_SUBFRAMES 1
-#define DEFAULT_LATM_SMC_REPEAT 8
-
-#define MAX_AAC_LAYERS 9
-
-#define LATM_MAX_PROGRAMS 1
-#define LATM_MAX_STREAM_ID 16
-
-#define LATM_MAX_LAYERS 1 /*MAX_AAC_LAYERS*/
-
-#define MAX_NR_OF_SUBFRAMES 2 /* set this carefully to avoid buffer overflows */
-
-typedef enum { LATMVAR_SIMPLE_SEQUENCE } LATM_VAR_MODE;
-
-typedef struct {
- signed int frameLengthType;
- signed int frameLengthBits;
- signed int varFrameLengthTable[4];
- signed int streamID;
-} LATM_LAYER_INFO;
-
-
-typedef struct {
- LATM_LAYER_INFO m_linfo[LATM_MAX_PROGRAMS][LATM_MAX_LAYERS];
- CODER_CONFIG *config[LATM_MAX_PROGRAMS][LATM_MAX_LAYERS];
-
- LATM_VAR_MODE varMode;
- TRANSPORT_TYPE tt;
-
- int audioMuxLengthBytes;
-
- int audioMuxLengthBytesPos;
- int taraBufferFullness; /* state of the bit reservoir */
- int varStreamCnt;
- unsigned int otherDataLenBytes;
-
- UCHAR latmFrameCounter; /* Current frame number. Counts modulo muxConfigPeriod */
- UCHAR muxConfigPeriod; /* Distance in frames between MuxConfig */
-
- UCHAR audioMuxVersion; /* AMV1 supports transmission of taraBufferFullness and ASC lengths */
- UCHAR audioMuxVersionA; /* for future extensions */
-
- UCHAR noProgram;
- UCHAR noLayer[LATM_MAX_PROGRAMS];
- UCHAR fractDelayPresent;
-
- UCHAR allStreamsSameTimeFraming;
- UCHAR subFrameCnt; /* Current Subframe frame */
- UCHAR noSubframes; /* Number of subframes */
- UINT latmSubframeStart; /* Position of current subframe start */
- UCHAR noSubframes_next;
-
- UCHAR fillBits; /* AudioMuxElement fill bits */
- UCHAR streamMuxConfigBits;
-
-} LATM_STREAM;
-
-typedef LATM_STREAM *HANDLE_LATM_STREAM;
-
-/**
- * \brief Initialize LATM_STREAM Handle. Creates automatically one program with one layer with
- * the given layerConfig. The layerConfig must be persisten because references to this pointer
- * are made at any time again.
- * Use transportEnc_Latm_AddLayer() to add more programs/layers.
- *
- * \param hLatmStreamInfo HANDLE_LATM_STREAM handle
- * \param hBs Bitstream handle
- * \param layerConfig a valid CODER_CONFIG struct containing the current audio configuration parameters
- * \param audioMuxVersion the LATM audioMuxVersion to be used
- * \param tt the specific TRANSPORT_TYPE to be used, either TT_MP4_LOAS, TT_MP4_LATM_MCP1 or TT_MP4_LATM_MCP0 LOAS
- * \param cb callback information structure.
- *
- * \return an TRANSPORTENC_ERROR error code
- */
-TRANSPORTENC_ERROR transportEnc_Latm_Init(
- HANDLE_LATM_STREAM hLatmStreamInfo,
- HANDLE_FDK_BITSTREAM hBs,
- CODER_CONFIG *layerConfig,
- UINT audioMuxVersion,
- TRANSPORT_TYPE tt,
- CSTpCallBacks *cb
- );
-
-/**
- * \brief Get bit demand of next LATM/LOAS header
- *
- * \param hAss HANDLE_LATM_STREAM handle
- * \param streamDataLength the length of the payload
- *
- * \return the number of bits required by the LATM/LOAS headers
- */
-unsigned int transportEnc_LatmCountTotalBitDemandHeader (
- HANDLE_LATM_STREAM hAss,
- unsigned int streamDataLength
- );
-
-/**
- * \brief Write LATM/LOAS header into given bitstream handle
- *
- * \param hLatmStreamInfo HANDLE_LATM_STREAM handle
- * \param hBitstream Bitstream handle
- * \param auBits amount of current payload bits
- * \param bufferFullness LATM buffer fullness value
- * \param cb callback information structure.
- *
- * \return an TRANSPORTENC_ERROR error code
- */
-TRANSPORTENC_ERROR
-transportEnc_LatmWrite (
- HANDLE_LATM_STREAM hAss,
- HANDLE_FDK_BITSTREAM hBitstream,
- int auBits,
- int bufferFullness,
- CSTpCallBacks *cb
- );
-
-/**
- * \brief Adjust bit count relative to current subframe
- *
- * \param hAss HANDLE_LATM_STREAM handle
- * \param pBits pointer to an int, where the current frame bit count is contained,
- * and where the subframe relative bit count will be returned into
- *
- * \return void
- */
-void transportEnc_LatmAdjustSubframeBits(HANDLE_LATM_STREAM hAss,
- int *pBits);
-
-/**
- * \brief Request an LATM frame, which may, or may not be available
- *
- * \param hAss HANDLE_LATM_STREAM handle
- * \param hBs Bitstream handle
- * \param pBytes pointer to an int, where the current frame byte count stored into.
- * A return value of zero means that currently no LATM/LOAS frame can be returned.
- * The latter is expected in case of multiple subframes being used.
- *
- * \return void
- */
-void transportEnc_LatmGetFrame(
- HANDLE_LATM_STREAM hAss,
- HANDLE_FDK_BITSTREAM hBs,
- int *pBytes
- );
-
-/**
- * \brief Write a StreamMuxConfig into the given bitstream handle
- *
- * \param hAss HANDLE_LATM_STREAM handle
- * \param hBs Bitstream handle
- * \param bufferFullness LATM buffer fullness value
- * \param cb callback information structure.
- *
- * \return void
- */
-TRANSPORTENC_ERROR
-CreateStreamMuxConfig(
- HANDLE_LATM_STREAM hAss,
- HANDLE_FDK_BITSTREAM hBs,
- int bufferFullness,
- CSTpCallBacks *cb
- );
-
-
-#endif /* TPENC_LATM_H */
diff --git a/libMpegTPEnc/src/tpenc_lib.cpp b/libMpegTPEnc/src/tpenc_lib.cpp
deleted file mode 100644
index 058b8d4..0000000
--- a/libMpegTPEnc/src/tpenc_lib.cpp
+++ /dev/null
@@ -1,685 +0,0 @@
-
-/* -----------------------------------------------------------------------------------------------------------
-Software License for The Fraunhofer FDK AAC Codec Library for Android
-
-© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
- All rights reserved.
-
- 1. INTRODUCTION
-The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
-the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
-This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
-
-AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
-audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
-independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
-of the MPEG specifications.
-
-Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
-may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
-individually for the purpose of encoding or decoding bit streams in products that are compliant with
-the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
-these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
-software may already be covered under those patent licenses when it is used for those licensed purposes only.
-
-Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
-are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
-applications information and documentation.
-
-2. COPYRIGHT LICENSE
-
-Redistribution and use in source and binary forms, with or without modification, are permitted without
-payment of copyright license fees provided that you satisfy the following conditions:
-
-You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
-your modifications thereto in source code form.
-
-You must retain the complete text of this software license in the documentation and/or other materials
-provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
-You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
-modifications thereto to recipients of copies in binary form.
-
-The name of Fraunhofer may not be used to endorse or promote products derived from this library without
-prior written permission.
-
-You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
-software or your modifications thereto.
-
-Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
-and the date of any change. For modified versions of the FDK AAC Codec, the term
-"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
-"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
-
-3. NO PATENT LICENSE
-
-NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
-ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
-respect to this software.
-
-You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
-by appropriate patent licenses.
-
-4. DISCLAIMER
-
-This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
-"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
-of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
-CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
-including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
-or business interruption, however caused and on any theory of liability, whether in contract, strict
-liability, or tort (including negligence), arising in any way out of the use of this software, even if
-advised of the possibility of such damage.
-
-5. CONTACT INFORMATION
-
-Fraunhofer Institute for Integrated Circuits IIS
-Attention: Audio and Multimedia Departments - FDK AAC LL
-Am Wolfsmantel 33
-91058 Erlangen, Germany
-
-www.iis.fraunhofer.de/amm
-amm-info@iis.fraunhofer.de
------------------------------------------------------------------------------------------------------------ */
-
-/************************** MPEG-4 Transport Encoder ************************
-
- Author(s): Manuel Jander
- Description: MPEG Transport encode
-
-******************************************************************************/
-
-#include "tpenc_lib.h"
-
-/* library info */
-#include "version"
-
-#define MODULE_NAME "transportEnc"
-
-#include "tpenc_asc.h"
-#include "conv_string.h"
-
-#include "tpenc_adts.h"
-
-#include "tpenc_adif.h"
-
-#include "tpenc_dab.h"
-
-#include "tpenc_latm.h"
-
-typedef struct {
- int curSubFrame;
- int nSubFrames;
- int prevBits;
-} RAWPACKETS_INFO;
-
-struct TRANSPORTENC
-{
- CODER_CONFIG config;
- TRANSPORT_TYPE transportFmt; /*!< MPEG4 transport type. */
-
- FDK_BITSTREAM bitStream;
- UCHAR *bsBuffer;
- INT bsBufferSize;
-
- INT pceFrameCounter; /*!< Indicates frame period when PCE must be written in raw_data_block.
- -1 means not to write a PCE in raw_dat_block. */
- union {
- STRUCT_ADTS adts;
-
- ADIF_INFO adif;
-
- STRUCT_DAB dab;
-
- LATM_STREAM latm;
-
- RAWPACKETS_INFO raw;
-
-
-
- } writer;
-
- CSTpCallBacks callbacks;
-};
-
-typedef struct _TRANSPORTENC_STRUCT TRANSPORTENC_STRUCT;
-
-
-/*
- * MEMORY Declaration
- */
-
-C_ALLOC_MEM(Ram_TransportEncoder, TRANSPORTENC, 1)
-
-TRANSPORTENC_ERROR transportEnc_Open( HANDLE_TRANSPORTENC *phTpEnc )
-{
- HANDLE_TRANSPORTENC hTpEnc;
-
- if ( phTpEnc == NULL ){
- return TRANSPORTENC_INVALID_PARAMETER;
- }
-
- hTpEnc = GetRam_TransportEncoder(0);
-
- if ( hTpEnc == NULL ) {
- return TRANSPORTENC_NO_MEM;
- }
-
- *phTpEnc = hTpEnc;
- return TRANSPORTENC_OK;
-}
-
-/**
- * \brief Get frame period of PCE in raw_data_block.
- *
- * - Write PCE only if necessary. PCE can be part of the ASC if chConfig==0 whererfore
- * no additonal PCE will be written in raw_data_block.
- * - A matrixMixdown coefficient can only be written if chConfig is 5.0 or 5.1.
- * - The PCE repetition rate in raw_data_block can be controlled via headerPeriod parameter.
- *
- * \param channelConfig Channel Configuration derived from Channel Mode
- * \param transportFmt Format of the transport to be written.
- * \param headerPeriod Chosen PCE frame repetition rate.
- * \param matrixMixdownA Indicates if a valid Matrix Mixdown coefficient is available.
- *
- * \return PCE frame repetition rate. -1 means no PCE present in raw_data_block.
- */
-static INT getPceRepetitionRate(
- const int channelConfig,
- const TRANSPORT_TYPE transportFmt,
- const int headerPeriod,
- const int matrixMixdownA
- )
-{
- INT pceFrameCounter = -1; /* variable to be returned */
-
- if (headerPeriod>0) {
- switch ( channelConfig ) {
- case 0:
- switch (transportFmt) {
- case TT_MP4_ADTS:
- case TT_MP4_LATM_MCP0:
- case TT_MP4_RAW:
- pceFrameCounter = headerPeriod;
- break;
- case TT_MP4_ADIF: /* ADIF header comprises PCE */
- case TT_MP4_LOAS: /* PCE in ASC if chChonfig==0 */
- case TT_MP4_LATM_MCP1: /* PCE in ASC if chChonfig==0 */
- case TT_DRM: /* PCE not allowed in DRM */
- case TT_DABPLUS:
- default:
- pceFrameCounter = -1; /* no PCE in raw_data_block */
- }
- break;
- case 5: /* MODE_1_2_2 */
- case 6: /* MODE_1_2_2_1 */
- /* matrixMixdownCoefficient can only be written if 5.0 and 5.1 config present. */
- if (matrixMixdownA!=0) {
- switch (transportFmt) {
- case TT_MP4_ADIF: /* ADIF header comprises PCE */
- case TT_MP4_ADTS:
- case TT_MP4_LOAS: /* no PCE in ASC because chConfig!=0 */
- case TT_MP4_LATM_MCP1: /* no PCE in ASC because chConfig!=0 */
- case TT_MP4_LATM_MCP0:
- case TT_MP4_RAW:
- pceFrameCounter = headerPeriod;
- break;
- case TT_DRM: /* PCE not allowed in DRM */
- case TT_DABPLUS:
- default:
- pceFrameCounter = -1; /* no PCE in raw_data_block */
- } /* switch transportFmt */
- } /* if matrixMixdownA!=0 */
- break;
- default:
- pceFrameCounter = -1; /* no PCE in raw_data_block */
- } /* switch getChannelConfig() */
- } /* if headerPeriod>0 */
- else {
- pceFrameCounter = -1; /* no PCE in raw_data_block */
- }
-
- return pceFrameCounter;
-}
-
-TRANSPORTENC_ERROR transportEnc_Init(
- HANDLE_TRANSPORTENC hTpEnc,
- UCHAR *bsBuffer,
- INT bsBufferSize,
- TRANSPORT_TYPE transportFmt,
- CODER_CONFIG *cconfig,
- UINT flags
- )
-{
- /* Copy configuration structure */
- FDKmemcpy(&hTpEnc->config, cconfig, sizeof(CODER_CONFIG));
-
- /* Init transportEnc struct. */
- hTpEnc->transportFmt = transportFmt;
-
- hTpEnc->bsBuffer = bsBuffer;
- hTpEnc->bsBufferSize = bsBufferSize;
-
- FDKinitBitStream(&hTpEnc->bitStream, hTpEnc->bsBuffer, hTpEnc->bsBufferSize, 0, BS_WRITER);
-
- switch (transportFmt) {
-
- case TT_MP4_ADIF:
- /* Sanity checks */
- if ( (hTpEnc->config.aot != AOT_AAC_LC)
- ||(hTpEnc->config.samplesPerFrame != 1024))
- {
- return TRANSPORTENC_INVALID_PARAMETER;
- }
- hTpEnc->writer.adif.headerWritten = 0;
- hTpEnc->writer.adif.samplingRate = hTpEnc->config.samplingRate;
- hTpEnc->writer.adif.bitRate = hTpEnc->config.bitRate;
- hTpEnc->writer.adif.profile = ((int)hTpEnc->config.aot) - 1;
- hTpEnc->writer.adif.cm = hTpEnc->config.channelMode;
- hTpEnc->writer.adif.bVariableRate = 0;
- hTpEnc->writer.adif.instanceTag = 0;
- break;
-
- case TT_MP4_ADTS:
- /* Sanity checks */
- if ( ( hTpEnc->config.aot != AOT_AAC_LC)
- ||(hTpEnc->config.samplesPerFrame != 1024) )
- {
- return TRANSPORTENC_INVALID_PARAMETER;
- }
- if ( adtsWrite_Init(&hTpEnc->writer.adts, &hTpEnc->config) != 0) {
- return TRANSPORTENC_INVALID_PARAMETER;
- }
- break;
-
- case TT_DABPLUS:
- /* Sanity checks */
- if ( ( hTpEnc->config.aot != AOT_AAC_LC)
- ||(hTpEnc->config.samplesPerFrame != 960) )
- {
- return TRANSPORTENC_INVALID_PARAMETER;
- }
- if ( dabWrite_Init(&hTpEnc->writer.dab, &hTpEnc->config) != 0) {
- return TRANSPORTENC_INVALID_PARAMETER;
- }
- break;
-
- case TT_MP4_LOAS:
- case TT_MP4_LATM_MCP0:
- case TT_MP4_LATM_MCP1:
- {
- TRANSPORTENC_ERROR error;
-
- error = transportEnc_Latm_Init(
- &hTpEnc->writer.latm,
- &hTpEnc->bitStream,
- &hTpEnc->config,
- flags & TP_FLAG_LATM_AMV,
- transportFmt,
- &hTpEnc->callbacks
- );
- if (error != TRANSPORTENC_OK) {
- return error;
- }
- }
- break;
-
- case TT_MP4_RAW:
- hTpEnc->writer.raw.curSubFrame = 0;
- hTpEnc->writer.raw.nSubFrames = hTpEnc->config.nSubFrames;
- break;
-
-
-
- default:
- return TRANSPORTENC_INVALID_PARAMETER;
- }
-
- /* pceFrameCounter indicates if PCE must be written in raw_data_block. */
- hTpEnc->pceFrameCounter = getPceRepetitionRate(
- getChannelConfig(hTpEnc->config.channelMode),
- transportFmt,
- hTpEnc->config.headerPeriod,
- hTpEnc->config.matrixMixdownA);
-
- return TRANSPORTENC_OK;
-}
-
-HANDLE_FDK_BITSTREAM transportEnc_GetBitstream( HANDLE_TRANSPORTENC hTp )
-{
- return &hTp->bitStream;
-}
-
-int transportEnc_RegisterSbrCallback( HANDLE_TRANSPORTENC hTpEnc, const cbSbr_t cbSbr, void* user_data)
-{
- if (hTpEnc == NULL) {
- return -1;
- }
- hTpEnc->callbacks.cbSbr = cbSbr;
- hTpEnc->callbacks.cbSbrData = user_data;
- return 0;
-}
-
-
-TRANSPORTENC_ERROR transportEnc_WriteAccessUnit(
- HANDLE_TRANSPORTENC hTp,
- INT frameUsedBits,
- int bufferFullness,
- int ncc
- )
-{
- TRANSPORTENC_ERROR err = TRANSPORTENC_OK;
-
- if (!hTp) {
- return TRANSPORTENC_INVALID_PARAMETER;
- }
- HANDLE_FDK_BITSTREAM hBs = &hTp->bitStream;
-
- /* In case of writing PCE in raw_data_block frameUsedBits must be adapted. */
- if (hTp->pceFrameCounter>=hTp->config.headerPeriod) {
- frameUsedBits += transportEnc_GetPCEBits(hTp->config.channelMode, hTp->config.matrixMixdownA, 3); /* Consider 3 bits ID signalling in alignment */
- }
-
- switch (hTp->transportFmt) {
- case TT_MP4_ADIF:
- FDKinitBitStream(&hTp->bitStream, hTp->bsBuffer, hTp->bsBufferSize, 0, BS_WRITER);
- adifWrite_EncodeHeader(
- &hTp->writer.adif,
- hBs,
- bufferFullness
- );
- break;
- case TT_MP4_ADTS:
- bufferFullness /= ncc; /* Number of Considered Channels */
- bufferFullness /= 32;
- bufferFullness = FDKmin(0x7FF, bufferFullness); /* Signal variable rate */
- adtsWrite_EncodeHeader(
- &hTp->writer.adts,
- &hTp->bitStream,
- bufferFullness,
- frameUsedBits
- );
- break;
- case TT_DABPLUS:
- bufferFullness /= ncc; /* Number of Considered Channels */
- bufferFullness /= 32;
- bufferFullness = FDKmin(0x7FF, bufferFullness); /* Signal variable rate */
- dabWrite_EncodeHeader(
- &hTp->writer.dab,
- &hTp->bitStream,
- bufferFullness,
- frameUsedBits
- );
- break;
- case TT_MP4_LOAS:
- case TT_MP4_LATM_MCP0:
- case TT_MP4_LATM_MCP1:
- bufferFullness /= ncc; /* Number of Considered Channels */
- bufferFullness /= 32;
- bufferFullness = FDKmin(0xFF, bufferFullness); /* Signal variable rate */
- transportEnc_LatmWrite(
- &hTp->writer.latm,
- hBs,
- frameUsedBits,
- bufferFullness,
- &hTp->callbacks
- );
- break;
- case TT_MP4_RAW:
- if (hTp->writer.raw.curSubFrame >= hTp->writer.raw.nSubFrames) {
- hTp->writer.raw.curSubFrame = 0;
- FDKinitBitStream(&hTp->bitStream, hTp->bsBuffer, hTp->bsBufferSize, 0, BS_WRITER);
- }
- hTp->writer.raw.prevBits = FDKgetValidBits(hBs);
- break;
- default:
- err = TRANSPORTENC_UNSUPPORTED_FORMAT;
- break;
- }
-
- /* Write PCE in raw_data_block if required */
- if (hTp->pceFrameCounter>=hTp->config.headerPeriod) {
- INT crcIndex = 0;
- /* Align inside PCE with repsect to the first bit of the raw_data_block() */
- UINT alignAnchor = FDKgetValidBits(&hTp->bitStream);
-
- /* Write PCE element ID bits */
- FDKwriteBits(&hTp->bitStream, ID_PCE, 3);
-
- if ( (hTp->transportFmt==TT_MP4_ADTS) && !hTp->writer.adts.protection_absent) {
- crcIndex = adtsWrite_CrcStartReg(&hTp->writer.adts, &hTp->bitStream, 0);
- }
-
- /* Write PCE as first raw_data_block element */
- transportEnc_writePCE(&hTp->bitStream, hTp->config.channelMode, hTp->config.samplingRate, 0, 1, hTp->config.matrixMixdownA, (hTp->config.flags&CC_PSEUDO_SURROUND)?1:0, alignAnchor);
-
- if ( (hTp->transportFmt==TT_MP4_ADTS) && !hTp->writer.adts.protection_absent) {
- adtsWrite_CrcEndReg(&hTp->writer.adts, &hTp->bitStream, crcIndex);
- }
- hTp->pceFrameCounter = 0; /* reset pce frame counter */
- }
-
- if (hTp->pceFrameCounter!=-1) {
- hTp->pceFrameCounter++; /* Update pceFrameCounter only if PCE writing is active. */
- }
-
- return err;
-}
-
-
-TRANSPORTENC_ERROR transportEnc_EndAccessUnit(HANDLE_TRANSPORTENC hTp, int *bits)
-{
- switch (hTp->transportFmt) {
- case TT_MP4_LATM_MCP0:
- case TT_MP4_LATM_MCP1:
- case TT_MP4_LOAS:
- transportEnc_LatmAdjustSubframeBits(&hTp->writer.latm, bits);
- break;
- case TT_MP4_ADTS:
- adtsWrite_EndRawDataBlock(&hTp->writer.adts, &hTp->bitStream, bits);
- break;
- case TT_DABPLUS:
- dabWrite_EndRawDataBlock(&hTp->writer.dab, &hTp->bitStream, bits);
- break;
- case TT_MP4_ADIF:
- /* Substract ADIF header from AU bits, not to be considered. */
- *bits -= adifWrite_GetHeaderBits(&hTp->writer.adif);
- hTp->writer.adif.headerWritten = 1;
- break;
- case TT_MP4_RAW:
- *bits -= hTp->writer.raw.prevBits;
- break;
- default:
- break;
- }
-
- return TRANSPORTENC_OK;
-}
-
-TRANSPORTENC_ERROR transportEnc_GetFrame(HANDLE_TRANSPORTENC hTpEnc, int *nbytes)
-{
- HANDLE_FDK_BITSTREAM hBs = &hTpEnc->bitStream;
-
- switch (hTpEnc->transportFmt) {
- case TT_MP4_LATM_MCP0:
- case TT_MP4_LATM_MCP1:
- case TT_MP4_LOAS:
- *nbytes = hTpEnc->bsBufferSize;
- transportEnc_LatmGetFrame(&hTpEnc->writer.latm, hBs, nbytes);
- break;
- case TT_MP4_ADTS:
- if (hTpEnc->writer.adts.currentBlock >= hTpEnc->writer.adts.num_raw_blocks+1) {
- *nbytes = (FDKgetValidBits(hBs) + 7)>>3;
- hTpEnc->writer.adts.currentBlock = 0;
- } else {
- *nbytes = 0;
- }
- break;
- case TT_DABPLUS:
- if (hTpEnc->writer.dab.currentBlock >= hTpEnc->writer.dab.num_raw_blocks+1) {
- *nbytes = (FDKgetValidBits(hBs) + 7)>>3;
- hTpEnc->writer.dab.currentBlock = 0;
- } else {
- *nbytes = 0;
- }
- break;
- case TT_MP4_ADIF:
- FDK_ASSERT((INT)FDKgetValidBits(hBs) >= 0);
- *nbytes = (FDKgetValidBits(hBs) + 7)>>3;
- break;
- case TT_MP4_RAW:
- FDKsyncCache(hBs);
- hTpEnc->writer.raw.curSubFrame++;
- *nbytes = ((FDKgetValidBits(hBs)-hTpEnc->writer.raw.prevBits) + 7)>>3;
- break;
- default:
- break;
- }
-
- return TRANSPORTENC_OK;
-}
-
-INT transportEnc_GetStaticBits( HANDLE_TRANSPORTENC hTp, int auBits )
-{
- INT nbits = 0, nPceBits = 0;
-
- /* Write PCE within raw_data_block in transport lib. */
- if (hTp->pceFrameCounter>=hTp->config.headerPeriod) {
- nPceBits = transportEnc_GetPCEBits(hTp->config.channelMode, hTp->config.matrixMixdownA, 3); /* Consider 3 bits ID signalling in alignment */
- auBits += nPceBits; /* Adapt required raw_data_block bit consumtpion for AU length information e.g. in LATM/LOAS configuration. */
- }
-
- switch (hTp->transportFmt) {
- case TT_MP4_ADIF:
- case TT_MP4_RAW:
- nbits = 0; /* Do not consider the ADIF header into the total bitrate */
- break;
- case TT_MP4_ADTS:
- nbits = adtsWrite_GetHeaderBits(&hTp->writer.adts);
- break;
- case TT_DABPLUS:
- nbits = dabWrite_CountTotalBitDemandHeader(&hTp->writer.dab, auBits);
- break;
- case TT_MP4_LOAS:
- case TT_MP4_LATM_MCP0:
- case TT_MP4_LATM_MCP1:
- nbits = transportEnc_LatmCountTotalBitDemandHeader( &hTp->writer.latm, auBits );
- break;
- default:
- nbits = 0;
- break;
- }
-
- /* PCE is written in the transport library therefore the bit consumption is part of the transport static bits. */
- nbits += nPceBits;
-
- return nbits;
-}
-
-void transportEnc_Close(HANDLE_TRANSPORTENC *phTp)
-{
- if (phTp != NULL)
- {
- if (*phTp != NULL) {
- FreeRam_TransportEncoder(phTp);
- }
- }
-}
-
-int transportEnc_CrcStartReg(HANDLE_TRANSPORTENC hTpEnc, int mBits)
-{
- int crcReg = 0;
-
- switch (hTpEnc->transportFmt) {
- case TT_MP4_ADTS:
- crcReg = adtsWrite_CrcStartReg(&hTpEnc->writer.adts, &hTpEnc->bitStream, mBits);
- break;
- case TT_DABPLUS:
- crcReg = dabWrite_CrcStartReg(&hTpEnc->writer.dab, &hTpEnc->bitStream, mBits);
- break;
- default:
- break;
- }
-
- return crcReg;
-}
-
-void transportEnc_CrcEndReg(HANDLE_TRANSPORTENC hTpEnc, int reg)
-{
- switch (hTpEnc->transportFmt) {
- case TT_MP4_ADTS:
- adtsWrite_CrcEndReg(&hTpEnc->writer.adts, &hTpEnc->bitStream, reg);
- break;
- case TT_DABPLUS:
- dabWrite_CrcEndReg(&hTpEnc->writer.dab, &hTpEnc->bitStream, reg);
- break;
- default:
- break;
- }
-}
-
-
-TRANSPORTENC_ERROR transportEnc_GetConf(HANDLE_TRANSPORTENC hTpEnc,
- CODER_CONFIG *cc,
- FDK_BITSTREAM *dataBuffer,
- UINT *confType)
-{
- TRANSPORTENC_ERROR tpErr = TRANSPORTENC_OK;
- HANDLE_LATM_STREAM hLatmConfig = &hTpEnc->writer.latm;
-
- *confType = 0; /* set confType variable to default */
-
- /* write StreamMuxConfig or AudioSpecificConfig depending on format used */
- switch (hTpEnc->transportFmt)
- {
- case TT_MP4_LATM_MCP0:
- case TT_MP4_LATM_MCP1:
- case TT_MP4_LOAS:
- tpErr = CreateStreamMuxConfig(hLatmConfig, dataBuffer, 0, &hTpEnc->callbacks);
- *confType = 1; /* config is SMC */
- break;
- default:
- if (transportEnc_writeASC(dataBuffer, cc, &hTpEnc->callbacks) != 0) {
- tpErr = TRANSPORTENC_UNKOWN_ERROR;
- }
- }
-
- return tpErr;
-
-}
-
-TRANSPORTENC_ERROR transportEnc_GetLibInfo( LIB_INFO *info )
-{
- int i;
-
- if (info == NULL) {
- return TRANSPORTENC_INVALID_PARAMETER;
- }
- /* search for next free tab */
- for (i = 0; i < FDK_MODULE_LAST; i++) {
- if (info[i].module_id == FDK_NONE) break;
- }
- if (i == FDK_MODULE_LAST) {
- return TRANSPORTENC_UNKOWN_ERROR;
- }
- info += i;
-
- info->module_id = FDK_TPENC;
- info->version = LIB_VERSION(TP_LIB_VL0, TP_LIB_VL1, TP_LIB_VL2);
- LIB_VERSION_STRING(info);
- info->build_date = __DATE__;
- info->build_time = __TIME__;
- info->title = TP_LIB_TITLE;
-
- /* Set flags */
- info->flags = 0
- | CAPF_ADIF
- | CAPF_ADTS
- | CAPF_LATM
- | CAPF_LOAS
- | CAPF_RAWPACKETS
- | CAPF_DAB_AAC
- ;
-
- return TRANSPORTENC_OK;
-}
-
diff --git a/libMpegTPEnc/src/version b/libMpegTPEnc/src/version
deleted file mode 100644
index 2803347..0000000
--- a/libMpegTPEnc/src/version
+++ /dev/null
@@ -1,8 +0,0 @@
-
-/* library info */
-#define TP_LIB_VL0 2
-#define TP_LIB_VL1 3
-#define TP_LIB_VL2 3
-#define TP_LIB_TITLE "MPEG Transport"
-#define TP_LIB_BUILD_DATE __DATE__
-#define TP_LIB_BUILD_TIME __TIME__