diff options
Diffstat (limited to 'libMpegTPEnc')
| -rw-r--r-- | libMpegTPEnc/include/mpegFileWrite.h | 140 | ||||
| -rw-r--r-- | libMpegTPEnc/include/tp_data.h | 340 | ||||
| -rw-r--r-- | libMpegTPEnc/include/tpenc_lib.h | 296 | ||||
| -rw-r--r-- | libMpegTPEnc/src/tpenc_adif.cpp | 182 | ||||
| -rw-r--r-- | libMpegTPEnc/src/tpenc_adif.h | 135 | ||||
| -rw-r--r-- | libMpegTPEnc/src/tpenc_adts.cpp | 315 | ||||
| -rw-r--r-- | libMpegTPEnc/src/tpenc_adts.h | 218 | ||||
| -rw-r--r-- | libMpegTPEnc/src/tpenc_asc.cpp | 576 | ||||
| -rw-r--r-- | libMpegTPEnc/src/tpenc_asc.h | 142 | ||||
| -rw-r--r-- | libMpegTPEnc/src/tpenc_dab.cpp | 467 | ||||
| -rw-r--r-- | libMpegTPEnc/src/tpenc_dab.h | 217 | ||||
| -rw-r--r-- | libMpegTPEnc/src/tpenc_latm.cpp | 882 | ||||
| -rw-r--r-- | libMpegTPEnc/src/tpenc_latm.h | 264 | ||||
| -rw-r--r-- | libMpegTPEnc/src/tpenc_lib.cpp | 685 | ||||
| -rw-r--r-- | libMpegTPEnc/src/version | 8 |
15 files changed, 0 insertions, 4867 deletions
diff --git a/libMpegTPEnc/include/mpegFileWrite.h b/libMpegTPEnc/include/mpegFileWrite.h deleted file mode 100644 index f886a0b..0000000 --- a/libMpegTPEnc/include/mpegFileWrite.h +++ /dev/null @@ -1,140 +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 Decoder ************************** - - Author(s): Manuel Jander - Description: Bitstream data provider for MP4 decoders - -******************************************************************************/ - -#include "machine_type.h" -#include "FDK_audio.h" - -/*!< If MPFWRITE_MP4FF_ENABLE is set, include support for MPEG ISO fileformat. - If not set, no .mp4, .m4a and .3gp files can be used for input. */ -/* #define MPFWRITE_MP4FF_ENABLE */ - -typedef struct STRUCT_FILEWRITE *HANDLE_FILEWRITE; - -#ifdef __cplusplus -extern "C" { -#endif - -/** - * \brief Open an MPEG audio file. - * \param mpegFileWrite_Filename String of the filename to be opened. - * \param fileFmt Transport format to use. - * \param conf - * \param confSize - * \return MPEG file write handle. - */ -HANDLE_FILEWRITE mpegFileWrite_Open( char *mpegFileWrite_Filename, - FILE_FORMAT fileFmt, - TRANSPORT_TYPE transportType, - UCHAR *conf, - UINT confSize - ); - -/** - * \brief Write to an MPEG audio file. - * \param inBuffer Buffer to write. - * \param bufferSize Size of buffer to write in bytes. - * \return 0 on sucess, -1 on unsupported file format or write error. - */ -int mpegFileWrite_Write( HANDLE_FILEWRITE hFileWrite, - UCHAR *inBuffer, - int bufferSize - ); - -/** - * \brief Deallocate memory and close file. - * \param hFileWrite MPEG file write handle. - * \return 0 on sucess. - */ -int mpegFileWrite_Close( HANDLE_FILEWRITE *hFileWrite ); - - -#ifdef __cplusplus -} -#endif diff --git a/libMpegTPEnc/include/tp_data.h b/libMpegTPEnc/include/tp_data.h deleted file mode 100644 index 5269858..0000000 --- a/libMpegTPEnc/include/tp_data.h +++ /dev/null @@ -1,340 +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 Decoder ************************** - - Author(s): Manuel Jander - Description: MPEG Transport data tables - -******************************************************************************/ - -#ifndef __TP_DATA_H__ -#define __TP_DATA_H__ - -#include "machine_type.h" -#include "FDK_audio.h" -#include "FDK_bitstream.h" - -/* - * Configuration - */ -#define TP_GA_ENABLE -/* #define TP_CELP_ENABLE */ -/* #define TP_HVXC_ENABLE */ -/* #define TP_SLS_ENABLE */ -#define TP_ELD_ENABLE -/* #define TP_USAC_ENABLE */ -/* #define TP_RSVD50_ENABLE */ - -#if defined(TP_GA_ENABLE) || defined(TP_SLS_ENABLE) -#define TP_PCE_ENABLE /**< Enable full PCE support */ -#endif - -/** - * ProgramConfig struct. - */ -/* ISO/IEC 14496-3 4.4.1.1 Table 4.2 Program config element */ -#define PC_FSB_CHANNELS_MAX 16 /* Front/Side/Back channels */ -#define PC_LFE_CHANNELS_MAX 4 -#define PC_ASSOCDATA_MAX 8 -#define PC_CCEL_MAX 16 /* CC elements */ -#define PC_COMMENTLENGTH 256 - -typedef struct -{ -#ifdef TP_PCE_ENABLE - /* PCE bitstream elements: */ - UCHAR ElementInstanceTag; - UCHAR Profile; - UCHAR SamplingFrequencyIndex; - UCHAR NumFrontChannelElements; - UCHAR NumSideChannelElements; - UCHAR NumBackChannelElements; - UCHAR NumLfeChannelElements; - UCHAR NumAssocDataElements; - UCHAR NumValidCcElements; - - UCHAR MonoMixdownPresent; - UCHAR MonoMixdownElementNumber; - - UCHAR StereoMixdownPresent; - UCHAR StereoMixdownElementNumber; - - UCHAR MatrixMixdownIndexPresent; - UCHAR MatrixMixdownIndex; - UCHAR PseudoSurroundEnable; - - UCHAR FrontElementIsCpe[PC_FSB_CHANNELS_MAX]; - UCHAR FrontElementTagSelect[PC_FSB_CHANNELS_MAX]; - - UCHAR SideElementIsCpe[PC_FSB_CHANNELS_MAX]; - UCHAR SideElementTagSelect[PC_FSB_CHANNELS_MAX]; - - UCHAR BackElementIsCpe[PC_FSB_CHANNELS_MAX]; - UCHAR BackElementTagSelect[PC_FSB_CHANNELS_MAX]; - - UCHAR LfeElementTagSelect[PC_LFE_CHANNELS_MAX]; - - UCHAR AssocDataElementTagSelect[PC_ASSOCDATA_MAX]; - - UCHAR CcElementIsIndSw[PC_CCEL_MAX]; - UCHAR ValidCcElementTagSelect[PC_CCEL_MAX]; - - UCHAR CommentFieldBytes; - UCHAR Comment[PC_COMMENTLENGTH]; -#endif /* TP_PCE_ENABLE */ - - /* Helper variables for administration: */ - UCHAR isValid; /*!< Flag showing if PCE has been read successfully. */ - UCHAR NumChannels; /*!< Amount of audio channels summing all channel elements including LFEs */ - UCHAR NumEffectiveChannels; /*!< Amount of audio channels summing only SCEs and CPEs */ - UCHAR elCounter; - -} CProgramConfig; - -typedef enum { - ASCEXT_UNKOWN = -1, - ASCEXT_SBR = 0x2b7, - ASCEXT_PS = 0x548, - ASCEXT_MPS = 0x76a, - ASCEXT_SAOC = 0x7cb, - ASCEXT_LDMPS = 0x7cc - -} TP_ASC_EXTENSION_ID; - -#ifdef TP_GA_ENABLE -/** - * GaSpecificConfig struct - */ -typedef struct { - UINT m_frameLengthFlag ; - UINT m_dependsOnCoreCoder ; - UINT m_coreCoderDelay ; - - UINT m_extensionFlag ; - UINT m_extensionFlag3 ; - - UINT m_layer; - UINT m_numOfSubFrame; - UINT m_layerLength; - -} CSGaSpecificConfig; -#endif /* TP_GA_ENABLE */ - - - - -#ifdef TP_ELD_ENABLE - -typedef enum { - ELDEXT_TERM = 0x0, /* Termination tag */ - ELDEXT_SAOC = 0x1, /* SAOC config */ - ELDEXT_LDSAC = 0x2 /* LD MPEG Surround config */ - /* reserved */ -} ASC_ELD_EXT_TYPE; - -typedef struct { - UCHAR m_frameLengthFlag; - - UCHAR m_sbrPresentFlag; - UCHAR m_useLdQmfTimeAlign; /* Use LD-MPS QMF in SBR to achive time alignment */ - UCHAR m_sbrSamplingRate; - UCHAR m_sbrCrcFlag; - -} CSEldSpecificConfig; -#endif /* TP_ELD_ENABLE */ - - - - -/** - * Audio configuration struct, suitable for encoder and decoder configuration. - */ -typedef struct { - - /* XYZ Specific Data */ - union { -#ifdef TP_GA_ENABLE - CSGaSpecificConfig m_gaSpecificConfig; /**< General audio specific configuration. */ -#endif /* TP_GA_ENABLE */ -#ifdef TP_ELD_ENABLE - CSEldSpecificConfig m_eldSpecificConfig; /**< ELD specific configuration. */ -#endif /* TP_ELD_ENABLE */ - } m_sc; - - /* Common ASC parameters */ -#ifdef TP_PCE_ENABLE - CProgramConfig m_progrConfigElement; /**< Program configuration. */ -#endif /* TP_PCE_ENABLE */ - - AUDIO_OBJECT_TYPE m_aot; /**< Audio Object Type. */ - UINT m_samplingFrequency; /**< Samplerate. */ - UINT m_samplesPerFrame; /**< Amount of samples per frame. */ - UINT m_directMapping; /**< Document this please !! */ - - AUDIO_OBJECT_TYPE m_extensionAudioObjectType; /**< Audio object type */ - UINT m_extensionSamplingFrequency; /**< Samplerate */ - - SCHAR m_channelConfiguration; /**< Channel configuration index */ - - SCHAR m_epConfig; /**< Error protection index */ - SCHAR m_vcb11Flag; /**< aacSectionDataResilienceFlag */ - SCHAR m_rvlcFlag; /**< aacScalefactorDataResilienceFlag */ - SCHAR m_hcrFlag; /**< aacSpectralDataResilienceFlag */ - - SCHAR m_sbrPresentFlag; /**< Flag indicating the presence of SBR data in the bitstream */ - SCHAR m_psPresentFlag; /**< Flag indicating the presence of parametric stereo data in the bitstream */ - UCHAR m_samplingFrequencyIndex; /**< Samplerate index */ - UCHAR m_extensionSamplingFrequencyIndex; /**< Samplerate index */ - SCHAR m_extensionChannelConfiguration; /**< Channel configuration index */ - -} CSAudioSpecificConfig; - -typedef INT (*cbUpdateConfig_t)(void*, const CSAudioSpecificConfig*); -typedef INT (*cbSsc_t)( - void*, HANDLE_FDK_BITSTREAM, - const AUDIO_OBJECT_TYPE coreCodec, - const INT samplingFrequency, - const INT muxMode, - const INT configBytes - ); -typedef INT (*cbSbr_t)( - void * self, - HANDLE_FDK_BITSTREAM hBs, - const INT sampleRateIn, - const INT sampleRateOut, - const INT samplesPerFrame, - const AUDIO_OBJECT_TYPE coreCodec, - const MP4_ELEMENT_ID elementID, - const INT elementIndex - ); - -typedef struct { - cbUpdateConfig_t cbUpdateConfig; /*!< Function pointer for Config change notify callback. */ - void *cbUpdateConfigData; /*!< User data pointer for Config change notify callback. */ - cbSsc_t cbSsc; /*!< Function pointer for SSC parser callback. */ - void *cbSscData; /*!< User data pointer for SSC parser callback. */ - cbSbr_t cbSbr; /*!< Function pointer for SBR header parser callback. */ - void *cbSbrData; /*!< User data pointer for SBR header parser callback. */ -} CSTpCallBacks; - -static const UINT SamplingRateTable[] = -{ 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050, 16000, 12000, 11025, 8000, 7350, 0, 0, - 0 -}; - -static inline -int getSamplingRateIndex( UINT samplingRate ) -{ - UINT sf_index, tableSize=sizeof(SamplingRateTable)/sizeof(UINT); - - for (sf_index=0; sf_index<tableSize; sf_index++) { - if( SamplingRateTable[sf_index] == samplingRate ) break; - } - - if (sf_index>tableSize-1) { - return tableSize-1; - } - - return sf_index; -} - -/* - * Get Channel count from channel configuration - */ -static inline int getNumberOfTotalChannels(int channelConfig) -{ - if (channelConfig > 0 && channelConfig < 8) - return (channelConfig == 7)?8:channelConfig; - else - return 0; -} - -static inline -int getNumberOfEffectiveChannels(const int channelConfig) -{ - const int n[] = {0,1,2,3,4,5,5,7}; - return n[channelConfig]; -} - -#endif /* __TP_DATA_H__ */ diff --git a/libMpegTPEnc/include/tpenc_lib.h b/libMpegTPEnc/include/tpenc_lib.h deleted file mode 100644 index 2833e82..0000000 --- a/libMpegTPEnc/include/tpenc_lib.h +++ /dev/null @@ -1,296 +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 - -******************************************************************************/ - -#ifndef __TPENC_LIB_H__ -#define __TPENC_LIB_H__ - -#include "tp_data.h" -#include "FDK_bitstream.h" - -#define TRANSPORTENC_INBUF_SIZE 8192 - -typedef enum { - TRANSPORTENC_OK = 0, /*!< All fine. */ - TRANSPORTENC_NO_MEM, /*!< Out of memory. */ - TRANSPORTENC_UNKOWN_ERROR = 1, /*!< Unknown error (embarrasing). */ - TRANSPORTENC_INVALID_PARAMETER, /*!< An invalid parameter was passed to a function . */ - TRANSPORTENC_PARSE_ERROR, /*!< Bitstream data contained inconsistencies (wrong syntax). */ - TRANSPORTENC_UNSUPPORTED_FORMAT, /*!< Unsupported transport format. */ - TRANSPORTENC_NOT_ENOUGH_BITS, /*!< Out of bits. Provide more bits and try again. */ - - TRANSPORTENC_INVALID_CONFIG, /*!< Error in configuration. */ - TRANSPORTENC_LATM_INVALID_NR_OF_SUBFRAMES, /*!< LATM: number of subframes out of range. */ - TRANSPORTENC_LOAS_NOT_AVAILABLE, /*!< LOAS format not supported. */ - TRANSPORTENC_INVALID_LATM_ALIGNMENT, /*!< AudioMuxElement length not aligned to 1 byte. */ - - TRANSPORTENC_INVALID_TRANSMISSION_FRAME_LENGTH, /*!< Invalid transmission frame length (< 0). */ - TRANSPORTENC_INVALID_CELP_FRAME_LENGTH, /*!< Invalid CELP frame length found (>= 62). */ - TRANSPORTENC_INVALID_FRAME_BITS, /*!< Frame bits is not 40 and not 80. */ - TRANSPORTENC_INVALID_AOT, /*!< Unknown AOT found. */ - TRANSPORTENC_INVALID_AU_LENGTH /*!< Invalid Access Unit length (not byte-aligned). */ - -} TRANSPORTENC_ERROR; - -typedef struct TRANSPORTENC *HANDLE_TRANSPORTENC; - -/** - * \brief Determine a reasonable channel configuration on the basis of channel_mode. - * \param noChannels Number of audio channels. - * \return CHANNEL_MODE value that matches the given amount of audio channels. - */ -CHANNEL_MODE transportEnc_GetChannelMode( int noChannels ); - -/** - * \brief Register SBR heaqder writer callback. - * \param hTp Handle of transport decoder. - * \param cbUpdateConfig Pointer to a callback function to handle SBR header writing. - * \param user_data void pointer for user data passed to the callback as first parameter. - * \return 0 on success. - */ -int transportEnc_RegisterSbrCallback ( - HANDLE_TRANSPORTENC hTpEnc, - const cbSbr_t cbSbr, - void* user_data - ); - -/** - * \brief Register SSC writer callback. - * \param hTp Handle of transport decoder. - * \param cbUpdateConfig Pointer to a callback function to handle SSC writing. - * \param user_data void pointer for user data passed to the callback as first parameter. - * \return 0 on success. - */ -int transportEnc_RegisterSscCallback ( - HANDLE_TRANSPORTENC hTpEnc, - const cbSsc_t cbSsc, - void* user_data - ); - -/** - * \brief Write ASC from given parameters. - * \param asc A HANDLE_FDK_BITSTREAM where the ASC is written to. - * \param config Structure containing the codec configuration settings. - * \param cb callback information structure. - * \return 0 on success. - */ -int transportEnc_writeASC ( - HANDLE_FDK_BITSTREAM asc, - CODER_CONFIG *config, - CSTpCallBacks *cb - ); - - -/* Defintion of flags that can be passed to transportEnc_Open() */ -#define TP_FLAG_MPEG4 1 /** MPEG4 (instead of MPEG2) */ -#define TP_FLAG_LATM_AMV 2 /** LATM AudioMuxVersion */ -#define TP_FLAG_LATM_AMVA 4 /** LATM AudioMuxVersionA */ - -/** - * \brief Allocate transport encoder. - * \param phTpEnc Pointer to transport encoder handle. - * \return Error code. - */ -TRANSPORTENC_ERROR transportEnc_Open( HANDLE_TRANSPORTENC *phTpEnc ); - -/** - * \brief Init transport encoder. - * \param bsBuffer Pointer to transport encoder. - * \param bsBuffer Pointer to bitstream buffer. - * \param bsBufferSize Size in bytes of bsBuffer. - * \param transportFmt Format of the transport to be written. - * \param config Pointer to a valid CODER_CONFIG struct. - * \param flags Transport encoder flags. - * \return Error code. - */ -TRANSPORTENC_ERROR transportEnc_Init( - HANDLE_TRANSPORTENC hTpEnc, - UCHAR *bsBuffer, - INT bsBufferSize, - TRANSPORT_TYPE transportFmt, - CODER_CONFIG *config, - UINT flags - ); - -/** - * \brief Get transport encoder bitstream. - * \param hTp Pointer to a transport encoder handle. - * \return The handle to the requested FDK bitstream. - */ -HANDLE_FDK_BITSTREAM transportEnc_GetBitstream( HANDLE_TRANSPORTENC hTp ); - -/** - * \brief Get amount of bits required by the transport headers. - * \param hTp Handle of transport encoder. - * \param auBits Amount of payload bits required for the current subframe. - * \return Error code. - */ -INT transportEnc_GetStaticBits( HANDLE_TRANSPORTENC hTp, int auBits ); - -/** - * \brief Close transport encoder. This function assures that all allocated memory is freed. - * \param phTp Pointer to a previously allocated transport encoder handle. - */ -void transportEnc_Close( HANDLE_TRANSPORTENC *phTp ); - -/** - * \brief Write one access unit. - * \param hTp Handle of transport encoder. - * \param total_bits Amount of total access unit bits. - * \param bufferFullness Value of current buffer fullness in bits. - * \param noConsideredChannels Number of bitrate wise considered channels (all minus LFE channels). - * \return Error code. - */ -TRANSPORTENC_ERROR transportEnc_WriteAccessUnit( HANDLE_TRANSPORTENC hTp, - INT total_bits, - int bufferFullness, - int noConsideredChannels ); - -/** - * \brief Inform the transportEnc layer that writing of access unit has finished. This function - * is required to be called when the encoder has finished writing one Access - * one Access Unit for bitstream housekeeping. - * \param hTp Transport handle. - * \param pBits Pointer to an int, where the current amount of frame bits is passed - * and where the current amount of subframe bits is returned. - * - * OR: This integer is modified by the amount of extra bit alignment that may occurr. - * - * \return Error code. - */ -TRANSPORTENC_ERROR transportEnc_EndAccessUnit( HANDLE_TRANSPORTENC hTp, int *pBits); - -/* - * \brief Get a payload frame. - * \param hTpEnc Transport encoder handle. - * \param nBytes Pointer to an int to hold the frame size in bytes. Returns zero - * if currently there is no complete frame for output (number of sub frames > 1). - * \return Error code. - */ -TRANSPORTENC_ERROR transportEnc_GetFrame(HANDLE_TRANSPORTENC hTpEnc, int *nbytes); - -/* ADTS CRC support */ - -/** - * \brief Set current bitstream position as start of a new data region. - * \param hTpEnc Transport encoder handle. - * \param mBits Size in bits of the data region. Set to 0 if it should not be of a fixed size. - * \return Data region ID, which should be used when calling transportEnc_CrcEndReg(). - */ -int transportEnc_CrcStartReg(HANDLE_TRANSPORTENC hTpEnc, int mBits); - -/** - * \brief Set end of data region. - * \param hTpEnc Transport encoder handle. - * \param reg Data region ID, opbtained from transportEnc_CrcStartReg(). - * \return void - */ -void transportEnc_CrcEndReg(HANDLE_TRANSPORTENC hTpEnc, int reg); - -/** - * \brief Get AudioSpecificConfig or StreamMuxConfig from transport encoder handle and write it to dataBuffer. - * \param hTpEnc Transport encoder handle. - * \param cc Pointer to the current and valid configuration contained in a CODER_CONFIG struct. - * \param dataBuffer Bitbuffer holding binary configuration. - * \param confType Pointer to an UINT where the configuration type is returned (0:ASC, 1:SMC). - * \return Error code. - */ -TRANSPORTENC_ERROR transportEnc_GetConf( HANDLE_TRANSPORTENC hTpEnc, - CODER_CONFIG *cc, - FDK_BITSTREAM *dataBuffer, - UINT *confType ); - -/** - * \brief Get information (version among other things) of the transport encoder library. - * \param info Pointer to an allocated LIB_INFO struct. - * \return Error code. - */ -TRANSPORTENC_ERROR transportEnc_GetLibInfo( LIB_INFO *info ); - -#endif /* #ifndef __TPENC_LIB_H__ */ diff --git a/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 - -ďż˝ Copyright 1995 - 2012 Fraunhofer-Gesellschaft zur Fďż˝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 - -ďż˝ Copyright 1995 - 2012 Fraunhofer-Gesellschaft zur Fďż˝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__ |