diff options
Diffstat (limited to 'libMpegTPDec/include')
-rw-r--r-- | libMpegTPDec/include/mpegFileRead.h | 194 | ||||
-rw-r--r-- | libMpegTPDec/include/tp_data.h | 487 | ||||
-rw-r--r-- | libMpegTPDec/include/tpdec_lib.h | 621 |
3 files changed, 683 insertions, 619 deletions
diff --git a/libMpegTPDec/include/mpegFileRead.h b/libMpegTPDec/include/mpegFileRead.h deleted file mode 100644 index 1fbfb58..0000000 --- a/libMpegTPDec/include/mpegFileRead.h +++ /dev/null @@ -1,194 +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" - -#define MPFREAD_MP4FF_DISABLE - -#ifndef MPFREAD_MP4FF_DISABLE - /*!< If MPFREAD_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 MPFREAD_MP4FF_ENABLE -#endif - -/* maximum number of layers which can be read */ -/* shall equal max number of layers read by iisisoff */ -#define FILEREAD_MAX_LAYERS (2) - -typedef struct STRUCT_FILEREAD *HANDLE_FILEREAD; - -#ifdef __cplusplus -extern "C" { -#endif - -/** - * \brief Open an MPEG audio file and try to detect its format. - * \param filename String of the filename to be opened. - * \param fileFormat Skip file format detection and use given format if fileFormat != FF_UNKNOWN. - Else store detected format into *fileFmt. - * \param transportType Skip transport type detection and use given format if transportType != TT_UNKNOWN. - Else store detected format into *fileFmt. - * \param conf Pointer to unsigned char to hold the AudioSpecificConfig of the input file, if - any (MPEG 4 file format). In case of RAW LATM it holds the StreamMuxConfig. - * \param confSize Pointer to an integer, where the length of the ASC or SMC (in case of RAW LATM) - is stored to. - * \return MPEG file read handle. - */ -HANDLE_FILEREAD mpegFileRead_Open( const char *filename, - FILE_FORMAT fileFormat, - TRANSPORT_TYPE transportType, - UCHAR *conf[], - UINT confSize[], - INT *noOfLayers - ); - -/** - * \brief Get the file format of the input file. - * \param hDataSrc MPEG file read handle. - * \return File format of the input file. - */ -FILE_FORMAT mpegFileRead_GetFileFormat(HANDLE_FILEREAD hDataSrc); - -/** - * \brief Get the transport type of the input file. - * \param hDataSrc MPEG file read handle. - * \return Transport type of the input file. - */ -TRANSPORT_TYPE mpegFileRead_GetTransportType(HANDLE_FILEREAD hDataSrc); - -/** - * \brief Read data from MPEG file. In case of packet file, read one packet, in case - * of streaming file with embedded synchronisation layer (LOAS/ADTS...), just - * fill the buffer. - * - * \param hMpegFile MPEG file read handle. - * \param inBuffer Pointer to input buffer. - * \param bufferSize Size of input buffer. - * \param bytesValid Number of bytes that were read. - * \return 0 on success, -1 if unsupported file format or file read error. - */ -int mpegFileRead_Read( HANDLE_FILEREAD hMpegFile, - UCHAR *inBuffer[], - UINT bufferSize, - UINT *bytesValid - ); - -/** - * \brief Seek in file from origin by given offset in frames. - * \param hMpegFile MPEG file read handle. - * \param origin If 0, the origin is the file beginning (absolute seek). - * If 1, the origin is the current position (relative seek). - * \param offset The amount of frames to seek from the given origin. - * \return 0 on sucess, -1 if offset < 0 or file read error. - */ -int mpegFileRead_seek( HANDLE_FILEREAD hMpegFile, - INT origin, - INT offset - ); - -/** - * \brief Get file position in percent. - * \param hMpegFile MPEG file read handle. - * \return File position in percent. - */ -int mpegFileRead_getPercent(HANDLE_FILEREAD hMpegFile); - - -/** - * \brief Close MPEG audio file. - * \param hMpegFile Mpeg file read handle. - * \return 0 on sucess. - */ -int mpegFileRead_Close(HANDLE_FILEREAD *hMpegFile); - -#ifdef __cplusplus -} -#endif diff --git a/libMpegTPDec/include/tp_data.h b/libMpegTPDec/include/tp_data.h index c6e89b5..b4ab802 100644 --- a/libMpegTPDec/include/tp_data.h +++ b/libMpegTPDec/include/tp_data.h @@ -1,74 +1,85 @@ - -/* ----------------------------------------------------------------------------------------------------------- +/* ----------------------------------------------------------------------------- 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. +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. +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: +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 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 +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. +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. +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." +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. +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. +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. +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 @@ -79,17 +90,18 @@ Am Wolfsmantel 33 www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ +----------------------------------------------------------------------------- */ + +/******************* MPEG transport format decoder library ********************* -/***************************** MPEG-4 AAC Decoder ************************** + Author(s): Manuel Jander - Author(s): Manuel Jander Description: MPEG Transport data tables -******************************************************************************/ +*******************************************************************************/ -#ifndef __TP_DATA_H__ -#define __TP_DATA_H__ +#ifndef TP_DATA_H +#define TP_DATA_H #include "machine_type.h" #include "FDK_audio.h" @@ -98,17 +110,35 @@ amm-info@iis.fraunhofer.de /* * 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 + +#define TP_USAC_MAX_SPEAKERS (24) + +#define TP_USAC_MAX_EXT_ELEMENTS ((24)) + +#define TP_USAC_MAX_ELEMENTS ((24) + TP_USAC_MAX_EXT_ELEMENTS) + +#define TP_USAC_MAX_CONFIG_LEN \ + 512 /* next power of two of maximum of escapedValue(hBs, 4, 4, 8) in \ + AudioPreRoll() (285) */ + +#define TPDEC_USAC_NUM_CONFIG_CHANGE_FRAMES \ + (1) /* Number of frames for config change in USAC */ + +enum { + TPDEC_FLUSH_OFF = 0, + TPDEC_RSV60_CFG_CHANGE_ATSC_FLUSH_ON = 1, + TPDEC_RSV60_DASH_IPF_ATSC_FLUSH_ON = 2, + TPDEC_USAC_DASH_IPF_FLUSH_ON = 3 +}; + +enum { + TPDEC_BUILD_UP_OFF = 0, + TPDEC_RSV60_BUILD_UP_ON = 1, + TPDEC_RSV60_BUILD_UP_ON_IN_BAND = 2, + TPDEC_USAC_BUILD_UP_ON = 3, + TPDEC_RSV60_BUILD_UP_IDLE = 4, + TPDEC_RSV60_BUILD_UP_IDLE_IN_BAND = 5 +}; /** * ProgramConfig struct. @@ -116,13 +146,12 @@ amm-info@iis.fraunhofer.de /* ISO/IEC 14496-3 4.4.1.1 Table 4.2 Program config element */ #define PC_FSB_CHANNELS_MAX 16 /* Front/Side/Back channels */ #define PC_LFE_CHANNELS_MAX 4 -#define PC_ASSOCDATA_MAX 8 -#define PC_CCEL_MAX 16 /* CC elements */ -#define PC_COMMENTLENGTH 256 +#define PC_ASSOCDATA_MAX 8 +#define PC_CCEL_MAX 16 /* CC elements */ +#define PC_COMMENTLENGTH 256 +#define PC_NUM_HEIGHT_LAYER 3 -typedef struct -{ -#ifdef TP_PCE_ENABLE +typedef struct { /* PCE bitstream elements: */ UCHAR ElementInstanceTag; UCHAR Profile; @@ -165,54 +194,50 @@ typedef struct 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 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 + 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_frameLengthFlag; + UINT m_dependsOnCoreCoder; + UINT m_coreCoderDelay; - UINT m_extensionFlag ; - UINT m_extensionFlag3 ; + 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 */ + ELDEXT_TERM = 0x0, /* Termination tag */ + ELDEXT_SAOC = 0x1, /* SAOC config */ + ELDEXT_LDSAC = 0x2, /* LD MPEG Surround config */ + ELDEXT_DOWNSCALEINFO = 0x3 /* ELD sample rate adaptation */ /* reserved */ } ASC_ELD_EXT_TYPE; @@ -220,103 +245,186 @@ typedef struct { UCHAR m_frameLengthFlag; UCHAR m_sbrPresentFlag; - UCHAR m_useLdQmfTimeAlign; /* Use LD-MPS QMF in SBR to achive time alignment */ + UCHAR + m_useLdQmfTimeAlign; /* Use LD-MPS QMF in SBR to achive time alignment */ UCHAR m_sbrSamplingRate; UCHAR m_sbrCrcFlag; + UINT m_downscaledSamplingFrequency; } CSEldSpecificConfig; -#endif /* TP_ELD_ENABLE */ +typedef struct { + USAC_EXT_ELEMENT_TYPE usacExtElementType; + USHORT usacExtElementConfigLength; + USHORT usacExtElementDefaultLength; + UCHAR usacExtElementPayloadFrag; + UCHAR usacExtElementHasAudioPreRoll; +} CSUsacExtElementConfig; +typedef struct { + MP4_ELEMENT_ID usacElementType; + UCHAR m_noiseFilling; + UCHAR m_harmonicSBR; + UCHAR m_interTes; + UCHAR m_pvc; + UCHAR m_stereoConfigIndex; + CSUsacExtElementConfig extElement; +} CSUsacElementConfig; +typedef struct { + UCHAR m_frameLengthFlag; + UCHAR m_coreSbrFrameLengthIndex; + UCHAR m_sbrRatioIndex; + UCHAR m_nUsacChannels; /* number of audio channels signaled in + UsacDecoderConfig() / rsv603daDecoderConfig() via + numElements and usacElementType */ + UCHAR m_channelConfigurationIndex; + UINT m_usacNumElements; + CSUsacElementConfig element[TP_USAC_MAX_ELEMENTS]; + + UCHAR numAudioChannels; + UCHAR m_usacConfigExtensionPresent; + UCHAR elementLengthPresent; + UCHAR UsacConfig[TP_USAC_MAX_CONFIG_LEN]; + USHORT UsacConfigBits; +} CSUsacConfig; /** * Audio configuration struct, suitable for encoder and decoder configuration. */ typedef struct { - /* XYZ Specific Data */ union { -#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 */ + CSGaSpecificConfig + m_gaSpecificConfig; /**< General audio specific configuration. */ + CSEldSpecificConfig m_eldSpecificConfig; /**< ELD specific configuration. */ + CSUsacConfig m_usacConfig; /**< USAC specific configuration */ } 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 */ + /* Common ASC parameters */ + CProgramConfig m_progrConfigElement; /**< Program configuration. */ + + AUDIO_OBJECT_TYPE m_aot; /**< Audio Object Type. */ + UINT m_samplingFrequency; /**< Samplerate. */ + UINT m_samplesPerFrame; /**< Amount of samples per frame. */ + UINT m_directMapping; /**< Document this please !! */ + + AUDIO_OBJECT_TYPE m_extensionAudioObjectType; /**< Audio object type */ + UINT m_extensionSamplingFrequency; /**< Samplerate */ + + SCHAR m_channelConfiguration; /**< Channel configuration index */ + + SCHAR m_epConfig; /**< Error protection index */ + SCHAR m_vcb11Flag; /**< aacSectionDataResilienceFlag */ + SCHAR m_rvlcFlag; /**< aacScalefactorDataResilienceFlag */ + SCHAR m_hcrFlag; /**< aacSpectralDataResilienceFlag */ + + SCHAR m_sbrPresentFlag; /**< Flag indicating the presence of SBR data in the + bitstream */ + SCHAR + m_psPresentFlag; /**< Flag indicating the presence of parametric stereo + data in the bitstream */ + UCHAR m_samplingFrequencyIndex; /**< Samplerate index */ + UCHAR m_extensionSamplingFrequencyIndex; /**< Samplerate index */ + SCHAR m_extensionChannelConfiguration; /**< Channel configuration index */ + + UCHAR + configMode; /**< The flag indicates if the callback shall work in memory + allocation mode or in config change detection mode */ + UCHAR AacConfigChanged; /**< The flag will be set if at least one aac config + parameter has changed that requires a memory + reconfiguration, otherwise it will be cleared */ + UCHAR SbrConfigChanged; /**< The flag will be set if at least one sbr config + parameter has changed that requires a memory + reconfiguration, otherwise it will be cleared */ + UCHAR SacConfigChanged; /**< The flag will be set if at least one sac config + parameter has changed that requires a memory + reconfiguration, otherwise it will be cleared */ + + UCHAR + config[TP_USAC_MAX_CONFIG_LEN]; /**< Configuration stored as bitstream */ + UINT configBits; /**< Configuration length in bits */ } CSAudioSpecificConfig; -typedef 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. */ +typedef struct { + SCHAR flushCnt; /**< Flush frame counter */ + UCHAR flushStatus; /**< Flag indicates flush mode: on|off */ + SCHAR buildUpCnt; /**< Build up frame counter */ + UCHAR buildUpStatus; /**< Flag indicates build up mode: on|off */ + UCHAR cfgChanged; /**< Flag indicates that the config changed and the decoder + needs to be initialized again via callback. Make sure + that memory is freed before initialization. */ + UCHAR contentChanged; /**< Flag indicates that the content changed i.e. a + right truncation occured before */ + UCHAR forceCfgChange; /**< Flag indicates if config change has to be forced + even if new config is the same */ +} CCtrlCFGChange; + +typedef INT (*cbUpdateConfig_t)(void *, const CSAudioSpecificConfig *, + const UCHAR configMode, UCHAR *configChanged); +typedef INT (*cbFreeMem_t)(void *, const CSAudioSpecificConfig *); +typedef INT (*cbCtrlCFGChange_t)(void *, const CCtrlCFGChange *); +typedef INT (*cbSsc_t)(void *, HANDLE_FDK_BITSTREAM, + const AUDIO_OBJECT_TYPE coreCodec, + const INT samplingRate, const INT stereoConfigIndex, + const INT coreSbrFrameLengthIndex, const INT configBytes, + const UCHAR configMode, UCHAR *configChanged); + +typedef INT (*cbSbr_t)(void *self, HANDLE_FDK_BITSTREAM hBs, + const INT sampleRateIn, const INT sampleRateOut, + const INT samplesPerFrame, + const AUDIO_OBJECT_TYPE coreCodec, + const MP4_ELEMENT_ID elementID, const INT elementIndex, + const UCHAR harmonicSbr, const UCHAR stereoConfigIndex, + const UCHAR configMode, UCHAR *configChanged, + const INT downscaleFactor); + +typedef INT (*cbUsac_t)(void *self, HANDLE_FDK_BITSTREAM hBs); + +typedef INT (*cbUniDrc_t)(void *self, HANDLE_FDK_BITSTREAM hBs, + const INT fullPayloadLength, const INT payloadType, + const INT subStreamIndex, const INT payloadStart, + const AUDIO_OBJECT_TYPE); + +typedef struct { + cbUpdateConfig_t cbUpdateConfig; /*!< Function pointer for Config change + notify callback. */ + void *cbUpdateConfigData; /*!< User data pointer for Config change notify + callback. */ + cbFreeMem_t cbFreeMem; /*!< Function pointer for free memory callback. */ + void *cbFreeMemData; /*!< User data pointer for free memory callback. */ + cbCtrlCFGChange_t cbCtrlCFGChange; /*!< Function pointer for config change + control callback. */ + void *cbCtrlCFGChangeData; /*!< User data pointer for config change control + callback. */ + cbSsc_t cbSsc; /*!< Function pointer for SSC parser callback. */ + void *cbSscData; /*!< User data pointer for SSC parser callback. */ + cbSbr_t cbSbr; /*!< Function pointer for SBR header parser callback. */ + void *cbSbrData; /*!< User data pointer for SBR header parser callback. */ + cbUsac_t cbUsac; + void *cbUsacData; + cbUniDrc_t cbUniDrc; /*!< Function pointer for uniDrcConfig and + loudnessInfoSet parser callback. */ + void *cbUniDrcData; /*!< User data pointer for uniDrcConfig and + loudnessInfoSet parser callback. */ } CSTpCallBacks; -static const UINT SamplingRateTable[] = -{ 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050, 16000, 12000, 11025, 8000, 7350, 0, 0, - 0 -}; +static const UINT SamplingRateTable[] = { + 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050, 16000, 12000, 11025, + 8000, 7350, 0, 0, 57600, 51200, 40000, 38400, 34150, 28800, 25600, + 20000, 19200, 17075, 14400, 12800, 9600, 0, 0, 0, 0}; -static inline -int getSamplingRateIndex( UINT samplingRate ) -{ - UINT sf_index, tableSize=sizeof(SamplingRateTable)/sizeof(UINT); +static inline int getSamplingRateIndex(UINT samplingRate, UINT nBits) { + UINT sf_index; + UINT tableSize = (1 << nBits) - 1; - for (sf_index=0; sf_index<tableSize; sf_index++) { - if( SamplingRateTable[sf_index] == samplingRate ) break; + for (sf_index = 0; sf_index < tableSize; sf_index++) { + if (SamplingRateTable[sf_index] == samplingRate) break; } - if (sf_index>tableSize-1) { - return tableSize-1; + if (sf_index > tableSize) { + return tableSize - 1; } return sf_index; @@ -325,26 +433,33 @@ int getSamplingRateIndex( UINT samplingRate ) /* * Get Channel count from channel configuration */ -static inline int getNumberOfTotalChannels(int channelConfig) -{ +static inline int getNumberOfTotalChannels(int channelConfig) { switch (channelConfig) { - case 1: case 2: case 3: - case 4: case 5: case 6: - return channelConfig; - case 7: case 12: case 14: - return 8; - case 11: - return 7; - default: - return 0; + case 1: + case 2: + case 3: + case 4: + case 5: + case 6: + return channelConfig; + case 7: + case 12: + case 14: + return 8; + case 11: + return 7; + case 13: + return 24; + default: + return 0; } } -static inline -int getNumberOfEffectiveChannels(const int channelConfig) -{ /* index: 0,1,2,3,4,5,6,7,8,9,10,11,12,13,14,15 */ - const int n[] = {0,1,2,3,4,5,5,7,0,0, 0, 6, 7, 0, 7, 0}; +static inline int getNumberOfEffectiveChannels( + const int + channelConfig) { /* index: 0,1,2,3,4,5,6,7,8,9,10,11,12,13,14,15 */ + const int n[] = {0, 1, 2, 3, 4, 5, 5, 7, 0, 0, 0, 6, 7, 22, 7, 0}; return n[channelConfig]; } -#endif /* __TP_DATA_H__ */ +#endif /* TP_DATA_H */ diff --git a/libMpegTPDec/include/tpdec_lib.h b/libMpegTPDec/include/tpdec_lib.h index 2ad397d..30e53c1 100644 --- a/libMpegTPDec/include/tpdec_lib.h +++ b/libMpegTPDec/include/tpdec_lib.h @@ -1,74 +1,85 @@ - -/* ----------------------------------------------------------------------------------------------------------- +/* ----------------------------------------------------------------------------- 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. +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. +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: +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 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 +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. +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. +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." +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. +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. +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. +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 @@ -79,114 +90,115 @@ Am Wolfsmantel 33 www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ +----------------------------------------------------------------------------- */ + +/******************* MPEG transport format decoder library ********************* -/************************** MPEG-4 Transport Decoder *********************** + Author(s): Manuel Jander - Author(s): Manuel Jander Description: MPEG Transport decoder -******************************************************************************/ +*******************************************************************************/ -#ifndef __TPDEC_LIB_H__ -#define __TPDEC_LIB_H__ +#ifndef TPDEC_LIB_H +#define TPDEC_LIB_H #include "tp_data.h" #include "FDK_bitstream.h" -#define TRANSPORTDEC_INBUF_SIZE ( 8192 ) /*!< Size is in bytes. - Set the transport input buffer size carefully and - assure that it fulfills the requirements of the - supported transport format(s). */ - typedef enum { - TRANSPORTDEC_OK = 0, /*!< All fine. */ + TRANSPORTDEC_OK = 0, /*!< All fine. */ /* Synchronization errors. Wait for new input data and try again. */ - tpdec_sync_error_start = 0x100, - TRANSPORTDEC_NOT_ENOUGH_BITS, /*!< Out of bits. Provide more bits and try again. */ - TRANSPORTDEC_SYNC_ERROR, /*!< No sync was found or sync got lost. Keep trying. */ + tpdec_sync_error_start = 0x100, + TRANSPORTDEC_NOT_ENOUGH_BITS, /*!< Out of bits. Provide more bits and try + again. */ + TRANSPORTDEC_SYNC_ERROR, /*!< No sync was found or sync got lost. Keep trying. + */ tpdec_sync_error_end, /* Decode errors. Mostly caused due to bit errors. */ tpdec_decode_error_start = 0x400, - TRANSPORTDEC_PARSE_ERROR, /*!< Bitstream data showed inconsistencies (wrong syntax). */ - TRANSPORTDEC_UNSUPPORTED_FORMAT, /*!< Unsupported format or feature found in the bitstream data. */ - TRANSPORTDEC_CRC_ERROR, /*!< CRC error encountered in bitstream data. */ + TRANSPORTDEC_PARSE_ERROR, /*!< Bitstream data showed inconsistencies (wrong + syntax). */ + TRANSPORTDEC_UNSUPPORTED_FORMAT, /*!< Unsupported format or feature found in + the bitstream data. */ + TRANSPORTDEC_CRC_ERROR, /*!< CRC error encountered in bitstream data. */ tpdec_decode_error_end, /* Fatal errors. Stop immediately on one of these errors! */ - tpdec_fatal_error_start = 0x200, - TRANSPORTDEC_UNKOWN_ERROR, /*!< An unknown error occured. */ - TRANSPORTDEC_INVALID_PARAMETER, /*!< An invalid parameter was passed to a function. */ - TRANSPORTDEC_NEED_TO_RESTART, /*!< The decoder needs to be restarted, since the requiered - configuration change cannot be performed. */ + tpdec_fatal_error_start = 0x200, + TRANSPORTDEC_UNKOWN_ERROR, /*!< An unknown error occured. */ + TRANSPORTDEC_INVALID_PARAMETER, /*!< An invalid parameter was passed to a + function. */ + TRANSPORTDEC_NEED_TO_RESTART, /*!< The decoder needs to be restarted, since + the requiered configuration change cannot + be performed. */ + TRANSPORTDEC_TOO_MANY_BITS, /*!< In case of packet based formats: Supplied + number of bits exceed the size of the + internal bit buffer. */ tpdec_fatal_error_end } TRANSPORTDEC_ERROR; - /** Macro to identify decode errors. */ -#define TPDEC_IS_DECODE_ERROR(err) ( ((err>=tpdec_decode_error_start) && (err<=tpdec_decode_error_end)) ? 1 : 0) +#define TPDEC_IS_DECODE_ERROR(err) \ + (((err >= tpdec_decode_error_start) && (err <= tpdec_decode_error_end)) ? 1 \ + : 0) /** Macro to identify fatal errors. */ -#define TPDEC_IS_FATAL_ERROR(err) ( ((err>=tpdec_fatal_error_start) && (err<=tpdec_fatal_error_end)) ? 1 : 0) - +#define TPDEC_IS_FATAL_ERROR(err) \ + (((err >= tpdec_fatal_error_start) && (err <= tpdec_fatal_error_end)) ? 1 : 0) /** * \brief Parameter identifiers for transportDec_SetParam() */ typedef enum { - TPDEC_PARAM_MINIMIZE_DELAY = 1, /** Delay minimization strategy. 0: none, 1: discard as many frames as possible. */ - TPDEC_PARAM_EARLY_CONFIG, /** Enable early config discovery. */ - TPDEC_PARAM_IGNORE_BUFFERFULLNESS, /** Ignore buffer fullness. */ - TPDEC_PARAM_SET_BITRATE, /** Set average bit rate for bit stream interruption frame misses estimation. */ - TPDEC_PARAM_RESET, /** Reset transport decoder instance status. */ - TPDEC_PARAM_BURST_PERIOD /** Set data reception burst period in mili seconds. */ + TPDEC_PARAM_MINIMIZE_DELAY = 1, /** Delay minimization strategy. 0: none, 1: + discard as many frames as possible. */ + TPDEC_PARAM_EARLY_CONFIG, /** Enable early config discovery. */ + TPDEC_PARAM_IGNORE_BUFFERFULLNESS, /** Ignore buffer fullness. */ + TPDEC_PARAM_SET_BITRATE, /** Set average bit rate for bit stream interruption + frame misses estimation. */ + TPDEC_PARAM_RESET, /** Reset transport decoder instance status. */ + TPDEC_PARAM_BURST_PERIOD, /** Set data reception burst period in mili seconds. + */ + TPDEC_PARAM_TARGETLAYOUT, /** Set CICP target layout */ + TPDEC_PARAM_FORCE_CONFIG_CHANGE, /** Force config change for next received + config */ + TPDEC_PARAM_USE_ELEM_SKIPPING } TPDEC_PARAM; -/* ISO/IEC 14496-3 4.4.1.1 Table 4.2 Program config element */ -#define PC_FSB_CHANNELS_MAX 16 /* Front/Side/Back channels */ -#define PC_LFE_CHANNELS_MAX 4 -#define PC_ASSOCDATA_MAX 8 -#define PC_CCEL_MAX 16 /* CC elements */ -#define PC_COMMENTLENGTH 256 -#define PC_NUM_HEIGHT_LAYER 3 - - /*! \brief Reset Program Config Element. \param pPce Program Config Element structure. \return void */ -void CProgramConfig_Reset ( CProgramConfig *pPce ); +void CProgramConfig_Reset(CProgramConfig *pPce); /*! \brief Initialize Program Config Element. \param pPce Program Config Element structure. \return void */ -void CProgramConfig_Init ( CProgramConfig *pPce ); +void CProgramConfig_Init(CProgramConfig *pPce); /*! - \brief Inquire state of present Program Config Element structure. - \param pPce Program Config Element structure. - \return 1 if the PCE structure is filled correct, - 0 if no valid PCE present. + \brief Inquire state of present Program Config Element + structure. \param pPce Program Config Element structure. \return + 1 if the PCE structure is filled correct, 0 if no valid PCE present. */ -int CProgramConfig_IsValid ( const CProgramConfig *pPce ); +int CProgramConfig_IsValid(const CProgramConfig *pPce); -#ifdef TP_PCE_ENABLE /*! \brief Read Program Config Element. \param pPce Program Config Element structure. \param bs Bitstream buffer to read from. - \param alignAnchor Align bitstream to alignAnchor bits after all read operations. - \return void + \param alignAnchor Align bitstream to alignAnchor bits after all read + operations. \return void */ -void CProgramConfig_Read ( CProgramConfig *pPce, - HANDLE_FDK_BITSTREAM bs, - UINT alignAnchor ); +void CProgramConfig_Read(CProgramConfig *pPce, HANDLE_FDK_BITSTREAM bs, + UINT alignAnchor); /*! \brief Compare two Program Config Elements. @@ -194,50 +206,57 @@ void CProgramConfig_Read ( CProgramConfig *pPce, \param pPce2 Pointer to second Program Config Element structure. \return -1 if PCEs are completely different, 0 if PCEs are completely equal, - 1 if PCEs are different but have the same channel config, - 2 if PCEs have different channel config but same number of channels. + 1 if PCEs are different but have the same channel + config, 2 if PCEs have different channel config but same number of channels. */ -int CProgramConfig_Compare ( const CProgramConfig * const pPce1, - const CProgramConfig * const pPce2 ); +int CProgramConfig_Compare(const CProgramConfig *const pPce1, + const CProgramConfig *const pPce2); /*! - \brief Get a Program Config Element that matches the predefined MPEG-4 channel configurations 1-14. - \param pPce Program Config Element structure. - \param channelConfig MPEG-4 channel configuration. - \return void + \brief Get a Program Config Element that matches the predefined + MPEG-4 channel configurations 1-14. \param pPce Program Config + Element structure. \param channelConfig MPEG-4 channel configuration. \return + void */ -void CProgramConfig_GetDefault ( CProgramConfig *pPce, - const UINT channelConfig ); -#endif /* TP_PCE_ENABLE */ +void CProgramConfig_GetDefault(CProgramConfig *pPce, const UINT channelConfig); /** * \brief Lookup and verify a given element. The decoder calls this * method with every new element ID found in the bitstream. * * \param pPce A valid Program config structure. + * \param chConfig MPEG-4 channel configuration. * \param tag Tag of the current element to be looked up. * \param channelIdx The current channel count of the decoder parser. * \param chMapping Array to store the canonical channel mapping indexes. * \param chType Array to store the audio channel type. * \param chIndex Array to store the individual audio channel type index. + * \param chDescrLen Length of the output channel description array. * \param elMapping Pointer where the canonical element index is stored. * \param elType The element id of the current element to be looked up. * - * \return Non-zero if the element belongs to the current program, zero - * if it does not. - */ -int CProgramConfig_LookupElement( - CProgramConfig *pPce, - UINT channelConfig, - const UINT tag, - const UINT channelIdx, - UCHAR chMapping[], - AUDIO_CHANNEL_TYPE chType[], - UCHAR chIndex[], - UCHAR *elMapping, - MP4_ELEMENT_ID elList[], - MP4_ELEMENT_ID elType - ); + * \return Non-zero if the element belongs to the current program, + * zero if it does not. + */ +int CProgramConfig_LookupElement(CProgramConfig *pPce, UINT chConfig, + const UINT tag, const UINT channelIdx, + UCHAR chMapping[], AUDIO_CHANNEL_TYPE chType[], + UCHAR chIndex[], const UINT chDescrLen, + UCHAR *elMapping, MP4_ELEMENT_ID elList[], + MP4_ELEMENT_ID elType); + +/** + * \brief Get table of channel indices in the order of their + * appearance in by the program config field. + * \param pPce A valid program config structure. + * \param pceChMap Array to store the channel mapping indices like they + * appear in the PCE. + * \param pceChMapLen Lenght of the channel mapping index array (pceChMap). + * + * \return Non-zero if any error occured otherwise zero. + */ +int CProgramConfig_GetPceChMap(const CProgramConfig *pPce, UCHAR pceChMap[], + const UINT pceChMapLen); /** * \brief Get table of elements in canonical order from a @@ -250,10 +269,22 @@ int CProgramConfig_LookupElement( * PCE. If none can be found it receives the value 0. * \return Total element count including all SCE, CPE and LFE. */ -int CProgramConfig_GetElementTable( const CProgramConfig *pPce, - MP4_ELEMENT_ID table[], - const INT elListSize, - UCHAR *pChMapIdx ); +int CProgramConfig_GetElementTable(const CProgramConfig *pPce, + MP4_ELEMENT_ID table[], const INT elListSize, + UCHAR *pChMapIdx); + +/** + * \brief Get channel description (type and index) for implicit + configurations (chConfig > 0) in MPEG canonical order. + * \param chConfig MPEG-4 channel configuration. + * \param chType Array to store the audio channel type. + * \param chIndex Array to store the individual audio channel type index. + * \return void + */ +void CProgramConfig_GetChannelDescription(const UINT chConfig, + const CProgramConfig *pPce, + AUDIO_CHANNEL_TYPE chType[], + UCHAR chIndex[]); /** * \brief Initialize a given AudioSpecificConfig structure. @@ -265,45 +296,50 @@ void AudioSpecificConfig_Init(CSAudioSpecificConfig *pAsc); /** * \brief Parse a AudioSpecificConfig from a given bitstream handle. * - * \param pAsc A pointer to an allocated CSAudioSpecificConfig struct. + * \param pAsc A pointer to an allocated + * CSAudioSpecificConfig struct. * \param hBs Bitstream handle. - * \param fExplicitBackwardCompatible Do explicit backward compatibility parsing if set (flag). + * \param fExplicitBackwardCompatible Do explicit backward compatibility + * parsing if set (flag). * \param cb pointer to structure holding callback information + * \param configMode Config modes: memory allocation mode or config change + * detection mode. + * \param configChanged Indicates a config change. + * \param m_aot in case of unequal AOT_NULL_OBJECT only the specific config is + * parsed. * * \return Total element count including all SCE, CPE and LFE. */ TRANSPORTDEC_ERROR AudioSpecificConfig_Parse( - CSAudioSpecificConfig *pAsc, - HANDLE_FDK_BITSTREAM hBs, - int fExplicitBackwardCompatible, - CSTpCallBacks *cb - ); + CSAudioSpecificConfig *pAsc, HANDLE_FDK_BITSTREAM hBs, + int fExplicitBackwardCompatible, CSTpCallBacks *cb, UCHAR configMode, + UCHAR configChanged, AUDIO_OBJECT_TYPE m_aot); /* CELP stuff */ -enum { - MPE = 0, - RPE = 1, - fs8KHz = 0, - fs16KHz = 1 -}; +enum { MPE = 0, RPE = 1, fs8KHz = 0, fs16KHz = 1 }; /* Defintion of flags that can be passed to transportDecOpen() */ #define TP_FLAG_MPEG4 1 /* Capability flags */ -#define CAPF_TPDEC_ADIF 0x00001000 /**< Flag indicating support for ADIF transport format. */ -#define CAPF_TPDEC_ADTS 0x00002000 /**< Flag indicating support for ADTS transport format. */ -#define CAPF_TPDEC_LOAS 0x00004000 /**< Flag indicating support for LOAS transport format. */ -#define CAPF_TPDEC_LATM 0x00008000 /**< Flag indicating support for LATM transport format. */ -#define CAPF_TPDEC_RAWPACKETS 0x00010000 /**< Flag indicating support for raw packets transport format. */ +#define CAPF_TPDEC_ADIF \ + 0x00001000 /**< Flag indicating support for ADIF transport format. */ +#define CAPF_TPDEC_ADTS \ + 0x00002000 /**< Flag indicating support for ADTS transport format. */ +#define CAPF_TPDEC_LOAS \ + 0x00004000 /**< Flag indicating support for LOAS transport format. */ +#define CAPF_TPDEC_LATM \ + 0x00008000 /**< Flag indicating support for LATM transport format. */ +#define CAPF_TPDEC_RAWPACKETS \ + 0x00010000 /**< Flag indicating support for raw packets transport format. */ typedef struct TRANSPORTDEC *HANDLE_TRANSPORTDEC; - /** - * \brief Configure Transport Decoder via a binary coded AudioSpecificConfig or StreamMuxConfig. - * The previously requested configuration callback will be called as well. The buffer conf - * must containt a SMC in case of LOAS/LATM transport format, and an ASC elseways. + * \brief Configure Transport Decoder via a binary coded AudioSpecificConfig or + * StreamMuxConfig. The previously requested configuration callback will be + * called as well. The buffer conf must containt a SMC in case of + * LOAS/LATM transport format, and an ASC elseways. * * \param hTp Handle of a transport decoder. * \param conf UCHAR buffer of the binary coded config (ASC or SMC). @@ -311,96 +347,174 @@ typedef struct TRANSPORTDEC *HANDLE_TRANSPORTDEC; * * \return Error code. */ -TRANSPORTDEC_ERROR transportDec_OutOfBandConfig( const HANDLE_TRANSPORTDEC hTp, - UCHAR *conf, - const UINT length, - const UINT layer ); +TRANSPORTDEC_ERROR transportDec_OutOfBandConfig(const HANDLE_TRANSPORTDEC hTp, + UCHAR *conf, const UINT length, + const UINT layer); + +/** + * \brief Configure Transport Decoder via a binary coded USAC/RSV603DA Config. + * The buffer newConfig contains a binary coded USAC/RSV603DA config of + * length newConfigLength bytes. If the new config and the previous config are + * different configChanged is set to 1 otherwise it is set to 0. + * + * \param hTp Handle of a transport decoder. + * \param newConfig buffer of the binary coded config. + * \param newConfigLength Length of new config in bytes. + * \param buildUpStatus Indicates build up status: off|on|idle. + * \param configChanged Indicates if config changed. + * \param layer Instance layer. + * + * \return Error code. + */ +TRANSPORTDEC_ERROR transportDec_InBandConfig( + const HANDLE_TRANSPORTDEC hTp, UCHAR *newConfig, const UINT newConfigLength, + const UCHAR buildUpStatus, UCHAR *configChanged, const UINT layer, + UCHAR *implicitExplicitCfgDiff); /** * \brief Open Transport medium for reading. * * \param transportDecFmt Format of the transport decoder medium to be accessed. - * \param flags Transport decoder flags. Currently only TP_FLAG_MPEG4, which signals a - * MPEG4 capable decoder (relevant for ADTS only). + * \param flags Transport decoder flags. Currently only TP_FLAG_MPEG4, + * which signals a MPEG4 capable decoder (relevant for ADTS only). * - * \return A pointer to a valid and allocated HANDLE_TRANSPORTDEC or a null pointer on failure. + * \return A pointer to a valid and allocated HANDLE_TRANSPORTDEC or a null + * pointer on failure. */ -HANDLE_TRANSPORTDEC transportDec_Open( TRANSPORT_TYPE transportDecFmt, - const UINT flags ); +HANDLE_TRANSPORTDEC transportDec_Open(TRANSPORT_TYPE transportDecFmt, + const UINT flags, const UINT nrOfLayer); /** * \brief Register configuration change callback. * \param hTp Handle of transport decoder. - * \param cbUpdateConfig Pointer to a callback function to handle audio config changes. - * \param user_data void pointer for user data passed to the callback as first parameter. + * \param cbUpdateConfig Pointer to a callback function to handle audio config + * changes. + * \param user_data void pointer for user data passed to the callback as + * first parameter. * \return 0 on success. */ -int transportDec_RegisterAscCallback ( - HANDLE_TRANSPORTDEC hTp, - const cbUpdateConfig_t cbUpdateConfig, - void* user_data ); +int transportDec_RegisterAscCallback(HANDLE_TRANSPORTDEC hTp, + const cbUpdateConfig_t cbUpdateConfig, + void *user_data); + +/** + * \brief Register free memory callback. + * \param hTp Handle of transport decoder. + * \param cbFreeMem Pointer to a callback function to free config dependent + * memory. + * \param user_data void pointer for user data passed to the callback as + * first parameter. + * \return 0 on success. + */ +int transportDec_RegisterFreeMemCallback(HANDLE_TRANSPORTDEC hTp, + const cbFreeMem_t cbFreeMem, + void *user_data); + +/** + * \brief Register config change control callback. + * \param hTp Handle of transport decoder. + * \param cbCtrlCFGChange Pointer to a callback function for config change + * control. + * \param user_data void pointer for user data passed to the callback as + * first parameter. + * \return 0 on success. + */ +int transportDec_RegisterCtrlCFGChangeCallback( + HANDLE_TRANSPORTDEC hTp, const cbCtrlCFGChange_t cbCtrlCFGChange, + void *user_data); /** * \brief Register SSC parser callback. * \param hTp Handle of transport decoder. * \param cbUpdateConfig Pointer to a callback function to handle SSC parsing. - * \param user_data void pointer for user data passed to the callback as first parameter. + * \param user_data void pointer for user data passed to the callback as + * first parameter. * \return 0 on success. */ -int transportDec_RegisterSscCallback ( - HANDLE_TRANSPORTDEC hTp, - const cbSsc_t cbSscParse, - void* user_data ); +int transportDec_RegisterSscCallback(HANDLE_TRANSPORTDEC hTp, + const cbSsc_t cbSscParse, void *user_data); /** * \brief Register SBR header parser callback. * \param hTp Handle of transport decoder. - * \param cbUpdateConfig Pointer to a callback function to handle SBR header parsing. - * \param user_data void pointer for user data passed to the callback as first parameter. - * \return 0 on success. + * \param cbUpdateConfig Pointer to a callback function to handle SBR header + * parsing. + * \param user_data void pointer for user data passed to the callback as + * first parameter. + * \return 0 on success. + */ +int transportDec_RegisterSbrCallback(HANDLE_TRANSPORTDEC hTpDec, + const cbSbr_t cbSbr, void *user_data); + +/** + * \brief Register USAC SC parser callback. + * \param hTp Handle of transport decoder. + * \param cbUpdateConfig Pointer to a callback function to handle USAC SC + * parsing. + * \param user_data void pointer for user data passed to the callback as + * first parameter. + * \return 0 on success. + */ +int transportDec_RegisterUsacCallback(HANDLE_TRANSPORTDEC hTpDec, + const cbUsac_t cbUsac, void *user_data); + +/** + * \brief Register uniDrcConfig and loudnessInfoSet parser + * callback. + * \param hTp Handle of transport decoder. + * \param cbUpdateConfig Pointer to a callback function to handle uniDrcConfig + * and loudnessInfoSet parsing. + * \param user_data void pointer for user data passed to the callback as + * first parameter. + * \return 0 on success. */ -int transportDec_RegisterSbrCallback( HANDLE_TRANSPORTDEC hTpDec, const cbSbr_t cbSbr, void* user_data); +int transportDec_RegisterUniDrcConfigCallback(HANDLE_TRANSPORTDEC hTpDec, + const cbUniDrc_t cbUniDrc, + void *user_data, + UINT *pLoudnessInfoSetPosition); /** - * \brief Fill internal input buffer with bitstream data from the external input buffer. - * The function only copies such data as long as the decoder-internal input buffer is not full. - * So it grabs whatever it can from pBuffer and returns information (bytesValid) so that at a - * subsequent call of %transportDec_FillData(), the right position in pBuffer can be determined to + * \brief Fill internal input buffer with bitstream data from the external input + * buffer. The function only copies such data as long as the decoder-internal + * input buffer is not full. So it grabs whatever it can from pBuffer and + * returns information (bytesValid) so that at a subsequent call of + * %transportDec_FillData(), the right position in pBuffer can be determined to * grab the next data. * * \param hTp Handle of transportDec. * \param pBuffer Pointer to external input buffer. - * \param bufferSize Size of external input buffer. This argument is required because decoder-internally - * we need the information to calculate the offset to pBuffer, where the next - * available data is, which is then fed into the decoder-internal buffer (as much - * as possible). Our example framework implementation fills the buffer at pBuffer - * again, once it contains no available valid bytes anymore (meaning bytesValid equal 0). - * \param bytesValid Number of bitstream bytes in the external bitstream buffer that have not yet been - * copied into the decoder's internal bitstream buffer by calling this function. - * The value is updated according to the amount of newly copied bytes. + * \param bufferSize Size of external input buffer. This argument is required + * because decoder-internally we need the information to calculate the offset to + * pBuffer, where the next available data is, which is then + * fed into the decoder-internal buffer (as much as + * possible). Our example framework implementation fills the + * buffer at pBuffer again, once it contains no available valid bytes anymore + * (meaning bytesValid equal 0). + * \param bytesValid Number of bitstream bytes in the external bitstream buffer + * that have not yet been copied into the decoder's internal bitstream buffer by + * calling this function. The value is updated according to + * the amount of newly copied bytes. * \param layer The layer the bitstream belongs to. * \return Error code. */ -TRANSPORTDEC_ERROR transportDec_FillData( - const HANDLE_TRANSPORTDEC hTp, - UCHAR *pBuffer, - const UINT bufferSize, - UINT *pBytesValid, - const INT layer ); +TRANSPORTDEC_ERROR transportDec_FillData(const HANDLE_TRANSPORTDEC hTp, + UCHAR *pBuffer, const UINT bufferSize, + UINT *pBytesValid, const INT layer); /** * \brief Get transportDec bitstream handle. * \param hTp Pointer to a transport decoder handle. * \return HANDLE_FDK_BITSTREAM bitstream handle. */ -HANDLE_FDK_BITSTREAM transportDec_GetBitstream ( const HANDLE_TRANSPORTDEC hTp, const UINT layer ); +HANDLE_FDK_BITSTREAM transportDec_GetBitstream(const HANDLE_TRANSPORTDEC hTp, + const UINT layer); /** * \brief Get transport format. * \param hTp Pointer to a transport decoder handle. * \return The transport format. */ -TRANSPORT_TYPE transportDec_GetFormat ( const HANDLE_TRANSPORTDEC hTp ); +TRANSPORT_TYPE transportDec_GetFormat(const HANDLE_TRANSPORTDEC hTp); /** * \brief Get the current buffer fullness value. @@ -409,60 +523,76 @@ TRANSPORT_TYPE transportDec_GetFormat ( const HANDLE_TRANSPORTDEC hTp ); * * \return Buffer fullness */ -INT transportDec_GetBufferFullness( const HANDLE_TRANSPORTDEC hTp ); +INT transportDec_GetBufferFullness(const HANDLE_TRANSPORTDEC hTp); /** * \brief Close and deallocate transportDec. * \param phTp Pointer to a previously allocated transport decoder handle. * \return void */ -void transportDec_Close ( HANDLE_TRANSPORTDEC *phTp ); +void transportDec_Close(HANDLE_TRANSPORTDEC *phTp); /** * \brief Read one access unit from the transportDec medium. * \param hTp Handle of transportDec. - * \param length On return, this value is overwritten with the actual access unit length in bits. - * Set to -1 if length is unknown. + * \param length On return, this value is overwritten with the actual access + * unit length in bits. Set to -1 if length is unknown. * \return Error code. */ -TRANSPORTDEC_ERROR transportDec_ReadAccessUnit ( const HANDLE_TRANSPORTDEC hTp, const UINT layer ); +TRANSPORTDEC_ERROR transportDec_ReadAccessUnit(const HANDLE_TRANSPORTDEC hTp, + const UINT layer); /** - * \brief Get the remaining amount of bits of the current access unit. The result - * can be below zero, meaning that too many bits have been read. + * \brief Get AudioSpecificConfig. + * \param hTp Handle of transportDec. + * \param layer Transport layer. + * \param asc Pointer to AudioSpecificConfig. + * \return Error code. + */ +TRANSPORTDEC_ERROR transportDec_GetAsc(const HANDLE_TRANSPORTDEC hTp, + const UINT layer, + CSAudioSpecificConfig *asc); + +/** + * \brief Get the remaining amount of bits of the current access unit. The + * result can be below zero, meaning that too many bits have been read. * \param hTp Handle of transportDec. * \return amount of remaining bits. */ -INT transportDec_GetAuBitsRemaining( const HANDLE_TRANSPORTDEC hTp, const UINT layer ); +INT transportDec_GetAuBitsRemaining(const HANDLE_TRANSPORTDEC hTp, + const UINT layer); /** * \brief Get the total amount of bits of the current access unit. * \param hTp Handle of transportDec. * \return amount of total bits. */ -INT transportDec_GetAuBitsTotal( const HANDLE_TRANSPORTDEC hTp, const UINT layer ); +INT transportDec_GetAuBitsTotal(const HANDLE_TRANSPORTDEC hTp, + const UINT layer); /** - * \brief This function is required to be called when the decoder has finished parsing - * one Access Unit for bitstream housekeeping. + * \brief This function is required to be called when the decoder has + * finished parsing one Access Unit for bitstream housekeeping. * \param hTp Transport Handle. * \return Error code. */ -TRANSPORTDEC_ERROR transportDec_EndAccessUnit ( const HANDLE_TRANSPORTDEC hTp ); +TRANSPORTDEC_ERROR transportDec_EndAccessUnit(const HANDLE_TRANSPORTDEC hTp); /** - * \brief Obtain the amount of missing access units if applicable in case of - * a bit stream synchronization error. Each time transportDec_ReadAccessUnit() - * returns TRANSPORTDEC_SYNC_ERROR this function can be called to retrieve an estimate - * of the amount of missing access units. This works only in case of constant average - * bit rate (has to be known) and if the parameter TPDEC_PARAM_SET_BITRATE has been set - * accordingly. + * \brief Obtain the amount of missing access units if applicable in case + * of a bit stream synchronization error. Each time + * transportDec_ReadAccessUnit() returns TRANSPORTDEC_SYNC_ERROR + * this function can be called to retrieve an estimate of the amount + * of missing access units. This works only in case of constant + * average bit rate (has to be known) and if the parameter + * TPDEC_PARAM_SET_BITRATE has been set accordingly. * \param hTp Transport Handle. - * \param pNAccessUnits pointer to a memory location where the estimated lost frame count will be stored into. + * \param pNAccessUnits pointer to a memory location where the estimated lost + * frame count will be stored into. * \return Error code. */ -TRANSPORTDEC_ERROR transportDec_GetMissingAccessUnitCount ( INT *pNAccessUnits, HANDLE_TRANSPORTDEC hTp ); - +TRANSPORTDEC_ERROR transportDec_GetMissingAccessUnitCount( + INT *pNAccessUnits, HANDLE_TRANSPORTDEC hTp); /** * \brief Set a given setting. @@ -471,35 +601,36 @@ TRANSPORTDEC_ERROR transportDec_GetMissingAccessUnitCount ( INT *pNAccessUnits, * \param value Value for the parameter to be changed. * \return Error code. */ -TRANSPORTDEC_ERROR transportDec_SetParam ( const HANDLE_TRANSPORTDEC hTp, - const TPDEC_PARAM param, - const INT value ); +TRANSPORTDEC_ERROR transportDec_SetParam(const HANDLE_TRANSPORTDEC hTp, + const TPDEC_PARAM param, + const INT value); /** * \brief Get number of subframes (for LATM or ADTS) * \param hTp Transport Handle. - * \return Number of ADTS/LATM subframes (return 1 for all other transport types). + * \return Number of ADTS/LATM subframes (return 1 for all other transport + * types). */ UINT transportDec_GetNrOfSubFrames(HANDLE_TRANSPORTDEC hTp); - /** * \brief Get info structure of transport decoder library. * \param info A pointer to an allocated LIB_INFO struct. * \return Error code. */ -TRANSPORTDEC_ERROR transportDec_GetLibInfo( LIB_INFO *info ); +TRANSPORTDEC_ERROR transportDec_GetLibInfo(LIB_INFO *info); /* ADTS CRC support */ /** * \brief Set current bitstream position as start of a new data region. * \param hTp Transport 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 transportDec_CrcEndReg(). + * \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 + * transportDec_CrcEndReg(). */ -int transportDec_CrcStartReg ( const HANDLE_TRANSPORTDEC hTp, - const INT mBits ); +int transportDec_CrcStartReg(const HANDLE_TRANSPORTDEC hTp, const INT mBits); /** * \brief Set end of data region. @@ -507,15 +638,27 @@ int transportDec_CrcStartReg ( const HANDLE_TRANSPORTDEC hTp, * \param reg Data region ID, opbtained from transportDec_CrcStartReg(). * \return void */ -void transportDec_CrcEndReg ( const HANDLE_TRANSPORTDEC hTp, - const INT reg ); +void transportDec_CrcEndReg(const HANDLE_TRANSPORTDEC hTp, const INT reg); /** - * \brief Calculate ADTS crc and check if it is correct. The ADTS checksum is held internally. + * \brief Calculate ADTS crc and check if it is correct. The ADTS checksum + * is held internally. * \param hTp Transport handle. - * \return Return TRANSPORTDEC_OK if the CRC is ok, or error if CRC is not correct. + * \return Return TRANSPORTDEC_OK if the CRC is ok, or error if CRC is not + * correct. */ -TRANSPORTDEC_ERROR transportDec_CrcCheck ( const HANDLE_TRANSPORTDEC hTp ); +TRANSPORTDEC_ERROR transportDec_CrcCheck(const HANDLE_TRANSPORTDEC hTp); +/** + * \brief Only check whether a given config seems to be valid without modifying + * internal states. + * + * \param conf UCHAR buffer of the binary coded config (SDC type 9). + * \param length The length in bytes of the conf buffer. + * + * \return Error code. + */ +TRANSPORTDEC_ERROR transportDec_DrmRawSdcAudioConfig_Check(UCHAR *conf, + const UINT length); -#endif /* #ifndef __TPDEC_LIB_H__ */ +#endif /* #ifndef TPDEC_LIB_H */ |