diff options
Diffstat (limited to 'fdk-aac/libMpegTPDec/include')
| -rw-r--r-- | fdk-aac/libMpegTPDec/include/tp_data.h | 466 | ||||
| -rw-r--r-- | fdk-aac/libMpegTPDec/include/tpdec_lib.h | 664 |
2 files changed, 1130 insertions, 0 deletions
diff --git a/fdk-aac/libMpegTPDec/include/tp_data.h b/fdk-aac/libMpegTPDec/include/tp_data.h new file mode 100644 index 0000000..b015332 --- /dev/null +++ b/fdk-aac/libMpegTPDec/include/tp_data.h @@ -0,0 +1,466 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/******************* MPEG transport format decoder library ********************* + + Author(s): Manuel Jander + + Description: MPEG Transport data tables + +*******************************************************************************/ + +#ifndef TP_DATA_H +#define TP_DATA_H + +#include "machine_type.h" +#include "FDK_audio.h" +#include "FDK_bitstream.h" + +/* + * Configuration + */ + +#define TP_USAC_MAX_SPEAKERS (24) + +#define TP_USAC_MAX_EXT_ELEMENTS ((24)) + +#define TP_USAC_MAX_ELEMENTS ((24) + TP_USAC_MAX_EXT_ELEMENTS) + +#define TP_USAC_MAX_CONFIG_LEN \ + 512 /* next power of two of maximum of escapedValue(hBs, 4, 4, 8) in \ + AudioPreRoll() (285) */ + +#define TPDEC_USAC_NUM_CONFIG_CHANGE_FRAMES \ + (1) /* Number of frames for config change in USAC */ + +enum { + TPDEC_FLUSH_OFF = 0, + TPDEC_RSV60_CFG_CHANGE_ATSC_FLUSH_ON = 1, + TPDEC_RSV60_DASH_IPF_ATSC_FLUSH_ON = 2, + TPDEC_USAC_DASH_IPF_FLUSH_ON = 3 +}; + +enum { + TPDEC_BUILD_UP_OFF = 0, + TPDEC_RSV60_BUILD_UP_ON = 1, + TPDEC_RSV60_BUILD_UP_ON_IN_BAND = 2, + TPDEC_USAC_BUILD_UP_ON = 3, + TPDEC_RSV60_BUILD_UP_IDLE = 4, + TPDEC_RSV60_BUILD_UP_IDLE_IN_BAND = 5 +}; + +/** + * ProgramConfig struct. + */ +/* ISO/IEC 14496-3 4.4.1.1 Table 4.2 Program config element */ +#define PC_FSB_CHANNELS_MAX 16 /* Front/Side/Back channels */ +#define PC_LFE_CHANNELS_MAX 4 +#define PC_ASSOCDATA_MAX 8 +#define PC_CCEL_MAX 16 /* CC elements */ +#define PC_COMMENTLENGTH 256 +#define PC_NUM_HEIGHT_LAYER 3 + +typedef struct { + /* PCE bitstream elements: */ + UCHAR ElementInstanceTag; + UCHAR Profile; + UCHAR SamplingFrequencyIndex; + UCHAR NumFrontChannelElements; + UCHAR NumSideChannelElements; + UCHAR NumBackChannelElements; + UCHAR NumLfeChannelElements; + UCHAR NumAssocDataElements; + UCHAR NumValidCcElements; + + UCHAR MonoMixdownPresent; + UCHAR MonoMixdownElementNumber; + + UCHAR StereoMixdownPresent; + UCHAR StereoMixdownElementNumber; + + UCHAR MatrixMixdownIndexPresent; + UCHAR MatrixMixdownIndex; + UCHAR PseudoSurroundEnable; + + UCHAR FrontElementIsCpe[PC_FSB_CHANNELS_MAX]; + UCHAR FrontElementTagSelect[PC_FSB_CHANNELS_MAX]; + UCHAR FrontElementHeightInfo[PC_FSB_CHANNELS_MAX]; + + UCHAR SideElementIsCpe[PC_FSB_CHANNELS_MAX]; + UCHAR SideElementTagSelect[PC_FSB_CHANNELS_MAX]; + UCHAR SideElementHeightInfo[PC_FSB_CHANNELS_MAX]; + + UCHAR BackElementIsCpe[PC_FSB_CHANNELS_MAX]; + UCHAR BackElementTagSelect[PC_FSB_CHANNELS_MAX]; + UCHAR BackElementHeightInfo[PC_FSB_CHANNELS_MAX]; + + UCHAR LfeElementTagSelect[PC_LFE_CHANNELS_MAX]; + + UCHAR AssocDataElementTagSelect[PC_ASSOCDATA_MAX]; + + UCHAR CcElementIsIndSw[PC_CCEL_MAX]; + UCHAR ValidCcElementTagSelect[PC_CCEL_MAX]; + + UCHAR CommentFieldBytes; + UCHAR Comment[PC_COMMENTLENGTH]; + + /* Helper variables for administration: */ + UCHAR isValid; /*!< Flag showing if PCE has been read successfully. */ + UCHAR + NumChannels; /*!< Amount of audio channels summing all channel elements + including LFEs */ + UCHAR NumEffectiveChannels; /*!< Amount of audio channels summing only SCEs + and CPEs */ + UCHAR elCounter; + +} CProgramConfig; + +typedef enum { + ASCEXT_UNKOWN = -1, + ASCEXT_SBR = 0x2b7, + ASCEXT_PS = 0x548, + ASCEXT_MPS = 0x76a, + ASCEXT_SAOC = 0x7cb, + ASCEXT_LDMPS = 0x7cc + +} TP_ASC_EXTENSION_ID; + +/** + * GaSpecificConfig struct + */ +typedef struct { + UINT m_frameLengthFlag; + UINT m_dependsOnCoreCoder; + UINT m_coreCoderDelay; + + UINT m_extensionFlag; + UINT m_extensionFlag3; + + UINT m_layer; + UINT m_numOfSubFrame; + UINT m_layerLength; + +} CSGaSpecificConfig; + +typedef enum { + ELDEXT_TERM = 0x0, /* Termination tag */ + ELDEXT_SAOC = 0x1, /* SAOC config */ + ELDEXT_LDSAC = 0x2, /* LD MPEG Surround config */ + ELDEXT_DOWNSCALEINFO = 0x3 /* ELD sample rate adaptation */ + /* reserved */ +} ASC_ELD_EXT_TYPE; + +typedef struct { + UCHAR m_frameLengthFlag; + + UCHAR m_sbrPresentFlag; + UCHAR + m_useLdQmfTimeAlign; /* Use LD-MPS QMF in SBR to achive time alignment */ + UCHAR m_sbrSamplingRate; + UCHAR m_sbrCrcFlag; + UINT m_downscaledSamplingFrequency; + +} CSEldSpecificConfig; + +typedef struct { + USAC_EXT_ELEMENT_TYPE usacExtElementType; + USHORT usacExtElementConfigLength; + USHORT usacExtElementDefaultLength; + UCHAR usacExtElementPayloadFrag; + UCHAR usacExtElementHasAudioPreRoll; +} CSUsacExtElementConfig; + +typedef struct { + MP4_ELEMENT_ID usacElementType; + UCHAR m_noiseFilling; + UCHAR m_harmonicSBR; + UCHAR m_interTes; + UCHAR m_pvc; + UCHAR m_stereoConfigIndex; + CSUsacExtElementConfig extElement; +} CSUsacElementConfig; + +typedef struct { + UCHAR m_frameLengthFlag; + UCHAR m_coreSbrFrameLengthIndex; + UCHAR m_sbrRatioIndex; + UCHAR m_nUsacChannels; /* number of audio channels signaled in + UsacDecoderConfig() / rsv603daDecoderConfig() via + numElements and usacElementType */ + UCHAR m_channelConfigurationIndex; + UINT m_usacNumElements; + CSUsacElementConfig element[TP_USAC_MAX_ELEMENTS]; + + UCHAR numAudioChannels; + UCHAR m_usacConfigExtensionPresent; + UCHAR elementLengthPresent; + UCHAR UsacConfig[TP_USAC_MAX_CONFIG_LEN]; + USHORT UsacConfigBits; +} CSUsacConfig; + +/** + * Audio configuration struct, suitable for encoder and decoder configuration. + */ +typedef struct { + /* XYZ Specific Data */ + union { + CSGaSpecificConfig + m_gaSpecificConfig; /**< General audio specific configuration. */ + CSEldSpecificConfig m_eldSpecificConfig; /**< ELD specific configuration. */ + CSUsacConfig m_usacConfig; /**< USAC specific configuration */ + } m_sc; + + /* Common ASC parameters */ + CProgramConfig m_progrConfigElement; /**< Program configuration. */ + + AUDIO_OBJECT_TYPE m_aot; /**< Audio Object Type. */ + UINT m_samplingFrequency; /**< Samplerate. */ + UINT m_samplesPerFrame; /**< Amount of samples per frame. */ + UINT m_directMapping; /**< Document this please !! */ + + AUDIO_OBJECT_TYPE m_extensionAudioObjectType; /**< Audio object type */ + UINT m_extensionSamplingFrequency; /**< Samplerate */ + + SCHAR m_channelConfiguration; /**< Channel configuration index */ + + SCHAR m_epConfig; /**< Error protection index */ + SCHAR m_vcb11Flag; /**< aacSectionDataResilienceFlag */ + SCHAR m_rvlcFlag; /**< aacScalefactorDataResilienceFlag */ + SCHAR m_hcrFlag; /**< aacSpectralDataResilienceFlag */ + + SCHAR m_sbrPresentFlag; /**< Flag indicating the presence of SBR data in the + bitstream */ + SCHAR + m_psPresentFlag; /**< Flag indicating the presence of parametric stereo + data in the bitstream */ + UCHAR m_samplingFrequencyIndex; /**< Samplerate index */ + UCHAR m_extensionSamplingFrequencyIndex; /**< Samplerate index */ + SCHAR m_extensionChannelConfiguration; /**< Channel configuration index */ + + UCHAR + configMode; /**< The flag indicates if the callback shall work in memory + allocation mode or in config change detection mode */ + UCHAR AacConfigChanged; /**< The flag will be set if at least one aac config + parameter has changed that requires a memory + reconfiguration, otherwise it will be cleared */ + UCHAR SbrConfigChanged; /**< The flag will be set if at least one sbr config + parameter has changed that requires a memory + reconfiguration, otherwise it will be cleared */ + UCHAR SacConfigChanged; /**< The flag will be set if at least one sac config + parameter has changed that requires a memory + reconfiguration, otherwise it will be cleared */ + + UCHAR + config[TP_USAC_MAX_CONFIG_LEN]; /**< Configuration stored as bitstream */ + UINT configBits; /**< Configuration length in bits */ + +} CSAudioSpecificConfig; + +typedef struct { + SCHAR flushCnt; /**< Flush frame counter */ + UCHAR flushStatus; /**< Flag indicates flush mode: on|off */ + SCHAR buildUpCnt; /**< Build up frame counter */ + UCHAR buildUpStatus; /**< Flag indicates build up mode: on|off */ + UCHAR cfgChanged; /**< Flag indicates that the config changed and the decoder + needs to be initialized again via callback. Make sure + that memory is freed before initialization. */ + UCHAR contentChanged; /**< Flag indicates that the content changed i.e. a + right truncation occured before */ + UCHAR forceCfgChange; /**< Flag indicates if config change has to be forced + even if new config is the same */ +} CCtrlCFGChange; + +typedef INT (*cbUpdateConfig_t)(void *, const CSAudioSpecificConfig *, + const UCHAR configMode, UCHAR *configChanged); +typedef INT (*cbFreeMem_t)(void *, const CSAudioSpecificConfig *); +typedef INT (*cbCtrlCFGChange_t)(void *, const CCtrlCFGChange *); +typedef INT (*cbSsc_t)(void *, HANDLE_FDK_BITSTREAM, + const AUDIO_OBJECT_TYPE coreCodec, + const INT samplingRate, const INT frameSize, + const INT stereoConfigIndex, + const INT coreSbrFrameLengthIndex, const INT configBytes, + const UCHAR configMode, UCHAR *configChanged); + +typedef INT (*cbSbr_t)(void *self, HANDLE_FDK_BITSTREAM hBs, + const INT sampleRateIn, const INT sampleRateOut, + const INT samplesPerFrame, + const AUDIO_OBJECT_TYPE coreCodec, + const MP4_ELEMENT_ID elementID, const INT elementIndex, + const UCHAR harmonicSbr, const UCHAR stereoConfigIndex, + const UCHAR configMode, UCHAR *configChanged, + const INT downscaleFactor); + +typedef INT (*cbUsac_t)(void *self, HANDLE_FDK_BITSTREAM hBs); + +typedef INT (*cbUniDrc_t)(void *self, HANDLE_FDK_BITSTREAM hBs, + const INT fullPayloadLength, const INT payloadType, + const INT subStreamIndex, const INT payloadStart, + const AUDIO_OBJECT_TYPE); + +typedef struct { + cbUpdateConfig_t cbUpdateConfig; /*!< Function pointer for Config change + notify callback. */ + void *cbUpdateConfigData; /*!< User data pointer for Config change notify + callback. */ + cbFreeMem_t cbFreeMem; /*!< Function pointer for free memory callback. */ + void *cbFreeMemData; /*!< User data pointer for free memory callback. */ + cbCtrlCFGChange_t cbCtrlCFGChange; /*!< Function pointer for config change + control callback. */ + void *cbCtrlCFGChangeData; /*!< User data pointer for config change control + callback. */ + cbSsc_t cbSsc; /*!< Function pointer for SSC parser callback. */ + void *cbSscData; /*!< User data pointer for SSC parser callback. */ + cbSbr_t cbSbr; /*!< Function pointer for SBR header parser callback. */ + void *cbSbrData; /*!< User data pointer for SBR header parser callback. */ + cbUsac_t cbUsac; + void *cbUsacData; + cbUniDrc_t cbUniDrc; /*!< Function pointer for uniDrcConfig and + loudnessInfoSet parser callback. */ + void *cbUniDrcData; /*!< User data pointer for uniDrcConfig and + loudnessInfoSet parser callback. */ +} CSTpCallBacks; + +static const UINT SamplingRateTable[] = { + 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050, 16000, 12000, 11025, + 8000, 7350, 0, 0, 57600, 51200, 40000, 38400, 34150, 28800, 25600, + 20000, 19200, 17075, 14400, 12800, 9600, 0, 0, 0, 0}; + +static inline int getSamplingRateIndex(UINT samplingRate, UINT nBits) { + UINT sf_index; + UINT tableSize = (1 << nBits) - 1; + + for (sf_index = 0; sf_index < tableSize; sf_index++) { + if (SamplingRateTable[sf_index] == samplingRate) break; + } + + if (sf_index > tableSize) { + return tableSize - 1; + } + + return sf_index; +} + +/* + * Get Channel count from channel configuration + */ +static inline int getNumberOfTotalChannels(int channelConfig) { + switch (channelConfig) { + case 1: + case 2: + case 3: + case 4: + case 5: + case 6: + return channelConfig; + case 7: + case 12: + case 14: + return 8; + case 11: + return 7; + case 13: + return 24; + default: + return 0; + } +} + +static inline int getNumberOfEffectiveChannels( + const int + channelConfig) { /* index: 0,1,2,3,4,5,6,7,8,9,10,11,12,13,14,15 */ + const int n[] = {0, 1, 2, 3, 4, 5, 5, 7, 0, 0, 0, 6, 7, 22, 7, 0}; + return n[channelConfig]; +} + +#endif /* TP_DATA_H */ diff --git a/fdk-aac/libMpegTPDec/include/tpdec_lib.h b/fdk-aac/libMpegTPDec/include/tpdec_lib.h new file mode 100644 index 0000000..30e53c1 --- /dev/null +++ b/fdk-aac/libMpegTPDec/include/tpdec_lib.h @@ -0,0 +1,664 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/******************* MPEG transport format decoder library ********************* + + Author(s): Manuel Jander + + Description: MPEG Transport decoder + +*******************************************************************************/ + +#ifndef TPDEC_LIB_H +#define TPDEC_LIB_H + +#include "tp_data.h" + +#include "FDK_bitstream.h" + +typedef enum { + 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_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. */ + 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. */ + 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) +/** Macro to identify fatal errors. */ +#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_TARGETLAYOUT, /** Set CICP target layout */ + TPDEC_PARAM_FORCE_CONFIG_CHANGE, /** Force config change for next received + config */ + TPDEC_PARAM_USE_ELEM_SKIPPING +} TPDEC_PARAM; + +/*! + \brief Reset Program Config Element. + \param pPce Program Config Element structure. + \return void +*/ +void CProgramConfig_Reset(CProgramConfig *pPce); + +/*! + \brief Initialize Program Config Element. + \param pPce Program Config Element structure. + \return void +*/ +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. +*/ +int CProgramConfig_IsValid(const CProgramConfig *pPce); + +/*! + \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 +*/ +void CProgramConfig_Read(CProgramConfig *pPce, HANDLE_FDK_BITSTREAM bs, + UINT alignAnchor); + +/*! + \brief Compare two Program Config Elements. + \param pPce1 Pointer to first Program Config Element structure. + \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. +*/ +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 +*/ +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 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 + * give program config field. + * \param pPce A valid program config structure. + * \param table An array where the element IDs are stored. + * \param elListSize The length of the table array. + * \param pChMapIdx Pointer to a field receiving the corresponding + * implicit channel configuration index of the given + * 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); + +/** + * \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. + * \param pAsc A pointer to an allocated CSAudioSpecificConfig struct. + * \return void + */ +void AudioSpecificConfig_Init(CSAudioSpecificConfig *pAsc); + +/** + * \brief Parse a AudioSpecificConfig from a given bitstream handle. + * + * \param pAsc A pointer to an allocated + * CSAudioSpecificConfig struct. + * \param hBs Bitstream handle. + * \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, UCHAR configMode, + UCHAR configChanged, AUDIO_OBJECT_TYPE m_aot); + +/* CELP stuff */ +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. */ + +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. + * + * \param hTp Handle of a transport decoder. + * \param conf UCHAR buffer of the binary coded config (ASC or SMC). + * \param length The length in bytes of the conf buffer. + * + * \return Error code. + */ +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). + * + * \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, 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. + * \return 0 on success. + */ +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. + * \return 0 on success. + */ +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. + */ +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_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 + * 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 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); + +/** + * \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); + +/** + * \brief Get transport format. + * \param hTp Pointer to a transport decoder handle. + * \return The transport format. + */ +TRANSPORT_TYPE transportDec_GetFormat(const HANDLE_TRANSPORTDEC hTp); + +/** + * \brief Get the current buffer fullness value. + * + * \param hTp Handle of a transport decoder. + * + * \return Buffer fullness + */ +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); + +/** + * \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. + * \return Error code. + */ +TRANSPORTDEC_ERROR transportDec_ReadAccessUnit(const HANDLE_TRANSPORTDEC hTp, + const UINT layer); + +/** + * \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); + +/** + * \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); + +/** + * \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); + +/** + * \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. + * \return Error code. + */ +TRANSPORTDEC_ERROR transportDec_GetMissingAccessUnitCount( + INT *pNAccessUnits, HANDLE_TRANSPORTDEC hTp); + +/** + * \brief Set a given setting. + * \param hTp Transport Handle. + * \param param Identifier of the parameter to be changed. + * \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); + +/** + * \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). + */ +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); + +/* 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(). + */ +int transportDec_CrcStartReg(const HANDLE_TRANSPORTDEC hTp, const INT mBits); + +/** + * \brief Set end of data region. + * \param hTp Transport handle. + * \param reg Data region ID, opbtained from transportDec_CrcStartReg(). + * \return void + */ +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. + * \param hTp Transport handle. + * \return Return TRANSPORTDEC_OK if the CRC is ok, or error if CRC is not + * correct. + */ +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 */ |