| /* ----------------------------------------------------------------------------- |
| 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 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 */ |
