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Diffstat (limited to 'fdk-aac/libAACdec/src/aacdecoder.h')
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diff --git a/fdk-aac/libAACdec/src/aacdecoder.h b/fdk-aac/libAACdec/src/aacdecoder.h new file mode 100644 index 0000000..20f4c45 --- /dev/null +++ b/fdk-aac/libAACdec/src/aacdecoder.h @@ -0,0 +1,465 @@ +/* ----------------------------------------------------------------------------- +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 +----------------------------------------------------------------------------- */ + +/**************************** AAC decoder library ****************************** + + Author(s): Josef Hoepfl + + Description: + +*******************************************************************************/ + +#ifndef AACDECODER_H +#define AACDECODER_H + +#include "common_fix.h" + +#include "FDK_bitstream.h" + +#include "channel.h" + +#include "tpdec_lib.h" +#include "FDK_audio.h" + +#include "block.h" + +#include "genericStds.h" + +#include "FDK_qmf_domain.h" + +#include "sbrdecoder.h" + +#include "aacdec_drc.h" + +#include "pcmdmx_lib.h" + +#include "FDK_drcDecLib.h" + +#include "limiter.h" + +#include "FDK_delay.h" + +#define TIME_DATA_FLUSH_SIZE (128) +#define TIME_DATA_FLUSH_SIZE_SF (7) + +#define AACDEC_MAX_NUM_PREROLL_AU_USAC (3) +#if (AACDEC_MAX_NUM_PREROLL_AU < 3) +#undef AACDEC_MAX_NUM_PREROLL_AU +#define AACDEC_MAX_NUM_PREROLL_AU AACDEC_MAX_NUM_PREROLL_AU_USAC +#endif + +typedef struct AAC_DECODER_INSTANCE *HANDLE_AACDECODER; + +enum { L = 0, R = 1 }; + +typedef struct { + unsigned char *buffer; + int bufferSize; + int offset[8]; + int nrElements; +} CAncData; + +typedef enum { NOT_DEFINED = -1, MODE_HQ = 0, MODE_LP = 1 } QMF_MODE; + +typedef struct { + int bsDelay; +} SBR_PARAMS; + +enum { + AACDEC_FLUSH_OFF = 0, + AACDEC_RSV60_CFG_CHANGE_ATSC_FLUSH_ON = 1, + AACDEC_RSV60_DASH_IPF_ATSC_FLUSH_ON = 2, + AACDEC_USAC_DASH_IPF_FLUSH_ON = 3 +}; + +enum { + AACDEC_BUILD_UP_OFF = 0, + AACDEC_RSV60_BUILD_UP_ON = 1, + AACDEC_RSV60_BUILD_UP_ON_IN_BAND = 2, + AACDEC_USAC_BUILD_UP_ON = 3, + AACDEC_RSV60_BUILD_UP_IDLE = 4, + AACDEC_RSV60_BUILD_UP_IDLE_IN_BAND = 5 +}; + +typedef struct { + /* Usac Extension Elements */ + USAC_EXT_ELEMENT_TYPE usacExtElementType[(3)]; + UINT usacExtElementDefaultLength[(3)]; + UCHAR usacExtElementPayloadFrag[(3)]; +} CUsacCoreExtensions; + +/* AAC decoder (opaque toward userland) struct declaration */ +struct AAC_DECODER_INSTANCE { + INT aacChannels; /*!< Amount of AAC decoder channels allocated. */ + INT ascChannels[(1 * + 1)]; /*!< Amount of AAC decoder channels signalled in ASC. */ + INT blockNumber; /*!< frame counter */ + + INT nrOfLayers; + + INT outputInterleaved; /*!< PCM output format (interleaved/none interleaved). + */ + + HANDLE_TRANSPORTDEC hInput; /*!< Transport layer handle. */ + + SamplingRateInfo + samplingRateInfo[(1 * 1)]; /*!< Sampling Rate information table */ + + UCHAR + frameOK; /*!< Will be unset if a consistency check, e.g. CRC etc. fails */ + + UINT flags[(1 * 1)]; /*!< Flags for internal decoder use. DO NOT USE + self::streaminfo::flags ! */ + UINT elFlags[(3 * ((8) * 2) + (((8) * 2)) / 2 + 4 * (1) + + 1)]; /*!< Flags for internal decoder use (element specific). DO + NOT USE self::streaminfo::flags ! */ + + MP4_ELEMENT_ID elements[(3 * ((8) * 2) + (((8) * 2)) / 2 + 4 * (1) + + 1)]; /*!< Table where the element Id's are listed */ + UCHAR elTags[(3 * ((8) * 2) + (((8) * 2)) / 2 + 4 * (1) + + 1)]; /*!< Table where the elements id Tags are listed */ + UCHAR chMapping[((8) * 2)]; /*!< Table of MPEG canonical order to bitstream + channel order mapping. */ + + AUDIO_CHANNEL_TYPE channelType[(8)]; /*!< Audio channel type of each output + audio channel (from 0 upto + numChannels). */ + UCHAR channelIndices[(8)]; /*!< Audio channel index for each output audio + channel (from 0 upto numChannels). */ + /* See ISO/IEC 13818-7:2005(E), 8.5.3.2 Explicit channel mapping using a + * program_config_element() */ + + FDK_channelMapDescr mapDescr; /*!< Describes the output channel mapping. */ + UCHAR chMapIndex; /*!< Index to access one line of the channelOutputMapping + table. This is required because not all 8 channel + configurations have the same output mapping. */ + INT sbrDataLen; /*!< Expected length of the SBR remaining in bitbuffer after + the AAC payload has been pared. */ + + CProgramConfig pce; + CStreamInfo + streamInfo; /*!< Pointer to StreamInfo data (read from the bitstream) */ + CAacDecoderChannelInfo + *pAacDecoderChannelInfo[(8)]; /*!< Temporal channel memory */ + CAacDecoderStaticChannelInfo + *pAacDecoderStaticChannelInfo[(8)]; /*!< Persistent channel memory */ + + FIXP_DBL *workBufferCore2; + PCM_DEC *pTimeData2; + INT timeData2Size; + + CpePersistentData *cpeStaticData[( + 3 * ((8) * 2) + (((8) * 2)) / 2 + 4 * (1) + + 1)]; /*!< Pointer to persistent data shared by both channels of a CPE. +This structure is allocated once for each CPE. */ + + CConcealParams concealCommonData; + CConcealmentMethod concealMethodUser; + + CUsacCoreExtensions usacCoreExt; /*!< Data and handles to extend USAC FD/LPD + core decoder (SBR, MPS, ...) */ + UINT numUsacElements[(1 * 1)]; + UCHAR usacStereoConfigIndex[(3 * ((8) * 2) + (((8) * 2)) / 2 + 4 * (1) + 1)]; + const CSUsacConfig *pUsacConfig[(1 * 1)]; + INT nbDiv; /*!< number of frame divisions in LPD-domain */ + + UCHAR useLdQmfTimeAlign; + + INT aacChannelsPrev; /*!< The amount of AAC core channels of the last + successful decode call. */ + AUDIO_CHANNEL_TYPE channelTypePrev[(8)]; /*!< Array holding the channelType + values of the last successful + decode call. */ + UCHAR + channelIndicesPrev[(8)]; /*!< Array holding the channelIndices values of + the last successful decode call. */ + + UCHAR + downscaleFactor; /*!< Variable to store a supported ELD downscale factor + of 1, 2, 3 or 4 */ + UCHAR downscaleFactorInBS; /*!< Variable to store the (not necessarily + supported) ELD downscale factor discovered in + the bitstream */ + + HANDLE_SBRDECODER hSbrDecoder; /*!< SBR decoder handle. */ + UCHAR sbrEnabled; /*!< flag to store if SBR has been detected */ + UCHAR sbrEnabledPrev; /*!< flag to store if SBR has been detected from + previous frame */ + UCHAR psPossible; /*!< flag to store if PS is possible */ + SBR_PARAMS sbrParams; /*!< struct to store all sbr parameters */ + + UCHAR *pDrmBsBuffer; /*!< Pointer to dynamic buffer which is used to reverse + the bits of the DRM SBR payload */ + USHORT drmBsBufferSize; /*!< Size of the dynamic buffer which is used to + reverse the bits of the DRM SBR payload */ + FDK_QMF_DOMAIN + qmfDomain; /*!< Instance of module for QMF domain data handling */ + + QMF_MODE qmfModeCurr; /*!< The current QMF mode */ + QMF_MODE qmfModeUser; /*!< The QMF mode requested by the library user */ + + HANDLE_AAC_DRC hDrcInfo; /*!< handle to DRC data structure */ + INT metadataExpiry; /*!< Metadata expiry time in milli-seconds. */ + + void *pMpegSurroundDecoder; /*!< pointer to mpeg surround decoder structure */ + UCHAR mpsEnableUser; /*!< MPS enable user flag */ + UCHAR mpsEnableCurr; /*!< MPS enable decoder state */ + UCHAR mpsApplicable; /*!< MPS applicable */ + SCHAR mpsOutputMode; /*!< setting: normal = 0, binaural = 1, stereo = 2, 5.1ch + = 3 */ + INT mpsOutChannelsLast; /*!< The amount of channels returned by the last + successful MPS decoder call. */ + INT mpsFrameSizeLast; /*!< The frame length returned by the last successful + MPS decoder call. */ + + CAncData ancData; /*!< structure to handle ancillary data */ + + HANDLE_PCM_DOWNMIX hPcmUtils; /*!< privat data for the PCM utils. */ + + TDLimiterPtr hLimiter; /*!< Handle of time domain limiter. */ + UCHAR limiterEnableUser; /*!< The limiter configuration requested by the + library user */ + UCHAR limiterEnableCurr; /*!< The current limiter configuration. */ + FIXP_DBL extGain[1]; /*!< Gain that must be applied to the output signal. */ + UINT extGainDelay; /*!< Delay that must be accounted for extGain. */ + + INT_PCM pcmOutputBuffer[(8) * (1024 * 2)]; + + HANDLE_DRC_DECODER hUniDrcDecoder; + UCHAR multibandDrcPresent; + UCHAR numTimeSlots; + UINT loudnessInfoSetPosition[3]; + SCHAR defaultTargetLoudness; + + INT_PCM + *pTimeDataFlush[((8) * 2)]; /*!< Pointer to the flushed time data which + will be used for the crossfade in case of + an USAC DASH IPF config change */ + + UCHAR flushStatus; /*!< Indicates flush status: on|off */ + SCHAR flushCnt; /*!< Flush frame counter */ + UCHAR buildUpStatus; /*!< Indicates build up status: on|off */ + SCHAR buildUpCnt; /*!< Build up frame counter */ + UCHAR hasAudioPreRoll; /*!< Indicates preRoll status: on|off */ + UINT prerollAULength[AACDEC_MAX_NUM_PREROLL_AU + 1]; /*!< Relative offset of + the prerollAU end + position to the AU + start position in the + bitstream */ + INT accessUnit; /*!< Number of the actual processed preroll accessUnit */ + UCHAR applyCrossfade; /*!< if set crossfade for seamless stream switching is + applied */ + + FDK_SignalDelay usacResidualDelay; /*!< Delay residual signal to compensate + for eSBR delay of DMX signal in case of + stereoConfigIndex==2. */ +}; + +#define AAC_DEBUG_EXTHLP \ + "\ +--- AAC-Core ---\n\ + 0x00010000 Header data\n\ + 0x00020000 CRC data\n\ + 0x00040000 Channel info\n\ + 0x00080000 Section data\n\ + 0x00100000 Scalefactor data\n\ + 0x00200000 Pulse data\n\ + 0x00400000 Tns data\n\ + 0x00800000 Quantized spectrum\n\ + 0x01000000 Requantized spectrum\n\ + 0x02000000 Time output\n\ + 0x04000000 Fatal errors\n\ + 0x08000000 Buffer fullness\n\ + 0x10000000 Average bitrate\n\ + 0x20000000 Synchronization\n\ + 0x40000000 Concealment\n\ + 0x7FFF0000 all AAC-Core-Info\n\ +" + +/** + * \brief Synchronise QMF mode for all modules using QMF data. + * \param self decoder handle + */ +void CAacDecoder_SyncQmfMode(HANDLE_AACDECODER self); + +/** + * \brief Signal a bit stream interruption to the decoder + * \param self decoder handle + */ +void CAacDecoder_SignalInterruption(HANDLE_AACDECODER self); + +/*! + \brief Initialize ancillary buffer + + \ancData Pointer to ancillary data structure + \buffer Pointer to (external) anc data buffer + \size Size of the buffer pointed on by buffer + + \return Error code +*/ +AAC_DECODER_ERROR CAacDecoder_AncDataInit(CAncData *ancData, + unsigned char *buffer, int size); + +/*! + \brief Get one ancillary data element + + \ancData Pointer to ancillary data structure + \index Index of the anc data element to get + \ptr Pointer to a buffer receiving a pointer to the requested anc data element + \size Pointer to a buffer receiving the length of the requested anc data + element + + \return Error code +*/ +AAC_DECODER_ERROR CAacDecoder_AncDataGet(CAncData *ancData, int index, + unsigned char **ptr, int *size); + +/* initialization of aac decoder */ +LINKSPEC_H HANDLE_AACDECODER CAacDecoder_Open(TRANSPORT_TYPE bsFormat); + +/* Initialization of channel elements */ +LINKSPEC_H AAC_DECODER_ERROR CAacDecoder_Init(HANDLE_AACDECODER self, + const CSAudioSpecificConfig *asc, + UCHAR configMode, + UCHAR *configChanged); +/*! + \brief Decodes one aac frame + + The function decodes one aac frame. The decoding of coupling channel + elements are not supported. The transport layer might signal, that the + data of the current frame is invalid, e.g. as a result of a packet + loss in streaming mode. + The bitstream position of transportDec_GetBitstream(self->hInput) must + be exactly the end of the access unit, including all byte alignment bits. + For this purpose, the variable auStartAnchor is used. + + \return error status +*/ +LINKSPEC_H AAC_DECODER_ERROR CAacDecoder_DecodeFrame( + HANDLE_AACDECODER self, const UINT flags, FIXP_PCM *pTimeData, + const INT timeDataSize, const int timeDataChannelOffset); + +/* Free config dependent AAC memory */ +LINKSPEC_H AAC_DECODER_ERROR CAacDecoder_FreeMem(HANDLE_AACDECODER self, + const int subStreamIndex); + +/* Prepare crossfade for USAC DASH IPF config change */ +LINKSPEC_H AAC_DECODER_ERROR CAacDecoder_PrepareCrossFade( + const INT_PCM *pTimeData, INT_PCM **pTimeDataFlush, const INT numChannels, + const INT frameSize, const INT interleaved); + +/* Apply crossfade for USAC DASH IPF config change */ +LINKSPEC_H AAC_DECODER_ERROR CAacDecoder_ApplyCrossFade( + INT_PCM *pTimeData, INT_PCM **pTimeDataFlush, const INT numChannels, + const INT frameSize, const INT interleaved); + +/* Set flush and build up mode */ +LINKSPEC_H AAC_DECODER_ERROR CAacDecoder_CtrlCFGChange(HANDLE_AACDECODER self, + UCHAR flushStatus, + SCHAR flushCnt, + UCHAR buildUpStatus, + SCHAR buildUpCnt); + +/* Parse preRoll Extension Payload */ +LINKSPEC_H AAC_DECODER_ERROR CAacDecoder_PreRollExtensionPayloadParse( + HANDLE_AACDECODER self, UINT *numPrerollAU, UINT *prerollAUOffset, + UINT *prerollAULength); + +/* Destroy aac decoder */ +LINKSPEC_H void CAacDecoder_Close(HANDLE_AACDECODER self); + +/* get streaminfo handle from decoder */ +LINKSPEC_H CStreamInfo *CAacDecoder_GetStreamInfo(HANDLE_AACDECODER self); + +#endif /* #ifndef AACDECODER_H */ |