From 0e5af65c467b2423a0b857ae3ad98c91acc1e190 Mon Sep 17 00:00:00 2001 From: "Matthias P. Braendli" Date: Mon, 11 Nov 2019 11:38:02 +0100 Subject: Include patched FDK-AAC in the repository The initial idea was to get the DAB+ patch into upstream, but since that follows the android source releases, there is no place for a custom DAB+ patch there. So instead of having to maintain a patched fdk-aac that has to have the same .so version as the distribution package on which it is installed, we prefer having a separate fdk-aac-dab library to avoid collision. At that point, there's no reason to keep fdk-aac in a separate repository, as odr-audioenc is the only tool that needs DAB+ encoding support. Including it here simplifies installation, and makes it consistent with toolame-dab, also shipped in this repository. DAB+ decoding support (needed by ODR-SourceCompanion, dablin, etisnoop, welle.io and others) can be done using upstream FDK-AAC. --- fdk-aac/libMpegTPDec/src/tpdec_latm.cpp | 676 ++++++++++++++++++++++++++++++++ 1 file changed, 676 insertions(+) create mode 100644 fdk-aac/libMpegTPDec/src/tpdec_latm.cpp (limited to 'fdk-aac/libMpegTPDec/src/tpdec_latm.cpp') diff --git a/fdk-aac/libMpegTPDec/src/tpdec_latm.cpp b/fdk-aac/libMpegTPDec/src/tpdec_latm.cpp new file mode 100644 index 0000000..2edf055 --- /dev/null +++ b/fdk-aac/libMpegTPDec/src/tpdec_latm.cpp @@ -0,0 +1,676 @@ +/* ----------------------------------------------------------------------------- +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): Daniel Homm + + Description: + +*******************************************************************************/ + +#include "tpdec_latm.h" + +#include "FDK_bitstream.h" + +#define TPDEC_TRACKINDEX(p, l) (1 * (p) + (l)) + +static UINT CLatmDemux_GetValue(HANDLE_FDK_BITSTREAM bs) { + UCHAR bytesForValue = 0, tmp = 0; + int value = 0; + + bytesForValue = (UCHAR)FDKreadBits(bs, 2); + + for (UINT i = 0; i <= bytesForValue; i++) { + value <<= 8; + tmp = (UCHAR)FDKreadBits(bs, 8); + value += tmp; + } + + return value; +} + +static TRANSPORTDEC_ERROR CLatmDemux_ReadAudioMuxElement( + HANDLE_FDK_BITSTREAM bs, CLatmDemux *pLatmDemux, int m_muxConfigPresent, + CSTpCallBacks *pTpDecCallbacks, CSAudioSpecificConfig *pAsc, + int *pfConfigFound) { + TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK; + + if (m_muxConfigPresent) { + pLatmDemux->m_useSameStreamMux = FDKreadBits(bs, 1); + + if (!pLatmDemux->m_useSameStreamMux) { + int i; + UCHAR configChanged = 0; + UCHAR configMode = 0; + + FDK_BITSTREAM bsAnchor; + + FDK_BITSTREAM bsAnchorDummyParse; + + if (!pLatmDemux->applyAsc) { + bsAnchorDummyParse = *bs; + pLatmDemux->newCfgHasAudioPreRoll = 0; + /* do dummy-parsing of ASC to determine if there is an audioPreRoll */ + configMode |= AC_CM_DET_CFG_CHANGE; + if (TRANSPORTDEC_OK != + (ErrorStatus = CLatmDemux_ReadStreamMuxConfig( + bs, pLatmDemux, pTpDecCallbacks, pAsc, pfConfigFound, + configMode, configChanged))) { + goto bail; + } + + /* Allow flushing only when audioPreroll functionality is enabled in + * current and new config otherwise the new config can be applied + * immediately. */ + if (pAsc->m_sc.m_usacConfig.element[0] + .extElement.usacExtElementHasAudioPreRoll && + pLatmDemux->newCfgHasAudioPreRoll) { + pLatmDemux->newCfgHasAudioPreRoll = 0; + /* with audioPreRoll we must flush before applying new cfg */ + pLatmDemux->applyAsc = 0; + } else { + *bs = bsAnchorDummyParse; + pLatmDemux->applyAsc = 1; /* apply new config immediate */ + } + } + + if (pLatmDemux->applyAsc) { + for (i = 0; i < 2; i++) { + configMode = 0; + + if (i == 0) { + configMode |= AC_CM_DET_CFG_CHANGE; + bsAnchor = *bs; + } else { + configMode |= AC_CM_ALLOC_MEM; + *bs = bsAnchor; + } + + if (TRANSPORTDEC_OK != + (ErrorStatus = CLatmDemux_ReadStreamMuxConfig( + bs, pLatmDemux, pTpDecCallbacks, pAsc, pfConfigFound, + configMode, configChanged))) { + goto bail; + } + + if (ErrorStatus == TRANSPORTDEC_OK) { + if ((i == 0) && (pAsc->AacConfigChanged || pAsc->SbrConfigChanged || + pAsc->SacConfigChanged)) { + int errC; + + configChanged = 1; + errC = pTpDecCallbacks->cbFreeMem(pTpDecCallbacks->cbFreeMemData, + pAsc); + if (errC != 0) { + ErrorStatus = TRANSPORTDEC_PARSE_ERROR; + goto bail; + } + } + } + } + } + } + } + + /* If there was no configuration read, its not possible to parse + * PayloadLengthInfo below. */ + if (!*pfConfigFound) { + ErrorStatus = TRANSPORTDEC_SYNC_ERROR; + goto bail; + } + + if (pLatmDemux->m_AudioMuxVersionA == 0) { + /* Do only once per call, because parsing and decoding is done in-line. */ + if (TRANSPORTDEC_OK != + (ErrorStatus = CLatmDemux_ReadPayloadLengthInfo(bs, pLatmDemux))) { + *pfConfigFound = 0; + goto bail; + } + } else { + /* audioMuxVersionA > 0 is reserved for future extensions */ + ErrorStatus = TRANSPORTDEC_UNSUPPORTED_FORMAT; + *pfConfigFound = 0; + goto bail; + } + +bail: + if (ErrorStatus != TRANSPORTDEC_OK) { + pLatmDemux->applyAsc = 1; + } + + return (ErrorStatus); +} + +TRANSPORTDEC_ERROR CLatmDemux_Read(HANDLE_FDK_BITSTREAM bs, + CLatmDemux *pLatmDemux, TRANSPORT_TYPE tt, + CSTpCallBacks *pTpDecCallbacks, + CSAudioSpecificConfig *pAsc, + int *pfConfigFound, + const INT ignoreBufferFullness) { + UINT cntBits; + UINT cmpBufferFullness; + UINT audioMuxLengthBytesLast = 0; + TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK; + + cntBits = FDKgetValidBits(bs); + + if ((INT)cntBits < MIN_LATM_HEADERLENGTH) { + return TRANSPORTDEC_NOT_ENOUGH_BITS; + } + + if (TRANSPORTDEC_OK != (ErrorStatus = CLatmDemux_ReadAudioMuxElement( + bs, pLatmDemux, (tt != TT_MP4_LATM_MCP0), + pTpDecCallbacks, pAsc, pfConfigFound))) + return (ErrorStatus); + + if (!ignoreBufferFullness) { + cmpBufferFullness = + 24 + audioMuxLengthBytesLast * 8 + + pLatmDemux->m_linfo[0][0].m_bufferFullness * + pAsc[TPDEC_TRACKINDEX(0, 0)].m_channelConfiguration * 32; + + /* evaluate buffer fullness */ + + if (pLatmDemux->m_linfo[0][0].m_bufferFullness != 0xFF) { + if (!pLatmDemux->BufferFullnessAchieved) { + if (cntBits < cmpBufferFullness) { + /* condition for start of decoding is not fulfilled */ + + /* the current frame will not be decoded */ + return TRANSPORTDEC_NOT_ENOUGH_BITS; + } else { + pLatmDemux->BufferFullnessAchieved = 1; + } + } + } + } + + return (ErrorStatus); +} + +TRANSPORTDEC_ERROR CLatmDemux_ReadStreamMuxConfig( + HANDLE_FDK_BITSTREAM bs, CLatmDemux *pLatmDemux, + CSTpCallBacks *pTpDecCallbacks, CSAudioSpecificConfig *pAsc, + int *pfConfigFound, UCHAR configMode, UCHAR configChanged) { + CSAudioSpecificConfig ascDummy; /* the actual config is needed for flushing, + after that new config can be parsed */ + CSAudioSpecificConfig *pAscDummy; + pAscDummy = &ascDummy; + pLatmDemux->usacExplicitCfgChanged = 0; + LATM_LAYER_INFO *p_linfo = NULL; + TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK; + UCHAR updateConfig[1 * 1] = {0}; + + pLatmDemux->m_AudioMuxVersion = FDKreadBits(bs, 1); + + if (pLatmDemux->m_AudioMuxVersion == 0) { + pLatmDemux->m_AudioMuxVersionA = 0; + } else { + pLatmDemux->m_AudioMuxVersionA = FDKreadBits(bs, 1); + } + + if (pLatmDemux->m_AudioMuxVersionA == 0) { + if (pLatmDemux->m_AudioMuxVersion == 1) { + pLatmDemux->m_taraBufferFullness = CLatmDemux_GetValue(bs); + } + pLatmDemux->m_allStreamsSameTimeFraming = FDKreadBits(bs, 1); + pLatmDemux->m_noSubFrames = FDKreadBits(bs, 6) + 1; + pLatmDemux->m_numProgram = FDKreadBits(bs, 4) + 1; + + if (pLatmDemux->m_numProgram > LATM_MAX_PROG) { + ErrorStatus = TRANSPORTDEC_UNSUPPORTED_FORMAT; + goto bail; + } + + int idCnt = 0; + for (UINT prog = 0; prog < pLatmDemux->m_numProgram; prog++) { + pLatmDemux->m_numLayer[prog] = FDKreadBits(bs, 3) + 1; + if (pLatmDemux->m_numLayer[prog] > LATM_MAX_LAYER) { + ErrorStatus = TRANSPORTDEC_UNSUPPORTED_FORMAT; + goto bail; + } + + for (UINT lay = 0; lay < pLatmDemux->m_numLayer[prog]; lay++) { + int useSameConfig; + p_linfo = &pLatmDemux->m_linfo[prog][lay]; + + p_linfo->m_streamID = idCnt++; + p_linfo->m_frameLengthInBits = 0; + + if ((prog == 0) && (lay == 0)) { + useSameConfig = 0; + } else { + useSameConfig = FDKreadBits(bs, 1); + } + + if (useSameConfig) { + if (lay > 0) { + FDKmemcpy(&pAsc[TPDEC_TRACKINDEX(prog, lay)], + &pAsc[TPDEC_TRACKINDEX(prog, lay - 1)], + sizeof(CSAudioSpecificConfig)); + } else { + ErrorStatus = TRANSPORTDEC_PARSE_ERROR; + goto bail; + } + } else { + UINT usacConfigLengthPrev = 0; + UCHAR usacConfigPrev[TP_USAC_MAX_CONFIG_LEN]; + + if (!(pLatmDemux->applyAsc) && + (pAsc[TPDEC_TRACKINDEX(prog, lay)].m_aot == AOT_USAC)) { + usacConfigLengthPrev = + (UINT)(pAsc[TPDEC_TRACKINDEX(prog, lay)] + .m_sc.m_usacConfig.UsacConfigBits + + 7) >> + 3; /* store previous USAC config length */ + if (usacConfigLengthPrev > TP_USAC_MAX_CONFIG_LEN) { + ErrorStatus = TRANSPORTDEC_PARSE_ERROR; + goto bail; + } + FDKmemclear(usacConfigPrev, TP_USAC_MAX_CONFIG_LEN); + FDKmemcpy( + usacConfigPrev, + &pAsc[TPDEC_TRACKINDEX(prog, lay)].m_sc.m_usacConfig.UsacConfig, + usacConfigLengthPrev); /* store previous USAC config */ + } + if (pLatmDemux->m_AudioMuxVersion == 1) { + FDK_BITSTREAM tmpBs; + UINT ascLen = 0; + ascLen = CLatmDemux_GetValue(bs); + /* The ascLen could be wrong, so check if validBits<=bufBits*/ + if (ascLen > FDKgetValidBits(bs)) { + ErrorStatus = TRANSPORTDEC_PARSE_ERROR; + goto bail; + } + FDKsyncCache(bs); + tmpBs = *bs; + tmpBs.hBitBuf.ValidBits = ascLen; + + /* Read ASC */ + if (pLatmDemux->applyAsc) { + if (TRANSPORTDEC_OK != + (ErrorStatus = AudioSpecificConfig_Parse( + &pAsc[TPDEC_TRACKINDEX(prog, lay)], &tmpBs, 1, + pTpDecCallbacks, configMode, configChanged, + AOT_NULL_OBJECT))) + goto bail; + } else { + if (TRANSPORTDEC_OK != + (ErrorStatus = AudioSpecificConfig_Parse( + pAscDummy, &tmpBs, 1, pTpDecCallbacks, configMode, + configChanged, AOT_NULL_OBJECT))) + goto bail; + } + + /* The field p_linfo->m_ascLen could be wrong, so check if */ + if (0 > (INT)FDKgetValidBits(&tmpBs)) { + ErrorStatus = TRANSPORTDEC_PARSE_ERROR; + goto bail; + } + FDKpushFor(bs, ascLen); /* position bitstream after ASC */ + } else { + /* Read ASC */ + if (pLatmDemux->applyAsc) { + if (TRANSPORTDEC_OK != (ErrorStatus = AudioSpecificConfig_Parse( + &pAsc[TPDEC_TRACKINDEX(prog, lay)], + bs, 0, pTpDecCallbacks, configMode, + configChanged, AOT_NULL_OBJECT))) + goto bail; + } else { + if (TRANSPORTDEC_OK != + (ErrorStatus = AudioSpecificConfig_Parse( + pAscDummy, bs, 0, pTpDecCallbacks, configMode, + configChanged, AOT_NULL_OBJECT))) + goto bail; + } + } + if (!pLatmDemux->applyAsc) { + updateConfig[TPDEC_TRACKINDEX(prog, lay)] = 0; + } else { + updateConfig[TPDEC_TRACKINDEX(prog, lay)] = 1; + } + + if (!pLatmDemux->applyAsc) { + if (pAscDummy[TPDEC_TRACKINDEX(prog, lay)].m_aot == + AOT_USAC) { /* flush in case SMC has changed */ + const UINT usacConfigLength = + (UINT)(pAscDummy->m_sc.m_usacConfig.UsacConfigBits + 7) >> 3; + if (usacConfigLength > TP_USAC_MAX_CONFIG_LEN) { + ErrorStatus = TRANSPORTDEC_PARSE_ERROR; + goto bail; + } + if (usacConfigLength != usacConfigLengthPrev) { + FDKmemclear(&pAsc[TPDEC_TRACKINDEX(prog, lay)] + .m_sc.m_usacConfig.UsacConfig, + TP_USAC_MAX_CONFIG_LEN); + FDKmemcpy(&pAsc[TPDEC_TRACKINDEX(prog, lay)] + .m_sc.m_usacConfig.UsacConfig, + &pAscDummy->m_sc.m_usacConfig.UsacConfig, + usacConfigLength); /* store new USAC config */ + pAsc[TPDEC_TRACKINDEX(prog, lay)] + .m_sc.m_usacConfig.UsacConfigBits = + pAscDummy->m_sc.m_usacConfig.UsacConfigBits; + pLatmDemux->usacExplicitCfgChanged = 1; + } else { + if (FDKmemcmp(usacConfigPrev, + pAscDummy->m_sc.m_usacConfig.UsacConfig, + usacConfigLengthPrev)) { + FDKmemclear(&pAsc[TPDEC_TRACKINDEX(prog, lay)] + .m_sc.m_usacConfig.UsacConfig, + TP_USAC_MAX_CONFIG_LEN); + FDKmemcpy(&pAsc[TPDEC_TRACKINDEX(prog, lay)] + .m_sc.m_usacConfig.UsacConfig, + &pAscDummy->m_sc.m_usacConfig.UsacConfig, + usacConfigLength); /* store new USAC config */ + pAsc[TPDEC_TRACKINDEX(prog, lay)] + .m_sc.m_usacConfig.UsacConfigBits = + pAscDummy->m_sc.m_usacConfig.UsacConfigBits; + pLatmDemux->usacExplicitCfgChanged = 1; + } + } + + if (pAscDummy[TPDEC_TRACKINDEX(prog, lay)] + .m_sc.m_usacConfig.m_usacNumElements) { + if (pAscDummy[TPDEC_TRACKINDEX(prog, lay)] + .m_sc.m_usacConfig.element[0] + .extElement.usacExtElementHasAudioPreRoll) { + pLatmDemux->newCfgHasAudioPreRoll = + 1; /* if dummy parsed cfg has audioPreRoll we first flush + before applying new cfg */ + } + } + } + } + } + + p_linfo->m_frameLengthType = FDKreadBits(bs, 3); + switch (p_linfo->m_frameLengthType) { + case 0: + p_linfo->m_bufferFullness = FDKreadBits(bs, 8); + + if (!pLatmDemux->m_allStreamsSameTimeFraming) { + if ((lay > 0) && + (pAsc[TPDEC_TRACKINDEX(prog, lay)].m_aot == AOT_AAC_SCAL || + pAsc[TPDEC_TRACKINDEX(prog, lay)].m_aot == + AOT_ER_AAC_SCAL) && + (pAsc[TPDEC_TRACKINDEX(prog, lay - 1)].m_aot == AOT_CELP || + pAsc[TPDEC_TRACKINDEX(prog, lay - 1)].m_aot == + AOT_ER_CELP)) { /* The layer maybe + ignored later so + read it anyway: */ + /* coreFrameOffset = */ FDKreadBits(bs, 6); + } + } + break; + case 1: + p_linfo->m_frameLengthInBits = FDKreadBits(bs, 9); + break; + case 3: + case 4: + case 5: + /* CELP */ + case 6: + case 7: + /* HVXC */ + default: + ErrorStatus = TRANSPORTDEC_PARSE_ERROR; + goto bail; + } /* switch framelengthtype*/ + + } /* layer loop */ + } /* prog loop */ + + pLatmDemux->m_otherDataPresent = FDKreadBits(bs, 1); + pLatmDemux->m_otherDataLength = 0; + + if (pLatmDemux->m_otherDataPresent) { + if (pLatmDemux->m_AudioMuxVersion == 1) { + pLatmDemux->m_otherDataLength = CLatmDemux_GetValue(bs); + } else { + int otherDataLenEsc = 0; + do { + pLatmDemux->m_otherDataLength <<= 8; // *= 256 + otherDataLenEsc = FDKreadBits(bs, 1); + pLatmDemux->m_otherDataLength += FDKreadBits(bs, 8); + } while (otherDataLenEsc); + } + if (pLatmDemux->m_audioMuxLengthBytes < + (pLatmDemux->m_otherDataLength >> 3)) { + ErrorStatus = TRANSPORTDEC_PARSE_ERROR; + goto bail; + } + } + + pLatmDemux->m_crcCheckPresent = FDKreadBits(bs, 1); + + if (pLatmDemux->m_crcCheckPresent) { + FDKreadBits(bs, 8); + } + + } else { + /* audioMuxVersionA > 0 is reserved for future extensions */ + ErrorStatus = TRANSPORTDEC_UNSUPPORTED_FORMAT; + } + + /* Configure source decoder: */ + if (ErrorStatus == TRANSPORTDEC_OK) { + UINT prog; + for (prog = 0; prog < pLatmDemux->m_numProgram; prog++) { + UINT lay; + for (lay = 0; lay < pLatmDemux->m_numLayer[prog]; lay++) { + if (updateConfig[TPDEC_TRACKINDEX(prog, lay)] != 0) { + int cbError; + cbError = pTpDecCallbacks->cbUpdateConfig( + pTpDecCallbacks->cbUpdateConfigData, + &pAsc[TPDEC_TRACKINDEX(prog, lay)], + pAsc[TPDEC_TRACKINDEX(prog, lay)].configMode, + &pAsc[TPDEC_TRACKINDEX(prog, lay)].AacConfigChanged); + if (cbError == TRANSPORTDEC_NEED_TO_RESTART) { + *pfConfigFound = 0; + ErrorStatus = TRANSPORTDEC_NEED_TO_RESTART; + goto bail; + } + if (cbError != 0) { + *pfConfigFound = 0; + if (lay == 0) { + ErrorStatus = TRANSPORTDEC_SYNC_ERROR; + goto bail; + } + } else { + *pfConfigFound = 1; + } + } else { + *pfConfigFound = 1; + } + } + } + } + +bail: + if (ErrorStatus != TRANSPORTDEC_OK) { + UCHAR applyAsc = pLatmDemux->applyAsc; + FDKmemclear(pLatmDemux, sizeof(CLatmDemux)); /* reset structure */ + pLatmDemux->applyAsc = applyAsc; + } else { + /* no error and config parsing is finished */ + if (configMode == AC_CM_ALLOC_MEM) pLatmDemux->applyAsc = 0; + } + + return (ErrorStatus); +} + +TRANSPORTDEC_ERROR CLatmDemux_ReadPayloadLengthInfo(HANDLE_FDK_BITSTREAM bs, + CLatmDemux *pLatmDemux) { + TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK; + int totalPayloadBits = 0; + + if (pLatmDemux->m_allStreamsSameTimeFraming == 1) { + FDK_ASSERT(pLatmDemux->m_numProgram <= LATM_MAX_PROG); + for (UINT prog = 0; prog < pLatmDemux->m_numProgram; prog++) { + FDK_ASSERT(pLatmDemux->m_numLayer[prog] <= LATM_MAX_LAYER); + for (UINT lay = 0; lay < pLatmDemux->m_numLayer[prog]; lay++) { + LATM_LAYER_INFO *p_linfo = &pLatmDemux->m_linfo[prog][lay]; + + switch (p_linfo->m_frameLengthType) { + case 0: + p_linfo->m_frameLengthInBits = CLatmDemux_ReadAuChunkLengthInfo(bs); + totalPayloadBits += p_linfo->m_frameLengthInBits; + break; + case 3: + case 5: + case 7: + default: + return TRANSPORTDEC_PARSE_ERROR; // AAC_DEC_LATM_INVALIDFRAMELENGTHTYPE; + } + } + } + } else { + ErrorStatus = TRANSPORTDEC_PARSE_ERROR; // AAC_DEC_LATM_TIMEFRAMING; + } + if (pLatmDemux->m_audioMuxLengthBytes > (UINT)0 && + totalPayloadBits > (int)pLatmDemux->m_audioMuxLengthBytes * 8) { + return TRANSPORTDEC_PARSE_ERROR; + } + + return (ErrorStatus); +} + +int CLatmDemux_ReadAuChunkLengthInfo(HANDLE_FDK_BITSTREAM bs) { + UCHAR endFlag; + int len = 0; + + do { + UCHAR tmp = (UCHAR)FDKreadBits(bs, 8); + endFlag = (tmp < 255); + + len += tmp; + + } while (endFlag == 0); + + len <<= 3; /* convert from bytes to bits */ + + return len; +} + +UINT CLatmDemux_GetFrameLengthInBits(CLatmDemux *pLatmDemux, const UINT prog, + const UINT layer) { + UINT nFrameLenBits = 0; + if (prog < pLatmDemux->m_numProgram) { + if (layer < pLatmDemux->m_numLayer[prog]) { + nFrameLenBits = pLatmDemux->m_linfo[prog][layer].m_frameLengthInBits; + } + } + return nFrameLenBits; +} + +UINT CLatmDemux_GetOtherDataPresentFlag(CLatmDemux *pLatmDemux) { + return pLatmDemux->m_otherDataPresent ? 1 : 0; +} + +UINT CLatmDemux_GetOtherDataLength(CLatmDemux *pLatmDemux) { + return pLatmDemux->m_otherDataLength; +} + +UINT CLatmDemux_GetNrOfSubFrames(CLatmDemux *pLatmDemux) { + return pLatmDemux->m_noSubFrames; +} + +UINT CLatmDemux_GetNrOfLayers(CLatmDemux *pLatmDemux, const UINT prog) { + UINT numLayer = 0; + if (prog < pLatmDemux->m_numProgram) { + numLayer = pLatmDemux->m_numLayer[prog]; + } + return numLayer; +} -- cgit v1.2.3