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Changes of Revision 12
fdk-aac.changes
Changed
x
1
2
-------------------------------------------------------------------3
+Sun Jul 25 17:02:42 UTC 2021 - Bjørn Lie <zaitor@opensuse.org>4
+5
+- Update to version 2.0.2:6
+ + This version adds files for building with CMake in addition to7
+ autotools, and contains a small number of upstream fixes and8
+ local fuzzing fixes.9
+10
+-------------------------------------------------------------------11
Tue Mar 24 17:47:21 UTC 2020 - Bjørn Lie <zaitor@opensuse.org>12
13
- Update to version 2.0.1:14
fdk-aac.spec
Changed
10
1
2
3
%define _sover 24
Name: fdk-aac5
-Version: 2.0.16
+Version: 2.0.27
Release: 08
Summary: A standalone library of the Fraunhofer FDK AAC code from Android9
License: NonFree10
fdk-aac-2.0.1.tar.gz/Makefile.vc
Deleted
201
1
2
-#3
-# Options:4
-# prefix=\path\to\install5
-#6
-# Compiling: nmake -f Makefile.vc7
-# Installing: nmake -f Makefile.vc prefix=\path\to\x install8
-#9
-10
-# Linker and librarian commands11
-LD = link12
-AR = lib13
-14
-!IFDEF HOME15
-# In case we are using a cross compiler shell.16
-MKDIR_FLAGS = -p17
-!ENDIF18
-19
-AM_CPPFLAGS = \20
- -Iwin32 \21
- -IlibAACdec/include \22
- -IlibAACenc/include \23
- -IlibArithCoding/include \24
- -IlibDRCdec/include \25
- -IlibSACdec/include \26
- -IlibSACenc/include \27
- -IlibSBRdec/include \28
- -IlibSBRenc/include \29
- -IlibMpegTPDec/include \30
- -IlibMpegTPEnc/include \31
- -IlibSYS/include \32
- -IlibFDK/include \33
- -IlibPCMutils/include34
-35
-AACDEC_SRC = \36
- libAACdec/src/FDK_delay.cpp \37
- libAACdec/src/aac_ram.cpp \38
- libAACdec/src/aac_rom.cpp \39
- libAACdec/src/aacdec_drc.cpp \40
- libAACdec/src/aacdec_hcr.cpp \41
- libAACdec/src/aacdec_hcr_bit.cpp \42
- libAACdec/src/aacdec_hcrs.cpp \43
- libAACdec/src/aacdec_pns.cpp \44
- libAACdec/src/aacdec_tns.cpp \45
- libAACdec/src/aacdecoder.cpp \46
- libAACdec/src/aacdecoder_lib.cpp \47
- libAACdec/src/block.cpp \48
- libAACdec/src/channel.cpp \49
- libAACdec/src/channelinfo.cpp \50
- libAACdec/src/conceal.cpp \51
- libAACdec/src/ldfiltbank.cpp \52
- libAACdec/src/pulsedata.cpp \53
- libAACdec/src/rvlc.cpp \54
- libAACdec/src/rvlcbit.cpp \55
- libAACdec/src/rvlcconceal.cpp \56
- libAACdec/src/stereo.cpp \57
- libAACdec/src/usacdec_ace_d4t64.cpp \58
- libAACdec/src/usacdec_ace_ltp.cpp \59
- libAACdec/src/usacdec_acelp.cpp \60
- libAACdec/src/usacdec_fac.cpp \61
- libAACdec/src/usacdec_lpc.cpp \62
- libAACdec/src/usacdec_lpd.cpp \63
- libAACdec/src/usacdec_rom.cpp64
-65
-AACENC_SRC = \66
- libAACenc/src/aacEnc_ram.cpp \67
- libAACenc/src/aacEnc_rom.cpp \68
- libAACenc/src/aacenc.cpp \69
- libAACenc/src/aacenc_lib.cpp \70
- libAACenc/src/aacenc_pns.cpp \71
- libAACenc/src/aacenc_tns.cpp \72
- libAACenc/src/adj_thr.cpp \73
- libAACenc/src/band_nrg.cpp \74
- libAACenc/src/bandwidth.cpp \75
- libAACenc/src/bit_cnt.cpp \76
- libAACenc/src/bitenc.cpp \77
- libAACenc/src/block_switch.cpp \78
- libAACenc/src/channel_map.cpp \79
- libAACenc/src/chaosmeasure.cpp \80
- libAACenc/src/dyn_bits.cpp \81
- libAACenc/src/grp_data.cpp \82
- libAACenc/src/intensity.cpp \83
- libAACenc/src/line_pe.cpp \84
- libAACenc/src/metadata_compressor.cpp \85
- libAACenc/src/metadata_main.cpp \86
- libAACenc/src/mps_main.cpp \87
- libAACenc/src/ms_stereo.cpp \88
- libAACenc/src/noisedet.cpp \89
- libAACenc/src/pnsparam.cpp \90
- libAACenc/src/pre_echo_control.cpp \91
- libAACenc/src/psy_configuration.cpp \92
- libAACenc/src/psy_main.cpp \93
- libAACenc/src/qc_main.cpp \94
- libAACenc/src/quantize.cpp \95
- libAACenc/src/sf_estim.cpp \96
- libAACenc/src/spreading.cpp \97
- libAACenc/src/tonality.cpp \98
- libAACenc/src/transform.cpp99
-100
-ARITHCODING_SRC = \101
- libArithCoding/src/ac_arith_coder.cpp102
-103
-DRCDEC_SRC = \104
- libDRCdec/src/FDK_drcDecLib.cpp \105
- libDRCdec/src/drcDec_gainDecoder.cpp \106
- libDRCdec/src/drcDec_reader.cpp \107
- libDRCdec/src/drcDec_rom.cpp \108
- libDRCdec/src/drcDec_selectionProcess.cpp \109
- libDRCdec/src/drcDec_tools.cpp \110
- libDRCdec/src/drcGainDec_init.cpp \111
- libDRCdec/src/drcGainDec_preprocess.cpp \112
- libDRCdec/src/drcGainDec_process.cpp113
-114
-FDK_SRC = \115
- libFDK/src/FDK_bitbuffer.cpp \116
- libFDK/src/FDK_core.cpp \117
- libFDK/src/FDK_crc.cpp \118
- libFDK/src/FDK_decorrelate.cpp \119
- libFDK/src/FDK_hybrid.cpp \120
- libFDK/src/FDK_lpc.cpp \121
- libFDK/src/FDK_matrixCalloc.cpp \122
- libFDK/src/FDK_qmf_domain.cpp \123
- libFDK/src/FDK_tools_rom.cpp \124
- libFDK/src/FDK_trigFcts.cpp \125
- libFDK/src/autocorr2nd.cpp \126
- libFDK/src/dct.cpp \127
- libFDK/src/fft.cpp \128
- libFDK/src/fft_rad2.cpp \129
- libFDK/src/fixpoint_math.cpp \130
- libFDK/src/huff_nodes.cpp \131
- libFDK/src/mdct.cpp \132
- libFDK/src/nlc_dec.cpp \133
- libFDK/src/qmf.cpp \134
- libFDK/src/scale.cpp135
-136
-MPEGTPDEC_SRC = \137
- libMpegTPDec/src/tpdec_adif.cpp \138
- libMpegTPDec/src/tpdec_adts.cpp \139
- libMpegTPDec/src/tpdec_asc.cpp \140
- libMpegTPDec/src/tpdec_drm.cpp \141
- libMpegTPDec/src/tpdec_latm.cpp \142
- libMpegTPDec/src/tpdec_lib.cpp143
-144
-MPEGTPENC_SRC = \145
- libMpegTPEnc/src/tpenc_adif.cpp \146
- libMpegTPEnc/src/tpenc_adts.cpp \147
- libMpegTPEnc/src/tpenc_asc.cpp \148
- libMpegTPEnc/src/tpenc_latm.cpp \149
- libMpegTPEnc/src/tpenc_lib.cpp150
-151
-PCMUTILS_SRC = \152
- libPCMutils/src/limiter.cpp \153
- libPCMutils/src/pcm_utils.cpp \154
- libPCMutils/src/pcmdmx_lib.cpp155
-156
-SACDEC_SRC = \157
- libSACdec/src/sac_bitdec.cpp \158
- libSACdec/src/sac_calcM1andM2.cpp \159
- libSACdec/src/sac_dec.cpp \160
- libSACdec/src/sac_dec_conceal.cpp \161
- libSACdec/src/sac_dec_lib.cpp \162
- libSACdec/src/sac_process.cpp \163
- libSACdec/src/sac_qmf.cpp \164
- libSACdec/src/sac_reshapeBBEnv.cpp \165
- libSACdec/src/sac_rom.cpp \166
- libSACdec/src/sac_smoothing.cpp \167
- libSACdec/src/sac_stp.cpp \168
- libSACdec/src/sac_tsd.cpp169
-170
-SACENC_SRC = \171
- libSACenc/src/sacenc_bitstream.cpp \172
- libSACenc/src/sacenc_delay.cpp \173
- libSACenc/src/sacenc_dmx_tdom_enh.cpp \174
- libSACenc/src/sacenc_filter.cpp \175
- libSACenc/src/sacenc_framewindowing.cpp \176
- libSACenc/src/sacenc_huff_tab.cpp \177
- libSACenc/src/sacenc_lib.cpp \178
- libSACenc/src/sacenc_nlc_enc.cpp \179
- libSACenc/src/sacenc_onsetdetect.cpp \180
- libSACenc/src/sacenc_paramextract.cpp \181
- libSACenc/src/sacenc_staticgain.cpp \182
- libSACenc/src/sacenc_tree.cpp \183
- libSACenc/src/sacenc_vectorfunctions.cpp184
-185
-SBRDEC_SRC = \186
- libSBRdec/src/HFgen_preFlat.cpp \187
- libSBRdec/src/env_calc.cpp \188
- libSBRdec/src/env_dec.cpp \189
- libSBRdec/src/env_extr.cpp \190
- libSBRdec/src/hbe.cpp \191
- libSBRdec/src/huff_dec.cpp \192
- libSBRdec/src/lpp_tran.cpp \193
- libSBRdec/src/psbitdec.cpp \194
- libSBRdec/src/psdec.cpp \195
- libSBRdec/src/psdec_drm.cpp \196
- libSBRdec/src/psdecrom_drm.cpp \197
- libSBRdec/src/pvc_dec.cpp \198
- libSBRdec/src/sbr_crc.cpp \199
- libSBRdec/src/sbr_deb.cpp \200
- libSBRdec/src/sbr_dec.cpp \201
fdk-aac-2.0.1.tar.gz/libSBRdec/src/sbr_crc.cpp
Deleted
194
1
2
-/* -----------------------------------------------------------------------------3
-Software License for The Fraunhofer FDK AAC Codec Library for Android4
-5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
-Forschung e.V. All rights reserved.7
-8
- 1. INTRODUCTION9
-The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software10
-that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding11
-scheme for digital audio. This FDK AAC Codec software is intended to be used on12
-a wide variety of Android devices.13
-14
-AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient15
-general perceptual audio codecs. AAC-ELD is considered the best-performing16
-full-bandwidth communications codec by independent studies and is widely17
-deployed. AAC has been standardized by ISO and IEC as part of the MPEG18
-specifications.19
-20
-Patent licenses for necessary patent claims for the FDK AAC Codec (including21
-those of Fraunhofer) may be obtained through Via Licensing22
-(www.vialicensing.com) or through the respective patent owners individually for23
-the purpose of encoding or decoding bit streams in products that are compliant24
-with the ISO/IEC MPEG audio standards. Please note that most manufacturers of25
-Android devices already license these patent claims through Via Licensing or26
-directly from the patent owners, and therefore FDK AAC Codec software may27
-already be covered under those patent licenses when it is used for those28
-licensed purposes only.29
-30
-Commercially-licensed AAC software libraries, including floating-point versions31
-with enhanced sound quality, are also available from Fraunhofer. Users are32
-encouraged to check the Fraunhofer website for additional applications33
-information and documentation.34
-35
-2. COPYRIGHT LICENSE36
-37
-Redistribution and use in source and binary forms, with or without modification,38
-are permitted without payment of copyright license fees provided that you39
-satisfy the following conditions:40
-41
-You must retain the complete text of this software license in redistributions of42
-the FDK AAC Codec or your modifications thereto in source code form.43
-44
-You must retain the complete text of this software license in the documentation45
-and/or other materials provided with redistributions of the FDK AAC Codec or46
-your modifications thereto in binary form. You must make available free of47
-charge copies of the complete source code of the FDK AAC Codec and your48
-modifications thereto to recipients of copies in binary form.49
-50
-The name of Fraunhofer may not be used to endorse or promote products derived51
-from this library without prior written permission.52
-53
-You may not charge copyright license fees for anyone to use, copy or distribute54
-the FDK AAC Codec software or your modifications thereto.55
-56
-Your modified versions of the FDK AAC Codec must carry prominent notices stating57
-that you changed the software and the date of any change. For modified versions58
-of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android"59
-must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK60
-AAC Codec Library for Android."61
-62
-3. NO PATENT LICENSE63
-64
-NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without65
-limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE.66
-Fraunhofer provides no warranty of patent non-infringement with respect to this67
-software.68
-69
-You may use this FDK AAC Codec software or modifications thereto only for70
-purposes that are authorized by appropriate patent licenses.71
-72
-4. DISCLAIMER73
-74
-This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright75
-holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES,76
-including but not limited to the implied warranties of merchantability and77
-fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR78
-CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary,79
-or consequential damages, including but not limited to procurement of substitute80
-goods or services; loss of use, data, or profits, or business interruption,81
-however caused and on any theory of liability, whether in contract, strict82
-liability, or tort (including negligence), arising in any way out of the use of83
-this software, even if advised of the possibility of such damage.84
-85
-5. CONTACT INFORMATION86
-87
-Fraunhofer Institute for Integrated Circuits IIS88
-Attention: Audio and Multimedia Departments - FDK AAC LL89
-Am Wolfsmantel 3390
-91058 Erlangen, Germany91
-92
-www.iis.fraunhofer.de/amm93
-amm-info@iis.fraunhofer.de94
------------------------------------------------------------------------------ */95
-96
-/**************************** SBR decoder library ******************************97
-98
- Author(s):99
-100
- Description:101
-102
-*******************************************************************************/103
-104
-/*!105
- \file106
- \brief CRC check coutines107
-*/108
-109
-#include "sbr_crc.h"110
-111
-#include "FDK_bitstream.h"112
-#include "transcendent.h"113
-114
-#define MAXCRCSTEP 16115
-#define MAXCRCSTEP_LD 4116
-117
-/*!118
- \brief crc calculation119
-*/120
-static ULONG calcCRC(HANDLE_CRC hCrcBuf, ULONG bValue, int nBits) {121
- int i;122
- ULONG bMask = (1UL << (nBits - 1));123
-124
- for (i = 0; i < nBits; i++, bMask >>= 1) {125
- USHORT flag = (hCrcBuf->crcState & hCrcBuf->crcMask) ? 1 : 0;126
- USHORT flag1 = (bMask & bValue) ? 1 : 0;127
-128
- flag ^= flag1;129
- hCrcBuf->crcState <<= 1;130
- if (flag) hCrcBuf->crcState ^= hCrcBuf->crcPoly;131
- }132
-133
- return (hCrcBuf->crcState);134
-}135
-136
-/*!137
- \brief crc138
-*/139
-static int getCrc(HANDLE_FDK_BITSTREAM hBs, ULONG NrBits) {140
- int i;141
- CRC_BUFFER CrcBuf;142
-143
- CrcBuf.crcState = SBR_CRC_START;144
- CrcBuf.crcPoly = SBR_CRC_POLY;145
- CrcBuf.crcMask = SBR_CRC_MASK;146
-147
- int CrcStep = NrBits >> MAXCRCSTEP_LD;148
-149
- int CrcNrBitsRest = (NrBits - CrcStep * MAXCRCSTEP);150
- ULONG bValue;151
-152
- for (i = 0; i < CrcStep; i++) {153
- bValue = FDKreadBits(hBs, MAXCRCSTEP);154
- calcCRC(&CrcBuf, bValue, MAXCRCSTEP);155
- }156
-157
- bValue = FDKreadBits(hBs, CrcNrBitsRest);158
- calcCRC(&CrcBuf, bValue, CrcNrBitsRest);159
-160
- return (CrcBuf.crcState & SBR_CRC_RANGE);161
-}162
-163
-/*!164
- \brief crc interface165
- \return 1: CRC OK, 0: CRC check failure166
-*/167
-int SbrCrcCheck(HANDLE_FDK_BITSTREAM hBs, /*!< handle to bit-buffer */168
- LONG NrBits) /*!< max. CRC length */169
-{170
- int crcResult = 1;171
- ULONG NrCrcBits;172
- ULONG crcCheckResult;173
- LONG NrBitsAvailable;174
- ULONG crcCheckSum;175
-176
- crcCheckSum = FDKreadBits(hBs, 10);177
-178
- NrBitsAvailable = FDKgetValidBits(hBs);179
- if (NrBitsAvailable <= 0) {180
- return 0;181
- }182
-183
- NrCrcBits = fixMin((INT)NrBits, (INT)NrBitsAvailable);184
-185
- crcCheckResult = getCrc(hBs, NrCrcBits);186
- FDKpushBack(hBs, (NrBitsAvailable - FDKgetValidBits(hBs)));187
-188
- if (crcCheckResult != crcCheckSum) {189
- crcResult = 0;190
- }191
-192
- return (crcResult);193
-}194
fdk-aac-2.0.1.tar.gz/libSBRdec/src/sbr_crc.h
Deleted
140
1
2
-/* -----------------------------------------------------------------------------3
-Software License for The Fraunhofer FDK AAC Codec Library for Android4
-5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
-Forschung e.V. All rights reserved.7
-8
- 1. INTRODUCTION9
-The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software10
-that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding11
-scheme for digital audio. This FDK AAC Codec software is intended to be used on12
-a wide variety of Android devices.13
-14
-AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient15
-general perceptual audio codecs. AAC-ELD is considered the best-performing16
-full-bandwidth communications codec by independent studies and is widely17
-deployed. AAC has been standardized by ISO and IEC as part of the MPEG18
-specifications.19
-20
-Patent licenses for necessary patent claims for the FDK AAC Codec (including21
-those of Fraunhofer) may be obtained through Via Licensing22
-(www.vialicensing.com) or through the respective patent owners individually for23
-the purpose of encoding or decoding bit streams in products that are compliant24
-with the ISO/IEC MPEG audio standards. Please note that most manufacturers of25
-Android devices already license these patent claims through Via Licensing or26
-directly from the patent owners, and therefore FDK AAC Codec software may27
-already be covered under those patent licenses when it is used for those28
-licensed purposes only.29
-30
-Commercially-licensed AAC software libraries, including floating-point versions31
-with enhanced sound quality, are also available from Fraunhofer. Users are32
-encouraged to check the Fraunhofer website for additional applications33
-information and documentation.34
-35
-2. COPYRIGHT LICENSE36
-37
-Redistribution and use in source and binary forms, with or without modification,38
-are permitted without payment of copyright license fees provided that you39
-satisfy the following conditions:40
-41
-You must retain the complete text of this software license in redistributions of42
-the FDK AAC Codec or your modifications thereto in source code form.43
-44
-You must retain the complete text of this software license in the documentation45
-and/or other materials provided with redistributions of the FDK AAC Codec or46
-your modifications thereto in binary form. You must make available free of47
-charge copies of the complete source code of the FDK AAC Codec and your48
-modifications thereto to recipients of copies in binary form.49
-50
-The name of Fraunhofer may not be used to endorse or promote products derived51
-from this library without prior written permission.52
-53
-You may not charge copyright license fees for anyone to use, copy or distribute54
-the FDK AAC Codec software or your modifications thereto.55
-56
-Your modified versions of the FDK AAC Codec must carry prominent notices stating57
-that you changed the software and the date of any change. For modified versions58
-of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android"59
-must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK60
-AAC Codec Library for Android."61
-62
-3. NO PATENT LICENSE63
-64
-NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without65
-limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE.66
-Fraunhofer provides no warranty of patent non-infringement with respect to this67
-software.68
-69
-You may use this FDK AAC Codec software or modifications thereto only for70
-purposes that are authorized by appropriate patent licenses.71
-72
-4. DISCLAIMER73
-74
-This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright75
-holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES,76
-including but not limited to the implied warranties of merchantability and77
-fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR78
-CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary,79
-or consequential damages, including but not limited to procurement of substitute80
-goods or services; loss of use, data, or profits, or business interruption,81
-however caused and on any theory of liability, whether in contract, strict82
-liability, or tort (including negligence), arising in any way out of the use of83
-this software, even if advised of the possibility of such damage.84
-85
-5. CONTACT INFORMATION86
-87
-Fraunhofer Institute for Integrated Circuits IIS88
-Attention: Audio and Multimedia Departments - FDK AAC LL89
-Am Wolfsmantel 3390
-91058 Erlangen, Germany91
-92
-www.iis.fraunhofer.de/amm93
-amm-info@iis.fraunhofer.de94
------------------------------------------------------------------------------ */95
-96
-/**************************** SBR decoder library ******************************97
-98
- Author(s):99
-100
- Description:101
-102
-*******************************************************************************/103
-104
-/*!105
- \file106
- \brief CRC checking routines107
-*/108
-#ifndef SBR_CRC_H109
-#define SBR_CRC_H110
-111
-#include "sbrdecoder.h"112
-113
-#include "FDK_bitstream.h"114
-115
-/* some useful crc polynoms:116
-117
-crc5: x^5+x^4+x^2+x^1+1118
-crc6: x^6+x^5+x^3+x^2+x+1119
-crc7: x^7+x^6+x^2+1120
-crc8: x^8+x^2+x+x+1121
-*/122
-123
-/* default SBR CRC */ /* G(x) = x^10 + x^9 + x^5 + x^4 + x + 1 */124
-#define SBR_CRC_POLY 0x0233125
-#define SBR_CRC_MASK 0x0200126
-#define SBR_CRC_START 0x0000127
-#define SBR_CRC_RANGE 0x03FF128
-129
-typedef struct {130
- USHORT crcState;131
- USHORT crcMask;132
- USHORT crcPoly;133
-} CRC_BUFFER;134
-135
-typedef CRC_BUFFER *HANDLE_CRC;136
-137
-int SbrCrcCheck(HANDLE_FDK_BITSTREAM hBitBuf, LONG NrCrcBits);138
-139
-#endif140
fdk-aac-2.0.1.tar.gz/.gitignore -> fdk-aac-2.0.2.tar.gz/.gitignore
Changed
20
1
2
stamp-h13
aac-enc4
compile5
+6
+CMakeCache.txt7
+CMakeFiles8
+CMakeScripts9
+Testing10
+cmake_install.cmake11
+install_manifest.txt12
+compile_commands.json13
+CTestTestfile.cmake14
+CMakeSettings.json15
+16
+*[Bb]uild*/17
+18
+.vs/19
+.vscode/20
fdk-aac-2.0.1.tar.gz/Android.bp -> fdk-aac-2.0.2.tar.gz/Android.bp
Changed
78
1
2
+// *** THIS PACKAGE HAS SPECIAL LICENSING CONDITIONS. PLEASE3
+// CONSULT THE OWNERS AND opensource-licensing@google.com BEFORE4
+// DEPENDING ON IT IN YOUR PROJECT. ***5
+package {6
+ default_applicable_licenses: ["external_aac_license"],7
+}8
+9
+// Added automatically by a large-scale-change that took the approach of10
+// 'apply every license found to every target'. While this makes sure we respect11
+// every license restriction, it may not be entirely correct.12
+//13
+// e.g. GPL in an MIT project might only apply to the contrib/ directory.14
+//15
+// Please consider splitting the single license below into multiple licenses,16
+// taking care not to lose any license_kind information, and overriding the17
+// default license using the 'licenses: [...]' property on targets as needed.18
+//19
+// For unused files, consider creating a 'fileGroup' with "//visibility:private"20
+// to attach the license to, and including a comment whether the files may be21
+// used in the current project.22
+// See: http://go/android-license-faq23
+license {24
+ name: "external_aac_license",25
+ visibility: [":__subpackages__"],26
+ license_kinds: [27
+ "SPDX-license-identifier-Apache-2.0",28
+ "legacy_by_exception_only", // by exception only29
+ ],30
+ license_text: [31
+ "NOTICE",32
+ ],33
+}34
+35
cc_library_static {36
name: "libFraunhoferAAC",37
vendor_available: true,38
+ host_supported: true,39
srcs: [40
"libAACdec/src/*.cpp",41
"libAACenc/src/*.cpp",42
43
"-Wuninitialized",44
"-Wno-self-assign",45
"-Wno-implicit-fallthrough",46
+ "-DSUPPRESS_BUILD_DATE_INFO",47
],48
sanitize: {49
- misc_undefined:[50
- "unsigned-integer-overflow",51
- "signed-integer-overflow",52
- "bounds",53
+ misc_undefined: [54
+ "unsigned-integer-overflow",55
+ "signed-integer-overflow",56
+ "bounds",57
],58
cfi: true,59
},60
61
"libSACdec/include",62
"libSACenc/include",63
],64
+65
+ target: {66
+ darwin: {67
+ enabled: false,68
+ },69
+ },70
+71
+ apex_available: [72
+ "//apex_available:platform",73
+ "com.android.bluetooth.updatable",74
+ "com.android.media.swcodec",75
+ ],76
+ min_sdk_version: "29",77
}78
fdk-aac-2.0.2.tar.gz/CMakeLists.txt
Added
201
1
2
+# # CMake project for fdk-aac3
+#4
+# ## Basic usage5
+#6
+# 1. Download and install CMake >= 3.10 (CMake >= 3.15 is recommended)7
+# 2. Clone fdk-aac repository8
+# 3. In fdk-aac directory create and go to `build` subdirectory:9
+#10
+# mkdir build11
+# cd build12
+#13
+# 4. Run cmake to configure project with desired build type (Release|Debug):14
+#15
+# cmake .. -DCMAKE_BUILD_TYPE=Release16
+#17
+# Note CMake configuration without CMAKE_BUILD_TYPE option will not set build18
+# type to some default and it is most likely is not what you want.19
+#20
+# If you want to build static (default is shared library), add BUILD_SHARED_LIBS=OFF option:21
+#22
+# # cmake .. -DBUILD_SHARED_LIBS=OFF23
+#24
+# You can combine options as well:25
+#26
+# cmake .. -DCMAKE_BUILD_TYPE=Release -DBUILD_SHARED_LIBS=OFF27
+#28
+# 5. Run cmake to build project:29
+#30
+# cmake --build . -c Release31
+32
+cmake_minimum_required(VERSION 3.5.1)33
+34
+# Policies35
+36
+## Enables CMAKE_MSVC_RUNTIME_LIBRARY option support for CMake >= 3.1537
+## if you want to use a MSVC multi-threaded statically-linked runtime library38
+## If you enable it, CMake will build fdk-acc.dll without external dependencies.39
+##40
+## Example usage:41
+##42
+## cmake .. -DCMAKE_MSVC_RUNTIME_LIBRARY="MultiThreaded$<$<CONFIG:Debug>:Debug>43
+if(POLICY CMP0091)44
+ cmake_policy(SET CMP0091 NEW)45
+endif()46
+47
+project(fdk-aac VERSION 2.0.2)48
+49
+# Includes50
+51
+include(CheckFunctionExists)52
+include(CheckLibraryExists)53
+include(GNUInstallDirs)54
+include(CMakePackageConfigHelpers)55
+56
+if(CMAKE_VERSION VERSION_LESS 3.11)57
+ set(CPACK_PACKAGE_VERSION_MAJOR ${PROJECT_VERSION_MAJOR})58
+ set(CPACK_PACKAGE_VERSION_MINOR ${PROJECT_VERSION_MINOR})59
+ set(CPACK_PACKAGE_VERSION_PATCH ${PROJECT_VERSION_PATCH})60
+endif()61
+include(CPack)62
+63
+# Options64
+65
+option(BUILD_SHARED_LIBS "Build shared library" ON)66
+option(BUILD_PROGRAMS "Build aac-enc utility" OFF)67
+option(FDK_AAC_INSTALL_CMAKE_CONFIG_MODULE "Install CMake package configuration file" ON)68
+option(FDK_AAC_INSTALL_PKGCONFIG_MODULE "Install pkg-config .pc file" ON)69
+70
+# Checks71
+72
+## Check if math functions are in separate library (Most of Linux distros, maybe some other OSes)73
+check_function_exists(sin HAVE_DEFAULT_MATH)74
+if(NOT HAVE_DEFAULT_MATH)75
+ check_library_exists(m sin "" HAVE_LIBM)76
+endif()77
+78
+# Library79
+80
+## Sources81
+82
+set(AACDEC_SRC83
+ libAACdec/src/FDK_delay.cpp84
+ libAACdec/src/aac_ram.cpp85
+ libAACdec/src/aac_rom.cpp86
+ libAACdec/src/aacdec_drc.cpp87
+ libAACdec/src/aacdec_hcr.cpp88
+ libAACdec/src/aacdec_hcr_bit.cpp89
+ libAACdec/src/aacdec_hcrs.cpp90
+ libAACdec/src/aacdec_pns.cpp91
+ libAACdec/src/aacdec_tns.cpp92
+ libAACdec/src/aacdecoder.cpp93
+ libAACdec/src/aacdecoder_lib.cpp94
+ libAACdec/src/block.cpp95
+ libAACdec/src/channel.cpp96
+ libAACdec/src/channelinfo.cpp97
+ libAACdec/src/conceal.cpp98
+ libAACdec/src/ldfiltbank.cpp99
+ libAACdec/src/pulsedata.cpp100
+ libAACdec/src/rvlc.cpp101
+ libAACdec/src/rvlcbit.cpp102
+ libAACdec/src/rvlcconceal.cpp103
+ libAACdec/src/stereo.cpp104
+ libAACdec/src/usacdec_ace_d4t64.cpp105
+ libAACdec/src/usacdec_ace_ltp.cpp106
+ libAACdec/src/usacdec_acelp.cpp107
+ libAACdec/src/usacdec_fac.cpp108
+ libAACdec/src/usacdec_lpc.cpp109
+ libAACdec/src/usacdec_lpd.cpp110
+ libAACdec/src/usacdec_rom.cpp111
+ libAACdec/src/aac_ram.h112
+ libAACdec/src/aac_rom.h113
+ libAACdec/src/aacdec_drc.h114
+ libAACdec/src/aacdec_drc_types.h115
+ libAACdec/src/aacdec_hcr.h116
+ libAACdec/src/aacdec_hcr_bit.h117
+ libAACdec/src/aacdec_hcr_types.h118
+ libAACdec/src/aacdec_hcrs.h119
+ libAACdec/src/aacdec_pns.h120
+ libAACdec/src/aacdec_tns.h121
+ libAACdec/src/aacdecoder.h122
+ libAACdec/src/block.h123
+ libAACdec/src/channel.h124
+ libAACdec/src/channelinfo.h125
+ libAACdec/src/conceal.h126
+ libAACdec/src/conceal_types.h127
+ libAACdec/src/FDK_delay.h128
+ libAACdec/src/ldfiltbank.h129
+ libAACdec/src/overlapadd.h130
+ libAACdec/src/pulsedata.h131
+ libAACdec/src/rvlc.h132
+ libAACdec/src/rvlc_info.h133
+ libAACdec/src/rvlcbit.h134
+ libAACdec/src/rvlcconceal.h135
+ libAACdec/src/stereo.h136
+ libAACdec/src/usacdec_ace_d4t64.h137
+ libAACdec/src/usacdec_ace_ltp.h138
+ libAACdec/src/usacdec_acelp.h139
+ libAACdec/src/usacdec_const.h140
+ libAACdec/src/usacdec_fac.h141
+ libAACdec/src/usacdec_lpc.h142
+ libAACdec/src/usacdec_lpd.h143
+ libAACdec/src/usacdec_rom.h)144
+145
+set(AACENC_SRC146
+ libAACenc/src/aacEnc_ram.cpp147
+ libAACenc/src/aacEnc_rom.cpp148
+ libAACenc/src/aacenc.cpp149
+ libAACenc/src/aacenc_lib.cpp150
+ libAACenc/src/aacenc_pns.cpp151
+ libAACenc/src/aacenc_tns.cpp152
+ libAACenc/src/adj_thr.cpp153
+ libAACenc/src/band_nrg.cpp154
+ libAACenc/src/bandwidth.cpp155
+ libAACenc/src/bit_cnt.cpp156
+ libAACenc/src/bitenc.cpp157
+ libAACenc/src/block_switch.cpp158
+ libAACenc/src/channel_map.cpp159
+ libAACenc/src/chaosmeasure.cpp160
+ libAACenc/src/dyn_bits.cpp161
+ libAACenc/src/grp_data.cpp162
+ libAACenc/src/intensity.cpp163
+ libAACenc/src/line_pe.cpp164
+ libAACenc/src/metadata_compressor.cpp165
+ libAACenc/src/metadata_main.cpp166
+ libAACenc/src/mps_main.cpp167
+ libAACenc/src/ms_stereo.cpp168
+ libAACenc/src/noisedet.cpp169
+ libAACenc/src/pnsparam.cpp170
+ libAACenc/src/pre_echo_control.cpp171
+ libAACenc/src/psy_configuration.cpp172
+ libAACenc/src/psy_main.cpp173
+ libAACenc/src/qc_main.cpp174
+ libAACenc/src/quantize.cpp175
+ libAACenc/src/sf_estim.cpp176
+ libAACenc/src/spreading.cpp177
+ libAACenc/src/tonality.cpp178
+ libAACenc/src/transform.cpp179
+ libAACenc/src/aacenc.h180
+ libAACenc/src/aacenc_pns.h181
+ libAACenc/src/aacEnc_ram.h182
+ libAACenc/src/aacEnc_rom.h183
+ libAACenc/src/aacenc_tns.h184
+ libAACenc/src/adj_thr.h185
+ libAACenc/src/adj_thr_data.h186
+ libAACenc/src/band_nrg.h187
+ libAACenc/src/bandwidth.h188
+ libAACenc/src/bit_cnt.h189
+ libAACenc/src/bitenc.h190
+ libAACenc/src/block_switch.h191
+ libAACenc/src/channel_map.h192
+ libAACenc/src/chaosmeasure.h193
+ libAACenc/src/dyn_bits.h194
+ libAACenc/src/grp_data.h195
+ libAACenc/src/intensity.h196
+ libAACenc/src/interface.h197
+ libAACenc/src/line_pe.h198
+ libAACenc/src/metadata_compressor.h199
+ libAACenc/src/metadata_main.h200
+ libAACenc/src/mps_main.h201
fdk-aac-2.0.1.tar.gz/ChangeLog -> fdk-aac-2.0.2.tar.gz/ChangeLog
Changed
11
1
2
+2.0.23
+ - Minor upstream updates4
+ - Lots of upstream and local fuzzing fixes5
+ - Added CMake project files6
+ - Removed the MSVC specific makefile7
+8
2.0.19
- Minor release with a number of crash/fuzz fixes, primarily for the decoder10
11
fdk-aac-2.0.2.tar.gz/METADATA
Added
5
1
2
+third_party {3
+ license_type: BY_EXCEPTION_ONLY4
+}5
fdk-aac-2.0.1.tar.gz/Makefile.am -> fdk-aac-2.0.2.tar.gz/Makefile.am
Changed
37
1
2
libSBRdec/src/psdec_drm.cpp \3
libSBRdec/src/psdecrom_drm.cpp \4
libSBRdec/src/pvc_dec.cpp \5
- libSBRdec/src/sbr_crc.cpp \6
libSBRdec/src/sbr_deb.cpp \7
libSBRdec/src/sbr_dec.cpp \8
libSBRdec/src/sbr_ram.cpp \9
10
$(PCMUTILS_SRC) $(FDK_SRC) $(SYS_SRC)11
12
EXTRA_DIST = \13
+ $(top_srcdir)/CMakeLists.txt \14
$(top_srcdir)/.clang-format \15
$(top_srcdir)/autogen.sh \16
$(top_srcdir)/MODULE_LICENSE_FRAUNHOFER \17
$(top_srcdir)/NOTICE \18
$(top_srcdir)/OWNERS \19
$(top_srcdir)/Android.bp \20
+ $(top_srcdir)/fdk-aac.def \21
$(top_srcdir)/fdk-aac.sym \22
- $(top_srcdir)/Makefile.vc \23
+ $(top_srcdir)/fdk-aac-config.cmake.in \24
$(top_srcdir)/documentation/*.pdf \25
+ $(top_srcdir)/fuzzer/* \26
$(top_srcdir)/libAACdec/src/*.h \27
$(top_srcdir)/libAACdec/src/arm/*.cpp \28
$(top_srcdir)/libAACenc/src/*.h \29
30
$(top_srcdir)/libFDK/include/x86/*.h \31
$(top_srcdir)/libFDK/src/arm/*.cpp \32
$(top_srcdir)/libFDK/src/mips/*.cpp \33
+ $(top_srcdir)/METADATA \34
+ $(top_srcdir)/PREUPLOAD.cfg \35
$(top_srcdir)/win32/*.h36
37
fdk-aac-2.0.2.tar.gz/PREUPLOAD.cfg
Added
4
1
2
+[Hook Scripts]3
+mainline_hook = ${REPO_ROOT}/frameworks/av/tools/mainline_hook_project.sh4
fdk-aac-2.0.1.tar.gz/configure.ac -> fdk-aac-2.0.2.tar.gz/configure.ac
Changed
19
1
2
dnl -*- Autoconf -*-3
dnl Process this file with autoconf to produce a configure script.4
5
-AC_INIT([fdk-aac], [2.0.1], [http://sourceforge.net/projects/opencore-amr/])6
+AC_INIT([fdk-aac], [2.0.2], [http://sourceforge.net/projects/opencore-amr/])7
AC_CONFIG_AUX_DIR(.)8
AC_CONFIG_MACRO_DIR([m4])9
AM_INIT_AUTOMAKE([tar-ustar foreign])10
11
dnl soname version to use12
dnl goes by ‘current[:revision[:age]]’ with the soname ending up as13
dnl current.age.revision14
-FDK_AAC_VERSION=2:1:015
+FDK_AAC_VERSION=2:2:016
17
AS_IF([test x$enable_shared = xyes], [LIBS_PRIVATE=$LIBS], [LIBS_PUBLIC=$LIBS])18
AC_SUBST(FDK_AAC_VERSION)19
fdk-aac-2.0.1.tar.gz/documentation/aacDecoder.pdf -> fdk-aac-2.0.2.tar.gz/documentation/aacDecoder.pdf
Changed
fdk-aac-2.0.1.tar.gz/documentation/aacEncoder.pdf -> fdk-aac-2.0.2.tar.gz/documentation/aacEncoder.pdf
Changed
fdk-aac-2.0.2.tar.gz/fdk-aac-config.cmake.in
Added
7
1
2
+@PACKAGE_INIT@3
+4
+include(${CMAKE_CURRENT_LIST_DIR}/fdk-aac-targets.cmake)5
+6
+check_required_components(FDK-AAC)7
fdk-aac-2.0.2.tar.gz/fdk-aac.def
Added
21
1
2
+EXPORTS3
+ aacDecoder_AncDataGet4
+ aacDecoder_AncDataInit5
+ aacDecoder_Close6
+ aacDecoder_ConfigRaw7
+ aacDecoder_DecodeFrame8
+ aacDecoder_Fill9
+ aacDecoder_GetFreeBytes10
+ aacDecoder_GetLibInfo11
+ aacDecoder_GetStreamInfo12
+ aacDecoder_Open13
+ aacDecoder_SetParam14
+ aacEncClose15
+ aacEncEncode16
+ aacEncGetLibInfo17
+ aacEncInfo18
+ aacEncOpen19
+ aacEncoder_GetParam20
+ aacEncoder_SetParam21
fdk-aac-2.0.2.tar.gz/fuzzer
Added
2
1
+(directory)2
fdk-aac-2.0.2.tar.gz/fuzzer/Android.bp
Added
84
1
2
+/******************************************************************************3
+ *4
+ * Copyright (C) 2020 The Android Open Source Project5
+ *6
+ * Licensed under the Apache License, Version 2.0 (the "License");7
+ * you may not use this file except in compliance with the License.8
+ * You may obtain a copy of the License at:9
+ *10
+ * http://www.apache.org/licenses/LICENSE-2.011
+ *12
+ * Unless required by applicable law or agreed to in writing, software13
+ * distributed under the License is distributed on an "AS IS" BASIS,14
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.15
+ * See the License for the specific language governing permissions and16
+ * limitations under the License.17
+ *18
+ *****************************************************************************19
+ * Originally developed and contributed by Ittiam Systems Pvt. Ltd, Bangalore20
+ */21
+22
+package {23
+ // See: http://go/android-license-faq24
+ // A large-scale-change added 'default_applicable_licenses' to import25
+ // all of the 'license_kinds' from "external_aac_license"26
+ // to get the below license kinds:27
+ // SPDX-license-identifier-Apache-2.028
+ default_applicable_licenses: ["external_aac_license"],29
+}30
+31
+cc_defaults {32
+ name: "aac_fuzz_defaults",33
+ host_supported: true,34
+35
+ static_libs: [36
+ "libFraunhoferAAC",37
+ ],38
+39
+ target: {40
+ darwin: {41
+ enabled: false,42
+ },43
+ },44
+45
+ fuzz_config: {46
+ cc: [47
+ "android-media-fuzzing-reports@google.com",48
+ ],49
+ componentid: 155276,50
+ },51
+}52
+53
+cc_fuzz {54
+ name: "aac_dec_fuzzer",55
+56
+ srcs: [57
+ "aac_dec_fuzzer.cpp",58
+ ],59
+60
+ static_libs: [61
+ "liblog",62
+ ],63
+64
+ defaults: [65
+ "aac_fuzz_defaults"66
+ ],67
+}68
+69
+cc_fuzz {70
+ name: "aac_enc_fuzzer",71
+72
+ srcs: [73
+ "aac_enc_fuzzer.cpp",74
+ ],75
+76
+ defaults: [77
+ "aac_fuzz_defaults"78
+ ],79
+80
+ include_dirs: [81
+ "external/aac/libAACenc/"82
+ ],83
+}84
fdk-aac-2.0.2.tar.gz/fuzzer/README.md
Added
152
1
2
+# Fuzzer for libFraunhoferAAC decoder3
+4
+## Plugin Design Considerations5
+The fuzzer plugin for aac decoder is designed based on the understanding of the6
+codec and tries to achieve the following:7
+8
+##### Maximize code coverage9
+10
+This fuzzer makes use of the following config parameters:11
+1. Transport type (parameter name: `TRANSPORT_TYPE`)12
+13
+| Parameter| Valid Values| Configured Value|14
+|------------- |-------------| ----- |15
+| `TRANSPORT_TYPE` | 0.`TT_UNKNOWN ` 1.`TT_MP4_RAW ` 2.`TT_MP4_ADIF ` 3.`TT_MP4_ADTS ` 4.`TT_MP4_LATM_MCP1 ` 5.`TT_MP4_LATM_MCP0 ` 6.`TT_MP4_LOAS ` 7.`TT_DRM ` | `TT_MP4_ADIF ` |16
+17
+Note: Value of `TRANSPORT_TYPE` could be set to any of these values.18
+It is set to `TT_MP4_ADIF` in the fuzzer plugin.19
+20
+##### Maximize utilization of input data21
+The plugin feeds the entire input data to the codec using a loop.22
+ * If the decode operation was successful, the input is advanced by an23
+ offset calculated using valid bytes.24
+ * If the decode operation was un-successful, the input is advanced by 1 byte25
+ till it reaches a valid frame or end of stream.26
+27
+This ensures that the plugin tolerates any kind of input (empty, huge,28
+malformed, etc) and doesnt `exit()` on any input and thereby increasing the29
+chance of identifying vulnerabilities.30
+31
+## Build32
+33
+This describes steps to build aac_dec_fuzzer binary.34
+35
+## Android36
+37
+### Steps to build38
+Build the fuzzer39
+```40
+ $ mm -j$(nproc) aac_dec_fuzzer41
+```42
+43
+### Steps to run44
+Create a directory CORPUS_DIR and copy some aac files to that folder.45
+Push this directory to device.46
+47
+To run on device48
+```49
+ $ adb sync data50
+ $ adb shell /data/fuzz/arm64/aac_dec_fuzzer/aac_dec_fuzzer CORPUS_DIR51
+```52
+To run on host53
+```54
+ $ $ANDROID_HOST_OUT/fuzz/x86_64/aac_dec_fuzzer/aac_dec_fuzzer CORPUS_DIR55
+```56
+57
+# Fuzzer for libFraunhoferAAC encoder58
+59
+## Plugin Design Considerations60
+The fuzzer plugin for aac encoder is designed based on the understanding of the61
+codec and tries to achieve the following:62
+63
+##### Maximize code coverage64
+65
+The configuration parameters are not hardcoded, but instead selected based on66
+incoming data. This ensures more code paths are reached by the fuzzer.67
+68
+Following arguments are passed to aacEncoder_SetParam to set the respective AACENC_PARAM parameter:69
+70
+| AACENC_PARAM param| Valid Values| Configured Value|71
+|-------------| ----- |----- |72
+|`AACENC_SBR_MODE` | `-1 ` `0 ` `1 ` `2 ` | Calculated using first byte of data |73
+|`AACENC_AOT` |`AOT_NONE ` `AOT_NULL_OBJECT ` `AOT_AAC_MAIN ` `AOT_AAC_LC ` `AOT_AAC_SSR ` `AOT_AAC_LTP ` `AOT_SBR ` `AOT_AAC_SCAL ` `AOT_TWIN_VQ ` `AOT_CELP ` `AOT_HVXC ` `AOT_RSVD_10 ` `AOT_RSVD_11 ` `AOT_TTSI ` `AOT_MAIN_SYNTH ` `AOT_WAV_TAB_SYNTH ` `AOT_GEN_MIDI ` `AOT_ALG_SYNTH_AUD_FX ` `AOT_ER_AAC_LC ` `AOT_RSVD_18 ` `AOT_ER_AAC_LTP ` `AOT_ER_AAC_SCAL ` `AOT_ER_TWIN_VQ ` `AOT_ER_BSAC ` `AOT_ER_AAC_LD ` `AOT_ER_CELP ` `AOT_ER_HVXC ` `AOT_ER_HILN ` `AOT_ER_PARA ` `AOT_RSVD_28 ` `AOT_PS ` `AOT_MPEGS ` `AOT_ESCAPE ` `AOT_MP3ONMP4_L1 ` `AOT_MP3ONMP4_L2 ` `AOT_MP3ONMP4_L3 ` `AOT_RSVD_35 ` `AOT_RSVD_36 ` `AOT_AAC_SLS ` `AOT_SLS ` `AOT_ER_AAC_ELD ` `AOT_USAC ` `AOT_SAOC ` `AOT_LD_MPEGS ` `AOT_MP2_AAC_LC ` `AOT_MP2_SBR ` `AOT_DRM_AAC ` `AOT_DRM_SBR ` `AOT_DRM_MPEG_PS ` `AOT_DRM_SURROUND ` `AOT_DRM_USAC ` | Calculated using second byte of data |74
+|`AACENC_SAMPLERATE` | `8000 ` `11025 ` `12000 ` `16000 ` `22050 ` `24000 ` `32000 ` `44100 ` `48000 ` `64000 ` `88200 ` `96000 `| Calculated using third byte of data |75
+|`AACENC_BITRATE` | In range `8000 ` to `960000 ` | Calculated using fourth, fifth and sixth byte of data |76
+|`AACENC_CHANNELMODE` | `MODE_1 ` `MODE_2 ` `MODE_1_2 ` `MODE_1_2_1 ` `MODE_1_2_2 ` `MODE_1_2_2_1 ` `MODE_1_2_2_2_1 ` `MODE_6_1 ` `MODE_7_1_BACK ` `MODE_7_1_TOP_FRONT ` `MODE_7_1_REAR_SURROUND ` `MODE_7_1_FRONT_CENTER ` `MODE_212 ` | Calculated using seventh byte of data |77
+|`AACENC_TRANSMUX` | `TT_MP4_RAW ` `TT_MP4_ADIF ` `TT_MP4_ADTS ` `TT_MP4_LATM_MCP1 ` `TT_MP4_LATM_MCP0 ` `TT_MP4_LOAS ` `TT_DRM ` | Calculated using eight byte of data |`AACENC_SBR_RATIO` |`-1 ` `0 ` `1 ` `2 ` | Calculated using ninth byte of data |78
+|`AACENC_BITRATEMODE` |`AACENC_BR_MODE_INVALID ` `AACENC_BR_MODE_CBR ` `AACENC_BR_MODE_VBR_1 ` `AACENC_BR_MODE_VBR_2 ` `AACENC_BR_MODE_VBR_3 ` `AACENC_BR_MODE_VBR_4 ` `AACENC_BR_MODE_VBR_5 ` `AACENC_BR_MODE_FF ` `AACENC_BR_MODE_SFR ` | Calculated using thirty-fourth byte of data |79
+|`AACENC_GRANULE_LENGTH` |`120 ` `128 ` `240 ` `256 ` `480 ` `512 ` `1024 ` | Calculated using thirty-fifth byte of data |80
+|`AACENC_CHANNELORDER` |`CH_ORDER_MPEG ` `CH_ORDER_WAV ` | Calculated using thirty-sixth byte of data |81
+|`AACENC_AFTERBURNER` |`0 ` `1 ` | Calculated using thirty-seventh byte of data |82
+|`AACENC_BANDWIDTH` |`0 ` `1` | Calculated using thirty-eigth byte of data |83
+|` AACENC_IDX_PEAK_BITRATE` | In range `8000 ` to `960000 ` | Calculated using thirty-ninth byte of data |84
+|` AACENC_HEADER_PERIOD` |In range `0 ` to `255 ` | Calculated using fortieth byte of data |85
+|` AACENC_SIGNALING_MODE` |`-1 ` `0 ` `1 ` `2 ` `3 ` | Calculated using forty-first byte of data |86
+|` AACENC_TPSUBFRAMES` |In range `0 ` to `255 ` | Calculated using forty-second byte of data |87
+|` AACENC_AUDIOMUXVER` |`-1 ` `0 ` `1 ` `2 ` | Calculated using forty-third byte of data |88
+|` AACENC_PROTECTION` |`0 ` `1 ` | Calculated using forty-fourth of data |89
+|`AACENC_ANCILLARY_BITRATE` |In range `0 ` to `960000 `| Calculated using forty-fifth byte of data |90
+|`AACENC_METADATA_MODE ` |`0 ` `1 ` `2 ` `3 ` | Calculated using forty-sixth byte of data |91
+92
+Following values are configured to set up the meta data represented by the class variable `mMetaData ` :93
+94
+| Variable name| Possible Values| Configured Value|95
+|------------- | ----- |----- |96
+| `drc_profile` | `AACENC_METADATA_DRC_NONE ` `AACENC_METADATA_DRC_FILMSTANDARD ` `AACENC_METADATA_DRC_FILMLIGHT ` `AACENC_METADATA_DRC_MUSICSTANDARD ` `AACENC_METADATA_DRC_MUSICLIGHT ` `AACENC_METADATA_DRC_SPEECH ` `AACENC_METADATA_DRC_NOT_PRESENT ` | Calculated using tenth byte of data |97
+| `comp_profile` | `AACENC_METADATA_DRC_NONE ` `AACENC_METADATA_DRC_FILMSTANDARD ` `AACENC_METADATA_DRC_FILMLIGHT ` `AACENC_METADATA_DRC_MUSICSTANDARD ` `AACENC_METADATA_DRC_MUSICLIGHT ` `AACENC_METADATA_DRC_SPEECH ` `AACENC_METADATA_DRC_NOT_PRESENT ` | Calculated using eleventh byte of data |98
+| `drc_TargetRefLevel` | In range `0 ` to `255 ` | Calculated using twelfth byte of data |99
+| `comp_TargetRefLevel` | In range `0 ` to `255 ` | Calculated using thirteenth byte of data |100
+| `prog_ref_level_present` | `0 ` `1 ` | Calculated using fourteenth byte of data |101
+| `prog_ref_level` | In range `0 ` to `255 ` | Calculated using fifteenth byte of data |102
+| `PCE_mixdown_idx_present` | `0 ` `1 ` | Calculated using sixteenth byte of data |103
+| `ETSI_DmxLvl_present` | `0 ` `1 ` | Calculated using seventeenth byte of data |104
+| `centerMixLevel` | In range `0 ` to `7 ` | Calculated using eighteenth byte of data |105
+| `surroundMixLevel` | In range `0 ` to `7 ` | Calculated using nineteenth byte of data |106
+| `dolbySurroundMode` | In range `0 ` to `2 ` | Calculated using twentieth byte of data |107
+| `drcPresentationMode` | In range `0 ` to `2 ` | Calculated using twenty-first byte of data |108
+| `extAncDataEnable` | `0 ` `1 ` | Calculated using twenty-second byte of data |109
+| `extDownmixLevelEnable` | `0 ` `1 ` | Calculated using twenty-third byte of data |110
+| `extDownmixLevel_A` | In range `0 ` to `7 ` | Calculated using twenty-fourth byte of data |111
+| `extDownmixLevel_B` | In range `0 ` to `7 ` | Calculated using twenty-fifth byte of data |112
+| `dmxGainEnable` | `0 ` `1 ` | Calculated using twenty-sixth byte of data |113
+| `dmxGain5` | In range `0 ` to `255 ` | Calculated using twenty-seventh byte of data |114
+| `dmxGain2` | In range `0 ` to `255 ` | Calculated using twenty-eighth byte of data |115
+| `lfeDmxEnable` | `0 ` `1 ` | Calculated using twenty-ninth byte of data |116
+| `lfeDmxLevel` | In range `0 ` to `15 ` | Calculated using thirtieth byte of data |117
+118
+Indexes `mInBufferIdx_1`, `mInBufferIdx_2` and `mInBufferIdx_3`(in range `0 ` to `2`) are calculated using the thirty-first, thirty-second and thirty-third byte respectively.119
+120
+##### Maximize utilization of input data121
+The plugin feeds the entire input data to the codec and continues with the encoding even on a failure. This ensures that the plugin tolerates any kind of input (empty, huge, malformed, etc) and doesnt `exit()` on any input and thereby increasing the chance of identifying vulnerabilities.122
+123
+## Build124
+125
+This describes steps to build aac_enc_fuzzer binary.126
+127
+## Android128
+129
+### Steps to build130
+Build the fuzzer131
+```132
+ $ mm -j$(nproc) aac_enc_fuzzer133
+```134
+135
+### Steps to run136
+Create a directory CORPUS_DIR and copy some raw files to that folder.137
+Push this directory to device.138
+139
+To run on device140
+```141
+ $ adb sync data142
+ $ adb shell /data/fuzz/arm64/aac_enc_fuzzer/aac_enc_fuzzer CORPUS_DIR143
+```144
+To run on host145
+```146
+ $ $ANDROID_HOST_OUT/fuzz/x86_64/aac_enc_fuzzer/aac_enc_fuzzer CORPUS_DIR147
+```148
+149
+## References:150
+ * http://llvm.org/docs/LibFuzzer.html151
+ * https://github.com/google/oss-fuzz152
fdk-aac-2.0.2.tar.gz/fuzzer/aac_dec_fuzzer.cpp
Added
142
1
2
+/******************************************************************************3
+ *4
+ * Copyright (C) 2020 The Android Open Source Project5
+ *6
+ * Licensed under the Apache License, Version 2.0 (the "License");7
+ * you may not use this file except in compliance with the License.8
+ * You may obtain a copy of the License at:9
+ *10
+ * http://www.apache.org/licenses/LICENSE-2.011
+ *12
+ * Unless required by applicable law or agreed to in writing, software13
+ * distributed under the License is distributed on an "AS IS" BASIS,14
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.15
+ * See the License for the specific language governing permissions and16
+ * limitations under the License.17
+ *18
+ *****************************************************************************19
+ * Originally developed and contributed by Ittiam Systems Pvt. Ltd, Bangalore20
+ */21
+22
+#include <stdint.h>23
+#include <string.h>24
+#include <algorithm>25
+#include "aacdecoder_lib.h"26
+27
+constexpr uint8_t kNumberOfLayers = 1;28
+constexpr uint8_t kMaxChannelCount = 8;29
+constexpr uint32_t kMaxConfigurationSize = 1024;30
+constexpr uint32_t kMaxOutBufferSize = 2048 * kMaxChannelCount;31
+32
+// Value indicating the start of AAC Header Segment33
+constexpr const char *kAacSegStartSeq = "AAC_STRT";34
+constexpr uint8_t kAacSegStartSeqLen = sizeof(kAacSegStartSeq);35
+// Value indicating the end of AAC Header Segment36
+constexpr const char *kAacSegEndSeq = "AAC_ENDS";37
+constexpr uint8_t kAacSegEndSeqLen = sizeof(kAacSegEndSeq);38
+39
+// Number of bytes used to signal the length of the header40
+constexpr uint8_t kHeaderLengthBytes = 2;41
+// Minimum size of an AAC header is 242
+// Minimum data required is43
+// strlen(AAC_STRT) + strlen(AAC_ENDS) + kHeaderLengthBytes + 2;44
+constexpr UINT kMinDataSize = kAacSegStartSeqLen + kAacSegEndSeqLen + kHeaderLengthBytes + 2;45
+46
+UINT getHeaderSize(UCHAR *data, UINT size) {47
+ if (size < kMinDataSize) {48
+ return 0;49
+ }50
+51
+ int32_t result = memcmp(data, kAacSegStartSeq, kAacSegStartSeqLen);52
+ if (result) {53
+ return 0;54
+ }55
+ data += kAacSegStartSeqLen;56
+ size -= kAacSegStartSeqLen;57
+58
+ uint32_t headerLengthInBytes = (data[0] << 8 | data[1]) & 0xFFFF;59
+ data += kHeaderLengthBytes;60
+ size -= kHeaderLengthBytes;61
+62
+ if (headerLengthInBytes + kAacSegEndSeqLen > size) {63
+ return 0;64
+ }65
+66
+ data += headerLengthInBytes;67
+ size -= headerLengthInBytes;68
+ result = memcmp(data, kAacSegEndSeq, kAacSegEndSeqLen);69
+ if (result) {70
+ return 0;71
+ }72
+73
+ return std::min(headerLengthInBytes, kMaxConfigurationSize);74
+}75
+76
+class Codec {77
+ public:78
+ Codec() = default;79
+ ~Codec() { deInitDecoder(); }80
+ bool initDecoder();81
+ void decodeFrames(UCHAR *data, UINT size);82
+ void deInitDecoder();83
+84
+ private:85
+ HANDLE_AACDECODER mAacDecoderHandle = nullptr;86
+ AAC_DECODER_ERROR mErrorCode = AAC_DEC_OK;87
+};88
+89
+bool Codec::initDecoder() {90
+ mAacDecoderHandle = aacDecoder_Open(TT_MP4_ADIF, kNumberOfLayers);91
+ if (!mAacDecoderHandle) {92
+ return false;93
+ }94
+ return true;95
+}96
+97
+void Codec::deInitDecoder() {98
+ aacDecoder_Close(mAacDecoderHandle);99
+ mAacDecoderHandle = nullptr;100
+}101
+102
+void Codec::decodeFrames(UCHAR *data, UINT size) {103
+ UINT headerSize = getHeaderSize(data, size);104
+ if (headerSize != 0) {105
+ data += kAacSegStartSeqLen + kHeaderLengthBytes;106
+ size -= kAacSegStartSeqLen + kHeaderLengthBytes;107
+ aacDecoder_ConfigRaw(mAacDecoderHandle, &data, &headerSize);108
+ data += headerSize + kAacSegEndSeqLen;109
+ size -= headerSize + kAacSegEndSeqLen;110
+ }111
+ while (size > 0) {112
+ UINT inputSize = size;113
+ UINT valid = size;114
+ mErrorCode = aacDecoder_Fill(mAacDecoderHandle, &data, &inputSize, &valid);115
+ if (mErrorCode != AAC_DEC_OK) {116
+ ++data;117
+ --size;118
+ } else {119
+ INT_PCM outputBuf[kMaxOutBufferSize];120
+ do {121
+ mErrorCode =122
+ aacDecoder_DecodeFrame(mAacDecoderHandle, outputBuf, sizeof(outputBuf), 0);123
+ } while (mErrorCode == AAC_DEC_OK);124
+ UINT offset = inputSize - valid;125
+ data += offset;126
+ size = valid;127
+ }128
+ }129
+}130
+131
+extern "C" int LLVMFuzzerTestOneInput(const uint8_t *data, size_t size) {132
+ Codec *codec = new Codec();133
+ if (!codec) {134
+ return 0;135
+ }136
+ if (codec->initDecoder()) {137
+ codec->decodeFrames((UCHAR *)(data), static_cast<UINT>(size));138
+ }139
+ delete codec;140
+ return 0;141
+}142
fdk-aac-2.0.2.tar.gz/fuzzer/aac_enc_fuzzer.cpp
Added
201
1
2
+/******************************************************************************3
+ *4
+ * Copyright (C) 2020 The Android Open Source Project5
+ *6
+ * Licensed under the Apache License, Version 2.0 (the "License");7
+ * you may not use this file except in compliance with the License.8
+ * You may obtain a copy of the License at:9
+ *10
+ * http://www.apache.org/licenses/LICENSE-2.011
+ *12
+ * Unless required by applicable law or agreed to in writing, software13
+ * distributed under the License is distributed on an "AS IS" BASIS,14
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.15
+ * See the License for the specific language governing permissions and16
+ * limitations under the License.17
+ *18
+ *****************************************************************************19
+ * Originally developed and contributed by Ittiam Systems Pvt. Ltd, Bangalore20
+ */21
+22
+#include <string>23
+#include "aacenc_lib.h"24
+#include "src/aacenc.h"25
+26
+using namespace std;27
+28
+// IN_AUDIO_DATA, IN_ANCILLRY_DATA and IN_METADATA_SETUP29
+constexpr size_t kMaxBuffers = 3;30
+31
+constexpr size_t kMaxOutputBufferSize = 8192;32
+33
+constexpr uint32_t kMinBitRate = 8000;34
+constexpr uint32_t kMaxBitRate = 960000;35
+36
+constexpr int32_t kSampleRates[] = {8000, 11025, 12000, 16000, 22050, 24000,37
+ 32000, 44100, 48000, 64000, 88200, 96000};38
+constexpr size_t kSampleRatesSize = size(kSampleRates);39
+40
+constexpr CHANNEL_MODE kChannelModes[] = {MODE_1,41
+ MODE_2,42
+ MODE_1_2,43
+ MODE_1_2_1,44
+ MODE_1_2_2,45
+ MODE_1_2_2_1,46
+ MODE_1_2_2_2_1,47
+ MODE_6_1,48
+ MODE_7_1_BACK,49
+ MODE_7_1_TOP_FRONT,50
+ MODE_7_1_REAR_SURROUND,51
+ MODE_7_1_FRONT_CENTER,52
+ MODE_212};53
+constexpr size_t kChannelModesSize = size(kChannelModes);54
+55
+constexpr TRANSPORT_TYPE kIdentifiers[] = {56
+ TT_MP4_RAW, TT_MP4_ADIF, TT_MP4_ADTS, TT_MP4_LATM_MCP1, TT_MP4_LATM_MCP0, TT_MP4_LOAS, TT_DRM};57
+constexpr size_t kIdentifiersSize = size(kIdentifiers);58
+59
+constexpr AUDIO_OBJECT_TYPE kAudioObjectTypes[] = {AOT_NONE, AOT_NULL_OBJECT,60
+ AOT_AAC_MAIN, AOT_AAC_LC,61
+ AOT_AAC_SSR, AOT_AAC_LTP,62
+ AOT_SBR, AOT_AAC_SCAL,63
+ AOT_TWIN_VQ, AOT_CELP,64
+ AOT_HVXC, AOT_RSVD_10,65
+ AOT_RSVD_11, AOT_TTSI,66
+ AOT_MAIN_SYNTH, AOT_WAV_TAB_SYNTH,67
+ AOT_GEN_MIDI, AOT_ALG_SYNTH_AUD_FX,68
+ AOT_ER_AAC_LC, AOT_RSVD_18,69
+ AOT_ER_AAC_LTP, AOT_ER_AAC_SCAL,70
+ AOT_ER_TWIN_VQ, AOT_ER_BSAC,71
+ AOT_ER_AAC_LD, AOT_ER_CELP,72
+ AOT_ER_HVXC, AOT_ER_HILN,73
+ AOT_ER_PARA, AOT_RSVD_28,74
+ AOT_PS, AOT_MPEGS,75
+ AOT_ESCAPE, AOT_MP3ONMP4_L1,76
+ AOT_MP3ONMP4_L2, AOT_MP3ONMP4_L3,77
+ AOT_RSVD_35, AOT_RSVD_36,78
+ AOT_AAC_SLS, AOT_SLS,79
+ AOT_ER_AAC_ELD, AOT_USAC,80
+ AOT_SAOC, AOT_LD_MPEGS,81
+ AOT_MP2_AAC_LC, AOT_MP2_SBR,82
+ AOT_DRM_AAC, AOT_DRM_SBR,83
+ AOT_DRM_MPEG_PS, AOT_DRM_SURROUND,84
+ AOT_DRM_USAC};85
+86
+constexpr size_t kAudioObjectTypesSize = size(kAudioObjectTypes);87
+88
+constexpr int32_t kSbrRatios[] = {-1, 0, 1, 2};89
+constexpr size_t kSbrRatiosSize = size(kSbrRatios);90
+91
+constexpr int32_t kBitRateModes[] = {92
+ AACENC_BR_MODE_INVALID, AACENC_BR_MODE_CBR, AACENC_BR_MODE_VBR_1,93
+ AACENC_BR_MODE_VBR_2, AACENC_BR_MODE_VBR_3, AACENC_BR_MODE_VBR_4,94
+ AACENC_BR_MODE_VBR_5, AACENC_BR_MODE_FF, AACENC_BR_MODE_SFR};95
+constexpr size_t kBitRateModesSize = size(kBitRateModes);96
+97
+constexpr int32_t kGranuleLengths[] = {120, 128, 240, 256, 480, 512, 1024};98
+constexpr size_t kGranuleLengthsSize = size(kGranuleLengths);99
+100
+constexpr int32_t kChannelOrder[] = {CH_ORDER_MPEG, CH_ORDER_WAV};101
+constexpr size_t kChannelOrderSize = size(kChannelOrder);102
+103
+constexpr int32_t kSignalingModes[] = {-1, 0, 1, 2, 3};104
+constexpr size_t kSignalingModesSize = size(kSignalingModes);105
+106
+constexpr int32_t kAudioMuxVer[] = {-1, 0, 1, 2};107
+constexpr size_t kAudioMuxVerSize = size(kAudioMuxVer);108
+109
+constexpr int32_t kSbrModes[] = {-1, 0, 1, 2};110
+constexpr size_t kSbrModesSize = size(kSbrModes);111
+112
+constexpr AACENC_METADATA_DRC_PROFILE kMetaDataDrcProfiles[] = {113
+ AACENC_METADATA_DRC_NONE, AACENC_METADATA_DRC_FILMSTANDARD,114
+ AACENC_METADATA_DRC_FILMLIGHT, AACENC_METADATA_DRC_MUSICSTANDARD,115
+ AACENC_METADATA_DRC_MUSICLIGHT, AACENC_METADATA_DRC_SPEECH,116
+ AACENC_METADATA_DRC_NOT_PRESENT};117
+constexpr size_t kMetaDataDrcProfilesSize = size(kMetaDataDrcProfiles);118
+119
+enum {120
+ IDX_SBR_MODE = 0,121
+ IDX_AAC_AOT,122
+ IDX_SAMPLE_RATE,123
+ IDX_BIT_RATE_1,124
+ IDX_BIT_RATE_2,125
+ IDX_BIT_RATE_3,126
+ IDX_CHANNEL,127
+ IDX_IDENTIFIER,128
+ IDX_SBR_RATIO,129
+ IDX_METADATA_DRC_PROFILE,130
+ IDX_METADATA_COMP_PROFILE,131
+ IDX_METADATA_DRC_TARGET_REF_LEVEL,132
+ IDX_METADATA_COMP_TARGET_REF_LEVEL,133
+ IDX_METADATA_PROG_LEVEL_PRESENT,134
+ IDX_METADATA_PROG_LEVEL,135
+ IDX_METADATA_PCE_MIXDOWN_IDX_PRESENT,136
+ IDX_METADATA_ETSI_DMXLVL_PRESENT,137
+ IDX_METADATA_CENTER_MIX_LEVEL,138
+ IDX_METADATA_SURROUND_MIX_LEVEL,139
+ IDX_METADATA_DOLBY_SURROUND_MODE,140
+ IDX_METADATA_DRC_PRESENTATION_MODE,141
+ IDX_METADATA_EXT_ANC_DATA_ENABLE,142
+ IDX_METADATA_EXT_DOWNMIX_LEVEL_ENABLE,143
+ IDX_METADATA_EXT_DOWNMIX_LEVEL_A,144
+ IDX_METADATA_EXT_DOWNMIX_LEVEL_B,145
+ IDX_METADATA_DMX_GAIN_ENABLE,146
+ IDX_METADATA_DMX_GAIN_5,147
+ IDX_METADATA_DMX_GAIN_2,148
+ IDX_METADATA_LFE_DMX_ENABLE,149
+ IDX_METADATA_LFE_DMX_LEVEL,150
+ IDX_IN_BUFFER_INDEX_1,151
+ IDX_IN_BUFFER_INDEX_2,152
+ IDX_IN_BUFFER_INDEX_3,153
+ IDX_BIT_RATE_MODE,154
+ IDX_GRANULE_LENGTH,155
+ IDX_CHANNELORDER,156
+ IDX_AFTERBURNER,157
+ IDX_BANDWIDTH,158
+ IDX_PEAK_BITRATE,159
+ IDX_HEADER_PERIOD,160
+ IDX_SIGNALING_MODE,161
+ IDX_TPSUBFRAMES,162
+ IDX_AUDIOMUXVER,163
+ IDX_PROTECTION,164
+ IDX_ANCILLARY_BITRATE,165
+ IDX_METADATA_MODE,166
+ IDX_LAST167
+};168
+169
+template <typename type1, typename type2, typename type3>170
+auto generateNumberInRangeFromData(type1 data, type2 min, type3 max) -> decltype(max) {171
+ return (data % (1 + max - min)) + min;172
+}173
+174
+class Codec {175
+ public:176
+ ~Codec() { deInitEncoder(); }177
+ bool initEncoder(uint8_t **dataPtr, size_t *sizePtr);178
+ void encodeFrames(const uint8_t *data, size_t size);179
+ void deInitEncoder();180
+181
+ private:182
+ template <typename type1, typename type2, typename type3>183
+ void setAACParam(type1 data, const AACENC_PARAM aacParam, type2 min, type2 max,184
+ const type3 *array = nullptr);185
+ void setupMetaData(uint8_t *data);186
+187
+ HANDLE_AACENCODER mEncoder = nullptr;188
+ AACENC_MetaData mMetaData = {};189
+ uint32_t mInBufferIdx_1 = 0;190
+ uint32_t mInBufferIdx_2 = 0;191
+ uint32_t mInBufferIdx_3 = 0;192
+};193
+194
+void Codec::setupMetaData(uint8_t *data) {195
+ uint32_t drcProfileIndex = generateNumberInRangeFromData(data[IDX_METADATA_DRC_PROFILE], 0,196
+ kMetaDataDrcProfilesSize - 1);197
+ AACENC_METADATA_DRC_PROFILE drcProfile = kMetaDataDrcProfiles[drcProfileIndex];198
+ mMetaData.drc_profile = drcProfile;199
+200
+ uint32_t compProfileIndex = generateNumberInRangeFromData(data[IDX_METADATA_COMP_PROFILE], 0,201
fdk-aac-2.0.1.tar.gz/libAACdec/include/aacdecoder_lib.h -> fdk-aac-2.0.2.tar.gz/libAACdec/include/aacdecoder_lib.h
Changed
201
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
files are provided for usage in specific C/C++ programs. The main AAC decoder12
library API functions are located in aacdecoder_lib.h header file.13
14
-In binary releases the decoder core resides in statically linkable libraries,15
-for example libAACdec.a.16
-17
18
\section Calling_Sequence Calling Sequence19
20
21
HE-AAC v2, or MPEG-D USAC bitstreams. In the following description, input stream22
read and output write function details are left out, since they may be23
implemented in a variety of configurations depending on the user's specific24
-requirements. The example implementation uses file-based input/output, and in25
-such case one may call mpegFileRead_Open() to open an input file and to allocate26
-memory for the required structures, and the corresponding mpegFileRead_Close()27
-to close opened files and to de-allocate associated structures.28
-mpegFileRead_Open() will attempt to detect the bitstream format and in case of29
-MPEG-4 file format or Raw Packets file format (a proprietary Fraunhofer IIS file30
-format suitable only for testing) it will read the Audio Specific Config data31
-(ASC). An unsuccessful attempt to recognize the bitstream format requires the32
-user to provide this information manually. For any other bitstream formats that33
-are usually applicable in streaming applications, the decoder itself will try to34
-synchronize and parse the given bitstream fragment using the FDK transport35
-library. Hence, for streaming applications (without file access) this step is36
-not necessary.37
+requirements.38
39
40
-# Call aacDecoder_Open() to open and retrieve a handle to a new AAC decoder41
42
working memory (a client-supplied input buffer "inBuffer" in framework). This43
buffer will be used to load AAC bitstream data to the decoder. Only when all44
data in this buffer has been processed will the decoder signal an empty buffer.45
-For file-based input, you may invoke mpegFileRead_Read() to acquire new46
-bitstream data.47
-# Call aacDecoder_Fill() to fill the decoder's internal bitstream input buffer48
with the client-supplied bitstream input buffer. Note, if the data loaded in to49
the internal buffer is not sufficient to decode a frame,50
aacDecoder_DecodeFrame() will return ::AAC_DEC_NOT_ENOUGH_BITS until a51
sufficient amount of data is loaded in to the internal buffer. For streaming52
formats (ADTS, LOAS), it is acceptable to load more than one frame to the53
-decoder. However, for RAW file format (Fraunhofer IIS proprietary format), only54
-one frame may be loaded to the decoder per aacDecoder_DecodeFrame() call. For55
-least amount of communication delay, fill and decode should be performed on a56
-frame by frame basis. \code ErrorStatus = aacDecoder_Fill(aacDecoderInfo,57
-inBuffer, bytesRead, bytesValid); \endcode58
+decoder. However, for packed based formats, only one frame may be loaded to the59
+decoder per aacDecoder_DecodeFrame() call. For least amount of communication60
+delay, fill and decode should be performed on a frame by frame basis. \code61
+ ErrorStatus = aacDecoder_Fill(aacDecoderInfo, inBuffer, bytesRead,62
+bytesValid); \endcode63
-# Call aacDecoder_DecodeFrame(). This function decodes one frame and writes64
decoded PCM audio data to a client-supplied buffer. It is the client's65
responsibility to allocate a buffer which is large enough to hold the decoded66
67
TimeData, OUT_BUF_SIZE, flags); \endcode If the bitstream configuration (number68
of channels, sample rate, frame size) is not known a priori, you may call69
aacDecoder_GetStreamInfo() to retrieve a structure that contains this70
-information. You may use this data to initialize an audio output device. In the71
-example program, if the number of channels or the sample rate has changed since72
-program start or the previously decoded frame, the audio output device is then73
-re-initialized. If WAVE file output is chosen, a new WAVE file for each new74
-stream configuration is be created. \code p_si =75
-aacDecoder_GetStreamInfo(aacDecoderInfo); \endcode76
+information. You may use this data to initialize an audio output device. \code77
+ p_si = aacDecoder_GetStreamInfo(aacDecoderInfo);78
+\endcode79
-# Repeat steps 5 to 7 until no data is available to decode any more, or in case80
of error. \code } while (bytesRead[0] > 0 || doFlush || doBsFlush ||81
forceContinue); \endcode82
83
84
\image latex decode.png "Decode calling sequence" width=11cm85
86
-\image latex change_source.png "Change data source sequence" width 5cm87
+\image latex change_source.png "Change data source sequence" width=5cm88
89
\image latex conceal.png "Error concealment sequence" width=14cm90
91
92
limited applications, the output buffer may be reused as an external input93
buffer prior to the subsequence aacDecoder_Fill() function call.94
95
-The external input buffer is set in the example program and its size is defined96
-by ::IN_BUF_SIZE. You may freely choose different buffer sizes. To feed the data97
-to the decoder-internal input buffer, use the function aacDecoder_Fill(). This98
-function returns important information regarding the number of bytes in the99
-external input buffer that have not yet been copied into the internal input100
-buffer (variable bytesValid). Once the external buffer has been fully copied, it101
-can be completely re-filled again. In case you wish to refill the buffer while102
-there are unprocessed bytes (bytesValid is unequal 0), you should preserve the103
-unconsumed data. However, we recommend to refill the buffer only when bytesValid104
-returns 0.105
+To feed the data to the decoder-internal input buffer, use the106
+function aacDecoder_Fill(). This function returns important information107
+regarding the number of bytes in the external input buffer that have not yet108
+been copied into the internal input buffer (variable bytesValid). Once the109
+external buffer has been fully copied, it can be completely re-filled again. In110
+case you wish to refill the buffer while there are unprocessed bytes (bytesValid111
+is unequal 0), you should preserve the unconsumed data. However, we recommend to112
+refill the buffer only when bytesValid returns 0.113
114
The bytesValid parameter is an input and output parameter to the FDK decoder. As115
an input, it signals how many valid bytes are available in the external buffer.116
117
In case a Program Config is included in the audio configuration, the channel118
mapping described within it will be adopted.119
120
-In case of MPEG-D Surround the channel mapping will follow the same criteria121
-described in ISO/IEC 13818-7:2005(E), but adding corresponding top channels (if122
-available) to the channel types in order to avoid ambiguity. The examples below123
-explain these aspects in detail.124
+The examples below explain these aspects in detail.125
126
\section OutputFormatChange Changing the audio output format127
128
129
#include "genericStds.h"130
131
#define AACDECODER_LIB_VL0 3132
-#define AACDECODER_LIB_VL1 0133
+#define AACDECODER_LIB_VL1 2134
#define AACDECODER_LIB_VL2 0135
136
/**137
138
2. If the parameter value is greater than that of139
::AAC_PCM_MAX_OUTPUT_CHANNELS both will be set to the same140
value. \n141
- 3. This parameter does not affect MPEG Surround processing.142
- \n143
- 4. This parameter will be ignored if the number of encoded144
+ 3. This parameter will be ignored if the number of encoded145
audio channels is greater than 8. */146
AAC_PCM_MAX_OUTPUT_CHANNELS =147
0x0012, /*!< Maximum number of PCM output channels. If lower than the148
149
2. If the parameter value is greater than zero but smaller150
than ::AAC_PCM_MIN_OUTPUT_CHANNELS both will be set to same151
value. \n152
- 3. The operating mode of the MPEG Surround module will be153
- set accordingly. \n154
- 4. Setting this parameter with any value will disable the155
- binaural processing of the MPEG Surround module156
- 5. This parameter will be ignored if the number of encoded157
+ 3. This parameter will be ignored if the number of encoded158
audio channels is greater than 8. */159
AAC_METADATA_PROFILE =160
0x0020, /*!< See ::AAC_MD_PROFILE for all available values. */161
162
::CONCEAL_INTER. only some AOTs are163
supported). \n */164
AAC_DRC_BOOST_FACTOR =165
- 0x0200, /*!< Dynamic Range Control: Scaling factor for boosting gain166
- values. Defines how the boosting DRC factors (conveyed in the167
- bitstream) will be applied to the decoded signal. The valid168
- values range from 0 (don't apply boost factors) to 127 (fully169
- apply boost factors). Default value is 0. */170
- AAC_DRC_ATTENUATION_FACTOR =171
- 0x0201, /*!< Dynamic Range Control: Scaling factor for attenuating gain172
- values. Same as173
- ::AAC_DRC_BOOST_FACTOR but for attenuating DRC factors. */174
+ 0x0200, /*!< MPEG-4 / MPEG-D Dynamic Range Control (DRC): Scaling factor175
+ for boosting gain values. Defines how the boosting DRC factors176
+ (conveyed in the bitstream) will be applied to the decoded177
+ signal. The valid values range from 0 (don't apply boost178
+ factors) to 127 (fully apply boost factors). Default value is 0179
+ for MPEG-4 DRC and 127 for MPEG-D DRC. */180
+ AAC_DRC_ATTENUATION_FACTOR = 0x0201, /*!< MPEG-4 / MPEG-D DRC: Scaling factor181
+ for attenuating gain values. Same as182
+ ::AAC_DRC_BOOST_FACTOR but for183
+ attenuating DRC factors. */184
AAC_DRC_REFERENCE_LEVEL =185
- 0x0202, /*!< Dynamic Range Control (DRC): Target reference level. Defines186
- the level below full-scale (quantized in steps of 0.25dB) to187
- which the output audio signal will be normalized to by the DRC188
- module. The parameter controls loudness normalization for both189
- MPEG-4 DRC and MPEG-D DRC. The valid values range from 40 (-10190
- dBFS) to 127 (-31.75 dBFS). Any value smaller than 0 switches191
- off loudness normalization and MPEG-4 DRC. By default, loudness192
- normalization and MPEG-4 DRC is switched off. */193
+ 0x0202, /*!< MPEG-4 / MPEG-D DRC: Target reference level / decoder target194
+ loudness.\n Defines the level below full-scale (quantized in195
+ steps of 0.25dB) to which the output audio signal will be196
+ normalized to by the DRC module.\n The parameter controls197
+ loudness normalization for both MPEG-4 DRC and MPEG-D DRC. The198
+ valid values range from 40 (-10 dBFS) to 127 (-31.75 dBFS).\n199
+ Example values:\n200
+ 124 (-31 dBFS) for audio/video receivers (AVR) or other201
fdk-aac-2.0.1.tar.gz/libAACdec/src/FDK_delay.cpp -> fdk-aac-2.0.2.tar.gz/libAACdec/src/FDK_delay.cpp
Changed
67
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
12
if (delay > 0) {13
data->delay_line =14
- (INT_PCM*)FDKcalloc(num_channels * delay, sizeof(INT_PCM));15
+ (PCM_DEC*)FDKcalloc(num_channels * delay, sizeof(PCM_DEC));16
if (data->delay_line == NULL) {17
return -1;18
}19
20
return 0;21
}22
23
-void FDK_Delay_Apply(FDK_SignalDelay* data, FIXP_PCM* time_buffer,24
+void FDK_Delay_Apply(FDK_SignalDelay* data, PCM_DEC* time_buffer,25
const UINT frame_length, const UCHAR channel) {26
FDK_ASSERT(data != NULL);27
28
if (data->delay > 0) {29
- C_ALLOC_SCRATCH_START(tmp, FIXP_PCM, MAX_FRAME_LENGTH)30
+ C_ALLOC_SCRATCH_START(tmp, PCM_DEC, MAX_FRAME_LENGTH)31
FDK_ASSERT(frame_length <= MAX_FRAME_LENGTH);32
FDK_ASSERT(channel < data->num_channels);33
FDK_ASSERT(time_buffer != NULL);34
if (frame_length >= data->delay) {35
FDKmemcpy(tmp, &time_buffer[frame_length - data->delay],36
- data->delay * sizeof(FIXP_PCM));37
+ data->delay * sizeof(PCM_DEC));38
FDKmemmove(&time_buffer[data->delay], &time_buffer[0],39
- (frame_length - data->delay) * sizeof(FIXP_PCM));40
+ (frame_length - data->delay) * sizeof(PCM_DEC));41
FDKmemcpy(&time_buffer[0], &data->delay_line[channel * data->delay],42
- data->delay * sizeof(FIXP_PCM));43
+ data->delay * sizeof(PCM_DEC));44
FDKmemcpy(&data->delay_line[channel * data->delay], tmp,45
- data->delay * sizeof(FIXP_PCM));46
+ data->delay * sizeof(PCM_DEC));47
} else {48
- FDKmemcpy(tmp, &time_buffer[0], frame_length * sizeof(FIXP_PCM));49
+ FDKmemcpy(tmp, &time_buffer[0], frame_length * sizeof(PCM_DEC));50
FDKmemcpy(&time_buffer[0], &data->delay_line[channel * data->delay],51
- frame_length * sizeof(FIXP_PCM));52
+ frame_length * sizeof(PCM_DEC));53
FDKmemcpy(&data->delay_line[channel * data->delay],54
&data->delay_line[channel * data->delay + frame_length],55
- (data->delay - frame_length) * sizeof(FIXP_PCM));56
+ (data->delay - frame_length) * sizeof(PCM_DEC));57
FDKmemcpy(&data->delay_line[channel * data->delay +58
(data->delay - frame_length)],59
- tmp, frame_length * sizeof(FIXP_PCM));60
+ tmp, frame_length * sizeof(PCM_DEC));61
}62
- C_ALLOC_SCRATCH_END(tmp, FIXP_PCM, MAX_FRAME_LENGTH)63
+ C_ALLOC_SCRATCH_END(tmp, PCM_DEC, MAX_FRAME_LENGTH)64
}65
66
return;67
fdk-aac-2.0.1.tar.gz/libAACdec/src/FDK_delay.h -> fdk-aac-2.0.2.tar.gz/libAACdec/src/FDK_delay.h
Changed
28
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
* Structure representing one delay element for multiple channels.12
*/13
typedef struct {14
- INT_PCM* delay_line; /*!< Pointer which stores allocated delay line. */15
+ PCM_DEC* delay_line; /*!< Pointer which stores allocated delay line. */16
USHORT delay; /*!< Delay required in samples (per channel). */17
UCHAR num_channels; /*!< Number of channels to delay. */18
} FDK_SignalDelay;19
20
*21
* \return void22
*/23
-void FDK_Delay_Apply(FDK_SignalDelay* data, FIXP_PCM* time_buffer,24
+void FDK_Delay_Apply(FDK_SignalDelay* data, PCM_DEC* time_buffer,25
const UINT frame_length, const UCHAR channel);26
27
/**28
fdk-aac-2.0.1.tar.gz/libAACdec/src/aac_ram.cpp -> fdk-aac-2.0.2.tar.gz/libAACdec/src/aac_ram.cpp
Changed
33
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
12
#define WORKBUFFER1_TAG 013
#define WORKBUFFER2_TAG 114
-15
-#define WORKBUFFER3_TAG 416
-#define WORKBUFFER4_TAG 517
-18
#define WORKBUFFER5_TAG 619
-20
#define WORKBUFFER6_TAG 721
22
/*! The structure AAC_DECODER_INSTANCE is the top level structure holding all23
24
C_ALLOC_MEM_OVERLAY(WorkBufferCore2, FIXP_DBL, ((8) * 1024), SECT_DATA_L2,25
WORKBUFFER2_TAG)26
27
-C_ALLOC_MEM_OVERLAY(WorkBufferCore3, FIXP_DBL, WB_SECTION_SIZE, SECT_DATA_L2,28
- WORKBUFFER3_TAG)29
-C_AALLOC_MEM(WorkBufferCore4, FIXP_DBL, WB_SECTION_SIZE)30
C_ALLOC_MEM_OVERLAY(WorkBufferCore6, SCHAR,31
fMax((INT)(sizeof(FIXP_DBL) * WB_SECTION_SIZE),32
(INT)sizeof(CAacDecoderCommonData)),33
fdk-aac-2.0.1.tar.gz/libAACdec/src/aac_ram.h -> fdk-aac-2.0.2.tar.gz/libAACdec/src/aac_ram.h
Changed
23
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
12
H_ALLOC_MEM_OVERLAY(WorkBufferCore1, CWorkBufferCore1)13
H_ALLOC_MEM_OVERLAY(WorkBufferCore2, FIXP_DBL)14
-15
-H_ALLOC_MEM_OVERLAY(WorkBufferCore3, FIXP_DBL)16
-H_ALLOC_MEM(WorkBufferCore4, FIXP_DBL)17
-18
H_ALLOC_MEM_OVERLAY(WorkBufferCore5, PCM_DEC)19
-20
H_ALLOC_MEM_OVERLAY(WorkBufferCore6, SCHAR)21
22
#endif /* #ifndef AAC_RAM_H */23
fdk-aac-2.0.1.tar.gz/libAACdec/src/aac_rom.h -> fdk-aac-2.0.2.tar.gz/libAACdec/src/aac_rom.h
Changed
18
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
#include "aacdec_hcr_types.h"12
#include "aacdec_hcrs.h"13
14
+#define PCM_AAC LONG15
#define PCM_DEC FIXP_DBL16
#define MAXVAL_PCM_DEC MAXVAL_DBL17
#define MINVAL_PCM_DEC MINVAL_DBL18
fdk-aac-2.0.1.tar.gz/libAACdec/src/aacdec_drc.cpp -> fdk-aac-2.0.2.tar.gz/libAACdec/src/aacdec_drc.cpp
Changed
201
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
}12
13
/*!14
+\brief Reset DRC information15
+16
+\self Handle of DRC info17
+18
+\return none19
+*/20
+void aacDecoder_drcReset(HANDLE_AAC_DRC self) {21
+ self->applyExtGain = 0;22
+ self->additionalGainPrev = AACDEC_DRC_GAIN_INIT_VALUE;23
+ self->additionalGainFilterState = AACDEC_DRC_GAIN_INIT_VALUE;24
+ self->additionalGainFilterState1 = AACDEC_DRC_GAIN_INIT_VALUE;25
+}26
+27
+/*!28
\brief Initialize DRC information29
30
\self Handle of DRC info31
32
pParams->usrCut = FL2FXCONST_DBL(0.0f);33
pParams->boost = FL2FXCONST_DBL(0.0f);34
pParams->usrBoost = FL2FXCONST_DBL(0.0f);35
- pParams->targetRefLevel = -1;36
+ pParams->targetRefLevel = 96;37
pParams->expiryFrame = AACDEC_DRC_DFLT_EXPIRY_FRAMES;38
- pParams->applyDigitalNorm = OFF;39
pParams->applyHeavyCompression = OFF;40
pParams->usrApplyHeavyCompression = OFF;41
42
43
self->progRefLevelPresent = 0;44
self->presMode = -1;45
self->uniDrcPrecedence = 0;46
+47
+ aacDecoder_drcReset(self);48
}49
50
/*!51
52
return AAC_DEC_INVALID_HANDLE;53
}54
if (value < 0) {55
- self->params.applyDigitalNorm = OFF;56
self->params.targetRefLevel = -1;57
} else {58
- /* ref_level must be between 0 and MAX_REFERENCE_LEVEL, inclusive */59
- self->params.applyDigitalNorm = ON;60
if (self->params.targetRefLevel != (SCHAR)value) {61
self->params.targetRefLevel = (SCHAR)value;62
self->progRefLevel = (SCHAR)value; /* Always set the program reference63
64
self->update = 1;65
}66
break;67
- case APPLY_NORMALIZATION:68
- if ((value != OFF) && (value != ON)) {69
- return AAC_DEC_SET_PARAM_FAIL;70
- }71
- if (self == NULL) {72
- return AAC_DEC_INVALID_HANDLE;73
- }74
- /* Store new parameter value */75
- self->params.applyDigitalNorm = (UCHAR)value;76
- break;77
case APPLY_HEAVY_COMPRESSION:78
if ((value != OFF) && (value != ON)) {79
return AAC_DEC_SET_PARAM_FAIL;80
81
FDK_ASSERT(0);82
}83
}84
- if (self->params.applyDigitalNorm == OFF) {85
- /* Reset normalization gain since this module must not apply it */86
- norm_mantissa = FL2FXCONST_DBL(0.5f);87
- norm_exponent = 1;88
- }89
+ /* Reset normalization gain since this module must not apply it */90
+ norm_mantissa = FL2FXCONST_DBL(0.5f);91
+ norm_exponent = 1;92
93
/* calc scale factors */94
for (band = 0; band < numBands; band++) {95
96
}97
}98
}99
+100
+/**101
+ * \brief Apply DRC Level Normalization.102
+ *103
+ * This function prepares/applies the gain values for the DRC Level104
+ * Normalization and returns the exponent of the time data. The following two105
+ * cases are handled:106
+ *107
+ * - Limiter enabled:108
+ * The input data must be interleaved.109
+ * One gain per sample is written to the buffer pGainPerSample.110
+ * If necessary the time data is rescaled.111
+ *112
+ * - Limiter disabled:113
+ * The input data can be interleaved or deinterleaved.114
+ * The gain values are applied to the time data.115
+ * If necessary the time data is rescaled.116
+ *117
+ * \param hDrcInfo [i/o] handle to drc data structure.118
+ * \param samplesIn [i/o] pointer to time data.119
+ * \param pGain [i ] pointer to gain to be applied to120
+ * the time data.121
+ * \param pGainPerSample [o ] pointer to the gain per sample to122
+ * be applied to the time data in the limiter.123
+ * \param gain_scale [i ] exponent to be applied to the time124
+ * data.125
+ * \param gain_delay [i ] delay[samples] with which the gains126
+ * in pGain shall be applied (gain_delay <= nSamples).127
+ * \param nSamples [i ] number of samples per frame.128
+ * \param channels [i ] number of channels.129
+ * \param stride [i ] channel stride of time data.130
+ * \param limiterEnabled [i ] 1 if limiter is enabled, otherwise131
+ * 0.132
+ *133
+ * \return exponent of time data134
+ */135
+INT applyDrcLevelNormalization(HANDLE_AAC_DRC hDrcInfo, PCM_DEC *samplesIn,136
+ FIXP_DBL *pGain, FIXP_DBL *pGainPerSample,137
+ const INT gain_scale, const UINT gain_delay,138
+ const UINT nSamples, const UINT channels,139
+ const UINT stride, const UINT limiterEnabled) {140
+ UINT i;141
+ INT additionalGain_scaling;142
+ FIXP_DBL additionalGain;143
+144
+ FDK_ASSERT(gain_delay <= nSamples);145
+146
+ FIXP_DBL additionalGainSmoothState = hDrcInfo->additionalGainFilterState;147
+ FIXP_DBL additionalGainSmoothState1 = hDrcInfo->additionalGainFilterState1;148
+149
+ if (!gain_delay) {150
+ additionalGain = pGain[0];151
+152
+ /* Apply the additional scaling gain_scale[0] that has no delay and no153
+ * smoothing */154
+ additionalGain_scaling =155
+ fMin(gain_scale, CntLeadingZeros(additionalGain) - 1);156
+ additionalGain = scaleValue(additionalGain, additionalGain_scaling);157
+158
+ /* if it's not possible to fully apply gain_scale to additionalGain, apply159
+ * it to the input signal */160
+ additionalGain_scaling -= gain_scale;161
+162
+ if (additionalGain_scaling) {163
+ scaleValuesSaturate(samplesIn, channels * nSamples,164
+ -additionalGain_scaling);165
+ }166
+167
+ if (limiterEnabled) {168
+ FDK_ASSERT(pGainPerSample != NULL);169
+170
+ for (i = 0; i < nSamples; i++) {171
+ pGainPerSample[i] = additionalGain;172
+ }173
+ } else {174
+ for (i = 0; i < channels * nSamples; i++) {175
+ samplesIn[i] = FIXP_DBL2PCM_DEC(fMult(samplesIn[i], additionalGain));176
+ }177
+ }178
+ } else {179
+ UINT inc;180
+ FIXP_DBL additionalGainUnfiltered;181
+182
+ inc = (stride == 1) ? channels : 1;183
+184
+ for (i = 0; i < nSamples; i++) {185
+ if (i < gain_delay) {186
+ additionalGainUnfiltered = hDrcInfo->additionalGainPrev;187
+ } else {188
+ additionalGainUnfiltered = pGain[0];189
+ }190
+191
+ /* Smooth additionalGain */192
+193
+ /* [b,a] = butter(1, 0.01) */194
+ static const FIXP_SGL b[] = {FL2FXCONST_SGL(0.015466 * 2.0),195
+ FL2FXCONST_SGL(0.015466 * 2.0)};196
+ static const FIXP_SGL a[] = {(FIXP_SGL)MAXVAL_SGL,197
+ FL2FXCONST_SGL(-0.96907)};198
+199
+ additionalGain = -fMult(additionalGainSmoothState, a[1]) +200
+ fMultDiv2(additionalGainUnfiltered, b[0]) +201
fdk-aac-2.0.1.tar.gz/libAACdec/src/aacdec_drc.h -> fdk-aac-2.0.2.tar.gz/libAACdec/src/aacdec_drc.h
Changed
85
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
#include "channel.h"12
#include "FDK_bitstream.h"13
14
+#define AACDEC_DRC_GAIN_SCALING (11) /* Scaling of DRC gains */15
+#define AACDEC_DRC_GAIN_INIT_VALUE \16
+ (FL2FXCONST_DBL( \17
+ 1.0f / (1 << AACDEC_DRC_GAIN_SCALING))) /* Init value for DRC gains */18
+19
#define AACDEC_DRC_DFLT_EXPIRY_FRAMES \20
(0) /* Default DRC data expiry time in AAC frames */21
22
23
TARGET_REF_LEVEL,24
DRC_BS_DELAY,25
DRC_DATA_EXPIRY_FRAME,26
- APPLY_NORMALIZATION,27
APPLY_HEAVY_COMPRESSION,28
DEFAULT_PRESENTATION_MODE,29
ENCODER_TARGET_LEVEL,30
31
/**32
* \brief DRC module interface functions33
*/34
+void aacDecoder_drcReset(HANDLE_AAC_DRC self);35
+36
void aacDecoder_drcInit(HANDLE_AAC_DRC self);37
38
void aacDecoder_drcInitChannelData(CDrcChannelData *pDrcChannel);39
40
void aacDecoder_drcGetInfo(HANDLE_AAC_DRC self, SCHAR *pPresMode,41
SCHAR *pProgRefLevel);42
43
+/**44
+ * \brief Apply DRC Level Normalization.45
+ *46
+ * This function prepares/applies the gain values for the DRC Level47
+ * Normalization and returns the exponent of the time data. The following two48
+ * cases are handled:49
+ *50
+ * - Limiter enabled:51
+ * The input data must be interleaved.52
+ * One gain per sample is written to the buffer pGainPerSample.53
+ * If necessary the time data is rescaled.54
+ *55
+ * - Limiter disabled:56
+ * The input data can be interleaved or deinterleaved.57
+ * The gain values are applied to the time data.58
+ * If necessary the time data is rescaled.59
+ *60
+ * \param hDrcInfo [i/o] handle to drc data structure.61
+ * \param samplesIn [i/o] pointer to time data.62
+ * \param pGain [i ] pointer to gain to be applied to63
+ * the time data.64
+ * \param pGainPerSample [o ] pointer to the gain per sample to65
+ * be applied to the time data in the limiter.66
+ * \param gain_scale [i ] exponent to be applied to the time67
+ * data.68
+ * \param gain_delay [i ] delay[samples] with which the gains69
+ * in pGain shall be applied (gain_delay <= nSamples).70
+ * \param nSamples [i ] number of samples per frame.71
+ * \param channels [i ] number of channels.72
+ * \param stride [i ] channel stride of time data.73
+ * \param limiterEnabled [i ] 1 if limiter is enabled, otherwise74
+ * 0.75
+ *76
+ * \return exponent of time data77
+ */78
+INT applyDrcLevelNormalization(HANDLE_AAC_DRC hDrcInfo, PCM_DEC *samplesIn,79
+ FIXP_DBL *pGain, FIXP_DBL *pGainPerSample,80
+ const INT gain_scale, const UINT gain_delay,81
+ const UINT nSamples, const UINT channels,82
+ const UINT stride, const UINT limiterEnabled);83
+84
#endif /* AACDEC_DRC_H */85
fdk-aac-2.0.1.tar.gz/libAACdec/src/aacdec_drc_types.h -> fdk-aac-2.0.2.tar.gz/libAACdec/src/aacdec_drc_types.h
Changed
32
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
12
UINT expiryFrame;13
UCHAR bsDelayEnable;14
- UCHAR applyDigitalNorm;15
16
AACDEC_DRC_PARAMETER_HANDLING defaultPresentationMode;17
UCHAR encoderTargetLevel;18
19
uniDrcPrecedence; /* Flag for signalling that uniDrc is active and takes20
precedence over legacy DRC */21
22
+ UCHAR applyExtGain; /* Flag is 1 if extGain has to be applied, otherwise 0. */23
+24
+ FIXP_DBL additionalGainPrev; /* Gain of previous frame to be applied to the25
+ time data */26
+ FIXP_DBL additionalGainFilterState; /* Filter state for the gain smoothing */27
+ FIXP_DBL additionalGainFilterState1; /* Filter state for the gain smoothing */28
+29
} CDrcInfo;30
31
typedef CDrcInfo *HANDLE_AAC_DRC;32
fdk-aac-2.0.1.tar.gz/libAACdec/src/aacdec_hcr.cpp -> fdk-aac-2.0.2.tar.gz/libAACdec/src/aacdec_hcr.cpp
Changed
85
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
static INT DecodeEscapeSequence(HANDLE_FDK_BITSTREAM bs, const INT bsAnchor,12
INT quantSpecCoef, INT *pLeftStartOfSegment,13
SCHAR *pRemainingBitsInSegment,14
- int *pNumDecodedBits);15
+ int *pNumDecodedBits, UINT *errorWord);16
17
static int DecodePCW_Sign(HANDLE_FDK_BITSTREAM bs, const INT bsAnchor,18
UINT codebookDim, const SCHAR *pQuantVal,19
20
bs, pHcr->decInOut.bitstreamAnchor,21
pQuantizedSpectralCoefficients22
[quantizedSpectralCoefficientsIdx],23
- pLeftStartOfSegment, pRemainingBitsInSegment,24
- &numDecodedBits);25
+ pLeftStartOfSegment, pRemainingBitsInSegment, &numDecodedBits,26
+ &pHcr->decInOut.errorLog);27
}28
quantizedSpectralCoefficientsIdx++;29
if (quantizedSpectralCoefficientsIdx >= 1024) {30
31
bs, pHcr->decInOut.bitstreamAnchor,32
pQuantizedSpectralCoefficients33
[quantizedSpectralCoefficientsIdx],34
- pLeftStartOfSegment, pRemainingBitsInSegment,35
- &numDecodedBits);36
+ pLeftStartOfSegment, pRemainingBitsInSegment, &numDecodedBits,37
+ &pHcr->decInOut.errorLog);38
}39
quantizedSpectralCoefficientsIdx++;40
if (quantizedSpectralCoefficientsIdx >= 1024) {41
42
static INT DecodeEscapeSequence(HANDLE_FDK_BITSTREAM bs, const INT bsAnchor,43
INT quantSpecCoef, INT *pLeftStartOfSegment,44
SCHAR *pRemainingBitsInSegment,45
- int *pNumDecodedBits) {46
+ int *pNumDecodedBits, UINT *errorWord) {47
UINT i;48
INT sign;49
UINT escapeOnesCounter = 0;50
51
FROM_LEFT_TO_RIGHT);52
*pRemainingBitsInSegment -= 1;53
*pNumDecodedBits += 1;54
+ if (*pRemainingBitsInSegment < 0) {55
+ return Q_VALUE_INVALID;56
+ }57
58
if (carryBit != 0) {59
escapeOnesCounter += 1;60
61
FROM_LEFT_TO_RIGHT);62
*pRemainingBitsInSegment -= 1;63
*pNumDecodedBits += 1;64
+ if (*pRemainingBitsInSegment < 0) {65
+ return Q_VALUE_INVALID;66
+ }67
68
escape_word <<= 1;69
escape_word = escape_word | carryBit;70
71
72
sign = (quantSpecCoef >= 0) ? 1 : -1;73
74
- quantSpecCoef = sign * (((INT)1 << escapeOnesCounter) + escape_word);75
-76
+ if (escapeOnesCounter < 13) {77
+ quantSpecCoef = sign * (((INT)1 << escapeOnesCounter) + escape_word);78
+ } else {79
+ *errorWord |= TOO_MANY_PCW_BODY_SIGN_ESC_BITS_DECODED;80
+ quantSpecCoef = Q_VALUE_INVALID;81
+ }82
return quantSpecCoef;83
}84
85
fdk-aac-2.0.1.tar.gz/libAACdec/src/aacdec_hcrs.cpp -> fdk-aac-2.0.2.tar.gz/libAACdec/src/aacdec_hcrs.cpp
Changed
21
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
/* count ones and store sum in escapePrefixUp */12
if (carryBit == 1) {13
escapePrefixUp += 1; /* update conter for ones */14
+ if (escapePrefixUp > 8) {15
+ pHcr->decInOut.errorLog |= STATE_ERROR_BODY_SIGN_ESC__ESC_PREFIX;16
+ return BODY_SIGN_ESC__ESC_PREFIX;17
+ }18
19
/* store updated counter in sideinfo of current codeword */20
pEscapeSequenceInfo[codewordOffset] &=21
fdk-aac-2.0.1.tar.gz/libAACdec/src/aacdecoder.cpp -> fdk-aac-2.0.2.tar.gz/libAACdec/src/aacdecoder.cpp
Changed
201
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
pStreamInfo->drcProgRefLev =12
-1; /* set program reference level to not indicated */13
pStreamInfo->drcPresMode = -1; /* default: presentation mode not indicated */14
+15
+ pStreamInfo->outputLoudness = -1; /* default: no loudness metadata present */16
}17
18
/*!19
20
/* Set default frame delay */21
aacDecoder_drcSetParam(self->hDrcInfo, DRC_BS_DELAY,22
CConcealment_GetDelay(&self->concealCommonData));23
+ self->workBufferCore1 = (FIXP_DBL *)GetWorkBufferCore1();24
25
self->workBufferCore2 = GetWorkBufferCore2();26
if (self->workBufferCore2 == NULL) goto bail;27
28
const int subStreamIndex) {29
int ch;30
int aacChannelOffset = 0, aacChannels = (8);31
- int numElements = (((8)) + (8)), elementOffset = 0;32
+ int numElements = (3 * ((8) * 2) + (((8) * 2)) / 2 + 4 * (1) + 1),33
+ elementOffset = 0;34
35
if (self == NULL) return;36
37
38
FreeDrcInfo(&self->hDrcInfo);39
}40
41
+ if (self->workBufferCore1 != NULL) {42
+ FreeWorkBufferCore1((CWorkBufferCore1 **)&self->workBufferCore1);43
+ }44
+45
/* Free WorkBufferCore2 */46
if (self->workBufferCore2 != NULL) {47
FreeWorkBufferCore2(&self->workBufferCore2);48
49
UCHAR downscaleFactor = self->downscaleFactor;50
UCHAR downscaleFactorInBS = self->downscaleFactorInBS;51
52
+ self->aacOutDataHeadroom = (3);53
+54
// set profile and check for supported aot55
// leave profile on default (=-1) for all other supported MPEG-4 aot's except56
// aot=2 (=AAC-LC)57
58
self->useLdQmfTimeAlign =59
asc->m_sc.m_eldSpecificConfig.m_useLdQmfTimeAlign;60
}61
+ if (self->sbrEnabled != asc->m_sbrPresentFlag) {62
+ ascChanged = 1;63
+ }64
}65
66
self->streamInfo.extAot = asc->m_extensionAudioObjectType;67
68
self->streamInfo.extSamplingRate / self->downscaleFactor;69
}70
}71
+ if ((asc->m_aot == AOT_AAC_LC) && (asc->m_sbrPresentFlag == 1) &&72
+ (asc->m_extensionSamplingFrequency > (2 * asc->m_samplingFrequency))) {73
+ return AAC_DEC_UNSUPPORTED_SAMPLINGRATE; /* Core decoder supports at most a74
+ 1:2 upsampling for HE-AAC and75
+ HE-AACv2 */76
+ }77
78
/* --------- vcb11 ------------ */79
self->flags[streamIndex] |= (asc->m_vcb11Flag) ? AC_ER_VCB11 : 0;80
81
self->samplingRateInfo[0].samplingRate / self->downscaleFactor;82
self->streamInfo.aacSamplesPerFrame =83
asc->m_samplesPerFrame / self->downscaleFactor;84
+ if (self->streamInfo.aacSampleRate <= 0) {85
+ return AAC_DEC_UNSUPPORTED_SAMPLINGRATE;86
+ }87
}88
}89
90
91
goto bail;92
}93
94
+ if (*configChanged) {95
+ if (asc->m_aot == AOT_USAC) {96
+ self->hDrcInfo->enable = 0;97
+ self->hDrcInfo->progRefLevelPresent = 0;98
+ }99
+ }100
+101
if (asc->m_aot == AOT_USAC) {102
pcmLimiter_SetAttack(self->hLimiter, (5));103
pcmLimiter_SetThreshold(self->hLimiter, FL2FXCONST_DBL(0.89125094f));104
105
}106
107
LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(108
- HANDLE_AACDECODER self, const UINT flags, FIXP_PCM *pTimeData,109
+ HANDLE_AACDECODER self, const UINT flags, PCM_DEC *pTimeData,110
const INT timeDataSize, const int timeDataChannelOffset) {111
AAC_DECODER_ERROR ErrorStatus = AAC_DEC_OK;112
113
114
FDKmemcpy(drcChMap, self->chMapping, (8) * sizeof(UCHAR));115
}116
117
- /* Turn on/off DRC modules level normalization in digital domain depending118
- * on the limiter status. */119
- aacDecoder_drcSetParam(self->hDrcInfo, APPLY_NORMALIZATION,120
- (self->limiterEnableCurr) ? 0 : 1);121
-122
/* deactivate legacy DRC in case uniDrc is active, i.e. uniDrc payload is123
* present and one of DRC or Loudness Normalization is switched on */124
aacDecoder_drcSetParam(125
126
self->hDrcInfo, bs, self->pAacDecoderStaticChannelInfo,127
pce->ElementInstanceTag, drcChMap, aacChannels);128
if (mapped > 0) {129
- /* If at least one DRC thread has been mapped to a channel threre was DRC130
- * data in the bitstream. */131
- self->flags[streamIndex] |= AC_DRC_PRESENT;132
+ if (!(self->flags[streamIndex] & (AC_USAC | AC_RSV603DA))) {133
+ /* If at least one DRC thread has been mapped to a channel there was DRC134
+ * data in the bitstream. */135
+ self->flags[streamIndex] |= AC_DRC_PRESENT;136
+ } else {137
+ self->hDrcInfo->enable = 0;138
+ self->hDrcInfo->progRefLevelPresent = 0;139
+ ErrorStatus = AAC_DEC_UNSUPPORTED_FORMAT;140
+ }141
}142
143
/* Create a reverse mapping table */144
145
&pAacDecoderStaticChannelInfo->drcData);146
}147
}148
+149
/* The DRC module demands to be called with the gain field holding the150
* gain scale. */151
- self->extGain[0] = (FIXP_DBL)TDL_GAIN_SCALING;152
+ self->extGain[0] = (FIXP_DBL)AACDEC_DRC_GAIN_SCALING;153
+154
/* DRC processing */155
aacDecoder_drcApply(156
self->hDrcInfo, self->hSbrDecoder, pAacDecoderChannelInfo,157
158
if (self->flushStatus && (self->flushCnt > 0) &&159
!(flags & AACDEC_CONCEAL)) {160
FDKmemclear(pTimeData + offset,161
- sizeof(FIXP_PCM) * self->streamInfo.aacSamplesPerFrame);162
+ sizeof(PCM_DEC) * self->streamInfo.aacSamplesPerFrame);163
} else164
switch (pAacDecoderChannelInfo->renderMode) {165
case AACDEC_RENDER_IMDCT:166
167
!frameOk_butConceal),168
pAacDecoderChannelInfo->pComStaticData->pWorkBufferCore1169
->mdctOutTemp,170
- self->elFlags[el], elCh);171
+ self->aacOutDataHeadroom, self->elFlags[el], elCh);172
173
self->extGainDelay = self->streamInfo.aacSamplesPerFrame;174
break;175
176
&self->samplingRateInfo[streamIndex],177
(self->frameOK && !(flags & AACDEC_CONCEAL) &&178
!frameOk_butConceal),179
- flags, self->flags[streamIndex]);180
+ self->aacOutDataHeadroom, flags, self->flags[streamIndex]);181
182
self->extGainDelay = self->streamInfo.aacSamplesPerFrame;183
break;184
185
if (!CConceal_TDFading_Applied[c]) {186
CConceal_TDFading_Applied[c] = CConcealment_TDFading(187
self->streamInfo.aacSamplesPerFrame,188
- &self->pAacDecoderStaticChannelInfo[c], pTimeData + offset, 0);189
+ &self->pAacDecoderStaticChannelInfo[c], self->aacOutDataHeadroom,190
+ pTimeData + offset, 0);191
if (c + 1 < (8) && c < aacChannels - 1) {192
/* update next TDNoise Seed to avoid muting in case of Parametric193
* Stereo */194
195
if ((aacChannels == 2) && bsPseudoLr) {196
int i, offset2;197
const FIXP_SGL invSqrt2 = FL2FXCONST_SGL(0.707106781186547f);198
- FIXP_PCM *pTD = pTimeData;199
+ PCM_DEC *pTD = pTimeData;200
201
fdk-aac-2.0.1.tar.gz/libAACdec/src/aacdecoder.h -> fdk-aac-2.0.2.tar.gz/libAACdec/src/aacdecoder.h
Changed
50
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
INT outputInterleaved; /*!< PCM output format (interleaved/none interleaved).12
*/13
14
+ INT aacOutDataHeadroom; /*!< Headroom of the output time signal to prevent15
+ clipping */16
+17
HANDLE_TRANSPORTDEC hInput; /*!< Transport layer handle. */18
19
SamplingRateInfo20
21
CAacDecoderStaticChannelInfo22
*pAacDecoderStaticChannelInfo[(8)]; /*!< Persistent channel memory */23
24
+ FIXP_DBL *workBufferCore1;25
FIXP_DBL *workBufferCore2;26
PCM_DEC *pTimeData2;27
INT timeData2Size;28
29
UCHAR limiterEnableUser; /*!< The limiter configuration requested by the30
library user */31
UCHAR limiterEnableCurr; /*!< The current limiter configuration. */32
+33
FIXP_DBL extGain[1]; /*!< Gain that must be applied to the output signal. */34
UINT extGainDelay; /*!< Delay that must be accounted for extGain. */35
36
- INT_PCM pcmOutputBuffer[(8) * (1024 * 2)];37
-38
HANDLE_DRC_DECODER hUniDrcDecoder;39
UCHAR multibandDrcPresent;40
UCHAR numTimeSlots;41
42
\return error status43
*/44
LINKSPEC_H AAC_DECODER_ERROR CAacDecoder_DecodeFrame(45
- HANDLE_AACDECODER self, const UINT flags, FIXP_PCM *pTimeData,46
+ HANDLE_AACDECODER self, const UINT flags, PCM_DEC *pTimeData,47
const INT timeDataSize, const int timeDataChannelOffset);48
49
/* Free config dependent AAC memory */50
fdk-aac-2.0.1.tar.gz/libAACdec/src/aacdecoder_lib.cpp -> fdk-aac-2.0.2.tar.gz/libAACdec/src/aacdecoder_lib.cpp
Changed
201
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
12
/* Decoder library info */13
#define AACDECODER_LIB_VL0 314
-#define AACDECODER_LIB_VL1 015
+#define AACDECODER_LIB_VL1 216
#define AACDECODER_LIB_VL2 017
#define AACDECODER_LIB_TITLE "AAC Decoder Lib"18
-#ifdef __ANDROID__19
+#ifdef SUPPRESS_BUILD_DATE_INFO20
#define AACDECODER_LIB_BUILD_DATE ""21
#define AACDECODER_LIB_BUILD_TIME ""22
#else23
24
case AAC_DRC_ATTENUATION_FACTOR:25
/* DRC compression factor (where 0 is no and 127 is max compression) */26
errorStatus = aacDecoder_drcSetParam(hDrcInfo, DRC_CUT_SCALE, value);27
+ uniDrcErr = FDK_drcDec_SetParam(self->hUniDrcDecoder, DRC_DEC_COMPRESS,28
+ value * (FL2FXCONST_DBL(0.5f / 127.0f)));29
break;30
31
case AAC_DRC_BOOST_FACTOR:32
/* DRC boost factor (where 0 is no and 127 is max boost) */33
errorStatus = aacDecoder_drcSetParam(hDrcInfo, DRC_BOOST_SCALE, value);34
+ uniDrcErr = FDK_drcDec_SetParam(self->hUniDrcDecoder, DRC_DEC_BOOST,35
+ value * (FL2FXCONST_DBL(0.5f / 127.0f)));36
break;37
38
case AAC_DRC_REFERENCE_LEVEL:39
40
uniDrcErr = FDK_drcDec_SetParam(self->hUniDrcDecoder, DRC_DEC_EFFECT_TYPE,41
(FIXP_DBL)value);42
break;43
+ case AAC_UNIDRC_ALBUM_MODE:44
+ uniDrcErr = FDK_drcDec_SetParam(self->hUniDrcDecoder, DRC_DEC_ALBUM_MODE,45
+ (FIXP_DBL)value);46
+ break;47
+48
case AAC_TPDEC_CLEAR_BUFFER:49
errTp = transportDec_SetParam(hTpDec, TPDEC_PARAM_RESET, 1);50
self->streamInfo.numLostAccessUnits = 0;51
52
return n;53
}54
55
-LINKSPEC_CPP AAC_DECODER_ERROR56
-aacDecoder_DecodeFrame(HANDLE_AACDECODER self, INT_PCM *pTimeData_extern,57
- const INT timeDataSize_extern, const UINT flags) {58
+LINKSPEC_CPP AAC_DECODER_ERROR aacDecoder_DecodeFrame(HANDLE_AACDECODER self,59
+ INT_PCM *pTimeData,60
+ const INT timeDataSize,61
+ const UINT flags) {62
AAC_DECODER_ERROR ErrorStatus;63
INT layer;64
INT nBits;65
+ INT timeData2Size;66
+ INT timeData3Size;67
+ INT timeDataHeadroom;68
HANDLE_FDK_BITSTREAM hBs;69
int fTpInterruption = 0; /* Transport originated interruption detection. */70
int fTpConceal = 0; /* Transport originated concealment. */71
- INT_PCM *pTimeData = NULL;72
- INT timeDataSize = 0;73
UINT accessUnit = 0;74
UINT numAccessUnits = 1;75
UINT numPrerollAU = 0;76
- int fEndAuNotAdjusted = 0; /* The end of the access unit was not adjusted */77
- int applyCrossfade = 1; /* flag indicates if flushing was possible */78
- FIXP_PCM *pTimeDataFixpPcm; /* Signal buffer for decoding process before PCM79
- processing */80
- INT timeDataFixpPcmSize;81
- PCM_DEC *pTimeDataPcmPost; /* Signal buffer for PCM post-processing */82
- INT timeDataPcmPostSize;83
+ int fEndAuNotAdjusted = 0; /* The end of the access unit was not adjusted */84
+ int applyCrossfade = 1; /* flag indicates if flushing was possible */85
+ PCM_DEC *pTimeData2;86
+ PCM_AAC *pTimeData3;87
88
if (self == NULL) {89
return AAC_DEC_INVALID_HANDLE;90
}91
92
- pTimeData = self->pcmOutputBuffer;93
- timeDataSize = sizeof(self->pcmOutputBuffer) / sizeof(*self->pcmOutputBuffer);94
-95
if (flags & AACDEC_INTR) {96
self->streamInfo.numLostAccessUnits = 0;97
}98
99
}100
101
/* Signal bit stream interruption to other modules if required. */102
- if (fTpInterruption || (flags & AACDEC_INTR)) {103
+ if (fTpInterruption || ((flags & AACDEC_INTR) && (accessUnit == 0))) {104
aacDecoder_SignalInterruption(self);105
- if (!(flags & AACDEC_INTR)) {106
+ if (!((flags & AACDEC_INTR) && (accessUnit == 0))) {107
ErrorStatus = AAC_DEC_TRANSPORT_SYNC_ERROR;108
goto bail;109
}110
111
/* Use limiter configuration as requested. */112
self->limiterEnableCurr = self->limiterEnableUser;113
}114
- /* reset limiter gain on a per frame basis */115
- self->extGain[0] = FL2FXCONST_DBL(1.0f / (float)(1 << TDL_GAIN_SCALING));116
117
- pTimeDataFixpPcm = pTimeData;118
- timeDataFixpPcmSize = timeDataSize;119
+ /* reset DRC level normalization gain on a per frame basis */120
+ self->extGain[0] = AACDEC_DRC_GAIN_INIT_VALUE;121
+122
+ pTimeData2 = self->pTimeData2;123
+ timeData2Size = self->timeData2Size / sizeof(PCM_DEC);124
+ pTimeData3 = (PCM_AAC *)self->pTimeData2;125
+ timeData3Size = self->timeData2Size / sizeof(PCM_AAC);126
127
ErrorStatus = CAacDecoder_DecodeFrame(128
self,129
flags | (fTpConceal ? AACDEC_CONCEAL : 0) |130
((self->flushStatus && !(flags & AACDEC_CONCEAL)) ? AACDEC_FLUSH131
: 0),132
- pTimeDataFixpPcm + 0, timeDataFixpPcmSize,133
- self->streamInfo.aacSamplesPerFrame + 0);134
+ pTimeData2 + 0, timeData2Size, self->streamInfo.aacSamplesPerFrame + 0);135
+136
+ timeDataHeadroom = self->aacOutDataHeadroom;137
138
/* if flushing for USAC DASH IPF was not possible go on with decoding139
* preroll */140
141
}142
}143
144
- /* If the current pTimeDataFixpPcm does not contain a valid signal, there145
+ /* If the current pTimeData2 does not contain a valid signal, there146
* nothing else we can do, so bail. */147
if (!IS_OUTPUT_VALID(ErrorStatus)) {148
goto bail;149
150
self->streamInfo.numChannels = self->streamInfo.aacNumChannels;151
152
{153
- FDK_Delay_Apply(&self->usacResidualDelay,154
- pTimeDataFixpPcm +155
- 1 * (self->streamInfo.aacSamplesPerFrame + 0) + 0,156
- self->streamInfo.frameSize, 0);157
+ FDK_Delay_Apply(158
+ &self->usacResidualDelay,159
+ pTimeData2 + 1 * (self->streamInfo.aacSamplesPerFrame + 0) + 0,160
+ self->streamInfo.frameSize, 0);161
}162
163
/* Setting of internal MPS state; may be reset in CAacDecoder_SyncQmfMode164
165
}166
}167
168
- self->qmfDomain.globalConf.TDinput = pTimeData;169
-170
switch (FDK_QmfDomain_Configure(&self->qmfDomain)) {171
default:172
case QMF_DOMAIN_INIT_ERROR:173
174
sbrDecoder_SetParam(self->hSbrDecoder, SBR_SKIP_QMF,175
(self->mpsEnableCurr) ? 2 : 0);176
177
- INT_PCM *input;178
- input = (INT_PCM *)self->workBufferCore2;179
- FDKmemcpy(input, pTimeData,180
- sizeof(INT_PCM) * (self->streamInfo.numChannels) *181
+ PCM_AAC *input;182
+ input = (PCM_AAC *)self->workBufferCore2;183
+ FDKmemcpy(input, pTimeData3,184
+ sizeof(PCM_AAC) * (self->streamInfo.numChannels) *185
(self->streamInfo.frameSize));186
187
/* apply SBR processing */188
- sbrError = sbrDecoder_Apply(self->hSbrDecoder, input, pTimeData,189
- timeDataSize, &self->streamInfo.numChannels,190
- &self->streamInfo.sampleRate,191
- &self->mapDescr, self->chMapIndex,192
- self->frameOK, &self->psPossible);193
+ sbrError = sbrDecoder_Apply(194
+ self->hSbrDecoder, input, pTimeData3, timeData3Size,195
+ &self->streamInfo.numChannels, &self->streamInfo.sampleRate,196
+ &self->mapDescr, self->chMapIndex, self->frameOK, &self->psPossible,197
+ self->aacOutDataHeadroom, &timeDataHeadroom);198
199
if (sbrError == SBRDEC_OK) {200
/* Update data in streaminfo structure. Assume that the SBR upsampling201
fdk-aac-2.0.1.tar.gz/libAACdec/src/block.cpp -> fdk-aac-2.0.2.tar.gz/libAACdec/src/block.cpp
Changed
49
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
12
void CBlock_FrequencyToTime(13
CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo,14
- CAacDecoderChannelInfo *pAacDecoderChannelInfo, FIXP_PCM outSamples[],15
+ CAacDecoderChannelInfo *pAacDecoderChannelInfo, PCM_DEC outSamples[],16
const SHORT frameLen, const int frameOk, FIXP_DBL *pWorkBuffer1,17
- UINT elFlags, INT elCh) {18
+ const INT aacOutDataHeadroom, UINT elFlags, INT elCh) {19
int fr, fl, tl, nSpec;20
21
#if defined(FDK_ASSERT_ENABLE)22
23
bass_pf_1sf_delay(p2_synth, pitch, pit_gain, frameLen,24
(LpdSfd + 2) * L_SUBFR + BPF_SFD * L_SUBFR,25
frameLen - (LpdSfd + 4) * L_SUBFR, outSamples,26
+ aacOutDataHeadroom,27
pAacDecoderStaticChannelInfo->mem_bpf);28
}29
30
31
? MLT_FLAG_CURR_ALIAS_SYMMETRY32
: 0);33
34
- scaleValuesSaturate(outSamples, tmp, frameLen, MDCT_OUT_HEADROOM);35
+ scaleValuesSaturate(outSamples, tmp, frameLen,36
+ MDCT_OUT_HEADROOM - aacOutDataHeadroom);37
}38
}39
40
41
#include "ldfiltbank.h"42
void CBlock_FrequencyToTimeLowDelay(43
CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo,44
- CAacDecoderChannelInfo *pAacDecoderChannelInfo, FIXP_PCM outSamples[],45
+ CAacDecoderChannelInfo *pAacDecoderChannelInfo, PCM_DEC outSamples[],46
const short frameLen) {47
InvMdctTransformLowDelay_fdk(48
SPEC_LONG(pAacDecoderChannelInfo->pSpectralCoefficient),49
fdk-aac-2.0.1.tar.gz/libAACdec/src/block.h -> fdk-aac-2.0.2.tar.gz/libAACdec/src/block.h
Changed
30
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
*/12
void CBlock_FrequencyToTime(13
CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo,14
- CAacDecoderChannelInfo *pAacDecoderChannelInfo, FIXP_PCM outSamples[],15
+ CAacDecoderChannelInfo *pAacDecoderChannelInfo, PCM_DEC outSamples[],16
const SHORT frameLen, const int frameOk, FIXP_DBL *pWorkBuffer1,17
- UINT elFlags, INT elCh);18
+ const INT aacOutDataHeadroom, UINT elFlags, INT elCh);19
20
/**21
* \brief Transform double lapped MDCT (AAC-ELD) spectral data into time domain.22
*/23
void CBlock_FrequencyToTimeLowDelay(24
CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo,25
- CAacDecoderChannelInfo *pAacDecoderChannelInfo, FIXP_PCM outSamples[],26
+ CAacDecoderChannelInfo *pAacDecoderChannelInfo, PCM_DEC outSamples[],27
const short frameLen);28
29
AAC_DECODER_ERROR CBlock_InverseQuantizeSpectralData(30
fdk-aac-2.0.1.tar.gz/libAACdec/src/conceal.cpp -> fdk-aac-2.0.2.tar.gz/libAACdec/src/conceal.cpp
Changed
152
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
12
/* TimeDomainFading */13
static void CConcealment_TDFadePcmAtt(int start, int len, FIXP_DBL fadeStart,14
- FIXP_DBL fadeStop, FIXP_PCM *pcmdata);15
+ FIXP_DBL fadeStop, PCM_DEC *pcmdata);16
static void CConcealment_TDFadeFillFadingStations(FIXP_DBL *fadingStations,17
int *fadingSteps,18
FIXP_DBL fadeStop,19
20
21
static int CConcealment_TDNoise_Random(ULONG *seed);22
static void CConcealment_TDNoise_Apply(CConcealmentInfo *const pConcealmentInfo,23
- const int len, FIXP_PCM *const pcmdata);24
+ const int len,25
+ const INT aacOutDataHeadroom,26
+ PCM_DEC *const pcmdata);27
28
static BLOCK_TYPE CConcealment_GetWinSeq(int prevWinSeq) {29
BLOCK_TYPE newWinSeq = BLOCK_LONG;30
31
int sfb, line = 0;32
int fac_shift;33
int fac_mod;34
- FIXP_DBL accu;35
36
for (sfb = 0; sfb < sfbCnt; sfb++) {37
fac_shift =38
39
fac_mod = fac_shift & 3;40
fac_shift = (fac_shift >> 2) + 1;41
fac_shift += *pSpecScalePrv - fixMax(*pSpecScalePrv, *pSpecScaleAct);42
+ fac_shift = fMax(fMin(fac_shift, DFRACT_BITS - 1), -(DFRACT_BITS - 1));43
44
for (; line < pSfbOffset[sfb + 1]; line++) {45
- accu = fMult(*(spectrum + line), facMod4Table[fac_mod]);46
- if (fac_shift < 0) {47
- accu >>= -fac_shift;48
- } else {49
- accu <<= fac_shift;50
- }51
- *(spectrum + line) = accu;52
+ FIXP_DBL accu = fMult(*(spectrum + line), facMod4Table[fac_mod]);53
+ *(spectrum + line) = scaleValue(accu, fac_shift);54
}55
}56
*pSpecScaleOut = fixMax(*pSpecScalePrv, *pSpecScaleAct);57
58
}59
60
if (packedSign & 0x1) {61
- spec[i] = -spec[i];62
+ spec[i] = -fMax(spec[i], (FIXP_DBL)(MINVAL_DBL + 1));63
}64
packedSign >>= 1;65
66
67
68
INT CConcealment_TDFading(69
int len, CAacDecoderStaticChannelInfo **ppAacDecoderStaticChannelInfo,70
- FIXP_PCM *pcmdata, FIXP_PCM *pcmdata_1) {71
+ const INT aacOutDataHeadroom, PCM_DEC *pcmdata, PCM_DEC *pcmdata_1) {72
/*73
Do the fading in Time domain based on concealment states and core mode74
*/75
76
start += len;77
}78
}79
- CConcealment_TDNoise_Apply(pConcealmentInfo, len, pcmdata);80
+ CConcealment_TDNoise_Apply(pConcealmentInfo, len, aacOutDataHeadroom,81
+ pcmdata);82
83
/* Save end-of-frame attenuation and fading type */84
pConcealmentInfo->lastFadingType = fadingType;85
86
87
/* attenuate pcmdata in Time Domain Fading process */88
static void CConcealment_TDFadePcmAtt(int start, int len, FIXP_DBL fadeStart,89
- FIXP_DBL fadeStop, FIXP_PCM *pcmdata) {90
+ FIXP_DBL fadeStop, PCM_DEC *pcmdata) {91
int i;92
FIXP_DBL dStep;93
FIXP_DBL dGain;94
FIXP_DBL dGain_apply;95
- int bitshift = (DFRACT_BITS - SAMPLE_BITS);96
97
/* set start energy */98
dGain = fadeStart;99
100
*/101
dGain_apply = fMax((FIXP_DBL)0, dGain);102
/* finally, attenuate samples */103
- pcmdata[i] = (FIXP_PCM)((fMult(pcmdata[i], (dGain_apply))) >> bitshift);104
+ pcmdata[i] = FIXP_DBL2PCM_DEC(fMult(pcmdata[i], dGain_apply));105
}106
}107
108
109
}110
111
static void CConcealment_TDNoise_Apply(CConcealmentInfo *const pConcealmentInfo,112
- const int len, FIXP_PCM *const pcmdata) {113
- FIXP_PCM *states = pConcealmentInfo->TDNoiseStates;114
- FIXP_PCM noiseVal;115
+ const int len,116
+ const INT aacOutDataHeadroom,117
+ PCM_DEC *const pcmdata) {118
+ PCM_DEC *states = pConcealmentInfo->TDNoiseStates;119
+ PCM_DEC noiseVal;120
FIXP_DBL noiseValLong;121
FIXP_SGL *coef = pConcealmentInfo->TDNoiseCoef;122
FIXP_DBL TDNoiseAtt;123
124
/* create filtered noise */125
states[2] = states[1];126
states[1] = states[0];127
- states[0] = ((FIXP_PCM)CConcealment_TDNoise_Random(&seed));128
+ states[0] =129
+ FIXP_DBL2PCM_DEC((FIXP_DBL)CConcealment_TDNoise_Random(&seed));130
noiseValLong = fMult(states[0], coef[0]) + fMult(states[1], coef[1]) +131
fMult(states[2], coef[2]);132
- noiseVal = FX_DBL2FX_PCM(fMult(noiseValLong, TDNoiseAtt));133
+ noiseVal = FIXP_DBL2PCM_DEC(fMult(noiseValLong, TDNoiseAtt) >>134
+ aacOutDataHeadroom);135
136
/* add filtered noise - check for clipping, before */137
- if (noiseVal > (FIXP_PCM)0 &&138
- pcmdata[ii] > (FIXP_PCM)MAXVAL_FIXP_PCM - noiseVal) {139
- noiseVal = noiseVal * (FIXP_PCM)-1;140
- } else if (noiseVal < (FIXP_PCM)0 &&141
- pcmdata[ii] < (FIXP_PCM)MINVAL_FIXP_PCM - noiseVal) {142
- noiseVal = noiseVal * (FIXP_PCM)-1;143
+ if (noiseVal > (PCM_DEC)0 &&144
+ pcmdata[ii] > (PCM_DEC)MAXVAL_PCM_DEC - noiseVal) {145
+ noiseVal = noiseVal * (PCM_DEC)-1;146
+ } else if (noiseVal < (PCM_DEC)0 &&147
+ pcmdata[ii] < (PCM_DEC)MINVAL_PCM_DEC - noiseVal) {148
+ noiseVal = noiseVal * (PCM_DEC)-1;149
}150
151
pcmdata[ii] += noiseVal;152
fdk-aac-2.0.1.tar.gz/libAACdec/src/conceal.h -> fdk-aac-2.0.2.tar.gz/libAACdec/src/conceal.h
Changed
18
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
12
INT CConcealment_TDFading(13
int len, CAacDecoderStaticChannelInfo **ppAacDecoderStaticChannelInfo,14
- FIXP_PCM *pcmdata, FIXP_PCM *pcmdata_1);15
+ const INT aacOutDataHeadroom, PCM_DEC *pcmdata, PCM_DEC *pcmdata_1);16
17
#endif /* #ifndef CONCEAL_H */18
fdk-aac-2.0.1.tar.gz/libAACdec/src/conceal_types.h -> fdk-aac-2.0.2.tar.gz/libAACdec/src/conceal_types.h
Changed
19
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
FIXP_DBL last_tcx_gain;12
INT last_tcx_gain_e;13
ULONG TDNoiseSeed;14
- FIXP_PCM TDNoiseStates[3];15
+ PCM_DEC TDNoiseStates[3];16
FIXP_SGL TDNoiseCoef[3];17
FIXP_SGL TDNoiseAtt;18
19
fdk-aac-2.0.1.tar.gz/libAACdec/src/ldfiltbank.cpp -> fdk-aac-2.0.2.tar.gz/libAACdec/src/ldfiltbank.cpp
Changed
138
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
#if defined(__arm__)12
#endif13
14
-static void multE2_DinvF_fdk(FIXP_PCM *output, FIXP_DBL *x, const FIXP_WTB *fb,15
+static void multE2_DinvF_fdk(PCM_DEC *output, FIXP_DBL *x, const FIXP_WTB *fb,16
FIXP_DBL *z, const int N) {17
int i;18
19
- /* scale for FIXP_DBL -> INT_PCM conversion. */20
- const int scale = (DFRACT_BITS - SAMPLE_BITS) - LDFB_HEADROOM;21
-#if ((DFRACT_BITS - SAMPLE_BITS - LDFB_HEADROOM) > 0)22
+ /* scale for FIXP_DBL -> PCM_DEC conversion: */23
+ const int scale = (DFRACT_BITS - PCM_OUT_BITS) - LDFB_HEADROOM + (3);24
+25
+#if ((DFRACT_BITS - PCM_OUT_BITS - LDFB_HEADROOM + (3) - 1) > 0)26
FIXP_DBL rnd_val_wts0 = (FIXP_DBL)0;27
FIXP_DBL rnd_val_wts1 = (FIXP_DBL)0;28
+#if ((DFRACT_BITS - PCM_OUT_BITS - LDFB_HEADROOM + (3) - WTS0 - 1) > 0)29
if (-WTS0 - 1 + scale)30
rnd_val_wts0 = (FIXP_DBL)(1 << (-WTS0 - 1 + scale - 1));31
+#endif32
if (-WTS1 - 1 + scale)33
rnd_val_wts1 = (FIXP_DBL)(1 << (-WTS1 - 1 + scale - 1));34
#endif35
36
FIXP_DBL z0, z2, tmp;37
38
z2 = x[N / 2 + i];39
- z0 = z2 + (fMultDiv2(z[N / 2 + i], fb[2 * N + i]) >> (-WTS2 - 1));40
+ z0 = fAddSaturate(z2,41
+ (fMultDiv2(z[N / 2 + i], fb[2 * N + i]) >> (-WTS2 - 1)));42
43
- z[N / 2 + i] = x[N / 2 - 1 - i] +44
- (fMultDiv2(z[N + i], fb[2 * N + N / 2 + i]) >> (-WTS2 - 1));45
+ z[N / 2 + i] = fAddSaturate(46
+ x[N / 2 - 1 - i],47
+ (fMultDiv2(z[N + i], fb[2 * N + N / 2 + i]) >> (-WTS2 - 1)));48
49
tmp = (fMultDiv2(z[N / 2 + i], fb[N + N / 2 - 1 - i]) +50
fMultDiv2(z[i], fb[N + N / 2 + i]));51
52
-#if ((DFRACT_BITS - SAMPLE_BITS - LDFB_HEADROOM) > 0)53
+#if ((DFRACT_BITS - PCM_OUT_BITS - LDFB_HEADROOM + (3) - 1) > 0)54
FDK_ASSERT((-WTS1 - 1 + scale) >= 0);55
FDK_ASSERT(tmp <= ((FIXP_DBL)0x7FFFFFFF -56
rnd_val_wts1)); /* rounding must not cause overflow */57
- output[(N * 3 / 4 - 1 - i)] = (FIXP_PCM)SATURATE_RIGHT_SHIFT(58
+ output[(N * 3 / 4 - 1 - i)] = (PCM_DEC)SATURATE_RIGHT_SHIFT(59
tmp + rnd_val_wts1, -WTS1 - 1 + scale, PCM_OUT_BITS);60
#else61
FDK_ASSERT((WTS1 + 1 - scale) >= 0);62
output[(N * 3 / 4 - 1 - i)] =63
- (FIXP_PCM)SATURATE_LEFT_SHIFT(tmp, WTS1 + 1 - scale, PCM_OUT_BITS);64
+ (PCM_DEC)SATURATE_LEFT_SHIFT(tmp, WTS1 + 1 - scale, PCM_OUT_BITS);65
#endif66
67
z[i] = z0;68
69
FIXP_DBL z0, z2, tmp0, tmp1;70
71
z2 = x[N / 2 + i];72
- z0 = z2 + (fMultDiv2(z[N / 2 + i], fb[2 * N + i]) >> (-WTS2 - 1));73
+ z0 = fAddSaturate(z2,74
+ (fMultDiv2(z[N / 2 + i], fb[2 * N + i]) >> (-WTS2 - 1)));75
76
- z[N / 2 + i] = x[N / 2 - 1 - i] +77
- (fMultDiv2(z[N + i], fb[2 * N + N / 2 + i]) >> (-WTS2 - 1));78
+ z[N / 2 + i] = fAddSaturate(79
+ x[N / 2 - 1 - i],80
+ (fMultDiv2(z[N + i], fb[2 * N + N / 2 + i]) >> (-WTS2 - 1)));81
82
tmp0 = (fMultDiv2(z[N / 2 + i], fb[N / 2 - 1 - i]) +83
fMultDiv2(z[i], fb[N / 2 + i]));84
tmp1 = (fMultDiv2(z[N / 2 + i], fb[N + N / 2 - 1 - i]) +85
fMultDiv2(z[i], fb[N + N / 2 + i]));86
87
-#if ((DFRACT_BITS - SAMPLE_BITS - LDFB_HEADROOM) > 0)88
+#if ((DFRACT_BITS - PCM_OUT_BITS - LDFB_HEADROOM + (3) - 1) > 0)89
FDK_ASSERT((-WTS0 - 1 + scale) >= 0);90
FDK_ASSERT(tmp0 <= ((FIXP_DBL)0x7FFFFFFF -91
rnd_val_wts0)); /* rounding must not cause overflow */92
FDK_ASSERT(tmp1 <= ((FIXP_DBL)0x7FFFFFFF -93
rnd_val_wts1)); /* rounding must not cause overflow */94
- output[(i - N / 4)] = (FIXP_PCM)SATURATE_RIGHT_SHIFT(95
+ output[(i - N / 4)] = (PCM_DEC)SATURATE_RIGHT_SHIFT(96
tmp0 + rnd_val_wts0, -WTS0 - 1 + scale, PCM_OUT_BITS);97
- output[(N * 3 / 4 - 1 - i)] = (FIXP_PCM)SATURATE_RIGHT_SHIFT(98
+ output[(N * 3 / 4 - 1 - i)] = (PCM_DEC)SATURATE_RIGHT_SHIFT(99
tmp1 + rnd_val_wts1, -WTS1 - 1 + scale, PCM_OUT_BITS);100
#else101
FDK_ASSERT((WTS0 + 1 - scale) >= 0);102
output[(i - N / 4)] =103
- (FIXP_PCM)SATURATE_LEFT_SHIFT(tmp0, WTS0 + 1 - scale, PCM_OUT_BITS);104
+ (PCM_DEC)SATURATE_LEFT_SHIFT(tmp0, WTS0 + 1 - scale, PCM_OUT_BITS);105
output[(N * 3 / 4 - 1 - i)] =106
- (FIXP_PCM)SATURATE_LEFT_SHIFT(tmp1, WTS1 + 1 - scale, PCM_OUT_BITS);107
+ (PCM_DEC)SATURATE_LEFT_SHIFT(tmp1, WTS1 + 1 - scale, PCM_OUT_BITS);108
#endif109
z[i] = z0;110
z[N + i] = z2;111
112
for (i = 0; i < N / 4; i++) {113
FIXP_DBL tmp0 = fMultDiv2(z[i], fb[N / 2 + i]);114
115
-#if ((DFRACT_BITS - SAMPLE_BITS - LDFB_HEADROOM) > 0)116
+#if ((DFRACT_BITS - PCM_OUT_BITS - LDFB_HEADROOM + (3) - 1) > 0)117
FDK_ASSERT((-WTS0 - 1 + scale) >= 0);118
FDK_ASSERT(tmp0 <= ((FIXP_DBL)0x7FFFFFFF -119
rnd_val_wts0)); /* rounding must not cause overflow */120
- output[(N * 3 / 4 + i)] = (FIXP_PCM)SATURATE_RIGHT_SHIFT(121
+ output[(N * 3 / 4 + i)] = (PCM_DEC)SATURATE_RIGHT_SHIFT(122
tmp0 + rnd_val_wts0, -WTS0 - 1 + scale, PCM_OUT_BITS);123
#else124
FDK_ASSERT((WTS0 + 1 - scale) >= 0);125
output[(N * 3 / 4 + i)] =126
- (FIXP_PCM)SATURATE_LEFT_SHIFT(tmp0, WTS0 + 1 - scale, PCM_OUT_BITS);127
+ (PCM_DEC)SATURATE_LEFT_SHIFT(tmp0, WTS0 + 1 - scale, PCM_OUT_BITS);128
#endif129
}130
}131
132
int InvMdctTransformLowDelay_fdk(FIXP_DBL *mdctData, const int mdctData_e,133
- FIXP_PCM *output, FIXP_DBL *fs_buffer,134
+ PCM_DEC *output, FIXP_DBL *fs_buffer,135
const int N) {136
const FIXP_WTB *coef;137
FIXP_DBL gain = (FIXP_DBL)0;138
fdk-aac-2.0.1.tar.gz/libAACdec/src/ldfiltbank.h -> fdk-aac-2.0.2.tar.gz/libAACdec/src/ldfiltbank.h
Changed
22
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
#define LDFILTBANK_H12
13
#include "common_fix.h"14
+#include "aac_rom.h"15
16
int InvMdctTransformLowDelay_fdk(FIXP_DBL *mdctdata_m, const int mdctdata_e,17
- FIXP_PCM *mdctOut, FIXP_DBL *fs_buffer,18
+ PCM_DEC *mdctOut, FIXP_DBL *fs_buffer,19
const int frameLength);20
21
#endif22
fdk-aac-2.0.1.tar.gz/libAACdec/src/stereo.cpp -> fdk-aac-2.0.2.tar.gz/libAACdec/src/stereo.cpp
Changed
125
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
for (int i = 0; i < windowLen; i++) {12
dmx_re_prev[i] =13
((staticSpectralCoeffsL[index_offset + i] >>14
- srLeftChan) +15
+ fMin(DFRACT_BITS - 1, srLeftChan + 1)) +16
(staticSpectralCoeffsR[index_offset + i] >>17
- srRightChan)) >>18
- 1;19
+ fMin(DFRACT_BITS - 1, srRightChan + 1)));20
}21
} else {22
for (int i = 0; i < windowLen; i++) {23
dmx_re_prev[i] =24
((staticSpectralCoeffsL[index_offset + i] >>25
- srLeftChan) -26
+ fMin(DFRACT_BITS - 1, srLeftChan + 1)) -27
(staticSpectralCoeffsR[index_offset + i] >>28
- srRightChan)) >>29
- 1;30
+ fMin(DFRACT_BITS - 1, srRightChan + 1)));31
}32
}33
}34
35
if (window == 0) {36
if (dmx_re_prev_e < frameMaxScale) {37
if (mainband_flag == 0) {38
- scaleValues(dmx_re_prev, store_dmx_re_prev, windowLen,39
- -(frameMaxScale - dmx_re_prev_e));40
+ scaleValues(41
+ dmx_re_prev, store_dmx_re_prev, windowLen,42
+ -fMin(DFRACT_BITS - 1, (frameMaxScale - dmx_re_prev_e)));43
} else {44
- for (int i = 0; i < windowLen; i++) {45
- dmx_re_prev[i] >>= (frameMaxScale - dmx_re_prev_e);46
- }47
+ scaleValues(48
+ dmx_re_prev, windowLen,49
+ -fMin(DFRACT_BITS - 1, (frameMaxScale - dmx_re_prev_e)));50
}51
} else {52
if (mainband_flag == 0) {53
54
FDK_ASSERT(pAacDecoderChannelInfo[L]->icsInfo.WindowSequence ==55
BLOCK_SHORT);56
if (specScaleL[window - 1] < frameMaxScale) {57
- for (int i = 0; i < windowLen; i++) {58
- dmx_re[windowLen * (window - 1) + i] >>=59
- (frameMaxScale - specScaleL[window - 1]);60
- }61
+ scaleValues(&dmx_re[windowLen * (window - 1)], windowLen,62
+ -fMin(DFRACT_BITS - 1,63
+ (frameMaxScale - specScaleL[window - 1])));64
} else {65
specScaleL[window] = specScaleL[window - 1];66
specScaleR[window] = specScaleR[window - 1];67
68
} /* if ( pJointStereoData->complex_coef == 1 ) */69
70
/* 4. upmix process */71
- INT pred_dir = cplxPredictionData->pred_dir ? -1 : 1;72
+ LONG pred_dir = cplxPredictionData->pred_dir ? -1 : 1;73
/* 0.1 in Q-3.34 */74
const FIXP_DBL pointOne = 0x66666666; /* 0.8 */75
/* Shift value for the downmix */76
77
the downmix. "dmx_re" and "specL" are two different pointers78
pointing to separate arrays, which may or may not contain the79
same data (with different scaling).80
- */81
-82
- /* help1: alpha_re[i] * dmx_re[i] */83
- FIXP_DBL help1 = fMultDiv2(alpha_re_tmp, *p2dmxRe++);84
85
- /* tmp: dmx_im[i] */86
- FIXP_DBL tmp = (*p2dmxIm++) << shift_dmx;87
-88
- /* help2: alpha_im[i] * dmx_im[i] */89
- FIXP_DBL help2 = fMultDiv2(alpha_im_tmp, tmp);90
-91
- /* help3: alpha_re[i] * dmx_re[i] + alpha_im[i] * dmx_im[i] */92
- FIXP_DBL help3 = help1 + help2;93
+ specL[i] = + (specL[i] + side);94
+ specR[i] = -/+ (specL[i] - side);95
+ */96
+ FIXP_DBL side, left, right;97
98
- /* side (= help4) = specR[i] - (dmx_re[i] * specL[i] + alpha_im[i]99
- * * dmx_im[i]) */100
- FIXP_DBL help4 = *p2CoeffR - scaleValue(help3, help3_shift);101
+ side = fMultAddDiv2(fMultDiv2(alpha_re_tmp, *p2dmxRe++),102
+ alpha_im_tmp, (*p2dmxIm++) << shift_dmx);103
+ side = ((*p2CoeffR) >> 2) -104
+ (FIXP_DBL)SATURATE_SHIFT(side, -(help3_shift - 2),105
+ DFRACT_BITS - 2);106
107
- /* We calculate the left and right output by using the helper108
- * function */109
- /* specR[i] = -/+ (specL[i] - side); */110
- *p2CoeffR =111
- (FIXP_DBL)((LONG)(*p2CoeffL - help4) * (LONG)pred_dir);112
- p2CoeffR++;113
+ left = ((*p2CoeffL) >> 2) + side;114
+ right = ((*p2CoeffL) >> 2) - side;115
+ right = (FIXP_DBL)((LONG)right * pred_dir);116
117
- /* specL[i] = specL[i] + side; */118
- *p2CoeffL = *p2CoeffL + help4;119
- p2CoeffL++;120
+ *p2CoeffL++ = SATURATE_LEFT_SHIFT_ALT(left, 2, DFRACT_BITS);121
+ *p2CoeffR++ = SATURATE_LEFT_SHIFT_ALT(right, 2, DFRACT_BITS);122
}123
}124
125
fdk-aac-2.0.1.tar.gz/libAACdec/src/usacdec_acelp.cpp -> fdk-aac-2.0.2.tar.gz/libAACdec/src/usacdec_acelp.cpp
Changed
82
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
int i;12
13
for (i = 0; i < L; i++) {14
- out[i] = in[i] - fMult(PREEMPH_FAC, in[i - 1]);15
+ out[i] = fAddSaturate(in[i], -fMult(PREEMPH_FAC, in[i - 1]));16
}17
18
return;19
20
/* Note: code[L_SUBFR] and exc2[L_SUBFR] share the same memory!21
If exc2[i] is written, code[i] will be destroyed!22
*/23
-#define SF (SF_CODE + SF_GAIN_C + 1 - SF_EXC)24
+#define SF_HEADROOM (1)25
+#define SF (SF_CODE + SF_GAIN_C + 1 - SF_EXC - SF_HEADROOM)26
+#define SF_GAIN_P2 (SF_GAIN_P - SF_HEADROOM)27
28
int i;29
FIXP_DBL tmp, cpe, code_smooth_prev, code_smooth;30
31
cpe = (period_fac >> (2 - SF_PFAC)) + FL2FXCONST_DBL(0.25f);32
33
/* u'(n) */34
- tmp = fMultDiv2(*exc, gain_pit) << (SF_GAIN_P + 1); /* v(0)*g_p */35
- *exc++ = tmp + (fMultDiv2(code[0], gain_code) << SF);36
+ tmp = fMultDiv2(*exc, gain_pit) << (SF_GAIN_P2 + 1); /* v(0)*g_p */37
+ *exc++ = (tmp + (fMultDiv2(code[0], gain_code) << SF)) << SF_HEADROOM;38
39
/* u(n) */40
code_smooth_prev = fMultDiv2(*code++, gain_code_smoothed)41
42
code_smooth = fMultDiv2(code_i, gain_code_smoothed) << SF; /* c(1) * g_sc */43
tmp += code_smooth_prev; /* tmp = v(0)*g_p + c(0)*g_sc */44
cpe_code_smooth = fMultDiv2(cpe, code_smooth);45
- *exc2++ = tmp - cpe_code_smooth;46
+ *exc2++ = (tmp - cpe_code_smooth) << SF_HEADROOM;47
cpe_code_smooth_prev = fMultDiv2(cpe, code_smooth_prev);48
49
i = L_SUBFR - 2;50
do /* ARM926: 22 cycles per iteration */51
{52
/* u'(n) */53
- tmp = fMultDiv2(*exc, gain_pit) << (SF_GAIN_P + 1);54
- *exc++ = tmp + (fMultDiv2(code_i, gain_code) << SF);55
+ tmp = fMultDiv2(*exc, gain_pit) << (SF_GAIN_P2 + 1);56
+ *exc++ = (tmp + (fMultDiv2(code_i, gain_code) << SF)) << SF_HEADROOM;57
/* u(n) */58
tmp += code_smooth; /* += g_sc * c(i) */59
tmp -= cpe_code_smooth_prev;60
61
code_i = *code++;62
code_smooth = fMultDiv2(code_i, gain_code_smoothed) << SF;63
cpe_code_smooth = fMultDiv2(cpe, code_smooth);64
- *exc2++ = tmp - cpe_code_smooth; /* tmp - c_pe * g_sc * c(i+1) */65
+ *exc2++ = (tmp - cpe_code_smooth)66
+ << SF_HEADROOM; /* tmp - c_pe * g_sc * c(i+1) */67
} while (--i != 0);68
69
/* u'(n) */70
- tmp = fMultDiv2(*exc, gain_pit) << (SF_GAIN_P + 1);71
- *exc = tmp + (fMultDiv2(code_i, gain_code) << SF);72
+ tmp = fMultDiv2(*exc, gain_pit) << (SF_GAIN_P2 + 1);73
+ *exc = (tmp + (fMultDiv2(code_i, gain_code) << SF)) << SF_HEADROOM;74
/* u(n) */75
tmp += code_smooth;76
tmp -= cpe_code_smooth_prev;77
- *exc2++ = tmp;78
+ *exc2++ = tmp << SF_HEADROOM;79
80
return;81
}82
fdk-aac-2.0.1.tar.gz/libAACdec/src/usacdec_fac.cpp -> fdk-aac-2.0.2.tar.gz/libAACdec/src/usacdec_fac.cpp
Changed
114
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
/* Overlap Add */12
x0 = -fMult(*pOvl--, pWindow[i].v.re);13
14
- *pOut0 += IMDCT_SCALE_DBL(x0);15
+ *pOut0 = fAddSaturate(*pOut0, IMDCT_SCALE_DBL(x0));16
pOut0++;17
}18
} else {19
20
/* Overlap Add */21
x0 = fMult(*pOvl--, pWindow[i].v.re);22
23
- *pOut0 += IMDCT_SCALE_DBL(x0);24
+ *pOut0 = fAddSaturate(*pOut0, IMDCT_SCALE_DBL(x0));25
pOut0++;26
}27
}28
29
0) { /* this should only happen for ACELP -> TCX20 -> ACELP transition */30
FIXP_DBL *pOut = pOut0 - fl / 2; /* fl/2 == fac_length */31
for (i = 0; i < fl / 2; i++) {32
- pOut[i] += IMDCT_SCALE_DBL(hMdct->pFacZir[i]);33
+ pOut[i] = fAddSaturate(pOut[i], IMDCT_SCALE_DBL(hMdct->pFacZir[i]));34
}35
hMdct->pFacZir = NULL;36
}37
38
/* Div2 is compensated by table scaling */39
x = fMultDiv2(pTmp2[i], FacWindowZir[w]);40
x += fMultDiv2(pTmp1[-i - 1], FacWindowSynth[w]);41
- x += pFAC_and_FAC_ZIR[i];42
- pOut1[i] = x;43
-44
+ pOut1[i] = fAddSaturate(x, pFAC_and_FAC_ZIR[i]);45
w++;46
}47
}48
49
FDK_ASSERT((pOut1 >= hMdct->overlap.time &&50
pOut1 < hMdct->overlap.time + hMdct->ov_size) ||51
(pOut1 >= output && pOut1 < output + 1024));52
- *pOut1 += IMDCT_SCALE_DBL(-x1);53
+ *pOut1 = fAddSaturate(*pOut1, IMDCT_SCALE_DBL(-x1));54
pOut1--;55
}56
57
58
FIXP_DBL x = -(*pCurr--);59
/* 5) (item 4) Synthesis filter Zir component, FAC ZIR (another one). */60
if (i < f_len) {61
- x += *pF++;62
+ x = fAddSaturate(x, *pF++);63
}64
65
FDK_ASSERT((pOut1 >= hMdct->overlap.time &&66
67
for (i = 0; i < fl / 2; i++) {68
FIXP_DBL x0, x1;69
70
- cplxMult(&x1, &x0, *pCurr++, -*pOvl--, pWindow_prev[i]);71
- *pOut0 = IMDCT_SCALE_DBL(x0);72
- *pOut1 = IMDCT_SCALE_DBL(-x1);73
+ cplxMultDiv2(&x1, &x0, *pCurr++, -*pOvl--, pWindow_prev[i]);74
+ *pOut0 = IMDCT_SCALE_DBL_LSH1(x0);75
+ *pOut1 = IMDCT_SCALE_DBL_LSH1(-x1);76
pOut0++;77
pOut1--;78
}79
80
for (i = 0; i < fl / 2; i++) {81
FIXP_DBL x0, x1;82
83
- cplxMult(&x1, &x0, *pCurr++, -*pOvl--, pWindow_prev[i]);84
- *pOut0 = IMDCT_SCALE_DBL(x0);85
- *pOut1 = IMDCT_SCALE_DBL(x1);86
+ cplxMultDiv2(&x1, &x0, *pCurr++, -*pOvl--, pWindow_prev[i]);87
+ *pOut0 = IMDCT_SCALE_DBL_LSH1(x0);88
+ *pOut1 = IMDCT_SCALE_DBL_LSH1(x1);89
pOut0++;90
pOut1--;91
}92
93
for (i = 0; i < fl / 2; i++) {94
FIXP_DBL x0, x1;95
96
- cplxMult(&x1, &x0, *pCurr++, *pOvl--, pWindow_prev[i]);97
- *pOut0 = IMDCT_SCALE_DBL(x0);98
- *pOut1 = IMDCT_SCALE_DBL(x1);99
+ cplxMultDiv2(&x1, &x0, *pCurr++, *pOvl--, pWindow_prev[i]);100
+ *pOut0 = IMDCT_SCALE_DBL_LSH1(x0);101
+ *pOut1 = IMDCT_SCALE_DBL_LSH1(x1);102
pOut0++;103
pOut1--;104
}105
106
FIXP_DBL *pOut = pOut0 - fl / 2;107
FDK_ASSERT(fl / 2 <= 128);108
for (i = 0; i < fl / 2; i++) {109
- pOut[i] += IMDCT_SCALE_DBL(hMdct->pFacZir[i]);110
+ pOut[i] = fAddSaturate(pOut[i], IMDCT_SCALE_DBL(hMdct->pFacZir[i]));111
}112
hMdct->pFacZir = NULL;113
}114
fdk-aac-2.0.1.tar.gz/libAACdec/src/usacdec_lpc.cpp -> fdk-aac-2.0.2.tar.gz/libAACdec/src/usacdec_lpc.cpp
Changed
98
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
void RE8_PPV(FIXP_ZF x[], SHORT y[], int r) {12
int i, y0[8], y1[8];13
FIXP_ZF x1[8], tmp;14
- FIXP_DBL e;15
+ INT64 e;16
17
/* find the nearest neighbor y0 of x in 2D8 */18
nearest_neighbor_2D8(x, y0);19
20
}21
22
/* compute e0=||x-y0||^2 and e1=||x-y1||^2 */23
- e = (FIXP_DBL)0;24
+ e = 0;25
for (i = 0; i < 8; i++) {26
tmp = x[i] - INT2ZF(y0[i], 0);27
- e += fPow2Div2(28
+ e += (INT64)fPow2Div2(29
tmp << r); /* shift left to ensure that no fract part bits get lost. */30
tmp = x[i] - INT2ZF(y1[i], 0);31
- e -= fPow2Div2(tmp << r);32
+ e -= (INT64)fPow2Div2(tmp << r);33
}34
/* select best candidate y0 or y1 to minimize distortion */35
- if (e < (FIXP_DBL)0) {36
+ if (e < 0) {37
for (i = 0; i < 8; i++) {38
y[i] = y0[i];39
}40
41
/* add non-weighted residual LSF vector to LSF1st */42
for (i = 0; i < M_LP_FILTER_ORDER; i++) {43
w = (LONG)fMultDiv2(factor, sqrtFixp(fMult(d[i], d[i + 1])));44
- lsfq[i] = fAddSaturate(lsfq[i], FX_DBL2FX_LPC((FIXP_DBL)(w * (LONG)xq[i])));45
+ lsfq[i] = fAddSaturate(lsfq[i],46
+ FX_DBL2FX_LPC((FIXP_DBL)((INT64)w * (LONG)xq[i])));47
}48
49
return;50
51
for (i = 2; i <= n; i++) {52
plsp += 2;53
b = -FX_LPC2FX_DBL(*plsp);54
- f[i] = ((fMultDiv2(b, f[i - 1]) << 1) + (f[i - 2])) << 1;55
+ f[i] = SATURATE_LEFT_SHIFT((fMultDiv2(b, f[i - 1]) + (f[i - 2] >> 1)), 2,56
+ DFRACT_BITS);57
for (j = i - 1; j > 1; j--) {58
- f[j] = f[j] + (fMultDiv2(b, f[j - 1]) << 2) + f[j - 2];59
+ f[j] = SATURATE_LEFT_SHIFT(60
+ ((f[j] >> 2) + fMultDiv2(b, f[j - 1]) + (f[j - 2] >> 2)), 2,61
+ DFRACT_BITS);62
}63
f[1] = f[1] + (b >> (SF_F - 1));64
}65
66
/*-----------------------------------------------------*67
* Multiply F1(z) by (1+z^-1) and F2(z) by (1-z^-1) *68
*-----------------------------------------------------*/69
+ scaleValues(f1, NC + 1, -2);70
+ scaleValues(f2, NC + 1, -2);71
+72
for (i = NC; i > 0; i--) {73
f1[i] += f1[i - 1];74
f2[i] -= f2[i - 1];75
76
FIXP_DBL aDBL[M_LP_FILTER_ORDER];77
78
for (i = 1, k = M_LP_FILTER_ORDER - 1; i <= NC; i++, k--) {79
- FIXP_DBL tmp1, tmp2;80
-81
- tmp1 = f1[i] >> 1;82
- tmp2 = f2[i] >> 1;83
-84
- aDBL[i - 1] = (tmp1 + tmp2);85
- aDBL[k] = (tmp1 - tmp2);86
+ aDBL[i - 1] = f1[i] + f2[i];87
+ aDBL[k] = f1[i] - f2[i];88
}89
90
int headroom_a = getScalefactor(aDBL, M_LP_FILTER_ORDER);91
92
a[i] = FX_DBL2FX_LPC(aDBL[i] << headroom_a);93
}94
95
- *a_exp = 8 - headroom_a;96
+ *a_exp = SF_F + (2 - 1) - headroom_a;97
}98
fdk-aac-2.0.1.tar.gz/libAACdec/src/usacdec_lpd.cpp -> fdk-aac-2.0.2.tar.gz/libAACdec/src/usacdec_lpd.cpp
Changed
123
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
12
#include "ac_arith_coder.h"13
14
-void filtLP(const FIXP_DBL *syn, FIXP_PCM *syn_out, FIXP_DBL *noise,15
- const FIXP_SGL *filt, INT stop, int len) {16
+void filtLP(const FIXP_DBL *syn, PCM_DEC *syn_out, FIXP_DBL *noise,17
+ const FIXP_SGL *filt, const INT aacOutDataHeadroom, INT stop,18
+ int len) {19
INT i, j;20
FIXP_DBL tmp;21
22
+ FDK_ASSERT((aacOutDataHeadroom - 1) >= -(MDCT_OUTPUT_SCALE));23
+24
for (i = 0; i < stop; i++) {25
tmp = fMultDiv2(noise[i], filt[0]); // Filt in Q-1.1626
for (j = 1; j <= len; j++) {27
- tmp += fMultDiv2((noise[i - j] + noise[i + j]), filt[j]);28
+ tmp += fMult((noise[i - j] >> 1) + (noise[i + j] >> 1), filt[j]);29
}30
- syn_out[i] = (FIXP_PCM)(IMDCT_SCALE(syn[i] - tmp));31
+ syn_out[i] = (PCM_DEC)(32
+ IMDCT_SCALE((syn[i] >> 1) - (tmp >> 1), aacOutDataHeadroom - 1));33
}34
}35
36
37
FIXP_DBL *pit_gain,38
const int frame_length, /* (i) : frame length (should be 768|1024) */39
const INT l_frame,40
- const INT l_next, /* (i) : look ahead for symmetric filtering */41
- FIXP_PCM *synth_out, /* (o) : filtered synthesis (with delay of 1 subfr) */42
+ const INT l_next, /* (i) : look ahead for symmetric filtering */43
+ PCM_DEC *synth_out, /* (o) : filtered synthesis (with delay of 1 subfr) */44
+ const INT aacOutDataHeadroom, /* (i) : headroom of the output time signal to45
+ prevent clipping */46
FIXP_DBL mem_bpf[]) /* i/o : memory state [L_FILT+L_SUBFR] */47
{48
INT i, sf, i_subfr, T, T2, lg;49
50
51
{52
for (i = 0; i < lg; i++) {53
- /* scaled with SF_SYNTH + gain_sf + 1 */54
+ /* scaled with SF_SYNTH + gain_sf + 1; composition of scalefactor 2:55
+ * one additional shift of syn values + fMult => fMultDiv2 */56
noise_in[i] =57
- (fMult(gainSGL, syn[i + i_subfr] - (syn[i + i_subfr - T] >> 1) -58
- (syn[i + i_subfr + T] >> 1))) >>59
- s1;60
+ scaleValue(fMultDiv2(gainSGL, (syn[i + i_subfr] >> 1) -61
+ (syn[i + i_subfr - T] >> 2) -62
+ (syn[i + i_subfr + T] >> 2)),63
+ 2 - s1);64
}65
66
for (i = lg; i < L_SUBFR; i++) {67
- /* scaled with SF_SYNTH + gain_sf + 1 */68
+ /* scaled with SF_SYNTH + gain_sf + 1; composition of scalefactor 2:69
+ * one additional shift of syn values + fMult => fMultDiv2 */70
noise_in[i] =71
- (fMult(gainSGL, syn[i + i_subfr] - syn[i + i_subfr - T])) >> s1;72
+ scaleValue(fMultDiv2(gainSGL, (syn[i + i_subfr] >> 1) -73
+ (syn[i + i_subfr - T] >> 1)),74
+ 2 - s1);75
}76
}77
} else {78
79
80
{81
filtLP(&syn[i_subfr - L_SUBFR], &synth_out[i_subfr], noise,82
- fdk_dec_filt_lp, L_SUBFR, L_FILT);83
+ fdk_dec_filt_lp, aacOutDataHeadroom, L_SUBFR, L_FILT);84
}85
}86
87
88
/* Output scaling of the BPF memory */89
scaleValues(mem_bpf, (L_FILT + L_SUBFR), -1);90
/* Copy the rest of the signal (after the fac) */91
- scaleValuesSaturate((FIXP_PCM *)&synth_out[l_frame],92
- (FIXP_DBL *)&syn[l_frame - L_SUBFR],93
- (frame_length - l_frame), MDCT_OUT_HEADROOM);94
+ scaleValuesSaturate(95
+ (PCM_DEC *)&synth_out[l_frame], (FIXP_DBL *)&syn[l_frame - L_SUBFR],96
+ (frame_length - l_frame), MDCT_OUT_HEADROOM - aacOutDataHeadroom);97
}98
99
return;100
101
102
AAC_DECODER_ERROR CLpd_RenderTimeSignal(103
CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo,104
- CAacDecoderChannelInfo *pAacDecoderChannelInfo, FIXP_PCM *pTimeData,105
- INT lFrame, SamplingRateInfo *pSamplingRateInfo, UINT frameOk, UINT flags,106
- UINT strmFlags) {107
+ CAacDecoderChannelInfo *pAacDecoderChannelInfo, PCM_DEC *pTimeData,108
+ INT lFrame, SamplingRateInfo *pSamplingRateInfo, UINT frameOk,109
+ const INT aacOutDataHeadroom, UINT flags, UINT strmFlags) {110
UCHAR *mod = pAacDecoderChannelInfo->data.usac.mod;111
AAC_DECODER_ERROR error = AAC_DEC_OK;112
int k, i_offset;113
114
{115
bass_pf_1sf_delay(p2_synth, pitch, pit_gain, lFrame, lFrame / facFB,116
mod[nbDiv - 1] ? (SynDelay - (lDiv / 2)) : SynDelay,117
- pTimeData, pAacDecoderStaticChannelInfo->mem_bpf);118
+ pTimeData, aacOutDataHeadroom,119
+ pAacDecoderStaticChannelInfo->mem_bpf);120
}121
}122
123
fdk-aac-2.0.1.tar.gz/libAACdec/src/usacdec_lpd.h -> fdk-aac-2.0.2.tar.gz/libAACdec/src/usacdec_lpd.h
Changed
57
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
* \param pTimeData pointer to output buffer12
* \param samplesPerFrame amount of output samples13
* \param pSamplingRateInfo holds the sampling rate information14
- * \param pWorkBuffer1 pointer to work buffer for temporal data15
+ * \param aacOutDataHeadroom headroom of the output time signal to prevent16
+ * clipping17
*/18
AAC_DECODER_ERROR CLpd_RenderTimeSignal(19
CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo,20
- CAacDecoderChannelInfo *pAacDecoderChannelInfo, FIXP_PCM *pTimeData,21
+ CAacDecoderChannelInfo *pAacDecoderChannelInfo, PCM_DEC *pTimeData,22
INT samplesPerFrame, SamplingRateInfo *pSamplingRateInfo, UINT frameOk,23
- UINT flags, UINT strmFlags);24
+ const INT aacOutDataHeadroom, UINT flags, UINT strmFlags);25
26
static inline INT CLpd_FAC_getLength(int fNotShortBlock, int fac_length_long) {27
if (fNotShortBlock) {28
29
}30
}31
32
-void filtLP(const FIXP_DBL *syn, FIXP_PCM *syn_out, FIXP_DBL *noise,33
- const FIXP_SGL *filt, INT stop, int len);34
+void filtLP(const FIXP_DBL *syn, PCM_DEC *syn_out, FIXP_DBL *noise,35
+ const FIXP_SGL *filt, const INT aacOutDataHeadroom, INT stop,36
+ int len);37
38
/**39
* \brief perform a low-frequency pitch enhancement on time domain signal40
41
* \param[in] l_frame length of filtering, must be multiple of L_SUBFR42
* \param[in] l_next length of allowed look ahead on syn[i], i < l_frame+l_next43
* \param[out] synth_out pointer to time domain output signal44
+ * \param[in] headroom of the output time signal to prevent clipping45
* \param[in,out] mem_bpf pointer to filter memory (L_FILT+L_SUBFR)46
*/47
48
void bass_pf_1sf_delay(FIXP_DBL syn[], const INT T_sf[], FIXP_DBL *pit_gain,49
const int frame_length, const INT l_frame,50
- const INT l_next, FIXP_PCM *synth_out,51
- FIXP_DBL mem_bpf[]);52
+ const INT l_next, PCM_DEC *synth_out,53
+ const INT aacOutDataHeadroom, FIXP_DBL mem_bpf[]);54
55
/**56
* \brief random sign generator for FD and TCX noise filling57
fdk-aac-2.0.1.tar.gz/libAACenc/include/aacenc_lib.h -> fdk-aac-2.0.2.tar.gz/libAACenc/include/aacenc_lib.h
Changed
168
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
All header files are provided for usage in C/C++ programs. The AAC encoder12
library API functions are located in aacenc_lib.h.13
14
-In binary releases the encoder core resides in statically linkable libraries15
-called for example libAACenc.a/libFDK.a (LINUX) or FDK_fastaaclib.lib (MS Visual16
-C++) for the plain AAC-LC core encoder and libSBRenc.a (LINUX) or17
-FDK_sbrEncLib.lib (MS Visual C++) for the SBR (Spectral Band Replication) and PS18
-(Parametric Stereo) modules.19
-20
\section CallingSequence Calling Sequence21
22
For encoding of ISO/MPEG-2/4 AAC bitstreams the following sequence is mandatory.23
24
\endcode25
26
\section writeOutData Output Bitstream Data27
-If any AAC bitstream data is available, write it to output file or device. This28
-can be done once the following condition is true: \code if29
-(outargs.numOutBytes>0) {30
-31
+If any AAC bitstream data is available, write it to output file or device as32
+follows. \code if (outargs.numOutBytes>0) { FDKfwrite(outputBuffer,33
+outargs.numOutBytes, 1, pOutFile);34
}35
\endcode36
37
-If you use file I/O then for example call mpegFileWrite_Write() from the library38
-libMpegFileWrite \code mpegFileWrite_Write(hMpegFile, outputBuffer,39
-outargs.numOutBytes, aacEncoder_GetParam(hAacEncoder, AACENC_GRANULE_LENGTH));40
-\endcode41
-42
\section cfgMetaData Meta Data Configuration43
44
If the present library is configured with Metadata support, it is possible to45
insert meta data side info into the generated audio bitstream while encoding.46
47
To work with meta data the encoder instance has to be \ref encOpen "allocated"48
-with meta data support. The meta data mode must be be configured with the49
+with meta data support. The meta data mode must be configured with the50
::AACENC_METADATA_MODE parameter and aacEncoder_SetParam() function. \code51
aacEncoder_SetParam(hAacEncoder, AACENC_METADATA_MODE, 0-3); \endcode52
53
54
return chMode;55
\endcode56
57
-\subsection bitreservoir Bitreservoir Configuration58
+\subsection peakbitrate Peak Bitrate Configuration59
In AAC, the default bitreservoir configuration depends on the chosen bitrate per60
frame and the number of effective channels. The size can be determined as below.61
\f[62
63
Due to audio quality concerns it is not recommended to change the bitreservoir64
size to a lower value than the default setting! However, for minimizing the65
delay for streaming applications or for achieving a constant size of the66
-bitstream packages in each frame, it may be necessaray to change the67
-bitreservoir size. This can be done with the ::AACENC_PEAK_BITRATE parameter.68
-\code69
+bitstream packages in each frame, it may be necessaray to limit the maximum bits70
+per frame size. This can be done with the ::AACENC_PEAK_BITRATE parameter. \code71
aacEncoder_SetParam(hAacEncoder, AACENC_PEAK_BITRATE, value);72
\endcode73
-By setting ::AACENC_BITRATEMODE to fixed framing, the bitreservoir is disabled.74
-A disabled bitreservoir results in a constant size for each bitstream package.75
-Please note that especially at lower bitrates a disabled bitreservoir can76
-downgrade the audio quality considerably! The default bitreservoir configuration77
-can be achieved as follows. \code aacEncoder_SetParam(hAacEncoder,78
-AACENC_BITRESERVOIR, -1); \endcode79
80
To achieve acceptable audio quality with a reduced bitreservoir size setting at81
least 1000 bits per audio channel is recommended. For a multichannel audio file82
83
84
85
\subsection vbrmode Variable Bitrate Mode86
-The encoder provides various Variable Bitrate Modes that differ in audio quality87
-and average overall bitrate. The given values are averages over time, different88
-encoder settings and strongly depend on the type of audio signal. The VBR89
-configurations can be adjusted via ::AACENC_BITRATEMODE encoder parameter.90
+The variable bitrate (VBR) mode coding adapts the bit consumption to the91
+psychoacoustic requirements of the signal. The encoder ignores the user-defined92
+bit rate and selects a suitable pre-defined configuration based on the provided93
+AOT. The VBR mode 1 is tuned for HE-AACv2, for VBR mode 2, HE-AACv1 should be94
+used. VBR modes 3-5 should be used with Low-Complexity AAC. When encoding95
+AAC-ELD, the best mode is selected automatically.96
+97
+The bitrates given in the table are averages over time and different encoder98
+settings. They strongly depend on the type of audio signal. The VBR99
+configurations can be adjusted with the ::AACENC_BITRATEMODE encoder parameter.100
\verbatim101
---------------------------------------------102
- VBR_MODE | Approx. Bitrate in kbps/channel103
- | AAC-LC | AAC-LD/AC_ELD104
-----------+---------------+-----------------105
- VBR_1 | 32 - 48 | 32 - 56106
- VBR_2 | 40 - 56 | 40 - 64107
- VBR_3 | 48 - 64 | 48 - 72108
- VBR_4 | 64 - 80 | 64 - 88109
- VBR_5 | 96 - 120 | 112 - 144110
+-----------------------------------------------111
+ VBR_MODE | Approx. Bitrate in kbps for stereo112
+ | AAC-LC | AAC-ELD113
+----------+---------------+--------------------114
+ VBR_1 | 32 (HE-AACv2) | 48115
+ VBR_2 | 72 (HE-AACv1) | 56116
+ VBR_3 | 112 | 72117
+ VBR_4 | 148 | 148118
+ VBR_5 | 228 | 224119
--------------------------------------------120
\endverbatim121
-The bitrate ranges apply for individual audio channels. In case of multichannel122
-configurations the average bitrate might be estimated by multiplying with the123
-number of effective channels. This corresponds to all audio input channels124
-exclusively the low frequency channel. At configurations which are making use of125
-downmix modules the AAC core channels respectively downmix channels shall be126
-considered. For ::AACENC_AOT which are using SBR, the average bitrate can be127
-estimated by using the ratio of 0.5 for dualrate SBR and 0.75 for downsampled128
-SBR configurations.129
-130
+Note that these figures are valid for stereo encoding only. VBR modes 2-5 will131
+yield much lower bit rates when encoding single-channel input. For132
+configurations which are making use of downmix modules the AAC core channels133
+respectively downmix channels shall be considered.134
135
\subsection encQual Audio Quality Considerations136
The default encoder configuration is suggested to be used. Encoder tools such as137
138
139
\subsection BEHAVIOUR_ESTIM_AVG_FRAMESIZES Estimating Average Frame Sizes140
141
-A HE-AAC v1 or v2 audio frame contains 2048 PCM samples per channel (there is142
-also one mode with 1920 samples per channel but this is only for special143
-purposes such as DAB+ digital radio).144
+A HE-AAC v1 or v2 audio frame contains 2048 PCM samples per channel.145
146
The number of HE-AAC frames \f$N\_FRAMES\f$ per second at 44.1 kHz is:147
148
149
150
#define AACENCODER_LIB_VL0 4151
#define AACENCODER_LIB_VL1 0152
-#define AACENCODER_LIB_VL2 0153
+#define AACENCODER_LIB_VL2 1154
155
/**156
* AAC encoder error codes.157
158
typedef struct {159
UINT maxOutBufBytes; /*!< Maximum number of encoder bitstream bytes within one160
frame. Size depends on maximum number of supported161
- channels in encoder instance. For superframing (as162
- used for example in DAB+), size has to be a multiple163
- accordingly. */164
+ channels in encoder instance. */165
166
UINT maxAncBytes; /*!< Maximum number of ancillary data bytes which can be167
inserted into bitstream within one frame. */168
fdk-aac-2.0.1.tar.gz/libAACenc/src/aacEnc_ram.cpp -> fdk-aac-2.0.2.tar.gz/libAACenc/src/aacEnc_ram.cpp
Changed
17
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
* (dynamic_RAM + P_BUF_0 + n*sizeof(QC_OUT_CHANNEL)) is sufficiently aligned,12
* so the cast is safe */13
return reinterpret_cast<QC_OUT_CHANNEL *>(reinterpret_cast<void *>(14
- dynamic_RAM + P_BUF_0 + n * sizeof(QC_OUT_CHANNEL)));15
+ dynamic_RAM + P_BUF_0 + n * ALIGN_SIZE(sizeof(QC_OUT_CHANNEL))));16
}17
fdk-aac-2.0.1.tar.gz/libAACenc/src/aacEnc_ram.h -> fdk-aac-2.0.2.tar.gz/libAACenc/src/aacEnc_ram.h
Changed
19
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
+++++++++++++++++++++++++++++++++++++++++++++++++++++12
*/13
14
-#define BUF_SIZE_0 (ALIGN_SIZE(sizeof(QC_OUT_CHANNEL) * (8)))15
+#define BUF_SIZE_0 (ALIGN_SIZE(sizeof(QC_OUT_CHANNEL)) * (8))16
#define BUF_SIZE_1 \17
(ALIGN_SIZE(maxSize(maxSize(sizeof(PSY_DYNAMIC), \18
(BIT_LOOK_UP_SIZE + MERGE_GAIN_LOOK_UP_SIZE)), \19
fdk-aac-2.0.1.tar.gz/libAACenc/src/aacenc.cpp -> fdk-aac-2.0.2.tar.gz/libAACenc/src/aacenc.cpp
Changed
104
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
12
#include "genericStds.h"13
14
+#define BITRES_MIN \15
+ 300 /* default threshold for using reduced/disabled bitres mode */16
#define BITRES_MAX_LD 400017
#define BITRES_MIN_LD 50018
#define BITRATE_MAX_LD 70000 /* Max assumed bitrate for bitres calculation */19
20
return bitrate;21
}22
23
+/*-----------------------------------------------------------------------------24
+25
+ functionname: FDKaacEnc_AdjustVBRBitrateMode26
+ description: Adjust bitrate mode to given bitrate parameter27
+ input params: int vbrQuality (VBR0, VBR1, VBR2)28
+ bitrate29
+ channelMode30
+ returns: vbr bitrate mode31
+32
+ ------------------------------------------------------------------------------*/33
+AACENC_BITRATE_MODE FDKaacEnc_AdjustVBRBitrateMode(34
+ AACENC_BITRATE_MODE bitrateMode, INT bitrate, CHANNEL_MODE channelMode) {35
+ AACENC_BITRATE_MODE newBitrateMode = bitrateMode;36
+37
+ if (bitrate != -1) {38
+ const INT monoStereoMode =39
+ (FDKaacEnc_GetMonoStereoMode(channelMode) == EL_MODE_STEREO) ? 1 : 0;40
+ const INT nChannelsEff =41
+ FDKaacEnc_GetChannelModeConfiguration(channelMode)->nChannelsEff;42
+ newBitrateMode = AACENC_BR_MODE_INVALID;43
+44
+ for (int idx = (int)(sizeof(configTabVBR) / sizeof(*configTabVBR)) - 1;45
+ idx >= 0; idx--) {46
+ if (bitrate >=47
+ configTabVBR[idx].chanBitrate[monoStereoMode] * nChannelsEff) {48
+ if (configTabVBR[idx].chanBitrate[monoStereoMode] * nChannelsEff <49
+ FDKaacEnc_GetVBRBitrate(bitrateMode, channelMode)) {50
+ newBitrateMode = configTabVBR[idx].bitrateMode;51
+ } else {52
+ newBitrateMode = bitrateMode;53
+ }54
+ break;55
+ }56
+ }57
+ }58
+59
+ return AACENC_BR_MODE_IS_VBR(newBitrateMode) ? newBitrateMode60
+ : AACENC_BR_MODE_INVALID;61
+}62
+63
/**64
* \brief Convert encoder bitreservoir value for transport library.65
*66
67
FIXP_DBL mbfac, bw_ratio;68
QC_INIT qcInit;69
INT averageBitsPerFrame = 0;70
- int bitresMin = 0; /* the bitreservoir is always big for AAC-LC */71
const CHANNEL_MODE prevChannelMode = hAacEnc->encoderMode;72
73
if (config == NULL) return AAC_ENC_INVALID_HANDLE;74
75
(config->minBitsPerFrame != -1) ? config->minBitsPerFrame : 0;76
qcInit.minBits = fixMin(qcInit.minBits, averageBitsPerFrame & ~7);77
} else {78
- INT bitreservoir = -1; /* default bitrservoir size*/79
+ INT bitreservoir = -1; /* default bitreservoir size*/80
if (isLowDelay(config->audioObjectType)) {81
INT brPerChannel = config->bitRate / config->nChannels;82
brPerChannel = fMin(BITRATE_MAX_LD, fMax(BITRATE_MIN_LD, brPerChannel));83
84
bitreservoir = fMultI(slope, (INT)(BITRES_MAX_LD - BITRES_MIN_LD)) +85
BITRES_MIN_LD; /* interpolate */86
bitreservoir = bitreservoir & ~7; /* align to bytes */87
- bitresMin = BITRES_MIN_LD;88
}89
90
int maxBitres;91
92
qcInit.nSubFrames = config->nSubFrames;93
qcInit.padding.paddingRest = config->sampleRate;94
95
- if (qcInit.bitRes >= bitresMin * config->nChannels) {96
+ if (qcInit.maxBits - qcInit.averageBits >=97
+ ((qcInit.isLowDelay) ? BITRES_MIN_LD : BITRES_MIN) * config->nChannels) {98
qcInit.bitResMode = AACENC_BR_MODE_FULL; /* full bitreservoir */99
- } else if (qcInit.bitRes > 0) {100
+ } else if (qcInit.maxBits > qcInit.averageBits) {101
qcInit.bitResMode = AACENC_BR_MODE_REDUCED; /* reduced bitreservoir */102
} else {103
qcInit.bitResMode = AACENC_BR_MODE_DISABLED; /* disabled bitreservoir */104
fdk-aac-2.0.1.tar.gz/libAACenc/src/aacenc.h -> fdk-aac-2.0.2.tar.gz/libAACenc/src/aacenc.h
Changed
30
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
12
/*-----------------------------------------------------------------------------13
14
+ functionname: FDKaacEnc_AdjustVBRBitrateMode15
+ description: Adjust bitrate mode to given bitrate parameter16
+ input params: int vbrQuality (VBR0, VBR1, VBR2)17
+ bitrate18
+ channelMode19
+ returns: vbr bitrate mode20
+21
+ ------------------------------------------------------------------------------*/22
+AACENC_BITRATE_MODE FDKaacEnc_AdjustVBRBitrateMode(23
+ AACENC_BITRATE_MODE bitrateMode, INT bitrate, CHANNEL_MODE channelMode);24
+25
+/*-----------------------------------------------------------------------------26
+27
functionname: FDKaacEnc_AacInitDefaultConfig28
description: gives reasonable default configuration29
returns: ---30
fdk-aac-2.0.1.tar.gz/libAACenc/src/aacenc_lib.cpp -> fdk-aac-2.0.2.tar.gz/libAACenc/src/aacenc_lib.cpp
Changed
89
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
/* Encoder library info */12
#define AACENCODER_LIB_VL0 413
#define AACENCODER_LIB_VL1 014
-#define AACENCODER_LIB_VL2 015
+#define AACENCODER_LIB_VL2 116
#define AACENCODER_LIB_TITLE "AAC Encoder"17
-#ifdef __ANDROID__18
+#ifdef SUPPRESS_BUILD_DATE_INFO19
#define AACENCODER_LIB_BUILD_DATE ""20
#define AACENCODER_LIB_BUILD_TIME ""21
#else22
23
return sbrSignaling;24
}25
26
+static inline INT getAssociatedChElement(SBR_ELEMENT_INFO *elInfoSbr,27
+ CHANNEL_MAPPING *channelMapping) {28
+ ELEMENT_INFO *elInfo = channelMapping->elInfo;29
+ INT nElements = channelMapping->nElements;30
+ INT associatedChElement = -1;31
+ int i;32
+33
+ for (i = 0; i < nElements; i++) {34
+ if (elInfoSbr->elType == elInfo[i].elType &&35
+ elInfoSbr->instanceTag == elInfo[i].instanceTag) {36
+ associatedChElement = i;37
+ break;38
+ }39
+ }40
+41
+ return associatedChElement;42
+}43
+44
/****************************************************************************45
Allocate Encoder46
****************************************************************************/47
48
case AACENC_BR_MODE_VBR_3:49
case AACENC_BR_MODE_VBR_4:50
case AACENC_BR_MODE_VBR_5:51
+ /* Adjust bitrate mode in case given peak bitrate is lower than expected52
+ * VBR bitrate. */53
+ if ((INT)config->userPeakBitrate != -1) {54
+ hAacConfig->bitrateMode = FDKaacEnc_AdjustVBRBitrateMode(55
+ hAacConfig->bitrateMode, config->userPeakBitrate,56
+ hAacConfig->channelMode);57
+ }58
/* Get bitrate in VBR configuration */59
/* In VBR mode; SBR-modul depends on bitrate, core encoder on bitrateMode.60
*/61
62
hAacEncoder->inputBuffer +63
(hAacEncoder->inputBufferOffset + hAacEncoder->nSamplesRead) /64
hAacEncoder->aacConfig.nChannels;65
+ newSamples -=66
+ (newSamples %67
+ hAacEncoder->extParam68
+ .nChannels); /* process multiple samples of input channels */69
70
/* Copy new input samples to internal buffer */71
if (inBufDesc->bufElSizes[idx] == (INT)sizeof(INT_PCM)) {72
73
{74
hAacEncoder->extPayload[nExtensions].dataSize =75
hAacEncoder->pSbrPayload->dataSize[nPayload][i];76
- hAacEncoder->extPayload[nExtensions].associatedChElement = i;77
+ hAacEncoder->extPayload[nExtensions].associatedChElement =78
+ getAssociatedChElement(79
+ &hAacEncoder->hEnvEnc->sbrElement[i]->elInfo,80
+ &hAacEncoder->hAacEnc->channelMapping);81
+ if (hAacEncoder->extPayload[nExtensions].associatedChElement ==82
+ -1) {83
+ err = AACENC_ENCODE_ERROR;84
+ goto bail;85
+ }86
}87
hAacEncoder->extPayload[nExtensions].dataType =88
EXT_SBR_DATA; /* Once SBR Encoder supports SBR CRC set89
fdk-aac-2.0.1.tar.gz/libAACenc/src/adj_thr.cpp -> fdk-aac-2.0.2.tar.gz/libAACenc/src/adj_thr.cpp
Changed
43
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
if (sfbEn > avgEn) {12
FIXP_DBL tmpMinSnrLdData;13
if (psyOutChannel[ch]->lastWindowSequence == LONG_WINDOW)14
- tmpMinSnrLdData =15
- fixMax(SnrLdFac + (FIXP_DBL)(avgEnLdData - sfbEnLdData),16
- (FIXP_DBL)SnrLdMin1);17
+ tmpMinSnrLdData = SnrLdFac + fixMax(avgEnLdData - sfbEnLdData,18
+ SnrLdMin1 - SnrLdFac);19
else20
- tmpMinSnrLdData =21
- fixMax(SnrLdFac + (FIXP_DBL)(avgEnLdData - sfbEnLdData),22
- (FIXP_DBL)SnrLdMin3);23
+ tmpMinSnrLdData = SnrLdFac + fixMax(avgEnLdData - sfbEnLdData,24
+ SnrLdMin3 - SnrLdFac);25
26
qcOutChan->sfbMinSnrLdData[sfbGrp + sfb] = fixMin(27
qcOutChan->sfbMinSnrLdData[sfbGrp + sfb], tmpMinSnrLdData);28
29
if (sfbThrReducedLdData < FL2FXCONST_DBL(-0.5f))30
sfbThrReducedLdData = FL2FXCONST_DBL(-1.f);31
32
- /* minimum of 29 dB Ratio for Thresholds */33
- if ((sfbEnLdData + FL2FXCONST_DBL(1.0f)) >34
- FL2FXCONST_DBL(9.6336206 / LD_DATA_SCALING)) {35
- sfbThrReducedLdData = fixMax(36
- sfbThrReducedLdData,37
- sfbEnLdData - FL2FXCONST_DBL(9.6336206 / LD_DATA_SCALING));38
- }39
-40
sfbThrReducedLdData = fixMax(MIN_LDTHRESH, sfbThrReducedLdData);41
42
qcOutChan->sfbThresholdLdData[sfbGrp + sfb] = sfbThrReducedLdData;43
fdk-aac-2.0.1.tar.gz/libAACenc/src/bandwidth.cpp -> fdk-aac-2.0.2.tar.gz/libAACenc/src/bandwidth.cpp
Changed
27
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
12
static const BANDWIDTH_TAB_VBR bandWidthTableVBR[] = {13
{AACENC_BR_MODE_CBR, 0, 0},14
- {AACENC_BR_MODE_VBR_1, 13050, 13050},15
- {AACENC_BR_MODE_VBR_2, 13050, 13050},16
- {AACENC_BR_MODE_VBR_3, 14260, 14260},17
- {AACENC_BR_MODE_VBR_4, 15500, 15500},18
- {AACENC_BR_MODE_VBR_5, 48000, 48000},19
+ {AACENC_BR_MODE_VBR_1, 13000, 13000},20
+ {AACENC_BR_MODE_VBR_2, 13000, 13000},21
+ {AACENC_BR_MODE_VBR_3, 15750, 15750},22
+ {AACENC_BR_MODE_VBR_4, 16500, 16500},23
+ {AACENC_BR_MODE_VBR_5, 19293, 19293},24
{AACENC_BR_MODE_SFR, 0, 0},25
{AACENC_BR_MODE_FF, 0, 0}26
27
fdk-aac-2.0.1.tar.gz/libAACenc/src/intensity.cpp -> fdk-aac-2.0.2.tar.gz/libAACenc/src/intensity.cpp
Changed
63
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
for (sfboffs = 0; sfboffs < maxSfbPerGroup; sfboffs++) {12
INT sL, sR;13
FIXP_DBL inv_n;14
+ INT mdct_spec_sf = MDCT_SPEC_SF;15
16
msMask[sfb + sfboffs] = 0;17
if (isMask[sfb + sfboffs] == 0) {18
19
}20
}21
22
+ if (sfbOffset[sfb + sfboffs + 1] - sfbOffset[sfb + sfboffs] >23
+ 1 << mdct_spec_sf) {24
+ mdct_spec_sf++; /* This is for rare cases where the number of bins in a25
+ scale factor band is > 64 */26
+ }27
+28
inv_n = GetInvInt(29
(sfbOffset[sfb + sfboffs + 1] - sfbOffset[sfb + sfboffs]) >>30
1); // scaled with 2 to compensate fMultDiv2() in subsequent loop31
32
j++) {33
d = ((mdctSpectrumLeft[j] << s0) >> 1) -34
((mdctSpectrumRight[j] << s0) >> 1);35
- ed += fMultDiv2(d, d) >> (MDCT_SPEC_SF - 1);36
+ ed += fMultDiv2(d, d) >> (mdct_spec_sf - 1);37
}38
msMask[sfb + sfboffs] = 1;39
tmp = fDivNorm(sfbEnergyLeft[sfb + sfboffs], ed, &s1);40
- s2 = (s1) + (2 * s0) - 2 - MDCT_SPEC_SF;41
+ s2 = (s1) + (2 * s0) - 2 - mdct_spec_sf;42
if (s2 & 1) {43
tmp = tmp >> 1;44
s2 = s2 + 1;45
46
j++) {47
s = ((mdctSpectrumLeft[j] << s0) >> 1) +48
((mdctSpectrumRight[j] << s0) >> 1);49
- es = fAddSaturate(es, fMultDiv2(s, s) >>50
- (MDCT_SPEC_SF -51
- 1)); // scaled 2*(mdctScale - s0 + 1) + MDCT_SPEC_SF52
+ es += fMultDiv2(s, s) >>53
+ (mdct_spec_sf -54
+ 1); // scaled 2*(mdctScale - s0 + 1) + mdct_spec_sf55
}56
msMask[sfb + sfboffs] = 0;57
tmp = fDivNorm(sfbEnergyLeft[sfb + sfboffs], es, &s1);58
- s2 = (s1) + (2 * s0) - 2 - MDCT_SPEC_SF;59
+ s2 = (s1) + (2 * s0) - 2 - mdct_spec_sf;60
if (s2 & 1) {61
tmp = tmp >> 1;62
s2 = s2 + 1;63
fdk-aac-2.0.1.tar.gz/libAACenc/src/qc_main.cpp -> fdk-aac-2.0.2.tar.gz/libAACenc/src/qc_main.cpp
Changed
113
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
12
static const TAB_VBR_QUAL_FACTOR tableVbrQualFactor[] = {13
{QCDATA_BR_MODE_VBR_1,14
- FL2FXCONST_DBL(0.160f)}, /* Approx. 32 - 48 (AC-LC), 32 - 5615
- (AAC-LD/ELD) kbps/channel */16
+ FL2FXCONST_DBL(0.150f)}, /* Approx. 32 kbps mono AAC-LC + SBR + PS */17
{QCDATA_BR_MODE_VBR_2,18
- FL2FXCONST_DBL(0.148f)}, /* Approx. 40 - 56 (AC-LC), 40 - 6419
- (AAC-LD/ELD) kbps/channel */20
+ FL2FXCONST_DBL(0.162f)}, /* Approx. 64 kbps stereo AAC-LC + SBR */21
{QCDATA_BR_MODE_VBR_3,22
- FL2FXCONST_DBL(0.135f)}, /* Approx. 48 - 64 (AC-LC), 48 - 7223
- (AAC-LD/ELD) kbps/channel */24
+ FL2FXCONST_DBL(0.176f)}, /* Approx. 96 kbps stereo AAC-LC */25
{QCDATA_BR_MODE_VBR_4,26
- FL2FXCONST_DBL(0.111f)}, /* Approx. 64 - 80 (AC-LC), 64 - 8827
- (AAC-LD/ELD) kbps/channel */28
+ FL2FXCONST_DBL(0.120f)}, /* Approx. 128 kbps stereo AAC-LC */29
{QCDATA_BR_MODE_VBR_5,30
- FL2FXCONST_DBL(0.070f)} /* Approx. 96 - 120 (AC-LC), 112 - 14431
- (AAC-LD/ELD) kbps/channel */32
+ FL2FXCONST_DBL(0.070f)} /* Approx. 192 kbps stereo AAC-LC */33
};34
35
static INT isConstantBitrateMode(const QCDATA_BR_MODE bitrateMode) {36
37
hQC->invQuant = init->invQuant;38
hQC->maxIterations = init->maxIterations;39
40
- if (isConstantBitrateMode(hQC->bitrateMode)) {41
- /* 0: full bitreservoir, 1: reduced bitreservoir, 2: disabled bitreservoir42
- */43
- hQC->bitResMode = init->bitResMode;44
- } else {45
- hQC->bitResMode = AACENC_BR_MODE_FULL; /* full bitreservoir */46
- }47
+ /* 0: full bitreservoir, 1: reduced bitreservoir, 2: disabled bitreservoir */48
+ hQC->bitResMode = init->bitResMode;49
50
hQC->padding.paddingRest = init->padding.paddingRest;51
52
53
INT avgTotalDynBits = 0; /* maximal allowed dynamic bits for all frames */54
INT totalAvailableBits = 0;55
INT nSubFrames = 1;56
+ const INT isCBRAdjustment = (isConstantBitrateMode(hQC->bitrateMode) ||57
+ (hQC->bitResMode != AACENC_BR_MODE_FULL))58
+ ? 159
+ : 0;60
61
/*-------------------------------------------- */62
/* redistribute total bitreservoir to elements */63
- ErrorStatus = FDKaacEnc_BitResRedistribution(hQC, cm, avgTotalBits);64
+ ErrorStatus = FDKaacEnc_BitResRedistribution(65
+ hQC, cm, (isCBRAdjustment == 0) ? hQC->maxBitsPerFrame : avgTotalBits);66
if (ErrorStatus != AAC_ENC_OK) {67
return ErrorStatus;68
}69
70
71
/*-------------------------------------------- */72
/*-------------------------------------------- */73
- if (isConstantBitrateMode(hQC->bitrateMode)) {74
- /* calc granted dynamic bits for sub frame and75
- distribute it to each element */76
- ErrorStatus = FDKaacEnc_prepareBitDistribution(77
- hQC, psyOut, qcOut, cm, qcElement, avgTotalBits, &totalAvailableBits,78
- &avgTotalDynBits);79
-80
- if (ErrorStatus != AAC_ENC_OK) {81
- return ErrorStatus;82
- }83
- } else {84
- qcOut[0]->grantedDynBits =85
- ((hQC->maxBitsPerFrame - (hQC->globHdrBits)) & ~7) -86
- (qcOut[0]->globalExtBits + qcOut[0]->staticBits +87
- qcOut[0]->elementExtBits);88
- qcOut[0]->maxDynBits = qcOut[0]->grantedDynBits;89
-90
- totalAvailableBits = hQC->maxBitsPerFrame;91
- avgTotalDynBits = 0;92
+ /* calc granted dynamic bits for sub frame and93
+ distribute it to each element */94
+ ErrorStatus = FDKaacEnc_prepareBitDistribution(95
+ hQC, psyOut, qcOut, cm, qcElement,96
+ (isCBRAdjustment == 0) ? hQC->maxBitsPerFrame : avgTotalBits,97
+ &totalAvailableBits, &avgTotalDynBits);98
+99
+ if (ErrorStatus != AAC_ENC_OK) {100
+ return ErrorStatus;101
}102
103
/* for ( all sub frames ) ... */104
for (c = 0; c < nSubFrames; c++) {105
/* for CBR and VBR mode */106
FDKaacEnc_AdjustThresholds(hQC->hAdjThr, qcElement[c], qcOut[c],107
- psyOut[c]->psyOutElement,108
- isConstantBitrateMode(hQC->bitrateMode), cm);109
+ psyOut[c]->psyOutElement, isCBRAdjustment, cm);110
111
} /* -end- sub frame counter */112
113
fdk-aac-2.0.1.tar.gz/libDRCdec/include/FDK_drcDecLib.h -> fdk-aac-2.0.2.tar.gz/libDRCdec/include/FDK_drcDecLib.h
Changed
51
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
extern "C" {12
#endif13
14
+#define DRC_DEC_LOUDNESS_NOT_PRESENT (LONG)0x7FFFFFFE15
+16
typedef struct s_drc_decoder* HANDLE_DRC_DECODER;17
typedef struct s_uni_drc_interface* HANDLE_UNI_DRC_INTERFACE;18
typedef struct s_selection_process_output* HANDLE_SEL_PROC_OUTPUT;19
20
DRC_DEC_EFFECT_TYPE,21
DRC_DEC_EFFECT_TYPE_FALLBACK_CODE,22
DRC_DEC_LOUDNESS_MEASUREMENT_METHOD,23
+ DRC_DEC_ALBUM_MODE,24
/* set only system (not user) parameters */25
DRC_DEC_DOWNMIX_ID,26
DRC_DEC_TARGET_CHANNEL_COUNT_REQUESTED, /**< number of output channels27
28
choosing an appropriate29
downmixInstruction */30
DRC_DEC_BASE_CHANNEL_COUNT,31
+ DRC_DEC_FRAME_SIZE,32
+ DRC_DEC_SAMPLE_RATE,33
/* get only system parameters */34
DRC_DEC_IS_MULTIBAND_DRC_1,35
DRC_DEC_IS_MULTIBAND_DRC_2,36
DRC_DEC_IS_ACTIVE, /**< MPEG-D DRC payload is present and at least one of37
Dynamic Range Control (DRC) or Loudness Normalization38
(LN) is activated */39
- DRC_DEC_TARGET_CHANNEL_COUNT_SELECTED /**< number of output channels if40
- appropriate downmixInstruction exists41
- */42
+ DRC_DEC_TARGET_CHANNEL_COUNT_SELECTED, /**< number of output channels if43
+ appropriate downmixInstruction44
+ exists */45
+ DRC_DEC_OUTPUT_LOUDNESS /**< output loudness in dB, with exponent e = 7, or46
+ DRC_DEC_LOUDNESS_NOT_PRESENT if no loudness is47
+ contained in the bitstream */48
} DRC_DEC_USERPARAM;49
50
typedef enum {51
fdk-aac-2.0.1.tar.gz/libDRCdec/src/FDK_drcDecLib.cpp -> fdk-aac-2.0.2.tar.gz/libDRCdec/src/FDK_drcDecLib.cpp
Changed
201
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
#define DRCDEC_LIB_VL1 112
#define DRCDEC_LIB_VL2 013
#define DRCDEC_LIB_TITLE "MPEG-D DRC Decoder Lib"14
-#ifdef __ANDROID__15
+#ifdef SUPPRESS_BUILD_DATE_INFO16
#define DRCDEC_LIB_BUILD_DATE ""17
#define DRCDEC_LIB_BUILD_TIME ""18
#else19
20
SEL_PROC_OUTPUT selProcOutput;21
} DRC_DECODER;22
23
+static int _getGainStatus(HANDLE_UNI_DRC_GAIN hUniDrcGain) {24
+ return hUniDrcGain->status;25
+}26
+27
static int isResetNeeded(HANDLE_DRC_DECODER hDrcDec,28
const SEL_PROC_OUTPUT oldSelProcOutput) {29
int i, resetNeeded = 0;30
31
return resetNeeded;32
}33
34
-static DRC_DEC_ERROR startSelectionProcess(HANDLE_DRC_DECODER hDrcDec) {35
- DRC_ERROR dErr = DE_OK;36
- DRCDEC_SELECTION_PROCESS_RETURN sErr = DRCDEC_SELECTION_PROCESS_NO_ERROR;37
+static void startSelectionProcess(HANDLE_DRC_DECODER hDrcDec) {38
int uniDrcConfigHasChanged = 0;39
SEL_PROC_OUTPUT oldSelProcOutput = hDrcDec->selProcOutput;40
41
- if (!hDrcDec->status) return DRC_DEC_NOT_READY;42
+ if (!hDrcDec->status) return;43
44
if (hDrcDec->functionalRange & DRC_DEC_SELECTION) {45
uniDrcConfigHasChanged = hDrcDec->uniDrcConfig.diff;46
47
*/48
hDrcDec->selProcOutput.numSelectedDrcSets = 0;49
50
- sErr = drcDec_SelectionProcess_Process(51
+ drcDec_SelectionProcess_Process(52
hDrcDec->hSelectionProc, &(hDrcDec->uniDrcConfig),53
&(hDrcDec->loudnessInfoSet), &(hDrcDec->selProcOutput));54
- if (sErr) return DRC_DEC_OK;55
56
hDrcDec->selProcInputDiff = 0;57
hDrcDec->uniDrcConfig.diff = 0;58
59
60
if (hDrcDec->functionalRange & DRC_DEC_GAIN) {61
if (isResetNeeded(hDrcDec, oldSelProcOutput) || uniDrcConfigHasChanged) {62
- dErr =63
- drcDec_GainDecoder_Config(hDrcDec->hGainDec, &(hDrcDec->uniDrcConfig),64
- hDrcDec->selProcOutput.numSelectedDrcSets,65
- hDrcDec->selProcOutput.selectedDrcSetIds,66
- hDrcDec->selProcOutput.selectedDownmixIds);67
- if (dErr) return DRC_DEC_OK;68
+ drcDec_GainDecoder_Config(hDrcDec->hGainDec, &(hDrcDec->uniDrcConfig),69
+ hDrcDec->selProcOutput.numSelectedDrcSets,70
+ hDrcDec->selProcOutput.selectedDrcSetIds,71
+ hDrcDec->selProcOutput.selectedDownmixIds);72
}73
}74
- return DRC_DEC_OK;75
}76
77
DRC_DEC_ERROR78
79
}80
81
if (hDrcDec->functionalRange & DRC_DEC_GAIN) {82
- dErr = drcDec_GainDecoder_Init(hDrcDec->hGainDec, frameSize, sampleRate);83
+ dErr = drcDec_GainDecoder_SetParam(hDrcDec->hGainDec, GAIN_DEC_FRAME_SIZE,84
+ frameSize);85
+ if (dErr) return DRC_DEC_NOT_OK;86
+ dErr = drcDec_GainDecoder_SetParam(hDrcDec->hGainDec, GAIN_DEC_SAMPLE_RATE,87
+ sampleRate);88
+ if (dErr) return DRC_DEC_NOT_OK;89
+ dErr = drcDec_GainDecoder_Init(hDrcDec->hGainDec);90
if (dErr) return DRC_DEC_NOT_OK;91
}92
93
94
FDK_drcDec_SetParam(HANDLE_DRC_DECODER hDrcDec,95
const DRC_DEC_USERPARAM requestType,96
const FIXP_DBL requestValue) {97
+ DRC_ERROR dErr = DE_OK;98
DRCDEC_SELECTION_PROCESS_RETURN sErr = DRCDEC_SELECTION_PROCESS_NO_ERROR;99
+ int invalidParameter = 0;100
101
if (hDrcDec == NULL) return DRC_DEC_NOT_OPENED;102
103
- if (hDrcDec->functionalRange == DRC_DEC_GAIN)104
- return DRC_DEC_NOT_OK; /* not supported for DRC_DEC_GAIN. All parameters are105
- handed over to selection process lib. */106
+ if (hDrcDec->functionalRange & DRC_DEC_GAIN) {107
+ switch (requestType) {108
+ case DRC_DEC_SAMPLE_RATE:109
+ dErr = drcDec_GainDecoder_SetParam(110
+ hDrcDec->hGainDec, GAIN_DEC_SAMPLE_RATE, (int)requestValue);111
+ if (dErr) return DRC_DEC_PARAM_OUT_OF_RANGE;112
+ break;113
+ case DRC_DEC_FRAME_SIZE:114
+ dErr = drcDec_GainDecoder_SetParam(115
+ hDrcDec->hGainDec, GAIN_DEC_FRAME_SIZE, (int)requestValue);116
+ if (dErr) return DRC_DEC_PARAM_OUT_OF_RANGE;117
+ break;118
+ default:119
+ invalidParameter |= DRC_DEC_GAIN;120
+ }121
+ }122
123
- switch (requestType) {124
- case DRC_DEC_BOOST:125
- sErr = drcDec_SelectionProcess_SetParam(hDrcDec->hSelectionProc,126
- SEL_PROC_BOOST, requestValue,127
- &(hDrcDec->selProcInputDiff));128
- if (sErr) return DRC_DEC_PARAM_OUT_OF_RANGE;129
- break;130
- case DRC_DEC_COMPRESS:131
- sErr = drcDec_SelectionProcess_SetParam(hDrcDec->hSelectionProc,132
- SEL_PROC_COMPRESS, requestValue,133
- &(hDrcDec->selProcInputDiff));134
- if (sErr) return DRC_DEC_PARAM_OUT_OF_RANGE;135
- break;136
- case DRC_DEC_LOUDNESS_NORMALIZATION_ON:137
- sErr = drcDec_SelectionProcess_SetParam(138
- hDrcDec->hSelectionProc, SEL_PROC_LOUDNESS_NORMALIZATION_ON,139
- requestValue, &(hDrcDec->selProcInputDiff));140
- if (sErr) return DRC_DEC_PARAM_OUT_OF_RANGE;141
- break;142
- case DRC_DEC_TARGET_LOUDNESS:143
- sErr = drcDec_SelectionProcess_SetParam(144
- hDrcDec->hSelectionProc, SEL_PROC_TARGET_LOUDNESS, requestValue,145
- &(hDrcDec->selProcInputDiff));146
- if (sErr) return DRC_DEC_PARAM_OUT_OF_RANGE;147
- break;148
- case DRC_DEC_EFFECT_TYPE:149
- sErr = drcDec_SelectionProcess_SetParam(150
- hDrcDec->hSelectionProc, SEL_PROC_EFFECT_TYPE, requestValue,151
- &(hDrcDec->selProcInputDiff));152
- if (sErr) return DRC_DEC_PARAM_OUT_OF_RANGE;153
- break;154
- case DRC_DEC_DOWNMIX_ID:155
- sErr = drcDec_SelectionProcess_SetParam(hDrcDec->hSelectionProc,156
- SEL_PROC_DOWNMIX_ID, requestValue,157
- &(hDrcDec->selProcInputDiff));158
- if (sErr) return DRC_DEC_PARAM_OUT_OF_RANGE;159
- break;160
- case DRC_DEC_TARGET_CHANNEL_COUNT_REQUESTED:161
- sErr = drcDec_SelectionProcess_SetParam(162
- hDrcDec->hSelectionProc, SEL_PROC_TARGET_CHANNEL_COUNT, requestValue,163
- &(hDrcDec->selProcInputDiff));164
- if (sErr) return DRC_DEC_PARAM_OUT_OF_RANGE;165
- break;166
- case DRC_DEC_BASE_CHANNEL_COUNT:167
- sErr = drcDec_SelectionProcess_SetParam(168
- hDrcDec->hSelectionProc, SEL_PROC_BASE_CHANNEL_COUNT, requestValue,169
- &(hDrcDec->selProcInputDiff));170
- if (sErr) return DRC_DEC_NOT_OK;171
- break;172
- case DRC_DEC_LOUDNESS_MEASUREMENT_METHOD:173
- sErr = drcDec_SelectionProcess_SetParam(174
- hDrcDec->hSelectionProc, SEL_PROC_LOUDNESS_MEASUREMENT_METHOD,175
- requestValue, &(hDrcDec->selProcInputDiff));176
- if (sErr) return DRC_DEC_PARAM_OUT_OF_RANGE;177
- break;178
- default:179
- return DRC_DEC_INVALID_PARAM;180
+ if (hDrcDec->functionalRange & DRC_DEC_SELECTION) {181
+ switch (requestType) {182
+ case DRC_DEC_BOOST:183
+ sErr = drcDec_SelectionProcess_SetParam(hDrcDec->hSelectionProc,184
+ SEL_PROC_BOOST, requestValue,185
+ &(hDrcDec->selProcInputDiff));186
+ if (sErr) return DRC_DEC_PARAM_OUT_OF_RANGE;187
+ break;188
+ case DRC_DEC_COMPRESS:189
+ sErr = drcDec_SelectionProcess_SetParam(hDrcDec->hSelectionProc,190
+ SEL_PROC_COMPRESS, requestValue,191
+ &(hDrcDec->selProcInputDiff));192
+ if (sErr) return DRC_DEC_PARAM_OUT_OF_RANGE;193
+ break;194
+ case DRC_DEC_LOUDNESS_NORMALIZATION_ON:195
+ sErr = drcDec_SelectionProcess_SetParam(196
+ hDrcDec->hSelectionProc, SEL_PROC_LOUDNESS_NORMALIZATION_ON,197
+ requestValue, &(hDrcDec->selProcInputDiff));198
+ if (sErr) return DRC_DEC_PARAM_OUT_OF_RANGE;199
+ break;200
+ case DRC_DEC_TARGET_LOUDNESS:201
fdk-aac-2.0.1.tar.gz/libDRCdec/src/drcDec_gainDecoder.cpp -> fdk-aac-2.0.2.tar.gz/libDRCdec/src/drcDec_gainDecoder.cpp
Changed
65
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
}12
13
DRC_ERROR14
-drcDec_GainDecoder_Init(HANDLE_DRC_GAIN_DECODER hGainDec, const int frameSize,15
- const int sampleRate) {16
+drcDec_GainDecoder_Init(HANDLE_DRC_GAIN_DECODER hGainDec) {17
DRC_ERROR err = DE_OK;18
19
- err = initGainDec(hGainDec, frameSize, sampleRate);20
+ err = initGainDec(hGainDec);21
if (err) return err;22
23
initDrcGainBuffers(hGainDec->frameSize, &hGainDec->drcGainBuffers);24
25
}26
27
DRC_ERROR28
+drcDec_GainDecoder_SetParam(HANDLE_DRC_GAIN_DECODER hGainDec,29
+ const GAIN_DEC_PARAM paramType,30
+ const int paramValue) {31
+ switch (paramType) {32
+ case GAIN_DEC_FRAME_SIZE:33
+ if (paramValue < 0) return DE_PARAM_OUT_OF_RANGE;34
+ hGainDec->frameSize = paramValue;35
+ break;36
+ case GAIN_DEC_SAMPLE_RATE:37
+ if (paramValue < 0) return DE_PARAM_OUT_OF_RANGE;38
+ hGainDec->deltaTminDefault = getDeltaTmin(paramValue);39
+ break;40
+ default:41
+ return DE_PARAM_INVALID;42
+ }43
+ return DE_OK;44
+}45
+46
+DRC_ERROR47
drcDec_GainDecoder_SetCodecDependentParameters(48
HANDLE_DRC_GAIN_DECODER hGainDec, const DELAY_MODE delayMode,49
const int timeDomainSupported,50
51
int seq, gainSequenceCount;52
DRC_COEFFICIENTS_UNI_DRC* pCoef =53
selectDrcCoefficients(hUniDrcConfig, LOCATION_SELECTED);54
- if (pCoef == NULL) return DE_OK;55
-56
- gainSequenceCount = fMin(pCoef->gainSequenceCount, (UCHAR)12);57
+ if (pCoef && pCoef->gainSequenceCount) {58
+ gainSequenceCount = fMin(pCoef->gainSequenceCount, (UCHAR)12);59
+ } else {60
+ gainSequenceCount = 1;61
+ }62
63
for (seq = 0; seq < gainSequenceCount; seq++) {64
int lastNodeIndex = 0;65
fdk-aac-2.0.1.tar.gz/libDRCdec/src/drcDec_gainDecoder.h -> fdk-aac-2.0.2.tar.gz/libDRCdec/src/drcDec_gainDecoder.h
Changed
34
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
GAIN_DEC_DRC2_DRC312
} GAIN_DEC_LOCATION;13
14
+typedef enum { GAIN_DEC_FRAME_SIZE, GAIN_DEC_SAMPLE_RATE } GAIN_DEC_PARAM;15
+16
typedef struct {17
FIXP_DBL gainLin; /* e = 7 */18
SHORT time;19
20
drcDec_GainDecoder_Open(HANDLE_DRC_GAIN_DECODER* phGainDec);21
22
DRC_ERROR23
-drcDec_GainDecoder_Init(HANDLE_DRC_GAIN_DECODER hGainDec, const int frameSize,24
- const int sampleRate);25
+drcDec_GainDecoder_Init(HANDLE_DRC_GAIN_DECODER hGainDec);26
+27
+DRC_ERROR28
+drcDec_GainDecoder_SetParam(HANDLE_DRC_GAIN_DECODER hGainDec,29
+ const GAIN_DEC_PARAM paramType,30
+ const int paramValue);31
32
DRC_ERROR33
drcDec_GainDecoder_SetCodecDependentParameters(34
fdk-aac-2.0.1.tar.gz/libDRCdec/src/drcDec_reader.cpp -> fdk-aac-2.0.2.tar.gz/libDRCdec/src/drcDec_reader.cpp
Changed
120
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
uniDrcConfigPresent = FDKreadBits(hBs, 1);12
if (uniDrcConfigPresent) {13
err = drcDec_readUniDrcConfig(hBs, hUniDrcConfig);14
- if (err) return err;15
+ if (err) {16
+ /* clear config, if parsing error occured */17
+ FDKmemclear(hUniDrcConfig, sizeof(UNI_DRC_CONFIG));18
+ hUniDrcConfig->diff = 1;19
+ }20
}21
err = drcDec_readLoudnessInfoSet(hBs, hLoudnessInfoSet);22
- if (err) return err;23
+ if (err) {24
+ /* clear config, if parsing error occured */25
+ FDKmemclear(hLoudnessInfoSet, sizeof(LOUDNESS_INFO_SET));26
+ hLoudnessInfoSet->diff = 1;27
+ }28
}29
30
- if (hUniDrcGain != NULL) {31
- err = drcDec_readUniDrcGain(hBs, hUniDrcConfig, frameSize, deltaTminDefault,32
- hUniDrcGain);33
- if (err) return err;34
- }35
+ err = drcDec_readUniDrcGain(hBs, hUniDrcConfig, frameSize, deltaTminDefault,36
+ hUniDrcGain);37
38
return err;39
}40
41
int seq, gainSequenceCount;42
DRC_COEFFICIENTS_UNI_DRC* pCoef =43
selectDrcCoefficients(hUniDrcConfig, LOCATION_SELECTED);44
- if (pCoef == NULL) return DE_OK;45
- if (hUniDrcGain == NULL) return DE_OK; /* hUniDrcGain not initialized yet */46
-47
- gainSequenceCount = fMin(pCoef->gainSequenceCount, (UCHAR)12);48
+ if (hUniDrcGain == NULL) return DE_NOT_OK;49
+ hUniDrcGain->status = 0;50
+ if (pCoef) {51
+ gainSequenceCount = fMin(pCoef->gainSequenceCount, (UCHAR)12);52
+ } else {53
+ gainSequenceCount = 0;54
+ }55
56
for (seq = 0; seq < gainSequenceCount; seq++) {57
UCHAR index = pCoef->gainSetIndexForGainSequence[seq];58
59
if (err) return err;60
}61
62
+ if (err == DE_OK && gainSequenceCount > 0) {63
+ hUniDrcGain->status = 1;64
+ }65
return err;66
}67
68
69
firFilterOrder;70
int uniqueEqSubbandGainsCount, eqSubbandGainRepresentation,71
eqSubbandGainCount;72
- EQ_SUBBAND_GAIN_FORMAT eqSubbandGainFormat;73
+ int eqSubbandGainFormat;74
75
eqDelayMaxPresent = FDKreadBits(hBs, 1);76
if (eqDelayMaxPresent) {77
78
uniqueEqSubbandGainsCount = FDKreadBits(hBs, 6);79
if (uniqueEqSubbandGainsCount > 0) {80
eqSubbandGainRepresentation = FDKreadBits(hBs, 1);81
- eqSubbandGainFormat = (EQ_SUBBAND_GAIN_FORMAT)FDKreadBits(hBs, 4);82
+ eqSubbandGainFormat = FDKreadBits(hBs, 4);83
switch (eqSubbandGainFormat) {84
case GF_QMF32:85
eqSubbandGainCount = 32;86
87
int additionalDrcSetIdPresent, additionalDrcSetIdCount;88
int dependsOnEqSetPresent, eqChannelGroupCount, tdFilterCascadePresent,89
subbandGainsPresent, eqTransitionDurationPresent;90
+ UCHAR eqChannelGroupForChannel[8];91
92
FDKpushFor(hBs, 6); /* eqSetId */93
FDKpushFor(hBs, 4); /* eqSetComplexityLevel */94
95
96
eqChannelGroupCount = 0;97
for (c = 0; c < channelCount; c++) {98
- UCHAR eqChannelGroupForChannel[8];99
int newGroup = 1;100
if (c >= 8) return DE_MEMORY_ERROR;101
eqChannelGroupForChannel[c] = FDKreadBits(hBs, 7);102
103
drcCharacteristicLeftPresent = FDKreadBits(hBs, 1);104
if (drcCharacteristicLeftPresent) {105
pCoef->characteristicLeftCount = FDKreadBits(hBs, 4);106
- if ((pCoef->characteristicLeftCount + 1) > 8) return DE_MEMORY_ERROR;107
+ if ((pCoef->characteristicLeftCount + 1) > 16) return DE_MEMORY_ERROR;108
for (i = 0; i < pCoef->characteristicLeftCount; i++) {109
err = _readCustomDrcCharacteristic(110
hBs, CS_LEFT, &(pCoef->characteristicLeftFormat[i + 1]),111
112
drcCharacteristicRightPresent = FDKreadBits(hBs, 1);113
if (drcCharacteristicRightPresent) {114
pCoef->characteristicRightCount = FDKreadBits(hBs, 4);115
- if ((pCoef->characteristicRightCount + 1) > 8) return DE_MEMORY_ERROR;116
+ if ((pCoef->characteristicRightCount + 1) > 16) return DE_MEMORY_ERROR;117
for (i = 0; i < pCoef->characteristicRightCount; i++) {118
err = _readCustomDrcCharacteristic(119
hBs, CS_RIGHT, &(pCoef->characteristicRightFormat[i + 1]),120
fdk-aac-2.0.1.tar.gz/libDRCdec/src/drcDec_selectionProcess.cpp -> fdk-aac-2.0.2.tar.gz/libDRCdec/src/drcDec_selectionProcess.cpp
Changed
201
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
#include "drcDec_selectionProcess.h"12
#include "drcDec_tools.h"13
14
-#define UNDEFINED_LOUDNESS_VALUE (FIXP_DBL) MAXVAL_DBL15
-16
typedef enum {17
DETR_NONE = 0,18
DETR_NIGHT = 1,19
20
diff |= _compAssign(&pSelProcInput->loudnessMeasurementMethod,21
requestValueInt);22
break;23
+ case SEL_PROC_ALBUM_MODE:24
+ if ((requestValueInt < 0) || (requestValueInt > 1))25
+ return DRCDEC_SELECTION_PROCESS_PARAM_OUT_OF_RANGE;26
+ diff |= _compAssign(&pSelProcInput->albumMode, requestValueInt);27
+ break;28
case SEL_PROC_DOWNMIX_ID:29
diff |=30
_compAssign(&pSelProcInput->targetConfigRequestType, TCRT_DOWNMIX_ID);31
32
if ((requestValue < (FIXP_DBL)0) ||33
(requestValue > FL2FXCONST_DBL(1.0f / (float)(1 << 1))))34
return DRCDEC_SELECTION_PROCESS_PARAM_OUT_OF_RANGE;35
- diff |= _compAssign(&pSelProcInput->boost, FX_DBL2FX_SGL(requestValue));36
+ diff |= _compAssign(37
+ &pSelProcInput->boost,38
+ FX_DBL2FX_SGL(39
+ requestValue +40
+ (FIXP_DBL)(1 << 15))); /* convert to FIXP_SGL with rounding */41
break;42
case SEL_PROC_COMPRESS:43
if ((requestValue < (FIXP_DBL)0) ||44
(requestValue > FL2FXCONST_DBL(1.0f / (float)(1 << 1))))45
return DRCDEC_SELECTION_PROCESS_PARAM_OUT_OF_RANGE;46
- diff |=47
- _compAssign(&pSelProcInput->compress, FX_DBL2FX_SGL(requestValue));48
+ diff |= _compAssign(49
+ &pSelProcInput->compress,50
+ FX_DBL2FX_SGL(51
+ requestValue +52
+ (FIXP_DBL)(1 << 15))); /* convert to FIXP_SGL with rounding */53
break;54
default:55
return DRCDEC_SELECTION_PROCESS_INVALID_PARAM;56
57
hSelProcInput->loudnessNormalizationOn = 1;58
hSelProcInput->targetLoudness = FL2FXCONST_DBL(-24.0f / (float)(1 << 7));59
hSelProcInput->loudnessDeviationMax = DEFAULT_LOUDNESS_DEVIATION_MAX;60
- hSelProcInput->loudnessMeasurementMethod = MDR_DEFAULT;61
- hSelProcInput->loudnessMeasurementSystem = MSR_DEFAULT;62
+ hSelProcInput->loudnessMeasurementMethod = MDR_ANCHOR_LOUDNESS;63
+ hSelProcInput->loudnessMeasurementSystem = MSR_EXPERT_PANEL;64
hSelProcInput->loudnessMeasurementPreProc = LPR_DEFAULT;65
hSelProcInput->deviceCutOffFrequency = 500;66
hSelProcInput->loudnessNormalizationGainDbMax =67
68
return DRCDEC_SELECTION_PROCESS_NO_ERROR;69
}70
71
-/* #5: The number of DRC bands is supported. */72
+/* #5: The number of DRC bands is supported. Moreover, gainSetIndex and73
+ * gainSequenceIndex are within the allowed range. */74
static DRCDEC_SELECTION_PROCESS_RETURN _preSelectionRequirement5(75
DRC_INSTRUCTIONS_UNI_DRC* pDrcInstructionUniDrc,76
DRC_COEFFICIENTS_UNI_DRC* pCoef, int* pMatchFound) {77
- int i;78
+ int b, i;79
80
*pMatchFound = 1;81
82
+ if (pDrcInstructionUniDrc->drcSetId < 0) /* virtual DRC sets are okay */83
+ {84
+ return DRCDEC_SELECTION_PROCESS_NO_ERROR;85
+ }86
+87
if (pCoef == NULL) /* check for parametricDRC */88
{89
- *pMatchFound = 1;90
+ *pMatchFound = 0; /* parametricDRC not supported */91
+ return DRCDEC_SELECTION_PROCESS_NO_ERROR;92
+ }93
+94
+ if (pCoef->drcLocation !=95
+ pDrcInstructionUniDrc96
+ ->drcLocation) /* drcLocation must be LOCATION_SELECTED */97
+ {98
+ *pMatchFound = 0;99
return DRCDEC_SELECTION_PROCESS_NO_ERROR;100
}101
102
103
int indexDrcCoeff = pDrcInstructionUniDrc->gainSetIndexForChannelGroup[i];104
int bandCount = 0;105
106
+ if (indexDrcCoeff >= 12) {107
+ *pMatchFound = 0;108
+ return DRCDEC_SELECTION_PROCESS_NO_ERROR;109
+ }110
+111
if (indexDrcCoeff > pCoef->gainSetCount - 1) /* check for parametricDRC */112
{113
- *pMatchFound = 1;114
- return DRCDEC_SELECTION_PROCESS_NO_ERROR;115
+ continue;116
}117
118
GAIN_SET* gainSet = &(pCoef->gainSet[indexDrcCoeff]);119
120
if (bandCount > 4) {121
*pMatchFound = 0;122
}123
+124
+ for (b = 0; b < bandCount; b++) {125
+ if ((gainSet->gainSequenceIndex[b] >= 12) ||126
+ (gainSet->gainSequenceIndex[b] >= pCoef->gainSequenceCount)) {127
+ *pMatchFound = 0;128
+ return DRCDEC_SELECTION_PROCESS_NO_ERROR;129
+ }130
+ }131
}132
133
return DRCDEC_SELECTION_PROCESS_NO_ERROR;134
135
return retVal;136
}137
138
+static int _drcSetIsUsable(HANDLE_UNI_DRC_CONFIG hUniDrcConfig,139
+ DRC_INSTRUCTIONS_UNI_DRC* pInst) {140
+ int usable = 0;141
+ DRC_COEFFICIENTS_UNI_DRC* pCoef =142
+ selectDrcCoefficients(hUniDrcConfig, LOCATION_SELECTED);143
+144
+ /* check if ID is unique */145
+ if (selectDrcInstructions(hUniDrcConfig, pInst->drcSetId) != pInst) return 0;146
+ /* sanity check on drcInstructions */147
+ _preSelectionRequirement5(pInst, pCoef, &usable);148
+ return usable;149
+}150
+151
/* #8: The range of the target loudness specified for a DRC set has to include152
* the requested decoder target loudness. */153
static DRCDEC_SELECTION_PROCESS_RETURN _preSelectionRequirement8(154
155
156
FIXP_DBL loudnessDeviationMax =157
((FIXP_DBL)hSelProcInput->loudnessDeviationMax) << (DFRACT_BITS - 1 - 7);158
- ;159
160
- if (hSelProcInput->loudnessNormalizationOn) {161
+ {162
retVal = _getLoudness(hLoudnessInfoSet, hSelProcInput->albumMode,163
hSelProcInput->loudnessMeasurementMethod,164
hSelProcInput->loudnessMeasurementSystem,165
166
hSelProcInput->downmixIdRequested[downmixIdIndex],167
&loudnessNormalizationGainDb, &loudness);168
if (retVal) return (retVal);169
- } else {170
+ }171
+172
+ if (!hSelProcInput->loudnessNormalizationOn) {173
loudnessNormalizationGainDb = (FIXP_DBL)0;174
- loudness = UNDEFINED_LOUDNESS_VALUE;175
}176
177
retVal = _getSignalPeakLevel(178
179
pSelectionData->loudnessNormalizationGainDbAdjusted +180
hSelProcInput->loudnessNormalizationGainModificationDb;181
hSelProcOutput->outputPeakLevelDb = pSelectionData->outputPeakLevel;182
+ hSelProcOutput->outputLoudness = pSelectionData->outputLoudness;183
184
hSelProcOutput->boost = boost;185
hSelProcOutput->compress = compress;186
187
int dependsOnDrcSetID = pSelectionData->pInst->dependsOnDrcSet;188
189
for (i = 0; i < hUniDrcConfig->drcInstructionsCountInclVirtual; i++) {190
- if (hUniDrcConfig->drcInstructionsUniDrc[i].drcSetId ==191
- dependsOnDrcSetID) {192
+ DRC_INSTRUCTIONS_UNI_DRC* pInst =193
+ &(hUniDrcConfig->drcInstructionsUniDrc[i]);194
+ if (!_drcSetIsUsable(hUniDrcConfig, pInst)) continue;195
+196
+ if (pInst->drcSetId == dependsOnDrcSetID) {197
hSelProcOutput->selectedDrcSetIds[hSelProcOutput->numSelectedDrcSets] =198
hUniDrcConfig->drcInstructionsUniDrc[i].drcSetId;199
hSelProcOutput->selectedDownmixIds[hSelProcOutput->numSelectedDrcSets] =200
201
fdk-aac-2.0.1.tar.gz/libDRCdec/src/drcDec_selectionProcess.h -> fdk-aac-2.0.2.tar.gz/libDRCdec/src/drcDec_selectionProcess.h
Changed
27
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
12
typedef struct s_drcdec_selection_process* HANDLE_DRC_SELECTION_PROCESS;13
14
+#define UNDEFINED_LOUDNESS_VALUE (FIXP_DBL)(MAXVAL_DBL - 1)15
+16
typedef enum {17
DRCDEC_SELECTION_PROCESS_NO_ERROR = 0,18
19
20
SEL_PROC_EFFECT_TYPE,21
SEL_PROC_EFFECT_TYPE_FALLBACK_CODE,22
SEL_PROC_LOUDNESS_MEASUREMENT_METHOD,23
+ SEL_PROC_ALBUM_MODE,24
SEL_PROC_DOWNMIX_ID,25
SEL_PROC_TARGET_LAYOUT,26
SEL_PROC_TARGET_CHANNEL_COUNT,27
fdk-aac-2.0.1.tar.gz/libDRCdec/src/drcDec_types.h -> fdk-aac-2.0.2.tar.gz/libDRCdec/src/drcDec_types.h
Changed
27
1
2
3
UCHAR uniDrcGainExtPresent;4
UNI_DRC_GAIN_EXTENSION uniDrcGainExtension;5
+6
+ /* derived data */7
+ UCHAR status;8
} UNI_DRC_GAIN, *HANDLE_UNI_DRC_GAIN;9
10
/****************/11
12
UCHAR drcFrameSizePresent;13
USHORT drcFrameSize;14
UCHAR characteristicLeftCount;15
- UCHAR characteristicLeftFormat[8];16
- CUSTOM_DRC_CHAR customCharacteristicLeft[8];17
+ UCHAR characteristicLeftFormat[16];18
+ CUSTOM_DRC_CHAR customCharacteristicLeft[16];19
UCHAR characteristicRightCount;20
- UCHAR characteristicRightFormat[8];21
- CUSTOM_DRC_CHAR customCharacteristicRight[8];22
+ UCHAR characteristicRightFormat[16];23
+ CUSTOM_DRC_CHAR customCharacteristicRight[16];24
UCHAR25
gainSequenceCount; /* unsaturated value, i.e. as provided in bitstream */26
UCHAR gainSetCount; /* saturated to 12 */27
fdk-aac-2.0.1.tar.gz/libDRCdec/src/drcGainDec_init.cpp -> fdk-aac-2.0.2.tar.gz/libDRCdec/src/drcGainDec_init.cpp
Changed
52
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
}12
13
DRC_ERROR14
-initGainDec(HANDLE_DRC_GAIN_DECODER hGainDec, const int frameSize,15
- const int sampleRate) {16
+initGainDec(HANDLE_DRC_GAIN_DECODER hGainDec) {17
int i, j, k;18
19
- if (frameSize < 1) {20
- return DE_NOT_OK;21
- }22
-23
- hGainDec->frameSize = frameSize;24
-25
- if (hGainDec->frameSize * 1000 < sampleRate) {26
- return DE_NOT_OK;27
- }28
-29
- hGainDec->deltaTminDefault = getDeltaTmin(sampleRate);30
- if (hGainDec->deltaTminDefault > hGainDec->frameSize) {31
- return DE_NOT_OK;32
- }33
+ /* sanity check */34
+ if (hGainDec->deltaTminDefault > hGainDec->frameSize) return DE_NOT_OK;35
36
for (i = 0; i < MAX_ACTIVE_DRCS; i++) {37
for (j = 0; j < 8; j++) {38
39
for (a = 0; a < hGainDec->nActiveDrcs; a++) {40
hGainDec->activeDrc[a].activeDrcOffset = accGainElementCount;41
accGainElementCount += hGainDec->activeDrc[a].gainElementCount;42
+ if (accGainElementCount > 12) {43
+ hGainDec->nActiveDrcs = a;44
+ return DE_NOT_OK;45
+ }46
}47
48
- if (accGainElementCount > 12) return DE_NOT_OK;49
-50
return DE_OK;51
}52
fdk-aac-2.0.1.tar.gz/libDRCdec/src/drcGainDec_init.h -> fdk-aac-2.0.2.tar.gz/libDRCdec/src/drcGainDec_init.h
Changed
20
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
#define DRCGAINDEC_INIT_H12
13
DRC_ERROR14
-initGainDec(HANDLE_DRC_GAIN_DECODER hGainDec, const int frameSize,15
- const int sampleRate);16
+initGainDec(HANDLE_DRC_GAIN_DECODER hGainDec);17
18
void initDrcGainBuffers(const int frameSize, DRC_GAIN_BUFFERS* drcGainBuffers);19
20
fdk-aac-2.0.1.tar.gz/libDRCdec/src/drcGainDec_preprocess.cpp -> fdk-aac-2.0.2.tar.gz/libDRCdec/src/drcGainDec_preprocess.cpp
Changed
36
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
&e_tmp2);12
invExp = fDivNorm(FL2FXCONST_DBL(1.0f / (float)(1 << 1)), exp, &e_invExp);13
e_invExp += 1 - 5;14
+ if (tmp2 < (FIXP_DBL)0) {15
+ return DE_NOT_OK;16
+ }17
denom = fPow(tmp2, e_tmp2, invExp, e_invExp, &e_denom);18
*out = fDivNormSigned(tmp, denom, &e_out);19
e_out += 7 - e_denom;20
21
nDrcBands = pActiveDrc->bandCountForChannelGroup[g];22
for (b = 0; b < nDrcBands; b++) {23
DRC_ERROR err = DE_OK;24
- if (gainSetIndex >= 12) return DE_PARAM_OUT_OF_RANGE;25
GAIN_SET* pGainSet = &(pCoef->gainSet[gainSetIndex]);26
int seq = pGainSet->gainSequenceIndex[b];27
DRC_CHARACTERISTIC* pDChar = &(pGainSet->drcCharacteristic[b]);28
29
err = _prepareDrcCharacteristic(pDChar, pCoef, b, &nodeMod);30
if (err) return err;31
32
- if (seq >= 12) return DE_PARAM_OUT_OF_RANGE;33
/* copy a node buffer and convert from dB to linear */34
pLnb->nNodes[lnbp] = fMin((int)hUniDrcGain->nNodes[seq], 16);35
for (i = 0; i < pLnb->nNodes[lnbp]; i++) {36
fdk-aac-2.0.1.tar.gz/libDRCdec/src/drcGainDec_process.cpp -> fdk-aac-2.0.2.tar.gz/libDRCdec/src/drcGainDec_process.cpp
Changed
10
1
2
pLinearNodeBuffer[pActiveDrc->lnbIndexForChannel[c][lnbIx] + b]);3
else4
pLnbPrevious = pDummyLnb;5
- if (pLnbPrevious->nNodes[lnbIx] <= 0 || pLnbPrevious->nNodes[lnbIx] > 16)6
- return DE_NOT_OK;7
nodePrevious =8
pLnbPrevious->linearNode[lnbIx][pLnbPrevious->nNodes[lnbIx] - 1];9
nodePrevious.time -= hGainDec->frameSize;10
fdk-aac-2.0.1.tar.gz/libFDK/include/FDK_qmf_domain.h -> fdk-aac-2.0.2.tar.gz/libFDK/include/FDK_qmf_domain.h
Changed
71
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
#define QMF_WB_SECTION_SIZE (1024 * 2)12
13
H_ALLOC_MEM_OVERLAY(QmfWorkBufferCore1, FIXP_DBL)14
-H_ALLOC_MEM_OVERLAY(QmfWorkBufferCore2, FIXP_DBL)15
H_ALLOC_MEM_OVERLAY(QmfWorkBufferCore3, FIXP_DBL)16
H_ALLOC_MEM_OVERLAY(QmfWorkBufferCore4, FIXP_DBL)17
-H_ALLOC_MEM_OVERLAY(QmfWorkBufferCore5, FIXP_DBL)18
H_ALLOC_MEM_OVERLAY(QmfWorkBufferCore6, FIXP_DBL)19
+H_ALLOC_MEM_OVERLAY(QmfWorkBufferCore7, FIXP_DBL)20
21
#define QMF_DOMAIN_MAX_ANALYSIS_QMF_BANDS (64)22
#define QMF_DOMAIN_MAX_SYNTHESIS_QMF_BANDS (QMF_MAX_SYNTHESIS_BANDS)23
24
#define QMF_DOMAIN_OV_TIMESLOTS_16 (3)25
#define QMF_DOMAIN_OV_TIMESLOTS_32 (6)26
27
-H_ALLOC_MEM(AnaQmfStates, FIXP_QAS)28
+H_ALLOC_MEM(AnaQmfStates, FIXP_DBL)29
H_ALLOC_MEM(SynQmfStates, FIXP_QSS)30
H_ALLOC_MEM(QmfSlotsReal, FIXP_DBL *)31
H_ALLOC_MEM(QmfSlotsImag, FIXP_DBL *)32
H_ALLOC_MEM(QmfOverlapBuffer, FIXP_DBL)33
34
-H_ALLOC_MEM(AnaQmfStates16, FIXP_QAS)35
-H_ALLOC_MEM(AnaQmfStates24, FIXP_QAS)36
-H_ALLOC_MEM(AnaQmfStates32, FIXP_QAS)37
+H_ALLOC_MEM(AnaQmfStates16, FIXP_DBL)38
+H_ALLOC_MEM(AnaQmfStates24, FIXP_DBL)39
+H_ALLOC_MEM(AnaQmfStates32, FIXP_DBL)40
H_ALLOC_MEM(QmfSlotsReal16, FIXP_DBL *)41
H_ALLOC_MEM(QmfSlotsReal32, FIXP_DBL *)42
H_ALLOC_MEM(QmfSlotsImag16, FIXP_DBL *)43
44
H_ALLOC_MEM(QmfOverlapBuffer16, FIXP_DBL)45
H_ALLOC_MEM(QmfOverlapBuffer32, FIXP_DBL)46
47
-#define QDOM_PCM INT_PCM48
-49
/**50
* Structure to hold the configuration data which is global whithin a QMF domain51
* instance.52
53
park a channel if only one processing channel is54
available. */55
UCHAR parkChannel_requested;56
- QDOM_PCM57
- *TDinput; /*!< Pointer to time domain data used as input for the QMF58
- analysis. */59
FIXP_DBL *60
pWorkBuffer[QMF_MAX_WB_SECTIONS]; /*!< Pointerarray to volatile memory. */61
UINT flags; /*!< Flags to be set on all QMF analysis/synthesis filter62
63
(workBuf_nTimeSlots * workBuf_nBands * CMPLX_MOD). */64
USHORT workBufferOffset; /*!< Offset within work buffer. */65
USHORT workBufferSectSize; /*!< Size of work buffer section. */66
- FIXP_QAS *67
+ FIXP_DBL *68
pAnaQmfStates; /*!< Pointer to QMF analysis states (persistent memory). */69
FIXP_DBL70
*pOverlapBuffer; /*!< Pointer to QMF overlap/delay memory (persistent71
fdk-aac-2.0.1.tar.gz/libFDK/include/fixpoint_math.h -> fdk-aac-2.0.2.tar.gz/libFDK/include/fixpoint_math.h
Changed
84
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
* \return non-zero if (a_m*2^a_e) < (b_m*2^b_e), 0 otherwise12
*/13
FDK_INLINE INT fIsLessThan(FIXP_DBL a_m, INT a_e, FIXP_DBL b_m, INT b_e) {14
+ INT n;15
+16
+ n = fixnorm_D(a_m);17
+ a_m <<= n;18
+ a_e -= n;19
+20
+ n = fixnorm_D(b_m);21
+ b_m <<= n;22
+ b_e -= n;23
+24
+ if (a_m == (FIXP_DBL)0) a_e = b_e;25
+ if (b_m == (FIXP_DBL)0) b_e = a_e;26
+27
if (a_e > b_e) {28
return ((b_m >> fMin(a_e - b_e, DFRACT_BITS - 1)) > a_m);29
} else {30
31
}32
33
FDK_INLINE INT fIsLessThan(FIXP_SGL a_m, INT a_e, FIXP_SGL b_m, INT b_e) {34
+ INT n;35
+36
+ n = fixnorm_S(a_m);37
+ a_m <<= n;38
+ a_e -= n;39
+40
+ n = fixnorm_S(b_m);41
+ b_m <<= n;42
+ b_e -= n;43
+44
+ if (a_m == (FIXP_SGL)0) a_e = b_e;45
+ if (b_m == (FIXP_SGL)0) b_e = a_e;46
+47
if (a_e > b_e) {48
return ((b_m >> fMin(a_e - b_e, FRACT_BITS - 1)) > a_m);49
} else {50
51
m = fMultNorm(a, (FIXP_DBL)b, &m_e);52
53
if (m_e < (INT)0) {54
- if (m_e > (INT)-DFRACT_BITS) {55
+ if (m_e > (INT) - (DFRACT_BITS - 1)) {56
mi = (m >> (-m_e));57
if ((LONG)m & ((1 << (-m_e)) - 1)) {58
mi = mi + (FIXP_DBL)1;59
}60
} else {61
- mi = (FIXP_DBL)1;62
- if (m < (FIXP_DBL)0) {63
- mi = (FIXP_DBL)0;64
+ if (m > (FIXP_DBL)0) {65
+ mi = (FIXP_DBL)1;66
+ } else {67
+ if ((m_e == -(DFRACT_BITS - 1)) && (m == (FIXP_DBL)MINVAL_DBL)) {68
+ mi = (FIXP_DBL)-1;69
+ } else {70
+ mi = (FIXP_DBL)0;71
+ }72
}73
}74
} else {75
76
77
/**78
* \brief return (base_m * 2^base_e) ^ N79
- * \param base_m mantissa of the base80
+ * \param base_m mantissa of the base. Must not be negative.81
* \param base_e exponent of the base82
* \param N power to be calculated of the base83
* \param result_e pointer to a INT where the exponent of the result will be84
fdk-aac-2.0.1.tar.gz/libFDK/include/mdct.h -> fdk-aac-2.0.2.tar.gz/libFDK/include/mdct.h
Changed
34
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
#include "common_fix.h"12
13
#define MDCT_OUT_HEADROOM 2 /* Output additional headroom */14
-#define PCM_OUT_BITS SAMPLE_BITS15
+16
+#define PCM_OUT_BITS DFRACT_BITS17
#define PCM_OUT_HEADROOM 8 /* Must have the same values as DMXH_HEADROOM */18
19
-#define MDCT_OUTPUT_SCALE (-MDCT_OUT_HEADROOM + (DFRACT_BITS - SAMPLE_BITS))20
+#define MDCT_OUTPUT_SCALE (-MDCT_OUT_HEADROOM + (DFRACT_BITS - PCM_OUT_BITS))21
/* Refer to "Output word length" in ISO/IEC 14496-3:2008(E) 23.2.3.6 */22
#define MDCT_OUTPUT_GAIN 1623
24
-#if (MDCT_OUTPUT_SCALE >= 0)25
-#define IMDCT_SCALE(x) SATURATE_RIGHT_SHIFT(x, MDCT_OUTPUT_SCALE, PCM_OUT_BITS)26
-#else27
-#define IMDCT_SCALE(x) SATURATE_LEFT_SHIFT(x, -MDCT_OUTPUT_SCALE, PCM_OUT_BITS)28
-#endif29
+#define IMDCT_SCALE(x, s) \30
+ SATURATE_RIGHT_SHIFT((x), ((s) + MDCT_OUTPUT_SCALE), PCM_OUT_BITS)31
#define IMDCT_SCALE_DBL(x) (FIXP_DBL)(x)32
#define IMDCT_SCALE_DBL_LSH1(x) SATURATE_LEFT_SHIFT_ALT((x), 1, DFRACT_BITS)33
34
fdk-aac-2.0.1.tar.gz/libFDK/include/mips/abs_mips.h -> fdk-aac-2.0.2.tar.gz/libFDK/include/mips/abs_mips.h
Changed
10
1
2
3
#if defined(__GNUC__) && defined(__mips__)4
5
-#if defined(__mips_dsp)6
+#if defined(__mips_dsp) && !defined(__mips16)7
#define FUNCTION_fixabs_D8
#define FUNCTION_fixabs_I9
#define FUNCTION_fixabs_S10
fdk-aac-2.0.1.tar.gz/libFDK/include/mips/scale_mips.h -> fdk-aac-2.0.2.tar.gz/libFDK/include/mips/scale_mips.h
Changed
10
1
2
#ifndef SCALE_MIPS_H3
#define SCALE_MIPS_H4
5
-#if defined(__mips_dsp)6
+#if defined(__mips_dsp) && !defined(__mips16)7
8
/*!9
*10
fdk-aac-2.0.1.tar.gz/libFDK/include/qmf.h -> fdk-aac-2.0.2.tar.gz/libFDK/include/qmf.h
Changed
177
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
12
#define FIXP_QAS FIXP_PCM13
#define QAS_BITS SAMPLE_BITS14
+#define INT_PCM_QMFIN INT_PCM15
16
#define FIXP_QSS FIXP_DBL17
#define QSS_BITS DFRACT_BITS18
19
20
typedef struct QMF_FILTER_BANK *HANDLE_QMF_FILTER_BANK;21
22
+int qmfInitAnalysisFilterBank(23
+ HANDLE_QMF_FILTER_BANK h_Qmf, /*!< QMF Handle */24
+ FIXP_QAS *pFilterStates, /*!< Pointer to filter state buffer */25
+ int noCols, /*!< Number of time slots */26
+ int lsb, /*!< Number of lower bands */27
+ int usb, /*!< Number of upper bands */28
+ int no_channels, /*!< Number of critically sampled bands */29
+ int flags); /*!< Flags */30
+#if SAMPLE_BITS == 1631
+32
+int qmfInitAnalysisFilterBank(33
+ HANDLE_QMF_FILTER_BANK h_Qmf, /*!< QMF Handle */34
+ FIXP_DBL *pFilterStates, /*!< Pointer to filter state buffer */35
+ int noCols, /*!< Number of time slots */36
+ int lsb, /*!< Number of lower bands */37
+ int usb, /*!< Number of upper bands */38
+ int no_channels, /*!< Number of critically sampled bands */39
+ int flags); /*!< Flags */40
+#endif41
+42
void qmfAnalysisFiltering(43
HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Analysis Bank */44
FIXP_DBL **qmfReal, /*!< Pointer to real subband slots */45
FIXP_DBL **qmfImag, /*!< Pointer to imag subband slots */46
QMF_SCALE_FACTOR *scaleFactor, /*!< Scale factors of QMF data */47
- const LONG *timeIn, /*!< Time signal */48
+ const INT_PCM *timeIn, /*!< Time signal */49
const int timeIn_e, /*!< Exponent of audio data */50
const int stride, /*!< Stride factor of audio data */51
- FIXP_DBL *pWorkBuffer /*!< pointer to temporary working buffer */52
+ FIXP_DBL *pWorkBuffer /*!< pointer to temporal working buffer */53
);54
+#if SAMPLE_BITS == 1655
56
void qmfAnalysisFiltering(57
HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Analysis Bank */58
FIXP_DBL **qmfReal, /*!< Pointer to real subband slots */59
FIXP_DBL **qmfImag, /*!< Pointer to imag subband slots */60
QMF_SCALE_FACTOR *scaleFactor, /*!< Scale factors of QMF data */61
- const INT_PCM *timeIn, /*!< Time signal */62
+ const LONG *timeIn, /*!< Time signal */63
const int timeIn_e, /*!< Exponent of audio data */64
const int stride, /*!< Stride factor of audio data */65
- FIXP_DBL *pWorkBuffer /*!< pointer to temporal working buffer */66
-);67
-68
-void qmfSynthesisFiltering(69
- HANDLE_QMF_FILTER_BANK synQmf, /*!< Handle of Qmf Synthesis Bank */70
- FIXP_DBL **QmfBufferReal, /*!< Pointer to real subband slots */71
- FIXP_DBL **QmfBufferImag, /*!< Pointer to imag subband slots */72
- const QMF_SCALE_FACTOR *scaleFactor, /*!< Scale factors of QMF data */73
- const int ov_len, /*!< Length of band overlap */74
- INT_PCM *timeOut, /*!< Time signal */75
- const INT stride, /*!< Stride factor of audio data */76
- FIXP_DBL *pWorkBuffer /*!< pointer to temporary working buffer. It must be77
- aligned */78
+ FIXP_DBL *pWorkBuffer /*!< pointer to temporary working buffer */79
);80
-81
-int qmfInitAnalysisFilterBank(82
- HANDLE_QMF_FILTER_BANK h_Qmf, /*!< QMF Handle */83
- FIXP_QAS *pFilterStates, /*!< Pointer to filter state buffer */84
- int noCols, /*!< Number of time slots */85
- int lsb, /*!< Number of lower bands */86
- int usb, /*!< Number of upper bands */87
- int no_channels, /*!< Number of critically sampled bands */88
- int flags); /*!< Flags */89
+#endif90
91
void qmfAnalysisFilteringSlot(92
HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Synthesis Bank */93
FIXP_DBL *qmfReal, /*!< Low and High band, real */94
FIXP_DBL *qmfImag, /*!< Low and High band, imag */95
- const LONG *timeIn, /*!< Pointer to input */96
+ const INT_PCM *timeIn, /*!< Pointer to input */97
const int stride, /*!< stride factor of input */98
- FIXP_DBL *pWorkBuffer /*!< pointer to temporary working buffer */99
+ FIXP_DBL *pWorkBuffer /*!< pointer to temporal working buffer */100
);101
+#if SAMPLE_BITS == 16102
103
void qmfAnalysisFilteringSlot(104
HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Synthesis Bank */105
FIXP_DBL *qmfReal, /*!< Low and High band, real */106
FIXP_DBL *qmfImag, /*!< Low and High band, imag */107
- const INT_PCM *timeIn, /*!< Pointer to input */108
+ const LONG *timeIn, /*!< Pointer to input */109
const int stride, /*!< stride factor of input */110
- FIXP_DBL *pWorkBuffer /*!< pointer to temporal working buffer */111
+ FIXP_DBL *pWorkBuffer /*!< pointer to temporary working buffer */112
);113
+#endif114
+115
int qmfInitSynthesisFilterBank(116
HANDLE_QMF_FILTER_BANK h_Qmf, /*!< QMF Handle */117
FIXP_QSS *pFilterStates, /*!< Pointer to filter state buffer */118
119
int no_channels, /*!< Number of critically sampled bands */120
int flags); /*!< Flags */121
122
+void qmfSynthesisFiltering(123
+ HANDLE_QMF_FILTER_BANK synQmf, /*!< Handle of Qmf Synthesis Bank */124
+ FIXP_DBL **QmfBufferReal, /*!< Pointer to real subband slots */125
+ FIXP_DBL **QmfBufferImag, /*!< Pointer to imag subband slots */126
+ const QMF_SCALE_FACTOR *scaleFactor, /*!< Scale factors of QMF data */127
+ const int ov_len, /*!< Length of band overlap */128
+ INT_PCM *timeOut, /*!< Time signal */129
+ const INT stride, /*!< Stride factor of audio data */130
+ FIXP_DBL *pWorkBuffer /*!< pointer to temporary working buffer. It must be131
+ aligned */132
+);133
+#if SAMPLE_BITS == 16134
+135
+void qmfSynthesisFiltering(136
+ HANDLE_QMF_FILTER_BANK synQmf, /*!< Handle of Qmf Synthesis Bank */137
+ FIXP_DBL **QmfBufferReal, /*!< Pointer to real subband slots */138
+ FIXP_DBL **QmfBufferImag, /*!< Pointer to imag subband slots */139
+ const QMF_SCALE_FACTOR *scaleFactor, /*!< Scale factors of QMF data */140
+ const int ov_len, /*!< Length of band overlap */141
+ LONG *timeOut, /*!< Time signal */142
+ const int timeOut_e, /*!< Target exponent for timeOut */143
+ FIXP_DBL *pWorkBuffer /*!< pointer to temporary working buffer */144
+);145
+#endif146
+147
void qmfSynthesisFilteringSlot(HANDLE_QMF_FILTER_BANK synQmf,148
const FIXP_DBL *realSlot,149
const FIXP_DBL *imagSlot,150
const int scaleFactorLowBand,151
const int scaleFactorHighBand, INT_PCM *timeOut,152
const int timeOut_e, FIXP_DBL *pWorkBuffer);153
+#if SAMPLE_BITS == 16154
+155
+void qmfSynthesisFilteringSlot(HANDLE_QMF_FILTER_BANK synQmf,156
+ const FIXP_DBL *realSlot,157
+ const FIXP_DBL *imagSlot,158
+ const int scaleFactorLowBand,159
+ const int scaleFactorHighBand, LONG *timeOut,160
+ const int timeOut_e, FIXP_DBL *pWorkBuffer);161
+#endif162
163
void qmfChangeOutScalefactor(164
HANDLE_QMF_FILTER_BANK synQmf, /*!< Handle of Qmf Synthesis Bank */165
166
FIXP_DBL outputGain, /*!< New gain for output data (mantissa) */167
int outputGainScale /*!< New gain for output data (exponent) */168
);169
-void qmfSynPrototypeFirSlot(170
- HANDLE_QMF_FILTER_BANK qmf,171
- FIXP_DBL *RESTRICT realSlot, /*!< Input: Pointer to real Slot */172
- FIXP_DBL *RESTRICT imagSlot, /*!< Input: Pointer to imag Slot */173
- INT_PCM *RESTRICT timeOut, /*!< Time domain data */174
- const int timeOut_e);175
176
#endif /*ifndef QMF_H */177
fdk-aac-2.0.1.tar.gz/libFDK/include/qmf_pcm.h -> fdk-aac-2.0.2.tar.gz/libFDK/include/qmf_pcm.h
Changed
201
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
timeOut + (i * L * stride), stride, pWorkBuffer);12
} /* no_col loop i */13
}14
+15
+/*!16
+ *17
+ * \brief Create QMF filter bank instance18
+ *19
+ *20
+ * \return 0 if successful21
+ *22
+ */23
+int qmfInitAnalysisFilterBank(24
+ HANDLE_QMF_FILTER_BANK h_Qmf, /*!< Returns handle */25
+ FIXP_QAS *pFilterStates, /*!< Handle to filter states */26
+ int noCols, /*!< Number of timeslots per frame */27
+ int lsb, /*!< lower end of QMF */28
+ int usb, /*!< upper end of QMF */29
+ int no_channels, /*!< Number of channels (bands) */30
+ int flags) /*!< Low Power flag */31
+{32
+ int err = qmfInitFilterBank(h_Qmf, pFilterStates, noCols, lsb, usb,33
+ no_channels, flags, 0);34
+ if (!(flags & QMF_FLAG_KEEP_STATES) && (h_Qmf->FilterStates != NULL)) {35
+ FDKmemclear(h_Qmf->FilterStates,36
+ (2 * QMF_NO_POLY - 1) * h_Qmf->no_channels * sizeof(FIXP_QAS));37
+ }38
+39
+ FDK_ASSERT(h_Qmf->no_channels >= h_Qmf->lsb);40
+41
+ return err;42
+}43
+44
+#ifndef FUNCTION_qmfAnaPrototypeFirSlot45
+/*!46
+ \brief Perform Analysis Prototype Filtering on a single slot of input data.47
+*/48
+static void qmfAnaPrototypeFirSlot(49
+ FIXP_DBL *analysisBuffer,50
+ INT no_channels, /*!< Number channels of analysis filter */51
+ const FIXP_PFT *p_filter, INT p_stride, /*!< Stride of analysis filter */52
+ FIXP_QAS *RESTRICT pFilterStates) {53
+ INT k;54
+55
+ FIXP_DBL accu;56
+ const FIXP_PFT *RESTRICT p_flt = p_filter;57
+ FIXP_DBL *RESTRICT pData_0 = analysisBuffer + 2 * no_channels - 1;58
+ FIXP_DBL *RESTRICT pData_1 = analysisBuffer;59
+60
+ FIXP_QAS *RESTRICT sta_0 = (FIXP_QAS *)pFilterStates;61
+ FIXP_QAS *RESTRICT sta_1 =62
+ (FIXP_QAS *)pFilterStates + (2 * QMF_NO_POLY * no_channels) - 1;63
+ INT pfltStep = QMF_NO_POLY * (p_stride);64
+ INT staStep1 = no_channels << 1;65
+ INT staStep2 = (no_channels << 3) - 1; /* Rewind one less */66
+67
+ /* FIR filters 127..64 0..63 */68
+ for (k = 0; k < no_channels; k++) {69
+ accu = fMultDiv2(p_flt[0], *sta_1);70
+ sta_1 -= staStep1;71
+ accu += fMultDiv2(p_flt[1], *sta_1);72
+ sta_1 -= staStep1;73
+ accu += fMultDiv2(p_flt[2], *sta_1);74
+ sta_1 -= staStep1;75
+ accu += fMultDiv2(p_flt[3], *sta_1);76
+ sta_1 -= staStep1;77
+ accu += fMultDiv2(p_flt[4], *sta_1);78
+ *pData_1++ = (accu << 1);79
+ sta_1 += staStep2;80
+81
+ p_flt += pfltStep;82
+ accu = fMultDiv2(p_flt[0], *sta_0);83
+ sta_0 += staStep1;84
+ accu += fMultDiv2(p_flt[1], *sta_0);85
+ sta_0 += staStep1;86
+ accu += fMultDiv2(p_flt[2], *sta_0);87
+ sta_0 += staStep1;88
+ accu += fMultDiv2(p_flt[3], *sta_0);89
+ sta_0 += staStep1;90
+ accu += fMultDiv2(p_flt[4], *sta_0);91
+ *pData_0-- = (accu << 1);92
+ sta_0 -= staStep2;93
+ }94
+}95
+#endif /* !defined(FUNCTION_qmfAnaPrototypeFirSlot) */96
+97
+#ifndef FUNCTION_qmfAnaPrototypeFirSlot_NonSymmetric98
+/*!99
+ \brief Perform Analysis Prototype Filtering on a single slot of input data.100
+*/101
+static void qmfAnaPrototypeFirSlot_NonSymmetric(102
+ FIXP_DBL *analysisBuffer,103
+ int no_channels, /*!< Number channels of analysis filter */104
+ const FIXP_PFT *p_filter, int p_stride, /*!< Stride of analysis filter */105
+ FIXP_QAS *RESTRICT pFilterStates) {106
+ const FIXP_PFT *RESTRICT p_flt = p_filter;107
+ int p, k;108
+109
+ for (k = 0; k < 2 * no_channels; k++) {110
+ FIXP_DBL accu = (FIXP_DBL)0;111
+112
+ p_flt += QMF_NO_POLY * (p_stride - 1);113
+114
+ /*115
+ Perform FIR-Filter116
+ */117
+ for (p = 0; p < QMF_NO_POLY; p++) {118
+ accu += fMultDiv2(*p_flt++, pFilterStates[2 * no_channels * p]);119
+ }120
+ analysisBuffer[2 * no_channels - 1 - k] = (accu << 1);121
+ pFilterStates++;122
+ }123
+}124
+#endif /* FUNCTION_qmfAnaPrototypeFirSlot_NonSymmetric */125
+126
+/*127
+ * \brief Perform one QMF slot analysis of the time domain data of timeIn128
+ * with specified stride and stores the real part of the subband129
+ * samples in rSubband, and the imaginary part in iSubband130
+ *131
+ * Note: anaQmf->lsb can be greater than anaQmf->no_channels in case132
+ * of implicit resampling (USAC with reduced 3/4 core frame length).133
+ */134
+void qmfAnalysisFilteringSlot(135
+ HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Synthesis Bank */136
+ FIXP_DBL *qmfReal, /*!< Low and High band, real */137
+ FIXP_DBL *qmfImag, /*!< Low and High band, imag */138
+ const INT_PCM_QMFIN *RESTRICT timeIn, /*!< Pointer to input */139
+ const int stride, /*!< stride factor of input */140
+ FIXP_DBL *pWorkBuffer /*!< pointer to temporal working buffer */141
+) {142
+ int offset = anaQmf->no_channels * (QMF_NO_POLY * 2 - 1);143
+ /*144
+ Feed time signal into oldest anaQmf->no_channels states145
+ */146
+ {147
+ FIXP_QAS *FilterStatesAnaTmp = ((FIXP_QAS *)anaQmf->FilterStates) + offset;148
+149
+ /* Feed and scale actual time in slot */150
+ for (int i = anaQmf->no_channels >> 1; i != 0; i--) {151
+ /* Place INT_PCM value left aligned in scaledTimeIn */152
+ *FilterStatesAnaTmp++ = (FIXP_QAS)*timeIn;153
+ timeIn += stride;154
+ *FilterStatesAnaTmp++ = (FIXP_QAS)*timeIn;155
+ timeIn += stride;156
+ }157
+ }158
+159
+ if (anaQmf->flags & QMF_FLAG_NONSYMMETRIC) {160
+ qmfAnaPrototypeFirSlot_NonSymmetric(pWorkBuffer, anaQmf->no_channels,161
+ anaQmf->p_filter, anaQmf->p_stride,162
+ (FIXP_QAS *)anaQmf->FilterStates);163
+ } else {164
+ qmfAnaPrototypeFirSlot(pWorkBuffer, anaQmf->no_channels, anaQmf->p_filter,165
+ anaQmf->p_stride, (FIXP_QAS *)anaQmf->FilterStates);166
+ }167
+168
+ if (anaQmf->flags & QMF_FLAG_LP) {169
+ if (anaQmf->flags & QMF_FLAG_CLDFB)170
+ qmfForwardModulationLP_odd(anaQmf, pWorkBuffer, qmfReal);171
+ else172
+ qmfForwardModulationLP_even(anaQmf, pWorkBuffer, qmfReal);173
+174
+ } else {175
+ qmfForwardModulationHQ(anaQmf, pWorkBuffer, qmfReal, qmfImag);176
+ }177
+ /*178
+ Shift filter states179
+180
+ Should be realized with modulo addressing on a DSP instead of a true buffer181
+ shift182
+ */183
+ FDKmemmove(anaQmf->FilterStates,184
+ (FIXP_QAS *)anaQmf->FilterStates + anaQmf->no_channels,185
+ offset * sizeof(FIXP_QAS));186
+}187
+188
+/*!189
+ *190
+ * \brief Perform complex-valued subband filtering of the time domain191
+ * data of timeIn and stores the real part of the subband192
+ * samples in rAnalysis, and the imaginary part in iAnalysis193
+ * The qmf coefficient table is symmetric. The symmetry is expoited by194
+ * shrinking the coefficient table to half the size. The addressing mode195
+ * takes care of the symmetries.196
+ *197
+ *198
+ * \sa PolyphaseFiltering199
+ */200
+void qmfAnalysisFiltering(201
fdk-aac-2.0.1.tar.gz/libFDK/include/scale.h -> fdk-aac-2.0.2.tar.gz/libFDK/include/scale.h
Changed
36
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
#if (SAMPLE_BITS == 16)12
void scaleValues(FIXP_PCM *dst, const FIXP_DBL *src, INT len, INT scalefactor);13
#endif14
-void scaleValues(FIXP_PCM *dst, const FIXP_SGL *src, INT len, INT scalefactor);15
+void scaleValues(FIXP_SGL *dst, const FIXP_SGL *src, INT len, INT scalefactor);16
void scaleCplxValues(FIXP_DBL *r_dst, FIXP_DBL *i_dst, const FIXP_DBL *r_src,17
const FIXP_DBL *i_src, INT len, INT scalefactor);18
void scaleValuesWithFactor(FIXP_DBL *vector, FIXP_DBL factor, INT len,19
INT scalefactor);20
void scaleValuesSaturate(FIXP_DBL *vector, INT len, INT scalefactor);21
-void scaleValuesSaturate(FIXP_DBL *dst, FIXP_DBL *src, INT len,22
+void scaleValuesSaturate(FIXP_DBL *dst, const FIXP_DBL *src, INT len,23
INT scalefactor);24
-void scaleValuesSaturate(FIXP_SGL *dst, FIXP_DBL *src, INT len,25
+void scaleValuesSaturate(FIXP_SGL *dst, const FIXP_DBL *src, INT len,26
INT scalefactor);27
-void scaleValuesSaturate(INT_PCM *dst, FIXP_DBL *src, INT len, INT scalefactor);28
void scaleValuesSaturate(FIXP_SGL *vector, INT len, INT scalefactor);29
-void scaleValuesSaturate(FIXP_SGL *dst, FIXP_SGL *src, INT len,30
+void scaleValuesSaturate(FIXP_SGL *dst, const FIXP_SGL *src, INT len,31
INT scalefactor);32
-void scaleValuesSaturate(INT_PCM *dst, INT_PCM *src, INT len, INT scalefactor);33
INT getScalefactorShort(const SHORT *vector, INT len);34
INT getScalefactorPCM(const INT_PCM *vector, INT len, INT stride);35
INT getScalefactor(const FIXP_DBL *vector, INT len);36
fdk-aac-2.0.1.tar.gz/libFDK/include/scramble.h -> fdk-aac-2.0.2.tar.gz/libFDK/include/scramble.h
Changed
10
1
2
#if defined(__arm__)3
#include "arm/scramble_arm.h"4
5
-#elif defined(__mips__) && defined(__mips_dsp)6
+#elif defined(__mips__) && defined(__mips_dsp) && !defined(__mips16)7
#include "mips/scramble_mips.h"8
9
#endif10
fdk-aac-2.0.1.tar.gz/libFDK/src/FDK_bitbuffer.cpp -> fdk-aac-2.0.2.tar.gz/libFDK/src/FDK_bitbuffer.cpp
Changed
31
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
12
UINT bTotal = 0;13
14
- UINT bToRead = (hBitBuf->bufBits - hBitBuf->ValidBits) >> 3;15
+ UINT bToRead =16
+ fMin(hBitBuf->bufBits,17
+ (UINT)fMax(0, ((INT)hBitBuf->bufBits - (INT)hBitBuf->ValidBits))) >>18
+ 3;19
UINT noOfBytes =20
fMin(bToRead,21
*bytesValid); //(bToRead < *bytesValid) ? bToRead : *bytesValid ;22
23
bToRead * sizeof(UCHAR));24
25
/* add noOfBits to number of valid bits in buffer */26
- hBitBuf->ValidBits += bToRead << 3;27
+ hBitBuf->ValidBits = (UINT)((INT)hBitBuf->ValidBits + (INT)(bToRead << 3));28
bTotal += bToRead;29
inputBuffer += bToRead;30
31
fdk-aac-2.0.1.tar.gz/libFDK/src/FDK_core.cpp -> fdk-aac-2.0.2.tar.gz/libFDK/src/FDK_core.cpp
Changed
23
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
12
/* FDK tools library info */13
#define FDK_TOOLS_LIB_VL0 314
-#define FDK_TOOLS_LIB_VL1 015
+#define FDK_TOOLS_LIB_VL1 116
#define FDK_TOOLS_LIB_VL2 017
#define FDK_TOOLS_LIB_TITLE "FDK Tools"18
-#ifdef __ANDROID__19
+#ifdef SUPPRESS_BUILD_DATE_INFO20
#define FDK_TOOLS_LIB_BUILD_DATE ""21
#define FDK_TOOLS_LIB_BUILD_TIME ""22
#else23
fdk-aac-2.0.1.tar.gz/libFDK/src/FDK_decorrelate.cpp -> fdk-aac-2.0.2.tar.gz/libFDK/src/FDK_decorrelate.cpp
Changed
158
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
}12
13
#define DUCKER_MAX_NRG_SCALE (24)14
-#define DUCKER_HEADROOM_BITS (3)15
+#define DUCKER_HEADROOM_BITS (2)16
17
#define FILTER_SF (2)18
19
20
dataImagIn += start;21
dataRealOut += start;22
dataImagOut += start;23
-#ifdef FUNCTION_DecorrFilterApplyPASS_func124
- DecorrFilterApplyPASS_func1(i, dataRealIn, dataImagIn, dataRealOut,25
- dataImagOut, pDelayBuffer, offset);26
-#else27
do {28
FIXP_DBL delay_re, delay_im, real, imag;29
30
31
*dataImagOut++ = delay_im;32
pDelayBuffer += offset;33
} while (--i != 0);34
-#endif35
}36
}37
38
39
FIXP_DBL maxVal = FL2FXCONST_DBL(-1.0f);40
41
if (maxVal == FL2FXCONST_DBL(-1.0f)) {42
-#ifdef FUNCTION_DuckerCalcEnergy_func243
- maxVal = DuckerCalcEnergy_func2(inputReal, inputImag, startHybBand,44
- maxHybBand, maxHybridBand);45
-#else46
- FIXP_DBL localMaxVal = FL2FXCONST_DBL(0.0f);47
- for (qs = startHybBand; qs <= maxHybBand; qs++) {48
- localMaxVal |= fAbs(inputReal[qs]);49
- localMaxVal |= fAbs(inputImag[qs]);50
- }51
- for (; qs <= maxHybridBand; qs++) {52
- localMaxVal |= fAbs(inputReal[qs]);53
- }54
- maxVal = localMaxVal;55
-#endif56
+ clz = fMin(getScalefactor(&inputReal[startHybBand],57
+ fMax(0, maxHybridBand - startHybBand + 1)),58
+ getScalefactor(&inputImag[startHybBand],59
+ fMax(0, maxHybBand - startHybBand + 1)));60
+ } else {61
+ clz = CntLeadingZeros(maxVal) - 1;62
}63
64
- clz = fixMax(0, CntLeadingZeros(maxVal) - DUCKER_HEADROOM_BITS);65
- clz = fixMin(clz, DUCKER_MAX_NRG_SCALE);66
+ clz = fMin(fMax(0, clz - DUCKER_HEADROOM_BITS), DUCKER_MAX_NRG_SCALE);67
*nrgScale = (SCHAR)clz << 1;68
69
/* Initialize pb since it would stay uninitialized for the case startHybBand70
71
pb = SpatialDecGetProcessingBand(maxHybBand, self->mapHybBands2ProcBands);72
for (qs = startHybBand; qs <= maxHybBand; qs++) {73
pb = SpatialDecGetProcessingBand(qs, self->mapHybBands2ProcBands);74
- energy[pb] =75
- fAddSaturate(energy[pb], fPow2Div2(inputReal[qs] << clz) +76
- fPow2Div2(inputImag[qs] << clz));77
+ energy[pb] = SATURATE_LEFT_SHIFT(78
+ (energy[pb] >> 1) + (fPow2Div2(inputReal[qs] << clz) >> 1) +79
+ (fPow2Div2(inputImag[qs] << clz) >> 1),80
+ 1, DFRACT_BITS);81
}82
pb++;83
84
85
maxVal = inputMaxVal;86
87
if (maxVal == FL2FXCONST_DBL(-1.0f)) {88
-#ifdef FUNCTION_DuckerCalcEnergy_func289
- maxVal = DuckerCalcEnergy_func2(inputReal, inputImag, startHybBand,90
- maxHybBand, maxHybridBand);91
-#else92
- FIXP_DBL localMaxVal = FL2FXCONST_DBL(0.0f);93
- for (qs = startHybBand; qs <= maxHybBand; qs++) {94
- localMaxVal |= fAbs(inputReal[qs]);95
- localMaxVal |= fAbs(inputImag[qs]);96
- }97
- for (; qs <= maxHybridBand; qs++) {98
- localMaxVal |= fAbs(inputReal[qs]);99
- }100
- maxVal = localMaxVal;101
-#endif102
+ clz = fMin(getScalefactor(&inputReal[startHybBand],103
+ fMax(0, maxHybridBand - startHybBand + 1)),104
+ getScalefactor(&inputImag[startHybBand],105
+ fMax(0, maxHybBand - startHybBand + 1)));106
+ } else {107
+ clz = CntLeadingZeros(maxVal) - 1;108
}109
110
- clz = fixMax(0, CntLeadingZeros(maxVal) - DUCKER_HEADROOM_BITS);111
- clz = fixMin(clz, DUCKER_MAX_NRG_SCALE);112
+ clz = fMin(fMax(0, clz - DUCKER_HEADROOM_BITS), DUCKER_MAX_NRG_SCALE);113
*nrgScale = (SCHAR)clz << 1;114
115
-#ifdef FUNCTION_DuckerCalcEnergy_func4116
- DuckerCalcEnergy_func4(inputReal, inputImag, energy,117
- self->mapHybBands2ProcBands, clz, startHybBand,118
- maxHybBand, maxHybridBand);119
-#else120
for (qs = startHybBand; qs <= maxHybBand; qs++) {121
int pb = SpatialDecGetProcessingBand(qs, self->mapHybBands2ProcBands);122
- energy[pb] =123
- fAddSaturate(energy[pb], fPow2Div2(inputReal[qs] << clz) +124
- fPow2Div2(inputImag[qs] << clz));125
+ energy[pb] = SATURATE_LEFT_SHIFT(126
+ (energy[pb] >> 1) + (fPow2Div2(inputReal[qs] << clz) >> 1) +127
+ (fPow2Div2(inputImag[qs] << clz) >> 1),128
+ 1, DFRACT_BITS);129
}130
131
for (; qs <= maxHybridBand; qs++) {132
int pb = SpatialDecGetProcessingBand(qs, self->mapHybBands2ProcBands);133
energy[pb] = fAddSaturate(energy[pb], fPow2Div2(inputReal[qs] << clz));134
}135
-#endif /* FUNCTION_DuckerCalcEnergy_func4 */136
}137
138
{139
140
}141
}142
143
-#ifdef FUNCTION_DuckerApply_func1144
- qs = DuckerApply_func1(qs, hybBands, qs_next, outputReal, outputImag,145
- duckGain);146
-#else147
/* general gain*output section */148
if (qs < hybBands) { /* true for about 39% */149
for (; qs < qs_next; qs++) { /* runs about 2 times */150
151
outputReal[qs] = fMultDiv2(outputReal[qs], duckGain) << 2;152
}153
}154
-#endif155
} /* pb */156
157
self->headroomSmoothDirRevNrg =158
fdk-aac-2.0.1.tar.gz/libFDK/src/FDK_hybrid.cpp -> fdk-aac-2.0.2.tar.gz/libFDK/src/FDK_hybrid.cpp
Changed
70
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
} /* How to arrange the packed values. */12
13
struct FDK_HYBRID_SETUP {14
- UCHAR nrQmfBands; /*!< Number of QMF bands to be converted to hybrid. */15
- UCHAR nHybBands[3]; /*!< Number of Hybrid bands generated by nrQmfBands. */16
- SCHAR kHybrid[3]; /*!< Filter configuration of each QMF band. */17
- UCHAR protoLen; /*!< Prototype filter length. */18
- UCHAR filterDelay; /*!< Delay caused by hybrid filter. */19
+ UCHAR nrQmfBands; /*!< Number of QMF bands to be converted to hybrid. */20
+ UCHAR nHybBands[3]; /*!< Number of Hybrid bands generated by nrQmfBands. */21
+ UCHAR synHybScale[3]; /*!< Headroom needed in hybrid synthesis filterbank. */22
+ SCHAR kHybrid[3]; /*!< Filter configuration of each QMF band. */23
+ UCHAR protoLen; /*!< Prototype filter length. */24
+ UCHAR filterDelay; /*!< Delay caused by hybrid filter. */25
const INT26
*pReadIdxTable; /*!< Helper table to access input data ringbuffer. */27
};28
29
9, 10, 11, 12, 0, 1, 2, 3, 4,30
5, 6, 7, 8, 9, 10, 11, 12};31
32
-static const FDK_HYBRID_SETUP setup_3_16 = {3, {8, 4, 4}, {8, 4, 4},33
- 13, (13 - 1) / 2, ringbuffIdxTab};34
-static const FDK_HYBRID_SETUP setup_3_12 = {3, {8, 2, 2}, {8, 2, 2},35
- 13, (13 - 1) / 2, ringbuffIdxTab};36
-static const FDK_HYBRID_SETUP setup_3_10 = {3, {6, 2, 2}, {-8, -2, 2},37
- 13, (13 - 1) / 2, ringbuffIdxTab};38
+static const FDK_HYBRID_SETUP setup_3_16 = {39
+ 3, {8, 4, 4}, {4, 3, 3}, {8, 4, 4}, 13, (13 - 1) / 2, ringbuffIdxTab};40
+static const FDK_HYBRID_SETUP setup_3_12 = {41
+ 3, {8, 2, 2}, {4, 2, 2}, {8, 2, 2}, 13, (13 - 1) / 2, ringbuffIdxTab};42
+static const FDK_HYBRID_SETUP setup_3_10 = {43
+ 3, {6, 2, 2}, {3, 2, 2}, {-8, -2, 2}, 13, (13 - 1) / 2, ringbuffIdxTab};44
45
static const FIXP_HTP HybFilterCoef8[] = {46
HTCP(0x10000000, 0x00000000), HTCP(0x0df26407, 0xfa391882),47
48
*/49
for (k = 0; k < nrQmfBandsLF; k++) {50
const int nHybBands = hSynthesisHybFilter->pSetup->nHybBands[k];51
+ const int scale = hSynthesisHybFilter->pSetup->synHybScale[k];52
53
FIXP_DBL accu1 = FL2FXCONST_DBL(0.f);54
FIXP_DBL accu2 = FL2FXCONST_DBL(0.f);55
56
/* Perform hybrid filtering. */57
for (n = 0; n < nHybBands; n++) {58
- accu1 += pHybridReal[hybOffset + n];59
- accu2 += pHybridImag[hybOffset + n];60
+ accu1 += pHybridReal[hybOffset + n] >> scale;61
+ accu2 += pHybridImag[hybOffset + n] >> scale;62
}63
- pQmfReal[k] = accu1;64
- pQmfImag[k] = accu2;65
+ pQmfReal[k] = SATURATE_LEFT_SHIFT(accu1, scale, DFRACT_BITS);66
+ pQmfImag[k] = SATURATE_LEFT_SHIFT(accu2, scale, DFRACT_BITS);67
68
hybOffset += nHybBands;69
}70
fdk-aac-2.0.1.tar.gz/libFDK/src/FDK_qmf_domain.cpp -> fdk-aac-2.0.2.tar.gz/libFDK/src/FDK_qmf_domain.cpp
Changed
126
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
#include "common_fix.h"12
13
#define WORKBUFFER1_TAG 014
-#define WORKBUFFER2_TAG 115
-16
#define WORKBUFFER3_TAG 417
#define WORKBUFFER4_TAG 518
-#define WORKBUFFER5_TAG 619
#define WORKBUFFER6_TAG 720
+#define WORKBUFFER7_TAG 821
22
C_ALLOC_MEM_OVERLAY(QmfWorkBufferCore1, FIXP_DBL, QMF_WB_SECTION_SIZE,23
SECT_DATA_L1, WORKBUFFER1_TAG)24
-C_ALLOC_MEM_OVERLAY(QmfWorkBufferCore2, FIXP_DBL, QMF_WB_SECTION_SIZE,25
- SECT_DATA_L2, WORKBUFFER2_TAG)26
C_ALLOC_MEM_OVERLAY(QmfWorkBufferCore3, FIXP_DBL, QMF_WB_SECTION_SIZE,27
SECT_DATA_L2, WORKBUFFER3_TAG)28
C_ALLOC_MEM_OVERLAY(QmfWorkBufferCore4, FIXP_DBL, QMF_WB_SECTION_SIZE,29
SECT_DATA_L2, WORKBUFFER4_TAG)30
-C_ALLOC_MEM_OVERLAY(QmfWorkBufferCore5, FIXP_DBL, QMF_WB_SECTION_SIZE,31
- SECT_DATA_L2, WORKBUFFER5_TAG)32
C_ALLOC_MEM_OVERLAY(QmfWorkBufferCore6, FIXP_DBL, QMF_WB_SECTION_SIZE,33
SECT_DATA_L2, WORKBUFFER6_TAG)34
+C_ALLOC_MEM_OVERLAY(QmfWorkBufferCore7, FIXP_DBL, QMF_WB_SECTION_SIZE,35
+ SECT_DATA_L2, WORKBUFFER7_TAG)36
37
/*! Analysis states buffer. <br>38
Dimension: #((8) + (1)) */39
-C_ALLOC_MEM2(AnaQmfStates, FIXP_QAS, 10 * QMF_DOMAIN_MAX_ANALYSIS_QMF_BANDS,40
- ((8) + (1)))41
+C_AALLOC_MEM2(AnaQmfStates, FIXP_DBL, 10 * QMF_DOMAIN_MAX_ANALYSIS_QMF_BANDS,42
+ ((8) + (1)))43
44
/*! Synthesis states buffer. <br>45
Dimension: #((8) + (1)) */46
-C_ALLOC_MEM2(SynQmfStates, FIXP_QSS, 9 * QMF_DOMAIN_MAX_SYNTHESIS_QMF_BANDS,47
- ((8) + (1)))48
+C_AALLOC_MEM2(SynQmfStates, FIXP_QSS, 9 * QMF_DOMAIN_MAX_SYNTHESIS_QMF_BANDS,49
+ ((8) + (1)))50
51
/*! Pointer to real qmf data for each time slot. <br>52
Dimension: #((8) + (1)) */53
54
55
/*! Analysis states buffer. <br>56
Dimension: #((8) + (1)) */57
-C_ALLOC_MEM2(AnaQmfStates16, FIXP_QAS, 10 * QMF_DOMAIN_ANALYSIS_QMF_BANDS_16,58
- ((8) + (1)))59
-60
+C_AALLOC_MEM2(AnaQmfStates16, FIXP_DBL, 10 * QMF_DOMAIN_ANALYSIS_QMF_BANDS_16,61
+ ((8) + (1)))62
/*! Analysis states buffer. <br>63
Dimension: #((8) + (1)) */64
-C_ALLOC_MEM2(AnaQmfStates24, FIXP_QAS, 10 * QMF_DOMAIN_ANALYSIS_QMF_BANDS_24,65
- ((8) + (1)))66
+C_AALLOC_MEM2(AnaQmfStates24, FIXP_DBL, 10 * QMF_DOMAIN_ANALYSIS_QMF_BANDS_24,67
+ ((8) + (1)))68
69
/*! Analysis states buffer. <br>70
Dimension: #((8) + (1)) */71
-C_ALLOC_MEM2(AnaQmfStates32, FIXP_QAS, 10 * QMF_DOMAIN_ANALYSIS_QMF_BANDS_32,72
- ((8) + (1)))73
+C_AALLOC_MEM2(AnaQmfStates32, FIXP_DBL, 10 * QMF_DOMAIN_ANALYSIS_QMF_BANDS_32,74
+ ((8) + (1)))75
76
/*! Pointer to real qmf data for each time slot. <br>77
Dimension: #((8) + (1)) */78
79
80
if (pQmfOutImag == NULL) {81
for (; b < fMin(lsb, stop_band); b++) {82
- pQmfOutReal[b] = scaleValue(real[b], lb_sf);83
+ pQmfOutReal[b] = scaleValueSaturate(real[b], lb_sf);84
}85
for (; b < fMin(usb, stop_band); b++) {86
- pQmfOutReal[b] = scaleValue(real[b], hb_sf);87
+ pQmfOutReal[b] = scaleValueSaturate(real[b], hb_sf);88
}89
for (; b < stop_band; b++) {90
pQmfOutReal[b] = (FIXP_DBL)0;91
92
} else {93
FDK_ASSERT(imag != NULL);94
for (; b < fMin(lsb, stop_band); b++) {95
- pQmfOutReal[b] = scaleValue(real[b], lb_sf);96
- pQmfOutImag[b] = scaleValue(imag[b], lb_sf);97
+ pQmfOutReal[b] = scaleValueSaturate(real[b], lb_sf);98
+ pQmfOutImag[b] = scaleValueSaturate(imag[b], lb_sf);99
}100
for (; b < fMin(usb, stop_band); b++) {101
- pQmfOutReal[b] = scaleValue(real[b], hb_sf);102
- pQmfOutImag[b] = scaleValue(imag[b], hb_sf);103
+ pQmfOutReal[b] = scaleValueSaturate(real[b], hb_sf);104
+ pQmfOutImag[b] = scaleValueSaturate(imag[b], hb_sf);105
}106
for (; b < stop_band; b++) {107
pQmfOutReal[b] = (FIXP_DBL)0;108
109
110
if ((size > 4 * QMF_WB_SECTION_SIZE) && (pWorkBuffer[4] == NULL)) {111
/* get work buffer of size QMF_WB_SECTION_SIZE */112
- pWorkBuffer[4] = GetQmfWorkBufferCore5();113
+ pWorkBuffer[4] = GetQmfWorkBufferCore7();114
}115
116
/* 8. distribute workbuffer over processing channels */117
118
if (pWorkBuffer[1]) FreeQmfWorkBufferCore1(&pWorkBuffer[1]);119
if (pWorkBuffer[2]) FreeQmfWorkBufferCore3(&pWorkBuffer[2]);120
if (pWorkBuffer[3]) FreeQmfWorkBufferCore4(&pWorkBuffer[3]);121
- if (pWorkBuffer[4]) FreeQmfWorkBufferCore5(&pWorkBuffer[4]);122
+ if (pWorkBuffer[4]) FreeQmfWorkBufferCore7(&pWorkBuffer[4]);123
}124
125
void FDK_QmfDomain_FreeMem(HANDLE_FDK_QMF_DOMAIN hqd) {126
fdk-aac-2.0.1.tar.gz/libFDK/src/dct.cpp -> fdk-aac-2.0.2.tar.gz/libFDK/src/dct.cpp
Changed
37
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
for (i = 0; i < M - 1; i += 2, pDat_0 += 2, pDat_1 -= 2) {12
FIXP_DBL accu1, accu2, accu3, accu4;13
14
- accu1 = pDat_1[1];15
- accu2 = -pDat_0[0];16
- accu3 = pDat_0[1];17
- accu4 = -pDat_1[0];18
+ accu1 = pDat_1[1] >> 1;19
+ accu2 = -(pDat_0[0] >> 1);20
+ accu3 = pDat_0[1] >> 1;21
+ accu4 = -(pDat_1[0] >> 1);22
23
cplxMultDiv2(&accu1, &accu2, accu1, accu2, twiddle[i]);24
cplxMultDiv2(&accu3, &accu4, accu4, accu3, twiddle[i + 1]);25
26
- pDat_0[0] = accu2 >> 1;27
- pDat_0[1] = accu1 >> 1;28
- pDat_1[0] = accu4 >> 1;29
- pDat_1[1] = -(accu3 >> 1);30
+ pDat_0[0] = accu2;31
+ pDat_0[1] = accu1;32
+ pDat_1[0] = accu4;33
+ pDat_1[1] = -accu3;34
}35
if (M & 1) {36
FIXP_DBL accu1, accu2;37
fdk-aac-2.0.1.tar.gz/libFDK/src/fft_rad2.cpp -> fdk-aac-2.0.2.tar.gz/libFDK/src/fft_rad2.cpp
Changed
10
1
2
#if defined(__arm__)3
#include "arm/fft_rad2_arm.cpp"4
5
-#elif defined(__GNUC__) && defined(__mips__) && defined(__mips_dsp)6
+#elif defined(__GNUC__) && defined(__mips__) && defined(__mips_dsp) && !defined(__mips16)7
#include "mips/fft_rad2_mips.cpp"8
9
#endif10
fdk-aac-2.0.1.tar.gz/libFDK/src/fixpoint_math.cpp -> fdk-aac-2.0.2.tar.gz/libFDK/src/fixpoint_math.cpp
Changed
23
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
INT ans_lg2_e, baselg2_e;12
FIXP_DBL base_lg2, ans_lg2, result;13
14
+ if (base_m <= (FIXP_DBL)0) {15
+ result = (FIXP_DBL)0;16
+ *result_e = 0;17
+ return result;18
+ }19
+20
/* Calc log2 of base */21
base_lg2 = fLog2(base_m, base_e, &baselg2_e);22
23
fdk-aac-2.0.1.tar.gz/libFDK/src/mdct.cpp -> fdk-aac-2.0.2.tar.gz/libFDK/src/mdct.cpp
Changed
28
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
*/12
for (i = 0; i < hMdct->prev_nr; i++) {13
FIXP_DBL x = -(*pOvl--);14
- *pOut0 = IMDCT_SCALE_DBL(x + hMdct->pFacZir[i]);15
+ *pOut0 = fAddSaturate(x, IMDCT_SCALE_DBL(hMdct->pFacZir[i]));16
pOut0++;17
}18
hMdct->pFacZir = NULL;19
20
FIXP_DBL *pOut = pOut0 - fl / 2;21
FDK_ASSERT(fl / 2 <= 128);22
for (i = 0; i < fl / 2; i++) {23
- pOut[i] += IMDCT_SCALE_DBL(hMdct->pFacZir[i]);24
+ pOut[i] = fAddSaturate(pOut[i], IMDCT_SCALE_DBL(hMdct->pFacZir[i]));25
}26
hMdct->pFacZir = NULL;27
}28
fdk-aac-2.0.1.tar.gz/libFDK/src/mips/scale_mips.cpp -> fdk-aac-2.0.2.tar.gz/libFDK/src/mips/scale_mips.cpp
Changed
10
1
2
3
*******************************************************************************/4
5
-#if defined(__mips_dsp)6
+#if defined(__mips_dsp) && !defined(__mips16)7
8
#ifndef FUNCTION_getScalefactor_DBL9
#define FUNCTION_getScalefactor_DBL10
fdk-aac-2.0.1.tar.gz/libFDK/src/nlc_dec.cpp -> fdk-aac-2.0.2.tar.gz/libFDK/src/nlc_dec.cpp
Changed
37
1
2
static ERROR_t huff_decode(HANDLE_FDK_BITSTREAM strm, SCHAR* out_data_1,3
SCHAR* out_data_2, DATA_TYPE data_type,4
DIFF_TYPE diff_type_1, DIFF_TYPE diff_type_2,5
- int num_val, CODING_SCHEME* cdg_scheme, int ldMode) {6
+ int num_val, int* cdg_scheme, int ldMode) {7
ERROR_t err = HUFFDEC_OK;8
DIFF_TYPE diff_type;9
10
11
12
/* Coding scheme */13
data = FDKreadBits(strm, 1);14
- *cdg_scheme = (CODING_SCHEME)(data << PAIR_SHIFT);15
+ *cdg_scheme = (data << PAIR_SHIFT);16
17
if (*cdg_scheme >> PAIR_SHIFT == HUFF_2D) {18
if ((out_data_1 != NULL) && (out_data_2 != NULL) && (ldMode == 0)) {19
data = FDKreadBits(strm, 1);20
- *cdg_scheme = (CODING_SCHEME)(*cdg_scheme | data);21
+ *cdg_scheme = (*cdg_scheme | data);22
} else {23
- *cdg_scheme = (CODING_SCHEME)(*cdg_scheme | FREQ_PAIR);24
+ *cdg_scheme = (*cdg_scheme | FREQ_PAIR);25
}26
}27
28
29
SCHAR* pDataVec[2] = {NULL, NULL};30
31
DIFF_TYPE diff_type[2] = {DIFF_FREQ, DIFF_FREQ};32
- CODING_SCHEME cdg_scheme = HUFF_1D;33
+ int cdg_scheme = HUFF_1D;34
DIRECTION direction = BACKWARDS;35
36
switch (data_type) {37
fdk-aac-2.0.1.tar.gz/libFDK/src/qmf.cpp -> fdk-aac-2.0.2.tar.gz/libFDK/src/qmf.cpp
Changed
201
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
/* moved to qmf_pcm.h: -> qmfSynPrototypeFirSlot_NonSymmetric */12
/* moved to qmf_pcm.h: -> qmfSynthesisFilteringSlot */13
14
-#ifndef FUNCTION_qmfAnaPrototypeFirSlot15
-/*!16
- \brief Perform Analysis Prototype Filtering on a single slot of input data.17
-*/18
-static void qmfAnaPrototypeFirSlot(19
- FIXP_DBL *analysisBuffer,20
- INT no_channels, /*!< Number channels of analysis filter */21
- const FIXP_PFT *p_filter, INT p_stride, /*!< Stride of analysis filter */22
- FIXP_QAS *RESTRICT pFilterStates) {23
- INT k;24
-25
- FIXP_DBL accu;26
- const FIXP_PFT *RESTRICT p_flt = p_filter;27
- FIXP_DBL *RESTRICT pData_0 = analysisBuffer + 2 * no_channels - 1;28
- FIXP_DBL *RESTRICT pData_1 = analysisBuffer;29
-30
- FIXP_QAS *RESTRICT sta_0 = (FIXP_QAS *)pFilterStates;31
- FIXP_QAS *RESTRICT sta_1 =32
- (FIXP_QAS *)pFilterStates + (2 * QMF_NO_POLY * no_channels) - 1;33
- INT pfltStep = QMF_NO_POLY * (p_stride);34
- INT staStep1 = no_channels << 1;35
- INT staStep2 = (no_channels << 3) - 1; /* Rewind one less */36
-37
- /* FIR filters 127..64 0..63 */38
- for (k = 0; k < no_channels; k++) {39
- accu = fMultDiv2(p_flt[0], *sta_1);40
- sta_1 -= staStep1;41
- accu += fMultDiv2(p_flt[1], *sta_1);42
- sta_1 -= staStep1;43
- accu += fMultDiv2(p_flt[2], *sta_1);44
- sta_1 -= staStep1;45
- accu += fMultDiv2(p_flt[3], *sta_1);46
- sta_1 -= staStep1;47
- accu += fMultDiv2(p_flt[4], *sta_1);48
- *pData_1++ = (accu << 1);49
- sta_1 += staStep2;50
-51
- p_flt += pfltStep;52
- accu = fMultDiv2(p_flt[0], *sta_0);53
- sta_0 += staStep1;54
- accu += fMultDiv2(p_flt[1], *sta_0);55
- sta_0 += staStep1;56
- accu += fMultDiv2(p_flt[2], *sta_0);57
- sta_0 += staStep1;58
- accu += fMultDiv2(p_flt[3], *sta_0);59
- sta_0 += staStep1;60
- accu += fMultDiv2(p_flt[4], *sta_0);61
- *pData_0-- = (accu << 1);62
- sta_0 -= staStep2;63
- }64
-}65
-#endif /* !defined(FUNCTION_qmfAnaPrototypeFirSlot) */66
-67
-#ifndef FUNCTION_qmfAnaPrototypeFirSlot_NonSymmetric68
-/*!69
- \brief Perform Analysis Prototype Filtering on a single slot of input data.70
-*/71
-static void qmfAnaPrototypeFirSlot_NonSymmetric(72
- FIXP_DBL *analysisBuffer,73
- int no_channels, /*!< Number channels of analysis filter */74
- const FIXP_PFT *p_filter, int p_stride, /*!< Stride of analysis filter */75
- FIXP_QAS *RESTRICT pFilterStates) {76
- const FIXP_PFT *RESTRICT p_flt = p_filter;77
- int p, k;78
-79
- for (k = 0; k < 2 * no_channels; k++) {80
- FIXP_DBL accu = (FIXP_DBL)0;81
-82
- p_flt += QMF_NO_POLY * (p_stride - 1);83
-84
- /*85
- Perform FIR-Filter86
- */87
- for (p = 0; p < QMF_NO_POLY; p++) {88
- accu += fMultDiv2(*p_flt++, pFilterStates[2 * no_channels * p]);89
- }90
- analysisBuffer[2 * no_channels - 1 - k] = (accu << 1);91
- pFilterStates++;92
- }93
-}94
-#endif /* FUNCTION_qmfAnaPrototypeFirSlot_NonSymmetric */95
-96
/*!97
*98
* \brief Perform real-valued forward modulation of the time domain99
100
int i;101
int L = anaQmf->no_channels;102
int M = L >> 1;103
- int scale;104
+ int scale = 0;105
FIXP_DBL accu;106
107
const FIXP_DBL *timeInTmp1 = (FIXP_DBL *)&timeIn[3 * M];108
109
}110
#endif /* FUNCTION_qmfForwardModulationHQ */111
112
-/*113
- * \brief Perform one QMF slot analysis of the time domain data of timeIn114
- * with specified stride and stores the real part of the subband115
- * samples in rSubband, and the imaginary part in iSubband116
- *117
- * Note: anaQmf->lsb can be greater than anaQmf->no_channels in case118
- * of implicit resampling (USAC with reduced 3/4 core frame length).119
- */120
-#if (SAMPLE_BITS != DFRACT_BITS) && (QAS_BITS == DFRACT_BITS)121
-void qmfAnalysisFilteringSlot(122
- HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Synthesis Bank */123
- FIXP_DBL *qmfReal, /*!< Low and High band, real */124
- FIXP_DBL *qmfImag, /*!< Low and High band, imag */125
- const LONG *RESTRICT timeIn, /*!< Pointer to input */126
- const int stride, /*!< stride factor of input */127
- FIXP_DBL *pWorkBuffer /*!< pointer to temporal working buffer */128
-) {129
- int offset = anaQmf->no_channels * (QMF_NO_POLY * 2 - 1);130
- /*131
- Feed time signal into oldest anaQmf->no_channels states132
- */133
- {134
- FIXP_DBL *FilterStatesAnaTmp = ((FIXP_DBL *)anaQmf->FilterStates) + offset;135
-136
- /* Feed and scale actual time in slot */137
- for (int i = anaQmf->no_channels >> 1; i != 0; i--) {138
- /* Place INT_PCM value left aligned in scaledTimeIn */139
-140
- *FilterStatesAnaTmp++ = (FIXP_QAS)*timeIn;141
- timeIn += stride;142
- *FilterStatesAnaTmp++ = (FIXP_QAS)*timeIn;143
- timeIn += stride;144
- }145
- }146
-147
- if (anaQmf->flags & QMF_FLAG_NONSYMMETRIC) {148
- qmfAnaPrototypeFirSlot_NonSymmetric(pWorkBuffer, anaQmf->no_channels,149
- anaQmf->p_filter, anaQmf->p_stride,150
- (FIXP_QAS *)anaQmf->FilterStates);151
- } else {152
- qmfAnaPrototypeFirSlot(pWorkBuffer, anaQmf->no_channels, anaQmf->p_filter,153
- anaQmf->p_stride, (FIXP_QAS *)anaQmf->FilterStates);154
- }155
-156
- if (anaQmf->flags & QMF_FLAG_LP) {157
- if (anaQmf->flags & QMF_FLAG_CLDFB)158
- qmfForwardModulationLP_odd(anaQmf, pWorkBuffer, qmfReal);159
- else160
- qmfForwardModulationLP_even(anaQmf, pWorkBuffer, qmfReal);161
-162
- } else {163
- qmfForwardModulationHQ(anaQmf, pWorkBuffer, qmfReal, qmfImag);164
- }165
- /*166
- Shift filter states167
-168
- Should be realized with modulo adressing on a DSP instead of a true buffer169
- shift170
- */171
- FDKmemmove(anaQmf->FilterStates,172
- (FIXP_QAS *)anaQmf->FilterStates + anaQmf->no_channels,173
- offset * sizeof(FIXP_QAS));174
-}175
-#endif176
-177
-void qmfAnalysisFilteringSlot(178
- HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Synthesis Bank */179
- FIXP_DBL *qmfReal, /*!< Low and High band, real */180
- FIXP_DBL *qmfImag, /*!< Low and High band, imag */181
- const INT_PCM *RESTRICT timeIn, /*!< Pointer to input */182
- const int stride, /*!< stride factor of input */183
- FIXP_DBL *pWorkBuffer /*!< pointer to temporal working buffer */184
-) {185
- int offset = anaQmf->no_channels * (QMF_NO_POLY * 2 - 1);186
- /*187
- Feed time signal into oldest anaQmf->no_channels states188
- */189
- {190
- FIXP_QAS *FilterStatesAnaTmp = ((FIXP_QAS *)anaQmf->FilterStates) + offset;191
-192
- /* Feed and scale actual time in slot */193
- for (int i = anaQmf->no_channels >> 1; i != 0; i--) {194
- /* Place INT_PCM value left aligned in scaledTimeIn */195
-#if (QAS_BITS == SAMPLE_BITS)196
- *FilterStatesAnaTmp++ = (FIXP_QAS)*timeIn;197
- timeIn += stride;198
- *FilterStatesAnaTmp++ = (FIXP_QAS)*timeIn;199
- timeIn += stride;200
-#elif (QAS_BITS > SAMPLE_BITS)201
fdk-aac-2.0.1.tar.gz/libFDK/src/scale.cpp -> fdk-aac-2.0.2.tar.gz/libFDK/src/scale.cpp
Changed
55
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
*/12
#define FUNCTION_scaleValuesSaturate_DBL_DBL13
SCALE_INLINE14
-void scaleValuesSaturate(FIXP_DBL *dst, /*!< Output */15
- FIXP_DBL *src, /*!< Input */16
- INT len, /*!< Length */17
- INT scalefactor /*!< Scalefactor */18
+void scaleValuesSaturate(FIXP_DBL *dst, /*!< Output */19
+ const FIXP_DBL *src, /*!< Input */20
+ INT len, /*!< Length */21
+ INT scalefactor /*!< Scalefactor */22
) {23
INT i;24
25
26
*/27
#define FUNCTION_scaleValuesSaturate_SGL_DBL28
SCALE_INLINE29
-void scaleValuesSaturate(FIXP_SGL *dst, /*!< Output */30
- FIXP_DBL *src, /*!< Input */31
- INT len, /*!< Length */32
- INT scalefactor) /*!< Scalefactor */33
+void scaleValuesSaturate(FIXP_SGL *dst, /*!< Output */34
+ const FIXP_DBL *src, /*!< Input */35
+ INT len, /*!< Length */36
+ INT scalefactor) /*!< Scalefactor */37
{38
INT i;39
scalefactor = fixmax_I(fixmin_I(scalefactor, (INT)DFRACT_BITS - 1),40
41
*/42
#define FUNCTION_scaleValuesSaturate_SGL_SGL43
SCALE_INLINE44
-void scaleValuesSaturate(FIXP_SGL *dst, /*!< Output */45
- FIXP_SGL *src, /*!< Input */46
- INT len, /*!< Length */47
- INT scalefactor /*!< Scalefactor */48
+void scaleValuesSaturate(FIXP_SGL *dst, /*!< Output */49
+ const FIXP_SGL *src, /*!< Input */50
+ INT len, /*!< Length */51
+ INT scalefactor /*!< Scalefactor */52
) {53
INT i;54
55
fdk-aac-2.0.1.tar.gz/libMpegTPDec/src/tpdec_adts.cpp -> fdk-aac-2.0.2.tar.gz/libMpegTPDec/src/tpdec_adts.cpp
Changed
78
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
goto bail;12
}13
14
+ FDKcrcReset(&pAdts->crcInfo);15
if (!bs.protection_absent) {16
- FDKcrcReset(&pAdts->crcInfo);17
FDKpushBack(hBs, 56); /* complete fixed and variable header! */18
crcReg = FDKcrcStartReg(&pAdts->crcInfo, hBs, 0);19
FDKpushFor(hBs, 56);20
21
if (bs.channel_config == 0) {22
int pceBits = 0;23
UINT alignAnchor = FDKgetValidBits(hBs);24
+ CProgramConfig tmpPce;25
26
if (FDKreadBits(hBs, 3) == ID_PCE) {27
/* Got luck! Parse the PCE */28
crcReg = adtsRead_CrcStartReg(pAdts, hBs, 0);29
30
- CProgramConfig_Read(&pAsc->m_progrConfigElement, hBs, alignAnchor);31
+ CProgramConfig_Init(&tmpPce);32
+ CProgramConfig_Read(&tmpPce, hBs, alignAnchor);33
+34
+ if (CProgramConfig_IsValid(&tmpPce)) {35
+ if (CProgramConfig_IsValid(&oldPce)) {36
+ /* Compare the new and the old PCE (tags ignored) */37
+ switch (CProgramConfig_Compare(&tmpPce, &oldPce)) {38
+ case 0: /* Nothing to do because PCE matches the old one exactly. */39
+ case 1: /* Channel configuration not changed. Just new metadata. */40
+ FDKmemcpy(&pAsc->m_progrConfigElement, &tmpPce,41
+ sizeof(CProgramConfig));42
+ break;43
+ case 2: /* The number of channels are identical but not the config44
+ */45
+ case -1: /* The channel configuration is completely different */46
+ default:47
+ FDKmemcpy(&pAsc->m_progrConfigElement, &oldPce,48
+ sizeof(CProgramConfig));49
+ FDKpushBack(hBs, adtsHeaderLength);50
+ return TRANSPORTDEC_PARSE_ERROR;51
+ }52
+ } else {53
+ FDKmemcpy(&pAsc->m_progrConfigElement, &tmpPce,54
+ sizeof(CProgramConfig));55
+ }56
+ } else {57
+ if (CProgramConfig_IsValid(&oldPce)) {58
+ FDKmemcpy(&pAsc->m_progrConfigElement, &oldPce,59
+ sizeof(CProgramConfig));60
+ } else {61
+ FDKpushBack(hBs, adtsHeaderLength);62
+ return TRANSPORTDEC_PARSE_ERROR;63
+ }64
+ }65
66
adtsRead_CrcEndReg(pAdts, hBs, crcReg);67
- pceBits = alignAnchor - FDKgetValidBits(hBs);68
+ pceBits = (INT)alignAnchor - (INT)FDKgetValidBits(hBs);69
+ adtsHeaderLength += pceBits;70
+71
+ if (pceBits > (INT)alignAnchor) {72
+ goto bail;73
+ }74
+75
/* store the number of PCE bits */76
bs.num_pce_bits = pceBits;77
} else {78
fdk-aac-2.0.1.tar.gz/libMpegTPDec/src/tpdec_asc.cpp -> fdk-aac-2.0.2.tar.gz/libMpegTPDec/src/tpdec_asc.cpp
Changed
146
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
CSTpCallBacks *cb) {12
TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK;13
CSEldSpecificConfig *esc = &asc->m_sc.m_eldSpecificConfig;14
- ASC_ELD_EXT_TYPE eldExtType;15
+ UINT eldExtType;16
int eldExtLen, len, cnt, ldSbrLen = 0, eldExtLenSum, numSbrHeader = 0,17
- sbrIndex;18
+ sbrIndex, eldExtCnt = 0;19
20
unsigned char downscale_fill_nibble;21
22
23
eldExtLenSum = FDKgetValidBits(hBs);24
esc->m_downscaledSamplingFrequency = asc->m_samplingFrequency;25
/* parse ExtTypeConfigData */26
- while (27
- ((eldExtType = (ASC_ELD_EXT_TYPE)FDKreadBits(hBs, 4)) != ELDEXT_TERM) &&28
- ((INT)FDKgetValidBits(hBs) >= 0)) {29
+ while (((eldExtType = FDKreadBits(hBs, 4)) != ELDEXT_TERM) &&30
+ ((INT)FDKgetValidBits(hBs) >= 0) && (eldExtCnt++ < 15)) {31
eldExtLen = len = FDKreadBits(hBs, 4);32
if (len == 0xf) {33
len = FDKreadBits(hBs, 8);34
35
UCHAR tmpDownscaleFreqIdx;36
esc->m_downscaledSamplingFrequency =37
getSampleRate(hBs, &tmpDownscaleFreqIdx, 4);38
- if (esc->m_downscaledSamplingFrequency == 0) {39
+ if (esc->m_downscaledSamplingFrequency == 0 ||40
+ esc->m_downscaledSamplingFrequency > 96000) {41
return TRANSPORTDEC_PARSE_ERROR;42
}43
downscale_fill_nibble = FDKreadBits(hBs, 4);44
45
break;46
}47
}48
+ if (eldExtType != ELDEXT_TERM) {49
+ return TRANSPORTDEC_PARSE_ERROR;50
+ }51
52
if ((INT)FDKgetValidBits(hBs) < 0) {53
return TRANSPORTDEC_PARSE_ERROR;54
55
const AUDIO_OBJECT_TYPE aot) {56
TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK;57
58
- USAC_EXT_ELEMENT_TYPE usacExtElementType =59
- (USAC_EXT_ELEMENT_TYPE)escapedValue(hBs, 4, 8, 16);60
+ int usacExtElementType = escapedValue(hBs, 4, 8, 16);61
62
/* recurve extension elements which are invalid for USAC */63
if (aot == AOT_USAC) {64
65
}66
}67
68
- extElement->usacExtElementType = usacExtElementType;69
+ extElement->usacExtElementType = (USAC_EXT_ELEMENT_TYPE)usacExtElementType;70
int usacExtElementConfigLength = escapedValue(hBs, 4, 8, 16);71
extElement->usacExtElementConfigLength = (USHORT)usacExtElementConfigLength;72
INT bsAnchor;73
74
TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK;75
76
int numConfigExtensions;77
- CONFIG_EXT_ID usacConfigExtType;78
+ int usacConfigExtType;79
int usacConfigExtLength;80
81
numConfigExtensions = (int)escapedValue(hBs, 2, 4, 8) + 1;82
for (int confExtIdx = 0; confExtIdx < numConfigExtensions; confExtIdx++) {83
INT nbits;84
int loudnessInfoSetConfigExtensionPosition = FDKgetValidBits(hBs);85
- usacConfigExtType = (CONFIG_EXT_ID)escapedValue(hBs, 4, 8, 16);86
+ usacConfigExtType = escapedValue(hBs, 4, 8, 16);87
usacConfigExtLength = (int)escapedValue(hBs, 4, 8, 16);88
89
/* Start bit position of config extension */90
91
INT nbits = (INT)FDKgetValidBits(hBs);92
93
usacSamplingFrequency = getSampleRate(hBs, &asc->m_samplingFrequencyIndex, 5);94
+ if (usacSamplingFrequency == 0 || usacSamplingFrequency > 96000) {95
+ return TRANSPORTDEC_PARSE_ERROR;96
+ }97
asc->m_samplingFrequency = (UINT)usacSamplingFrequency;98
99
coreSbrFrameLengthIndex = FDKreadBits(hBs, 3);100
101
self->m_extensionSamplingFrequency = getSampleRate(102
bs, &self->m_extensionSamplingFrequencyIndex, 4);103
104
- if ((INT)self->m_extensionSamplingFrequency <= 0) {105
+ if (self->m_extensionSamplingFrequency == 0 ||106
+ self->m_extensionSamplingFrequency > 96000) {107
return TRANSPORTDEC_PARSE_ERROR;108
}109
}110
111
112
self->m_channelConfiguration = FDKreadBits(bs, 4);113
114
+ /* MPEG-04 standard ISO/IEC 14496-3: channelConfiguration == 0 is reserved115
+ in er_raw_data_block (table 4.19) and er_raw_data_block_eld (table 4.75)116
+ MPEG-04 conformance ISO/IEC 14496-4: channelConfiguration == 0 is not117
+ permitted for AOT_ER_AAC_LC, AOT_ER_AAC_LTP, AOT_ER_AAC_LD,118
+ AOT_ER_AAC_SCAL (chapter 6.6.4.1.2.1.1) */119
+ if ((self->m_channelConfiguration == 0) &&120
+ ((self->m_aot == AOT_ER_AAC_LC) || (self->m_aot == AOT_ER_AAC_LTP) ||121
+ (self->m_aot == AOT_ER_AAC_LD) || (self->m_aot == AOT_ER_AAC_SCAL) ||122
+ (self->m_aot == AOT_ER_AAC_ELD))) {123
+ return TRANSPORTDEC_UNSUPPORTED_FORMAT;124
+ }125
+ /* MPEG-04 conformance ISO/IEC 14496-4: channelConfiguration > 2 is not126
+ * permitted for AOT_AAC_SCAL and AOT_ER_AAC_SCAL (chapter 6.6.4.1.2.1.1) */127
+ if ((self->m_channelConfiguration > 2) &&128
+ ((self->m_aot == AOT_AAC_SCAL) || (self->m_aot == AOT_ER_AAC_SCAL))) {129
+ return TRANSPORTDEC_UNSUPPORTED_FORMAT;130
+ }131
+132
/* SBR extension ( explicit non-backwards compatible mode ) */133
self->m_sbrPresentFlag = 0;134
self->m_psPresentFlag = 0;135
136
137
self->m_extensionSamplingFrequency =138
getSampleRate(bs, &self->m_extensionSamplingFrequencyIndex, 4);139
+ if (self->m_extensionSamplingFrequency == 0 ||140
+ self->m_extensionSamplingFrequency > 96000) {141
+ return TRANSPORTDEC_PARSE_ERROR;142
+ }143
self->m_aot = getAOT(bs);144
145
switch (self->m_aot) {146
fdk-aac-2.0.1.tar.gz/libMpegTPDec/src/tpdec_latm.cpp -> fdk-aac-2.0.2.tar.gz/libMpegTPDec/src/tpdec_latm.cpp
Changed
23
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
}12
if (pLatmDemux->m_AudioMuxVersion == 1) {13
FDK_BITSTREAM tmpBs;14
- UINT ascLen = 0;15
+ INT ascLen = 0;16
ascLen = CLatmDemux_GetValue(bs);17
/* The ascLen could be wrong, so check if validBits<=bufBits*/18
- if (ascLen > FDKgetValidBits(bs)) {19
+ if (ascLen < 0 || ascLen > (INT)FDKgetValidBits(bs)) {20
ErrorStatus = TRANSPORTDEC_PARSE_ERROR;21
goto bail;22
}23
fdk-aac-2.0.1.tar.gz/libMpegTPDec/src/tpdec_lib.cpp -> fdk-aac-2.0.2.tar.gz/libMpegTPDec/src/tpdec_lib.cpp
Changed
130
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
UCHAR configChanged = 0;12
UCHAR configMode = AC_CM_DET_CFG_CHANGE;13
14
- UCHAR tmpConf[1024];15
+ UCHAR tmpConf[1024] = {0};16
if (length > 1024) {17
return TRANSPORTDEC_UNSUPPORTED_FORMAT;18
}19
20
21
for (int i = 0; i < 2; i++) {22
if (i > 0) {23
- FDKpushBack(hBs, newConfigLength * 8 - FDKgetValidBits(hBs));24
+ FDKpushBack(hBs,25
+ (INT)newConfigLength * 8 - (INT)FDKgetValidBits(hBs));26
configMode = AC_CM_ALLOC_MEM;27
}28
/* config transport decoder */29
30
if (*pBytesValid == 0) {31
/* nothing to do */32
return TRANSPORTDEC_OK;33
- }34
-35
- if (hTp->numberOfRawDataBlocks <= 0) {36
+ } else {37
+ const int bytesValid = *pBytesValid;38
FDKfeedBuffer(hBs, pBuffer, bufferSize, pBytesValid);39
+40
+ if (hTp->numberOfRawDataBlocks > 0) {41
+ hTp->globalFramePos += (bytesValid - *pBytesValid) * 8;42
+ hTp->accessUnitAnchor[layer] = FDKgetValidBits(hBs);43
+ }44
}45
}46
47
48
}49
}50
}51
+ /* if an error is detected terminate config parsing to avoid that an52
+ * invalid config is accepted in the second pass */53
+ if (err != TRANSPORTDEC_OK) {54
+ break;55
+ }56
}57
} else {58
/* Reset CRC because the next bits are the beginning of a59
60
CLatmDemux_GetNrOfSubFrames(&hTp->parser.latm);61
if (hTp->transportFmt == TT_MP4_LOAS) {62
syncLayerFrameBits -= startPos - (INT)FDKgetValidBits(hBs) - (13);63
+ if (syncLayerFrameBits <= 0) {64
+ err = TRANSPORTDEC_SYNC_ERROR;65
+ }66
}67
}68
} else {69
70
&rawDataBlockLength, &fTraverseMoreFrames,71
&syncLayerFrameBits, &fConfigFound,72
&headerBits);73
+ if (headerBits > bitsAvail) {74
+ err = (headerBits < (INT)hBs->hBitBuf.bufBits)75
+ ? TRANSPORTDEC_NOT_ENOUGH_BITS76
+ : TRANSPORTDEC_SYNC_ERROR;77
+ }78
if (TPDEC_IS_FATAL_ERROR(err)) {79
/* Rewind - TPDEC_SYNCSKIP, in order to look for a synch one bit ahead80
* next time. Ensure that the bit amount lands at a multiple of81
82
}83
84
if (err == TRANSPORTDEC_NOT_ENOUGH_BITS) {85
- /* Enforce reading of new data */86
- hTp->numberOfRawDataBlocks = 0;87
break;88
}89
90
91
if (!(hTp->flags & (TPDEC_LOST_FRAMES_PENDING | TPDEC_IGNORE_BUFFERFULLNESS |92
TPDEC_SYNCOK)) &&93
err == TRANSPORTDEC_OK) {94
- err = additionalHoldOffNeeded(hTp, transportDec_GetBufferFullness(hTp),95
- FDKgetValidBits(hBs) - syncLayerFrameBits);96
+ err =97
+ additionalHoldOffNeeded(hTp, transportDec_GetBufferFullness(hTp),98
+ (INT)FDKgetValidBits(hBs) - syncLayerFrameBits);99
if (err == TRANSPORTDEC_NOT_ENOUGH_BITS) {100
hTp->holdOffFrames++;101
}102
103
/* Rewind for retry because of not enough bits */104
if (err == TRANSPORTDEC_NOT_ENOUGH_BITS) {105
FDKpushBack(hBs, headerBits);106
+ hTp->numberOfRawDataBlocks = numRawDataBlocksPrevious;107
headerBits = 0;108
+ rawDataBlockLength = rawDataBlockLengthPrevious;109
} else {110
/* reset hold off frame counter */111
hTp->holdOffFrames = 0;112
113
114
for (i = 0; i < 2; i++) {115
if (i > 0) {116
- FDKpushBack(hBs, bsStart - FDKgetValidBits(hBs));117
+ FDKpushBack(hBs, bsStart - (INT)FDKgetValidBits(hBs));118
configMode = AC_CM_ALLOC_MEM;119
}120
121
122
info += i;123
124
info->module_id = FDK_TPDEC;125
-#ifdef __ANDROID__126
+#ifdef SUPPRESS_BUILD_DATE_INFO127
info->build_date = "";128
info->build_time = "";129
#else130
fdk-aac-2.0.1.tar.gz/libMpegTPEnc/src/tpenc_lib.cpp -> fdk-aac-2.0.2.tar.gz/libMpegTPEnc/src/tpenc_lib.cpp
Changed
10
1
2
info->module_id = FDK_TPENC;3
info->version = LIB_VERSION(TP_LIB_VL0, TP_LIB_VL1, TP_LIB_VL2);4
LIB_VERSION_STRING(info);5
-#ifdef __ANDROID__6
+#ifdef SUPPRESS_BUILD_DATE_INFO7
info->build_date = "";8
info->build_time = "";9
#else10
fdk-aac-2.0.1.tar.gz/libPCMutils/include/limiter.h -> fdk-aac-2.0.2.tar.gz/libPCMutils/include/limiter.h
Changed
67
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
#define TDL_ATTACK_DEFAULT_MS (15) /* default attack time in ms */12
#define TDL_RELEASE_DEFAULT_MS (50) /* default release time in ms */13
14
-#define TDL_GAIN_SCALING (15) /* scaling of gain value. */15
-16
#ifdef __cplusplus17
extern "C" {18
#endif19
20
unsigned int maxBufIdx, delayBufIdx;21
FIXP_DBL smoothState0;22
FIXP_DBL minGain;23
-24
- FIXP_DBL additionalGainPrev;25
- FIXP_DBL additionalGainFilterState;26
- FIXP_DBL additionalGainFilterState1;27
+ INT scaling;28
};29
30
typedef enum {31
32
33
/******************************************************************************34
* pcmLimiter_Apply *35
- * limiter: limiter handle *36
- * pGain : pointer to gains to be applied to the signal before limiting, *37
- * which are downscaled by TDL_GAIN_SCALING bit. *38
- * These gains are delayed by gain_delay, and smoothed. *39
- * Smoothing is done by a butterworth lowpass filter with a cutoff *40
- * frequency which is fixed with respect to the sampling rate. *41
- * It is a substitute for the smoothing due to windowing and *42
- * overlap/add, if a gain is applied in frequency domain. *43
- * gain_scale: pointer to scaling exponents to be applied to the signal before *44
- * limiting, without delay and without smoothing *45
- * gain_size: number of elements in pGain, currently restricted to 1 *46
- * gain_delay: delay [samples] with which the gains in pGain shall be applied *47
- * gain_delay <= nSamples *48
- * samples: input/output buffer containing interleaved samples *49
- * precision of output will be DFRACT_BITS-TDL_GAIN_SCALING bits *50
- * nSamples: number of samples per channel *51
+ * limiter: limiter handle *52
+ * samplesIn: pointer to input buffer containing interleaved samples *53
+ * samplesOut: pointer to output buffer containing interleaved samples *54
+ * pGainPerSample: pointer to gains for each sample *55
+ * scaling: scaling of output samples *56
+ * nSamples: number of samples per channel *57
* returns: error code *58
******************************************************************************/59
TDLIMITER_ERROR pcmLimiter_Apply(TDLimiterPtr limiter, PCM_LIM* samplesIn,60
- INT_PCM* samplesOut, FIXP_DBL* pGain,61
- const INT* gain_scale, const UINT gain_size,62
- const UINT gain_delay, const UINT nSamples);63
+ INT_PCM* samplesOut, FIXP_DBL* pGainPerSample,64
+ const INT scaling, const UINT nSamples);65
66
#endif /* #ifndef LIMITER_H */67
fdk-aac-2.0.1.tar.gz/libPCMutils/src/limiter.cpp -> fdk-aac-2.0.2.tar.gz/libPCMutils/src/limiter.cpp
Changed
201
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
limiter->attack = attack;12
limiter->attackConst = attackConst;13
limiter->releaseConst = releaseConst;14
- limiter->threshold = threshold >> TDL_GAIN_SCALING;15
+ limiter->threshold = threshold;16
limiter->channels = maxChannels;17
limiter->maxChannels = maxChannels;18
limiter->sampleRate = maxSampleRate;19
20
21
/* apply limiter */22
TDLIMITER_ERROR pcmLimiter_Apply(TDLimiterPtr limiter, PCM_LIM* samplesIn,23
- INT_PCM* samplesOut, FIXP_DBL* RESTRICT pGain,24
- const INT* RESTRICT gain_scale,25
- const UINT gain_size, const UINT gain_delay,26
- const UINT nSamples) {27
+ INT_PCM* samplesOut, FIXP_DBL* pGainPerSample,28
+ const INT scaling, const UINT nSamples) {29
unsigned int i, j;30
- FIXP_DBL tmp1;31
FIXP_DBL tmp2;32
- FIXP_DBL tmp, old, gain, additionalGain = 0, additionalGainUnfiltered;33
+ FIXP_DBL tmp, old, gain, additionalGain = 0;34
FIXP_DBL minGain = FL2FXCONST_DBL(1.0f / (1 << 1));35
-36
- FDK_ASSERT(gain_size == 1);37
- FDK_ASSERT(gain_delay <= nSamples);38
+ UINT additionalGainAvailable = 1;39
40
if (limiter == NULL) return TDLIMIT_INVALID_HANDLE;41
42
43
unsigned int attack = limiter->attack;44
FIXP_DBL attackConst = limiter->attackConst;45
FIXP_DBL releaseConst = limiter->releaseConst;46
- FIXP_DBL threshold = limiter->threshold;47
+ FIXP_DBL threshold = limiter->threshold >> scaling;48
49
FIXP_DBL max = limiter->max;50
FIXP_DBL* maxBuf = limiter->maxBuf;51
52
unsigned int delayBufIdx = limiter->delayBufIdx;53
54
FIXP_DBL smoothState0 = limiter->smoothState0;55
- FIXP_DBL additionalGainSmoothState = limiter->additionalGainFilterState;56
- FIXP_DBL additionalGainSmoothState1 = limiter->additionalGainFilterState1;57
58
- if (!gain_delay) {59
- additionalGain = pGain[0];60
- if (gain_scale[0] > 0) {61
- additionalGain <<= gain_scale[0];62
- } else {63
- additionalGain >>= -gain_scale[0];64
- }65
+ if (limiter->scaling != scaling) {66
+ scaleValuesSaturate(delayBuf, attack * channels,67
+ limiter->scaling - scaling);68
+ scaleValuesSaturate(maxBuf, attack + 1, limiter->scaling - scaling);69
+ max = scaleValueSaturate(max, limiter->scaling - scaling);70
+ limiter->scaling = scaling;71
}72
73
- for (i = 0; i < nSamples; i++) {74
- if (gain_delay) {75
- if (i < gain_delay) {76
- additionalGainUnfiltered = limiter->additionalGainPrev;77
- } else {78
- additionalGainUnfiltered = pGain[0];79
- }80
+ if (pGainPerSample == NULL) {81
+ additionalGainAvailable = 0;82
+ }83
84
- /* Smooth additionalGain */85
- /* [b,a] = butter(1, 0.01) */86
- static const FIXP_SGL b[] = {FL2FXCONST_SGL(0.015466 * 2.0),87
- FL2FXCONST_SGL(0.015466 * 2.0)};88
- static const FIXP_SGL a[] = {(FIXP_SGL)MAXVAL_SGL,89
- FL2FXCONST_SGL(-0.96907)};90
- additionalGain = -fMult(additionalGainSmoothState, a[1]) +91
- fMultDiv2(additionalGainUnfiltered, b[0]) +92
- fMultDiv2(additionalGainSmoothState1, b[1]);93
- additionalGainSmoothState1 = additionalGainUnfiltered;94
- additionalGainSmoothState = additionalGain;95
-96
- /* Apply the additional scaling that has no delay and no smoothing */97
- if (gain_scale[0] > 0) {98
- additionalGain <<= gain_scale[0];99
- } else {100
- additionalGain >>= -gain_scale[0];101
- }102
- }103
+ for (i = 0; i < nSamples; i++) {104
/* get maximum absolute sample value of all channels, including the105
* additional gain. */106
- tmp1 = (FIXP_DBL)0;107
+ tmp = (FIXP_DBL)0;108
for (j = 0; j < channels; j++) {109
tmp2 = PCM_LIM2FIXP_DBL(samplesIn[j]);110
- tmp2 = fAbs(tmp2);111
- tmp2 = FIXP_DBL(INT(tmp2) ^ INT((tmp2 >> (SAMPLE_BITS_LIM - 1))));112
- tmp1 = fMax(tmp1, tmp2);113
+ tmp2 =114
+ (tmp2 == (FIXP_DBL)MINVAL_DBL) ? (FIXP_DBL)MAXVAL_DBL : fAbs(tmp2);115
+ tmp = fMax(tmp, tmp2);116
+ }117
+118
+ if (additionalGainAvailable) {119
+ additionalGain = pGainPerSample[i];120
+ tmp = fMult(tmp, additionalGain);121
}122
- tmp = fMult(tmp1, additionalGain);123
124
/* set threshold as lower border to save calculations in running maximum125
* algorithm */126
127
/* lookahead delay, apply gain */128
for (j = 0; j < channels; j++) {129
tmp = p_delayBuf[j];130
- p_delayBuf[j] = fMult((FIXP_PCM_LIM)samplesIn[j], additionalGain);131
+132
+ if (additionalGainAvailable) {133
+ p_delayBuf[j] = fMult((FIXP_PCM_LIM)samplesIn[j], additionalGain);134
+ } else {135
+ p_delayBuf[j] = PCM_LIM2FIXP_DBL(samplesIn[j]);136
+ }137
138
/* Apply gain to delayed signal */139
tmp = fMultDiv2(tmp, gain);140
-141
+#if (SAMPLE_BITS == DFRACT_BITS)142
+ samplesOut[j] = (INT_PCM)FX_DBL2FX_PCM(143
+ (FIXP_DBL)SATURATE_LEFT_SHIFT(tmp, scaling + 1, DFRACT_BITS));144
+#else145
samplesOut[j] = (INT_PCM)FX_DBL2FX_PCM((FIXP_DBL)SATURATE_LEFT_SHIFT(146
- tmp, TDL_GAIN_SCALING + 1, DFRACT_BITS));147
+ tmp + ((FIXP_DBL)0x8000 >> (scaling + 1)), scaling + 1,148
+ DFRACT_BITS));149
+#endif150
}151
gain >>= 1;152
} else {153
/* lookahead delay, apply gain=1.0f */154
for (j = 0; j < channels; j++) {155
tmp = p_delayBuf[j];156
- p_delayBuf[j] = fMult((FIXP_PCM_LIM)samplesIn[j], additionalGain);157
+ if (additionalGainAvailable) {158
+ p_delayBuf[j] = fMult((FIXP_PCM_LIM)samplesIn[j], additionalGain);159
+ } else {160
+ p_delayBuf[j] = PCM_LIM2FIXP_DBL(samplesIn[j]);161
+ }162
+163
+#if (SAMPLE_BITS == DFRACT_BITS)164
+ samplesOut[j] = (INT_PCM)FX_DBL2FX_PCM(165
+ (FIXP_DBL)SATURATE_LEFT_SHIFT(tmp, scaling, DFRACT_BITS));166
+#else167
samplesOut[j] = (INT_PCM)FX_DBL2FX_PCM((FIXP_DBL)SATURATE_LEFT_SHIFT(168
- tmp, TDL_GAIN_SCALING, DFRACT_BITS));169
+ tmp + ((FIXP_DBL)0x8000 >> scaling), scaling, DFRACT_BITS));170
+#endif171
}172
}173
174
175
limiter->delayBufIdx = delayBufIdx;176
177
limiter->smoothState0 = smoothState0;178
- limiter->additionalGainFilterState = additionalGainSmoothState;179
- limiter->additionalGainFilterState1 = additionalGainSmoothState1;180
181
limiter->minGain = minGain;182
183
- limiter->additionalGainPrev = pGain[0];184
-185
return TDLIMIT_OK;186
}187
}188
189
FIXP_DBL threshold) {190
if (limiter == NULL) return TDLIMIT_INVALID_HANDLE;191
192
- limiter->threshold = threshold >> TDL_GAIN_SCALING;193
+ limiter->threshold = threshold;194
195
return TDLIMIT_OK;196
}197
198
limiter->cor = FL2FXCONST_DBL(1.0f / (1 << 1));199
limiter->smoothState0 = FL2FXCONST_DBL(1.0f / (1 << 1));200
limiter->minGain = FL2FXCONST_DBL(1.0f / (1 << 1));201
fdk-aac-2.0.1.tar.gz/libPCMutils/src/pcmdmx_lib.cpp -> fdk-aac-2.0.2.tar.gz/libPCMutils/src/pcmdmx_lib.cpp
Changed
9
1
2
3
if (channelIndices[ch] >= numCh[channelType[ch] >> 4][chGrp])4
return PCMDMX_INVALID_CH_CONFIG;5
-6
spkrPos[ch] = getSpeakerPos(channelType[ch], channelIndices[ch],7
numCh[channelType[ch] >> 4][chGrp]);8
9
fdk-aac-2.0.1.tar.gz/libPCMutils/src/version.h -> fdk-aac-2.0.2.tar.gz/libPCMutils/src/version.h
Changed
23
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
12
/* library info */13
#define PCMUTIL_LIB_VL0 314
-#define PCMUTIL_LIB_VL1 015
+#define PCMUTIL_LIB_VL1 116
#define PCMUTIL_LIB_VL2 017
#define PCMUTIL_LIB_TITLE "PCM Utility Lib"18
-#ifdef __ANDROID__19
+#ifdef SUPPRESS_BUILD_DATE_INFO20
#define PCMUTIL_LIB_BUILD_DATE ""21
#define PCMUTIL_LIB_BUILD_TIME ""22
#else23
fdk-aac-2.0.1.tar.gz/libSACdec/include/sac_dec_lib.h -> fdk-aac-2.0.2.tar.gz/libSACdec/include/sac_dec_lib.h
Changed
44
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
scenario. Default parameter value is 3 frames. */12
} SACDEC_PARAM;13
14
-#define PCM_MPS INT_PCM15
+#define PCM_MPS LONG16
17
/**18
* \brief MPEG Surround decoder handle.19
20
* for each output audio channel is stored into.21
* \param mapDescr Channep map descriptor for output channel mapping22
* to be used (From MPEG PCE ordering to whatever is required).23
+ * \param inDataHeadroom Headroom of SAC input time signal to prevent24
+ * clipping.25
+ * \param outDataHeadroom Pointer to headroom of SAC output time signal to26
+ * prevent clipping.27
*28
* \return Error code.29
*/30
int mpegSurroundDecoder_Apply(CMpegSurroundDecoder *pMpegSurroundDecoder,31
- INT_PCM *input, PCM_MPS *pTimeData,32
+ PCM_MPS *input, PCM_MPS *pTimeData,33
const int timeDataSize, int timeDataFrameSize,34
int *nChannels, int *frameSize, int sampleRate,35
AUDIO_OBJECT_TYPE coreCodec,36
AUDIO_CHANNEL_TYPE channelType[],37
UCHAR channelIndices[],38
- const FDK_channelMapDescr *const mapDescr);39
+ const FDK_channelMapDescr *const mapDescr,40
+ const INT inDataHeadroom, INT *outDataHeadroom);41
42
/**43
* \brief Deallocate a MPEG Surround decoder instance.44
fdk-aac-2.0.1.tar.gz/libSACdec/src/sac_bitdec.cpp -> fdk-aac-2.0.2.tar.gz/libSACdec/src/sac_bitdec.cpp
Changed
99
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
int bsFreqRes, b3DaudioMode = 0;12
int numHeaderBits;13
int cfgStartPos, bitsAvailable;14
+ int treeConfig;15
16
FDKmemclear(pSpatialSpecificConfig, sizeof(SPATIAL_SPECIFIC_CONFIG));17
18
19
pSpatialSpecificConfig->freqRes =20
(SPATIALDEC_FREQ_RES)freqResTable_LD[bsFreqRes];21
22
- pSpatialSpecificConfig->treeConfig =23
- (SPATIALDEC_TREE_CONFIG)FDKreadBits(bitstream, 4);24
+ treeConfig = FDKreadBits(bitstream, 4);25
26
- if (pSpatialSpecificConfig->treeConfig != SPATIALDEC_MODE_RSVD7) {27
+ if (treeConfig != SPATIALDEC_MODE_RSVD7) {28
err = MPS_UNSUPPORTED_CONFIG;29
goto bail;30
}31
+ pSpatialSpecificConfig->treeConfig = (SPATIALDEC_TREE_CONFIG) treeConfig;32
33
{34
pSpatialSpecificConfig->nOttBoxes =35
36
FIXP_DBL (*pOttVsTotDb1)[MAX_PARAMETER_SETS][MAX_PARAMETER_BANDS],37
FIXP_DBL (*pOttVsTotDb2)[MAX_PARAMETER_SETS][MAX_PARAMETER_BANDS]) {38
int aParamSlots[MAX_PARAMETER_SETS];39
- int aInterpolate[MAX_PARAMETER_SETS];40
+ int aInterpolate[MAX_PARAMETER_SETS] = {0};41
42
int dataSets;43
int aMap[MAX_PARAMETER_BANDS + 1];44
45
/* Interpolate */46
i1 = 0;47
for (i = 0; i < numParameterSets; i++) {48
- int xi, i2, x1, x2;49
-50
if (aInterpolate[i] != 1) {51
i1 = i;52
- }53
- i2 = i;54
- while (aInterpolate[i2] == 1) {55
- i2++;56
- }57
- x1 = paramSlot[i1];58
- xi = paramSlot[i];59
- x2 = paramSlot[i2];60
+ } else {61
+ int xi, i2, x1, x2;62
63
- if (aInterpolate[i] == 1) {64
+ for (i2 = i; i2 < numParameterSets; i2++) {65
+ if (aInterpolate[i2] != 1) break;66
+ }67
if (i2 >= numParameterSets) return MPS_WRONG_PARAMETERSETS;68
+69
+ x1 = paramSlot[i1];70
+ xi = paramSlot[i];71
+ x2 = paramSlot[i2];72
+73
for (band = startBand; band < stopBand; band++) {74
int yi, y1, y2;75
y1 = outputIdxData[xttIdx][i1][band];76
77
for (ps = 0; ps < numParameterSets; ps++) {78
if (quantMode && (paramType == t_CLD)) {79
if (pOttVsTotDbIn == 0) return MPS_WRONG_OTT;80
- if ((pOttVsTotDb1 == 0) && (ottVsTotDbMode == ottVsTotDb1Activ))81
+ if ((pOttVsTotDb1 == 0) && (ottVsTotDbMode & ottVsTotDb1Activ))82
return MPS_WRONG_OTT;83
- if ((pOttVsTotDb2 == 0) && (ottVsTotDbMode == ottVsTotDb2Activ))84
+ if ((pOttVsTotDb2 == 0) && (ottVsTotDbMode & ottVsTotDb2Activ))85
return MPS_WRONG_OTT;86
87
for (pb = startBand; pb < stopBand; pb++) {88
89
} /* for( i = 0 ; i < numParameterSets; i++ ) */90
91
if (extendFrame) {92
+ if (paramType == t_IPD) {93
+ llData->bsQuantCoarseXXX[numParameterSets] =94
+ llData->bsQuantCoarseXXX[numParameterSets - 1];95
+ }96
for (band = startBand; band < stopBand; band++) {97
outputDataIdx[xttIdx][numParameterSets][band] =98
outputDataIdx[xttIdx][numParameterSets - 1][band];99
fdk-aac-2.0.1.tar.gz/libSACdec/src/sac_calcM1andM2.cpp -> fdk-aac-2.0.2.tar.gz/libSACdec/src/sac_calcM1andM2.cpp
Changed
108
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
FL2FXCONST_DBL(0.888178419700125),12
};13
14
-static FIXP_DBL dequantIPD_CLD_ICC_splitAngle__FDK_Function(INT ipdIdx,15
- INT cldIdx,16
- INT iccIdx) {17
- FIXP_DBL cld;18
- SpatialDequantGetCLD2Values(cldIdx, &cld);19
-20
- /*const FIXP_DBL one_m = (FIXP_DBL)MAXVAL_DBL;21
- const int one_e = 0;*/22
- const FIXP_DBL one_m = FL2FXCONST_DBL(0.5f);23
- const int one_e = 1;24
- /* iidLin = sqrt(cld); */25
- FIXP_DBL iidLin_m = sqrt_CLD_m[cldIdx];26
- int iidLin_e = sqrt_CLD_e[cldIdx];27
- /* iidLin2 = cld; */28
- FIXP_DBL iidLin2_m = CLD_m[cldIdx];29
- int iidLin2_e = sqrt_CLD_e[cldIdx] << 1;30
- /* iidLin21 = iidLin2 + 1.0f; */31
- int iidLin21_e;32
- FIXP_DBL iidLin21_m =33
- fAddNorm(iidLin2_m, iidLin2_e, one_m, one_e, &iidLin21_e);34
- /* iidIcc2 = iidLin * icc * 2.0f; */35
- FIXP_CFG icc = dequantICC__FDK[iccIdx];36
- FIXP_DBL temp1_m, temp1c_m;37
- int temp1_e, temp1c_e;38
- temp1_m = fMult(iidLin_m, icc);39
- temp1_e = iidLin_e + 1;40
-41
- FIXP_DBL cosIpd, sinIpd;42
- cosIpd = COS_IPD(ipdIdx);43
- sinIpd = SIN_IPD(ipdIdx);44
-45
- temp1c_m = fMult(temp1_m, cosIpd);46
- temp1c_e = temp1_e; //+cosIpd_e;47
-48
- int temp2_e, temp3_e, inv_temp3_e, ratio_e;49
- FIXP_DBL temp2_m =50
- fAddNorm(iidLin21_m, iidLin21_e, temp1c_m, temp1c_e, &temp2_e);51
- FIXP_DBL temp3_m =52
- fAddNorm(iidLin21_m, iidLin21_e, temp1_m, temp1_e, &temp3_e);53
- /* calculate 1/temp3 needed later */54
- inv_temp3_e = temp3_e;55
- FIXP_DBL inv_temp3_m = invFixp(temp3_m, &inv_temp3_e);56
- FIXP_DBL ratio_m =57
- fAddNorm(fMult(inv_temp3_m, temp2_m), (inv_temp3_e + temp2_e),58
- FL2FXCONST_DBL(1e-9f), 0, &ratio_e);59
-60
- int weight2_e, tempb_atan2_e;61
- FIXP_DBL weight2_m =62
- fPow(ratio_m, ratio_e, FL2FXCONST_DBL(0.5f), -1, &weight2_e);63
- /* atan2(w2*sinIpd, w1*iidLin + w2*cosIpd) = atan2(w2*sinIpd, (2 - w2)*iidLin64
- * + w2*cosIpd) = atan2(w2*sinIpd, 2*iidLin + w2*(cosIpd - iidLin)); */65
- /* tmpa_atan2 = w2*sinIpd; tmpb_atan2 = 2*iidLin + w2*(cosIpd - iidLin); */66
- FIXP_DBL tempb_atan2_m = iidLin_m;67
- tempb_atan2_e = iidLin_e + 1;68
- int add_tmp1_e = 0;69
- FIXP_DBL add_tmp1_m = fAddNorm(cosIpd, 0, -iidLin_m, iidLin_e, &add_tmp1_e);70
- FIXP_DBL add_tmp2_m = fMult(add_tmp1_m, weight2_m);71
- int add_tmp2_e = add_tmp1_e + weight2_e;72
- tempb_atan2_m = fAddNorm(tempb_atan2_m, tempb_atan2_e, add_tmp2_m, add_tmp2_e,73
- &tempb_atan2_e);74
-75
- FIXP_DBL tempa_atan2_m = fMult(weight2_m, sinIpd);76
- int tempa_atan2_e = weight2_e; // + sinIpd_e;77
-78
- if (tempa_atan2_e > tempb_atan2_e) {79
- tempb_atan2_m = (tempb_atan2_m >> (tempa_atan2_e - tempb_atan2_e));80
- tempb_atan2_e = tempa_atan2_e;81
- } else if (tempb_atan2_e > tempa_atan2_e) {82
- tempa_atan2_m = (tempa_atan2_m >> (tempb_atan2_e - tempa_atan2_e));83
- }84
-85
- return fixp_atan2(tempa_atan2_m, tempb_atan2_m);86
-}87
-88
static void calculateOpd(spatialDec* self, INT ottBoxIndx, INT parameterSetIndx,89
FIXP_DBL opd[MAX_PARAMETER_BANDS]) {90
INT band;91
92
SpatialDequantGetCLD2Values(idxCld, &cld);93
94
/* ipd(idxIpd==8) == PI */95
- if ((cld == FL2FXCONST_DBL(0.0f)) && (idxIpd == 8)) {96
+ if (((cld == FL2FXCONST_DBL(0.0f)) && (idxIpd == 8)) || (idxIpd == 0)) {97
opd[2 * band] = FL2FXCONST_DBL(0.0f);98
} else {99
- opd[2 * band] = (dequantIPD_CLD_ICC_splitAngle__FDK_Function(100
- idxIpd, idxCld, idxIcc) >>101
- (IPD_SCALE - AT2O_SF));102
+ FDK_ASSERT(idxIpd > 0);103
+ opd[2 * band] =104
+ dequantIPD_CLD_ICC_splitAngle__FDK[idxIpd - 1][idxCld][idxIcc];105
}106
opd[2 * band + 1] = opd[2 * band] - ipd;107
}108
fdk-aac-2.0.1.tar.gz/libSACdec/src/sac_calcM1andM2.h -> fdk-aac-2.0.2.tar.gz/libSACdec/src/sac_calcM1andM2.h
Changed
20
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
/* Scaling of spectral data after applying M2 matrix, but only for binaural12
upmix type Scaling is compensated later in synthesis qmf filterbank */13
#define SCALE_DATA_APPLY_M2 (1)14
+/* Applying M2 parameter in combination with phase coding needs 2 bits headroom15
+ * because up to a maximum of 4 spectral values can be added for USAC */16
+#define SCALE_DATA_APPLY_M2_PC (2)17
18
SACDEC_ERROR initM1andM2(spatialDec* self, int initStatesFlag,19
int configChanged);20
fdk-aac-2.0.1.tar.gz/libSACdec/src/sac_dec.cpp -> fdk-aac-2.0.2.tar.gz/libSACdec/src/sac_dec.cpp
Changed
107
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
12
/* parameter smoothing tool set to off */13
bsFrame->bsSmoothMode[0] = 0;14
+ initParameterSmoothing(self);15
16
/* reset residual data */17
{18
19
20
/* output scaling */21
for (nCh = 0; nCh < self->numOutputChannelsAT; nCh++) {22
- int outputScale = 0, outputGain_e = 0, scale = 0;23
+ int outputScale = 0, outputGain_e = 0, scale = -(8) + (1);24
FIXP_DBL outputGain_m = getChGain(self, nCh, &outputGain_e);25
26
if (!isTwoChMode(self->upmixType) && !bypassMode) {27
28
synthesis qmf */29
}30
31
- scale = outputScale;32
+ scale += outputScale;33
34
qmfChangeOutScalefactor(&self->pQmfDomain->QmfDomainOut[nCh].fb, scale);35
qmfChangeOutGain(&self->pQmfDomain->QmfDomainOut[nCh].fb, outputGain_m,36
37
!(self->stereoConfigIndex == 3)) {38
for (i = 0; i < self->qmfBands; i++) {39
self_qmfResidualReal__FDK_0_0[i] =40
- fMult(self_qmfResidualReal__FDK_0_0[i] << 1,41
+ fMult(scaleValueSaturate(self_qmfResidualReal__FDK_0_0[i],42
+ 1 + self->sacInDataHeadroom - (1)),43
self->clipProtectGain__FDK);44
self_qmfResidualImag__FDK_0_0[i] =45
- fMult(self_qmfResidualImag__FDK_0_0[i] << 1,46
+ fMult(scaleValueSaturate(self_qmfResidualImag__FDK_0_0[i],47
+ 1 + self->sacInDataHeadroom - (1)),48
self->clipProtectGain__FDK);49
}50
} else {51
for (i = 0; i < self->qmfBands; i++) {52
- self_qmfResidualReal__FDK_0_0[i] = fMult(53
- self_qmfResidualReal__FDK_0_0[i], self->clipProtectGain__FDK);54
- self_qmfResidualImag__FDK_0_0[i] = fMult(55
- self_qmfResidualImag__FDK_0_0[i], self->clipProtectGain__FDK);56
+ self_qmfResidualReal__FDK_0_0[i] =57
+ fMult(scaleValueSaturate(self_qmfResidualReal__FDK_0_0[i],58
+ self->sacInDataHeadroom - (1)),59
+ self->clipProtectGain__FDK);60
+ self_qmfResidualImag__FDK_0_0[i] =61
+ fMult(scaleValueSaturate(self_qmfResidualImag__FDK_0_0[i],62
+ self->sacInDataHeadroom - (1)),63
+ self->clipProtectGain__FDK);64
}65
}66
}67
68
if ((self->tempShapeConfig == 1) && (!isTwoChMode(self->upmixType))) {69
for (ch = 0; ch < self->numOutputChannels; ch++) {70
for (hyb = 0; hyb < self->tp_hybBandBorder; hyb++) {71
- self->hybOutputRealDry__FDK[ch][hyb] +=72
- self->hybOutputRealWet__FDK[ch][hyb];73
- self->hybOutputImagDry__FDK[ch][hyb] +=74
- self->hybOutputImagWet__FDK[ch][hyb];75
+ self->hybOutputRealDry__FDK[ch][hyb] =76
+ fAddSaturate(self->hybOutputRealDry__FDK[ch][hyb],77
+ self->hybOutputRealWet__FDK[ch][hyb]);78
+ self->hybOutputImagDry__FDK[ch][hyb] =79
+ fAddSaturate(self->hybOutputImagDry__FDK[ch][hyb],80
+ self->hybOutputImagWet__FDK[ch][hyb]);81
} /* loop hyb */82
} /* loop ch */83
err = subbandTPApply(84
85
FIXP_DBL *RESTRICT pRealWet = self->hybOutputRealWet__FDK[ch];86
FIXP_DBL *RESTRICT pImagWet = self->hybOutputImagWet__FDK[ch];87
for (hyb = 0; hyb < nHybBands; hyb++) {88
- pRealDry[hyb] += pRealWet[hyb];89
- pImagDry[hyb] += pImagWet[hyb];90
+ pRealDry[hyb] = fAddSaturate(pRealDry[hyb], pRealWet[hyb]);91
+ pImagDry[hyb] = fAddSaturate(pImagDry[hyb], pImagWet[hyb]);92
} /* loop hyb */93
for (; hyb < self->hybridBands; hyb++) {94
- pRealDry[hyb] += pRealWet[hyb];95
+ pRealDry[hyb] = fAddSaturate(pRealDry[hyb], pRealWet[hyb]);96
} /* loop hyb */97
} /* loop ch */98
} /* ( self->tempShapeConfig == 1 ) || ( self->tempShapeConfig == 2 ) */99
100
FDK_ASSERT(self != NULL);101
FDK_ASSERT(pControlFlags != NULL);102
FDK_ASSERT(pcmOutBuf != NULL);103
+ FDK_ASSERT(self->sacInDataHeadroom >= (1));104
105
self->errInt = err; /* Init internal error */106
107
fdk-aac-2.0.1.tar.gz/libSACdec/src/sac_dec.h -> fdk-aac-2.0.2.tar.gz/libSACdec/src/sac_dec.h
Changed
20
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
new frame after SSC change (aka12
decodeAfterConfigHasChangedFlag). */13
SpatialDecConcealmentInfo concealInfo;14
+15
+ INT sacInDataHeadroom; /* Headroom of the SAC input time signal to prevent16
+ clipping */17
};18
19
#define SACDEC_SYNTAX_MPS 120
fdk-aac-2.0.1.tar.gz/libSACdec/src/sac_dec_interface.h -> fdk-aac-2.0.2.tar.gz/libSACdec/src/sac_dec_interface.h
Changed
24
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
12
} MEM_REQUIREMENTS;13
14
-#define PCM_MPS INT_PCM15
-#define PCM_MPSF FIXP_PCM16
+#define PCM_MPS LONG17
+#define PCM_MPSF FIXP_DBL18
19
-#define FIXP_DBL2PCM_MPS(x) ((INT_PCM)FX_DBL2FX_PCM(x))20
+#define FIXP_DBL2PCM_MPS(x) ((LONG)(x))21
22
/* exposed functions (library interface) */23
24
fdk-aac-2.0.1.tar.gz/libSACdec/src/sac_dec_lib.cpp -> fdk-aac-2.0.2.tar.gz/libSACdec/src/sac_dec_lib.cpp
Changed
90
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
}12
13
int mpegSurroundDecoder_Apply(CMpegSurroundDecoder *pMpegSurroundDecoder,14
- INT_PCM *input, PCM_MPS *pTimeData,15
+ PCM_MPS *input, PCM_MPS *pTimeData,16
const int timeDataSize, int timeDataFrameSize,17
int *nChannels, int *frameSize, int sampleRate,18
AUDIO_OBJECT_TYPE coreCodec,19
AUDIO_CHANNEL_TYPE channelType[],20
UCHAR channelIndices[],21
- const FDK_channelMapDescr *const mapDescr) {22
+ const FDK_channelMapDescr *const mapDescr,23
+ const INT inDataHeadroom, INT *outDataHeadroom) {24
SACDEC_ERROR err = MPS_OK;25
PCM_MPS *pTimeOut = pTimeData;26
+ PCM_MPS *TDinput = NULL;27
UINT initControlFlags = 0, controlFlags = 0;28
int timeDataRequiredSize = 0;29
int newData;30
31
return MPS_NOTOK;32
}33
34
+ pMpegSurroundDecoder->pSpatialDec->sacInDataHeadroom = inDataHeadroom;35
+ *outDataHeadroom = (INT)(8);36
+37
pMpegSurroundDecoder->pSpatialDec->pConfigCurrent =38
&pMpegSurroundDecoder39
->spatialSpecificConfig[pMpegSurroundDecoder->bsFrameDecode];40
41
(timeDataFrameSize *42
pMpegSurroundDecoder->pQmfDomain->globalConf.nBandsSynthesis) /43
pMpegSurroundDecoder->pQmfDomain->globalConf.nBandsAnalysis;44
- pMpegSurroundDecoder->pQmfDomain->globalConf.TDinput =45
- pTimeData + timeDataFrameSizeOut - timeDataFrameSize;46
+ TDinput = pTimeData + timeDataFrameSizeOut - timeDataFrameSize;47
for (int i = *nChannels - 1; i >= 0; i--) {48
FDKmemmove(pTimeData + (i + 1) * timeDataFrameSizeOut - timeDataFrameSize,49
pTimeData + timeDataFrameSize * i,50
51
} else {52
if (pMpegSurroundDecoder->mpegSurroundUseTimeInterface) {53
FDKmemcpy(input, pTimeData,54
- sizeof(INT_PCM) * (*nChannels) * (*frameSize));55
- pMpegSurroundDecoder->pQmfDomain->globalConf.TDinput = input;56
+ sizeof(PCM_MPS) * (*nChannels) * (*frameSize));57
+ TDinput = input;58
}59
}60
61
62
&pMpegSurroundDecoder->bsFrames[pMpegSurroundDecoder->bsFrameDecode],63
pMpegSurroundDecoder->mpegSurroundUseTimeInterface ? INPUTMODE_TIME64
: INPUTMODE_QMF_SBR,65
- pMpegSurroundDecoder->pQmfDomain->globalConf.TDinput, NULL, NULL,66
- pTimeOut, *frameSize, &controlFlags, *nChannels, mapDescr);67
+ TDinput, NULL, NULL, pTimeOut, *frameSize, &controlFlags, *nChannels,68
+ mapDescr);69
*nChannels = pMpegSurroundDecoder->pSpatialDec->numOutputChannelsAT;70
71
if (err !=72
73
}74
75
#define SACDEC_VL0 276
-#define SACDEC_VL1 077
+#define SACDEC_VL1 178
#define SACDEC_VL2 079
80
int mpegSurroundDecoder_GetLibInfo(LIB_INFO *info) {81
82
info += i;83
84
info->module_id = FDK_MPSDEC;85
-#ifdef __ANDROID__86
+#ifdef SUPPRESS_BUILD_DATE_INFO87
info->build_date = "";88
info->build_time = "";89
#else90
fdk-aac-2.0.1.tar.gz/libSACdec/src/sac_process.cpp -> fdk-aac-2.0.2.tar.gz/libSACdec/src/sac_process.cpp
Changed
201
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
#include "FDK_trigFcts.h"12
#include "FDK_decorrelate.h"13
14
+#define SAC_DEC_APPLY_M2_SCALE(spec, s) ((spec) >> (-(s)))15
+16
/**17
* \brief Linear interpolation between two parameter values.18
* a*alpha + b*(1-alpha)19
20
if (!isTwoChMode(self->upmixType) && !bypassMode) {21
int i;22
for (i = 0; i < self->qmfBands; i++) {23
- qmfReal[ch][i] = fMult(qmfReal[ch][i], self->clipProtectGain__FDK);24
- qmfImag[ch][i] = fMult(qmfImag[ch][i], self->clipProtectGain__FDK);25
+ qmfReal[ch][i] = fMult(26
+ scaleValueSaturate(qmfReal[ch][i], self->sacInDataHeadroom - (1)),27
+ self->clipProtectGain__FDK);28
+ qmfImag[ch][i] = fMult(29
+ scaleValueSaturate(qmfImag[ch][i], self->sacInDataHeadroom - (1)),30
+ self->clipProtectGain__FDK);31
}32
}33
}34
35
36
/* Write Input data to pQmfRealAnalysis. */37
if (self->bShareDelayWithSBR) {38
- FDK_QmfDomain_GetSlot(39
- &self->pQmfDomain->QmfDomainIn[ch], ts + HYBRID_FILTER_DELAY, 0,40
- MAX_QMF_BANDS_TO_HYBRID, pQmfRealAnalysis, pQmfImagAnalysis, 15);41
+ FDK_QmfDomain_GetSlot(&self->pQmfDomain->QmfDomainIn[ch],42
+ ts + HYBRID_FILTER_DELAY, 0,43
+ MAX_QMF_BANDS_TO_HYBRID, pQmfRealAnalysis,44
+ pQmfImagAnalysis, 15 + (1));45
FDK_QmfDomain_GetSlot(&self->pQmfDomain->QmfDomainIn[ch], ts,46
MAX_QMF_BANDS_TO_HYBRID, self->qmfBands,47
- pQmfRealAnalysis, pQmfImagAnalysis, 15);48
+ pQmfRealAnalysis, pQmfImagAnalysis, 15 + (1));49
} else {50
FDK_QmfDomain_GetSlot(&self->pQmfDomain->QmfDomainIn[ch], ts, 0,51
self->qmfBands, pQmfRealAnalysis,52
- pQmfImagAnalysis, 15);53
+ pQmfImagAnalysis, 15 + (1));54
}55
if (ts == self->pQmfDomain->globalConf.nQmfTimeSlots - 1) {56
/* Is currently also needed in case we dont have any overlap. We need to57
58
for (pb = 0, qs = 3; pb < 2; pb++) {59
INT s;60
FIXP_DBL maxVal;61
- FIXP_SGL mReal1;62
- FIXP_SGL mReal0, mImag0;63
+ FIXP_DBL mReal1;64
+ FIXP_DBL mReal0, mImag0;65
FIXP_DBL iReal0, iImag0, iReal1;66
67
iReal0 = interpolateParameter(alpha, MReal0[pb], MRealPrev0[pb]);68
69
s = fMax(CntLeadingZeros(maxVal) - 1, 0);70
s = fMin(s, scale_param_m2);71
72
- mReal0 = FX_DBL2FX_SGL(iReal0 << s);73
- mImag0 = FX_DBL2FX_SGL(iImag0 << s);74
- mReal1 = FX_DBL2FX_SGL(iReal1 << s);75
+ mReal0 = iReal0 << s;76
+ mImag0 = iImag0 << s;77
+ mReal1 = iReal1 << s;78
79
s = scale_param_m2 - s;80
81
82
}83
84
if (self->phaseCoding == 3) {85
- /* + SCALE_DATA_APPLY_M2 to compensate for Div2 below ?! */86
- scale_param_m2 = SCALE_PARAM_M2_212_PRED + SCALE_DATA_APPLY_M2;87
+ scale_param_m2 = -(SCALE_DATA_APPLY_M2_PC - 1);88
}89
90
for (row = 0; row < self->numM2rows; row++) {91
92
} else { /* isBinauralMode(self->upmixType) */93
94
for (qs = 0; qs < complexHybBands; qs++) {95
- pHybOutRealDry[qs] += fMultDiv2(pWReal[qs], pKernel[qs])96
- << (scale_param_m2);97
- pHybOutImagDry[qs] += fMultDiv2(pWImag[qs], pKernel[qs])98
- << (scale_param_m2);99
+ pHybOutRealDry[qs] += SAC_DEC_APPLY_M2_SCALE(100
+ fMultDiv2(pWReal[qs], pKernel[qs]), scale_param_m2);101
+ pHybOutImagDry[qs] += SAC_DEC_APPLY_M2_SCALE(102
+ fMultDiv2(pWImag[qs], pKernel[qs]), scale_param_m2);103
}104
105
M2ParamToKernelMult(pKernel, self->M2Imag__FDK[row][col],106
107
self->kernels_width, alpha, complexParBands);108
109
/* direct signals sign is -1 for qs = 0,2 */110
- pHybOutRealDry[0] += fMultDiv2(pWImag[0], pKernel[0])111
- << (scale_param_m2);112
- pHybOutImagDry[0] -= fMultDiv2(pWReal[0], pKernel[0])113
- << (scale_param_m2);114
+ pHybOutRealDry[0] += SAC_DEC_APPLY_M2_SCALE(115
+ fMultDiv2(pWImag[0], pKernel[0]), scale_param_m2);116
+ pHybOutImagDry[0] -= SAC_DEC_APPLY_M2_SCALE(117
+ fMultDiv2(pWReal[0], pKernel[0]), scale_param_m2);118
119
- pHybOutRealDry[2] += fMultDiv2(pWImag[2], pKernel[2])120
- << (scale_param_m2);121
- pHybOutImagDry[2] -= fMultDiv2(pWReal[2], pKernel[2])122
- << (scale_param_m2);123
+ pHybOutRealDry[2] += SAC_DEC_APPLY_M2_SCALE(124
+ fMultDiv2(pWImag[2], pKernel[2]), scale_param_m2);125
+ pHybOutImagDry[2] -= SAC_DEC_APPLY_M2_SCALE(126
+ fMultDiv2(pWReal[2], pKernel[2]), scale_param_m2);127
128
/* direct signals sign is +1 for qs = 1,3,4,5,...,complexHybBands */129
- pHybOutRealDry[1] -= fMultDiv2(pWImag[1], pKernel[1])130
- << (scale_param_m2);131
- pHybOutImagDry[1] += fMultDiv2(pWReal[1], pKernel[1])132
- << (scale_param_m2);133
+ pHybOutRealDry[1] -= SAC_DEC_APPLY_M2_SCALE(134
+ fMultDiv2(pWImag[1], pKernel[1]), scale_param_m2);135
+ pHybOutImagDry[1] += SAC_DEC_APPLY_M2_SCALE(136
+ fMultDiv2(pWReal[1], pKernel[1]), scale_param_m2);137
138
for (qs = 3; qs < complexHybBands; qs++) {139
- pHybOutRealDry[qs] -= fMultDiv2(pWImag[qs], pKernel[qs])140
- << (scale_param_m2);141
- pHybOutImagDry[qs] += fMultDiv2(pWReal[qs], pKernel[qs])142
- << (scale_param_m2);143
+ pHybOutRealDry[qs] -= SAC_DEC_APPLY_M2_SCALE(144
+ fMultDiv2(pWImag[qs], pKernel[qs]), scale_param_m2);145
+ pHybOutImagDry[qs] += SAC_DEC_APPLY_M2_SCALE(146
+ fMultDiv2(pWReal[qs], pKernel[qs]), scale_param_m2);147
}148
} /* self->upmixType */149
} /* if (activParamBands) */150
151
FIXP_DBL *RESTRICT pHybOutImag;152
153
for (qs = 0; qs < resHybIndex; qs++) {154
- pHybOutRealDry[qs] += fMultDiv2(pWReal[qs], pKernel[qs])155
- << (scale_param_m2);156
- pHybOutImagDry[qs] += fMultDiv2(pWImag[qs], pKernel[qs])157
- << (scale_param_m2);158
+ pHybOutRealDry[qs] += SAC_DEC_APPLY_M2_SCALE(159
+ fMultDiv2(pWReal[qs], pKernel[qs]), scale_param_m2);160
+ pHybOutImagDry[qs] += SAC_DEC_APPLY_M2_SCALE(161
+ fMultDiv2(pWImag[qs], pKernel[qs]), scale_param_m2);162
}163
/* decor signals */164
for (; qs < complexHybBands; qs++) {165
- pHybOutRealWet[qs] += fMultDiv2(pWReal[qs], pKernel[qs])166
- << (scale_param_m2);167
- pHybOutImagWet[qs] += fMultDiv2(pWImag[qs], pKernel[qs])168
- << (scale_param_m2);169
+ pHybOutRealWet[qs] += SAC_DEC_APPLY_M2_SCALE(170
+ fMultDiv2(pWReal[qs], pKernel[qs]), scale_param_m2);171
+ pHybOutImagWet[qs] += SAC_DEC_APPLY_M2_SCALE(172
+ fMultDiv2(pWImag[qs], pKernel[qs]), scale_param_m2);173
}174
175
M2ParamToKernelMult(pKernel, self->M2Imag__FDK[row][col],176
177
/* direct signals sign is -1 for qs = 0,2 */178
/* direct signals sign is +1 for qs = 1,3.. */179
if (toolsDisabled) {180
- pHybOutRealDry[0] += fMultDiv2(pWImag[0], pKernel[0])181
- << (scale_param_m2);182
- pHybOutImagDry[0] -= fMultDiv2(pWReal[0], pKernel[0])183
- << (scale_param_m2);184
-185
- pHybOutRealDry[1] -= fMultDiv2(pWImag[1], pKernel[1])186
- << (scale_param_m2);187
- pHybOutImagDry[1] += fMultDiv2(pWReal[1], pKernel[1])188
- << (scale_param_m2);189
-190
- pHybOutRealDry[2] += fMultDiv2(pWImag[2], pKernel[2])191
- << (scale_param_m2);192
- pHybOutImagDry[2] -= fMultDiv2(pWReal[2], pKernel[2])193
- << (scale_param_m2);194
+ pHybOutRealDry[0] += SAC_DEC_APPLY_M2_SCALE(195
+ fMultDiv2(pWImag[0], pKernel[0]), scale_param_m2);196
+ pHybOutImagDry[0] -= SAC_DEC_APPLY_M2_SCALE(197
+ fMultDiv2(pWReal[0], pKernel[0]), scale_param_m2);198
+199
+ pHybOutRealDry[1] -= SAC_DEC_APPLY_M2_SCALE(200
+ fMultDiv2(pWImag[1], pKernel[1]), scale_param_m2);201
fdk-aac-2.0.1.tar.gz/libSACdec/src/sac_qmf.cpp -> fdk-aac-2.0.2.tar.gz/libSACdec/src/sac_qmf.cpp
Changed
19
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
12
SACDEC_ERROR CalculateSpaceSynthesisQmf(13
const HANDLE_FDK_QMF_DOMAIN_OUT hQmfDomainOutCh, const FIXP_DBL *Sr,14
- const FIXP_DBL *Si, const INT stride, INT_PCM *timeSig) {15
+ const FIXP_DBL *Si, const INT stride, PCM_MPS *timeSig) {16
SACDEC_ERROR err = MPS_OK;17
18
if (hQmfDomainOutCh == NULL) {19
fdk-aac-2.0.1.tar.gz/libSACdec/src/sac_qmf.h -> fdk-aac-2.0.2.tar.gz/libSACdec/src/sac_qmf.h
Changed
19
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
*/12
SACDEC_ERROR CalculateSpaceSynthesisQmf(13
const HANDLE_FDK_QMF_DOMAIN_OUT hQmfDomainOutCh, const FIXP_DBL *Sr,14
- const FIXP_DBL *Si, const INT stride, INT_PCM *timeSig);15
+ const FIXP_DBL *Si, const INT stride, PCM_MPS *timeSig);16
17
/**18
* \brief Convert audio input data to qmf representation.19
fdk-aac-2.0.1.tar.gz/libSACdec/src/sac_reshapeBBEnv.cpp -> fdk-aac-2.0.2.tar.gz/libSACdec/src/sac_reshapeBBEnv.cpp
Changed
201
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
FIXP_DBL nrg;12
13
/* qs = 12, 13, 14 */14
- slotNrg[0] = ((fPow2Div2((*pReal++) << maxValSF) +15
- fPow2Div2((*pImag++) << maxValSF)) >>16
- (SF_FACTOR_SLOT - 1));17
- slotNrg[1] = ((fPow2Div2((*pReal++) << maxValSF) +18
- fPow2Div2((*pImag++) << maxValSF)) >>19
- (SF_FACTOR_SLOT - 1));20
- slotNrg[2] = ((fPow2Div2((*pReal++) << maxValSF) +21
- fPow2Div2((*pImag++) << maxValSF)) >>22
- (SF_FACTOR_SLOT - 1));23
+ slotNrg[0] = ((fPow2Div2((*pReal++) << maxValSF) >> (SF_FACTOR_SLOT - 1)) +24
+ (fPow2Div2((*pImag++) << maxValSF) >> (SF_FACTOR_SLOT - 1)));25
+ slotNrg[1] = ((fPow2Div2((*pReal++) << maxValSF) >> (SF_FACTOR_SLOT - 1)) +26
+ (fPow2Div2((*pImag++) << maxValSF) >> (SF_FACTOR_SLOT - 1)));27
+ slotNrg[2] = ((fPow2Div2((*pReal++) << maxValSF) >> (SF_FACTOR_SLOT - 1)) +28
+ (fPow2Div2((*pImag++) << maxValSF) >> (SF_FACTOR_SLOT - 1)));29
/* qs = 15 */30
- slotNrg[3] = ((fPow2Div2((*pReal++) << maxValSF) +31
- fPow2Div2((*pImag++) << maxValSF)) >>32
- (SF_FACTOR_SLOT - 1));33
+ slotNrg[3] = ((fPow2Div2((*pReal++) << maxValSF) >> (SF_FACTOR_SLOT - 1)) +34
+ (fPow2Div2((*pImag++) << maxValSF) >> (SF_FACTOR_SLOT - 1)));35
/* qs = 16, 17 */36
- nrg = ((fPow2Div2((*pReal++) << maxValSF) +37
- fPow2Div2((*pImag++) << maxValSF)) >>38
- (SF_FACTOR_SLOT - 1));39
- slotNrg[4] = nrg + ((fPow2Div2((*pReal++) << maxValSF) +40
- fPow2Div2((*pImag++) << maxValSF)) >>41
- (SF_FACTOR_SLOT - 1));42
+ nrg = ((fPow2Div2((*pReal++) << maxValSF) >> (SF_FACTOR_SLOT - 1)) +43
+ (fPow2Div2((*pImag++) << maxValSF) >> (SF_FACTOR_SLOT - 1)));44
+ slotNrg[4] =45
+ nrg + ((fPow2Div2((*pReal++) << maxValSF) >> (SF_FACTOR_SLOT - 1)) +46
+ (fPow2Div2((*pImag++) << maxValSF) >> (SF_FACTOR_SLOT - 1)));47
/* qs = 18, 19, 20 */48
- nrg = ((fPow2Div2((*pReal++) << maxValSF) +49
- fPow2Div2((*pImag++) << maxValSF)) >>50
- (SF_FACTOR_SLOT - 1));51
- nrg += ((fPow2Div2((*pReal++) << maxValSF) +52
- fPow2Div2((*pImag++) << maxValSF)) >>53
- (SF_FACTOR_SLOT - 1));54
- slotNrg[5] = nrg + ((fPow2Div2((*pReal++) << maxValSF) +55
- fPow2Div2((*pImag++) << maxValSF)) >>56
- (SF_FACTOR_SLOT - 1));57
+ nrg = ((fPow2Div2((*pReal++) << maxValSF) >> (SF_FACTOR_SLOT - 1)) +58
+ (fPow2Div2((*pImag++) << maxValSF) >> (SF_FACTOR_SLOT - 1)));59
+ nrg += ((fPow2Div2((*pReal++) << maxValSF) >> (SF_FACTOR_SLOT - 1)) +60
+ (fPow2Div2((*pImag++) << maxValSF) >> (SF_FACTOR_SLOT - 1)));61
+ slotNrg[5] =62
+ nrg + ((fPow2Div2((*pReal++) << maxValSF) >> (SF_FACTOR_SLOT - 1)) +63
+ (fPow2Div2((*pImag++) << maxValSF) >> (SF_FACTOR_SLOT - 1)));64
/* qs = 21, 22 */65
- nrg = ((fPow2Div2((*pReal++) << maxValSF) +66
- fPow2Div2((*pImag++) << maxValSF)) >>67
- (SF_FACTOR_SLOT - 1));68
- slotNrg[6] = nrg + ((fPow2Div2((*pReal++) << maxValSF) +69
- fPow2Div2((*pImag++) << maxValSF)) >>70
- (SF_FACTOR_SLOT - 1));71
+ nrg = ((fPow2Div2((*pReal++) << maxValSF) >> (SF_FACTOR_SLOT - 1)) +72
+ (fPow2Div2((*pImag++) << maxValSF) >> (SF_FACTOR_SLOT - 1)));73
+ slotNrg[6] =74
+ nrg + ((fPow2Div2((*pReal++) << maxValSF) >> (SF_FACTOR_SLOT - 1)) +75
+ (fPow2Div2((*pImag++) << maxValSF) >> (SF_FACTOR_SLOT - 1)));76
/* qs = 23, 24 */77
if (hybBands > 23) {78
- slotNrg[6] += ((fPow2Div2((*pReal++) << maxValSF) +79
- fPow2Div2((*pImag++) << maxValSF)) >>80
- (SF_FACTOR_SLOT - 1));81
- slotNrg[6] += ((fPow2Div2((*pReal++) << maxValSF) +82
- fPow2Div2((*pImag++) << maxValSF)) >>83
- (SF_FACTOR_SLOT - 1));84
+ slotNrg[6] += ((fPow2Div2((*pReal++) << maxValSF) >> (SF_FACTOR_SLOT - 1)) +85
+ (fPow2Div2((*pImag++) << maxValSF) >> (SF_FACTOR_SLOT - 1)));86
+ slotNrg[6] += ((fPow2Div2((*pReal++) << maxValSF) >> (SF_FACTOR_SLOT - 1)) +87
+ (fPow2Div2((*pImag++) << maxValSF) >> (SF_FACTOR_SLOT - 1)));88
/* qs = 25, 26, 29, 28, 29 */89
- nrg = ((fPow2Div2((*pReal++) << maxValSF) +90
- fPow2Div2((*pImag++) << maxValSF)) >>91
- (SF_FACTOR_SLOT - 1));92
- nrg += ((fPow2Div2((*pReal++) << maxValSF) +93
- fPow2Div2((*pImag++) << maxValSF)) >>94
- (SF_FACTOR_SLOT - 1));95
- nrg += ((fPow2Div2((*pReal++) << maxValSF) +96
- fPow2Div2((*pImag++) << maxValSF)) >>97
- (SF_FACTOR_SLOT - 1));98
- nrg += ((fPow2Div2((*pReal++) << maxValSF) +99
- fPow2Div2((*pImag++) << maxValSF)) >>100
- (SF_FACTOR_SLOT - 1));101
- slotNrg[7] = nrg + ((fPow2Div2((*pReal++) << maxValSF) +102
- fPow2Div2((*pImag++) << maxValSF)) >>103
- (SF_FACTOR_SLOT - 1));104
+ nrg = ((fPow2Div2((*pReal++) << maxValSF) >> (SF_FACTOR_SLOT - 1)) +105
+ (fPow2Div2((*pImag++) << maxValSF) >> (SF_FACTOR_SLOT - 1)));106
+ nrg += ((fPow2Div2((*pReal++) << maxValSF) >> (SF_FACTOR_SLOT - 1)) +107
+ (fPow2Div2((*pImag++) << maxValSF) >> (SF_FACTOR_SLOT - 1)));108
+ nrg += ((fPow2Div2((*pReal++) << maxValSF) >> (SF_FACTOR_SLOT - 1)) +109
+ (fPow2Div2((*pImag++) << maxValSF) >> (SF_FACTOR_SLOT - 1)));110
+ nrg += ((fPow2Div2((*pReal++) << maxValSF) >> (SF_FACTOR_SLOT - 1)) +111
+ (fPow2Div2((*pImag++) << maxValSF) >> (SF_FACTOR_SLOT - 1)));112
+ slotNrg[7] =113
+ nrg + ((fPow2Div2((*pReal++) << maxValSF) >> (SF_FACTOR_SLOT - 1)) +114
+ (fPow2Div2((*pImag++) << maxValSF) >> (SF_FACTOR_SLOT - 1)));115
/* qs = 30 ... min(41,hybBands-1) */116
- nrg = ((fPow2Div2((*pReal++) << maxValSF) +117
- fPow2Div2((*pImag++) << maxValSF)) >>118
- (SF_FACTOR_SLOT - 1));119
+ nrg = ((fPow2Div2((*pReal++) << maxValSF) >> (SF_FACTOR_SLOT - 1)) +120
+ (fPow2Div2((*pImag++) << maxValSF) >> (SF_FACTOR_SLOT - 1)));121
for (qs = 31; qs < hybBands; qs++) {122
- nrg += ((fPow2Div2((*pReal++) << maxValSF) +123
- fPow2Div2((*pImag++) << maxValSF)) >>124
- (SF_FACTOR_SLOT - 1));125
+ nrg += ((fPow2Div2((*pReal++) << maxValSF) >> (SF_FACTOR_SLOT - 1)) +126
+ (fPow2Div2((*pImag++) << maxValSF) >> (SF_FACTOR_SLOT - 1)));127
}128
slotNrg[8] = nrg;129
} else {130
131
}132
}133
134
-static inline INT getMaxValDmx(FIXP_DBL *RESTRICT pReal,135
- FIXP_DBL *RESTRICT pImag, INT cplxBands,136
- INT hybBands) {137
- INT qs, clz;138
- FIXP_DBL maxVal = FL2FXCONST_DBL(0.0f);139
-140
- for (qs = 12; qs < cplxBands; qs++) {141
- maxVal |= fAbs(pReal[qs]);142
- maxVal |= fAbs(pImag[qs]);143
- }144
- for (; qs < hybBands; qs++) {145
- maxVal |= fAbs(pReal[qs]);146
- }147
-148
- clz = fixMax(0, CntLeadingZeros(maxVal) - 1);149
-150
- return (clz);151
-}152
-153
-static inline INT getMaxValDryWet(FIXP_DBL *RESTRICT pReal,154
- FIXP_DBL *RESTRICT pImag,155
- FIXP_DBL *RESTRICT pHybOutputRealDry,156
- FIXP_DBL *RESTRICT pHybOutputImagDry,157
- FIXP_DBL *RESTRICT pHybOutputRealWet,158
- FIXP_DBL *RESTRICT pHybOutputImagWet,159
- INT cplxBands, INT hybBands) {160
- INT qs, clz;161
- FIXP_DBL maxVal = FL2FXCONST_DBL(0.0f);162
+static inline void combineDryWet(FIXP_DBL *RESTRICT pReal,163
+ FIXP_DBL *RESTRICT pImag,164
+ FIXP_DBL *RESTRICT pHybOutputRealDry,165
+ FIXP_DBL *RESTRICT pHybOutputImagDry,166
+ FIXP_DBL *RESTRICT pHybOutputRealWet,167
+ FIXP_DBL *RESTRICT pHybOutputImagWet,168
+ INT cplxBands, INT hybBands) {169
+ INT qs;170
171
for (qs = 12; qs < cplxBands; qs++) {172
- pReal[qs] = pHybOutputRealDry[qs] + pHybOutputRealWet[qs];173
- maxVal |= fAbs(pReal[qs]);174
- pImag[qs] = pHybOutputImagDry[qs] + pHybOutputImagWet[qs];175
- maxVal |= fAbs(pImag[qs]);176
+ pReal[qs] = (pHybOutputRealDry[qs] >> 1) + (pHybOutputRealWet[qs] >> 1);177
+ pImag[qs] = (pHybOutputImagDry[qs] >> 1) + (pHybOutputImagWet[qs] >> 1);178
}179
for (; qs < hybBands; qs++) {180
- pReal[qs] = pHybOutputRealDry[qs] + pHybOutputRealWet[qs];181
- maxVal |= fAbs(pReal[qs]);182
+ pReal[qs] = (pHybOutputRealDry[qs] >> 1) + (pHybOutputRealWet[qs] >> 1);183
}184
-185
- clz = fixMax(0, CntLeadingZeros(maxVal) - 1);186
-187
- return (clz);188
}189
190
static inline void slotAmp(FIXP_DBL *RESTRICT slotAmp_dry,191
192
193
dry = wet = FL2FXCONST_DBL(0.0f);194
for (qs = 0; qs < cplxBands; qs++) {195
- dry = fAddSaturate(dry, fPow2Div2(pHybOutputRealDry[qs]) +196
- fPow2Div2(pHybOutputImagDry[qs]));197
- wet = fAddSaturate(wet, fPow2Div2(pHybOutputRealWet[qs]) +198
- fPow2Div2(pHybOutputImagWet[qs]));199
+ dry = fAddSaturate(dry, fPow2Div2(pHybOutputRealDry[qs] << (1)) +200
+ fPow2Div2(pHybOutputImagDry[qs] << (1)));201
fdk-aac-2.0.1.tar.gz/libSACdec/src/sac_rom.cpp -> fdk-aac-2.0.2.tar.gz/libSACdec/src/sac_rom.cpp
Changed
201
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
SCALE_IPD(4.71238898038469f), SCALE_IPD(5.105088062f),12
SCALE_IPD(5.49778714378214f), SCALE_IPD(5.890486225f)};13
14
-#define SCALE_SPLIT_ANGLE(a) (FL2FXCONST_CFG(a / (float)(1 << IPD_SCALE)))15
+#define SCALE_SPLIT_ANGLE(a) (FL2FXCONST_DBL(a / (float)(1 << IPD_SCALE)))16
/*17
Generate table dequantIPD_CLD_ICC_splitAngle__FDK[16][31][8]:18
19
20
atan2(w2 * sinIpd, w1 * iidLin + w2 * cosIpd);21
}22
*/23
+const FIXP_DBL dequantIPD_CLD_ICC_splitAngle__FDK[15][31][8] = {24
+ {25
+ {SCALE_SPLIT_ANGLE(0.39269909262657),26
+ SCALE_SPLIT_ANGLE(0.39269909262657),27
+ SCALE_SPLIT_ANGLE(0.39269909262657),28
+ SCALE_SPLIT_ANGLE(0.39269909262657),29
+ SCALE_SPLIT_ANGLE(0.39269909262657),30
+ SCALE_SPLIT_ANGLE(0.39269909262657),31
+ SCALE_SPLIT_ANGLE(0.39269909262657),32
+ SCALE_SPLIT_ANGLE(0.39269909262657)},33
+ {SCALE_SPLIT_ANGLE(0.39055734872818),34
+ SCALE_SPLIT_ANGLE(0.39055740833282),35
+ SCALE_SPLIT_ANGLE(0.39055749773979),36
+ SCALE_SPLIT_ANGLE(0.39055770635605),37
+ SCALE_SPLIT_ANGLE(0.39055791497231),38
+ SCALE_SPLIT_ANGLE(0.39055824279785),39
+ SCALE_SPLIT_ANGLE(0.39055877923965),40
+ SCALE_SPLIT_ANGLE(0.39055916666985)},41
+ {SCALE_SPLIT_ANGLE(0.38890451192856),42
+ SCALE_SPLIT_ANGLE(0.38890469074249),43
+ SCALE_SPLIT_ANGLE(0.38890495896339),44
+ SCALE_SPLIT_ANGLE(0.38890564441681),45
+ SCALE_SPLIT_ANGLE(0.38890630006790),46
+ SCALE_SPLIT_ANGLE(0.38890734314919),47
+ SCALE_SPLIT_ANGLE(0.38890904188156),48
+ SCALE_SPLIT_ANGLE(0.38891020417213)},49
+ {SCALE_SPLIT_ANGLE(0.38599541783333),50
+ SCALE_SPLIT_ANGLE(0.38599592447281),51
+ SCALE_SPLIT_ANGLE(0.38599672913551),52
+ SCALE_SPLIT_ANGLE(0.38599878549576),53
+ SCALE_SPLIT_ANGLE(0.38600078225136),54
+ SCALE_SPLIT_ANGLE(0.38600403070450),55
+ SCALE_SPLIT_ANGLE(0.38600939512253),56
+ SCALE_SPLIT_ANGLE(0.38601315021515)},57
+ {SCALE_SPLIT_ANGLE(0.38091576099396),58
+ SCALE_SPLIT_ANGLE(0.38091731071472),59
+ SCALE_SPLIT_ANGLE(0.38091966509819),60
+ SCALE_SPLIT_ANGLE(0.38092571496964),61
+ SCALE_SPLIT_ANGLE(0.38093179464340),62
+ SCALE_SPLIT_ANGLE(0.38094165921211),63
+ SCALE_SPLIT_ANGLE(0.38095843791962),64
+ SCALE_SPLIT_ANGLE(0.38097056746483)},65
+ {SCALE_SPLIT_ANGLE(0.37216997146606),66
+ SCALE_SPLIT_ANGLE(0.37217426300049),67
+ SCALE_SPLIT_ANGLE(0.37218090891838),68
+ SCALE_SPLIT_ANGLE(0.37219807505608),69
+ SCALE_SPLIT_ANGLE(0.37221556901932),70
+ SCALE_SPLIT_ANGLE(0.37224492430687),71
+ SCALE_SPLIT_ANGLE(0.37229704856873),72
+ SCALE_SPLIT_ANGLE(0.37233674526215)},73
+ {SCALE_SPLIT_ANGLE(0.36424967646599),74
+ SCALE_SPLIT_ANGLE(0.36425727605820),75
+ SCALE_SPLIT_ANGLE(0.36426907777786),76
+ SCALE_SPLIT_ANGLE(0.36430004239082),77
+ SCALE_SPLIT_ANGLE(0.36433213949203),78
+ SCALE_SPLIT_ANGLE(0.36438730359077),79
+ SCALE_SPLIT_ANGLE(0.36448970437050),80
+ SCALE_SPLIT_ANGLE(0.36457163095474)},81
+ {SCALE_SPLIT_ANGLE(0.35356888175011),82
+ SCALE_SPLIT_ANGLE(0.35358175635338),83
+ SCALE_SPLIT_ANGLE(0.35360190272331),84
+ SCALE_SPLIT_ANGLE(0.35365560650826),85
+ SCALE_SPLIT_ANGLE(0.35371258854866),86
+ SCALE_SPLIT_ANGLE(0.35381385684013),87
+ SCALE_SPLIT_ANGLE(0.35401365160942),88
+ SCALE_SPLIT_ANGLE(0.35418578982353)},89
+ {SCALE_SPLIT_ANGLE(0.33942583203316),90
+ SCALE_SPLIT_ANGLE(0.33944645524025),91
+ SCALE_SPLIT_ANGLE(0.33947896957397),92
+ SCALE_SPLIT_ANGLE(0.33956742286682),93
+ SCALE_SPLIT_ANGLE(0.33966416120529),94
+ SCALE_SPLIT_ANGLE(0.33984372019768),95
+ SCALE_SPLIT_ANGLE(0.34023007750511),96
+ SCALE_SPLIT_ANGLE(0.34060221910477)},97
+ {SCALE_SPLIT_ANGLE(0.32115450501442),98
+ SCALE_SPLIT_ANGLE(0.32118520140648),99
+ SCALE_SPLIT_ANGLE(0.32123416662216),100
+ SCALE_SPLIT_ANGLE(0.32137048244476),101
+ SCALE_SPLIT_ANGLE(0.32152509689331),102
+ SCALE_SPLIT_ANGLE(0.32182863354683),103
+ SCALE_SPLIT_ANGLE(0.32256561517715),104
+ SCALE_SPLIT_ANGLE(0.32340893149376)},105
+ {SCALE_SPLIT_ANGLE(0.29830521345139),106
+ SCALE_SPLIT_ANGLE(0.29834723472595),107
+ SCALE_SPLIT_ANGLE(0.29841500520706),108
+ SCALE_SPLIT_ANGLE(0.29860860109329),109
+ SCALE_SPLIT_ANGLE(0.29883742332458),110
+ SCALE_SPLIT_ANGLE(0.29931786656380),111
+ SCALE_SPLIT_ANGLE(0.30069106817245),112
+ SCALE_SPLIT_ANGLE(0.30275771021843)},113
+ {SCALE_SPLIT_ANGLE(0.28050789237022),114
+ SCALE_SPLIT_ANGLE(0.28055712580681),115
+ SCALE_SPLIT_ANGLE(0.28063708543777),116
+ SCALE_SPLIT_ANGLE(0.28086942434311),117
+ SCALE_SPLIT_ANGLE(0.28115189075470),118
+ SCALE_SPLIT_ANGLE(0.28177404403687),119
+ SCALE_SPLIT_ANGLE(0.28380545973778),120
+ SCALE_SPLIT_ANGLE(0.28782638907433)},121
+ {SCALE_SPLIT_ANGLE(0.26083287596703),122
+ SCALE_SPLIT_ANGLE(0.26088824868202),123
+ SCALE_SPLIT_ANGLE(0.26097872853279),124
+ SCALE_SPLIT_ANGLE(0.26124578714371),125
+ SCALE_SPLIT_ANGLE(0.26157897710800),126
+ SCALE_SPLIT_ANGLE(0.26234793663025),127
+ SCALE_SPLIT_ANGLE(0.26525345444679),128
+ SCALE_SPLIT_ANGLE(0.27373576164246)},129
+ {SCALE_SPLIT_ANGLE(0.23960022628307),130
+ SCALE_SPLIT_ANGLE(0.23966011404991),131
+ SCALE_SPLIT_ANGLE(0.23975846171379),132
+ SCALE_SPLIT_ANGLE(0.24005253612995),133
+ SCALE_SPLIT_ANGLE(0.24042735993862),134
+ SCALE_SPLIT_ANGLE(0.24132782220840),135
+ SCALE_SPLIT_ANGLE(0.24525125324726),136
+ SCALE_SPLIT_ANGLE(0.26534587144852)},137
+ {SCALE_SPLIT_ANGLE(0.21727463603020),138
+ SCALE_SPLIT_ANGLE(0.21733720600605),139
+ SCALE_SPLIT_ANGLE(0.21744030714035),140
+ SCALE_SPLIT_ANGLE(0.21775111556053),141
+ SCALE_SPLIT_ANGLE(0.21815299987793),142
+ SCALE_SPLIT_ANGLE(0.21914559602737),143
+ SCALE_SPLIT_ANGLE(0.22397418320179),144
+ SCALE_SPLIT_ANGLE(0.27997341752052)},145
+ {SCALE_SPLIT_ANGLE(0.19442924857140),146
+ SCALE_SPLIT_ANGLE(0.19449260830879),147
+ SCALE_SPLIT_ANGLE(0.19459712505341),148
+ SCALE_SPLIT_ANGLE(0.19491320848465),149
+ SCALE_SPLIT_ANGLE(0.19532410800457),150
+ SCALE_SPLIT_ANGLE(0.19634956121445),151
+ SCALE_SPLIT_ANGLE(0.20156545937061),152
+ SCALE_SPLIT_ANGLE(0.33650875091553)},153
+ {SCALE_SPLIT_ANGLE(0.17168679833412),154
+ SCALE_SPLIT_ANGLE(0.17174908518791),155
+ SCALE_SPLIT_ANGLE(0.17185173928738),156
+ SCALE_SPLIT_ANGLE(0.17216140031815),157
+ SCALE_SPLIT_ANGLE(0.17256212234497),158
+ SCALE_SPLIT_ANGLE(0.17355351150036),159
+ SCALE_SPLIT_ANGLE(0.17841057479382),160
+ SCALE_SPLIT_ANGLE(0.23922468721867)},161
+ {SCALE_SPLIT_ANGLE(0.14965108036995),162
+ SCALE_SPLIT_ANGLE(0.14971046149731),163
+ SCALE_SPLIT_ANGLE(0.14980803430080),164
+ SCALE_SPLIT_ANGLE(0.15010002255440),165
+ SCALE_SPLIT_ANGLE(0.15047283470631),166
+ SCALE_SPLIT_ANGLE(0.15137128531933),167
+ SCALE_SPLIT_ANGLE(0.15533345937729),168
+ SCALE_SPLIT_ANGLE(0.17691856622696)},169
+ {SCALE_SPLIT_ANGLE(0.12884500622749),170
+ SCALE_SPLIT_ANGLE(0.12889973819256),171
+ SCALE_SPLIT_ANGLE(0.12898921966553),172
+ SCALE_SPLIT_ANGLE(0.12925371527672),173
+ SCALE_SPLIT_ANGLE(0.12958450615406),174
+ SCALE_SPLIT_ANGLE(0.13035117089748),175
+ SCALE_SPLIT_ANGLE(0.13329000771046),176
+ SCALE_SPLIT_ANGLE(0.14226883649826)},177
+ {SCALE_SPLIT_ANGLE(0.10966771841049),178
+ SCALE_SPLIT_ANGLE(0.10971628874540),179
+ SCALE_SPLIT_ANGLE(0.10979522019625),180
+ SCALE_SPLIT_ANGLE(0.11002497375011),181
+ SCALE_SPLIT_ANGLE(0.11030506342649),182
+ SCALE_SPLIT_ANGLE(0.11092507839203),183
+ SCALE_SPLIT_ANGLE(0.11297956109047),184
+ SCALE_SPLIT_ANGLE(0.11718676239252)},185
+ {SCALE_SPLIT_ANGLE(0.09237616509199),186
+ SCALE_SPLIT_ANGLE(0.09241759032011),187
+ SCALE_SPLIT_ANGLE(0.09248441457748),188
+ SCALE_SPLIT_ANGLE(0.09267570823431),189
+ SCALE_SPLIT_ANGLE(0.09290253371000),190
+ SCALE_SPLIT_ANGLE(0.09338124841452),191
+ SCALE_SPLIT_ANGLE(0.09476862102747),192
+ SCALE_SPLIT_ANGLE(0.09691140055656)},193
+ {SCALE_SPLIT_ANGLE(0.07020132243633),194
+ SCALE_SPLIT_ANGLE(0.07023159414530),195
+ SCALE_SPLIT_ANGLE(0.07027988880873),196
+ SCALE_SPLIT_ANGLE(0.07041462510824),197
+ SCALE_SPLIT_ANGLE(0.07056795060635),198
+ SCALE_SPLIT_ANGLE(0.07087047398090),199
+ SCALE_SPLIT_ANGLE(0.07161350548267),200
+ SCALE_SPLIT_ANGLE(0.07247893512249)},201
fdk-aac-2.0.1.tar.gz/libSACdec/src/sac_rom.h -> fdk-aac-2.0.2.tar.gz/libSACdec/src/sac_rom.h
Changed
63
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
#include "machine_type.h"12
13
/* Global ROM table data type: */14
-#ifndef ARCH_PREFER_MULT_32x3215
-#define FIXP_CFG FIXP_SGL16
-#define FX_CFG2FX_DBL FX_SGL2FX_DBL17
-#define FX_CFG2FX_SGL18
-#define CFG(a) (FX_DBL2FXCONST_SGL(a))19
-#define FL2FXCONST_CFG FL2FXCONST_SGL20
-#define FX_DBL2FX_CFG(x) FX_DBL2FX_SGL((FIXP_DBL)(x))21
-#else22
#define FIXP_CFG FIXP_DBL23
#define FX_CFG2FX_DBL24
#define FX_CFG2FX_SGL FX_DBL2FX_SGL25
#define CFG(a) FIXP_DBL(a)26
#define FL2FXCONST_CFG FL2FXCONST_DBL27
#define FX_DBL2FX_CFG(x) ((FIXP_DBL)(x))28
-#endif29
30
/* others */31
#define SCALE_INV_ICC (2)32
33
34
#define QCC_SCALE 135
#define M1M2_DATA FIXP_DBL36
-#ifndef ARCH_PREFER_MULT_32x3237
-#define M1M2_CDATA FIXP_SGL38
-#define M1M2_CDATA2FX_DBL(a) FX_SGL2FX_DBL(a)39
-#define FX_DBL2M1M2_CDATA(a) FX_DBL2FX_SGL(a)40
-#else41
#define M1M2_CDATA FIXP_DBL42
#define M1M2_CDATA2FX_DBL(a) (a)43
#define FX_DBL2M1M2_CDATA(a) (a)44
-#endif45
46
#define CLIP_PROTECT_GAIN_0(x) FL2FXCONST_CFG(((x) / (float)(1 << 0)))47
#define CLIP_PROTECT_GAIN_1(x) FL2FXCONST_CFG(((x) / (float)(1 << 1)))48
#define CLIP_PROTECT_GAIN_2(x) FL2FXCONST_CFG(((x) / (float)(1 << 2)))49
50
-#define SF_CLD_C1C2 (8)51
+#define SF_CLD_C1C2 (9)52
53
extern const FIXP_CFG dequantCPC__FDK[];54
extern const FIXP_CFG dequantICC__FDK[8];55
56
(FL2FXCONST_DBL(3.1415926535897932f / (float)(1 << (IPD_SCALE - 1))))57
58
extern const FIXP_CFG dequantIPD__FDK[16];59
+extern const FIXP_DBL dequantIPD_CLD_ICC_splitAngle__FDK[15][31][8];60
61
extern const FIXP_CFG H11_nc[31][8];62
extern const FIXP_CFG H12_nc[31][8];63
fdk-aac-2.0.1.tar.gz/libSACdec/src/sac_stp.cpp -> fdk-aac-2.0.2.tar.gz/libSACdec/src/sac_stp.cpp
Changed
201
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
#include "FDK_matrixCalloc.h"12
#include "sac_rom.h"13
14
+#define SF_FREQ_DOMAIN_HEADROOM (2 * (1))15
+16
#define BP_GF_START 617
#define BP_GF_SIZE 2518
#define HP_SIZE 919
20
#define SF_WET 521
#define SF_DRY \22
3 /* SF_DRY == 2 would produce good conformance test results as well */23
+#define SF_DRY_NRG \24
+ (4 - 1) /* 8.495f = sum(BP_GF__FDK[i]) \25
+ i=0,..,(sizeof(BP_GF__FDK)/sizeof(FIXP_CFG)-1) => energy \26
+ calculation needs 4 bits headroom, headroom can be reduced by 1 \27
+ bit due to fPow2Div2() usage */28
+#define SF_WET_NRG \29
+ (4 - 1) /* 8.495f = sum(BP_GF__FDK[i]) \30
+ i=0,..,(sizeof(BP_GF__FDK)/sizeof(FIXP_CFG)-1) => energy \31
+ calculation needs 4 bits headroom, headroom can be reduced by 1 \32
+ bit due to fPow2Div2() usage */33
#define SF_PRODUCT_BP_GF 1334
#define SF_PRODUCT_BP_GF_GF 2635
#define SF_SCALE 236
37
STP_SCALE_LIMIT_LO_LD64 = LD64(STP_SCALE_LIMIT_LO*STP_SCALE_LIMIT_LO)38
*/39
40
-#define DRY_ENER_WEIGHT(DryEner) DryEner = DryEner >> dry_scale_dmx41
-42
-#define WET_ENER_WEIGHT(WetEner) WetEner = WetEner << wet_scale_dmx43
-44
-#define DRY_ENER_SUM_REAL(DryEner, dmxReal, n) \45
- DryEner += \46
- fMultDiv2(fPow2Div2(dmxReal << SF_DRY), pBP[n]) >> ((2 * SF_DRY) - 2)47
-48
-#define DRY_ENER_SUM_CPLX(DryEner, dmxReal, dmxImag, n) \49
- DryEner += fMultDiv2( \50
- fPow2Div2(dmxReal << SF_DRY) + fPow2Div2(dmxImag << SF_DRY), pBP[n])51
-52
#define CALC_WET_SCALE(dryIdx, wetIdx) \53
if ((DryEnerLD64[dryIdx] - STP_SCALE_LIMIT_HI_LD64) > WetEnerLD64[wetIdx]) { \54
scale[wetIdx] = STP_SCALE_LIMIT_HI; \55
56
int update_old_ener;57
};58
59
-inline void combineSignalReal(FIXP_DBL *hybOutputRealDry,60
- FIXP_DBL *hybOutputRealWet, int bands) {61
- int n;62
-63
- for (n = bands - 1; n >= 0; n--) {64
- *hybOutputRealDry = *hybOutputRealDry + *hybOutputRealWet;65
- hybOutputRealDry++, hybOutputRealWet++;66
- }67
-}68
-69
-inline void combineSignalRealScale1(FIXP_DBL *hybOutputRealDry,70
- FIXP_DBL *hybOutputRealWet, FIXP_DBL scaleX,71
- int bands) {72
- int n;73
-74
- for (n = bands - 1; n >= 0; n--) {75
- *hybOutputRealDry =76
- *hybOutputRealDry +77
- (fMultDiv2(*hybOutputRealWet, scaleX) << (SF_SCALE + 1));78
- hybOutputRealDry++, hybOutputRealWet++;79
- }80
-}81
-82
inline void combineSignalCplx(FIXP_DBL *hybOutputRealDry,83
FIXP_DBL *hybOutputImagDry,84
FIXP_DBL *hybOutputRealWet,85
86
int n;87
88
for (n = bands - 1; n >= 0; n--) {89
- *hybOutputRealDry = *hybOutputRealDry + *hybOutputRealWet;90
- *hybOutputImagDry = *hybOutputImagDry + *hybOutputImagWet;91
+ *hybOutputRealDry = fAddSaturate(*hybOutputRealDry, *hybOutputRealWet);92
+ *hybOutputImagDry = fAddSaturate(*hybOutputImagDry, *hybOutputImagWet);93
hybOutputRealDry++, hybOutputRealWet++;94
hybOutputImagDry++, hybOutputImagWet++;95
}96
97
FIXP_DBL scaleY;98
for (n = bands - 1; n >= 0; n--) {99
scaleY = fMultDiv2(scaleX, *pBP);100
- *hybOutputRealDry =101
- *hybOutputRealDry +102
- (fMultDiv2(*hybOutputRealWet, scaleY) << (SF_SCALE + 2));103
- *hybOutputImagDry =104
- *hybOutputImagDry +105
- (fMultDiv2(*hybOutputImagWet, scaleY) << (SF_SCALE + 2));106
+ *hybOutputRealDry = SATURATE_LEFT_SHIFT(107
+ (*hybOutputRealDry >> 1) +108
+ (fMultDiv2(*hybOutputRealWet, scaleY) << (SF_SCALE + 1)),109
+ 1, DFRACT_BITS);110
+ *hybOutputImagDry = SATURATE_LEFT_SHIFT(111
+ (*hybOutputImagDry >> 1) +112
+ (fMultDiv2(*hybOutputImagWet, scaleY) << (SF_SCALE + 1)),113
+ 1, DFRACT_BITS);114
hybOutputRealDry++, hybOutputRealWet++;115
hybOutputImagDry++, hybOutputImagWet++;116
pBP++;117
118
119
for (ch = 0; ch < MAX_OUTPUT_CHANNELS; ch++) {120
self->prev_tp_scale[ch] = FL2FXCONST_DBL(1.0f / (1 << SF_SCALE));121
- self->oldWetEnerLD64[ch] =122
- FL2FXCONST_DBL(0.34375f); /* 32768.0*32768.0/2^(44-26-10) */123
+ self->oldWetEnerLD64[ch] = FL2FXCONST_DBL(0.0);124
}125
for (ch = 0; ch < MAX_INPUT_CHANNELS; ch++) {126
- self->oldDryEnerLD64[ch] =127
- FL2FXCONST_DBL(0.1875f); /* 32768.0*32768.0/2^(44-26) */128
+ self->oldDryEnerLD64[ch] = FL2FXCONST_DBL(0.0);129
}130
131
self->BP = BP__FDK;132
133
{134
cplxBands = BP_GF_SIZE;135
cplxHybBands = self->hybridBands;136
- dry_scale_dmx = (2 * SF_DRY) - 2;137
+ if (self->treeConfig == TREE_212) {138
+ dry_scale_dmx = 2; /* 2 bits to compensate fMultDiv2() and fPow2Div2()139
+ used in energy calculation */140
+ } else {141
+ dry_scale_dmx = (2 * SF_DRY) - 2;142
+ }143
wet_scale_dmx = 2;144
}145
146
147
CalcLdData(hStpDec->runDryEner[ch] + ABS_THR__FDK);148
}149
for (ch = 0; ch < self->numOutputChannels; ch++) {150
- hStpDec->oldWetEnerLD64[ch] =151
- CalcLdData(hStpDec->runWetEner[ch] + ABS_THR2__FDK);152
+ if (self->treeConfig == TREE_212)153
+ hStpDec->oldWetEnerLD64[ch] =154
+ CalcLdData(hStpDec->runWetEner[ch] + ABS_THR__FDK);155
+ else156
+ hStpDec->oldWetEnerLD64[ch] =157
+ CalcLdData(hStpDec->runWetEner[ch] + ABS_THR2__FDK);158
}159
} else {160
hStpDec->update_old_ener++;161
162
pBP = hStpDec->BP_GF - BP_GF_START;163
switch (self->treeConfig) {164
case TREE_212:165
+ INT sMin, sNorm, sReal, sImag;166
+167
+ sReal = fMin(getScalefactor(&qmfOutputRealDry[i_LF][BP_GF_START],168
+ cplxBands - BP_GF_START),169
+ getScalefactor(&qmfOutputRealDry[i_RF][BP_GF_START],170
+ cplxBands - BP_GF_START));171
+ sImag = fMin(getScalefactor(&qmfOutputImagDry[i_LF][BP_GF_START],172
+ cplxBands - BP_GF_START),173
+ getScalefactor(&qmfOutputImagDry[i_RF][BP_GF_START],174
+ cplxBands - BP_GF_START));175
+ sMin = fMin(sReal, sImag) - 1;176
+177
for (n = BP_GF_START; n < cplxBands; n++) {178
- dmxReal0 = qmfOutputRealDry[i_LF][n] + qmfOutputRealDry[i_RF][n];179
- dmxImag0 = qmfOutputImagDry[i_LF][n] + qmfOutputImagDry[i_RF][n];180
- DRY_ENER_SUM_CPLX(DryEner0, dmxReal0, dmxImag0, n);181
+ dmxReal0 = scaleValue(qmfOutputRealDry[i_LF][n], sMin) +182
+ scaleValue(qmfOutputRealDry[i_RF][n], sMin);183
+ dmxImag0 = scaleValue(qmfOutputImagDry[i_LF][n], sMin) +184
+ scaleValue(qmfOutputImagDry[i_RF][n], sMin);185
+186
+ DryEner0 += (fMultDiv2(fPow2Div2(dmxReal0), pBP[n]) +187
+ fMultDiv2(fPow2Div2(dmxImag0), pBP[n])) >>188
+ SF_DRY_NRG;189
}190
- DRY_ENER_WEIGHT(DryEner0);191
+192
+ sNorm = SF_FREQ_DOMAIN_HEADROOM + SF_DRY_NRG + dry_scale_dmx -193
+ (2 * sMin) + nrgScale;194
+ DryEner0 = scaleValueSaturate(195
+ DryEner0, fMax(fMin(sNorm, DFRACT_BITS - 1), -(DFRACT_BITS - 1)));196
break;197
default:;198
}199
200
201
fdk-aac-2.0.1.tar.gz/libSACdec/src/sac_tsd.cpp -> fdk-aac-2.0.2.tar.gz/libSACdec/src/sac_tsd.cpp
Changed
56
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
12
RAM_ALIGN13
LNK_SECTION_CONSTDATA14
-static const FIXP_STP phiTsd[8] = {15
- STCP(0x7fffffff, 0x00000000), STCP(0x5a82799a, 0x5a82799a),16
- STCP(0x00000000, 0x7fffffff), STCP(0xa57d8666, 0x5a82799a),17
- STCP(0x80000000, 0x00000000), STCP(0xa57d8666, 0xa57d8666),18
- STCP(0x00000000, 0x80000000), STCP(0x5a82799a, 0xa57d8666),19
-};20
+static const FIXP_DPK phiTsd[8] = {21
+ {{(FIXP_DBL)0x7fffffff, (FIXP_DBL)0x00000000}},22
+ {{(FIXP_DBL)0x5a82799a, (FIXP_DBL)0x5a82799a}},23
+ {{(FIXP_DBL)0x00000000, (FIXP_DBL)0x7fffffff}},24
+ {{(FIXP_DBL)0xa57d8666, (FIXP_DBL)0x5a82799a}},25
+ {{(FIXP_DBL)0x80000000, (FIXP_DBL)0x00000000}},26
+ {{(FIXP_DBL)0xa57d8666, (FIXP_DBL)0xa57d8666}},27
+ {{(FIXP_DBL)0x00000000, (FIXP_DBL)0x80000000}},28
+ {{(FIXP_DBL)0x5a82799a, (FIXP_DBL)0xa57d8666}}};29
30
/*** Static Functions ***/31
static void longmult1(USHORT a[], USHORT b, USHORT d[], int len) {32
33
34
if (isTrSlot(pTsdData, ts)) {35
int k;36
- const FIXP_STP *phi = &phiTsd[pTsdData->bsTsdTrPhaseData[ts]];37
+ const FIXP_DPK *phi = &phiTsd[pTsdData->bsTsdTrPhaseData[ts]];38
FDK_ASSERT((pTsdData->bsTsdTrPhaseData[ts] >= 0) &&39
(pTsdData->bsTsdTrPhaseData[ts] < 8));40
41
/* d = d_nonTr + v_direct * exp(j * bsTsdTrPhaseData[ts]/4 * pi ) */42
for (k = TSD_START_BAND; k < numHybridBands; k++) {43
FIXP_DBL tempReal, tempImag;44
- cplxMult(&tempReal, &tempImag, pVdirectReal[k], pVdirectImag[k], *phi);45
- pDnonTrReal[k] += tempReal;46
- pDnonTrImag[k] += tempImag;47
+ cplxMultDiv2(&tempReal, &tempImag, pVdirectReal[k], pVdirectImag[k],48
+ *phi);49
+ pDnonTrReal[k] = SATURATE_LEFT_SHIFT(50
+ (pDnonTrReal[k] >> 2) + (tempReal >> 1), 2, DFRACT_BITS);51
+ pDnonTrImag[k] = SATURATE_LEFT_SHIFT(52
+ (pDnonTrImag[k] >> 2) + (tempImag >> 1), 2, DFRACT_BITS);53
}54
}55
56
fdk-aac-2.0.1.tar.gz/libSACenc/src/sacenc_lib.cpp -> fdk-aac-2.0.2.tar.gz/libSACenc/src/sacenc_lib.cpp
Changed
10
1
2
#define SACENC_LIB_VL1 03
#define SACENC_LIB_VL2 04
#define SACENC_LIB_TITLE "MPEG Surround Encoder"5
-#ifdef __ANDROID__6
+#ifdef SUPPRESS_BUILD_DATE_INFO7
#define SACENC_LIB_BUILD_DATE ""8
#define SACENC_LIB_BUILD_TIME ""9
#else10
fdk-aac-2.0.1.tar.gz/libSBRdec/include/sbrdecoder.h -> fdk-aac-2.0.2.tar.gz/libSBRdec/include/sbrdecoder.h
Changed
35
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
* error (0: core decoder found errors, 1: no errors).12
* \param psDecoded Pointer to a buffer holding a flag. Input: PS is13
* possible, Output: PS has been rendered.14
+ * \param inDataHeadroom Headroom of the SBR input time signal to prevent15
+ * clipping.16
+ * \param outDataHeadroom Pointer to headroom of the SBR output time signal to17
+ * prevent clipping.18
*19
* \return Error code.20
*/21
-SBR_ERROR sbrDecoder_Apply(HANDLE_SBRDECODER self, INT_PCM *input,22
- INT_PCM *timeData, const int timeDataSize,23
- int *numChannels, int *sampleRate,24
+SBR_ERROR sbrDecoder_Apply(HANDLE_SBRDECODER self, LONG *input, LONG *timeData,25
+ const int timeDataSize, int *numChannels,26
+ int *sampleRate,27
const FDK_channelMapDescr *const mapDescr,28
const int mapIdx, const int coreDecodedOk,29
- UCHAR *psDecoded);30
+ UCHAR *psDecoded, const INT inDataHeadroom,31
+ INT *outDataHeadroom);32
33
/**34
* \brief Close SBR decoder instance and free memory.35
fdk-aac-2.0.1.tar.gz/libSBRdec/src/HFgen_preFlat.cpp -> fdk-aac-2.0.2.tar.gz/libSBRdec/src/HFgen_preFlat.cpp
Changed
50
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
for (i = startSample; i < stopSample; i++) {12
maxVal |=13
(FIXP_DBL)((LONG)(sourceBufferReal[i][loBand]) ^14
- ((LONG)sourceBufferReal[i][loBand] >> (SAMPLE_BITS - 1)));15
+ ((LONG)sourceBufferReal[i][loBand] >> (DFRACT_BITS - 1)));16
maxVal |=17
(FIXP_DBL)((LONG)(sourceBufferImag[i][loBand]) ^18
- ((LONG)sourceBufferImag[i][loBand] >> (SAMPLE_BITS - 1)));19
+ ((LONG)sourceBufferImag[i][loBand] >> (DFRACT_BITS - 1)));20
}21
22
if (maxVal != FL2FX_DBL(0.0f)) {23
- reserve = fixMax(0, CntLeadingZeros(maxVal) - 2);24
+ reserve = CntLeadingZeros(maxVal) - 2;25
}26
27
nrg_ov = nrg = (FIXP_DBL)0;28
if (scale_nrg_ov > -31) {29
for (i = startSample; i < overlap; i++) {30
- nrg_ov += (fPow2Div2(sourceBufferReal[i][loBand] << reserve) +31
- fPow2Div2(sourceBufferImag[i][loBand] << reserve)) >>32
- sum_scale_ov;33
+ nrg_ov +=34
+ (fPow2Div2(scaleValue(sourceBufferReal[i][loBand], reserve)) +35
+ fPow2Div2(scaleValue(sourceBufferImag[i][loBand], reserve))) >>36
+ sum_scale_ov;37
}38
} else {39
scale_nrg_ov = 0;40
}41
if (scale_nrg > -31) {42
for (i = overlap; i < stopSample; i++) {43
- nrg += (fPow2Div2(sourceBufferReal[i][loBand] << reserve) +44
- fPow2Div2(sourceBufferImag[i][loBand] << reserve)) >>45
+ nrg += (fPow2Div2(scaleValue(sourceBufferReal[i][loBand], reserve)) +46
+ fPow2Div2(scaleValue(sourceBufferImag[i][loBand], reserve))) >>47
sum_scale;48
}49
} else {50
fdk-aac-2.0.1.tar.gz/libSBRdec/src/env_calc.cpp -> fdk-aac-2.0.2.tar.gz/libSBRdec/src/env_calc.cpp
Changed
201
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
12
#include "genericStds.h" /* need FDKpow() for debug outputs */13
14
+#define MAX_SFB_NRG_HEADROOM (1)15
+#define MAX_VAL_NRG_HEADROOM ((((FIXP_DBL)MAXVAL_DBL) >> MAX_SFB_NRG_HEADROOM))16
+17
typedef struct {18
FIXP_DBL nrgRef[MAX_FREQ_COEFFS];19
FIXP_DBL nrgEst[MAX_FREQ_COEFFS];20
21
}22
}23
24
-#define INTER_TES_SF_CHANGE 325
+#define INTER_TES_SF_CHANGE 426
27
typedef struct {28
FIXP_DBL subsample_power_low[(((1024) / (32) * (4) / 2) + (3 * (4)))];29
30
gain_sf[i] += gamma_sf + 1; /* +1 because of fMultDiv2() */31
32
/* set gain to at least 0.2f */33
- FIXP_DBL point_two = FL2FXCONST_DBL(0.8f); /* scaled up by 2 */34
- int point_two_sf = -2;35
-36
- FIXP_DBL tmp = gain[i];37
- if (point_two_sf < gain_sf[i]) {38
- point_two >>= gain_sf[i] - point_two_sf;39
- } else {40
- tmp >>= point_two_sf - gain_sf[i];41
- }42
-43
/* limit and calculate gain[i]^2 too */44
FIXP_DBL gain_pow2;45
int gain_pow2_sf;46
- if (tmp < point_two) {47
+48
+ if (fIsLessThan(gain[i], gain_sf[i], FL2FXCONST_DBL(0.2f), 0)) {49
gain[i] = FL2FXCONST_DBL(0.8f);50
gain_sf[i] = -2;51
gain_pow2 = FL2FXCONST_DBL(0.64f);52
53
fMin(DFRACT_BITS - 1, new_summand_sf - total_power_high_after_sf);54
total_power_high_after_sf = new_summand_sf;55
} else if (new_summand_sf < total_power_high_after_sf) {56
- subsample_power_high[i] >>= total_power_high_after_sf - new_summand_sf;57
+ subsample_power_high[i] >>=58
+ fMin(DFRACT_BITS - 1, total_power_high_after_sf - new_summand_sf);59
}60
total_power_high_after += subsample_power_high[i] >> preShift2;61
}62
63
*/64
if (!useLP)65
adj_e = h_sbr_cal_env->filtBufferNoise_e -66
- getScalefactor(h_sbr_cal_env->filtBufferNoise, noSubbands);67
+ getScalefactor(h_sbr_cal_env->filtBufferNoise, noSubbands) +68
+ (INT)MAX_SFB_NRG_HEADROOM;69
70
/*71
Scan for maximum reference energy to be able72
73
- Smoothing can smear high gains of the previous envelope into the74
current75
*/76
- maxSfbNrg_e += 6;77
+ maxSfbNrg_e += (6 + MAX_SFB_NRG_HEADROOM);78
79
adj_e = maxSfbNrg_e;80
// final_e should not exist for PVC fixfix framing81
82
- Smoothing can smear high gains of the previous envelope into the83
current84
*/85
- maxSfbNrg_e += 6;86
+ maxSfbNrg_e += (6 + MAX_SFB_NRG_HEADROOM);87
88
if (borders[i] < hHeaderData->numberTimeSlots)89
/* This envelope affects timeslots that belong to the output frame */90
91
92
for (k = 0; k < noSubbands; k++) {93
int sc = scale_change - pNrgs->nrgGain_e[k] + (sc_change - 1);94
- pNrgs->nrgGain[k] >>= sc;95
+ pNrgs->nrgGain[k] >>= fixMin(sc, DFRACT_BITS - 1);96
pNrgs->nrgGain_e[k] += sc;97
}98
99
100
for (k = 0; k < noSubbands; k++) {101
int sc =102
scale_change - h_sbr_cal_env->filtBuffer_e[k] + (sc_change - 1);103
- h_sbr_cal_env->filtBuffer[k] >>= sc;104
+ h_sbr_cal_env->filtBuffer[k] >>= fixMin(sc, DFRACT_BITS - 1);105
}106
}107
108
109
FDK_ASSERT(!iTES_enable); /* not supported */110
if (flags & SBRDEC_ELD_GRID) {111
/* FDKmemset(analysBufferReal[j], 0, 64 * sizeof(FIXP_DBL)); */112
- adjustTimeSlot_EldGrid(&analysBufferReal[j][lowSubband], pNrgs,113
- &h_sbr_cal_env->harmIndex, lowSubband,114
- noSubbands,115
- fMin(scale_change, DFRACT_BITS - 1),116
- noNoiseFlag, &h_sbr_cal_env->phaseIndex,117
- EXP2SCALE(adj_e) - sbrScaleFactor->lb_scale);118
+ adjustTimeSlot_EldGrid(119
+ &analysBufferReal[j][lowSubband], pNrgs,120
+ &h_sbr_cal_env->harmIndex, lowSubband, noSubbands,121
+ fMin(scale_change, DFRACT_BITS - 1), noNoiseFlag,122
+ &h_sbr_cal_env->phaseIndex,123
+ fMax(EXP2SCALE(adj_e) - sbrScaleFactor->lb_scale,124
+ -(DFRACT_BITS - 1)));125
} else {126
adjustTimeSlotLC(&analysBufferReal[j][lowSubband], pNrgs,127
&h_sbr_cal_env->harmIndex, lowSubband, noSubbands,128
129
diff = (int)(nrgGain_e[band] - filtBuffer_e[band]);130
if (diff > 0) {131
filtBuffer[band] >>=132
- diff; /* Compensate for the scale change by shifting the mantissa. */133
+ fMin(diff, DFRACT_BITS - 1); /* Compensate for the scale change by134
+ shifting the mantissa. */135
filtBuffer_e[band] += diff; /* New gain is bigger, use its exponent */136
} else if (diff < 0) {137
/* The buffered gains seem to be larger, but maybe there138
139
filtBuffer_e[band] -= reserve; /* Compensate in the exponent: */140
141
/* For the remaining difference, change the new gain value */142
- diff = fixMin(-(reserve + diff), DFRACT_BITS - 1);143
- nrgGain[band] >>= diff;144
+ diff = -(reserve + diff);145
+ nrgGain[band] >>= fMin(diff, DFRACT_BITS - 1);146
nrgGain_e[band] += diff;147
}148
}149
150
}151
152
/* gain = nrgRef / B */153
- FDK_divide_MantExp(nrgRef, nrgRef_e, b, b_e, ptrNrgGain, ptrNrgGain_e);154
+ INT result_exp = 0;155
+ *ptrNrgGain = fDivNorm(nrgRef, b, &result_exp);156
+ *ptrNrgGain_e = (SCHAR)result_exp + (nrgRef_e - b_e);157
+158
+ /* There could be a one bit diffs. This is important to compensate,159
+ because later in the code values are compared by exponent only. */160
+ int headroom = CountLeadingBits(*ptrNrgGain);161
+ *ptrNrgGain <<= headroom;162
+ *ptrNrgGain_e -= headroom;163
}164
}165
166
167
const FIXP_DBL *p_harmonicPhaseX = &harmonicPhaseX[harmIndex][0];168
const INT *p_harmonicPhase = &harmonicPhase[harmIndex][0];169
170
+ const FIXP_DBL max_val = MAX_VAL_NRG_HEADROOM >> scale_change;171
+ const FIXP_DBL min_val = -max_val;172
+173
*(ptrReal - 1) = fAddSaturate(174
*(ptrReal - 1),175
SATURATE_SHIFT(fMultDiv2(p_harmonicPhaseX[lowSubband & 1], pSineLevel[0]),176
177
FIXP_DBL sineLevel_curr = *pSineLevel++;178
phaseIndex = (phaseIndex + 1) & (SBR_NF_NO_RANDOM_VAL - 1);179
180
- signalReal = fMultDiv2(*ptrReal, *pGain++) << ((int)scale_change);181
+ signalReal = fMax(fMin(fMultDiv2(*ptrReal, *pGain++), max_val), min_val)182
+ << scale_change;183
sbNoise = *pNoiseLevel++;184
if (((INT)sineLevel_curr | noNoiseFlag) == 0) {185
signalReal +=186
- (fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[phaseIndex][0], sbNoise)187
- << 4);188
+ fMult(FDK_sbrDecoder_sbr_randomPhase[phaseIndex][0], sbNoise);189
}190
signalReal += sineLevel_curr * p_harmonicPhase[0];191
signalReal =192
193
FIXP_DBL sineLevel_curr = *pSineLevel++;194
phaseIndex = (phaseIndex + 1) & (SBR_NF_NO_RANDOM_VAL - 1);195
196
- signalReal = fMultDiv2(*ptrReal, *pGain++) << ((int)scale_change);197
+ signalReal = fMax(fMin(fMultDiv2(*ptrReal, *pGain++), max_val), min_val)198
+ << scale_change;199
sbNoise = *pNoiseLevel++;200
if (((INT)sineLevel_curr | noNoiseFlag) == 0) {201
fdk-aac-2.0.1.tar.gz/libSBRdec/src/env_extr.cpp -> fdk-aac-2.0.2.tar.gz/libSBRdec/src/env_extr.cpp
Changed
12
1
2
int nEnv, i, tranIdx;3
const int *pTable;4
5
+ if (tranPosInternal >= numberTimeSlots) {6
+ return 0;7
+ }8
+9
switch (numberTimeSlots) {10
case 8:11
pTable = FDK_sbrDecoder_envelopeTable_8[tranPosInternal];12
fdk-aac-2.0.1.tar.gz/libSBRdec/src/hbe.cpp -> fdk-aac-2.0.2.tar.gz/libSBRdec/src/hbe.cpp
Changed
28
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
hQmfTran->qmfOutBufSize = 2 * (hQmfTran->noCols / 2 + QMF_WIN_LEN - 1);12
13
hQmfTran->inBuf_F =14
- (INT_PCM*)FDKcalloc(QMF_SYNTH_CHANNELS + 20 + 1, sizeof(INT_PCM));15
+ (LONG*)FDKcalloc(QMF_SYNTH_CHANNELS + 20 + 1, sizeof(LONG));16
/* buffered time signal needs to be delayed by synthesis_size; max17
* synthesis_size = 20; */18
if (hQmfTran->inBuf_F == NULL) {19
20
g_r_m = fMultDiv2(tmp_r, factor_m) << shift;21
g_i_m = fMultDiv2(tmp_i, factor_m) << shift;22
g_e = scale_factor_hbe - (g_e + factor_e + gammaCenter_e + add);23
- fMax((INT)0, g_e);24
+ g_e = fMax((INT)0, g_e);25
*qmfHBEBufReal_F += g_r_m >> g_e;26
*qmfHBEBufImag_F += g_i_m >> g_e;27
}28
fdk-aac-2.0.1.tar.gz/libSBRdec/src/hbe.h -> fdk-aac-2.0.2.tar.gz/libSBRdec/src/hbe.h
Changed
39
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
} KEEP_STATES_SYNCED_MODE;12
13
struct hbeTransposer {14
+ FIXP_DBL anaQmfStates[HBE_QMF_FILTER_STATE_ANA_SIZE];15
+ FIXP_QSS synQmfStates[HBE_QMF_FILTER_STATE_SYN_SIZE];16
+17
int xOverQmf[MAX_NUM_PATCHES_HBE];18
19
int maxStretch;20
21
int stopBand;22
int bSbr41;23
24
- INT_PCM *inBuf_F;25
+ LONG *inBuf_F;26
FIXP_DBL **qmfInBufReal_F;27
FIXP_DBL **qmfInBufImag_F;28
29
30
FIXP_DBL const *synthesisQmfPreModCos_F;31
FIXP_DBL const *synthesisQmfPreModSin_F;32
33
- FIXP_QAS anaQmfStates[HBE_QMF_FILTER_STATE_ANA_SIZE];34
- FIXP_QSS synQmfStates[HBE_QMF_FILTER_STATE_SYN_SIZE];35
-36
FIXP_DBL **qmfHBEBufReal_F;37
FIXP_DBL **qmfHBEBufImag_F;38
39
fdk-aac-2.0.1.tar.gz/libSBRdec/src/lpp_tran.cpp -> fdk-aac-2.0.2.tar.gz/libSBRdec/src/lpp_tran.cpp
Changed
72
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
pSettings->nCols) +12
lowBandShift);13
}14
- dynamicScale = fixMax(15
- 0, dynamicScale - 1); /* one additional bit headroom to prevent -1.0 */16
+17
+ if (dynamicScale == 0) {18
+ /* In this special case the available headroom bits as well as19
+ ovLowBandShift and lowBandShift are zero. The spectrum is limited to20
+ prevent -1.0, so negative values for dynamicScale can be avoided. */21
+ for (i = 0; i < (LPC_ORDER + pSettings->overlap + pSettings->nCols);22
+ i++) {23
+ lowBandReal[i] = fixMax(lowBandReal[i], (FIXP_DBL)0x80000001);24
+ }25
+ if (!useLP) {26
+ for (i = 0; i < (LPC_ORDER + pSettings->overlap + pSettings->nCols);27
+ i++) {28
+ lowBandImag[i] = fixMax(lowBandImag[i], (FIXP_DBL)0x80000001);29
+ }30
+ }31
+ } else {32
+ dynamicScale =33
+ fixMax(0, dynamicScale -34
+ 1); /* one additional bit headroom to prevent -1.0 */35
+ }36
37
/*38
Scale temporal QMF buffer.39
40
pSettings->nCols) +41
lowBandShift);42
43
- dynamicScale = fixMax(44
- 0, dynamicScale - 1); /* one additional bit headroom to prevent -1.0 */45
+ dynamicScale =46
+ dynamicScale - 1; /* one additional bit headroom to prevent -1.0 */47
48
/*49
Scale temporal QMF buffer.50
51
} else { /* bw <= 0 */52
53
int descale = fixMin(DFRACT_BITS - 1, (LPC_SCALE_FACTOR + dynamicScale));54
+ dynamicScale +=55
+ 1; /* prevent negativ scale factor due to 'one additional bit56
+ headroom' */57
58
for (i = startSample; i < stopSample; i++) {59
FIXP_DBL accu1, accu2;60
61
dynamicScale;62
63
qmfBufferReal[i][loBand] =64
- (lowBandReal[LPC_ORDER + i] >> descale) + (accu1 << 1);65
+ (lowBandReal[LPC_ORDER + i] >> descale) + (accu1 << (1 + 1));66
qmfBufferImag[i][loBand] =67
- (lowBandImag[LPC_ORDER + i] >> descale) + (accu2 << 1);68
+ (lowBandImag[LPC_ORDER + i] >> descale) + (accu2 << (1 + 1));69
}70
} /* bw <= 0 */71
72
fdk-aac-2.0.1.tar.gz/libSBRdec/src/psdec.cpp -> fdk-aac-2.0.2.tar.gz/libSBRdec/src/psdec.cpp
Changed
58
1
2
int env, int usb) {3
INT group = 0;4
INT bin = 0;5
- INT noIidSteps, noFactors;6
+ INT noIidSteps;7
8
FIXP_SGL invL;9
FIXP_DBL ScaleL, ScaleR;10
- FIXP_DBL Alpha, Beta, AlphasValue;11
+ FIXP_DBL Alpha, Beta;12
FIXP_DBL h11r, h12r, h21r, h22r;13
14
const FIXP_DBL *PScaleFactors;15
16
if (h_ps_d->bsData[h_ps_d->processSlot].mpeg.bFineIidQ) {17
PScaleFactors = ScaleFactorsFine; /* values are shiftet right by one */18
noIidSteps = NO_IID_STEPS_FINE;19
- noFactors = NO_IID_LEVELS_FINE;20
} else {21
PScaleFactors = ScaleFactors; /* values are shiftet right by one */22
noIidSteps = NO_IID_STEPS;23
- noFactors = NO_IID_LEVELS;24
}25
26
/* dequantize and decode */27
28
29
/* ScaleR and ScaleL are scaled by 1 shift right */30
31
- ScaleL = ScaleR = 0;32
- if (noIidSteps + h_ps_d->specificTo.mpeg.pCoef->aaIidIndexMapped[env][bin] >= 0 && noIidSteps + h_ps_d->specificTo.mpeg.pCoef->aaIidIndexMapped[env][bin] < noFactors)33
- ScaleR = PScaleFactors[noIidSteps + h_ps_d->specificTo.mpeg.pCoef34
- ->aaIidIndexMapped[env][bin]];35
- if (noIidSteps - h_ps_d->specificTo.mpeg.pCoef->aaIidIndexMapped[env][bin] >= 0 && noIidSteps - h_ps_d->specificTo.mpeg.pCoef->aaIidIndexMapped[env][bin] < noFactors)36
- ScaleL = PScaleFactors[noIidSteps - h_ps_d->specificTo.mpeg.pCoef37
- ->aaIidIndexMapped[env][bin]];38
-39
- AlphasValue = 0;40
- if (h_ps_d->specificTo.mpeg.pCoef->aaIccIndexMapped[env][bin] >= 0)41
- AlphasValue = Alphas[h_ps_d->specificTo.mpeg.pCoef->aaIccIndexMapped[env][bin]];42
+ ScaleR = PScaleFactors[noIidSteps + h_ps_d->specificTo.mpeg.pCoef43
+ ->aaIidIndexMapped[env][bin]];44
+ ScaleL = PScaleFactors[noIidSteps - h_ps_d->specificTo.mpeg.pCoef45
+ ->aaIidIndexMapped[env][bin]];46
+47
Beta = fMult(48
- fMult(AlphasValue,49
+ fMult(Alphas[h_ps_d->specificTo.mpeg.pCoef->aaIccIndexMapped[env][bin]],50
(ScaleR - ScaleL)),51
FIXP_SQRT05);52
Alpha =53
- AlphasValue >> 1;54
+ Alphas[h_ps_d->specificTo.mpeg.pCoef->aaIccIndexMapped[env][bin]] >> 1;55
56
/* Alpha and Beta are now both scaled by 2 shifts right */57
58
fdk-aac-2.0.1.tar.gz/libSBRdec/src/pvc_dec.cpp -> fdk-aac-2.0.2.tar.gz/libSBRdec/src/pvc_dec.cpp
Changed
70
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
for (ksg = ksg_start; ksg < PVC_NBLOW; ksg++) {12
for (band = sg_borders[ksg]; band < sg_borders[ksg + 1]; band++) {13
/* The division by 8 == (RATE*lbw) is required algorithmically */14
- E[ksg] += (fPow2Div2(qmfR[band]) + fPow2Div2(qmfI[band])) >> 2;15
+ E[ksg] +=16
+ ((fPow2Div2(qmfR[band]) >> 1) + (fPow2Div2(qmfI[band]) >> 1)) >> 3;17
}18
}19
}20
21
if (E[ksg] > (FIXP_DBL)0) {22
/* 10/log2(10) = 0.752574989159953 * 2^2 */23
int exp_log;24
- FIXP_DBL nrg = CalcLog2(E[ksg], 2 * qmfExponent, &exp_log);25
+ FIXP_DBL nrg = CalcLog2(E[ksg], 2 * qmfExponent + 2, &exp_log);26
nrg = fMult(nrg, FL2FXCONST_SGL(LOG10FAC));27
nrg = scaleValue(nrg, exp_log - PVC_ESG_EXP + 2);28
pEsg[ksg] = fMax(nrg, FL2FXCONST_DBL(-10.0 / (1 << PVC_ESG_EXP)));29
30
E_high_exp[ksg] = 0;31
32
/* residual part */33
- accu = ((LONG)(SCHAR)*pTab2++) << (DFRACT_BITS - 8 - PVC_ESG_EXP +34
+ accu = ((LONG)(SCHAR)*pTab2++) << (DFRACT_BITS - 8 - PVC_ESG_EXP - 2 +35
pPvcDynamicData->pScalingCoef[3]);36
37
/* linear combination of lower grouped energies part */38
for (kb = 0; kb < PVC_NBLOW; kb++) {39
predCoeff = (FIXP_SGL)(40
(SHORT)(SCHAR)pTab1[kb * pPvcDynamicData->nbHigh + ksg] << 8);41
- predCoeff_exp = pPvcDynamicData->pScalingCoef[kb] +42
- 1; /* +1 to compensate for Div2 */43
- accu += fMultDiv2(E[kb], predCoeff) << predCoeff_exp;44
+ predCoeff_exp = -(pPvcDynamicData->pScalingCoef[kb] + 1 -45
+ 2); /* +1 to compensate for Div2; -2 for accu */46
+ accu += fMultDiv2(E[kb], predCoeff) >> predCoeff_exp;47
}48
/* convert back to linear domain */49
accu = fMult(accu, FL2FXCONST_SGL(LOG10FAC_INV));50
- accu = f2Pow(51
- accu, PVC_ESG_EXP - 1,52
- &predCoeff_exp); /* -1 compensates for exponent of LOG10FAC_INV */53
+ accu = f2Pow(accu, PVC_ESG_EXP - 1 + 2,54
+ &predCoeff_exp); /* -1 compensates for exponent of55
+ LOG10FAC_INV; +2 for accu */56
predictedEsgSlot[ksg] = accu;57
E_high_exp[ksg] = predCoeff_exp;58
if (predCoeff_exp > E_high_exp_max) {59
60
61
/* rescale output vector according to largest exponent */62
for (ksg = 0; ksg < pPvcDynamicData->nbHigh; ksg++) {63
- int scale = E_high_exp[ksg] - E_high_exp_max;64
- predictedEsgSlot[ksg] = scaleValue(predictedEsgSlot[ksg], scale);65
+ int scale = fMin(E_high_exp_max - E_high_exp[ksg], DFRACT_BITS - 1);66
+ predictedEsgSlot[ksg] = predictedEsgSlot[ksg] >> scale;67
}68
*predictedEsg_exp = E_high_exp_max;69
}70
fdk-aac-2.0.1.tar.gz/libSBRdec/src/sbr_dec.cpp -> fdk-aac-2.0.2.tar.gz/libSBRdec/src/sbr_dec.cpp
Changed
90
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
12
void sbr_dec(13
HANDLE_SBR_DEC hSbrDec, /*!< handle to Decoder channel */14
- INT_PCM *timeIn, /*!< pointer to input time signal */15
- INT_PCM *timeOut, /*!< pointer to output time signal */16
+ LONG *timeIn, /*!< pointer to input time signal */17
+ LONG *timeOut, /*!< pointer to output time signal */18
HANDLE_SBR_DEC hSbrDecRight, /*!< handle to Decoder channel right */19
- INT_PCM *timeOutRight, /*!< pointer to output time signal */20
+ LONG *timeOutRight, /*!< pointer to output time signal */21
const int strideOut, /*!< Time data traversal strideOut */22
HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */23
HANDLE_SBR_FRAME_DATA hFrameData, /*!< Control data of current frame */24
HANDLE_SBR_PREV_FRAME_DATA25
hPrevFrameData, /*!< Some control data of last frame */26
const int applyProcessing, /*!< Flag for SBR operation */27
- HANDLE_PS_DEC h_ps_d, const UINT flags, const int codecFrameSize) {28
+ HANDLE_PS_DEC h_ps_d, const UINT flags, const int codecFrameSize,29
+ const INT sbrInDataHeadroom) {30
int i, slot, reserve;31
int saveLbScale;32
int lastSlotOffs;33
34
/* temporary pointer / variable for QMF;35
required as we want to use temporary buffer36
creating one frame delay for HBE in LP mode */37
- INT_PCM *pTimeInQmf = timeIn;38
+ LONG *pTimeInQmf = timeIn;39
40
/* Number of QMF timeslots in the overlap buffer: */41
int ov_len = hSbrDec->LppTrans.pSettings->overlap;42
43
} else {44
C_AALLOC_SCRATCH_START(qmfTemp, FIXP_DBL, 2 * (64));45
qmfAnalysisFiltering(&hSbrDec->qmfDomainInCh->fb, pReal, pImag,46
- &hSbrDec->qmfDomainInCh->scaling, pTimeInQmf, 0, 1,47
- qmfTemp);48
+ &hSbrDec->qmfDomainInCh->scaling, pTimeInQmf,49
+ 0 + sbrInDataHeadroom, 1, qmfTemp);50
51
C_AALLOC_SCRATCH_END(qmfTemp, FIXP_DBL, 2 * (64));52
}53
54
55
if (!(flags & SBRDEC_PS_DECODED)) {56
if (!(flags & SBRDEC_SKIP_QMF_SYN)) {57
- int outScalefactor = 0;58
+ int outScalefactor = -(8);59
60
if (h_ps_d != NULL) {61
h_ps_d->procFrameBased = 1; /* we here do frame based processing */62
63
*/64
FDK_ASSERT(hSbrDec->qmfDomainInCh->pGlobalConf->nBandsSynthesis <=65
QMF_MAX_SYNTHESIS_BANDS);66
+ qmfChangeOutScalefactor(synQmfRight, -(8));67
FDKmemcpy(synQmfRight->FilterStates, synQmf->FilterStates,68
9 * hSbrDec->qmfDomainInCh->pGlobalConf->nBandsSynthesis *69
sizeof(FIXP_QSS));70
71
: scaleFactorLowBand_no_ov,72
scaleFactorHighBand, synQmf->lsb, synQmf->usb);73
74
- outScalefactorL = outScalefactorR = 1; /* psDiffScale! (MPEG-PS) */75
+ outScalefactorL = outScalefactorR =76
+ 1 + sbrInDataHeadroom; /* psDiffScale! (MPEG-PS) */77
}78
79
sbrDecoder_drcApplySlot(/* right channel */80
81
outScalefactorL += maxShift;82
83
if (!(flags & SBRDEC_SKIP_QMF_SYN)) {84
+ qmfChangeOutScalefactor(synQmf, -(8));85
+ qmfChangeOutScalefactor(synQmfRight, -(8));86
+87
qmfSynthesisFilteringSlot(88
synQmfRight, rQmfReal, /* QMF real buffer */89
rQmfImag, /* QMF imag buffer */90
fdk-aac-2.0.1.tar.gz/libSBRdec/src/sbr_dec.h -> fdk-aac-2.0.2.tar.gz/libSBRdec/src/sbr_dec.h
Changed
33
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
12
void sbr_dec(13
HANDLE_SBR_DEC hSbrDec, /*!< handle to Decoder channel */14
- INT_PCM *timeIn, /*!< pointer to input time signal */15
- INT_PCM *timeOut, /*!< pointer to output time signal */16
+ LONG *timeIn, /*!< pointer to input time signal */17
+ LONG *timeOut, /*!< pointer to output time signal */18
HANDLE_SBR_DEC hSbrDecRight, /*!< handle to Decoder channel right */19
- INT_PCM *timeOutRight, /*!< pointer to output time signal */20
+ LONG *timeOutRight, /*!< pointer to output time signal */21
INT strideOut, /*!< Time data traversal strideOut */22
HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */23
HANDLE_SBR_FRAME_DATA hFrameData, /*!< Control data of current frame */24
HANDLE_SBR_PREV_FRAME_DATA25
hPrevFrameData, /*!< Some control data of last frame */26
const int applyProcessing, /*!< Flag for SBR operation */27
- HANDLE_PS_DEC h_ps_d, const UINT flags, const int codecFrameSize);28
+ HANDLE_PS_DEC h_ps_d, const UINT flags, const int codecFrameSize,29
+ const INT sbrInDataHeadroom);30
31
SBR_ERROR32
createSbrDec(SBR_CHANNEL *hSbrChannel, HANDLE_SBR_HEADER_DATA hHeaderData,33
fdk-aac-2.0.1.tar.gz/libSBRdec/src/sbr_ram.cpp -> fdk-aac-2.0.2.tar.gz/libSBRdec/src/sbr_ram.cpp
Changed
20
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
12
#include "sbr_ram.h"13
14
-#define WORKBUFFER1_TAG 215
-#define WORKBUFFER2_TAG 316
-17
/*!18
\name StaticSbrData19
20
fdk-aac-2.0.1.tar.gz/libSBRdec/src/sbr_ram.h -> fdk-aac-2.0.2.tar.gz/libSBRdec/src/sbr_ram.h
Changed
20
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
flushed consecutively. */12
13
UINT flags;14
+15
+ INT sbrInDataHeadroom; /* Headroom of the SBR input time signal to prevent16
+ clipping */17
};18
19
H_ALLOC_MEM(Ram_SbrDecElement, SBR_DECODER_ELEMENT)20
fdk-aac-2.0.1.tar.gz/libSBRdec/src/sbr_rom.cpp -> fdk-aac-2.0.2.tar.gz/libSBRdec/src/sbr_rom.cpp
Changed
201
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
random numbers.12
*/13
const FIXP_SGL FDK_sbrDecoder_sbr_randomPhase[SBR_NF_NO_RANDOM_VAL][2] = {14
- {FL2FXCONST_SGL(-0.99948153278296f / 8.0),15
- FL2FXCONST_SGL(-0.59483417516607f / 8.0)},16
- {FL2FXCONST_SGL(0.97113454393991f / 8.0),17
- FL2FXCONST_SGL(-0.67528515225647f / 8.0)},18
- {FL2FXCONST_SGL(0.14130051758487f / 8.0),19
- FL2FXCONST_SGL(-0.95090983575689f / 8.0)},20
- {FL2FXCONST_SGL(-0.47005496701697f / 8.0),21
- FL2FXCONST_SGL(-0.37340549728647f / 8.0)},22
- {FL2FXCONST_SGL(0.80705063769351f / 8.0),23
- FL2FXCONST_SGL(0.29653668284408f / 8.0)},24
- {FL2FXCONST_SGL(-0.38981478896926f / 8.0),25
- FL2FXCONST_SGL(0.89572605717087f / 8.0)},26
- {FL2FXCONST_SGL(-0.01053049862020f / 8.0),27
- FL2FXCONST_SGL(-0.66959058036166f / 8.0)},28
- {FL2FXCONST_SGL(-0.91266367957293f / 8.0),29
- FL2FXCONST_SGL(-0.11522938140034f / 8.0)},30
- {FL2FXCONST_SGL(0.54840422910309f / 8.0),31
- FL2FXCONST_SGL(0.75221367176302f / 8.0)},32
- {FL2FXCONST_SGL(0.40009252867955f / 8.0),33
- FL2FXCONST_SGL(-0.98929400334421f / 8.0)},34
- {FL2FXCONST_SGL(-0.99867974711855f / 8.0),35
- FL2FXCONST_SGL(-0.88147068645358f / 8.0)},36
- {FL2FXCONST_SGL(-0.95531076805040f / 8.0),37
- FL2FXCONST_SGL(0.90908757154593f / 8.0)},38
- {FL2FXCONST_SGL(-0.45725933317144f / 8.0),39
- FL2FXCONST_SGL(-0.56716323646760f / 8.0)},40
- {FL2FXCONST_SGL(-0.72929675029275f / 8.0),41
- FL2FXCONST_SGL(-0.98008272727324f / 8.0)},42
- {FL2FXCONST_SGL(0.75622801399036f / 8.0),43
- FL2FXCONST_SGL(0.20950329995549f / 8.0)},44
- {FL2FXCONST_SGL(0.07069442601050f / 8.0),45
- FL2FXCONST_SGL(-0.78247898470706f / 8.0)},46
- {FL2FXCONST_SGL(0.74496252926055f / 8.0),47
- FL2FXCONST_SGL(-0.91169004445807f / 8.0)},48
- {FL2FXCONST_SGL(-0.96440182703856f / 8.0),49
- FL2FXCONST_SGL(-0.94739918296622f / 8.0)},50
- {FL2FXCONST_SGL(0.30424629369539f / 8.0),51
- FL2FXCONST_SGL(-0.49438267012479f / 8.0)},52
- {FL2FXCONST_SGL(0.66565033746925f / 8.0),53
- FL2FXCONST_SGL(0.64652935542491f / 8.0)},54
- {FL2FXCONST_SGL(0.91697008020594f / 8.0),55
- FL2FXCONST_SGL(0.17514097332009f / 8.0)},56
- {FL2FXCONST_SGL(-0.70774918760427f / 8.0),57
- FL2FXCONST_SGL(0.52548653416543f / 8.0)},58
- {FL2FXCONST_SGL(-0.70051415345560f / 8.0),59
- FL2FXCONST_SGL(-0.45340028808763f / 8.0)},60
- {FL2FXCONST_SGL(-0.99496513054797f / 8.0),61
- FL2FXCONST_SGL(-0.90071908066973f / 8.0)},62
- {FL2FXCONST_SGL(0.98164490790123f / 8.0),63
- FL2FXCONST_SGL(-0.77463155528697f / 8.0)},64
- {FL2FXCONST_SGL(-0.54671580548181f / 8.0),65
- FL2FXCONST_SGL(-0.02570928536004f / 8.0)},66
- {FL2FXCONST_SGL(-0.01689629065389f / 8.0),67
- FL2FXCONST_SGL(0.00287506445732f / 8.0)},68
- {FL2FXCONST_SGL(-0.86110349531986f / 8.0),69
- FL2FXCONST_SGL(0.42548583726477f / 8.0)},70
- {FL2FXCONST_SGL(-0.98892980586032f / 8.0),71
- FL2FXCONST_SGL(-0.87881132267556f / 8.0)},72
- {FL2FXCONST_SGL(0.51756627678691f / 8.0),73
- FL2FXCONST_SGL(0.66926784710139f / 8.0)},74
- {FL2FXCONST_SGL(-0.99635026409640f / 8.0),75
- FL2FXCONST_SGL(-0.58107730574765f / 8.0)},76
- {FL2FXCONST_SGL(-0.99969370862163f / 8.0),77
- FL2FXCONST_SGL(0.98369989360250f / 8.0)},78
- {FL2FXCONST_SGL(0.55266258627194f / 8.0),79
- FL2FXCONST_SGL(0.59449057465591f / 8.0)},80
- {FL2FXCONST_SGL(0.34581177741673f / 8.0),81
- FL2FXCONST_SGL(0.94879421061866f / 8.0)},82
- {FL2FXCONST_SGL(0.62664209577999f / 8.0),83
- FL2FXCONST_SGL(-0.74402970906471f / 8.0)},84
- {FL2FXCONST_SGL(-0.77149701404973f / 8.0),85
- FL2FXCONST_SGL(-0.33883658042801f / 8.0)},86
- {FL2FXCONST_SGL(-0.91592244254432f / 8.0),87
- FL2FXCONST_SGL(0.03687901376713f / 8.0)},88
- {FL2FXCONST_SGL(-0.76285492357887f / 8.0),89
- FL2FXCONST_SGL(-0.91371867919124f / 8.0)},90
- {FL2FXCONST_SGL(0.79788337195331f / 8.0),91
- FL2FXCONST_SGL(-0.93180971199849f / 8.0)},92
- {FL2FXCONST_SGL(0.54473080610200f / 8.0),93
- FL2FXCONST_SGL(-0.11919206037186f / 8.0)},94
- {FL2FXCONST_SGL(-0.85639281671058f / 8.0),95
- FL2FXCONST_SGL(0.42429854760451f / 8.0)},96
- {FL2FXCONST_SGL(-0.92882402971423f / 8.0),97
- FL2FXCONST_SGL(0.27871809078609f / 8.0)},98
- {FL2FXCONST_SGL(-0.11708371046774f / 8.0),99
- FL2FXCONST_SGL(-0.99800843444966f / 8.0)},100
- {FL2FXCONST_SGL(0.21356749817493f / 8.0),101
- FL2FXCONST_SGL(-0.90716295627033f / 8.0)},102
- {FL2FXCONST_SGL(-0.76191692573909f / 8.0),103
- FL2FXCONST_SGL(0.99768118356265f / 8.0)},104
- {FL2FXCONST_SGL(0.98111043100884f / 8.0),105
- FL2FXCONST_SGL(-0.95854459734407f / 8.0)},106
- {FL2FXCONST_SGL(-0.85913269895572f / 8.0),107
- FL2FXCONST_SGL(0.95766566168880f / 8.0)},108
- {FL2FXCONST_SGL(-0.93307242253692f / 8.0),109
- FL2FXCONST_SGL(0.49431757696466f / 8.0)},110
- {FL2FXCONST_SGL(0.30485754879632f / 8.0),111
- FL2FXCONST_SGL(-0.70540034357529f / 8.0)},112
- {FL2FXCONST_SGL(0.85289650925190f / 8.0),113
- FL2FXCONST_SGL(0.46766131791044f / 8.0)},114
- {FL2FXCONST_SGL(0.91328082618125f / 8.0),115
- FL2FXCONST_SGL(-0.99839597361769f / 8.0)},116
- {FL2FXCONST_SGL(-0.05890199924154f / 8.0),117
- FL2FXCONST_SGL(0.70741827819497f / 8.0)},118
- {FL2FXCONST_SGL(0.28398686150148f / 8.0),119
- FL2FXCONST_SGL(0.34633555702188f / 8.0)},120
- {FL2FXCONST_SGL(0.95258164539612f / 8.0),121
- FL2FXCONST_SGL(-0.54893416026939f / 8.0)},122
- {FL2FXCONST_SGL(-0.78566324168507f / 8.0),123
- FL2FXCONST_SGL(-0.75568541079691f / 8.0)},124
- {FL2FXCONST_SGL(-0.95789495447877f / 8.0),125
- FL2FXCONST_SGL(-0.20423194696966f / 8.0)},126
- {FL2FXCONST_SGL(0.82411158711197f / 8.0),127
- FL2FXCONST_SGL(0.96654618432562f / 8.0)},128
- {FL2FXCONST_SGL(-0.65185446735885f / 8.0),129
- FL2FXCONST_SGL(-0.88734990773289f / 8.0)},130
- {FL2FXCONST_SGL(-0.93643603134666f / 8.0),131
- FL2FXCONST_SGL(0.99870790442385f / 8.0)},132
- {FL2FXCONST_SGL(0.91427159529618f / 8.0),133
- FL2FXCONST_SGL(-0.98290505544444f / 8.0)},134
- {FL2FXCONST_SGL(-0.70395684036886f / 8.0),135
- FL2FXCONST_SGL(0.58796798221039f / 8.0)},136
- {FL2FXCONST_SGL(0.00563771969365f / 8.0),137
- FL2FXCONST_SGL(0.61768196727244f / 8.0)},138
- {FL2FXCONST_SGL(0.89065051931895f / 8.0),139
- FL2FXCONST_SGL(0.52783352697585f / 8.0)},140
- {FL2FXCONST_SGL(-0.68683707712762f / 8.0),141
- FL2FXCONST_SGL(0.80806944710339f / 8.0)},142
- {FL2FXCONST_SGL(0.72165342518718f / 8.0),143
- FL2FXCONST_SGL(-0.69259857349564f / 8.0)},144
- {FL2FXCONST_SGL(-0.62928247730667f / 8.0),145
- FL2FXCONST_SGL(0.13627037407335f / 8.0)},146
- {FL2FXCONST_SGL(0.29938434065514f / 8.0),147
- FL2FXCONST_SGL(-0.46051329682246f / 8.0)},148
- {FL2FXCONST_SGL(-0.91781958879280f / 8.0),149
- FL2FXCONST_SGL(-0.74012716684186f / 8.0)},150
- {FL2FXCONST_SGL(0.99298717043688f / 8.0),151
- FL2FXCONST_SGL(0.40816610075661f / 8.0)},152
- {FL2FXCONST_SGL(0.82368298622748f / 8.0),153
- FL2FXCONST_SGL(-0.74036047190173f / 8.0)},154
- {FL2FXCONST_SGL(-0.98512833386833f / 8.0),155
- FL2FXCONST_SGL(-0.99972330709594f / 8.0)},156
- {FL2FXCONST_SGL(-0.95915368242257f / 8.0),157
- FL2FXCONST_SGL(-0.99237800466040f / 8.0)},158
- {FL2FXCONST_SGL(-0.21411126572790f / 8.0),159
- FL2FXCONST_SGL(-0.93424819052545f / 8.0)},160
- {FL2FXCONST_SGL(-0.68821476106884f / 8.0),161
- FL2FXCONST_SGL(-0.26892306315457f / 8.0)},162
- {FL2FXCONST_SGL(0.91851997982317f / 8.0),163
- FL2FXCONST_SGL(0.09358228901785f / 8.0)},164
- {FL2FXCONST_SGL(-0.96062769559127f / 8.0),165
- FL2FXCONST_SGL(0.36099095133739f / 8.0)},166
- {FL2FXCONST_SGL(0.51646184922287f / 8.0),167
- FL2FXCONST_SGL(-0.71373332873917f / 8.0)},168
- {FL2FXCONST_SGL(0.61130721139669f / 8.0),169
- FL2FXCONST_SGL(0.46950141175917f / 8.0)},170
- {FL2FXCONST_SGL(0.47336129371299f / 8.0),171
- FL2FXCONST_SGL(-0.27333178296162f / 8.0)},172
- {FL2FXCONST_SGL(0.90998308703519f / 8.0),173
- FL2FXCONST_SGL(0.96715662938132f / 8.0)},174
- {FL2FXCONST_SGL(0.44844799194357f / 8.0),175
- FL2FXCONST_SGL(0.99211574628306f / 8.0)},176
- {FL2FXCONST_SGL(0.66614891079092f / 8.0),177
- FL2FXCONST_SGL(0.96590176169121f / 8.0)},178
- {FL2FXCONST_SGL(0.74922239129237f / 8.0),179
- FL2FXCONST_SGL(-0.89879858826087f / 8.0)},180
- {FL2FXCONST_SGL(-0.99571588506485f / 8.0),181
- FL2FXCONST_SGL(0.52785521494349f / 8.0)},182
- {FL2FXCONST_SGL(0.97401082477563f / 8.0),183
- FL2FXCONST_SGL(-0.16855870075190f / 8.0)},184
- {FL2FXCONST_SGL(0.72683747733879f / 8.0),185
- FL2FXCONST_SGL(-0.48060774432251f / 8.0)},186
- {FL2FXCONST_SGL(0.95432193457128f / 8.0),187
- FL2FXCONST_SGL(0.68849603408441f / 8.0)},188
- {FL2FXCONST_SGL(-0.72962208425191f / 8.0),189
- FL2FXCONST_SGL(-0.76608443420917f / 8.0)},190
- {FL2FXCONST_SGL(-0.85359479233537f / 8.0),191
- FL2FXCONST_SGL(0.88738125901579f / 8.0)},192
- {FL2FXCONST_SGL(-0.81412430338535f / 8.0),193
- FL2FXCONST_SGL(-0.97480768049637f / 8.0)},194
- {FL2FXCONST_SGL(-0.87930772356786f / 8.0),195
- FL2FXCONST_SGL(0.74748307690436f / 8.0)},196
- {FL2FXCONST_SGL(-0.71573331064977f / 8.0),197
- FL2FXCONST_SGL(-0.98570608178923f / 8.0)},198
- {FL2FXCONST_SGL(0.83524300028228f / 8.0),199
- FL2FXCONST_SGL(0.83702537075163f / 8.0)},200
- {FL2FXCONST_SGL(-0.48086065601423f / 8.0),201
fdk-aac-2.0.1.tar.gz/libSBRdec/src/sbrdec_freq_sca.cpp -> fdk-aac-2.0.2.tar.gz/libSBRdec/src/sbrdec_freq_sca.cpp
Changed
29
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
}12
}13
14
+ stopMin = fMin(stopMin, 64);15
+16
/*17
Choose a stop band between k1 and 64 depending on stopFreq (0..13),18
based on a logarithmic scale.19
20
step = FL2FXCONST_DBL(0.0f);21
}22
}23
- return FX_DBL2FX_SGL(bandfactor << 1);24
+ return (bandfactor >= FL2FXCONST_DBL(0.5)) ? (FIXP_SGL)MAXVAL_SGL25
+ : FX_DBL2FX_SGL(bandfactor << 1);26
}27
28
/*!29
fdk-aac-2.0.1.tar.gz/libSBRdec/src/sbrdecoder.cpp -> fdk-aac-2.0.2.tar.gz/libSBRdec/src/sbrdecoder.cpp
Changed
201
1
2
/* -----------------------------------------------------------------------------3
Software License for The Fraunhofer FDK AAC Codec Library for Android4
5
-© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten6
+© Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten7
Forschung e.V. All rights reserved.8
9
1. INTRODUCTION10
11
#include "env_extr.h"12
#include "sbr_dec.h"13
#include "env_dec.h"14
-#include "sbr_crc.h"15
+#include "FDK_crc.h"16
#include "sbr_ram.h"17
#include "sbr_rom.h"18
#include "lpp_tran.h"19
#include "transcendent.h"20
21
-#include "FDK_crc.h"22
-23
#include "sbrdec_drc.h"24
25
#include "psbitdec.h"26
27
/* Decoder library info */28
#define SBRDECODER_LIB_VL0 329
-#define SBRDECODER_LIB_VL1 030
+#define SBRDECODER_LIB_VL1 131
#define SBRDECODER_LIB_VL2 032
#define SBRDECODER_LIB_TITLE "SBR Decoder"33
-#ifdef __ANDROID__34
+#ifdef SUPPRESS_BUILD_DATE_INFO35
#define SBRDECODER_LIB_BUILD_DATE ""36
#define SBRDECODER_LIB_BUILD_TIME ""37
#else38
39
self->numSbrChannels -= self->pSbrElement[elementIndex]->nChannels;40
}41
42
- /* Save element ID for sanity checks and to have a fallback for concealment.43
- */44
- self->pSbrElement[elementIndex]->elementID = elementID;45
-46
/* Determine amount of channels for this element */47
switch (elementID) {48
case ID_NONE:49
50
}51
52
/* Sanity check to avoid memory leaks */53
- if (elChannels < self->pSbrElement[elementIndex]->nChannels) {54
+ if (elChannels < self->pSbrElement[elementIndex]->nChannels ||55
+ (self->numSbrChannels + elChannels) > (8) + (1)) {56
self->numSbrChannels += self->pSbrElement[elementIndex]->nChannels;57
sbrError = SBRDEC_PARSE_ERROR;58
goto bail;59
}60
61
+ /* Save element ID for sanity checks and to have a fallback for concealment.62
+ */63
+ self->pSbrElement[elementIndex]->elementID = elementID;64
self->pSbrElement[elementIndex]->nChannels = elChannels;65
66
for (ch = 0; ch < elChannels; ch++) {67
68
if (sbrError == SBRDEC_OK) {69
hSbrHeader->syncState = SBR_HEADER;70
hSbrHeader->status |= SBRDEC_HDR_STAT_UPDATE;71
+ } else {72
+ hSbrHeader->syncState = SBR_NOT_INITIALIZED;73
+ hSbrHeader->status = HEADER_ERROR;74
}75
- /* else {76
- Since we already have overwritten the old SBR header the only way out77
- is UPSAMPLING! This will be prepared in the next step.78
- } */79
}80
}81
}82
83
SBR_HEADER_STATUS headerStatus = HEADER_NOT_PRESENT;84
85
INT startPos = FDKgetValidBits(hBs);86
- INT CRCLen = 0;87
+ FDK_CRCINFO crcInfo;88
+ INT crcReg = 0;89
+ USHORT sbrCrc = 0;90
+ UINT crcPoly;91
+ UINT crcStartValue = 0;92
+ UINT crcLen;93
+94
HANDLE_FDK_BITSTREAM hBsOriginal = hBs;95
FDK_BITSTREAM bsBwd;96
97
- FDK_CRCINFO crcInfo;98
- INT crcReg = 0;99
- USHORT drmSbrCrc = 0;100
const int fGlobalIndependencyFlag = acFlags & AC_INDEP;101
const int bs_pvc = acElFlags[elementIndex] & AC_EL_USAC_PVC;102
const int bs_interTes = acElFlags[elementIndex] & AC_EL_USAC_ITES;103
int stereo;104
int fDoDecodeSbrData = 1;105
+ int alignBits = 0;106
107
int lastSlot, lastHdrSlot = 0, thisHdrSlot = 0;108
109
110
if (fDoDecodeSbrData) {111
if (crcFlag) {112
switch (self->coreCodec) {113
- case AOT_ER_AAC_ELD:114
- FDKpushFor(hBs, 10);115
- /* check sbrcrc later: we don't know the payload length now */116
- break;117
case AOT_DRM_AAC:118
case AOT_DRM_SURROUND:119
- drmSbrCrc = (USHORT)FDKreadBits(hBs, 8);120
- /* Setup CRC decoder */121
- FDKcrcInit(&crcInfo, 0x001d, 0xFFFF, 8);122
- /* Start CRC region */123
- crcReg = FDKcrcStartReg(&crcInfo, hBs, 0);124
+ crcPoly = 0x001d;125
+ crcLen = 8;126
+ crcStartValue = 0x000000ff;127
break;128
default:129
- CRCLen = bsPayLen - 10; /* change: 0 => i */130
- if (CRCLen < 0) {131
- fDoDecodeSbrData = 0;132
- } else {133
- fDoDecodeSbrData = SbrCrcCheck(hBs, CRCLen);134
- }135
+ crcPoly = 0x0633;136
+ crcLen = 10;137
+ crcStartValue = 0x00000000;138
break;139
}140
+ sbrCrc = (USHORT)FDKreadBits(hBs, crcLen);141
+ /* Setup CRC decoder */142
+ FDKcrcInit(&crcInfo, crcPoly, crcStartValue, crcLen);143
+ /* Start CRC region */144
+ crcReg = FDKcrcStartReg(&crcInfo, hBs, 0);145
}146
} /* if (fDoDecodeSbrData) */147
148
149
valBits = (INT)FDKgetValidBits(hBs);150
}151
152
- if (crcFlag) {153
- switch (self->coreCodec) {154
- case AOT_ER_AAC_ELD: {155
- /* late crc check for eld */156
- INT payloadbits =157
- (INT)startPos - (INT)FDKgetValidBits(hBs) - startPos;158
- INT crcLen = payloadbits - 10;159
- FDKpushBack(hBs, payloadbits);160
- fDoDecodeSbrData = SbrCrcCheck(hBs, crcLen);161
- FDKpushFor(hBs, crcLen);162
- } break;163
- case AOT_DRM_AAC:164
- case AOT_DRM_SURROUND:165
- /* End CRC region */166
- FDKcrcEndReg(&crcInfo, hBs, crcReg);167
- /* Check CRC */168
- if ((FDKcrcGetCRC(&crcInfo) ^ 0xFF) != drmSbrCrc) {169
- fDoDecodeSbrData = 0;170
- if (headerStatus != HEADER_NOT_PRESENT) {171
- headerStatus = HEADER_ERROR;172
- hSbrHeader->syncState = SBR_NOT_INITIALIZED;173
- }174
- }175
- break;176
- default:177
- break;178
- }179
- }180
-181
/* sanity check of remaining bits */182
if (valBits < 0) {183
fDoDecodeSbrData = 0;184
185
case AOT_AAC_LC: {186
/* This sanity check is only meaningful with General Audio187
* bitstreams */188
- int alignBits = valBits & 0x7;189
+ alignBits = valBits & 0x7;190
191
if (valBits > alignBits) {192
fDoDecodeSbrData = 0;193
194
errorStatus = SBRDEC_PARSE_ERROR;195
}196
197
+ if (crcFlag && (hSbrHeader->syncState >= SBR_HEADER) && fDoDecodeSbrData) {198
+ FDKpushFor(hBs, alignBits);199
+ FDKcrcEndReg(&crcInfo, hBs, crcReg); /* End CRC region */200
+ FDKpushBack(hBs, alignBits);201
fdk-aac-2.0.1.tar.gz/libSBRenc/src/sbr_encoder.cpp -> fdk-aac-2.0.2.tar.gz/libSBRenc/src/sbr_encoder.cpp
Changed
10
1
2
info->version =3
LIB_VERSION(SBRENCODER_LIB_VL0, SBRENCODER_LIB_VL1, SBRENCODER_LIB_VL2);4
LIB_VERSION_STRING(info);5
-#ifdef __ANDROID__6
+#ifdef SUPPRESS_BUILD_DATE_INFO7
info->build_date = "";8
info->build_time = "";9
#else10