Request 6530 (new)

fdk-aac.changes Changed

fdk-aac.spec Changed

fdk-aac-2.0.2.tar.gz/libSBRdec/src/arm Deleted

fdk-aac-2.0.2.tar.gz/libSBRdec/src/arm/lpp_tran_arm.cpp Deleted

@@ -1,159 +0,0 @@
-/* -----------------------------------------------------------------------------
-Software License for The Fraunhofer FDK AAC Codec Library for Android
-
-© Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
-Forschung e.V. All rights reserved.
-
- 1.    INTRODUCTION
-The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software
-that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding
-scheme for digital audio. This FDK AAC Codec software is intended to be used on
-a wide variety of Android devices.
-
-AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient
-general perceptual audio codecs. AAC-ELD is considered the best-performing
-full-bandwidth communications codec by independent studies and is widely
-deployed. AAC has been standardized by ISO and IEC as part of the MPEG
-specifications.
-
-Patent licenses for necessary patent claims for the FDK AAC Codec (including
-those of Fraunhofer) may be obtained through Via Licensing
-(www.vialicensing.com) or through the respective patent owners individually for
-the purpose of encoding or decoding bit streams in products that are compliant
-with the ISO/IEC MPEG audio standards. Please note that most manufacturers of
-Android devices already license these patent claims through Via Licensing or
-directly from the patent owners, and therefore FDK AAC Codec software may
-already be covered under those patent licenses when it is used for those
-licensed purposes only.
-
-Commercially-licensed AAC software libraries, including floating-point versions
-with enhanced sound quality, are also available from Fraunhofer. Users are
-encouraged to check the Fraunhofer website for additional applications
-information and documentation.
-
-2.    COPYRIGHT LICENSE
-
-Redistribution and use in source and binary forms, with or without modification,
-are permitted without payment of copyright license fees provided that you
-satisfy the following conditions:
-
-You must retain the complete text of this software license in redistributions of
-the FDK AAC Codec or your modifications thereto in source code form.
-
-You must retain the complete text of this software license in the documentation
-and/or other materials provided with redistributions of the FDK AAC Codec or
-your modifications thereto in binary form. You must make available free of
-charge copies of the complete source code of the FDK AAC Codec and your
-modifications thereto to recipients of copies in binary form.
-
-The name of Fraunhofer may not be used to endorse or promote products derived
-from this library without prior written permission.
-
-You may not charge copyright license fees for anyone to use, copy or distribute
-the FDK AAC Codec software or your modifications thereto.
-
-Your modified versions of the FDK AAC Codec must carry prominent notices stating
-that you changed the software and the date of any change. For modified versions
-of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android"
-must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK
-AAC Codec Library for Android."
-
-3.    NO PATENT LICENSE
-
-NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without
-limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE.
-Fraunhofer provides no warranty of patent non-infringement with respect to this
-software.
-
-You may use this FDK AAC Codec software or modifications thereto only for
-purposes that are authorized by appropriate patent licenses.
-
-4.    DISCLAIMER
-
-This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright
-holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES,
-including but not limited to the implied warranties of merchantability and
-fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
-CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary,
-or consequential damages, including but not limited to procurement of substitute
-goods or services; loss of use, data, or profits, or business interruption,
-however caused and on any theory of liability, whether in contract, strict
-liability, or tort (including negligence), arising in any way out of the use of
-this software, even if advised of the possibility of such damage.
-
-5.    CONTACT INFORMATION
-
-Fraunhofer Institute for Integrated Circuits IIS
-Attention: Audio and Multimedia Departments - FDK AAC LL
-Am Wolfsmantel 33
-91058 Erlangen, Germany
-
-www.iis.fraunhofer.de/amm
-amm-info@iis.fraunhofer.de
------------------------------------------------------------------------------ */
-
-/**************************** SBR decoder library ******************************
-
-   Author(s):   Arthur Tritthart
-
-   Description: (ARM optimised) LPP transposer subroutines
-
-*******************************************************************************/
-
-#if defined(__arm__)
-
-#define FUNCTION_LPPTRANSPOSER_func1
-
-#ifdef FUNCTION_LPPTRANSPOSER_func1
-
-/* Note: This code requires only 43 cycles per iteration instead of 61 on
- * ARM926EJ-S */
-static void lppTransposer_func1(FIXP_DBL *lowBandReal, FIXP_DBL *lowBandImag,
-                                FIXP_DBL **qmfBufferReal,
-                                FIXP_DBL **qmfBufferImag, int loops, int hiBand,
-                                int dynamicScale, int descale, FIXP_SGL a0r,
-                                FIXP_SGL a0i, FIXP_SGL a1r, FIXP_SGL a1i,
-                                const int fPreWhitening,
-                                FIXP_DBL preWhiteningGain,
-                                int preWhiteningGains_sf) {
-  FIXP_DBL real1, real2, imag1, imag2, accu1, accu2;
-
-  real2 = lowBandReal-2;
-  real1 = lowBandReal-1;
-  imag2 = lowBandImag-2;
-  imag1 = lowBandImag-1;
-  for (int i = 0; i < loops; i++) {
-    accu1 = fMultDiv2(a0r, real1);
-    accu2 = fMultDiv2(a0i, imag1);
-    accu1 = fMultAddDiv2(accu1, a1r, real2);
-    accu2 = fMultAddDiv2(accu2, a1i, imag2);
-    real2 = fMultDiv2(a1i, real2);
-    accu1 = accu1 - accu2;
-    accu1 = accu1 >> dynamicScale;
-
-    accu2 = fMultAddDiv2(real2, a1r, imag2);
-    real2 = real1;
-    imag2 = imag1;
-    accu2 = fMultAddDiv2(accu2, a0i, real1);
-    real1 = lowBandReali;
-    accu2 = fMultAddDiv2(accu2, a0r, imag1);
-    imag1 = lowBandImagi;
-    accu2 = accu2 >> dynamicScale;
-
-    accu1 <<= 1;
-    accu2 <<= 1;
-    accu1 += (real1 >> descale);
-    accu2 += (imag1 >> descale);
-    if (fPreWhitening) {
-      accu1 = scaleValueSaturate(fMultDiv2(accu1, preWhiteningGain),
-                                 preWhiteningGains_sf);
-      accu2 = scaleValueSaturate(fMultDiv2(accu2, preWhiteningGain),
-                                 preWhiteningGains_sf);
-    }
-    qmfBufferRealihiBand = accu1;
-    qmfBufferImagihiBand = accu2;
-  }
-}
-#endif /* #ifdef FUNCTION_LPPTRANSPOSER_func1 */
-
-#endif /* __arm__ */

fdk-aac-2.0.3.tar.gz/.github Added

fdk-aac-2.0.3.tar.gz/.github/workflows Added

fdk-aac-2.0.3.tar.gz/.github/workflows/ci.yml Added

@@ -0,0 +1,215 @@
+name: CI
+on: push, pull_request
+
+jobs:
+  linux-autotools:
+    runs-on: ubuntu-latest
+    steps:
+      - name: Checkout sources
+        uses: actions/checkout@v4
+      - name: Build
+        run: |
+          ./autogen.sh
+          ./configure --enable-example
+          make -j$(nproc) CFLAGS="-g -O2 -Werror" CXXFLAGS="-g -O2 -Werror"
+      - name: Run tests
+        run: |
+          sudo apt-get update && sudo apt-get install --no-install-recommends ffmpeg
+          ./test/run-test.sh
+
+  linux-cmake:
+    runs-on: ubuntu-latest
+    steps:
+      - name: Checkout sources
+        uses: actions/checkout@v4
+      - name: Build
+        run: |
+          cmake . -B build -DCMAKE_BUILD_TYPE=Release -DBUILD_PROGRAMS=YES -DCMAKE_C_FLAGS="-Werror" -DCMAKE_CXX_FLAGS="-Werror -Wno-stringop-overflow"
+          cmake --build build
+      - name: Run tests
+        run: |
+          sudo apt-get update && sudo apt-get install --no-install-recommends ffmpeg
+          cd build
+          ../test/run-test.sh
+
+  linux-sanitizers-gcc-autotools:
+    runs-on: ubuntu-latest
+    steps:
+      - name: Checkout sources
+        uses: actions/checkout@v4
+      - name: Build
+        run: |
+          ./autogen.sh
+          FLAGS="-fsanitize=address,undefined -fno-omit-frame-pointer -fno-sanitize=shift-base -DDEBUG -fno-sanitize-recover=all"
+          CC="gcc $FLAGS" CXX="g++ $FLAGS" ./configure --enable-example
+          make -j$(nproc)
+      - name: Run tests
+        run: |
+          sudo apt-get update && sudo apt-get install --no-install-recommends ffmpeg
+          ./test/run-test.sh
+
+  linux-sanitizers-clang-cmake:
+    runs-on: ubuntu-latest
+    steps:
+      - name: Checkout sources
+        uses: actions/checkout@v4
+      - name: Build
+        run: |
+          FLAGS="-fsanitize=address,undefined -fno-omit-frame-pointer -fno-sanitize=shift-base -DDEBUG -fno-sanitize-recover=all -fno-sanitize=vptr -fno-sanitize=function"
+          cmake . -B build -DCMAKE_BUILD_TYPE=Release -DBUILD_PROGRAMS=YES -DCMAKE_C_COMPILER=clang -DCMAKE_CXX_COMPILER=clang++ -DCMAKE_C_FLAGS="$FLAGS" -DCMAKE_CXX_FLAGS="$FLAGS"
+          cmake --build build
+      - name: Run tests
+        run: |
+          sudo apt-get update && sudo apt-get install --no-install-recommends ffmpeg
+          cd build
+          ../test/run-test.sh
+
+  macos-autotools:
+    runs-on: macos-latest
+    steps:
+      - name: Install autotools
+        run: |
+          brew install automake
+      - name: Checkout sources
+        uses: actions/checkout@v4
+      - name: Build
+        run: |
+          ./autogen.sh
+          ./configure --enable-example
+          make -j$(sysctl -n hw.ncpu) CFLAGS="-g -O2 -Werror" CXXFLAGS="-g -O2 -Werror -Wno-deprecated-declarations"
+
+  macos-cmake:
+    runs-on: macos-latest
+    steps:
+      - name: Checkout sources
+        uses: actions/checkout@v4
+      - name: Build
+        run: |
+          cmake . -B build -DCMAKE_BUILD_TYPE=Release -DBUILD_PROGRAMS=YES -DCMAKE_OSX_ARCHITECTURES="arm64;x86_64" -DCMAKE_C_FLAGS="-Werror" -DCMAKE_CXX_FLAGS="-Werror -Wno-deprecated-declarations"
+          cmake --build build
+
+  mingw-cross-autotools:
+    runs-on: ubuntu-latest
+    steps:
+      - name: Install cross compiler
+        run: |
+          sudo apt-get update && sudo apt-get install g++-mingw-w64-x86-64
+      - name: Checkout sources
+        uses: actions/checkout@v4
+      - name: Build
+        run: |
+          ./autogen.sh
+          ./configure --enable-example --host=x86_64-w64-mingw32
+          make -j$(nproc) CFLAGS="-g -O2 -Werror" CXXFLAGS="-g -O2 -Werror"
+
+  mingw-cross-cmake:
+    runs-on: ubuntu-latest
+    steps:
+      - name: Install cross compiler
+        run: |
+          sudo apt-get update && sudo apt-get install g++-mingw-w64-x86-64
+      - name: Checkout sources
+        uses: actions/checkout@v4
+      - name: Build
+        run: |
+          cmake . -B build -DCMAKE_BUILD_TYPE=Release -DBUILD_PROGRAMS=YES -DCMAKE_C_COMPILER=x86_64-w64-mingw32-gcc -DCMAKE_CXX_COMPILER=x86_64-w64-mingw32-g++ -DCMAKE_CROSSCOMPILING=TRUE -DCMAKE_SYSTEM_NAME=Windows -DCMAKE_C_FLAGS="-Werror" -DCMAKE_CXX_FLAGS="-Werror -Wno-stringop-overflow"
+          cmake --build build
+
+  msys-cmake:
+    runs-on: windows-latest
+    defaults:
+      run:
+        shell: msys2 {0}
+    steps:
+      - uses: msys2/setup-msys2@v2
+        with:
+          msystem: mingw64
+          update: true
+          install: >-
+            git
+            mingw-w64-x86_64-toolchain
+            mingw-w64-x86_64-cmake
+            mingw-w64-x86_64-ninja
+      - name: Checkout sources
+        uses: actions/checkout@v4
+      - name: Build
+        run: |
+          cmake . -G Ninja -B build -DCMAKE_BUILD_TYPE=Release -DBUILD_PROGRAMS=YES
+          cmake --build build
+
+  msvc-cmake:
+    runs-on: windows-latest
+    steps:
+      - name: Checkout sources
+        uses: actions/checkout@v4
+      - name: Build
+        run: |
+          cmake . -B build -DBUILD_PROGRAMS=YES
+          cmake --build build --config Release
+
+  msvc-cmake-ninja-arm:
+    runs-on: windows-latest
+    steps:
+      - name: Checkout sources
+        uses: actions/checkout@v4
+      - name: Set up the environment
+        uses: ilammy/msvc-dev-cmd@v1
+        with:
+          arch: amd64_arm
+      - name: Build
+        run: |
+          cmake . -G Ninja -B build -DCMAKE_BUILD_TYPE=Release -DBUILD_PROGRAMS=YES
+          cmake --build build
+
+  msvc-cmake-arm64:
+    runs-on: windows-latest
+    steps:
+      - name: Checkout sources
+        uses: actions/checkout@v4
+      - name: Build
+        run: |
+          cmake . -B build -DBUILD_PROGRAMS=YES -A ARM64
+          cmake --build build --config Release
+
+  llvm-mingw:
+    runs-on: ubuntu-latest
+    steps:
+      - name: Install llvm-mingw
+        run: |
+          wget https://github.com/mstorsjo/llvm-mingw/releases/download/20231003/llvm-mingw-20231003-ucrt-ubuntu-20.04-x86_64.tar.xz
+          tar -Jxvf llvm-mingw-*-ucrt-ubuntu-*-x86_64.tar.xz
+          rm llvm-mingw-*.tar.xz
+          sudo mv llvm-mingw-* /opt/llvm-mingw
+          echo /opt/llvm-mingw/bin >> $GITHUB_PATH
+      - name: Checkout sources
+        uses: actions/checkout@v4
+      - name: Build fdk-aac
+        run: |
+          ./autogen.sh
+          ./configure --host=aarch64-w64-mingw32 --enable-example
+          make -j$(nproc) CFLAGS="-g -O2 -Werror" CXXFLAGS="-g -O2 -Werror"
+
+  linux-ffmpeg:
+    runs-on: ubuntu-latest
+    steps:
+      - name: Checkout sources
+        uses: actions/checkout@v4
+      - name: Build fdk-aac
+        run: |
+          ./autogen.sh
+          mkdir build
+          cd build
+          ../configure --enable-example --prefix=$(pwd)/../prefix
+          make -j$(nproc)
+          make -j$(nproc) install
+      - name: Checkout FFmpeg
+        uses: actions/checkout@v4
+        with:
+          repository: ffmpeg/ffmpeg
+          path: ffmpeg
+      - name: Build FFmpeg
+        run: |
+          sudo apt-get update && sudo apt-get install -y nasm
+          cd ffmpeg
+          PKG_CONFIG_PATH=$(pwd)/../prefix/lib/pkgconfig ./configure --enable-libfdk-aac --enable-nonfree
+          make -j$(nproc)

fdk-aac-2.0.2.tar.gz/.gitignore -> fdk-aac-2.0.3.tar.gz/.gitignore Changed

fdk-aac-2.0.2.tar.gz/Android.bp -> fdk-aac-2.0.3.tar.gz/Android.bp Changed

fdk-aac-2.0.2.tar.gz/CMakeLists.txt -> fdk-aac-2.0.3.tar.gz/CMakeLists.txt Changed

@@ -43,7 +43,7 @@
   cmake_policy(SET CMP0091 NEW)
 endif()
 
-project(fdk-aac VERSION 2.0.2)
+project(fdk-aac VERSION 2.0.3)
 
 # Includes
 
@@ -62,7 +62,7 @@
 # Options
 
 option(BUILD_SHARED_LIBS "Build shared library" ON)
-option(BUILD_PROGRAMS "Build aac-enc utility" OFF)
+option(BUILD_PROGRAMS "Build extra utilities" OFF)
 option(FDK_AAC_INSTALL_CMAKE_CONFIG_MODULE "Install CMake package configuration file" ON)
 option(FDK_AAC_INSTALL_PKGCONFIG_MODULE "Install pkg-config .pc file" ON)
 
@@ -487,10 +487,11 @@
 ### Set public headers and shared library version. Version info is critical for Unix-like OSes.
 set_target_properties(fdk-aac PROPERTIES
   PUBLIC_HEADER "${fdk_aacinclude_HEADERS}"
-  VERSION 2.0.2
+  VERSION 2.0.3
   SOVERSION 2
   MACHO_COMPATIBILITY_VERSION 3.0.0
-  MACHO_CURRENT_VERSION 3.2.0)
+  MACHO_CURRENT_VERSION 3.3.0
+  LINKER_LANGUAGE C)
 
 ### Some compiler options from Makefile.am
 if(MSVC)
@@ -585,4 +586,15 @@
   ## Program target installation
   install(TARGETS aac-enc RUNTIME DESTINATION ${CMAKE_INSTALL_BINDIR})
 
+
+  ## Tool for testing the library
+  set(test_encode_decode_SOURCES
+    test-encode-decode.c
+    wavreader.c
+    wavreader.h
+    sha1.c
+    sha1.h)
+
+  add_executable(test-encode-decode ${test_encode_decode_SOURCES})
+  target_link_libraries(test-encode-decode PRIVATE fdk-aac)
 endif()

fdk-aac-2.0.2.tar.gz/ChangeLog -> fdk-aac-2.0.3.tar.gz/ChangeLog Changed

fdk-aac-2.0.2.tar.gz/METADATA -> fdk-aac-2.0.3.tar.gz/METADATA Changed

fdk-aac-2.0.2.tar.gz/Makefile.am -> fdk-aac-2.0.3.tar.gz/Makefile.am Changed

fdk-aac-2.0.2.tar.gz/aac-enc.c -> fdk-aac-2.0.3.tar.gz/aac-enc.c Changed

fdk-aac-2.0.2.tar.gz/configure.ac -> fdk-aac-2.0.3.tar.gz/configure.ac Changed

fdk-aac-2.0.2.tar.gz/documentation/aacDecoder.pdf -> fdk-aac-2.0.3.tar.gz/documentation/aacDecoder.pdf Changed

fdk-aac-2.0.2.tar.gz/documentation/aacEncoder.pdf -> fdk-aac-2.0.3.tar.gz/documentation/aacEncoder.pdf Changed

fdk-aac-2.0.2.tar.gz/fuzzer/Android.bp -> fdk-aac-2.0.3.tar.gz/fuzzer/Android.bp Changed

fdk-aac-2.0.2.tar.gz/fuzzer/aac_dec_fuzzer.cpp -> fdk-aac-2.0.3.tar.gz/fuzzer/aac_dec_fuzzer.cpp Changed

fdk-aac-2.0.2.tar.gz/libAACdec/include/aacdecoder_lib.h -> fdk-aac-2.0.3.tar.gz/libAACdec/include/aacdecoder_lib.h Changed

fdk-aac-2.0.2.tar.gz/libAACdec/src/aac_ram.cpp -> fdk-aac-2.0.3.tar.gz/libAACdec/src/aac_ram.cpp Changed

fdk-aac-2.0.2.tar.gz/libAACdec/src/aac_ram.h -> fdk-aac-2.0.3.tar.gz/libAACdec/src/aac_ram.h Changed

fdk-aac-2.0.2.tar.gz/libAACdec/src/aacdec_drc.cpp -> fdk-aac-2.0.3.tar.gz/libAACdec/src/aacdec_drc.cpp Changed

@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2023 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -150,6 +150,19 @@
 }
 
 /*!
+\brief  Disable DRC
+
+\self Handle of DRC info
+
+\return none
+*/
+void aacDecoder_drcDisable(HANDLE_AAC_DRC self) {
+  self->enable = 0;
+  self->applyExtGain = 0;
+  self->progRefLevelPresent = 0;
+}
+
+/*!
 \brief Reset DRC information
 
 \self Handle of DRC info
@@ -204,7 +217,6 @@
   self->progRefLevel = pParams->targetRefLevel;
   self->progRefLevelPresent = 0;
   self->presMode = -1;
-  self->uniDrcPrecedence = 0;
 
   aacDecoder_drcReset(self);
 }
@@ -340,12 +352,6 @@
       self->numOutChannels = (INT)value;
       self->update = 1;
       break;
-    case UNIDRC_PRECEDENCE:
-      if (self == NULL) {
-        return AAC_DEC_INVALID_HANDLE;
-      }
-      self->uniDrcPrecedence = (UCHAR)value;
-      break;
     default:
       return AAC_DEC_SET_PARAM_FAIL;
   } /* switch(param) */
@@ -1245,7 +1251,6 @@
   /* switch on/off processing */
   self->enable = ((p->boost > (FIXP_DBL)0) || (p->cut > (FIXP_DBL)0) ||
                   (p->applyHeavyCompression == ON) || (p->targetRefLevel >= 0));
-  self->enable = (self->enable && !self->uniDrcPrecedence);
 
   self->prevAacNumChannels = aacNumChannels;
   self->update = 0;

fdk-aac-2.0.2.tar.gz/libAACdec/src/aacdec_drc.h -> fdk-aac-2.0.3.tar.gz/libAACdec/src/aacdec_drc.h Changed

fdk-aac-2.0.2.tar.gz/libAACdec/src/aacdec_drc_types.h -> fdk-aac-2.0.3.tar.gz/libAACdec/src/aacdec_drc_types.h Changed

fdk-aac-2.0.2.tar.gz/libAACdec/src/aacdec_hcrs.cpp -> fdk-aac-2.0.3.tar.gz/libAACdec/src/aacdec_hcrs.cpp Changed

fdk-aac-2.0.2.tar.gz/libAACdec/src/aacdecoder.cpp -> fdk-aac-2.0.3.tar.gz/libAACdec/src/aacdecoder.cpp Changed

@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2023 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -494,6 +494,75 @@
   return error;
 }
 
+static INT findElementInstanceTag(
+    INT elementTag, MP4_ELEMENT_ID elementId,
+    CAacDecoderChannelInfo **pAacDecoderChannelInfo, INT nChannels,
+    MP4_ELEMENT_ID *pElementIdTab, INT nElements) {
+  int el, chCnt = 0;
+
+  for (el = 0; el < nElements; el++) {
+    switch (pElementIdTabel) {
+      case ID_CPE:
+      case ID_SCE:
+      case ID_LFE:
+        if ((elementTag == pAacDecoderChannelInfochCnt->ElementInstanceTag) &&
+            (elementId == pElementIdTabel)) {
+          return 1; /* element instance tag found */
+        }
+        chCnt += (pElementIdTabel == ID_CPE) ? 2 : 1;
+        break;
+      default:
+        break;
+    }
+    if (chCnt >= nChannels) break;
+    if (pElementIdTabel == ID_END) break;
+  }
+
+  return 0; /* element instance tag not found */
+}
+
+static INT validateElementInstanceTags(
+    CProgramConfig *pce, CAacDecoderChannelInfo **pAacDecoderChannelInfo,
+    INT nChannels, MP4_ELEMENT_ID *pElementIdTab, INT nElements) {
+  if (nChannels >= pce->NumChannels) {
+    for (int el = 0; el < pce->NumFrontChannelElements; el++) {
+      if (!findElementInstanceTag(pce->FrontElementTagSelectel,
+                                  pce->FrontElementIsCpeel ? ID_CPE : ID_SCE,
+                                  pAacDecoderChannelInfo, nChannels,
+                                  pElementIdTab, nElements)) {
+        return 0; /* element instance tag not in raw_data_block() */
+      }
+    }
+    for (int el = 0; el < pce->NumSideChannelElements; el++) {
+      if (!findElementInstanceTag(pce->SideElementTagSelectel,
+                                  pce->SideElementIsCpeel ? ID_CPE : ID_SCE,
+                                  pAacDecoderChannelInfo, nChannels,
+                                  pElementIdTab, nElements)) {
+        return 0; /* element instance tag not in raw_data_block() */
+      }
+    }
+    for (int el = 0; el < pce->NumBackChannelElements; el++) {
+      if (!findElementInstanceTag(pce->BackElementTagSelectel,
+                                  pce->BackElementIsCpeel ? ID_CPE : ID_SCE,
+                                  pAacDecoderChannelInfo, nChannels,
+                                  pElementIdTab, nElements)) {
+        return 0; /* element instance tag not in raw_data_block() */
+      }
+    }
+    for (int el = 0; el < pce->NumLfeChannelElements; el++) {
+      if (!findElementInstanceTag(pce->LfeElementTagSelectel, ID_LFE,
+                                  pAacDecoderChannelInfo, nChannels,
+                                  pElementIdTab, nElements)) {
+        return 0; /* element instance tag not in raw_data_block() */
+      }
+    }
+  } else {
+    return 0; /* too less decoded audio channels */
+  }
+
+  return 1; /* all element instance tags found in raw_data_block() */
+}
+
 /*!
   \brief Read Program Config Element
 
@@ -568,7 +637,7 @@
   \return  Error code
 */
 LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_PrepareCrossFade(
-    const INT_PCM *pTimeData, INT_PCM **pTimeDataFlush, const INT numChannels,
+    const PCM_DEC *pTimeData, PCM_DEC **pTimeDataFlush, const INT numChannels,
     const INT frameSize, const INT interleaved) {
   int i, ch, s1, s2;
   AAC_DECODER_ERROR ErrorStatus;
@@ -584,7 +653,7 @@
   }
 
   for (ch = 0; ch < numChannels; ch++) {
-    const INT_PCM *pIn = &pTimeDatach * s1;
+    const PCM_DEC *pIn = &pTimeDatach * s1;
     for (i = 0; i < TIME_DATA_FLUSH_SIZE; i++) {
       pTimeDataFlushchi = *pIn;
       pIn += s2;
@@ -606,7 +675,7 @@
   \return  Error code
 */
 LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_ApplyCrossFade(
-    INT_PCM *pTimeData, INT_PCM **pTimeDataFlush, const INT numChannels,
+    PCM_DEC *pTimeData, PCM_DEC **pTimeDataFlush, const INT numChannels,
     const INT frameSize, const INT interleaved) {
   int i, ch, s1, s2;
   AAC_DECODER_ERROR ErrorStatus;
@@ -622,15 +691,15 @@
   }
 
   for (ch = 0; ch < numChannels; ch++) {
-    INT_PCM *pIn = &pTimeDatach * s1;
+    PCM_DEC *pIn = &pTimeDatach * s1;
     for (i = 0; i < TIME_DATA_FLUSH_SIZE; i++) {
       FIXP_SGL alpha = (FIXP_SGL)i
                        << (FRACT_BITS - 1 - TIME_DATA_FLUSH_SIZE_SF);
-      FIXP_DBL time = FX_PCM2FX_DBL(*pIn);
-      FIXP_DBL timeFlush = FX_PCM2FX_DBL(pTimeDataFlushchi);
+      FIXP_DBL time = PCM_DEC2FIXP_DBL(*pIn);
+      FIXP_DBL timeFlush = PCM_DEC2FIXP_DBL(pTimeDataFlushchi);
 
-      *pIn = (INT_PCM)(FIXP_PCM)FX_DBL2FX_PCM(
-          timeFlush - fMult(timeFlush, alpha) + fMult(time, alpha));
+      *pIn = FIXP_DBL2PCM_DEC(timeFlush - fMult(timeFlush, alpha) +
+                              fMult(time, alpha));
       pIn += s2;
     }
   }
@@ -753,7 +822,12 @@
     /* We are interested in preroll AUs if an explicit or an implicit config
      * change is signalized in other words if the build up status is set. */
     if (self->buildUpStatus == AACDEC_USAC_BUILD_UP_ON) {
-      self->applyCrossfade |= FDKreadBit(hBs);
+      UCHAR applyCrossfade = FDKreadBit(hBs);
+      if (applyCrossfade) {
+        self->applyCrossfade |= AACDEC_CROSSFADE_BITMASK_PREROLL;
+      } else {
+        self->applyCrossfade &= ~AACDEC_CROSSFADE_BITMASK_PREROLL;
+      }
       FDKreadBit(hBs); /* reserved */
       /* Read num preroll AU's */
       *numPrerollAU = escapedValue(hBs, 2, 4, 0);
@@ -838,43 +912,6 @@
       }
       *count -= readBits;
     } break;
-    case EXT_UNI_DRC: {
-      DRC_DEC_ERROR drcErr = DRC_DEC_OK;
-      DRC_DEC_CODEC_MODE drcDecCodecMode = DRC_DEC_CODEC_MODE_UNDEFINED;
-      INT nBitsRemaining = FDKgetValidBits(hBs);
-      INT readBits;
-
-      switch (self->streamInfo.aot) {
-        case AOT_AAC_LC:
-        case AOT_SBR:
-        case AOT_PS:
-          drcDecCodecMode = DRC_DEC_MPEG_4_AAC;
-          break;
-        default:
-          error = AAC_DEC_PARSE_ERROR;
-          goto bail;
-      }
-
-      drcErr = FDK_drcDec_SetCodecMode(self->hUniDrcDecoder, drcDecCodecMode);
-      if (drcErr) {
-        error = AAC_DEC_PARSE_ERROR;
-        goto bail;
-      }
-
-      drcErr = FDK_drcDec_ReadUniDrc(self->hUniDrcDecoder, hBs);
-      if (drcErr) {
-        error = AAC_DEC_PARSE_ERROR;
-        goto bail;
-      }
-      readBits = (INT)nBitsRemaining - (INT)FDKgetValidBits(hBs);
-      if (readBits > *count) { /* Read too much. Something went wrong! */
-        error = AAC_DEC_PARSE_ERROR;
-      }
-      *count -= readBits;
-      /* Skip any trailing bits */
-      FDKpushFor(hBs, *count);
-      *count = 0;
-    } break;
     case EXT_LDSAC_DATA:
     case EXT_SAC_DATA:
       /* Read MPEG Surround Extension payload */
@@ -1397,6 +1434,27 @@
 }
 
 /*!
+ * \brief CAacDecoder_AcceptFlags Accept flags and element flags
+ *
+ * \param self          o   handle to AACDECODER structure
+ * \param asc           i   handle to ASC structure
+ * \param flags         i   flags
+ * \param elFlags       i   pointer to element flags
+ * \param streamIndex   i   stream index
+ * \param elementOffset i   element offset
+ *
+ * \return void
+ */
+static void CAacDecoder_AcceptFlags(HANDLE_AACDECODER self,
+                                    const CSAudioSpecificConfig *asc,
+                                    UINT flags, UINT *elFlags, int streamIndex,
+                                    int elementOffset) {
+  FDKmemcpy(self->elFlags, elFlags, sizeof(self->elFlags));
+
+  self->flagsstreamIndex = flags;
+}
+
+/*!
  * \brief CAacDecoder_CtrlCFGChange Set config change parameters.
  *
  * \param self           i   handle to AACDECODER structure
@@ -1493,6 +1551,15 @@
   const int streamIndex = 0;
   INT flushChannels = 0;
 
+  UINT flags;
+  /* elFlags(3*MAX_CHANNELS + (MAX_CHANNELS)/2 + 4 * (MAX_TRACKS) + 1
+     where MAX_CHANNELS is (8*2) and MAX_TRACKS is 1 */
+  UINT elFlags(3 * ((8) * 2) + (((8) * 2)) / 2 + 4 * (1) + 1);
+
+  UCHAR sbrEnabled = self->sbrEnabled;
+  UCHAR sbrEnabledPrev = self->sbrEnabledPrev;
+  UCHAR mpsEnableCurr = self->mpsEnableCurr;
+
   if (!self) return AAC_DEC_INVALID_HANDLE;
 
   UCHAR downscaleFactor = self->downscaleFactor;
@@ -1649,8 +1716,8 @@
   }
 
   /* Set syntax flags */
-  self->flagsstreamIndex = 0;
-  { FDKmemclear(self->elFlags, sizeof(self->elFlags)); }
+  flags = 0;
+  { FDKmemclear(elFlags, sizeof(elFlags)); }
 
   if ((asc->m_channelConfiguration > 0) || IS_USAC(asc->m_aot)) {
     if (IS_USAC(asc->m_aot)) {
@@ -1676,7 +1743,7 @@
             asc->m_sc.m_usacConfig.m_usacNumElements;
       }
 
-      self->mpsEnableCurr = 0;
+      mpsEnableCurr = 0;
       for (int _el = 0;
            _el < (int)self->pUsacConfigstreamIndex->m_usacNumElements;
            _el++) {
@@ -1696,35 +1763,34 @@
           self->usacStereoConfigIndexel =
               asc->m_sc.m_usacConfig.element_el.m_stereoConfigIndex;
           if (self->elementsel == ID_USAC_CPE) {
-            self->mpsEnableCurr |= self->usacStereoConfigIndexel ? 1 : 0;
+            mpsEnableCurr |= self->usacStereoConfigIndexel ? 1 : 0;
           }
         }
 
-        self->elFlagsel |=
-            (asc->m_sc.m_usacConfig.element_el.m_noiseFilling)
-                ? AC_EL_USAC_NOISE
-                : 0;
-        self->elFlagsel |=
+        elFlagsel |= (asc->m_sc.m_usacConfig.element_el.m_noiseFilling)
+                           ? AC_EL_USAC_NOISE
+                           : 0;
+        elFlagsel |=
             (asc->m_sc.m_usacConfig.element_el.m_stereoConfigIndex > 0)
                 ? AC_EL_USAC_MPS212
                 : 0;
-        self->elFlagsel |= (asc->m_sc.m_usacConfig.element_el.m_interTes)
-                                 ? AC_EL_USAC_ITES
-                                 : 0;
-        self->elFlagsel |=
+        elFlagsel |= (asc->m_sc.m_usacConfig.element_el.m_interTes)
+                           ? AC_EL_USAC_ITES
+                           : 0;
+        elFlagsel |=
             (asc->m_sc.m_usacConfig.element_el.m_pvc) ? AC_EL_USAC_PVC : 0;
-        self->elFlagsel |=
+        elFlagsel |=
             (asc->m_sc.m_usacConfig.element_el.usacElementType == ID_USAC_LFE)
                 ? AC_EL_USAC_LFE
                 : 0;
-        self->elFlagsel |=
+        elFlagsel |=
             (asc->m_sc.m_usacConfig.element_el.usacElementType == ID_USAC_LFE)
                 ? AC_EL_LFE
                 : 0;
         if ((asc->m_sc.m_usacConfig.element_el.usacElementType ==
              ID_USAC_CPE) &&
             ((self->usacStereoConfigIndexel == 0))) {
-          self->elFlagsel |= AC_EL_USAC_CP_POSSIBLE;
+          elFlagsel |= AC_EL_USAC_CP_POSSIBLE;
         }
       }
 
@@ -1791,9 +1857,17 @@
       downscaleFactorInBS =
           asc->m_samplingFrequency /
           asc->m_sc.m_eldSpecificConfig.m_downscaledSamplingFrequency;
-      if (downscaleFactorInBS == 1 || downscaleFactorInBS == 2 ||
-          downscaleFactorInBS == 3 || downscaleFactorInBS == 4) {
+      if ((downscaleFactorInBS == 1 || downscaleFactorInBS == 2 ||
+           (downscaleFactorInBS == 3 &&
+            asc->m_sc.m_eldSpecificConfig.m_frameLengthFlag) ||
+           downscaleFactorInBS == 4) &&
+          ((asc->m_samplingFrequency %
+            asc->m_sc.m_eldSpecificConfig.m_downscaledSamplingFrequency) ==
+           0)) {
         downscaleFactor = downscaleFactorInBS;
+      } else {
+        downscaleFactorInBS = 1;
+        downscaleFactor = 1;
       }
     }
   } else {
@@ -1825,7 +1899,7 @@
       self->useLdQmfTimeAlign =
           asc->m_sc.m_eldSpecificConfig.m_useLdQmfTimeAlign;
     }
-    if (self->sbrEnabled != asc->m_sbrPresentFlag) {
+    if (sbrEnabled != asc->m_sbrPresentFlag) {
       ascChanged = 1;
     }
   }
@@ -1838,16 +1912,16 @@
   if (configMode & AC_CM_ALLOC_MEM) {
     self->streamInfo.extSamplingRate = asc->m_extensionSamplingFrequency;
   }
-  self->flagsstreamIndex |= (asc->m_sbrPresentFlag) ? AC_SBR_PRESENT : 0;
-  self->flagsstreamIndex |= (asc->m_psPresentFlag) ? AC_PS_PRESENT : 0;
+  flags |= (asc->m_sbrPresentFlag) ? AC_SBR_PRESENT : 0;
+  flags |= (asc->m_psPresentFlag) ? AC_PS_PRESENT : 0;
   if (asc->m_sbrPresentFlag) {
-    self->sbrEnabled = 1;
-    self->sbrEnabledPrev = 1;
+    sbrEnabled = 1;
+    sbrEnabledPrev = 1;
   } else {
-    self->sbrEnabled = 0;
-    self->sbrEnabledPrev = 0;
+    sbrEnabled = 0;
+    sbrEnabledPrev = 0;
   }
-  if (self->sbrEnabled && asc->m_extensionSamplingFrequency) {
+  if (sbrEnabled && asc->m_extensionSamplingFrequency) {
     if (downscaleFactor != 1 && (downscaleFactor)&1) {
       return AAC_DEC_UNSUPPORTED_SAMPLINGRATE; /* SBR needs an even downscale
                                                   factor */
@@ -1865,51 +1939,47 @@
   }
 
   /* --------- vcb11 ------------ */
-  self->flagsstreamIndex |= (asc->m_vcb11Flag) ? AC_ER_VCB11 : 0;
+  flags |= (asc->m_vcb11Flag) ? AC_ER_VCB11 : 0;
 
   /* ---------- rvlc ------------ */
-  self->flagsstreamIndex |= (asc->m_rvlcFlag) ? AC_ER_RVLC : 0;
+  flags |= (asc->m_rvlcFlag) ? AC_ER_RVLC : 0;
 
   /* ----------- hcr ------------ */
-  self->flagsstreamIndex |= (asc->m_hcrFlag) ? AC_ER_HCR : 0;
+  flags |= (asc->m_hcrFlag) ? AC_ER_HCR : 0;
 
   if (asc->m_aot == AOT_ER_AAC_ELD) {
-    self->mpsEnableCurr = 0;
-    self->flagsstreamIndex |= AC_ELD;
-    self->flagsstreamIndex |=
-        (asc->m_sbrPresentFlag)
-            ? AC_SBR_PRESENT
-            : 0; /* Need to set the SBR flag for backward-compatibility
-       reasons. Even if SBR is not supported. */
-    self->flagsstreamIndex |=
-        (asc->m_sc.m_eldSpecificConfig.m_sbrCrcFlag) ? AC_SBRCRC : 0;
-    self->flagsstreamIndex |=
-        (asc->m_sc.m_eldSpecificConfig.m_useLdQmfTimeAlign) ? AC_MPS_PRESENT
-                                                            : 0;
+    mpsEnableCurr = 0;
+    flags |= AC_ELD;
+    flags |= (asc->m_sbrPresentFlag)
+                 ? AC_SBR_PRESENT
+                 : 0; /* Need to set the SBR flag for backward-compatibility
+                               reasons. Even if SBR is not supported. */
+    flags |= (asc->m_sc.m_eldSpecificConfig.m_sbrCrcFlag) ? AC_SBRCRC : 0;
+    flags |= (asc->m_sc.m_eldSpecificConfig.m_useLdQmfTimeAlign)
+                 ? AC_MPS_PRESENT
+                 : 0;
     if (self->mpsApplicable) {
-      self->mpsEnableCurr = asc->m_sc.m_eldSpecificConfig.m_useLdQmfTimeAlign;
+      mpsEnableCurr = asc->m_sc.m_eldSpecificConfig.m_useLdQmfTimeAlign;
     }
   }
-  self->flagsstreamIndex |= (asc->m_aot == AOT_ER_AAC_LD) ? AC_LD : 0;
-  self->flagsstreamIndex |= (asc->m_epConfig >= 0) ? AC_ER : 0;
+  flags |= (asc->m_aot == AOT_ER_AAC_LD) ? AC_LD : 0;
+  flags |= (asc->m_epConfig >= 0) ? AC_ER : 0;
 
   if (asc->m_aot == AOT_USAC) {
-    self->flagsstreamIndex |= AC_USAC;
-    self->flagsstreamIndex |=
-        (asc->m_sc.m_usacConfig.element0.m_stereoConfigIndex > 0)
-            ? AC_MPS_PRESENT
-            : 0;
+    flags |= AC_USAC;
+    flags |= (asc->m_sc.m_usacConfig.element0.m_stereoConfigIndex > 0)
+                 ? AC_MPS_PRESENT
+                 : 0;
   }
   if (asc->m_aot == AOT_DRM_AAC) {
-    self->flagsstreamIndex |= AC_DRM | AC_SBRCRC | AC_SCALABLE;
+    flags |= AC_DRM | AC_SBRCRC | AC_SCALABLE;
   }
   if (asc->m_aot == AOT_DRM_SURROUND) {
-    self->flagsstreamIndex |=
-        AC_DRM | AC_SBRCRC | AC_SCALABLE | AC_MPS_PRESENT;
+    flags |= AC_DRM | AC_SBRCRC | AC_SCALABLE | AC_MPS_PRESENT;
     FDK_ASSERT(!asc->m_psPresentFlag);
   }
   if ((asc->m_aot == AOT_AAC_SCAL) || (asc->m_aot == AOT_ER_AAC_SCAL)) {
-    self->flagsstreamIndex |= AC_SCALABLE;
+    flags |= AC_SCALABLE;
   }
 
   if ((asc->m_epConfig >= 0) && (asc->m_channelConfiguration <= 0)) {
@@ -1960,13 +2030,17 @@
     if (ascChanged != 0) {
       *configChanged = 1;
     }
+
+    CAacDecoder_AcceptFlags(self, asc, flags, elFlags, streamIndex,
+                            elementOffset);
+
     return err;
   }
 
   /* set AC_USAC_SCFGI3 globally if any usac element uses */
   switch (asc->m_aot) {
     case AOT_USAC:
-      if (self->sbrEnabled) {
+      if (sbrEnabled) {
         for (int _el = 0;
              _el < (int)self->pUsacConfigstreamIndex->m_usacNumElements;
              _el++) {
@@ -1988,7 +2062,7 @@
         }
 
         if (usacStereoConfigIndex == 3) {
-          self->flagsstreamIndex |= AC_USAC_SCFGI3;
+          flags |= AC_USAC_SCFGI3;
         }
       }
       break;
@@ -2003,7 +2077,7 @@
     */
     switch (asc->m_aot) {
       case AOT_USAC:
-        if (self->sbrEnabled) {
+        if (sbrEnabled) {
           const UCHAR map_sbrRatio_2_nAnaBands = {16, 24, 32};
 
           FDK_ASSERT(asc->m_sc.m_usacConfig.m_sbrRatioIndex > 0);
@@ -2031,11 +2105,11 @@
         }
         break;
       case AOT_ER_AAC_ELD:
-        if (self->mpsEnableCurr &&
+        if (mpsEnableCurr &&
             asc->m_sc.m_eldSpecificConfig.m_useLdQmfTimeAlign) {
-          SAC_INPUT_CONFIG sac_interface =
-              (self->sbrEnabled && self->hSbrDecoder) ? SAC_INTERFACE_QMF
-                                                      : SAC_INTERFACE_TIME;
+          SAC_INPUT_CONFIG sac_interface = (sbrEnabled && self->hSbrDecoder)
+                                               ? SAC_INTERFACE_QMF
+                                               : SAC_INTERFACE_TIME;
           mpegSurroundDecoder_ConfigureQmfDomain(
               (CMpegSurroundDecoder *)self->pMpegSurroundDecoder, sac_interface,
               (UINT)self->streamInfo.aacSampleRate, asc->m_aot);
@@ -2069,14 +2143,14 @@
       ch = aacChannelsOffset;
       int _numElements;
       _numElements = (((8)) + (8));
-      if (self->flagsstreamIndex & (AC_RSV603DA | AC_USAC)) {
+      if (flags & (AC_RSV603DA | AC_USAC)) {
         _numElements = (int)asc->m_sc.m_usacConfig.m_usacNumElements;
       }
       for (int _el = 0; _el < _numElements; _el++) {
         int el_channels = 0;
         int el = elementOffset + _el;
 
-        if (self->flagsstreamIndex &
+        if (flags &
             (AC_ER | AC_LD | AC_ELD | AC_RSV603DA | AC_USAC | AC_RSVD50)) {
           if (ch >= ascChannels) {
             break;
@@ -2176,15 +2250,14 @@
           if (self->pAacDecoderStaticChannelInfoch->pOverlapBuffer == NULL) {
             goto bail;
           }
-          if (self->flagsstreamIndex &
-              (AC_USAC | AC_RSVD50 | AC_RSV603DA /*|AC_BSAC*/)) {
+          if (flags & (AC_USAC | AC_RSVD50 | AC_RSV603DA /*|AC_BSAC*/)) {
             self->pAacDecoderStaticChannelInfoch->hArCo = CArco_Create();
             if (self->pAacDecoderStaticChannelInfoch->hArCo == NULL) {
               goto bail;
             }
           }
 
-          if (!(self->flagsstreamIndex & (AC_USAC | AC_RSV603DA))) {
+          if (!(flags & (AC_USAC | AC_RSV603DA))) {
             CPns_UpdateNoiseState(
                 &self->pAacDecoderChannelInfoch->data.aac.PnsData,
                 &self->pAacDecoderStaticChannelInfoch->pnsCurrentSeed,
@@ -2195,7 +2268,7 @@
         chIdx++;
       }
 
-      if (self->flagsstreamIndex & AC_USAC) {
+      if (flags & AC_USAC) {
         for (int _ch = 0; _ch < flushChannels; _ch++) {
           ch = aacChannelsOffset + _ch;
           if (self->pTimeDataFlushch == NULL) {
@@ -2207,7 +2280,7 @@
         }
       }
 
-      if (self->flagsstreamIndex & (AC_USAC | AC_RSV603DA)) {
+      if (flags & (AC_USAC | AC_RSV603DA)) {
         int complexStereoPredPossible = 0;
         ch = aacChannelsOffset;
         chIdx = aacChannelsOffsetIdx;
@@ -2223,7 +2296,7 @@
             elCh = 1;
           }
 
-          if (self->elFlagsel2 & AC_EL_USAC_CP_POSSIBLE) {
+          if (elFlagsel2 & AC_EL_USAC_CP_POSSIBLE) {
             complexStereoPredPossible = 1;
             if (self->cpeStaticDatael2 == NULL) {
               self->cpeStaticDatael2 = GetCpePersistentData();
@@ -2360,9 +2433,6 @@
     }
   }
 
-  /* Update externally visible copy of flags */
-  self->streamInfo.flags = self->flags0;
-
   if (*configChanged) {
     int drcDecSampleRate, drcDecFrameSize;
 
@@ -2383,8 +2453,7 @@
 
   if (*configChanged) {
     if (asc->m_aot == AOT_USAC) {
-      self->hDrcInfo->enable = 0;
-      self->hDrcInfo->progRefLevelPresent = 0;
+      aacDecoder_drcDisable(self->hDrcInfo);
     }
   }
 
@@ -2393,6 +2462,15 @@
     pcmLimiter_SetThreshold(self->hLimiter, FL2FXCONST_DBL(0.89125094f));
   }
 
+  CAacDecoder_AcceptFlags(self, asc, flags, elFlags, streamIndex,
+                          elementOffset);
+  self->sbrEnabled = sbrEnabled;
+  self->sbrEnabledPrev = sbrEnabledPrev;
+  self->mpsEnableCurr = mpsEnableCurr;
+
+  /* Update externally visible copy of flags */
+  self->streamInfo.flags = self->flags0;
+
   return err;
 
 bail:
@@ -2927,6 +3005,24 @@
 
   } /* while ( (type != ID_END) ... ) */
 
+  if (!(self->flagsstreamIndex &
+        (AC_USAC | AC_RSVD50 | AC_RSV603DA | AC_BSAC | AC_LD | AC_ELD | AC_ER |
+         AC_SCALABLE)) &&
+      (self->streamInfo.channelConfig == 0) && pce->isValid &&
+      (ErrorStatus == AAC_DEC_OK) && self->frameOK &&
+      !(flags & (AACDEC_CONCEAL | AACDEC_FLUSH))) {
+    /* Check whether all PCE listed element instance tags are present in
+     * raw_data_block() */
+    if (!validateElementInstanceTags(
+            &self->pce, self->pAacDecoderChannelInfo, aacChannels,
+            channel_elements,
+            fMin(channel_element_count, (int)(sizeof(channel_elements) /
+                                              sizeof(*channel_elements))))) {
+      ErrorStatus = AAC_DEC_DECODE_FRAME_ERROR;
+      self->frameOK = 0;
+    }
+  }
+
   if (!(flags & (AACDEC_CONCEAL | AACDEC_FLUSH))) {
     /* float decoder checks if bitsLeft is in range 0-7; only prerollAUs are
      * byteAligned with respect to the first bit */
@@ -3177,12 +3273,6 @@
       FDKmemcpy(drcChMap, self->chMapping, (8) * sizeof(UCHAR));
     }
 
-    /* deactivate legacy DRC in case uniDrc is active, i.e. uniDrc payload is
-     * present and one of DRC or Loudness Normalization is switched on */
-    aacDecoder_drcSetParam(
-        self->hDrcInfo, UNIDRC_PRECEDENCE,
-        FDK_drcDec_GetParam(self->hUniDrcDecoder, DRC_DEC_IS_ACTIVE));
-
     /* Extract DRC control data and map it to channels (without bitstream delay)
      */
     mapped = aacDecoder_drcProlog(
@@ -3194,11 +3284,12 @@
          * data in the bitstream. */
         self->flagsstreamIndex |= AC_DRC_PRESENT;
       } else {
-        self->hDrcInfo->enable = 0;
-        self->hDrcInfo->progRefLevelPresent = 0;
         ErrorStatus = AAC_DEC_UNSUPPORTED_FORMAT;
       }
     }
+    if (self->flagsstreamIndex & (AC_USAC | AC_RSV603DA)) {
+      aacDecoder_drcDisable(self->hDrcInfo);
+    }
 
     /* Create a reverse mapping table */
     UCHAR Reverse_chMapping((8) * 2);
@@ -3441,11 +3532,12 @@
          * data in the bitstream. */
         self->flagsstreamIndex |= AC_DRC_PRESENT;
       } else {
-        self->hDrcInfo->enable = 0;
-        self->hDrcInfo->progRefLevelPresent = 0;
         ErrorStatus = AAC_DEC_UNSUPPORTED_FORMAT;
       }
     }
+    if (self->flagsstreamIndex & (AC_USAC | AC_RSV603DA)) {
+      aacDecoder_drcDisable(self->hDrcInfo);
+    }
   }
 
   /* Add additional concealment delay */

fdk-aac-2.0.2.tar.gz/libAACdec/src/aacdecoder.h -> fdk-aac-2.0.3.tar.gz/libAACdec/src/aacdecoder.h Changed

@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -172,6 +172,12 @@
   AACDEC_RSV60_BUILD_UP_IDLE_IN_BAND = 5
 };
 
+#define AACDEC_CROSSFADE_BITMASK_OFF                                    \
+  ((UCHAR)0) /*!< No cross-fade between frames shall be applied at next \
+                config change. */
+#define AACDEC_CROSSFADE_BITMASK_PREROLL \
+  ((UCHAR)1 << 1) /*!< applyCrossfade is signaled in AudioPreRoll */
+
 typedef struct {
   /* Usac Extension Elements */
   USAC_EXT_ELEMENT_TYPE usacExtElementType(3);
@@ -325,7 +331,7 @@
   UINT loudnessInfoSetPosition3;
   SCHAR defaultTargetLoudness;
 
-  INT_PCM
+  PCM_DEC
   *pTimeDataFlush((8) * 2); /*!< Pointer to the flushed time data which
                                  will be used for the crossfade in case of
                                  an USAC DASH IPF config change */
@@ -341,8 +347,8 @@
                                                           start position in the
                                                           bitstream */
   INT accessUnit; /*!< Number of the actual processed preroll accessUnit */
-  UCHAR applyCrossfade; /*!< if set crossfade for seamless stream switching is
-                           applied */
+  UCHAR applyCrossfade; /*!< If any bit is set, cross-fade for seamless stream
+                           switching is applied */
 
   FDK_SignalDelay usacResidualDelay; /*!< Delay residual signal to compensate
                                         for eSBR delay of DMX signal in case of
@@ -439,12 +445,12 @@
 
 /* Prepare crossfade for USAC DASH IPF config change */
 LINKSPEC_H AAC_DECODER_ERROR CAacDecoder_PrepareCrossFade(
-    const INT_PCM *pTimeData, INT_PCM **pTimeDataFlush, const INT numChannels,
+    const PCM_DEC *pTimeData, PCM_DEC **pTimeDataFlush, const INT numChannels,
     const INT frameSize, const INT interleaved);
 
 /* Apply crossfade for USAC DASH IPF config change */
 LINKSPEC_H AAC_DECODER_ERROR CAacDecoder_ApplyCrossFade(
-    INT_PCM *pTimeData, INT_PCM **pTimeDataFlush, const INT numChannels,
+    PCM_DEC *pTimeData, PCM_DEC **pTimeDataFlush, const INT numChannels,
     const INT frameSize, const INT interleaved);
 
 /* Set flush and build up mode */

fdk-aac-2.0.2.tar.gz/libAACdec/src/aacdecoder_lib.cpp -> fdk-aac-2.0.3.tar.gz/libAACdec/src/aacdecoder_lib.cpp Changed

@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2023 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -385,21 +385,19 @@
   return errTp;
 }
 
-static INT aacDecoder_SscCallback(void *handle, HANDLE_FDK_BITSTREAM hBs,
-                                  const AUDIO_OBJECT_TYPE coreCodec,
-                                  const INT samplingRate, const INT frameSize,
-                                  const INT stereoConfigIndex,
-                                  const INT coreSbrFrameLengthIndex,
-                                  const INT configBytes, const UCHAR configMode,
-                                  UCHAR *configChanged) {
+static INT aacDecoder_SscCallback(
+    void *handle, HANDLE_FDK_BITSTREAM hBs, const AUDIO_OBJECT_TYPE coreCodec,
+    const INT samplingRate, const INT frameSize, const INT numChannels,
+    const INT stereoConfigIndex, const INT coreSbrFrameLengthIndex,
+    const INT configBytes, const UCHAR configMode, UCHAR *configChanged) {
   SACDEC_ERROR err;
   TRANSPORTDEC_ERROR errTp;
   HANDLE_AACDECODER hAacDecoder = (HANDLE_AACDECODER)handle;
 
   err = mpegSurroundDecoder_Config(
       (CMpegSurroundDecoder *)hAacDecoder->pMpegSurroundDecoder, hBs, coreCodec,
-      samplingRate, frameSize, stereoConfigIndex, coreSbrFrameLengthIndex,
-      configBytes, configMode, configChanged);
+      samplingRate, frameSize, numChannels, stereoConfigIndex,
+      coreSbrFrameLengthIndex, configBytes, configMode, configChanged);
 
   switch (err) {
     case MPS_UNSUPPORTED_CONFIG:
@@ -443,12 +441,23 @@
   TRANSPORTDEC_ERROR errTp;
   HANDLE_AACDECODER hAacDecoder = (HANDLE_AACDECODER)handle;
   DRC_DEC_CODEC_MODE drcDecCodecMode = DRC_DEC_CODEC_MODE_UNDEFINED;
+  UCHAR dummyBuffer4 = {0};
+  FDK_BITSTREAM dummyBs;
+  HANDLE_FDK_BITSTREAM hReadBs;
 
   if (subStreamIndex != 0) {
     return TRANSPORTDEC_OK;
   }
 
-  else if (aot == AOT_USAC) {
+  if (hBs == NULL) {
+    /* use dummy zero payload to clear memory */
+    hReadBs = &dummyBs;
+    FDKinitBitStream(hReadBs, dummyBuffer, 4, 24);
+  } else {
+    hReadBs = hBs;
+  }
+
+  if (aot == AOT_USAC) {
     drcDecCodecMode = DRC_DEC_MPEG_D_USAC;
   }
 
@@ -457,10 +466,10 @@
 
   if (payloadType == 0) /* uniDrcConfig */
   {
-    err = FDK_drcDec_ReadUniDrcConfig(hAacDecoder->hUniDrcDecoder, hBs);
+    err = FDK_drcDec_ReadUniDrcConfig(hAacDecoder->hUniDrcDecoder, hReadBs);
   } else /* loudnessInfoSet */
   {
-    err = FDK_drcDec_ReadLoudnessInfoSet(hAacDecoder->hUniDrcDecoder, hBs);
+    err = FDK_drcDec_ReadLoudnessInfoSet(hAacDecoder->hUniDrcDecoder, hReadBs);
     hAacDecoder->loudnessInfoSetPosition1 = payloadStart;
     hAacDecoder->loudnessInfoSetPosition2 = fullPayloadLength;
   }
@@ -822,6 +831,9 @@
 
     case AAC_DRC_ATTENUATION_FACTOR:
       /* DRC compression factor (where 0 is no and 127 is max compression) */
+      if ((value < 0) || (value > 127)) {
+        return AAC_DEC_SET_PARAM_FAIL;
+      }
       errorStatus = aacDecoder_drcSetParam(hDrcInfo, DRC_CUT_SCALE, value);
       uniDrcErr = FDK_drcDec_SetParam(self->hUniDrcDecoder, DRC_DEC_COMPRESS,
                                       value * (FL2FXCONST_DBL(0.5f / 127.0f)));
@@ -829,6 +841,9 @@
 
     case AAC_DRC_BOOST_FACTOR:
       /* DRC boost factor (where 0 is no and 127 is max boost) */
+      if ((value < 0) || (value > 127)) {
+        return AAC_DEC_SET_PARAM_FAIL;
+      }
       errorStatus = aacDecoder_drcSetParam(hDrcInfo, DRC_BOOST_SCALE, value);
       uniDrcErr = FDK_drcDec_SetParam(self->hUniDrcDecoder, DRC_DEC_BOOST,
                                       value * (FL2FXCONST_DBL(0.5f / 127.0f)));
@@ -1153,6 +1168,8 @@
   int applyCrossfade = 1;    /* flag indicates if flushing was possible */
   PCM_DEC *pTimeData2;
   PCM_AAC *pTimeData3;
+  INT pcmLimiterScale = 0;
+  INT interleaved = 0;
 
   if (self == NULL) {
     return AAC_DEC_INVALID_HANDLE;
@@ -1175,8 +1192,10 @@
       aacDecoder_FreeMemCallback(self, &asc);
       self->streamInfo.numChannels = 0;
       /* 3) restore AudioSpecificConfig */
-      transportDec_OutOfBandConfig(self->hInput, asc.config,
-                                   (asc.configBits + 7) >> 3, 0);
+      if (asc.configBits <= (TP_USAC_MAX_CONFIG_LEN << 3)) {
+        transportDec_OutOfBandConfig(self->hInput, asc.config,
+                                     (asc.configBits + 7) >> 3, 0);
+      }
     }
   }
 
@@ -1609,6 +1628,11 @@
         /* set params */
         sbrDecoder_SetParam(self->hSbrDecoder, SBR_SYSTEM_BITSTREAM_DELAY,
                             self->sbrParams.bsDelay);
+        sbrDecoder_SetParam(
+            self->hSbrDecoder, SBR_FLUSH_DATA,
+            (flags & AACDEC_FLUSH) |
+                ((self->flushStatus && !(flags & AACDEC_CONCEAL)) ? AACDEC_FLUSH
+                                                                  : 0));
 
         sbrDecoder_SetParam(self->hSbrDecoder, SBR_SKIP_QMF, 1);
 
@@ -1659,7 +1683,7 @@
 
       {
         if ((FDK_drcDec_GetParam(self->hUniDrcDecoder, DRC_DEC_IS_ACTIVE)) &&
-            !(self->flags0 & AC_RSV603DA)) {
+            (self->flags0 & AC_USAC)) {
           /* Apply DRC gains*/
           int ch, drcDelay = 0;
           int needsDeinterleaving = 0;
@@ -1667,8 +1691,6 @@
           FIXP_DBL channelGain(8);
           int reverseInChannelMap(8);
           int reverseOutChannelMap(8);
-          int numDrcOutChannels = FDK_drcDec_GetParam(
-              self->hUniDrcDecoder, DRC_DEC_TARGET_CHANNEL_COUNT_SELECTED);
           FDKmemclear(channelGain, sizeof(channelGain));
           for (ch = 0; ch < (8); ch++) {
             reverseInChannelMapch = ch;
@@ -1691,17 +1713,6 @@
           drcDelay += CConcealment_GetDelay(&self->concealCommonData) *
                       self->streamInfo.frameSize;
 
-          for (ch = 0; ch < self->streamInfo.numChannels; ch++) {
-            UCHAR mapValue = FDK_chMapDescr_getMapValue(
-                &self->mapDescr, (UCHAR)ch, self->chMapIndex);
-            if (mapValue < (8)) reverseInChannelMapmapValue = ch;
-          }
-          for (ch = 0; ch < (int)numDrcOutChannels; ch++) {
-            UCHAR mapValue = FDK_chMapDescr_getMapValue(
-                &self->mapDescr, (UCHAR)ch, numDrcOutChannels);
-            if (mapValue < (8)) reverseOutChannelMapmapValue = ch;
-          }
-
           /* The output of SBR and MPS is interleaved. Deinterleaving may be
            * necessary for FDK_drcDec_ProcessTime, which accepts deinterleaved
            * audio only. */
@@ -1736,11 +1747,9 @@
           FDK_drcDec_Preprocess(self->hUniDrcDecoder);
 
           /* apply DRC1 gain sequence */
-          for (ch = 0; ch < self->streamInfo.numChannels; ch++) {
-            FDK_drcDec_ProcessTime(self->hUniDrcDecoder, drcDelay, DRC_DEC_DRC1,
-                                   ch, reverseInChannelMapch - ch, 1,
-                                   drcWorkBuffer, self->streamInfo.frameSize);
-          }
+          FDK_drcDec_ProcessTime(self->hUniDrcDecoder, drcDelay, DRC_DEC_DRC1,
+                                 0, 0, self->streamInfo.numChannels,
+                                 drcWorkBuffer, self->streamInfo.frameSize);
           /* apply downmix */
           FDK_drcDec_ApplyDownmix(
               self->hUniDrcDecoder, reverseInChannelMap, reverseOutChannelMap,
@@ -1748,12 +1757,10 @@
               &self->streamInfo.numChannels); /* self->streamInfo.numChannels
                                                  may change here */
           /* apply DRC2/3 gain sequence */
-          for (ch = 0; ch < self->streamInfo.numChannels; ch++) {
-            FDK_drcDec_ProcessTime(self->hUniDrcDecoder, drcDelay,
-                                   DRC_DEC_DRC2_DRC3, ch,
-                                   reverseOutChannelMapch - ch, 1,
-                                   drcWorkBuffer, self->streamInfo.frameSize);
-          }
+          FDK_drcDec_ProcessTime(self->hUniDrcDecoder, drcDelay,
+                                 DRC_DEC_DRC2_DRC3, 0, 0,
+                                 self->streamInfo.numChannels, drcWorkBuffer,
+                                 self->streamInfo.frameSize);
 
           if (needsDeinterleaving) {
             FDK_interleave(
@@ -1796,8 +1803,7 @@
       }
 
       if (self->streamInfo.extAot != AOT_AAC_SLS) {
-        INT pcmLimiterScale = 0;
-        INT interleaved = 0;
+        interleaved = 0;
         interleaved |= (self->sbrEnabled) ? 1 : 0;
         interleaved |= (self->mpsEnableCurr) ? 1 : 0;
         PCMDMX_ERROR dmxErr = PCMDMX_OK;
@@ -1828,145 +1834,38 @@
            * predictable behavior and thus maybe produce strange output. */
           ErrorStatus = AAC_DEC_DECODE_FRAME_ERROR;
         }
-
-        pcmLimiterScale += PCM_OUT_HEADROOM;
-
-        if (flags & AACDEC_CLRHIST) {
-          if (!(self->flags0 & AC_USAC)) {
-            /* Reset DRC data */
-            aacDecoder_drcReset(self->hDrcInfo);
-            /* Delete the delayed signal. */
-            pcmLimiter_Reset(self->hLimiter);
-          }
-        }
-
-        /* Set applyExtGain if DRC processing is enabled and if
-           progRefLevelPresent is present for the first time. Consequences: The
-           headroom of the output signal can be set to AACDEC_DRC_GAIN_SCALING
-           only for audio formats which support legacy DRC Level Normalization.
-                         For all other audio formats the headroom of the output
-           signal is set to PCM_OUT_HEADROOM. */
-        if (self->hDrcInfo->enable &&
-            (self->hDrcInfo->progRefLevelPresent == 1)) {
-          self->hDrcInfo->applyExtGain |= 1;
-        }
-
-        /* Check whether time data buffer is large enough. */
-        if (timeDataSize <
-            (self->streamInfo.numChannels * self->streamInfo.frameSize)) {
-          ErrorStatus = AAC_DEC_OUTPUT_BUFFER_TOO_SMALL;
-          goto bail;
-        }
-
-        if (self->limiterEnableCurr) {
-          /* use workBufferCore2 buffer for interleaving */
-          PCM_LIM *pInterleaveBuffer;
-          int blockLength = self->streamInfo.frameSize;
-
-          /* Set actual signal parameters */
-          pcmLimiter_SetNChannels(self->hLimiter, self->streamInfo.numChannels);
-          pcmLimiter_SetSampleRate(self->hLimiter, self->streamInfo.sampleRate);
-
-          if ((self->streamInfo.numChannels == 1) || (self->sbrEnabled) ||
-              (self->mpsEnableCurr)) {
-            pInterleaveBuffer = (PCM_LIM *)pTimeData2;
-          } else {
-            pInterleaveBuffer = (PCM_LIM *)self->workBufferCore2;
-
-            /* applyLimiter requests for interleaved data */
-            /* Interleave ouput buffer */
-            FDK_interleave(pTimeData2, pInterleaveBuffer,
-                           self->streamInfo.numChannels, blockLength,
-                           self->streamInfo.frameSize);
-          }
-
-          FIXP_DBL *pGainPerSample = NULL;
-
-          if (self->hDrcInfo->enable && self->hDrcInfo->applyExtGain) {
-            pGainPerSample = self->workBufferCore1;
-
-            if ((INT)GetRequiredMemWorkBufferCore1() <
-                (INT)(self->streamInfo.frameSize * sizeof(FIXP_DBL))) {
-              ErrorStatus = AAC_DEC_UNKNOWN;
-              goto bail;
-            }
-
-            pcmLimiterScale = applyDrcLevelNormalization(
-                self->hDrcInfo, (PCM_DEC *)pInterleaveBuffer, self->extGain,
-                pGainPerSample, pcmLimiterScale, self->extGainDelay,
-                self->streamInfo.frameSize, self->streamInfo.numChannels, 1, 1);
-          }
-
-          pcmLimiter_Apply(self->hLimiter, pInterleaveBuffer, pTimeData,
-                           pGainPerSample, pcmLimiterScale,
-                           self->streamInfo.frameSize);
-
-          {
-            /* Announce the additional limiter output delay */
-            self->streamInfo.outputDelay += pcmLimiter_GetDelay(self->hLimiter);
-          }
-        } else {
-          if (self->hDrcInfo->enable && self->hDrcInfo->applyExtGain) {
-            pcmLimiterScale = applyDrcLevelNormalization(
-                self->hDrcInfo, pTimeData2, self->extGain, NULL,
-                pcmLimiterScale, self->extGainDelay, self->streamInfo.frameSize,
-                self->streamInfo.numChannels,
-                (interleaved || (self->streamInfo.numChannels == 1))
-                    ? 1
-                    : self->streamInfo.frameSize,
-                0);
-          }
-
-          /* If numChannels = 1 we do not need interleaving. The same applies if
-          SBR or MPS are used, since their output is interleaved already
-          (resampled or not) */
-          if ((self->streamInfo.numChannels == 1) || (self->sbrEnabled) ||
-              (self->mpsEnableCurr)) {
-            scaleValuesSaturate(
-                pTimeData, pTimeData2,
-                self->streamInfo.frameSize * self->streamInfo.numChannels,
-                pcmLimiterScale);
-
-          } else {
-            scaleValuesSaturate(
-                (INT_PCM *)self->workBufferCore2, pTimeData2,
-                self->streamInfo.frameSize * self->streamInfo.numChannels,
-                pcmLimiterScale);
-            /* Interleave ouput buffer */
-            FDK_interleave((INT_PCM *)self->workBufferCore2, pTimeData,
-                           self->streamInfo.numChannels,
-                           self->streamInfo.frameSize,
-                           self->streamInfo.frameSize);
-          }
-        }
-      } /* if (self->streamInfo.extAot != AOT_AAC_SLS)*/
+      }
 
       if (self->flags0 & AC_USAC) {
         if (self->flushStatus == AACDEC_USAC_DASH_IPF_FLUSH_ON &&
             !(flags & AACDEC_CONCEAL)) {
-          CAacDecoder_PrepareCrossFade(pTimeData, self->pTimeDataFlush,
+          CAacDecoder_PrepareCrossFade(pTimeData2, self->pTimeDataFlush,
                                        self->streamInfo.numChannels,
-                                       self->streamInfo.frameSize, 1);
+                                       self->streamInfo.frameSize, interleaved);
         }
 
         /* prepare crossfade buffer for fade in */
-        if (!applyCrossfade && self->applyCrossfade &&
+        if (!applyCrossfade &&
+            (self->applyCrossfade != AACDEC_CROSSFADE_BITMASK_OFF) &&
             !(flags & AACDEC_CONCEAL)) {
           for (int ch = 0; ch < self->streamInfo.numChannels; ch++) {
             for (int i = 0; i < TIME_DATA_FLUSH_SIZE; i++) {
-              self->pTimeDataFlushchi = 0;
+              self->pTimeDataFlushchi = (PCM_DEC)0;
             }
           }
           applyCrossfade = 1;
         }
 
-        if (applyCrossfade && self->applyCrossfade &&
+        if (applyCrossfade &&
+            (self->applyCrossfade != AACDEC_CROSSFADE_BITMASK_OFF) &&
             !(accessUnit < numPrerollAU) &&
             (self->buildUpStatus == AACDEC_USAC_BUILD_UP_ON)) {
-          CAacDecoder_ApplyCrossFade(pTimeData, self->pTimeDataFlush,
+          CAacDecoder_ApplyCrossFade(pTimeData2, self->pTimeDataFlush,
                                      self->streamInfo.numChannels,
-                                     self->streamInfo.frameSize, 1);
-          self->applyCrossfade = 0;
+                                     self->streamInfo.frameSize, interleaved);
+          self->applyCrossfade =
+              AACDEC_CROSSFADE_BITMASK_OFF; /* disable cross-fade between frames
+                                               at nect config change */
         }
       }
 
@@ -2008,6 +1907,116 @@
            ((self->flushStatus == AACDEC_USAC_DASH_IPF_FLUSH_ON) &&
             !(flags & AACDEC_CONCEAL)));
 
+  if (self->streamInfo.extAot != AOT_AAC_SLS) {
+    pcmLimiterScale += PCM_OUT_HEADROOM;
+
+    if (flags & AACDEC_CLRHIST) {
+      if (!(self->flags0 & AC_USAC)) {
+        /* Reset DRC data */
+        aacDecoder_drcReset(self->hDrcInfo);
+        /* Delete the delayed signal. */
+        pcmLimiter_Reset(self->hLimiter);
+      }
+    }
+
+    /* Set applyExtGain if DRC processing is enabled and if progRefLevelPresent
+       is present for the first time. Consequences: The headroom of the output
+       signal can be set to AACDEC_DRC_GAIN_SCALING only for audio formats which
+       support legacy DRC Level Normalization. For all other audio formats the
+       headroom of the output signal is set to PCM_OUT_HEADROOM. */
+    if (self->hDrcInfo->enable && (self->hDrcInfo->progRefLevelPresent == 1)) {
+      self->hDrcInfo->applyExtGain |= 1;
+    }
+
+    /* Check whether time data buffer is large enough. */
+    if (timeDataSize <
+        (self->streamInfo.numChannels * self->streamInfo.frameSize)) {
+      ErrorStatus = AAC_DEC_OUTPUT_BUFFER_TOO_SMALL;
+      goto bail;
+    }
+
+    if (self->limiterEnableCurr) {
+      /* use workBufferCore2 buffer for interleaving */
+      PCM_LIM *pInterleaveBuffer;
+      int blockLength = self->streamInfo.frameSize;
+
+      /* Set actual signal parameters */
+      pcmLimiter_SetNChannels(self->hLimiter, self->streamInfo.numChannels);
+      pcmLimiter_SetSampleRate(self->hLimiter, self->streamInfo.sampleRate);
+
+      if ((self->streamInfo.numChannels == 1) || (self->sbrEnabled) ||
+          (self->mpsEnableCurr)) {
+        pInterleaveBuffer = (PCM_LIM *)pTimeData2;
+      } else {
+        pInterleaveBuffer = (PCM_LIM *)self->workBufferCore2;
+
+        /* applyLimiter requests for interleaved data */
+        /* Interleave ouput buffer */
+        FDK_interleave(pTimeData2, pInterleaveBuffer,
+                       self->streamInfo.numChannels, blockLength,
+                       self->streamInfo.frameSize);
+      }
+
+      FIXP_DBL *pGainPerSample = NULL;
+
+      if (self->hDrcInfo->enable && self->hDrcInfo->applyExtGain) {
+        pGainPerSample = self->workBufferCore1;
+
+        if ((INT)GetRequiredMemWorkBufferCore1() <
+            (INT)(self->streamInfo.frameSize * sizeof(FIXP_DBL))) {
+          ErrorStatus = AAC_DEC_UNKNOWN;
+          goto bail;
+        }
+
+        pcmLimiterScale = applyDrcLevelNormalization(
+            self->hDrcInfo, (PCM_DEC *)pInterleaveBuffer, self->extGain,
+            pGainPerSample, pcmLimiterScale, self->extGainDelay,
+            self->streamInfo.frameSize, self->streamInfo.numChannels, 1, 1);
+      }
+
+      pcmLimiter_Apply(self->hLimiter, pInterleaveBuffer, pTimeData,
+                       pGainPerSample, pcmLimiterScale,
+                       self->streamInfo.frameSize);
+
+      {
+        /* Announce the additional limiter output delay */
+        self->streamInfo.outputDelay += pcmLimiter_GetDelay(self->hLimiter);
+      }
+    } else {
+      if (self->hDrcInfo->enable && self->hDrcInfo->applyExtGain) {
+        pcmLimiterScale = applyDrcLevelNormalization(
+            self->hDrcInfo, pTimeData2, self->extGain, NULL, pcmLimiterScale,
+            self->extGainDelay, self->streamInfo.frameSize,
+            self->streamInfo.numChannels,
+            (interleaved || (self->streamInfo.numChannels == 1))
+                ? 1
+                : self->streamInfo.frameSize,
+            0);
+      }
+
+      /* If numChannels = 1 we do not need interleaving. The same applies if SBR
+      or MPS are used, since their output is interleaved already (resampled or
+      not) */
+      if ((self->streamInfo.numChannels == 1) || (self->sbrEnabled) ||
+          (self->mpsEnableCurr)) {
+        scaleValuesSaturate(
+            pTimeData, pTimeData2,
+            self->streamInfo.frameSize * self->streamInfo.numChannels,
+            pcmLimiterScale);
+
+      } else {
+        scaleValuesSaturate(
+            (INT_PCM *)self->workBufferCore2, pTimeData2,
+            self->streamInfo.frameSize * self->streamInfo.numChannels,
+            pcmLimiterScale);
+        /* Interleave ouput buffer */
+        FDK_interleave((INT_PCM *)self->workBufferCore2, pTimeData,
+                       self->streamInfo.numChannels, self->streamInfo.frameSize,
+                       self->streamInfo.frameSize);
+      }
+    }
+  } /* if (self->streamInfo.extAot != AOT_AAC_SLS)*/
+
 bail:
 
   /* error in renderer part occurred, ErrorStatus was set to invalid output */

fdk-aac-2.0.2.tar.gz/libAACdec/src/channel.cpp -> fdk-aac-2.0.3.tar.gz/libAACdec/src/channel.cpp Changed

fdk-aac-2.0.2.tar.gz/libAACdec/src/rvlc.cpp -> fdk-aac-2.0.3.tar.gz/libAACdec/src/rvlc.cpp Changed

@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -628,7 +628,7 @@
 
   SHORT *pScfBwd = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfBwd;
   SHORT *pScfEsc = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfEsc;
-  UCHAR *pEscEscCnt = &(pRvlc->numDecodedEscapeWordsEsc);
+  UCHAR escEscCnt = pRvlc->numDecodedEscapeWordsEsc;
   UCHAR *pEscBwdCnt = &(pRvlc->numDecodedEscapeWordsBwd);
 
   pRvlc->pRvlBitCnt_RVL = &(pRvlc->length_of_rvlc_sf_bwd);
@@ -636,7 +636,7 @@
 
   *pEscBwdCnt = 0;
   pRvlc->direction = BWD;
-  pScfEsc += *pEscEscCnt - 1; /* set pScfEsc to last entry */
+  pScfEsc += escEscCnt - 1; /* set pScfEsc to last entry */
   pRvlc->firstScf = 0;
   pRvlc->firstNrg = 0;
   pRvlc->firstIs = 0;
@@ -651,7 +651,7 @@
     }
     dpcm -= TABLE_OFFSET;
     if ((dpcm == MIN_RVL) || (dpcm == MAX_RVL)) {
-      if (pRvlc->length_of_rvlc_escapes) {
+      if ((pRvlc->length_of_rvlc_escapes) || (*pEscBwdCnt >= escEscCnt)) {
         pRvlc->conceal_min = bnds;
         return;
       } else {
@@ -694,7 +694,7 @@
           }
           dpcm -= TABLE_OFFSET;
           if ((dpcm == MIN_RVL) || (dpcm == MAX_RVL)) {
-            if (pRvlc->length_of_rvlc_escapes) {
+            if ((pRvlc->length_of_rvlc_escapes) || (*pEscBwdCnt >= escEscCnt)) {
               pScfBwdbnds = position;
               pRvlc->conceal_min = fMax(0, bnds - offset);
               return;
@@ -731,7 +731,8 @@
             }
             dpcm -= TABLE_OFFSET;
             if ((dpcm == MIN_RVL) || (dpcm == MAX_RVL)) {
-              if (pRvlc->length_of_rvlc_escapes) {
+              if ((pRvlc->length_of_rvlc_escapes) ||
+                  (*pEscBwdCnt >= escEscCnt)) {
                 pScfBwdbnds = noisenrg;
                 pRvlc->conceal_min = fMax(0, bnds - offset);
                 return;
@@ -762,7 +763,7 @@
           }
           dpcm -= TABLE_OFFSET;
           if ((dpcm == MIN_RVL) || (dpcm == MAX_RVL)) {
-            if (pRvlc->length_of_rvlc_escapes) {
+            if ((pRvlc->length_of_rvlc_escapes) || (*pEscBwdCnt >= escEscCnt)) {
               pScfBwdbnds = factor;
               pRvlc->conceal_min = fMax(0, bnds - offset);
               return;

fdk-aac-2.0.2.tar.gz/libAACdec/src/usacdec_acelp.cpp -> fdk-aac-2.0.3.tar.gz/libAACdec/src/usacdec_acelp.cpp Changed

fdk-aac-2.0.2.tar.gz/libAACenc/include/aacenc_lib.h -> fdk-aac-2.0.3.tar.gz/libAACenc/include/aacenc_lib.h Changed

@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -1090,7 +1090,7 @@
                       write edit lists for gapless playback. The decoder may not
                       know how much delay is introdcued by SBR, since it may not
                       know if SBR is active at all (implicit signaling),
-                      therefore the deocder must take into account any delay
+                      therefore the decoder must take into account any delay
                       caused by the SBR module. */
 
   UCHAR confBuf64; /*!< Configuration buffer in binary format as an
@@ -1647,7 +1647,7 @@
  *
  * \return
  *          - AACENC_OK, on succes.
- *          - AACENC_INIT_ERROR, on failure.
+ *          - AACENC_INVALID_HANDLE, AACENC_INIT_ERROR, on failure.
  */
 AACENC_ERROR aacEncInfo(const HANDLE_AACENCODER hAacEncoder,
                         AACENC_InfoStruct *pInfo);

fdk-aac-2.0.2.tar.gz/libAACenc/src/aacenc_lib.cpp -> fdk-aac-2.0.3.tar.gz/libAACenc/src/aacenc_lib.cpp Changed

@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -1242,7 +1242,7 @@
 INT aacenc_SscCallback(void *self, HANDLE_FDK_BITSTREAM hBs,
                        const AUDIO_OBJECT_TYPE coreCodec,
                        const INT samplingRate, const INT frameSize,
-                       const INT stereoConfigIndex,
+                       const INT numChannels, const INT stereoConfigIndex,
                        const INT coreSbrFrameLengthIndex, const INT configBytes,
                        const UCHAR configMode, UCHAR *configChanged) {
   HANDLE_AACENCODER hAacEncoder = (HANDLE_AACENCODER)self;
@@ -1784,8 +1784,8 @@
                                                    hAacEncoder->nSamplesRead));
     INT_PCM *pIn =
         hAacEncoder->inputBuffer +
-        (hAacEncoder->inputBufferOffset + hAacEncoder->nSamplesRead) /
-            hAacEncoder->aacConfig.nChannels;
+        hAacEncoder->inputBufferOffset / hAacEncoder->aacConfig.nChannels +
+        hAacEncoder->nSamplesRead / hAacEncoder->extParam.nChannels;
     newSamples -=
         (newSamples %
          hAacEncoder->extParam
@@ -1827,12 +1827,13 @@
 
         /* clear out until end-of-buffer */
         if (nZeros) {
+          INT_PCM *pIn =
+              hAacEncoder->inputBuffer +
+              hAacEncoder->inputBufferOffset /
+                  hAacEncoder->aacConfig.nChannels +
+              hAacEncoder->nSamplesRead / hAacEncoder->extParam.nChannels;
           for (i = 0; i < (int)hAacEncoder->extParam.nChannels; i++) {
-            FDKmemclear(hAacEncoder->inputBuffer +
-                            i * hAacEncoder->inputBufferSizePerChannel +
-                            (hAacEncoder->inputBufferOffset +
-                             hAacEncoder->nSamplesRead) /
-                                hAacEncoder->extParam.nChannels,
+            FDKmemclear(pIn + i * hAacEncoder->inputBufferSizePerChannel,
                         sizeof(INT_PCM) * nZeros);
           }
           hAacEncoder->nZerosAppended += nZeros;
@@ -2520,6 +2521,11 @@
                         AACENC_InfoStruct *pInfo) {
   AACENC_ERROR err = AACENC_OK;
 
+  if ((hAacEncoder == NULL) || (pInfo == NULL)) {
+    err = AACENC_INVALID_HANDLE;
+    goto bail;
+  }
+
   FDKmemclear(pInfo, sizeof(AACENC_InfoStruct));
   pInfo->confSize = 64; /* pre-initialize */

fdk-aac-2.0.2.tar.gz/libDRCdec/src/drcDec_reader.cpp -> fdk-aac-2.0.3.tar.gz/libDRCdec/src/drcDec_reader.cpp Changed

@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -512,10 +512,13 @@
               fMin(tmpNNodes, (UCHAR)16) * sizeof(GAIN_NODE));
   }
 
-  hUniDrcGain->uniDrcGainExtPresent = FDKreadBits(hBs, 1);
-  if (hUniDrcGain->uniDrcGainExtPresent == 1) {
-    err = _readUniDrcGainExtension(hBs, &(hUniDrcGain->uniDrcGainExtension));
-    if (err) return err;
+  if (pCoef && (gainSequenceCount ==
+                pCoef->gainSequenceCount)) { /* all sequences have been read */
+    hUniDrcGain->uniDrcGainExtPresent = FDKreadBits(hBs, 1);
+    if (hUniDrcGain->uniDrcGainExtPresent == 1) {
+      err = _readUniDrcGainExtension(hBs, &(hUniDrcGain->uniDrcGainExtension));
+      if (err) return err;
+    }
   }
 
   if (err == DE_OK && gainSequenceCount > 0) {

fdk-aac-2.0.2.tar.gz/libFDK/include/FDK_archdef.h -> fdk-aac-2.0.3.tar.gz/libFDK/include/FDK_archdef.h Changed

fdk-aac-2.0.2.tar.gz/libFDK/include/fft.h -> fdk-aac-2.0.3.tar.gz/libFDK/include/fft.h Changed

fdk-aac-2.0.2.tar.gz/libFDK/include/nlc_dec.h -> fdk-aac-2.0.3.tar.gz/libFDK/include/nlc_dec.h Changed

fdk-aac-2.0.2.tar.gz/libFDK/src/FDK_hybrid.cpp -> fdk-aac-2.0.3.tar.gz/libFDK/src/FDK_hybrid.cpp Changed

@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -539,11 +539,11 @@
   i6 = pQmfImagpReadIdx6 >> 2;
 
   FDK_ASSERT((invert == 0) || (invert == 1));
-  mHybridReal0 + invert = (r6 + r1) << 1;
-  mHybridImag0 + invert = (i6 + i1) << 1;
+  mHybridReal0 + invert = SATURATE_LEFT_SHIFT((r6 + r1), 1, DFRACT_BITS);
+  mHybridImag0 + invert = SATURATE_LEFT_SHIFT((i6 + i1), 1, DFRACT_BITS);
 
-  mHybridReal1 - invert = (r6 - r1) << 1;
-  mHybridImag1 - invert = (i6 - i1) << 1;
+  mHybridReal1 - invert = SATURATE_LEFT_SHIFT((r6 - r1), 1, DFRACT_BITS);
+  mHybridImag1 - invert = SATURATE_LEFT_SHIFT((i6 - i1), 1, DFRACT_BITS);
 }
 
 static void fourChannelFiltering(const FIXP_DBL *const pQmfReal,
@@ -766,15 +766,15 @@
     mHybridReal3 = pfftFFT_IDX_R(1) << sc;
     mHybridImag3 = pfftFFT_IDX_I(1) << sc;
 
-    mHybridReal4 = pfftFFT_IDX_R(2) << sc;
-    mHybridReal4 += pfftFFT_IDX_R(5) << sc;
-    mHybridImag4 = pfftFFT_IDX_I(2) << sc;
-    mHybridImag4 += pfftFFT_IDX_I(5) << sc;
+    mHybridReal4 = SATURATE_LEFT_SHIFT(
+        (pfftFFT_IDX_R(2) + pfftFFT_IDX_R(5)), sc, DFRACT_BITS);
+    mHybridImag4 = SATURATE_LEFT_SHIFT(
+        (pfftFFT_IDX_I(2) + pfftFFT_IDX_I(5)), sc, DFRACT_BITS);
 
-    mHybridReal5 = pfftFFT_IDX_R(3) << sc;
-    mHybridReal5 += pfftFFT_IDX_R(4) << sc;
-    mHybridImag5 = pfftFFT_IDX_I(3) << sc;
-    mHybridImag5 += pfftFFT_IDX_I(4) << sc;
+    mHybridReal5 = SATURATE_LEFT_SHIFT(
+        (pfftFFT_IDX_R(3) + pfftFFT_IDX_R(4)), sc, DFRACT_BITS);
+    mHybridImag5 = SATURATE_LEFT_SHIFT(
+        (pfftFFT_IDX_I(3) + pfftFFT_IDX_I(4)), sc, DFRACT_BITS);
   } else {
     for (k = 0; k < 8; k++) {
       mHybridRealk = pfftFFT_IDX_R(k) << sc;

fdk-aac-2.0.2.tar.gz/libFDK/src/autocorr2nd.cpp -> fdk-aac-2.0.3.tar.gz/libFDK/src/autocorr2nd.cpp Changed

@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -102,11 +102,6 @@
 
 #include "autocorr2nd.h"
 
-/*  If the accumulator does not provide enough overflow bits,
-    products have to be shifted down in the autocorrelation below. */
-#define SHIFT_FACTOR (5)
-#define SHIFT >> (SHIFT_FACTOR)
-
 /*!
  *
  * \brief Calculate second order autocorrelation using 2 accumulators
@@ -126,45 +121,49 @@
 
   const FIXP_DBL *realBuf = reBuffer;
 
+  const int len_scale = fMax(DFRACT_BITS - fNormz((FIXP_DBL)(len / 2)), 1);
   /*
     r11r,r22r
     r01r,r12r
     r02r
   */
   pReBuf = realBuf - 2;
-  accu5 = ((fMultDiv2(pReBuf0, pReBuf2) + fMultDiv2(pReBuf1, pReBuf3))
-               SHIFT);
+  accu5 =
+      ((fMultDiv2(pReBuf0, pReBuf2) + fMultDiv2(pReBuf1, pReBuf3)) >>
+       len_scale);
   pReBuf++;
 
   /* len must be even */
-  accu1 = fPow2Div2(pReBuf0) SHIFT;
-  accu3 = fMultDiv2(pReBuf0, pReBuf1) SHIFT;
+  accu1 = fPow2Div2(pReBuf0) >> len_scale;
+  accu3 = fMultDiv2(pReBuf0, pReBuf1) >> len_scale;
   pReBuf++;
 
   for (j = (len - 2) >> 1; j != 0; j--, pReBuf += 2) {
-    accu1 += ((fPow2Div2(pReBuf0) + fPow2Div2(pReBuf1)) SHIFT);
+    accu1 += ((fPow2Div2(pReBuf0) + fPow2Div2(pReBuf1)) >> len_scale);
 
-    accu3 += ((fMultDiv2(pReBuf0, pReBuf1) +
-               fMultDiv2(pReBuf1, pReBuf2)) SHIFT);
+    accu3 +=
+        ((fMultDiv2(pReBuf0, pReBuf1) + fMultDiv2(pReBuf1, pReBuf2)) >>
+         len_scale);
 
-    accu5 += ((fMultDiv2(pReBuf0, pReBuf2) +
-               fMultDiv2(pReBuf1, pReBuf3)) SHIFT);
+    accu5 +=
+        ((fMultDiv2(pReBuf0, pReBuf2) + fMultDiv2(pReBuf1, pReBuf3)) >>
+         len_scale);
   }
 
-  accu2 = (fPow2Div2(realBuf-2) SHIFT);
+  accu2 = (fPow2Div2(realBuf-2) >> len_scale);
   accu2 += accu1;
 
-  accu1 += (fPow2Div2(realBuflen - 2) SHIFT);
+  accu1 += (fPow2Div2(realBuflen - 2) >> len_scale);
 
-  accu4 = (fMultDiv2(realBuf-1, realBuf-2) SHIFT);
+  accu4 = (fMultDiv2(realBuf-1, realBuf-2) >> len_scale);
   accu4 += accu3;
 
-  accu3 += (fMultDiv2(realBuflen - 1, realBuflen - 2) SHIFT);
+  accu3 += (fMultDiv2(realBuflen - 1, realBuflen - 2) >> len_scale);
 
   mScale = CntLeadingZeros(
                (accu1 | accu2 | fAbs(accu3) | fAbs(accu4) | fAbs(accu5))) -
            1;
-  autoCorrScaling = mScale - 1 - SHIFT_FACTOR; /* -1 because of fMultDiv2*/
+  autoCorrScaling = mScale - 1 - len_scale; /* -1 because of fMultDiv2*/
 
   /* Scale to common scale factor */
   ac->r11r = accu1 << mScale;
@@ -190,7 +189,7 @@
     const FIXP_DBL *imBuffer, /*!< Pointer to imag part of input samples */
     const int len /*!< Number of input samples (should be smaller than 128) */
 ) {
-  int j, autoCorrScaling, mScale, len_scale;
+  int j, autoCorrScaling, mScale;
 
   FIXP_DBL accu0, accu1, accu2, accu3, accu4, accu5, accu6, accu7, accu8;
 
@@ -199,7 +198,7 @@
   const FIXP_DBL *realBuf = reBuffer;
   const FIXP_DBL *imagBuf = imBuffer;
 
-  (len > 64) ? (len_scale = 6) : (len_scale = 5);
+  const int len_scale = fMax(DFRACT_BITS - fNormz((FIXP_DBL)len), 1);
   /*
     r00r,
     r11r,r22r

fdk-aac-2.0.2.tar.gz/libFDK/src/dct.cpp -> fdk-aac-2.0.3.tar.gz/libFDK/src/dct.cpp Changed

@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -305,9 +305,8 @@
 
   {
     for (i = 0; i < M; i++) {
-      tmpi = pDat2 * i >> 1; /* dit_fft expects 1 bit scaled input values */
-      tmpL - 1 - i =
-          pDat2 * i + 1 >> 1; /* dit_fft expects 1 bit scaled input values */
+      tmpi = pDat2 * i >> 2;
+      tmpL - 1 - i = pDat2 * i + 1 >> 2;
     }
   }
 
@@ -337,15 +336,14 @@
     a1 = ((pTmp_00 >> 1) + (pTmp_10 >> 1));
     a2 = ((pTmp_01 >> 1) - (pTmp_11 >> 1));
 
-    cplxMultDiv2(&accu3, &accu4, (a1 + accu2), -(accu1 + a2),
-                 sin_twiddlei * inc);
-    pDatL - i = accu4;
-    pDati = accu3;
+    cplxMult(&accu3, &accu4, (accu1 + a2), (a1 + accu2), sin_twiddlei * inc);
+    pDatL - i = -accu3;
+    pDati = accu4;
 
-    cplxMultDiv2(&accu3, &accu4, (a1 - accu2), -(accu1 - a2),
-                 sin_twiddle(M - i) * inc);
-    pDatM + i = accu4;
-    pDatM - i = accu3;
+    cplxMult(&accu3, &accu4, (accu1 - a2), (a1 - accu2),
+             sin_twiddle(M - i) * inc);
+    pDatM + i = -accu3;
+    pDatM - i = accu4;
 
     /* Create index helper variables for (4*i)*inc indexed equivalent values of
      * short tables. */
@@ -356,12 +354,12 @@
     }
   }
 
-  cplxMultDiv2(&accu1, &accu2, tmpM, tmpM + 1, sin_twiddle(M / 2) * inc);
+  cplxMult(&accu1, &accu2, tmpM, tmpM + 1, sin_twiddle(M / 2) * inc);
   pDatL - (M / 2) = accu2;
   pDatM / 2 = accu1;
 
-  pDat0 = (tmp0 >> 1) + (tmp1 >> 1);
-  pDatM = fMult(((tmp0 >> 1) - (tmp1 >> 1)),
+  pDat0 = tmp0 + tmp1;
+  pDatM = fMult(tmp0 - tmp1,
                   sin_twiddleM * inc.v.re); /* cos((PI/(2*L))*M); */
 
   *pDat_e += 2;

fdk-aac-2.0.2.tar.gz/libFDK/src/fft.cpp -> fdk-aac-2.0.3.tar.gz/libFDK/src/fft.cpp Changed

fdk-aac-2.0.2.tar.gz/libFDK/src/nlc_dec.cpp -> fdk-aac-2.0.3.tar.gz/libFDK/src/nlc_dec.cpp Changed

@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -568,12 +568,12 @@
 static ERROR_t huff_decode(HANDLE_FDK_BITSTREAM strm, SCHAR* out_data_1,
                            SCHAR* out_data_2, DATA_TYPE data_type,
                            DIFF_TYPE diff_type_1, DIFF_TYPE diff_type_2,
-                           int num_val, int* cdg_scheme, int ldMode) {
+                           int num_val, PAIRING* pairing_scheme, int ldMode) {
   ERROR_t err = HUFFDEC_OK;
+  CODING_SCHEME coding_scheme = HUFF_1D;
   DIFF_TYPE diff_type;
 
   int i = 0;
-  ULONG data = 0;
 
   SCHAR pair_vec282;
 
@@ -596,15 +596,13 @@
   int hufYY;
 
   /* Coding scheme */
-  data = FDKreadBits(strm, 1);
-  *cdg_scheme = (data << PAIR_SHIFT);
+  coding_scheme = (CODING_SCHEME)FDKreadBits(strm, 1);
 
-  if (*cdg_scheme >> PAIR_SHIFT == HUFF_2D) {
+  if (coding_scheme == HUFF_2D) {
     if ((out_data_1 != NULL) && (out_data_2 != NULL) && (ldMode == 0)) {
-      data = FDKreadBits(strm, 1);
-      *cdg_scheme = (*cdg_scheme | data);
+      *pairing_scheme = (PAIRING)FDKreadBits(strm, 1);
     } else {
-      *cdg_scheme = (*cdg_scheme | FREQ_PAIR);
+      *pairing_scheme = FREQ_PAIR;
     }
   }
 
@@ -613,7 +611,7 @@
     hufYY2 = diff_type_2;
   }
 
-  switch (*cdg_scheme >> PAIR_SHIFT) {
+  switch (coding_scheme) {
     case HUFF_1D:
       p0_flag0 = (diff_type_1 == DIFF_FREQ);
       p0_flag1 = (diff_type_2 == DIFF_FREQ);
@@ -634,7 +632,7 @@
 
     case HUFF_2D:
 
-      switch (*cdg_scheme & PAIR_MASK) {
+      switch (*pairing_scheme) {
         case FREQ_PAIR:
 
           if (out_data_1 != NULL) {
@@ -843,7 +841,7 @@
   SCHAR* pDataVec2 = {NULL, NULL};
 
   DIFF_TYPE diff_type2 = {DIFF_FREQ, DIFF_FREQ};
-  int cdg_scheme = HUFF_1D;
+  PAIRING pairing = FREQ_PAIR;
   DIRECTION direction = BACKWARDS;
 
   switch (data_type) {
@@ -959,7 +957,7 @@
     }
     /* Huffman decoding */
     err = huff_decode(strm, pDataVec0, pDataVec1, data_type, diff_type0,
-                      diff_type1, dataBands, &cdg_scheme,
+                      diff_type1, dataBands, &pairing,
                       (DECODER == SAOC_DECODER));
     if (err != HUFFDEC_OK) {
       return HUFFDEC_NOTOK;
@@ -986,8 +984,8 @@
         }
       }
 
-      mixed_time_pair = (diff_type0 != diff_type1) &&
-                        ((cdg_scheme & PAIR_MASK) == TIME_PAIR);
+      mixed_time_pair =
+          (diff_type0 != diff_type1) && (pairing == TIME_PAIR);
 
       if (direction == BACKWARDS) {
         if (diff_type0 == DIFF_FREQ) {

fdk-aac-2.0.2.tar.gz/libMpegTPDec/include/tp_data.h -> fdk-aac-2.0.3.tar.gz/libMpegTPDec/include/tp_data.h Changed

fdk-aac-2.0.2.tar.gz/libMpegTPDec/src/tpdec_asc.cpp -> fdk-aac-2.0.3.tar.gz/libMpegTPDec/src/tpdec_asc.cpp Changed

@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -266,11 +266,118 @@
   return (err);
 }
 
+/**
+ * \brief Sanity checks for program config element.
+ *        Check order of elements according to ISO/IEC 13818-7:2003(E),
+ * chapter 8.5.1
+ *
+ * \param pPce  pointer to program config element.
+ *
+ * \return  0 if successful, otherwise 1.
+ */
+static int CProgramConfig_Check(CProgramConfig *pPce) {
+  INT i;
+  INT err = 0;
+  INT numBackChannels3 = {0};
+  INT numSideChannels3 = {0};
+  INT numFrontChannels3 = {0};
+  UCHAR *pCpeFront = pPce->FrontElementIsCpe;
+  UCHAR *pCpeSide = pPce->SideElementIsCpe;
+  UCHAR *pCpeBack = pPce->BackElementIsCpe;
+  UCHAR *pHeight;
+
+  pHeight = pPce->BackElementHeightInfo;
+  for (i = 0; i < pPce->NumBackChannelElements; i++) {
+    numBackChannels*pHeight += pPce->BackElementIsCpei ? 2 : 1;
+    pHeight++;
+  }
+  pHeight = pPce->SideElementHeightInfo;
+  for (i = 0; i < pPce->NumSideChannelElements; i++) {
+    numSideChannels*pHeight += pPce->SideElementIsCpei ? 2 : 1;
+    pHeight++;
+  }
+  pHeight = pPce->FrontElementHeightInfo;
+  for (i = 0; i < pPce->NumFrontChannelElements; i++) {
+    numFrontChannels*pHeight += pPce->FrontElementIsCpei ? 2 : 1;
+    pHeight++;
+  }
+
+  /* 0 = normal height channels, 1 = top height channels, 2 = bottom height
+   * channels */
+  for (i = 0; i < 3; i++) {
+    /* if number of channels is odd => first element must be a SCE (front center
+     * channel) */
+    if (numFrontChannelsi & 1) {
+      if (*pCpeFront++ == ID_CPE) {
+        err = 1;
+        goto bail;
+      }
+      numFrontChannelsi--;
+    }
+    while (numFrontChannelsi > 0) {
+      /* must be CPE or paired SCE */
+      if (*pCpeFront++ == ID_SCE) {
+        if (*pCpeFront++ == ID_CPE) {
+          err = 1;
+          goto bail;
+        }
+      }
+      numFrontChannelsi -= 2;
+    };
+
+    /* in case that a top center surround channel (Ts) is transmitted the number
+     * of channels can be odd */
+    if (i != 1) {
+      /* number of channels must be even */
+      if (numSideChannelsi & 1) {
+        err = 1;
+        goto bail;
+      }
+      while (numSideChannelsi > 0) {
+        /* must be CPE or paired SCE */
+        if (*pCpeSide++ == ID_SCE) {
+          if (*pCpeSide++ == ID_CPE) {
+            err = 1;
+            goto bail;
+          }
+        }
+        numSideChannelsi -= 2;
+      };
+    }
+
+    while (numBackChannelsi > 1) {
+      /* must be CPE or paired SCE */
+      if (*pCpeBack++ == ID_SCE) {
+        if (*pCpeBack++ == ID_CPE) {
+          err = 1;
+          goto bail;
+        }
+      }
+      numBackChannelsi -= 2;
+    };
+    /* if number of channels is odd => last element must be a SCE (back center
+     * channel) */
+    if (numBackChannelsi) {
+      if (*pCpeBack++ == ID_CPE) {
+        err = 1;
+        goto bail;
+      }
+    }
+  }
+
+bail:
+
+  return err;
+}
+
 void CProgramConfig_Read(CProgramConfig *pPce, HANDLE_FDK_BITSTREAM bs,
                          UINT alignmentAnchor) {
-  int i, err = 0;
+  int i;
   int commentBytes;
+  UCHAR tag, isCpe;
+  UCHAR checkElementTagSelect3PC_FSB_CHANNELS_MAX = {{0}};
 
+  pPce->isValid = 1;
   pPce->NumEffectiveChannels = 0;
   pPce->NumChannels = 0;
   pPce->ElementInstanceTag = (UCHAR)FDKreadBits(bs, 4);
@@ -297,28 +404,60 @@
   }
 
   for (i = 0; i < pPce->NumFrontChannelElements; i++) {
-    pPce->FrontElementIsCpei = (UCHAR)FDKreadBits(bs, 1);
-    pPce->FrontElementTagSelecti = (UCHAR)FDKreadBits(bs, 4);
+    pPce->FrontElementIsCpei = isCpe = (UCHAR)FDKreadBits(bs, 1);
+    pPce->FrontElementTagSelecti = tag = (UCHAR)FDKreadBits(bs, 4);
     pPce->NumChannels += pPce->FrontElementIsCpei ? 2 : 1;
+
+    /* Check element instance tag according to ISO/IEC 13818-7:2003(E),
+     * chapter 8.2.1.1 */
+    if (checkElementTagSelectisCpetag == 0) {
+      checkElementTagSelectisCpetag = 1;
+    } else {
+      pPce->isValid = 0;
+    }
   }
 
   for (i = 0; i < pPce->NumSideChannelElements; i++) {
-    pPce->SideElementIsCpei = (UCHAR)FDKreadBits(bs, 1);
-    pPce->SideElementTagSelecti = (UCHAR)FDKreadBits(bs, 4);
+    pPce->SideElementIsCpei = isCpe = (UCHAR)FDKreadBits(bs, 1);
+    pPce->SideElementTagSelecti = tag = (UCHAR)FDKreadBits(bs, 4);
     pPce->NumChannels += pPce->SideElementIsCpei ? 2 : 1;
+
+    /* Check element instance tag according to ISO/IEC 13818-7:2003(E),
+     * chapter 8.2.1.1 */
+    if (checkElementTagSelectisCpetag == 0) {
+      checkElementTagSelectisCpetag = 1;
+    } else {
+      pPce->isValid = 0;
+    }
   }
 
   for (i = 0; i < pPce->NumBackChannelElements; i++) {
-    pPce->BackElementIsCpei = (UCHAR)FDKreadBits(bs, 1);
-    pPce->BackElementTagSelecti = (UCHAR)FDKreadBits(bs, 4);
+    pPce->BackElementIsCpei = isCpe = (UCHAR)FDKreadBits(bs, 1);
+    pPce->BackElementTagSelecti = tag = (UCHAR)FDKreadBits(bs, 4);
     pPce->NumChannels += pPce->BackElementIsCpei ? 2 : 1;
+
+    /* Check element instance tag according to ISO/IEC 13818-7:2003(E),
+     * chapter 8.2.1.1 */
+    if (checkElementTagSelectisCpetag == 0) {
+      checkElementTagSelectisCpetag = 1;
+    } else {
+      pPce->isValid = 0;
+    }
   }
 
   pPce->NumEffectiveChannels = pPce->NumChannels;
 
   for (i = 0; i < pPce->NumLfeChannelElements; i++) {
-    pPce->LfeElementTagSelecti = (UCHAR)FDKreadBits(bs, 4);
+    pPce->LfeElementTagSelecti = tag = (UCHAR)FDKreadBits(bs, 4);
     pPce->NumChannels += 1;
+
+    /* Check element instance tag according to ISO/IEC 13818-7:2003(E),
+     * chapter 8.2.1.1 */
+    if (checkElementTagSelect2tag == 0) {
+      checkElementTagSelect2tag = 1;
+    } else {
+      pPce->isValid = 0;
+    }
   }
 
   for (i = 0; i < pPce->NumAssocDataElements; i++) {
@@ -336,7 +475,15 @@
   commentBytes = pPce->CommentFieldBytes;
 
   /* Search for height info extension and read it if available */
-  err = CProgramConfig_ReadHeightExt(pPce, bs, &commentBytes, alignmentAnchor);
+  if (CProgramConfig_ReadHeightExt(pPce, bs, &commentBytes, alignmentAnchor)) {
+    pPce->isValid = 0;
+  }
+
+  /* Check order of elements according to ISO / IEC 13818 - 7:2003(E),
+   * chapter 8.5.1 */
+  if (CProgramConfig_Check(pPce)) {
+    pPce->isValid = 0;
+  }
 
   for (i = 0; i < commentBytes; i++) {
     UCHAR text;
@@ -347,8 +494,6 @@
       pPce->Commenti = text;
     }
   }
-
-  pPce->isValid = (err) ? 0 : 1;
 }
 
 /*
@@ -1415,7 +1560,7 @@
               cb->cbSscData, hBs, asc->m_aot,
               asc->m_samplingFrequency << esc->m_sbrSamplingRate,
               asc->m_samplesPerFrame << esc->m_sbrSamplingRate,
-              1,  /* stereoConfigIndex */
+              asc->m_channelConfiguration, 1, /* stereoConfigIndex */
               -1, /* nTimeSlots: read from bitstream */
               eldExtLen, asc->configMode, &asc->SacConfigChanged);
           if (ErrorStatus != TRANSPORTDEC_OK) {
@@ -1549,7 +1694,7 @@
                                            const AUDIO_OBJECT_TYPE aot) {
   TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK;
 
-  int usacExtElementType = escapedValue(hBs, 4, 8, 16);
+  UINT usacExtElementType = escapedValue(hBs, 4, 8, 16);
 
   /* recurve extension elements which are invalid for USAC */
   if (aot == AOT_USAC) {
@@ -1566,7 +1711,6 @@
     }
   }
 
-  extElement->usacExtElementType = (USAC_EXT_ELEMENT_TYPE)usacExtElementType;
   int usacExtElementConfigLength = escapedValue(hBs, 4, 8, 16);
   extElement->usacExtElementConfigLength = (USHORT)usacExtElementConfigLength;
   INT bsAnchor;
@@ -1600,8 +1744,10 @@
       }
     } break;
     default:
+      usacExtElementType = ID_EXT_ELE_UNKNOWN;
       break;
   }
+  extElement->usacExtElementType = (USAC_EXT_ELEMENT_TYPE)usacExtElementType;
 
   /* Adjust bit stream position. This is required because of byte alignment and
    * unhandled extensions. */
@@ -1630,8 +1776,12 @@
   TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK;
 
   int numConfigExtensions;
-  int usacConfigExtType;
+  UINT usacConfigExtType;
   int usacConfigExtLength;
+  int loudnessInfoSetIndex =
+      -1; /* index of loudnessInfoSet config extension. -1 if not contained. */
+  int tmp_subStreamIndex = 0;
+  AUDIO_OBJECT_TYPE tmp_aot = AOT_USAC;
 
   numConfigExtensions = (int)escapedValue(hBs, 2, 4, 8) + 1;
   for (int confExtIdx = 0; confExtIdx < numConfigExtensions; confExtIdx++) {
@@ -1661,10 +1811,12 @@
           ErrorStatus = (TRANSPORTDEC_ERROR)cb->cbUniDrc(
               cb->cbUniDrcData, hBs, usacConfigExtLength,
               1, /* loudnessInfoSet */
-              0, loudnessInfoSetConfigExtensionPosition, AOT_USAC);
+              tmp_subStreamIndex, loudnessInfoSetConfigExtensionPosition,
+              tmp_aot);
           if (ErrorStatus != TRANSPORTDEC_OK) {
             return ErrorStatus;
           }
+          loudnessInfoSetIndex = confExtIdx;
         }
       } break;
       default:
@@ -1680,6 +1832,17 @@
     FDKpushFor(hBs, usacConfigExtLength);
   }
 
+  if (loudnessInfoSetIndex == -1 && cb->cbUniDrc != NULL) {
+    /* no loudnessInfoSet contained. Clear the loudnessInfoSet struct by feeding
+     * an empty config extension */
+    ErrorStatus = (TRANSPORTDEC_ERROR)cb->cbUniDrc(
+        cb->cbUniDrcData, NULL, 0, 1 /* loudnessInfoSet */, tmp_subStreamIndex,
+        0, tmp_aot);
+    if (ErrorStatus != TRANSPORTDEC_OK) {
+      return ErrorStatus;
+    }
+  }
+
   return ErrorStatus;
 }
 
@@ -1696,6 +1859,8 @@
   int channelElementIdx =
       0; /* index for elements which contain audio channels (sce, cpe, lfe) */
   SC_CHANNEL_CONFIG sc_chan_config = {0, 0, 0, 0};
+  int uniDrcElement =
+      -1; /* index of uniDrc extension element. -1 if not contained. */
 
   numberOfElements = (int)escapedValue(hBs, 4, 8, 16) + 1;
   usc->m_usacNumElements = numberOfElements;
@@ -1826,6 +1991,8 @@
               /* Mps212Config() ISO/IEC FDIS 23003-3 */
               if (cb->cbSsc(cb->cbSscData, hBs, asc->m_aot,
                             asc->m_extensionSamplingFrequency, samplesPerFrame,
+                            1, /* only downmix channels (residual channels are
+                                  not counted) */
                             usc->elementi.m_stereoConfigIndex,
                             usc->m_coreSbrFrameLengthIndex,
                             0, /* don't know the length */
@@ -1869,6 +2036,10 @@
       case ID_USAC_EXT:
         ErrorStatus = extElementConfig(&usc->elementi.extElement, hBs, cb, 0,
                                        asc->m_samplesPerFrame, 0, asc->m_aot);
+        if (usc->elementi.extElement.usacExtElementType ==
+            ID_EXT_ELE_UNI_DRC) {
+          uniDrcElement = i;
+        }
 
         if (ErrorStatus) {
           return ErrorStatus;
@@ -1897,6 +2068,18 @@
     }
   }
 
+  if (uniDrcElement == -1 && cb->cbUniDrc != NULL) {
+    /* no uniDrcConfig contained. Clear the uniDrcConfig struct by feeding an
+     * empty extension element */
+    int subStreamIndex = 0;
+    ErrorStatus = (TRANSPORTDEC_ERROR)cb->cbUniDrc(
+        cb->cbUniDrcData, NULL, 0, 0 /* uniDrcConfig */, subStreamIndex, 0,
+        asc->m_aot);
+    if (ErrorStatus != TRANSPORTDEC_OK) {
+      return ErrorStatus;
+    }
+  }
+
   return ErrorStatus;
 }
 
@@ -1983,6 +2166,14 @@
     if (err != TRANSPORTDEC_OK) {
       return err;
     }
+  } else if (cb->cbUniDrc != NULL) {
+    /* no loudnessInfoSet contained. Clear the loudnessInfoSet struct by feeding
+     * an empty config extension */
+    err = (TRANSPORTDEC_ERROR)cb->cbUniDrc(
+        cb->cbUniDrcData, NULL, 0, 1 /* loudnessInfoSet */, 0, 0, asc->m_aot);
+    if (err != TRANSPORTDEC_OK) {
+      return err;
+    }
   }
 
   /* sanity check whether number of channels signaled in UsacDecoderConfig()
@@ -1995,9 +2186,11 @@
 
   /* Copy UsacConfig() to asc->m_sc.m_usacConfig.UsacConfig buffer. */
   INT configSize_bits = (INT)FDKgetValidBits(hBs) - nbits;
-  StoreConfigAsBitstream(hBs, configSize_bits,
-                         asc->m_sc.m_usacConfig.UsacConfig,
-                         TP_USAC_MAX_CONFIG_LEN);
+  if (StoreConfigAsBitstream(hBs, configSize_bits,
+                             asc->m_sc.m_usacConfig.UsacConfig,
+                             TP_USAC_MAX_CONFIG_LEN)) {
+    return TRANSPORTDEC_PARSE_ERROR;
+  }
   asc->m_sc.m_usacConfig.UsacConfigBits = fAbs(configSize_bits);
 
   return err;
@@ -2218,7 +2411,7 @@
     case AOT_MPEGS:
       if (cb->cbSsc != NULL) {
         if (cb->cbSsc(cb->cbSscData, bs, self->m_aot, self->m_samplingFrequency,
-                      self->m_samplesPerFrame, 1,
+                      self->m_samplesPerFrame, self->m_channelConfiguration, 1,
                       -1, /* nTimeSlots: read from bitstream */
                       0,  /* don't know the length */
                       self->configMode, &self->SacConfigChanged)) {
@@ -2299,8 +2492,10 @@
   /* Copy config() to asc->config buffer. */
   if ((ErrorStatus == TRANSPORTDEC_OK) && (self->m_aot == AOT_USAC)) {
     INT configSize_bits = (INT)FDKgetValidBits(bs) - (INT)ascStartAnchor;
-    StoreConfigAsBitstream(bs, configSize_bits, self->config,
-                           TP_USAC_MAX_CONFIG_LEN);
+    if (StoreConfigAsBitstream(bs, configSize_bits, self->config,
+                               TP_USAC_MAX_CONFIG_LEN)) {
+      return TRANSPORTDEC_PARSE_ERROR;
+    }
     self->configBits = fAbs(configSize_bits);
   }
 
@@ -2414,6 +2609,8 @@
                 cb->cbSscData, hBs,
                 AOT_DRM_USAC, /* syntax differs from MPEG Mps212Config() */
                 asc->m_extensionSamplingFrequency, samplesPerFrame,
+                1, /* only downmix channels (residual channels are not
+                      counted) */
                 usc->elementelemIdx.m_stereoConfigIndex,
                 usc->m_coreSbrFrameLengthIndex, 0, /* don't know the length */
                 asc->configMode, &asc->SacConfigChanged);

fdk-aac-2.0.2.tar.gz/libMpegTPDec/src/tpdec_latm.cpp -> fdk-aac-2.0.3.tar.gz/libMpegTPDec/src/tpdec_latm.cpp Changed

@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -591,6 +591,18 @@
   return (ErrorStatus);
 }
 
+static int CLatmDemux_ReadAuChunkLengthInfo(HANDLE_FDK_BITSTREAM bs) {
+  int len = 0, tmp = 255;
+  int validBytes = (int)FDKgetValidBits(bs) >> 3;
+
+  while (tmp == 255 && validBytes-- > 0) {
+    tmp = (int)FDKreadBits(bs, 8);
+    len += tmp;
+  }
+
+  return ((tmp == 255) ? -1 : (len << 3));
+}
+
 TRANSPORTDEC_ERROR CLatmDemux_ReadPayloadLengthInfo(HANDLE_FDK_BITSTREAM bs,
                                                     CLatmDemux *pLatmDemux) {
   TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK;
@@ -602,11 +614,17 @@
       FDK_ASSERT(pLatmDemux->m_numLayerprog <= LATM_MAX_LAYER);
       for (UINT lay = 0; lay < pLatmDemux->m_numLayerprog; lay++) {
         LATM_LAYER_INFO *p_linfo = &pLatmDemux->m_linfoproglay;
+        int auChunkLengthInfo = 0;
 
         switch (p_linfo->m_frameLengthType) {
           case 0:
-            p_linfo->m_frameLengthInBits = CLatmDemux_ReadAuChunkLengthInfo(bs);
-            totalPayloadBits += p_linfo->m_frameLengthInBits;
+            auChunkLengthInfo = CLatmDemux_ReadAuChunkLengthInfo(bs);
+            if (auChunkLengthInfo >= 0) {
+              p_linfo->m_frameLengthInBits = (UINT)auChunkLengthInfo;
+              totalPayloadBits += p_linfo->m_frameLengthInBits;
+            } else {
+              return TRANSPORTDEC_PARSE_ERROR;
+            }
             break;
           case 3:
           case 5:
@@ -627,23 +645,6 @@
   return (ErrorStatus);
 }
 
-int CLatmDemux_ReadAuChunkLengthInfo(HANDLE_FDK_BITSTREAM bs) {
-  UCHAR endFlag;
-  int len = 0;
-
-  do {
-    UCHAR tmp = (UCHAR)FDKreadBits(bs, 8);
-    endFlag = (tmp < 255);
-
-    len += tmp;
-
-  } while (endFlag == 0);
-
-  len <<= 3; /* convert from bytes to bits */
-
-  return len;
-}
-
 UINT CLatmDemux_GetFrameLengthInBits(CLatmDemux *pLatmDemux, const UINT prog,
                                      const UINT layer) {
   UINT nFrameLenBits = 0;

fdk-aac-2.0.2.tar.gz/libMpegTPDec/src/tpdec_latm.h -> fdk-aac-2.0.3.tar.gz/libMpegTPDec/src/tpdec_latm.h Changed

fdk-aac-2.0.2.tar.gz/libMpegTPDec/src/tpdec_lib.cpp -> fdk-aac-2.0.3.tar.gz/libMpegTPDec/src/tpdec_lib.cpp Changed

fdk-aac-2.0.2.tar.gz/libMpegTPEnc/include/tp_data.h -> fdk-aac-2.0.3.tar.gz/libMpegTPEnc/include/tp_data.h Changed

fdk-aac-2.0.2.tar.gz/libMpegTPEnc/src/tpenc_asc.cpp -> fdk-aac-2.0.3.tar.gz/libMpegTPEnc/src/tpenc_asc.cpp Changed

fdk-aac-2.0.2.tar.gz/libPCMutils/src/limiter.cpp -> fdk-aac-2.0.3.tar.gz/libPCMutils/src/limiter.cpp Changed

fdk-aac-2.0.2.tar.gz/libPCMutils/src/pcmdmx_lib.cpp -> fdk-aac-2.0.3.tar.gz/libPCMutils/src/pcmdmx_lib.cpp Changed

@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -494,13 +494,40 @@
   return plainChMode;
 }
 
-static inline UINT getIdxSum(UCHAR numCh) {
-  UINT result = 0;
-  int i;
-  for (i = 1; i < numCh; i += 1) {
-    result += i;
+/** Validates the channel indices of all channels present in the bitstream.
+ * The channel indices have to be consecutive and unique for each audio channel
+ *type.
+ * @param in The total number of channels of the given configuration.
+ * @param in The total number of channels of the current audio channel type of
+ *the given configuration.
+ * @param in Audio channel type to be examined.
+ * @param in Array holding the corresponding channel types for each channel.
+ * @param in Array holding the corresponding channel type indices for each
+ *channel.
+ * @returns Returns 1 on success, returns 0 on error.
+ **/
+static UINT validateIndices(UINT numChannels, UINT numChannelsPlaneAndGrp,
+                            AUDIO_CHANNEL_TYPE aChType,
+                            const AUDIO_CHANNEL_TYPE channelType,
+                            const UCHAR channelIndices) {
+  for (UINT reqValue = 0; reqValue < numChannelsPlaneAndGrp; reqValue++) {
+    int found = FALSE;
+    for (UINT i = 0; i < numChannels; i++) {
+      if (channelTypei == aChType) {
+        if (channelIndicesi == reqValue) {
+          if (found == TRUE) {
+            return 0; /* Found channel index a second time */
+          } else {
+            found = TRUE; /* Found channel index */
+          }
+        }
+      }
+    }
+    if (found == FALSE) {
+      return 0; /* Did not find channel index */
+    }
   }
-  return result;
+  return 1; /* Successfully validated channel indices */
 }
 
 /** Evaluate a given channel configuration and extract a packed channel mode. In
@@ -523,7 +550,6 @@
     UCHAR offsetTable(8),                 /* out */
     PCM_DMX_CHANNEL_MODE *chMode            /* out */
 ) {
-  UINT idxSum(3)(4);
   UCHAR numCh(3)(4);
   UCHAR mapped(8);
   PCM_DMX_SPEAKER_POSITION spkrPos(8);
@@ -538,7 +564,6 @@
   FDK_ASSERT(chMode != NULL);
 
   /* For details see ISO/IEC 13818-7:2005(E), 8.5.3 Channel configuration */
-  FDKmemclear(idxSum, (3) * (4) * sizeof(UINT));
   FDKmemclear(numCh, (3) * (4) * sizeof(UCHAR));
   FDKmemclear(mapped, (8) * sizeof(UCHAR));
   FDKmemclear(spkrPos, (8) * sizeof(PCM_DMX_SPEAKER_POSITION));
@@ -552,19 +577,22 @@
         (channelTypech & 0x0F) - 1,
         0); /* Assign all undefined channels (ACT_NONE) to front channels. */
     numChchannelTypech >> 4chGrp += 1;
-    idxSumchannelTypech >> 4chGrp += channelIndicesch;
   }
-  if (numChannels > TWO_CHANNEL) {
+
+  {
     int chGrp;
     /* Sanity check on the indices */
     for (chGrp = 0; chGrp < (4); chGrp += 1) {
       int plane;
       for (plane = 0; plane < (3); plane += 1) {
-        if (idxSumplanechGrp != getIdxSum(numChplanechGrp)) {
+        if (numChplanechGrp == 0) continue;
+        AUDIO_CHANNEL_TYPE aChType =
+            (AUDIO_CHANNEL_TYPE)((plane << 4) | ((chGrp + 1) & 0xF));
+        if (!validateIndices(numChannels, numChplanechGrp, aChType,
+                             channelType, channelIndices)) {
           unsigned idxCnt = 0;
           for (ch = 0; ch < numChannels; ch += 1) {
-            if (channelTypech ==
-                (AUDIO_CHANNEL_TYPE)((plane << 4) | ((chGrp + 1) & 0xF))) {
+            if (channelTypech == aChType) {
               channelIndicesch = idxCnt++;
             }
           }
@@ -583,8 +611,6 @@
         (channelTypech & 0x0F) - 1,
         0); /* Assign all undefined channels (ACT_NONE) to front channels. */
 
-    if (channelIndicesch >= numChchannelTypech >> 4chGrp)
-      return PCMDMX_INVALID_CH_CONFIG;
     spkrPosch = getSpeakerPos(channelTypech, channelIndicesch,
                                 numChchannelTypech >> 4chGrp);

fdk-aac-2.0.2.tar.gz/libSACdec/include/sac_dec_lib.h -> fdk-aac-2.0.3.tar.gz/libSACdec/include/sac_dec_lib.h Changed

fdk-aac-2.0.2.tar.gz/libSACdec/src/sac_bitdec.cpp -> fdk-aac-2.0.3.tar.gz/libSACdec/src/sac_bitdec.cpp Changed

@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -448,7 +448,6 @@
   int bsFreqRes, b3DaudioMode = 0;
   int numHeaderBits;
   int cfgStartPos, bitsAvailable;
-  int treeConfig;
 
   FDKmemclear(pSpatialSpecificConfig, sizeof(SPATIAL_SPECIFIC_CONFIG));
 
@@ -489,13 +488,18 @@
   pSpatialSpecificConfig->freqRes =
       (SPATIALDEC_FREQ_RES)freqResTable_LDbsFreqRes;
 
-  treeConfig = FDKreadBits(bitstream, 4);
+  {
+    UINT treeConfig = FDKreadBits(bitstream, 4);
 
-  if (treeConfig != SPATIALDEC_MODE_RSVD7) {
-    err = MPS_UNSUPPORTED_CONFIG;
-    goto bail;
+    switch (treeConfig) {
+      case SPATIALDEC_MODE_RSVD7:
+        pSpatialSpecificConfig->treeConfig = (SPATIALDEC_TREE_CONFIG)treeConfig;
+        break;
+      default:
+        err = MPS_UNSUPPORTED_CONFIG;
+        goto bail;
+    }
   }
-  pSpatialSpecificConfig->treeConfig = (SPATIALDEC_TREE_CONFIG) treeConfig;
 
   {
     pSpatialSpecificConfig->nOttBoxes =

fdk-aac-2.0.2.tar.gz/libSACdec/src/sac_dec.cpp -> fdk-aac-2.0.3.tar.gz/libSACdec/src/sac_dec.cpp Changed

@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -1098,6 +1098,28 @@
   }
 }
 
+/**
+ * \brief Set internal error and reset error status
+ *
+ * \param self         spatialDec handle.
+ * \param bypassMode   pointer to bypassMode.
+ * \param err          error status.
+ *
+ * \return  error status.
+ */
+static SACDEC_ERROR SpatialDecSetInternalError(spatialDec *self,
+                                               int *bypassMode,
+                                               SACDEC_ERROR err) {
+  *bypassMode = 1;
+
+  if (self->errInt == MPS_OK) {
+    /* store internal error before it gets overwritten */
+    self->errInt = err;
+  }
+
+  return MPS_OK;
+}
+
 /*******************************************************************************
  Functionname: SpatialDecApplyParameterSets
  *******************************************************************************
@@ -1118,7 +1140,7 @@
     const FDK_channelMapDescr *const mapDescr) {
   SACDEC_ERROR err = MPS_OK;
 
-  FIXP_SGL alpha;
+  FIXP_SGL alpha = FL2FXCONST_SGL(0.0);
 
   int ts;
   int ch;
@@ -1141,20 +1163,22 @@
        ts++, ts_io++) {
     int currSlot = frame->paramSlotps;
 
+    err = (currSlot < ts) ? MPS_WRONG_PARAMETERSETS : MPS_OK;
+    if (err != MPS_OK) {
+      err = SpatialDecSetInternalError(self, &bypassMode, err);
+    }
+
     /*
      * Get new parameter set
      */
     if (ts == prevSlot + 1) {
-      err = SpatialDecCalculateM1andM2(self, ps,
-                                       frame); /* input: ottCLD, ottICC, ... */
-      /* output: M1param(Real/Imag), M2(Real/Imag) */
-      if (err != MPS_OK) {
-        bypassMode = 1;
-        if (self->errInt == MPS_OK) {
-          /* store internal error befor it gets overwritten */
-          self->errInt = err;
+      if (bypassMode == 0) {
+        err = SpatialDecCalculateM1andM2(
+            self, ps, frame); /* input: ottCLD, ottICC, ... */
+                              /* output: M1param(Real/Imag), M2(Real/Imag) */
+        if (err != MPS_OK) {
+          err = SpatialDecSetInternalError(self, &bypassMode, err);
         }
-        err = MPS_OK;
       }
 
       if ((ps == 0) && (self->bOverwriteM1M2prev != 0)) {
@@ -1168,13 +1192,16 @@
         self->bOverwriteM1M2prev = 0;
       }
 
-      SpatialDecSmoothM1andM2(
-          self, frame,
-          ps); /* input: M1param(Real/Imag)(Prev), M2(Real/Imag)(Prev) */
-               /* output: M1param(Real/Imag), M2(Real/Imag) */
+      if (bypassMode == 0) {
+        SpatialDecSmoothM1andM2(
+            self, frame,
+            ps); /* input: M1param(Real/Imag)(Prev), M2(Real/Imag)(Prev) */
+      }          /* output: M1param(Real/Imag), M2(Real/Imag) */
     }
 
-    alpha = FX_DBL2FX_SGL(fDivNorm(ts - prevSlot, currSlot - prevSlot));
+    if (bypassMode == 0) {
+      alpha = FX_DBL2FX_SGL(fDivNorm(ts - prevSlot, currSlot - prevSlot));
+    }
 
     switch (mode) {
       case INPUTMODE_QMF_SBR:
@@ -1182,15 +1209,17 @@
           self->bShareDelayWithSBR = 0; /* We got no hybrid delay */
         else
           self->bShareDelayWithSBR = 1;
-        SpatialDecFeedQMF(self, qmfInDataReal, qmfInDataImag, ts_io, bypassMode,
-                          self->qmfInputReal__FDK, self->qmfInputImag__FDK,
-                          self->numInputChannels);
+        SpatialDecFeedQMF(
+            self, qmfInDataReal, qmfInDataImag, ts_io, bypassMode,
+            self->qmfInputReal__FDK, self->qmfInputImag__FDK,
+            (bypassMode) ? numInputChannels : self->numInputChannels);
         break;
       case INPUTMODE_TIME:
         self->bShareDelayWithSBR = 0;
-        SpatialDecQMFAnalysis(self, inData, ts_io, bypassMode,
-                              self->qmfInputReal__FDK, self->qmfInputImag__FDK,
-                              self->numInputChannels);
+        SpatialDecQMFAnalysis(
+            self, inData, ts_io, bypassMode, self->qmfInputReal__FDK,
+            self->qmfInputImag__FDK,
+            (bypassMode) ? numInputChannels : self->numInputChannels);
         break;
       default:
         break;
@@ -1360,7 +1389,7 @@
       }   /* !self->tempShapeConfig == 1 */
     }     /*  !bypassMode */
 
-    if (self->phaseCoding == 1) {
+    if ((self->phaseCoding == 1) && (bypassMode == 0)) {
       /* only if bsPhaseCoding == 1 and bsResidualCoding == 0 */
 
       SpatialDecApplyPhase(

fdk-aac-2.0.2.tar.gz/libSACdec/src/sac_dec_lib.cpp -> fdk-aac-2.0.3.tar.gz/libSACdec/src/sac_dec_lib.cpp Changed

@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -700,9 +700,10 @@
 SACDEC_ERROR mpegSurroundDecoder_Config(
     CMpegSurroundDecoder *pMpegSurroundDecoder, HANDLE_FDK_BITSTREAM hBs,
     AUDIO_OBJECT_TYPE coreCodec, INT samplingRate, INT frameSize,
-    INT stereoConfigIndex, INT coreSbrFrameLengthIndex, INT configBytes,
-    const UCHAR configMode, UCHAR *configChanged) {
+    INT numChannels, INT stereoConfigIndex, INT coreSbrFrameLengthIndex,
+    INT configBytes, const UCHAR configMode, UCHAR *configChanged) {
   SACDEC_ERROR err = MPS_OK;
+  INT nInputChannels;
   SPATIAL_SPECIFIC_CONFIG spatialSpecificConfig;
   SPATIAL_SPECIFIC_CONFIG *pSsc =
       &pMpegSurroundDecoder->spatialSpecificConfigBackup;
@@ -716,12 +717,18 @@
         err = SpatialDecParseMps212Config(
             hBs, &spatialSpecificConfig, samplingRate, coreCodec,
             stereoConfigIndex, coreSbrFrameLengthIndex);
+        nInputChannels = spatialSpecificConfig.nInputChannels;
         pSsc = &spatialSpecificConfig;
       } else {
         err = SpatialDecParseMps212Config(
             hBs, &pMpegSurroundDecoder->spatialSpecificConfigBackup,
             samplingRate, coreCodec, stereoConfigIndex,
             coreSbrFrameLengthIndex);
+        nInputChannels =
+            pMpegSurroundDecoder->spatialSpecificConfigBackup.nInputChannels;
+      }
+      if ((err == MPS_OK) && (numChannels != nInputChannels)) {
+        err = MPS_PARSE_ERROR;
       }
       break;
     case AOT_ER_AAC_ELD:
@@ -731,11 +738,19 @@
          * into temporarily allocated structure */
         err = SpatialDecParseSpecificConfig(hBs, &spatialSpecificConfig,
                                             configBytes, coreCodec);
+        nInputChannels = spatialSpecificConfig.nInputChannels;
         pSsc = &spatialSpecificConfig;
       } else {
         err = SpatialDecParseSpecificConfig(
             hBs, &pMpegSurroundDecoder->spatialSpecificConfigBackup,
             configBytes, coreCodec);
+        nInputChannels =
+            pMpegSurroundDecoder->spatialSpecificConfigBackup.nInputChannels;
+      }
+      /* check number of channels for channel_configuration > 0  */
+      if ((err == MPS_OK) && (numChannels > 0) &&
+          (numChannels != nInputChannels)) {
+        err = MPS_PARSE_ERROR;
       }
       break;
     default:

fdk-aac-2.0.2.tar.gz/libSACdec/src/sac_process.cpp -> fdk-aac-2.0.3.tar.gz/libSACdec/src/sac_process.cpp Changed

@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -517,12 +517,11 @@
       maxVal = fAbs(iReal0) | fAbs(iImag0);
       maxVal |= fAbs(iReal1);
 
-      s = fMax(CntLeadingZeros(maxVal) - 1, 0);
-      s = fMin(s, scale_param_m2);
+      s = fMin(CntLeadingZeros(maxVal) - 2, scale_param_m2);
 
-      mReal0 = iReal0 << s;
-      mImag0 = iImag0 << s;
-      mReal1 = iReal1 << s;
+      mReal0 = scaleValue(iReal0, s);
+      mImag0 = scaleValue(iImag0, s);
+      mReal1 = scaleValue(iReal1, s);
 
       s = scale_param_m2 - s;
 
@@ -562,12 +561,11 @@
       maxVal = fAbs(iReal0) | fAbs(iImag0);
       maxVal |= fAbs(iReal1);
 
-      s = fMax(CntLeadingZeros(maxVal) - 1, 0);
-      s = fMin(s, scale_param_m2);
+      s = fMin(CntLeadingZeros(maxVal) - 2, scale_param_m2);
 
-      mReal0 = FX_DBL2FX_SGL(iReal0 << s);
-      mImag0 = FX_DBL2FX_SGL(iImag0 << s);
-      mReal1 = FX_DBL2FX_SGL(iReal1 << s);
+      mReal0 = FX_DBL2FX_SGL(scaleValue(iReal0, s));
+      mImag0 = FX_DBL2FX_SGL(scaleValue(iImag0, s));
+      mReal1 = FX_DBL2FX_SGL(scaleValue(iReal1, s));
 
       s = scale_param_m2 - s;

fdk-aac-2.0.2.tar.gz/libSACdec/src/sac_reshapeBBEnv.cpp -> fdk-aac-2.0.3.tar.gz/libSACdec/src/sac_reshapeBBEnv.cpp Changed

@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -241,29 +241,56 @@
   }
 }
 
-static inline void slotAmp(FIXP_DBL *RESTRICT slotAmp_dry,
-                           FIXP_DBL *RESTRICT slotAmp_wet,
-                           FIXP_DBL *RESTRICT pHybOutputRealDry,
-                           FIXP_DBL *RESTRICT pHybOutputImagDry,
-                           FIXP_DBL *RESTRICT pHybOutputRealWet,
-                           FIXP_DBL *RESTRICT pHybOutputImagWet, INT cplxBands,
-                           INT hybBands) {
-  INT qs;
+static inline void slotAmp(
+    FIXP_DBL *RESTRICT slotAmp_dry, INT *RESTRICT slotAmp_dry_e,
+    FIXP_DBL *RESTRICT slotAmp_wet, INT *RESTRICT slotAmp_wet_e,
+    FIXP_DBL *RESTRICT pHybOutputRealDry, FIXP_DBL *RESTRICT pHybOutputImagDry,
+    FIXP_DBL *RESTRICT pHybOutputRealWet, FIXP_DBL *RESTRICT pHybOutputImagWet,
+    INT cplxBands, INT hybBands) {
+  INT qs, s1, s2, headroom_dry, headroom_wet;
   FIXP_DBL dry, wet;
 
+  /* headroom can be reduced by 1 bit due to use of fPow2Div2 */
+  s1 = DFRACT_BITS - 1 - CntLeadingZeros(hybBands + cplxBands);
+  headroom_dry = fMin(getScalefactor(pHybOutputRealDry, hybBands),
+                      getScalefactor(pHybOutputImagDry, cplxBands));
+  headroom_wet = fMin(getScalefactor(pHybOutputRealWet, hybBands),
+                      getScalefactor(pHybOutputImagWet, cplxBands));
+
   dry = wet = FL2FXCONST_DBL(0.0f);
   for (qs = 0; qs < cplxBands; qs++) {
-    dry = fAddSaturate(dry, fPow2Div2(pHybOutputRealDryqs << (1)) +
-                                fPow2Div2(pHybOutputImagDryqs << (1)));
-    wet = fAddSaturate(wet, fPow2Div2(pHybOutputRealWetqs << (1)) +
-                                fPow2Div2(pHybOutputImagWetqs << (1)));
+    /* sum up dry part */
+    dry += (fPow2Div2(pHybOutputRealDryqs << headroom_dry) >> s1);
+    dry += (fPow2Div2(pHybOutputImagDryqs << headroom_dry) >> s1);
+    /* sum up wet part */
+    wet += (fPow2Div2(pHybOutputRealWetqs << headroom_wet) >> s1);
+    wet += (fPow2Div2(pHybOutputImagWetqs << headroom_wet) >> s1);
   }
   for (; qs < hybBands; qs++) {
-    dry = fAddSaturate(dry, fPow2Div2(pHybOutputRealDryqs << (1)));
-    wet = fAddSaturate(wet, fPow2Div2(pHybOutputRealWetqs << (1)));
+    dry += (fPow2Div2(pHybOutputRealDryqs << headroom_dry) >> s1);
+    wet += (fPow2Div2(pHybOutputRealWetqs << headroom_wet) >> s1);
+  }
+
+  /* consider fPow2Div2() */
+  s1 += 1;
+
+  /* normalize dry part, ensure that exponent is even */
+  s2 = fixMax(0, CntLeadingZeros(dry) - 1);
+  *slotAmp_dry = dry << s2;
+  *slotAmp_dry_e = s1 - s2 - 2 * headroom_dry;
+  if (*slotAmp_dry_e & 1) {
+    *slotAmp_dry = *slotAmp_dry >> 1;
+    *slotAmp_dry_e += 1;
+  }
+
+  /* normalize wet part, ensure that exponent is even */
+  s2 = fixMax(0, CntLeadingZeros(wet) - 1);
+  *slotAmp_wet = wet << s2;
+  *slotAmp_wet_e = s1 - s2 - 2 * headroom_wet;
+  if (*slotAmp_wet_e & 1) {
+    *slotAmp_wet = *slotAmp_wet >> 1;
+    *slotAmp_wet_e += 1;
   }
-  *slotAmp_dry = dry >> (2 * (1));
-  *slotAmp_wet = wet >> (2 * (1));
 }
 
 #if defined(__aarch64__)
@@ -533,6 +560,7 @@
                             INT ts) {
   INT ch, scale;
   INT dryFacSF, slotAmpSF;
+  INT slotAmp_dry_e, slotAmp_wet_e;
   FIXP_DBL tmp, dryFac, envShape;
   FIXP_DBL slotAmp_dry, slotAmp_wet, slotAmp_ratio;
   FIXP_DBL envDryMAX_OUTPUT_CHANNELS, envDmx2;
@@ -594,22 +622,25 @@
         dryFacSF = SF_SHAPE + 2 * dryFacSF;
       }
 
+      slotAmp_dry_e = slotAmp_wet_e = 0;
+
       /* calculate slotAmp_dry and slotAmp_wet */
-      slotAmp(&slotAmp_dry, &slotAmp_wet, &self->hybOutputRealDry__FDKch6,
+      slotAmp(&slotAmp_dry, &slotAmp_dry_e, &slotAmp_wet, &slotAmp_wet_e,
+              &self->hybOutputRealDry__FDKch6,
               &self->hybOutputImagDry__FDKch6,
               &self->hybOutputRealWet__FDKch6,
               &self->hybOutputImagWet__FDKch6, cplxBands, hybBands);
 
+      /* exponents must be even due to subsequent square root calculation */
+      FDK_ASSERT(((slotAmp_dry_e & 1) == 0) && ((slotAmp_wet_e & 1) == 0));
+
       /* slotAmp_ratio will be scaled by slotAmpSF bits */
       if (slotAmp_dry != FL2FXCONST_DBL(0.0f)) {
-        sc = fixMax(0, CntLeadingZeros(slotAmp_wet) - 1);
-        sc = sc - (sc & 1);
-
-        slotAmp_wet = sqrtFixp(slotAmp_wet << sc);
+        slotAmp_wet = sqrtFixp(slotAmp_wet);
         slotAmp_dry = invSqrtNorm2(slotAmp_dry, &slotAmpSF);
 
         slotAmp_ratio = fMult(slotAmp_wet, slotAmp_dry);
-        slotAmpSF = slotAmpSF - (sc >> 1);
+        slotAmpSF = slotAmpSF + (slotAmp_wet_e >> 1) - (slotAmp_dry_e >> 1);
       }
 
       /* calculate common scale factor */

fdk-aac-2.0.2.tar.gz/libSACdec/src/sac_stp.cpp -> fdk-aac-2.0.3.tar.gz/libSACdec/src/sac_stp.cpp Changed

@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -229,15 +229,13 @@
   int n;
   FIXP_DBL scaleY;
   for (n = bands - 1; n >= 0; n--) {
-    scaleY = fMultDiv2(scaleX, *pBP);
+    scaleY = fMult(scaleX, *pBP);
     *hybOutputRealDry = SATURATE_LEFT_SHIFT(
-        (*hybOutputRealDry >> 1) +
-            (fMultDiv2(*hybOutputRealWet, scaleY) << (SF_SCALE + 1)),
-        1, DFRACT_BITS);
+        (*hybOutputRealDry >> SF_SCALE) + fMult(*hybOutputRealWet, scaleY),
+        SF_SCALE, DFRACT_BITS);
     *hybOutputImagDry = SATURATE_LEFT_SHIFT(
-        (*hybOutputImagDry >> 1) +
-            (fMultDiv2(*hybOutputImagWet, scaleY) << (SF_SCALE + 1)),
-        1, DFRACT_BITS);
+        (*hybOutputImagDry >> SF_SCALE) + fMult(*hybOutputImagWet, scaleY),
+        SF_SCALE, DFRACT_BITS);
     hybOutputRealDry++, hybOutputRealWet++;
     hybOutputImagDry++, hybOutputImagWet++;
     pBP++;
@@ -252,12 +250,12 @@
   int n;
 
   for (n = bands - 1; n >= 0; n--) {
-    *hybOutputRealDry =
-        *hybOutputRealDry +
-        (fMultDiv2(*hybOutputRealWet, scaleX) << (SF_SCALE + 1));
-    *hybOutputImagDry =
-        *hybOutputImagDry +
-        (fMultDiv2(*hybOutputImagWet, scaleX) << (SF_SCALE + 1));
+    *hybOutputRealDry = SATURATE_LEFT_SHIFT(
+        (*hybOutputRealDry >> SF_SCALE) + fMult(*hybOutputRealWet, scaleX),
+        SF_SCALE, DFRACT_BITS);
+    *hybOutputImagDry = SATURATE_LEFT_SHIFT(
+        (*hybOutputImagDry >> SF_SCALE) + fMult(*hybOutputImagWet, scaleX),
+        SF_SCALE, DFRACT_BITS);
     hybOutputRealDry++, hybOutputRealWet++;
     hybOutputImagDry++, hybOutputImagWet++;
   }
@@ -369,15 +367,15 @@
     hStpDec->update_old_ener = 1;
     for (ch = 0; ch < self->numInputChannels; ch++) {
       hStpDec->oldDryEnerLD64ch =
-          CalcLdData(hStpDec->runDryEnerch + ABS_THR__FDK);
+          CalcLdData(fAddSaturate(hStpDec->runDryEnerch, ABS_THR__FDK));
     }
     for (ch = 0; ch < self->numOutputChannels; ch++) {
       if (self->treeConfig == TREE_212)
         hStpDec->oldWetEnerLD64ch =
-            CalcLdData(hStpDec->runWetEnerch + ABS_THR__FDK);
+            CalcLdData(fAddSaturate(hStpDec->runWetEnerch, ABS_THR__FDK));
       else
         hStpDec->oldWetEnerLD64ch =
-            CalcLdData(hStpDec->runWetEnerch + ABS_THR2__FDK);
+            CalcLdData(fAddSaturate(hStpDec->runWetEnerch, ABS_THR2__FDK));
     }
   } else {
     hStpDec->update_old_ener++;

fdk-aac-2.0.2.tar.gz/libSBRdec/src/env_calc.cpp -> fdk-aac-2.0.3.tar.gz/libSBRdec/src/env_calc.cpp Changed

@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -664,7 +664,7 @@
         gain_sfi = mult_sf - total_power_low_sf + sf2;
         gaini = sqrtFixp_lookup(gaini, &gain_sfi);
         if (gain_sfi < 0) {
-          gaini >>= -gain_sfi;
+          gaini >>= fMin(DFRACT_BITS - 1, -gain_sfi);
           gain_sfi = 0;
         }
       } else {
@@ -683,11 +683,6 @@
 
     /* gaini = g_interi */
     for (i = 0; i < nbSubsample; ++i) {
-      if (gain_sfi < 0) {
-        gaini >>= -gain_sfi;
-        gain_sfi = 0;
-      }
-
       /* calculate: gaini = 1.0f + gamma * (gaini - 1.0f); */
       FIXP_DBL one = (FIXP_DBL)MAXVAL_DBL >>
                      gain_sfi; /* to substract this from gaini */
@@ -755,23 +750,15 @@
     int gain_adj_sf = gain_adj_2_sf;
 
     for (i = 0; i < nbSubsample; ++i) {
-      gaini = fMult(gaini, gain_adj);
-      gain_sfi += gain_adj_sf;
-
-      /* limit gain */
-      if (gain_sfi > INTER_TES_SF_CHANGE) {
-        gaini = (FIXP_DBL)MAXVAL_DBL;
-        gain_sfi = INTER_TES_SF_CHANGE;
-      }
-    }
-
-    for (i = 0; i < nbSubsample; ++i) {
-      /* equalize gain's scale factors */
-      gaini >>= INTER_TES_SF_CHANGE - gain_sfi;
+      int gain_e = fMax(
+          fMin(gain_sfi + gain_adj_sf - INTER_TES_SF_CHANGE, DFRACT_BITS - 1),
+          -(DFRACT_BITS - 1));
+      FIXP_DBL gain_final = fMult(gaini, gain_adj);
+      gain_final = scaleValueSaturate(gain_final, gain_e);
 
       for (j = lowSubband; j < highSubband; j++) {
-        qmfRealstartPos + ij = fMult(qmfRealstartPos + ij, gaini);
-        qmfImagstartPos + ij = fMult(qmfImagstartPos + ij, gaini);
+        qmfRealstartPos + ij = fMult(qmfRealstartPos + ij, gain_final);
+        qmfImagstartPos + ij = fMult(qmfImagstartPos + ij, gain_final);
       }
     }
   } else { /* gamma_idx == 0 */
@@ -1398,6 +1385,17 @@
     */
     noise_e = (start_pos < no_cols) ? adj_e : final_e;
 
+    if (start_pos >= no_cols) {
+      int diff = h_sbr_cal_env->filtBufferNoise_e - noise_e;
+      if (diff > 0) {
+        int s = getScalefactor(h_sbr_cal_env->filtBufferNoise, noSubbands);
+        if (diff > s) {
+          final_e += diff - s;
+          noise_e = final_e;
+        }
+      }
+    }
+
     /*
       Convert energies to amplitude levels
     */
@@ -2741,6 +2739,9 @@
             fMult(direct_ratio, noiseLevelk);
       }
 
+      smoothedNoise = fMax(fMin(smoothedNoise, (FIXP_DBL)(MAXVAL_DBL / 2)),
+                           (FIXP_DBL)(MINVAL_DBL / 2));
+
       /*
         The next 2 multiplications constitute the actual envelope adjustment
         of the signal and should be carried out with full accuracy
@@ -2930,6 +2931,9 @@
             fMult(direct_ratio, noiseLevelk);
       }
 
+      smoothedNoise = fMax(fMin(smoothedNoise, (FIXP_DBL)(MAXVAL_DBL / 2)),
+                           (FIXP_DBL)(MINVAL_DBL / 2));
+
       /*
         The next 2 multiplications constitute the actual envelope adjustment
         of the signal and should be carried out with full accuracy

fdk-aac-2.0.2.tar.gz/libSBRdec/src/hbe.cpp -> fdk-aac-2.0.3.tar.gz/libSBRdec/src/hbe.cpp Changed

@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -1400,42 +1400,27 @@
 
   if (shift_ov != 0) {
     for (i = 0; i < HBE_MAX_OUT_SLOTS; i++) {
-      for (band = 0; band < QMF_SYNTH_CHANNELS; band++) {
-        if (shift_ov >= 0) {
-          hQmfTransposer->qmfHBEBufReal_Fiband <<= shift_ov;
-          hQmfTransposer->qmfHBEBufImag_Fiband <<= shift_ov;
-        } else {
-          hQmfTransposer->qmfHBEBufReal_Fiband >>= (-shift_ov);
-          hQmfTransposer->qmfHBEBufImag_Fiband >>= (-shift_ov);
-        }
-      }
+      scaleValuesSaturate(&hQmfTransposer->qmfHBEBufReal_Fi0,
+                          QMF_SYNTH_CHANNELS, shift_ov);
+      scaleValuesSaturate(&hQmfTransposer->qmfHBEBufImag_Fi0,
+                          QMF_SYNTH_CHANNELS, shift_ov);
     }
-  }
 
-  if ((keepStatesSyncedMode == KEEP_STATES_SYNCED_OFF) && shift_ov != 0) {
-    for (i = timeStep * firstSlotOffsset; i < ov_len; i++) {
-      for (band = hQmfTransposer->startBand; band < hQmfTransposer->stopBand;
-           band++) {
-        if (shift_ov >= 0) {
-          ppQmfBufferOutReal_Fiband <<= shift_ov;
-          ppQmfBufferOutImag_Fiband <<= shift_ov;
-        } else {
-          ppQmfBufferOutReal_Fiband >>= (-shift_ov);
-          ppQmfBufferOutImag_Fiband >>= (-shift_ov);
-        }
+    if (keepStatesSyncedMode == KEEP_STATES_SYNCED_OFF) {
+      int nBands =
+          fMax(0, hQmfTransposer->stopBand - hQmfTransposer->startBand);
+
+      for (i = timeStep * firstSlotOffsset; i < ov_len; i++) {
+        scaleValuesSaturate(&ppQmfBufferOutReal_FihQmfTransposer->startBand,
+                            nBands, shift_ov);
+        scaleValuesSaturate(&ppQmfBufferOutImag_FihQmfTransposer->startBand,
+                            nBands, shift_ov);
       }
-    }
 
-    /* shift lpc filterstates */
-    for (i = 0; i < timeStep * firstSlotOffsset + LPC_ORDER; i++) {
-      for (band = 0; band < (64); band++) {
-        if (shift_ov >= 0) {
-          lpcFilterStatesRealiband <<= shift_ov;
-          lpcFilterStatesImagiband <<= shift_ov;
-        } else {
-          lpcFilterStatesRealiband >>= (-shift_ov);
-          lpcFilterStatesImagiband >>= (-shift_ov);
-        }
+      /* shift lpc filterstates */
+      for (i = 0; i < timeStep * firstSlotOffsset + LPC_ORDER; i++) {
+        scaleValuesSaturate(&lpcFilterStatesReali0, (64), shift_ov);
+        scaleValuesSaturate(&lpcFilterStatesImagi0, (64), shift_ov);
       }
     }
   }

fdk-aac-2.0.2.tar.gz/libSBRdec/src/lpp_tran.cpp -> fdk-aac-2.0.3.tar.gz/libSBRdec/src/lpp_tran.cpp Changed

@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -132,10 +132,6 @@
 
 #include "HFgen_preFlat.h"
 
-#if defined(__arm__)
-#include "arm/lpp_tran_arm.cpp"
-#endif
-
 #define LPC_SCALE_FACTOR 2
 
 /*!
@@ -220,19 +216,21 @@
                                       const FIXP_DBL *const lowBandReal,
                                       const int startSample,
                                       const int stopSample, const UCHAR hiBand,
-                                      const int dynamicScale, const int descale,
+                                      const int dynamicScale,
                                       const FIXP_SGL a0r, const FIXP_SGL a1r) {
-  FIXP_DBL accu1, accu2;
-  int i;
+  const int dynscale = fixMax(0, dynamicScale - 1) + 1;
+  const int rescale = -fixMin(0, dynamicScale - 1) + 1;
+  const int descale =
+      fixMin(DFRACT_BITS - 1, LPC_SCALE_FACTOR + dynamicScale + rescale);
+
+  for (int i = 0; i < stopSample - startSample; i++) {
+    FIXP_DBL accu;
 
-  for (i = 0; i < stopSample - startSample; i++) {
-    accu1 = fMultDiv2(a1r, lowBandReali);
-    accu1 = (fMultDiv2(a0r, lowBandReali + 1) + accu1);
-    accu1 = accu1 >> dynamicScale;
+    accu = fMultDiv2(a1r, lowBandReali) + fMultDiv2(a0r, lowBandReali + 1);
+    accu = (lowBandReali + 2 >> descale) + (accu >> dynscale);
 
-    accu1 <<= 1;
-    accu2 = (lowBandReali + 2 >> descale);
-    qmfBufferReali + startSamplehiBand = accu1 + accu2;
+    qmfBufferReali + startSamplehiBand =
+        SATURATE_LEFT_SHIFT(accu, rescale, DFRACT_BITS);
   }
 }
 
@@ -529,7 +527,7 @@
         if ((scale > 0) && (result >= (FIXP_DBL)MAXVAL_DBL >> scale)) {
           resetLPCCoeffs = 1;
         } else {
-          alphar1 = FX_DBL2FX_SGL(scaleValue(result, scale));
+          alphar1 = FX_DBL2FX_SGL(scaleValueSaturate(result, scale));
           if ((tmp < FL2FX_DBL(0.0f)) ^ (ac.det < FL2FX_DBL(0.0f))) {
             alphar1 = -alphar1;
           }
@@ -557,7 +555,7 @@
                scale)) {
             resetLPCCoeffs = 1;
           } else {
-            alphai1 = FX_DBL2FX_SGL(scaleValue(result, scale));
+            alphai1 = FX_DBL2FX_SGL(scaleValueSaturate(result, scale));
             if ((tmp < FL2FX_DBL(0.0f)) ^ (ac.det < FL2FX_DBL(0.0f))) {
               alphai1 = -alphai1;
             }
@@ -596,7 +594,7 @@
       } else {
         INT scale;
         FIXP_DBL result = fDivNorm(absTmp, fixp_abs(ac.r11r), &scale);
-        alphar0 = FX_DBL2FX_SGL(scaleValue(result, scale + 1));
+        alphar0 = FX_DBL2FX_SGL(scaleValueSaturate(result, scale + 1));
 
         if ((tmp > FL2FX_DBL(0.0f)) ^ (ac.r11r < FL2FX_DBL(0.0f)))
           alphar0 = -alphar0;
@@ -616,7 +614,7 @@
         } else {
           INT scale;
           FIXP_DBL result = fDivNorm(absTmp, fixp_abs(ac.r11r), &scale);
-          alphai0 = FX_DBL2FX_SGL(scaleValue(result, scale + 1));
+          alphai0 = FX_DBL2FX_SGL(scaleValueSaturate(result, scale + 1));
           if ((tmp > FL2FX_DBL(0.0f)) ^ (ac.r11r < FL2FX_DBL(0.0f)))
             alphai0 = -alphai0;
         }
@@ -659,7 +657,7 @@
           INT scale;
           FIXP_DBL result =
               fDivNorm(fixp_abs(ac.r01r), fixp_abs(ac.r11r), &scale);
-          k1 = scaleValue(result, scale);
+          k1 = scaleValueSaturate(result, scale);
 
           if (!((ac.r01r < FL2FX_DBL(0.0f)) ^ (ac.r11r < FL2FX_DBL(0.0f)))) {
             k1 = -k1;
@@ -771,52 +769,50 @@
       } else { /* bw <= 0 */
 
         if (!useLP) {
-          int descale =
-              fixMin(DFRACT_BITS - 1, (LPC_SCALE_FACTOR + dynamicScale));
-#ifdef FUNCTION_LPPTRANSPOSER_func1
-          lppTransposer_func1(
-              lowBandReal + LPC_ORDER + startSample,
-              lowBandImag + LPC_ORDER + startSample,
-              qmfBufferReal + startSample, qmfBufferImag + startSample,
-              stopSample - startSample, (int)hiBand, dynamicScale, descale, a0r,
-              a0i, a1r, a1i, fPreWhitening, preWhiteningGainsloBand,
-              preWhiteningGains_exploBand + 1);
-#else
+          const int dynscale = fixMax(0, dynamicScale - 2) + 1;
+          const int rescale = -fixMin(0, dynamicScale - 2) + 1;
+          const int descale = fixMin(DFRACT_BITS - 1,
+                                     LPC_SCALE_FACTOR + dynamicScale + rescale);
+
           for (i = startSample; i < stopSample; i++) {
             FIXP_DBL accu1, accu2;
 
-            accu1 = (fMultDiv2(a0r, lowBandRealLPC_ORDER + i - 1) -
-                     fMultDiv2(a0i, lowBandImagLPC_ORDER + i - 1) +
-                     fMultDiv2(a1r, lowBandRealLPC_ORDER + i - 2) -
-                     fMultDiv2(a1i, lowBandImagLPC_ORDER + i - 2)) >>
-                    dynamicScale;
-            accu2 = (fMultDiv2(a0i, lowBandRealLPC_ORDER + i - 1) +
-                     fMultDiv2(a0r, lowBandImagLPC_ORDER + i - 1) +
-                     fMultDiv2(a1i, lowBandRealLPC_ORDER + i - 2) +
-                     fMultDiv2(a1r, lowBandImagLPC_ORDER + i - 2)) >>
-                    dynamicScale;
-
-            accu1 = (lowBandRealLPC_ORDER + i >> descale) + (accu1 << 1);
-            accu2 = (lowBandImagLPC_ORDER + i >> descale) + (accu2 << 1);
+            accu1 = ((fMultDiv2(a0r, lowBandRealLPC_ORDER + i - 1) -
+                      fMultDiv2(a0i, lowBandImagLPC_ORDER + i - 1)) >>
+                     1) +
+                    ((fMultDiv2(a1r, lowBandRealLPC_ORDER + i - 2) -
+                      fMultDiv2(a1i, lowBandImagLPC_ORDER + i - 2)) >>
+                     1);
+            accu2 = ((fMultDiv2(a0i, lowBandRealLPC_ORDER + i - 1) +
+                      fMultDiv2(a0r, lowBandImagLPC_ORDER + i - 1)) >>
+                     1) +
+                    ((fMultDiv2(a1i, lowBandRealLPC_ORDER + i - 2) +
+                      fMultDiv2(a1r, lowBandImagLPC_ORDER + i - 2)) >>
+                     1);
+
+            accu1 =
+                (lowBandRealLPC_ORDER + i >> descale) + (accu1 >> dynscale);
+            accu2 =
+                (lowBandImagLPC_ORDER + i >> descale) + (accu2 >> dynscale);
             if (fPreWhitening) {
-              accu1 = scaleValueSaturate(
+              qmfBufferRealihiBand = scaleValueSaturate(
                   fMultDiv2(accu1, preWhiteningGainsloBand),
-                  preWhiteningGains_exploBand + 1);
-              accu2 = scaleValueSaturate(
+                  preWhiteningGains_exploBand + 1 + rescale);
+              qmfBufferImagihiBand = scaleValueSaturate(
                   fMultDiv2(accu2, preWhiteningGainsloBand),
-                  preWhiteningGains_exploBand + 1);
+                  preWhiteningGains_exploBand + 1 + rescale);
+            } else {
+              qmfBufferRealihiBand =
+                  SATURATE_LEFT_SHIFT(accu1, rescale, DFRACT_BITS);
+              qmfBufferImagihiBand =
+                  SATURATE_LEFT_SHIFT(accu2, rescale, DFRACT_BITS);
             }
-            qmfBufferRealihiBand = accu1;
-            qmfBufferImagihiBand = accu2;
           }
-#endif
         } else {
           FDK_ASSERT(dynamicScale >= 0);
           calc_qmfBufferReal(
               qmfBufferReal, &(lowBandRealLPC_ORDER + startSample - 2),
-              startSample, stopSample, hiBand, dynamicScale,
-              fMin(DFRACT_BITS - 1, (LPC_SCALE_FACTOR + dynamicScale)), a0r,
-              a1r);
+              startSample, stopSample, hiBand, dynamicScale, a0r, a1r);
         }
       } /* bw <= 0 */
 
@@ -1066,7 +1062,7 @@
         if ((scale > 0) && (result >= (FIXP_DBL)MAXVAL_DBL >> scale)) {
           resetLPCCoeffs = 1;
         } else {
-          alphar1 = FX_DBL2FX_SGL(scaleValue(result, scale));
+          alphar1 = FX_DBL2FX_SGL(scaleValueSaturate(result, scale));
           if ((tmp < FL2FX_DBL(0.0f)) ^ (ac.det < FL2FX_DBL(0.0f))) {
             alphar1 = -alphar1;
           }
@@ -1092,7 +1088,7 @@
             (result >= /*FL2FXCONST_DBL(1.f)*/ (FIXP_DBL)MAXVAL_DBL >> scale)) {
           resetLPCCoeffs = 1;
         } else {
-          alphai1 = FX_DBL2FX_SGL(scaleValue(result, scale));
+          alphai1 = FX_DBL2FX_SGL(scaleValueSaturate(result, scale));
           if ((tmp < FL2FX_DBL(0.0f)) ^ (ac.det < FL2FX_DBL(0.0f))) {
             alphai1 = -alphai1;
           }
@@ -1121,7 +1117,7 @@
       } else {
         INT scale;
         FIXP_DBL result = fDivNorm(absTmp, fixp_abs(ac.r11r), &scale);
-        alphar0 = FX_DBL2FX_SGL(scaleValue(result, scale + 1));
+        alphar0 = FX_DBL2FX_SGL(scaleValueSaturate(result, scale + 1));
 
         if ((tmp > FL2FX_DBL(0.0f)) ^ (ac.r11r < FL2FX_DBL(0.0f)))
           alphar0 = -alphar0;
@@ -1140,7 +1136,7 @@
       } else {
         INT scale;
         FIXP_DBL result = fDivNorm(absTmp, fixp_abs(ac.r11r), &scale);
-        alphai0 = FX_DBL2FX_SGL(scaleValue(result, scale + 1));
+        alphai0 = FX_DBL2FX_SGL(scaleValueSaturate(result, scale + 1));
         if ((tmp > FL2FX_DBL(0.0f)) ^ (ac.r11r < FL2FX_DBL(0.0f))) {
           alphai0 = -alphai0;
         }

fdk-aac-2.0.2.tar.gz/libSBRdec/src/lpp_tran.h -> fdk-aac-2.0.3.tar.gz/libSBRdec/src/lpp_tran.h Changed

fdk-aac-2.0.2.tar.gz/libSBRdec/src/sbr_dec.cpp -> fdk-aac-2.0.3.tar.gz/libSBRdec/src/sbr_dec.cpp Changed

@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -713,7 +713,8 @@
 
   } else { /* (flags & SBRDEC_PS_DECODED) */
     INT sdiff;
-    INT scaleFactorHighBand, scaleFactorLowBand_ov, scaleFactorLowBand_no_ov;
+    INT scaleFactorHighBand, scaleFactorLowBand_ov, scaleFactorLowBand_no_ov,
+        outScalefactor, outScalefactorR, outScalefactorL;
 
     HANDLE_QMF_FILTER_BANK synQmf = &hSbrDec->qmfDomainOutCh->fb;
     HANDLE_QMF_FILTER_BANK synQmfRight = &hSbrDecRight->qmfDomainOutCh->fb;
@@ -744,7 +745,7 @@
       */
       FDK_ASSERT(hSbrDec->qmfDomainInCh->pGlobalConf->nBandsSynthesis <=
                  QMF_MAX_SYNTHESIS_BANDS);
-      qmfChangeOutScalefactor(synQmfRight, -(8));
+      synQmfRight->outScalefactor = synQmf->outScalefactor;
       FDKmemcpy(synQmfRight->FilterStates, synQmf->FilterStates,
                 9 * hSbrDec->qmfDomainInCh->pGlobalConf->nBandsSynthesis *
                     sizeof(FIXP_QSS));
@@ -788,9 +789,11 @@
       FDKmemcpy(&hSbrDecRight->sbrDrcChannel, &hSbrDec->sbrDrcChannel,
                 sizeof(SBRDEC_DRC_CHANNEL));
 
-      for (i = 0; i < synQmf->no_col; i++) { /* ----- no_col loop ----- */
+      outScalefactor = maxShift - (8);
+      outScalefactorL = outScalefactorR =
+          sbrInDataHeadroom + 1; /* +1: psDiffScale! (MPEG-PS) */
 
-        INT outScalefactorR, outScalefactorL;
+      for (i = 0; i < synQmf->no_col; i++) { /* ----- no_col loop ----- */
 
         /* qmf timeslot of right channel */
         FIXP_DBL *rQmfReal = pWorkBuffer;
@@ -815,27 +818,20 @@
                   ? scaleFactorLowBand_ov
                   : scaleFactorLowBand_no_ov,
               scaleFactorHighBand, synQmf->lsb, synQmf->usb);
-
-          outScalefactorL = outScalefactorR =
-              1 + sbrInDataHeadroom; /* psDiffScale! (MPEG-PS) */
         }
 
         sbrDecoder_drcApplySlot(/* right channel */
                                 &hSbrDecRight->sbrDrcChannel, rQmfReal,
                                 rQmfImag, i, synQmfRight->no_col, maxShift);
 
-        outScalefactorR += maxShift;
-
         sbrDecoder_drcApplySlot(/* left channel */
                                 &hSbrDec->sbrDrcChannel, *(pLowBandReal + i),
                                 *(pLowBandImag + i), i, synQmf->no_col,
                                 maxShift);
 
-        outScalefactorL += maxShift;
-
         if (!(flags & SBRDEC_SKIP_QMF_SYN)) {
-          qmfChangeOutScalefactor(synQmf, -(8));
-          qmfChangeOutScalefactor(synQmfRight, -(8));
+          qmfChangeOutScalefactor(synQmf, outScalefactor);
+          qmfChangeOutScalefactor(synQmfRight, outScalefactor);
 
           qmfSynthesisFilteringSlot(
               synQmfRight, rQmfReal, /* QMF real buffer */

fdk-aac-2.0.2.tar.gz/libSBRdec/src/sbrdec_drc.cpp -> fdk-aac-2.0.3.tar.gz/libSBRdec/src/sbrdec_drc.cpp Changed

@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -233,14 +233,19 @@
     if (hDrcData->winSequenceCurr != 2) { /* long window */
       int j = col + (numQmfSubSamples >> 1);
 
-      if (hDrcData->drcInterpolationSchemeCurr == 0) {
-        INT k = (frameLenFlag) ? 0x4444445 : 0x4000000;
+      if (j < winBorderToColMap15) {
+        if (hDrcData->drcInterpolationSchemeCurr == 0) {
+          INT k = (frameLenFlag) ? 0x4444445 : 0x4000000;
 
-        alphaValue = (FIXP_DBL)(j * k);
-      } else {
-        if (j >= (int)winBorderToColMaphDrcData->drcInterpolationSchemeCurr) {
-          alphaValue = (FIXP_DBL)MAXVAL_DBL;
+          alphaValue = (FIXP_DBL)(j * k);
+        } else {
+          if (j >=
+              (int)winBorderToColMaphDrcData->drcInterpolationSchemeCurr) {
+            alphaValue = (FIXP_DBL)MAXVAL_DBL;
+          }
         }
+      } else {
+        alphaValue = (FIXP_DBL)MAXVAL_DBL;
       }
     } else { /* short windows */
       shortDrc = 1;
@@ -254,14 +259,19 @@
     if (hDrcData->winSequenceNext != 2) { /* next: long window */
       int j = col - (numQmfSubSamples >> 1);
 
-      if (hDrcData->drcInterpolationSchemeNext == 0) {
-        INT k = (frameLenFlag) ? 0x4444445 : 0x4000000;
+      if (j < winBorderToColMap15) {
+        if (hDrcData->drcInterpolationSchemeNext == 0) {
+          INT k = (frameLenFlag) ? 0x4444445 : 0x4000000;
 
-        alphaValue = (FIXP_DBL)(j * k);
-      } else {
-        if (j >= (int)winBorderToColMaphDrcData->drcInterpolationSchemeNext) {
-          alphaValue = (FIXP_DBL)MAXVAL_DBL;
+          alphaValue = (FIXP_DBL)(j * k);
+        } else {
+          if (j >=
+              (int)winBorderToColMaphDrcData->drcInterpolationSchemeNext) {
+            alphaValue = (FIXP_DBL)MAXVAL_DBL;
+          }
         }
+      } else {
+        alphaValue = (FIXP_DBL)MAXVAL_DBL;
       }
 
       fact_mag = hDrcData->nextFact_mag;
@@ -289,14 +299,19 @@
     if (hDrcData->winSequenceNext != 2) { /* long window */
       int j = col - (numQmfSubSamples >> 1);
 
-      if (hDrcData->drcInterpolationSchemeNext == 0) {
-        INT k = (frameLenFlag) ? 0x4444445 : 0x4000000;
+      if (j < winBorderToColMap15) {
+        if (hDrcData->drcInterpolationSchemeNext == 0) {
+          INT k = (frameLenFlag) ? 0x4444445 : 0x4000000;
 
-        alphaValue = (FIXP_DBL)(j * k);
-      } else {
-        if (j >= (int)winBorderToColMaphDrcData->drcInterpolationSchemeNext) {
-          alphaValue = (FIXP_DBL)MAXVAL_DBL;
+          alphaValue = (FIXP_DBL)(j * k);
+        } else {
+          if (j >=
+              (int)winBorderToColMaphDrcData->drcInterpolationSchemeNext) {
+            alphaValue = (FIXP_DBL)MAXVAL_DBL;
+          }
         }
+      } else {
+        alphaValue = (FIXP_DBL)MAXVAL_DBL;
       }
     } else { /* short windows */
       shortDrc = 1;

fdk-aac-2.0.2.tar.gz/libSBRdec/src/sbrdec_freq_sca.cpp -> fdk-aac-2.0.3.tar.gz/libSBRdec/src/sbrdec_freq_sca.cpp Changed

@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -765,9 +765,6 @@
   sbrdecUpdateLoRes(hFreq->freqBandTable0, &nBandsLo, hFreq->freqBandTable1,
                     nBandsHi);
 
-  hFreq->nSfb0 = nBandsLo;
-  hFreq->nSfb1 = nBandsHi;
-
   /* Check index to freqBandTable0 */
   if (!(nBandsLo > 0) ||
       (nBandsLo > (((hHeaderData->numberOfAnalysisBands == 16)
@@ -777,6 +774,9 @@
     return SBRDEC_UNSUPPORTED_CONFIG;
   }
 
+  hFreq->nSfb0 = nBandsLo;
+  hFreq->nSfb1 = nBandsHi;
+
   lsb = hFreq->freqBandTable00;
   usb = hFreq->freqBandTable0nBandsLo;
 
@@ -814,15 +814,15 @@
 
     if (intTemp == 0) intTemp = 1;
 
+    if (intTemp > MAX_NOISE_COEFFS) {
+      return SBRDEC_UNSUPPORTED_CONFIG;
+    }
+
     hFreq->nNfb = intTemp;
   }
 
   hFreq->nInvfBands = hFreq->nNfb;
 
-  if (hFreq->nNfb > MAX_NOISE_COEFFS) {
-    return SBRDEC_UNSUPPORTED_CONFIG;
-  }
-
   /* Get noise bands */
   sbrdecDownSampleLoRes(hFreq->freqBandTableNoise, hFreq->nNfb,
                         hFreq->freqBandTable0, nBandsLo);

fdk-aac-2.0.2.tar.gz/libSBRdec/src/sbrdecoder.cpp -> fdk-aac-2.0.3.tar.gz/libSBRdec/src/sbrdecoder.cpp Changed

@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2021 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -961,8 +961,10 @@
 
           /* Set sync state UPSAMPLING for the corresponding slot.
              This switches off bitstream parsing until a new header arrives. */
-          hSbrHeader->syncState = UPSAMPLING;
-          hSbrHeader->status |= SBRDEC_HDR_STAT_UPDATE;
+          if (hSbrHeader->syncState != SBR_NOT_INITIALIZED) {
+            hSbrHeader->syncState = UPSAMPLING;
+            hSbrHeader->status |= SBRDEC_HDR_STAT_UPDATE;
+          }
         }
       }
     } break;
@@ -1371,7 +1373,9 @@
       }
       if (headerStatus == HEADER_ERROR) {
         /* Corrupt SBR info data, do not decode and switch to UPSAMPLING */
-        hSbrHeader->syncState = UPSAMPLING;
+        hSbrHeader->syncState = hSbrHeader->syncState > UPSAMPLING
+                                    ? UPSAMPLING
+                                    : hSbrHeader->syncState;
         fDoDecodeSbrData = 0;
         sbrHeaderPresent = 0;
       }
@@ -1610,7 +1614,9 @@
       /* No valid SBR payload available, hence switch to upsampling (in all
        * headers) */
       for (hdrIdx = 0; hdrIdx < ((1) + 1); hdrIdx += 1) {
-        self->sbrHeaderelementIndexhdrIdx.syncState = UPSAMPLING;
+        if (self->sbrHeaderelementIndexhdrIdx.syncState > UPSAMPLING) {
+          self->sbrHeaderelementIndexhdrIdx.syncState = UPSAMPLING;
+        }
       }
     } else {
       /* Move frame pointer to the next slot which is up to be decoded/applied

fdk-aac-2.0.2.tar.gz/libSBRenc/src/env_est.cpp -> fdk-aac-2.0.3.tar.gz/libSBRenc/src/env_est.cpp Changed

@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -637,8 +637,8 @@
     for (; l < stop_pos; l++) {
       nrg2 += YBufferl >> YBufferSzShiftk >> sc1;
     }
-    accu1 += (nrg1 >> dynScale1);
-    accu2 += (nrg2 >> dynScale2);
+    accu1 = fAddSaturate(accu1, (nrg1 >> dynScale1));
+    accu2 = fAddSaturate(accu2, (nrg2 >> dynScale2));
   }
   /* This shift factor is always positive. See comment above. */
   nrgSum +=
@@ -1267,6 +1267,7 @@
     sbrExtrEnv->pre_transient_info1 = ed->transient_info1; /* tran_flag */
     hEnvChan->encEnvData.noOfEnvelopes = ed->nEnvelopes =
         ed->frame_info->nEnvelopes; /* number of envelopes of current frame */
+    hEnvChan->encEnvData.currentAmpResFF = (AMP_RES)h_con->initAmpResFF;
 
     /*
       Check if the current frame is divided into one envelope only. If so, set
@@ -1274,8 +1275,9 @@
     */
     if ((hEnvChan->encEnvData.hSbrBSGrid->frameClass == FIXFIX) &&
         (ed->nEnvelopes == 1)) {
+      AMP_RES currentAmpResFF = SBR_AMP_RES_1_5;
       if (h_con->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) {
-        /* Note: global_tonaliy_float_value ==
+        /* Note: global_tonality_float_value ==
            ((float)hEnvChan->encEnvData.global_tonality/((INT64)(1)<<(31-(19+2)))/0.524288*(2.0/3.0)));
                  threshold_float_value ==
            ((float)h_con->thresholdAmpResFF_m/((INT64)(1)<<(31-(h_con->thresholdAmpResFF_e)))/0.524288*(2.0/3.0)));
@@ -1289,14 +1291,13 @@
         } else {
           hEnvChan->encEnvData.currentAmpResFF = SBR_AMP_RES_3_0;
         }
-      } else
-        hEnvChan->encEnvData.currentAmpResFF = SBR_AMP_RES_1_5;
+        currentAmpResFF = hEnvChan->encEnvData.currentAmpResFF;
+      }
 
-      if (hEnvChan->encEnvData.currentAmpResFF !=
-          hEnvChan->encEnvData.init_sbr_amp_res) {
+      if (currentAmpResFF != hEnvChan->encEnvData.init_sbr_amp_res) {
         FDKsbrEnc_InitSbrHuffmanTables(
             &hEnvChan->encEnvData, &hEnvChan->sbrCodeEnvelope,
-            &hEnvChan->sbrCodeNoiseFloor, hEnvChan->encEnvData.currentAmpResFF);
+            &hEnvChan->sbrCodeNoiseFloor, currentAmpResFF);
       }
     } else {
       if (sbrHeaderData->sbr_amp_res != hEnvChan->encEnvData.init_sbr_amp_res) {
@@ -1355,6 +1356,13 @@
     }
   }
 
+  if (h_con->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY &&
+      stereoMode == SBR_SWITCH_LRC &&
+      h_envChan0->encEnvData.currentAmpResFF !=
+          h_envChan1->encEnvData.currentAmpResFF) {
+    stereoMode = SBR_LEFT_RIGHT;
+  }
+
   /*
     Extract envelope of current frame.
   */

fdk-aac-2.0.2.tar.gz/libSBRenc/src/sbr_encoder.cpp -> fdk-aac-2.0.3.tar.gz/libSBRenc/src/sbr_encoder.cpp Changed

@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -1450,8 +1450,6 @@
                                     params->deltaTAcrossFrames, 0, 0))
     return (1);
 
-  sbrConfigData->initAmpResFF = params->init_amp_res_FF;
-
   if (FDKsbrEnc_InitSbrHuffmanTables(&hEnv->encEnvData, &hEnv->sbrCodeEnvelope,
                                      &hEnv->sbrCodeNoiseFloor,
                                      sbrHeaderData->sbr_amp_res))
@@ -1749,6 +1747,7 @@
     hSbrElement->sbrHeaderData.sbr_data_extra = 1;
 
   hSbrElement->sbrHeaderData.sbr_amp_res = (AMP_RES)params->amp_res;
+  hSbrElement->sbrConfigData.initAmpResFF = params->init_amp_res_FF;
 
   /* header_extra_1 */
   hSbrElement->sbrHeaderData.freqScale = params->freqScale;

fdk-aac-2.0.3.tar.gz/sha1.c Added

@@ -0,0 +1,399 @@
+/*!
+ * \copy
+ *     Copyright (c)  1998, 2009 Paul E. Jones <paulej@packetizer.com>
+ *     All rights reserved.
+ *
+ *     Redistribution and use in source and binary forms, with or without
+ *     modification, are permitted provided that the following conditions
+ *     are met:
+ *
+ *        * Redistributions of source code must retain the above copyright
+ *          notice, this list of conditions and the following disclaimer.
+ *
+ *        * Redistributions in binary form must reproduce the above copyright
+ *          notice, this list of conditions and the following disclaimer in
+ *          the documentation and/or other materials provided with the
+ *          distribution.
+ *
+ *     THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ *     "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ *     LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS
+ *     FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE
+ *     COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT,
+ *     INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING,
+ *     BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ *     LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
+ *     CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
+ *     LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN
+ *     ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+ *     POSSIBILITY OF SUCH DAMAGE.
+ *
+ */
+
+/*
+ *  Description:
+ *      This file implements the Secure Hashing Standard as defined
+ *      in FIPS PUB 180-1 published April 17, 1995.
+ *
+ *      The Secure Hashing Standard, which uses the Secure Hashing
+ *      Algorithm (SHA), produces a 160-bit message digest for a
+ *      given data stream.  In theory, it is highly improbable that
+ *      two messages will produce the same message digest.  Therefore,
+ *      this algorithm can serve as a means of providing a "fingerprint"
+ *      for a message.
+ *
+ *  Portability Issues:
+ *      SHA-1 is defined in terms of 32-bit "words".  This code was
+ *      written with the expectation that the processor has at least
+ *      a 32-bit machine word size.  If the machine word size is larger,
+ *      the code should still function properly.  One caveat to that
+ *      is that the input functions taking characters and character
+ *      arrays assume that only 8 bits of information are stored in each
+ *      character.
+ *
+ *  Caveats:
+ *      SHA-1 is designed to work with messages less than 2^64 bits
+ *      long. Although SHA-1 allows a message digest to be generated for
+ *      messages of any number of bits less than 2^64, this
+ *      implementation only works with messages with a length that is a
+ *      multiple of the size of an 8-bit character.
+ *
+ */
+
+#include "sha1.h"
+
+/*
+ *  Define the circular shift macro
+ */
+#define SHA1CircularShift(bits,word) \
+                ((((word) << (bits)) & 0xFFFFFFFF) | \
+                ((word) >> (32-(bits))))
+
+/* Function prototypes */
+void SHA1ProcessMessageBlock(SHA1Context *);
+void SHA1PadMessage(SHA1Context *);
+
+/*
+ *  SHA1Reset
+ *
+ *  Description:
+ *      This function will initialize the SHA1Context in preparation
+ *      for computing a new message digest.
+ *
+ *  Parameters:
+ *      context: in/out
+ *          The context to reset.
+ *
+ *  Returns:
+ *      Nothing.
+ *
+ *  Comments:
+ *
+ */
+void SHA1Reset(SHA1Context *context)
+{
+    context->Length_Low             = 0;
+    context->Length_High            = 0;
+    context->Message_Block_Index    = 0;
+
+    context->Message_Digest0      = 0x67452301;
+    context->Message_Digest1      = 0xEFCDAB89;
+    context->Message_Digest2      = 0x98BADCFE;
+    context->Message_Digest3      = 0x10325476;
+    context->Message_Digest4      = 0xC3D2E1F0;
+
+    context->Computed   = 0;
+    context->Corrupted  = 0;
+}
+
+/*
+ *  SHA1Result
+ *
+ *  Description:
+ *      This function will return the 160-bit message digest into the
+ *      digest array provided as a parameter.
+ *
+ *  Parameters:
+ *      context: in/out
+ *          The context to use to calculate the SHA-1 hash.
+ *      digest: out
+ *          An array of characters where the digest is written.
+ *
+ *  Returns:
+ *      1 if successful, 0 if it failed.
+ *
+ *  Comments:
+ *
+ */
+int SHA1Result(SHA1Context *context, unsigned char *digest)
+{
+    int i;
+
+    if (context->Corrupted)
+    {
+        return 0;
+    }
+
+    if (!context->Computed)
+    {
+        SHA1PadMessage(context);
+        context->Computed = 1;
+    }
+
+    for (i = 0; i < SHA_DIGEST_LENGTH; i++)
+        digesti = context->Message_Digesti / 4 >> (8 * (3 - (i % 4)));
+
+    return 1;
+}
+
+/*
+ *  SHA1Input
+ *
+ *  Description:
+ *      This function accepts an array of octets as the next portion of
+ *      the message.
+ *
+ *  Parameters:
+ *      context: in/out
+ *          The SHA-1 context to update
+ *      message_array: in
+ *          An array of characters representing the next portion of the
+ *          message.
+ *      length: in
+ *          The length of the message in message_array
+ *
+ *  Returns:
+ *      Nothing.
+ *
+ *  Comments:
+ *
+ */
+void SHA1Input(     SHA1Context         *context,
+                    const unsigned char *message_array,
+                    unsigned            length)
+{
+    if (!length)
+    {
+        return;
+    }
+
+    if (context->Computed || context->Corrupted)
+    {
+        context->Corrupted = 1;
+        return;
+    }
+
+    while(length-- && !context->Corrupted)
+    {
+        context->Message_Blockcontext->Message_Block_Index++ =
+                                                (*message_array & 0xFF);
+
+        context->Length_Low += 8;
+        /* Force it to 32 bits */
+        context->Length_Low &= 0xFFFFFFFF;
+        if (context->Length_Low == 0)
+        {
+            context->Length_High++;
+            /* Force it to 32 bits */
+            context->Length_High &= 0xFFFFFFFF;
+            if (context->Length_High == 0)
+            {
+                /* Message is too long */
+                context->Corrupted = 1;
+            }
+        }
+
+        if (context->Message_Block_Index == 64)
+        {
+            SHA1ProcessMessageBlock(context);
+        }
+
+        message_array++;
+    }
+}
+
+/*
+ *  SHA1ProcessMessageBlock
+ *
+ *  Description:
+ *      This function will process the next 512 bits of the message
+ *      stored in the Message_Block array.
+ *
+ *  Parameters:
+ *      None.
+ *
+ *  Returns:
+ *      Nothing.
+ *
+ *  Comments:
+ *      Many of the variable names in the SHAContext, especially the
+ *      single character names, were used because those were the names
+ *      used in the publication.
+ *
+ *
+ */
+void SHA1ProcessMessageBlock(SHA1Context *context)
+{
+    const unsigned K =            /* Constants defined in SHA-1   */
+    {
+        0x5A827999,
+        0x6ED9EBA1,
+        0x8F1BBCDC,
+        0xCA62C1D6
+    };
+    int         t;                  /* Loop counter                 */
+    unsigned    temp;               /* Temporary word value         */
+    unsigned    W80;              /* Word sequence                */
+    unsigned    A, B, C, D, E;      /* Word buffers                 */
+
+    /*
+     *  Initialize the first 16 words in the array W
+     */
+    for(t = 0; t < 16; t++)
+    {
+        Wt = ((unsigned) context->Message_Blockt * 4) << 24;
+        Wt |= ((unsigned) context->Message_Blockt * 4 + 1) << 16;
+        Wt |= ((unsigned) context->Message_Blockt * 4 + 2) << 8;
+        Wt |= ((unsigned) context->Message_Blockt * 4 + 3);
+    }
+
+    for(t = 16; t < 80; t++)
+    {
+       Wt = SHA1CircularShift(1,Wt-3 ^ Wt-8 ^ Wt-14 ^ Wt-16);
+    }
+
+    A = context->Message_Digest0;
+    B = context->Message_Digest1;
+    C = context->Message_Digest2;
+    D = context->Message_Digest3;
+    E = context->Message_Digest4;
+
+    for(t = 0; t < 20; t++)
+    {
+        temp =  SHA1CircularShift(5,A) +
+                ((B & C) | ((~B) & D)) + E + Wt + K0;
+        temp &= 0xFFFFFFFF;
+        E = D;
+        D = C;
+        C = SHA1CircularShift(30,B);
+        B = A;
+        A = temp;
+    }
+
+    for(t = 20; t < 40; t++)
+    {
+        temp = SHA1CircularShift(5,A) + (B ^ C ^ D) + E + Wt + K1;
+        temp &= 0xFFFFFFFF;
+        E = D;
+        D = C;
+        C = SHA1CircularShift(30,B);
+        B = A;
+        A = temp;
+    }
+
+    for(t = 40; t < 60; t++)
+    {
+        temp = SHA1CircularShift(5,A) +
+               ((B & C) | (B & D) | (C & D)) + E + Wt + K2;
+        temp &= 0xFFFFFFFF;
+        E = D;
+        D = C;
+        C = SHA1CircularShift(30,B);
+        B = A;
+        A = temp;
+    }
+
+    for(t = 60; t < 80; t++)
+    {
+        temp = SHA1CircularShift(5,A) + (B ^ C ^ D) + E + Wt + K3;
+        temp &= 0xFFFFFFFF;
+        E = D;
+        D = C;
+        C = SHA1CircularShift(30,B);
+        B = A;
+        A = temp;
+    }
+
+    context->Message_Digest0 =
+                        (context->Message_Digest0 + A) & 0xFFFFFFFF;
+    context->Message_Digest1 =
+                        (context->Message_Digest1 + B) & 0xFFFFFFFF;
+    context->Message_Digest2 =
+                        (context->Message_Digest2 + C) & 0xFFFFFFFF;
+    context->Message_Digest3 =
+                        (context->Message_Digest3 + D) & 0xFFFFFFFF;
+    context->Message_Digest4 =
+                        (context->Message_Digest4 + E) & 0xFFFFFFFF;
+
+    context->Message_Block_Index = 0;
+}
+
+/*
+ *  SHA1PadMessage
+ *
+ *  Description:
+ *      According to the standard, the message must be padded to an even
+ *      512 bits.  The first padding bit must be a '1'.  The last 64
+ *      bits represent the length of the original message.  All bits in
+ *      between should be 0.  This function will pad the message
+ *      according to those rules by filling the Message_Block array
+ *      accordingly.  It will also call SHA1ProcessMessageBlock()
+ *      appropriately.  When it returns, it can be assumed that the
+ *      message digest has been computed.
+ *
+ *  Parameters:
+ *      context: in/out
+ *          The context to pad
+ *
+ *  Returns:
+ *      Nothing.
+ *
+ *  Comments:
+ *
+ */
+void SHA1PadMessage(SHA1Context *context)
+{
+    /*
+     *  Check to see if the current message block is too small to hold
+     *  the initial padding bits and length.  If so, we will pad the
+     *  block, process it, and then continue padding into a second
+     *  block.
+     */
+    if (context->Message_Block_Index > 55)
+    {
+        context->Message_Blockcontext->Message_Block_Index++ = 0x80;
+        while(context->Message_Block_Index < 64)
+        {
+            context->Message_Blockcontext->Message_Block_Index++ = 0;
+        }
+
+        SHA1ProcessMessageBlock(context);
+
+        while(context->Message_Block_Index < 56)
+        {
+            context->Message_Blockcontext->Message_Block_Index++ = 0;
+        }
+    }
+    else
+    {
+        context->Message_Blockcontext->Message_Block_Index++ = 0x80;
+        while(context->Message_Block_Index < 56)
+        {
+            context->Message_Blockcontext->Message_Block_Index++ = 0;
+        }
+    }
+
+    /*
+     *  Store the message length as the last 8 octets
+     */
+    context->Message_Block56 = (context->Length_High >> 24) & 0xFF;
+    context->Message_Block57 = (context->Length_High >> 16) & 0xFF;
+    context->Message_Block58 = (context->Length_High >> 8) & 0xFF;
+    context->Message_Block59 = (context->Length_High) & 0xFF;
+    context->Message_Block60 = (context->Length_Low >> 24) & 0xFF;
+    context->Message_Block61 = (context->Length_Low >> 16) & 0xFF;
+    context->Message_Block62 = (context->Length_Low >> 8) & 0xFF;
+    context->Message_Block63 = (context->Length_Low) & 0xFF;
+
+    SHA1ProcessMessageBlock(context);
+}

fdk-aac-2.0.3.tar.gz/sha1.h Added

@@ -0,0 +1,85 @@
+/*!
+ * \copy
+ *     Copyright (c)  1998, 2009 Paul E. Jones <paulej@packetizer.com>
+ *     All rights reserved.
+ *
+ *     Redistribution and use in source and binary forms, with or without
+ *     modification, are permitted provided that the following conditions
+ *     are met:
+ *
+ *        * Redistributions of source code must retain the above copyright
+ *          notice, this list of conditions and the following disclaimer.
+ *
+ *        * Redistributions in binary form must reproduce the above copyright
+ *          notice, this list of conditions and the following disclaimer in
+ *          the documentation and/or other materials provided with the
+ *          distribution.
+ *
+ *     THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ *     "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ *     LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS
+ *     FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE
+ *     COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT,
+ *     INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING,
+ *     BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ *     LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
+ *     CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
+ *     LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN
+ *     ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+ *     POSSIBILITY OF SUCH DAMAGE.
+ *
+ */
+
+/*
+ *  Description:
+ *      This class implements the Secure Hashing Standard as defined
+ *      in FIPS PUB 180-1 published April 17, 1995.
+ *
+ *      Many of the variable names in the SHA1Context, especially the
+ *      single character names, were used because those were the names
+ *      used in the publication.
+ *
+ *      Please read the file sha1.c for more information.
+ *
+ */
+
+#ifndef _SHA1_H_
+#define _SHA1_H_
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/*
+ *  This structure will hold context information for the hashing
+ *  operation
+ */
+typedef struct SHA1Context {
+  unsigned Message_Digest5; /* Message Digest (output)          */
+
+  unsigned Length_Low;        /* Message length in bits           */
+  unsigned Length_High;       /* Message length in bits           */
+
+  unsigned char Message_Block64; /* 512-bit message blocks      */
+  int Message_Block_Index;    /* Index into message block array   */
+
+  int Computed;               /* Is the digest computed?          */
+  int Corrupted;              /* Is the message digest corruped?  */
+} SHA1Context;
+
+/*
+ *  Function Prototypes
+ */
+void SHA1Reset (SHA1Context*);
+int SHA1Result (SHA1Context*, unsigned char*);
+void SHA1Input (SHA1Context*,
+                const unsigned char*,
+                unsigned);
+
+#define SHA_DIGEST_LENGTH 20
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif

fdk-aac-2.0.3.tar.gz/test Added

fdk-aac-2.0.3.tar.gz/test-encode-decode.c Added

@@ -0,0 +1,383 @@
+/* ------------------------------------------------------------------
+ * Copyright (C) 2017 Martin Storsjo
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *	  http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either
+ * express or implied.
+ * See the License for the specific language governing permissions
+ * and limitations under the License.
+ * -------------------------------------------------------------------
+ */
+
+#include <stdio.h>
+#include <stdint.h>
+#include <stdlib.h>
+#include <string.h>
+#include "aacenc_lib.h"
+#include "aacdecoder_lib.h"
+#include "wavreader.h"
+#include "sha1.h"
+
+
+static int encoder_input_samples, encoder_input_size;
+static int decoder_output_skip;
+static int16_t *encoder_input;
+static int max_diff;
+static uint64_t diff_sum, diff_samples;
+static SHA1Context encode_hash, decode_hash;
+
+static void init_encoder_input(void) {
+	encoder_input_samples = 0;
+	max_diff = 0;
+	diff_sum = diff_samples = 0;
+}
+
+static void free_encoder_input(void) {
+	free(encoder_input);
+	encoder_input = NULL;
+	encoder_input_size = 0;
+}
+
+static void append_encoder_input(const int16_t *input, int samples) {
+	if (encoder_input_samples + samples > encoder_input_size) {
+		int size = 2*(encoder_input_samples + samples);
+		int16_t *ptr = realloc(encoder_input, size * sizeof(*encoder_input));
+		if (!ptr)
+			abort();
+		encoder_input = ptr;
+		encoder_input_size = size;
+	}
+	memcpy(encoder_input + encoder_input_samples, input, samples * sizeof(*input));
+	encoder_input_samples += samples;
+}
+
+static void compare_decoder_output(const int16_t *output, int samples) {
+	int i;
+	// TODO: Stereo upconvert?
+	SHA1Input(&decode_hash, (const unsigned char*) output, samples * sizeof(*output));
+	if (decoder_output_skip > 0) {
+		int n = samples;
+		if (n > decoder_output_skip)
+			n = decoder_output_skip;
+		output += n;
+		samples -= n;
+		decoder_output_skip -= n;
+		if (samples <= 0)
+			return;
+	}
+	if (samples > encoder_input_samples)
+		samples = encoder_input_samples;
+	for (i = 0; i < samples; i++) {
+		int diff = abs(encoder_inputi - outputi);
+		if (diff > max_diff)
+			max_diff = diff;
+		diff_sum += diff;
+		diff_samples++;
+	}
+	memmove(encoder_input, encoder_input + samples, (encoder_input_samples - samples) * sizeof(*encoder_input));
+	encoder_input_samples -= samples;
+}
+
+static int decode(HANDLE_AACDECODER decoder, const uint8_t *ptr, int size, uint8_t *decoder_buffer, int decoder_buffer_size, int channels) {
+	AAC_DECODER_ERROR err;
+	CStreamInfo *info;
+	UINT valid, buffer_size;
+	SHA1Input(&encode_hash, ptr, size);
+	do {
+		valid = buffer_size = size;
+		err = aacDecoder_Fill(decoder, (UCHAR**) &ptr, &buffer_size, &valid);
+		ptr += buffer_size - valid;
+		size -= buffer_size - valid;
+		if (err == AAC_DEC_NOT_ENOUGH_BITS)
+			continue;
+		if (err != AAC_DEC_OK)
+			break;
+		err = aacDecoder_DecodeFrame(decoder, (INT_PCM *) decoder_buffer, decoder_buffer_size / sizeof(INT_PCM), 0);
+		if (!ptr && err != AAC_DEC_OK)
+			break;
+		if (err == AAC_DEC_NOT_ENOUGH_BITS)
+			continue;
+		if (err != AAC_DEC_OK) {
+			fprintf(stderr, "Decoding failed\n");
+			return 1;
+		}
+		info = aacDecoder_GetStreamInfo(decoder);
+		if (info->numChannels != channels) {
+			fprintf(stderr, "Mismatched number of channels, input %d, output %d\n", channels, info->numChannels);
+			return 1;
+		}
+		compare_decoder_output((int16_t*) decoder_buffer, info->numChannels * info->frameSize);
+	} while (size > 0);
+	return 0;
+}
+
+static int test_encode_decode(const char *infile, int aot, int afterburner, int eld_sbr, int vbr, int bitrate, int adts) {
+	void *wav;
+	int format, sample_rate, channels, bits_per_sample;
+	int input_size;
+	uint8_t* input_buf;
+	int16_t* convert_buf;
+	HANDLE_AACENCODER encoder;
+	CHANNEL_MODE mode;
+	AACENC_InfoStruct info = { 0 };
+	HANDLE_AACDECODER decoder;
+	int ret = 0;
+	int decoder_buffer_size = 2048 * 2 * 8;
+	uint8_t* decoder_buffer = malloc(decoder_buffer_size);
+	int avg_diff;
+
+	fprintf(stderr, "Testing encoding with aot %d afterburner %d eld_sbr %d vbr %d bitrate %d adts %d\n", aot, afterburner, eld_sbr, vbr, bitrate, adts);
+	init_encoder_input();
+
+	wav = wav_read_open(infile);
+	if (!wav) {
+		fprintf(stderr, "Unable to open wav file %s\n", infile);
+		return 1;
+	}
+	if (!wav_get_header(wav, &format, &channels, &sample_rate, &bits_per_sample, NULL)) {
+		fprintf(stderr, "Bad wav file %s\n", infile);
+		return 1;
+	}
+	if (format != 1) {
+		fprintf(stderr, "Unsupported WAV format %d\n", format);
+		return 1;
+	}
+	if (bits_per_sample != 16) {
+		fprintf(stderr, "Unsupported WAV sample depth %d\n", bits_per_sample);
+		return 1;
+	}
+	switch (channels) {
+	case 1: mode = MODE_1;       break;
+	case 2: mode = MODE_2;       break;
+	case 3: mode = MODE_1_2;     break;
+	case 4: mode = MODE_1_2_1;   break;
+	case 5: mode = MODE_1_2_2;   break;
+	case 6: mode = MODE_1_2_2_1; break;
+	default:
+		fprintf(stderr, "Unsupported WAV channels %d\n", channels);
+		return 1;
+	}
+	if (aacEncOpen(&encoder, 0, channels) != AACENC_OK) {
+		fprintf(stderr, "Unable to open encoder\n");
+		return 1;
+	}
+	if (aacEncoder_SetParam(encoder, AACENC_AOT, aot) != AACENC_OK) {
+		fprintf(stderr, "Unable to set the AOT\n");
+		return 1;
+	}
+	if (aot == 39 && eld_sbr) {
+		if (aacEncoder_SetParam(encoder, AACENC_SBR_MODE, 1) != AACENC_OK) {
+			fprintf(stderr, "Unable to set SBR mode for ELD\n");
+			return 1;
+		}
+	}
+	if (aacEncoder_SetParam(encoder, AACENC_SAMPLERATE, sample_rate) != AACENC_OK) {
+		fprintf(stderr, "Unable to set the AOT\n");
+		return 1;
+	}
+	if (aacEncoder_SetParam(encoder, AACENC_CHANNELMODE, mode) != AACENC_OK) {
+		fprintf(stderr, "Unable to set the channel mode\n");
+		return 1;
+	}
+	if (aacEncoder_SetParam(encoder, AACENC_CHANNELORDER, 1) != AACENC_OK) {
+		fprintf(stderr, "Unable to set the wav channel order\n");
+		return 1;
+	}
+	if (vbr) {
+		if (aacEncoder_SetParam(encoder, AACENC_BITRATEMODE, vbr) != AACENC_OK) {
+			fprintf(stderr, "Unable to set the VBR bitrate mode\n");
+			return 1;
+		}
+	} else {
+		if (aacEncoder_SetParam(encoder, AACENC_BITRATE, bitrate) != AACENC_OK) {
+			fprintf(stderr, "Unable to set the bitrate\n");
+			return 1;
+		}
+	}
+	if (aacEncoder_SetParam(encoder, AACENC_TRANSMUX, adts ? 2 : 0) != AACENC_OK) {
+		fprintf(stderr, "Unable to set the ADTS transmux\n");
+		return 1;
+	}
+	if (aacEncoder_SetParam(encoder, AACENC_AFTERBURNER, afterburner) != AACENC_OK) {
+		fprintf(stderr, "Unable to set the afterburner mode\n");
+		return 1;
+	}
+	if (aacEncEncode(encoder, NULL, NULL, NULL, NULL) != AACENC_OK) {
+		fprintf(stderr, "Unable to initialize the encoder\n");
+		return 1;
+	}
+	if (aacEncInfo(encoder, &info) != AACENC_OK) {
+		fprintf(stderr, "Unable to get the encoder info\n");
+		return 1;
+	}
+
+	input_size = channels*2*info.frameLength;
+	input_buf = (uint8_t*) malloc(input_size);
+	convert_buf = (int16_t*) malloc(input_size);
+
+	decoder_output_skip = channels * info.nDelay;
+
+	decoder = aacDecoder_Open(adts ? TT_MP4_ADTS : TT_MP4_RAW, 1);
+	if (!adts) {
+		UCHAR *bufArray = { info.confBuf };
+		if (aacDecoder_ConfigRaw(decoder, (UCHAR**) bufArray, &info.confSize) != AAC_DEC_OK) {
+			fprintf(stderr, "Unable to set ASC\n");
+			ret = 1;
+			goto end;
+		}
+	}
+	aacDecoder_SetParam(decoder, AAC_CONCEAL_METHOD, 1);
+	aacDecoder_SetParam(decoder, AAC_PCM_LIMITER_ENABLE, 0);
+	while (1) {
+		AACENC_BufDesc in_buf = { 0 }, out_buf = { 0 };
+		AACENC_InArgs in_args = { 0 };
+		AACENC_OutArgs out_args = { 0 };
+		int in_identifier = IN_AUDIO_DATA;
+		int in_size, in_elem_size;
+		int out_identifier = OUT_BITSTREAM_DATA;
+		int out_size, out_elem_size;
+		int read, i;
+		void *in_ptr, *out_ptr;
+		uint8_t outbuf20480;
+		AACENC_ERROR err;
+
+		read = wav_read_data(wav, input_buf, input_size);
+		for (i = 0; i < read/2; i++) {
+			const uint8_t* in = &input_buf2*i;
+			convert_bufi = in0 | (in1 << 8);
+		}
+		in_ptr = convert_buf;
+		in_size = read;
+		in_elem_size = 2;
+
+		in_buf.numBufs = 1;
+		in_buf.bufs = &in_ptr;
+		in_buf.bufferIdentifiers = &in_identifier;
+		in_buf.bufSizes = &in_size;
+		in_buf.bufElSizes = &in_elem_size;
+
+		if (read <= 0) {
+			in_args.numInSamples = -1;
+		} else {
+			in_args.numInSamples = read/2;
+			append_encoder_input(convert_buf, in_args.numInSamples);
+		}
+		out_ptr = outbuf;
+		out_size = sizeof(outbuf);
+		out_elem_size = 1;
+		out_buf.numBufs = 1;
+		out_buf.bufs = &out_ptr;
+		out_buf.bufferIdentifiers = &out_identifier;
+		out_buf.bufSizes = &out_size;
+		out_buf.bufElSizes = &out_elem_size;
+
+		if ((err = aacEncEncode(encoder, &in_buf, &out_buf, &in_args, &out_args)) != AACENC_OK) {
+			if (err == AACENC_ENCODE_EOF)
+				break;
+			fprintf(stderr, "Encoding failed\n");
+			ret = 1;
+			goto end;
+		}
+		if (out_args.numOutBytes == 0)
+			continue;
+
+		if (decode(decoder, outbuf, out_args.numOutBytes, decoder_buffer, decoder_buffer_size, channels)) {
+			ret = 1;
+			goto end;
+		}
+	}
+
+	if (encoder_input_samples > 0) {
+		fprintf(stderr, "%d unmatched samples left at the end\n", encoder_input_samples);
+		ret = 1;
+		goto end;
+	}
+	avg_diff = 0;
+	if (diff_samples > 0)
+		avg_diff = diff_sum / diff_samples;
+	if (/*max_diff > 10000 ||*/ avg_diff > ((aot == 23) ? 2500 : (aot == 29) ? 1500 : 300)) {
+		fprintf(stderr, "max_diff %d, avg_diff %d\n", max_diff, avg_diff);
+		ret = 1;
+		goto end;
+	}
+
+end:
+	free(input_buf);
+	free(convert_buf);
+	wav_read_close(wav);
+	aacEncClose(&encoder);
+	free(decoder_buffer);
+	aacDecoder_Close(decoder);
+
+	return ret;
+}
+
+int main(int argc, char *argv) {
+	const char* infile;
+	void *wav;
+	int sample_rate, channels;
+	int failures = 0;
+	int i;
+	unsigned char encode_digestSHA_DIGEST_LENGTH, decode_digestSHA_DIGEST_LENGTH;
+	if (argc < 2) {
+		printf("%s input.wav\n", argv0);
+		return 1;
+	}
+	infile = argv1;
+
+	wav = wav_read_open(infile);
+	if (!wav) {
+		fprintf(stderr, "Unable to open wav file %s\n", infile);
+		return 1;
+	}
+	if (!wav_get_header(wav, NULL, &channels, &sample_rate, NULL, NULL)) {
+		fprintf(stderr, "Bad wav file %s\n", infile);
+		return 1;
+	}
+	wav_read_close(wav);
+
+	SHA1Reset(&encode_hash);
+	SHA1Reset(&decode_hash);
+
+	failures += test_encode_decode(infile,  2, 0, 0, 0, 64000, 0); // AAC-LC, without afterburner
+	for (i = 0; i < 2; i++)
+		failures += test_encode_decode(infile, 2, 1, 0, 0, 64000, i); // AAC-LC
+	for (i = 1; i <= 5; i++)
+		failures += test_encode_decode(infile, 2, 1, 0, i, 0, 0); // AAC-LC VBR
+	if (channels == 2) {
+		// HE-AACv2 only works for stereo; HE-AACv1 gets upconverted to stereo (which we don't match properly)
+		for (i = 0; i < 2; i++)
+			failures += test_encode_decode(infile, 5, 1, 0, 0, 64000, i); // HE-AAC
+		for (i = 1; i <= 5; i++)
+			failures += test_encode_decode(infile, 5, 1, 0, i, 0, 0); // HE-AAC VBR
+		for (i = 0; i < 2; i++)
+			failures += test_encode_decode(infile, 29, 1, 0, 0, 64000, i); // HE-AACv2
+		for (i = 1; i <= 5; i++)
+			failures += test_encode_decode(infile, 29, 1, 0, i, 0, 0); // HE-AACv2 VBR
+	}
+	if (channels == 1)
+		failures += test_encode_decode(infile, 23, 1, 0, 0, 64000, 0); // AAC-LD
+	failures += test_encode_decode(infile, 39, 1, 0, 0, 64000, 0); // AAC-ELD
+	failures += test_encode_decode(infile, 39, 1, 1, 0, 64000, 0); // AAC-ELD with SBR
+
+	free_encoder_input();
+	fprintf(stderr, "%d failures\n", failures);
+	SHA1Result(&encode_hash, encode_digest);
+	SHA1Result(&decode_hash, decode_digest);
+	printf("encode hash: ");
+	for (i = 0; i < SHA_DIGEST_LENGTH; i++)
+		printf("%02x", encode_digesti);
+	printf("\n");
+	printf("decode hash: ");
+	for (i = 0; i < SHA_DIGEST_LENGTH; i++)
+		printf("%02x", decode_digesti);
+	printf("\n");
+	return failures;
+}

fdk-aac-2.0.3.tar.gz/test/ref-mono.txt Added

fdk-aac-2.0.3.tar.gz/test/ref-stereo.txt Added

fdk-aac-2.0.3.tar.gz/test/run-test.sh Added

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