ci: Don't bother with static or 32-bit

Static builds are mostly pointless as the compilers used differ quite a lot. We also shouldn't bother with 32-bit anymore, as it has been replaced by 64-bit on Windows.

.github/workflows/build.yml

@@ -0,0 +1,270 @@
+name: Build
+on:
+  workflow_dispatch:
+    inputs:
+      ref:
+        description: 'Branch, Tag or Commit to build'
+        required: false
+        default: 'master'
+      apply_patches:
+        description: 'Apply custom patches (Boolean)'
+        required: false
+        default: false
+
+env:
+  X264_VERSION: "0.161.3049"
+  FFNVCODEC_VERSION: "n11.0.10.0"
+  AMF_VERSION: "v1.4.18"
+  ZLIB_NG_VERSION: "2.0.3"
+
+jobs:
+  cc:
+    runs-on: ubuntu-20.04
+    strategy:
+      matrix:
+        license_version: [ 3, 2 ]
+        license: [ "GPL", "LGPL" ]
+        type: [ "shared" ]
+        bits: [ 64 ]
+    name: "Windows (${{ matrix.bits }}bit, ${{ matrix.type }}, ${{ matrix.license }}v${{ matrix.license_version}})"
+    steps:
+    - name: "automation: Check out"
+      uses: actions/checkout@v2
+      with:
+        submodules: "recursive"
+        fetch-depth: 0
+
+    - name: "automation: Copy patches to safe directory"
+      if: ${{ github.event.inputs.apply_patches == 'true' }}
+      shell: bash
+      run: |
+        if [[ -d ./patches ]]; then
+          cp -a ./patches /tmp/
+        fi
+
+    - name: "automation: Gather Information"
+      id: data
+      shell: bash
+      run: |
+        # Bitness
+        if [ "${{ matrix.bits }}" == "32" ]; then
+          echo "::set-output name=arch::i686"
+          echo "::set-output name=target_os::mingw32"
+          echo "::set-output name=cross_prefix::i686-w64-mingw32"
+        else
+          echo "::set-output name=arch::x86_64"
+          echo "::set-output name=target_os::mingw64"
+          echo "::set-output name=cross_prefix::x86_64-w64-mingw32"
+        fi
+
+        # License (GPL vs LGPL, v2 vs v3)
+        if  [ "${{ matrix.license }}" == "GPL" ]; then
+          echo "::set-output name=flags_license::--enable-gpl"
+        fi
+        if [ "${{ matrix.license_version }}" == "3" ]; then
+          echo "::set-output name=flags_license_version::--enable-version3"
+        fi
+
+        # Build Type
+        if [ "${{ matrix.type }}" == "static" ]; then
+          echo "::set-output name=flags_type::--enable-static --disable-shared"
+        else
+          echo "::set-output name=flags_type::--disable-static --enable-shared"
+        fi
+
+    - name: "ffmpeg: Check out ${{ github.event.inputs.ref }}"
+      uses: actions/checkout@v2
+      with:
+        ref: "${{ github.event.inputs.ref }}"
+        submodules: "recursive"
+        fetch-depth: 0
+
+    - name: "automation: Detect version (and apply patches if necessary)"
+      id: version
+      shell: bash
+      run: |
+        # Detect Major.Minor.Patch version
+        VERSION=( $(cat RELEASE | sed -E 's/\./ /gi') )
+        VERSION_MAJOR=${VERSION[0]}
+        VERSION_MINOR=${VERSION[1]}
+        if (( ${#VERSION[@]} == 3 )); then
+          VERSION_PATCH=${VERSION[2]}
+        else
+          VERSION_PATCH=0
+        fi
+
+        COMMIT="$(git rev-parse --short=8 HEAD)"
+        echo "FFmpeg v${VERSION_MAJOR}.${VERSION_MINOR}.${VERSION_PATCH}-${COMMIT}"
+
+        # Apply available patches
+        if [[ "${{ github.event.inputs.apply_patches}}" == "true" ]]; then
+          echo "Applying custom patches:"
+
+          declare -a PATCHES
+          if [[ "${{ github.event.inputs.ref }}" == "master" ]]; then
+            PATCHES[${#PATCHES[@]}]="master"
+          else
+            PATCHES[${#PATCHES[@]}]="${VERSION_MAJOR}.${VERSION_MINOR}.${VERSION_PATCH}"
+            PATCHES[${#PATCHES[@]}]="${VERSION_MAJOR}.${VERSION_MINOR}"
+            PATCHES[${#PATCHES[@]}]="${VERSION_MAJOR}"
+          fi
+
+          for p in ${PATCHES[@]}; do
+            if [[ -d "/tmp/patches/${p}" ]]; then
+              echo "  Found patches for ${p}:"
+              for f in /tmp/patches/${p}/*.patch; do
+                echo "    ${f}..."
+                [ -e "$f" ] || continue
+                git apply "$f"
+              done
+            else
+              echo "  No patches for ${p}."
+            fi
+          done
+
+          VERSION_PATCH="${VERSION_PATCH}.patched"
+        fi
+
+        # Set Outputs
+        echo "::set-output name=major::${VERSION_MAJOR}"
+        echo "::set-output name=minor::${VERSION_MINOR}"
+        echo "::set-output name=patch::${VERSION_PATCH}"
+        echo "::set-output name=commit::${COMMIT}"
+
+        # Create distrib directory
+        mkdir distrib
+        mkdir distrib/bin
+        mkdir distrib/lib
+        mkdir distrib/include
+        mkdir distrib/share
+
+    - name: "dependency: cmake, make, pkg-config, mingw, nasm"
+      shell: bash
+      run: |
+        sudo apt-get update
+        sudo apt-get install \
+          build-essential git \
+          cmake make ninja-build \
+          pkg-config \
+          mingw-w64 mingw-w64-tools gcc-mingw-w64 g++-mingw-w64 \
+          nasm
+
+# zlib-ng
+    - name: "dependency: zlib (zlib-ng v${{ env.ZLIB_NG_VERSION }}, Zlib license, shared)"
+      id: zlib
+      shell: bash
+      run: |
+        git clone --depth 1 --branch ${ZLIB_NG_VERSION} "https://github.com/zlib-ng/zlib-ng" /tmp/zlib-ng
+        pushd "/tmp/zlib-ng" > /dev/null
+          cmake -H. -Bbuild/build \
+            -DCMAKE_TOOLCHAIN_FILE=./cmake/toolchain-mingw-${{ steps.data.outputs.arch }}.cmake \
+            -DCMAKE_BUILD_TYPE=RELEASE -DZLIB_COMPAT=ON -DZLIB_ENABLE_TESTS=OFF -DBUILD_SHARED_LIBS=ON \
+            -DCMAKE_INSTALL_PREFIX=./build/distrib/
+          cmake --build build/build --target install
+          pushd "./build/distrib" > /dev/null
+            # Fix ZLIB_COMPAT=ON still adding a suffix.
+            cp ./lib/libzlib.dll.a ./lib/libz.dll.a
+
+            # Generate MSVC compatible .lib file
+            gendef - ./bin/libzlib1.dll > ./lib/libzlib.def
+            ${{ steps.data.outputs.cross_prefix }}-dlltool -d ./lib/libzlib.def -l ./lib/libzlib.lib
+            cp ./lib/libzlib.lib ./lib/libz.lib
+          popd > /dev/null
+        popd > /dev/null
+        sudo cp -a /tmp/zlib-ng/build/distrib/. /usr/${{ steps.data.outputs.cross_prefix }}
+        cp -a /tmp/zlib-ng/build/distrib/bin/*.dll ./distrib/bin/
+
+# libx264 (FFmpeg 0.5 and up, arbitrarily limited to 1.0 because I'm lazy)
+    - name: "dependency: x264 v${{ env.X264_VERSION }} (GPLv2, shared)"
+      if: ${{ (steps.version.outputs.major >= 1) && startsWith(matrix.license, 'GPL') }}
+      id: x264
+      shell: bash
+      run: |
+        curl -L -o "/tmp/x264.zip" "https://github.com/Xaymar/x264/releases/download/${X264_VERSION}/x264-${{ matrix.bits }}-shared-GPLv2.zip"
+        7z x -o/tmp/x264/ "/tmp/x264.zip"
+        sudo cp -a /tmp/x264/. /usr/${{ steps.data.outputs.cross_prefix }}/
+        cp -a /tmp/x264/bin/*.dll ./distrib/bin/
+        echo "::set-output name=flags::--enable-libx264"
+
+# NVIDIA Codec Headers (FFmpeg 3.0 and up)
+    - name: "dependency: NVIDIA Codec Headers v${{ env.FFNVCODEC_VERSION }} (MIT, shared)"
+      if: ${{ steps.version.outputs.major >= 3 }}
+      id: ffnvcodec
+      shell: bash
+      run: |
+        git clone --depth 1 --branch ${FFNVCODEC_VERSION} "https://git.videolan.org/git/ffmpeg/nv-codec-headers.git" /tmp/nv-codec-headers
+        pushd "/tmp/nv-codec-headers" > /dev/null
+        make PREFIX=/usr/${{ steps.data.outputs.cross_prefix }}
+        sudo make PREFIX=/usr/${{ steps.data.outputs.cross_prefix }} install
+        popd > /dev/null
+
+        if (( "${{ steps.version.outputs.major }}" >= 4 )); then
+          echo "::set-output name=flags::--enable-ffnvcodec --enable-nvdec --enable-cuvid --enable-nvenc"
+        elif (( "${{ steps.version.outputs.major }}" >= 3 )); then
+          if (( "${{ steps.version.outputs.minor }}" >= 2 )); then
+            # 3.2+ has cuda, cuvid, nvenc
+            echo "::set-output name=flags::--enable-cuvid --enable-nvenc"
+          elif (( "${{ steps.version.outputs.minor }}" >= 1 )); then
+            # 3.1 has cuda, nvenc
+            echo "::set-output name=flags::--enable-nvenc"
+          elif (( "${{ steps.version.outputs.minor }}" >= 0 )); then
+            # 3.0 has nvenc
+            echo "::set-output name=flags::--enable-nvenc"
+          fi
+        fi
+
+# AMD AMF (FFmpeg 4.0 and up)
+    - name: "dependency: AMD AMF (v${{ env.AMF_VERSION }}, MIT, shared)"
+      if: ${{ steps.version.outputs.major >= 4 }}
+      id: amf
+      shell: bash
+      run: |
+        git clone --depth 1 --branch ${AMF_VERSION} "https://github.com/GPUOpen-LibrariesAndSDKs/AMF.git" /tmp/amd-amf
+        pushd "/tmp/amd-amf"
+        sudo cp -R amf/public/include/ /usr/${{ steps.data.outputs.cross_prefix }}/include/AMF
+        popd
+
+        if (( "${{ steps.version.outputs.major }}" >= 4 )); then
+          echo "::set-output name=flags::--enable-amf"
+        fi
+
+# Configure FFmpeg
+    - name: "ffmpeg: Configure"
+      shell: bash
+      run: |
+        export PKG_CONFIG_PATH=/usr/${{ steps.data.outputs.cross_prefix }}/lib/pkgconfig:${PKG_CONFIG_PATH}
+        ./configure \
+          --arch=${{ steps.data.outputs.arch }} \
+          --target-os=${{ steps.data.outputs.target_os }} \
+          --cross-prefix=${{ steps.data.outputs.cross_prefix }}- \
+          --prefix="${{ github.workspace }}/distrib" \
+          --bindir="${{ github.workspace }}/distrib/bin" \
+          --libdir="${{ github.workspace }}/distrib/lib" \
+          --shlibdir="${{ github.workspace }}/distrib/bin" \
+          --pkg-config=pkg-config \
+          --extra-cflags=-O3 --extra-cflags=-mmmx --extra-cflags=-msse --extra-cflags=-msse2 --extra-cflags=-msse3 --extra-cflags=-mssse3 \
+          --extra-cflags=-msse4.1 --extra-cflags=-msse4.2 --extra-cflags=-mavx --extra-cflags=-maes --extra-cflags=-mpclmul \
+          --pkg-config=pkg-config \
+          ${{ steps.data.outputs.flags_license }} ${{ steps.data.outputs.flags_license_version }} \
+          ${{ steps.data.outputs.flags_type }} \
+          ${{ steps.x264.outputs.flags }} \
+          ${{ steps.ffnvcodec.outputs.flags }} \
+          ${{ steps.amf.outputs.flags }}
+
+    - name: "ffmpeg: Compile"
+      shell: bash
+      run: |
+        make -j 4
+
+    - name: "ffmpeg: Install"
+      shell: bash
+      run: |
+        make install
+        # Move .lib files which are in the wrong place.
+        mv ./distrib/bin/*.lib ./distrib/lib/
+
+    - name: "automation: Upload Artifacts"
+      uses: actions/upload-artifact@v1
+      with:
+        name: ffmpeg-${{ matrix.bits }}-${{ matrix.type }}-${{ matrix.license }}v${{ matrix.license_version }}-${{ steps.version.outputs.major }}.${{ steps.version.outputs.minor }}.${{ steps.version.outputs.patch }}-${{ steps.version.outputs.commit }}
+        path: distrib

.github/workflows/refresh.yml

@@ -0,0 +1,125 @@
+name: Refresh
+on:
+  schedule:
+    - cron: '0 0 * * 0'
+  push:
+    branches:
+      - 'automation'
+      - 'automation-test'
+
+jobs:
+  update:
+    runs-on: ubuntu-latest
+    name: "Update Mirror"
+    steps:
+    - uses: actions/checkout@v2
+      name: "Checkout"
+    - name: "Configure"
+      shell: bash
+      run: |
+        git config --global user.name 'GitHub Actions'
+        git config --global user.email 'xaymar@users.noreply.github.com'
+        git config pull.ff only
+        git config pull.rebase true
+    - name: "Remotes"
+      if: ${{ github.event_name == 'schedule' }}
+      shell: bash
+      run: |
+        git remote set-url origin https://x-access-token:${{ secrets.GITHUB_TOKEN }}@github.com/${{ github.repository }}
+        git remote add -f --tags remote https://git.ffmpeg.org/ffmpeg.git
+        git fetch --all
+    - name: "Synchronize with Remote and trigger Builds"
+      if: ${{ github.event_name == 'schedule' }}
+      shell: bash
+      run: |
+        declare -a BRANCHES
+        BRANCHES[${#BRANCHES[@]}]="master"
+        BRANCHES[${#BRANCHES[@]}]="release/4.4"
+        BRANCHES[${#BRANCHES[@]}]="release/4.3"
+        BRANCHES[${#BRANCHES[@]}]="release/4.2"
+        BRANCHES[${#BRANCHES[@]}]="release/4.1"
+        BRANCHES[${#BRANCHES[@]}]="release/4.0"
+        BRANCHES[${#BRANCHES[@]}]="release/3.4"
+        BRANCHES[${#BRANCHES[@]}]="release/3.3"
+        BRANCHES[${#BRANCHES[@]}]="release/3.2"
+        BRANCHES[${#BRANCHES[@]}]="release/3.1"
+        BRANCHES[${#BRANCHES[@]}]="release/3.0"
+        BRANCHES[${#BRANCHES[@]}]="release/2.8"
+        BRANCHES[${#BRANCHES[@]}]="release/2.7"
+        BRANCHES[${#BRANCHES[@]}]="release/2.6"
+        BRANCHES[${#BRANCHES[@]}]="release/2.5"
+        BRANCHES[${#BRANCHES[@]}]="release/2.4"
+        BRANCHES[${#BRANCHES[@]}]="release/2.3"
+        BRANCHES[${#BRANCHES[@]}]="release/2.2"
+        BRANCHES[${#BRANCHES[@]}]="release/2.1"
+        BRANCHES[${#BRANCHES[@]}]="release/2.0"
+        BRANCHES[${#BRANCHES[@]}]="release/1.2"
+        BRANCHES[${#BRANCHES[@]}]="release/1.1"
+        BRANCHES[${#BRANCHES[@]}]="release/1.0"
+        BRANCHES[${#BRANCHES[@]}]="release/0.11"
+        BRANCHES[${#BRANCHES[@]}]="release/0.10"
+        BRANCHES[${#BRANCHES[@]}]="release/0.9"
+        BRANCHES[${#BRANCHES[@]}]="release/0.8"
+        BRANCHES[${#BRANCHES[@]}]="release/0.7"
+        BRANCHES[${#BRANCHES[@]}]="release/0.6"
+        BRANCHES[${#BRANCHES[@]}]="release/0.5"
+        BRANCHES[${#BRANCHES[@]}]="oldabi"
+
+        echo "Testing branches for differences..."
+        for d in ${BRANCHES[@]}; do
+          BRANCH_REQUIRES_UPDATE=false
+          if ! git branch -a | grep origin/${d} > /dev/null; then
+            echo "  '${d}' is missing, creating..."
+            BRANCH_REQUIRES_UPDATE=true
+          elif ! git diff -s --exit-code origin/${d} remote/${d} > /dev/null; then
+            echo "  '${d}' is out of date, updating..."
+            BRANCH_REQUIRES_UPDATE=true
+          fi
+
+          # Always check out the remote branch.
+          git checkout -b "${d}" "remote/${d}" > /dev/null
+
+          if ${BRANCH_REQUIRES_UPDATE}; then
+            git push --follow-tags --set-upstream origin ${d}
+
+            # Trigger build without custom patches
+            curl -s --show-error \
+              -X POST \
+              -H "Authorization: token ${{ secrets.WORKFLOW_TOKEN }}" \
+              -H "Accept: application/vnd.github.v3+json" \
+              -d "{\"ref\":\"${{ github.ref }}\",\"inputs\":{\"ref\":\"${d}\"}}" \
+              "https://api.github.com/repos/${{ github.repository }}/actions/workflows/build.yml/dispatches"
+
+            # Trigger build with custom patches
+            curl -s --show-error \
+              -X POST \
+              -H "Authorization: token ${{ secrets.WORKFLOW_TOKEN }}" \
+              -H "Accept: application/vnd.github.v3+json" \
+              -d "{\"ref\":\"${{ github.ref }}\",\"inputs\":{\"ref\":\"${d}\",\"apply_patches\":\"true\"}}" \
+              "https://api.github.com/repos/${{ github.repository }}/actions/workflows/build.yml/dispatches"
+          fi
+        done
+    - name: "Only trigger build on manual push"
+      if: ${{ github.event_name == 'push' }}
+      shell: bash
+      run: |
+        declare -a BRANCHES
+        BRANCHES[${#BRANCHES[@]}]="master"
+
+        for d in ${BRANCHES[@]}; do
+          # Trigger build without custom patches
+          curl -s --show-error \
+            -X POST \
+            -H "Authorization: token ${{ secrets.WORKFLOW_TOKEN }}" \
+            -H "Accept: application/vnd.github.v3+json" \
+            -d "{\"ref\":\"${{ github.ref }}\",\"inputs\":{\"ref\":\"${d}\"}}" \
+            "https://api.github.com/repos/${{ github.repository }}/actions/workflows/build.yml/dispatches"
+
+          # Trigger build with custom patches
+          curl -s --show-error \
+            -X POST \
+            -H "Authorization: token ${{ secrets.WORKFLOW_TOKEN }}" \
+            -H "Accept: application/vnd.github.v3+json" \
+            -d "{\"ref\":\"${{ github.ref }}\",\"inputs\":{\"ref\":\"${d}\",\"apply_patches\":\"true\"}}" \
+            "https://api.github.com/repos/${{ github.repository }}/actions/workflows/build.yml/dispatches"
+        done

.gitignore

@@ -1,46 +0,0 @@
-.config
-.version
-*.o
-*.so
-*.d
-*.exe
-*.ho
-*-example
-*-test
-*_g
-config.*
-doc/*.1
-doc/*.html
-doc/*.pod
-doxy
-ffmpeg
-ffplay
-ffprobe
-ffserver
-libavcodec/libavcodec*
-libavcore/libavcore*
-libavdevice/libavdevice*
-libavfilter/libavfilter*
-libavformat/libavformat*
-libavutil/avconfig.h
-libavutil/libavutil*
-libpostproc/libpostproc*
-libswscale/libswscale*
-tests/audiogen
-tests/base64
-tests/data
-tests/rotozoom
-tests/seek_test
-tests/tiny_psnr
-tests/videogen
-tests/vsynth1
-tests/vsynth2
-tools/cws2fws
-tools/graph2dot
-tools/lavfi-showfiltfmts
-tools/pktdumper
-tools/probetest
-tools/qt-faststart
-tools/trasher
-tools/trasher*.d
-version.h

COPYING.GPLv2

@@ -1,339 +0,0 @@
-                    GNU GENERAL PUBLIC LICENSE
-                       Version 2, June 1991
-
- Copyright (C) 1989, 1991 Free Software Foundation, Inc.,
- 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- Everyone is permitted to copy and distribute verbatim copies
- of this license document, but changing it is not allowed.
-
-                            Preamble
-
-  The licenses for most software are designed to take away your
-freedom to share and change it.  By contrast, the GNU General Public
-License is intended to guarantee your freedom to share and change free
-software--to make sure the software is free for all its users.  This
-General Public License applies to most of the Free Software
-Foundation's software and to any other program whose authors commit to
-using it.  (Some other Free Software Foundation software is covered by
-the GNU Lesser General Public License instead.)  You can apply it to
-your programs, too.
-
-  When we speak of free software, we are referring to freedom, not
-price.  Our General Public Licenses are designed to make sure that you
-have the freedom to distribute copies of free software (and charge for
-this service if you wish), that you receive source code or can get it
-if you want it, that you can change the software or use pieces of it
-in new free programs; and that you know you can do these things.
-
-  To protect your rights, we need to make restrictions that forbid
-anyone to deny you these rights or to ask you to surrender the rights.
-These restrictions translate to certain responsibilities for you if you
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-
-  For example, if you distribute copies of such a program, whether
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-you have.  You must make sure that they, too, receive or can get the
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-
-  We protect your rights with two steps: (1) copyright the software, and
-(2) offer you this license which gives you legal permission to copy,
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-
-  Also, for each author's protection and ours, we want to make certain
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-   TERMS AND CONDITIONS FOR COPYING, DISTRIBUTION AND MODIFICATION
-
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-
-  <signature of Ty Coon>, 1 April 1989
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COPYING.GPLv3

@@ -1,674 +0,0 @@
-                    GNU GENERAL PUBLIC LICENSE
-                       Version 3, 29 June 2007
-
- Copyright (C) 2007 Free Software Foundation, Inc. <http://fsf.org/>
- Everyone is permitted to copy and distribute verbatim copies
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-                            Preamble
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COPYING.LGPLv2.1

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COPYING.LGPLv3

@@ -1,165 +0,0 @@
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CREDITS

@@ -1,50 +0,0 @@
-This file contains the name of the people who have contributed to
-FFmpeg. The names are sorted alphabetically by last name.
-
-Dénes Balatoni
-Michel Bardiaux
-Fabrice Bellard
-Patrice Bensoussan
-Alex Beregszaszi
-BERO
-Mario Brito
-Ronald Bultje
-Maarten Daniels
-Reimar Doeffinger
-Tim Ferguson
-Brian Foley
-Arpad Gereoffy
-Philip Gladstone
-Vladimir Gneushev
-Roine Gustafsson
-David Hammerton
-Wolfgang Hesseler
-Marc Hoffman
-Falk Hueffner
-Aurélien Jacobs
-Steven Johnson
-Zdenek Kabelac
-Robin Kay
-Todd Kirby
-Nick Kurshev
-Benjamin Larsson
-Loïc Le Loarer
-Daniel Maas
-Mike Melanson
-Loren Merritt
-Jeff Muizelaar
-Michael Niedermayer
-François Revol
-Peter Ross
-Måns Rullgård
-Roman Shaposhnik
-Oded Shimon
-Dieter Shirley
-Konstantin Shishkov
-Juan J. Sierralta
-Ewald Snel
-Sascha Sommer
-Leon van Stuivenberg
-Roberto Togni
-Lionel Ulmer
-Reynaldo Verdejo

Changelog

@@ -1,629 +0,0 @@
-Entries are sorted chronologically from oldest to youngest within each release,
-releases are sorted from youngest to oldest.
-
-version 0.5.10:
-
-- mpeg12: do not decode extradata more than once (CVE-2012-2803)
-- vp6: properly fail on unsupported feature (CVE-2012-2783)
-- vp56: release frames on error (CVE-2012-2783)
-- shorten: Use separate pointers for the allocated memory for decoded samples (CVE-2012-0858)
-- shorten: check for realloc failure
-- h264: check context state before decoding slice data partitions
-- oggdec: check memory allocation
-- Fix uninitialized reads on malformed Ogg files
-- lavf: avoid integer overflow in ff_compute_frame_duration()
-- yuv4mpeg: reject unsupported codecs
-- tiffenc: Check av_malloc() results
-- mpegaudiodec: fix short_start calculation
-- h264: avoid stuck buffer pointer in decode_nal_units
-- yuv4mpeg: return proper error codes (Bug 373)
-- avidec: return 0, not packet size from read_packet()
-- cavsdec: check for changing w/h (CVE-2012-2777 and CVE-2012-2784)
-- avidec: use actually read size instead of requested size CVE-2012-2788
-- bytestream: add a new set of bytestream functions with overread checking
-- avsdec: Set dimensions instead of relying on the demuxer (CVE-2012-2801)
-- lavfi: avfilter_merge_formats: handle case where inputs are same
-- bmpdec: only initialize palette for pal8 (Bug 367)
-- Bump version number for the 0.5.10 release
-- lavfi: avfilter_merge_formats: handle case where inputs are same
-- mpegvideo: Don't use ff_mspel_motion() for vc1
-- imgconvert: avoid undefined left shift in avcodec_find_best_pix_fmt
-- nuv: check RTjpeg header for validity
-- vc1dec: add flush function for WMV9 and VC-1 decoders
-
-
-version 0.5.9:
-
-- dpcm: ignore extra unpaired bytes in stereo streams (CVE-2011-3951)
-- h264: Add check for invalid chroma_format_idc (CVE-2012-0851)
-- adpcm: ADPCM Electronic Arts has always two channels (CVE-2012-0852)
-- kmvc: Check palsize (CVE-2011-3952)
-- qdm2: clip array indices returned by qdm2_get_vlc()
-- configure: properly check for mingw-w64 through installed headers
-- Replace every usage of -lvfw32 with what is particularly necessary for that case
-- mingw32: properly check if vfw capture is supported by the system headers
-- mingw32: merge checks for mingw-w64 and mingw32-runtime >= 3.15 into one
-- vfwcap: Include windows.h before vfw.h since the latter requires defines from the former
-- ea: check chunk_size for validity
-- eatqi: move "block" variable into context to ensure sufficient alignment for idct_put
-- tqi: Pass errors from the MB decoder
-- png: check bit depth for PAL8/Y400A pixel formats.
-
-
-version 0.5.8:
-
-- id3v2: fix skipping extended header in id3v2.4
-- nsvdec: Several bugfixes related to CVE-2011-3940
-- dv: check stype
-- dv: Fix null pointer dereference due to ach=0
-- dv: Fix small stack overread related to CVE-2011-3929 and CVE-2011-3936.
-- atrac3: Fix crash in tonal component decoding, fixes CVE-2012-0853
-- mjpegbdec: Fix overflow in SOS, fixes CVE-2011-3947
-- motionpixels: Clip YUV values after applying a gradient.
-- vqavideo: return error if image size is not a multiple of block size,
-  fixes CVE-2012-0947.
-
-
-version 0.5.7:
-- vorbis: An additional defense in the Vorbis codec. (CVE-2011-3895)
-- vorbisdec: Fix decoding bug with channel handling.
-- matroskadec: Fix a bug where a pointer was cached to an array that might
-  later move due to a realloc(). (CVE-2011-3893)
-- vorbis: Avoid some out-of-bounds reads. (CVE-2011-3893)
-- vp3: fix oob read for negative tokens and memleaks on error, (CVE-2011-3892)
-- vp3: fix streams with non-zero last coefficient.
-
-
-version 0.5.6:
-- svq1dec: call avcodec_set_dimensions() after dimensions changed. (NGS00148, CVE-2011-4579)
-- vmd: fix segfaults on corruped streams (CVE-2011-4364)
-- commits related to CVE-2011-4353:
-  - vp6: partially propagate huffman tree building errors during coeff model parsing and fix misspelling
-  - Plug some memory leaks in the VP6 decoder
-  - vp6: Reset the internal state when aborting key frames header parsing
-  - vp6: Fix illegal read.
-  - vp6: Fix illegal read.
-  - Fix out of bound reads in the QDM2 decoder.
-- commits related to CVE-2011-4351:
-  - Check for out of bound writes in the QDM2 decoder.
-  - qdm2: check output buffer size before decoding
-  - Fix qdm2 decoder packet handling to match the api
-
-
-version 0.5.5:
-
-- Fix memory (re)allocation in matroskadec.c (MSVR11-011/CVE-2011-3504)
-- Fix some crashes with invalid bitstreams in the CAVS decoder
-  (CVE-2011-3362, CVE-2011-3973, CVE-2011-3974)
-- Compilation fixes for gcc-4.6, testsuite now passes again
-- Detect and handle overreads in the MJPEG decoder.
-- multiple other security fixes.
-
-
-
-version 0.5.4:
-
-- Fix memory corruption in WMV parsing (addresses CVE-2010-3908)
-- Fix heap corruption crashes (addresses CVE-2011-0722)
-- Fix crashes in Vorbis decoding found by zzuf (addresses CVE-2010-4704)
-- Fix another crash in Vorbis decoding (addresses CVE-2011-0480, Chrome issue 68115)
-- Fix invalid reads in VC-1 decoding (related to CVE-2011-0723)
-- Do not attempt to decode APE file with no frames
-  (adresses http://packetstorm.linuxsecurity.com/1103-exploits/vlc105-dos.txt)
-
-
-
-version 0.5.3:
-
-- build system improvements
-- performance fix for seekable HTTP
-- fix several potentially exploitable issues in the FLIC decoder
-  (addresses CVE-2010-3429)
-
-
-
-version 0.5.2:
-
-- Hurd support
-- PowerPC without AltiVec compilation issues
-- validate channels and samplerate in the Vorbis decoder
-
-
-
-version 0.5.1:
-
-- build system updates
-- documentation updates
-- libswscale now is LGPL except for x86 optimizations
-- fix for GPL code in libswscale that was erroneously activated
-- AltiVec code in libswscale is now LGPL
-- remaining GPL parts in AC-3 decoder converted to LGPL
-- (L)GPL license upgrade support
-- AMR-NB decoding/encoding, AMR-WB decoding via OpenCORE libraries
-- enable symbol versioning by default for linkers that support it
-- backport av_lockmgr_register(), see doc/APIchanges for details
-- security fixes for:
-  - ASF, Ogg and MOV demuxers
-  - FFv1, H.264, HuffYUV, MLP, MPEG audio and Snow decoders
-
-
-
-version 0.5:
-
-- The "device" muxers and demuxers are now in a new libavdevice library
-- DV50 AKA DVCPRO50 encoder, decoder, muxer and demuxer
-- DV100 AKA DVCPRO HD decoder and demuxer
-- TechSmith Camtasia (TSCC) video decoder
-- IBM Ultimotion (ULTI) video decoder
-- Sierra Online audio file demuxer and decoder
-- Apple QuickDraw (qdrw) video decoder
-- Creative ADPCM audio decoder (16 bits as well as 8 bits schemes)
-- Electronic Arts Multimedia (WVE/UV2/etc.) file demuxer
-- Miro VideoXL (VIXL) video decoder
-- H.261 video encoder
-- QPEG video decoder
-- Nullsoft Video (NSV) file demuxer
-- Shorten audio decoder
-- LOCO video decoder
-- Apple Lossless Audio Codec (ALAC) decoder
-- Winnov WNV1 video decoder
-- Autodesk Animator Studio Codec (AASC) decoder
-- Indeo 2 video decoder
-- Fraps FPS1 video decoder
-- Snow video encoder/decoder
-- Sonic audio encoder/decoder
-- Vorbis audio encoder/decoder
-- Macromedia ADPCM decoder
-- Duck TrueMotion 2 video decoder
-- support for decoding FLX and DTA extensions in FLIC files
-- H.264 custom quantization matrices support
-- ffserver fixed, it should now be usable again
-- QDM2 audio decoder
-- Real Cooker audio decoder
-- TrueSpeech audio decoder
-- WMA2 audio decoder fixed, now all files should play correctly
-- RealAudio 14.4 and 28.8 decoders fixed
-- JPEG-LS encoder and decoder
-- CamStudio video decoder
-- build system improvements
-- tabs and trailing whitespace removed from the codebase
-- AIFF/AIFF-C audio format, encoding and decoding
-- ADTS AAC file reading and writing
-- Creative VOC file reading and writing
-- American Laser Games multimedia (*.mm) playback system
-- Zip Blocks Motion Video decoder and encoder
-- improved Theora/VP3 decoder
-- True Audio (TTA) decoder
-- AVS demuxer and video decoder
-- Smacker demuxer and decoder
-- NuppelVideo/MythTV demuxer and RTjpeg decoder
-- KMVC decoder
-- MPEG-2 intra VLC support
-- MPEG-2 4:2:2 encoder
-- Flash Screen Video decoder
-- GXF demuxer
-- Chinese AVS decoder
-- GXF muxer
-- MXF demuxer
-- VC-1/WMV3/WMV9 video decoder
-- MacIntel support
-- AVISynth support
-- VMware video decoder
-- VP5 video decoder
-- VP6 video decoder
-- WavPack lossless audio decoder
-- Targa (.TGA) picture decoder
-- Delphine Software .cin demuxer/audio and video decoder
-- Tiertex .seq demuxer/video decoder
-- MTV demuxer
-- TIFF picture encoder and decoder
-- GIF picture decoder
-- Intel Music Coder decoder
-- Musepack decoder
-- Flash Screen Video encoder
-- Theora encoding via libtheora
-- BMP encoder
-- WMA encoder
-- GSM-MS encoder and decoder
-- DCA decoder
-- DXA demuxer and decoder
-- DNxHD decoder
-- Gamecube movie (.THP) playback system
-- Blackfin optimizations
-- Interplay C93 demuxer and video decoder
-- Bethsoft VID demuxer and video decoder
-- CRYO APC demuxer
-- Atrac3 decoder
-- V.Flash PTX decoder
-- RoQ muxer, RoQ audio encoder
-- Renderware TXD demuxer and decoder
-- extern C declarations for C++ removed from headers
-- sws_flags command line option
-- codebook generator
-- RoQ video encoder
-- QTRLE encoder
-- OS/2 support removed and restored again
-- AC-3 decoder
-- NUT muxer
-- Matroska muxer
-- slice-based parallel H.264 decoding
-- Monkey's Audio demuxer and decoder
-- additional SPARC (VIS) optimizations
-- AMV audio and video decoder
-- DNxHD encoder
-- H.264 PAFF decoding
-- Nellymoser ASAO decoder
-- Beam Software SIFF demuxer and decoder
-- libvorbis Vorbis decoding removed in favor of native decoder
-- IntraX8 (J-Frame) subdecoder for WMV2 and VC-1
-- Ogg (Theora, Vorbis and FLAC) muxer
-- PC Paintbrush PCX decoder
-- Sun Rasterfile decoder
-- TechnoTrend PVA demuxer
-- Linux Media Labs MPEG-4 (LMLM4) demuxer
-- AVM2 (Flash 9) SWF muxer
-- QT variant of IMA ADPCM encoder
-- VFW grabber
-- iPod/iPhone compatible mp4 muxer
-- Mimic decoder
-- MSN TCP Webcam stream demuxer
-- RL2 demuxer / decoder
-- IFF demuxer
-- 8SVX audio decoder
-- non-recursive Makefiles
-- BFI demuxer
-- MAXIS EA XA (.xa) demuxer / decoder
-- BFI video decoder
-- OMA demuxer
-- MLP/TrueHD decoder
-- Electronic Arts CMV decoder
-- Motion Pixels Video decoder
-- Motion Pixels MVI demuxer
-- removed animated GIF decoder/demuxer
-- D-Cinema audio muxer
-- Electronic Arts TGV decoder
-- Apple Lossless Audio Codec (ALAC) encoder
-- AAC decoder
-- floating point PCM encoder/decoder
-- MXF muxer
-- E-AC-3 support added to AC-3 decoder
-- Nellymoser ASAO encoder
-- ASS and SSA demuxer and muxer
-- liba52 wrapper removed
-- SVQ3 watermark decoding support
-- Speex decoding via libspeex
-- Electronic Arts TGQ decoder
-- RV30 and RV40 decoder
-- QCELP / PureVoice decoder
-- hybrid WavPack support
-- R3D REDCODE demuxer
-- ALSA support for playback and record
-- Electronic Arts TQI decoder
-- OpenJPEG based JPEG 2000 decoder
-- NC (NC4600) camera file demuxer
-- Gopher client support
-- MXF D-10 muxer
-- generic metadata API
-
-
-version 0.4.9-pre1:
-
-- DV encoder, DV muxer
-- Microsoft RLE video decoder
-- Microsoft Video-1 decoder
-- Apple Animation (RLE) decoder
-- Apple Graphics (SMC) decoder
-- Apple Video (RPZA) decoder
-- Cinepak decoder
-- Sega FILM (CPK) file demuxer
-- Westwood multimedia support (VQA & AUD files)
-- Id Quake II CIN playback support
-- 8BPS video decoder
-- FLIC playback support
-- RealVideo 2.0 (RV20) decoder
-- Duck TrueMotion v1 (DUCK) video decoder
-- Sierra VMD demuxer and video decoder
-- MSZH and ZLIB decoder support
-- SVQ1 video encoder
-- AMR-WB support
-- PPC optimizations
-- rate distortion optimal cbp support
-- rate distorted optimal ac prediction for MPEG-4
-- rate distorted optimal lambda->qp support
-- AAC encoding with libfaac
-- Sunplus JPEG codec (SP5X) support
-- use Lagrange multipler instead of QP for ratecontrol
-- Theora/VP3 decoding support
-- XA and ADX ADPCM codecs
-- export MPEG-2 active display area / pan scan
-- Add support for configuring with IBM XLC
-- floating point AAN DCT
-- initial support for zygo video (not complete)
-- RGB ffv1 support
-- new audio/video parser API
-- av_log() system
-- av_read_frame() and av_seek_frame() support
-- missing last frame fixes
-- seek by mouse in ffplay
-- noise reduction of DCT coefficients
-- H.263 OBMC & 4MV support
-- H.263 alternative inter vlc support
-- H.263 loop filter
-- H.263 slice structured mode
-- interlaced DCT support for MPEG-2 encoding
-- stuffing to stay above min_bitrate
-- MB type & QP visualization
-- frame stepping for ffplay
-- interlaced motion estimation
-- alternate scantable support
-- SVCD scan offset support
-- closed GOP support
-- SSE2 FDCT
-- quantizer noise shaping
-- G.726 ADPCM audio codec
-- MS ADPCM encoding
-- multithreaded/SMP motion estimation
-- multithreaded/SMP encoding for MPEG-1/MPEG-2/MPEG-4/H.263
-- multithreaded/SMP decoding for MPEG-2
-- FLAC decoder
-- Metrowerks CodeWarrior suppport
-- H.263+ custom pcf support
-- nicer output for 'ffmpeg -formats'
-- Matroska demuxer
-- SGI image format, encoding and decoding
-- H.264 loop filter support
-- H.264 CABAC support
-- nicer looking arrows for the motion vector visualization
-- improved VCD support
-- audio timestamp drift compensation
-- MPEG-2 YUV 422/444 support
-- polyphase kaiser windowed sinc and blackman nuttall windowed sinc audio resample
-- better image scaling
-- H.261 support
-- correctly interleave packets during encoding
-- VIS optimized motion compensation
-- intra_dc_precision>0 encoding support
-- support reuse of motion vectors/MB types/field select values of the source video
-- more accurate deblock filter
-- padding support
-- many optimizations and bugfixes
-- FunCom ISS audio file demuxer and according ADPCM decoding
-
-version 0.4.8:
-
-- MPEG-2 video encoding (Michael)
-- Id RoQ playback subsystem (Mike Melanson and Tim Ferguson)
-- Wing Commander III Movie (.mve) file playback subsystem (Mike Melanson
-  and Mario Brito)
-- Xan DPCM audio decoder (Mario Brito)
-- Interplay MVE playback subsystem (Mike Melanson)
-- Duck DK3 and DK4 ADPCM audio decoders (Mike Melanson)
-
-version 0.4.7:
-
-- RealAudio 1.0 (14_4) and 2.0 (28_8) native decoders. Author unknown, code from mplayerhq
-  (originally from public domain player for Amiga at http://www.honeypot.net/audio)
-- current version now also compiles with older GCC (Fabrice)
-- 4X multimedia playback system including 4xm file demuxer (Mike
-  Melanson), and 4X video and audio codecs (Michael)
-- Creative YUV (CYUV) decoder (Mike Melanson)
-- FFV1 codec (our very simple lossless intra only codec, compresses much better
-  than HuffYUV) (Michael)
-- ASV1 (Asus), H.264, Intel indeo3 codecs have been added (various)
-- tiny PNG encoder and decoder, tiny GIF decoder, PAM decoder (PPM with
-  alpha support), JPEG YUV colorspace support. (Fabrice Bellard)
-- ffplay has been replaced with a newer version which uses SDL (optionally)
-  for multiplatform support (Fabrice)
-- Sorenson Version 3 codec (SVQ3) support has been added (decoding only) - donated
-  by anonymous
-- AMR format has been added (Johannes Carlsson)
-- 3GP support has been added (Johannes Carlsson)
-- VP3 codec has been added (Mike Melanson)
-- more MPEG-1/2 fixes
-- better multiplatform support, MS Visual Studio fixes (various)
-- AltiVec optimizations (Magnus Damn and others)
-- SH4 processor support has been added (BERO)
-- new public interfaces (avcodec_get_pix_fmt) (Roman Shaposhnick)
-- VOB streaming support (Brian Foley)
-- better MP3 autodetection (Andriy Rysin)
-- qpel encoding (Michael)
-- 4mv+b frames encoding finally fixed (Michael)
-- chroma ME (Michael)
-- 5 comparison functions for ME (Michael)
-- B-frame encoding speedup (Michael)
-- WMV2 codec (unfinished - Michael)
-- user specified diamond size for EPZS (Michael)
-- Playstation STR playback subsystem, still experimental (Mike and Michael)
-- ASV2 codec (Michael)
-- CLJR decoder (Alex)
-
-.. And lots more new enhancements and fixes.
-
-version 0.4.6:
-
-- completely new integer only MPEG audio layer 1/2/3 decoder rewritten
-  from scratch
-- Recoded DCT and motion vector search with gcc (no longer depends on nasm)
-- fix quantization bug in AC3 encoder
-- added PCM codecs and format. Corrected WAV/AVI/ASF PCM issues
-- added prototype ffplay program
-- added GOB header parsing on H.263/H.263+ decoder (Juanjo)
-- bug fix on MCBPC tables of H.263 (Juanjo)
-- bug fix on DC coefficients of H.263 (Juanjo)
-- added Advanced Prediction Mode on H.263/H.263+ decoder (Juanjo)
-- now we can decode H.263 streams found in QuickTime files (Juanjo)
-- now we can decode H.263 streams found in VIVO v1 files(Juanjo)
-- preliminary RTP "friendly" mode for H.263/H.263+ coding. (Juanjo)
-- added GOB header for H.263/H.263+ coding on RTP mode (Juanjo)
-- now H.263 picture size is returned on the first decoded frame (Juanjo)
-- added first regression tests
-- added MPEG-2 TS demuxer
-- new demux API for libav
-- more accurate and faster IDCT (Michael)
-- faster and entropy-controlled motion search (Michael)
-- two pass video encoding (Michael)
-- new video rate control (Michael)
-- added MSMPEG4V1, MSMPEGV2 and WMV1 support (Michael)
-- great performance improvement of video encoders and decoders (Michael)
-- new and faster bit readers and vlc parsers (Michael)
-- high quality encoding mode: tries all macroblock/VLC types (Michael)
-- added DV video decoder
-- preliminary RTP/RTSP support in ffserver and libavformat
-- H.263+ AIC decoding/encoding support (Juanjo)
-- VCD MPEG-PS mode (Juanjo)
-- PSNR stuff (Juanjo)
-- simple stats output (Juanjo)
-- 16-bit and 15-bit RGB/BGR/GBR support (Bisqwit)
-
-version 0.4.5:
-
-- some header fixes (Zdenek Kabelac <kabi at informatics.muni.cz>)
-- many MMX optimizations (Nick Kurshev <nickols_k at mail.ru>)
-- added configure system (actually a small shell script)
-- added MPEG audio layer 1/2/3 decoding using LGPL'ed mpglib by
-  Michael Hipp (temporary solution - waiting for integer only
-  decoder)
-- fixed VIDIOCSYNC interrupt
-- added Intel H.263 decoding support ('I263' AVI fourCC)
-- added Real Video 1.0 decoding (needs further testing)
-- simplified image formats again. Added PGM format (=grey
-  pgm). Renamed old PGM to PGMYUV.
-- fixed msmpeg4 slice issues (tell me if you still find problems)
-- fixed OpenDivX bugs with newer versions (added VOL header decoding)
-- added support for MPlayer interface
-- added macroblock skip optimization
-- added MJPEG decoder
-- added mmx/mmxext IDCT from libmpeg2
-- added pgmyuvpipe, ppm, and ppm_pipe formats (original patch by Celer
-  <celer at shell.scrypt.net>)
-- added pixel format conversion layer (e.g. for MJPEG or PPM)
-- added deinterlacing option
-- MPEG-1/2 fixes
-- MPEG-4 vol header fixes (Jonathan Marsden <snmjbm at pacbell.net>)
-- ARM optimizations (Lionel Ulmer <lionel.ulmer at free.fr>).
-- Windows porting of file converter
-- added MJPEG raw format (input/ouput)
-- added JPEG image format support (input/output)
-
-version 0.4.4:
-
-- fixed some std header definitions (Bjorn Lindgren
-  <bjorn.e.lindgren at telia.com>).
-- added MPEG demuxer (MPEG-1 and 2 compatible).
-- added ASF demuxer
-- added prototype RM demuxer
-- added AC3 decoding (done with libac3 by Aaron Holtzman)
-- added decoding codec parameter guessing (.e.g. for MPEG, because the
-  header does not include them)
-- fixed header generation in MPEG-1, AVI and ASF muxer: wmplayer can now
-  play them (only tested video)
-- fixed H.263 white bug
-- fixed phase rounding in img resample filter
-- add MMX code for polyphase img resample filter
-- added CPU autodetection
-- added generic title/author/copyright/comment string handling (ASF and RM
-  use them)
-- added SWF demux to extract MP3 track (not usable yet because no MP3
-  decoder)
-- added fractional frame rate support
-- codecs are no longer searched by read_header() (should fix ffserver
-  segfault)
-
-version 0.4.3:
-
-- BGR24 patch (initial patch by Jeroen Vreeken <pe1rxq at amsat.org>)
-- fixed raw yuv output
-- added motion rounding support in MPEG-4
-- fixed motion bug rounding in MSMPEG4
-- added B-frame handling in video core
-- added full MPEG-1 decoding support
-- added partial (frame only) MPEG-2 support
-- changed the FOURCC code for H.263 to "U263" to be able to see the
-  +AVI/H.263 file with the UB Video H.263+ decoder. MPlayer works with
-  this +codec ;) (JuanJo).
-- Halfpel motion estimation after MB type selection (JuanJo)
-- added pgm and .Y.U.V output format
-- suppressed 'img:' protocol. Simply use: /tmp/test%d.[pgm|Y] as input or
-  output.
-- added pgmpipe I/O format (original patch from Martin Aumueller
-  <lists at reserv.at>, but changed completely since we use a format
-  instead of a protocol)
-
-version 0.4.2:
-
-- added H.263/MPEG-4/MSMPEG4 decoding support. MPEG-4 decoding support
-  (for OpenDivX) is almost complete: 8x8 MVs and rounding are
-  missing. MSMPEG4 support is complete.
-- added prototype MPEG-1 decoder. Only I- and P-frames handled yet (it
-  can decode ffmpeg MPEGs :-)).
-- added libavcodec API documentation (see apiexample.c).
-- fixed image polyphase bug (the bottom of some images could be
-  greenish)
-- added support for non clipped motion vectors (decoding only)
-  and image sizes non-multiple of 16
-- added support for AC prediction (decoding only)
-- added file overwrite confirmation (can be disabled with -y)
-- added custom size picture to H.263 using H.263+ (Juanjo)
-
-version 0.4.1:
-
-- added MSMPEG4 (aka DivX) compatible encoder. Changed default codec
-  of AVI and ASF to DIV3.
-- added -me option to set motion estimation method
-  (default=log). suppressed redundant -hq option.
-- added options -acodec and -vcodec to force a given codec (useful for
-  AVI for example)
-- fixed -an option
-- improved dct_quantize speed
-- factorized some motion estimation code
-
-version 0.4.0:
-
-- removing grab code from ffserver and moved it to ffmpeg. Added
-  multistream support to ffmpeg.
-- added timeshifting support for live feeds (option ?date=xxx in the
-  URL)
-- added high quality image resize code with polyphase filter (need
-  mmx/see optimization). Enable multiple image size support in ffserver.
-- added multi live feed support in ffserver
-- suppressed master feature from ffserver (it should be done with an
-  external program which opens the .ffm url and writes it to another
-  ffserver)
-- added preliminary support for video stream parsing (WAV and AVI half
-  done). Added proper support for audio/video file conversion in
-  ffmpeg.
-- added preliminary support for video file sending from ffserver
-- redesigning I/O subsystem: now using URL based input and output
-  (see avio.h)
-- added WAV format support
-- added "tty user interface" to ffmpeg to stop grabbing gracefully
-- added MMX/SSE optimizations to SAD (Sums of Absolutes Differences)
-  (Juan J. Sierralta P. a.k.a. "Juanjo" <juanjo at atmlab.utfsm.cl>)
-- added MMX DCT from mpeg2_movie 1.5 (Juanjo)
-- added new motion estimation algorithms, log and phods (Juanjo)
-- changed directories: libav for format handling, libavcodec for
-  codecs
-
-version 0.3.4:
-
-- added stereo in MPEG audio encoder
-
-version 0.3.3:
-
-- added 'high quality' mode which use motion vectors. It can be used in
-  real time at low resolution.
-- fixed rounding problems which caused quality problems at high
-  bitrates and large GOP size
-
-version 0.3.2: small fixes
-
-- ASF fixes
-- put_seek bug fix
-
-version 0.3.1: added avi/divx support
-
-- added AVI support
-- added MPEG-4 codec compatible with OpenDivX. It is based on the H.263 codec
-- added sound for flash format (not tested)
-
-version 0.3: initial public release

INSTALL

@@ -1,11 +0,0 @@
-
-1) Type './configure' to create the configuration. A list of configure
-options is printed by running 'configure --help'.
-
-'configure' can be launched from a directory different from the FFmpeg
-sources to build the objects out of tree. To do this, use an absolute
-path when launching 'configure', e.g. '/ffmpegdir/ffmpeg/configure'.
-
-2) Then type 'make' to build FFmpeg. GNU Make 3.81 or later is required.
-
-3) Type 'make install' to install all binaries and libraries you built.

LICENSE

@@ -1,50 +0,0 @@
-FFmpeg:
--------
-
-Most files in FFmpeg are under the GNU Lesser General Public License version 2.1
-or later (LGPL v2.1+). Read the file COPYING.LGPLv2.1 for details. Some other
-files have MIT/X11/BSD-style licenses. In combination the LGPL v2.1+ applies to
-FFmpeg.
-
-Some optional parts of FFmpeg are licensed under the GNU General Public License
-version 2 or later (GPL v2+). See the file COPYING.GPLv2 for details. None of
-these parts are used by default, you have to explicitly pass --enable-gpl to
-configure to activate them. In this case, FFmpeg's license changes to GPL v2+.
-
-Specifically, the GPL parts of FFmpeg are
-
-- libpostproc
-- some x86 optimizations in libswscale
-- optional x86 optimizations in the files
-  libavcodec/x86/h264_deblock_sse2.asm
-  libavcodec/x86/h264_idct_sse2.asm
-  libavcodec/x86/idct_mmx.c
-- the X11 grabber in libavdevice/x11grab.c
-
-There are a handful of files under other licensing terms, namely:
-
-* The files libavcodec/jfdctfst.c, libavcodec/jfdctint.c, libavcodec/jrevdct.c
-  are taken from libjpeg, see the top of the files for licensing details.
-
-Should you, for whatever reason, prefer to use version 3 of the (L)GPL, then
-the configure parameter --enable-version3 will activate this licensing option
-for you. Read the file COPYING.LGPLv3 or, if you have enabled GPL parts,
-COPYING.GPLv3 to learn the exact legal terms that apply in this case.
-
-
-external libraries:
--------------------
-
-Some external libraries, e.g. libx264, are under GPL and can be used in
-conjunction with FFmpeg. They require --enable-gpl to be passed to configure
-as well.
-
-The OpenCORE external libraries are under the Apache License 2.0. That license
-is incompatible with the LGPL v2.1 and the GPL v2, but not with version 3 of
-those licenses. So to combine the OpenCORE libraries with FFmpeg, the license
-version needs to be upgraded by passing --enable-version3 to configure.
-
-The nonfree external libraries libamrnb, libamrwb and libfaac can be hooked up
-in FFmpeg. You need to pass --enable-nonfree to configure to enable them. Employ
-this option with care as FFmpeg then becomes nonfree and unredistributable.
-Note that libfaac claims to be LGPL, but is not.

MAINTAINERS

@@ -1,334 +0,0 @@
-FFmpeg maintainers
-==================
-
-Below is a list of the people maintaining different parts of the
-FFmpeg code.
-
-
-Project Leader
-==============
-
-  final design decisions
-
-
-Applications
-============
-
-ffmpeg:
-  ffmpeg.c                              Michael Niedermayer
-
-  Video Hooks:
-    vhook
-    vhook/watermark.c                   Marcus Engene
-    vhook/ppm.c
-    vhook/drawtext.c
-    vhook/fish.c
-    vhook/null.c
-    vhook/imlib2.c
-
-ffplay:
-  ffplay.c                              Michael Niedermayer
-
-ffserver:
-  ffserver.c, ffserver.h                Baptiste Coudurier
-
-Commandline utility code:
-  cmdutils.c, cmdutils.h                Michael Niedermayer
-
-QuickTime faststart:
-  qt-faststart.c                        Mike Melanson
-
-
-Miscellaneous Areas
-===================
-
-documentation                           Mike Melanson, Diego Biurrun
-website                                 Robert Swain
-build system (configure,Makefiles)      Diego Biurrun, Mans Rullgard
-project server                          Diego Biurrun, Mans Rullgard
-mailinglists                            Michael Niedermayer, Baptiste Coudurier
-presets                                 Robert Swain
-release management                      Diego Biurrun, Reinhard Tartler
-
-
-libavutil
-=========
-
-External Interfaces:
-  libavutil/avutil.h                    Michael Niedermayer
-Internal Interfaces:
-  libavutil/common.h                    Michael Niedermayer
-
-Other:
-  intfloat*                             Michael Niedermayer
-  rational.c, rational.h                Michael Niedermayer
-  mathematics.c, mathematics.h          Michael Niedermayer
-  integer.c, integer.h                  Michael Niedermayer
-  bswap.h
-
-
-libavcodec
-==========
-
-Generic Parts:
-  External Interfaces:
-    avcodec.h                           Michael Niedermayer
-  utility code:
-    utils.c                             Michael Niedermayer
-    mem.c                               Michael Niedermayer
-    opt.c, opt.h                        Michael Niedermayer
-  arithmetic expression evaluator:
-    eval.c                              Michael Niedermayer
-  audio and video frame extraction:
-    parser.c                            Michael Niedermayer
-  bitstream reading:
-    bitstream.c, bitstream.h            Michael Niedermayer
-  CABAC:
-    cabac.h, cabac.c                    Michael Niedermayer
-  DSP utilities:
-    dsputils.c, dsputils.h              Michael Niedermayer
-  entropy coding:
-    rangecoder.c, rangecoder.h          Michael Niedermayer
-    lzw.*                               Michael Niedermayer
-  floating point AAN DCT:
-    faandct.c, faandct.h                Michael Niedermayer
-  Golomb coding:
-    golomb.c, golomb.h                  Michael Niedermayer
-  LPC:
-    lpc.c, lpc.h                        Justin Ruggles
-  motion estimation:
-    motion*                             Michael Niedermayer
-  rate control:
-    ratecontrol.c                       Michael Niedermayer
-    libxvid_rc.c                        Michael Niedermayer
-  simple IDCT:
-    simple_idct.c, simple_idct.h        Michael Niedermayer
-  postprocessing:
-    libpostproc/*                       Michael Niedermayer
-  vdpau:
-    vdpau*                              Carl Eugen Hoyos
-
-Codecs:
-  4xm.c                                 Michael Niedermayer
-  8bps.c                                Roberto Togni
-  8svx.c                                Jaikrishnan Menon
-  aasc.c                                Kostya Shishkov
-  aac.[ch], aactab.[ch], aacdectab.h    Robert Swain
-  ac3*                                  Justin Ruggles
-  alacenc.c                             Jaikrishnan Menon
-  apedec.c                              Kostya Shishkov
-  asv*                                  Michael Niedermayer
-  atrac3*                               Benjamin Larsson
-  bmp.c                                 Mans Rullgard, Kostya Shishkov
-  cavs*                                 Stefan Gehrer
-  cinepak.c                             Roberto Togni
-  cljr                                  Alex Beregszaszi
-  cook.c, cookdata.h                    Benjamin Larsson
-  cscd.c                                Reimar Doeffinger
-  dca.c                                 Kostya Shishkov, Benjamin Larsson
-  dnxhd*                                Baptiste Coudurier
-  dpcm.c                                Mike Melanson
-  dxa.c                                 Kostya Shishkov
-  dv.c                                  Roman Shaposhnik
-  eacmv*, eaidct*, eat*                 Peter Ross
-  ffv1.c                                Michael Niedermayer
-  flacdec.c                             Alex Beregszaszi, Justin Ruggles
-  flacenc.c                             Justin Ruggles
-  flashsv*                              Benjamin Larsson
-  flicvideo.c                           Mike Melanson
-  g726.c                                Roman Shaposhnik
-  gifdec.c                              Baptiste Coudurier
-  h264*                                 Loren Merritt, Michael Niedermayer
-  h261*                                 Michael Niedermayer
-  h263*                                 Michael Niedermayer
-  huffyuv.c                             Michael Niedermayer
-  idcinvideo.c                          Mike Melanson
-  imc*                                  Benjamin Larsson
-  indeo2*                               Kostya Shishkov
-  interplayvideo.c                      Mike Melanson
-  jpeg_ls.c                             Kostya Shishkov
-  kmvc.c                                Kostya Shishkov
-  lcl*.c                                Roberto Togni
-  libgsm.c                              Michel Bardiaux
-  libopenjpeg.c                         Jaikrishnan Menon
-  libx264.c                             Mans Rullgard, Jason Garrett-Glaser
-  loco.c                                Kostya Shishkov
-  lzo.h, lzo.c                          Reimar Doeffinger
-  mdec.c                                Michael Niedermayer
-  mimic.c                               Ramiro Polla
-  mjpeg.c                               Michael Niedermayer
-  mmvideo.c                             Peter Ross
-  mpc*                                  Kostya Shishkov
-  mpeg12.c, mpeg12data.h                Michael Niedermayer
-  mpegvideo.c, mpegvideo.h              Michael Niedermayer
-  msmpeg4.c, msmpeg4data.h              Michael Niedermayer
-  msrle.c                               Mike Melanson
-  msvideo1.c                            Mike Melanson
-  nellymoserdec.c                       Benjamin Larsson
-  nuv.c                                 Reimar Doeffinger
-  pcx.c                                 Ivo van Poorten
-  ptx.c                                 Ivo van Poorten
-  qcelp*                                Reynaldo H. Verdejo Pinochet
-  qdm2.c, qdm2data.h                    Roberto Togni, Benjamin Larsson
-  qdrw.c                                Kostya Shishkov
-  qpeg.c                                Kostya Shishkov
-  qtrle.c                               Mike Melanson
-  ra144.c, ra144.h, ra288.c, ra288.h    Roberto Togni
-  resample2.c                           Michael Niedermayer
-  rl2.c                                 Sascha Sommer
-  rpza.c                                Roberto Togni
-  rtjpeg.c, rtjpeg.h                    Reimar Doeffinger
-  rv10.c                                Michael Niedermayer
-  rv3*                                  Kostya Shishkov
-  rv4*                                  Kostya Shishkov
-  s3tc*                                 Ivo van Poorten
-  smc.c                                 Mike Melanson
-  snow.c                                Michael Niedermayer, Loren Merritt
-  sonic.c                               Alex Beregszaszi
-  sunrast.c                             Ivo van Poorten
-  svq3.c                                Michael Niedermayer
-  targa.c                               Kostya Shishkov
-  tiff.c                                Kostya Shishkov
-  truemotion1*                          Mike Melanson
-  truemotion2*                          Kostya Shishkov
-  truespeech.c                          Kostya Shishkov
-  tscc.c                                Kostya Shishkov
-  tta.c                                 Alex Beregszaszi
-  txd.c                                 Ivo van Poorten
-  ulti*                                 Kostya Shishkov
-  vb.c                                  Kostya Shishkov
-  vc1*                                  Kostya Shishkov
-  vcr1.c                                Michael Niedermayer
-  vmnc.c                                Kostya Shishkov
-  vorbis_enc.c                          Oded Shimon
-  vorbis_dec.c                          Denes Balatoni
-  vp3*                                  Mike Melanson
-  vp5                                   Aurelien Jacobs
-  vp6                                   Aurelien Jacobs
-  vqavideo.c                            Mike Melanson
-  wavpack.c                             Kostya Shishkov
-  wmv2.c                                Michael Niedermayer
-  wnv1.c                                Kostya Shishkov
-  xan.c                                 Mike Melanson
-  xl.c                                  Kostya Shishkov
-  xvmc.c                                Ivan Kalvachev
-  zmbv*                                 Kostya Shishkov
-
-
-libavdevice
-===========
-  External Interface:
-    libavdevice/avdevice.h
-
-
-  libdc1394.c                           Roman Shaposhnik
-  v4l2.c                                Luca Abeni
-  vfwcap.c                              Ramiro Polla
-
-
-libavformat
-===========
-
-Generic parts:
-  External Interface:
-    libavformat/avformat.h              Michael Niedermayer
-  Utility Code:
-    libavformat/utils.c                 Michael Niedermayer
-
-
-Muxers/Demuxers:
-  4xm.c                                 Mike Melanson
-  adtsenc.c                             Robert Swain
-  aiff.c                                Baptiste Coudurier
-  ape.c                                 Kostya Shishkov
-  avi*                                  Michael Niedermayer
-  crc.c                                 Michael Niedermayer
-  daud.c                                Reimar Doeffinger
-  dv.c                                  Roman Shaposhnik
-  dxa.c                                 Kostya Shishkov
-  ffm*                                  Baptiste Coudurier
-  flac*                                 Justin Ruggles
-  flic.c                                Mike Melanson
-  flvdec.c, flvenc.c                    Michael Niedermayer
-  gxf.c                                 Reimar Doeffinger
-  gxfenc.c                              Baptiste Coudurier
-  idcin.c                               Mike Melanson
-  idroq.c                               Mike Melanson
-  iff.c                                 Jaikrishnan Menon
-  ipmovie.c                             Mike Melanson
-  img2.c                                Michael Niedermayer
-  iss.c                                 Stefan Gehrer
-  libnut.c                              Oded Shimon
-  lmlm4.c                               Ivo van Poorten
-  matroska.c                            Aurelien Jacobs
-  matroskaenc.c                         David Conrad
-  mm.c                                  Peter Ross
-  mov.c                                 Michael Niedermayer, Baptiste Coudurier
-  movenc.c                              Michael Niedermayer, Baptiste Coudurier
-  mpc.c                                 Kostya Shishkov
-  mpeg.c                                Michael Niedermayer
-  mpegenc.c                             Michael Niedermayer
-  mpegts*                               Mans Rullgard
-  msnwc_tcp.c                           Ramiro Polla
-  mtv.c                                 Reynaldo H. Verdejo Pinochet
-  mxf*                                  Baptiste Coudurier
-  nsvdec.c                              Francois Revol
-  nut.c                                 Michael Niedermayer
-  nuv.c                                 Reimar Doeffinger
-  oggdec.c, oggdec.h                    Mans Rullgard
-  oggenc.c                              Baptiste Coudurier
-  oggparsevorbis.c                      Mans Rullgard
-  oggparseogm.c                         Mans Rullgard
-  psxstr.c                              Mike Melanson
-  pva.c                                 Ivo van Poorten
-  r3d.c                                 Baptiste Coudurier
-  raw.c                                 Michael Niedermayer
-  rl2.c                                 Sascha Sommer
-  rm.c                                  Roberto Togni
-  rtp.c, rtpenc.c                       Luca Abeni
-  rtp_mpv.*, rtp_aac.*                  Luca Abeni
-  rtsp.c                                Luca Barbato
-  sdp.c                                 Luca Abeni
-  segafilm.c                            Mike Melanson
-  siff.c                                Kostya Shishkov
-  swf.c                                 Baptiste Coudurier
-  tta.c                                 Alex Beregszaszi
-  txd.c                                 Ivo van Poorten
-  voc.c                                 Aurelien Jacobs
-  wav.c                                 Michael Niedermayer
-  wc3movie.c                            Mike Melanson
-  westwood.c                            Mike Melanson
-  wv.c                                  Kostya Shishkov
-
-Protocols:
-  udp.c                                 Luca Abeni
-
-
-Operating systems / CPU architectures
-=====================================
-
-Alpha                                   Mans Rullgard, Falk Hueffner
-ARM                                     Mans Rullgard
-BeOS                                    Francois Revol
-Mac OS X / PowerPC                      Romain Dolbeau, Guillaume Poirier
-Amiga / PowerPC                         Colin Ward
-Linux / PowerPC                         Luca Barbato
-Windows MinGW                           Alex Beregszaszi, Ramiro Polla
-Windows Cygwin                          Victor Paesa
-ADI/Blackfin DSP                        Marc Hoffman
-Sparc                                   Roman Shaposhnik
-x86                                     Michael Niedermayer
-
-
-GnuPG Fingerprints of maintainers and others who have svn write access
-======================================================================
-
-Benoit Fouet                  B22A 4F4F 43EF 636B BB66 FCDC 0023 AE1E 2985 49C8
-Diego Biurrun                 8227 1E31 B6D9 4994 7427 E220 9CAE D6CC 4757 FCC5
-Loren Merritt                 ABD9 08F4 C920 3F65 D8BE 35D7 1540 DAA7 060F 56DE
-Michael Niedermayer           9FF2 128B 147E F673 0BAD F133 611E C787 040B 0FAB
-Peter Ross                    A907 E02F A6E5 0CD2 34CD 20D2 6760 79C5 AC40 DD6B
-Reimar Döffinger              C61D 16E5 9E2C D10C 8958 38A4 0899 A2B9 06D4 D9C7
-Reinhard Tartler              9300 5DC2 7E87 6C37 ED7B CA9A 9808 3544 9453 48A4
-Reynaldo H. Verdejo Pinochet  6E27 CD34 170C C78E 4D4F 5F40 C18E 077F 3114 452A
-Sascha Sommer                 38A0 F88B 868E 9D3A 97D4 D6A0 E823 706F 1E07 0D3C

Makefile

@@ -1,360 +0,0 @@
-include config.mak
-
-SRC_DIR = $(SRC_PATH_BARE)
-
-vpath %.texi $(SRC_PATH_BARE)
-
-PROGS-$(CONFIG_FFMPEG)   += ffmpeg
-PROGS-$(CONFIG_FFPLAY)   += ffplay
-PROGS-$(CONFIG_FFSERVER) += ffserver
-
-PROGS       = $(addsuffix   $(EXESUF), $(PROGS-yes))
-PROGS_G     = $(addsuffix _g$(EXESUF), $(PROGS-yes))
-OBJS        = $(addsuffix .o,          $(PROGS-yes)) cmdutils.o
-MANPAGES    = $(addprefix doc/, $(addsuffix .1, $(PROGS-yes)))
-
-BASENAMES   = ffmpeg ffplay ffserver
-ALLPROGS    = $(addsuffix   $(EXESUF), $(BASENAMES))
-ALLPROGS_G  = $(addsuffix _g$(EXESUF), $(BASENAMES))
-ALLMANPAGES = $(addsuffix .1, $(BASENAMES))
-
-FFLIBS-$(CONFIG_AVFILTER) += avfilter
-FFLIBS-$(CONFIG_POSTPROC) += postproc
-FFLIBS-$(CONFIG_SWSCALE)  += swscale
-
-FFLIBS := avdevice avformat avcodec avutil
-
-DATA_FILES := $(wildcard $(SRC_DIR)/ffpresets/*.ffpreset)
-
-include common.mak
-
-FF_LDFLAGS   := $(FFLDFLAGS)
-FF_EXTRALIBS := $(FFEXTRALIBS)
-FF_DEP_LIBS  := $(DEP_LIBS)
-
-ALL_TARGETS-$(CONFIG_VHOOK) += videohook
-ALL_TARGETS-$(BUILD_DOC)    += documentation
-
-INSTALL_TARGETS-$(CONFIG_VHOOK) += install-vhook
-ifneq ($(PROGS),)
-INSTALL_TARGETS-yes             += install-progs install-data
-INSTALL_TARGETS-$(BUILD_DOC)    += install-man
-endif
-INSTALL_PROGS_TARGETS-$(BUILD_SHARED) = install-libs
-
-all: $(FF_DEP_LIBS) $(PROGS) $(ALL_TARGETS-yes)
-
-$(PROGS): %$(EXESUF): %_g$(EXESUF)
-	cp -p $< $@
-	$(STRIP) $@
-
-SUBDIR_VARS := OBJS FFLIBS CLEANFILES DIRS TESTS
-
-define RESET
-$(1) :=
-$(1)-yes :=
-endef
-
-define DOSUBDIR
-$(foreach V,$(SUBDIR_VARS),$(eval $(call RESET,$(V))))
-SUBDIR := $(1)/
-include $(1)/Makefile
-endef
-
-$(foreach D,$(FFLIBS),$(eval $(call DOSUBDIR,lib$(D))))
-
-ffplay_g$(EXESUF): FF_EXTRALIBS += $(SDL_LIBS)
-ffserver_g$(EXESUF): FF_LDFLAGS += $(FFSERVERLDFLAGS)
-
-%_g$(EXESUF): %.o cmdutils.o $(FF_DEP_LIBS)
-	$(CC) $(FF_LDFLAGS) -o $@ $< cmdutils.o $(FF_EXTRALIBS)
-
-output_example$(EXESUF): output_example.o $(FF_DEP_LIBS)
-	$(CC) $(CFLAGS) $(FF_LDFLAGS) -o $@ $< $(FF_EXTRALIBS)
-
-tools/%$(EXESUF): tools/%.c
-	$(CC) $(CFLAGS) $(FF_LDFLAGS) -o $@ $< $(FF_EXTRALIBS)
-
-ffplay.o ffplay.d: CFLAGS += $(SDL_CFLAGS)
-
-cmdutils.o cmdutils.d: version.h
-
-alltools: $(addsuffix $(EXESUF),$(addprefix tools/, cws2fws pktdumper qt-faststart trasher))
-
-VHOOKCFLAGS += $(filter-out -mdynamic-no-pic,$(CFLAGS))
-
-BASEHOOKS = fish null watermark
-ALLHOOKS = $(BASEHOOKS) drawtext imlib2 ppm
-ALLHOOKS_SRCS = $(addprefix vhook/, $(addsuffix .c, $(ALLHOOKS)))
-
-HOOKS-$(HAVE_FORK)      += ppm
-HOOKS-$(HAVE_IMLIB2)    += imlib2
-HOOKS-$(HAVE_FREETYPE2) += drawtext
-
-HOOKS = $(addprefix vhook/, $(addsuffix $(SLIBSUF), $(BASEHOOKS) $(HOOKS-yes)))
-
-VHOOKCFLAGS-$(HAVE_IMLIB2) += `imlib2-config --cflags`
-LIBS_imlib2$(SLIBSUF)       = `imlib2-config --libs`
-
-VHOOKCFLAGS-$(HAVE_FREETYPE2) += `freetype-config --cflags`
-LIBS_drawtext$(SLIBSUF)        = `freetype-config --libs`
-
-VHOOKCFLAGS += $(VHOOKCFLAGS-yes)
-
-vhook/%.o vhook/%.d: CFLAGS:=$(VHOOKCFLAGS)
-
-# vhooks compile fine without libav*, but need them nonetheless.
-videohook: $(FF_DEP_LIBS) $(HOOKS)
-
-$(eval VHOOKSHFLAGS=$(VHOOKSHFLAGS))
-vhook/%$(SLIBSUF): vhook/%.o
-	$(CC) $(LDFLAGS) -o $@ $(VHOOKSHFLAGS) $< $(VHOOKLIBS) $(LIBS_$(@F))
-
-VHOOK_DEPS = $(HOOKS:$(SLIBSUF)=.d)
-depend dep: $(VHOOK_DEPS)
-
-documentation: $(addprefix doc/, ffmpeg-doc.html faq.html ffserver-doc.html \
-                                 ffplay-doc.html general.html hooks.html \
-                                 $(ALLMANPAGES))
-
-doc/%.html: doc/%.texi
-	texi2html -monolithic -number $<
-	mv $(@F) $@
-
-doc/%.pod: doc/%-doc.texi
-	doc/texi2pod.pl $< $@
-
-doc/%.1: doc/%.pod
-	pod2man --section=1 --center=" " --release=" " $< > $@
-
-install: $(INSTALL_TARGETS-yes)
-
-install-progs: $(PROGS) $(INSTALL_PROGS_TARGETS-yes)
-	install -d "$(BINDIR)"
-	install -c -m 755 $(PROGS) "$(BINDIR)"
-
-install-data: $(DATA_FILES)
-	install -d "$(DATADIR)"
-	install -m 644 $(DATA_FILES) "$(DATADIR)"
-
-install-man: $(MANPAGES)
-	install -d "$(MANDIR)/man1"
-	install -m 644 $(MANPAGES) "$(MANDIR)/man1"
-
-install-vhook: videohook
-	install -d "$(SHLIBDIR)/vhook"
-	install -m 755 $(HOOKS) "$(SHLIBDIR)/vhook"
-
-uninstall: uninstall-progs uninstall-data uninstall-man uninstall-vhook
-
-uninstall-progs:
-	rm -f $(addprefix "$(BINDIR)/", $(ALLPROGS))
-
-uninstall-data:
-	rm -rf "$(DATADIR)"
-
-uninstall-man:
-	rm -f $(addprefix "$(MANDIR)/man1/",$(ALLMANPAGES))
-
-uninstall-vhook:
-	rm -f $(addprefix "$(SHLIBDIR)/",$(ALLHOOKS_SRCS:.c=$(SLIBSUF)))
-	-rmdir "$(SHLIBDIR)/vhook/"
-
-testclean:
-	rm -rf tests/vsynth1 tests/vsynth2 tests/data tests/asynth1.sw tests/*~
-
-clean:: testclean
-	rm -f $(ALLPROGS) $(ALLPROGS_G) output_example$(EXESUF)
-	rm -f doc/*.html doc/*.pod doc/*.1
-	rm -f $(addprefix tests/,$(addsuffix $(EXESUF),audiogen videogen rotozoom seek_test tiny_psnr))
-	rm -f $(addprefix tools/,$(addsuffix $(EXESUF),cws2fws pktdumper qt-faststart trasher))
-	rm -f vhook/*.o vhook/*~ vhook/*.so vhook/*.dylib vhook/*.dll
-
-distclean::
-	rm -f version.h config.* vhook/*.d
-
-# regression tests
-
-check: test checkheaders
-
-fulltest test: codectest libavtest seektest
-
-FFMPEG_REFFILE   = $(SRC_PATH)/tests/ffmpeg.regression.ref
-FFSERVER_REFFILE = $(SRC_PATH)/tests/ffserver.regression.ref
-LIBAV_REFFILE    = $(SRC_PATH)/tests/libav.regression.ref
-ROTOZOOM_REFFILE = $(SRC_PATH)/tests/rotozoom.regression.ref
-SEEK_REFFILE     = $(SRC_PATH)/tests/seek.regression.ref
-
-CODEC_TESTS = $(addprefix regtest-,             \
-        mpeg                                    \
-        mpeg2                                   \
-        mpeg2thread                             \
-        msmpeg4v2                               \
-        msmpeg4                                 \
-        wmv1                                    \
-        wmv2                                    \
-        h261                                    \
-        h263                                    \
-        h263p                                   \
-        mpeg4                                   \
-        huffyuv                                 \
-        rc                                      \
-        mpeg4adv                                \
-        mpeg4thread                             \
-        error                                   \
-        mpeg4nr                                 \
-        mpeg1b                                  \
-        mjpeg                                   \
-        ljpeg                                   \
-        jpegls                                  \
-        rv10                                    \
-        rv20                                    \
-        asv1                                    \
-        asv2                                    \
-        flv                                     \
-        ffv1                                    \
-        snow                                    \
-        snowll                                  \
-        dv                                      \
-        dv50                                    \
-        svq1                                    \
-        flashsv                                 \
-        mp2                                     \
-        ac3                                     \
-        g726                                    \
-        adpcm_ima_wav                           \
-        adpcm_ima_qt                            \
-        adpcm_ms                                \
-        adpcm_yam                               \
-        adpcm_swf                               \
-        flac                                    \
-        wma                                     \
-        pcm                                     \
-    )
-
-LAVF_TESTS = $(addprefix regtest-,              \
-        avi                                     \
-        asf                                     \
-        rm                                      \
-        mpg                                     \
-        ts                                      \
-        swf                                     \
-        ffm                                     \
-        flv_fmt                                 \
-        mov                                     \
-        dv_fmt                                  \
-        gxf                                     \
-        nut                                     \
-        mkv                                     \
-        pbmpipe                                 \
-        pgmpipe                                 \
-        ppmpipe                                 \
-        gif                                     \
-        yuv4mpeg                                \
-        pgm                                     \
-        ppm                                     \
-        bmp                                     \
-        tga                                     \
-        tiff                                    \
-        sgi                                     \
-        jpg                                     \
-        wav                                     \
-        alaw                                    \
-        mulaw                                   \
-        au                                      \
-        mmf                                     \
-        aiff                                    \
-        voc                                     \
-        ogg                                     \
-        pixfmt                                  \
-    )
-
-REGFILES = $(addprefix tests/data/,$(addsuffix .$(1),$(2:regtest-%=%)))
-
-CODEC_ROTOZOOM = $(call REGFILES,rotozoom.regression,$(CODEC_TESTS))
-CODEC_VSYNTH   = $(call REGFILES,vsynth.regression,$(CODEC_TESTS))
-
-LAVF_REGFILES = $(call REGFILES,lavf.regression,$(LAVF_TESTS))
-
-LAVF_REG     = tests/data/lavf.regression
-ROTOZOOM_REG = tests/data/rotozoom.regression
-VSYNTH_REG   = tests/data/vsynth.regression
-
-ifneq ($(CONFIG_SWSCALE),yes)
-servertest codectest $(CODEC_TESTS) libavtest: swscale-error
-swscale-error:
-	@echo
-	@echo "This regression test requires --enable-swscale."
-	@echo
-	@exit 1
-endif
-
-ifneq ($(CONFIG_ZLIB),yes)
-regtest-flashsv codectest: zlib-error
-endif
-zlib-error:
-	@echo
-	@echo "This regression test requires zlib."
-	@echo
-	@exit 1
-
-codectest: $(VSYNTH_REG) $(ROTOZOOM_REG)
-	diff -u -w $(FFMPEG_REFFILE)   $(VSYNTH_REG)
-	diff -u -w $(ROTOZOOM_REFFILE) $(ROTOZOOM_REG)
-
-libavtest: $(LAVF_REG)
-	diff -u -w $(LIBAV_REFFILE) $(LAVF_REG)
-
-$(VSYNTH_REG) $(ROTOZOOM_REG) $(LAVF_REG):
-	cat $^ > $@
-
-$(LAVF_REG):     $(LAVF_REGFILES)
-$(ROTOZOOM_REG): $(CODEC_ROTOZOOM)
-$(VSYNTH_REG):   $(CODEC_VSYNTH)
-
-$(CODEC_VSYNTH) $(CODEC_ROTOZOOM): $(CODEC_TESTS)
-
-$(LAVF_REGFILES): $(LAVF_TESTS)
-
-$(CODEC_TESTS) $(LAVF_TESTS): regtest-ref
-
-regtest-ref: ffmpeg$(EXESUF) tests/vsynth1/00.pgm tests/vsynth2/00.pgm tests/asynth1.sw
-
-$(CODEC_TESTS) regtest-ref: tests/tiny_psnr$(EXESUF)
-	$(SRC_PATH)/tests/regression.sh $@ vsynth   tests/vsynth1 a "$(TARGET_EXEC)" "$(TARGET_PATH)"
-	$(SRC_PATH)/tests/regression.sh $@ rotozoom tests/vsynth2 a "$(TARGET_EXEC)" "$(TARGET_PATH)"
-
-$(LAVF_TESTS):
-	$(SRC_PATH)/tests/regression.sh $@ lavf tests/vsynth1 b "$(TARGET_EXEC)" "$(TARGET_PATH)"
-
-seektest: codectest libavtest tests/seek_test$(EXESUF)
-	$(SRC_PATH)/tests/seek_test.sh $(SEEK_REFFILE) "$(TARGET_EXEC)" "$(TARGET_PATH)"
-
-servertest: ffserver$(EXESUF) tests/vsynth1/00.pgm tests/asynth1.sw
-	@echo
-	@echo "Unfortunately ffserver is broken and therefore its regression"
-	@echo "test fails randomly. Treat the results accordingly."
-	@echo
-	$(SRC_PATH)/tests/server-regression.sh $(FFSERVER_REFFILE) $(SRC_PATH)/tests/test.conf
-
-tests/vsynth1/00.pgm: tests/videogen$(EXESUF)
-	mkdir -p tests/vsynth1
-	$(BUILD_ROOT)/$< 'tests/vsynth1/'
-
-tests/vsynth2/00.pgm: tests/rotozoom$(EXESUF)
-	mkdir -p tests/vsynth2
-	$(BUILD_ROOT)/$< 'tests/vsynth2/' $(SRC_PATH)/tests/lena.pnm
-
-tests/asynth1.sw: tests/audiogen$(EXESUF)
-	$(BUILD_ROOT)/$< $@
-
-tests/%$(EXESUF): tests/%.c
-	$(HOSTCC) $(HOSTCFLAGS) $(HOSTLDFLAGS) -o $@ $< $(HOSTLIBS)
-
-tests/seek_test$(EXESUF): tests/seek_test.c $(FF_DEP_LIBS)
-	$(CC) $(FF_LDFLAGS) $(CFLAGS) -o $@ $< $(FF_EXTRALIBS)
-
-
-.PHONY: lib videohook documentation *test regtest-* swscale-error zlib-error alltools check
-
--include $(VHOOK_DEPS)

README

@@ -1,12 +0,0 @@
-FFmpeg README
--------------
-
-1) Documentation
-----------------
-
-* Read the documentation in the doc/ directory.
-
-2) Licensing
-------------
-
-* See the LICENSE file.

README.md

@@ -0,0 +1,2 @@
+# FFmpeg
+Unofficial mirror of the FFmpeg repository on GitHub. Primary goal is to provide easily reproducible builds that run on Windows.

RELEASE

@@ -1,254 +0,0 @@
-Release Notes
-=============
-
-* 0.5 "Bike Shed World Domination" March 3, 2009
-
-General notes
--------------
-
-It has been so long since the last release that this should be considered the
-first FFmpeg release of recent times. Because of the way things have unfolded to
-date, the notes for this version cannot be entirely conventional.
-
-See the Changelog file for a list of significant changes.
-
-Please note that our policy on bug reports has not changed. We still only accept
-bug reports against HEAD of the FFmpeg trunk repository. If you are experiencing
-any issues with any formally released version of FFmpeg, please try a current
-version of the development code to check if the issue still exists. If it does,
-make your report against the development code following the usual bug reporting
-guidelines.
-
-API notes
----------
-
-In the next release, it is intended to remove a number of deprecated APIs. We
-decided to put out a release that includes said APIs for the benefit of third
-party software.
-
-As such, this release:
-- provides a sync point for said APIs
-- increases awareness of API changes
-- allows the next release to detail how to transition from the old to the new
-
-The deprecated APIs to be removed are:
-- imgconvert (to be replaced by libswscale)
-- vhook (to be replaced by libavfilter)
-
-If at all possible, do not use the deprecated APIs. All notes on API changes
-should appear in doc/APIchanges.
-
-
-
-* 0.5.1 March 2, 2010
-
-General notes
--------------
-
-This point release includes some minor updates to make the 0.5 release series
-usable for users that need to retain the existing behavior as closely as
-possible. The changes follow below:
-
-Security fixes
---------------
-
-Various programming errors in container and codec implementations
-may lead to denial of service or the execution of arbitrary code
-if the user is tricked into opening a malformed media file or stream.
-
-Affected and updated have been the implementations of the following
-codecs and container formats:
-
- - the Vorbis audio codec
- - the FF Video 1 codec
- - the MPEG audio codec
- - the H264 video codec
- - the MLP codec
- - the HuffYUV codec
- - the ASF demuxer
- - the Ogg container implementation
- - the MOV container implementation
-
-Symbol Versioning enabled
--------------------------
-
-The backported symbol versioning change is enabled on platforms that support
-it. This allows users to upgrade from 0.5.1 to the upcoming 0.6 release
-without having to recompile their applications. Please note that distributors
-have to recompile applications against 0.5.1 before upgrading to 0.6.
-
-libx264.c backport
-------------------
-
-This release includes a backport to the libx264 wrapper that allows FFmpeg to
-be compiled against newer versions of libx264 up to API version 85.
-
-licensing changes
------------------
-
-Previously both libswscale and our AC-3 decoder had GPLed parts. These have
-been replaced by fresh LGPL code. x86 optimizations for libswscale remain GPL,
-but the C code is fully functional. Optimizations for other architectures have
-been relicensed to LGPL.
-
-AMR-NB decoding/encoding and AMR-WB decoding is now possible through the free
-software OpenCORE libraries as an alternative to the non-free libamr libraries.
-
-We found out that libfaac contains non-free parts and is not LGPL as previously
-claimed. We have changed configure to reflect this. You now have to pass the
---enable-nonfree option if you wish to compile with libfaac support enabled.
-
-Furthermore the non-free bits in libavcodec/fdctref.c have been rewritten. Note
-well that they were only used in a test program and never compiled into any
-FFmpeg library.
-
-
-
-* 0.5.2 May 25, 2010
-
-General notes
--------------
-
-This is a maintenance-only release that addresses a small number of security
-and portability issues. Distributors and system integrators are encouraged
-to update and share their patches against this branch.
-
-
-
-* 0.5.3 Oct 18, 2010
-
-General notes
--------------
-
-This is (again) another maintenance-only release that addresses a fix
-for seekable HTTP and an exploitable bug in the FLIC decoder
-(cf. CVE-2010-3429 for details). Distributors and system integrators are
-encouraged to update and share their patches against this branch.
-
-
-
-* 0.5.4 Mar 17, 2011
-
-General notes
--------------
-
-This is the first release that we cut after git migration. It is another
-maintenance-only release that addresses several security issues that were
-brought to our attention. In detail, fixes for RV30/40, WMV, Vorbis and
-VC-1 have been backported from trunk. Distributors and system integrators
-are encouraged to update and share their patches against this branch.
-
-
-
-* 0.5.5 Nov 6, 2011
-
-General notes
--------------
-
-This maintenance-only release addresses several security issues that
-were brought to our attention. In detail, fixes for the MJPEG decoder,
-the CAVS decoder (CVE-2011-3362, CVE-2011-3973, CVE-2011-3974), and the
-Matroska decoder (MSVR11-011/CVE-2011-3504) and many others have been
-corrected. Additional, this release contains fixes for compilation with
-gcc-4.6. Distributors and system integrators are encouraged to update
-and share their patches against this branch.
-
-
-
-* 0.5.6 Nov 21, 2011
-
-General notes
--------------
-
-This maintenance-only release addresses several security issues that
-were brought to our attention.
-
-
-
-* 0.5.7 Dec 25, 2011
-
-General notes
--------------
-
-This maintenance-only release addresses several security issues that
-were brought to our attention. In details, it features fixes for the
-QDM2 decoder (CVE-2011-4351), DoS in the VP5/VP6 decoders
-(CVE-2011-4353), and a buffer overflow in the Sierra VMD decoder
-CVE-2011-4364, and a safety fix in the SVQ1 decoder (CVE-2011-4579).
-CVE-2011-4352, a bug in the VP3 decoder, is not known to affect this
-release.
-
-Distributors and system integrators are encouraged to update and share
-their patches against this branch.
-
-
-
-* 0.5.8 Jan 12, 2012
-
-General notes
--------------
-
-This mostly maintenance-only release that addresses a number a number of
-bugs such as security and compilation issues that have been brought to
-our attention. Among other (rather minor) fixes, this release features
-fixes for the VP3 decoder (CVE-2011-3892), vorbis decoder, and matroska
-demuxer (CVE-2011-3893 and CVE-2011-3895).
-
-Distributors and system integrators are encouraged
-to update and share their patches against this branch.  For a full list
-of changes please see the Changelog file.
-
-
-
-* 0.5.9 May 11, 2012
-
-General notes
--------------
-
-This maintenance-only release that addresses a number a number of
-security issues that have been brought to our attention. Among other
-(rather minor) fixes, this release features fixes for the DV decoder
-(CVE-2011-3929 and CVE-2011-3936), nsvdec (CVE-2011-3940), Atrac3
-(CVE-2012-0853), mjpegdec (CVE-2011-3947) and the VQA video decoder
-(CVE-2012-0947).
-
-Distributors and system integrators are encouraged
-to update and share their patches against this branch.  For a full list
-of changes please see the Changelog file.
-
-
-
-* 0.5.10 Jun 09, 2012
-
-General notes
--------------
-
-This mostly maintenance-only release addresses a number of bugs such as
-security and compilation issues that have been brought to our
-attention. Among other fixes, this release includes security updates for
-the DPCM codecs (CVE-2011-3951), H.264 (CVE-2012-0851), ADPCM
-(CVE-2012-0852), and the KMVC decoder (CVE-2011-3952).
-
-Distributors and system integrators are encouraged to update and share
-their patches against this branch. For a full list of changes please see
-the Changelog file or the Git commit history.
-
-
-
-* 0.5.11 Feb 17, 2013
-
-General notes
--------------
-
-This maintenance-only release addresses a number of bugs such as
-security and compilation issues that have been brought to our
-attention. Among other fixes, this release includes security updates for
-the mpeg12 codecs (CVE-2012-2803), H.264, VP5/VP6 (CVE-2012-2783,
-CVE-2012-2783), shorten (CVE-2012-0858), CAVS (CVE-2012-2777 and
-CVE-2012-2784), AVS (CVE-2012-2801) and a number of additional safe but
-important bugs in other decoders. Additionally, reported bugs in the
-yuv4mpeg (Bug 373) and BMP decoder (Bug 367) have been addressed.
-
-Distributors and system integrators are encouraged to update and share
-their patches against this branch. For a full list of changes please
-see the Changelog file or the Git commit history.

VERSION

@@ -1 +0,0 @@
-0.5.15

cmdutils.c

@@ -1,500 +0,0 @@
-/*
- * Various utilities for command line tools
- * Copyright (c) 2000-2003 Fabrice Bellard
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#include <string.h>
-#include <stdlib.h>
-#include <errno.h>
-#include <math.h>
-
-/* Include only the enabled headers since some compilers (namely, Sun
-   Studio) will not omit unused inline functions and create undefined
-   references to libraries that are not being built. */
-
-#include "config.h"
-#include "libavformat/avformat.h"
-#include "libavfilter/avfilter.h"
-#include "libavdevice/avdevice.h"
-#include "libswscale/swscale.h"
-#include "libpostproc/postprocess.h"
-#include "libavutil/avstring.h"
-#include "libavcodec/opt.h"
-#include "cmdutils.h"
-#include "version.h"
-#if CONFIG_NETWORK
-#include "libavformat/network.h"
-#endif
-
-#undef exit
-
-const char **opt_names;
-static int opt_name_count;
-AVCodecContext *avctx_opts[CODEC_TYPE_NB];
-AVFormatContext *avformat_opts;
-struct SwsContext *sws_opts;
-
-const int this_year = 2014;
-
-double parse_number_or_die(const char *context, const char *numstr, int type, double min, double max)
-{
-    char *tail;
-    const char *error;
-    double d = strtod(numstr, &tail);
-    if (*tail)
-        error= "Expected number for %s but found: %s\n";
-    else if (d < min || d > max)
-        error= "The value for %s was %s which is not within %f - %f\n";
-    else if(type == OPT_INT64 && (int64_t)d != d)
-        error= "Expected int64 for %s but found %s\n";
-    else
-        return d;
-    fprintf(stderr, error, context, numstr, min, max);
-    exit(1);
-}
-
-int64_t parse_time_or_die(const char *context, const char *timestr, int is_duration)
-{
-    int64_t us = parse_date(timestr, is_duration);
-    if (us == INT64_MIN) {
-        fprintf(stderr, "Invalid %s specification for %s: %s\n",
-                is_duration ? "duration" : "date", context, timestr);
-        exit(1);
-    }
-    return us;
-}
-
-void show_help_options(const OptionDef *options, const char *msg, int mask, int value)
-{
-    const OptionDef *po;
-    int first;
-
-    first = 1;
-    for(po = options; po->name != NULL; po++) {
-        char buf[64];
-        if ((po->flags & mask) == value) {
-            if (first) {
-                printf("%s", msg);
-                first = 0;
-            }
-            av_strlcpy(buf, po->name, sizeof(buf));
-            if (po->flags & HAS_ARG) {
-                av_strlcat(buf, " ", sizeof(buf));
-                av_strlcat(buf, po->argname, sizeof(buf));
-            }
-            printf("-%-17s  %s\n", buf, po->help);
-        }
-    }
-}
-
-static const OptionDef* find_option(const OptionDef *po, const char *name){
-    while (po->name != NULL) {
-        if (!strcmp(name, po->name))
-            break;
-        po++;
-    }
-    return po;
-}
-
-void parse_options(int argc, char **argv, const OptionDef *options,
-                   void (* parse_arg_function)(const char*))
-{
-    const char *opt, *arg;
-    int optindex, handleoptions=1;
-    const OptionDef *po;
-
-    /* parse options */
-    optindex = 1;
-    while (optindex < argc) {
-        opt = argv[optindex++];
-
-        if (handleoptions && opt[0] == '-' && opt[1] != '\0') {
-            if (opt[1] == '-' && opt[2] == '\0') {
-                handleoptions = 0;
-                continue;
-            }
-            po= find_option(options, opt + 1);
-            if (!po->name)
-                po= find_option(options, "default");
-            if (!po->name) {
-unknown_opt:
-                fprintf(stderr, "%s: unrecognized option '%s'\n", argv[0], opt);
-                exit(1);
-            }
-            arg = NULL;
-            if (po->flags & HAS_ARG) {
-                arg = argv[optindex++];
-                if (!arg) {
-                    fprintf(stderr, "%s: missing argument for option '%s'\n", argv[0], opt);
-                    exit(1);
-                }
-            }
-            if (po->flags & OPT_STRING) {
-                char *str;
-                str = av_strdup(arg);
-                *po->u.str_arg = str;
-            } else if (po->flags & OPT_BOOL) {
-                *po->u.int_arg = 1;
-            } else if (po->flags & OPT_INT) {
-                *po->u.int_arg = parse_number_or_die(opt+1, arg, OPT_INT64, INT_MIN, INT_MAX);
-            } else if (po->flags & OPT_INT64) {
-                *po->u.int64_arg = parse_number_or_die(opt+1, arg, OPT_INT64, INT64_MIN, INT64_MAX);
-            } else if (po->flags & OPT_FLOAT) {
-                *po->u.float_arg = parse_number_or_die(opt+1, arg, OPT_FLOAT, -1.0/0.0, 1.0/0.0);
-            } else if (po->flags & OPT_FUNC2) {
-                if(po->u.func2_arg(opt+1, arg)<0)
-                    goto unknown_opt;
-            } else {
-                po->u.func_arg(arg);
-            }
-            if(po->flags & OPT_EXIT)
-                exit(0);
-        } else {
-            if (parse_arg_function)
-                parse_arg_function(opt);
-        }
-    }
-}
-
-int opt_default(const char *opt, const char *arg){
-    int type;
-    int ret= 0;
-    const AVOption *o= NULL;
-    int opt_types[]={AV_OPT_FLAG_VIDEO_PARAM, AV_OPT_FLAG_AUDIO_PARAM, 0, AV_OPT_FLAG_SUBTITLE_PARAM, 0};
-
-    for(type=0; type<CODEC_TYPE_NB && ret>= 0; type++){
-        const AVOption *o2 = av_find_opt(avctx_opts[0], opt, NULL, opt_types[type], opt_types[type]);
-        if(o2)
-            ret = av_set_string3(avctx_opts[type], opt, arg, 1, &o);
-    }
-    if(!o)
-        ret = av_set_string3(avformat_opts, opt, arg, 1, &o);
-    if(!o)
-        ret = av_set_string3(sws_opts, opt, arg, 1, &o);
-    if(!o){
-        if(opt[0] == 'a')
-            ret = av_set_string3(avctx_opts[CODEC_TYPE_AUDIO], opt+1, arg, 1, &o);
-        else if(opt[0] == 'v')
-            ret = av_set_string3(avctx_opts[CODEC_TYPE_VIDEO], opt+1, arg, 1, &o);
-        else if(opt[0] == 's')
-            ret = av_set_string3(avctx_opts[CODEC_TYPE_SUBTITLE], opt+1, arg, 1, &o);
-    }
-    if (o && ret < 0) {
-        fprintf(stderr, "Invalid value '%s' for option '%s'\n", arg, opt);
-        exit(1);
-    }
-    if(!o)
-        return -1;
-
-//    av_log(NULL, AV_LOG_ERROR, "%s:%s: %f 0x%0X\n", opt, arg, av_get_double(avctx_opts, opt, NULL), (int)av_get_int(avctx_opts, opt, NULL));
-
-    //FIXME we should always use avctx_opts, ... for storing options so there will not be any need to keep track of what i set over this
-    opt_names= av_realloc(opt_names, sizeof(void*)*(opt_name_count+1));
-    opt_names[opt_name_count++]= o->name;
-
-    if(avctx_opts[0]->debug || avformat_opts->debug)
-        av_log_set_level(AV_LOG_DEBUG);
-    return 0;
-}
-
-void set_context_opts(void *ctx, void *opts_ctx, int flags)
-{
-    int i;
-    for(i=0; i<opt_name_count; i++){
-        char buf[256];
-        const AVOption *opt;
-        const char *str= av_get_string(opts_ctx, opt_names[i], &opt, buf, sizeof(buf));
-        /* if an option with name opt_names[i] is present in opts_ctx then str is non-NULL */
-        if(str && ((opt->flags & flags) == flags))
-            av_set_string3(ctx, opt_names[i], str, 1, NULL);
-    }
-}
-
-void print_error(const char *filename, int err)
-{
-    switch(err) {
-    case AVERROR_NUMEXPECTED:
-        fprintf(stderr, "%s: Incorrect image filename syntax.\n"
-                "Use '%%d' to specify the image number:\n"
-                "  for img1.jpg, img2.jpg, ..., use 'img%%d.jpg';\n"
-                "  for img001.jpg, img002.jpg, ..., use 'img%%03d.jpg'.\n",
-                filename);
-        break;
-    case AVERROR_INVALIDDATA:
-        fprintf(stderr, "%s: Error while parsing header\n", filename);
-        break;
-    case AVERROR_NOFMT:
-        fprintf(stderr, "%s: Unknown format\n", filename);
-        break;
-    case AVERROR(EIO):
-        fprintf(stderr, "%s: I/O error occurred\n"
-                "Usually that means that input file is truncated and/or corrupted.\n",
-                filename);
-        break;
-    case AVERROR(ENOMEM):
-        fprintf(stderr, "%s: memory allocation error occurred\n", filename);
-        break;
-    case AVERROR(ENOENT):
-        fprintf(stderr, "%s: no such file or directory\n", filename);
-        break;
-#if CONFIG_NETWORK
-    case AVERROR(FF_NETERROR(EPROTONOSUPPORT)):
-        fprintf(stderr, "%s: Unsupported network protocol\n", filename);
-        break;
-#endif
-    default:
-        fprintf(stderr, "%s: Error while opening file\n", filename);
-        break;
-    }
-}
-
-#define PRINT_LIB_VERSION(outstream,libname,LIBNAME,indent) \
-    version= libname##_version(); \
-    fprintf(outstream, "%slib%-10s %2d.%2d.%2d / %2d.%2d.%2d\n", indent? "  " : "", #libname, \
-            LIB##LIBNAME##_VERSION_MAJOR, LIB##LIBNAME##_VERSION_MINOR, LIB##LIBNAME##_VERSION_MICRO, \
-            version >> 16, version >> 8 & 0xff, version & 0xff);
-
-static void print_all_lib_versions(FILE* outstream, int indent)
-{
-    unsigned int version;
-    PRINT_LIB_VERSION(outstream, avutil,   AVUTIL,   indent);
-    PRINT_LIB_VERSION(outstream, avcodec,  AVCODEC,  indent);
-    PRINT_LIB_VERSION(outstream, avformat, AVFORMAT, indent);
-    PRINT_LIB_VERSION(outstream, avdevice, AVDEVICE, indent);
-#if CONFIG_AVFILTER
-    PRINT_LIB_VERSION(outstream, avfilter, AVFILTER, indent);
-#endif
-#if CONFIG_SWSCALE
-    PRINT_LIB_VERSION(outstream, swscale,  SWSCALE,  indent);
-#endif
-#if CONFIG_POSTPROC
-    PRINT_LIB_VERSION(outstream, postproc, POSTPROC, indent);
-#endif
-}
-
-void show_banner(void)
-{
-    fprintf(stderr, "%s version " FFMPEG_VERSION ", Copyright (c) %d-%d Fabrice Bellard, et al.\n",
-            program_name, program_birth_year, this_year);
-    fprintf(stderr, "  configuration: " FFMPEG_CONFIGURATION "\n");
-    print_all_lib_versions(stderr, 1);
-    fprintf(stderr, "  built on " __DATE__ " " __TIME__);
-#ifdef __GNUC__
-    fprintf(stderr, ", gcc: " __VERSION__ "\n");
-#else
-    fprintf(stderr, ", using a non-gcc compiler\n");
-#endif
-}
-
-void show_version(void) {
-    printf("%s " FFMPEG_VERSION "\n", program_name);
-    print_all_lib_versions(stdout, 0);
-}
-
-void show_license(void)
-{
-    printf(
-#if CONFIG_NONFREE
-    "This version of %s has nonfree parts compiled in.\n"
-    "Therefore it is not legally redistributable.\n",
-    program_name
-#elif CONFIG_GPLV3
-    "%s is free software; you can redistribute it and/or modify\n"
-    "it under the terms of the GNU General Public License as published by\n"
-    "the Free Software Foundation; either version 3 of the License, or\n"
-    "(at your option) any later version.\n"
-    "\n"
-    "%s is distributed in the hope that it will be useful,\n"
-    "but WITHOUT ANY WARRANTY; without even the implied warranty of\n"
-    "MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the\n"
-    "GNU General Public License for more details.\n"
-    "\n"
-    "You should have received a copy of the GNU General Public License\n"
-    "along with %s.  If not, see <http://www.gnu.org/licenses/>.\n",
-    program_name, program_name, program_name
-#elif CONFIG_GPL
-    "%s is free software; you can redistribute it and/or modify\n"
-    "it under the terms of the GNU General Public License as published by\n"
-    "the Free Software Foundation; either version 2 of the License, or\n"
-    "(at your option) any later version.\n"
-    "\n"
-    "%s is distributed in the hope that it will be useful,\n"
-    "but WITHOUT ANY WARRANTY; without even the implied warranty of\n"
-    "MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the\n"
-    "GNU General Public License for more details.\n"
-    "\n"
-    "You should have received a copy of the GNU General Public License\n"
-    "along with %s; if not, write to the Free Software\n"
-    "Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA\n",
-    program_name, program_name, program_name
-#elif CONFIG_LGPLV3
-    "%s is free software; you can redistribute it and/or modify\n"
-    "it under the terms of the GNU Lesser General Public License as published by\n"
-    "the Free Software Foundation; either version 3 of the License, or\n"
-    "(at your option) any later version.\n"
-    "\n"
-    "%s is distributed in the hope that it will be useful,\n"
-    "but WITHOUT ANY WARRANTY; without even the implied warranty of\n"
-    "MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the\n"
-    "GNU Lesser General Public License for more details.\n"
-    "\n"
-    "You should have received a copy of the GNU Lesser General Public License\n"
-    "along with %s.  If not, see <http://www.gnu.org/licenses/>.\n",
-    program_name, program_name, program_name
-#else
-    "%s is free software; you can redistribute it and/or\n"
-    "modify it under the terms of the GNU Lesser General Public\n"
-    "License as published by the Free Software Foundation; either\n"
-    "version 2.1 of the License, or (at your option) any later version.\n"
-    "\n"
-    "%s is distributed in the hope that it will be useful,\n"
-    "but WITHOUT ANY WARRANTY; without even the implied warranty of\n"
-    "MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU\n"
-    "Lesser General Public License for more details.\n"
-    "\n"
-    "You should have received a copy of the GNU Lesser General Public\n"
-    "License along with %s; if not, write to the Free Software\n"
-    "Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA\n",
-    program_name, program_name, program_name
-#endif
-    );
-}
-
-void show_formats(void)
-{
-    AVInputFormat *ifmt=NULL;
-    AVOutputFormat *ofmt=NULL;
-    URLProtocol *up=NULL;
-    AVCodec *p=NULL, *p2;
-    AVBitStreamFilter *bsf=NULL;
-    const char *last_name;
-
-    printf("File formats:\n");
-    last_name= "000";
-    for(;;){
-        int decode=0;
-        int encode=0;
-        const char *name=NULL;
-        const char *long_name=NULL;
-
-        while((ofmt= av_oformat_next(ofmt))) {
-            if((name == NULL || strcmp(ofmt->name, name)<0) &&
-                strcmp(ofmt->name, last_name)>0){
-                name= ofmt->name;
-                long_name= ofmt->long_name;
-                encode=1;
-            }
-        }
-        while((ifmt= av_iformat_next(ifmt))) {
-            if((name == NULL || strcmp(ifmt->name, name)<0) &&
-                strcmp(ifmt->name, last_name)>0){
-                name= ifmt->name;
-                long_name= ifmt->long_name;
-                encode=0;
-            }
-            if(name && strcmp(ifmt->name, name)==0)
-                decode=1;
-        }
-        if(name==NULL)
-            break;
-        last_name= name;
-
-        printf(
-            " %s%s %-15s %s\n",
-            decode ? "D":" ",
-            encode ? "E":" ",
-            name,
-            long_name ? long_name:" ");
-    }
-    printf("\n");
-
-    printf("Codecs:\n");
-    last_name= "000";
-    for(;;){
-        int decode=0;
-        int encode=0;
-        int cap=0;
-        const char *type_str;
-
-        p2=NULL;
-        while((p= av_codec_next(p))) {
-            if((p2==NULL || strcmp(p->name, p2->name)<0) &&
-                strcmp(p->name, last_name)>0){
-                p2= p;
-                decode= encode= cap=0;
-            }
-            if(p2 && strcmp(p->name, p2->name)==0){
-                if(p->decode) decode=1;
-                if(p->encode) encode=1;
-                cap |= p->capabilities;
-            }
-        }
-        if(p2==NULL)
-            break;
-        last_name= p2->name;
-
-        switch(p2->type) {
-        case CODEC_TYPE_VIDEO:
-            type_str = "V";
-            break;
-        case CODEC_TYPE_AUDIO:
-            type_str = "A";
-            break;
-        case CODEC_TYPE_SUBTITLE:
-            type_str = "S";
-            break;
-        default:
-            type_str = "?";
-            break;
-        }
-        printf(
-            " %s%s%s%s%s%s %-15s %s",
-            decode ? "D": (/*p2->decoder ? "d":*/" "),
-            encode ? "E":" ",
-            type_str,
-            cap & CODEC_CAP_DRAW_HORIZ_BAND ? "S":" ",
-            cap & CODEC_CAP_DR1 ? "D":" ",
-            cap & CODEC_CAP_TRUNCATED ? "T":" ",
-            p2->name,
-            p2->long_name ? p2->long_name : "");
-       /* if(p2->decoder && decode==0)
-            printf(" use %s for decoding", p2->decoder->name);*/
-        printf("\n");
-    }
-    printf("\n");
-
-    printf("Bitstream filters:\n");
-    while((bsf = av_bitstream_filter_next(bsf)))
-        printf(" %s", bsf->name);
-    printf("\n");
-
-    printf("Supported file protocols:\n");
-    while((up = av_protocol_next(up)))
-        printf(" %s:", up->name);
-    printf("\n");
-
-    printf("Frame size, frame rate abbreviations:\n ntsc pal qntsc qpal sntsc spal film ntsc-film sqcif qcif cif 4cif\n");
-    printf("\n");
-    printf(
-"Note, the names of encoders and decoders do not always match, so there are\n"
-"several cases where the above table shows encoder only or decoder only entries\n"
-"even though both encoding and decoding are supported. For example, the h263\n"
-"decoder corresponds to the h263 and h263p encoders, for file formats it is even\n"
-"worse.\n");
-}

cmdutils.h

@@ -1,155 +0,0 @@
-/*
- * Various utilities for command line tools
- * copyright (c) 2003 Fabrice Bellard
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#ifndef FFMPEG_CMDUTILS_H
-#define FFMPEG_CMDUTILS_H
-
-#include <inttypes.h>
-#include "libavcodec/avcodec.h"
-#include "libavformat/avformat.h"
-#include "libswscale/swscale.h"
-
-/**
- * program name, defined by the program for show_version().
- */
-extern const char program_name[];
-
-/**
- * program birth year, defined by the program for show_banner()
- */
-extern const int program_birth_year;
-
-extern const int this_year;
-
-extern const char **opt_names;
-extern AVCodecContext *avctx_opts[CODEC_TYPE_NB];
-extern AVFormatContext *avformat_opts;
-extern struct SwsContext *sws_opts;
-
-/**
- * Fallback for options that are not explicitly handled, these will be
- * parsed through AVOptions.
- */
-int opt_default(const char *opt, const char *arg);
-
-/**
- * Parses a string and returns its corresponding value as a double.
- * Exits from the application if the string cannot be correctly
- * parsed or the corresponding value is invalid.
- *
- * @param context the context of the value to be set (e.g. the
- * corresponding commandline option name)
- * @param numstr the string to be parsed
- * @param type the type (OPT_INT64 or OPT_FLOAT) as which the
- * string should be parsed
- * @param min the minimum valid accepted value
- * @param max the maximum valid accepted value
- */
-double parse_number_or_die(const char *context, const char *numstr, int type, double min, double max);
-
-/**
- * Parses a string specifying a time and returns its corresponding
- * value as a number of microseconds. Exits from the application if
- * the string cannot be correctly parsed.
- *
- * @param context the context of the value to be set (e.g. the
- * corresponding commandline option name)
- * @param timestr the string to be parsed
- * @param is_duration a flag which tells how to interpret \p timestr, if
- * not zero \p timestr is interpreted as a duration, otherwise as a
- * date
- *
- * @see parse_date()
- */
-int64_t parse_time_or_die(const char *context, const char *timestr, int is_duration);
-
-typedef struct {
-    const char *name;
-    int flags;
-#define HAS_ARG    0x0001
-#define OPT_BOOL   0x0002
-#define OPT_EXPERT 0x0004
-#define OPT_STRING 0x0008
-#define OPT_VIDEO  0x0010
-#define OPT_AUDIO  0x0020
-#define OPT_GRAB   0x0040
-#define OPT_INT    0x0080
-#define OPT_FLOAT  0x0100
-#define OPT_SUBTITLE 0x0200
-#define OPT_FUNC2  0x0400
-#define OPT_INT64  0x0800
-#define OPT_EXIT   0x1000
-     union {
-        void (*func_arg)(const char *); //FIXME passing error code as int return would be nicer then exit() in the func
-        int *int_arg;
-        char **str_arg;
-        float *float_arg;
-        int (*func2_arg)(const char *, const char *);
-        int64_t *int64_arg;
-    } u;
-    const char *help;
-    const char *argname;
-} OptionDef;
-
-void show_help_options(const OptionDef *options, const char *msg, int mask, int value);
-
-/**
- * Parses the command line arguments.
- * @param options Array with the definitions required to interpret every
- * option of the form: -<option_name> [<argument>]
- * @param parse_arg_function Name of the function called to process every
- * argument without a leading option name flag. NULL if such arguments do
- * not have to be processed.
- */
-void parse_options(int argc, char **argv, const OptionDef *options,
-                   void (* parse_arg_function)(const char*));
-
-void set_context_opts(void *ctx, void *opts_ctx, int flags);
-
-void print_error(const char *filename, int err);
-
-/**
- * Prints the program banner to stderr. The banner contents depend on the
- * current version of the repository and of the libav* libraries used by
- * the program.
- */
-void show_banner(void);
-
-/**
- * Prints the version of the program to stdout. The version message
- * depends on the current versions of the repository and of the libav*
- * libraries.
- */
-void show_version(void);
-
-/**
- * Prints the license of the program to stdout. The license depends on
- * the license of the libraries compiled into the program.
- */
-void show_license(void);
-
-/**
- * Prints a listing containing all the formats supported by the
- * program.
- */
-void show_formats(void);
-
-#endif /* FFMPEG_CMDUTILS_H */

common.mak

@@ -1,117 +0,0 @@
-#
-# common bits used by all libraries
-#
-
-all: # make "all" default target
-
-ifndef SUBDIR
-vpath %.c $(SRC_DIR)
-vpath %.h $(SRC_DIR)
-vpath %.S $(SRC_DIR)
-vpath %.asm $(SRC_DIR)
-vpath %.v   $(SRC_DIR)
-
-ifeq ($(SRC_DIR),$(SRC_PATH_BARE))
-BUILD_ROOT_REL = .
-else
-BUILD_ROOT_REL = ..
-endif
-
-ALLFFLIBS = avcodec avdevice avfilter avformat avutil postproc swscale
-
-CFLAGS := -DHAVE_AV_CONFIG_H -D_FILE_OFFSET_BITS=64 -D_LARGEFILE_SOURCE \
-          -I$(BUILD_ROOT_REL) -I$(SRC_PATH) $(OPTFLAGS)
-
-%.o: %.c
-	$(CC) $(CFLAGS) $(LIBOBJFLAGS) -c -o $@ $<
-
-%.o: %.S
-	$(CC) $(CFLAGS) $(LIBOBJFLAGS) -c -o $@ $<
-
-%.ho: %.h
-	$(CC) $(CFLAGS) $(LIBOBJFLAGS) -Wno-unused -c -o $@ -x c $<
-
-%.d: %.c
-	$(DEPEND_CMD) > $@
-
-%.d: %.S
-	$(DEPEND_CMD) > $@
-
-%.d: %.cpp
-	$(DEPEND_CMD) > $@
-
-%.o: %.d
-
-%$(EXESUF): %.c
-
-%.ver: %.v
-	sed 's/$$MAJOR/$($(basename $(@F))_VERSION_MAJOR)/' $^ > $@
-
-SVN_ENTRIES = $(SRC_PATH_BARE)/.svn/entries
-ifeq ($(wildcard $(SVN_ENTRIES)),$(SVN_ENTRIES))
-$(BUILD_ROOT_REL)/version.h: $(SVN_ENTRIES)
-endif
-
-$(BUILD_ROOT_REL)/version.h: $(SRC_PATH_BARE)/version.sh
-	$< $(SRC_PATH) $@ $(EXTRA_VERSION)
-
-install: install-libs install-headers
-
-uninstall: uninstall-libs uninstall-headers
-
-.PHONY: all depend dep clean distclean install* uninstall* tests
-endif
-
-CFLAGS   += $(CFLAGS-yes)
-OBJS     += $(OBJS-yes)
-FFLIBS   := $(FFLIBS-yes) $(FFLIBS)
-TESTS    += $(TESTS-yes)
-
-FFEXTRALIBS := $(addprefix -l,$(addsuffix $(BUILDSUF),$(FFLIBS))) $(EXTRALIBS)
-FFLDFLAGS   := $(addprefix -L$(BUILD_ROOT)/lib,$(FFLIBS)) $(LDFLAGS)
-
-OBJS  := $(addprefix $(SUBDIR),$(OBJS))
-TESTS := $(addprefix $(SUBDIR),$(TESTS))
-
-DEP_LIBS:=$(foreach NAME,$(FFLIBS),lib$(NAME)/$($(BUILD_SHARED:yes=S)LIBNAME))
-
-ALLHEADERS := $(subst $(SRC_DIR)/,$(SUBDIR),$(wildcard $(SRC_DIR)/*.h $(SRC_DIR)/$(ARCH)/*.h))
-checkheaders: $(filter-out %_template.ho,$(ALLHEADERS:.h=.ho))
-
-DEPS := $(OBJS:.o=.d)
-depend dep: $(DEPS)
-
-CLEANSUFFIXES = *.o *~ *.ho *.ver
-LIBSUFFIXES   = *.a *.lib *.so *.so.* *.dylib *.dll *.def *.dll.a *.exp *.map
-DISTCLEANSUFFIXES = *.d *.pc
-
-define RULES
-$(SUBDIR)%$(EXESUF): $(SUBDIR)%.o
-	$(CC) $(FFLDFLAGS) -o $$@ $$^ $(SUBDIR)$(LIBNAME) $(FFEXTRALIBS)
-
-$(SUBDIR)%-test.o: $(SUBDIR)%.c
-	$(CC) $(CFLAGS) -DTEST -c -o $$@ $$^
-
-$(SUBDIR)%-test.o: $(SUBDIR)%-test.c
-	$(CC) $(CFLAGS) -DTEST -c -o $$@ $$^
-
-$(SUBDIR)x86/%.o: $(SUBDIR)x86/%.asm
-	$(YASM) $(YASMFLAGS) -I $$(<D)/ -o $$@ $$<
-
-$(SUBDIR)x86/%.d: $(SUBDIR)x86/%.asm
-	$(YASM) $(YASMFLAGS) -I $$(<D)/ -M -o $$(@:%.d=%.o) $$< > $$@
-
-clean::
-	rm -f $(TESTS) $(addprefix $(SUBDIR),$(CLEANFILES) $(CLEANSUFFIXES) $(LIBSUFFIXES)) \
-	    $(addprefix $(SUBDIR), $(foreach suffix,$(CLEANSUFFIXES),$(addsuffix /$(suffix),$(DIRS))))
-
-distclean:: clean
-	rm -f  $(addprefix $(SUBDIR),$(DISTCLEANSUFFIXES)) \
-            $(addprefix $(SUBDIR), $(foreach suffix,$(DISTCLEANSUFFIXES),$(addsuffix /$(suffix),$(DIRS))))
-endef
-
-$(eval $(RULES))
-
-tests: $(TESTS)
-
--include $(DEPS)

doc/APIchanges

@@ -1,25 +0,0 @@
-20090601 - r19025 - lavc 52.30.0 - av_lockmgr_register()
-  av_lockmgr_register() can be used to register a callback function
-  that lavc (and in the future, libraries that depend on lavc) can use
-  to implement mutexes. The application should provide a callback function
-  the implements the AV_LOCK_* operations described in avcodec.h.
-  When the lock manager is registered FFmpeg is guaranteed to behave
-  correct also in a multi-threaded application.
-
-20090301 - r17682 - lavf 52.31.0 - Generic metadata API
-  This version introduce a new metadata API (see av_metadata_get() and friends).
-  The old API is now deprecated and shouldn't be used anymore. This especially
-  include the following structure fields:
-    - AVFormatContext.title
-    - AVFormatContext.author
-    - AVFormatContext.copyright
-    - AVFormatContext.comment
-    - AVFormatContext.album
-    - AVFormatContext.year
-    - AVFormatContext.track
-    - AVFormatContext.genre
-    - AVStream.language
-    - AVStream.filename
-    - AVProgram.provider_name
-    - AVProgram.name
-    - AVChapter.title

doc/TODO

@@ -1,90 +0,0 @@
-ffmpeg TODO list:
-----------------
-
-Fabrice's TODO list: (unordered)
--------------------
-Short term:
-
-- use AVFMTCTX_DISCARD_PKT in ffplay so that DV has a chance to work
-- add RTSP regression test (both client and server)
-- make ffserver allocate AVFormatContext
-- clean up (incompatible change, for 0.5.0):
-    * AVStream -> AVComponent
-    * AVFormatContext -> AVInputStream/AVOutputStream
-    * suppress rate_emu from AVCodecContext
-- add new float/integer audio filterting and conversion : suppress
-  CODEC_ID_PCM_xxc and use CODEC_ID_RAWAUDIO.
-- fix telecine and frame rate conversion
-
-Long term (ask me if you want to help):
-
-- commit new imgconvert API and new PIX_FMT_xxx alpha formats
-- commit new LGPL'ed float and integer-only AC3 decoder
-- add WMA integer-only decoder
-- add new MPEG4-AAC audio decoder (both integer-only and float version)
-
-Michael's TODO list: (unordered) (if anyone wanna help with sth, just ask)
--------------------
-- optimize H264 CABAC
-- more optimizations
-- simper rate control
-
-Francois' TODO list: (unordered, without any timeframe)
--------------------
-- test MACE decoder against the openquicktime one as suggested by A'rpi
-- BeOS audio input grabbing backend
-- BeOS video input grabbing backend
-- publish my BeOS libposix on BeBits so I can officially support ffserver :)
-- check the whole code for thread-safety (global and init stuff)
-
-Philip'a TODO list: (alphabetically ordered) (please help)
-------------------
-- Add a multi-ffm filetype so that feeds can be recorded into multiple files rather
-  than one big file.
-- Authenticated users support -- where the authentication is in the URL
-- Change ASF files so that the embedded timestamp in the frames is right rather
-  than being an offset from the start of the stream
-- Make ffm files more resilient to changes in the codec structures so that you
-  can play old ffm files.
-
-Baptiste's TODO list:
------------------
-- mov edit list support (AVEditList)
-- YUV 10 bit per component support "2vuy"
-- mxf muxer
-- mpeg2 non linear quantizer
-
-unassigned TODO: (unordered)
----------------
-- use AVFrame for audio codecs too
-- rework aviobuf.c buffering strategy and fix url_fskip
-- generate optimal huffman tables for mjpeg encoding
-- fix ffserver regression tests
-- support xvids motion estimation
-- support x264s motion estimation
-- support x264s rate control
-- SNOW: non translational motion compensation
-- SNOW: more optimal quantization
-- SNOW: 4x4 block support
-- SNOW: 1/8 pel motion compensation support
-- SNOW: iterative motion estimation based on subsampled images
-- SNOW: try B frames and MCTF and see how their PSNR/bitrate/complexity behaves
-- SNOW: try to use the wavelet transformed MC-ed reference frame as context for the entropy coder
-- SNOW: think about/analyize how to make snow use multiple cpus/threads
-- SNOW: finish spec
-- FLAC: lossy encoding (viterbi and naive scalar quantization)
-- libavfilter
-- JPEG2000 decoder & encoder
-- MPEG4 GMC encoding support
-- macroblock based pixel format (better cache locality, somewhat complex, one paper claimed it faster for high res)
-- regression tests for codecs which do not have an encoder (I+P-frame bitstream in svn)
-- add support for using mplayers video filters to ffmpeg
-- H264 encoder
-- per MB ratecontrol (so VCD and such do work better)
-- write a script which iteratively changes all functions between always_inline and noinline and benchmarks the result to find the best set of inlined functions
-- convert all the non SIMD asm into small asm vs. C testcases and submit them to the gcc devels so they can improve gcc
-- generic audio mixing API
-- extract PES packetizer from PS muxer and use it for new TS muxer
-- implement automatic AVBistreamFilter activation
-- make cabac encoder use bytestream (see http://trac.videolan.org/x264/changeset/?format=diff&new=651)
-- merge imdct and windowing, the current code does considerable amounts of redundant work

doc/avutil.txt

@@ -1,37 +0,0 @@
-AVUtil
-======
-libavutil is a small lightweight library of generally useful functions.
-It is not a library for code needed by both libavcodec and libavformat.
-
-
-Overview:
-=========
-adler32.c               adler32 checksum
-aes.c                   AES encryption and decryption
-fifo.c                  resizeable first in first out buffer
-intfloat_readwrite.c    portable reading and writing of floating point values
-log.c                   "printf" with context and level
-md5.c                   MD5 Message-Digest Algorithm
-rational.c              code to perform exact calculations with rational numbers
-tree.c                  generic AVL tree
-crc.c                   generic CRC checksumming code
-integer.c               128bit integer math
-lls.c
-mathematics.c           greatest common divisor, integer sqrt, integer log2, ...
-mem.c                   memory allocation routines with guaranteed alignment
-softfloat.c
-
-Headers:
-bswap.h                 big/little/native-endian conversion code
-x86_cpu.h               a few useful macros for unifying x86-64 and x86-32 code
-avutil.h
-common.h
-intreadwrite.h          reading and writing of unaligned big/little/native-endian integers
-
-
-Goals:
-======
-* Modular (few interdependencies and the possibility of disabling individual parts during ./configure)
-* Small (source and object)
-* Efficient (low CPU and memory usage)
-* Useful (avoid useless features almost no one needs)

doc/faq.texi

@@ -1,488 +0,0 @@
-\input texinfo @c -*- texinfo -*-
-
-@settitle FFmpeg FAQ
-@titlepage
-@sp 7
-@center @titlefont{FFmpeg FAQ}
-@sp 3
-@end titlepage
-
-
-@chapter General Questions
-
-@section When will the next FFmpeg version be released? / Why are FFmpeg releases so few and far between?
-
-Like most open source projects FFmpeg suffers from a certain lack of
-manpower. For this reason the developers have to prioritize the work
-they do and putting out releases is not at the top of the list, fixing
-bugs and reviewing patches takes precedence. Please don't complain or
-request more timely and/or frequent releases unless you are willing to
-help out creating them.
-
-@section I have a problem with an old version of FFmpeg; where should I report it?
-Nowhere. Upgrade to the latest release or if there is no recent release upgrade
-to Subversion HEAD. You could also try to report it. Maybe you will get lucky and
-become the first person in history to get an answer different from "upgrade
-to Subversion HEAD".
-
-@section Why doesn't FFmpeg support feature [xyz]?
-
-Because no one has taken on that task yet. FFmpeg development is
-driven by the tasks that are important to the individual developers.
-If there is a feature that is important to you, the best way to get
-it implemented is to undertake the task yourself or sponsor a developer.
-
-@section FFmpeg does not support codec XXX. Can you include a Windows DLL loader to support it?
-
-No. Windows DLLs are not portable, bloated and often slow.
-Moreover FFmpeg strives to support all codecs natively.
-A DLL loader is not conducive to that goal.
-
-@section My bug report/mail to ffmpeg-devel/user has not received any replies.
-
-Likely reasons
-@itemize
-@item We are busy and haven't had time yet to read your report or
-investigate the issue.
-@item You didn't follow bugreports.html.
-@item You didn't use Subversion HEAD.
-@item You reported a segmentation fault without gdb output.
-@item You describe a problem but not how to reproduce it.
-@item It's unclear if you use ffmpeg as command line tool or use
-libav* from another application.
-@item You speak about a video having problems on playback but
-not what you use to play it.
-@item We have no faint clue what you are talking about besides
-that it is related to FFmpeg.
-@end itemize
-
-@section Is there a forum for FFmpeg? I do not like mailing lists.
-
-You may view our mailing lists with a more forum-alike look here:
-@url{http://dir.gmane.org/gmane.comp.video.ffmpeg.user},
-but, if you post, please remember that our mailing list rules still apply there.
-
-@section I cannot read this file although this format seems to be supported by ffmpeg.
-
-Even if ffmpeg can read the container format, it may not support all its
-codecs. Please consult the supported codec list in the ffmpeg
-documentation.
-
-@section Which codecs are supported by Windows?
-
-Windows does not support standard formats like MPEG very well, unless you
-install some additional codecs.
-
-The following list of video codecs should work on most Windows systems:
-@table @option
-@item msmpeg4v2
-.avi/.asf
-@item msmpeg4
-.asf only
-@item wmv1
-.asf only
-@item wmv2
-.asf only
-@item mpeg4
-Only if you have some MPEG-4 codec like ffdshow or Xvid installed.
-@item mpeg1
-.mpg only
-@end table
-Note, ASF files often have .wmv or .wma extensions in Windows. It should also
-be mentioned that Microsoft claims a patent on the ASF format, and may sue
-or threaten users who create ASF files with non-Microsoft software. It is
-strongly advised to avoid ASF where possible.
-
-The following list of audio codecs should work on most Windows systems:
-@table @option
-@item adpcm_ima_wav
-@item adpcm_ms
-@item pcm
-always
-@item mp3
-If some MP3 codec like LAME is installed.
-@end table
-
-
-@chapter Compilation
-
-@section @code{error: can't find a register in class 'GENERAL_REGS' while reloading 'asm'}
-
-This is a bug in gcc. Do not report it to us. Instead, please report it to
-the gcc developers. Note that we will not add workarounds for gcc bugs.
-
-Also note that (some of) the gcc developers believe this is not a bug or
-not a bug they should fix:
-@url{http://gcc.gnu.org/bugzilla/show_bug.cgi?id=11203}.
-Then again, some of them do not know the difference between an undecidable
-problem and an NP-hard problem...
-
-@chapter Usage
-
-@section ffmpeg does not work; what is wrong?
-
-Try a @code{make distclean} in the ffmpeg source directory before the build. If this does not help see
-(@url{http://ffmpeg.org/bugreports.html}).
-
-@section How do I encode single pictures into movies?
-
-First, rename your pictures to follow a numerical sequence.
-For example, img1.jpg, img2.jpg, img3.jpg,...
-Then you may run:
-
-@example
-  ffmpeg -f image2 -i img%d.jpg /tmp/a.mpg
-@end example
-
-Notice that @samp{%d} is replaced by the image number.
-
-@file{img%03d.jpg} means the sequence @file{img001.jpg}, @file{img002.jpg}, etc...
-
-The same logic is used for any image format that ffmpeg reads.
-
-@section How do I encode movie to single pictures?
-
-Use:
-
-@example
-  ffmpeg -i movie.mpg movie%d.jpg
-@end example
-
-The @file{movie.mpg} used as input will be converted to
-@file{movie1.jpg}, @file{movie2.jpg}, etc...
-
-Instead of relying on file format self-recognition, you may also use
-@table @option
-@item -vcodec ppm
-@item -vcodec png
-@item -vcodec mjpeg
-@end table
-to force the encoding.
-
-Applying that to the previous example:
-@example
-  ffmpeg -i movie.mpg -f image2 -vcodec mjpeg menu%d.jpg
-@end example
-
-Beware that there is no "jpeg" codec. Use "mjpeg" instead.
-
-@section Why do I see a slight quality degradation with multithreaded MPEG* encoding?
-
-For multithreaded MPEG* encoding, the encoded slices must be independent,
-otherwise thread n would practically have to wait for n-1 to finish, so it's
-quite logical that there is a small reduction of quality. This is not a bug.
-
-@section How can I read from the standard input or write to the standard output?
-
-Use @file{-} as file name.
-
-@section Why does FFmpeg not decode audio in VOB files?
-
-The audio is AC-3 (a.k.a. A/52). AC-3 decoding is an optional component in FFmpeg
-as the component that handles AC-3 decoding is currently released under the GPL.
-Enable AC-3 decoding with @code{./configure --enable-gpl}. Take care: By
-enabling AC-3, you automatically change the license of libavcodec from
-LGPL to GPL.
-
-@section Why does the chrominance data seem to be sampled at a different time from the luminance data on bt8x8 captures on Linux?
-
-This is a well-known bug in the bt8x8 driver. For 2.4.26 there is a patch at
-(@url{http://svn.ffmpeg.org/michael/trunk/patches/bttv-420-2.4.26.patch?view=co}). This may also
-apply cleanly to other 2.4-series kernels.
-
-@section How do I avoid the ugly aliasing artifacts in bt8x8 captures on Linux?
-
-Pass 'combfilter=1 lumafilter=1' to the bttv driver. Note though that 'combfilter=1'
-will cause somewhat too strong filtering. A fix is to apply (@url{http://svn.ffmpeg.org/michael/trunk/patches/bttv-comb-2.4.26.patch?view=co})
-or (@url{http://svn.ffmpeg.org/michael/trunk/patches/bttv-comb-2.6.6.patch?view=co})
-and pass 'combfilter=2'.
-
-@section -f jpeg doesn't work.
-
-Try '-f image2 test%d.jpg'.
-
-@section Why can I not change the framerate?
-
-Some codecs, like MPEG-1/2, only allow a small number of fixed framerates.
-Choose a different codec with the -vcodec command line option.
-
-@section How do I encode Xvid or DivX video with ffmpeg?
-
-Both Xvid and DivX (version 4+) are implementations of the ISO MPEG-4
-standard (note that there are many other coding formats that use this
-same standard). Thus, use '-vcodec mpeg4' to encode in these formats. The
-default fourcc stored in an MPEG-4-coded file will be 'FMP4'. If you want
-a different fourcc, use the '-vtag' option. E.g., '-vtag xvid' will
-force the fourcc 'xvid' to be stored as the video fourcc rather than the
-default.
-
-@section How do I encode videos which play on the iPod?
-
-@table @option
-@item needed stuff
--acodec libfaac -vcodec mpeg4 width<=320 height<=240
-@item working stuff
-4mv, title
-@item non-working stuff
-B-frames
-@item example command line
-ffmpeg -i input -acodec libfaac -ab 128kb -vcodec mpeg4 -b 1200kb -mbd 2 -flags +4mv -trellis 2 -aic 2 -cmp 2 -subcmp 2 -s 320x180 -title X output.mp4
-@end table
-
-@section How do I encode videos which play on the PSP?
-
-@table @option
-@item needed stuff
--acodec libfaac -vcodec mpeg4 width*height<=76800 width%16=0 height%16=0 -ar 24000 -r 30000/1001 or 15000/1001 -f psp
-@item working stuff
-4mv, title
-@item non-working stuff
-B-frames
-@item example command line
-ffmpeg -i input -acodec libfaac -ab 128kb -vcodec mpeg4 -b 1200kb -ar 24000 -mbd 2 -flags +4mv -trellis 2 -aic 2 -cmp 2 -subcmp 2 -s 368x192 -r 30000/1001 -title X -f psp output.mp4
-@item needed stuff for H.264
--acodec libfaac -vcodec libx264 width*height<=76800 width%16=0? height%16=0? -ar 48000 -coder 1 -r 30000/1001 or 15000/1001 -f psp
-@item working stuff for H.264
-title, loop filter
-@item non-working stuff for H.264
-CAVLC
-@item example command line
-ffmpeg -i input -acodec libfaac -ab 128kb -vcodec libx264 -b 1200kb -ar 48000 -mbd 2 -coder 1 -cmp 2 -subcmp 2 -s 368x192 -r 30000/1001 -title X -f psp -flags loop -trellis 2 -partitions parti4x4+parti8x8+partp4x4+partp8x8+partb8x8 output.mp4
-@item higher resolution for newer PSP firmwares, width<=480, height<=272
--vcodec libx264 -level 21 -coder 1 -f psp
-@item example command line
-ffmpeg -i input -acodec libfaac -ab 128kb -ac 2 -ar 48000 -vcodec libx264 -level 21 -b 640kb -coder 1 -f psp -flags +loop -trellis 2 -partitions +parti4x4+parti8x8+partp4x4+partp8x8+partb8x8 -g 250 -s 480x272 output.mp4
-@end table
-
-@section Which are good parameters for encoding high quality MPEG-4?
-
-'-mbd rd -flags +4mv+aic -trellis 2 -cmp 2 -subcmp 2 -g 300 -pass 1/2',
-things to try: '-bf 2', '-flags qprd', '-flags mv0', '-flags skiprd'.
-
-@section Which are good parameters for encoding high quality MPEG-1/MPEG-2?
-
-'-mbd rd -trellis 2 -cmp 2 -subcmp 2 -g 100 -pass 1/2'
-but beware the '-g 100' might cause problems with some decoders.
-Things to try: '-bf 2', '-flags qprd', '-flags mv0', '-flags skiprd.
-
-@section Interlaced video looks very bad when encoded with ffmpeg, what is wrong?
-
-You should use '-flags +ilme+ildct' and maybe '-flags +alt' for interlaced
-material, and try '-top 0/1' if the result looks really messed-up.
-
-@section How can I read DirectShow files?
-
-If you have built FFmpeg with @code{./configure --enable-avisynth}
-(only possible on MinGW/Cygwin platforms),
-then you may use any file that DirectShow can read as input.
-(Be aware that this feature has been recently added,
-so you will need to help yourself in case of problems.)
-
-Just create an "input.avs" text file with this single line ...
-@example
-  DirectShowSource("C:\path to your file\yourfile.asf")
-@end example
-... and then feed that text file to FFmpeg:
-@example
-  ffmpeg -i input.avs
-@end example
-
-For ANY other help on Avisynth, please visit @url{http://www.avisynth.org/}.
-
-@section How can I join video files?
-
-A few multimedia containers (MPEG-1, MPEG-2 PS, DV) allow to join video files by
-merely concatenating them.
-
-Hence you may concatenate your multimedia files by first transcoding them to
-these privileged formats, then using the humble @code{cat} command (or the
-equally humble @code{copy} under Windows), and finally transcoding back to your
-format of choice.
-
-@example
-ffmpeg -i input1.avi -sameq intermediate1.mpg
-ffmpeg -i input2.avi -sameq intermediate2.mpg
-cat intermediate1.mpg intermediate2.mpg > intermediate_all.mpg
-ffmpeg -i intermediate_all.mpg -sameq output.avi
-@end example
-
-Notice that you should either use @code{-sameq} or set a reasonably high
-bitrate for your intermediate and output files, if you want to preserve
-video quality.
-
-Also notice that you may avoid the huge intermediate files by taking advantage
-of named pipes, should your platform support it:
-
-@example
-mkfifo intermediate1.mpg
-mkfifo intermediate2.mpg
-ffmpeg -i input1.avi -sameq -y intermediate1.mpg < /dev/null &
-ffmpeg -i input2.avi -sameq -y intermediate2.mpg < /dev/null &
-cat intermediate1.mpg intermediate2.mpg |\
-ffmpeg -f mpeg -i - -sameq -vcodec mpeg4 -acodec libmp3lame output.avi
-@end example
-
-Similarly, the yuv4mpegpipe format, and the raw video, raw audio codecs also
-allow concatenation, and the transcoding step is almost lossless.
-
-For example, let's say we want to join two FLV files into an output.flv file:
-
-@example
-mkfifo temp1.a
-mkfifo temp1.v
-mkfifo temp2.a
-mkfifo temp2.v
-mkfifo all.a
-mkfifo all.v
-ffmpeg -i input1.flv -vn -f u16le -acodec pcm_s16le -ac 2 -ar 44100 - > temp1.a < /dev/null &
-ffmpeg -i input2.flv -vn -f u16le -acodec pcm_s16le -ac 2 -ar 44100 - > temp2.a < /dev/null &
-ffmpeg -i input1.flv -an -f yuv4mpegpipe - > temp1.v < /dev/null &
-ffmpeg -i input2.flv -an -f yuv4mpegpipe - > temp2.v < /dev/null &
-cat temp1.a temp2.a > all.a &
-cat temp1.v temp2.v > all.v &
-ffmpeg -f u16le -acodec pcm_s16le -ac 2 -ar 44100 -i all.a \
-       -f yuv4mpegpipe -i all.v \
-       -sameq -y output.flv
-rm temp[12].[av] all.[av]
-@end example
-
-@section FFmpeg does not adhere to the -maxrate setting, some frames are bigger than maxrate/fps.
-
-Read the MPEG spec about video buffer verifier.
-
-@section I want CBR, but no matter what I do frame sizes differ.
-
-You do not understand what CBR is, please read the MPEG spec.
-Read about video buffer verifier and constant bitrate.
-The one sentence summary is that there is a buffer and the input rate is
-constant, the output can vary as needed.
-
-@section How do I check if a stream is CBR?
-
-To quote the MPEG-2 spec:
-"There is no way to tell that a bitstream is constant bitrate without
-examining all of the vbv_delay values and making complicated computations."
-
-
-@chapter Development
-
-@section Are there examples illustrating how to use the FFmpeg libraries, particularly libavcodec and libavformat?
-
-Yes. Read the Developers Guide of the FFmpeg documentation. Alternatively,
-examine the source code for one of the many open source projects that
-already incorporate FFmpeg at (@url{projects.html}).
-
-@section Can you support my C compiler XXX?
-
-It depends. If your compiler is C99-compliant, then patches to support
-it are likely to be welcome if they do not pollute the source code
-with @code{#ifdef}s related to the compiler.
-
-@section Is Microsoft Visual C++ supported?
-
-No. Microsoft Visual C++ is not compliant to the C99 standard and does
-not - among other things - support the inline assembly used in FFmpeg.
-If you wish to use MSVC++ for your
-project then you can link the MSVC++ code with libav* as long as
-you compile the latter with a working C compiler. For more information, see
-the @emph{Microsoft Visual C++ compatibility} section in the FFmpeg
-documentation.
-
-There have been efforts to make FFmpeg compatible with MSVC++ in the
-past. However, they have all been rejected as too intrusive, especially
-since MinGW does the job adequately. None of the core developers
-work with MSVC++ and thus this item is low priority. Should you find
-the silver bullet that solves this problem, feel free to shoot it at us.
-
-We strongly recommend you to move over from MSVC++ to MinGW tools.
-
-@section Can I use FFmpeg or libavcodec under Windows?
-
-Yes, but the Cygwin or MinGW tools @emph{must} be used to compile FFmpeg.
-Read the @emph{Windows} section in the FFmpeg documentation to find more
-information.
-
-To get help and instructions for building FFmpeg under Windows, check out
-the FFmpeg Windows Help Forum at
-@url{http://ffmpeg.arrozcru.org/}.
-
-@section Can you add automake, libtool or autoconf support?
-
-No. These tools are too bloated and they complicate the build.
-
-@section Why not rewrite ffmpeg in object-oriented C++?
-
-FFmpeg is already organized in a highly modular manner and does not need to
-be rewritten in a formal object language. Further, many of the developers
-favor straight C; it works for them. For more arguments on this matter,
-read "Programming Religion" at (@url{http://www.tux.org/lkml/#s15}).
-
-@section Why are the ffmpeg programs devoid of debugging symbols?
-
-The build process creates ffmpeg_g, ffplay_g, etc. which contain full debug
-information. Those binaries are stripped to create ffmpeg, ffplay, etc. If
-you need the debug information, used the *_g versions.
-
-@section I do not like the LGPL, can I contribute code under the GPL instead?
-
-Yes, as long as the code is optional and can easily and cleanly be placed
-under #if CONFIG_GPL without breaking anything. So for example a new codec
-or filter would be OK under GPL while a bug fix to LGPL code would not.
-
-@section I want to compile xyz.c alone but my compiler produced many errors.
-
-Common code is in its own files in libav* and is used by the individual
-codecs. They will not work without the common parts, you have to compile
-the whole libav*. If you wish, disable some parts with configure switches.
-You can also try to hack it and remove more, but if you had problems fixing
-the compilation failure then you are probably not qualified for this.
-
-@section I'm using libavcodec from within my C++ application but the linker complains about missing symbols which seem to be available.
-
-FFmpeg is a pure C project, so to use the libraries within your C++ application
-you need to explicitly state that you are using a C library. You can do this by
-encompassing your FFmpeg includes using @code{extern "C"}.
-
-See @url{http://www.parashift.com/c++-faq-lite/mixing-c-and-cpp.html#faq-32.3}
-
-@section I have a file in memory / a API different from *open/*read/ libc how do I use it with libavformat?
-
-You have to implement a URLProtocol, see libavformat/file.c in FFmpeg
-and libmpdemux/demux_lavf.c in MPlayer sources.
-
-@section I get "No compatible shell script interpreter found." in MSys.
-
-The standard MSys bash (2.04) is broken. You need to install 2.05 or later.
-
-@section I get "./configure: line <xxx>: pr: command not found" in MSys.
-
-The standard MSys install doesn't come with pr. You need to get it from the coreutils package.
-
-@section I tried to pass RTP packets into a decoder, but it doesn't work.
-
-RTP is a container format like any other, you must first depacketize the
-codec frames/samples stored in RTP and then feed to the decoder.
-
-@section Where can I find libav* headers for Pascal/Delphi?
-
-see @url{http://www.iversenit.dk/dev/ffmpeg-headers/}
-
-@section Where is the documentation about ffv1, msmpeg4, asv1, 4xm?
-
-see @url{http://svn.ffmpeg.org/michael/trunk/docs/}
-
-@section How do I feed H.263-RTP (and other codecs in RTP) to libavcodec?
-
-Even if peculiar since it is network oriented, RTP is a container like any
-other. You have to @emph{demux} RTP before feeding the payload to libavcodec.
-In this specific case please look at RFC 4629 to see how it should be done.
-
-@section AVStream.r_frame_rate is wrong, it is much larger than the framerate.
-
-r_frame_rate is NOT the average framerate, it is the smallest framerate
-that can accurately represent all timestamps. So no, it is not
-wrong if it is larger than the average!
-For example, if you have mixed 25 and 30 fps content, then r_frame_rate
-will be 150.
-
-@bye

doc/ffmpeg-doc.texi

@@ -1,969 +0,0 @@
-\input texinfo @c -*- texinfo -*-
-
-@settitle FFmpeg Documentation
-@titlepage
-@sp 7
-@center @titlefont{FFmpeg Documentation}
-@sp 3
-@end titlepage
-
-
-@chapter Introduction
-
-FFmpeg is a very fast video and audio converter. It can also grab from
-a live audio/video source.
-
-The command line interface is designed to be intuitive, in the sense
-that FFmpeg tries to figure out all parameters that can possibly be
-derived automatically. You usually only have to specify the target
-bitrate you want.
-
-FFmpeg can also convert from any sample rate to any other, and resize
-video on the fly with a high quality polyphase filter.
-
-@chapter Quick Start
-
-@c man begin EXAMPLES
-@section Video and Audio grabbing
-
-FFmpeg can grab video and audio from devices given that you specify the input
-format and device.
-
-@example
-ffmpeg -f oss -i /dev/dsp -f video4linux2 -i /dev/video0 /tmp/out.mpg
-@end example
-
-Note that you must activate the right video source and channel before
-launching FFmpeg with any TV viewer such as xawtv
-(@url{http://linux.bytesex.org/xawtv/}) by Gerd Knorr. You also
-have to set the audio recording levels correctly with a
-standard mixer.
-
-@section X11 grabbing
-
-FFmpeg can grab the X11 display.
-
-@example
-ffmpeg -f x11grab -s cif -i :0.0 /tmp/out.mpg
-@end example
-
-0.0 is display.screen number of your X11 server, same as
-the DISPLAY environment variable.
-
-@example
-ffmpeg -f x11grab -s cif -i :0.0+10,20 /tmp/out.mpg
-@end example
-
-0.0 is display.screen number of your X11 server, same as the DISPLAY environment
-variable. 10 is the x-offset and 20 the y-offset for the grabbing.
-
-@section Video and Audio file format conversion
-
-* FFmpeg can use any supported file format and protocol as input:
-
-Examples:
-
-* You can use YUV files as input:
-
-@example
-ffmpeg -i /tmp/test%d.Y /tmp/out.mpg
-@end example
-
-It will use the files:
-@example
-/tmp/test0.Y, /tmp/test0.U, /tmp/test0.V,
-/tmp/test1.Y, /tmp/test1.U, /tmp/test1.V, etc...
-@end example
-
-The Y files use twice the resolution of the U and V files. They are
-raw files, without header. They can be generated by all decent video
-decoders. You must specify the size of the image with the @option{-s} option
-if FFmpeg cannot guess it.
-
-* You can input from a raw YUV420P file:
-
-@example
-ffmpeg -i /tmp/test.yuv /tmp/out.avi
-@end example
-
-test.yuv is a file containing raw YUV planar data. Each frame is composed
-of the Y plane followed by the U and V planes at half vertical and
-horizontal resolution.
-
-* You can output to a raw YUV420P file:
-
-@example
-ffmpeg -i mydivx.avi hugefile.yuv
-@end example
-
-* You can set several input files and output files:
-
-@example
-ffmpeg -i /tmp/a.wav -s 640x480 -i /tmp/a.yuv /tmp/a.mpg
-@end example
-
-Converts the audio file a.wav and the raw YUV video file a.yuv
-to MPEG file a.mpg.
-
-* You can also do audio and video conversions at the same time:
-
-@example
-ffmpeg -i /tmp/a.wav -ar 22050 /tmp/a.mp2
-@end example
-
-Converts a.wav to MPEG audio at 22050 Hz sample rate.
-
-* You can encode to several formats at the same time and define a
-mapping from input stream to output streams:
-
-@example
-ffmpeg -i /tmp/a.wav -ab 64k /tmp/a.mp2 -ab 128k /tmp/b.mp2 -map 0:0 -map 0:0
-@end example
-
-Converts a.wav to a.mp2 at 64 kbits and to b.mp2 at 128 kbits. '-map
-file:index' specifies which input stream is used for each output
-stream, in the order of the definition of output streams.
-
-* You can transcode decrypted VOBs:
-
-@example
-ffmpeg -i snatch_1.vob -f avi -vcodec mpeg4 -b 800k -g 300 -bf 2 -acodec libmp3lame -ab 128k snatch.avi
-@end example
-
-This is a typical DVD ripping example; the input is a VOB file, the
-output an AVI file with MPEG-4 video and MP3 audio. Note that in this
-command we use B-frames so the MPEG-4 stream is DivX5 compatible, and
-GOP size is 300 which means one intra frame every 10 seconds for 29.97fps
-input video. Furthermore, the audio stream is MP3-encoded so you need
-to enable LAME support by passing @code{--enable-libmp3lame} to configure.
-The mapping is particularly useful for DVD transcoding
-to get the desired audio language.
-
-NOTE: To see the supported input formats, use @code{ffmpeg -formats}.
-
-* You can extract images from a video, or create a video from many images:
-
-For extracting images from a video:
-@example
-ffmpeg -i foo.avi -r 1 -s WxH -f image2 foo-%03d.jpeg
-@end example
-
-This will extract one video frame per second from the video and will
-output them in files named @file{foo-001.jpeg}, @file{foo-002.jpeg},
-etc. Images will be rescaled to fit the new WxH values.
-
-If you want to extract just a limited number of frames, you can use the
-above command in combination with the -vframes or -t option, or in
-combination with -ss to start extracting from a certain point in time.
-
-For creating a video from many images:
-@example
-ffmpeg -f image2 -i foo-%03d.jpeg -r 12 -s WxH foo.avi
-@end example
-
-The syntax @code{foo-%03d.jpeg} specifies to use a decimal number
-composed of three digits padded with zeroes to express the sequence
-number. It is the same syntax supported by the C printf function, but
-only formats accepting a normal integer are suitable.
-
-* You can put many streams of the same type in the output:
-
-@example
-ffmpeg -i test1.avi -i test2.avi -vcodec copy -acodec copy -vcodec copy -acodec copy test12.avi -newvideo -newaudio
-@end example
-
-In addition to the first video and audio streams, the resulting
-output file @file{test12.avi} will contain the second video
-and the second audio stream found in the input streams list.
-
-The @code{-newvideo}, @code{-newaudio} and @code{-newsubtitle}
-options have to be specified immediately after the name of the output
-file to which you want to add them.
-@c man end
-
-@chapter Invocation
-
-@section Syntax
-
-The generic syntax is:
-
-@example
-@c man begin SYNOPSIS
-ffmpeg [[infile options][@option{-i} @var{infile}]]... @{[outfile options] @var{outfile}@}...
-@c man end
-@end example
-@c man begin DESCRIPTION
-As a general rule, options are applied to the next specified
-file. Therefore, order is important, and you can have the same
-option on the command line multiple times. Each occurrence is
-then applied to the next input or output file.
-
-* To set the video bitrate of the output file to 64kbit/s:
-@example
-ffmpeg -i input.avi -b 64k output.avi
-@end example
-
-* To force the frame rate of the output file to 24 fps:
-@example
-ffmpeg -i input.avi -r 24 output.avi
-@end example
-
-* To force the frame rate of the input file (valid for raw formats only)
-to 1 fps and the frame rate of the output file to 24 fps:
-@example
-ffmpeg -r 1 -i input.m2v -r 24 output.avi
-@end example
-
-The format option may be needed for raw input files.
-
-By default, FFmpeg tries to convert as losslessly as possible: It
-uses the same audio and video parameters for the outputs as the one
-specified for the inputs.
-@c man end
-
-@c man begin OPTIONS
-@section Main options
-
-@table @option
-@item -L
-Show license.
-
-@item -h
-Show help.
-
-@item -version
-Show version.
-
-@item -formats
-Show available formats, codecs, bitstream filters, protocols, and frame size and frame rate abbreviations.
-
-The fields preceding the format and codec names have the following meanings:
-@table @samp
-@item D
-Decoding available
-@item E
-Encoding available
-@item V/A/S
-Video/audio/subtitle codec
-@item S
-Codec supports slices
-@item D
-Codec supports direct rendering
-@item T
-Codec can handle input truncated at random locations instead of only at frame boundaries
-@end table
-
-@item -f @var{fmt}
-Force format.
-
-@item -i @var{filename}
-input file name
-
-@item -y
-Overwrite output files.
-
-@item -t @var{duration}
-Restrict the transcoded/captured video sequence
-to the duration specified in seconds.
-@code{hh:mm:ss[.xxx]} syntax is also supported.
-
-@item -fs @var{limit_size}
-Set the file size limit.
-
-@item -ss @var{position}
-Seek to given time position in seconds.
-@code{hh:mm:ss[.xxx]} syntax is also supported.
-
-@item -itsoffset @var{offset}
-Set the input time offset in seconds.
-@code{[-]hh:mm:ss[.xxx]} syntax is also supported.
-This option affects all the input files that follow it.
-The offset is added to the timestamps of the input files.
-Specifying a positive offset means that the corresponding
-streams are delayed by 'offset' seconds.
-
-@item -timestamp @var{time}
-Set the timestamp.
-
-@item -metadata @var{key}=@var{value}
-Set a metadata key/value pair.
-
-For example, for setting the title in the output file:
-@example
-ffmpeg -i in.avi -metadata title="my title" out.flv
-@end example
-
-@item -v @var{number}
-Set the logging verbosity level.
-
-@item -target @var{type}
-Specify target file type ("vcd", "svcd", "dvd", "dv", "dv50", "pal-vcd",
-"ntsc-svcd", ... ). All the format options (bitrate, codecs,
-buffer sizes) are then set automatically. You can just type:
-
-@example
-ffmpeg -i myfile.avi -target vcd /tmp/vcd.mpg
-@end example
-
-Nevertheless you can specify additional options as long as you know
-they do not conflict with the standard, as in:
-
-@example
-ffmpeg -i myfile.avi -target vcd -bf 2 /tmp/vcd.mpg
-@end example
-
-@item -dframes @var{number}
-Set the number of data frames to record.
-
-@item -scodec @var{codec}
-Force subtitle codec ('copy' to copy stream).
-
-@item -newsubtitle
-Add a new subtitle stream to the current output stream.
-
-@item -slang @var{code}
-Set the ISO 639 language code (3 letters) of the current subtitle stream.
-
-@end table
-
-@section Video Options
-
-@table @option
-@item -b @var{bitrate}
-Set the video bitrate in bit/s (default = 200 kb/s).
-@item -vframes @var{number}
-Set the number of video frames to record.
-@item -r @var{fps}
-Set frame rate (Hz value, fraction or abbreviation), (default = 25).
-@item -s @var{size}
-Set frame size. The format is @samp{wxh} (ffserver default = 160x128, ffmpeg default = same as source).
-The following abbreviations are recognized:
-@table @samp
-@item sqcif
-128x96
-@item qcif
-176x144
-@item cif
-352x288
-@item 4cif
-704x576
-@item qqvga
-160x120
-@item qvga
-320x240
-@item vga
-640x480
-@item svga
-800x600
-@item xga
-1024x768
-@item uxga
-1600x1200
-@item qxga
-2048x1536
-@item sxga
-1280x1024
-@item qsxga
-2560x2048
-@item hsxga
-5120x4096
-@item wvga
-852x480
-@item wxga
-1366x768
-@item wsxga
-1600x1024
-@item wuxga
-1920x1200
-@item woxga
-2560x1600
-@item wqsxga
-3200x2048
-@item wquxga
-3840x2400
-@item whsxga
-6400x4096
-@item whuxga
-7680x4800
-@item cga
-320x200
-@item ega
-640x350
-@item hd480
-852x480
-@item hd720
-1280x720
-@item hd1080
-1920x1080
-@end table
-
-@item -aspect @var{aspect}
-Set aspect ratio (4:3, 16:9 or 1.3333, 1.7777).
-@item -croptop @var{size}
-Set top crop band size (in pixels).
-@item -cropbottom @var{size}
-Set bottom crop band size (in pixels).
-@item -cropleft @var{size}
-Set left crop band size (in pixels).
-@item -cropright @var{size}
-Set right crop band size (in pixels).
-@item -padtop @var{size}
-Set top pad band size (in pixels).
-@item -padbottom @var{size}
-Set bottom pad band size (in pixels).
-@item -padleft @var{size}
-Set left pad band size (in pixels).
-@item -padright @var{size}
-Set right pad band size (in pixels).
-@item -padcolor @var{hex_color}
-Set color of padded bands. The value for padcolor is expressed
-as a six digit hexadecimal number where the first two digits
-represent red, the middle two digits green and last two digits
-blue (default = 000000 (black)).
-@item -vn
-Disable video recording.
-@item -bt @var{tolerance}
-Set video bitrate tolerance (in bits, default 4000k).
-Has a minimum value of: (target_bitrate/target_framerate).
-In 1-pass mode, bitrate tolerance specifies how far ratecontrol is
-willing to deviate from the target average bitrate value. This is
-not related to min/max bitrate. Lowering tolerance too much has
-an adverse effect on quality.
-@item -maxrate @var{bitrate}
-Set max video bitrate (in bit/s).
-Requires -bufsize to be set.
-@item -minrate @var{bitrate}
-Set min video bitrate (in bit/s).
-Most useful in setting up a CBR encode:
-@example
-ffmpeg -i myfile.avi -b 4000k -minrate 4000k -maxrate 4000k -bufsize 1835k out.m2v
-@end example
-It is of little use elsewise.
-@item -bufsize @var{size}
-Set video buffer verifier buffer size (in bits).
-@item -vcodec @var{codec}
-Force video codec to @var{codec}. Use the @code{copy} special value to
-tell that the raw codec data must be copied as is.
-@item -sameq
-Use same video quality as source (implies VBR).
-
-@item -pass @var{n}
-Select the pass number (1 or 2). It is used to do two-pass
-video encoding. The statistics of the video are recorded in the first
-pass into a log file (see also the option -passlogfile),
-and in the second pass that log file is used to generate the video
-at the exact requested bitrate.
-On pass 1, you may just deactivate audio and set output to null,
-examples for Windows and Unix:
-@example
-ffmpeg -i foo.mov -vcodec libxvid -pass 1 -an -f rawvideo -y NUL
-ffmpeg -i foo.mov -vcodec libxvid -pass 1 -an -f rawvideo -y /dev/null
-@end example
-
-@item -passlogfile @var{prefix}
-Set two-pass log file name prefix to @var{prefix}, the default file name
-prefix is ``ffmpeg2pass''. The complete file name will be
-@file{PREFIX-N.log}, where N is a number specific to the output
-stream.
-
-@item -newvideo
-Add a new video stream to the current output stream.
-
-@end table
-
-@section Advanced Video Options
-
-@table @option
-@item -pix_fmt @var{format}
-Set pixel format. Use 'list' as parameter to show all the supported
-pixel formats.
-@item -sws_flags @var{flags}
-Set SwScaler flags (only available when compiled with swscale support).
-@item -g @var{gop_size}
-Set the group of pictures size.
-@item -intra
-Use only intra frames.
-@item -vdt @var{n}
-Discard threshold.
-@item -qscale @var{q}
-Use fixed video quantizer scale (VBR).
-@item -qmin @var{q}
-minimum video quantizer scale (VBR)
-@item -qmax @var{q}
-maximum video quantizer scale (VBR)
-@item -qdiff @var{q}
-maximum difference between the quantizer scales (VBR)
-@item -qblur @var{blur}
-video quantizer scale blur (VBR) (range 0.0 - 1.0)
-@item -qcomp @var{compression}
-video quantizer scale compression (VBR) (default 0.5).
-Constant of ratecontrol equation. Recommended range for default rc_eq: 0.0-1.0
-
-@item -lmin @var{lambda}
-minimum video lagrange factor (VBR)
-@item -lmax @var{lambda}
-max video lagrange factor (VBR)
-@item -mblmin @var{lambda}
-minimum macroblock quantizer scale (VBR)
-@item -mblmax @var{lambda}
-maximum macroblock quantizer scale (VBR)
-
-These four options (lmin, lmax, mblmin, mblmax) use 'lambda' units,
-but you may use the QP2LAMBDA constant to easily convert from 'q' units:
-@example
-ffmpeg -i src.ext -lmax 21*QP2LAMBDA dst.ext
-@end example
-
-@item -rc_init_cplx @var{complexity}
-initial complexity for single pass encoding
-@item -b_qfactor @var{factor}
-qp factor between P- and B-frames
-@item -i_qfactor @var{factor}
-qp factor between P- and I-frames
-@item -b_qoffset @var{offset}
-qp offset between P- and B-frames
-@item -i_qoffset @var{offset}
-qp offset between P- and I-frames
-@item -rc_eq @var{equation}
-Set rate control equation (@pxref{FFmpeg formula
-evaluator}) (default = @code{tex^qComp}).
-@item -rc_override @var{override}
-rate control override for specific intervals
-@item -me_method @var{method}
-Set motion estimation method to @var{method}.
-Available methods are (from lowest to best quality):
-@table @samp
-@item zero
-Try just the (0, 0) vector.
-@item phods
-@item log
-@item x1
-@item hex
-@item umh
-@item epzs
-(default method)
-@item full
-exhaustive search (slow and marginally better than epzs)
-@end table
-
-@item -dct_algo @var{algo}
-Set DCT algorithm to @var{algo}. Available values are:
-@table @samp
-@item 0
-FF_DCT_AUTO (default)
-@item 1
-FF_DCT_FASTINT
-@item 2
-FF_DCT_INT
-@item 3
-FF_DCT_MMX
-@item 4
-FF_DCT_MLIB
-@item 5
-FF_DCT_ALTIVEC
-@end table
-
-@item -idct_algo @var{algo}
-Set IDCT algorithm to @var{algo}. Available values are:
-@table @samp
-@item 0
-FF_IDCT_AUTO (default)
-@item 1
-FF_IDCT_INT
-@item 2
-FF_IDCT_SIMPLE
-@item 3
-FF_IDCT_SIMPLEMMX
-@item 4
-FF_IDCT_LIBMPEG2MMX
-@item 5
-FF_IDCT_PS2
-@item 6
-FF_IDCT_MLIB
-@item 7
-FF_IDCT_ARM
-@item 8
-FF_IDCT_ALTIVEC
-@item 9
-FF_IDCT_SH4
-@item 10
-FF_IDCT_SIMPLEARM
-@end table
-
-@item -er @var{n}
-Set error resilience to @var{n}.
-@table @samp
-@item 1
-FF_ER_CAREFUL (default)
-@item 2
-FF_ER_COMPLIANT
-@item 3
-FF_ER_AGGRESSIVE
-@item 4
-FF_ER_VERY_AGGRESSIVE
-@end table
-
-@item -ec @var{bit_mask}
-Set error concealment to @var{bit_mask}. @var{bit_mask} is a bit mask of
-the following values:
-@table @samp
-@item 1
-FF_EC_GUESS_MVS (default = enabled)
-@item 2
-FF_EC_DEBLOCK (default = enabled)
-@end table
-
-@item -bf @var{frames}
-Use 'frames' B-frames (supported for MPEG-1, MPEG-2 and MPEG-4).
-@item -mbd @var{mode}
-macroblock decision
-@table @samp
-@item 0
-FF_MB_DECISION_SIMPLE: Use mb_cmp (cannot change it yet in FFmpeg).
-@item 1
-FF_MB_DECISION_BITS: Choose the one which needs the fewest bits.
-@item 2
-FF_MB_DECISION_RD: rate distortion
-@end table
-
-@item -4mv
-Use four motion vector by macroblock (MPEG-4 only).
-@item -part
-Use data partitioning (MPEG-4 only).
-@item -bug @var{param}
-Work around encoder bugs that are not auto-detected.
-@item -strict @var{strictness}
-How strictly to follow the standards.
-@item -aic
-Enable Advanced intra coding (h263+).
-@item -umv
-Enable Unlimited Motion Vector (h263+)
-
-@item -deinterlace
-Deinterlace pictures.
-@item -ilme
-Force interlacing support in encoder (MPEG-2 and MPEG-4 only).
-Use this option if your input file is interlaced and you want
-to keep the interlaced format for minimum losses.
-The alternative is to deinterlace the input stream with
-@option{-deinterlace}, but deinterlacing introduces losses.
-@item -psnr
-Calculate PSNR of compressed frames.
-@item -vstats
-Dump video coding statistics to @file{vstats_HHMMSS.log}.
-@item -vstats_file @var{file}
-Dump video coding statistics to @var{file}.
-@item -vhook @var{module}
-Insert video processing @var{module}. @var{module} contains the module
-name and its parameters separated by spaces.
-@item -top @var{n}
-top=1/bottom=0/auto=-1 field first
-@item -dc @var{precision}
-Intra_dc_precision.
-@item -vtag @var{fourcc/tag}
-Force video tag/fourcc.
-@item -qphist
-Show QP histogram.
-@item -vbsf @var{bitstream_filter}
-Bitstream filters available are "dump_extra", "remove_extra", "noise", "h264_mp4toannexb", "imxdump", "mjpegadump".
-@example
-ffmpeg -i h264.mp4 -vcodec copy -vbsf h264_mp4toannexb -an out.h264
-@end example
-@end table
-
-@section Audio Options
-
-@table @option
-@item -aframes @var{number}
-Set the number of audio frames to record.
-@item -ar @var{freq}
-Set the audio sampling frequency (default = 44100 Hz).
-@item -ab @var{bitrate}
-Set the audio bitrate in bit/s (default = 64k).
-@item -aq @var{q}
-Set the audio quality (codec-specific, VBR).
-@item -ac @var{channels}
-Set the number of audio channels (default = 1).
-@item -an
-Disable audio recording.
-@item -acodec @var{codec}
-Force audio codec to @var{codec}. Use the @code{copy} special value to
-specify that the raw codec data must be copied as is.
-@item -newaudio
-Add a new audio track to the output file. If you want to specify parameters,
-do so before @code{-newaudio} (@code{-acodec}, @code{-ab}, etc..).
-
-Mapping will be done automatically, if the number of output streams is equal to
-the number of input streams, else it will pick the first one that matches. You
-can override the mapping using @code{-map} as usual.
-
-Example:
-@example
-ffmpeg -i file.mpg -vcodec copy -acodec ac3 -ab 384k test.mpg -acodec mp2 -ab 192k -newaudio
-@end example
-@item -alang @var{code}
-Set the ISO 639 language code (3 letters) of the current audio stream.
-@end table
-
-@section Advanced Audio options:
-
-@table @option
-@item -atag @var{fourcc/tag}
-Force audio tag/fourcc.
-@item -absf @var{bitstream_filter}
-Bitstream filters available are "dump_extra", "remove_extra", "noise", "mp3comp", "mp3decomp".
-@end table
-
-@section Subtitle options:
-
-@table @option
-@item -scodec @var{codec}
-Force subtitle codec ('copy' to copy stream).
-@item -newsubtitle
-Add a new subtitle stream to the current output stream.
-@item -slang @var{code}
-Set the ISO 639 language code (3 letters) of the current subtitle stream.
-@item -sn
-Disable subtitle recording.
-@item -sbsf @var{bitstream_filter}
-Bitstream filters available are "mov2textsub", "text2movsub".
-@example
-ffmpeg -i file.mov -an -vn -sbsf mov2textsub -scodec copy -f rawvideo sub.txt
-@end example
-@end table
-
-@section Audio/Video grab options
-
-@table @option
-@item -vc @var{channel}
-Set video grab channel (DV1394 only).
-@item -tvstd @var{standard}
-Set television standard (NTSC, PAL (SECAM)).
-@item -isync
-Synchronize read on input.
-@end table
-
-@section Advanced options
-
-@table @option
-@item -map @var{input_stream_id}[:@var{sync_stream_id}]
-Set stream mapping from input streams to output streams.
-Just enumerate the input streams in the order you want them in the output.
-@var{sync_stream_id} if specified sets the input stream to sync
-against.
-@item -map_meta_data @var{outfile}:@var{infile}
-Set meta data information of @var{outfile} from @var{infile}.
-@item -debug
-Print specific debug info.
-@item -benchmark
-Add timings for benchmarking.
-@item -dump
-Dump each input packet.
-@item -hex
-When dumping packets, also dump the payload.
-@item -bitexact
-Only use bit exact algorithms (for codec testing).
-@item -ps @var{size}
-Set packet size in bits.
-@item -re
-Read input at native frame rate. Mainly used to simulate a grab device.
-@item -loop_input
-Loop over the input stream. Currently it works only for image
-streams. This option is used for automatic FFserver testing.
-@item -loop_output @var{number_of_times}
-Repeatedly loop output for formats that support looping such as animated GIF
-(0 will loop the output infinitely).
-@item -threads @var{count}
-Thread count.
-@item -vsync @var{parameter}
-Video sync method. Video will be stretched/squeezed to match the timestamps,
-it is done by duplicating and dropping frames. With -map you can select from
-which stream the timestamps should be taken. You can leave either video or
-audio unchanged and sync the remaining stream(s) to the unchanged one.
-@item -async @var{samples_per_second}
-Audio sync method. "Stretches/squeezes" the audio stream to match the timestamps,
-the parameter is the maximum samples per second by which the audio is changed.
--async 1 is a special case where only the start of the audio stream is corrected
-without any later correction.
-@item -copyts
-Copy timestamps from input to output.
-@item -shortest
-Finish encoding when the shortest input stream ends.
-@item -dts_delta_threshold
-Timestamp discontinuity delta threshold.
-@item -muxdelay @var{seconds}
-Set the maximum demux-decode delay.
-@item -muxpreload @var{seconds}
-Set the initial demux-decode delay.
-@end table
-
-@section Preset files
-
-A preset file contains a sequence of @var{option}=@var{value} pairs,
-one for each line, specifying a sequence of options which would be
-awkward to specify on the command line. Lines starting with the hash
-('#') character are ignored and are used to provide comments. Check
-the @file{ffpresets} directory in the FFmpeg source tree for examples.
-
-Preset files are specified with the @code{vpre}, @code{apre} and
-@code{spre} options. The options specified in a preset file are
-applied to the currently selected codec of the same type as the preset
-option.
-
-The argument passed to the preset options identifies the preset file
-to use according to the following rules.
-
-First ffmpeg searches for a file named @var{arg}.ffpreset in the
-directories @file{$HOME/.ffmpeg}, and in the datadir defined at
-configuration time (usually @file{PREFIX/share/ffmpeg}) in that
-order. For example, if the argument is @code{libx264-max}, it will
-search for the file @file{libx264-max.ffpreset}.
-
-If no such file is found, then ffmpeg will search for a file named
-@var{codec_name}-@var{arg}.ffpreset in the above-mentioned
-directories, where @var{codec_name} is the name of the codec to which
-the preset file options will be applied. For example, if you select
-the video codec with @code{-vcodec libx264} and use @code{-vpre max},
-then it will search for the file @file{libx264-max.ffpreset}.
-
-Finally, if the above rules failed and the argument specifies an
-absolute pathname, ffmpeg will search for that filename. This way you
-can specify the absolute and complete filename of the preset file, for
-example @file{./ffpresets/libx264-max.ffpreset}.
-
-@node FFmpeg formula evaluator
-@section FFmpeg formula evaluator
-
-When evaluating a rate control string, FFmpeg uses an internal formula
-evaluator.
-
-The following binary operators are available: @code{+}, @code{-},
-@code{*}, @code{/}, @code{^}.
-
-The following unary operators are available: @code{+}, @code{-},
-@code{(...)}.
-
-The following statements are available: @code{ld}, @code{st},
-@code{while}.
-
-The following functions are available:
-@table @var
-@item sinh(x)
-@item cosh(x)
-@item tanh(x)
-@item sin(x)
-@item cos(x)
-@item tan(x)
-@item atan(x)
-@item asin(x)
-@item acos(x)
-@item exp(x)
-@item log(x)
-@item abs(x)
-@item squish(x)
-@item gauss(x)
-@item mod(x, y)
-@item max(x, y)
-@item min(x, y)
-@item eq(x, y)
-@item gte(x, y)
-@item gt(x, y)
-@item lte(x, y)
-@item lt(x, y)
-@item bits2qp(bits)
-@item qp2bits(qp)
-@end table
-
-The following constants are available:
-@table @var
-@item PI
-@item E
-@item iTex
-@item pTex
-@item tex
-@item mv
-@item fCode
-@item iCount
-@item mcVar
-@item var
-@item isI
-@item isP
-@item isB
-@item avgQP
-@item qComp
-@item avgIITex
-@item avgPITex
-@item avgPPTex
-@item avgBPTex
-@item avgTex
-@end table
-
-@c man end
-
-@ignore
-
-@setfilename ffmpeg
-@settitle FFmpeg video converter
-
-@c man begin SEEALSO
-ffserver(1), ffplay(1) and the HTML documentation of @file{ffmpeg}.
-@c man end
-
-@c man begin AUTHOR
-Fabrice Bellard
-@c man end
-
-@end ignore
-
-@section Protocols
-
-The file name can be @file{-} to read from standard input or to write
-to standard output.
-
-FFmpeg also handles many protocols specified with an URL syntax.
-
-Use 'ffmpeg -formats' to see a list of the supported protocols.
-
-The protocol @code{http:} is currently used only to communicate with
-FFserver (see the FFserver documentation). When FFmpeg will be a
-video player it will also be used for streaming :-)
-
-@chapter Tips
-
-@itemize
-@item For streaming at very low bitrate application, use a low frame rate
-and a small GOP size. This is especially true for RealVideo where
-the Linux player does not seem to be very fast, so it can miss
-frames. An example is:
-
-@example
-ffmpeg -g 3 -r 3 -t 10 -b 50k -s qcif -f rv10 /tmp/b.rm
-@end example
-
-@item  The parameter 'q' which is displayed while encoding is the current
-quantizer. The value 1 indicates that a very good quality could
-be achieved. The value 31 indicates the worst quality. If q=31 appears
-too often, it means that the encoder cannot compress enough to meet
-your bitrate. You must either increase the bitrate, decrease the
-frame rate or decrease the frame size.
-
-@item If your computer is not fast enough, you can speed up the
-compression at the expense of the compression ratio. You can use
-'-me zero' to speed up motion estimation, and '-intra' to disable
-motion estimation completely (you have only I-frames, which means it
-is about as good as JPEG compression).
-
-@item To have very low audio bitrates, reduce the sampling frequency
-(down to 22050 Hz for MPEG audio, 22050 or 11025 for AC-3).
-
-@item To have a constant quality (but a variable bitrate), use the option
-'-qscale n' when 'n' is between 1 (excellent quality) and 31 (worst
-quality).
-
-@item When converting video files, you can use the '-sameq' option which
-uses the same quality factor in the encoder as in the decoder.
-It allows almost lossless encoding.
-
-@end itemize
-
-@bye

doc/ffmpeg_powerpc_performance_evaluation_howto.txt

@@ -1,172 +0,0 @@
-FFmpeg & evaluating performance on the PowerPC Architecture HOWTO
-
-(c) 2003-2004 Romain Dolbeau <romain@dolbeau.org>
-
-
-
-I - Introduction
-
-The PowerPC architecture and its SIMD extension AltiVec offer some
-interesting tools to evaluate performance and improve the code.
-This document tries to explain how to use those tools with FFmpeg.
-
-The architecture itself offers two ways to evaluate the performance of
-a given piece of code:
-
-1) The Time Base Registers (TBL)
-2) The Performance Monitor Counter Registers (PMC)
-
-The first ones are always available, always active, but they're not very
-accurate: the registers increment by one every four *bus* cycles. On
-my 667 Mhz tiBook (ppc7450), this means once every twenty *processor*
-cycles. So we won't use that.
-
-The PMC are much more useful: not only can they report cycle-accurate
-timing, but they can also be used to monitor many other parameters,
-such as the number of AltiVec stalls for every kind of instruction,
-or instruction cache misses. The downside is that not all processors
-support the PMC (all G3, all G4 and the 970 do support them), and
-they're inactive by default - you need to activate them with a
-dedicated tool. Also, the number of available PMC depends on the
-procesor: the various 604 have 2, the various 75x (aka. G3) have 4,
-and the various 74xx (aka G4) have 6.
-
-*WARNING*: The PowerPC 970 is not very well documented, and its PMC
-registers are 64 bits wide. To properly notify the code, you *must*
-tune for the 970 (using --tune=970), or the code will assume 32 bit
-registers.
-
-
-II - Enabling FFmpeg PowerPC performance support
-
-This needs to be done by hand. First, you need to configure FFmpeg as
-usual, but add the "--powerpc-perf-enable" option. For instance:
-
-#####
-./configure --prefix=/usr/local/ffmpeg-svn --cc=gcc-3.3 --tune=7450 --powerpc-perf-enable
-#####
-
-This will configure FFmpeg to install inside /usr/local/ffmpeg-svn,
-compiling with gcc-3.3 (you should try to use this one or a newer
-gcc), and tuning for the PowerPC 7450 (i.e. the newer G4; as a rule of
-thumb, those at 550Mhz and more). It will also enable the PMC.
-
-You may also edit the file "config.h" to enable the following line:
-
-#####
-// #define ALTIVEC_USE_REFERENCE_C_CODE 1
-#####
-
-If you enable this line, then the code will not make use of AltiVec,
-but will use the reference C code instead. This is useful to compare
-performance between two versions of the code.
-
-Also, the number of enabled PMC is defined in "libavcodec/ppc/dsputil_ppc.h":
-
-#####
-#define POWERPC_NUM_PMC_ENABLED 4
-#####
-
-If you have a G4 CPU, you can enable all 6 PMC. DO NOT enable more
-PMC than available on your CPU!
-
-Then, simply compile FFmpeg as usual (make && make install).
-
-
-
-III - Using FFmpeg PowerPC performance support
-
-This FFmeg can be used exactly as usual. But before exiting, FFmpeg
-will dump a per-function report that looks like this:
-
-#####
-PowerPC performance report
- Values are from the PMC registers, and represent whatever the
- registers are set to record.
- Function "gmc1_altivec" (pmc1):
-        min: 231
-        max: 1339867
-        avg: 558.25 (255302)
- Function "gmc1_altivec" (pmc2):
-        min: 93
-        max: 2164
-        avg: 267.31 (255302)
- Function "gmc1_altivec" (pmc3):
-        min: 72
-        max: 1987
-        avg: 276.20 (255302)
-(...)
-#####
-
-In this example, PMC1 was set to record CPU cycles, PMC2 was set to
-record AltiVec Permute Stall Cycles, and PMC3 was set to record AltiVec
-Issue Stalls.
-
-The function "gmc1_altivec" was monitored 255302 times, and the
-minimum execution time was 231 processor cycles. The max and average
-aren't much use, as it's very likely the OS interrupted execution for
-reasons of its own :-(
-
-With the exact same settings and source file, but using the reference C
-code we get:
-
-#####
-PowerPC performance report
- Values are from the PMC registers, and represent whatever the
- registers are set to record.
- Function "gmc1_altivec" (pmc1):
-        min: 592
-        max: 2532235
-        avg: 962.88 (255302)
- Function "gmc1_altivec" (pmc2):
-        min: 0
-        max: 33
-        avg: 0.00 (255302)
- Function "gmc1_altivec" (pmc3):
-        min: 0
-        max: 350
-        avg: 0.03 (255302)
-(...)
-#####
-
-592 cycles, so the fastest AltiVec execution is about 2.5x faster than
-the fastest C execution in this example. It's not perfect but it's not
-bad (well I wrote this function so I can't say otherwise :-).
-
-Once you have that kind of report, you can try to improve things by
-finding what goes wrong and fixing it; in the example above, one
-should try to diminish the number of AltiVec stalls, as this *may*
-improve performance.
-
-
-
-IV) Enabling the PMC in Mac OS X
-
-This is easy. Use "Monster" and "monster". Those tools come from
-Apple's CHUD package, and can be found hidden in the developer web
-site & FTP site. "MONster" is the graphical application, use it to
-generate a config file specifying what each register should
-monitor. Then use the command-line application "monster" to use that
-config file, and enjoy the results.
-
-Note that "MONster" can be used for many other things, but it's
-documented by Apple, it's not my subject.
-
-If you are using CHUD 4.4.2 or later, you'll notice that MONster is
-no longer available. It's been superseeded by Shark, where
-configuration of PMCs is available as a plugin.
-
-
-
-V) Enabling the PMC on Linux
-
-On linux you may use oprofile from http://oprofile.sf.net, depending on the
-version and the cpu you may need to apply a patch[1] to access a set of the
-possibile counters from the userspace application. You can always define them
-using the kernel interface /dev/oprofile/* .
-
-[1] http://dev.gentoo.org/~lu_zero/development/oprofile-g4-20060423.patch
-
---
-Romain Dolbeau <romain@dolbeau.org>
-Luca Barbato <lu_zero@gentoo.org>

doc/ffplay-doc.texi

@@ -1,159 +0,0 @@
-\input texinfo @c -*- texinfo -*-
-
-@settitle FFplay Documentation
-@titlepage
-@sp 7
-@center @titlefont{FFplay Documentation}
-@sp 3
-@end titlepage
-
-
-@chapter Introduction
-
-@c man begin DESCRIPTION
-FFplay is a very simple and portable media player using the FFmpeg
-libraries and the SDL library. It is mostly used as a testbed for the
-various FFmpeg APIs.
-@c man end
-
-@chapter Invocation
-
-@section Syntax
-@example
-@c man begin SYNOPSIS
-ffplay [options] @file{input_file}
-@c man end
-@end example
-
-@c man begin OPTIONS
-@section Main options
-
-@table @option
-@item -h
-Show help.
-@item -version
-Show version.
-@item -L
-Show license.
-@item -formats
-Show available formats, codecs, protocols, ...
-@item -x @var{width}
-Force displayed width.
-@item -y @var{height}
-Force displayed height.
-@item -s @var{size}
-Set frame size (WxH or abbreviation), needed for videos which don't
-contain a header with the frame size like raw YUV.
-@item -an
-Disable audio.
-@item -vn
-Disable video.
-@item -ss @var{pos}
-Seek to a given position in seconds.
-@item -bytes
-Seek by bytes.
-@item -nodisp
-Disable graphical display.
-@item -f @var{fmt}
-Force format.
-@end table
-
-@section Advanced options
-@table @option
-@item -pix_fmt @var{format}
-Set pixel format.
-@item -stats
-Show the stream duration, the codec parameters, the current position in
-the stream and the audio/video synchronisation drift.
-@item -debug
-Print specific debug info.
-@item -bug
-Work around bugs.
-@item -vismv
-Visualize motion vectors.
-@item -fast
-Non-spec-compliant optimizations.
-@item -genpts
-Generate pts.
-@item -rtp_tcp
-Force RTP/TCP protocol usage instead of RTP/UDP. It is only meaningful
-if you are streaming with the RTSP protocol.
-@item -sync @var{type}
-Set the master clock to audio (@code{type=audio}), video
-(@code{type=video}) or external (@code{type=ext}). Default is audio. The
-master clock is used to control audio-video synchronization. Most media
-players use audio as master clock, but in some cases (streaming or high
-quality broadcast) it is necessary to change that. This option is mainly
-used for debugging purposes.
-@item -threads @var{count}
-Set the thread count.
-@item -ast @var{audio_stream_number}
-Select the desired audio stream number, counting from 0. The number
-refers to the list of all the input audio streams. If it is greater
-than the number of audio streams minus one, then the last one is
-selected, if it is negative the audio playback is disabled.
-@item -vst @var{video_stream_number}
-Select the desired video stream number, counting from 0. The number
-refers to the list of all the input video streams. If it is greater
-than the number of video streams minus one, then the last one is
-selected, if it is negative the video playback is disabled.
-@item -sst @var{subtitle_stream_number}
-Select the desired subtitle stream number, counting from 0. The number
-refers to the list of all the input subtitle streams. If it is greater
-than the number of subtitle streams minus one, then the last one is
-selected, if it is negative the subtitle rendering is disabled.
-@end table
-
-@section While playing
-
-@table @key
-@item q, ESC
-Quit.
-
-@item f
-Toggle full screen.
-
-@item p, SPC
-Pause.
-
-@item a
-Cycle audio channel.
-
-@item v
-Cycle video channel.
-
-@item t
-Cycle subtitle channel.
-
-@item w
-Show audio waves.
-
-@item left/right
-Seek backward/forward 10 seconds.
-
-@item down/up
-Seek backward/forward 1 minute.
-
-@item mouse click
-Seek to percentage in file corresponding to fraction of width.
-
-@end table
-
-@c man end
-
-@ignore
-
-@setfilename ffplay
-@settitle FFplay media player
-
-@c man begin SEEALSO
-ffmpeg(1), ffserver(1) and the HTML documentation of @file{ffmpeg}.
-@c man end
-
-@c man begin AUTHOR
-Fabrice Bellard
-@c man end
-
-@end ignore
-
-@bye

doc/ffserver-doc.texi

@@ -1,277 +0,0 @@
-\input texinfo @c -*- texinfo -*-
-
-@settitle FFserver Documentation
-@titlepage
-@sp 7
-@center @titlefont{FFserver Documentation}
-@sp 3
-@end titlepage
-
-
-@chapter Introduction
-
-@c man begin DESCRIPTION
-FFserver is a streaming server for both audio and video. It supports
-several live feeds, streaming from files and time shifting on live feeds
-(you can seek to positions in the past on each live feed, provided you
-specify a big enough feed storage in ffserver.conf).
-
-FFserver runs in daemon mode by default; that is, it puts itself in
-the background and detaches from its TTY, unless it is launched in
-debug mode or a NoDaemon option is specified in the configuration
-file.
-
-This documentation covers only the streaming aspects of ffserver /
-ffmpeg. All questions about parameters for ffmpeg, codec questions,
-etc. are not covered here. Read @file{ffmpeg-doc.html} for more
-information.
-
-@section How does it work?
-
-FFserver receives prerecorded files or FFM streams from some ffmpeg
-instance as input, then streams them over RTP/RTSP/HTTP.
-
-An ffserver instance will listen on some port as specified in the
-configuration file. You can launch one or more instances of ffmpeg and
-send one or more FFM streams to the port where ffserver is expecting
-to receive them. Alternately, you can make ffserver launch such ffmpeg
-instances at startup.
-
-Input streams are called feeds, and each one is specified by a <Feed>
-section in the configuration file.
-
-For each feed you can have different output streams in various
-formats, each one specified by a <Stream> section in the configuration
-file.
-
-@section Status stream
-
-FFserver supports an HTTP interface which exposes the current status
-of the server.
-
-Simply point your browser to the address of the special status stream
-specified in the configuration file.
-
-For example if you have:
-@example
-<Stream status.html>
-Format status
-
-# Only allow local people to get the status
-ACL allow localhost
-ACL allow 192.168.0.0 192.168.255.255
-</Stream>
-@end example
-
-then the server will post a page with the status information when
-the special stream @file{status.html} is requested.
-
-@section What can this do?
-
-When properly configured and running, you can capture video and audio in real
-time from a suitable capture card, and stream it out over the Internet to
-either Windows Media Player or RealAudio player (with some restrictions).
-
-It can also stream from files, though that is currently broken. Very often, a
-web server can be used to serve up the files just as well.
-
-It can stream prerecorded video from .ffm files, though it is somewhat tricky
-to make it work correctly.
-
-@section What do I need?
-
-I use Linux on a 900 MHz Duron with a cheapo Bt848 based TV capture card. I'm
-using stock Linux 2.4.17 with the stock drivers. [Actually that isn't true,
-I needed some special drivers for my motherboard-based sound card.]
-
-I understand that FreeBSD systems work just fine as well.
-
-@section How do I make it work?
-
-First, build the kit. It *really* helps to have installed LAME first. Then when
-you run the ffserver ./configure, make sure that you have the
-@code{--enable-libmp3lame} flag turned on.
-
-LAME is important as it allows for streaming audio to Windows Media Player.
-Don't ask why the other audio types do not work.
-
-As a simple test, just run the following two command lines where INPUTFILE
-is some file which you can decode with ffmpeg:
-
-@example
-./ffserver -f doc/ffserver.conf &
-./ffmpeg -i INPUTFILE http://localhost:8090/feed1.ffm
-@end example
-
-At this point you should be able to go to your Windows machine and fire up
-Windows Media Player (WMP). Go to Open URL and enter
-
-@example
-    http://<linuxbox>:8090/test.asf
-@end example
-
-You should (after a short delay) see video and hear audio.
-
-WARNING: trying to stream test1.mpg doesn't work with WMP as it tries to
-transfer the entire file before starting to play.
-The same is true of AVI files.
-
-@section What happens next?
-
-You should edit the ffserver.conf file to suit your needs (in terms of
-frame rates etc). Then install ffserver and ffmpeg, write a script to start
-them up, and off you go.
-
-@section Troubleshooting
-
-@subsection I don't hear any audio, but video is fine.
-
-Maybe you didn't install LAME, or got your ./configure statement wrong. Check
-the ffmpeg output to see if a line referring to MP3 is present. If not, then
-your configuration was incorrect. If it is, then maybe your wiring is not
-set up correctly. Maybe the sound card is not getting data from the right
-input source. Maybe you have a really awful audio interface (like I do)
-that only captures in stereo and also requires that one channel be flipped.
-If you are one of these people, then export 'AUDIO_FLIP_LEFT=1' before
-starting ffmpeg.
-
-@subsection The audio and video loose sync after a while.
-
-Yes, they do.
-
-@subsection After a long while, the video update rate goes way down in WMP.
-
-Yes, it does. Who knows why?
-
-@subsection WMP 6.4 behaves differently to WMP 7.
-
-Yes, it does. Any thoughts on this would be gratefully received. These
-differences extend to embedding WMP into a web page. [There are two
-object IDs that you can use: The old one, which does not play well, and
-the new one, which does (both tested on the same system). However,
-I suspect that the new one is not available unless you have installed WMP 7].
-
-@section What else can it do?
-
-You can replay video from .ffm files that was recorded earlier.
-However, there are a number of caveats, including the fact that the
-ffserver parameters must match the original parameters used to record the
-file. If they do not, then ffserver deletes the file before recording into it.
-(Now that I write this, it seems broken).
-
-You can fiddle with many of the codec choices and encoding parameters, and
-there are a bunch more parameters that you cannot control. Post a message
-to the mailing list if there are some 'must have' parameters. Look in
-ffserver.conf for a list of the currently available controls.
-
-It will automatically generate the ASX or RAM files that are often used
-in browsers. These files are actually redirections to the underlying ASF
-or RM file. The reason for this is that the browser often fetches the
-entire file before starting up the external viewer. The redirection files
-are very small and can be transferred quickly. [The stream itself is
-often 'infinite' and thus the browser tries to download it and never
-finishes.]
-
-@section Tips
-
-* When you connect to a live stream, most players (WMP, RA, etc) want to
-buffer a certain number of seconds of material so that they can display the
-signal continuously. However, ffserver (by default) starts sending data
-in realtime. This means that there is a pause of a few seconds while the
-buffering is being done by the player. The good news is that this can be
-cured by adding a '?buffer=5' to the end of the URL. This means that the
-stream should start 5 seconds in the past -- and so the first 5 seconds
-of the stream are sent as fast as the network will allow. It will then
-slow down to real time. This noticeably improves the startup experience.
-
-You can also add a 'Preroll 15' statement into the ffserver.conf that will
-add the 15 second prebuffering on all requests that do not otherwise
-specify a time. In addition, ffserver will skip frames until a key_frame
-is found. This further reduces the startup delay by not transferring data
-that will be discarded.
-
-* You may want to adjust the MaxBandwidth in the ffserver.conf to limit
-the amount of bandwidth consumed by live streams.
-
-@section Why does the ?buffer / Preroll stop working after a time?
-
-It turns out that (on my machine at least) the number of frames successfully
-grabbed is marginally less than the number that ought to be grabbed. This
-means that the timestamp in the encoded data stream gets behind realtime.
-This means that if you say 'Preroll 10', then when the stream gets 10
-or more seconds behind, there is no Preroll left.
-
-Fixing this requires a change in the internals of how timestamps are
-handled.
-
-@section Does the @code{?date=} stuff work.
-
-Yes (subject to the limitation outlined above). Also note that whenever you
-start ffserver, it deletes the ffm file (if any parameters have changed),
-thus wiping out what you had recorded before.
-
-The format of the @code{?date=xxxxxx} is fairly flexible. You should use one
-of the following formats (the 'T' is literal):
-
-@example
-* YYYY-MM-DDTHH:MM:SS     (localtime)
-* YYYY-MM-DDTHH:MM:SSZ    (UTC)
-@end example
-
-You can omit the YYYY-MM-DD, and then it refers to the current day. However
-note that @samp{?date=16:00:00} refers to 16:00 on the current day -- this
-may be in the future and so is unlikely to be useful.
-
-You use this by adding the ?date= to the end of the URL for the stream.
-For example:   @samp{http://localhost:8080/test.asf?date=2002-07-26T23:05:00}.
-@c man end
-
-@chapter Invocation
-@section Syntax
-@example
-@c man begin SYNOPSIS
-ffserver [options]
-@c man end
-@end example
-
-@section Options
-@c man begin OPTIONS
-@table @option
-@item -version
-Show version.
-@item -L
-Show license.
-@item -formats
-Show available formats, codecs, protocols, ...
-@item -h
-Show help.
-@item -f @var{configfile}
-Use @file{configfile} instead of @file{/etc/ffserver.conf}.
-@item -n
-Enable no-launch mode. This option disables all the Launch directives
-within the various <Stream> sections. FFserver will not launch any
-ffmpeg instance, so you will have to launch them manually.
-@item -d
-Enable debug mode. This option increases log verbosity, directs log
-messages to stdout and causes ffserver to run in the foreground
-rather than as a daemon.
-@end table
-@c man end
-
-@ignore
-
-@setfilename ffserver
-@settitle FFserver video server
-
-@c man begin SEEALSO
-ffmpeg(1), ffplay(1), the @file{ffmpeg/doc/ffserver.conf} example and
-the HTML documentation of @file{ffmpeg}.
-@c man end
-
-@c man begin AUTHOR
-Fabrice Bellard
-@c man end
-
-@end ignore
-
-@bye

doc/ffserver.conf

@@ -1,356 +0,0 @@
-# Port on which the server is listening. You must select a different
-# port from your standard HTTP web server if it is running on the same
-# computer.
-Port 8090
-
-# Address on which the server is bound. Only useful if you have
-# several network interfaces.
-BindAddress 0.0.0.0
-
-# Number of simultaneous HTTP connections that can be handled. It has
-# to be defined *before* the MaxClients parameter, since it defines the
-# MaxClients maximum limit.
-MaxHTTPConnections 2000
-
-# Number of simultaneous requests that can be handled. Since FFServer
-# is very fast, it is more likely that you will want to leave this high
-# and use MaxBandwidth, below.
-MaxClients 1000
-
-# This the maximum amount of kbit/sec that you are prepared to
-# consume when streaming to clients.
-MaxBandwidth 1000
-
-# Access log file (uses standard Apache log file format)
-# '-' is the standard output.
-CustomLog -
-
-# Suppress that if you want to launch ffserver as a daemon.
-NoDaemon
-
-
-##################################################################
-# Definition of the live feeds. Each live feed contains one video
-# and/or audio sequence coming from an ffmpeg encoder or another
-# ffserver. This sequence may be encoded simultaneously with several
-# codecs at several resolutions.
-
-<Feed feed1.ffm>
-
-# You must use 'ffmpeg' to send a live feed to ffserver. In this
-# example, you can type:
-#
-# ffmpeg http://localhost:8090/feed1.ffm
-
-# ffserver can also do time shifting. It means that it can stream any
-# previously recorded live stream. The request should contain:
-# "http://xxxx?date=[YYYY-MM-DDT][[HH:]MM:]SS[.m...]".You must specify
-# a path where the feed is stored on disk. You also specify the
-# maximum size of the feed, where zero means unlimited. Default:
-# File=/tmp/feed_name.ffm FileMaxSize=5M
-File /tmp/feed1.ffm
-FileMaxSize 200K
-
-# You could specify
-# ReadOnlyFile /saved/specialvideo.ffm
-# This marks the file as readonly and it will not be deleted or updated.
-
-# Specify launch in order to start ffmpeg automatically.
-# First ffmpeg must be defined with an appropriate path if needed,
-# after that options can follow, but avoid adding the http:// field
-#Launch ffmpeg
-
-# Only allow connections from localhost to the feed.
-ACL allow 127.0.0.1
-
-</Feed>
-
-
-##################################################################
-# Now you can define each stream which will be generated from the
-# original audio and video stream. Each format has a filename (here
-# 'test1.mpg'). FFServer will send this stream when answering a
-# request containing this filename.
-
-<Stream test1.mpg>
-
-# coming from live feed 'feed1'
-Feed feed1.ffm
-
-# Format of the stream : you can choose among:
-# mpeg       : MPEG-1 multiplexed video and audio
-# mpegvideo  : only MPEG-1 video
-# mp2        : MPEG-2 audio (use AudioCodec to select layer 2 and 3 codec)
-# ogg        : Ogg format (Vorbis audio codec)
-# rm         : RealNetworks-compatible stream. Multiplexed audio and video.
-# ra         : RealNetworks-compatible stream. Audio only.
-# mpjpeg     : Multipart JPEG (works with Netscape without any plugin)
-# jpeg       : Generate a single JPEG image.
-# asf        : ASF compatible streaming (Windows Media Player format).
-# swf        : Macromedia Flash compatible stream
-# avi        : AVI format (MPEG-4 video, MPEG audio sound)
-Format mpeg
-
-# Bitrate for the audio stream. Codecs usually support only a few
-# different bitrates.
-AudioBitRate 32
-
-# Number of audio channels: 1 = mono, 2 = stereo
-AudioChannels 1
-
-# Sampling frequency for audio. When using low bitrates, you should
-# lower this frequency to 22050 or 11025. The supported frequencies
-# depend on the selected audio codec.
-AudioSampleRate 44100
-
-# Bitrate for the video stream
-VideoBitRate 64
-
-# Ratecontrol buffer size
-VideoBufferSize 40
-
-# Number of frames per second
-VideoFrameRate 3
-
-# Size of the video frame: WxH (default: 160x128)
-# The following abbreviations are defined: sqcif, qcif, cif, 4cif, qqvga,
-# qvga, vga, svga, xga, uxga, qxga, sxga, qsxga, hsxga, wvga, wxga, wsxga,
-# wuxga, woxga, wqsxga, wquxga, whsxga, whuxga, cga, ega, hd480, hd720,
-# hd1080
-VideoSize 160x128
-
-# Transmit only intra frames (useful for low bitrates, but kills frame rate).
-#VideoIntraOnly
-
-# If non-intra only, an intra frame is transmitted every VideoGopSize
-# frames. Video synchronization can only begin at an intra frame.
-VideoGopSize 12
-
-# More MPEG-4 parameters
-# VideoHighQuality
-# Video4MotionVector
-
-# Choose your codecs:
-#AudioCodec mp2
-#VideoCodec mpeg1video
-
-# Suppress audio
-#NoAudio
-
-# Suppress video
-#NoVideo
-
-#VideoQMin 3
-#VideoQMax 31
-
-# Set this to the number of seconds backwards in time to start. Note that
-# most players will buffer 5-10 seconds of video, and also you need to allow
-# for a keyframe to appear in the data stream.
-#Preroll 15
-
-# ACL:
-
-# You can allow ranges of addresses (or single addresses)
-#ACL ALLOW <first address> <last address>
-
-# You can deny ranges of addresses (or single addresses)
-#ACL DENY <first address> <last address>
-
-# You can repeat the ACL allow/deny as often as you like. It is on a per
-# stream basis. The first match defines the action. If there are no matches,
-# then the default is the inverse of the last ACL statement.
-#
-# Thus 'ACL allow localhost' only allows access from localhost.
-# 'ACL deny 1.0.0.0 1.255.255.255' would deny the whole of network 1 and
-# allow everybody else.
-
-</Stream>
-
-
-##################################################################
-# Example streams
-
-
-# Multipart JPEG
-
-#<Stream test.mjpg>
-#Feed feed1.ffm
-#Format mpjpeg
-#VideoFrameRate 2
-#VideoIntraOnly
-#NoAudio
-#Strict -1
-#</Stream>
-
-
-# Single JPEG
-
-#<Stream test.jpg>
-#Feed feed1.ffm
-#Format jpeg
-#VideoFrameRate 2
-#VideoIntraOnly
-##VideoSize 352x240
-#NoAudio
-#Strict -1
-#</Stream>
-
-
-# Flash
-
-#<Stream test.swf>
-#Feed feed1.ffm
-#Format swf
-#VideoFrameRate 2
-#VideoIntraOnly
-#NoAudio
-#</Stream>
-
-
-# ASF compatible
-
-<Stream test.asf>
-Feed feed1.ffm
-Format asf
-VideoFrameRate 15
-VideoSize 352x240
-VideoBitRate 256
-VideoBufferSize 40
-VideoGopSize 30
-AudioBitRate 64
-StartSendOnKey
-</Stream>
-
-
-# MP3 audio
-
-#<Stream test.mp3>
-#Feed feed1.ffm
-#Format mp2
-#AudioCodec mp3
-#AudioBitRate 64
-#AudioChannels 1
-#AudioSampleRate 44100
-#NoVideo
-#</Stream>
-
-
-# Ogg Vorbis audio
-
-#<Stream test.ogg>
-#Feed feed1.ffm
-#Title "Stream title"
-#AudioBitRate 64
-#AudioChannels 2
-#AudioSampleRate 44100
-#NoVideo
-#</Stream>
-
-
-# Real with audio only at 32 kbits
-
-#<Stream test.ra>
-#Feed feed1.ffm
-#Format rm
-#AudioBitRate 32
-#NoVideo
-#NoAudio
-#</Stream>
-
-
-# Real with audio and video at 64 kbits
-
-#<Stream test.rm>
-#Feed feed1.ffm
-#Format rm
-#AudioBitRate 32
-#VideoBitRate 128
-#VideoFrameRate 25
-#VideoGopSize 25
-#NoAudio
-#</Stream>
-
-
-##################################################################
-# A stream coming from a file: you only need to set the input
-# filename and optionally a new format. Supported conversions:
-#    AVI -> ASF
-
-#<Stream file.rm>
-#File "/usr/local/httpd/htdocs/tlive.rm"
-#NoAudio
-#</Stream>
-
-#<Stream file.asf>
-#File "/usr/local/httpd/htdocs/test.asf"
-#NoAudio
-#Author "Me"
-#Copyright "Super MegaCorp"
-#Title "Test stream from disk"
-#Comment "Test comment"
-#</Stream>
-
-
-##################################################################
-# RTSP examples
-#
-# You can access this stream with the RTSP URL:
-#   rtsp://localhost:5454/test1-rtsp.mpg
-#
-# A non-standard RTSP redirector is also created. Its URL is:
-#   http://localhost:8090/test1-rtsp.rtsp
-
-#<Stream test1-rtsp.mpg>
-#Format rtp
-#File "/usr/local/httpd/htdocs/test1.mpg"
-#</Stream>
-
-
-##################################################################
-# SDP/multicast examples
-#
-# If you want to send your stream in multicast, you must set the
-# multicast address with MulticastAddress. The port and the TTL can
-# also be set.
-#
-# An SDP file is automatically generated by ffserver by adding the
-# 'sdp' extension to the stream name (here
-# http://localhost:8090/test1-sdp.sdp). You should usually give this
-# file to your player to play the stream.
-#
-# The 'NoLoop' option can be used to avoid looping when the stream is
-# terminated.
-
-#<Stream test1-sdp.mpg>
-#Format rtp
-#File "/usr/local/httpd/htdocs/test1.mpg"
-#MulticastAddress 224.124.0.1
-#MulticastPort 5000
-#MulticastTTL 16
-#NoLoop
-#</Stream>
-
-
-##################################################################
-# Special streams
-
-# Server status
-
-<Stream stat.html>
-Format status
-
-# Only allow local people to get the status
-ACL allow localhost
-ACL allow 192.168.0.0 192.168.255.255
-
-#FaviconURL http://pond1.gladstonefamily.net:8080/favicon.ico
-</Stream>
-
-
-# Redirect index.html to the appropriate site
-
-<Redirect index.html>
-URL http://www.ffmpeg.org/
-</Redirect>
-
-

doc/hooks.texi

@@ -1,299 +0,0 @@
-\input texinfo @c -*- texinfo -*-
-
-@settitle Video Hook Documentation
-@titlepage
-@sp 7
-@center @titlefont{Video Hook Documentation}
-@sp 3
-@end titlepage
-
-
-@chapter Introduction
-
-@var{Please be aware that vhook is deprecated, and hence its development is
-frozen (bug fixes are still accepted).
-The substitute will be 'libavfilter', the result of our 'Video Filter API'
-Google Summer of Code project. You may monitor its progress by subscribing to
-the ffmpeg-soc mailing list at
-@url{http://lists.mplayerhq.hu/mailman/listinfo/ffmpeg-soc}.}
-
-The video hook functionality is designed (mostly) for live video. It allows
-the video to be modified or examined between the decoder and the encoder.
-
-Any number of hook modules can be placed inline, and they are run in the
-order that they were specified on the ffmpeg command line.
-
-The video hook modules are provided for use as a base for your own modules,
-and are described below.
-
-Modules are loaded using the -vhook option to ffmpeg. The value of this parameter
-is a space separated list of arguments. The first is the module name, and the rest
-are passed as arguments to the Configure function of the module.
-
-The modules are dynamic libraries: They have different suffixes (.so, .dll, .dylib)
-depending on your platform. And your platform dictates if they need to be
-somewhere in your PATH, or in your LD_LIBRARY_PATH. Otherwise you will need to
-specify the full path of the vhook file that you are using.
-
-@section null.c
-
-This does nothing. Actually it converts the input image to RGB24 and then converts
-it back again. This is meant as a sample that you can use to test your setup.
-
-@section fish.c
-
-This implements a 'fish detector'. Essentially it converts the image into HSV
-space and tests whether more than a certain percentage of the pixels fall into
-a specific HSV cuboid. If so, then the image is saved into a file for processing
-by other bits of code.
-
-Why use HSV? It turns out that HSV cuboids represent a more compact range of
-colors than would an RGB cuboid.
-
-@section imlib2.c
-
-This module implements a text overlay for a video image. Currently it
-supports a fixed overlay or reading the text from a file. The string
-is passed through strftime() so that it is easy to imprint the date and
-time onto the image.
-
-This module depends on the external library imlib2, available on
-Sourceforge, among other places, if it is not already installed on
-your system.
-
-You may also overlay an image (even semi-transparent) like TV stations do.
-You may move either the text or the image around your video to create
-scrolling credits, for example.
-
-The font file used is looked for in a FONTPATH environment variable, and
-prepended to the point size as a command line option and can be specified
-with the full path to the font file, as in:
-@example
--F /usr/X11R6/lib/X11/fonts/TTF/VeraBd.ttf/20
-@end example
-where 20 is the point size.
-
-You can specify the filename to read RGB color names from. If it is not
-specified, these defaults are used: @file{/usr/share/X11/rgb.txt} and
-@file{/usr/lib/X11/rgb.txt}
-
-Options:
-@multitable @columnfractions .2 .8
-@item @option{-C <rgb.txt>}   @tab The filename to read RGB color names from
-@item @option{-c <color>}     @tab The color of the text
-@item @option{-F <fontname>}  @tab The font face and size
-@item @option{-t <text>}      @tab The text
-@item @option{-f <filename>}  @tab The filename to read text from
-@item @option{-x <expression>}@tab x coordinate of text or image
-@item @option{-y <expression>}@tab y coordinate of text or image
-@item @option{-i <filename>}  @tab The filename to read a image from
-@item @option{-R <expression>}@tab Value for R color
-@item @option{-G <expression>}@tab Value for G color
-@item @option{-B <expression>}@tab Value for B color
-@item @option{-A <expression>}@tab Value for Alpha channel
-@end multitable
-
-Expressions are functions of these variables:
-@multitable @columnfractions .2 .8
-@item @var{N} @tab frame number (starting at zero)
-@item @var{H} @tab frame height
-@item @var{W} @tab frame width
-@item @var{h} @tab image height
-@item @var{w} @tab image width
-@item @var{X} @tab previous x coordinate of text or image
-@item @var{Y} @tab previous y coordinate of text or image
-@end multitable
-
-You may also use the constants @var{PI}, @var{E}, and the math functions available at the
-FFmpeg formula evaluator at (@url{ffmpeg-doc.html#SEC13}), except @var{bits2qp(bits)}
-and @var{qp2bits(qp)}.
-
-Usage examples:
-
-@example
-   # Remember to set the path to your fonts
-   FONTPATH="/cygdrive/c/WINDOWS/Fonts/"
-   FONTPATH="$FONTPATH:/usr/share/imlib2/data/fonts/"
-   FONTPATH="$FONTPATH:/usr/X11R6/lib/X11/fonts/TTF/"
-   export FONTPATH
-
-   # Bulb dancing in a Lissajous pattern
-   ffmpeg -i input.avi -vhook \
-     'vhook/imlib2.dll -x W*(0.5+0.25*sin(N/47*PI))-w/2 -y H*(0.5+0.50*cos(N/97*PI))-h/2 -i /usr/share/imlib2/data/images/bulb.png' \
-     -acodec copy -sameq output.avi
-
-   # Text scrolling
-   ffmpeg -i input.avi -vhook \
-     'vhook/imlib2.dll -c red -F Vera.ttf/20 -x 150+0.5*N -y 70+0.25*N -t Hello' \
-     -acodec copy -sameq output.avi
-
-   # Date and time stamp, security-camera style:
-   ffmpeg -r 29.97 -s 320x256 -f video4linux -i /dev/video0 \
-     -vhook 'vhook/imlib2.so -x 0 -y 0 -i black-260x20.png' \
-     -vhook 'vhook/imlib2.so -c white -F VeraBd.ttf/12 -x 0 -y 0 -t %A-%D-%T' \
-     output.avi
-
-     In this example the video is captured from the first video capture card as a
-     320x256 AVI, and a black 260 by 20 pixel PNG image is placed in the upper
-     left corner, with the day, date and time overlaid on it in Vera Bold 12
-     point font. A simple black PNG file 260 pixels wide and 20 pixels tall
-     was created in the GIMP for this purpose.
-
-   # Scrolling credits from a text file
-   ffmpeg -i input.avi -vhook \
-     'vhook/imlib2.so -c white -F VeraBd.ttf/16 -x 100 -y -1.0*N -f credits.txt' \
-     -sameq output.avi
-
-     In this example, the text is stored in a file, and is positioned 100
-     pixels from the left hand edge of the video. The text is scrolled from the
-     bottom up. Making the y factor positive will scroll from the top down.
-     Increasing the magnitude of the y factor makes the text scroll faster,
-     decreasing it makes it scroll slower. Hint: Blank lines containing only
-     a newline are treated as end-of-file. To create blank lines, use lines
-     that consist of space characters only.
-
-   # Scrolling credits with custom color from a text file
-   ffmpeg -i input.avi -vhook \
-     'vhook/imlib2.so -C rgb.txt -c CustomColor1 -F VeraBd.ttf/16 -x 100 -y -1.0*N -f credits.txt' \
-     -sameq output.avi
-
-     This example does the same as the one above, but specifies an rgb.txt file
-     to be used, which has a custom-made color in it.
-
-   # Variable colors
-   ffmpeg -i input.avi -vhook \
-     'vhook/imlib2.so -t Hello -R abs(255*sin(N/47*PI)) -G abs(255*sin(N/47*PI)) -B abs(255*sin(N/47*PI))' \
-     -sameq output.avi
-
-     In this example, the color for the text goes up and down from black to
-     white.
-
-   # Text fade-out
-   ffmpeg -i input.avi -vhook \
-     'vhook/imlib2.so -t Hello -A max(0,255-exp(N/47))' \
-     -sameq output.avi
-
-     In this example, the text fades out in about 10 seconds for a 25 fps input
-     video file.
-
-   # scrolling credits from a graphics file
-   ffmpeg -sameq -i input.avi \
-     -vhook 'vhook/imlib2.so -x 0 -y -1.0*N -i credits.png' output.avi
-
-     In this example, a transparent PNG file the same width as the video
-     (e.g. 320 pixels), but very long, (e.g. 3000 pixels), was created, and
-     text, graphics, brushstrokes, etc, were added to the image. The image
-     is then scrolled up, from the bottom of the frame.
-
-@end example
-
-@section ppm.c
-
-It's basically a launch point for a PPM pipe, so you can use any
-executable (or script) which consumes a PPM on stdin and produces a PPM
-on stdout (and flushes each frame). The Netpbm utilities are a series of
-such programs.
-
-A list of them is here:
-
-@url{http://netpbm.sourceforge.net/doc/directory.html}
-
-Usage example:
-
-@example
-ffmpeg -i input -vhook "/path/to/ppm.so some-ppm-filter args" output
-@end example
-
-@section drawtext.c
-
-This module implements a text overlay for a video image. Currently it
-supports a fixed overlay or reading the text from a file. The string
-is passed through strftime() so that it is easy to imprint the date and
-time onto the image.
-
-Features:
-@itemize @minus
-@item TrueType, Type1 and others via the FreeType2 library
-@item Font kerning (better output)
-@item Line Wrap (put the text that doesn't fit one line on the next line)
-@item Background box (currently in development)
-@item Outline
-@end itemize
-
-Options:
-@multitable @columnfractions .2 .8
-@item @option{-c <color>}          @tab Foreground color of the text ('internet' way) <#RRGGBB> [default #FFFFFF]
-@item @option{-C <color>}          @tab Background color of the text ('internet' way) <#RRGGBB> [default #000000]
-@item @option{-f <font-filename>}  @tab font file to use
-@item @option{-t <text>}           @tab text to display
-@item @option{-T <filename>}       @tab file to read text from
-@item @option{-x <pos>}            @tab x coordinate of the start of the text
-@item @option{-y <pos>}            @tab y coordinate of the start of the text
-@end multitable
-
-Text fonts are being looked for in a FONTPATH environment variable.
-If the FONTPATH environment variable is not available, or is not checked by
-your target (i.e. Cygwin), then specify the full path to the font file as in:
-@example
--f /usr/X11R6/lib/X11/fonts/TTF/VeraBd.ttf
-@end example
-
-Usage Example:
-@example
-   # Remember to set the path to your fonts
-   FONTPATH="/cygdrive/c/WINDOWS/Fonts/"
-   FONTPATH="$FONTPATH:/usr/share/imlib2/data/fonts/"
-   FONTPATH="$FONTPATH:/usr/X11R6/lib/X11/fonts/TTF/"
-   export FONTPATH
-
-   # Time and date display
-   ffmpeg -f video4linux2 -i /dev/video0 \
-   -vhook 'vhook/drawtext.so -f VeraBd.ttf -t %A-%D-%T' movie.mpg
-
-     This example grabs video from the first capture card and outputs it to an
-     MPEG video, and places "Weekday-dd/mm/yy-hh:mm:ss" at the top left of the
-     frame, updated every second, using the Vera Bold TrueType Font, which
-     should exist in: /usr/X11R6/lib/X11/fonts/TTF/
-@end example
-
-Check the man page for strftime() for all the various ways you can format
-the date and time.
-
-@section watermark.c
-
-Command Line options:
-@multitable @columnfractions .2 .8
-@item @option{-m [0|1]}            @tab Mode (default: 0, see below)
-@item @option{-t 000000 - FFFFFF}  @tab Threshold, six digit hex number
-@item @option{-f <filename>}       @tab Watermark image filename, must be specified!
-@end multitable
-
-MODE 0:
- The watermark picture works like this (assuming color intensities 0..0xFF):
- Per color do this:
- If mask color is 0x80, no change to the original frame.
- If mask color is < 0x80 the absolute difference is subtracted from the
- frame. If result < 0, result = 0.
- If mask color is > 0x80 the absolute difference is added to the
- frame. If result > 0xFF, result = 0xFF.
-
- You can override the 0x80 level with the -t flag. E.g. if threshold is
- 000000 the color value of watermark is added to the destination.
-
- This way a mask that is visible both in light and dark pictures can be made
- (e.g. by using a picture generated by the Gimp and the bump map tool).
-
- An example watermark file is at:
- @url{http://engene.se/ffmpeg_watermark.gif}
-
-MODE 1:
- Per color do this:
- If mask color > threshold color then the watermark pixel is used.
-
-Example usage:
-@example
-   ffmpeg -i infile -vhook '/path/watermark.so -f wm.gif' -an out.mov
-   ffmpeg -i infile -vhook '/path/watermark.so -f wm.gif -m 1 -t 222222' -an out.mov
-@end example
-
-@bye

doc/issue_tracker.txt

@@ -1,228 +0,0 @@
-FFmpeg's bug/patch/feature request tracker manual
-=================================================
-
-NOTE: This is a draft.
-
-Overview:
----------
-FFmpeg uses Roundup for tracking issues, new issues and changes to
-existing issues can be done through a web interface and through email.
-It is possible to subscribe to individual issues by adding yourself to the
-nosy list or to subscribe to the ffmpeg-issues mailing list which receives
-a mail for every change to every issue. Replies to such mails will also
-be properly added to the respective issue.
-(the above does all work already after light testing)
-The subscription URL for the ffmpeg-issues list is:
-http://live.polito/mailman/listinfo/ffmpeg-issues
-The URL of the webinterface of the tracker is:
-http(s)://roundup.ffmpeg/roundup/ffmpeg/
-Note the URLs in this document are obfuscated, you must append the top level
-domain for non-profit organizations to the tracker, and of Italy to the
-mailing list.
-
-Email Interface:
-----------------
-There is a mailing list to which all new issues and changes to existing issues
-are sent. You can subscribe through
-http://live.polito/mailman/listinfo/ffmpeg-issues
-Replies to messages there will have their text added to the specific issues.
-Attachments will be added as if they had been uploaded via the web interface.
-You can change the status, substatus, topic, ... by changing the subject in
-your reply like:
-Re: [issue94] register_avcodec and allcodecs.h [type=patch;status=open;substatus=approved]
-Roundup will then change things as you requested and remove the [...] from
-the subject before forwarding the mail to the mailing list.
-
-
-NOTE: issue = (bug report || patch || feature request)
-
-Type:
------
-bug
-    An error, flaw, mistake, failure, or fault in FFmpeg or libav* that
-    prevents it from behaving as intended.
-
-feature request
-    Request of support for encoding or decoding of a new codec, container
-    or variant.
-    Request of support for more, less or plain different output or behavior
-    where the current implementation cannot be considered wrong.
-
-patch
-    A patch as generated by diff which conforms to the patch submission and
-    development policy.
-
-
-Priority:
----------
-critical
-    Bugs and patches which deal with data loss and security issues.
-    No feature request can be critical.
-
-important
-    Bugs which make FFmpeg unusable for a significant number of users, and
-    patches fixing them.
-    Examples here might be completely broken MPEG-4 decoding or a build issue
-    on Linux.
-    While broken 4xm decoding or a broken OS/2 build would not be important,
-    the separation to normal is somewhat fuzzy.
-    For feature requests this priority would be used for things many people
-    want.
-
-normal
-
-
-minor
-    Bugs and patches about things like spelling errors, "mp2" instead of
-    "mp3" being shown and such.
-    Feature requests about things few people want or which do not make a big
-    difference.
-
-wish
-    Something that is desirable to have but that there is no urgency at
-    all to implement, e.g. something completely cosmetic like a website
-    restyle or a personalized doxy template or the FFmpeg logo.
-    This priority is not valid for bugs.
-
-
-Status:
--------
-new
-    initial state
-
-open
-    intermediate states
-
-closed
-    final state
-
-
-Type/Status/Substatus:
-----------
-*/new/new
-    Initial state of new bugs, patches and feature requests submitted by
-    users.
-
-*/open/open
-    Issues which have been briefly looked at and which did not look outright
-    invalid.
-    This implicates that no real more detailed state applies yet. Conversely,
-    the more detailed states below implicate that the issue has been briefly
-    looked at.
-
-*/closed/duplicate
-    Bugs, patches or feature requests which are duplicates.
-    Note that patches dealing with the same thing in a different way are not
-    duplicates.
-    Note, if you mark something as duplicate, do not forget setting the
-    superseder so bug reports are properly linked.
-
-*/closed/invalid
-    Bugs caused by user errors, random ineligible or otherwise nonsense stuff.
-
-*/closed/needs_more_info
-    Issues for which some information has been requested by the developers,
-    but which has not been provided by anyone within reasonable time.
-
-bug/open/reproduced
-    Bugs which have been reproduced.
-
-bug/open/analyzed
-    Bugs which have been analyzed and where it is understood what causes them
-    and which exact chain of events triggers them. This analysis should be
-    available as a message in the bug report.
-    Note, do not change the status to analyzed without also providing a clear
-    and understandable analysis.
-    This state implicates that the bug either has been reproduced or that
-    reproduction is not needed as the bug is already understood.
-
-bug/open/needs_more_info
-    Bug reports which are incomplete and or where more information is needed
-    from the submitter or another person who can provide it.
-    This state implicates that the bug has not been analyzed or reproduced.
-    Note, the idea behind needs_more_info is to offload work from the
-    developers to the users whenever possible.
-
-bug/closed/fixed
-    Bugs which have to the best of our knowledge been fixed.
-
-bug/closed/wont_fix
-    Bugs which we will not fix. Possible reasons include legality, high
-    complexity for the sake of supporting obscure corner cases, speed loss
-    for similarly esoteric purposes, et cetera.
-    This also means that we would reject a patch.
-    If we are just too lazy to fix a bug then the correct state is open
-    and unassigned. Closed means that the case is closed which is not
-    the case if we are just waiting for a patch.
-
-bug/closed/works_for_me
-    Bugs for which sufficient information was provided to reproduce but
-    reproduction failed - that is the code seems to work correctly to the
-    best of our knowledge.
-
-patch/open/approved
-    Patches which have been reviewed and approved by a developer.
-    Such patches can be applied anytime by any other developer after some
-    reasonable testing (compile + regression tests + does the patch do
-    what the author claimed).
-
-patch/open/needs_changes
-    Patches which have been reviewed and need changes to be accepted.
-
-patch/closed/applied
-    Patches which have been applied.
-
-patch/closed/rejected
-    Patches which have been rejected.
-
-feature_request/open/needs_more_info
-    Feature requests where it is not clear what exactly is wanted
-    (these also could be closed as invalid ...).
-
-feature_request/closed/implemented
-    Feature requests which have been implemented.
-
-feature_request/closed/wont_implement
-    Feature requests which will not be implemented. The reasons here could
-    be legal, philosophical or others.
-
-Note, please do not use type-status-substatus combinations other than the
-above without asking on ffmpeg-dev first!
-
-Note2, if you provide the requested info do not forget to remove the
-needs_more_info substate.
-
-Topic:
-------
-A topic is a tag you should add to your issue in order to make grouping them
-easier.
-
-avcodec
-    issues in libavcodec/*
-
-avformat
-    issues in libavformat/*
-
-avutil
-    issues in libavutil/*
-
-regression test
-    issues in tests/*
-
-ffmpeg
-    issues in or related to ffmpeg.c
-
-ffplay
-    issues in or related to ffplay.c
-
-ffserver
-    issues in or related to ffserver.c
-
-build system
-    issues in or related to configure/Makefile
-
-regression
-    bugs which were working in a past revision
-
-roundup
-    issues related to our issue tracker

doc/optimization.txt

@@ -1,235 +0,0 @@
-optimization Tips (for libavcodec):
-===================================
-
-What to optimize:
------------------
-If you plan to do non-x86 architecture specific optimizations (SIMD normally),
-then take a look in the x86/ directory, as most important functions are
-already optimized for MMX.
-
-If you want to do x86 optimizations then you can either try to finetune the
-stuff in the x86 directory or find some other functions in the C source to
-optimize, but there aren't many left.
-
-
-Understanding these overoptimized functions:
---------------------------------------------
-As many functions tend to be a bit difficult to understand because
-of optimizations, it can be hard to optimize them further, or write
-architecture-specific versions. It is recommended to look at older
-revisions of the interesting files (for a web frontend try ViewVC at
-http://svn.ffmpeg.org/ffmpeg/trunk/).
-Alternatively, look into the other architecture-specific versions in
-the x86/, ppc/, alpha/ subdirectories. Even if you don't exactly
-comprehend the instructions, it could help understanding the functions
-and how they can be optimized.
-
-NOTE: If you still don't understand some function, ask at our mailing list!!!
-(http://lists.mplayerhq.hu/mailman/listinfo/ffmpeg-devel)
-
-
-When is an optimization justified?
-----------------------------------
-Normally, clean and simple optimizations for widely used codecs are
-justified even if they only achieve an overall speedup of 0.1%. These
-speedups accumulate and can make a big difference after awhile. Also, if
-none of the following factors get worse due to an optimization -- speed,
-binary code size, source size, source readability -- and at least one
-factor improves, then an optimization is always a good idea even if the
-overall gain is less than 0.1%. For obscure codecs that are not often
-used, the goal is more toward keeping the code clean, small, and
-readable instead of making it 1% faster.
-
-
-WTF is that function good for ....:
------------------------------------
-The primary purpose of this list is to avoid wasting time optimizing functions
-which are rarely used.
-
-put(_no_rnd)_pixels{,_x2,_y2,_xy2}
-    Used in motion compensation (en/decoding).
-
-avg_pixels{,_x2,_y2,_xy2}
-    Used in motion compensation of B-frames.
-    These are less important than the put*pixels functions.
-
-avg_no_rnd_pixels*
-    unused
-
-pix_abs16x16{,_x2,_y2,_xy2}
-    Used in motion estimation (encoding) with SAD.
-
-pix_abs8x8{,_x2,_y2,_xy2}
-    Used in motion estimation (encoding) with SAD of MPEG-4 4MV only.
-    These are less important than the pix_abs16x16* functions.
-
-put_mspel8_mc* / wmv2_mspel8*
-    Used only in WMV2.
-    it is not recommended that you waste your time with these, as WMV2
-    is an ugly and relatively useless codec.
-
-mpeg4_qpel* / *qpel_mc*
-    Used in MPEG-4 qpel motion compensation (encoding & decoding).
-    The qpel8 functions are used only for 4mv,
-    the avg_* functions are used only for B-frames.
-    Optimizing them should have a significant impact on qpel
-    encoding & decoding.
-
-qpel{8,16}_mc??_old_c / *pixels{8,16}_l4
-    Just used to work around a bug in an old libavcodec encoder version.
-    Don't optimize them.
-
-tpel_mc_func {put,avg}_tpel_pixels_tab
-    Used only for SVQ3, so only optimize them if you need fast SVQ3 decoding.
-
-add_bytes/diff_bytes
-    For huffyuv only, optimize if you want a faster ffhuffyuv codec.
-
-get_pixels / diff_pixels
-    Used for encoding, easy.
-
-clear_blocks
-    easiest to optimize
-
-gmc
-    Used for MPEG-4 gmc.
-    Optimizing this should have a significant effect on the gmc decoding
-    speed.
-
-gmc1
-    Used for chroma blocks in MPEG-4 gmc with 1 warp point
-    (there are 4 luma & 2 chroma blocks per macroblock, so
-    only 1/3 of the gmc blocks use this, the other 2/3
-    use the normal put_pixel* code, but only if there is
-    just 1 warp point).
-    Note: DivX5 gmc always uses just 1 warp point.
-
-pix_sum
-    Used for encoding.
-
-hadamard8_diff / sse / sad == pix_norm1 / dct_sad / quant_psnr / rd / bit
-    Specific compare functions used in encoding, it depends upon the
-    command line switches which of these are used.
-    Don't waste your time with dct_sad & quant_psnr, they aren't
-    really useful.
-
-put_pixels_clamped / add_pixels_clamped
-    Used for en/decoding in the IDCT, easy.
-    Note, some optimized IDCTs have the add/put clamped code included and
-    then put_pixels_clamped / add_pixels_clamped will be unused.
-
-idct/fdct
-    idct (encoding & decoding)
-    fdct (encoding)
-    difficult to optimize
-
-dct_quantize_trellis
-    Used for encoding with trellis quantization.
-    difficult to optimize
-
-dct_quantize
-    Used for encoding.
-
-dct_unquantize_mpeg1
-    Used in MPEG-1 en/decoding.
-
-dct_unquantize_mpeg2
-    Used in MPEG-2 en/decoding.
-
-dct_unquantize_h263
-    Used in MPEG-4/H.263 en/decoding.
-
-FIXME remaining functions?
-BTW, most of these functions are in dsputil.c/.h, some are in mpegvideo.c/.h.
-
-
-
-Alignment:
-Some instructions on some architectures have strict alignment restrictions,
-for example most SSE/SSE2 instructions on x86.
-The minimum guaranteed alignment is written in the .h files, for example:
-    void (*put_pixels_clamped)(const DCTELEM *block/*align 16*/, UINT8 *pixels/*align 8*/, int line_size);
-
-
-General Tips:
--------------
-Use asm loops like:
-__asm__(
-    "1: ....
-    ...
-    "jump_instruciton ....
-Do not use C loops:
-do{
-    __asm__(
-        ...
-}while()
-
-Use __asm__() instead of intrinsics. The latter requires a good optimizing compiler
-which gcc is not.
-
-
-Links:
-======
-http://www.aggregate.org/MAGIC/
-
-x86-specific:
--------------
-http://developer.intel.com/design/pentium4/manuals/248966.htm
-
-The IA-32 Intel Architecture Software Developer's Manual, Volume 2:
-Instruction Set Reference
-http://developer.intel.com/design/pentium4/manuals/245471.htm
-
-http://www.agner.org/assem/
-
-AMD Athlon Processor x86 Code Optimization Guide:
-http://www.amd.com/us-en/assets/content_type/white_papers_and_tech_docs/22007.pdf
-
-
-ARM-specific:
--------------
-ARM Architecture Reference Manual (up to ARMv5TE):
-http://www.arm.com/community/university/eulaarmarm.html
-
-Procedure Call Standard for the ARM Architecture:
-http://www.arm.com/pdfs/aapcs.pdf
-
-Optimization guide for ARM9E (used in Nokia 770 Internet Tablet):
-http://infocenter.arm.com/help/topic/com.arm.doc.ddi0240b/DDI0240A.pdf
-Optimization guide for ARM11 (used in Nokia N800 Internet Tablet):
-http://infocenter.arm.com/help/topic/com.arm.doc.ddi0211j/DDI0211J_arm1136_r1p5_trm.pdf
-Optimization guide for Intel XScale (used in Sharp Zaurus PDA):
-http://download.intel.com/design/intelxscale/27347302.pdf
-Intel Wireless MMX2 Coprocessor: Programmers Reference Manual
-http://download.intel.com/design/intelxscale/31451001.pdf
-
-PowerPC-specific:
------------------
-PowerPC32/AltiVec PIM:
-www.freescale.com/files/32bit/doc/ref_manual/ALTIVECPEM.pdf
-
-PowerPC32/AltiVec PEM:
-www.freescale.com/files/32bit/doc/ref_manual/ALTIVECPIM.pdf
-
-CELL/SPU:
-http://www-01.ibm.com/chips/techlib/techlib.nsf/techdocs/30B3520C93F437AB87257060006FFE5E/$file/Language_Extensions_for_CBEA_2.4.pdf
-http://www-01.ibm.com/chips/techlib/techlib.nsf/techdocs/9F820A5FFA3ECE8C8725716A0062585F/$file/CBE_Handbook_v1.1_24APR2007_pub.pdf
-
-SPARC-specific:
----------------
-SPARC Joint Programming Specification (JPS1): Commonality
-http://www.fujitsu.com/downloads/PRMPWR/JPS1-R1.0.4-Common-pub.pdf
-
-UltraSPARC III Processor User's Manual (contains instruction timings)
-http://www.sun.com/processors/manuals/USIIIv2.pdf
-
-VIS Whitepaper (contains optimization guidelines)
-http://www.sun.com/processors/vis/download/vis/vis_whitepaper.pdf
-
-GCC asm links:
---------------
-official doc but quite ugly
-http://gcc.gnu.org/onlinedocs/gcc/Extended-Asm.html
-
-a bit old (note "+" is valid for input-output, even though the next disagrees)
-http://www.cs.virginia.edu/~clc5q/gcc-inline-asm.pdf

doc/snow.txt

@@ -1,630 +0,0 @@
-=============================================
-Snow Video Codec Specification Draft 20080110
-=============================================
-
-Introduction:
-=============
-This specification describes the Snow bitstream syntax and semantics as
-well as the formal Snow decoding process.
-
-The decoding process is described precisely and any compliant decoder
-MUST produce the exact same output for a spec-conformant Snow stream.
-For encoding, though, any process which generates a stream compliant to
-the syntactical and semantic requirements and which is decodable by
-the process described in this spec shall be considered a conformant
-Snow encoder.
-
-Definitions:
-============
-
-MUST    the specific part must be done to conform to this standard
-SHOULD  it is recommended to be done that way, but not strictly required
-
-ilog2(x) is the rounded down logarithm of x with basis 2
-ilog2(0) = 0
-
-Type definitions:
-=================
-
-b   1-bit range coded
-u   unsigned scalar value range coded
-s   signed scalar value range coded
-
-
-Bitstream syntax:
-=================
-
-frame:
-    header
-    prediction
-    residual
-
-header:
-    keyframe                            b   MID_STATE
-    if(keyframe || always_reset)
-        reset_contexts
-    if(keyframe){
-        version                         u   header_state
-        always_reset                    b   header_state
-        temporal_decomposition_type     u   header_state
-        temporal_decomposition_count    u   header_state
-        spatial_decomposition_count     u   header_state
-        colorspace_type                 u   header_state
-        chroma_h_shift                  u   header_state
-        chroma_v_shift                  u   header_state
-        spatial_scalability             b   header_state
-        max_ref_frames-1                u   header_state
-        qlogs
-    }
-    if(!keyframe){
-        update_mc                       b   header_state
-        if(update_mc){
-            for(plane=0; plane<2; plane++){
-                diag_mc                 b   header_state
-                htaps/2-1               u   header_state
-                for(i= p->htaps/2; i; i--)
-                    |hcoeff[i]|         u   header_state
-            }
-        }
-        update_qlogs                    b   header_state
-        if(update_qlogs){
-            spatial_decomposition_count u   header_state
-            qlogs
-        }
-    }
-
-    spatial_decomposition_type          s   header_state
-    qlog                                s   header_state
-    mv_scale                            s   header_state
-    qbias                               s   header_state
-    block_max_depth                     s   header_state
-
-qlogs:
-    for(plane=0; plane<2; plane++){
-        quant_table[plane][0][0]        s   header_state
-        for(level=0; level < spatial_decomposition_count; level++){
-            quant_table[plane][level][1]s   header_state
-            quant_table[plane][level][3]s   header_state
-        }
-    }
-
-reset_contexts
-    *_state[*]= MID_STATE
-
-prediction:
-    for(y=0; y<block_count_vertical; y++)
-        for(x=0; x<block_count_horizontal; x++)
-            block(0)
-
-block(level):
-    mvx_diff=mvy_diff=y_diff=cb_diff=cr_diff=0
-    if(keyframe){
-        intra=1
-    }else{
-        if(level!=max_block_depth){
-            s_context= 2*left->level + 2*top->level + topleft->level + topright->level
-            leaf                        b   block_state[4 + s_context]
-        }
-        if(level==max_block_depth || leaf){
-            intra                       b   block_state[1 + left->intra + top->intra]
-            if(intra){
-                y_diff                  s   block_state[32]
-                cb_diff                 s   block_state[64]
-                cr_diff                 s   block_state[96]
-            }else{
-                ref_context= ilog2(2*left->ref) + ilog2(2*top->ref)
-                if(ref_frames > 1)
-                    ref                 u   block_state[128 + 1024 + 32*ref_context]
-                mx_context= ilog2(2*abs(left->mx - top->mx))
-                my_context= ilog2(2*abs(left->my - top->my))
-                mvx_diff                s   block_state[128 + 32*(mx_context + 16*!!ref)]
-                mvy_diff                s   block_state[128 + 32*(my_context + 16*!!ref)]
-            }
-        }else{
-            block(level+1)
-            block(level+1)
-            block(level+1)
-            block(level+1)
-        }
-    }
-
-
-residual:
-    residual2(luma)
-    residual2(chroma_cr)
-    residual2(chroma_cb)
-
-residual2:
-    for(level=0; level<spatial_decomposition_count; level++){
-        if(level==0)
-            subband(LL, 0)
-        subband(HL, level)
-        subband(LH, level)
-        subband(HH, level)
-    }
-
-subband:
-    FIXME
-
-
-
-Tag description:
-----------------
-
-version
-    0
-    this MUST NOT change within a bitstream
-
-always_reset
-    if 1 then the range coder contexts will be reset after each frame
-
-temporal_decomposition_type
-    0
-
-temporal_decomposition_count
-    0
-
-spatial_decomposition_count
-    FIXME
-
-colorspace_type
-    0
-    this MUST NOT change within a bitstream
-
-chroma_h_shift
-    log2(luma.width / chroma.width)
-    this MUST NOT change within a bitstream
-
-chroma_v_shift
-    log2(luma.height / chroma.height)
-    this MUST NOT change within a bitstream
-
-spatial_scalability
-    0
-
-max_ref_frames
-    maximum number of reference frames
-    this MUST NOT change within a bitstream
-
-update_mc
-    indicates that motion compensation filter parameters are stored in the
-    header
-
-diag_mc
-    flag to enable faster diagonal interpolation
-    this SHOULD be 1 unless it turns out to be covered by a valid patent
-
-htaps
-    number of half pel interpolation filter taps, MUST be even, >0 and <10
-
-hcoeff
-    half pel interpolation filter coefficients, hcoeff[0] are the 2 middle
-    coefficients [1] are the next outer ones and so on, resulting in a filter
-    like: ...eff[2], hcoeff[1], hcoeff[0], hcoeff[0], hcoeff[1], hcoeff[2] ...
-    the sign of the coefficients is not explicitly stored but alternates
-    after each coeff and coeff[0] is positive, so ...,+,-,+,-,+,+,-,+,-,+,...
-    hcoeff[0] is not explicitly stored but found by subtracting the sum
-    of all stored coefficients with signs from 32
-    hcoeff[0]= 32 - hcoeff[1] - hcoeff[2] - ...
-    a good choice for hcoeff and htaps is
-    htaps= 6
-    hcoeff={40,-10,2}
-    an alternative which requires more computations at both encoder and
-    decoder side and may or may not be better is
-    htaps= 8
-    hcoeff={42,-14,6,-2}
-
-
-ref_frames
-    minimum of the number of available reference frames and max_ref_frames
-    for example the first frame after a key frame always has ref_frames=1
-
-spatial_decomposition_type
-    wavelet type
-    0 is a 9/7 symmetric compact integer wavelet
-    1 is a 5/3 symmetric compact integer wavelet
-    others are reserved
-    stored as delta from last, last is reset to 0 if always_reset || keyframe
-
-qlog
-    quality (logarthmic quantizer scale)
-    stored as delta from last, last is reset to 0 if always_reset || keyframe
-
-mv_scale
-    stored as delta from last, last is reset to 0 if always_reset || keyframe
-    FIXME check that everything works fine if this changes between frames
-
-qbias
-    dequantization bias
-    stored as delta from last, last is reset to 0 if always_reset || keyframe
-
-block_max_depth
-    maximum depth of the block tree
-    stored as delta from last, last is reset to 0 if always_reset || keyframe
-
-quant_table
-    quantiztation table
-
-
-Highlevel bitstream structure:
-=============================
- --------------------------------------------
-|                   Header                   |
- --------------------------------------------
-|    ------------------------------------    |
-|   |               Block0               |   |
-|   |             split?                 |   |
-|   |     yes              no            |   |
-|   |  .........         intra?          |   |
-|   | : Block01 :    yes         no      |   |
-|   | : Block02 :  .......   ..........  |   |
-|   | : Block03 : :  y DC : : ref index: |   |
-|   | : Block04 : : cb DC : : motion x : |   |
-|   |  .........  : cr DC : : motion y : |   |
-|   |              .......   ..........  |   |
-|    ------------------------------------    |
-|    ------------------------------------    |
-|   |               Block1               |   |
-|                    ...                     |
- --------------------------------------------
-| ------------   ------------   ------------ |
-|| Y subbands | | Cb subbands| | Cr subbands||
-||  ---  ---  | |  ---  ---  | |  ---  ---  ||
-|| |LL0||HL0| | | |LL0||HL0| | | |LL0||HL0| ||
-||  ---  ---  | |  ---  ---  | |  ---  ---  ||
-||  ---  ---  | |  ---  ---  | |  ---  ---  ||
-|| |LH0||HH0| | | |LH0||HH0| | | |LH0||HH0| ||
-||  ---  ---  | |  ---  ---  | |  ---  ---  ||
-||  ---  ---  | |  ---  ---  | |  ---  ---  ||
-|| |HL1||LH1| | | |HL1||LH1| | | |HL1||LH1| ||
-||  ---  ---  | |  ---  ---  | |  ---  ---  ||
-||  ---  ---  | |  ---  ---  | |  ---  ---  ||
-|| |HH1||HL2| | | |HH1||HL2| | | |HH1||HL2| ||
-||    ...     | |    ...     | |    ...     ||
-| ------------   ------------   ------------ |
- --------------------------------------------
-
-Decoding process:
-=================
-
-                                         ------------
-                                        |            |
-                                        |  Subbands  |
-                   ------------         |            |
-                  |            |         ------------
-                  |  Intra DC  |               |
-                  |            |    LL0 subband prediction
-                   ------------                |
-                                \        Dequantizaton
- -------------------             \             |
-|  Reference frames |             \           IDWT
-| -------   ------- |    Motion    \           |
-||Frame 0| |Frame 1|| Compensation  .   OBMC   v      -------
-| -------   ------- | --------------. \------> + --->|Frame n|-->output
-| -------   ------- |                                 -------
-||Frame 2| |Frame 3||<----------------------------------/
-|        ...        |
- -------------------
-
-
-Range Coder:
-============
-
-Binary Range Coder:
--------------------
-The implemented range coder is an adapted version based upon "Range encoding:
-an algorithm for removing redundancy from a digitised message." by G. N. N.
-Martin.
-The symbols encoded by the Snow range coder are bits (0|1). The
-associated probabilities are not fix but change depending on the symbol mix
-seen so far.
-
-
-bit seen | new state
----------+-----------------------------------------------
-    0    | 256 - state_transition_table[256 - old_state];
-    1    |       state_transition_table[      old_state];
-
-state_transition_table = {
-  0,   0,   0,   0,   0,   0,   0,   0,  20,  21,  22,  23,  24,  25,  26,  27,
- 28,  29,  30,  31,  32,  33,  34,  35,  36,  37,  37,  38,  39,  40,  41,  42,
- 43,  44,  45,  46,  47,  48,  49,  50,  51,  52,  53,  54,  55,  56,  56,  57,
- 58,  59,  60,  61,  62,  63,  64,  65,  66,  67,  68,  69,  70,  71,  72,  73,
- 74,  75,  75,  76,  77,  78,  79,  80,  81,  82,  83,  84,  85,  86,  87,  88,
- 89,  90,  91,  92,  93,  94,  94,  95,  96,  97,  98,  99, 100, 101, 102, 103,
-104, 105, 106, 107, 108, 109, 110, 111, 112, 113, 114, 114, 115, 116, 117, 118,
-119, 120, 121, 122, 123, 124, 125, 126, 127, 128, 129, 130, 131, 132, 133, 133,
-134, 135, 136, 137, 138, 139, 140, 141, 142, 143, 144, 145, 146, 147, 148, 149,
-150, 151, 152, 152, 153, 154, 155, 156, 157, 158, 159, 160, 161, 162, 163, 164,
-165, 166, 167, 168, 169, 170, 171, 171, 172, 173, 174, 175, 176, 177, 178, 179,
-180, 181, 182, 183, 184, 185, 186, 187, 188, 189, 190, 190, 191, 192, 194, 194,
-195, 196, 197, 198, 199, 200, 201, 202, 202, 204, 205, 206, 207, 208, 209, 209,
-210, 211, 212, 213, 215, 215, 216, 217, 218, 219, 220, 220, 222, 223, 224, 225,
-226, 227, 227, 229, 229, 230, 231, 232, 234, 234, 235, 236, 237, 238, 239, 240,
-241, 242, 243, 244, 245, 246, 247, 248, 248,   0,   0,   0,   0,   0,   0,   0};
-
-FIXME
-
-
-Range Coding of integers:
--------------------------
-FIXME
-
-
-Neighboring Blocks:
-===================
-left and top are set to the respective blocks unless they are outside of
-the image in which case they are set to the Null block
-
-top-left is set to the top left block unless it is outside of the image in
-which case it is set to the left block
-
-if this block has no larger parent block or it is at the left side of its
-parent block and the top right block is not outside of the image then the
-top right block is used for top-right else the top-left block is used
-
-Null block
-y,cb,cr are 128
-level, ref, mx and my are 0
-
-
-Motion Vector Prediction:
-=========================
-1. the motion vectors of all the neighboring blocks are scaled to
-compensate for the difference of reference frames
-
-scaled_mv= (mv * (256 * (current_reference+1) / (mv.reference+1)) + 128)>>8
-
-2. the median of the scaled left, top and top-right vectors is used as
-motion vector prediction
-
-3. the used motion vector is the sum of the predictor and
-   (mvx_diff, mvy_diff)*mv_scale
-
-
-Intra DC Predicton:
-======================
-the luma and chroma values of the left block are used as predictors
-
-the used luma and chroma is the sum of the predictor and y_diff, cb_diff, cr_diff
-to reverse this in the decoder apply the following:
-block[y][x].dc[0] = block[y][x-1].dc[0] +  y_diff;
-block[y][x].dc[1] = block[y][x-1].dc[1] + cb_diff;
-block[y][x].dc[2] = block[y][x-1].dc[2] + cr_diff;
-block[*][-1].dc[*]= 128;
-
-
-Motion Compensation:
-====================
-
-Halfpel interpolation:
-----------------------
-halfpel interpolation is done by convolution with the halfpel filter stored
-in the header:
-
-horizontal halfpel samples are found by
-H1[y][x] =    hcoeff[0]*(F[y][x  ] + F[y][x+1])
-            + hcoeff[1]*(F[y][x-1] + F[y][x+2])
-            + hcoeff[2]*(F[y][x-2] + F[y][x+3])
-            + ...
-h1[y][x] = (H1[y][x] + 32)>>6;
-
-vertical halfpel samples are found by
-H2[y][x] =    hcoeff[0]*(F[y  ][x] + F[y+1][x])
-            + hcoeff[1]*(F[y-1][x] + F[y+2][x])
-            + ...
-h2[y][x] = (H2[y][x] + 32)>>6;
-
-vertical+horizontal halfpel samples are found by
-H3[y][x] =    hcoeff[0]*(H2[y][x  ] + H2[y][x+1])
-            + hcoeff[1]*(H2[y][x-1] + H2[y][x+2])
-            + ...
-H3[y][x] =    hcoeff[0]*(H1[y  ][x] + H1[y+1][x])
-            + hcoeff[1]*(H1[y+1][x] + H1[y+2][x])
-            + ...
-h3[y][x] = (H3[y][x] + 2048)>>12;
-
-
-                   F   H1  F
-                   |   |   |
-                   |   |   |
-                   |   |   |
-                   F   H1  F
-                   |   |   |
-                   |   |   |
-                   |   |   |
-   F-------F-------F-> H1<-F-------F-------F
-                   v   v   v
-                  H2   H3  H2
-                   ^   ^   ^
-   F-------F-------F-> H1<-F-------F-------F
-                   |   |   |
-                   |   |   |
-                   |   |   |
-                   F   H1  F
-                   |   |   |
-                   |   |   |
-                   |   |   |
-                   F   H1  F
-
-
-unavailable fullpel samples (outside the picture for example) shall be equal
-to the closest available fullpel sample
-
-
-Smaller pel interpolation:
---------------------------
-if diag_mc is set then points which lie on a line between 2 vertically,
-horiziontally or diagonally adjacent halfpel points shall be interpolated
-linearls with rounding to nearest and halfway values rounded up.
-points which lie on 2 diagonals at the same time should only use the one
-diagonal not containing the fullpel point
-
-
-
-           F-->O---q---O<--h1->O---q---O<--F
-           v \           / v \           / v
-           O   O       O   O   O       O   O
-           |         /     |     \         |
-           q       q       q       q       q
-           |     /         |         \     |
-           O   O       O   O   O       O   O
-           ^ /           \ ^ /           \ ^
-          h2-->O---q---O<--h3->O---q---O<--h2
-           v \           / v \           / v
-           O   O       O   O   O       O   O
-           |     \         |         /     |
-           q       q       q       q       q
-           |         \     |     /         |
-           O   O       O   O   O       O   O
-           ^ /           \ ^ /           \ ^
-           F-->O---q---O<--h1->O---q---O<--F
-
-
-
-the remaining points shall be bilinearly interpolated from the
-up to 4 surrounding halfpel and fullpel points, again rounding should be to
-nearest and halfway values rounded up
-
-compliant Snow decoders MUST support 1-1/8 pel luma and 1/2-1/16 pel chroma
-interpolation at least
-
-
-Overlapped block motion compensation:
--------------------------------------
-FIXME
-
-LL band prediction:
-===================
-Each sample in the LL0 subband is predicted by the median of the left, top and
-left+top-topleft samples, samples outside the subband shall be considered to
-be 0. To reverse this prediction in the decoder apply the following.
-for(y=0; y<height; y++){
-    for(x=0; x<width; x++){
-        sample[y][x] += median(sample[y-1][x],
-                               sample[y][x-1],
-                               sample[y-1][x]+sample[y][x-1]-sample[y-1][x-1]);
-    }
-}
-sample[-1][*]=sample[*][-1]= 0;
-width,height here are the width and height of the LL0 subband not of the final
-video
-
-
-Dequantizaton:
-==============
-FIXME
-
-Wavelet Transform:
-==================
-
-Snow supports 2 wavelet transforms, the symmetric biorthogonal 5/3 integer
-transform and a integer approximation of the symmetric biorthogonal 9/7
-daubechies wavelet.
-
-2D IDWT (inverse discrete wavelet transform)
---------------------------------------------
-The 2D IDWT applies a 2D filter recursively, each time combining the
-4 lowest frequency subbands into a single subband until only 1 subband
-remains.
-The 2D filter is done by first applying a 1D filter in the vertical direction
-and then applying it in the horizontal one.
- ---------------    ---------------    ---------------    ---------------
-|LL0|HL0|       |  |   |   |       |  |       |       |  |       |       |
-|---+---|  HL1  |  | L0|H0 |  HL1  |  |  LL1  |  HL1  |  |       |       |
-|LH0|HH0|       |  |   |   |       |  |       |       |  |       |       |
-|-------+-------|->|-------+-------|->|-------+-------|->|   L1  |  H1   |->...
-|       |       |  |       |       |  |       |       |  |       |       |
-|  LH1  |  HH1  |  |  LH1  |  HH1  |  |  LH1  |  HH1  |  |       |       |
-|       |       |  |       |       |  |       |       |  |       |       |
- ---------------    ---------------    ---------------    ---------------
-
-
-1D Filter:
-----------
-1. interleave the samples of the low and high frequency subbands like
-s={L0, H0, L1, H1, L2, H2, L3, H3, ... }
-note, this can end with a L or a H, the number of elements shall be w
-s[-1] shall be considered equivalent to s[1  ]
-s[w ] shall be considered equivalent to s[w-2]
-
-2. perform the lifting steps in order as described below
-
-5/3 Integer filter:
-1. s[i] -= (s[i-1] + s[i+1] + 2)>>2; for all even i < w
-2. s[i] += (s[i-1] + s[i+1]    )>>1; for all odd  i < w
-
-\ | /|\ | /|\ | /|\ | /|\
- \|/ | \|/ | \|/ | \|/ |
-  +  |  +  |  +  |  +  |   -1/4
- /|\ | /|\ | /|\ | /|\ |
-/ | \|/ | \|/ | \|/ | \|/
-  |  +  |  +  |  +  |  +   +1/2
-
-
-Snow's 9/7 Integer filter:
-1. s[i] -= (3*(s[i-1] + s[i+1])         + 4)>>3; for all even i < w
-2. s[i] -=     s[i-1] + s[i+1]                 ; for all odd  i < w
-3. s[i] += (   s[i-1] + s[i+1] + 4*s[i] + 8)>>4; for all even i < w
-4. s[i] += (3*(s[i-1] + s[i+1])            )>>1; for all odd  i < w
-
-\ | /|\ | /|\ | /|\ | /|\
- \|/ | \|/ | \|/ | \|/ |
-  +  |  +  |  +  |  +  |   -3/8
- /|\ | /|\ | /|\ | /|\ |
-/ | \|/ | \|/ | \|/ | \|/
- (|  + (|  + (|  + (|  +   -1
-\ + /|\ + /|\ + /|\ + /|\  +1/4
- \|/ | \|/ | \|/ | \|/ |
-  +  |  +  |  +  |  +  |   +1/16
- /|\ | /|\ | /|\ | /|\ |
-/ | \|/ | \|/ | \|/ | \|/
-  |  +  |  +  |  +  |  +   +3/2
-
-optimization tips:
-following are exactly identical
-(3a)>>1 == a + (a>>1)
-(a + 4b + 8)>>4 == ((a>>2) + b + 2)>>2
-
-16bit implementation note:
-The IDWT can be implemented with 16bits, but this requires some care to
-prevent overflows, the following list, lists the minimum number of bits needed
-for some terms
-1. lifting step
-A= s[i-1] + s[i+1]                              16bit
-3*A + 4                                         18bit
-A + (A>>1) + 2                                  17bit
-
-3. lifting step
-s[i-1] + s[i+1]                                 17bit
-
-4. lifiting step
-3*(s[i-1] + s[i+1])                             17bit
-
-
-TODO:
-=====
-Important:
-finetune initial contexts
-flip wavelet?
-try to use the wavelet transformed predicted image (motion compensated image) as context for coding the residual coefficients
-try the MV length as context for coding the residual coefficients
-use extradata for stuff which is in the keyframes now?
-the MV median predictor is patented IIRC
-implement per picture halfpel interpolation
-try different range coder state transition tables for different contexts
-
-Not Important:
-compare the 6 tap and 8 tap hpel filters (psnr/bitrate and subjective quality)
-spatial_scalability b vs u (!= 0 breaks syntax anyway so we can add a u later)
-
-
-Credits:
-========
-Michael Niedermayer
-Loren Merritt
-
-
-Copyright:
-==========
-GPL + GFDL + whatever is needed to make this a RFC

doc/soc.txt

@@ -1,24 +0,0 @@
-Google Summer of Code and similar project guidelines
-
-Summer of Code is a project by Google in which students are paid to implement
-some nice new features for various participating open source projects ...
-
-This text is a collection of things to take care of for the next soc as
-it's a little late for this year's soc (2006).
-
-The Goal:
-Our goal in respect to soc is and must be of course exactly one thing and
-that is to improve FFmpeg, to reach this goal, code must
-* conform to the svn policy and patch submission guidelines
-* must improve FFmpeg somehow (faster, smaller, "better",
-  more codecs supported, fewer bugs, cleaner, ...)
-
-for mentors and other developers to help students to reach that goal it is
-essential that changes to their codebase are publicly visible, clean and
-easy reviewable that again leads us to:
-* use of a revision control system like svn
-* separation of cosmetic from non-cosmetic changes (this is almost entirely
-  ignored by mentors and students in soc 2006 which might lead to a suprise
-  when the code will be reviewed at the end before a possible inclusion in
-  FFmpeg, individual changes were generally not reviewable due to cosmetics).
-* frequent commits, so that comments can be provided early

doc/swscale.txt

@@ -1,99 +0,0 @@
-    The official guide to swscale for confused developers.
-   ========================================================
-
-Current (simplified) Architecture:
----------------------------------
-                        Input
-                          v
-                   _______OR_________
-                 /                   \
-               /                       \
-       special converter     [Input to YUV converter]
-              |                         |
-              |          (8bit YUV 4:4:4 / 4:2:2 / 4:2:0 / 4:0:0 )
-              |                         |
-              |                         v
-              |                  Horizontal scaler
-              |                         |
-              |      (15bit YUV 4:4:4 / 4:2:2 / 4:2:0 / 4:1:1 / 4:0:0 )
-              |                         |
-              |                         v
-              |          Vertical scaler and output converter
-              |                         |
-              v                         v
-                         output
-
-
-Swscale has 2 scaler paths. Each side must be capable of handling
-slices, that is, consecutive non-overlapping rectangles of dimension
-(0,slice_top) - (picture_width, slice_bottom).
-
-special converter
-    These generally are unscaled converters of common
-    formats, like YUV 4:2:0/4:2:2 -> RGB15/16/24/32. Though it could also
-    in principle contain scalers optimized for specific common cases.
-
-Main path
-    The main path is used when no special converter can be used. The code
-    is designed as a destination line pull architecture. That is, for each
-    output line the vertical scaler pulls lines from a ring buffer. When
-    the ring buffer does not contain the wanted line, then it is pulled from
-    the input slice through the input converter and horizontal scaler.
-    The result is also stored in the ring buffer to serve future vertical
-    scaler requests.
-    When no more output can be generated because lines from a future slice
-    would be needed, then all remaining lines in the current slice are
-    converted, horizontally scaled and put in the ring buffer.
-    [This is done for luma and chroma, each with possibly different numbers
-     of lines per picture.]
-
-Input to YUV Converter
-    When the input to the main path is not planar 8 bits per component YUV or
-    8-bit gray, it is converted to planar 8-bit YUV. Two sets of converters
-    exist for this currently: One performs horizontal downscaling by 2
-    before the conversion, the other leaves the full chroma resolution,
-    but is slightly slower. The scaler will try to preserve full chroma
-    when the output uses it. It is possible to force full chroma with
-    SWS_FULL_CHR_H_INP even for cases where the scaler thinks it is useless.
-
-Horizontal scaler
-    There are several horizontal scalers. A special case worth mentioning is
-    the fast bilinear scaler that is made of runtime-generated MMX2 code
-    using specially tuned pshufw instructions.
-    The remaining scalers are specially-tuned for various filter lengths.
-    They scale 8-bit unsigned planar data to 16-bit signed planar data.
-    Future >8 bits per component inputs will need to add a new horizontal
-    scaler that preserves the input precision.
-
-Vertical scaler and output converter
-    There is a large number of combined vertical scalers + output converters.
-    Some are:
-    * unscaled output converters
-    * unscaled output converters that average 2 chroma lines
-    * bilinear converters                (C, MMX and accurate MMX)
-    * arbitrary filter length converters (C, MMX and accurate MMX)
-    And
-    * Plain C  8-bit 4:2:2 YUV -> RGB converters using LUTs
-    * Plain C 17-bit 4:4:4 YUV -> RGB converters using multiplies
-    * MMX     11-bit 4:2:2 YUV -> RGB converters
-    * Plain C 16-bit Y -> 16-bit gray
-      ...
-
-    RGB with less than 8 bits per component uses dither to improve the
-    subjective quality and low-frequency accuracy.
-
-
-Filter coefficients:
---------------------
-There are several different scalers (bilinear, bicubic, lanczos, area,
-sinc, ...). Their coefficients are calculated in initFilter().
-Horizontal filter coefficients have a 1.0 point at 1 << 14, vertical ones at
-1 << 12. The 1.0 points have been chosen to maximize precision while leaving
-a little headroom for convolutional filters like sharpening filters and
-minimizing SIMD instructions needed to apply them.
-It would be trivial to use a different 1.0 point if some specific scaler
-would benefit from it.
-Also, as already hinted at, initFilter() accepts an optional convolutional
-filter as input that can be used for contrast, saturation, blur, sharpening
-shift, chroma vs. luma shift, ...
-

doc/texi2pod.pl

@@ -1,427 +0,0 @@
-#! /usr/bin/perl -w
-
-#   Copyright (C) 1999, 2000, 2001 Free Software Foundation, Inc.
-
-# This file is part of GNU CC.
-
-# GNU CC is free software; you can redistribute it and/or modify
-# it under the terms of the GNU General Public License as published by
-# the Free Software Foundation; either version 2, or (at your option)
-# any later version.
-
-# GNU CC is distributed in the hope that it will be useful,
-# but WITHOUT ANY WARRANTY; without even the implied warranty of
-# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
-# GNU General Public License for more details.
-
-# You should have received a copy of the GNU General Public License
-# along with GNU CC; see the file COPYING.  If not, write to
-# the Free Software Foundation, 51 Franklin Street, Fifth Floor,
-# Boston, MA 02110-1301 USA
-
-# This does trivial (and I mean _trivial_) conversion of Texinfo
-# markup to Perl POD format.  It's intended to be used to extract
-# something suitable for a manpage from a Texinfo document.
-
-$output = 0;
-$skipping = 0;
-%sects = ();
-$section = "";
-@icstack = ();
-@endwstack = ();
-@skstack = ();
-@instack = ();
-$shift = "";
-%defs = ();
-$fnno = 1;
-$inf = "";
-$ibase = "";
-
-while ($_ = shift) {
-    if (/^-D(.*)$/) {
-        if ($1 ne "") {
-            $flag = $1;
-        } else {
-            $flag = shift;
-        }
-        $value = "";
-        ($flag, $value) = ($flag =~ /^([^=]+)(?:=(.+))?/);
-        die "no flag specified for -D\n"
-            unless $flag ne "";
-        die "flags may only contain letters, digits, hyphens, dashes and underscores\n"
-            unless $flag =~ /^[a-zA-Z0-9_-]+$/;
-        $defs{$flag} = $value;
-    } elsif (/^-/) {
-        usage();
-    } else {
-        $in = $_, next unless defined $in;
-        $out = $_, next unless defined $out;
-        usage();
-    }
-}
-
-if (defined $in) {
-    $inf = gensym();
-    open($inf, "<$in") or die "opening \"$in\": $!\n";
-    $ibase = $1 if $in =~ m|^(.+)/[^/]+$|;
-} else {
-    $inf = \*STDIN;
-}
-
-if (defined $out) {
-    open(STDOUT, ">$out") or die "opening \"$out\": $!\n";
-}
-
-while(defined $inf) {
-while(<$inf>) {
-    # Certain commands are discarded without further processing.
-    /^\@(?:
-         [a-z]+index            # @*index: useful only in complete manual
-         |need                  # @need: useful only in printed manual
-         |(?:end\s+)?group      # @group .. @end group: ditto
-         |page                  # @page: ditto
-         |node                  # @node: useful only in .info file
-         |(?:end\s+)?ifnottex   # @ifnottex .. @end ifnottex: use contents
-        )\b/x and next;
-
-    chomp;
-
-    # Look for filename and title markers.
-    /^\@setfilename\s+([^.]+)/ and $fn = $1, next;
-    /^\@settitle\s+([^.]+)/ and $tl = postprocess($1), next;
-
-    # Identify a man title but keep only the one we are interested in.
-    /^\@c\s+man\s+title\s+([A-Za-z0-9-]+)\s+(.+)/ and do {
-        if (exists $defs{$1}) {
-            $fn = $1;
-            $tl = postprocess($2);
-        }
-        next;
-    };
-
-    # Look for blocks surrounded by @c man begin SECTION ... @c man end.
-    # This really oughta be @ifman ... @end ifman and the like, but such
-    # would require rev'ing all other Texinfo translators.
-    /^\@c\s+man\s+begin\s+([A-Z]+)\s+([A-Za-z0-9-]+)/ and do {
-        $output = 1 if exists $defs{$2};
-        $sect = $1;
-        next;
-    };
-    /^\@c\s+man\s+begin\s+([A-Z]+)/ and $sect = $1, $output = 1, next;
-    /^\@c\s+man\s+end/ and do {
-        $sects{$sect} = "" unless exists $sects{$sect};
-        $sects{$sect} .= postprocess($section);
-        $section = "";
-        $output = 0;
-        next;
-    };
-
-    # handle variables
-    /^\@set\s+([a-zA-Z0-9_-]+)\s*(.*)$/ and do {
-        $defs{$1} = $2;
-        next;
-    };
-    /^\@clear\s+([a-zA-Z0-9_-]+)/ and do {
-        delete $defs{$1};
-        next;
-    };
-
-    next unless $output;
-
-    # Discard comments.  (Can't do it above, because then we'd never see
-    # @c man lines.)
-    /^\@c\b/ and next;
-
-    # End-block handler goes up here because it needs to operate even
-    # if we are skipping.
-    /^\@end\s+([a-z]+)/ and do {
-        # Ignore @end foo, where foo is not an operation which may
-        # cause us to skip, if we are presently skipping.
-        my $ended = $1;
-        next if $skipping && $ended !~ /^(?:ifset|ifclear|ignore|menu|iftex)$/;
-
-        die "\@end $ended without \@$ended at line $.\n" unless defined $endw;
-        die "\@$endw ended by \@end $ended at line $.\n" unless $ended eq $endw;
-
-        $endw = pop @endwstack;
-
-        if ($ended =~ /^(?:ifset|ifclear|ignore|menu|iftex)$/) {
-            $skipping = pop @skstack;
-            next;
-        } elsif ($ended =~ /^(?:example|smallexample|display)$/) {
-            $shift = "";
-            $_ = "";        # need a paragraph break
-        } elsif ($ended =~ /^(?:itemize|enumerate|[fv]?table)$/) {
-            $_ = "\n=back\n";
-            $ic = pop @icstack;
-        } else {
-            die "unknown command \@end $ended at line $.\n";
-        }
-    };
-
-    # We must handle commands which can cause skipping even while we
-    # are skipping, otherwise we will not process nested conditionals
-    # correctly.
-    /^\@ifset\s+([a-zA-Z0-9_-]+)/ and do {
-        push @endwstack, $endw;
-        push @skstack, $skipping;
-        $endw = "ifset";
-        $skipping = 1 unless exists $defs{$1};
-        next;
-    };
-
-    /^\@ifclear\s+([a-zA-Z0-9_-]+)/ and do {
-        push @endwstack, $endw;
-        push @skstack, $skipping;
-        $endw = "ifclear";
-        $skipping = 1 if exists $defs{$1};
-        next;
-    };
-
-    /^\@(ignore|menu|iftex)\b/ and do {
-        push @endwstack, $endw;
-        push @skstack, $skipping;
-        $endw = $1;
-        $skipping = 1;
-        next;
-    };
-
-    next if $skipping;
-
-    # Character entities.  First the ones that can be replaced by raw text
-    # or discarded outright:
-    s/\@copyright\{\}/(c)/g;
-    s/\@dots\{\}/.../g;
-    s/\@enddots\{\}/..../g;
-    s/\@([.!? ])/$1/g;
-    s/\@[:-]//g;
-    s/\@bullet(?:\{\})?/*/g;
-    s/\@TeX\{\}/TeX/g;
-    s/\@pounds\{\}/\#/g;
-    s/\@minus(?:\{\})?/-/g;
-    s/\\,/,/g;
-
-    # Now the ones that have to be replaced by special escapes
-    # (which will be turned back into text by unmunge())
-    s/&/&amp;/g;
-    s/\@\{/&lbrace;/g;
-    s/\@\}/&rbrace;/g;
-    s/\@\@/&at;/g;
-
-    # Inside a verbatim block, handle @var specially.
-    if ($shift ne "") {
-        s/\@var\{([^\}]*)\}/<$1>/g;
-    }
-
-    # POD doesn't interpret E<> inside a verbatim block.
-    if ($shift eq "") {
-        s/</&lt;/g;
-        s/>/&gt;/g;
-    } else {
-        s/</&LT;/g;
-        s/>/&GT;/g;
-    }
-
-    # Single line command handlers.
-
-    /^\@include\s+(.+)$/ and do {
-        push @instack, $inf;
-        $inf = gensym();
-
-        # Try cwd and $ibase.
-        open($inf, "<" . $1)
-            or open($inf, "<" . $ibase . "/" . $1)
-                or die "cannot open $1 or $ibase/$1: $!\n";
-        next;
-    };
-
-    /^\@(?:section|unnumbered|unnumberedsec|center)\s+(.+)$/
-        and $_ = "\n=head2 $1\n";
-    /^\@subsection\s+(.+)$/
-        and $_ = "\n=head3 $1\n";
-
-    # Block command handlers:
-    /^\@itemize\s+(\@[a-z]+|\*|-)/ and do {
-        push @endwstack, $endw;
-        push @icstack, $ic;
-        $ic = $1;
-        $_ = "\n=over 4\n";
-        $endw = "itemize";
-    };
-
-    /^\@enumerate(?:\s+([a-zA-Z0-9]+))?/ and do {
-        push @endwstack, $endw;
-        push @icstack, $ic;
-        if (defined $1) {
-            $ic = $1 . ".";
-        } else {
-            $ic = "1.";
-        }
-        $_ = "\n=over 4\n";
-        $endw = "enumerate";
-    };
-
-    /^\@([fv]?table)\s+(\@[a-z]+)/ and do {
-        push @endwstack, $endw;
-        push @icstack, $ic;
-        $endw = $1;
-        $ic = $2;
-        $ic =~ s/\@(?:samp|strong|key|gcctabopt|option|env)/B/;
-        $ic =~ s/\@(?:code|kbd)/C/;
-        $ic =~ s/\@(?:dfn|var|emph|cite|i)/I/;
-        $ic =~ s/\@(?:file)/F/;
-        $_ = "\n=over 4\n";
-    };
-
-    /^\@((?:small)?example|display)/ and do {
-        push @endwstack, $endw;
-        $endw = $1;
-        $shift = "\t";
-        $_ = "";        # need a paragraph break
-    };
-
-    /^\@itemx?\s*(.+)?$/ and do {
-        if (defined $1) {
-            # Entity escapes prevent munging by the <> processing below.
-            $_ = "\n=item $ic\&LT;$1\&GT;\n";
-        } else {
-            $_ = "\n=item $ic\n";
-            $ic =~ y/A-Ya-y/B-Zb-z/;
-            $ic =~ s/(\d+)/$1 + 1/eg;
-        }
-    };
-
-    $section .= $shift.$_."\n";
-}
-# End of current file.
-close($inf);
-$inf = pop @instack;
-}
-
-die "No filename or title\n" unless defined $fn && defined $tl;
-
-$sects{NAME} = "$fn \- $tl\n";
-$sects{FOOTNOTES} .= "=back\n" if exists $sects{FOOTNOTES};
-
-for $sect (qw(NAME SYNOPSIS DESCRIPTION OPTIONS EXAMPLES ENVIRONMENT FILES
-              BUGS NOTES FOOTNOTES SEEALSO AUTHOR COPYRIGHT)) {
-    if(exists $sects{$sect}) {
-        $head = $sect;
-        $head =~ s/SEEALSO/SEE ALSO/;
-        print "=head1 $head\n\n";
-        print scalar unmunge ($sects{$sect});
-        print "\n";
-    }
-}
-
-sub usage
-{
-    die "usage: $0 [-D toggle...] [infile [outfile]]\n";
-}
-
-sub postprocess
-{
-    local $_ = $_[0];
-
-    # @value{foo} is replaced by whatever 'foo' is defined as.
-    while (m/(\@value\{([a-zA-Z0-9_-]+)\})/g) {
-        if (! exists $defs{$2}) {
-            print STDERR "Option $2 not defined\n";
-            s/\Q$1\E//;
-        } else {
-            $value = $defs{$2};
-            s/\Q$1\E/$value/;
-        }
-    }
-
-    # Formatting commands.
-    # Temporary escape for @r.
-    s/\@r\{([^\}]*)\}/R<$1>/g;
-    s/\@(?:dfn|var|emph|cite|i)\{([^\}]*)\}/I<$1>/g;
-    s/\@(?:code|kbd)\{([^\}]*)\}/C<$1>/g;
-    s/\@(?:gccoptlist|samp|strong|key|option|env|command|b)\{([^\}]*)\}/B<$1>/g;
-    s/\@sc\{([^\}]*)\}/\U$1/g;
-    s/\@file\{([^\}]*)\}/F<$1>/g;
-    s/\@w\{([^\}]*)\}/S<$1>/g;
-    s/\@(?:dmn|math)\{([^\}]*)\}/$1/g;
-
-    # Cross references are thrown away, as are @noindent and @refill.
-    # (@noindent is impossible in .pod, and @refill is unnecessary.)
-    # @* is also impossible in .pod; we discard it and any newline that
-    # follows it.  Similarly, our macro @gol must be discarded.
-
-    s/\(?\@xref\{(?:[^\}]*)\}(?:[^.<]|(?:<[^<>]*>))*\.\)?//g;
-    s/\s+\(\@pxref\{(?:[^\}]*)\}\)//g;
-    s/;\s+\@pxref\{(?:[^\}]*)\}//g;
-    s/\@noindent\s*//g;
-    s/\@refill//g;
-    s/\@gol//g;
-    s/\@\*\s*\n?//g;
-
-    # @uref can take one, two, or three arguments, with different
-    # semantics each time.  @url and @email are just like @uref with
-    # one argument, for our purposes.
-    s/\@(?:uref|url|email)\{([^\},]*)\}/&lt;B<$1>&gt;/g;
-    s/\@uref\{([^\},]*),([^\},]*)\}/$2 (C<$1>)/g;
-    s/\@uref\{([^\},]*),([^\},]*),([^\},]*)\}/$3/g;
-
-    # Turn B<blah I<blah> blah> into B<blah> I<blah> B<blah> to
-    # match Texinfo semantics of @emph inside @samp.  Also handle @r
-    # inside bold.
-    s/&LT;/</g;
-    s/&GT;/>/g;
-    1 while s/B<((?:[^<>]|I<[^<>]*>)*)R<([^>]*)>/B<$1>${2}B</g;
-    1 while (s/B<([^<>]*)I<([^>]+)>/B<$1>I<$2>B</g);
-    1 while (s/I<([^<>]*)B<([^>]+)>/I<$1>B<$2>I</g);
-    s/[BI]<>//g;
-    s/([BI])<(\s+)([^>]+)>/$2$1<$3>/g;
-    s/([BI])<([^>]+?)(\s+)>/$1<$2>$3/g;
-
-    # Extract footnotes.  This has to be done after all other
-    # processing because otherwise the regexp will choke on formatting
-    # inside @footnote.
-    while (/\@footnote/g) {
-        s/\@footnote\{([^\}]+)\}/[$fnno]/;
-        add_footnote($1, $fnno);
-        $fnno++;
-    }
-
-    return $_;
-}
-
-sub unmunge
-{
-    # Replace escaped symbols with their equivalents.
-    local $_ = $_[0];
-
-    s/&lt;/E<lt>/g;
-    s/&gt;/E<gt>/g;
-    s/&lbrace;/\{/g;
-    s/&rbrace;/\}/g;
-    s/&at;/\@/g;
-    s/&amp;/&/g;
-    return $_;
-}
-
-sub add_footnote
-{
-    unless (exists $sects{FOOTNOTES}) {
-        $sects{FOOTNOTES} = "\n=over 4\n\n";
-    }
-
-    $sects{FOOTNOTES} .= "=item $fnno.\n\n"; $fnno++;
-    $sects{FOOTNOTES} .= $_[0];
-    $sects{FOOTNOTES} .= "\n\n";
-}
-
-# stolen from Symbol.pm
-{
-    my $genseq = 0;
-    sub gensym
-    {
-        my $name = "GEN" . $genseq++;
-        my $ref = \*{$name};
-        delete $::{$name};
-        return $ref;
-    }
-}

ffpresets/libx264-baseline.ffpreset

@@ -1,4 +0,0 @@
-coder=0
-bf=0
-flags2=-wpred-dct8x8+mbtree
-wpredp=0

ffpresets/libx264-default.ffpreset

@@ -1,22 +0,0 @@
-coder=1
-flags=+loop
-cmp=+chroma
-partitions=+parti8x8+parti4x4+partp8x8+partb8x8
-me_method=hex
-subq=7
-me_range=16
-g=250
-keyint_min=25
-sc_threshold=40
-i_qfactor=0.71
-b_strategy=1
-qcomp=0.6
-qmin=10
-qmax=51
-qdiff=4
-bf=3
-refs=3
-directpred=1
-trellis=1
-flags2=+mixed_refs+wpred+dct8x8+fastpskip+mbtree
-wpredp=2

ffpresets/libx264-fastfirstpass.ffpreset

@@ -1,22 +0,0 @@
-coder=1
-flags=+loop
-cmp=+chroma
-partitions=-parti8x8-parti4x4-partp8x8-partp4x4-partb8x8
-me_method=dia
-subq=2
-me_range=16
-g=250
-keyint_min=25
-sc_threshold=40
-i_qfactor=0.71
-b_strategy=1
-qcomp=0.6
-qmin=10
-qmax=51
-qdiff=4
-bf=3
-refs=1
-directpred=3
-trellis=0
-flags2=-bpyramid-wpred-mixed_refs-dct8x8+fastpskip+mbtree
-wpredp=2

ffpresets/libx264-hq.ffpreset

@@ -1,22 +0,0 @@
-coder=1
-flags=+loop
-cmp=+chroma
-partitions=+parti8x8+parti4x4+partp8x8+partb8x8
-me_method=umh
-subq=8
-me_range=16
-g=250
-keyint_min=25
-sc_threshold=40
-i_qfactor=0.71
-b_strategy=2
-qcomp=0.6
-qmin=10
-qmax=51
-qdiff=4
-bf=3
-refs=4
-directpred=3
-trellis=1
-flags2=+wpred+mixed_refs+dct8x8+fastpskip+mbtree
-wpredp=2

ffpresets/libx264-ipod320.ffpreset

@@ -1,7 +0,0 @@
-coder=0
-bf=0
-flags2=-wpred-dct8x8+mbtree
-level=13
-maxrate=768000
-bufsize=3000000
-wpredp=0

ffpresets/libx264-ipod640.ffpreset

@@ -1,8 +0,0 @@
-coder=0
-bf=0
-refs=1
-flags2=-wpred-dct8x8+mbtree
-level=30
-maxrate=10000000
-bufsize=10000000
-wpredp=0

ffpresets/libx264-lossless_fast.ffpreset

@@ -1,20 +0,0 @@
-coder=0
-flags=+loop
-cmp=+chroma
-partitions=-parti8x8+parti4x4+partp8x8-partp4x4-partb8x8
-me_method=hex
-subq=3
-me_range=16
-g=250
-keyint_min=25
-sc_threshold=40
-i_qfactor=0.71
-b_strategy=1
-qcomp=0.6
-qmin=10
-qmax=51
-qdiff=4
-directpred=1
-flags2=+fastpskip+mbtree
-cqp=0
-wpredp=0

ffpresets/libx264-lossless_max.ffpreset

@@ -1,21 +0,0 @@
-coder=1
-flags=+loop
-cmp=+chroma
-partitions=+parti8x8+parti4x4+partp8x8+partp4x4-partb8x8
-me_method=esa
-subq=8
-me_range=16
-g=250
-keyint_min=25
-sc_threshold=40
-i_qfactor=0.71
-b_strategy=1
-qcomp=0.6
-qmin=10
-qmax=51
-qdiff=4
-refs=16
-directpred=1
-flags2=+mixed_refs+dct8x8+fastpskip+mbtree
-cqp=0
-wpredp=2

ffpresets/libx264-lossless_medium.ffpreset

@@ -1,20 +0,0 @@
-coder=1
-flags=+loop
-cmp=+chroma
-partitions=-parti8x8+parti4x4+partp8x8+partp4x4-partb8x8
-me_method=hex
-subq=5
-me_range=16
-g=250
-keyint_min=25
-sc_threshold=40
-i_qfactor=0.71
-b_strategy=1
-qcomp=0.6
-qmin=10
-qmax=51
-qdiff=4
-directpred=1
-flags2=+fastpskip+mbtree
-cqp=0
-wpredp=2

ffpresets/libx264-lossless_slow.ffpreset

@@ -1,21 +0,0 @@
-coder=1
-flags=+loop
-cmp=+chroma
-partitions=+parti8x8+parti4x4+partp8x8+partp4x4-partb8x8
-me_method=umh
-subq=6
-me_range=16
-g=250
-keyint_min=25
-sc_threshold=40
-i_qfactor=0.71
-b_strategy=1
-qcomp=0.6
-qmin=10
-qmax=51
-qdiff=4
-refs=2
-directpred=1
-flags2=+dct8x8+fastpskip+mbtree
-cqp=0
-wpredp=2

ffpresets/libx264-lossless_slower.ffpreset

@@ -1,21 +0,0 @@
-coder=1
-flags=+loop
-cmp=+chroma
-partitions=+parti8x8+parti4x4+partp8x8+partp4x4-partb8x8
-me_method=umh
-subq=8
-me_range=16
-g=250
-keyint_min=25
-sc_threshold=40
-i_qfactor=0.71
-b_strategy=1
-qcomp=0.6
-qmin=10
-qmax=51
-qdiff=4
-refs=4
-directpred=1
-flags2=+mixed_refs+dct8x8+fastpskip+mbtree
-cqp=0
-wpredp=2

ffpresets/libx264-lossless_ultrafast.ffpreset

@@ -1,19 +0,0 @@
-coder=0
-flags=+loop
-cmp=+chroma
-partitions=-parti8x8-parti4x4-partp8x8-partp4x4-partb8x8
-me_method=dia
-subq=0
-me_range=16
-g=250
-keyint_min=25
-sc_threshold=40
-i_qfactor=0.71
-b_strategy=1
-qcomp=0.6
-qmin=10
-qmax=51
-qdiff=4
-directpred=1
-flags2=+fastpskip+mbtree
-cqp=0

ffpresets/libx264-main.ffpreset

@@ -1 +0,0 @@
-flags2=-dct8x8+mbtree

ffpresets/libx264-max.ffpreset

@@ -1,22 +0,0 @@
-coder=1
-flags=+loop
-cmp=+chroma
-partitions=+parti8x8+parti4x4+partp8x8+partp4x4+partb8x8
-me_method=tesa
-subq=10
-me_range=24
-g=250
-keyint_min=25
-sc_threshold=40
-i_qfactor=0.71
-b_strategy=2
-qcomp=0.6
-qmin=10
-qmax=51
-qdiff=4
-bf=3
-refs=16
-directpred=3
-trellis=2
-flags2=+wpred+mixed_refs+dct8x8-fastpskip+mbtree
-wpredp=2

ffpresets/libx264-normal.ffpreset

@@ -1,22 +0,0 @@
-coder=1
-flags=+loop
-cmp=+chroma
-partitions=+parti8x8+parti4x4+partp8x8+partb8x8
-me_method=hex
-subq=6
-me_range=16
-g=250
-keyint_min=25
-sc_threshold=40
-i_qfactor=0.71
-b_strategy=1
-qcomp=0.6
-qmin=10
-qmax=51
-qdiff=4
-bf=3
-refs=2
-directpred=3
-trellis=0
-flags2=+wpred+dct8x8+fastpskip+mbtree
-wpredp=2

ffpresets/libx264-slowfirstpass.ffpreset

@@ -1,22 +0,0 @@
-coder=1
-flags=+loop
-cmp=+chroma
-partitions=+parti8x8+parti4x4+partp8x8+partb8x8
-me_method=hex
-subq=6
-me_range=16
-g=250
-keyint_min=25
-sc_threshold=40
-i_qfactor=0.71
-b_strategy=2
-qcomp=0.6
-qmin=10
-qmax=51
-qdiff=4
-bf=3
-refs=1
-directpred=3
-trellis=0
-flags2=+wpred+dct8x8+fastpskip+mbtree
-wpredp=2

ffserver.h

@@ -1,28 +0,0 @@
-/*
- * Multiple format streaming server
- * copyright (c) 2002 Fabrice Bellard
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-#ifndef FFMPEG_FFSERVER_H
-#define FFMPEG_FFSERVER_H
-
-/* interface between ffserver and modules */
-
-void ffserver_module_init(void);
-
-#endif /* FFMPEG_FFSERVER_H */

libavcodec/4xm.c

@@ -1,875 +0,0 @@
-/*
- * 4XM codec
- * Copyright (c) 2003 Michael Niedermayer
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/**
- * @file libavcodec/4xm.c
- * 4XM codec.
- */
-
-#include "libavutil/intreadwrite.h"
-#include "avcodec.h"
-#include "dsputil.h"
-#include "bitstream.h"
-#include "bytestream.h"
-
-//#undef NDEBUG
-//#include <assert.h>
-
-#define BLOCK_TYPE_VLC_BITS 5
-#define ACDC_VLC_BITS 9
-
-#define CFRAME_BUFFER_COUNT 100
-
-static const uint8_t block_type_tab[2][4][8][2]={
- {
-  {   //{8,4,2}x{8,4,2}
-    { 0,1}, { 2,2}, { 6,3}, {14,4}, {30,5}, {31,5}, { 0,0}
-  },{ //{8,4}x1
-    { 0,1}, { 0,0}, { 2,2}, { 6,3}, {14,4}, {15,4}, { 0,0}
-  },{ //1x{8,4}
-    { 0,1}, { 2,2}, { 0,0}, { 6,3}, {14,4}, {15,4}, { 0,0}
-  },{ //1x2, 2x1
-    { 0,1}, { 0,0}, { 0,0}, { 2,2}, { 6,3}, {14,4}, {15,4}
-  }
- },{
-  {  //{8,4,2}x{8,4,2}
-    { 1,2}, { 4,3}, { 5,3}, {0,2}, {6,3}, {7,3}, {0,0}
-  },{//{8,4}x1
-    { 1,2}, { 0,0}, { 2,2}, {0,2}, {6,3}, {7,3}, {0,0}
-  },{//1x{8,4}
-    { 1,2}, { 2,2}, { 0,0}, {0,2}, {6,3}, {7,3}, {0,0}
-  },{//1x2, 2x1
-    { 1,2}, { 0,0}, { 0,0}, {0,2}, {2,2}, {6,3}, {7,3}
-  }
- }
-};
-
-static const uint8_t size2index[4][4]={
-  {-1, 3, 1, 1},
-  { 3, 0, 0, 0},
-  { 2, 0, 0, 0},
-  { 2, 0, 0, 0},
-};
-
-static const int8_t mv[256][2]={
-{  0,  0},{  0, -1},{ -1,  0},{  1,  0},{  0,  1},{ -1, -1},{  1, -1},{ -1,  1},
-{  1,  1},{  0, -2},{ -2,  0},{  2,  0},{  0,  2},{ -1, -2},{  1, -2},{ -2, -1},
-{  2, -1},{ -2,  1},{  2,  1},{ -1,  2},{  1,  2},{ -2, -2},{  2, -2},{ -2,  2},
-{  2,  2},{  0, -3},{ -3,  0},{  3,  0},{  0,  3},{ -1, -3},{  1, -3},{ -3, -1},
-{  3, -1},{ -3,  1},{  3,  1},{ -1,  3},{  1,  3},{ -2, -3},{  2, -3},{ -3, -2},
-{  3, -2},{ -3,  2},{  3,  2},{ -2,  3},{  2,  3},{  0, -4},{ -4,  0},{  4,  0},
-{  0,  4},{ -1, -4},{  1, -4},{ -4, -1},{  4, -1},{  4,  1},{ -1,  4},{  1,  4},
-{ -3, -3},{ -3,  3},{  3,  3},{ -2, -4},{ -4, -2},{  4, -2},{ -4,  2},{ -2,  4},
-{  2,  4},{ -3, -4},{  3, -4},{  4, -3},{ -5,  0},{ -4,  3},{ -3,  4},{  3,  4},
-{ -1, -5},{ -5, -1},{ -5,  1},{ -1,  5},{ -2, -5},{  2, -5},{  5, -2},{  5,  2},
-{ -4, -4},{ -4,  4},{ -3, -5},{ -5, -3},{ -5,  3},{  3,  5},{ -6,  0},{  0,  6},
-{ -6, -1},{ -6,  1},{  1,  6},{  2, -6},{ -6,  2},{  2,  6},{ -5, -4},{  5,  4},
-{  4,  5},{ -6, -3},{  6,  3},{ -7,  0},{ -1, -7},{  5, -5},{ -7,  1},{ -1,  7},
-{  4, -6},{  6,  4},{ -2, -7},{ -7,  2},{ -3, -7},{  7, -3},{  3,  7},{  6, -5},
-{  0, -8},{ -1, -8},{ -7, -4},{ -8,  1},{  4,  7},{  2, -8},{ -2,  8},{  6,  6},
-{ -8,  3},{  5, -7},{ -5,  7},{  8, -4},{  0, -9},{ -9, -1},{  1,  9},{  7, -6},
-{ -7,  6},{ -5, -8},{ -5,  8},{ -9,  3},{  9, -4},{  7, -7},{  8, -6},{  6,  8},
-{ 10,  1},{-10,  2},{  9, -5},{ 10, -3},{ -8, -7},{-10, -4},{  6, -9},{-11,  0},
-{ 11,  1},{-11, -2},{ -2, 11},{  7, -9},{ -7,  9},{ 10,  6},{ -4, 11},{  8, -9},
-{  8,  9},{  5, 11},{  7,-10},{ 12, -3},{ 11,  6},{ -9, -9},{  8, 10},{  5, 12},
-{-11,  7},{ 13,  2},{  6,-12},{ 10,  9},{-11,  8},{ -7, 12},{  0, 14},{ 14, -2},
-{ -9, 11},{ -6, 13},{-14, -4},{ -5,-14},{  5, 14},{-15, -1},{-14, -6},{  3,-15},
-{ 11,-11},{ -7, 14},{ -5, 15},{  8,-14},{ 15,  6},{  3, 16},{  7,-15},{-16,  5},
-{  0, 17},{-16, -6},{-10, 14},{-16,  7},{ 12, 13},{-16,  8},{-17,  6},{-18,  3},
-{ -7, 17},{ 15, 11},{ 16, 10},{  2,-19},{  3,-19},{-11,-16},{-18,  8},{-19, -6},
-{  2,-20},{-17,-11},{-10,-18},{  8, 19},{-21, -1},{-20,  7},{ -4, 21},{ 21,  5},
-{ 15, 16},{  2,-22},{-10,-20},{-22,  5},{ 20,-11},{ -7,-22},{-12, 20},{ 23, -5},
-{ 13,-20},{ 24, -2},{-15, 19},{-11, 22},{ 16, 19},{ 23,-10},{-18,-18},{ -9,-24},
-{ 24,-10},{ -3, 26},{-23, 13},{-18,-20},{ 17, 21},{ -4, 27},{ 27,  6},{  1,-28},
-{-11, 26},{-17,-23},{  7, 28},{ 11,-27},{ 29,  5},{-23,-19},{-28,-11},{-21, 22},
-{-30,  7},{-17, 26},{-27, 16},{ 13, 29},{ 19,-26},{ 10,-31},{-14,-30},{ 20,-27},
-{-29, 18},{-16,-31},{-28,-22},{ 21,-30},{-25, 28},{ 26,-29},{ 25,-32},{-32,-32}
-};
-
-// this is simply the scaled down elementwise product of the standard jpeg quantizer table and the AAN premul table
-static const uint8_t dequant_table[64]={
- 16, 15, 13, 19, 24, 31, 28, 17,
- 17, 23, 25, 31, 36, 63, 45, 21,
- 18, 24, 27, 37, 52, 59, 49, 20,
- 16, 28, 34, 40, 60, 80, 51, 20,
- 18, 31, 48, 66, 68, 86, 56, 21,
- 19, 38, 56, 59, 64, 64, 48, 20,
- 27, 48, 55, 55, 56, 51, 35, 15,
- 20, 35, 34, 32, 31, 22, 15,  8,
-};
-
-static VLC block_type_vlc[2][4];
-
-
-typedef struct CFrameBuffer{
-    unsigned int allocated_size;
-    unsigned int size;
-    int id;
-    uint8_t *data;
-}CFrameBuffer;
-
-typedef struct FourXContext{
-    AVCodecContext *avctx;
-    DSPContext dsp;
-    AVFrame current_picture, last_picture;
-    GetBitContext pre_gb;          ///< ac/dc prefix
-    GetBitContext gb;
-    const uint8_t *bytestream;
-    const uint8_t *bytestream_end;
-    const uint16_t *wordstream;
-    const uint16_t *wordstream_end;
-    int mv[256];
-    VLC pre_vlc;
-    int last_dc;
-    DECLARE_ALIGNED_8(DCTELEM, block[6][64]);
-    uint8_t *bitstream_buffer;
-    unsigned int bitstream_buffer_size;
-    int version;
-    CFrameBuffer cfrm[CFRAME_BUFFER_COUNT];
-} FourXContext;
-
-
-#define FIX_1_082392200  70936
-#define FIX_1_414213562  92682
-#define FIX_1_847759065 121095
-#define FIX_2_613125930 171254
-
-#define MULTIPLY(var,const)  (((var)*(const)) >> 16)
-
-static void idct(DCTELEM block[64]){
-    int tmp0, tmp1, tmp2, tmp3, tmp4, tmp5, tmp6, tmp7;
-    int tmp10, tmp11, tmp12, tmp13;
-    int z5, z10, z11, z12, z13;
-    int i;
-    int temp[64];
-
-    for(i=0; i<8; i++){
-        tmp10 = block[8*0 + i] + block[8*4 + i];
-        tmp11 = block[8*0 + i] - block[8*4 + i];
-
-        tmp13 =          block[8*2 + i] + block[8*6 + i];
-        tmp12 = MULTIPLY(block[8*2 + i] - block[8*6 + i], FIX_1_414213562) - tmp13;
-
-        tmp0 = tmp10 + tmp13;
-        tmp3 = tmp10 - tmp13;
-        tmp1 = tmp11 + tmp12;
-        tmp2 = tmp11 - tmp12;
-
-        z13 = block[8*5 + i] + block[8*3 + i];
-        z10 = block[8*5 + i] - block[8*3 + i];
-        z11 = block[8*1 + i] + block[8*7 + i];
-        z12 = block[8*1 + i] - block[8*7 + i];
-
-        tmp7  =          z11 + z13;
-        tmp11 = MULTIPLY(z11 - z13, FIX_1_414213562);
-
-        z5    = MULTIPLY(z10 + z12, FIX_1_847759065);
-        tmp10 = MULTIPLY(z12, FIX_1_082392200) - z5;
-        tmp12 = MULTIPLY(z10, - FIX_2_613125930) + z5;
-
-        tmp6 = tmp12 - tmp7;
-        tmp5 = tmp11 - tmp6;
-        tmp4 = tmp10 + tmp5;
-
-        temp[8*0 + i] = tmp0 + tmp7;
-        temp[8*7 + i] = tmp0 - tmp7;
-        temp[8*1 + i] = tmp1 + tmp6;
-        temp[8*6 + i] = tmp1 - tmp6;
-        temp[8*2 + i] = tmp2 + tmp5;
-        temp[8*5 + i] = tmp2 - tmp5;
-        temp[8*4 + i] = tmp3 + tmp4;
-        temp[8*3 + i] = tmp3 - tmp4;
-    }
-
-    for(i=0; i<8*8; i+=8){
-        tmp10 = temp[0 + i] + temp[4 + i];
-        tmp11 = temp[0 + i] - temp[4 + i];
-
-        tmp13 = temp[2 + i] + temp[6 + i];
-        tmp12 = MULTIPLY(temp[2 + i] - temp[6 + i], FIX_1_414213562) - tmp13;
-
-        tmp0 = tmp10 + tmp13;
-        tmp3 = tmp10 - tmp13;
-        tmp1 = tmp11 + tmp12;
-        tmp2 = tmp11 - tmp12;
-
-        z13 = temp[5 + i] + temp[3 + i];
-        z10 = temp[5 + i] - temp[3 + i];
-        z11 = temp[1 + i] + temp[7 + i];
-        z12 = temp[1 + i] - temp[7 + i];
-
-        tmp7 = z11 + z13;
-        tmp11 = MULTIPLY(z11 - z13, FIX_1_414213562);
-
-        z5 = MULTIPLY(z10 + z12, FIX_1_847759065);
-        tmp10 = MULTIPLY(z12, FIX_1_082392200) - z5;
-        tmp12 = MULTIPLY(z10, - FIX_2_613125930) + z5;
-
-        tmp6 = tmp12 - tmp7;
-        tmp5 = tmp11 - tmp6;
-        tmp4 = tmp10 + tmp5;
-
-        block[0 + i] = (tmp0 + tmp7)>>6;
-        block[7 + i] = (tmp0 - tmp7)>>6;
-        block[1 + i] = (tmp1 + tmp6)>>6;
-        block[6 + i] = (tmp1 - tmp6)>>6;
-        block[2 + i] = (tmp2 + tmp5)>>6;
-        block[5 + i] = (tmp2 - tmp5)>>6;
-        block[4 + i] = (tmp3 + tmp4)>>6;
-        block[3 + i] = (tmp3 - tmp4)>>6;
-    }
-}
-
-static av_cold void init_vlcs(FourXContext *f){
-    int i;
-
-    for(i=0; i<8; i++){
-        init_vlc(&block_type_vlc[0][i], BLOCK_TYPE_VLC_BITS, 7,
-                 &block_type_tab[0][i][0][1], 2, 1,
-                 &block_type_tab[0][i][0][0], 2, 1, 1);
-    }
-}
-
-static void init_mv(FourXContext *f){
-    int i;
-
-    for(i=0; i<256; i++){
-        if(f->version>1)
-            f->mv[i] = mv[i][0]   + mv[i][1]  *f->current_picture.linesize[0]/2;
-        else
-            f->mv[i] = (i&15) - 8 + ((i>>4)-8)*f->current_picture.linesize[0]/2;
-    }
-}
-
-static inline void mcdc(uint16_t *dst, uint16_t *src, int log2w, int h, int stride, int scale, int dc){
-   int i;
-   dc*= 0x10001;
-
-   switch(log2w){
-   case 0:
-        for(i=0; i<h; i++){
-            dst[0] = scale*src[0] + dc;
-            if(scale) src += stride;
-            dst += stride;
-        }
-        break;
-    case 1:
-        for(i=0; i<h; i++){
-            ((uint32_t*)dst)[0] = scale*((uint32_t*)src)[0] + dc;
-            if(scale) src += stride;
-            dst += stride;
-        }
-        break;
-    case 2:
-        for(i=0; i<h; i++){
-            ((uint32_t*)dst)[0] = scale*((uint32_t*)src)[0] + dc;
-            ((uint32_t*)dst)[1] = scale*((uint32_t*)src)[1] + dc;
-            if(scale) src += stride;
-            dst += stride;
-        }
-        break;
-    case 3:
-        for(i=0; i<h; i++){
-            ((uint32_t*)dst)[0] = scale*((uint32_t*)src)[0] + dc;
-            ((uint32_t*)dst)[1] = scale*((uint32_t*)src)[1] + dc;
-            ((uint32_t*)dst)[2] = scale*((uint32_t*)src)[2] + dc;
-            ((uint32_t*)dst)[3] = scale*((uint32_t*)src)[3] + dc;
-            if(scale) src += stride;
-            dst += stride;
-        }
-        break;
-    default: assert(0);
-    }
-}
-
-static void decode_p_block(FourXContext *f, uint16_t *dst, uint16_t *src, int log2w, int log2h, int stride){
-    const int index= size2index[log2h][log2w];
-    const int h= 1<<log2h;
-    int code= get_vlc2(&f->gb, block_type_vlc[1-(f->version>1)][index].table, BLOCK_TYPE_VLC_BITS, 1);
-    uint16_t *start= (uint16_t*)f->last_picture.data[0];
-    uint16_t *end= start + stride*(f->avctx->height-h+1) - (1<<log2w);
-
-    assert(code>=0 && code<=6);
-
-    if(code == 0){
-        if (f->bytestream_end - f->bytestream < 1)
-            return;
-        src += f->mv[ *f->bytestream++ ];
-        if(start > src || src > end){
-            av_log(f->avctx, AV_LOG_ERROR, "mv out of pic\n");
-            return;
-        }
-        mcdc(dst, src, log2w, h, stride, 1, 0);
-    }else if(code == 1){
-        log2h--;
-        decode_p_block(f, dst                  , src                  , log2w, log2h, stride);
-        decode_p_block(f, dst + (stride<<log2h), src + (stride<<log2h), log2w, log2h, stride);
-    }else if(code == 2){
-        log2w--;
-        decode_p_block(f, dst             , src             , log2w, log2h, stride);
-        decode_p_block(f, dst + (1<<log2w), src + (1<<log2w), log2w, log2h, stride);
-    }else if(code == 3 && f->version<2){
-        mcdc(dst, src, log2w, h, stride, 1, 0);
-    }else if(code == 4){
-        if (f->bytestream_end - f->bytestream < 1)
-            return;
-        src += f->mv[ *f->bytestream++ ];
-        if(start > src || src > end){
-            av_log(f->avctx, AV_LOG_ERROR, "mv out of pic\n");
-            return;
-        }
-        if (f->wordstream_end - f->wordstream < 1)
-            return;
-        mcdc(dst, src, log2w, h, stride, 1, le2me_16(*f->wordstream++));
-    }else if(code == 5){
-        if (f->wordstream_end - f->wordstream < 1)
-            return;
-        mcdc(dst, src, log2w, h, stride, 0, le2me_16(*f->wordstream++));
-    }else if(code == 6){
-        if (f->wordstream_end - f->wordstream < 2)
-            return;
-        if(log2w){
-            dst[0] = le2me_16(*f->wordstream++);
-            dst[1] = le2me_16(*f->wordstream++);
-        }else{
-            dst[0     ] = le2me_16(*f->wordstream++);
-            dst[stride] = le2me_16(*f->wordstream++);
-        }
-    }
-}
-
-static int decode_p_frame(FourXContext *f, const uint8_t *buf, int length){
-    int x, y;
-    const int width= f->avctx->width;
-    const int height= f->avctx->height;
-    uint16_t *src= (uint16_t*)f->last_picture.data[0];
-    uint16_t *dst= (uint16_t*)f->current_picture.data[0];
-    const int stride= f->current_picture.linesize[0]>>1;
-    unsigned int bitstream_size, bytestream_size, wordstream_size, extra;
-
-    if(f->version>1){
-        extra=20;
-        if (length < extra)
-            return -1;
-        bitstream_size= AV_RL32(buf+8);
-        wordstream_size= AV_RL32(buf+12);
-        bytestream_size= AV_RL32(buf+16);
-    }else{
-        extra=0;
-        bitstream_size = AV_RL16(buf-4);
-        wordstream_size= AV_RL16(buf-2);
-        bytestream_size= FFMAX(length - bitstream_size - wordstream_size, 0);
-    }
-
-    if (bitstream_size > length ||
-        bytestream_size > length - bitstream_size ||
-        wordstream_size > length - bytestream_size - bitstream_size ||
-        extra > length - bytestream_size - bitstream_size - wordstream_size){
-        av_log(f->avctx, AV_LOG_ERROR, "lengths %d %d %d %d\n", bitstream_size, bytestream_size, wordstream_size,
-        bitstream_size+ bytestream_size+ wordstream_size - length);
-        return -1;
-    }
-
-    f->bitstream_buffer= av_fast_realloc(f->bitstream_buffer, &f->bitstream_buffer_size, bitstream_size + FF_INPUT_BUFFER_PADDING_SIZE);
-    f->dsp.bswap_buf((uint32_t*)f->bitstream_buffer, (const uint32_t*)(buf + extra), bitstream_size/4);
-    init_get_bits(&f->gb, f->bitstream_buffer, 8*bitstream_size);
-
-    f->wordstream= (const uint16_t*)(buf + extra + bitstream_size);
-    f->wordstream_end= f->wordstream + wordstream_size/2;
-    f->bytestream= buf + extra + bitstream_size + wordstream_size;
-    f->bytestream_end = f->bytestream + bytestream_size;
-
-    init_mv(f);
-
-    for(y=0; y<height; y+=8){
-        for(x=0; x<width; x+=8){
-            decode_p_block(f, dst + x, src + x, 3, 3, stride);
-        }
-        src += 8*stride;
-        dst += 8*stride;
-    }
-
-    if(   bitstream_size != (get_bits_count(&f->gb)+31)/32*4
-       || (((const char*)f->wordstream - (const char*)buf + 2)&~2) != extra + bitstream_size + wordstream_size
-       || (((const char*)f->bytestream - (const char*)buf + 3)&~3) != extra + bitstream_size + wordstream_size + bytestream_size)
-        av_log(f->avctx, AV_LOG_ERROR, " %d %td %td bytes left\n",
-            bitstream_size - (get_bits_count(&f->gb)+31)/32*4,
-            -(((const char*)f->bytestream - (const char*)buf + 3)&~3) + (extra + bitstream_size + wordstream_size + bytestream_size),
-            -(((const char*)f->wordstream - (const char*)buf + 2)&~2) + (extra + bitstream_size + wordstream_size)
-        );
-
-    return 0;
-}
-
-/**
- * decode block and dequantize.
- * Note this is almost identical to MJPEG.
- */
-static int decode_i_block(FourXContext *f, DCTELEM *block){
-    int code, i, j, level, val;
-
-    /* DC coef */
-    val = get_vlc2(&f->pre_gb, f->pre_vlc.table, ACDC_VLC_BITS, 3);
-    if (val>>4){
-        av_log(f->avctx, AV_LOG_ERROR, "error dc run != 0\n");
-    }
-
-    if(val)
-        val = get_xbits(&f->gb, val);
-
-    val = val * dequant_table[0] + f->last_dc;
-    f->last_dc =
-    block[0] = val;
-    /* AC coefs */
-    i = 1;
-    for(;;) {
-        code = get_vlc2(&f->pre_gb, f->pre_vlc.table, ACDC_VLC_BITS, 3);
-
-        /* EOB */
-        if (code == 0)
-            break;
-        if (code == 0xf0) {
-            i += 16;
-        } else {
-            level = get_xbits(&f->gb, code & 0xf);
-            i += code >> 4;
-            if (i >= 64) {
-                av_log(f->avctx, AV_LOG_ERROR, "run %d oveflow\n", i);
-                return 0;
-            }
-
-            j= ff_zigzag_direct[i];
-            block[j] = level * dequant_table[j];
-            i++;
-            if (i >= 64)
-                break;
-        }
-    }
-
-    return 0;
-}
-
-static inline void idct_put(FourXContext *f, int x, int y){
-    DCTELEM (*block)[64]= f->block;
-    int stride= f->current_picture.linesize[0]>>1;
-    int i;
-    uint16_t *dst = ((uint16_t*)f->current_picture.data[0]) + y * stride + x;
-
-    for(i=0; i<4; i++){
-        block[i][0] += 0x80*8*8;
-        idct(block[i]);
-    }
-
-    if(!(f->avctx->flags&CODEC_FLAG_GRAY)){
-        for(i=4; i<6; i++) idct(block[i]);
-    }
-
-/* Note transform is:
-y= ( 1b + 4g + 2r)/14
-cb=( 3b - 2g - 1r)/14
-cr=(-1b - 4g + 5r)/14
-*/
-    for(y=0; y<8; y++){
-        for(x=0; x<8; x++){
-            DCTELEM *temp= block[(x>>2) + 2*(y>>2)] + 2*(x&3) + 2*8*(y&3); //FIXME optimize
-            int cb= block[4][x + 8*y];
-            int cr= block[5][x + 8*y];
-            int cg= (cb + cr)>>1;
-            int y;
-
-            cb+=cb;
-
-            y = temp[0];
-            dst[0       ]= ((y+cb)>>3) + (((y-cg)&0xFC)<<3) + (((y+cr)&0xF8)<<8);
-            y = temp[1];
-            dst[1       ]= ((y+cb)>>3) + (((y-cg)&0xFC)<<3) + (((y+cr)&0xF8)<<8);
-            y = temp[8];
-            dst[  stride]= ((y+cb)>>3) + (((y-cg)&0xFC)<<3) + (((y+cr)&0xF8)<<8);
-            y = temp[9];
-            dst[1+stride]= ((y+cb)>>3) + (((y-cg)&0xFC)<<3) + (((y+cr)&0xF8)<<8);
-            dst += 2;
-        }
-        dst += 2*stride - 2*8;
-    }
-}
-
-static int decode_i_mb(FourXContext *f){
-    int i;
-
-    f->dsp.clear_blocks(f->block[0]);
-
-    for(i=0; i<6; i++){
-        if(decode_i_block(f, f->block[i]) < 0)
-            return -1;
-    }
-
-    return 0;
-}
-
-static const uint8_t *read_huffman_tables(FourXContext *f, const uint8_t * const buf, int buf_size){
-    int frequency[512];
-    uint8_t flag[512];
-    int up[512];
-    uint8_t len_tab[257];
-    int bits_tab[257];
-    int start, end;
-    const uint8_t *ptr= buf;
-    const uint8_t *ptr_end = buf + buf_size;
-    int j;
-
-    memset(frequency, 0, sizeof(frequency));
-    memset(up, -1, sizeof(up));
-
-    start= *ptr++;
-    end= *ptr++;
-    for(;;){
-        int i;
-
-        if (start <= end && ptr_end - ptr < end - start + 1 + 1)
-            return NULL;
-        for(i=start; i<=end; i++){
-            frequency[i]= *ptr++;
-        }
-        start= *ptr++;
-        if(start==0) break;
-
-        end= *ptr++;
-    }
-    frequency[256]=1;
-
-    while((ptr - buf)&3) ptr++; // 4byte align
-
-    for(j=257; j<512; j++){
-        int min_freq[2]= {256*256, 256*256};
-        int smallest[2]= {0, 0};
-        int i;
-        for(i=0; i<j; i++){
-            if(frequency[i] == 0) continue;
-            if(frequency[i] < min_freq[1]){
-                if(frequency[i] < min_freq[0]){
-                    min_freq[1]= min_freq[0]; smallest[1]= smallest[0];
-                    min_freq[0]= frequency[i];smallest[0]= i;
-                }else{
-                    min_freq[1]= frequency[i];smallest[1]= i;
-                }
-            }
-        }
-        if(min_freq[1] == 256*256) break;
-
-        frequency[j]= min_freq[0] + min_freq[1];
-        flag[ smallest[0] ]= 0;
-        flag[ smallest[1] ]= 1;
-        up[ smallest[0] ]=
-        up[ smallest[1] ]= j;
-        frequency[ smallest[0] ]= frequency[ smallest[1] ]= 0;
-    }
-
-    for(j=0; j<257; j++){
-        int node;
-        int len=0;
-        int bits=0;
-
-        for(node= j; up[node] != -1; node= up[node]){
-            bits += flag[node]<<len;
-            len++;
-            if(len > 31) av_log(f->avctx, AV_LOG_ERROR, "vlc length overflow\n"); //can this happen at all ?
-        }
-
-        bits_tab[j]= bits;
-        len_tab[j]= len;
-    }
-
-    init_vlc(&f->pre_vlc, ACDC_VLC_BITS, 257,
-             len_tab , 1, 1,
-             bits_tab, 4, 4, 0);
-
-    return ptr;
-}
-
-static int mix(int c0, int c1){
-    int blue = 2*(c0&0x001F) + (c1&0x001F);
-    int green= (2*(c0&0x03E0) + (c1&0x03E0))>>5;
-    int red  = 2*(c0>>10) + (c1>>10);
-    return red/3*1024 + green/3*32 + blue/3;
-}
-
-static int decode_i2_frame(FourXContext *f, const uint8_t *buf, int length){
-    int x, y, x2, y2;
-    const int width= f->avctx->width;
-    const int height= f->avctx->height;
-    uint16_t *dst= (uint16_t*)f->current_picture.data[0];
-    const int stride= f->current_picture.linesize[0]>>1;
-    const uint8_t *buf_end = buf + length;
-
-    for(y=0; y<height; y+=16){
-        for(x=0; x<width; x+=16){
-            unsigned int color[4], bits;
-            if (buf_end - buf < 8)
-                return -1;
-            memset(color, 0, sizeof(color));
-//warning following is purely guessed ...
-            color[0]= bytestream_get_le16(&buf);
-            color[1]= bytestream_get_le16(&buf);
-
-            if(color[0]&0x8000) av_log(NULL, AV_LOG_ERROR, "unk bit 1\n");
-            if(color[1]&0x8000) av_log(NULL, AV_LOG_ERROR, "unk bit 2\n");
-
-            color[2]= mix(color[0], color[1]);
-            color[3]= mix(color[1], color[0]);
-
-            bits= bytestream_get_le32(&buf);
-            for(y2=0; y2<16; y2++){
-                for(x2=0; x2<16; x2++){
-                    int index= 2*(x2>>2) + 8*(y2>>2);
-                    dst[y2*stride+x2]= color[(bits>>index)&3];
-                }
-            }
-            dst+=16;
-        }
-        dst += 16*stride - width;
-    }
-
-    return 0;
-}
-
-static int decode_i_frame(FourXContext *f, const uint8_t *buf, int length){
-    int x, y;
-    const int width= f->avctx->width;
-    const int height= f->avctx->height;
-    uint16_t *dst= (uint16_t*)f->current_picture.data[0];
-    const int stride= f->current_picture.linesize[0]>>1;
-    const unsigned int bitstream_size= AV_RL32(buf);
-    unsigned int prestream_size;
-    const uint8_t *prestream;
-
-    if (bitstream_size > (1<<26) || length < bitstream_size + 12)
-        return -1;
-    prestream_size = 4*AV_RL32(buf + bitstream_size + 4);
-    prestream = buf + bitstream_size + 12;
-
-    if (prestream_size > (1<<26) ||
-        prestream_size != length - (bitstream_size + 12)){
-        av_log(f->avctx, AV_LOG_ERROR, "size mismatch %d %d %d\n", prestream_size, bitstream_size, length);
-        return -1;
-    }
-
-    prestream= read_huffman_tables(f, prestream, buf + length - prestream);
-    if (!prestream)
-        return -1;
-
-    init_get_bits(&f->gb, buf + 4, 8*bitstream_size);
-
-    prestream_size= length + buf - prestream;
-
-    f->bitstream_buffer= av_fast_realloc(f->bitstream_buffer, &f->bitstream_buffer_size, prestream_size + FF_INPUT_BUFFER_PADDING_SIZE);
-    f->dsp.bswap_buf((uint32_t*)f->bitstream_buffer, (const uint32_t*)prestream, prestream_size/4);
-    init_get_bits(&f->pre_gb, f->bitstream_buffer, 8*prestream_size);
-
-    f->last_dc= 0*128*8*8;
-
-    for(y=0; y<height; y+=16){
-        for(x=0; x<width; x+=16){
-            if(decode_i_mb(f) < 0)
-                return -1;
-
-            idct_put(f, x, y);
-        }
-        dst += 16*stride;
-    }
-
-    if(get_vlc2(&f->pre_gb, f->pre_vlc.table, ACDC_VLC_BITS, 3) != 256)
-        av_log(f->avctx, AV_LOG_ERROR, "end mismatch\n");
-
-    return 0;
-}
-
-static int decode_frame(AVCodecContext *avctx,
-                        void *data, int *data_size,
-                        const uint8_t *buf, int buf_size)
-{
-    FourXContext * const f = avctx->priv_data;
-    AVFrame *picture = data;
-    AVFrame *p, temp;
-    int i, frame_4cc, frame_size;
-
-    if (buf_size < 12)
-        return AVERROR_INVALIDDATA;
-    frame_4cc= AV_RL32(buf);
-    if(buf_size != AV_RL32(buf+4)+8 || buf_size < 20){
-        av_log(f->avctx, AV_LOG_ERROR, "size mismatch %d %d\n", buf_size, AV_RL32(buf+4));
-    }
-
-    if(frame_4cc == AV_RL32("cfrm")){
-        int free_index=-1;
-        const int data_size= buf_size - 20;
-        const int id= AV_RL32(buf+12);
-        const int whole_size= AV_RL32(buf+16);
-        CFrameBuffer *cfrm;
-
-        if (data_size < 0 || whole_size < 0)
-            return AVERROR_INVALIDDATA;
-
-        for(i=0; i<CFRAME_BUFFER_COUNT; i++){
-            if(f->cfrm[i].id && f->cfrm[i].id < avctx->frame_number)
-                av_log(f->avctx, AV_LOG_ERROR, "lost c frame %d\n", f->cfrm[i].id);
-        }
-
-        for(i=0; i<CFRAME_BUFFER_COUNT; i++){
-            if(f->cfrm[i].id   == id) break;
-            if(f->cfrm[i].size == 0 ) free_index= i;
-        }
-
-        if(i>=CFRAME_BUFFER_COUNT){
-            i= free_index;
-            f->cfrm[i].id= id;
-        }
-        cfrm= &f->cfrm[i];
-
-        if (data_size > UINT_MAX -  cfrm->size - FF_INPUT_BUFFER_PADDING_SIZE)
-            return AVERROR_INVALIDDATA;
-        cfrm->data= av_fast_realloc(cfrm->data, &cfrm->allocated_size, cfrm->size + data_size + FF_INPUT_BUFFER_PADDING_SIZE);
-        if(!cfrm->data){ //explicit check needed as memcpy below might not catch a NULL
-            av_log(f->avctx, AV_LOG_ERROR, "realloc falure");
-            return -1;
-        }
-
-        memcpy(cfrm->data + cfrm->size, buf+20, data_size);
-        cfrm->size += data_size;
-
-        if(cfrm->size >= whole_size){
-            buf= cfrm->data;
-            frame_size= cfrm->size;
-
-            if(id != avctx->frame_number){
-                av_log(f->avctx, AV_LOG_ERROR, "cframe id mismatch %d %d\n", id, avctx->frame_number);
-            }
-
-            cfrm->size= cfrm->id= 0;
-            frame_4cc= AV_RL32("pfrm");
-        }else
-            return buf_size;
-    }else{
-        buf= buf + 12;
-        frame_size= buf_size - 12;
-    }
-
-    temp= f->current_picture;
-    f->current_picture= f->last_picture;
-    f->last_picture= temp;
-
-    p= &f->current_picture;
-    avctx->coded_frame= p;
-
-    avctx->flags |= CODEC_FLAG_EMU_EDGE; // alternatively we would have to use our own buffer management
-
-    if(p->data[0])
-        avctx->release_buffer(avctx, p);
-
-    p->reference= 1;
-    if(avctx->get_buffer(avctx, p) < 0){
-        av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
-        return -1;
-    }
-
-    if(frame_4cc == AV_RL32("ifr2")){
-        p->pict_type= FF_I_TYPE;
-        if(decode_i2_frame(f, buf-4, frame_size) < 0)
-            return -1;
-    }else if(frame_4cc == AV_RL32("ifrm")){
-        p->pict_type= FF_I_TYPE;
-        if(decode_i_frame(f, buf, frame_size) < 0)
-            return -1;
-    }else if(frame_4cc == AV_RL32("pfrm") || frame_4cc == AV_RL32("pfr2")){
-        p->pict_type= FF_P_TYPE;
-        if(decode_p_frame(f, buf, frame_size) < 0)
-            return -1;
-    }else if(frame_4cc == AV_RL32("snd_")){
-        av_log(avctx, AV_LOG_ERROR, "ignoring snd_ chunk length:%d\n", buf_size);
-    }else{
-        av_log(avctx, AV_LOG_ERROR, "ignoring unknown chunk length:%d\n", buf_size);
-    }
-
-    p->key_frame= p->pict_type == FF_I_TYPE;
-
-    *picture= *p;
-    *data_size = sizeof(AVPicture);
-
-    emms_c();
-
-    return buf_size;
-}
-
-
-static av_cold void common_init(AVCodecContext *avctx){
-    FourXContext * const f = avctx->priv_data;
-
-    dsputil_init(&f->dsp, avctx);
-
-    f->avctx= avctx;
-}
-
-static av_cold int decode_init(AVCodecContext *avctx){
-    FourXContext * const f = avctx->priv_data;
-
-    if(avctx->extradata_size != 4 || !avctx->extradata) {
-        av_log(avctx, AV_LOG_ERROR, "extradata wrong or missing\n");
-        return 1;
-    }
-
-    f->version= AV_RL32(avctx->extradata)>>16;
-    common_init(avctx);
-    init_vlcs(f);
-
-    if(f->version>2) avctx->pix_fmt= PIX_FMT_RGB565;
-    else             avctx->pix_fmt= PIX_FMT_RGB555;
-
-    return 0;
-}
-
-
-static av_cold int decode_end(AVCodecContext *avctx){
-    FourXContext * const f = avctx->priv_data;
-    int i;
-
-    av_freep(&f->bitstream_buffer);
-    f->bitstream_buffer_size=0;
-    for(i=0; i<CFRAME_BUFFER_COUNT; i++){
-        av_freep(&f->cfrm[i].data);
-        f->cfrm[i].allocated_size= 0;
-    }
-    free_vlc(&f->pre_vlc);
-
-    return 0;
-}
-
-AVCodec fourxm_decoder = {
-    "4xm",
-    CODEC_TYPE_VIDEO,
-    CODEC_ID_4XM,
-    sizeof(FourXContext),
-    decode_init,
-    NULL,
-    decode_end,
-    decode_frame,
-    /*CODEC_CAP_DR1,*/
-    .long_name = NULL_IF_CONFIG_SMALL("4X Movie"),
-};
-

libavcodec/8bps.c

@@ -1,235 +0,0 @@
-/*
- * Quicktime Planar RGB (8BPS) Video Decoder
- * Copyright (C) 2003 Roberto Togni
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/**
- * @file libavcodec/8bps.c
- * QT 8BPS Video Decoder by Roberto Togni
- * For more information about the 8BPS format, visit:
- *   http://www.pcisys.net/~melanson/codecs/
- *
- * Supports: PAL8 (RGB 8bpp, paletted)
- *         : BGR24 (RGB 24bpp) (can also output it as RGB32)
- *         : RGB32 (RGB 32bpp, 4th plane is probably alpha and it's ignored)
- *
- */
-
-#include <stdio.h>
-#include <stdlib.h>
-
-#include "libavutil/intreadwrite.h"
-#include "avcodec.h"
-
-
-static const enum PixelFormat pixfmt_rgb24[] = {PIX_FMT_BGR24, PIX_FMT_RGB32, PIX_FMT_NONE};
-
-/*
- * Decoder context
- */
-typedef struct EightBpsContext {
-
-        AVCodecContext *avctx;
-        AVFrame pic;
-
-        unsigned char planes;
-        unsigned char planemap[4];
-} EightBpsContext;
-
-
-/*
- *
- * Decode a frame
- *
- */
-static int decode_frame(AVCodecContext *avctx, void *data, int *data_size, const uint8_t *buf, int buf_size)
-{
-        EightBpsContext * const c = avctx->priv_data;
-        const unsigned char *encoded = buf;
-        unsigned char *pixptr, *pixptr_end;
-        unsigned int height = avctx->height; // Real image height
-        unsigned int dlen, p, row;
-        const unsigned char *lp, *dp;
-        unsigned char count;
-        unsigned int px_inc;
-        unsigned int planes = c->planes;
-        unsigned char *planemap = c->planemap;
-
-        if(c->pic.data[0])
-                avctx->release_buffer(avctx, &c->pic);
-
-        c->pic.reference = 0;
-        c->pic.buffer_hints = FF_BUFFER_HINTS_VALID;
-        if(avctx->get_buffer(avctx, &c->pic) < 0){
-                av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
-                return -1;
-        }
-
-        /* Set data pointer after line lengths */
-        dp = encoded + planes * (height << 1);
-
-        /* Ignore alpha plane, don't know what to do with it */
-        if (planes == 4)
-                planes--;
-
-        px_inc = planes + (avctx->pix_fmt == PIX_FMT_RGB32);
-
-        for (p = 0; p < planes; p++) {
-                /* Lines length pointer for this plane */
-                lp = encoded + p * (height << 1);
-
-                /* Decode a plane */
-                for(row = 0; row < height; row++) {
-                        pixptr = c->pic.data[0] + row * c->pic.linesize[0] + planemap[p];
-                        pixptr_end = pixptr + c->pic.linesize[0];
-                        dlen = be2me_16(*(const unsigned short *)(lp+row*2));
-                        /* Decode a row of this plane */
-                        while(dlen > 0) {
-                                if(dp + 1 >= buf+buf_size) return -1;
-                                if ((count = *dp++) <= 127) {
-                                        count++;
-                                        dlen -= count + 1;
-                                        if (pixptr + count * px_inc > pixptr_end)
-                                            break;
-                                        if(dp + count > buf+buf_size) return -1;
-                                        while(count--) {
-                                                *pixptr = *dp++;
-                                                pixptr += px_inc;
-                                        }
-                                } else {
-                                        count = 257 - count;
-                                        if (pixptr + count * px_inc > pixptr_end)
-                                            break;
-                                        while(count--) {
-                                                *pixptr = *dp;
-                                                pixptr += px_inc;
-                                        }
-                                        dp++;
-                                        dlen -= 2;
-                                }
-                        }
-                }
-        }
-
-        if (avctx->palctrl) {
-                memcpy (c->pic.data[1], avctx->palctrl->palette, AVPALETTE_SIZE);
-                if (avctx->palctrl->palette_changed) {
-                        c->pic.palette_has_changed = 1;
-                        avctx->palctrl->palette_changed = 0;
-                } else
-                        c->pic.palette_has_changed = 0;
-        }
-
-        *data_size = sizeof(AVFrame);
-        *(AVFrame*)data = c->pic;
-
-        /* always report that the buffer was completely consumed */
-        return buf_size;
-}
-
-
-/*
- *
- * Init 8BPS decoder
- *
- */
-static av_cold int decode_init(AVCodecContext *avctx)
-{
-        EightBpsContext * const c = avctx->priv_data;
-
-        c->avctx = avctx;
-
-        c->pic.data[0] = NULL;
-
-    if (avcodec_check_dimensions(avctx, avctx->width, avctx->height) < 0) {
-        return 1;
-    }
-
-        switch (avctx->bits_per_coded_sample) {
-                case 8:
-                        avctx->pix_fmt = PIX_FMT_PAL8;
-                        c->planes = 1;
-                        c->planemap[0] = 0; // 1st plane is palette indexes
-                        if (avctx->palctrl == NULL) {
-                                av_log(avctx, AV_LOG_ERROR, "Error: PAL8 format but no palette from demuxer.\n");
-                                return -1;
-                        }
-                        break;
-                case 24:
-                        avctx->pix_fmt = avctx->get_format(avctx, pixfmt_rgb24);
-                        c->planes = 3;
-                        c->planemap[0] = 2; // 1st plane is red
-                        c->planemap[1] = 1; // 2nd plane is green
-                        c->planemap[2] = 0; // 3rd plane is blue
-                        break;
-                case 32:
-                        avctx->pix_fmt = PIX_FMT_RGB32;
-                        c->planes = 4;
-#ifdef WORDS_BIGENDIAN
-                        c->planemap[0] = 1; // 1st plane is red
-                        c->planemap[1] = 2; // 2nd plane is green
-                        c->planemap[2] = 3; // 3rd plane is blue
-                        c->planemap[3] = 0; // 4th plane is alpha???
-#else
-                        c->planemap[0] = 2; // 1st plane is red
-                        c->planemap[1] = 1; // 2nd plane is green
-                        c->planemap[2] = 0; // 3rd plane is blue
-                        c->planemap[3] = 3; // 4th plane is alpha???
-#endif
-                        break;
-                default:
-                        av_log(avctx, AV_LOG_ERROR, "Error: Unsupported color depth: %u.\n", avctx->bits_per_coded_sample);
-                        return -1;
-        }
-
-  return 0;
-}
-
-
-
-
-/*
- *
- * Uninit 8BPS decoder
- *
- */
-static av_cold int decode_end(AVCodecContext *avctx)
-{
-        EightBpsContext * const c = avctx->priv_data;
-
-        if (c->pic.data[0])
-                avctx->release_buffer(avctx, &c->pic);
-
-        return 0;
-}
-
-
-
-AVCodec eightbps_decoder = {
-        "8bps",
-        CODEC_TYPE_VIDEO,
-        CODEC_ID_8BPS,
-        sizeof(EightBpsContext),
-        decode_init,
-        NULL,
-        decode_end,
-        decode_frame,
-        CODEC_CAP_DR1,
-        .long_name = NULL_IF_CONFIG_SMALL("QuickTime 8BPS video"),
-};

libavcodec/8svx.c

@@ -1,111 +0,0 @@
-/*
- * 8SVX audio decoder
- * Copyright (C) 2008 Jaikrishnan Menon
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/**
- * @file libavcodec/8svx.c
- * 8svx audio decoder
- * @author Jaikrishnan Menon
- * supports: fibonacci delta encoding
- *         : exponential encoding
- */
-
-#include "avcodec.h"
-
-/** decoder context */
-typedef struct EightSvxContext {
-    int16_t fib_acc;
-    const int16_t *table;
-} EightSvxContext;
-
-static const int16_t fibonacci[16]   = { -34<<8, -21<<8, -13<<8,  -8<<8, -5<<8, -3<<8, -2<<8, -1<<8,
-                                          0, 1<<8, 2<<8, 3<<8, 5<<8, 8<<8, 13<<8, 21<<8 };
-static const int16_t exponential[16] = { -128<<8, -64<<8, -32<<8, -16<<8, -8<<8, -4<<8, -2<<8, -1<<8,
-                                          0, 1<<8, 2<<8, 4<<8, 8<<8, 16<<8, 32<<8, 64<<8 };
-
-/** decode a frame */
-static int eightsvx_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
-                                 const uint8_t *buf, int buf_size)
-{
-    EightSvxContext *esc = avctx->priv_data;
-    int16_t *out_data = data;
-    int consumed = buf_size;
-    const uint8_t *buf_end = buf + buf_size;
-
-    if((*data_size >> 2) < buf_size)
-        return -1;
-
-    if(avctx->frame_number == 0) {
-        esc->fib_acc = buf[1] << 8;
-        buf_size -= 2;
-        buf += 2;
-    }
-
-    *data_size = buf_size << 2;
-
-    while(buf < buf_end) {
-        uint8_t d = *buf++;
-        esc->fib_acc += esc->table[d & 0x0f];
-        *out_data++ = esc->fib_acc;
-        esc->fib_acc += esc->table[d >> 4];
-        *out_data++ = esc->fib_acc;
-    }
-
-    return consumed;
-}
-
-/** initialize 8svx decoder */
-static av_cold int eightsvx_decode_init(AVCodecContext *avctx)
-{
-    EightSvxContext *esc = avctx->priv_data;
-
-    switch(avctx->codec->id) {
-        case CODEC_ID_8SVX_FIB:
-          esc->table = fibonacci;
-          break;
-        case CODEC_ID_8SVX_EXP:
-          esc->table = exponential;
-          break;
-        default:
-          return -1;
-    }
-    avctx->sample_fmt = SAMPLE_FMT_S16;
-    return 0;
-}
-
-AVCodec eightsvx_fib_decoder = {
-  .name           = "8svx_fib",
-  .type           = CODEC_TYPE_AUDIO,
-  .id             = CODEC_ID_8SVX_FIB,
-  .priv_data_size = sizeof (EightSvxContext),
-  .init           = eightsvx_decode_init,
-  .decode         = eightsvx_decode_frame,
-  .long_name      = NULL_IF_CONFIG_SMALL("8SVX fibonacci"),
-};
-
-AVCodec eightsvx_exp_decoder = {
-  .name           = "8svx_exp",
-  .type           = CODEC_TYPE_AUDIO,
-  .id             = CODEC_ID_8SVX_EXP,
-  .priv_data_size = sizeof (EightSvxContext),
-  .init           = eightsvx_decode_init,
-  .decode         = eightsvx_decode_frame,
-  .long_name      = NULL_IF_CONFIG_SMALL("8SVX exponential"),
-};

libavcodec/Makefile

@@ -1,538 +0,0 @@
-include $(SUBDIR)../config.mak
-
-NAME = avcodec
-FFLIBS = avutil
-
-HEADERS = avcodec.h opt.h vdpau.h xvmc.h
-
-OBJS = allcodecs.o                                                      \
-       audioconvert.o                                                   \
-       bitstream.o                                                      \
-       bitstream_filter.o                                               \
-       dsputil.o                                                        \
-       eval.o                                                           \
-       faanidct.o                                                       \
-       imgconvert.o                                                     \
-       jrevdct.o                                                        \
-       opt.o                                                            \
-       options.o                                                        \
-       parser.o                                                         \
-       raw.o                                                            \
-       resample.o                                                       \
-       resample2.o                                                      \
-       simple_idct.o                                                    \
-       utils.o                                                          \
-
-# parts needed for many different codecs
-OBJS-$(CONFIG_AANDCT)                  += aandcttab.o
-OBJS-$(CONFIG_ENCODERS)                += faandct.o jfdctfst.o jfdctint.o
-OBJS-$(CONFIG_FFT)                     += fft.o
-OBJS-$(CONFIG_GOLOMB)                  += golomb.o
-OBJS-$(CONFIG_MDCT)                    += mdct.o
-OBJS-$(CONFIG_OLDSCALER)               += imgresample.o
-OBJS-$(CONFIG_RDFT)                    += rdft.o
-
-# decoders/encoders
-OBJS-$(CONFIG_AAC_DECODER)             += aac.o aactab.o mpeg4audio.o aac_parser.o aac_ac3_parser.o
-OBJS-$(CONFIG_AASC_DECODER)            += aasc.o msrledec.o
-OBJS-$(CONFIG_AC3_DECODER)             += eac3dec.o ac3dec.o ac3tab.o ac3dec_data.o ac3.o
-OBJS-$(CONFIG_AC3_ENCODER)             += ac3enc.o ac3tab.o ac3.o
-OBJS-$(CONFIG_ALAC_DECODER)            += alac.o
-OBJS-$(CONFIG_ALAC_ENCODER)            += alacenc.o lpc.o
-OBJS-$(CONFIG_AMV_DECODER)             += sp5xdec.o mjpegdec.o mjpeg.o
-OBJS-$(CONFIG_APE_DECODER)             += apedec.o
-OBJS-$(CONFIG_ASV1_DECODER)            += asv1.o mpeg12data.o
-OBJS-$(CONFIG_ASV1_ENCODER)            += asv1.o mpeg12data.o
-OBJS-$(CONFIG_ASV2_DECODER)            += asv1.o mpeg12data.o
-OBJS-$(CONFIG_ASV2_ENCODER)            += asv1.o mpeg12data.o
-OBJS-$(CONFIG_ATRAC3_DECODER)          += atrac3.o
-OBJS-$(CONFIG_AVS_DECODER)             += avs.o
-OBJS-$(CONFIG_BETHSOFTVID_DECODER)     += bethsoftvideo.o
-OBJS-$(CONFIG_BFI_DECODER)             += bfi.o
-OBJS-$(CONFIG_BMP_DECODER)             += bmp.o msrledec.o
-OBJS-$(CONFIG_BMP_ENCODER)             += bmpenc.o
-OBJS-$(CONFIG_C93_DECODER)             += c93.o
-OBJS-$(CONFIG_CAVS_DECODER)            += cavs.o cavsdec.o cavsdsp.o mpeg12data.o mpegvideo.o
-OBJS-$(CONFIG_CINEPAK_DECODER)         += cinepak.o
-OBJS-$(CONFIG_CLJR_DECODER)            += cljr.o
-OBJS-$(CONFIG_CLJR_ENCODER)            += cljr.o
-OBJS-$(CONFIG_COOK_DECODER)            += cook.o
-OBJS-$(CONFIG_CSCD_DECODER)            += cscd.o
-OBJS-$(CONFIG_CYUV_DECODER)            += cyuv.o
-OBJS-$(CONFIG_DCA_DECODER)             += dca.o
-OBJS-$(CONFIG_DNXHD_DECODER)           += dnxhddec.o dnxhddata.o
-OBJS-$(CONFIG_DNXHD_ENCODER)           += dnxhdenc.o dnxhddata.o mpegvideo_enc.o motion_est.o ratecontrol.o mpeg12data.o mpegvideo.o
-OBJS-$(CONFIG_DSICINAUDIO_DECODER)     += dsicinav.o
-OBJS-$(CONFIG_DSICINVIDEO_DECODER)     += dsicinav.o
-OBJS-$(CONFIG_DVBSUB_DECODER)          += dvbsubdec.o
-OBJS-$(CONFIG_DVBSUB_ENCODER)          += dvbsub.o
-OBJS-$(CONFIG_DVDSUB_DECODER)          += dvdsubdec.o
-OBJS-$(CONFIG_DVDSUB_ENCODER)          += dvdsubenc.o
-OBJS-$(CONFIG_DVVIDEO_DECODER)         += dv.o
-OBJS-$(CONFIG_DVVIDEO_ENCODER)         += dv.o
-OBJS-$(CONFIG_DXA_DECODER)             += dxa.o
-OBJS-$(CONFIG_EAC3_DECODER)            += eac3dec.o ac3dec.o ac3tab.o ac3dec_data.o ac3.o
-OBJS-$(CONFIG_EACMV_DECODER)           += eacmv.o
-OBJS-$(CONFIG_EATGQ_DECODER)           += eatgq.o eaidct.o
-OBJS-$(CONFIG_EATGV_DECODER)           += eatgv.o
-OBJS-$(CONFIG_EATQI_DECODER)           += eatqi.o eaidct.o mpeg12.o mpeg12data.o mpegvideo.o error_resilience.o
-OBJS-$(CONFIG_EIGHTBPS_DECODER)        += 8bps.o
-OBJS-$(CONFIG_EIGHTSVX_EXP_DECODER)    += 8svx.o
-OBJS-$(CONFIG_EIGHTSVX_FIB_DECODER)    += 8svx.o
-OBJS-$(CONFIG_ESCAPE124_DECODER)       += escape124.o
-OBJS-$(CONFIG_FFV1_DECODER)            += ffv1.o rangecoder.o
-OBJS-$(CONFIG_FFV1_ENCODER)            += ffv1.o rangecoder.o
-OBJS-$(CONFIG_FFVHUFF_DECODER)         += huffyuv.o
-OBJS-$(CONFIG_FFVHUFF_ENCODER)         += huffyuv.o
-OBJS-$(CONFIG_FLAC_DECODER)            += flacdec.o
-OBJS-$(CONFIG_FLAC_ENCODER)            += flacenc.o lpc.o
-OBJS-$(CONFIG_FLASHSV_DECODER)         += flashsv.o
-OBJS-$(CONFIG_FLASHSV_ENCODER)         += flashsvenc.o
-OBJS-$(CONFIG_FLIC_DECODER)            += flicvideo.o
-OBJS-$(CONFIG_FLV_DECODER)             += h263dec.o h263.o mpeg12data.o mpegvideo.o error_resilience.o
-OBJS-$(CONFIG_FLV_ENCODER)             += mpegvideo_enc.o motion_est.o ratecontrol.o h263.o mpeg12data.o mpegvideo.o error_resilience.o
-OBJS-$(CONFIG_FOURXM_DECODER)          += 4xm.o
-OBJS-$(CONFIG_FRAPS_DECODER)           += fraps.o huffman.o
-OBJS-$(CONFIG_GIF_DECODER)             += gifdec.o lzw.o
-OBJS-$(CONFIG_GIF_ENCODER)             += gif.o
-OBJS-$(CONFIG_H261_DECODER)            += h261dec.o h261.o mpegvideo.o error_resilience.o
-OBJS-$(CONFIG_H261_ENCODER)            += h261enc.o h261.o mpegvideo_enc.o motion_est.o ratecontrol.o mpeg12data.o mpegvideo.o
-OBJS-$(CONFIG_H263_DECODER)            += h263dec.o h263.o h263_parser.o mpeg12data.o mpegvideo.o error_resilience.o
-OBJS-$(CONFIG_H263I_DECODER)           += h263dec.o h263.o mpeg12data.o mpegvideo.o error_resilience.o
-OBJS-$(CONFIG_H263_ENCODER)            += mpegvideo_enc.o motion_est.o ratecontrol.o h263.o mpeg12data.o mpegvideo.o error_resilience.o
-OBJS-$(CONFIG_H263P_ENCODER)           += mpegvideo_enc.o motion_est.o ratecontrol.o h263.o mpeg12data.o mpegvideo.o error_resilience.o
-OBJS-$(CONFIG_H264_DECODER)            += h264.o h264idct.o h264pred.o h264_parser.o cabac.o mpegvideo.o error_resilience.o
-OBJS-$(CONFIG_H264_ENCODER)            += h264enc.o h264dspenc.o
-OBJS-$(CONFIG_H264_VDPAU_DECODER)      += vdpau.o h264.o h264idct.o h264pred.o h264_parser.o cabac.o mpegvideo.o error_resilience.o
-OBJS-$(CONFIG_HUFFYUV_DECODER)         += huffyuv.o
-OBJS-$(CONFIG_HUFFYUV_ENCODER)         += huffyuv.o
-OBJS-$(CONFIG_IDCIN_DECODER)           += idcinvideo.o
-OBJS-$(CONFIG_IMC_DECODER)             += imc.o
-OBJS-$(CONFIG_INDEO2_DECODER)          += indeo2.o
-OBJS-$(CONFIG_INDEO3_DECODER)          += indeo3.o
-OBJS-$(CONFIG_INTERPLAY_DPCM_DECODER)  += dpcm.o
-OBJS-$(CONFIG_INTERPLAY_VIDEO_DECODER) += interplayvideo.o
-OBJS-$(CONFIG_JPEGLS_DECODER)          += jpeglsdec.o jpegls.o mjpegdec.o mjpeg.o
-OBJS-$(CONFIG_JPEGLS_ENCODER)          += jpeglsenc.o jpegls.o
-OBJS-$(CONFIG_KMVC_DECODER)            += kmvc.o
-OBJS-$(CONFIG_LJPEG_ENCODER)           += ljpegenc.o mjpegenc.o mjpeg.o mpegvideo_enc.o motion_est.o ratecontrol.o mpeg12data.o mpegvideo.o
-OBJS-$(CONFIG_LOCO_DECODER)            += loco.o
-OBJS-$(CONFIG_MACE3_DECODER)           += mace.o
-OBJS-$(CONFIG_MACE6_DECODER)           += mace.o
-OBJS-$(CONFIG_MDEC_DECODER)            += mdec.o mpeg12.o mpeg12data.o mpegvideo.o error_resilience.o
-OBJS-$(CONFIG_MIMIC_DECODER)           += mimic.o
-OBJS-$(CONFIG_MJPEG_DECODER)           += mjpegdec.o mjpeg.o
-OBJS-$(CONFIG_MJPEG_ENCODER)           += mjpegenc.o mjpeg.o mpegvideo_enc.o motion_est.o ratecontrol.o mpeg12data.o mpegvideo.o
-OBJS-$(CONFIG_MJPEGB_DECODER)          += mjpegbdec.o mjpegdec.o mjpeg.o
-OBJS-$(CONFIG_MLP_DECODER)             += mlpdec.o mlp_parser.o mlp.o
-OBJS-$(CONFIG_MMVIDEO_DECODER)         += mmvideo.o
-OBJS-$(CONFIG_MOTIONPIXELS_DECODER)    += motionpixels.o
-OBJS-$(CONFIG_MP1_DECODER)             += mpegaudiodec.o mpegaudiodecheader.o mpegaudio.o mpegaudiodata.o
-OBJS-$(CONFIG_MP2_DECODER)             += mpegaudiodec.o mpegaudiodecheader.o mpegaudio.o mpegaudiodata.o
-OBJS-$(CONFIG_MP2_ENCODER)             += mpegaudioenc.o mpegaudio.o mpegaudiodata.o
-OBJS-$(CONFIG_MP3ADU_DECODER)          += mpegaudiodec.o mpegaudiodecheader.o mpegaudio.o mpegaudiodata.o
-OBJS-$(CONFIG_MP3ON4_DECODER)          += mpegaudiodec.o mpegaudiodecheader.o mpegaudio.o mpegaudiodata.o mpeg4audio.o
-OBJS-$(CONFIG_MP3_DECODER)             += mpegaudiodec.o mpegaudiodecheader.o mpegaudio.o mpegaudiodata.o
-OBJS-$(CONFIG_MPC7_DECODER)            += mpc7.o mpc.o mpegaudiodec.o mpegaudiodecheader.o mpegaudio.o mpegaudiodata.o
-OBJS-$(CONFIG_MPC8_DECODER)            += mpc8.o mpc.o mpegaudiodec.o mpegaudiodecheader.o mpegaudio.o mpegaudiodata.o
-OBJS-$(CONFIG_MPEG_VDPAU_DECODER)      += vdpau.o mpeg12.o mpeg12data.o mpegvideo.o error_resilience.o
-OBJS-$(CONFIG_MPEG1_VDPAU_DECODER)     += vdpau.o mpeg12.o mpeg12data.o mpegvideo.o error_resilience.o
-OBJS-$(CONFIG_MPEG_XVMC_DECODER)       += mpegvideo_xvmc.o mpeg12.o mpeg12data.o mpegvideo.o error_resilience.o
-OBJS-$(CONFIG_MPEGVIDEO_DECODER)       += mpeg12.o mpeg12data.o mpegvideo.o error_resilience.o
-OBJS-$(CONFIG_MPEG1VIDEO_DECODER)      += mpeg12.o mpeg12data.o mpegvideo.o error_resilience.o
-OBJS-$(CONFIG_MPEG1VIDEO_ENCODER)      += mpeg12enc.o mpeg12data.o mpegvideo_enc.o motion_est.o ratecontrol.o mpeg12.o mpeg12data.o mpegvideo.o error_resilience.o
-OBJS-$(CONFIG_MPEG2VIDEO_DECODER)      += mpeg12.o mpeg12data.o mpegvideo.o error_resilience.o
-OBJS-$(CONFIG_MPEG2VIDEO_ENCODER)      += mpeg12enc.o mpeg12data.o mpegvideo_enc.o motion_est.o ratecontrol.o mpeg12.o mpeg12data.o mpegvideo.o error_resilience.o
-OBJS-$(CONFIG_MPEG4_DECODER)           += h263dec.o h263.o mpeg4video_parser.o mpeg12data.o mpegvideo.o error_resilience.o
-OBJS-$(CONFIG_MPEG4_ENCODER)           += mpegvideo_enc.o motion_est.o ratecontrol.o h263.o mpeg12data.o mpegvideo.o error_resilience.o
-OBJS-$(CONFIG_MSMPEG4V1_DECODER)       += msmpeg4.o msmpeg4data.o h263dec.o h263.o mpeg12data.o mpegvideo.o error_resilience.o
-OBJS-$(CONFIG_MSMPEG4V1_ENCODER)       += msmpeg4.o msmpeg4data.o mpegvideo_enc.o motion_est.o ratecontrol.o h263.o mpeg12data.o mpegvideo.o error_resilience.o
-OBJS-$(CONFIG_MSMPEG4V2_DECODER)       += msmpeg4.o msmpeg4data.o h263dec.o h263.o mpeg12data.o mpegvideo.o error_resilience.o
-OBJS-$(CONFIG_MSMPEG4V2_ENCODER)       += msmpeg4.o msmpeg4data.o mpegvideo_enc.o motion_est.o ratecontrol.o h263.o mpeg12data.o mpegvideo.o error_resilience.o
-OBJS-$(CONFIG_MSMPEG4V3_DECODER)       += msmpeg4.o msmpeg4data.o h263dec.o h263.o mpeg12data.o mpegvideo.o error_resilience.o
-OBJS-$(CONFIG_MSMPEG4V3_ENCODER)       += msmpeg4.o msmpeg4data.o mpegvideo_enc.o motion_est.o ratecontrol.o h263.o mpeg12data.o mpegvideo.o error_resilience.o
-OBJS-$(CONFIG_MSRLE_DECODER)           += msrle.o msrledec.o
-OBJS-$(CONFIG_MSVIDEO1_DECODER)        += msvideo1.o
-OBJS-$(CONFIG_MSZH_DECODER)            += lcldec.o
-OBJS-$(CONFIG_NELLYMOSER_DECODER)      += nellymoserdec.o nellymoser.o
-OBJS-$(CONFIG_NELLYMOSER_ENCODER)      += nellymoserenc.o nellymoser.o
-OBJS-$(CONFIG_NUV_DECODER)             += nuv.o rtjpeg.o
-OBJS-$(CONFIG_PAM_ENCODER)             += pnmenc.o pnm.o
-OBJS-$(CONFIG_PBM_ENCODER)             += pnmenc.o pnm.o
-OBJS-$(CONFIG_PCX_DECODER)             += pcx.o
-OBJS-$(CONFIG_PGM_ENCODER)             += pnmenc.o pnm.o
-OBJS-$(CONFIG_PGMYUV_ENCODER)          += pnmenc.o pnm.o
-OBJS-$(CONFIG_PNG_DECODER)             += png.o pngdec.o
-OBJS-$(CONFIG_PNG_ENCODER)             += png.o pngenc.o
-OBJS-$(CONFIG_PPM_ENCODER)             += pnmenc.o pnm.o
-OBJS-$(CONFIG_PTX_DECODER)             += ptx.o
-OBJS-$(CONFIG_QCELP_DECODER)           += qcelpdec.o qcelp_lsp.o celp_math.o celp_filters.o
-OBJS-$(CONFIG_QDM2_DECODER)            += qdm2.o mpegaudiodec.o mpegaudiodecheader.o mpegaudio.o mpegaudiodata.o
-OBJS-$(CONFIG_QDRAW_DECODER)           += qdrw.o
-OBJS-$(CONFIG_QPEG_DECODER)            += qpeg.o
-OBJS-$(CONFIG_QTRLE_DECODER)           += qtrle.o
-OBJS-$(CONFIG_QTRLE_ENCODER)           += qtrleenc.o
-OBJS-$(CONFIG_RA_144_DECODER)          += ra144.o celp_filters.o
-OBJS-$(CONFIG_RA_288_DECODER)          += ra288.o celp_math.o celp_filters.o
-OBJS-$(CONFIG_RAWVIDEO_DECODER)        += rawdec.o
-OBJS-$(CONFIG_RAWVIDEO_ENCODER)        += rawenc.o
-OBJS-$(CONFIG_RL2_DECODER)             += rl2.o
-OBJS-$(CONFIG_ROQ_DECODER)             += roqvideodec.o roqvideo.o
-OBJS-$(CONFIG_ROQ_ENCODER)             += roqvideoenc.o roqvideo.o elbg.o
-OBJS-$(CONFIG_ROQ_DPCM_DECODER)        += dpcm.o
-OBJS-$(CONFIG_ROQ_DPCM_ENCODER)        += roqaudioenc.o
-OBJS-$(CONFIG_RPZA_DECODER)            += rpza.o
-OBJS-$(CONFIG_RV10_DECODER)            += rv10.o h263.o mpeg12data.o mpegvideo.o error_resilience.o
-OBJS-$(CONFIG_RV10_ENCODER)            += rv10.o mpegvideo_enc.o motion_est.o ratecontrol.o h263.o mpeg12data.o mpegvideo.o error_resilience.o
-OBJS-$(CONFIG_RV20_DECODER)            += rv10.o h263.o mpeg12data.o mpegvideo.o error_resilience.o
-OBJS-$(CONFIG_RV20_ENCODER)            += rv10.o mpegvideo_enc.o motion_est.o ratecontrol.o h263.o mpeg12data.o mpegvideo.o error_resilience.o
-OBJS-$(CONFIG_RV30_DECODER)            += rv30.o rv34.o h264pred.o rv30dsp.o mpegvideo.o error_resilience.o
-OBJS-$(CONFIG_RV40_DECODER)            += rv40.o rv34.o h264pred.o rv40dsp.o mpegvideo.o error_resilience.o
-OBJS-$(CONFIG_SGI_DECODER)             += sgidec.o
-OBJS-$(CONFIG_SGI_ENCODER)             += sgienc.o rle.o
-OBJS-$(CONFIG_SHORTEN_DECODER)         += shorten.o
-OBJS-$(CONFIG_SMACKAUD_DECODER)        += smacker.o
-OBJS-$(CONFIG_SMACKER_DECODER)         += smacker.o
-OBJS-$(CONFIG_SMC_DECODER)             += smc.o
-OBJS-$(CONFIG_SNOW_DECODER)            += snow.o rangecoder.o
-OBJS-$(CONFIG_SNOW_ENCODER)            += snow.o rangecoder.o motion_est.o ratecontrol.o h263.o mpeg12data.o mpegvideo.o error_resilience.o
-OBJS-$(CONFIG_SOL_DPCM_DECODER)        += dpcm.o
-OBJS-$(CONFIG_SONIC_DECODER)           += sonic.o
-OBJS-$(CONFIG_SONIC_ENCODER)           += sonic.o
-OBJS-$(CONFIG_SONIC_LS_ENCODER)        += sonic.o
-OBJS-$(CONFIG_SP5X_DECODER)            += sp5xdec.o mjpegdec.o mjpeg.o
-OBJS-$(CONFIG_SUNRAST_DECODER)         += sunrast.o
-OBJS-$(CONFIG_SVQ1_DECODER)            += svq1dec.o svq1.o h263.o mpeg12data.o mpegvideo.o error_resilience.o
-OBJS-$(CONFIG_SVQ1_ENCODER)            += svq1enc.o svq1.o motion_est.o h263.o mpeg12data.o mpegvideo.o error_resilience.o
-OBJS-$(CONFIG_SVQ3_DECODER)            += h264.o h264idct.o h264pred.o h264_parser.o cabac.o mpegvideo.o error_resilience.o svq1dec.o svq1.o h263.o
-OBJS-$(CONFIG_TARGA_DECODER)           += targa.o
-OBJS-$(CONFIG_TARGA_ENCODER)           += targaenc.o rle.o
-OBJS-$(CONFIG_THEORA_DECODER)          += vp3.o xiph.o vp3dsp.o
-OBJS-$(CONFIG_THP_DECODER)             += mjpegdec.o mjpeg.o
-OBJS-$(CONFIG_TIERTEXSEQVIDEO_DECODER) += tiertexseqv.o
-OBJS-$(CONFIG_TIFF_DECODER)            += tiff.o lzw.o faxcompr.o
-OBJS-$(CONFIG_TIFF_ENCODER)            += tiffenc.o rle.o lzwenc.o
-OBJS-$(CONFIG_TRUEMOTION1_DECODER)     += truemotion1.o
-OBJS-$(CONFIG_TRUEMOTION2_DECODER)     += truemotion2.o
-OBJS-$(CONFIG_TRUESPEECH_DECODER)      += truespeech.o
-OBJS-$(CONFIG_TSCC_DECODER)            += tscc.o msrledec.o
-OBJS-$(CONFIG_TTA_DECODER)             += tta.o
-OBJS-$(CONFIG_TXD_DECODER)             += txd.o s3tc.o
-OBJS-$(CONFIG_ULTI_DECODER)            += ulti.o
-OBJS-$(CONFIG_VB_DECODER)              += vb.o
-OBJS-$(CONFIG_VC1_DECODER)             += vc1.o vc1data.o vc1dsp.o msmpeg4data.o h263dec.o h263.o intrax8.o intrax8dsp.o error_resilience.o mpegvideo.o msmpeg4.o
-OBJS-$(CONFIG_VC1_VDPAU_DECODER)       += vdpau.o vc1.o vc1data.o vc1dsp.o msmpeg4data.o h263dec.o h263.o intrax8.o intrax8dsp.o error_resilience.o mpegvideo.o msmpeg4.o
-OBJS-$(CONFIG_VCR1_DECODER)            += vcr1.o
-OBJS-$(CONFIG_VCR1_ENCODER)            += vcr1.o
-OBJS-$(CONFIG_VMDAUDIO_DECODER)        += vmdav.o
-OBJS-$(CONFIG_VMDVIDEO_DECODER)        += vmdav.o
-OBJS-$(CONFIG_VMNC_DECODER)            += vmnc.o
-OBJS-$(CONFIG_VORBIS_DECODER)          += vorbis_dec.o vorbis.o vorbis_data.o xiph.o
-OBJS-$(CONFIG_VORBIS_ENCODER)          += vorbis_enc.o vorbis.o vorbis_data.o
-OBJS-$(CONFIG_VP3_DECODER)             += vp3.o vp3dsp.o
-OBJS-$(CONFIG_VP5_DECODER)             += vp5.o vp56.o vp56data.o vp3dsp.o
-OBJS-$(CONFIG_VP6_DECODER)             += vp6.o vp56.o vp56data.o vp3dsp.o vp6dsp.o huffman.o
-OBJS-$(CONFIG_VP6A_DECODER)            += vp6.o vp56.o vp56data.o vp3dsp.o vp6dsp.o huffman.o
-OBJS-$(CONFIG_VP6F_DECODER)            += vp6.o vp56.o vp56data.o vp3dsp.o vp6dsp.o huffman.o
-OBJS-$(CONFIG_VQA_DECODER)             += vqavideo.o
-OBJS-$(CONFIG_WAVPACK_DECODER)         += wavpack.o
-OBJS-$(CONFIG_WMAV1_DECODER)           += wmadec.o wma.o
-OBJS-$(CONFIG_WMAV1_ENCODER)           += wmaenc.o wma.o
-OBJS-$(CONFIG_WMAV2_DECODER)           += wmadec.o wma.o
-OBJS-$(CONFIG_WMAV2_ENCODER)           += wmaenc.o wma.o
-OBJS-$(CONFIG_WMV1_DECODER)            += h263dec.o h263.o mpeg12data.o mpegvideo.o error_resilience.o
-OBJS-$(CONFIG_WMV1_ENCODER)            += mpegvideo_enc.o motion_est.o ratecontrol.o h263.o mpeg12data.o mpegvideo.o error_resilience.o
-OBJS-$(CONFIG_WMV2_DECODER)            += wmv2dec.o wmv2.o msmpeg4.o msmpeg4data.o h263dec.o h263.o intrax8.o intrax8dsp.o mpeg12data.o mpegvideo.o error_resilience.o
-OBJS-$(CONFIG_WMV2_ENCODER)            += wmv2enc.o wmv2.o msmpeg4.o msmpeg4data.o mpegvideo_enc.o motion_est.o ratecontrol.o h263.o mpeg12data.o mpegvideo.o error_resilience.o
-OBJS-$(CONFIG_WMV3_DECODER)            += vc1.o vc1data.o vc1dsp.o msmpeg4data.o h263dec.o h263.o intrax8.o intrax8dsp.o error_resilience.o mpegvideo.o msmpeg4.o
-OBJS-$(CONFIG_WMV3_VDPAU_DECODER)      += vdpau.o vc1.o vc1data.o vc1dsp.o msmpeg4data.o h263dec.o h263.o intrax8.o intrax8dsp.o error_resilience.o mpegvideo.o msmpeg4.o
-OBJS-$(CONFIG_WNV1_DECODER)            += wnv1.o
-OBJS-$(CONFIG_WS_SND1_DECODER)         += ws-snd1.o
-OBJS-$(CONFIG_XAN_DPCM_DECODER)        += dpcm.o
-OBJS-$(CONFIG_XAN_WC3_DECODER)         += xan.o
-OBJS-$(CONFIG_XAN_WC4_DECODER)         += xan.o
-OBJS-$(CONFIG_XL_DECODER)              += xl.o
-OBJS-$(CONFIG_XSUB_DECODER)            += xsubdec.o
-OBJS-$(CONFIG_ZLIB_DECODER)            += lcldec.o
-OBJS-$(CONFIG_ZLIB_ENCODER)            += lclenc.o
-OBJS-$(CONFIG_ZMBV_DECODER)            += zmbv.o
-OBJS-$(CONFIG_ZMBV_ENCODER)            += zmbvenc.o
-
-# (AD)PCM decoders/encoders
-OBJS-$(CONFIG_PCM_ALAW_DECODER)           += pcm.o
-OBJS-$(CONFIG_PCM_ALAW_ENCODER)           += pcm.o
-OBJS-$(CONFIG_PCM_DVD_DECODER)            += pcm.o
-OBJS-$(CONFIG_PCM_DVD_ENCODER)            += pcm.o
-OBJS-$(CONFIG_PCM_F32BE_DECODER)          += pcm.o
-OBJS-$(CONFIG_PCM_F32BE_ENCODER)          += pcm.o
-OBJS-$(CONFIG_PCM_F32LE_DECODER)          += pcm.o
-OBJS-$(CONFIG_PCM_F32LE_ENCODER)          += pcm.o
-OBJS-$(CONFIG_PCM_F64BE_DECODER)          += pcm.o
-OBJS-$(CONFIG_PCM_F64BE_ENCODER)          += pcm.o
-OBJS-$(CONFIG_PCM_F64LE_DECODER)          += pcm.o
-OBJS-$(CONFIG_PCM_F64LE_ENCODER)          += pcm.o
-OBJS-$(CONFIG_PCM_MULAW_DECODER)          += pcm.o
-OBJS-$(CONFIG_PCM_MULAW_ENCODER)          += pcm.o
-OBJS-$(CONFIG_PCM_S8_DECODER)             += pcm.o
-OBJS-$(CONFIG_PCM_S8_ENCODER)             += pcm.o
-OBJS-$(CONFIG_PCM_S16BE_DECODER)          += pcm.o
-OBJS-$(CONFIG_PCM_S16BE_ENCODER)          += pcm.o
-OBJS-$(CONFIG_PCM_S16LE_DECODER)          += pcm.o
-OBJS-$(CONFIG_PCM_S16LE_ENCODER)          += pcm.o
-OBJS-$(CONFIG_PCM_S16LE_PLANAR_DECODER)   += pcm.o
-OBJS-$(CONFIG_PCM_S24BE_DECODER)          += pcm.o
-OBJS-$(CONFIG_PCM_S24BE_ENCODER)          += pcm.o
-OBJS-$(CONFIG_PCM_S24DAUD_DECODER)        += pcm.o
-OBJS-$(CONFIG_PCM_S24DAUD_ENCODER)        += pcm.o
-OBJS-$(CONFIG_PCM_S24LE_DECODER)          += pcm.o
-OBJS-$(CONFIG_PCM_S24LE_ENCODER)          += pcm.o
-OBJS-$(CONFIG_PCM_S32BE_DECODER)          += pcm.o
-OBJS-$(CONFIG_PCM_S32BE_ENCODER)          += pcm.o
-OBJS-$(CONFIG_PCM_S32LE_DECODER)          += pcm.o
-OBJS-$(CONFIG_PCM_S32LE_ENCODER)          += pcm.o
-OBJS-$(CONFIG_PCM_U8_DECODER)             += pcm.o
-OBJS-$(CONFIG_PCM_U8_ENCODER)             += pcm.o
-OBJS-$(CONFIG_PCM_U16BE_DECODER)          += pcm.o
-OBJS-$(CONFIG_PCM_U16BE_ENCODER)          += pcm.o
-OBJS-$(CONFIG_PCM_U16LE_DECODER)          += pcm.o
-OBJS-$(CONFIG_PCM_U16LE_ENCODER)          += pcm.o
-OBJS-$(CONFIG_PCM_U24BE_DECODER)          += pcm.o
-OBJS-$(CONFIG_PCM_U24BE_ENCODER)          += pcm.o
-OBJS-$(CONFIG_PCM_U24LE_DECODER)          += pcm.o
-OBJS-$(CONFIG_PCM_U24LE_ENCODER)          += pcm.o
-OBJS-$(CONFIG_PCM_U32BE_DECODER)          += pcm.o
-OBJS-$(CONFIG_PCM_U32BE_ENCODER)          += pcm.o
-OBJS-$(CONFIG_PCM_U32LE_DECODER)          += pcm.o
-OBJS-$(CONFIG_PCM_U32LE_ENCODER)          += pcm.o
-OBJS-$(CONFIG_PCM_ZORK_DECODER)           += pcm.o
-OBJS-$(CONFIG_PCM_ZORK_ENCODER)           += pcm.o
-
-OBJS-$(CONFIG_ADPCM_4XM_DECODER)          += adpcm.o
-OBJS-$(CONFIG_ADPCM_ADX_DECODER)          += adxdec.o
-OBJS-$(CONFIG_ADPCM_ADX_ENCODER)          += adxenc.o
-OBJS-$(CONFIG_ADPCM_CT_DECODER)           += adpcm.o
-OBJS-$(CONFIG_ADPCM_EA_DECODER)           += adpcm.o
-OBJS-$(CONFIG_ADPCM_EA_MAXIS_XA_DECODER)  += adpcm.o
-OBJS-$(CONFIG_ADPCM_EA_R1_DECODER)        += adpcm.o
-OBJS-$(CONFIG_ADPCM_EA_R2_DECODER)        += adpcm.o
-OBJS-$(CONFIG_ADPCM_EA_R3_DECODER)        += adpcm.o
-OBJS-$(CONFIG_ADPCM_EA_XAS_DECODER)       += adpcm.o
-OBJS-$(CONFIG_ADPCM_G726_DECODER)         += g726.o
-OBJS-$(CONFIG_ADPCM_G726_ENCODER)         += g726.o
-OBJS-$(CONFIG_ADPCM_IMA_AMV_DECODER)      += adpcm.o
-OBJS-$(CONFIG_ADPCM_IMA_DK3_DECODER)      += adpcm.o
-OBJS-$(CONFIG_ADPCM_IMA_DK4_DECODER)      += adpcm.o
-OBJS-$(CONFIG_ADPCM_IMA_EA_EACS_DECODER)  += adpcm.o
-OBJS-$(CONFIG_ADPCM_IMA_EA_SEAD_DECODER)  += adpcm.o
-OBJS-$(CONFIG_ADPCM_IMA_ISS_DECODER)      += adpcm.o
-OBJS-$(CONFIG_ADPCM_IMA_QT_DECODER)       += adpcm.o
-OBJS-$(CONFIG_ADPCM_IMA_QT_ENCODER)       += adpcm.o
-OBJS-$(CONFIG_ADPCM_IMA_SMJPEG_DECODER)   += adpcm.o
-OBJS-$(CONFIG_ADPCM_IMA_WAV_DECODER)      += adpcm.o
-OBJS-$(CONFIG_ADPCM_IMA_WAV_ENCODER)      += adpcm.o
-OBJS-$(CONFIG_ADPCM_IMA_WS_DECODER)       += adpcm.o
-OBJS-$(CONFIG_ADPCM_MS_DECODER)           += adpcm.o
-OBJS-$(CONFIG_ADPCM_MS_ENCODER)           += adpcm.o
-OBJS-$(CONFIG_ADPCM_SBPRO_2_DECODER)      += adpcm.o
-OBJS-$(CONFIG_ADPCM_SBPRO_3_DECODER)      += adpcm.o
-OBJS-$(CONFIG_ADPCM_SBPRO_4_DECODER)      += adpcm.o
-OBJS-$(CONFIG_ADPCM_SWF_DECODER)          += adpcm.o
-OBJS-$(CONFIG_ADPCM_SWF_ENCODER)          += adpcm.o
-OBJS-$(CONFIG_ADPCM_THP_DECODER)          += adpcm.o
-OBJS-$(CONFIG_ADPCM_XA_DECODER)           += adpcm.o
-OBJS-$(CONFIG_ADPCM_YAMAHA_DECODER)       += adpcm.o
-OBJS-$(CONFIG_ADPCM_YAMAHA_ENCODER)       += adpcm.o
-
-# libavformat dependencies
-OBJS-$(CONFIG_EAC3_DEMUXER)            += ac3_parser.o ac3tab.o aac_ac3_parser.o
-OBJS-$(CONFIG_FLAC_MUXER)              += flacdec.o
-OBJS-$(CONFIG_GXF_DEMUXER)             += mpeg12data.o
-OBJS-$(CONFIG_MATROSKA_AUDIO_MUXER)    += xiph.o mpeg4audio.o flacdec.o
-OBJS-$(CONFIG_MATROSKA_DEMUXER)        += mpeg4audio.o
-OBJS-$(CONFIG_MATROSKA_MUXER)          += xiph.o mpeg4audio.o flacdec.o
-OBJS-$(CONFIG_MOV_DEMUXER)             += mpeg4audio.o mpegaudiodata.o
-OBJS-$(CONFIG_MPEGTS_MUXER)            += mpegvideo.o
-OBJS-$(CONFIG_NUT_MUXER)               += mpegaudiodata.o
-OBJS-$(CONFIG_OGG_DEMUXER)             += flacdec.o
-OBJS-$(CONFIG_OGG_MUXER)               += xiph.o flacdec.o
-OBJS-$(CONFIG_RTP_MUXER)               += mpegvideo.o
-
-# external codec libraries
-OBJS-$(CONFIG_LIBAMR_NB)               += libamr.o
-OBJS-$(CONFIG_LIBAMR_WB)               += libamr.o
-OBJS-$(CONFIG_LIBDIRAC_DECODER)        += libdiracdec.o
-OBJS-$(CONFIG_LIBDIRAC_ENCODER)        += libdiracenc.o libdirac_libschro.o
-OBJS-$(CONFIG_LIBFAAC)                 += libfaac.o
-OBJS-$(CONFIG_LIBFAAD)                 += libfaad.o
-OBJS-$(CONFIG_LIBGSM)                  += libgsm.o
-OBJS-$(CONFIG_LIBMP3LAME)              += libmp3lame.o
-OBJS-$(CONFIG_LIBOPENCORE_AMRNB)       += libopencore-amr.o
-OBJS-$(CONFIG_LIBOPENCORE_AMRWB)       += libopencore-amr.o
-OBJS-$(CONFIG_LIBOPENJPEG)             += libopenjpeg.o
-OBJS-$(CONFIG_LIBSCHROEDINGER_DECODER) += libschroedingerdec.o libschroedinger.o libdirac_libschro.o
-OBJS-$(CONFIG_LIBSCHROEDINGER_ENCODER) += libschroedingerenc.o libschroedinger.o libdirac_libschro.o
-OBJS-$(CONFIG_LIBSPEEX)                += libspeexdec.o
-OBJS-$(CONFIG_LIBTHEORA)               += libtheoraenc.o
-OBJS-$(CONFIG_LIBVORBIS)               += libvorbis.o
-OBJS-$(CONFIG_LIBX264)                 += libx264.o
-OBJS-$(CONFIG_LIBXVID)                 += libxvidff.o libxvid_rc.o
-
-# parsers
-OBJS-$(CONFIG_AAC_PARSER)              += aac_parser.o aac_ac3_parser.o mpeg4audio.o
-OBJS-$(CONFIG_AC3_PARSER)              += ac3_parser.o ac3tab.o aac_ac3_parser.o
-OBJS-$(CONFIG_CAVSVIDEO_PARSER)        += cavs_parser.o
-OBJS-$(CONFIG_DCA_PARSER)              += dca_parser.o
-OBJS-$(CONFIG_DIRAC_PARSER)            += dirac_parser.o
-OBJS-$(CONFIG_DNXHD_PARSER)            += dnxhd_parser.o
-OBJS-$(CONFIG_DVBSUB_PARSER)           += dvbsub_parser.o
-OBJS-$(CONFIG_DVDSUB_PARSER)           += dvdsub_parser.o
-OBJS-$(CONFIG_H261_PARSER)             += h261_parser.o
-OBJS-$(CONFIG_H263_PARSER)             += h263_parser.o
-OBJS-$(CONFIG_H264_PARSER)             += h264_parser.o
-OBJS-$(CONFIG_MJPEG_PARSER)            += mjpeg_parser.o
-OBJS-$(CONFIG_MLP_PARSER)              += mlp_parser.o mlp.o
-OBJS-$(CONFIG_MPEG4VIDEO_PARSER)       += mpeg4video_parser.o h263.o mpeg12data.o mpegvideo.o error_resilience.o
-OBJS-$(CONFIG_MPEGAUDIO_PARSER)        += mpegaudio_parser.o mpegaudiodecheader.o mpegaudiodata.o
-OBJS-$(CONFIG_MPEGVIDEO_PARSER)        += mpegvideo_parser.o mpeg12.o mpeg12data.o mpegvideo.o error_resilience.o
-OBJS-$(CONFIG_PNM_PARSER)              += pnm_parser.o pnm.o
-OBJS-$(CONFIG_VC1_PARSER)              += vc1_parser.o
-OBJS-$(CONFIG_VP3_PARSER)              += vp3_parser.o
-
-# bitstream filters
-OBJS-$(CONFIG_DUMP_EXTRADATA_BSF)         += dump_extradata_bsf.o
-OBJS-$(CONFIG_H264_MP4TOANNEXB_BSF)       += h264_mp4toannexb_bsf.o
-OBJS-$(CONFIG_IMX_DUMP_HEADER_BSF)        += imx_dump_header_bsf.o
-OBJS-$(CONFIG_MJPEGA_DUMP_HEADER_BSF)     += mjpega_dump_header_bsf.o
-OBJS-$(CONFIG_MOV2TEXTSUB_BSF)            += movsub_bsf.o
-OBJS-$(CONFIG_MP3_HEADER_COMPRESS_BSF)    += mp3_header_compress_bsf.o
-OBJS-$(CONFIG_MP3_HEADER_DECOMPRESS_BSF)  += mp3_header_decompress_bsf.o mpegaudiodata.o
-OBJS-$(CONFIG_NOISE_BSF)                  += noise_bsf.o
-OBJS-$(CONFIG_REMOVE_EXTRADATA_BSF)       += remove_extradata_bsf.o
-OBJS-$(CONFIG_TEXT2MOVSUB_BSF)            += movsub_bsf.o
-
-# thread libraries
-OBJS-$(HAVE_BEOSTHREADS)               += beosthread.o
-OBJS-$(HAVE_OS2THREADS)                += os2thread.o
-OBJS-$(HAVE_PTHREADS)                  += pthread.o
-OBJS-$(HAVE_W32THREADS)                += w32thread.o
-
-# processor-specific code
-YASM-OBJS-FFT-$(HAVE_AMD3DNOW)         += x86/fft_3dn.o
-YASM-OBJS-FFT-$(HAVE_AMD3DNOWEXT)      += x86/fft_3dn2.o
-YASM-OBJS-FFT-$(HAVE_SSE)              += x86/fft_sse.o
-YASM-OBJS-$(CONFIG_FFT)                += x86/fft_mmx.o $(YASM-OBJS-FFT-yes)
-YASM-OBJS-$(CONFIG_GPL)                += x86/h264_deblock_sse2.o       \
-                                          x86/h264_idct_sse2.o          \
-
-MMX-OBJS-$(CONFIG_CAVS_DECODER)        += x86/cavsdsp_mmx.o
-MMX-OBJS-$(CONFIG_ENCODERS)            += x86/dsputilenc_mmx.o
-MMX-OBJS-$(CONFIG_FLAC_ENCODER)        += x86/flacdsp_mmx.o
-MMX-OBJS-$(CONFIG_GPL)                 += x86/idct_mmx.o
-MMX-OBJS-$(CONFIG_SNOW_DECODER)        += x86/snowdsp_mmx.o
-MMX-OBJS-$(CONFIG_THEORA_DECODER)      += x86/vp3dsp_mmx.o x86/vp3dsp_sse2.o
-MMX-OBJS-$(CONFIG_VC1_DECODER)         += x86/vc1dsp_mmx.o
-MMX-OBJS-$(CONFIG_VP3_DECODER)         += x86/vp3dsp_mmx.o x86/vp3dsp_sse2.o
-MMX-OBJS-$(CONFIG_VP5_DECODER)         += x86/vp3dsp_mmx.o x86/vp3dsp_sse2.o
-MMX-OBJS-$(CONFIG_VP6_DECODER)         += x86/vp3dsp_mmx.o x86/vp3dsp_sse2.o \
-                                          x86/vp6dsp_mmx.o x86/vp6dsp_sse2.o
-MMX-OBJS-$(CONFIG_VP6A_DECODER)        += x86/vp3dsp_mmx.o x86/vp3dsp_sse2.o \
-                                          x86/vp6dsp_mmx.o x86/vp6dsp_sse2.o
-MMX-OBJS-$(CONFIG_VP6F_DECODER)        += x86/vp3dsp_mmx.o x86/vp3dsp_sse2.o \
-                                          x86/vp6dsp_mmx.o x86/vp6dsp_sse2.o
-MMX-OBJS-$(CONFIG_WMV3_DECODER)        += x86/vc1dsp_mmx.o
-MMX-OBJS-$(HAVE_YASM)                  += x86/dsputil_yasm.o            \
-                                          $(YASM-OBJS-yes)
-
-OBJS-$(HAVE_MMX)                       += x86/cpuid.o                   \
-                                          x86/dnxhd_mmx.o               \
-                                          x86/dsputil_mmx.o             \
-                                          x86/fdct_mmx.o                \
-                                          x86/idct_mmx_xvid.o           \
-                                          x86/idct_sse2_xvid.o          \
-                                          x86/motion_est_mmx.o          \
-                                          x86/mpegvideo_mmx.o           \
-                                          x86/simple_idct_mmx.o         \
-                                          $(MMX-OBJS-yes)
-
-OBJS-$(ARCH_ALPHA)                     += alpha/dsputil_alpha.o         \
-                                          alpha/dsputil_alpha_asm.o     \
-                                          alpha/motion_est_alpha.o      \
-                                          alpha/motion_est_mvi_asm.o    \
-                                          alpha/mpegvideo_alpha.o       \
-                                          alpha/simple_idct_alpha.o     \
-
-OBJS-$(ARCH_ARM)                       += arm/dsputil_arm.o             \
-                                          arm/dsputil_arm_s.o           \
-                                          arm/jrevdct_arm.o             \
-                                          arm/mpegvideo_arm.o           \
-                                          arm/simple_idct_arm.o         \
-
-OBJS-$(HAVE_ARMV5TE)                   += arm/mpegvideo_armv5te.o       \
-                                          arm/mpegvideo_armv5te_s.o     \
-                                          arm/simple_idct_armv5te.o     \
-
-OBJS-$(HAVE_ARMV6)                     += arm/simple_idct_armv6.o       \
-
-OBJS-$(HAVE_ARMVFP)                    += arm/dsputil_vfp.o             \
-                                          arm/float_arm_vfp.o           \
-
-OBJS-$(HAVE_IWMMXT)                    += arm/dsputil_iwmmxt.o          \
-                                          arm/mpegvideo_iwmmxt.o        \
-
-OBJS-$(HAVE_NEON)                      += arm/dsputil_neon.o            \
-                                          arm/dsputil_neon_s.o          \
-                                          arm/h264dsp_neon.o            \
-                                          arm/h264idct_neon.o           \
-                                          arm/simple_idct_neon.o        \
-
-OBJS-$(ARCH_BFIN)                      += bfin/dsputil_bfin.o           \
-                                          bfin/fdct_bfin.o              \
-                                          bfin/idct_bfin.o              \
-                                          bfin/mpegvideo_bfin.o         \
-                                          bfin/pixels_bfin.o            \
-                                          bfin/vp3_bfin.o               \
-                                          bfin/vp3_idct_bfin.o          \
-
-OBJS-$(ARCH_PPC)                       += ppc/dsputil_ppc.o             \
-
-ALTIVEC-OBJS-$(CONFIG_H264_DECODER)    += ppc/h264_altivec.o
-ALTIVEC-OBJS-$(CONFIG_OLDSCALER)       += ppc/imgresample_altivec.o
-ALTIVEC-OBJS-$(CONFIG_SNOW_DECODER)    += ppc/snow_altivec.o
-ALTIVEC-OBJS-$(CONFIG_VC1_DECODER)     += ppc/vc1dsp_altivec.o
-ALTIVEC-OBJS-$(CONFIG_WMV3_DECODER)    += ppc/vc1dsp_altivec.o
-
-OBJS-$(HAVE_ALTIVEC)                   += ppc/check_altivec.o           \
-                                          ppc/dsputil_altivec.o         \
-                                          ppc/fdct_altivec.o            \
-                                          ppc/fft_altivec.o             \
-                                          ppc/float_altivec.o           \
-                                          ppc/gmc_altivec.o             \
-                                          ppc/idct_altivec.o            \
-                                          ppc/int_altivec.o             \
-                                          ppc/mpegvideo_altivec.o       \
-                                          $(ALTIVEC-OBJS-yes)
-
-OBJS-$(ARCH_SH4)                       += sh4/dsputil_align.o           \
-                                          sh4/dsputil_sh4.o             \
-                                          sh4/idct_sh4.o                \
-
-OBJS-$(CONFIG_MLIB)                    += mlib/dsputil_mlib.o           \
-
-OBJS-$(HAVE_MMI)                       += ps2/dsputil_mmi.o             \
-                                          ps2/idct_mmi.o                \
-                                          ps2/mpegvideo_mmi.o           \
-
-OBJS-$(HAVE_VIS)                       += sparc/dsputil_vis.o           \
-                                          sparc/simple_idct_vis.o       \
-
-
-TESTS = $(addsuffix -test$(EXESUF), cabac dct eval fft h264 rangecoder snow)
-TESTS-$(CONFIG_OLDSCALER) += imgresample-test$(EXESUF)
-TESTS-$(ARCH_X86) += x86/cpuid-test$(EXESUF) motion-test$(EXESUF)
-
-CLEANFILES = apiexample$(EXESUF)
-DIRS = alpha arm bfin mlib ppc ps2 sh4 sparc x86
-
-include $(SUBDIR)../subdir.mak
-
-$(SUBDIR)dct-test$(EXESUF): $(SUBDIR)dctref.o $(SUBDIR)aandcttab.o

libavcodec/aac.h

@@ -1,297 +0,0 @@
-/*
- * AAC definitions and structures
- * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
- * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/**
- * @file libavcodec/aac.h
- * AAC definitions and structures
- * @author Oded Shimon  ( ods15 ods15 dyndns org )
- * @author Maxim Gavrilov ( maxim.gavrilov gmail com )
- */
-
-#ifndef AVCODEC_AAC_H
-#define AVCODEC_AAC_H
-
-#include "libavutil/internal.h"
-#include "avcodec.h"
-#include "dsputil.h"
-#include "mpeg4audio.h"
-
-#include <stdint.h>
-
-#define AAC_INIT_VLC_STATIC(num, size) \
-    INIT_VLC_STATIC(&vlc_spectral[num], 6, ff_aac_spectral_sizes[num], \
-         ff_aac_spectral_bits[num], sizeof( ff_aac_spectral_bits[num][0]), sizeof( ff_aac_spectral_bits[num][0]), \
-        ff_aac_spectral_codes[num], sizeof(ff_aac_spectral_codes[num][0]), sizeof(ff_aac_spectral_codes[num][0]), \
-        size);
-
-#define MAX_CHANNELS 64
-#define MAX_ELEM_ID 16
-
-#define TNS_MAX_ORDER 20
-
-enum AudioObjectType {
-    AOT_NULL,
-                               // Support?                Name
-    AOT_AAC_MAIN,              ///< Y                       Main
-    AOT_AAC_LC,                ///< Y                       Low Complexity
-    AOT_AAC_SSR,               ///< N (code in SoC repo)    Scalable Sample Rate
-    AOT_AAC_LTP,               ///< N (code in SoC repo)    Long Term Prediction
-    AOT_SBR,                   ///< N (in progress)         Spectral Band Replication
-    AOT_AAC_SCALABLE,          ///< N                       Scalable
-    AOT_TWINVQ,                ///< N                       Twin Vector Quantizer
-    AOT_CELP,                  ///< N                       Code Excited Linear Prediction
-    AOT_HVXC,                  ///< N                       Harmonic Vector eXcitation Coding
-    AOT_TTSI             = 12, ///< N                       Text-To-Speech Interface
-    AOT_MAINSYNTH,             ///< N                       Main Synthesis
-    AOT_WAVESYNTH,             ///< N                       Wavetable Synthesis
-    AOT_MIDI,                  ///< N                       General MIDI
-    AOT_SAFX,                  ///< N                       Algorithmic Synthesis and Audio Effects
-    AOT_ER_AAC_LC,             ///< N                       Error Resilient Low Complexity
-    AOT_ER_AAC_LTP       = 19, ///< N                       Error Resilient Long Term Prediction
-    AOT_ER_AAC_SCALABLE,       ///< N                       Error Resilient Scalable
-    AOT_ER_TWINVQ,             ///< N                       Error Resilient Twin Vector Quantizer
-    AOT_ER_BSAC,               ///< N                       Error Resilient Bit-Sliced Arithmetic Coding
-    AOT_ER_AAC_LD,             ///< N                       Error Resilient Low Delay
-    AOT_ER_CELP,               ///< N                       Error Resilient Code Excited Linear Prediction
-    AOT_ER_HVXC,               ///< N                       Error Resilient Harmonic Vector eXcitation Coding
-    AOT_ER_HILN,               ///< N                       Error Resilient Harmonic and Individual Lines plus Noise
-    AOT_ER_PARAM,              ///< N                       Error Resilient Parametric
-    AOT_SSC,                   ///< N                       SinuSoidal Coding
-};
-
-enum RawDataBlockType {
-    TYPE_SCE,
-    TYPE_CPE,
-    TYPE_CCE,
-    TYPE_LFE,
-    TYPE_DSE,
-    TYPE_PCE,
-    TYPE_FIL,
-    TYPE_END,
-};
-
-enum ExtensionPayloadID {
-    EXT_FILL,
-    EXT_FILL_DATA,
-    EXT_DATA_ELEMENT,
-    EXT_DYNAMIC_RANGE = 0xb,
-    EXT_SBR_DATA      = 0xd,
-    EXT_SBR_DATA_CRC  = 0xe,
-};
-
-enum WindowSequence {
-    ONLY_LONG_SEQUENCE,
-    LONG_START_SEQUENCE,
-    EIGHT_SHORT_SEQUENCE,
-    LONG_STOP_SEQUENCE,
-};
-
-enum BandType {
-    ZERO_BT        = 0,     ///< Scalefactors and spectral data are all zero.
-    FIRST_PAIR_BT  = 5,     ///< This and later band types encode two values (rather than four) with one code word.
-    ESC_BT         = 11,    ///< Spectral data are coded with an escape sequence.
-    NOISE_BT       = 13,    ///< Spectral data are scaled white noise not coded in the bitstream.
-    INTENSITY_BT2  = 14,    ///< Scalefactor data are intensity stereo positions.
-    INTENSITY_BT   = 15,    ///< Scalefactor data are intensity stereo positions.
-};
-
-#define IS_CODEBOOK_UNSIGNED(x) ((x - 1) & 10)
-
-enum ChannelPosition {
-    AAC_CHANNEL_FRONT = 1,
-    AAC_CHANNEL_SIDE  = 2,
-    AAC_CHANNEL_BACK  = 3,
-    AAC_CHANNEL_LFE   = 4,
-    AAC_CHANNEL_CC    = 5,
-};
-
-/**
- * The point during decoding at which channel coupling is applied.
- */
-enum CouplingPoint {
-    BEFORE_TNS,
-    BETWEEN_TNS_AND_IMDCT,
-    AFTER_IMDCT = 3,
-};
-
-/**
- * Predictor State
- */
-typedef struct {
-    float cor0;
-    float cor1;
-    float var0;
-    float var1;
-    float r0;
-    float r1;
-} PredictorState;
-
-#define MAX_PREDICTORS 672
-
-/**
- * Individual Channel Stream
- */
-typedef struct {
-    uint8_t max_sfb;            ///< number of scalefactor bands per group
-    enum WindowSequence window_sequence[2];
-    uint8_t use_kb_window[2];   ///< If set, use Kaiser-Bessel window, otherwise use a sinus window.
-    int num_window_groups;
-    uint8_t group_len[8];
-    const uint16_t *swb_offset; ///< table of offsets to the lowest spectral coefficient of a scalefactor band, sfb, for a particular window
-    int num_swb;                ///< number of scalefactor window bands
-    int num_windows;
-    int tns_max_bands;
-    int predictor_present;
-    int predictor_initialized;
-    int predictor_reset_group;
-    uint8_t prediction_used[41];
-} IndividualChannelStream;
-
-/**
- * Temporal Noise Shaping
- */
-typedef struct {
-    int present;
-    int n_filt[8];
-    int length[8][4];
-    int direction[8][4];
-    int order[8][4];
-    float coef[8][4][TNS_MAX_ORDER];
-} TemporalNoiseShaping;
-
-/**
- * Dynamic Range Control - decoded from the bitstream but not processed further.
- */
-typedef struct {
-    int pce_instance_tag;                           ///< Indicates with which program the DRC info is associated.
-    int dyn_rng_sgn[17];                            ///< DRC sign information; 0 - positive, 1 - negative
-    int dyn_rng_ctl[17];                            ///< DRC magnitude information
-    int exclude_mask[MAX_CHANNELS];                 ///< Channels to be excluded from DRC processing.
-    int band_incr;                                  ///< Number of DRC bands greater than 1 having DRC info.
-    int interpolation_scheme;                       ///< Indicates the interpolation scheme used in the SBR QMF domain.
-    int band_top[17];                               ///< Indicates the top of the i-th DRC band in units of 4 spectral lines.
-    int prog_ref_level;                             /**< A reference level for the long-term program audio level for all
-                                                     *   channels combined.
-                                                     */
-} DynamicRangeControl;
-
-typedef struct {
-    int num_pulse;
-    int pos[4];
-    int amp[4];
-} Pulse;
-
-/**
- * coupling parameters
- */
-typedef struct {
-    enum CouplingPoint coupling_point;  ///< The point during decoding at which coupling is applied.
-    int num_coupled;       ///< number of target elements
-    enum RawDataBlockType type[8];   ///< Type of channel element to be coupled - SCE or CPE.
-    int id_select[8];      ///< element id
-    int ch_select[8];      /**< [0] shared list of gains; [1] list of gains for right channel;
-                            *   [2] list of gains for left channel; [3] lists of gains for both channels
-                            */
-    float gain[16][120];
-} ChannelCoupling;
-
-/**
- * Single Channel Element - used for both SCE and LFE elements.
- */
-typedef struct {
-    IndividualChannelStream ics;
-    TemporalNoiseShaping tns;
-    enum BandType band_type[120];             ///< band types
-    int band_type_run_end[120];               ///< band type run end points
-    float sf[120];                            ///< scalefactors
-    DECLARE_ALIGNED_16(float, coeffs[1024]);  ///< coefficients for IMDCT
-    DECLARE_ALIGNED_16(float, saved[512]);    ///< overlap
-    DECLARE_ALIGNED_16(float, ret[1024]);     ///< PCM output
-    PredictorState predictor_state[MAX_PREDICTORS];
-} SingleChannelElement;
-
-/**
- * channel element - generic struct for SCE/CPE/CCE/LFE
- */
-typedef struct {
-    // CPE specific
-    uint8_t ms_mask[120];     ///< Set if mid/side stereo is used for each scalefactor window band
-    // shared
-    SingleChannelElement ch[2];
-    // CCE specific
-    ChannelCoupling coup;
-} ChannelElement;
-
-/**
- * main AAC context
- */
-typedef struct {
-    AVCodecContext * avccontext;
-
-    MPEG4AudioConfig m4ac;
-
-    int is_saved;                 ///< Set if elements have stored overlap from previous frame.
-    DynamicRangeControl che_drc;
-
-    /**
-     * @defgroup elements Channel element related data.
-     * @{
-     */
-    enum ChannelPosition che_pos[4][MAX_ELEM_ID]; /**< channel element channel mapping with the
-                                                   *   first index as the first 4 raw data block types
-                                                   */
-    ChannelElement * che[4][MAX_ELEM_ID];
-    ChannelElement * tag_che_map[4][MAX_ELEM_ID];
-    int tags_mapped;
-    /** @} */
-
-    /**
-     * @defgroup temporary aligned temporary buffers (We do not want to have these on the stack.)
-     * @{
-     */
-    DECLARE_ALIGNED_16(float, buf_mdct[1024]);
-    /** @} */
-
-    /**
-     * @defgroup tables   Computed / set up during initialization.
-     * @{
-     */
-    MDCTContext mdct;
-    MDCTContext mdct_small;
-    DSPContext dsp;
-    int random_state;
-    /** @} */
-
-    /**
-     * @defgroup output   Members used for output interleaving.
-     * @{
-     */
-    float *output_data[MAX_CHANNELS];                 ///< Points to each element's 'ret' buffer (PCM output).
-    float add_bias;                                   ///< offset for dsp.float_to_int16
-    float sf_scale;                                   ///< Pre-scale for correct IMDCT and dsp.float_to_int16.
-    int sf_offset;                                    ///< offset into pow2sf_tab as appropriate for dsp.float_to_int16
-    /** @} */
-
-    DECLARE_ALIGNED(16, float, temp[128]);
-} AACContext;
-
-#endif /* AVCODEC_AAC_H */

libavcodec/aac_ac3_parser.c

@@ -1,93 +0,0 @@
-/*
- * Common AAC and AC-3 parser
- * Copyright (c) 2003 Fabrice Bellard
- * Copyright (c) 2003 Michael Niedermayer
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#include "parser.h"
-#include "aac_ac3_parser.h"
-
-int ff_aac_ac3_parse(AVCodecParserContext *s1,
-                     AVCodecContext *avctx,
-                     const uint8_t **poutbuf, int *poutbuf_size,
-                     const uint8_t *buf, int buf_size)
-{
-    AACAC3ParseContext *s = s1->priv_data;
-    ParseContext *pc = &s->pc;
-    int len, i;
-    int new_frame_start;
-
-get_next:
-    i=END_NOT_FOUND;
-    if(s->remaining_size <= buf_size){
-        if(s->remaining_size && !s->need_next_header){
-            i= s->remaining_size;
-            s->remaining_size = 0;
-        }else{ //we need a header first
-            len=0;
-            for(i=s->remaining_size; i<buf_size; i++){
-                s->state = (s->state<<8) + buf[i];
-                if((len=s->sync(s->state, s, &s->need_next_header, &new_frame_start)))
-                    break;
-            }
-            if(len<=0){
-                i=END_NOT_FOUND;
-            }else{
-                s->state=0;
-                i-= s->header_size -1;
-                s->remaining_size = len;
-                if(!new_frame_start || pc->index+i<=0){
-                    s->remaining_size += i;
-                    goto get_next;
-                }
-            }
-        }
-    }
-
-    if(ff_combine_frame(pc, i, &buf, &buf_size)<0){
-        s->remaining_size -= FFMIN(s->remaining_size, buf_size);
-        *poutbuf = NULL;
-        *poutbuf_size = 0;
-        return buf_size;
-    }
-
-    *poutbuf = buf;
-    *poutbuf_size = buf_size;
-
-    /* update codec info */
-    avctx->sample_rate = s->sample_rate;
-    if(s->codec_id)
-        avctx->codec_id = s->codec_id;
-
-    /* allow downmixing to stereo (or mono for AC-3) */
-    if(avctx->request_channels > 0 &&
-            avctx->request_channels < s->channels &&
-            (avctx->request_channels <= 2 ||
-            (avctx->request_channels == 1 &&
-            (avctx->codec_id == CODEC_ID_AC3 ||
-             avctx->codec_id == CODEC_ID_EAC3)))) {
-        avctx->channels = avctx->request_channels;
-    } else {
-        avctx->channels = s->channels;
-    }
-    avctx->bit_rate = s->bit_rate;
-    avctx->frame_size = s->samples;
-
-    return i;
-}

libavcodec/aac_ac3_parser.h

@@ -1,64 +0,0 @@
-/*
- * Common AAC and AC-3 parser prototypes
- * Copyright (c) 2003 Fabrice Bellard
- * Copyright (c) 2003 Michael Niedermayer
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#ifndef AVCODEC_AAC_AC3_PARSER_H
-#define AVCODEC_AAC_AC3_PARSER_H
-
-#include <stdint.h>
-#include "avcodec.h"
-#include "parser.h"
-
-typedef enum {
-    AAC_AC3_PARSE_ERROR_SYNC        = -1,
-    AAC_AC3_PARSE_ERROR_BSID        = -2,
-    AAC_AC3_PARSE_ERROR_SAMPLE_RATE = -3,
-    AAC_AC3_PARSE_ERROR_FRAME_SIZE  = -4,
-    AAC_AC3_PARSE_ERROR_FRAME_TYPE  = -5,
-    AAC_AC3_PARSE_ERROR_CRC         = -6,
-    AAC_AC3_PARSE_ERROR_CHANNEL_CFG = -7,
-} AACAC3ParseError;
-
-typedef struct AACAC3ParseContext {
-    ParseContext pc;
-    int frame_size;
-    int header_size;
-    int (*sync)(uint64_t state, struct AACAC3ParseContext *hdr_info,
-            int *need_next_header, int *new_frame_start);
-
-    int channels;
-    int sample_rate;
-    int bit_rate;
-    int samples;
-
-    int remaining_size;
-    uint64_t state;
-
-    int need_next_header;
-    enum CodecID codec_id;
-} AACAC3ParseContext;
-
-int ff_aac_ac3_parse(AVCodecParserContext *s1,
-                     AVCodecContext *avctx,
-                     const uint8_t **poutbuf, int *poutbuf_size,
-                     const uint8_t *buf, int buf_size);
-
-#endif /* AVCODEC_AAC_AC3_PARSER_H */

libavcodec/aac_parser.c

@@ -1,117 +0,0 @@
-/*
- * Audio and Video frame extraction
- * Copyright (c) 2003 Fabrice Bellard
- * Copyright (c) 2003 Michael Niedermayer
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#include "parser.h"
-#include "aac_ac3_parser.h"
-#include "aac_parser.h"
-#include "bitstream.h"
-#include "mpeg4audio.h"
-
-#define AAC_HEADER_SIZE 7
-
-int ff_aac_parse_header(GetBitContext *gbc, AACADTSHeaderInfo *hdr)
-{
-    int size, rdb, ch, sr;
-    int aot, crc_abs;
-
-    if(get_bits(gbc, 12) != 0xfff)
-        return AAC_AC3_PARSE_ERROR_SYNC;
-
-    skip_bits1(gbc);             /* id */
-    skip_bits(gbc, 2);           /* layer */
-    crc_abs = get_bits1(gbc);    /* protection_absent */
-    aot     = get_bits(gbc, 2);  /* profile_objecttype */
-    sr      = get_bits(gbc, 4);  /* sample_frequency_index */
-    if(!ff_mpeg4audio_sample_rates[sr])
-        return AAC_AC3_PARSE_ERROR_SAMPLE_RATE;
-    skip_bits1(gbc);             /* private_bit */
-    ch      = get_bits(gbc, 3);  /* channel_configuration */
-
-    if(!ff_mpeg4audio_channels[ch])
-        return AAC_AC3_PARSE_ERROR_CHANNEL_CFG;
-
-    skip_bits1(gbc);             /* original/copy */
-    skip_bits1(gbc);             /* home */
-
-    /* adts_variable_header */
-    skip_bits1(gbc);             /* copyright_identification_bit */
-    skip_bits1(gbc);             /* copyright_identification_start */
-    size    = get_bits(gbc, 13); /* aac_frame_length */
-    if(size < AAC_HEADER_SIZE)
-        return AAC_AC3_PARSE_ERROR_FRAME_SIZE;
-
-    skip_bits(gbc, 11);          /* adts_buffer_fullness */
-    rdb = get_bits(gbc, 2);      /* number_of_raw_data_blocks_in_frame */
-
-    hdr->object_type    = aot + 1;
-    hdr->chan_config    = ch;
-    hdr->crc_absent     = crc_abs;
-    hdr->num_aac_frames = rdb + 1;
-    hdr->sampling_index = sr;
-    hdr->sample_rate    = ff_mpeg4audio_sample_rates[sr];
-    hdr->samples        = (rdb + 1) * 1024;
-    hdr->bit_rate       = size * 8 * hdr->sample_rate / hdr->samples;
-
-    return size;
-}
-
-static int aac_sync(uint64_t state, AACAC3ParseContext *hdr_info,
-        int *need_next_header, int *new_frame_start)
-{
-    GetBitContext bits;
-    AACADTSHeaderInfo hdr;
-    int size;
-    union {
-        uint64_t u64;
-        uint8_t  u8[8];
-    } tmp;
-
-    tmp.u64 = be2me_64(state);
-    init_get_bits(&bits, tmp.u8+8-AAC_HEADER_SIZE, AAC_HEADER_SIZE * 8);
-
-    if ((size = ff_aac_parse_header(&bits, &hdr)) < 0)
-        return 0;
-    *need_next_header = 0;
-    *new_frame_start  = 1;
-    hdr_info->sample_rate = hdr.sample_rate;
-    hdr_info->channels    = ff_mpeg4audio_channels[hdr.chan_config];
-    hdr_info->samples     = hdr.samples;
-    hdr_info->bit_rate    = hdr.bit_rate;
-    return size;
-}
-
-static av_cold int aac_parse_init(AVCodecParserContext *s1)
-{
-    AACAC3ParseContext *s = s1->priv_data;
-    s->header_size = AAC_HEADER_SIZE;
-    s->sync = aac_sync;
-    return 0;
-}
-
-
-AVCodecParser aac_parser = {
-    { CODEC_ID_AAC },
-    sizeof(AACAC3ParseContext),
-    aac_parse_init,
-    ff_aac_ac3_parse,
-    ff_parse_close,
-};

libavcodec/aac_parser.h

@@ -1,53 +0,0 @@
-/*
- * AAC parser prototypes
- * Copyright (c) 2003 Fabrice Bellard
- * Copyright (c) 2003 Michael Niedermayer
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#ifndef AVCODEC_AAC_PARSER_H
-#define AVCODEC_AAC_PARSER_H
-
-#include <stdint.h>
-#include "aac_ac3_parser.h"
-#include "bitstream.h"
-
-typedef struct {
-    uint32_t sample_rate;
-    uint32_t samples;
-    uint32_t bit_rate;
-    uint8_t  crc_absent;
-    uint8_t  object_type;
-    uint8_t  sampling_index;
-    uint8_t  chan_config;
-    uint8_t  num_aac_frames;
-} AACADTSHeaderInfo;
-
-/**
- * Parses AAC frame header.
- * Parses the ADTS frame header to the end of the variable header, which is
- * the first 54 bits.
- * @param gbc[in] BitContext containing the first 54 bits of the frame.
- * @param hdr[out] Pointer to struct where header info is written.
- * @return Returns 0 on success, -1 if there is a sync word mismatch,
- * -2 if the version element is invalid, -3 if the sample rate
- * element is invalid, or -4 if the bit rate element is invalid.
- */
-int ff_aac_parse_header(GetBitContext *gbc, AACADTSHeaderInfo *hdr);
-
-#endif /* AVCODEC_AAC_PARSER_H */

libavcodec/aacdectab.h

@@ -1,211 +0,0 @@
-/*
- * AAC decoder data
- * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
- * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/**
- * @file libavcodec/aacdectab.h
- * AAC decoder data
- * @author Oded Shimon  ( ods15 ods15 dyndns org )
- * @author Maxim Gavrilov ( maxim.gavrilov gmail com )
- */
-
-#ifndef AVCODEC_AACDECTAB_H
-#define AVCODEC_AACDECTAB_H
-
-#include "aac.h"
-
-#include <stdint.h>
-
-/* @name swb_offsets
- * Sample offset into the window indicating the beginning of a scalefactor
- * window band
- *
- * scalefactor window band - term for scalefactor bands within a window,
- * given in Table 4.110 to Table 4.128.
- *
- * scalefactor band - a set of spectral coefficients which are scaled by one
- * scalefactor. In case of EIGHT_SHORT_SEQUENCE and grouping a scalefactor band
- * may contain several scalefactor window bands of corresponding frequency. For
- * all other window_sequences scalefactor bands and scalefactor window bands are
- * identical.
- * @{
- */
-
-static const uint16_t swb_offset_1024_96[] = {
-      0,   4,   8,  12,  16,  20,  24,  28,
-     32,  36,  40,  44,  48,  52,  56,  64,
-     72,  80,  88,  96, 108, 120, 132, 144,
-    156, 172, 188, 212, 240, 276, 320, 384,
-    448, 512, 576, 640, 704, 768, 832, 896,
-    960, 1024
-};
-
-static const uint16_t swb_offset_128_96[] = {
-    0, 4, 8, 12, 16, 20, 24, 32, 40, 48, 64, 92, 128
-};
-
-static const uint16_t swb_offset_1024_64[] = {
-      0,   4,   8,  12,  16,  20,  24,  28,
-     32,  36,  40,  44,  48,  52,  56,  64,
-     72,  80,  88, 100, 112, 124, 140, 156,
-    172, 192, 216, 240, 268, 304, 344, 384,
-    424, 464, 504, 544, 584, 624, 664, 704,
-    744, 784, 824, 864, 904, 944, 984, 1024
-};
-
-static const uint16_t swb_offset_1024_48[] = {
-      0,   4,   8,  12,  16,  20,  24,  28,
-     32,  36,  40,  48,  56,  64,  72,  80,
-     88,  96, 108, 120, 132, 144, 160, 176,
-    196, 216, 240, 264, 292, 320, 352, 384,
-    416, 448, 480, 512, 544, 576, 608, 640,
-    672, 704, 736, 768, 800, 832, 864, 896,
-    928, 1024
-};
-
-static const uint16_t swb_offset_128_48[] = {
-     0,   4,   8,  12,  16,  20,  28,  36,
-    44,  56,  68,  80,  96, 112, 128
-};
-
-static const uint16_t swb_offset_1024_32[] = {
-      0,   4,   8,  12,  16,  20,  24,  28,
-     32,  36,  40,  48,  56,  64,  72,  80,
-     88,  96, 108, 120, 132, 144, 160, 176,
-    196, 216, 240, 264, 292, 320, 352, 384,
-    416, 448, 480, 512, 544, 576, 608, 640,
-    672, 704, 736, 768, 800, 832, 864, 896,
-    928, 960, 992, 1024
-};
-
-static const uint16_t swb_offset_1024_24[] = {
-      0,   4,   8,  12,  16,  20,  24,  28,
-     32,  36,  40,  44,  52,  60,  68,  76,
-     84,  92, 100, 108, 116, 124, 136, 148,
-    160, 172, 188, 204, 220, 240, 260, 284,
-    308, 336, 364, 396, 432, 468, 508, 552,
-    600, 652, 704, 768, 832, 896, 960, 1024
-};
-
-static const uint16_t swb_offset_128_24[] = {
-     0,   4,   8,  12,  16,  20,  24,  28,
-    36,  44,  52,  64,  76,  92, 108, 128
-};
-
-static const uint16_t swb_offset_1024_16[] = {
-      0,   8,  16,  24,  32,  40,  48,  56,
-     64,  72,  80,  88, 100, 112, 124, 136,
-    148, 160, 172, 184, 196, 212, 228, 244,
-    260, 280, 300, 320, 344, 368, 396, 424,
-    456, 492, 532, 572, 616, 664, 716, 772,
-    832, 896, 960, 1024
-};
-
-static const uint16_t swb_offset_128_16[] = {
-     0,   4,   8,  12,  16,  20,  24,  28,
-    32,  40,  48,  60,  72,  88, 108, 128
-};
-
-static const uint16_t swb_offset_1024_8[] = {
-      0,  12,  24,  36,  48,  60,  72,  84,
-     96, 108, 120, 132, 144, 156, 172, 188,
-    204, 220, 236, 252, 268, 288, 308, 328,
-    348, 372, 396, 420, 448, 476, 508, 544,
-    580, 620, 664, 712, 764, 820, 880, 944,
-    1024
-};
-
-static const uint16_t swb_offset_128_8[] = {
-     0,   4,   8,  12,  16,  20,  24,  28,
-    36,  44,  52,  60,  72,  88, 108, 128
-};
-
-static const uint16_t *swb_offset_1024[] = {
-    swb_offset_1024_96, swb_offset_1024_96, swb_offset_1024_64,
-    swb_offset_1024_48, swb_offset_1024_48, swb_offset_1024_32,
-    swb_offset_1024_24, swb_offset_1024_24, swb_offset_1024_16,
-    swb_offset_1024_16, swb_offset_1024_16, swb_offset_1024_8,
-    swb_offset_1024_8
-};
-
-static const uint16_t *swb_offset_128[] = {
-    /* The last entry on the following row is swb_offset_128_64 but is a
-       duplicate of swb_offset_128_96. */
-    swb_offset_128_96, swb_offset_128_96, swb_offset_128_96,
-    swb_offset_128_48, swb_offset_128_48, swb_offset_128_48,
-    swb_offset_128_24, swb_offset_128_24, swb_offset_128_16,
-    swb_offset_128_16, swb_offset_128_16, swb_offset_128_8,
-    swb_offset_128_8
-};
-
-// @}
-
-/* @name tns_max_bands
- * The maximum number of scalefactor bands on which TNS can operate for the long
- * and short transforms respectively. The index to these tables is related to
- * the sample rate of the audio.
- * @{
- */
-static const uint8_t tns_max_bands_1024[] = {
-    31, 31, 34, 40, 42, 51, 46, 46, 42, 42, 42, 39, 39
-};
-
-static const uint8_t tns_max_bands_128[] = {
-    9, 9, 10, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14
-};
-// @}
-
-/* @name tns_tmp2_map
- * Tables of the tmp2[] arrays of LPC coefficients used for TNS.
- * The suffix _M_N[] indicate the values of coef_compress and coef_res
- * respectively.
- * @{
- */
-static const float tns_tmp2_map_1_3[4] = {
-     0.00000000, -0.43388373,  0.64278758,  0.34202015,
-};
-
-static const float tns_tmp2_map_0_3[8] = {
-     0.00000000, -0.43388373, -0.78183150, -0.97492790,
-     0.98480773,  0.86602539,  0.64278758,  0.34202015,
-};
-
-static const float tns_tmp2_map_1_4[8] = {
-     0.00000000, -0.20791170, -0.40673664, -0.58778524,
-     0.67369562,  0.52643216,  0.36124167,  0.18374951,
-};
-
-static const float tns_tmp2_map_0_4[16] = {
-     0.00000000, -0.20791170, -0.40673664, -0.58778524,
-    -0.74314481, -0.86602539, -0.95105654, -0.99452192,
-     0.99573416,  0.96182561,  0.89516330,  0.79801720,
-     0.67369562,  0.52643216,  0.36124167,  0.18374951,
-};
-
-static const float * const tns_tmp2_map[4] = {
-    tns_tmp2_map_0_3,
-    tns_tmp2_map_0_4,
-    tns_tmp2_map_1_3,
-    tns_tmp2_map_1_4
-};
-// @}
-
-#endif /* AVCODEC_AACDECTAB_H */

libavcodec/aacenc.c

@@ -1,365 +0,0 @@
-/*
- * AAC encoder
- * Copyright (C) 2008 Konstantin Shishkov
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/**
- * @file libavcodec/aacenc.c
- * AAC encoder
- */
-
-/***********************************
- *              TODOs:
- * psy model selection with some option
- * add sane pulse detection
- * add temporal noise shaping
- ***********************************/
-
-#include "avcodec.h"
-#include "bitstream.h"
-#include "dsputil.h"
-#include "mpeg4audio.h"
-
-#include "aacpsy.h"
-#include "aac.h"
-#include "aactab.h"
-
-static const uint8_t swb_size_1024_96[] = {
-    4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8,
-    12, 12, 12, 12, 12, 16, 16, 24, 28, 36, 44,
-    64, 64, 64, 64, 64, 64, 64, 64, 64, 64, 64
-};
-
-static const uint8_t swb_size_1024_64[] = {
-    4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8,
-    12, 12, 12, 16, 16, 16, 20, 24, 24, 28, 36,
-    40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40
-};
-
-static const uint8_t swb_size_1024_48[] = {
-    4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
-    12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,
-    32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32,
-    96
-};
-
-static const uint8_t swb_size_1024_32[] = {
-    4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
-    12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,
-    32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32
-};
-
-static const uint8_t swb_size_1024_24[] = {
-    4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,
-    12, 12, 12, 12, 16, 16, 16, 20, 20, 24, 24, 28, 28,
-    32, 36, 36, 40, 44, 48, 52, 52, 64, 64, 64, 64, 64
-};
-
-static const uint8_t swb_size_1024_16[] = {
-    8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,
-    12, 12, 12, 12, 12, 12, 12, 12, 12, 16, 16, 16, 16, 20, 20, 20, 24, 24, 28, 28,
-    32, 36, 40, 40, 44, 48, 52, 56, 60, 64, 64, 64
-};
-
-static const uint8_t swb_size_1024_8[] = {
-    12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12,
-    16, 16, 16, 16, 16, 16, 16, 20, 20, 20, 20, 24, 24, 24, 28, 28,
-    32, 36, 36, 40, 44, 48, 52, 56, 60, 64, 80
-};
-
-static const uint8_t * const swb_size_1024[] = {
-    swb_size_1024_96, swb_size_1024_96, swb_size_1024_64,
-    swb_size_1024_48, swb_size_1024_48, swb_size_1024_32,
-    swb_size_1024_24, swb_size_1024_24, swb_size_1024_16,
-    swb_size_1024_16, swb_size_1024_16, swb_size_1024_8
-};
-
-static const uint8_t swb_size_128_96[] = {
-    4, 4, 4, 4, 4, 4, 8, 8, 8, 16, 28, 36
-};
-
-static const uint8_t swb_size_128_48[] = {
-    4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 12, 16, 16, 16
-};
-
-static const uint8_t swb_size_128_24[] = {
-    4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 16, 16, 20
-};
-
-static const uint8_t swb_size_128_16[] = {
-    4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 12, 12, 16, 20, 20
-};
-
-static const uint8_t swb_size_128_8[] = {
-    4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 12, 16, 20, 20
-};
-
-static const uint8_t * const swb_size_128[] = {
-    /* the last entry on the following row is swb_size_128_64 but is a
-       duplicate of swb_size_128_96 */
-    swb_size_128_96, swb_size_128_96, swb_size_128_96,
-    swb_size_128_48, swb_size_128_48, swb_size_128_48,
-    swb_size_128_24, swb_size_128_24, swb_size_128_16,
-    swb_size_128_16, swb_size_128_16, swb_size_128_8
-};
-
-/** bits needed to code codebook run value for long windows */
-static const uint8_t run_value_bits_long[64] = {
-     5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,
-     5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5, 10,
-    10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10,
-    10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 15
-};
-
-/** bits needed to code codebook run value for short windows */
-static const uint8_t run_value_bits_short[16] = {
-    3, 3, 3, 3, 3, 3, 3, 6, 6, 6, 6, 6, 6, 6, 6, 9
-};
-
-static const uint8_t* const run_value_bits[2] = {
-    run_value_bits_long, run_value_bits_short
-};
-
-/** default channel configurations */
-static const uint8_t aac_chan_configs[6][5] = {
- {1, TYPE_SCE},                               // 1 channel  - single channel element
- {1, TYPE_CPE},                               // 2 channels - channel pair
- {2, TYPE_SCE, TYPE_CPE},                     // 3 channels - center + stereo
- {3, TYPE_SCE, TYPE_CPE, TYPE_SCE},           // 4 channels - front center + stereo + back center
- {3, TYPE_SCE, TYPE_CPE, TYPE_CPE},           // 5 channels - front center + stereo + back stereo
- {4, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_LFE}, // 6 channels - front center + stereo + back stereo + LFE
-};
-
-/**
- * structure used in optimal codebook search
- */
-typedef struct BandCodingPath {
-    int prev_idx; ///< pointer to the previous path point
-    int codebook; ///< codebook for coding band run
-    int bits;     ///< number of bit needed to code given number of bands
-} BandCodingPath;
-
-/**
- * AAC encoder context
- */
-typedef struct {
-    PutBitContext pb;
-    MDCTContext mdct1024;                        ///< long (1024 samples) frame transform context
-    MDCTContext mdct128;                         ///< short (128 samples) frame transform context
-    DSPContext  dsp;
-    DECLARE_ALIGNED_16(FFTSample, output[2048]); ///< temporary buffer for MDCT input coefficients
-    int16_t* samples;                            ///< saved preprocessed input
-
-    int samplerate_index;                        ///< MPEG-4 samplerate index
-
-    ChannelElement *cpe;                         ///< channel elements
-    AACPsyContext psy;                           ///< psychoacoustic model context
-    int last_frame;
-} AACEncContext;
-
-/**
- * Make AAC audio config object.
- * @see 1.6.2.1 "Syntax - AudioSpecificConfig"
- */
-static void put_audio_specific_config(AVCodecContext *avctx)
-{
-    PutBitContext pb;
-    AACEncContext *s = avctx->priv_data;
-
-    init_put_bits(&pb, avctx->extradata, avctx->extradata_size*8);
-    put_bits(&pb, 5, 2); //object type - AAC-LC
-    put_bits(&pb, 4, s->samplerate_index); //sample rate index
-    put_bits(&pb, 4, avctx->channels);
-    //GASpecificConfig
-    put_bits(&pb, 1, 0); //frame length - 1024 samples
-    put_bits(&pb, 1, 0); //does not depend on core coder
-    put_bits(&pb, 1, 0); //is not extension
-    flush_put_bits(&pb);
-}
-
-static av_cold int aac_encode_init(AVCodecContext *avctx)
-{
-    AACEncContext *s = avctx->priv_data;
-    int i;
-
-    avctx->frame_size = 1024;
-
-    for(i = 0; i < 16; i++)
-        if(avctx->sample_rate == ff_mpeg4audio_sample_rates[i])
-            break;
-    if(i == 16){
-        av_log(avctx, AV_LOG_ERROR, "Unsupported sample rate %d\n", avctx->sample_rate);
-        return -1;
-    }
-    if(avctx->channels > 6){
-        av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels: %d\n", avctx->channels);
-        return -1;
-    }
-    s->samplerate_index = i;
-
-    dsputil_init(&s->dsp, avctx);
-    ff_mdct_init(&s->mdct1024, 11, 0);
-    ff_mdct_init(&s->mdct128,   8, 0);
-    // window init
-    ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
-    ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
-    ff_sine_window_init(ff_sine_1024, 1024);
-    ff_sine_window_init(ff_sine_128, 128);
-
-    s->samples = av_malloc(2 * 1024 * avctx->channels * sizeof(s->samples[0]));
-    s->cpe = av_mallocz(sizeof(ChannelElement) * aac_chan_configs[avctx->channels-1][0]);
-    if(ff_aac_psy_init(&s->psy, avctx, AAC_PSY_3GPP,
-                       aac_chan_configs[avctx->channels-1][0], 0,
-                       swb_size_1024[i], ff_aac_num_swb_1024[i], swb_size_128[i], ff_aac_num_swb_128[i]) < 0){
-        av_log(avctx, AV_LOG_ERROR, "Cannot initialize selected model.\n");
-        return -1;
-    }
-    avctx->extradata = av_malloc(2);
-    avctx->extradata_size = 2;
-    put_audio_specific_config(avctx);
-    return 0;
-}
-
-/**
- * Encode ics_info element.
- * @see Table 4.6 (syntax of ics_info)
- */
-static void put_ics_info(AACEncContext *s, IndividualChannelStream *info)
-{
-    int i;
-
-    put_bits(&s->pb, 1, 0);                // ics_reserved bit
-    put_bits(&s->pb, 2, info->window_sequence[0]);
-    put_bits(&s->pb, 1, info->use_kb_window[0]);
-    if(info->window_sequence[0] != EIGHT_SHORT_SEQUENCE){
-        put_bits(&s->pb, 6, info->max_sfb);
-        put_bits(&s->pb, 1, 0);            // no prediction
-    }else{
-        put_bits(&s->pb, 4, info->max_sfb);
-        for(i = 1; i < info->num_windows; i++)
-            put_bits(&s->pb, 1, info->group_len[i]);
-    }
-}
-
-/**
- * Calculate the number of bits needed to code all coefficient signs in current band.
- */
-static int calculate_band_sign_bits(AACEncContext *s, SingleChannelElement *sce,
-                                    int group_len, int start, int size)
-{
-    int bits = 0;
-    int i, w;
-    for(w = 0; w < group_len; w++){
-        for(i = 0; i < size; i++){
-            if(sce->icoefs[start + i])
-                bits++;
-        }
-        start += 128;
-    }
-    return bits;
-}
-
-/**
- * Encode pulse data.
- */
-static void encode_pulses(AACEncContext *s, Pulse *pulse)
-{
-    int i;
-
-    put_bits(&s->pb, 1, !!pulse->num_pulse);
-    if(!pulse->num_pulse) return;
-
-    put_bits(&s->pb, 2, pulse->num_pulse - 1);
-    put_bits(&s->pb, 6, pulse->start);
-    for(i = 0; i < pulse->num_pulse; i++){
-        put_bits(&s->pb, 5, pulse->pos[i]);
-        put_bits(&s->pb, 4, pulse->amp[i]);
-    }
-}
-
-/**
- * Encode spectral coefficients processed by psychoacoustic model.
- */
-static void encode_spectral_coeffs(AACEncContext *s, SingleChannelElement *sce)
-{
-    int start, i, w, w2, wg;
-
-    w = 0;
-    for(wg = 0; wg < sce->ics.num_window_groups; wg++){
-        start = 0;
-        for(i = 0; i < sce->ics.max_sfb; i++){
-            if(sce->zeroes[w*16 + i]){
-                start += sce->ics.swb_sizes[i];
-                continue;
-            }
-            for(w2 = w; w2 < w + sce->ics.group_len[wg]; w2++){
-                encode_band_coeffs(s, sce, start + w2*128,
-                                   sce->ics.swb_sizes[i],
-                                   sce->band_type[w*16 + i]);
-            }
-            start += sce->ics.swb_sizes[i];
-        }
-        w += sce->ics.group_len[wg];
-    }
-}
-
-/**
- * Write some auxiliary information about the created AAC file.
- */
-static void put_bitstream_info(AVCodecContext *avctx, AACEncContext *s, const char *name)
-{
-    int i, namelen, padbits;
-
-    namelen = strlen(name) + 2;
-    put_bits(&s->pb, 3, TYPE_FIL);
-    put_bits(&s->pb, 4, FFMIN(namelen, 15));
-    if(namelen >= 15)
-        put_bits(&s->pb, 8, namelen - 16);
-    put_bits(&s->pb, 4, 0); //extension type - filler
-    padbits = 8 - (put_bits_count(&s->pb) & 7);
-    align_put_bits(&s->pb);
-    for(i = 0; i < namelen - 2; i++)
-        put_bits(&s->pb, 8, name[i]);
-    put_bits(&s->pb, 12 - padbits, 0);
-}
-
-static av_cold int aac_encode_end(AVCodecContext *avctx)
-{
-    AACEncContext *s = avctx->priv_data;
-
-    ff_mdct_end(&s->mdct1024);
-    ff_mdct_end(&s->mdct128);
-    ff_aac_psy_end(&s->psy);
-    av_freep(&s->samples);
-    av_freep(&s->cpe);
-    return 0;
-}
-
-AVCodec aac_encoder = {
-    "aac",
-    CODEC_TYPE_AUDIO,
-    CODEC_ID_AAC,
-    sizeof(AACEncContext),
-    aac_encode_init,
-    aac_encode_frame,
-    aac_encode_end,
-    .capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY,
-    .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
-    .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
-};

libavcodec/aacpsy.c

@@ -1,104 +0,0 @@
-/*
- * AAC encoder psychoacoustic model
- * Copyright (C) 2008 Konstantin Shishkov
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/**
- * @file libavcodec/aacpsy.c
- * AAC encoder psychoacoustic model
- */
-
-#include "avcodec.h"
-#include "aacpsy.h"
-#include "aactab.h"
-
-/***********************************
- *              TODOs:
- * General:
- * better audio preprocessing (add DC highpass filter?)
- * more psy models
- * maybe improve coefficient quantization function in some way
- *
- * 3GPP-based psy model:
- * thresholds linearization after their modifications for attaining given bitrate
- * try other bitrate controlling mechanism (maybe use ratecontrol.c?)
- * control quality for quality-based output
- **********************************/
-
-/**
- * Quantize one coefficient.
- * @return absolute value of the quantized coefficient
- * @see 3GPP TS26.403 5.6.2 "Scalefactor determination"
- */
-static av_always_inline int quant(float coef, const float Q)
-{
-    return av_clip((int)(pow(fabsf(coef) * Q, 0.75) + 0.4054), 0, 8191);
-}
-
-static inline float get_approximate_quant_error(float *c, int size, int scale_idx)
-{
-    int i;
-    int q;
-    float coef, unquant, sum = 0.0f;
-    const float Q  = ff_aac_pow2sf_tab[200 - scale_idx + SCALE_ONE_POS - SCALE_DIV_512];
-    const float IQ = ff_aac_pow2sf_tab[200 + scale_idx - SCALE_ONE_POS + SCALE_DIV_512];
-    for(i = 0; i < size; i++){
-        coef = fabs(c[i]);
-        q = quant(c[i], Q);
-        unquant = (q * cbrt(q)) * IQ;
-        sum += (coef - unquant) * (coef - unquant);
-    }
-    return sum;
-}
-
-/**
- * constants for 3GPP AAC psychoacoustic model
- * @{
- */
-#define PSY_3GPP_SPREAD_LOW  1.5f // spreading factor for ascending threshold spreading  (15 dB/Bark)
-#define PSY_3GPP_SPREAD_HI   3.0f // spreading factor for descending threshold spreading (30 dB/Bark)
-/**
- * @}
- */
-
-/**
- * information for single band used by 3GPP TS26.403-inspired psychoacoustic model
- */
-typedef struct Psy3gppBand{
-    float energy;    ///< band energy
-    float ffac;      ///< form factor
-}Psy3gppBand;
-
-/**
- * psychoacoustic model frame type-dependent coefficients
- */
-typedef struct Psy3gppCoeffs{
-    float ath       [64]; ///< absolute threshold of hearing per bands
-    float barks     [64]; ///< Bark value for each spectral band in long frame
-    float spread_low[64]; ///< spreading factor for low-to-high threshold spreading in long frame
-    float spread_hi [64]; ///< spreading factor for high-to-low threshold spreading in long frame
-}Psy3gppCoeffs;
-
-/**
- * Calculate Bark value for given line.
- */
-static inline float calc_bark(float f)
-{
-    return 13.3f * atanf(0.00076f * f) + 3.5f * atanf((f / 7500.0f) * (f / 7500.0f));
-}

libavcodec/aacpsy.h

@@ -1,50 +0,0 @@
-/*
- * AAC encoder psychoacoustic model
- * Copyright (C) 2008 Konstantin Shishkov
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#ifndef AVCODEC_AACPSY_H
-#define AVCODEC_AACPSY_H
-
-#include "avcodec.h"
-#include "aac.h"
-//#include "lowpass.h"
-
-enum AACPsyModelType{
-    AAC_PSY_TEST,              ///< a sample model to exercise encoder
-    AAC_PSY_3GPP,              ///< model following recommendations from 3GPP TS 26.403
-
-    AAC_NB_PSY_MODELS          ///< total number of psychoacoustic models, since it's not a part of the ABI new models can be added freely
-};
-
-/**
- * context used by psychoacoustic model
- */
-typedef struct AACPsyContext {
-    AVCodecContext *avctx;            ///< encoder context
-}AACPsyContext;
-
-/**
- * Cleanup model context at the end.
- *
- * @param ctx model context
- */
-void ff_aac_psy_end(AACPsyContext *ctx);
-
-#endif /* AVCODEC_AACPSY_H */

libavcodec/aactab.h

@@ -1,74 +0,0 @@
-/*
- * AAC data declarations
- * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
- * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/**
- * @file libavcodec/aactab.h
- * AAC data declarations
- * @author Oded Shimon  ( ods15 ods15 dyndns org )
- * @author Maxim Gavrilov ( maxim.gavrilov gmail com )
- */
-
-#ifndef AVCODEC_AACTAB_H
-#define AVCODEC_AACTAB_H
-
-#include "libavutil/mem.h"
-#include "aac.h"
-
-#include <stdint.h>
-
-/* NOTE:
- * Tables in this file are used by the AAC decoder and will be used by the AAC
- * encoder.
- */
-
-/* @name window coefficients
- * @{
- */
-DECLARE_ALIGNED(16, extern float,  ff_aac_kbd_long_1024[1024]);
-DECLARE_ALIGNED(16, extern float,  ff_aac_kbd_short_128[128]);
-// @}
-
-/* @name number of scalefactor window bands for long and short transform windows respectively
- * @{
- */
-extern const uint8_t ff_aac_num_swb_1024[];
-extern const uint8_t ff_aac_num_swb_128 [];
-// @}
-
-extern const uint8_t ff_aac_pred_sfb_max [];
-
-extern const uint32_t ff_aac_scalefactor_code[121];
-extern const uint8_t  ff_aac_scalefactor_bits[121];
-
-extern const uint16_t * const ff_aac_spectral_codes[11];
-extern const uint8_t  * const ff_aac_spectral_bits [11];
-extern const uint16_t  ff_aac_spectral_sizes[11];
-
-extern const float *ff_aac_codebook_vectors[];
-
-#if CONFIG_HARDCODED_TABLES
-extern const float ff_aac_pow2sf_tab[428];
-#else
-extern       float ff_aac_pow2sf_tab[428];
-#endif /* CONFIG_HARDCODED_TABLES */
-
-#endif /* AVCODEC_AACTAB_H */

libavcodec/aandcttab.c

@@ -1,47 +0,0 @@
-/*
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/**
- * @file libavcodec/aandcttab.c
- * AAN (Arai Agui Aakajima) (I)DCT tables
- */
-
-#include <stdint.h>
-
-const uint16_t ff_aanscales[64] = {
-    /* precomputed values scaled up by 14 bits */
-    16384, 22725, 21407, 19266, 16384, 12873,  8867,  4520,
-    22725, 31521, 29692, 26722, 22725, 17855, 12299,  6270,
-    21407, 29692, 27969, 25172, 21407, 16819, 11585,  5906,
-    19266, 26722, 25172, 22654, 19266, 15137, 10426,  5315,
-    16384, 22725, 21407, 19266, 16384, 12873,  8867,  4520,
-    12873, 17855, 16819, 15137, 12873, 10114,  6967,  3552,
-    8867 , 12299, 11585, 10426,  8867,  6967,  4799,  2446,
-    4520 ,  6270,  5906,  5315,  4520,  3552,  2446,  1247
-};
-
-const uint16_t ff_inv_aanscales[64] = {
-  4096,  2953,  3135,  3483,  4096,  5213,  7568, 14846,
-  2953,  2129,  2260,  2511,  2953,  3759,  5457, 10703,
-  3135,  2260,  2399,  2666,  3135,  3990,  5793, 11363,
-  3483,  2511,  2666,  2962,  3483,  4433,  6436, 12625,
-  4096,  2953,  3135,  3483,  4096,  5213,  7568, 14846,
-  5213,  3759,  3990,  4433,  5213,  6635,  9633, 18895,
-  7568,  5457,  5793,  6436,  7568,  9633, 13985, 27432,
- 14846, 10703, 11363, 12625, 14846, 18895, 27432, 53809,
-};

libavcodec/aandcttab.h

@@ -1,32 +0,0 @@
-/*
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/**
- * @file libavcodec/aandcttab.h
- * AAN (Arai Agui Nakajima) (I)DCT tables
- */
-
-#ifndef AVCODEC_AANDCTTAB_H
-#define AVCODEC_AANDCTTAB_H
-
-#include <stdint.h>
-
-extern const uint16_t ff_aanscales[64];
-extern const uint16_t ff_inv_aanscales[64];
-
-#endif /* AVCODEC_AANDCTTAB_H */

libavcodec/aasc.c

@@ -1,122 +0,0 @@
-/*
- * Autodesk RLE Decoder
- * Copyright (C) 2005 the ffmpeg project
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/**
- * @file libavcodec/aasc.c
- * Autodesk RLE Video Decoder by Konstantin Shishkov
- */
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-
-#include "avcodec.h"
-#include "dsputil.h"
-#include "msrledec.h"
-
-typedef struct AascContext {
-    AVCodecContext *avctx;
-    AVFrame frame;
-} AascContext;
-
-#define FETCH_NEXT_STREAM_BYTE() \
-    if (stream_ptr >= buf_size) \
-    { \
-      av_log(s->avctx, AV_LOG_ERROR, " AASC: stream ptr just went out of bounds (fetch)\n"); \
-      break; \
-    } \
-    stream_byte = buf[stream_ptr++];
-
-static av_cold int aasc_decode_init(AVCodecContext *avctx)
-{
-    AascContext *s = avctx->priv_data;
-
-    s->avctx = avctx;
-
-    avctx->pix_fmt = PIX_FMT_BGR24;
-    s->frame.data[0] = NULL;
-
-    return 0;
-}
-
-static int aasc_decode_frame(AVCodecContext *avctx,
-                              void *data, int *data_size,
-                              const uint8_t *buf, int buf_size)
-{
-    AascContext *s = avctx->priv_data;
-    int compr, i, stride;
-
-    s->frame.reference = 1;
-    s->frame.buffer_hints = FF_BUFFER_HINTS_VALID | FF_BUFFER_HINTS_PRESERVE | FF_BUFFER_HINTS_REUSABLE;
-    if (avctx->reget_buffer(avctx, &s->frame)) {
-        av_log(avctx, AV_LOG_ERROR, "reget_buffer() failed\n");
-        return -1;
-    }
-
-    compr = AV_RL32(buf);
-    buf += 4;
-    buf_size -= 4;
-    switch(compr){
-    case 0:
-        stride = (avctx->width * 3 + 3) & ~3;
-        for(i = avctx->height - 1; i >= 0; i--){
-            memcpy(s->frame.data[0] + i*s->frame.linesize[0], buf, avctx->width*3);
-            buf += stride;
-        }
-        break;
-    case 1:
-        ff_msrle_decode(avctx, (AVPicture*)&s->frame, 8, buf - 4, buf_size + 4);
-        break;
-    default:
-        av_log(avctx, AV_LOG_ERROR, "Unknown compression type %d\n", compr);
-        return -1;
-    }
-
-    *data_size = sizeof(AVFrame);
-    *(AVFrame*)data = s->frame;
-
-    /* report that the buffer was completely consumed */
-    return buf_size;
-}
-
-static av_cold int aasc_decode_end(AVCodecContext *avctx)
-{
-    AascContext *s = avctx->priv_data;
-
-    /* release the last frame */
-    if (s->frame.data[0])
-        avctx->release_buffer(avctx, &s->frame);
-
-    return 0;
-}
-
-AVCodec aasc_decoder = {
-    "aasc",
-    CODEC_TYPE_VIDEO,
-    CODEC_ID_AASC,
-    sizeof(AascContext),
-    aasc_decode_init,
-    NULL,
-    aasc_decode_end,
-    aasc_decode_frame,
-    CODEC_CAP_DR1,
-    .long_name = NULL_IF_CONFIG_SMALL("Autodesk RLE"),
-};

libavcodec/ac3.c

@@ -1,246 +0,0 @@
-/*
- * Common code between the AC-3 encoder and decoder
- * Copyright (c) 2000 Fabrice Bellard
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/**
- * @file libavcodec/ac3.c
- * Common code between the AC-3 encoder and decoder.
- */
-
-#include "avcodec.h"
-#include "ac3.h"
-#include "bitstream.h"
-
-static uint8_t band_start_tab[51];
-static uint8_t bin_to_band_tab[253];
-
-static inline int calc_lowcomp1(int a, int b0, int b1, int c)
-{
-    if ((b0 + 256) == b1) {
-        a = c;
-    } else if (b0 > b1) {
-        a = FFMAX(a - 64, 0);
-    }
-    return a;
-}
-
-static inline int calc_lowcomp(int a, int b0, int b1, int bin)
-{
-    if (bin < 7) {
-        return calc_lowcomp1(a, b0, b1, 384);
-    } else if (bin < 20) {
-        return calc_lowcomp1(a, b0, b1, 320);
-    } else {
-        return FFMAX(a - 128, 0);
-    }
-}
-
-void ff_ac3_bit_alloc_calc_psd(int8_t *exp, int start, int end, int16_t *psd,
-                               int16_t *band_psd)
-{
-    int bin, i, j, k, end1, v;
-
-    /* exponent mapping to PSD */
-    for(bin=start;bin<end;bin++) {
-        psd[bin]=(3072 - (exp[bin] << 7));
-    }
-
-    /* PSD integration */
-    j=start;
-    k=bin_to_band_tab[start];
-    do {
-        v=psd[j];
-        j++;
-        end1 = FFMIN(band_start_tab[k+1], end);
-        for(i=j;i<end1;i++) {
-            /* logadd */
-            int adr = FFMIN(FFABS(v - psd[j]) >> 1, 255);
-            v = FFMAX(v, psd[j]) + ff_ac3_log_add_tab[adr];
-            j++;
-        }
-        band_psd[k]=v;
-        k++;
-    } while (end > band_start_tab[k]);
-}
-
-int ff_ac3_bit_alloc_calc_mask(AC3BitAllocParameters *s, int16_t *band_psd,
-                               int start, int end, int fast_gain, int is_lfe,
-                               int dba_mode, int dba_nsegs, uint8_t *dba_offsets,
-                               uint8_t *dba_lengths, uint8_t *dba_values,
-                               int16_t *mask)
-{
-    int16_t excite[50]; /* excitation */
-    int bin, k;
-    int bndstrt, bndend, begin, end1, tmp;
-    int lowcomp, fastleak, slowleak;
-
-    /* excitation function */
-    bndstrt = bin_to_band_tab[start];
-    bndend = bin_to_band_tab[end-1] + 1;
-
-    if (bndstrt == 0) {
-        lowcomp = 0;
-        lowcomp = calc_lowcomp1(lowcomp, band_psd[0], band_psd[1], 384);
-        excite[0] = band_psd[0] - fast_gain - lowcomp;
-        lowcomp = calc_lowcomp1(lowcomp, band_psd[1], band_psd[2], 384);
-        excite[1] = band_psd[1] - fast_gain - lowcomp;
-        begin = 7;
-        for (bin = 2; bin < 7; bin++) {
-            if (!(is_lfe && bin == 6))
-                lowcomp = calc_lowcomp1(lowcomp, band_psd[bin], band_psd[bin+1], 384);
-            fastleak = band_psd[bin] - fast_gain;
-            slowleak = band_psd[bin] - s->slow_gain;
-            excite[bin] = fastleak - lowcomp;
-            if (!(is_lfe && bin == 6)) {
-                if (band_psd[bin] <= band_psd[bin+1]) {
-                    begin = bin + 1;
-                    break;
-                }
-            }
-        }
-
-        end1=bndend;
-        if (end1 > 22) end1=22;
-
-        for (bin = begin; bin < end1; bin++) {
-            if (!(is_lfe && bin == 6))
-                lowcomp = calc_lowcomp(lowcomp, band_psd[bin], band_psd[bin+1], bin);
-
-            fastleak = FFMAX(fastleak - s->fast_decay, band_psd[bin] - fast_gain);
-            slowleak = FFMAX(slowleak - s->slow_decay, band_psd[bin] - s->slow_gain);
-            excite[bin] = FFMAX(fastleak - lowcomp, slowleak);
-        }
-        begin = 22;
-    } else {
-        /* coupling channel */
-        begin = bndstrt;
-
-        fastleak = (s->cpl_fast_leak << 8) + 768;
-        slowleak = (s->cpl_slow_leak << 8) + 768;
-    }
-
-    for (bin = begin; bin < bndend; bin++) {
-        fastleak = FFMAX(fastleak - s->fast_decay, band_psd[bin] - fast_gain);
-        slowleak = FFMAX(slowleak - s->slow_decay, band_psd[bin] - s->slow_gain);
-        excite[bin] = FFMAX(fastleak, slowleak);
-    }
-
-    /* compute masking curve */
-
-    for (bin = bndstrt; bin < bndend; bin++) {
-        tmp = s->db_per_bit - band_psd[bin];
-        if (tmp > 0) {
-            excite[bin] += tmp >> 2;
-        }
-        mask[bin] = FFMAX(ff_ac3_hearing_threshold_tab[bin >> s->sr_shift][s->sr_code], excite[bin]);
-    }
-
-    /* delta bit allocation */
-
-    if (dba_mode == DBA_REUSE || dba_mode == DBA_NEW) {
-        int band, seg, delta;
-        if (dba_nsegs >= 8)
-            return -1;
-        band = 0;
-        for (seg = 0; seg < dba_nsegs; seg++) {
-            band += dba_offsets[seg];
-            if (band >= 50 || dba_lengths[seg] > 50-band)
-                return -1;
-            if (dba_values[seg] >= 4) {
-                delta = (dba_values[seg] - 3) << 7;
-            } else {
-                delta = (dba_values[seg] - 4) << 7;
-            }
-            for (k = 0; k < dba_lengths[seg]; k++) {
-                mask[band] += delta;
-                band++;
-            }
-        }
-    }
-    return 0;
-}
-
-void ff_ac3_bit_alloc_calc_bap(int16_t *mask, int16_t *psd, int start, int end,
-                               int snr_offset, int floor,
-                               const uint8_t *bap_tab, uint8_t *bap)
-{
-    int i, j, k, end1, v, address;
-
-    /* special case, if snr offset is -960, set all bap's to zero */
-    if(snr_offset == -960) {
-        memset(bap, 0, 256);
-        return;
-    }
-
-    i = start;
-    j = bin_to_band_tab[start];
-    do {
-        v = (FFMAX(mask[j] - snr_offset - floor, 0) & 0x1FE0) + floor;
-        end1 = FFMIN(band_start_tab[j] + ff_ac3_critical_band_size_tab[j], end);
-        for (k = i; k < end1; k++) {
-            address = av_clip((psd[i] - v) >> 5, 0, 63);
-            bap[i] = bap_tab[address];
-            i++;
-        }
-    } while (end > band_start_tab[j++]);
-}
-
-/* AC-3 bit allocation. The algorithm is the one described in the AC-3
-   spec. */
-void ac3_parametric_bit_allocation(AC3BitAllocParameters *s, uint8_t *bap,
-                                   int8_t *exp, int start, int end,
-                                   int snr_offset, int fast_gain, int is_lfe,
-                                   int dba_mode, int dba_nsegs,
-                                   uint8_t *dba_offsets, uint8_t *dba_lengths,
-                                   uint8_t *dba_values)
-{
-    int16_t psd[256];   /* scaled exponents */
-    int16_t band_psd[50]; /* interpolated exponents */
-    int16_t mask[50];   /* masking value */
-
-    ff_ac3_bit_alloc_calc_psd(exp, start, end, psd, band_psd);
-
-    ff_ac3_bit_alloc_calc_mask(s, band_psd, start, end, fast_gain, is_lfe,
-                               dba_mode, dba_nsegs, dba_offsets, dba_lengths, dba_values,
-                               mask);
-
-    ff_ac3_bit_alloc_calc_bap(mask, psd, start, end, snr_offset, s->floor,
-                              ff_ac3_bap_tab, bap);
-}
-
-/**
- * Initializes some tables.
- * note: This function must remain thread safe because it is called by the
- *       AVParser init code.
- */
-av_cold void ac3_common_init(void)
-{
-    int i, j, k, l, v;
-    /* compute bndtab and masktab from bandsz */
-    k = 0;
-    l = 0;
-    for(i=0;i<50;i++) {
-        band_start_tab[i] = l;
-        v = ff_ac3_critical_band_size_tab[i];
-        for(j=0;j<v;j++) bin_to_band_tab[k++]=i;
-        l += v;
-    }
-    band_start_tab[50] = l;
-}

libavcodec/ac3.h

@@ -1,186 +0,0 @@
-/*
- * Common code between the AC-3 encoder and decoder
- * Copyright (c) 2000, 2001, 2002 Fabrice Bellard
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/**
- * @file libavcodec/ac3.h
- * Common code between the AC-3 encoder and decoder.
- */
-
-#ifndef AVCODEC_AC3_H
-#define AVCODEC_AC3_H
-
-#include "ac3tab.h"
-
-#define AC3_MAX_CODED_FRAME_SIZE 3840 /* in bytes */
-#define AC3_MAX_CHANNELS 6 /* including LFE channel */
-
-#define NB_BLOCKS 6 /* number of PCM blocks inside an AC-3 frame */
-#define AC3_FRAME_SIZE (NB_BLOCKS * 256)
-
-/* exponent encoding strategy */
-#define EXP_REUSE 0
-#define EXP_NEW   1
-
-#define EXP_D15   1
-#define EXP_D25   2
-#define EXP_D45   3
-
-/** Delta bit allocation strategy */
-typedef enum {
-    DBA_REUSE = 0,
-    DBA_NEW,
-    DBA_NONE,
-    DBA_RESERVED
-} AC3DeltaStrategy;
-
-/** Channel mode (audio coding mode) */
-typedef enum {
-    AC3_CHMODE_DUALMONO = 0,
-    AC3_CHMODE_MONO,
-    AC3_CHMODE_STEREO,
-    AC3_CHMODE_3F,
-    AC3_CHMODE_2F1R,
-    AC3_CHMODE_3F1R,
-    AC3_CHMODE_2F2R,
-    AC3_CHMODE_3F2R
-} AC3ChannelMode;
-
-typedef struct AC3BitAllocParameters {
-    int sr_code;
-    int sr_shift;
-    int slow_gain, slow_decay, fast_decay, db_per_bit, floor;
-    int cpl_fast_leak, cpl_slow_leak;
-} AC3BitAllocParameters;
-
-/**
- * @struct AC3HeaderInfo
- * Coded AC-3 header values up to the lfeon element, plus derived values.
- */
-typedef struct {
-    /** @defgroup coded Coded elements
-     * @{
-     */
-    uint16_t sync_word;
-    uint16_t crc1;
-    uint8_t sr_code;
-    uint8_t bitstream_id;
-    uint8_t channel_mode;
-    uint8_t lfe_on;
-    uint8_t frame_type;
-    int substreamid;                        ///< substream identification
-    int center_mix_level;                   ///< Center mix level index
-    int surround_mix_level;                 ///< Surround mix level index
-    uint16_t channel_map;
-    int num_blocks;                         ///< number of audio blocks
-    /** @} */
-
-    /** @defgroup derived Derived values
-     * @{
-     */
-    uint8_t sr_shift;
-    uint16_t sample_rate;
-    uint32_t bit_rate;
-    uint8_t channels;
-    uint16_t frame_size;
-    /** @} */
-} AC3HeaderInfo;
-
-typedef enum {
-    EAC3_FRAME_TYPE_INDEPENDENT = 0,
-    EAC3_FRAME_TYPE_DEPENDENT,
-    EAC3_FRAME_TYPE_AC3_CONVERT,
-    EAC3_FRAME_TYPE_RESERVED
-} EAC3FrameType;
-
-void ac3_common_init(void);
-
-/**
- * Calculates the log power-spectral density of the input signal.
- * This gives a rough estimate of signal power in the frequency domain by using
- * the spectral envelope (exponents).  The psd is also separately grouped
- * into critical bands for use in the calculating the masking curve.
- * 128 units in psd = -6 dB.  The dbknee parameter in AC3BitAllocParameters
- * determines the reference level.
- *
- * @param[in]  exp        frequency coefficient exponents
- * @param[in]  start      starting bin location
- * @param[in]  end        ending bin location
- * @param[out] psd        signal power for each frequency bin
- * @param[out] band_psd   signal power for each critical band
- */
-void ff_ac3_bit_alloc_calc_psd(int8_t *exp, int start, int end, int16_t *psd,
-                               int16_t *band_psd);
-
-/**
- * Calculates the masking curve.
- * First, the excitation is calculated using parameters in \p s and the signal
- * power in each critical band.  The excitation is compared with a predefined
- * hearing threshold table to produce the masking curve.  If delta bit
- * allocation information is provided, it is used for adjusting the masking
- * curve, usually to give a closer match to a better psychoacoustic model.
- *
- * @param[in]  s            adjustable bit allocation parameters
- * @param[in]  band_psd     signal power for each critical band
- * @param[in]  start        starting bin location
- * @param[in]  end          ending bin location
- * @param[in]  fast_gain    fast gain (estimated signal-to-mask ratio)
- * @param[in]  is_lfe       whether or not the channel being processed is the LFE
- * @param[in]  dba_mode     delta bit allocation mode (none, reuse, or new)
- * @param[in]  dba_nsegs    number of delta segments
- * @param[in]  dba_offsets  location offsets for each segment
- * @param[in]  dba_lengths  length of each segment
- * @param[in]  dba_values   delta bit allocation for each segment
- * @param[out] mask         calculated masking curve
- * @return returns 0 for success, non-zero for error
- */
-int ff_ac3_bit_alloc_calc_mask(AC3BitAllocParameters *s, int16_t *band_psd,
-                               int start, int end, int fast_gain, int is_lfe,
-                               int dba_mode, int dba_nsegs, uint8_t *dba_offsets,
-                               uint8_t *dba_lengths, uint8_t *dba_values,
-                               int16_t *mask);
-
-/**
- * Calculates bit allocation pointers.
- * The SNR is the difference between the masking curve and the signal.  AC-3
- * uses this value for each frequency bin to allocate bits.  The \p snroffset
- * parameter is a global adjustment to the SNR for all bins.
- *
- * @param[in]  mask       masking curve
- * @param[in]  psd        signal power for each frequency bin
- * @param[in]  start      starting bin location
- * @param[in]  end        ending bin location
- * @param[in]  snr_offset SNR adjustment
- * @param[in]  floor      noise floor
- * @param[in]  bap_tab    look-up table for bit allocation pointers
- * @param[out] bap        bit allocation pointers
- */
-void ff_ac3_bit_alloc_calc_bap(int16_t *mask, int16_t *psd, int start, int end,
-                               int snr_offset, int floor,
-                               const uint8_t *bap_tab, uint8_t *bap);
-
-void ac3_parametric_bit_allocation(AC3BitAllocParameters *s, uint8_t *bap,
-                                   int8_t *exp, int start, int end,
-                                   int snr_offset, int fast_gain, int is_lfe,
-                                   int dba_mode, int dba_nsegs,
-                                   uint8_t *dba_offsets, uint8_t *dba_lengths,
-                                   uint8_t *dba_values);
-
-#endif /* AVCODEC_AC3_H */

libavcodec/ac3_parser.c

@@ -1,203 +0,0 @@
-/*
- * AC-3 parser
- * Copyright (c) 2003 Fabrice Bellard
- * Copyright (c) 2003 Michael Niedermayer
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#include "parser.h"
-#include "ac3_parser.h"
-#include "aac_ac3_parser.h"
-#include "bitstream.h"
-
-
-#define AC3_HEADER_SIZE 7
-
-
-static const uint8_t eac3_blocks[4] = {
-    1, 2, 3, 6
-};
-
-
-int ff_ac3_parse_header(GetBitContext *gbc, AC3HeaderInfo *hdr)
-{
-    int frame_size_code;
-
-    memset(hdr, 0, sizeof(*hdr));
-
-    hdr->sync_word = get_bits(gbc, 16);
-    if(hdr->sync_word != 0x0B77)
-        return AAC_AC3_PARSE_ERROR_SYNC;
-
-    /* read ahead to bsid to distinguish between AC-3 and E-AC-3 */
-    hdr->bitstream_id = show_bits_long(gbc, 29) & 0x1F;
-    if(hdr->bitstream_id > 16)
-        return AAC_AC3_PARSE_ERROR_BSID;
-
-    hdr->num_blocks = 6;
-
-    /* set default mix levels */
-    hdr->center_mix_level   = 1;  // -4.5dB
-    hdr->surround_mix_level = 1;  // -6.0dB
-
-    if(hdr->bitstream_id <= 10) {
-        /* Normal AC-3 */
-        hdr->crc1 = get_bits(gbc, 16);
-        hdr->sr_code = get_bits(gbc, 2);
-        if(hdr->sr_code == 3)
-            return AAC_AC3_PARSE_ERROR_SAMPLE_RATE;
-
-        frame_size_code = get_bits(gbc, 6);
-        if(frame_size_code > 37)
-            return AAC_AC3_PARSE_ERROR_FRAME_SIZE;
-
-        skip_bits(gbc, 5); // skip bsid, already got it
-
-        skip_bits(gbc, 3); // skip bitstream mode
-        hdr->channel_mode = get_bits(gbc, 3);
-
-        if(hdr->channel_mode == AC3_CHMODE_STEREO) {
-            skip_bits(gbc, 2); // skip dsurmod
-        } else {
-            if((hdr->channel_mode & 1) && hdr->channel_mode != AC3_CHMODE_MONO)
-                hdr->center_mix_level = get_bits(gbc, 2);
-            if(hdr->channel_mode & 4)
-                hdr->surround_mix_level = get_bits(gbc, 2);
-        }
-        hdr->lfe_on = get_bits1(gbc);
-
-        hdr->sr_shift = FFMAX(hdr->bitstream_id, 8) - 8;
-        hdr->sample_rate = ff_ac3_sample_rate_tab[hdr->sr_code] >> hdr->sr_shift;
-        hdr->bit_rate = (ff_ac3_bitrate_tab[frame_size_code>>1] * 1000) >> hdr->sr_shift;
-        hdr->channels = ff_ac3_channels_tab[hdr->channel_mode] + hdr->lfe_on;
-        hdr->frame_size = ff_ac3_frame_size_tab[frame_size_code][hdr->sr_code] * 2;
-        hdr->frame_type = EAC3_FRAME_TYPE_AC3_CONVERT; //EAC3_FRAME_TYPE_INDEPENDENT;
-        hdr->substreamid = 0;
-    } else {
-        /* Enhanced AC-3 */
-        hdr->crc1 = 0;
-        hdr->frame_type = get_bits(gbc, 2);
-        if(hdr->frame_type == EAC3_FRAME_TYPE_RESERVED)
-            return AAC_AC3_PARSE_ERROR_FRAME_TYPE;
-
-        hdr->substreamid = get_bits(gbc, 3);
-
-        hdr->frame_size = (get_bits(gbc, 11) + 1) << 1;
-        if(hdr->frame_size < AC3_HEADER_SIZE)
-            return AAC_AC3_PARSE_ERROR_FRAME_SIZE;
-
-        hdr->sr_code = get_bits(gbc, 2);
-        if (hdr->sr_code == 3) {
-            int sr_code2 = get_bits(gbc, 2);
-            if(sr_code2 == 3)
-                return AAC_AC3_PARSE_ERROR_SAMPLE_RATE;
-            hdr->sample_rate = ff_ac3_sample_rate_tab[sr_code2] / 2;
-            hdr->sr_shift = 1;
-        } else {
-            hdr->num_blocks = eac3_blocks[get_bits(gbc, 2)];
-            hdr->sample_rate = ff_ac3_sample_rate_tab[hdr->sr_code];
-            hdr->sr_shift = 0;
-        }
-
-        hdr->channel_mode = get_bits(gbc, 3);
-        hdr->lfe_on = get_bits1(gbc);
-
-        hdr->bit_rate = (uint32_t)(8.0 * hdr->frame_size * hdr->sample_rate /
-                        (hdr->num_blocks * 256.0));
-        hdr->channels = ff_ac3_channels_tab[hdr->channel_mode] + hdr->lfe_on;
-    }
-
-    return 0;
-}
-
-int ff_ac3_parse_header_full(GetBitContext *gbc, AC3HeaderInfo *hdr){
-    int ret, i;
-    ret = ff_ac3_parse_header(gbc, hdr);
-    if(!ret){
-        if(hdr->bitstream_id>10){
-            /* Enhanced AC-3 */
-            skip_bits(gbc, 5); // skip bitstream id
-
-            /* skip dialog normalization and compression gain */
-            for (i = 0; i < (hdr->channel_mode ? 1 : 2); i++) {
-                skip_bits(gbc, 5); // skip dialog normalization
-                if (get_bits1(gbc)) {
-                    skip_bits(gbc, 8); //skip Compression gain word
-                }
-            }
-            /* dependent stream channel map */
-            if (hdr->frame_type == EAC3_FRAME_TYPE_DEPENDENT && get_bits1(gbc)) {
-                    hdr->channel_map = get_bits(gbc, 16); //custom channel map
-                    return 0;
-            }
-        }
-        //default channel map based on acmod and lfeon
-        hdr->channel_map = ff_eac3_default_chmap[hdr->channel_mode];
-        if(hdr->lfe_on)
-            hdr->channel_map |= AC3_CHMAP_LFE;
-    }
-    return ret;
-}
-
-static int ac3_sync(uint64_t state, AACAC3ParseContext *hdr_info,
-        int *need_next_header, int *new_frame_start)
-{
-    int err;
-    union {
-        uint64_t u64;
-        uint8_t  u8[8];
-    } tmp = { be2me_64(state) };
-    AC3HeaderInfo hdr;
-    GetBitContext gbc;
-
-    init_get_bits(&gbc, tmp.u8+8-AC3_HEADER_SIZE, 54);
-    err = ff_ac3_parse_header(&gbc, &hdr);
-
-    if(err < 0)
-        return 0;
-
-    hdr_info->sample_rate = hdr.sample_rate;
-    hdr_info->bit_rate = hdr.bit_rate;
-    hdr_info->channels = hdr.channels;
-    hdr_info->samples = hdr.num_blocks * 256;
-    if(hdr.bitstream_id>10)
-        hdr_info->codec_id = CODEC_ID_EAC3;
-    else
-        hdr_info->codec_id = CODEC_ID_AC3;
-
-    *need_next_header = (hdr.frame_type != EAC3_FRAME_TYPE_AC3_CONVERT);
-    *new_frame_start  = (hdr.frame_type != EAC3_FRAME_TYPE_DEPENDENT);
-    return hdr.frame_size;
-}
-
-static av_cold int ac3_parse_init(AVCodecParserContext *s1)
-{
-    AACAC3ParseContext *s = s1->priv_data;
-    s->header_size = AC3_HEADER_SIZE;
-    s->sync = ac3_sync;
-    return 0;
-}
-
-
-AVCodecParser ac3_parser = {
-    { CODEC_ID_AC3, CODEC_ID_EAC3 },
-    sizeof(AACAC3ParseContext),
-    ac3_parse_init,
-    ff_aac_ac3_parse,
-    ff_parse_close,
-};

libavcodec/ac3_parser.h

@@ -1,52 +0,0 @@
-/*
- * AC-3 parser prototypes
- * Copyright (c) 2003 Fabrice Bellard
- * Copyright (c) 2003 Michael Niedermayer
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#ifndef AVCODEC_AC3_PARSER_H
-#define AVCODEC_AC3_PARSER_H
-
-#include "ac3.h"
-#include "bitstream.h"
-
-/**
- * Parses AC-3 frame header.
- * Parses the header up to the lfeon element, which is the first 52 or 54 bits
- * depending on the audio coding mode.
- * @param gbc[in] BitContext containing the first 54 bits of the frame.
- * @param hdr[out] Pointer to struct where header info is written.
- * @return Returns 0 on success, -1 if there is a sync word mismatch,
- * -2 if the bsid (version) element is invalid, -3 if the fscod (sample rate)
- * element is invalid, or -4 if the frmsizecod (bit rate) element is invalid.
- */
-int ff_ac3_parse_header(GetBitContext *gbc, AC3HeaderInfo *hdr);
-
-/**
- * Parses AC-3 frame header and sets channel_map
- * Parses the header up to the lfeon (channel_map in E-AC-3)
- * element, which is the first 52, 54 or 104 bits depending
- * on the audio coding mode.
- * @param gbc[in] BitContext containing the first 54 bits of the frame.
- * @param hdr[out] Pointer to struct where header info is written.
- * @return value returned by ff_ac3_parse_header
- */
-int ff_ac3_parse_header_full(GetBitContext *gbc, AC3HeaderInfo *hdr);
-
-#endif /* AVCODEC_AC3_PARSER_H */

libavcodec/ac3dec.h

@@ -1,183 +0,0 @@
-/*
- * Common code between the AC-3 and E-AC-3 decoders
- * Copyright (c) 2007 Bartlomiej Wolowiec <bartek.wolowiec@gmail.com>
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/**
- * @file libavcodec/ac3.h
- * Common code between the AC-3 and E-AC-3 decoders.
- */
-
-#ifndef AVCODEC_AC3DEC_H
-#define AVCODEC_AC3DEC_H
-
-#include "libavutil/internal.h"
-#include "libavutil/lfg.h"
-#include "ac3.h"
-#include "bitstream.h"
-#include "dsputil.h"
-
-/* override ac3.h to include coupling channel */
-#undef AC3_MAX_CHANNELS
-#define AC3_MAX_CHANNELS 7
-#define CPL_CH 0
-
-#define AC3_OUTPUT_LFEON  8
-
-#define AC3_MAX_COEFS   256
-#define AC3_BLOCK_SIZE  256
-#define MAX_BLOCKS        6
-
-typedef struct {
-    AVCodecContext *avctx;                  ///< parent context
-    GetBitContext gbc;                      ///< bitstream reader
-    uint8_t *input_buffer;                  ///< temp buffer to prevent overread
-
-///@defgroup bsi bit stream information
-///@{
-    int frame_type;                         ///< frame type                             (strmtyp)
-    int substreamid;                        ///< substream identification
-    int frame_size;                         ///< current frame size, in bytes
-    int bit_rate;                           ///< stream bit rate, in bits-per-second
-    int sample_rate;                        ///< sample frequency, in Hz
-    int num_blocks;                         ///< number of audio blocks
-    int channel_mode;                       ///< channel mode                           (acmod)
-    int lfe_on;                             ///< lfe channel in use
-    int channel_map;                        ///< custom channel map
-    int center_mix_level;                   ///< Center mix level index
-    int surround_mix_level;                 ///< Surround mix level index
-    int eac3;                               ///< indicates if current frame is E-AC-3
-///@}
-
-///@defgroup audfrm frame syntax parameters
-    int snr_offset_strategy;                ///< SNR offset strategy                    (snroffststr)
-    int block_switch_syntax;                ///< block switch syntax enabled            (blkswe)
-    int dither_flag_syntax;                 ///< dither flag syntax enabled             (dithflage)
-    int bit_allocation_syntax;              ///< bit allocation model syntax enabled    (bamode)
-    int fast_gain_syntax;                   ///< fast gain codes enabled                (frmfgaincode)
-    int dba_syntax;                         ///< delta bit allocation syntax enabled    (dbaflde)
-    int skip_syntax;                        ///< skip field syntax enabled              (skipflde)
- ///@}
-
-///@defgroup cpl standard coupling
-    int cpl_in_use[MAX_BLOCKS];             ///< coupling in use                        (cplinu)
-    int cpl_strategy_exists[MAX_BLOCKS];    ///< coupling strategy exists               (cplstre)
-    int channel_in_cpl[AC3_MAX_CHANNELS];   ///< channel in coupling                    (chincpl)
-    int phase_flags_in_use;                 ///< phase flags in use                     (phsflginu)
-    int phase_flags[18];                    ///< phase flags                            (phsflg)
-    int num_cpl_subbands;                   ///< number of coupling sub bands           (ncplsubnd)
-    int num_cpl_bands;                      ///< number of coupling bands               (ncplbnd)
-    uint8_t cpl_band_struct[18];            ///< coupling band structure                (cplbndstrc)
-    int firstchincpl;                       ///< first channel in coupling
-    int first_cpl_coords[AC3_MAX_CHANNELS]; ///< first coupling coordinates states      (firstcplcos)
-    int cpl_coords[AC3_MAX_CHANNELS][18];   ///< coupling coordinates                   (cplco)
-///@}
-
-///@defgroup aht adaptive hybrid transform
-    int channel_uses_aht[AC3_MAX_CHANNELS];                         ///< channel AHT in use (chahtinu)
-    int pre_mantissa[AC3_MAX_CHANNELS][AC3_MAX_COEFS][MAX_BLOCKS];  ///< pre-IDCT mantissas
-///@}
-
-///@defgroup channel channel
-    int fbw_channels;                           ///< number of full-bandwidth channels
-    int channels;                               ///< number of total channels
-    int lfe_ch;                                 ///< index of LFE channel
-    float downmix_coeffs[AC3_MAX_CHANNELS][2];  ///< stereo downmix coefficients
-    int downmixed;                              ///< indicates if coeffs are currently downmixed
-    int output_mode;                            ///< output channel configuration
-    int out_channels;                           ///< number of output channels
-///@}
-
-///@defgroup dynrng dynamic range
-    float dynamic_range[2];                 ///< dynamic range
-///@}
-
-///@defgroup bandwidth bandwidth
-    int start_freq[AC3_MAX_CHANNELS];       ///< start frequency bin                    (strtmant)
-    int end_freq[AC3_MAX_CHANNELS];         ///< end frequency bin                      (endmant)
-///@}
-
-///@defgroup rematrixing rematrixing
-    int num_rematrixing_bands;              ///< number of rematrixing bands            (nrematbnd)
-    int rematrixing_flags[4];               ///< rematrixing flags                      (rematflg)
-///@}
-
-///@defgroup exponents exponents
-    int num_exp_groups[AC3_MAX_CHANNELS];           ///< Number of exponent groups      (nexpgrp)
-    int8_t dexps[AC3_MAX_CHANNELS][AC3_MAX_COEFS];  ///< decoded exponents
-    int exp_strategy[MAX_BLOCKS][AC3_MAX_CHANNELS]; ///< exponent strategies            (expstr)
-///@}
-
-///@defgroup bitalloc bit allocation
-    AC3BitAllocParameters bit_alloc_params;         ///< bit allocation parameters
-    int first_cpl_leak;                             ///< first coupling leak state      (firstcplleak)
-    int snr_offset[AC3_MAX_CHANNELS];               ///< signal-to-noise ratio offsets  (snroffst)
-    int fast_gain[AC3_MAX_CHANNELS];                ///< fast gain values/SMR's         (fgain)
-    uint8_t bap[AC3_MAX_CHANNELS][AC3_MAX_COEFS];   ///< bit allocation pointers
-    int16_t psd[AC3_MAX_CHANNELS][AC3_MAX_COEFS];   ///< scaled exponents
-    int16_t band_psd[AC3_MAX_CHANNELS][50];         ///< interpolated exponents
-    int16_t mask[AC3_MAX_CHANNELS][50];             ///< masking curve values
-    int dba_mode[AC3_MAX_CHANNELS];                 ///< delta bit allocation mode
-    int dba_nsegs[AC3_MAX_CHANNELS];                ///< number of delta segments
-    uint8_t dba_offsets[AC3_MAX_CHANNELS][8];       ///< delta segment offsets
-    uint8_t dba_lengths[AC3_MAX_CHANNELS][8];       ///< delta segment lengths
-    uint8_t dba_values[AC3_MAX_CHANNELS][8];        ///< delta values for each segment
-///@}
-
-///@defgroup dithering zero-mantissa dithering
-    int dither_flag[AC3_MAX_CHANNELS];      ///< dither flags                           (dithflg)
-    AVLFG dith_state;                       ///< for dither generation
-///@}
-
-///@defgroup imdct IMDCT
-    int block_switch[AC3_MAX_CHANNELS];     ///< block switch flags                     (blksw)
-    MDCTContext imdct_512;                  ///< for 512 sample IMDCT
-    MDCTContext imdct_256;                  ///< for 256 sample IMDCT
-///@}
-
-///@defgroup opt optimization
-    DSPContext dsp;                         ///< for optimization
-    float add_bias;                         ///< offset for float_to_int16 conversion
-    float mul_bias;                         ///< scaling for float_to_int16 conversion
-///@}
-
-    DECLARE_ALIGNED_16(int, fixed_coeffs[AC3_MAX_CHANNELS][AC3_MAX_COEFS]);  ///> fixed-point transform coefficients
-
-///@defgroup arrays aligned arrays
-    DECLARE_ALIGNED_16(float, transform_coeffs[AC3_MAX_CHANNELS][AC3_MAX_COEFS]);   ///< transform coefficients
-    DECLARE_ALIGNED_16(float, delay[AC3_MAX_CHANNELS][AC3_BLOCK_SIZE]);             ///< delay - added to the next block
-    DECLARE_ALIGNED_16(float, window[AC3_BLOCK_SIZE]);                              ///< window coefficients
-    DECLARE_ALIGNED_16(float, tmp_output[AC3_BLOCK_SIZE]);                          ///< temporary storage for output before windowing
-    DECLARE_ALIGNED_16(float, output[AC3_MAX_CHANNELS][AC3_BLOCK_SIZE]);            ///< output after imdct transform and windowing
-///@}
-} AC3DecodeContext;
-
-/**
- * Parse the E-AC-3 frame header.
- * This parses both the bit stream info and audio frame header.
- */
-int ff_eac3_parse_header(AC3DecodeContext *s);
-
-/**
- * Decode mantissas in a single channel for the entire frame.
- * This is used when AHT mode is enabled.
- */
-void ff_eac3_decode_transform_coeffs_aht_ch(AC3DecodeContext *s, int ch);
-
-#endif /* AVCODEC_AC3DEC_H */

libavcodec/ac3dec_data.h

@@ -1,40 +0,0 @@
-/*
- * AC-3 and E-AC-3 decoder tables
- * Copyright (c) 2007 Bartlomiej Wolowiec <bartek.wolowiec@gmail.com>
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#ifndef AVCODEC_AC3DEC_DATA_H
-#define AVCODEC_AC3DEC_DATA_H
-
-#include "libavutil/common.h"
-
-extern const uint8_t ff_ac3_ungroup_3_in_5_bits_tab[32][3];
-extern const uint8_t ff_eac3_hebap_tab[64];
-extern const uint8_t ff_eac3_bits_vs_hebap[20];
-extern const int16_t ff_eac3_gaq_remap_1[12];
-extern const int16_t ff_eac3_gaq_remap_2_4_a[9][2];
-extern const int8_t  ff_eac3_gaq_remap_2_4_b[9][2];
-
-extern const int16_t (* const ff_eac3_mantissa_vq[8])[6];
-extern const uint8_t ff_eac3_frm_expstr[32][6];
-extern const uint8_t ff_eac3_default_cpl_band_struct[18];
-
-extern const uint8_t ff_ac3_rematrix_band_tab[5];
-
-#endif /* AVCODEC_AC3DEC_DATA_H */

libavcodec/ac3tab.c

@@ -1,262 +0,0 @@
-/*
- * AC-3 tables
- * copyright (c) 2001 Fabrice Bellard
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/**
- * @file libavcodec/ac3tab.c
- * tables taken directly from the AC-3 spec.
- */
-
-#include "ac3tab.h"
-
-/**
- * Possible frame sizes.
- * from ATSC A/52 Table 5.18 Frame Size Code Table.
- */
-const uint16_t ff_ac3_frame_size_tab[38][3] = {
-    { 64,   69,   96   },
-    { 64,   70,   96   },
-    { 80,   87,   120  },
-    { 80,   88,   120  },
-    { 96,   104,  144  },
-    { 96,   105,  144  },
-    { 112,  121,  168  },
-    { 112,  122,  168  },
-    { 128,  139,  192  },
-    { 128,  140,  192  },
-    { 160,  174,  240  },
-    { 160,  175,  240  },
-    { 192,  208,  288  },
-    { 192,  209,  288  },
-    { 224,  243,  336  },
-    { 224,  244,  336  },
-    { 256,  278,  384  },
-    { 256,  279,  384  },
-    { 320,  348,  480  },
-    { 320,  349,  480  },
-    { 384,  417,  576  },
-    { 384,  418,  576  },
-    { 448,  487,  672  },
-    { 448,  488,  672  },
-    { 512,  557,  768  },
-    { 512,  558,  768  },
-    { 640,  696,  960  },
-    { 640,  697,  960  },
-    { 768,  835,  1152 },
-    { 768,  836,  1152 },
-    { 896,  975,  1344 },
-    { 896,  976,  1344 },
-    { 1024, 1114, 1536 },
-    { 1024, 1115, 1536 },
-    { 1152, 1253, 1728 },
-    { 1152, 1254, 1728 },
-    { 1280, 1393, 1920 },
-    { 1280, 1394, 1920 },
-};
-
-/**
- * Maps audio coding mode (acmod) to number of full-bandwidth channels.
- * from ATSC A/52 Table 5.8 Audio Coding Mode
- */
-const uint8_t ff_ac3_channels_tab[8] = {
-    2, 1, 2, 3, 3, 4, 4, 5
-};
-
-/* possible frequencies */
-const uint16_t ff_ac3_sample_rate_tab[3] = { 48000, 44100, 32000 };
-
-/* possible bitrates */
-const uint16_t ff_ac3_bitrate_tab[19] = {
-    32, 40, 48, 56, 64, 80, 96, 112, 128,
-    160, 192, 224, 256, 320, 384, 448, 512, 576, 640
-};
-
-/* AC-3 MDCT window */
-
-/* MDCT window */
-const int16_t ff_ac3_window[256] = {
-    4,    7,   12,   16,   21,   28,   34,   42,
-   51,   61,   72,   84,   97,  111,  127,  145,
-  164,  184,  207,  231,  257,  285,  315,  347,
-  382,  419,  458,  500,  544,  591,  641,  694,
-  750,  810,  872,  937, 1007, 1079, 1155, 1235,
- 1318, 1406, 1497, 1593, 1692, 1796, 1903, 2016,
- 2132, 2253, 2379, 2509, 2644, 2783, 2927, 3076,
- 3230, 3389, 3552, 3721, 3894, 4072, 4255, 4444,
- 4637, 4835, 5038, 5246, 5459, 5677, 5899, 6127,
- 6359, 6596, 6837, 7083, 7334, 7589, 7848, 8112,
- 8380, 8652, 8927, 9207, 9491, 9778,10069,10363,
-10660,10960,11264,11570,11879,12190,12504,12820,
-13138,13458,13780,14103,14427,14753,15079,15407,
-15735,16063,16392,16720,17049,17377,17705,18032,
-18358,18683,19007,19330,19651,19970,20287,20602,
-20914,21225,21532,21837,22139,22438,22733,23025,
-23314,23599,23880,24157,24430,24699,24964,25225,
-25481,25732,25979,26221,26459,26691,26919,27142,
-27359,27572,27780,27983,28180,28373,28560,28742,
-28919,29091,29258,29420,29577,29729,29876,30018,
-30155,30288,30415,30538,30657,30771,30880,30985,
-31086,31182,31274,31363,31447,31528,31605,31678,
-31747,31814,31877,31936,31993,32046,32097,32145,
-32190,32232,32272,32310,32345,32378,32409,32438,
-32465,32490,32513,32535,32556,32574,32592,32608,
-32623,32636,32649,32661,32671,32681,32690,32698,
-32705,32712,32718,32724,32729,32733,32737,32741,
-32744,32747,32750,32752,32754,32756,32757,32759,
-32760,32761,32762,32763,32764,32764,32765,32765,
-32766,32766,32766,32766,32767,32767,32767,32767,
-32767,32767,32767,32767,32767,32767,32767,32767,
-32767,32767,32767,32767,32767,32767,32767,32767,
-};
-
-const uint8_t ff_ac3_log_add_tab[260]= {
-0x40,0x3f,0x3e,0x3d,0x3c,0x3b,0x3a,0x39,0x38,0x37,
-0x36,0x35,0x34,0x34,0x33,0x32,0x31,0x30,0x2f,0x2f,
-0x2e,0x2d,0x2c,0x2c,0x2b,0x2a,0x29,0x29,0x28,0x27,
-0x26,0x26,0x25,0x24,0x24,0x23,0x23,0x22,0x21,0x21,
-0x20,0x20,0x1f,0x1e,0x1e,0x1d,0x1d,0x1c,0x1c,0x1b,
-0x1b,0x1a,0x1a,0x19,0x19,0x18,0x18,0x17,0x17,0x16,
-0x16,0x15,0x15,0x15,0x14,0x14,0x13,0x13,0x13,0x12,
-0x12,0x12,0x11,0x11,0x11,0x10,0x10,0x10,0x0f,0x0f,
-0x0f,0x0e,0x0e,0x0e,0x0d,0x0d,0x0d,0x0d,0x0c,0x0c,
-0x0c,0x0c,0x0b,0x0b,0x0b,0x0b,0x0a,0x0a,0x0a,0x0a,
-0x0a,0x09,0x09,0x09,0x09,0x09,0x08,0x08,0x08,0x08,
-0x08,0x08,0x07,0x07,0x07,0x07,0x07,0x07,0x06,0x06,
-0x06,0x06,0x06,0x06,0x06,0x06,0x05,0x05,0x05,0x05,
-0x05,0x05,0x05,0x05,0x04,0x04,0x04,0x04,0x04,0x04,
-0x04,0x04,0x04,0x04,0x04,0x03,0x03,0x03,0x03,0x03,
-0x03,0x03,0x03,0x03,0x03,0x03,0x03,0x03,0x03,0x02,
-0x02,0x02,0x02,0x02,0x02,0x02,0x02,0x02,0x02,0x02,
-0x02,0x02,0x02,0x02,0x02,0x02,0x02,0x02,0x01,0x01,
-0x01,0x01,0x01,0x01,0x01,0x01,0x01,0x01,0x01,0x01,
-0x01,0x01,0x01,0x01,0x01,0x01,0x01,0x01,0x01,0x01,
-0x01,0x01,0x01,0x01,0x01,0x01,0x01,0x01,0x01,0x01,
-0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,
-0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,
-0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,
-0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,
-0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,
-};
-
-const uint16_t ff_ac3_hearing_threshold_tab[50][3]= {
-{ 0x04d0,0x04f0,0x0580 },
-{ 0x04d0,0x04f0,0x0580 },
-{ 0x0440,0x0460,0x04b0 },
-{ 0x0400,0x0410,0x0450 },
-{ 0x03e0,0x03e0,0x0420 },
-{ 0x03c0,0x03d0,0x03f0 },
-{ 0x03b0,0x03c0,0x03e0 },
-{ 0x03b0,0x03b0,0x03d0 },
-{ 0x03a0,0x03b0,0x03c0 },
-{ 0x03a0,0x03a0,0x03b0 },
-{ 0x03a0,0x03a0,0x03b0 },
-{ 0x03a0,0x03a0,0x03b0 },
-{ 0x03a0,0x03a0,0x03a0 },
-{ 0x0390,0x03a0,0x03a0 },
-{ 0x0390,0x0390,0x03a0 },
-{ 0x0390,0x0390,0x03a0 },
-{ 0x0380,0x0390,0x03a0 },
-{ 0x0380,0x0380,0x03a0 },
-{ 0x0370,0x0380,0x03a0 },
-{ 0x0370,0x0380,0x03a0 },
-{ 0x0360,0x0370,0x0390 },
-{ 0x0360,0x0370,0x0390 },
-{ 0x0350,0x0360,0x0390 },
-{ 0x0350,0x0360,0x0390 },
-{ 0x0340,0x0350,0x0380 },
-{ 0x0340,0x0350,0x0380 },
-{ 0x0330,0x0340,0x0380 },
-{ 0x0320,0x0340,0x0370 },
-{ 0x0310,0x0320,0x0360 },
-{ 0x0300,0x0310,0x0350 },
-{ 0x02f0,0x0300,0x0340 },
-{ 0x02f0,0x02f0,0x0330 },
-{ 0x02f0,0x02f0,0x0320 },
-{ 0x02f0,0x02f0,0x0310 },
-{ 0x0300,0x02f0,0x0300 },
-{ 0x0310,0x0300,0x02f0 },
-{ 0x0340,0x0320,0x02f0 },
-{ 0x0390,0x0350,0x02f0 },
-{ 0x03e0,0x0390,0x0300 },
-{ 0x0420,0x03e0,0x0310 },
-{ 0x0460,0x0420,0x0330 },
-{ 0x0490,0x0450,0x0350 },
-{ 0x04a0,0x04a0,0x03c0 },
-{ 0x0460,0x0490,0x0410 },
-{ 0x0440,0x0460,0x0470 },
-{ 0x0440,0x0440,0x04a0 },
-{ 0x0520,0x0480,0x0460 },
-{ 0x0800,0x0630,0x0440 },
-{ 0x0840,0x0840,0x0450 },
-{ 0x0840,0x0840,0x04e0 },
-};
-
-const uint8_t ff_ac3_bap_tab[64]= {
-    0, 1, 1, 1, 1, 1, 2, 2, 3, 3,
-    3, 4, 4, 5, 5, 6, 6, 6, 6, 7,
-    7, 7, 7, 8, 8, 8, 8, 9, 9, 9,
-    9, 10, 10, 10, 10, 11, 11, 11, 11, 12,
-    12, 12, 12, 13, 13, 13, 13, 14, 14, 14,
-    14, 14, 14, 14, 14, 15, 15, 15, 15, 15,
-    15, 15, 15, 15,
-};
-
-const uint8_t ff_ac3_slow_decay_tab[4]={
-    0x0f, 0x11, 0x13, 0x15,
-};
-
-const uint8_t ff_ac3_fast_decay_tab[4]={
-    0x3f, 0x53, 0x67, 0x7b,
-};
-
-const uint16_t ff_ac3_slow_gain_tab[4]= {
-    0x540, 0x4d8, 0x478, 0x410,
-};
-
-const uint16_t ff_ac3_db_per_bit_tab[4]= {
-    0x000, 0x700, 0x900, 0xb00,
-};
-
-const int16_t ff_ac3_floor_tab[8]= {
-    0x2f0, 0x2b0, 0x270, 0x230, 0x1f0, 0x170, 0x0f0, 0xf800,
-};
-
-const uint16_t ff_ac3_fast_gain_tab[8]= {
-    0x080, 0x100, 0x180, 0x200, 0x280, 0x300, 0x380, 0x400,
-};
-
-const uint8_t ff_ac3_critical_band_size_tab[50]={
-    1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1,
-    1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 3, 3, 3, 3, 3, 3,
-    3, 6, 6, 6, 6, 6, 6, 12, 12, 12, 12, 24, 24, 24, 24, 24
-};
-/**
- * Default channel map for a dependent substream defined by acmod
- */
-const uint16_t ff_eac3_default_chmap[8] = {
-    AC3_CHMAP_L |               AC3_CHMAP_R, // FIXME Ch1+Ch2
-                  AC3_CHMAP_C,
-    AC3_CHMAP_L |               AC3_CHMAP_R,
-    AC3_CHMAP_L | AC3_CHMAP_C | AC3_CHMAP_R,
-    AC3_CHMAP_L |               AC3_CHMAP_R |                   AC3_CHMAP_C_SUR,
-    AC3_CHMAP_L | AC3_CHMAP_C | AC3_CHMAP_R |                   AC3_CHMAP_C_SUR,
-    AC3_CHMAP_L |               AC3_CHMAP_R | AC3_CHMAP_L_SUR |                  AC3_CHMAP_R_SUR,
-    AC3_CHMAP_L | AC3_CHMAP_C | AC3_CHMAP_R | AC3_CHMAP_L_SUR |                  AC3_CHMAP_R_SUR
-};

libavcodec/ac3tab.h

@@ -1,59 +0,0 @@
-/*
- * AC-3 tables
- * Copyright (c) 2000, 2001, 2002 Fabrice Bellard
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#ifndef AVCODEC_AC3TAB_H
-#define AVCODEC_AC3TAB_H
-
-#include "libavutil/common.h"
-
-extern const uint16_t ff_ac3_frame_size_tab[38][3];
-extern const uint8_t  ff_ac3_channels_tab[8];
-extern const uint16_t ff_ac3_sample_rate_tab[3];
-extern const uint16_t ff_ac3_bitrate_tab[19];
-extern const int16_t  ff_ac3_window[256];
-extern const uint8_t  ff_ac3_log_add_tab[260];
-extern const uint16_t ff_ac3_hearing_threshold_tab[50][3];
-extern const uint8_t  ff_ac3_bap_tab[64];
-extern const uint8_t  ff_ac3_slow_decay_tab[4];
-extern const uint8_t  ff_ac3_fast_decay_tab[4];
-extern const uint16_t ff_ac3_slow_gain_tab[4];
-extern const uint16_t ff_ac3_db_per_bit_tab[4];
-extern const int16_t  ff_ac3_floor_tab[8];
-extern const uint16_t ff_ac3_fast_gain_tab[8];
-extern const uint8_t  ff_ac3_critical_band_size_tab[50];
-extern const uint16_t ff_eac3_default_chmap[8];
-
-/** Custom channel map locations bitmask
- *  Other channels described in documentation:
- *      Lc/Rc pair, Lrs/Rrs pair, Ts, Lsd/Rsd pair,
- *      Lw/Rw pair, Lvh/Rvh pair, Cvh, Reserved, LFE2
- */
-enum CustomChannelMapLocation{
-    AC3_CHMAP_L=        1<<(15-0),
-    AC3_CHMAP_C=        1<<(15-1),
-    AC3_CHMAP_R=        1<<(15-2),
-    AC3_CHMAP_L_SUR=    1<<(15-3),
-    AC3_CHMAP_R_SUR =   1<<(15-4),
-    AC3_CHMAP_C_SUR=    1<<(15-7),
-    AC3_CHMAP_LFE =     1<<(15-15)
-};
-
-#endif /* AVCODEC_AC3TAB_H */

libavcodec/acelp_filters.c

@@ -1,107 +0,0 @@
-/*
- * various filters for ACELP-based codecs
- *
- * Copyright (c) 2008 Vladimir Voroshilov
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#include <inttypes.h>
-
-#include "avcodec.h"
-#include "acelp_filters.h"
-
-const int16_t ff_acelp_interp_filter[61] =
-{ /* (0.15) */
-  29443, 28346, 25207, 20449, 14701,  8693,
-   3143, -1352, -4402, -5865, -5850, -4673,
-  -2783,  -672,  1211,  2536,  3130,  2991,
-   2259,  1170,     0, -1001, -1652, -1868,
-  -1666, -1147,  -464,   218,   756,  1060,
-   1099,   904,   550,   135,  -245,  -514,
-   -634,  -602,  -451,  -231,     0,   191,
-    308,   340,   296,   198,    78,   -36,
-   -120,  -163,  -165,  -132,   -79,   -19,
-     34,    73,    91,    89,    70,    38,
-      0,
-};
-
-void ff_acelp_interpolate(
-        int16_t* out,
-        const int16_t* in,
-        const int16_t* filter_coeffs,
-        int precision,
-        int frac_pos,
-        int filter_length,
-        int length)
-{
-    int n, i;
-
-    assert(pitch_delay_frac >= 0 && pitch_delay_frac < precision);
-
-    for(n=0; n<length; n++)
-    {
-        int idx = 0;
-        int v = 0x4000;
-
-        for(i=0; i<filter_length;)
-        {
-
-            /* The reference G.729 and AMR fixed point code performs clipping after
-               each of the two following accumulations.
-               Since clipping affects only the synthetic OVERFLOW test without
-               causing an int type overflow, it was moved outside the loop. */
-
-            /*  R(x):=ac_v[-k+x]
-                v += R(n-i)*ff_acelp_interp_filter(t+6i)
-                v += R(n+i+1)*ff_acelp_interp_filter(6-t+6i) */
-
-            v += in[n + i] * filter_coeffs[idx + frac_pos];
-            idx += precision;
-            i++;
-            v += in[n - i] * filter_coeffs[idx - frac_pos];
-        }
-        if(av_clip_int16(v>>15) != (v>>15))
-            av_log(NULL, AV_LOG_WARNING, "overflow that would need cliping in ff_acelp_interpolate()\n");
-        out[n] = v >> 15;
-    }
-}
-
-
-void ff_acelp_high_pass_filter(
-        int16_t* out,
-        int hpf_f[2],
-        const int16_t* in,
-        int length)
-{
-    int i;
-    int tmp;
-
-    for(i=0; i<length; i++)
-    {
-        tmp =  (hpf_f[0]* 15836LL)>>13;
-        tmp += (hpf_f[1]* -7667LL)>>13;
-        tmp += 7699 * (in[i] - 2*in[i-1] + in[i-2]);
-
-        /* With "+0x800" rounding, clipping is needed
-           for ALGTHM and SPEECH tests. */
-        out[i] = av_clip_int16((tmp + 0x800) >> 12);
-
-        hpf_f[1] = hpf_f[0];
-        hpf_f[0] = tmp;
-    }
-}

libavcodec/acelp_filters.h

@@ -1,93 +0,0 @@
-/*
- * various filters for ACELP-based codecs
- *
- * Copyright (c) 2008 Vladimir Voroshilov
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#ifndef AVCODEC_ACELP_FILTERS_H
-#define AVCODEC_ACELP_FILTERS_H
-
-#include <stdint.h>
-
-/**
- * low-pass Finite Impulse Response filter coefficients.
- *
- * Hamming windowed sinc filter with cutoff freq 3/40 of the sampling freq,
- * the coefficients are scaled by 2^15.
- * This array only contains the right half of the filter.
- * This filter is likely identical to the one used in G.729, though this
- * could not be determined from the original comments with certainity.
- */
-extern const int16_t ff_acelp_interp_filter[61];
-
-/**
- * Generic FIR interpolation routine.
- * @param out [out] buffer for interpolated data
- * @param in input data
- * @param filter_coeffs interpolation filter coefficients (0.15)
- * @param precision sub sample factor, that is the precision of the position
- * @param frac_pos fractional part of position [0..precision-1]
- * @param filter_length filter length
- * @param length length of output
- *
- * filter_coeffs contains coefficients of the right half of the symmetric
- * interpolation filter. filter_coeffs[0] should the central (unpaired) coefficient.
- * See ff_acelp_interp_filter for an example.
- *
- */
-void ff_acelp_interpolate(
-        int16_t* out,
-        const int16_t* in,
-        const int16_t* filter_coeffs,
-        int precision,
-        int frac_pos,
-        int filter_length,
-        int length);
-
-
-/**
- * high-pass filtering and upscaling (4.2.5 of G.729).
- * @param out [out] output buffer for filtered speech data
- * @param hpf_f [in/out] past filtered data from previous (2 items long)
- *                       frames (-0x20000000 <= (14.13) < 0x20000000)
- * @param in speech data to process
- * @param length input data size
- *
- * out[i] = 0.93980581 * in[i] - 1.8795834 * in[i-1] + 0.93980581 * in[i-2] +
- *          1.9330735 * out[i-1] - 0.93589199 * out[i-2]
- *
- * The filter has a cut-off frequency of 1/80 of the sampling freq
- *
- * @note Two items before the top of the out buffer must contain two items from the
- *       tail of the previous subframe.
- *
- * @remark It is safe to pass the same array in in and out parameters.
- *
- * @remark AMR uses mostly the same filter (cut-off frequency 60Hz, same formula,
- *         but constants differs in 5th sign after comma). Fortunately in
- *         fixed-point all coefficients are the same as in G.729. Thus this
- *         routine can be used for the fixed-point AMR decoder, too.
- */
-void ff_acelp_high_pass_filter(
-        int16_t* out,
-        int hpf_f[2],
-        const int16_t* in,
-        int length);
-
-#endif /* AVCODEC_ACELP_FILTERS_H */

libavcodec/acelp_pitch_delay.c

@@ -1,121 +0,0 @@
-/*
- * gain code, gain pitch and pitch delay decoding
- *
- * Copyright (c) 2008 Vladimir Voroshilov
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#include "avcodec.h"
-#include "dsputil.h"
-#include "acelp_pitch_delay.h"
-#include "celp_math.h"
-
-int ff_acelp_decode_8bit_to_1st_delay3(int ac_index)
-{
-    ac_index += 58;
-    if(ac_index > 254)
-        ac_index = 3 * ac_index - 510;
-    return ac_index;
-}
-
-int ff_acelp_decode_4bit_to_2nd_delay3(
-        int ac_index,
-        int pitch_delay_min)
-{
-    if(ac_index < 4)
-        return 3 * (ac_index + pitch_delay_min);
-    else if(ac_index < 12)
-        return 3 * pitch_delay_min + ac_index + 6;
-    else
-        return 3 * (ac_index + pitch_delay_min) - 18;
-}
-
-int ff_acelp_decode_5_6_bit_to_2nd_delay3(
-        int ac_index,
-        int pitch_delay_min)
-{
-        return 3 * pitch_delay_min + ac_index - 2;
-}
-
-int ff_acelp_decode_9bit_to_1st_delay6(int ac_index)
-{
-    if(ac_index < 463)
-        return ac_index + 105;
-    else
-        return 6 * (ac_index - 368);
-}
-int ff_acelp_decode_6bit_to_2nd_delay6(
-        int ac_index,
-        int pitch_delay_min)
-{
-    return 6 * pitch_delay_min + ac_index - 3;
-}
-
-void ff_acelp_update_past_gain(
-    int16_t* quant_energy,
-    int gain_corr_factor,
-    int log2_ma_pred_order,
-    int erasure)
-{
-    int i;
-    int avg_gain=quant_energy[(1 << log2_ma_pred_order) - 1]; // (5.10)
-
-    for(i=(1 << log2_ma_pred_order) - 1; i>0; i--)
-    {
-        avg_gain       += quant_energy[i-1];
-        quant_energy[i] = quant_energy[i-1];
-    }
-
-    if(erasure)
-        quant_energy[0] = FFMAX(avg_gain >> log2_ma_pred_order, -10240) - 4096; // -10 and -4 in (5.10)
-    else
-        quant_energy[0] = (6165 * ((ff_log2(gain_corr_factor) >> 2) - (13 << 13))) >> 13;
-}
-
-int16_t ff_acelp_decode_gain_code(
-    DSPContext *dsp,
-    int gain_corr_factor,
-    const int16_t* fc_v,
-    int mr_energy,
-    const int16_t* quant_energy,
-    const int16_t* ma_prediction_coeff,
-    int subframe_size,
-    int ma_pred_order)
-{
-    int i;
-
-    mr_energy <<= 10;
-
-    for(i=0; i<ma_pred_order; i++)
-        mr_energy += quant_energy[i] * ma_prediction_coeff[i];
-
-#ifdef G729_BITEXACT
-    mr_energy += (((-6165LL * ff_log2(dsp->scalarproduct_int16(fc_v, fc_v, subframe_size, 0))) >> 3) & ~0x3ff);
-
-    mr_energy = (5439 * (mr_energy >> 15)) >> 8;           // (0.15) = (0.15) * (7.23)
-
-    return bidir_sal(
-               ((ff_exp2(mr_energy & 0x7fff) + 16) >> 5) * (gain_corr_factor >> 1),
-               (mr_energy >> 15) - 25
-           );
-#else
-    mr_energy = gain_corr_factor * exp(M_LN10 / (20 << 23) * mr_energy) /
-                sqrt(dsp->scalarproduct_int16(fc_v, fc_v, subframe_size, 0));
-    return mr_energy >> 12;
-#endif
-}

libavcodec/acelp_pitch_delay.h

@@ -1,223 +0,0 @@
-/*
- * gain code, gain pitch and pitch delay decoding
- *
- * Copyright (c) 2008 Vladimir Voroshilov
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#ifndef AVCODEC_ACELP_PITCH_DELAY_H
-#define AVCODEC_ACELP_PITCH_DELAY_H
-
-#include <stdint.h>
-#include "dsputil.h"
-
-#define PITCH_DELAY_MIN             20
-#define PITCH_DELAY_MAX             143
-
-/**
- * \brief Decode pitch delay of the first subframe encoded by 8 bits with 1/3
- *        resolution.
- * \param ac_index adaptive codebook index (8 bits)
- *
- * \return pitch delay in 1/3 units
- *
- * Pitch delay is coded:
- *    with 1/3 resolution, 19  < pitch_delay <  85
- *    integers only,       85 <= pitch_delay <= 143
- */
-int ff_acelp_decode_8bit_to_1st_delay3(int ac_index);
-
-/**
- * \brief Decode pitch delay of the second subframe encoded by 5 or 6 bits
- *        with 1/3 precision.
- * \param ac_index adaptive codebook index (5 or 6 bits)
- * \param pitch_delay_min lower bound (integer) of pitch delay interval
- *                      for second subframe
- *
- * \return pitch delay in 1/3 units
- *
- * Pitch delay is coded:
- *    with 1/3 resolution, -6 < pitch_delay - int(prev_pitch_delay) < 5
- *
- * \remark The routine is used in G.729 @8k, AMR @10.2k, AMR @7.95k,
- *         AMR @7.4k for the second subframe.
- */
-int ff_acelp_decode_5_6_bit_to_2nd_delay3(
-        int ac_index,
-        int pitch_delay_min);
-
-/**
- * \brief Decode pitch delay with 1/3 precision.
- * \param ac_index adaptive codebook index (4 bits)
- * \param pitch_delay_min lower bound (integer) of pitch delay interval for
- *                      second subframe
- *
- * \return pitch delay in 1/3 units
- *
- * Pitch delay is coded:
- *    integers only,          -6  < pitch_delay - int(prev_pitch_delay) <= -2
- *    with 1/3 resolution,    -2  < pitch_delay - int(prev_pitch_delay) <  1
- *    integers only,           1 <= pitch_delay - int(prev_pitch_delay) <  5
- *
- * \remark The routine is used in G.729 @6.4k, AMR @6.7k, AMR @5.9k,
- *         AMR @5.15k, AMR @4.75k for the second subframe.
- */
-int ff_acelp_decode_4bit_to_2nd_delay3(
-        int ac_index,
-        int pitch_delay_min);
-
-/**
- * \brief Decode pitch delay of the first subframe encoded by 9 bits
- *        with 1/6 precision.
- * \param ac_index adaptive codebook index (9 bits)
- * \param pitch_delay_min lower bound (integer) of pitch delay interval for
- *                      second subframe
- *
- * \return pitch delay in 1/6 units
- *
- * Pitch delay is coded:
- *    with 1/6 resolution,  17  < pitch_delay <  95
- *    integers only,        95 <= pitch_delay <= 143
- *
- * \remark The routine is used in AMR @12.2k for the first and third subframes.
- */
-int ff_acelp_decode_9bit_to_1st_delay6(int ac_index);
-
-/**
- * \brief Decode pitch delay of the second subframe encoded by 6 bits
- *        with 1/6 precision.
- * \param ac_index adaptive codebook index (6 bits)
- * \param pitch_delay_min lower bound (integer) of pitch delay interval for
- *                      second subframe
- *
- * \return pitch delay in 1/6 units
- *
- * Pitch delay is coded:
- *    with 1/6 resolution, -6 < pitch_delay - int(prev_pitch_delay) < 5
- *
- * \remark The routine is used in AMR @12.2k for the second and fourth subframes.
- */
-int ff_acelp_decode_6bit_to_2nd_delay6(
-        int ac_index,
-        int pitch_delay_min);
-
-/**
- * \brief Update past quantized energies
- * \param quant_energy [in/out] past quantized energies (5.10)
- * \param gain_corr_factor gain correction factor
- * \param log2_ma_pred_order log2() of MA prediction order
- * \param erasure frame erasure flag
- *
- * If frame erasure flag is not equal to zero, memory is updated with
- * averaged energy, attenuated by 4dB:
- *     max(avg(quant_energy[i])-4, -14), i=0,ma_pred_order
- *
- * In normal mode memory is updated with
- *     Er - Ep = 20 * log10(gain_corr_factor)
- *
- * \remark The routine is used in G.729 and AMR (all modes).
- */
-void ff_acelp_update_past_gain(
-        int16_t* quant_energy,
-        int gain_corr_factor,
-        int log2_ma_pred_order,
-        int erasure);
-
-/**
- * \brief Decode the adaptive codebook gain and add
- *        correction (4.1.5 and 3.9.1 of G.729).
- * \param dsp initialized dsputil context
- * \param gain_corr_factor gain correction factor (2.13)
- * \param fc_v fixed-codebook vector (2.13)
- * \param mr_energy mean innovation energy and fixed-point correction (7.13)
- * \param quant_energy [in/out] past quantized energies (5.10)
- * \param subframe_size length of subframe
- * \param ma_pred_order MA prediction order
- *
- * \return quantized fixed-codebook gain (14.1)
- *
- * The routine implements equations 69, 66 and 71 of the G.729 specification (3.9.1)
- *
- *    Em   - mean innovation energy (dB, constant, depends on decoding algorithm)
- *    Ep   - mean-removed predicted energy (dB)
- *    Er   - mean-removed innovation energy (dB)
- *    Ei   - mean energy of the fixed-codebook contribution (dB)
- *    N    - subframe_size
- *    M    - MA (Moving Average) prediction order
- *    gc   - fixed-codebook gain
- *    gc_p - predicted fixed-codebook gain
- *
- *    Fixed codebook gain is computed using predicted gain gc_p and
- *    correction factor gain_corr_factor as shown below:
- *
- *        gc = gc_p * gain_corr_factor
- *
- *    The predicted fixed codebook gain gc_p is found by predicting
- *    the energy of the fixed-codebook contribution from the energy
- *    of previous fixed-codebook contributions.
- *
- *        mean = 1/N * sum(i,0,N){ fc_v[i] * fc_v[i] }
- *
- *        Ei = 10log(mean)
- *
- *        Er = 10log(1/N * gc^2 * mean) - Em = 20log(gc) + Ei - Em
- *
- *    Replacing Er with Ep and gc with gc_p we will receive:
- *
- *        Ep = 10log(1/N * gc_p^2 * mean) - Em = 20log(gc_p) + Ei - Em
- *
- *    and from above:
- *
- *        gc_p = 10^((Ep - Ei + Em) / 20)
- *
- *    Ep is predicted using past energies and prediction coefficients:
- *
- *        Ep = sum(i,0,M){ ma_prediction_coeff[i] * quant_energy[i] }
- *
- *    gc_p in fixed-point arithmetic is calculated as following:
- *
- *        mean = 1/N * sum(i,0,N){ (fc_v[i] / 2^13) * (fc_v[i] / 2^13) } =
- *        = 1/N * sum(i,0,N) { fc_v[i] * fc_v[i] } / 2^26
- *
- *        Ei = 10log(mean) = -10log(N) - 10log(2^26) +
- *        + 10log(sum(i,0,N) { fc_v[i] * fc_v[i] })
- *
- *        Ep - Ei + Em = Ep + Em + 10log(N) + 10log(2^26) -
- *        - 10log(sum(i,0,N) { fc_v[i] * fc_v[i] }) =
- *        = Ep + mr_energy - 10log(sum(i,0,N) { fc_v[i] * fc_v[i] })
- *
- *        gc_p = 10 ^ ((Ep - Ei + Em) / 20) =
- *        = 2 ^ (3.3219 * (Ep - Ei + Em) / 20) = 2 ^ (0.166 * (Ep - Ei + Em))
- *
- *    where
- *
- *        mr_energy = Em + 10log(N) + 10log(2^26)
- *
- * \remark The routine is used in G.729 and AMR (all modes).
- */
-int16_t ff_acelp_decode_gain_code(
-    DSPContext *dsp,
-    int gain_corr_factor,
-    const int16_t* fc_v,
-    int mr_energy,
-    const int16_t* quant_energy,
-    const int16_t* ma_prediction_coeff,
-    int subframe_size,
-    int max_pred_order);
-
-#endif /* AVCODEC_ACELP_PITCH_DELAY_H */

libavcodec/acelp_vectors.c

@@ -1,147 +0,0 @@
-/*
- * adaptive and fixed codebook vector operations for ACELP-based codecs
- *
- * Copyright (c) 2008 Vladimir Voroshilov
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#include <inttypes.h>
-#include "avcodec.h"
-#include "acelp_vectors.h"
-
-const uint8_t ff_fc_2pulses_9bits_track1[16] =
-{
-    1,  3,
-    6,  8,
-    11, 13,
-    16, 18,
-    21, 23,
-    26, 28,
-    31, 33,
-    36, 38
-};
-const uint8_t ff_fc_2pulses_9bits_track1_gray[16] =
-{
-  1,  3,
-  8,  6,
-  18, 16,
-  11, 13,
-  38, 36,
-  31, 33,
-  21, 23,
-  28, 26,
-};
-
-const uint8_t ff_fc_2pulses_9bits_track2_gray[32] =
-{
-  0,  2,
-  5,  4,
-  12, 10,
-  7,  9,
-  25, 24,
-  20, 22,
-  14, 15,
-  19, 17,
-  36, 31,
-  21, 26,
-  1,  6,
-  16, 11,
-  27, 29,
-  32, 30,
-  39, 37,
-  34, 35,
-};
-
-const uint8_t ff_fc_4pulses_8bits_tracks_13[16] =
-{
-  0, 5, 10, 15, 20, 25, 30, 35, 40, 45, 50, 55, 60, 65, 70, 75,
-};
-
-const uint8_t ff_fc_4pulses_8bits_track_4[32] =
-{
-    3,  4,
-    8,  9,
-    13, 14,
-    18, 19,
-    23, 24,
-    28, 29,
-    33, 34,
-    38, 39,
-    43, 44,
-    48, 49,
-    53, 54,
-    58, 59,
-    63, 64,
-    68, 69,
-    73, 74,
-    78, 79,
-};
-
-#if 0
-static uint8_t gray_decode[32] =
-{
-    0,  1,  3,  2,  7,  6,  4,  5,
-   15, 14, 12, 13,  8,  9, 11, 10,
-   31, 30, 28, 29, 24, 25, 27, 26,
-   16, 17, 19, 18, 23, 22, 20, 21
-};
-#endif
-
-void ff_acelp_fc_pulse_per_track(
-        int16_t* fc_v,
-        const uint8_t *tab1,
-        const uint8_t *tab2,
-        int pulse_indexes,
-        int pulse_signs,
-        int pulse_count,
-        int bits)
-{
-    int mask = (1 << bits) - 1;
-    int i;
-
-    for(i=0; i<pulse_count; i++)
-    {
-        fc_v[i + tab1[pulse_indexes & mask]] +=
-                (pulse_signs & 1) ? 8191 : -8192; // +/-1 in (2.13)
-
-        pulse_indexes >>= bits;
-        pulse_signs >>= 1;
-    }
-
-    fc_v[tab2[pulse_indexes]] += (pulse_signs & 1) ? 8191 : -8192;
-}
-
-void ff_acelp_weighted_vector_sum(
-        int16_t* out,
-        const int16_t *in_a,
-        const int16_t *in_b,
-        int16_t weight_coeff_a,
-        int16_t weight_coeff_b,
-        int16_t rounder,
-        int shift,
-        int length)
-{
-    int i;
-
-    // Clipping required here; breaks OVERFLOW test.
-    for(i=0; i<length; i++)
-        out[i] = av_clip_int16((
-                 in_a[i] * weight_coeff_a +
-                 in_b[i] * weight_coeff_b +
-                 rounder) >> shift);
-}

libavcodec/acelp_vectors.h

@@ -1,153 +0,0 @@
-/*
- * adaptive and fixed codebook vector operations for ACELP-based codecs
- *
- * Copyright (c) 2008 Vladimir Voroshilov
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#ifndef AVCODEC_ACELP_VECTORS_H
-#define AVCODEC_ACELP_VECTORS_H
-
-#include <stdint.h>
-
-/**
- * Track|Pulse|        Positions
- * -------------------------------------------------------------------------
- *  1   | 0   | 0, 5, 10, 15, 20, 25, 30, 35, 40, 45, 50, 55, 60, 65, 70, 75
- * -------------------------------------------------------------------------
- *  2   | 1   | 1, 6, 11, 16, 21, 26, 31, 36, 41, 46, 51, 56, 61, 66, 71, 76
- * -------------------------------------------------------------------------
- *  3   | 2   | 2, 7, 12, 17, 22, 27, 32, 37, 42, 47, 52, 57, 62, 67, 72, 77
- * -------------------------------------------------------------------------
- *
- * Table contains only first the pulse indexes.
- *
- * Used in G.729 @8k, G.729 @4.4k, AMR @7.95k, AMR @7.40k
- */
-extern const uint8_t ff_fc_4pulses_8bits_tracks_13[16];
-
-/**
- * Track|Pulse|        Positions
- * -------------------------------------------------------------------------
- *  4   | 3   | 3, 8, 13, 18, 23, 28, 33, 38, 43, 48, 53, 58, 63, 68, 73, 78
- *      |     | 4, 9, 14, 19, 24, 29, 34, 39, 44, 49, 54, 59, 64, 69, 74, 79
- * -------------------------------------------------------------------------
- *
- * @remark Track in the table should be read top-to-bottom, left-to-right.
- *
- * Used in G.729 @8k, G.729 @4.4k, AMR @7.95k, AMR @7.40k
- */
-extern const uint8_t ff_fc_4pulses_8bits_track_4[32];
-
-/**
- * Track|Pulse|        Positions
- * -----------------------------------------
- *  1   | 0   | 1, 6, 11, 16, 21, 26, 31, 36
- *      |     | 3, 8, 13, 18, 23, 28, 33, 38
- * -----------------------------------------
- *
- * @remark Track in the table should be read top-to-bottom, left-to-right.
- *
- * @note (EE) Reference G.729D code also uses gray decoding for each
- *            pulse index before looking up the value in the table.
- *
- * Used in G.729 @6.4k (with gray coding), AMR @5.9k (without gray coding)
- */
-extern const uint8_t ff_fc_2pulses_9bits_track1[16];
-extern const uint8_t ff_fc_2pulses_9bits_track1_gray[16];
-
-/**
- * Track|Pulse|        Positions
- * -----------------------------------------
- *  2   | 1   | 0, 7, 14, 20, 27, 34,  1, 21
- *      |     | 2, 9, 15, 22, 29, 35,  6, 26
- *      |     | 4,10, 17, 24, 30, 37, 11, 31
- *      |     | 5,12, 19, 25, 32, 39, 16, 36
- * -----------------------------------------
- *
- * @remark Track in the table should be read top-to-bottom, left-to-right.
- *
- * @note (EE.1) This table (from the reference code) does not comply with
- *              the specification.
- *              The specification contains the following table:
- *
- * Track|Pulse|        Positions
- * -----------------------------------------
- *  2   | 1   | 0, 5, 10, 15, 20, 25, 30, 35
- *      |     | 1, 6, 11, 16, 21, 26, 31, 36
- *      |     | 2, 7, 12, 17, 22, 27, 32, 37
- *      |     | 4, 9, 14, 19, 24, 29, 34, 39
- *
- * -----------------------------------------
- *
- * @note (EE.2) Reference G.729D code also uses gray decoding for each
- *              pulse index before looking up the value in the table.
- *
- * Used in G.729 @6.4k (with gray coding)
- */
-extern const uint8_t ff_fc_2pulses_9bits_track2_gray[32];
-
-/**
- * Decode fixed-codebook vector (3.8 and D.5.8 of G.729, 5.7.1 of AMR).
- * @param fc_v [out] decoded fixed codebook vector (2.13)
- * @param tab1 table used for first pulse_count pulses
- * @param tab2 table used for last pulse
- * @param pulse_indexes fixed codebook indexes
- * @param pulse_signs signs of the excitation pulses (0 bit value
- *                     means negative sign)
- * @param bits number of bits per one pulse index
- * @param pulse_count number of pulses decoded using first table
- * @param bits length of one pulse index in bits
- *
- * Used in G.729 @8k, G.729 @4.4k, G.729 @6.4k, AMR @7.95k, AMR @7.40k
- */
-void ff_acelp_fc_pulse_per_track(
-        int16_t* fc_v,
-        const uint8_t *tab1,
-        const uint8_t *tab2,
-        int pulse_indexes,
-        int pulse_signs,
-        int pulse_count,
-        int bits);
-
-/**
- * weighted sum of two vectors with rounding.
- * @param out [out] result of addition
- * @param in_a first vector
- * @param in_b second vector
- * @param weight_coeff_a first vector weight coefficient
- * @param weight_coeff_a second vector weight coefficient
- * @param rounder this value will be added to the sum of the two vectors
- * @param shift result will be shifted to right by this value
- * @param length vectors length
- *
- * @note It is safe to pass the same buffer for out and in_a or in_b.
- *
- *  out[i] = (in_a[i]*weight_a + in_b[i]*weight_b + rounder) >> shift
- */
-void ff_acelp_weighted_vector_sum(
-        int16_t* out,
-        const int16_t *in_a,
-        const int16_t *in_b,
-        int16_t weight_coeff_a,
-        int16_t weight_coeff_b,
-        int16_t rounder,
-        int shift,
-        int length);
-
-#endif /* AVCODEC_ACELP_VECTORS_H */

libavcodec/adx.h

@@ -1,49 +0,0 @@
-/*
- * ADX ADPCM codecs
- * Copyright (c) 2001,2003 BERO
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/**
- * @file libavcodec/adx.h
- * SEGA CRI adx codecs.
- *
- * Reference documents:
- * http://ku-www.ss.titech.ac.jp/~yatsushi/adx.html
- * adx2wav & wav2adx http://www.geocities.co.jp/Playtown/2004/
- */
-
-#ifndef AVCODEC_ADX_H
-#define AVCODEC_ADX_H
-
-typedef struct {
-    int s1,s2;
-} PREV;
-
-typedef struct {
-    PREV prev[2];
-    int header_parsed;
-    unsigned char dec_temp[18*2];
-    int in_temp;
-} ADXContext;
-
-#define    BASEVOL   0x4000
-#define    SCALE1    0x7298
-#define    SCALE2    0x3350
-
-#endif /* AVCODEC_ADX_H */

libavcodec/adxdec.c

@@ -1,178 +0,0 @@
-/*
- * ADX ADPCM codecs
- * Copyright (c) 2001,2003 BERO
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#include "libavutil/intreadwrite.h"
-#include "avcodec.h"
-#include "adx.h"
-
-/**
- * @file libavcodec/adxdec.c
- * SEGA CRI adx codecs.
- *
- * Reference documents:
- * http://ku-www.ss.titech.ac.jp/~yatsushi/adx.html
- * adx2wav & wav2adx http://www.geocities.co.jp/Playtown/2004/
- */
-
-static av_cold int adx_decode_init(AVCodecContext *avctx)
-{
-    avctx->sample_fmt = SAMPLE_FMT_S16;
-    return 0;
-}
-
-/* 18 bytes <-> 32 samples */
-
-static void adx_decode(short *out,const unsigned char *in,PREV *prev)
-{
-    int scale = AV_RB16(in);
-    int i;
-    int s0,s1,s2,d;
-
-//    printf("%x ",scale);
-
-    in+=2;
-    s1 = prev->s1;
-    s2 = prev->s2;
-    for(i=0;i<16;i++) {
-        d = in[i];
-        // d>>=4; if (d&8) d-=16;
-        d = ((signed char)d >> 4);
-        s0 = (BASEVOL*d*scale + SCALE1*s1 - SCALE2*s2)>>14;
-        s2 = s1;
-        s1 = av_clip_int16(s0);
-        *out++=s1;
-
-        d = in[i];
-        //d&=15; if (d&8) d-=16;
-        d = ((signed char)(d<<4) >> 4);
-        s0 = (BASEVOL*d*scale + SCALE1*s1 - SCALE2*s2)>>14;
-        s2 = s1;
-        s1 = av_clip_int16(s0);
-        *out++=s1;
-    }
-    prev->s1 = s1;
-    prev->s2 = s2;
-
-}
-
-static void adx_decode_stereo(short *out,const unsigned char *in,PREV *prev)
-{
-    short tmp[32*2];
-    int i;
-
-    adx_decode(tmp   ,in   ,prev);
-    adx_decode(tmp+32,in+18,prev+1);
-    for(i=0;i<32;i++) {
-        out[i*2]   = tmp[i];
-        out[i*2+1] = tmp[i+32];
-    }
-}
-
-/* return data offset or 0 */
-static int adx_decode_header(AVCodecContext *avctx,const unsigned char *buf,size_t bufsize)
-{
-    int offset;
-
-    if (buf[0]!=0x80) return 0;
-    offset = (AV_RB32(buf)^0x80000000)+4;
-    if (bufsize<offset || memcmp(buf+offset-6,"(c)CRI",6)) return 0;
-
-    avctx->channels    = buf[7];
-    avctx->sample_rate = AV_RB32(buf+8);
-    avctx->bit_rate    = avctx->sample_rate*avctx->channels*18*8/32;
-
-    return offset;
-}
-
-static int adx_decode_frame(AVCodecContext *avctx,
-                void *data, int *data_size,
-                const uint8_t *buf0, int buf_size)
-{
-    ADXContext *c = avctx->priv_data;
-    short *samples = data;
-    const uint8_t *buf = buf0;
-    int rest = buf_size;
-
-    if (!c->header_parsed) {
-        int hdrsize = adx_decode_header(avctx,buf,rest);
-        if (hdrsize==0) return -1;
-        c->header_parsed = 1;
-        buf  += hdrsize;
-        rest -= hdrsize;
-    }
-
-    /* 18 bytes of data are expanded into 32*2 bytes of audio,
-       so guard against buffer overflows */
-    if(rest/18 > *data_size/64)
-        rest = (*data_size/64) * 18;
-
-    if (c->in_temp) {
-        int copysize = 18*avctx->channels - c->in_temp;
-        memcpy(c->dec_temp+c->in_temp,buf,copysize);
-        rest -= copysize;
-        buf  += copysize;
-        if (avctx->channels==1) {
-            adx_decode(samples,c->dec_temp,c->prev);
-            samples += 32;
-        } else {
-            adx_decode_stereo(samples,c->dec_temp,c->prev);
-            samples += 32*2;
-        }
-    }
-    //
-    if (avctx->channels==1) {
-        while(rest>=18) {
-            adx_decode(samples,buf,c->prev);
-            rest-=18;
-            buf+=18;
-            samples+=32;
-        }
-    } else {
-        while(rest>=18*2) {
-            adx_decode_stereo(samples,buf,c->prev);
-            rest-=18*2;
-            buf+=18*2;
-            samples+=32*2;
-        }
-    }
-    //
-    c->in_temp = rest;
-    if (rest) {
-        memcpy(c->dec_temp,buf,rest);
-        buf+=rest;
-    }
-    *data_size = (uint8_t*)samples - (uint8_t*)data;
-//    printf("%d:%d ",buf-buf0,*data_size); fflush(stdout);
-    return buf-buf0;
-}
-
-AVCodec adpcm_adx_decoder = {
-    "adpcm_adx",
-    CODEC_TYPE_AUDIO,
-    CODEC_ID_ADPCM_ADX,
-    sizeof(ADXContext),
-    adx_decode_init,
-    NULL,
-    NULL,
-    adx_decode_frame,
-    .long_name = NULL_IF_CONFIG_SMALL("SEGA CRI ADX ADPCM"),
-};
-

libavcodec/adxenc.c

@@ -1,197 +0,0 @@
-/*
- * ADX ADPCM codecs
- * Copyright (c) 2001,2003 BERO
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#include "libavutil/intreadwrite.h"
-#include "avcodec.h"
-#include "adx.h"
-
-/**
- * @file libavcodec/adxenc.c
- * SEGA CRI adx codecs.
- *
- * Reference documents:
- * http://ku-www.ss.titech.ac.jp/~yatsushi/adx.html
- * adx2wav & wav2adx http://www.geocities.co.jp/Playtown/2004/
- */
-
-/* 18 bytes <-> 32 samples */
-
-static void adx_encode(unsigned char *adx,const short *wav,PREV *prev)
-{
-    int scale;
-    int i;
-    int s0,s1,s2,d;
-    int max=0;
-    int min=0;
-    int data[32];
-
-    s1 = prev->s1;
-    s2 = prev->s2;
-    for(i=0;i<32;i++) {
-        s0 = wav[i];
-        d = ((s0<<14) - SCALE1*s1 + SCALE2*s2)/BASEVOL;
-        data[i]=d;
-        if (max<d) max=d;
-        if (min>d) min=d;
-        s2 = s1;
-        s1 = s0;
-    }
-    prev->s1 = s1;
-    prev->s2 = s2;
-
-    /* -8..+7 */
-
-    if (max==0 && min==0) {
-        memset(adx,0,18);
-        return;
-    }
-
-    if (max/7>-min/8) scale = max/7;
-    else scale = -min/8;
-
-    if (scale==0) scale=1;
-
-    AV_WB16(adx, scale);
-
-    for(i=0;i<16;i++) {
-        adx[i+2] = ((data[i*2]/scale)<<4) | ((data[i*2+1]/scale)&0xf);
-    }
-}
-
-static int adx_encode_header(AVCodecContext *avctx,unsigned char *buf,size_t bufsize)
-{
-#if 0
-    struct {
-        uint32_t offset; /* 0x80000000 + sample start - 4 */
-        unsigned char unknown1[3]; /* 03 12 04 */
-        unsigned char channel; /* 1 or 2 */
-        uint32_t freq;
-        uint32_t size;
-        uint32_t unknown2; /* 01 f4 03 00 */
-        uint32_t unknown3; /* 00 00 00 00 */
-        uint32_t unknown4; /* 00 00 00 00 */
-
-    /* if loop
-        unknown3 00 15 00 01
-        unknown4 00 00 00 01
-        long loop_start_sample;
-        long loop_start_byte;
-        long loop_end_sample;
-        long loop_end_byte;
-        long
-    */
-    } adxhdr; /* big endian */
-    /* offset-6 "(c)CRI" */
-#endif
-    AV_WB32(buf+0x00,0x80000000|0x20);
-    AV_WB32(buf+0x04,0x03120400|avctx->channels);
-    AV_WB32(buf+0x08,avctx->sample_rate);
-    AV_WB32(buf+0x0c,0); /* FIXME: set after */
-    AV_WB32(buf+0x10,0x01040300);
-    AV_WB32(buf+0x14,0x00000000);
-    AV_WB32(buf+0x18,0x00000000);
-    memcpy(buf+0x1c,"\0\0(c)CRI",8);
-    return 0x20+4;
-}
-
-static av_cold int adx_encode_init(AVCodecContext *avctx)
-{
-    if (avctx->channels > 2)
-        return -1; /* only stereo or mono =) */
-    avctx->frame_size = 32;
-
-    avctx->coded_frame= avcodec_alloc_frame();
-    avctx->coded_frame->key_frame= 1;
-
-//    avctx->bit_rate = avctx->sample_rate*avctx->channels*18*8/32;
-
-    av_log(avctx, AV_LOG_DEBUG, "adx encode init\n");
-
-    return 0;
-}
-
-static av_cold int adx_encode_close(AVCodecContext *avctx)
-{
-    av_freep(&avctx->coded_frame);
-
-    return 0;
-}
-
-static int adx_encode_frame(AVCodecContext *avctx,
-                uint8_t *frame, int buf_size, void *data)
-{
-    ADXContext *c = avctx->priv_data;
-    const short *samples = data;
-    unsigned char *dst = frame;
-    int rest = avctx->frame_size;
-
-/*
-    input data size =
-    ffmpeg.c: do_audio_out()
-    frame_bytes = enc->frame_size * 2 * enc->channels;
-*/
-
-//    printf("sz=%d ",buf_size); fflush(stdout);
-    if (!c->header_parsed) {
-        int hdrsize = adx_encode_header(avctx,dst,buf_size);
-        dst+=hdrsize;
-        c->header_parsed = 1;
-    }
-
-    if (avctx->channels==1) {
-        while(rest>=32) {
-            adx_encode(dst,samples,c->prev);
-            dst+=18;
-            samples+=32;
-            rest-=32;
-        }
-    } else {
-        while(rest>=32*2) {
-            short tmpbuf[32*2];
-            int i;
-
-            for(i=0;i<32;i++) {
-                tmpbuf[i] = samples[i*2];
-                tmpbuf[i+32] = samples[i*2+1];
-            }
-
-            adx_encode(dst,tmpbuf,c->prev);
-            adx_encode(dst+18,tmpbuf+32,c->prev+1);
-            dst+=18*2;
-            samples+=32*2;
-            rest-=32*2;
-        }
-    }
-    return dst-frame;
-}
-
-AVCodec adpcm_adx_encoder = {
-    "adpcm_adx",
-    CODEC_TYPE_AUDIO,
-    CODEC_ID_ADPCM_ADX,
-    sizeof(ADXContext),
-    adx_encode_init,
-    adx_encode_frame,
-    adx_encode_close,
-    NULL,
-    .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
-    .long_name = NULL_IF_CONFIG_SMALL("SEGA CRI ADX ADPCM"),
-};

libavcodec/alac.c

@@ -1,629 +0,0 @@
-/*
- * ALAC (Apple Lossless Audio Codec) decoder
- * Copyright (c) 2005 David Hammerton
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/**
- * @file libavcodec/alac.c
- * ALAC (Apple Lossless Audio Codec) decoder
- * @author 2005 David Hammerton
- *
- * For more information on the ALAC format, visit:
- *  http://crazney.net/programs/itunes/alac.html
- *
- * Note: This decoder expects a 36- (0x24-)byte QuickTime atom to be
- * passed through the extradata[_size] fields. This atom is tacked onto
- * the end of an 'alac' stsd atom and has the following format:
- *  bytes 0-3   atom size (0x24), big-endian
- *  bytes 4-7   atom type ('alac', not the 'alac' tag from start of stsd)
- *  bytes 8-35  data bytes needed by decoder
- *
- * Extradata:
- * 32bit  size
- * 32bit  tag (=alac)
- * 32bit  zero?
- * 32bit  max sample per frame
- *  8bit  ?? (zero?)
- *  8bit  sample size
- *  8bit  history mult
- *  8bit  initial history
- *  8bit  kmodifier
- *  8bit  channels?
- * 16bit  ??
- * 32bit  max coded frame size
- * 32bit  bitrate?
- * 32bit  samplerate
- */
-
-
-#include "avcodec.h"
-#include "bitstream.h"
-#include "bytestream.h"
-#include "unary.h"
-
-#define ALAC_EXTRADATA_SIZE 36
-#define MAX_CHANNELS 2
-
-typedef struct {
-
-    AVCodecContext *avctx;
-    GetBitContext gb;
-    /* init to 0; first frame decode should initialize from extradata and
-     * set this to 1 */
-    int context_initialized;
-
-    int numchannels;
-    int bytespersample;
-
-    /* buffers */
-    int32_t *predicterror_buffer[MAX_CHANNELS];
-
-    int32_t *outputsamples_buffer[MAX_CHANNELS];
-
-    /* stuff from setinfo */
-    uint32_t setinfo_max_samples_per_frame; /* 0x1000 = 4096 */    /* max samples per frame? */
-    uint8_t setinfo_sample_size; /* 0x10 */
-    uint8_t setinfo_rice_historymult; /* 0x28 */
-    uint8_t setinfo_rice_initialhistory; /* 0x0a */
-    uint8_t setinfo_rice_kmodifier; /* 0x0e */
-    /* end setinfo stuff */
-
-} ALACContext;
-
-static void allocate_buffers(ALACContext *alac)
-{
-    int chan;
-    for (chan = 0; chan < MAX_CHANNELS; chan++) {
-        alac->predicterror_buffer[chan] =
-            av_malloc(alac->setinfo_max_samples_per_frame * 4);
-
-        alac->outputsamples_buffer[chan] =
-            av_malloc(alac->setinfo_max_samples_per_frame * 4);
-    }
-}
-
-static int alac_set_info(ALACContext *alac)
-{
-    const unsigned char *ptr = alac->avctx->extradata;
-
-    ptr += 4; /* size */
-    ptr += 4; /* alac */
-    ptr += 4; /* 0 ? */
-
-    if(AV_RB32(ptr) >= UINT_MAX/4){
-        av_log(alac->avctx, AV_LOG_ERROR, "setinfo_max_samples_per_frame too large\n");
-        return -1;
-    }
-
-    /* buffer size / 2 ? */
-    alac->setinfo_max_samples_per_frame = bytestream_get_be32(&ptr);
-    ptr++;                          /* ??? */
-    alac->setinfo_sample_size           = *ptr++;
-    if (alac->setinfo_sample_size > 32) {
-        av_log(alac->avctx, AV_LOG_ERROR, "setinfo_sample_size too large\n");
-        return -1;
-    }
-    alac->setinfo_rice_historymult      = *ptr++;
-    alac->setinfo_rice_initialhistory   = *ptr++;
-    alac->setinfo_rice_kmodifier        = *ptr++;
-    ptr++;                         /* channels? */
-    bytestream_get_be16(&ptr);      /* ??? */
-    bytestream_get_be32(&ptr);      /* max coded frame size */
-    bytestream_get_be32(&ptr);      /* bitrate ? */
-    bytestream_get_be32(&ptr);      /* samplerate */
-
-    allocate_buffers(alac);
-
-    return 0;
-}
-
-static inline int decode_scalar(GetBitContext *gb, int k, int limit, int readsamplesize){
-    /* read x - number of 1s before 0 represent the rice */
-    int x = get_unary_0_9(gb);
-
-    if (x > 8) { /* RICE THRESHOLD */
-        /* use alternative encoding */
-        x = get_bits(gb, readsamplesize);
-    } else {
-        if (k >= limit)
-            k = limit;
-
-        if (k != 1) {
-            int extrabits = show_bits(gb, k);
-
-            /* multiply x by 2^k - 1, as part of their strange algorithm */
-            x = (x << k) - x;
-
-            if (extrabits > 1) {
-                x += extrabits - 1;
-                skip_bits(gb, k);
-            } else
-                skip_bits(gb, k - 1);
-        }
-    }
-    return x;
-}
-
-static void bastardized_rice_decompress(ALACContext *alac,
-                                 int32_t *output_buffer,
-                                 int output_size,
-                                 int readsamplesize, /* arg_10 */
-                                 int rice_initialhistory, /* arg424->b */
-                                 int rice_kmodifier, /* arg424->d */
-                                 int rice_historymult, /* arg424->c */
-                                 int rice_kmodifier_mask /* arg424->e */
-        )
-{
-    int output_count;
-    unsigned int history = rice_initialhistory;
-    int sign_modifier = 0;
-
-    for (output_count = 0; output_count < output_size; output_count++) {
-        int32_t x;
-        int32_t x_modified;
-        int32_t final_val;
-
-        /* standard rice encoding */
-        int k; /* size of extra bits */
-
-        /* read k, that is bits as is */
-        k = av_log2((history >> 9) + 3);
-        x= decode_scalar(&alac->gb, k, rice_kmodifier, readsamplesize);
-
-        x_modified = sign_modifier + x;
-        final_val = (x_modified + 1) / 2;
-        if (x_modified & 1) final_val *= -1;
-
-        output_buffer[output_count] = final_val;
-
-        sign_modifier = 0;
-
-        /* now update the history */
-        history += x_modified * rice_historymult
-                   - ((history * rice_historymult) >> 9);
-
-        if (x_modified > 0xffff)
-            history = 0xffff;
-
-        /* special case: there may be compressed blocks of 0 */
-        if ((history < 128) && (output_count+1 < output_size)) {
-            int k;
-            unsigned int block_size;
-
-            sign_modifier = 1;
-
-            k = 7 - av_log2(history) + ((history + 16) >> 6 /* / 64 */);
-
-            block_size= decode_scalar(&alac->gb, k, rice_kmodifier, 16);
-
-            if (block_size > 0) {
-                if(block_size >= output_size - output_count){
-                    av_log(alac->avctx, AV_LOG_ERROR, "invalid zero block size of %d %d %d\n", block_size, output_size, output_count);
-                    block_size= output_size - output_count - 1;
-                }
-                memset(&output_buffer[output_count+1], 0, block_size * 4);
-                output_count += block_size;
-            }
-
-            if (block_size > 0xffff)
-                sign_modifier = 0;
-
-            history = 0;
-        }
-    }
-}
-
-static inline int32_t extend_sign32(int32_t val, int bits)
-{
-    return (val << (32 - bits)) >> (32 - bits);
-}
-
-static inline int sign_only(int v)
-{
-    return v ? FFSIGN(v) : 0;
-}
-
-static void predictor_decompress_fir_adapt(int32_t *error_buffer,
-                                           int32_t *buffer_out,
-                                           int output_size,
-                                           int readsamplesize,
-                                           int16_t *predictor_coef_table,
-                                           int predictor_coef_num,
-                                           int predictor_quantitization)
-{
-    int i;
-
-    /* first sample always copies */
-    *buffer_out = *error_buffer;
-
-    if (!predictor_coef_num) {
-        if (output_size <= 1)
-            return;
-
-        memcpy(buffer_out+1, error_buffer+1, (output_size-1) * 4);
-        return;
-    }
-
-    if (predictor_coef_num == 0x1f) { /* 11111 - max value of predictor_coef_num */
-      /* second-best case scenario for fir decompression,
-       * error describes a small difference from the previous sample only
-       */
-        if (output_size <= 1)
-            return;
-        for (i = 0; i < output_size - 1; i++) {
-            int32_t prev_value;
-            int32_t error_value;
-
-            prev_value = buffer_out[i];
-            error_value = error_buffer[i+1];
-            buffer_out[i+1] =
-                extend_sign32((prev_value + error_value), readsamplesize);
-        }
-        return;
-    }
-
-    /* read warm-up samples */
-    if (predictor_coef_num > 0)
-        for (i = 0; i < predictor_coef_num; i++) {
-            int32_t val;
-
-            val = buffer_out[i] + error_buffer[i+1];
-            val = extend_sign32(val, readsamplesize);
-            buffer_out[i+1] = val;
-        }
-
-#if 0
-    /* 4 and 8 are very common cases (the only ones i've seen). these
-     * should be unrolled and optimized
-     */
-    if (predictor_coef_num == 4) {
-        /* FIXME: optimized general case */
-        return;
-    }
-
-    if (predictor_coef_table == 8) {
-        /* FIXME: optimized general case */
-        return;
-    }
-#endif
-
-    /* general case */
-    if (predictor_coef_num > 0) {
-        for (i = predictor_coef_num + 1; i < output_size; i++) {
-            int j;
-            int sum = 0;
-            int outval;
-            int error_val = error_buffer[i];
-
-            for (j = 0; j < predictor_coef_num; j++) {
-                sum += (buffer_out[predictor_coef_num-j] - buffer_out[0]) *
-                       predictor_coef_table[j];
-            }
-
-            outval = (1 << (predictor_quantitization-1)) + sum;
-            outval = outval >> predictor_quantitization;
-            outval = outval + buffer_out[0] + error_val;
-            outval = extend_sign32(outval, readsamplesize);
-
-            buffer_out[predictor_coef_num+1] = outval;
-
-            if (error_val > 0) {
-                int predictor_num = predictor_coef_num - 1;
-
-                while (predictor_num >= 0 && error_val > 0) {
-                    int val = buffer_out[0] - buffer_out[predictor_coef_num - predictor_num];
-                    int sign = sign_only(val);
-
-                    predictor_coef_table[predictor_num] -= sign;
-
-                    val *= sign; /* absolute value */
-
-                    error_val -= ((val >> predictor_quantitization) *
-                                  (predictor_coef_num - predictor_num));
-
-                    predictor_num--;
-                }
-            } else if (error_val < 0) {
-                int predictor_num = predictor_coef_num - 1;
-
-                while (predictor_num >= 0 && error_val < 0) {
-                    int val = buffer_out[0] - buffer_out[predictor_coef_num - predictor_num];
-                    int sign = - sign_only(val);
-
-                    predictor_coef_table[predictor_num] -= sign;
-
-                    val *= sign; /* neg value */
-
-                    error_val -= ((val >> predictor_quantitization) *
-                                  (predictor_coef_num - predictor_num));
-
-                    predictor_num--;
-                }
-            }
-
-            buffer_out++;
-        }
-    }
-}
-
-static void reconstruct_stereo_16(int32_t *buffer[MAX_CHANNELS],
-                                  int16_t *buffer_out,
-                                  int numchannels, int numsamples,
-                                  uint8_t interlacing_shift,
-                                  uint8_t interlacing_leftweight)
-{
-    int i;
-    if (numsamples <= 0)
-        return;
-
-    /* weighted interlacing */
-    if (interlacing_leftweight) {
-        for (i = 0; i < numsamples; i++) {
-            int32_t a, b;
-
-            a = buffer[0][i];
-            b = buffer[1][i];
-
-            a -= (b * interlacing_leftweight) >> interlacing_shift;
-            b += a;
-
-            buffer_out[i*numchannels] = b;
-            buffer_out[i*numchannels + 1] = a;
-        }
-
-        return;
-    }
-
-    /* otherwise basic interlacing took place */
-    for (i = 0; i < numsamples; i++) {
-        int16_t left, right;
-
-        left = buffer[0][i];
-        right = buffer[1][i];
-
-        buffer_out[i*numchannels] = left;
-        buffer_out[i*numchannels + 1] = right;
-    }
-}
-
-static int alac_decode_frame(AVCodecContext *avctx,
-                             void *outbuffer, int *outputsize,
-                             const uint8_t *inbuffer, int input_buffer_size)
-{
-    ALACContext *alac = avctx->priv_data;
-
-    int channels;
-    unsigned int outputsamples;
-    int hassize;
-    unsigned int readsamplesize;
-    int wasted_bytes;
-    int isnotcompressed;
-    uint8_t interlacing_shift;
-    uint8_t interlacing_leftweight;
-
-    /* short-circuit null buffers */
-    if (!inbuffer || !input_buffer_size)
-        return input_buffer_size;
-
-    /* initialize from the extradata */
-    if (!alac->context_initialized) {
-        if (alac->avctx->extradata_size != ALAC_EXTRADATA_SIZE) {
-            av_log(avctx, AV_LOG_ERROR, "alac: expected %d extradata bytes\n",
-                ALAC_EXTRADATA_SIZE);
-            return input_buffer_size;
-        }
-        if (alac_set_info(alac)) {
-            av_log(avctx, AV_LOG_ERROR, "alac: set_info failed\n");
-            return input_buffer_size;
-        }
-        alac->context_initialized = 1;
-    }
-
-    init_get_bits(&alac->gb, inbuffer, input_buffer_size * 8);
-
-    channels = get_bits(&alac->gb, 3) + 1;
-    if (channels > MAX_CHANNELS) {
-        av_log(avctx, AV_LOG_ERROR, "channels > %d not supported\n",
-               MAX_CHANNELS);
-        return input_buffer_size;
-    }
-
-    /* 2^result = something to do with output waiting.
-     * perhaps matters if we read > 1 frame in a pass?
-     */
-    skip_bits(&alac->gb, 4);
-
-    skip_bits(&alac->gb, 12); /* unknown, skip 12 bits */
-
-    /* the output sample size is stored soon */
-    hassize = get_bits1(&alac->gb);
-
-    wasted_bytes = get_bits(&alac->gb, 2); /* unknown ? */
-
-    /* whether the frame is compressed */
-    isnotcompressed = get_bits1(&alac->gb);
-
-    if (hassize) {
-        /* now read the number of samples as a 32bit integer */
-        outputsamples = get_bits_long(&alac->gb, 32);
-        if(outputsamples > alac->setinfo_max_samples_per_frame){
-            av_log(avctx, AV_LOG_ERROR, "outputsamples %d > %d\n", outputsamples, alac->setinfo_max_samples_per_frame);
-            return -1;
-        }
-    } else
-        outputsamples = alac->setinfo_max_samples_per_frame;
-
-    if(outputsamples > *outputsize / alac->bytespersample){
-        av_log(avctx, AV_LOG_ERROR, "sample buffer too small\n");
-        return -1;
-    }
-
-    *outputsize = outputsamples * alac->bytespersample;
-    readsamplesize = alac->setinfo_sample_size - (wasted_bytes * 8) + channels - 1;
-    if (readsamplesize > MIN_CACHE_BITS) {
-        av_log(avctx, AV_LOG_ERROR, "readsamplesize too big (%d)\n", readsamplesize);
-        return -1;
-    }
-
-    if (!isnotcompressed) {
-        /* so it is compressed */
-        int16_t predictor_coef_table[channels][32];
-        int predictor_coef_num[channels];
-        int prediction_type[channels];
-        int prediction_quantitization[channels];
-        int ricemodifier[channels];
-        int i, chan;
-
-        interlacing_shift = get_bits(&alac->gb, 8);
-        interlacing_leftweight = get_bits(&alac->gb, 8);
-
-        for (chan = 0; chan < channels; chan++) {
-            prediction_type[chan] = get_bits(&alac->gb, 4);
-            prediction_quantitization[chan] = get_bits(&alac->gb, 4);
-
-            ricemodifier[chan] = get_bits(&alac->gb, 3);
-            predictor_coef_num[chan] = get_bits(&alac->gb, 5);
-
-            /* read the predictor table */
-            for (i = 0; i < predictor_coef_num[chan]; i++)
-                predictor_coef_table[chan][i] = (int16_t)get_bits(&alac->gb, 16);
-        }
-
-        if (wasted_bytes)
-            av_log(avctx, AV_LOG_ERROR, "FIXME: unimplemented, unhandling of wasted_bytes\n");
-
-        for (chan = 0; chan < channels; chan++) {
-            bastardized_rice_decompress(alac,
-                                        alac->predicterror_buffer[chan],
-                                        outputsamples,
-                                        readsamplesize,
-                                        alac->setinfo_rice_initialhistory,
-                                        alac->setinfo_rice_kmodifier,
-                                        ricemodifier[chan] * alac->setinfo_rice_historymult / 4,
-                                        (1 << alac->setinfo_rice_kmodifier) - 1);
-
-            if (prediction_type[chan] == 0) {
-                /* adaptive fir */
-                predictor_decompress_fir_adapt(alac->predicterror_buffer[chan],
-                                               alac->outputsamples_buffer[chan],
-                                               outputsamples,
-                                               readsamplesize,
-                                               predictor_coef_table[chan],
-                                               predictor_coef_num[chan],
-                                               prediction_quantitization[chan]);
-            } else {
-                av_log(avctx, AV_LOG_ERROR, "FIXME: unhandled prediction type: %i\n", prediction_type[chan]);
-                /* I think the only other prediction type (or perhaps this is
-                 * just a boolean?) runs adaptive fir twice.. like:
-                 * predictor_decompress_fir_adapt(predictor_error, tempout, ...)
-                 * predictor_decompress_fir_adapt(predictor_error, outputsamples ...)
-                 * little strange..
-                 */
-            }
-        }
-    } else {
-        /* not compressed, easy case */
-        int i, chan;
-        for (i = 0; i < outputsamples; i++)
-            for (chan = 0; chan < channels; chan++) {
-                int32_t audiobits;
-
-                audiobits = get_bits_long(&alac->gb, alac->setinfo_sample_size);
-                audiobits = extend_sign32(audiobits, alac->setinfo_sample_size);
-
-                alac->outputsamples_buffer[chan][i] = audiobits;
-            }
-        /* wasted_bytes = 0; */
-        interlacing_shift = 0;
-        interlacing_leftweight = 0;
-    }
-    if (get_bits(&alac->gb, 3) != 7)
-        av_log(avctx, AV_LOG_ERROR, "Error : Wrong End Of Frame\n");
-
-    switch(alac->setinfo_sample_size) {
-    case 16:
-        if (channels == 2) {
-            reconstruct_stereo_16(alac->outputsamples_buffer,
-                                  (int16_t*)outbuffer,
-                                  alac->numchannels,
-                                  outputsamples,
-                                  interlacing_shift,
-                                  interlacing_leftweight);
-        } else {
-            int i;
-            for (i = 0; i < outputsamples; i++) {
-                int16_t sample = alac->outputsamples_buffer[0][i];
-                ((int16_t*)outbuffer)[i * alac->numchannels] = sample;
-            }
-        }
-        break;
-    case 20:
-    case 24:
-        // It is not clear if there exist any encoder that creates 24 bit ALAC
-        // files. iTunes convert 24 bit raw files to 16 bit before encoding.
-    case 32:
-        av_log(avctx, AV_LOG_ERROR, "FIXME: unimplemented sample size %i\n", alac->setinfo_sample_size);
-        break;
-    default:
-        break;
-    }
-
-    if (input_buffer_size * 8 - get_bits_count(&alac->gb) > 8)
-        av_log(avctx, AV_LOG_ERROR, "Error : %d bits left\n", input_buffer_size * 8 - get_bits_count(&alac->gb));
-
-    return input_buffer_size;
-}
-
-static av_cold int alac_decode_init(AVCodecContext * avctx)
-{
-    ALACContext *alac = avctx->priv_data;
-    alac->avctx = avctx;
-    alac->context_initialized = 0;
-
-    alac->numchannels = alac->avctx->channels;
-    alac->bytespersample = 2 * alac->numchannels;
-    avctx->sample_fmt = SAMPLE_FMT_S16;
-
-    return 0;
-}
-
-static av_cold int alac_decode_close(AVCodecContext *avctx)
-{
-    ALACContext *alac = avctx->priv_data;
-
-    int chan;
-    for (chan = 0; chan < MAX_CHANNELS; chan++) {
-        av_free(alac->predicterror_buffer[chan]);
-        av_free(alac->outputsamples_buffer[chan]);
-    }
-
-    return 0;
-}
-
-AVCodec alac_decoder = {
-    "alac",
-    CODEC_TYPE_AUDIO,
-    CODEC_ID_ALAC,
-    sizeof(ALACContext),
-    alac_decode_init,
-    NULL,
-    alac_decode_close,
-    alac_decode_frame,
-    .long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),
-};