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58 Commits

Author SHA1 Message Date
Michael Fabian 'Xaymar' Dirks 4c09db4eac ci: Add patch for updated AMF 2022-01-13 00:16:44 +01:00
Michael Fabian 'Xaymar' Dirks e13759d272 ci: Use hidden .env file to store environmental updates 2022-01-10 01:29:48 +01:00
Michael Fabian 'Xaymar' Dirks a2e7cca590 ci: Figure out why scripts aren't working right 2022-01-10 01:12:45 +01:00
Michael Fabian 'Xaymar' Dirks f2e7cd16c5 ci: Add AOM 2022-01-10 00:52:47 +01:00
Michael Fabian 'Xaymar' Dirks 9a0470f317 ci: Add libx264 2022-01-10 00:42:01 +01:00
Michael Fabian 'Xaymar' Dirks d99e9e9c74 ci: Add zLib 2022-01-10 00:41:44 +01:00
Michael Fabian 'Xaymar' Dirks ba8761110f ci: Always use twice the "ideal" amount of threads 2022-01-10 00:31:30 +01:00
Michael Fabian 'Xaymar' Dirks 8adf994e36 ci: Don't limit parallel builds 2022-01-10 00:31:30 +01:00
Michael Fabian 'Xaymar' Dirks be09d8675e ci: Add NVIDIA NVENC 2022-01-10 00:31:30 +01:00
Michael Fabian 'Xaymar' Dirks 8aa7731030 ci: Fix uploaded file names 2022-01-10 00:31:30 +01:00
Michael Fabian 'Xaymar' Dirks e357430b22 ci: Add AMD AMF v1.4.23 2022-01-10 00:31:19 +01:00
Michael Fabian 'Xaymar' Dirks 5c9a68281a ci: Move to a script based system for easier local repro
Also limit parallel execution of automation commands so they don't break each other randomly.
2022-01-10 00:31:19 +01:00
Michael Fabian 'Xaymar' Dirks aef6bb2532 ci: Let FFmpeg figure things out by itself 2022-01-10 00:29:26 +01:00
Michael Fabian 'Xaymar' Dirks c8a19476c0 patches: Remove incompatible patch from master 2022-01-10 00:29:23 +01:00
Michael Fabian 'Xaymar' Dirks 1394ca0edb ci: Add AOM and improve some dependencies 2022-01-10 00:29:23 +01:00
Michael Fabian 'Xaymar' Dirks 64daba4a66 ci: x264 0.161.3049 -> 0.163.3060 2021-09-07 18:29:38 +02:00
Michael Fabian 'Xaymar' Dirks 3cfe4b67a2 ci: Use matrix instead of parameter for patches
The matrix build type can take care of most if not all optional configurations, so there is no point in having this as an argument. In any case we'd want both normal and patched builds anyway.
2021-05-23 22:28:20 +02:00
Michael Fabian 'Xaymar' Dirks 8a11745d31 ci: Prefer pre-built zlib-ng 2021-05-23 22:24:11 +02:00
Michael Fabian 'Xaymar' Dirks c7f4b1690a 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 We also shouldn't bother with 32-bit anymore, as it has been replaced by 64-bit on Windows.
2021-05-23 21:57:06 +02:00
Michael Fabian 'Xaymar' Dirks cc54016f3a ci: Only package necessary binaries 2021-05-23 20:54:20 +02:00
Michael Fabian 'Xaymar' Dirks 07af655f36 patches: Fix 45be7d3194 for 4.2 2021-05-23 20:54:20 +02:00
Michael Fabian 'Xaymar' Dirks 45be7d3194 patches: Add "avformat/matroskaenc: Allow changing the time stamp precision via option"
Adds "timestamp_precision" to the available options for Matroska muxing.
The option enables users and developers to change the precision of the
time stamps in the Matroska container up to 1 nanosecond, which can aid
with the proper detection of constant and variable rate content.

Work-around fix for: 259, 6406, 7927, 8909 and 9124.

Signed-off-by: Michael Fabian 'Xaymar' Dirks <michael.dirks@xaymar.com>
2021-05-23 16:35:56 +02:00
Michael Fabian 'Xaymar' Dirks b0d8f23b7b ci: Add custom patches and improve version detection 2021-05-23 16:35:56 +02:00
Michael Fabian 'Xaymar' Dirks 96e3d9f125 ci: Don't enable non-free CUDA 2021-05-23 16:35:31 +02:00
Michael Fabian 'Xaymar' Dirks 7750b0ddab ci: Don't add remotes if not scheduled run 2021-05-23 00:00:21 +02:00
Michael Fabian 'Xaymar' Dirks 53d6f50a0f ci: Make zlib-ng binaries compatible with MSVC 2021-05-22 07:20:10 +02:00
Michael Fabian 'Xaymar' Dirks 9b9751ba31 ci: Add zlib via zlib-ng 2021-05-22 06:27:44 +02:00
Michael Fabian 'Xaymar' Dirks d7c1d4d7f8 ci: Add version information to ffnvcodec and amf 2021-05-22 05:01:59 +02:00
Michael Fabian 'Xaymar' Dirks 4fd8be88e7 ci: Don't limit parallel builds 2021-05-22 03:15:56 +02:00
Michael Fabian 'Xaymar' Dirks d42a6366d3 ci: Only build master on push 2021-05-22 03:15:38 +02:00
Michael Fabian 'Xaymar' Dirks 6ef38695a2 ci: Add AMD AMF for FFmpeg 4.0+ 2021-05-22 03:06:30 +02:00
Michael Fabian 'Xaymar' Dirks a48b1e39f6 ci: Enable CUVID for 4.0+ 2021-05-22 02:37:50 +02:00
Michael Fabian 'Xaymar' Dirks 32330c349c ci: Upgrade x264 to 0.161.3049 2021-05-22 02:34:49 +02:00
Michael Fabian 'Xaymar' Dirks f77f194f2e ci: Always check out the remote 2021-05-22 02:34:49 +02:00
Michael Fabian 'Xaymar' Dirks c44ac7171c ci: Build GPL first, then LGPL 2021-05-22 02:34:49 +02:00
Michael Fabian 'Xaymar' Dirks 89507374e7 ci: Actually check for version numbers 2021-05-22 02:34:49 +02:00
Michael Fabian 'Xaymar' Dirks 547173be40 ci: Add build version to artifacts 2021-05-22 00:30:33 +02:00
Michael Fabian 'Xaymar' Dirks 634234677b ci: Protect against configure issues from version difference 2021-05-19 05:32:17 +02:00
Michael Fabian 'Xaymar' Dirks c7f5b99d34 ci: Improve build priorities with matrix order changes
We want to prioritize bitness over type over license over license version.
2021-05-19 05:06:27 +02:00
Michael Fabian 'Xaymar' Dirks f5857e38f5 ci: Only trigger builds on manual push 2021-05-19 05:01:11 +02:00
Michael Fabian 'Xaymar' Dirks c65e769e09 ci: Also push tags provided by FFmpeg remote 2021-05-19 04:52:39 +02:00
Michael Fabian 'Xaymar' Dirks 8ca8a76581 ci: Limit the number of concurrent builds to 4 2021-05-19 04:52:35 +02:00
Michael Fabian 'Xaymar' Dirks ea7406c8bc ci: Don't fail if the branch is missing 2021-05-19 04:50:32 +02:00
Michael Fabian 'Xaymar' Dirks 571bc81584 ci: Simplify NVIDIA CUDA/CUVID/NVDEC/NVENC logic 2021-05-19 04:50:32 +02:00
Michael Fabian 'Xaymar' Dirks 803fc7df14 ci: Only update once per week 2021-05-19 04:50:32 +02:00
Michael Fabian 'Xaymar' Dirks 224421162c ci: Track versions 4.0 and 4.4 2021-05-19 04:19:36 +02:00
Michael Fabian 'Xaymar' Dirks 71f6bcb3d1 ci: Add support for NVIDIA Codec Headers (v11.0.10.0)
Encoders
- h264_nvenc
- hevc_nvenc

Decoders:
- av1_cuvid
- h264_cuvid
- hevc_cuvid
- mjpeg_cuvid
- mpeg1_cuvid
- mpeg2_cuvid
- mpeg4_cuvid
- vc1_cuvid
- vp8_cuvid
- vp9_cuvid

Filters:
- hwupload_cuda
- yadif_cuda
- scale_cuda
- thumbnail_cuda
- overlay_cuda
2020-11-22 23:13:16 +01:00
Michael Fabian 'Xaymar' Dirks e20853ea67 ci: Fix missing license version in artifacts 2020-11-22 23:05:59 +01:00
Michael Fabian 'Xaymar' Dirks d1f3ff282a ci: Use system pkg-config instead of MinGW one 2020-11-22 22:42:44 +01:00
Michael Fabian 'Xaymar' Dirks 84537d4761 ci: Split License from License Version 2020-11-22 18:30:28 +01:00
Michael Fabian 'Xaymar' Dirks edb102cf45 ci: Run pushed builds on the correct branch 2020-11-22 04:37:14 +01:00
Michael Fabian 'Xaymar' Dirks 97a2ea1135 ci: Update x264 to 0.161.3020 2020-11-22 02:55:19 +01:00
Michael Fabian 'Xaymar' Dirks a15e077819 ci: Reduce refresh rate to once a day 2020-11-01 12:16:06 +01:00
Michael Fabian 'Xaymar' Dirks c6611ca1b6 ci: Fix refresh failure on branch differences 2020-10-18 03:01:08 +02:00
Michael Fabian 'Xaymar' Dirks 141b529c73 ci: Fix incorrect output reference 2020-10-18 02:47:30 +02:00
Michael Fabian 'Xaymar' Dirks f1929abc8b ci: Add support for x264 (v0.160.3011) 2020-10-18 02:24:44 +02:00
Michael Fabian 'Xaymar' Dirks 66f68f5ce9 ci: Build FFmpeg based on trigger 2020-10-18 02:23:47 +02:00
Michael Fabian 'Xaymar' Dirks edb8c65d0f ci: Update branches hourly 2020-10-18 02:22:07 +02:00
2723 changed files with 1836 additions and 742188 deletions
+110
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name: Build
on:
workflow_dispatch:
inputs:
ref:
description: 'Branch, Tag or Commit to build'
required: false
default: 'master'
env:
AMDAMF_VERSION: "v1.4.23"
AOM_VERSION: "v3.1.2.115"
MINGW_VERSION: "v9.0.0"
NVIDIANVENC_VERSION: "n11.0.10.0"
X264_VERSION: "0.163.3060"
ZLIB_VERSION: "2.0.3"
jobs:
cc:
runs-on: ubuntu-20.04
strategy:
matrix:
license: [ "LGPL", "GPL" ]
license_version: [ 2, 3 ]
type: [ "shared" ]
bits: [ 64 ]
patches: [ false, true ]
name: "Windows (${{ matrix.bits }}bit, ${{ matrix.type }}, ${{ matrix.license }}v${{ matrix.license_version}}, patched=${{ matrix.patches }})"
env:
BUILD_TYPE: ${{ matrix.type }}
BUILD_BITS: ${{ matrix.bits }}
LICENSE: ${{ matrix.license }}
LICENSE_VERSION: ${{ matrix.license_version }}
SCRIPTROOT: "/tmp"
steps:
- name: "automation: Check-out"
uses: actions/checkout@v2
with:
submodules: "recursive"
fetch-depth: 1
- name: "automation: Copy things to safety"
shell: bash
run: |
# Copy important scripts to "safe" directory".
cp -R -f ./patches ${SCRIPTROOT}/patches
cp -R -f ./scripts ${SCRIPTROOT}/scripts
chmod +x ${SCRIPTROOT}/scripts/*.sh
cp -R -f ./addons ${SCRIPTROOT}/addons
chmod +x ${SCRIPTROOT}/addons/*.sh
- name: "dependency: cmake, make, pkg-config, mingw, nasm"
shell: bash
run: |
sudo apt-get update
sudo apt-get install \
build-essential git \
cmake make ninja-build \
pkg-config \
mingw-w64 mingw-w64-tools gcc-mingw-w64 g++-mingw-w64 \
nasm
- name: "ffmpeg: Check out '${{ github.event.inputs.ref }}'"
uses: actions/checkout@v2
with:
ref: "${{ github.event.inputs.ref }}"
submodules: "recursive"
fetch-depth: 0
- name: "automation: Bootstrap"
id: bootstrap
shell: bash
run: |
. ${SCRIPTROOT}/addons/bootstrap.sh
echo "FFmpeg v${VERSION_MAJOR}.${VERSION_MINOR}.${VERSION_PATCH}.${VERSION_TWEAK}-${COMMIT}"
- name: "automation: Apply Patches"
if: ${{ (matrix.patches == true) }}
id: patch
env:
VERSION_MAJOR: ${{ steps.bootstrap.outputs.VERSION_MAJOR }}
VERSION_MINOR: ${{ steps.bootstrap.outputs.VERSION_MINOR }}
VERSION_PATCH: ${{ steps.bootstrap.outputs.VERSION_PATCH }}
VERSION_TWEAK: ${{ steps.bootstrap.outputs.VERSION_TWEAK }}
COMMIT: ${{ steps.bootstrap.outputs.COMMIT }}
shell: bash
run: |
. ${SCRIPTROOT}/addons/patch.sh
echo "::set-output name=name::-patched"
- name: "automation: Run scripts"
env:
VERSION_MAJOR: ${{ steps.bootstrap.outputs.VERSION_MAJOR }}
VERSION_MINOR: ${{ steps.bootstrap.outputs.VERSION_MINOR }}
VERSION_PATCH: ${{ steps.bootstrap.outputs.VERSION_PATCH }}
VERSION_TWEAK: ${{ steps.bootstrap.outputs.VERSION_TWEAK }}
COMMIT: ${{ steps.bootstrap.outputs.COMMIT }}
BUILD_ARCH: ${{ steps.bootstrap.outputs.BUILD_ARCH }}
BUILD_TARGET: ${{ steps.bootstrap.outputs.BUILD_TARGET }}
BUILD_PREFIX: ${{ steps.bootstrap.outputs.BUILD_PREFIX }}
BUILD_FLAGS: ${{ steps.bootstrap.outputs.BUILD_FLAGS }}
shell: bash
run: |
. ${SCRIPTROOT}/addons/run.sh
- name: "automation: Upload Artifacts"
uses: actions/upload-artifact@v1
with:
name: "ffmpeg-${{ matrix.license }}v${{ matrix.license_version }}-${{ matrix.bits}}-${{ matrix.type }}-${{ steps.bootstrap.outputs.VERSION_MAJOR }}.${{ steps.bootstrap.outputs.VERSION_MINOR }}.${{ steps.bootstrap.outputs.VERSION_PATCH }}.${{ steps.bootstrap.outputs.VERSION_TWEAK }}-${{ steps.bootstrap.outputs.COMMIT }}${{ steps.patch.outputs.name }}"
path: distrib
+110
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name: Refresh
on:
schedule:
- cron: '0 0 * * 0'
push:
branches:
- 'automation'
- 'automation-test'
concurrency:
group: "refresh"
cancel-in-progress: false
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}
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"
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
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"
done
-56
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@@ -1,56 +0,0 @@
.config
.version
*.o
*.d
*.exe
*.ho
*-example
*-test
*_g
*.def
*.dll
*.lib
*.exp
config.*
doc/*.1
doc/*.html
doc/*.pod
doc/fate.txt
doxy
ffmpeg
ffplay
ffprobe
ffserver
avconv
libavcodec/*_tablegen
libavcodec/*_tables.c
libavcodec/*_tables.h
libavcodec/codec_names.h
libavcodec/libavcodec*
libavcore/libavcore*
libavdevice/libavdevice*
libavfilter/libavfilter*
libavformat/libavformat*
libavutil/avconfig.h
libavutil/libavutil*
libpostproc/libpostproc*
libswresample/libswresample*
libswscale/libswscale*
tests/audiogen
tests/base64
tests/data
tests/rotozoom
tests/tiny_psnr
tests/videogen
tests/vsynth1
tests/vsynth2
tools/aviocat
tools/cws2fws
tools/graph2dot
tools/ismindex
tools/lavfi-showfiltfmts
tools/pktdumper
tools/probetest
tools/qt-faststart
tools/trasher
version.h
-339
View File
@@ -1,339 +0,0 @@
GNU GENERAL PUBLIC LICENSE
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Copyright (C) 1989, 1991 Free Software Foundation, Inc.,
51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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To protect your rights, we need to make restrictions that forbid
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control compilation and installation of the executable. However, as a
special exception, the source code distributed need not include
anything that is normally distributed (in either source or binary
form) with the major components (compiler, kernel, and so on) of the
operating system on which the executable runs, unless that component
itself accompanies the executable.
If distribution of executable or object code is made by offering
access to copy from a designated place, then offering equivalent
access to copy the source code from the same place counts as
distribution of the source code, even though third parties are not
compelled to copy the source along with the object code.
4. You may not copy, modify, sublicense, or distribute the Program
except as expressly provided under this License. Any attempt
otherwise to copy, modify, sublicense or distribute the Program is
void, and will automatically terminate your rights under this License.
However, parties who have received copies, or rights, from you under
this License will not have their licenses terminated so long as such
parties remain in full compliance.
5. You are not required to accept this License, since you have not
signed it. However, nothing else grants you permission to modify or
distribute the Program or its derivative works. These actions are
prohibited by law if you do not accept this License. Therefore, by
modifying or distributing the Program (or any work based on the
Program), you indicate your acceptance of this License to do so, and
all its terms and conditions for copying, distributing or modifying
the Program or works based on it.
6. Each time you redistribute the Program (or any work based on the
Program), the recipient automatically receives a license from the
original licensor to copy, distribute or modify the Program subject to
these terms and conditions. You may not impose any further
restrictions on the recipients' exercise of the rights granted herein.
You are not responsible for enforcing compliance by third parties to
this License.
7. If, as a consequence of a court judgment or allegation of patent
infringement or for any other reason (not limited to patent issues),
conditions are imposed on you (whether by court order, agreement or
otherwise) that contradict the conditions of this License, they do not
excuse you from the conditions of this License. If you cannot
distribute so as to satisfy simultaneously your obligations under this
License and any other pertinent obligations, then as a consequence you
may not distribute the Program at all. For example, if a patent
license would not permit royalty-free redistribution of the Program by
all those who receive copies directly or indirectly through you, then
the only way you could satisfy both it and this License would be to
refrain entirely from distribution of the Program.
If any portion of this section is held invalid or unenforceable under
any particular circumstance, the balance of the section is intended to
apply and the section as a whole is intended to apply in other
circumstances.
It is not the purpose of this section to induce you to infringe any
patents or other property right claims or to contest validity of any
such claims; this section has the sole purpose of protecting the
integrity of the free software distribution system, which is
implemented by public license practices. Many people have made
generous contributions to the wide range of software distributed
through that system in reliance on consistent application of that
system; it is up to the author/donor to decide if he or she is willing
to distribute software through any other system and a licensee cannot
impose that choice.
This section is intended to make thoroughly clear what is believed to
be a consequence of the rest of this License.
8. If the distribution and/or use of the Program is restricted in
certain countries either by patents or by copyrighted interfaces, the
original copyright holder who places the Program under this License
may add an explicit geographical distribution limitation excluding
those countries, so that distribution is permitted only in or among
countries not thus excluded. In such case, this License incorporates
the limitation as if written in the body of this License.
9. The Free Software Foundation may publish revised and/or new versions
of the General Public License from time to time. Such new versions will
be similar in spirit to the present version, but may differ in detail to
address new problems or concerns.
Each version is given a distinguishing version number. If the Program
specifies a version number of this License which applies to it and "any
later version", you have the option of following the terms and conditions
either of that version or of any later version published by the Free
Software Foundation. If the Program does not specify a version number of
this License, you may choose any version ever published by the Free Software
Foundation.
10. If you wish to incorporate parts of the Program into other free
programs whose distribution conditions are different, write to the author
to ask for permission. For software which is copyrighted by the Free
Software Foundation, write to the Free Software Foundation; we sometimes
make exceptions for this. Our decision will be guided by the two goals
of preserving the free status of all derivatives of our free software and
of promoting the sharing and reuse of software generally.
NO WARRANTY
11. BECAUSE THE PROGRAM IS LICENSED FREE OF CHARGE, THERE IS NO WARRANTY
FOR THE PROGRAM, TO THE EXTENT PERMITTED BY APPLICABLE LAW. EXCEPT WHEN
OTHERWISE STATED IN WRITING THE COPYRIGHT HOLDERS AND/OR OTHER PARTIES
PROVIDE THE PROGRAM "AS IS" WITHOUT WARRANTY OF ANY KIND, EITHER EXPRESSED
OR IMPLIED, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE. THE ENTIRE RISK AS
TO THE QUALITY AND PERFORMANCE OF THE PROGRAM IS WITH YOU. SHOULD THE
PROGRAM PROVE DEFECTIVE, YOU ASSUME THE COST OF ALL NECESSARY SERVICING,
REPAIR OR CORRECTION.
12. IN NO EVENT UNLESS REQUIRED BY APPLICABLE LAW OR AGREED TO IN WRITING
WILL ANY COPYRIGHT HOLDER, OR ANY OTHER PARTY WHO MAY MODIFY AND/OR
REDISTRIBUTE THE PROGRAM AS PERMITTED ABOVE, BE LIABLE TO YOU FOR DAMAGES,
INCLUDING ANY GENERAL, SPECIAL, INCIDENTAL OR CONSEQUENTIAL DAMAGES ARISING
OUT OF THE USE OR INABILITY TO USE THE PROGRAM (INCLUDING BUT NOT LIMITED
TO LOSS OF DATA OR DATA BEING RENDERED INACCURATE OR LOSSES SUSTAINED BY
YOU OR THIRD PARTIES OR A FAILURE OF THE PROGRAM TO OPERATE WITH ANY OTHER
PROGRAMS), EVEN IF SUCH HOLDER OR OTHER PARTY HAS BEEN ADVISED OF THE
POSSIBILITY OF SUCH DAMAGES.
END OF TERMS AND CONDITIONS
How to Apply These Terms to Your New Programs
If you develop a new program, and you want it to be of the greatest
possible use to the public, the best way to achieve this is to make it
free software which everyone can redistribute and change under these terms.
To do so, attach the following notices to the program. It is safest
to attach them to the start of each source file to most effectively
convey the exclusion of warranty; and each file should have at least
the "copyright" line and a pointer to where the full notice is found.
<one line to give the program's name and a brief idea of what it does.>
Copyright (C) <year> <name of author>
This program 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 of the License, or
(at your option) any later version.
This program 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 this program; if not, write to the Free Software Foundation, Inc.,
51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
Also add information on how to contact you by electronic and paper mail.
If the program is interactive, make it output a short notice like this
when it starts in an interactive mode:
Gnomovision version 69, Copyright (C) year name of author
Gnomovision comes with ABSOLUTELY NO WARRANTY; for details type `show w'.
This is free software, and you are welcome to redistribute it
under certain conditions; type `show c' for details.
The hypothetical commands `show w' and `show c' should show the appropriate
parts of the General Public License. Of course, the commands you use may
be called something other than `show w' and `show c'; they could even be
mouse-clicks or menu items--whatever suits your program.
You should also get your employer (if you work as a programmer) or your
school, if any, to sign a "copyright disclaimer" for the program, if
necessary. Here is a sample; alter the names:
Yoyodyne, Inc., hereby disclaims all copyright interest in the program
`Gnomovision' (which makes passes at compilers) written by James Hacker.
<signature of Ty Coon>, 1 April 1989
Ty Coon, President of Vice
This General Public License does not permit incorporating your program into
proprietary programs. If your program is a subroutine library, you may
consider it more useful to permit linking proprietary applications with the
library. If this is what you want to do, use the GNU Lesser General
Public License instead of this License.
-674
View File
@@ -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
of this license document, but changing it is not allowed.
Preamble
The GNU General Public License is a free, copyleft license for
software and other kinds of works.
The licenses for most software and other practical works are designed
to take away your freedom to share and change the works. By contrast,
the GNU General Public License is intended to guarantee your freedom to
share and change all versions of a program--to make sure it remains free
software for all its users. We, the Free Software Foundation, use the
GNU General Public License for most of our software; it applies also to
any other work released this way by its authors. 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
them 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 prevent others from denying you
these rights or asking you to surrender the rights. Therefore, you have
certain responsibilities if you distribute copies of the software, or if
you modify it: responsibilities to respect the freedom of others.
For example, if you distribute copies of such a program, whether
gratis or for a fee, you must pass on to the recipients the same
freedoms that you received. You must make sure that they, too, receive
or can get the source code. And you must show them these terms so they
know their rights.
Developers that use the GNU GPL protect your rights with two steps:
(1) assert copyright on the software, and (2) offer you this License
giving you legal permission to copy, distribute and/or modify it.
For the developers' and authors' protection, the GPL clearly explains
that there is no warranty for this free software. For both users' and
authors' sake, the GPL requires that modified versions be marked as
changed, so that their problems will not be attributed erroneously to
authors of previous versions.
Some devices are designed to deny users access to install or run
modified versions of the software inside them, although the manufacturer
can do so. This is fundamentally incompatible with the aim of
protecting users' freedom to change the software. The systematic
pattern of such abuse occurs in the area of products for individuals to
use, which is precisely where it is most unacceptable. Therefore, we
have designed this version of the GPL to prohibit the practice for those
products. If such problems arise substantially in other domains, we
stand ready to extend this provision to those domains in future versions
of the GPL, as needed to protect the freedom of users.
Finally, every program is threatened constantly by software patents.
States should not allow patents to restrict development and use of
software on general-purpose computers, but in those that do, we wish to
avoid the special danger that patents applied to a free program could
make it effectively proprietary. To prevent this, the GPL assures that
patents cannot be used to render the program non-free.
The precise terms and conditions for copying, distribution and
modification follow.
TERMS AND CONDITIONS
0. Definitions.
"This License" refers to version 3 of the GNU General Public License.
"Copyright" also means copyright-like laws that apply to other kinds of
works, such as semiconductor masks.
"The Program" refers to any copyrightable work licensed under this
License. Each licensee is addressed as "you". "Licensees" and
"recipients" may be individuals or organizations.
To "modify" a work means to copy from or adapt all or part of the work
in a fashion requiring copyright permission, other than the making of an
exact copy. The resulting work is called a "modified version" of the
earlier work or a work "based on" the earlier work.
A "covered work" means either the unmodified Program or a work based
on the Program.
To "propagate" a work means to do anything with it that, without
permission, would make you directly or secondarily liable for
infringement under applicable copyright law, except executing it on a
computer or modifying a private copy. Propagation includes copying,
distribution (with or without modification), making available to the
public, and in some countries other activities as well.
To "convey" a work means any kind of propagation that enables other
parties to make or receive copies. Mere interaction with a user through
a computer network, with no transfer of a copy, is not conveying.
An interactive user interface displays "Appropriate Legal Notices"
to the extent that it includes a convenient and prominently visible
feature that (1) displays an appropriate copyright notice, and (2)
tells the user that there is no warranty for the work (except to the
extent that warranties are provided), that licensees may convey the
work under this License, and how to view a copy of this License. If
the interface presents a list of user commands or options, such as a
menu, a prominent item in the list meets this criterion.
1. Source Code.
The "source code" for a work means the preferred form of the work
for making modifications to it. "Object code" means any non-source
form of a work.
A "Standard Interface" means an interface that either is an official
standard defined by a recognized standards body, or, in the case of
interfaces specified for a particular programming language, one that
is widely used among developers working in that language.
The "System Libraries" of an executable work include anything, other
than the work as a whole, that (a) is included in the normal form of
packaging a Major Component, but which is not part of that Major
Component, and (b) serves only to enable use of the work with that
Major Component, or to implement a Standard Interface for which an
implementation is available to the public in source code form. A
"Major Component", in this context, means a major essential component
(kernel, window system, and so on) of the specific operating system
(if any) on which the executable work runs, or a compiler used to
produce the work, or an object code interpreter used to run it.
The "Corresponding Source" for a work in object code form means all
the source code needed to generate, install, and (for an executable
work) run the object code and to modify the work, including scripts to
control those activities. However, it does not include the work's
System Libraries, or general-purpose tools or generally available free
programs which are used unmodified in performing those activities but
which are not part of the work. For example, Corresponding Source
includes interface definition files associated with source files for
the work, and the source code for shared libraries and dynamically
linked subprograms that the work is specifically designed to require,
such as by intimate data communication or control flow between those
subprograms and other parts of the work.
The Corresponding Source need not include anything that users
can regenerate automatically from other parts of the Corresponding
Source.
The Corresponding Source for a work in source code form is that
same work.
2. Basic Permissions.
All rights granted under this License are granted for the term of
copyright on the Program, and are irrevocable provided the stated
conditions are met. This License explicitly affirms your unlimited
permission to run the unmodified Program. The output from running a
covered work is covered by this License only if the output, given its
content, constitutes a covered work. This License acknowledges your
rights of fair use or other equivalent, as provided by copyright law.
You may make, run and propagate covered works that you do not
convey, without conditions so long as your license otherwise remains
in force. You may convey covered works to others for the sole purpose
of having them make modifications exclusively for you, or provide you
with facilities for running those works, provided that you comply with
the terms of this License in conveying all material for which you do
not control copyright. Those thus making or running the covered works
for you must do so exclusively on your behalf, under your direction
and control, on terms that prohibit them from making any copies of
your copyrighted material outside their relationship with you.
Conveying under any other circumstances is permitted solely under
the conditions stated below. Sublicensing is not allowed; section 10
makes it unnecessary.
3. Protecting Users' Legal Rights From Anti-Circumvention Law.
No covered work shall be deemed part of an effective technological
measure under any applicable law fulfilling obligations under article
11 of the WIPO copyright treaty adopted on 20 December 1996, or
similar laws prohibiting or restricting circumvention of such
measures.
When you convey a covered work, you waive any legal power to forbid
circumvention of technological measures to the extent such circumvention
is effected by exercising rights under this License with respect to
the covered work, and you disclaim any intention to limit operation or
modification of the work as a means of enforcing, against the work's
users, your or third parties' legal rights to forbid circumvention of
technological measures.
4. Conveying Verbatim Copies.
You may convey verbatim copies of the Program's source code as you
receive it, in any medium, provided that you conspicuously and
appropriately publish on each copy an appropriate copyright notice;
keep intact all notices stating that this License and any
non-permissive terms added in accord with section 7 apply to the code;
keep intact all notices of the absence of any warranty; and give all
recipients a copy of this License along with the Program.
You may charge any price or no price for each copy that you convey,
and you may offer support or warranty protection for a fee.
5. Conveying Modified Source Versions.
You may convey a work based on the Program, or the modifications to
produce it from the Program, in the form of source code under the
terms of section 4, provided that you also meet all of these conditions:
a) The work must carry prominent notices stating that you modified
it, and giving a relevant date.
b) The work must carry prominent notices stating that it is
released under this License and any conditions added under section
7. This requirement modifies the requirement in section 4 to
"keep intact all notices".
c) You must license the entire work, as a whole, under this
License to anyone who comes into possession of a copy. This
License will therefore apply, along with any applicable section 7
additional terms, to the whole of the work, and all its parts,
regardless of how they are packaged. This License gives no
permission to license the work in any other way, but it does not
invalidate such permission if you have separately received it.
d) If the work has interactive user interfaces, each must display
Appropriate Legal Notices; however, if the Program has interactive
interfaces that do not display Appropriate Legal Notices, your
work need not make them do so.
A compilation of a covered work with other separate and independent
works, which are not by their nature extensions of the covered work,
and which are not combined with it such as to form a larger program,
in or on a volume of a storage or distribution medium, is called an
"aggregate" if the compilation and its resulting copyright are not
used to limit the access or legal rights of the compilation's users
beyond what the individual works permit. Inclusion of a covered work
in an aggregate does not cause this License to apply to the other
parts of the aggregate.
6. Conveying Non-Source Forms.
You may convey a covered work in object code form under the terms
of sections 4 and 5, provided that you also convey the
machine-readable Corresponding Source under the terms of this License,
in one of these ways:
a) Convey the object code in, or embodied in, a physical product
(including a physical distribution medium), accompanied by the
Corresponding Source fixed on a durable physical medium
customarily used for software interchange.
b) Convey the object code in, or embodied in, a physical product
(including a physical distribution medium), accompanied by a
written offer, valid for at least three years and valid for as
long as you offer spare parts or customer support for that product
model, to give anyone who possesses the object code either (1) a
copy of the Corresponding Source for all the software in the
product that is covered by this License, on a durable physical
medium customarily used for software interchange, for a price no
more than your reasonable cost of physically performing this
conveying of source, or (2) access to copy the
Corresponding Source from a network server at no charge.
c) Convey individual copies of the object code with a copy of the
written offer to provide the Corresponding Source. This
alternative is allowed only occasionally and noncommercially, and
only if you received the object code with such an offer, in accord
with subsection 6b.
d) Convey the object code by offering access from a designated
place (gratis or for a charge), and offer equivalent access to the
Corresponding Source in the same way through the same place at no
further charge. You need not require recipients to copy the
Corresponding Source along with the object code. If the place to
copy the object code is a network server, the Corresponding Source
may be on a different server (operated by you or a third party)
that supports equivalent copying facilities, provided you maintain
clear directions next to the object code saying where to find the
Corresponding Source. Regardless of what server hosts the
Corresponding Source, you remain obligated to ensure that it is
available for as long as needed to satisfy these requirements.
e) Convey the object code using peer-to-peer transmission, provided
you inform other peers where the object code and Corresponding
Source of the work are being offered to the general public at no
charge under subsection 6d.
A separable portion of the object code, whose source code is excluded
from the Corresponding Source as a System Library, need not be
included in conveying the object code work.
A "User Product" is either (1) a "consumer product", which means any
tangible personal property which is normally used for personal, family,
or household purposes, or (2) anything designed or sold for incorporation
into a dwelling. In determining whether a product is a consumer product,
doubtful cases shall be resolved in favor of coverage. For a particular
product received by a particular user, "normally used" refers to a
typical or common use of that class of product, regardless of the status
of the particular user or of the way in which the particular user
actually uses, or expects or is expected to use, the product. A product
is a consumer product regardless of whether the product has substantial
commercial, industrial or non-consumer uses, unless such uses represent
the only significant mode of use of the product.
"Installation Information" for a User Product means any methods,
procedures, authorization keys, or other information required to install
and execute modified versions of a covered work in that User Product from
a modified version of its Corresponding Source. The information must
suffice to ensure that the continued functioning of the modified object
code is in no case prevented or interfered with solely because
modification has been made.
If you convey an object code work under this section in, or with, or
specifically for use in, a User Product, and the conveying occurs as
part of a transaction in which the right of possession and use of the
User Product is transferred to the recipient in perpetuity or for a
fixed term (regardless of how the transaction is characterized), the
Corresponding Source conveyed under this section must be accompanied
by the Installation Information. But this requirement does not apply
if neither you nor any third party retains the ability to install
modified object code on the User Product (for example, the work has
been installed in ROM).
The requirement to provide Installation Information does not include a
requirement to continue to provide support service, warranty, or updates
for a work that has been modified or installed by the recipient, or for
the User Product in which it has been modified or installed. Access to a
network may be denied when the modification itself materially and
adversely affects the operation of the network or violates the rules and
protocols for communication across the network.
Corresponding Source conveyed, and Installation Information provided,
in accord with this section must be in a format that is publicly
documented (and with an implementation available to the public in
source code form), and must require no special password or key for
unpacking, reading or copying.
7. Additional Terms.
"Additional permissions" are terms that supplement the terms of this
License by making exceptions from one or more of its conditions.
Additional permissions that are applicable to the entire Program shall
be treated as though they were included in this License, to the extent
that they are valid under applicable law. If additional permissions
apply only to part of the Program, that part may be used separately
under those permissions, but the entire Program remains governed by
this License without regard to the additional permissions.
When you convey a copy of a covered work, you may at your option
remove any additional permissions from that copy, or from any part of
it. (Additional permissions may be written to require their own
removal in certain cases when you modify the work.) You may place
additional permissions on material, added by you to a covered work,
for which you have or can give appropriate copyright permission.
Notwithstanding any other provision of this License, for material you
add to a covered work, you may (if authorized by the copyright holders of
that material) supplement the terms of this License with terms:
a) Disclaiming warranty or limiting liability differently from the
terms of sections 15 and 16 of this License; or
b) Requiring preservation of specified reasonable legal notices or
author attributions in that material or in the Appropriate Legal
Notices displayed by works containing it; or
c) Prohibiting misrepresentation of the origin of that material, or
requiring that modified versions of such material be marked in
reasonable ways as different from the original version; or
d) Limiting the use for publicity purposes of names of licensors or
authors of the material; or
e) Declining to grant rights under trademark law for use of some
trade names, trademarks, or service marks; or
f) Requiring indemnification of licensors and authors of that
material by anyone who conveys the material (or modified versions of
it) with contractual assumptions of liability to the recipient, for
any liability that these contractual assumptions directly impose on
those licensors and authors.
All other non-permissive additional terms are considered "further
restrictions" within the meaning of section 10. If the Program as you
received it, or any part of it, contains a notice stating that it is
governed by this License along with a term that is a further
restriction, you may remove that term. If a license document contains
a further restriction but permits relicensing or conveying under this
License, you may add to a covered work material governed by the terms
of that license document, provided that the further restriction does
not survive such relicensing or conveying.
If you add terms to a covered work in accord with this section, you
must place, in the relevant source files, a statement of the
additional terms that apply to those files, or a notice indicating
where to find the applicable terms.
Additional terms, permissive or non-permissive, may be stated in the
form of a separately written license, or stated as exceptions;
the above requirements apply either way.
8. Termination.
You may not propagate or modify a covered work except as expressly
provided under this License. Any attempt otherwise to propagate or
modify it is void, and will automatically terminate your rights under
this License (including any patent licenses granted under the third
paragraph of section 11).
However, if you cease all violation of this License, then your
license from a particular copyright holder is reinstated (a)
provisionally, unless and until the copyright holder explicitly and
finally terminates your license, and (b) permanently, if the copyright
holder fails to notify you of the violation by some reasonable means
prior to 60 days after the cessation.
Moreover, your license from a particular copyright holder is
reinstated permanently if the copyright holder notifies you of the
violation by some reasonable means, this is the first time you have
received notice of violation of this License (for any work) from that
copyright holder, and you cure the violation prior to 30 days after
your receipt of the notice.
Termination of your rights under this section does not terminate the
licenses of parties who have received copies or rights from you under
this License. If your rights have been terminated and not permanently
reinstated, you do not qualify to receive new licenses for the same
material under section 10.
9. Acceptance Not Required for Having Copies.
You are not required to accept this License in order to receive or
run a copy of the Program. Ancillary propagation of a covered work
occurring solely as a consequence of using peer-to-peer transmission
to receive a copy likewise does not require acceptance. However,
nothing other than this License grants you permission to propagate or
modify any covered work. These actions infringe copyright if you do
not accept this License. Therefore, by modifying or propagating a
covered work, you indicate your acceptance of this License to do so.
10. Automatic Licensing of Downstream Recipients.
Each time you convey a covered work, the recipient automatically
receives a license from the original licensors, to run, modify and
propagate that work, subject to this License. You are not responsible
for enforcing compliance by third parties with this License.
An "entity transaction" is a transaction transferring control of an
organization, or substantially all assets of one, or subdividing an
organization, or merging organizations. If propagation of a covered
work results from an entity transaction, each party to that
transaction who receives a copy of the work also receives whatever
licenses to the work the party's predecessor in interest had or could
give under the previous paragraph, plus a right to possession of the
Corresponding Source of the work from the predecessor in interest, if
the predecessor has it or can get it with reasonable efforts.
You may not impose any further restrictions on the exercise of the
rights granted or affirmed under this License. For example, you may
not impose a license fee, royalty, or other charge for exercise of
rights granted under this License, and you may not initiate litigation
(including a cross-claim or counterclaim in a lawsuit) alleging that
any patent claim is infringed by making, using, selling, offering for
sale, or importing the Program or any portion of it.
11. Patents.
A "contributor" is a copyright holder who authorizes use under this
License of the Program or a work on which the Program is based. The
work thus licensed is called the contributor's "contributor version".
A contributor's "essential patent claims" are all patent claims
owned or controlled by the contributor, whether already acquired or
hereafter acquired, that would be infringed by some manner, permitted
by this License, of making, using, or selling its contributor version,
but do not include claims that would be infringed only as a
consequence of further modification of the contributor version. For
purposes of this definition, "control" includes the right to grant
patent sublicenses in a manner consistent with the requirements of
this License.
Each contributor grants you a non-exclusive, worldwide, royalty-free
patent license under the contributor's essential patent claims, to
make, use, sell, offer for sale, import and otherwise run, modify and
propagate the contents of its contributor version.
In the following three paragraphs, a "patent license" is any express
agreement or commitment, however denominated, not to enforce a patent
(such as an express permission to practice a patent or covenant not to
sue for patent infringement). To "grant" such a patent license to a
party means to make such an agreement or commitment not to enforce a
patent against the party.
If you convey a covered work, knowingly relying on a patent license,
and the Corresponding Source of the work is not available for anyone
to copy, free of charge and under the terms of this License, through a
publicly available network server or other readily accessible means,
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You should also get your employer (if you work as a programmer) or your
school, if any, to sign a "copyright disclaimer" for the library, if
necessary. Here is a sample; alter the names:
Yoyodyne, Inc., hereby disclaims all copyright interest in the
library `Frob' (a library for tweaking knobs) written by James Random Hacker.
<signature of Ty Coon>, 1 April 1990
Ty Coon, President of Vice
That's all there is to it!
-165
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GNU LESSER 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
of this license document, but changing it is not allowed.
This version of the GNU Lesser General Public License incorporates
the terms and conditions of version 3 of the GNU General Public
License, supplemented by the additional permissions listed below.
0. Additional Definitions.
As used herein, "this License" refers to version 3 of the GNU Lesser
General Public License, and the "GNU GPL" refers to version 3 of the GNU
General Public License.
"The Library" refers to a covered work governed by this License,
other than an Application or a Combined Work as defined below.
An "Application" is any work that makes use of an interface provided
by the Library, but which is not otherwise based on the Library.
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A "Combined Work" is a work produced by combining or linking an
Application with the Library. The particular version of the Library
with which the Combined Work was made is also called the "Linked
Version".
The "Minimal Corresponding Source" for a Combined Work means the
Corresponding Source for the Combined Work, excluding any source code
for portions of the Combined Work that, considered in isolation, are
based on the Application, and not on the Linked Version.
The "Corresponding Application Code" for a Combined Work means the
object code and/or source code for the Application, including any data
and utility programs needed for reproducing the Combined Work from the
Application, but excluding the System Libraries of the Combined Work.
1. Exception to Section 3 of the GNU GPL.
You may convey a covered work under sections 3 and 4 of this License
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4. Combined Works.
You may convey a Combined Work under terms of your choice that,
taken together, effectively do not restrict modification of the
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d) Do one of the following:
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the Linked Version to produce a modified Combined Work, in the
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5. Combined Libraries.
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This file contains the names of some of the people who have contributed to
FFmpeg. The names are sorted alphabetically by last name. As this file is
currently quite outdated and git serves as a much better tool for determining
authorship, it remains here for historical reasons only.
Dénes Balatoni
Michel Bardiaux
Fabrice Bellard
Patrice Bensoussan
Alex Beregszaszi
BERO
Thilo Borgmann
Mario Brito
Ronald Bultje
Alex Converse
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
Stefano Sabatini
Roman Shaposhnik
Oded Shimon
Dieter Shirley
Konstantin Shishkov
Juan J. Sierralta
Ewald Snel
Sascha Sommer
Leon van Stuivenberg
Roberto Togni
Lionel Ulmer
Reynaldo Verdejo
-881
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Entries are sorted chronologically from oldest to youngest within each release,
releases are sorted from youngest to oldest.
version next:
version 0.10:
- Fixes: CVE-2011-3929, CVE-2011-3934, CVE-2011-3935, CVE-2011-3936,
CVE-2011-3937, CVE-2011-3940, CVE-2011-3941, CVE-2011-3944,
CVE-2011-3945, CVE-2011-3946, CVE-2011-3947, CVE-2011-3949,
CVE-2011-3950, CVE-2011-3951, CVE-2011-3952
- v410 Quicktime Uncompressed 4:4:4 10-bit encoder and decoder
- SBaGen (SBG) binaural beats script demuxer
- OpenMG Audio muxer
- Timecode extraction in DV and MOV
- thumbnail video filter
- XML output in ffprobe
- asplit audio filter
- tinterlace video filter
- astreamsync audio filter
- amerge audio filter
- ISMV (Smooth Streaming) muxer
- GSM audio parser
- SMJPEG muxer
- XWD encoder and decoder
- Automatic thread count based on detection number of (available) CPU cores
- y41p Brooktree Uncompressed 4:1:1 12-bit encoder and decoder
- ffprobe -show_error option
- Avid 1:1 10-bit RGB Packer codec
- v308 Quicktime Uncompressed 4:4:4 encoder and decoder
- yuv4 libquicktime packed 4:2:0 encoder and decoder
- ffprobe -show_frames option
- silencedetect audio filter
- ffprobe -show_program_version, -show_library_versions, -show_versions options
- rv34: frame-level multi-threading
- optimized iMDCT transform on x86 using SSE for for mpegaudiodec
- Improved PGS subtitle decoder
- dumpgraph option to lavfi device
- r210 and r10k encoders
- ffwavesynth decoder
- aviocat tool
- ffeval tool
version 0.9:
- openal input device added
- boxblur filter added
- BWF muxer
- Flash Screen Video 2 decoder
- lavfi input device added
- added avconv, which is almost the same for now, except
for a few incompatible changes in the options, which will hopefully make them
easier to use. The changes are:
* The options placement is now strictly enforced! While in theory the
options for ffmpeg should be given in [input options] -i INPUT [output
options] OUTPUT order, in practice it was possible to give output options
before the -i and it mostly worked. Except when it didn't - the behavior was
a bit inconsistent. In avconv, it is not possible to mix input and output
options. All non-global options are reset after an input or output filename.
* All per-file options are now truly per-file - they apply only to the next
input or output file and specifying different values for different files
will now work properly (notably -ss and -t options).
* All per-stream options are now truly per-stream - it is possible to
specify which stream(s) should a given option apply to. See the Stream
specifiers section in the avconv manual for details.
* In ffmpeg some options (like -newvideo/-newaudio/...) are irregular in the
sense that they're specified after the output filename instead of before,
like all other options. In avconv this irregularity is removed, all options
apply to the next input or output file.
* -newvideo/-newaudio/-newsubtitle options were removed. Not only were they
irregular and highly confusing, they were also redundant. In avconv the -map
option will create new streams in the output file and map input streams to
them. E.g. avconv -i INPUT -map 0 OUTPUT will create an output stream for
each stream in the first input file.
* The -map option now has slightly different and more powerful syntax:
+ Colons (':') are used to separate file index/stream type/stream index
instead of dots. Comma (',') is used to separate the sync stream instead
of colon.. This is done for consistency with other options.
+ It's possible to specify stream type. E.g. -map 0:a:2 creates an
output stream from the third input audio stream.
+ Omitting the stream index now maps all the streams of the given type,
not just the first. E.g. -map 0:s creates output streams for all the
subtitle streams in the first input file.
+ Since -map can now match multiple streams, negative mappings were
introduced. Negative mappings disable some streams from an already
defined map. E.g. '-map 0 -map -0:a:1' means 'create output streams for
all the stream in the first input file, except for the second audio
stream'.
* There is a new option -c (or -codec) for choosing the decoder/encoder to
use, which allows to precisely specify target stream(s) consistently with
other options. E.g. -c:v lib264 sets the codec for all video streams, -c:a:0
libvorbis sets the codec for the first audio stream and -c copy copies all
the streams without reencoding. Old -vcodec/-acodec/-scodec options are now
aliases to -c:v/a/s
* It is now possible to precisely specify which stream should an AVOption
apply to. E.g. -b:v:0 2M sets the bitrate for the first video stream, while
-b:a 128k sets the bitrate for all audio streams. Note that the old -ab 128k
syntax is deprecated and will stop working soon.
* -map_chapters now takes only an input file index and applies to the next
output file. This is consistent with how all the other options work.
* -map_metadata now takes only an input metadata specifier and applies to
the next output file. Output metadata specifier is now part of the option
name, similarly to the AVOptions/map/codec feature above.
* -metadata can now be used to set metadata on streams and chapters, e.g.
-metadata:s:1 language=eng sets the language of the first stream to 'eng'.
This made -vlang/-alang/-slang options redundant, so they were removed.
* -qscale option now uses stream specifiers and applies to all streams, not
just video. I.e. plain -qscale number would now apply to all streams. To get
the old behavior, use -qscale:v. Also there is now a shortcut -q for -qscale
and -aq is now an alias for -q:a.
* -vbsf/-absf/-sbsf options were removed and replaced by a -bsf option which
uses stream specifiers. Use -bsf:v/a/s instead of the old options.
* -itsscale option now uses stream specifiers, so its argument is only the
scale parameter.
* -intra option was removed, use -g 0 for the same effect.
* -psnr option was removed, use -flags +psnr for the same effect.
* -vf option is now an alias to the new -filter option, which uses stream specifiers.
* -vframes/-aframes/-dframes options are now aliases to the new -frames option.
* -vtag/-atag/-stag options are now aliases to the new -tag option.
- XMV demuxer
- LOAS demuxer
- ashowinfo filter added
- Windows Media Image decoder
- amovie source added
- LATM muxer/demuxer
- Speex encoder via libspeex
- JSON output in ffprobe
- WTV muxer
- Optional C++ Support (needed for libstagefright)
- H.264 Decoding on Android via Stagefright
- Prores decoder
- BIN/XBIN/ADF/IDF text file decoder
- aconvert audio filter added
- audio support to lavfi input device added
- libcdio-paranoia input device for audio CD grabbing
- Apple ProRes decoder
- CELT in Ogg demuxing
- G.723.1 demuxer and decoder
- libmodplug support (--enable-libmodplug)
- VC-1 interlaced decoding
- libutvideo wrapper (--enable-libutvideo)
- aevalsrc audio source added
- Ut Video decoder
- Speex encoding via libspeex
- 4:2:2 H.264 decoding support
- 4:2:2 and 4:4:4 H.264 encoding with libx264
- Pulseaudio input device
- Prores encoder
- Video Decoder Acceleration (VDA) HWAccel module.
- replacement Indeo 3 decoder
- new ffmpeg option: -map_channel
- volume audio filter added
- earwax audio filter added
- libv4l2 support (--enable-libv4l2)
- TLS/SSL and HTTPS protocol support
- AVOptions API rewritten and documented
- most of CODEC_FLAG2_*, some CODEC_FLAG_* and many codec-specific fields in
AVCodecContext deprecated. Codec private options should be used instead.
- Properly working defaults in libx264 wrapper, support for native presets.
- Encrypted OMA files support
- Discworld II BMV decoding support
- VBLE Decoder
- OS X Video Decoder Acceleration (VDA) support
- compact and csv output in ffprobe
- pan audio filter
- IFF Amiga Continuous Bitmap (ACBM) decoder
- ass filter
- CRI ADX audio format muxer and demuxer
- Playstation Portable PMP format demuxer
- Microsoft Windows ICO demuxer
- life source
- PCM format support in OMA demuxer
- CLJR encoder
- new option: -report
- Dxtory capture format decoder
- cellauto source
- Simple segmenting muxer
- Indeo 4 decoder
- SMJPEG demuxer
version 0.8:
- many many things we forgot because we rather write code than changelogs
- WebM support in Matroska de/muxer
- low overhead Ogg muxing
- MMS-TCP support
- VP8 de/encoding via libvpx
- Demuxer for On2's IVF format
- Pictor/PC Paint decoder
- HE-AAC v2 decoder
- HE-AAC v2 encoding with libaacplus
- libfaad2 wrapper removed
- DTS-ES extension (XCh) decoding support
- native VP8 decoder
- RTSP tunneling over HTTP
- RTP depacketization of SVQ3
- -strict inofficial replaced by -strict unofficial
- ffplay -exitonkeydown and -exitonmousedown options added
- native GSM / GSM MS decoder
- RTP depacketization of QDM2
- ANSI/ASCII art playback system
- Lego Mindstorms RSO de/muxer
- libavcore added (and subsequently removed)
- SubRip subtitle file muxer and demuxer
- Chinese AVS encoding via libxavs
- ffprobe -show_packets option added
- RTP packetization of Theora and Vorbis
- RTP depacketization of MP4A-LATM
- RTP packetization and depacketization of VP8
- hflip filter
- Apple HTTP Live Streaming demuxer
- a64 codec
- MMS-HTTP support
- G.722 ADPCM audio encoder/decoder
- R10k video decoder
- ocv_smooth filter
- frei0r wrapper filter
- change crop filter syntax to width:height:x:y
- make the crop filter accept parametric expressions
- make ffprobe accept AVFormatContext options
- yadif filter
- blackframe filter
- Demuxer for Leitch/Harris' VR native stream format (LXF)
- RTP depacketization of the X-QT QuickTime format
- SAP (Session Announcement Protocol, RFC 2974) muxer and demuxer
- cropdetect filter
- ffmpeg -crop* options removed
- transpose filter added
- ffmpeg -force_key_frames option added
- demuxer for receiving raw rtp:// URLs without an SDP description
- single stream LATM/LOAS decoder
- setpts filter added
- Win64 support for optimized x86 assembly functions
- MJPEG/AVI1 to JPEG/JFIF bitstream filter
- ASS subtitle encoder and decoder
- IEC 61937 encapsulation for E-AC-3, TrueHD, DTS-HD (for HDMI passthrough)
- overlay filter added
- rename aspect filter to setdar, and pixelaspect to setsar
- IEC 61937 demuxer
- Mobotix .mxg demuxer
- frei0r source added
- hqdn3d filter added
- RTP depacketization of QCELP
- FLAC parser added
- gradfun filter added
- AMR-WB decoder
- replace the ocv_smooth filter with a more generic ocv filter
- Windows Televison (WTV) demuxer
- FFmpeg metadata format muxer and demuxer
- SubRip (srt) subtitle encoder and decoder
- floating-point AC-3 encoder added
- Lagarith decoder
- ffmpeg -copytb option added
- IVF muxer added
- Wing Commander IV movies decoder added
- movie source added
- Bink version 'b' audio and video decoder
- Bitmap Brothers JV playback system
- Apple HTTP Live Streaming protocol handler
- sndio support for playback and record
- Linux framebuffer input device added
- Chronomaster DFA decoder
- DPX image encoder
- MicroDVD subtitle file muxer and demuxer
- Playstation Portable PMP format demuxer
- fieldorder video filter added
- AAC encoding via libvo-aacenc
- AMR-WB encoding via libvo-amrwbenc
- xWMA demuxer
- Mobotix MxPEG decoder
- VP8 frame-multithreading
- NEON optimizations for VP8
- Lots of deprecated API cruft removed
- fft and imdct optimizations for AVX (Sandy Bridge) processors
- showinfo filter added
- SMPTE 302M AES3 audio decoder
- Apple Core Audio Format muxer
- 9bit and 10bit per sample support in the H.264 decoder
- 9bit and 10bit FFV1 encoding / decoding
- split filter added
- select filter added
- sdl output device added
- libmpcodecs video filter support (3 times as many filters than before)
- mpeg2 aspect ratio dection fixed
- libxvid aspect pickiness fixed
- Frame multithreaded decoding
- E-AC-3 audio encoder
- ac3enc: add channel coupling support
- floating-point sample format support to the ac3, eac3, dca, aac, and vorbis decoders.
- H264/MPEG frame-level multi-threading
- All av_metadata_* functions renamed to av_dict_* and moved to libavutil
- 4:4:4 H.264 decoding support
- 10-bit H.264 optimizations for x86
- lut, lutrgb, and lutyuv filters added
- buffersink libavfilter sink added
- Bump libswscale for recently reported ABI break
- New J2K encoder (via OpenJPEG)
version 0.7:
- all the changes for 0.8, but keeping API/ABI compatibility with the 0.6 release
version 0.6:
- PB-frame decoding for H.263
- deprecated vhook subsystem removed
- deprecated old scaler removed
- VQF demuxer
- Alpha channel scaler
- PCX encoder
- RTP packetization of H.263
- RTP packetization of AMR
- RTP depacketization of Vorbis
- CorePNG decoding support
- Cook multichannel decoding support
- introduced avlanguage helpers in libavformat
- 8088flex TMV demuxer and decoder
- per-stream language-tags extraction in asfdec
- V210 decoder and encoder
- remaining GPL parts in AC-3 decoder converted to LGPL
- QCP demuxer
- SoX native format muxer and demuxer
- AMR-NB decoding/encoding, AMR-WB decoding via OpenCORE libraries
- DPX image decoder
- Electronic Arts Madcow decoder
- DivX (XSUB) subtitle encoder
- nonfree libamr support for AMR-NB/WB decoding/encoding removed
- experimental AAC encoder
- RTP depacketization of ASF and RTSP from WMS servers
- RTMP support in libavformat
- noX handling for OPT_BOOL X options
- Wave64 demuxer
- IEC-61937 compatible Muxer
- TwinVQ decoder
- Bluray (PGS) subtitle decoder
- LPCM support in MPEG-TS (HDMV RID as found on Blu-ray disks)
- WMA Pro decoder
- Core Audio Format demuxer
- Atrac1 decoder
- MD STUDIO audio demuxer
- RF64 support in WAV demuxer
- MPEG-4 Audio Lossless Coding (ALS) decoder
- -formats option split into -formats, -codecs, -bsfs, and -protocols
- IV8 demuxer
- CDG demuxer and decoder
- R210 decoder
- Auravision Aura 1 and 2 decoders
- Deluxe Paint Animation playback system
- SIPR decoder
- Adobe Filmstrip muxer and demuxer
- RTP depacketization of H.263
- Bink demuxer and audio/video decoders
- enable symbol versioning by default for linkers that support it
- IFF PBM/ILBM bitmap decoder
- concat protocol
- Indeo 5 decoder
- RTP depacketization of AMR
- WMA Voice decoder
- ffprobe tool
- AMR-NB decoder
- RTSP muxer
- HE-AAC v1 decoder
- Kega Game Video (KGV1) decoder
- VorbisComment writing for FLAC, Ogg FLAC and Ogg Speex files
- RTP depacketization of Theora
- HTTP Digest authentication
- RTMP/RTMPT/RTMPS/RTMPE/RTMPTE protocol support via librtmp
- Psygnosis YOP demuxer and video decoder
- spectral extension support in the E-AC-3 decoder
- unsharp video filter
- RTP hinting in the mov/3gp/mp4 muxer
- Dirac in Ogg demuxing
- seek to keyframes in Ogg
- 4:2:2 and 4:4:4 Theora decoding
- 35% faster VP3/Theora decoding
- faster AAC decoding
- faster H.264 decoding
- RealAudio 1.0 (14.4K) encoder
version 0.5:
- DV50 AKA DVCPRO50 encoder, decoder, muxer 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 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 decoder
- build system improvements
- tabs and trailing whitespace removed from the codebase
- CamStudio video decoder
- 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 Motion Blocks Video decoder
- improved Theora/VP3 decoder
- True Audio (TTA) decoder
- AVS demuxer and video decoder
- JPEG-LS encoder
- 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
- Vorbis audio encoder
- 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
- Zip Motion Blocks Video encoder
- 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
- additional SPARC (VIS) optimizations
- Matroska muxer
- slice-based parallel H.264 decoding
- Monkey's Audio demuxer and decoder
- 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
- The "device" muxers and demuxers are now in a new libavdevice library
- 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
- DV100 AKA DVCPRO HD decoder and demuxer
- 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
- RV40 decoder
- QCELP / PureVoice decoder
- RV30 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
- flash ScreenVideo2 encoder
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
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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.
NOTICE
- Non system dependencies (e.g. libx264, libvpx) are disabled by default.
-46
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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
- optional x86 optimizations in the files
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 libfaac and libaacplus can be hooked up in FFmpeg.
You need to pass --enable-nonfree to configure to enable it. Employ this option
with care as FFmpeg then becomes nonfree and unredistributable.
-429
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FFmpeg maintainers
==================
Below is a list of the people maintaining different parts of the
FFmpeg code.
Please try to keep entries where you are the maintainer upto date!
Names in () mean that the maintainer currently has no time to maintain the code.
A CC after the name means that the maintainer prefers to be CC-ed on patches
and related discussions.
Project Leader
==============
Michael Niedermayer
final design decisions
Applications
============
ffmpeg:
ffmpeg.c Michael Niedermayer
ffplay:
ffplay.c Marton Balint
ffprobe:
ffprobe.c Stefano Sabatini
ffserver:
ffserver.c, ffserver.h Baptiste Coudurier
Commandline utility code:
cmdutils.c, cmdutils.h Michael Niedermayer
QuickTime faststart:
tools/qt-faststart.c Baptiste Coudurier
Miscellaneous Areas
===================
documentation Mike Melanson
website Robert Swain, Lou Logan
build system (configure,Makefiles) Diego Biurrun, Mans Rullgard
project server Árpád Gereöffy, Michael Niedermayer, Reimar Döffinger
mailinglists Michael Niedermayer, Baptiste Coudurier, Lou Logan
presets Robert Swain
metadata subsystem Aurelien Jacobs
release management Michael Niedermayer
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
codec names:
codec_names.sh Nicolas George
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
table generation:
tableprint.c, tableprint.h Reimar Doeffinger
Codecs:
4xm.c Michael Niedermayer
8bps.c Roberto Togni
8svx.c Jaikrishnan Menon
aasc.c Kostya Shishkov
ac3* Justin Ruggles
alacenc.c Jaikrishnan Menon
alsdec.c Thilo Borgmann
apedec.c Kostya Shishkov
ass* Aurelien Jacobs
asv* Michael Niedermayer
atrac3* Benjamin Larsson
bgmc.c, bgmc.h Thilo Borgmann
bink.c Kostya Shishkov
binkaudio.c Peter Ross
bmp.c Mans Rullgard, Kostya Shishkov
cavs* Stefan Gehrer
celp_filters.* Vitor Sessak
cinepak.c Roberto Togni
cljr Alex Beregszaszi
cook.c, cookdata.h Benjamin Larsson
crystalhd.c Philip Langdale
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
ffwavesynth.c Nicolas George
flac* Justin Ruggles
flashsv* Benjamin Larsson
flicvideo.c Mike Melanson
g722.c Martin Storsjo
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
indeo5* Kostya Shishkov
interplayvideo.c Mike Melanson
ivi* Kostya Shishkov
jpeg_ls.c Kostya Shishkov
jvdec.c Peter Ross
kmvc.c Kostya Shishkov
lcl*.c Roberto Togni, Reimar Doeffinger
libcelt_dec.c Nicolas George
libgsm.c Michel Bardiaux
libdirac* David Conrad
libopenjpeg.c Jaikrishnan Menon
libopenjpegenc.c Michael Bradshaw
libschroedinger* David Conrad
libspeexdec.c Justin Ruggles
libtheoraenc.c David Conrad
libvorbis.c David Conrad
libxavs.c Stefan Gehrer
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
mlp* Ramiro Polla
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
pgssubdec.c Reimar Doeffinger
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
smacker.c Kostya Shishkov
smc.c Mike Melanson
snow.c Michael Niedermayer, Loren Merritt
sonic.c Alex Beregszaszi
srt* Aurelien Jacobs
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, Jaikrishnan Menon
txd.c Ivo van Poorten
ulti* Kostya Shishkov
v410*.c Derek Buitenhuis
vb.c Kostya Shishkov
vble.c Derek Buitenhuis
vc1* Kostya Shishkov
vcr1.c Michael Niedermayer
vmnc.c Kostya Shishkov
vorbis_enc.c Oded Shimon
vorbis_dec.c Denes Balatoni, David Conrad
vp3* Mike Melanson
vp5 Aurelien Jacobs
vp6 Aurelien Jacobs
vp8 David Conrad, Jason Garrett-Glaser, Ronald Bultje
vqavideo.c Mike Melanson
wavpack.c Kostya Shishkov
wmaprodec.c Sascha Sommer
wmavoice.c Ronald S. Bultje
wmv2.c Michael Niedermayer
wnv1.c Kostya Shishkov
xan.c Mike Melanson
xl.c Kostya Shishkov
xvmc.c Ivan Kalvachev
zmbv* Kostya Shishkov
Hardware acceleration:
crystalhd.c Philip Langdale
dxva2* Laurent Aimar
libstagefright.cpp Mohamed Naufal
vaapi* Gwenole Beauchesne
vda* Sebastien Zwickert
vdpau* Carl Eugen Hoyos
libavdevice
===========
External Interface:
libavdevice/avdevice.h
libdc1394.c Roman Shaposhnik
v4l2.c Luca Abeni
vfwcap.c Ramiro Polla
libavfilter
===========
Video filters:
graphdump.c Nicolas George
af_amerge.c Nicolas George
af_astreamsync.c Nicolas George
af_pan.c Nicolas George
vsrc_mandelbrot.c Michael Niedermayer
vf_yadif.c Michael Niedermayer
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
ass* Aurelien Jacobs
avi* Michael Niedermayer
bink.c Peter Ross
caf* Peter Ross
crc.c Michael Niedermayer
daud.c Reimar Doeffinger
dv.c Roman Shaposhnik
dxa.c Kostya Shishkov
electronicarts.c Peter Ross
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
idroqdec.c Mike Melanson
iff.c Jaikrishnan Menon
ipmovie.c Mike Melanson
img2.c Michael Niedermayer
iss.c Stefan Gehrer
jvdec.c Peter Ross
libmodplug.c Clément Bœsch
libnut.c Oded Shimon
lmlm4.c Ivo van Poorten
lxfdec.c Tomas Härdin
matroska.c Aurelien Jacobs
matroskadec.c Aurelien Jacobs
matroskaenc.c David Conrad
metadata* Aurelien Jacobs
microdvd* Aurelien Jacobs
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* Baptiste Coudurier
msnwc_tcp.c Ramiro Polla
mtv.c Reynaldo H. Verdejo Pinochet
mxf* Baptiste Coudurier
mxfdec.c Tomas Härdin
nsvdec.c Francois Revol
nut.c Michael Niedermayer
nuv.c Reimar Doeffinger
oggdec.c, oggdec.h David Conrad
oggenc.c Baptiste Coudurier
oggparse*.c David Conrad
oma.c Maxim Poliakovski
psxstr.c Mike Melanson
pva.c Ivo van Poorten
r3d.c Baptiste Coudurier
raw.c Michael Niedermayer
rdt.c Ronald S. Bultje
rl2.c Sascha Sommer
rmdec.c, rmenc.c Ronald S. Bultje, Kostya Shishkov
rtmp* Kostya Shishkov
rtp.c, rtpenc.c Martin Storsjo
rtpdec_asf.* Ronald S. Bultje
rtpenc_mpv.*, rtpenc_aac.* Martin Storsjo
rtsp.c Luca Barbato
sbgdec.c Nicolas George
sdp.c Martin Storsjo
segafilm.c Mike Melanson
siff.c Kostya Shishkov
smacker.c Kostya Shishkov
srtdec.c Aurelien Jacobs
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
wtv.c Peter Ross
wv.c Kostya Shishkov
Protocols:
http.c Ronald S. Bultje
mms*.c Ronald S. Bultje
udp.c Luca Abeni
Operating systems / CPU architectures
=====================================
Alpha Mans Rullgard, Falk Hueffner
ARM Mans Rullgard
AVR32 Mans Rullgard
MIPS Mans Rullgard
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
Releases
========
0.9 Michael Niedermayer
GnuPG Fingerprints of maintainers and contributors
==================================================
Anssi Hannula 1A92 FF42 2DD9 8D2E 8AF7 65A9 4278 C520 513D F3CB
Anton Khirnov 6D0C 6625 56F8 65D1 E5F5 814B B50A 1241 C067 07AB
Attila Kinali 11F0 F9A6 A1D2 11F6 C745 D10C 6520 BCDD F2DF E765
Baptiste Coudurier 8D77 134D 20CC 9220 201F C5DB 0AC9 325C 5C1A BAAA
Ben Littler 3EE3 3723 E560 3214 A8CD 4DEB 2CDB FCE7 768C 8D2C
Benoit Fouet B22A 4F4F 43EF 636B BB66 FCDC 0023 AE1E 2985 49C8
Daniel Verkamp 78A6 07ED 782C 653E C628 B8B9 F0EB 8DD8 2F0E 21C7
Diego Biurrun 8227 1E31 B6D9 4994 7427 E220 9CAE D6CC 4757 FCC5
Gwenole Beauchesne 2E63 B3A6 3E44 37E2 017D 2704 53C7 6266 B153 99C4
Jaikrishnan Menon 61A1 F09F 01C9 2D45 78E1 C862 25DC 8831 AF70 D368
Justin Ruggles 3136 ECC0 C10D 6C04 5F43 CA29 FCBE CD2A 3787 1EBF
Loren Merritt ABD9 08F4 C920 3F65 D8BE 35D7 1540 DAA7 060F 56DE
Lou Logan 7D68 DC73 CBEF EABB 671A B6CF 621C 2E28 82F8 DC3A
Luca Barbato 6677 4209 213C 8843 5B67 29E7 E84C 78C2 84E9 0E34
Michael Niedermayer 9FF2 128B 147E F673 0BAD F133 611E C787 040B 0FAB
Nicolas George 24CE 01CE 9ACC 5CEB 74D8 8D9D B063 D997 36E5 4C93
Panagiotis Issaris 6571 13A3 33D9 3726 F728 AA98 F643 B12E ECF3 E029
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
Robert Swain EE7A 56EA 4A81 A7B5 2001 A521 67FA 362D A2FC 3E71
Sascha Sommer 38A0 F88B 868E 9D3A 97D4 D6A0 E823 706F 1E07 0D3C
Stefano Sabatini 9A43 10F8 D32C D33C 48E7 C52C 5DF2 8E4D B2EE 066B
Tomas Härdin D133 29CA 4EEC 9DB4 7076 F697 B04B 7403 3313 41FD
-176
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@@ -1,176 +0,0 @@
MAIN_MAKEFILE=1
include config.mak
vpath %.c $(SRC_PATH)
vpath %.cpp $(SRC_PATH)
vpath %.h $(SRC_PATH)
vpath %.S $(SRC_PATH)
vpath %.asm $(SRC_PATH)
vpath %.v $(SRC_PATH)
vpath %.texi $(SRC_PATH)
vpath %/fate_config.sh.template $(SRC_PATH)
PROGS-$(CONFIG_FFMPEG) += ffmpeg
PROGS-$(CONFIG_FFPLAY) += ffplay
PROGS-$(CONFIG_FFPROBE) += ffprobe
PROGS-$(CONFIG_FFSERVER) += ffserver
PROGS := $(PROGS-yes:%=%$(EXESUF))
INSTPROGS = $(PROGS-yes:%=%$(PROGSSUF)$(EXESUF))
OBJS = $(PROGS-yes:%=%.o) cmdutils.o
TESTTOOLS = audiogen videogen rotozoom tiny_psnr base64
HOSTPROGS := $(TESTTOOLS:%=tests/%)
TOOLS = qt-faststart trasher
TOOLS-$(CONFIG_ZLIB) += cws2fws
BASENAMES = ffmpeg ffplay ffprobe ffserver
ALLPROGS = $(BASENAMES:%=%$(PROGSSUF)$(EXESUF))
ALLPROGS_G = $(BASENAMES:%=%$(PROGSSUF)_g$(EXESUF))
ALLMANPAGES = $(BASENAMES:%=%.1)
FFLIBS-$(CONFIG_AVDEVICE) += avdevice
FFLIBS-$(CONFIG_AVFILTER) += avfilter
FFLIBS-$(CONFIG_AVFORMAT) += avformat
FFLIBS-$(CONFIG_AVCODEC) += avcodec
FFLIBS-$(CONFIG_POSTPROC) += postproc
FFLIBS-$(CONFIG_SWRESAMPLE)+= swresample
FFLIBS-$(CONFIG_SWSCALE) += swscale
FFLIBS := avutil
DATA_FILES := $(wildcard $(SRC_PATH)/presets/*.ffpreset) $(SRC_PATH)/doc/ffprobe.xsd
SKIPHEADERS = cmdutils_common_opts.h
include $(SRC_PATH)/common.mak
FF_EXTRALIBS := $(FFEXTRALIBS)
FF_DEP_LIBS := $(DEP_LIBS)
all: $(PROGS)
$(PROGS): %$(EXESUF): %$(PROGSSUF)_g$(EXESUF)
$(CP) $< $@$(PROGSSUF)
$(STRIP) $@$(PROGSSUF)
$(TOOLS): %$(EXESUF): %.o
$(LD) $(LDFLAGS) -o $@ $< $(ELIBS)
tools/cws2fws$(EXESUF): ELIBS = -lz
config.h: .config
.config: $(wildcard $(FFLIBS:%=$(SRC_PATH)/lib%/all*.c))
@-tput bold 2>/dev/null
@-printf '\nWARNING: $(?F) newer than config.h, rerun configure\n\n'
@-tput sgr0 2>/dev/null
SUBDIR_VARS := OBJS FFLIBS CLEANFILES DIRS TESTPROGS EXAMPLES SKIPHEADERS \
ALTIVEC-OBJS MMX-OBJS NEON-OBJS X86-OBJS YASM-OBJS-FFT YASM-OBJS \
HOSTPROGS BUILT_HEADERS TESTOBJS ARCH_HEADERS ARMV6-OBJS TOOLS
define RESET
$(1) :=
$(1)-yes :=
endef
define DOSUBDIR
$(foreach V,$(SUBDIR_VARS),$(eval $(call RESET,$(V))))
SUBDIR := $(1)/
include $(SRC_PATH)/$(1)/Makefile
-include $(SRC_PATH)/$(1)/$(ARCH)/Makefile
include $(SRC_PATH)/library.mak
endef
$(foreach D,$(FFLIBS),$(eval $(call DOSUBDIR,lib$(D))))
ffplay.o: CFLAGS += $(SDL_CFLAGS)
ffplay_g$(EXESUF): FF_EXTRALIBS += $(SDL_LIBS)
ffserver_g$(EXESUF): LDFLAGS += $(FFSERVERLDFLAGS)
%$(PROGSSUF)_g$(EXESUF): %.o cmdutils.o $(FF_DEP_LIBS)
$(LD) $(LDFLAGS) -o $@ $< cmdutils.o $(FF_EXTRALIBS)
OBJDIRS += tools
-include $(wildcard tools/*.d)
VERSION_SH = $(SRC_PATH)/version.sh
GIT_LOG = $(SRC_PATH)/.git/logs/HEAD
.version: $(wildcard $(GIT_LOG)) $(VERSION_SH) config.mak
.version: M=@
version.h .version:
$(M)$(VERSION_SH) $(SRC_PATH) version.h $(EXTRA_VERSION)
$(Q)touch .version
# force version.sh to run whenever version might have changed
-include .version
ifdef PROGS
install: install-progs install-data
endif
install: install-libs install-headers
install-libs: install-libs-yes
install-progs-yes:
install-progs-$(CONFIG_SHARED): install-libs
install-progs: install-progs-yes $(PROGS)
$(Q)mkdir -p "$(BINDIR)"
$(INSTALL) -c -m 755 $(INSTPROGS) "$(BINDIR)"
install-data: $(DATA_FILES)
$(Q)mkdir -p "$(DATADIR)"
$(INSTALL) -m 644 $(DATA_FILES) "$(DATADIR)"
uninstall: uninstall-libs uninstall-headers uninstall-progs uninstall-data
uninstall-progs:
$(RM) $(addprefix "$(BINDIR)/", $(ALLPROGS))
uninstall-data:
$(RM) -r "$(DATADIR)"
clean::
$(RM) $(ALLPROGS) $(ALLPROGS_G)
$(RM) $(CLEANSUFFIXES)
$(RM) $(TOOLS)
$(RM) $(CLEANSUFFIXES:%=tools/%)
$(RM) coverage.info
$(RM) -r coverage-html
distclean::
$(RM) $(DISTCLEANSUFFIXES)
$(RM) config.* .version version.h libavutil/avconfig.h
config:
$(SRC_PATH)/configure $(value FFMPEG_CONFIGURATION)
# Without the sed genthml thinks "libavutil" and "./libavutil" are two different things
coverage.info: $(wildcard *.gcda *.gcno */*.gcda */*.gcno */*/*.gcda */*/*.gcno)
$(Q)lcov -c -d . -b . | sed -e 's#/./#/#g' > $@
coverage-html: coverage.info
$(Q)mkdir -p $@
$(Q)genhtml -o $@ $<
$(Q)touch $@
include $(SRC_PATH)/doc/Makefile
include $(SRC_PATH)/tests/Makefile
$(sort $(OBJDIRS)):
$(Q)mkdir -p $@
# Dummy rule to stop make trying to rebuild removed or renamed headers
%.h:
@:
# Disable suffix rules. Most of the builtin rules are suffix rules,
# so this saves some time on slow systems.
.SUFFIXES:
.PHONY: all all-yes alltools *clean config examples install*
.PHONY: testprogs uninstall*
-12
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@@ -1,12 +0,0 @@
FFmpeg README
-------------
1) Documentation
----------------
* Read the documentation in the doc/ directory.
2) Licensing
------------
* See the LICENSE file.
+2
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@@ -0,0 +1,2 @@
# FFmpeg
Unofficial mirror of the FFmpeg repository on GitHub. Primary goal is to provide easily reproducible builds that run on Windows.
-1
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@@ -1 +0,0 @@
0.9.1.git
+66
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@@ -0,0 +1,66 @@
# Ask git for a description of the current commit.
DESCRIPTION=`git describe --long HEAD`
if [ "${DESCRIPTION:0:1}" == "n" ] || [ "${DESCRIPTION:0:1}" == "v" ]; then
DESCRIPTION=${DESCRIPTION:1}
fi
#echo "echo FFmpeg v${DESCRIPTION}"
# Extract version information from description.
VERSION=${DESCRIPTION//./ }
VERSION=${VERSION//-/ }
VERSION=( ${VERSION} )
VERSION_MAJOR=${VERSION[0]}
VERSION_MINOR=${VERSION[1]}
VERSION_PATCH=${VERSION[2]}
VERSION_TWEAK=${VERSION[3]}
COMMIT=${VERSION[4]:1}
#echo "echo FFmpeg v${VERSION_MAJOR}.${VERSION_MINOR}.${VERSION_PATCH}.${VERSION_TWEAK}-${COMMIT}"
. ${SCRIPTROOT}/addons/export.sh VERSION_MAJOR "${VERSION_MAJOR}"
. ${SCRIPTROOT}/addons/export.sh VERSION_MINOR "${VERSION_MINOR}"
. ${SCRIPTROOT}/addons/export.sh VERSION_PATCH "${VERSION_PATCH}"
. ${SCRIPTROOT}/addons/export.sh VERSION_TWEAK "${VERSION_TWEAK}"
. ${SCRIPTROOT}/addons/export.sh COMMIT "${COMMIT}"
# Set up some baseline directories
if [ ! -d distrib ]; then mkdir distrib; fi
if [ ! -d distrib/bin ]; then mkdir distrib/bin; fi
if [ ! -d distrib/lib ]; then mkdir distrib/lib; fi
if [ ! -d distrib/include ]; then mkdir distrib/include; fi
if [ ! -d distrib/share ]; then mkdir distrib/share; fi
# Target Architecture
if [ "${BUILD_BITS}" == "32" ]; then
BUILD_ARCH="i686"
BUILD_TARGET="mingw32"
BUILD_PREFIX="i686-w64-mingw32"
else
BUILD_ARCH="x86_64"
BUILD_TARGET="mingw64"
BUILD_PREFIX="x86_64-w64-mingw32"
fi
. ${SCRIPTROOT}/addons/export.sh BUILD_ARCH "${BUILD_ARCH}"
. ${SCRIPTROOT}/addons/export.sh BUILD_TARGET "${BUILD_TARGET}"
. ${SCRIPTROOT}/addons/export.sh BUILD_PREFIX "${BUILD_PREFIX}"
# License (GPL vs LGPL, v2 vs v3)
declare -a BUILD_FLAGS
if [ "${LICENSE}" == "GPL" ]; then
#echo "::set-output name=flags_license::--enable-gpl"
BUILD_FLAGS+=("--enable-gpl")
fi
if [ "${LICENSE_VERSION}" == "3" ]; then
#echo "::set-output name=flags_license_version::--enable-version3"
BUILD_FLAGS+=("--enable-version3")
fi
# Build Type
if [ "${BUILD_TYPE}" == "static" ]; then
#echo "::set-output name=flags_type::--enable-static --disable-shared"
BUILD_FLAGS+=("--enable-static")
BUILD_FLAGS+=("--disable-shared")
else
#echo "::set-output name=flags_type::--disable-static --enable-shared"
BUILD_FLAGS+=("--disable-static")
BUILD_FLAGS+=("--enable-shared")
fi
. ${SCRIPTROOT}/addons/export.sh BUILD_FLAGS "`echo ${BUILD_FLAGS[@]}`"
+5
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@@ -0,0 +1,5 @@
export $1="$2"
if [[ "${CI}" == "true" ]]; then
echo "::set-output name=$1::$2"
fi
echo "export $1=\"$2\"" >> ${SCRIPTROOT}/.env
+18
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@@ -0,0 +1,18 @@
echo "Applying custom patches..."
declare -a PATCHES
PATCHES[${#PATCHES[@]}]="${VERSION_MAJOR}.${VERSION_MINOR}.${VERSION_TWEAK}"
PATCHES[${#PATCHES[@]}]="${VERSION_MAJOR}.${VERSION_MINOR}"
PATCHES[${#PATCHES[@]}]="${VERSION_MAJOR}"
for patchset in ${PATCHES[@]}; do
if [[ -d "${SCRIPTROOT}/patches/${patchset}" ]]; then
echo " Found ${patchset}..."
for file in ${SCRIPTROOT}/patches/${patchset}/*.patch; do
# Skip files that don't actually exist.
[ -e "${file}" ] || continue
echo " Applying '${file}'..."
git apply "${file}"
done
fi
done
+9
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@@ -0,0 +1,9 @@
for file in ${SCRIPTROOT}/scripts/*.sh; do
[ -e "${file}" ] || continue
echo "================================================================================"
echo ">> ${file}"
echo "================================================================================"
. ${SCRIPTROOT}/.env
${file}
if [[ $? -ne 0 ]]; then exit 1; fi
done
-1192
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File diff suppressed because it is too large Load Diff
-380
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@@ -1,380 +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 <stdint.h>
#include "libavcodec/avcodec.h"
#include "libavfilter/avfilter.h"
#include "libavformat/avformat.h"
#include "libswscale/swscale.h"
#ifdef __MINGW32__
#undef main /* We don't want SDL to override our main() */
#endif
/**
* 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;
/**
* this year, defined by the program for show_banner()
*/
extern const int this_year;
extern AVCodecContext *avcodec_opts[AVMEDIA_TYPE_NB];
extern AVFormatContext *avformat_opts;
extern struct SwsContext *sws_opts;
extern AVDictionary *format_opts, *codec_opts;
/**
* Initialize the cmdutils option system, in particular
* allocate the *_opts contexts.
*/
void init_opts(void);
/**
* Uninitialize the cmdutils option system, in particular
* free the *_opts contexts and their contents.
*/
void uninit_opts(void);
/**
* Trivial log callback.
* Only suitable for opt_help and similar since it lacks prefix handling.
*/
void log_callback_help(void* ptr, int level, const char* fmt, va_list vl);
/**
* Fallback for options that are not explicitly handled, these will be
* parsed through AVOptions.
*/
int opt_default(const char *opt, const char *arg);
/**
* Set the libav* libraries log level.
*/
int opt_loglevel(const char *opt, const char *arg);
int opt_report(const char *opt);
int opt_max_alloc(const char *opt, const char *arg);
int opt_codec_debug(const char *opt, const char *arg);
/**
* Limit the execution time.
*/
int opt_timelimit(const char *opt, const char *arg);
/**
* Parse a string and return its corresponding value as a double.
* Exit 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 command line 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);
/**
* Parse a string specifying a time and return its corresponding
* value as a number of microseconds. Exit from the application if
* the string cannot be correctly parsed.
*
* @param context the context of the value to be set (e.g. the
* corresponding command line option name)
* @param timestr the string to be parsed
* @param is_duration a flag which tells how to interpret timestr, if
* not zero 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 SpecifierOpt {
char *specifier; /**< stream/chapter/program/... specifier */
union {
uint8_t *str;
int i;
int64_t i64;
float f;
double dbl;
} u;
} SpecifierOpt;
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_INT64 0x0400
#define OPT_EXIT 0x0800
#define OPT_DATA 0x1000
#define OPT_FUNC2 0x2000
#define OPT_OFFSET 0x4000 /* option is specified as an offset in a passed optctx */
#define OPT_SPEC 0x8000 /* option is to be stored in an array of SpecifierOpt.
Implies OPT_OFFSET. Next element after the offset is
an int containing element count in the array. */
#define OPT_TIME 0x10000
#define OPT_DOUBLE 0x20000
union {
void *dst_ptr;
int (*func_arg)(const char *, const char *);
int (*func2_arg)(void *, const char *, const char *);
size_t off;
} u;
const char *help;
const char *argname;
} OptionDef;
void show_help_options(const OptionDef *options, const char *msg, int mask,
int value);
/**
* Show help for all options with given flags in class and all its
* children.
*/
void show_help_children(const AVClass *class, int flags);
/**
* Parse the command line arguments.
*
* @param optctx an opaque options context
* @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(void *optctx, int argc, char **argv, const OptionDef *options,
void (* parse_arg_function)(void *optctx, const char*));
/**
* Parse one given option.
*
* @return on success 1 if arg was consumed, 0 otherwise; negative number on error
*/
int parse_option(void *optctx, const char *opt, const char *arg,
const OptionDef *options);
/**
* Find the '-loglevel' option in the command line args and apply it.
*/
void parse_loglevel(int argc, char **argv, const OptionDef *options);
/**
* Check if the given stream matches a stream specifier.
*
* @param s Corresponding format context.
* @param st Stream from s to be checked.
* @param spec A stream specifier of the [v|a|s|d]:[\<stream index\>] form.
*
* @return 1 if the stream matches, 0 if it doesn't, <0 on error
*/
int check_stream_specifier(AVFormatContext *s, AVStream *st, const char *spec);
/**
* Filter out options for given codec.
*
* Create a new options dictionary containing only the options from
* opts which apply to the codec with ID codec_id.
*
* @param s Corresponding format context.
* @param st A stream from s for which the options should be filtered.
* @return a pointer to the created dictionary
*/
AVDictionary *filter_codec_opts(AVDictionary *opts, AVCodec *codec,
AVFormatContext *s, AVStream *st);
/**
* Setup AVCodecContext options for avformat_find_stream_info().
*
* Create an array of dictionaries, one dictionary for each stream
* contained in s.
* Each dictionary will contain the options from codec_opts which can
* be applied to the corresponding stream codec context.
*
* @return pointer to the created array of dictionaries, NULL if it
* cannot be created
*/
AVDictionary **setup_find_stream_info_opts(AVFormatContext *s,
AVDictionary *codec_opts);
/**
* Print an error message to stderr, indicating filename and a human
* readable description of the error code err.
*
* If strerror_r() is not available the use of this function in a
* multithreaded application may be unsafe.
*
* @see av_strerror()
*/
void print_error(const char *filename, int err);
/**
* Print 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(int argc, char **argv, const OptionDef *options);
/**
* Print the version of the program to stdout. The version message
* depends on the current versions of the repository and of the libav*
* libraries.
* This option processing function does not utilize the arguments.
*/
int opt_version(const char *opt, const char *arg);
/**
* Print the license of the program to stdout. The license depends on
* the license of the libraries compiled into the program.
* This option processing function does not utilize the arguments.
*/
int opt_license(const char *opt, const char *arg);
/**
* Print a listing containing all the formats supported by the
* program.
* This option processing function does not utilize the arguments.
*/
int opt_formats(const char *opt, const char *arg);
/**
* Print a listing containing all the codecs supported by the
* program.
* This option processing function does not utilize the arguments.
*/
int opt_codecs(const char *opt, const char *arg);
/**
* Print a listing containing all the filters supported by the
* program.
* This option processing function does not utilize the arguments.
*/
int opt_filters(const char *opt, const char *arg);
/**
* Print a listing containing all the bit stream filters supported by the
* program.
* This option processing function does not utilize the arguments.
*/
int opt_bsfs(const char *opt, const char *arg);
/**
* Print a listing containing all the protocols supported by the
* program.
* This option processing function does not utilize the arguments.
*/
int opt_protocols(const char *opt, const char *arg);
/**
* Print a listing containing all the pixel formats supported by the
* program.
* This option processing function does not utilize the arguments.
*/
int opt_pix_fmts(const char *opt, const char *arg);
/**
* Print a listing containing all the sample formats supported by the
* program.
*/
int show_sample_fmts(const char *opt, const char *arg);
/**
* Return a positive value if a line read from standard input
* starts with [yY], otherwise return 0.
*/
int read_yesno(void);
/**
* Read the file with name filename, and put its content in a newly
* allocated 0-terminated buffer.
*
* @param bufptr location where pointer to buffer is returned
* @param size location where size of buffer is returned
* @return 0 in case of success, a negative value corresponding to an
* AVERROR error code in case of failure.
*/
int cmdutils_read_file(const char *filename, char **bufptr, size_t *size);
/**
* Get a file corresponding to a preset file.
*
* If is_path is non-zero, look for the file in the path preset_name.
* Otherwise search for a file named arg.ffpreset in the directories
* $FFMPEG_DATADIR (if set), $HOME/.ffmpeg, and in the datadir defined
* at configuration time or in a "ffpresets" folder along the executable
* on win32, in that order. If no such file is found and
* codec_name is defined, then search for a file named
* codec_name-preset_name.ffpreset in the above-mentioned directories.
*
* @param filename buffer where the name of the found filename is written
* @param filename_size size in bytes of the filename buffer
* @param preset_name name of the preset to search
* @param is_path tell if preset_name is a filename path
* @param codec_name name of the codec for which to look for the
* preset, may be NULL
*/
FILE *get_preset_file(char *filename, size_t filename_size,
const char *preset_name, int is_path, const char *codec_name);
/**
* Do all the necessary cleanup and abort.
* This function is implemented in the avtools, not cmdutils.
*/
void exit_program(int ret);
/**
* Realloc array to hold new_size elements of elem_size.
* Calls exit_program() on failure.
*
* @param elem_size size in bytes of each element
* @param size new element count will be written here
* @return reallocated array
*/
void *grow_array(void *array, int elem_size, int *size, int new_size);
#endif /* CMDUTILS_H */
-18
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@@ -1,18 +0,0 @@
{ "L", OPT_EXIT, {(void*)opt_license}, "show license" },
{ "h", OPT_EXIT, {(void*)opt_help}, "show help" },
{ "?", OPT_EXIT, {(void*)opt_help}, "show help" },
{ "help", OPT_EXIT, {(void*)opt_help}, "show help" },
{ "-help", OPT_EXIT, {(void*)opt_help}, "show help" },
{ "version", OPT_EXIT, {(void*)opt_version}, "show version" },
{ "formats" , OPT_EXIT, {(void*)opt_formats }, "show available formats" },
{ "codecs" , OPT_EXIT, {(void*)opt_codecs }, "show available codecs" },
{ "bsfs" , OPT_EXIT, {(void*)opt_bsfs }, "show available bit stream filters" },
{ "protocols", OPT_EXIT, {(void*)opt_protocols}, "show available protocols" },
{ "filters", OPT_EXIT, {(void*)opt_filters }, "show available filters" },
{ "pix_fmts" , OPT_EXIT, {(void*)opt_pix_fmts }, "show available pixel formats" },
{ "sample_fmts", OPT_EXIT, {.func_arg = show_sample_fmts }, "show available audio sample formats" },
{ "loglevel", HAS_ARG, {(void*)opt_loglevel}, "set libav* logging level", "loglevel" },
{ "v", HAS_ARG, {(void*)opt_loglevel}, "set libav* logging level", "loglevel" },
{ "debug", HAS_ARG, {(void*)opt_codec_debug}, "set debug flags", "flags" },
{ "report", 0, {(void*)opt_report}, "generate a report" },
{ "max_alloc", HAS_ARG, {(void*)opt_max_alloc}, "set maximum size of a single allocated block", "bytes" },
-120
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#
# common bits used by all libraries
#
# first so "all" becomes default target
all: all-yes
ifndef SUBDIR
ifndef V
Q = @
ECHO = printf "$(1)\t%s\n" $(2)
BRIEF = CC CXX AS YASM AR LD HOSTCC STRIP CP
SILENT = DEPCC YASMDEP RM RANLIB
MSG = $@
M = @$(call ECHO,$(TAG),$@);
$(foreach VAR,$(BRIEF), \
$(eval override $(VAR) = @$$(call ECHO,$(VAR),$$(MSG)); $($(VAR))))
$(foreach VAR,$(SILENT),$(eval override $(VAR) = @$($(VAR))))
$(eval INSTALL = @$(call ECHO,INSTALL,$$(^:$(SRC_DIR)/%=%)); $(INSTALL))
endif
ALLFFLIBS = avcodec avdevice avfilter avformat avutil postproc swscale swresample
# NASM requires -I path terminated with /
IFLAGS := -I. -I$(SRC_PATH)/
CPPFLAGS := $(IFLAGS) $(CPPFLAGS)
CFLAGS += $(ECFLAGS)
CCFLAGS = $(CFLAGS)
CXXFLAGS := $(CFLAGS) $(CXXFLAGS)
YASMFLAGS += $(IFLAGS) -I$(SRC_PATH)/libavutil/x86/ -Pconfig.asm
HOSTCFLAGS += $(IFLAGS)
LDFLAGS := $(ALLFFLIBS:%=-Llib%) $(LDFLAGS)
define COMPILE
$($(1)DEP)
$($(1)) $(CPPFLAGS) $($(1)FLAGS) $($(1)_DEPFLAGS) -c $($(1)_O) $<
endef
COMPILE_C = $(call COMPILE,CC)
COMPILE_CXX = $(call COMPILE,CXX)
COMPILE_S = $(call COMPILE,AS)
%.o: %.c
$(COMPILE_C)
%.o: %.cpp
$(COMPILE_CXX)
%.s: %.c
$(CC) $(CPPFLAGS) $(CFLAGS) -S -o $@ $<
%.o: %.S
$(COMPILE_S)
%.ho: %.h
$(CC) $(CPPFLAGS) $(CFLAGS) -Wno-unused -c -o $@ -x c $<
%.ver: %.v
$(Q)sed 's/$$MAJOR/$($(basename $(@F))_VERSION_MAJOR)/' $^ > $@
%.c %.h: TAG = GEN
# Dummy rule to stop make trying to rebuild removed or renamed headers
%.h:
@:
# Disable suffix rules. Most of the builtin rules are suffix rules,
# so this saves some time on slow systems.
.SUFFIXES:
# Do not delete intermediate files from chains of implicit rules
$(OBJS):
endif
OBJS-$(HAVE_MMX) += $(MMX-OBJS-yes)
OBJS += $(OBJS-yes)
FFLIBS := $(FFLIBS-yes) $(FFLIBS)
TESTPROGS += $(TESTPROGS-yes)
FFEXTRALIBS := $(FFLIBS:%=-l%$(BUILDSUF)) $(EXTRALIBS)
EXAMPLES := $(EXAMPLES:%=$(SUBDIR)%-example$(EXESUF))
OBJS := $(sort $(OBJS:%=$(SUBDIR)%))
TESTOBJS := $(TESTOBJS:%=$(SUBDIR)%) $(TESTPROGS:%=$(SUBDIR)%-test.o)
TESTPROGS := $(TESTPROGS:%=$(SUBDIR)%-test$(EXESUF))
HOSTOBJS := $(HOSTPROGS:%=$(SUBDIR)%.o)
HOSTPROGS := $(HOSTPROGS:%=$(SUBDIR)%$(HOSTEXESUF))
TOOLS += $(TOOLS-yes)
TOOLOBJS := $(TOOLS:%=tools/%.o)
TOOLS := $(TOOLS:%=tools/%$(EXESUF))
DEP_LIBS := $(foreach NAME,$(FFLIBS),lib$(NAME)/$($(CONFIG_SHARED:yes=S)LIBNAME))
ALLHEADERS := $(subst $(SRC_DIR)/,$(SUBDIR),$(wildcard $(SRC_DIR)/*.h $(SRC_DIR)/$(ARCH)/*.h))
SKIPHEADERS += $(ARCH_HEADERS:%=$(ARCH)/%) $(SKIPHEADERS-)
SKIPHEADERS := $(SKIPHEADERS:%=$(SUBDIR)%)
checkheaders: $(filter-out $(SKIPHEADERS:.h=.ho),$(ALLHEADERS:.h=.ho))
alltools: $(TOOLS)
$(HOSTOBJS): %.o: %.c
$(HOSTCC) $(HOSTCFLAGS) -c -o $@ $<
$(HOSTPROGS): %$(HOSTEXESUF): %.o
$(HOSTCC) $(HOSTLDFLAGS) -o $@ $< $(HOSTLIBS)
$(OBJS): | $(sort $(dir $(OBJS)))
$(HOSTOBJS): | $(sort $(dir $(HOSTOBJS)))
$(TESTOBJS): | $(sort $(dir $(TESTOBJS)))
$(TOOLOBJS): | tools
OBJDIRS := $(OBJDIRS) $(dir $(OBJS) $(HOSTOBJS) $(TESTOBJS))
CLEANSUFFIXES = *.d *.o *~ *.ho *.map *.ver *.gcno *.gcda
DISTCLEANSUFFIXES = *.pc
LIBSUFFIXES = *.a *.lib *.so *.so.* *.dylib *.dll *.def *.dll.a *.exp
-include $(wildcard $(OBJS:.o=.d) $(TESTOBJS:.o=.d))
Vendored
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-64
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MANPAGES = $(PROGS-yes:%=doc/%.1)
PODPAGES = $(PROGS-yes:%=doc/%.pod)
HTMLPAGES = $(PROGS-yes:%=doc/%.html) \
doc/developer.html \
doc/faq.html \
doc/fate.html \
doc/general.html \
doc/git-howto.html \
doc/libavfilter.html \
doc/platform.html \
TXTPAGES = doc/fate.txt \
DOCS = $(HTMLPAGES) $(MANPAGES) $(PODPAGES)
ifdef HAVE_MAKEINFO
DOCS += $(TXTPAGES)
endif
all-$(CONFIG_DOC): documentation
documentation: $(DOCS)
TEXIDEP = awk '/^@(verbatim)?include/ { printf "$@: $(@D)/%s\n", $$2 }' <$< >$(@:%=%.d)
doc/%.txt: TAG = TXT
doc/%.txt: doc/%.texi
$(Q)$(TEXIDEP)
$(M)makeinfo --force --no-headers -o $@ $< 2>/dev/null
doc/%.html: TAG = HTML
doc/%.html: doc/%.texi $(SRC_PATH)/doc/t2h.init
$(Q)$(TEXIDEP)
$(M)texi2html -monolithic --init-file $(SRC_PATH)/doc/t2h.init --output $@ $<
doc/%.pod: TAG = POD
doc/%.pod: doc/%.texi
$(Q)$(TEXIDEP)
$(M)$(SRC_PATH)/doc/texi2pod.pl $< $@
doc/%.1: TAG = MAN
doc/%.1: doc/%.pod
$(M)pod2man --section=1 --center=" " --release=" " $< > $@
$(DOCS): | doc
OBJDIRS += doc
install-progs-$(CONFIG_DOC): install-man
install-man: $(MANPAGES)
$(Q)mkdir -p "$(MANDIR)/man1"
$(INSTALL) -m 644 $(MANPAGES) "$(MANDIR)/man1"
uninstall: uninstall-man
uninstall-man:
$(RM) $(addprefix "$(MANDIR)/man1/",$(ALLMANPAGES))
clean::
$(RM) $(TXTPAGES) doc/*.html doc/*.pod doc/*.1 $(CLEANSUFFIXES:%=doc/%)
-include $(wildcard $(DOCS:%=%.d))
.PHONY: documentation
-49
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@@ -1,49 +0,0 @@
Release Notes
=============
* 0.10 "Freedom" January, 2012
General notes
-------------
This release is binary compatible with 0.8 and 0.9.
See the Changelog file for a list of significant changes. Note, there
are many more new features and bugfixes than whats listed there.
Bugreports against FFmpeg git master or the most recent FFmpeg release are
accepted. If you are experiencing issues with any formally released version of
FFmpeg, please try git master to check if the issue still exists. If it does,
make your report against the development code following the usual bug reporting
guidelines.
API changes
-----------
A number of additional APIs have been introduced and some existing
functions have been deprecated and are scheduled for removal in the next
release. Significant API changes include:
* new audio decoding API which decodes from an AVPacket to an AVFrame and
is able to use AVCodecContext.get_buffer() in the similar way as video decoding.
* new audio encoding API which encodes from an AVFrame to an AVPacket, thus
allowing it to properly output timing information and side data.
Please see the git history and the file doc/APIchanges for details.
Other notable changes
---------------------
Libavcodec and libavformat built as shared libraries now hide non-public
symbols. This will break applications using those symbols. Possible solutions
are, in order of preference:
1) Try finding a way of accomplishing the same with public API.
2) If there is no corresponding public API, but you think there should be,
post a request on the developer mailing list or IRC channel.
3) Finally if your program needs access to FFmpeg / libavcodec / libavformat
internals for some special reason then the best solution is to link statically.
Please see the Changelog file and git history for a more detailed list of changes.
-168
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All the numerical options, if not specified otherwise, accept in input
a string representing a number, which may contain one of the
International System number postfixes, for example 'K', 'M', 'G'.
If 'i' is appended after the postfix, powers of 2 are used instead of
powers of 10. The 'B' postfix multiplies the value for 8, and can be
appended after another postfix or used alone. This allows using for
example 'KB', 'MiB', 'G' and 'B' as postfix.
Options which do not take arguments are boolean options, and set the
corresponding value to true. They can be set to false by prefixing
with "no" the option name, for example using "-nofoo" in the
command line will set to false the boolean option with name "foo".
@anchor{Stream specifiers}
@section Stream specifiers
Some options are applied per-stream, e.g. bitrate or codec. Stream specifiers
are used to precisely specify which stream(s) does a given option belong to.
A stream specifier is a string generally appended to the option name and
separated from it by a colon. E.g. @code{-codec:a:1 ac3} option contains
@code{a:1} stream specifer, which matches the second audio stream. Therefore it
would select the ac3 codec for the second audio stream.
A stream specifier can match several stream, the option is then applied to all
of them. E.g. the stream specifier in @code{-b:a 128k} matches all audio
streams.
An empty stream specifier matches all streams, for example @code{-codec copy}
or @code{-codec: copy} would copy all the streams without reencoding.
Possible forms of stream specifiers are:
@table @option
@item @var{stream_index}
Matches the stream with this index. E.g. @code{-threads:1 4} would set the
thread count for the second stream to 4.
@item @var{stream_type}[:@var{stream_index}]
@var{stream_type} is one of: 'v' for video, 'a' for audio, 's' for subtitle,
'd' for data and 't' for attachments. If @var{stream_index} is given, then
matches stream number @var{stream_index} of this type. Otherwise matches all
streams of this type.
@item p:@var{program_id}[:@var{stream_index}]
If @var{stream_index} is given, then matches stream number @var{stream_index} in
program with id @var{program_id}. Otherwise matches all streams in this program.
@end table
@section Generic options
These options are shared amongst the av* tools.
@table @option
@item -L
Show license.
@item -h, -?, -help, --help
Show help.
@item -version
Show version.
@item -formats
Show available formats.
The fields preceding the format names have the following meanings:
@table @samp
@item D
Decoding available
@item E
Encoding available
@end table
@item -codecs
Show available codecs.
The fields preceding the 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 -bsfs
Show available bitstream filters.
@item -protocols
Show available protocols.
@item -filters
Show available libavfilter filters.
@item -pix_fmts
Show available pixel formats.
@item -sample_fmts
Show available sample formats.
@item -loglevel @var{loglevel} | -v @var{loglevel}
Set the logging level used by the library.
@var{loglevel} is a number or a string containing one of the following values:
@table @samp
@item quiet
@item panic
@item fatal
@item error
@item warning
@item info
@item verbose
@item debug
@end table
By default the program logs to stderr, if coloring is supported by the
terminal, colors are used to mark errors and warnings. Log coloring
can be disabled setting the environment variable
@env{AV_LOG_FORCE_NOCOLOR} or @env{NO_COLOR}, or can be forced setting
the environment variable @env{AV_LOG_FORCE_COLOR}.
The use of the environment variable @env{NO_COLOR} is deprecated and
will be dropped in a following FFmpeg version.
@item -report
Dump full command line and console output to a file named
@code{@var{program}-@var{YYYYMMDD}-@var{HHMMSS}.log} in the current
directory.
This file can be useful for bug reports.
It also implies @code{-loglevel verbose}.
Note: setting the environment variable @code{FFREPORT} to any value has the
same effect.
@end table
@section AVOptions
These options are provided directly by the libavformat, libavdevice and
libavcodec libraries. To see the list of available AVOptions, use the
@option{-help} option. They are separated into two categories:
@table @option
@item generic
These options can be set for any container, codec or device. Generic options
are listed under AVFormatContext options for containers/devices and under
AVCodecContext options for codecs.
@item private
These options are specific to the given container, device or codec. Private
options are listed under their corresponding containers/devices/codecs.
@end table
For example to write an ID3v2.3 header instead of a default ID3v2.4 to
an MP3 file, use the @option{id3v2_version} private option of the MP3
muxer:
@example
ffmpeg -i input.flac -id3v2_version 3 out.mp3
@end example
All codec AVOptions are obviously per-stream, so the chapter on stream
specifiers applies to them
Note @option{-nooption} syntax cannot be used for boolean AVOptions,
use @option{-option 0}/@option{-option 1}.
Note2 old undocumented way of specifying per-stream AVOptions by prepending
v/a/s to the options name is now obsolete and will be removed soon.
-36
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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
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)
-91
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@chapter Bitstream Filters
@c man begin BITSTREAM FILTERS
When you configure your FFmpeg build, all the supported bitstream
filters are enabled by default. You can list all available ones using
the configure option @code{--list-bsfs}.
You can disable all the bitstream filters using the configure option
@code{--disable-bsfs}, and selectively enable any bitstream filter using
the option @code{--enable-bsf=BSF}, or you can disable a particular
bitstream filter using the option @code{--disable-bsf=BSF}.
The option @code{-bsfs} of the ff* tools will display the list of
all the supported bitstream filters included in your build.
Below is a description of the currently available bitstream filters.
@section aac_adtstoasc
@section chomp
@section dump_extradata
@section h264_mp4toannexb
Convert an H.264 bitstream from length prefixed mode to start code
prefixed mode (as defined in the Annex B of the ITU-T H.264
specification).
This is required by some streaming formats, typically the MPEG-2
transport stream format ("mpegts").
For example to remux an MP4 file containing an H.264 stream to mpegts
format with @command{ffmpeg}, you can use the command:
@example
ffmpeg -i INPUT.mp4 -codec copy -bsf:v h264_mp4toannexb OUTPUT.ts
@end example
@section imx_dump_header
@section mjpeg2jpeg
Convert MJPEG/AVI1 packets to full JPEG/JFIF packets.
MJPEG is a video codec wherein each video frame is essentially a
JPEG image. The individual frames can be extracted without loss,
e.g. by
@example
ffmpeg -i ../some_mjpeg.avi -c:v copy frames_%d.jpg
@end example
Unfortunately, these chunks are incomplete JPEG images, because
they lack the DHT segment required for decoding. Quoting from
@url{http://www.digitalpreservation.gov/formats/fdd/fdd000063.shtml}:
Avery Lee, writing in the rec.video.desktop newsgroup in 2001,
commented that "MJPEG, or at least the MJPEG in AVIs having the
MJPG fourcc, is restricted JPEG with a fixed -- and *omitted* --
Huffman table. The JPEG must be YCbCr colorspace, it must be 4:2:2,
and it must use basic Huffman encoding, not arithmetic or
progressive. . . . You can indeed extract the MJPEG frames and
decode them with a regular JPEG decoder, but you have to prepend
the DHT segment to them, or else the decoder won't have any idea
how to decompress the data. The exact table necessary is given in
the OpenDML spec."
This bitstream filter patches the header of frames extracted from an MJPEG
stream (carrying the AVI1 header ID and lacking a DHT segment) to
produce fully qualified JPEG images.
@example
ffmpeg -i mjpeg-movie.avi -c:v copy -vbsf mjpeg2jpeg frame_%d.jpg
exiftran -i -9 frame*.jpg
ffmpeg -i frame_%d.jpg -c:v copy rotated.avi
@end example
@section mjpega_dump_header
@section movsub
@section mp3_header_compress
@section mp3_header_decompress
@section noise
@section remove_extradata
@c man end BITSTREAM FILTERS
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FFmpeg currently uses a custom build system, this text attempts to document
some of its obscure features and options.
Makefile variables:
V
Disable the default terse mode, the full command issued by make and its
output will be shown on the screen.
DESTDIR
Destination directory for the install targets, useful to prepare packages
or install FFmpeg in cross-environments.
Makefile targets:
all
Default target, builds all the libraries and the executables.
fate
Run the fate test suite, note you must have installed it
fate-list
Will list all fate/regression test targets
install
Install headers, libraries and programs.
libavformat/output-example
Build the libavformat basic example.
libavcodec/api-example
Build the libavcodec basic example.
libswscale/swscale-test
Build the swscale self-test (useful also as example).
Useful standard make commands:
make -t <target>
Touch all files that otherwise would be build, this is useful to reduce
unneeded rebuilding when changing headers, but note you must force rebuilds
of files that actually need it by hand then.
make -j<num>
rebuild with multiple jobs at the same time. Faster on multi processor systems
make -k
continue build in case of errors, this is useful for the regression tests
sometimes but note it will still not run all reg tests.
-63
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@chapter Decoders
@c man begin DECODERS
Decoders are configured elements in FFmpeg which allow the decoding of
multimedia streams.
When you configure your FFmpeg build, all the supported native decoders
are enabled by default. Decoders requiring an external library must be enabled
manually via the corresponding @code{--enable-lib} option. You can list all
available decoders using the configure option @code{--list-decoders}.
You can disable all the decoders with the configure option
@code{--disable-decoders} and selectively enable / disable single decoders
with the options @code{--enable-decoder=@var{DECODER}} /
@code{--disable-decoder=@var{DECODER}}.
The option @code{-codecs} of the ff* tools will display the list of
enabled decoders.
@c man end DECODERS
@chapter Video Decoders
@c man begin VIDEO DECODERS
A description of some of the currently available video decoders
follows.
@section rawvideo
Raw video decoder.
This decoder decodes rawvideo streams.
@subsection Options
@table @option
@item top @var{top_field_first}
Specify the assumed field type of the input video.
@table @option
@item -1
the video is assumed to be progressive (default)
@item 0
bottom-field-first is assumed
@item 1
top-field-first is assumed
@end table
@end table
@c man end VIDEO DECODERS
@chapter Audio Decoders
@c man begin AUDIO DECODERS
@section ffwavesynth
Internal wave synthetizer.
This decoder generates wave patterns according to predefined sequences. Its
use is purely internal and the format of the data it accepts is not publicly
documented.
@c man end AUDIO DECODERS
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@chapter Demuxers
@c man begin DEMUXERS
Demuxers are configured elements in FFmpeg which allow to read the
multimedia streams from a particular type of file.
When you configure your FFmpeg build, all the supported demuxers
are enabled by default. You can list all available ones using the
configure option "--list-demuxers".
You can disable all the demuxers using the configure option
"--disable-demuxers", and selectively enable a single demuxer with
the option "--enable-demuxer=@var{DEMUXER}", or disable it
with the option "--disable-demuxer=@var{DEMUXER}".
The option "-formats" of the ff* tools will display the list of
enabled demuxers.
The description of some of the currently available demuxers follows.
@section image2
Image file demuxer.
This demuxer reads from a list of image files specified by a pattern.
The pattern may contain the string "%d" or "%0@var{N}d", which
specifies the position of the characters representing a sequential
number in each filename matched by the pattern. If the form
"%d0@var{N}d" is used, the string representing the number in each
filename is 0-padded and @var{N} is the total number of 0-padded
digits representing the number. The literal character '%' can be
specified in the pattern with the string "%%".
If the pattern contains "%d" or "%0@var{N}d", the first filename of
the file list specified by the pattern must contain a number
inclusively contained between 0 and 4, all the following numbers must
be sequential. This limitation may be hopefully fixed.
The pattern may contain a suffix which is used to automatically
determine the format of the images contained in the files.
For example the pattern "img-%03d.bmp" will match a sequence of
filenames of the form @file{img-001.bmp}, @file{img-002.bmp}, ...,
@file{img-010.bmp}, etc.; the pattern "i%%m%%g-%d.jpg" will match a
sequence of filenames of the form @file{i%m%g-1.jpg},
@file{i%m%g-2.jpg}, ..., @file{i%m%g-10.jpg}, etc.
The size, the pixel format, and the format of each image must be the
same for all the files in the sequence.
The following example shows how to use @command{ffmpeg} for creating a
video from the images in the file sequence @file{img-001.jpeg},
@file{img-002.jpeg}, ..., assuming an input frame rate of 10 frames per
second:
@example
ffmpeg -i 'img-%03d.jpeg' -r 10 out.mkv
@end example
Note that the pattern must not necessarily contain "%d" or
"%0@var{N}d", for example to convert a single image file
@file{img.jpeg} you can employ the command:
@example
ffmpeg -i img.jpeg img.png
@end example
@section applehttp
Apple HTTP Live Streaming demuxer.
This demuxer presents all AVStreams from all variant streams.
The id field is set to the bitrate variant index number. By setting
the discard flags on AVStreams (by pressing 'a' or 'v' in ffplay),
the caller can decide which variant streams to actually receive.
The total bitrate of the variant that the stream belongs to is
available in a metadata key named "variant_bitrate".
@section sbg
SBaGen script demuxer.
This demuxer reads the script language used by SBaGen
@url{http://uazu.net/sbagen/} to generate binaural beats sessions. A SBG
script looks like that:
@example
-SE
a: 300-2.5/3 440+4.5/0
b: 300-2.5/0 440+4.5/3
off: -
NOW == a
+0:07:00 == b
+0:14:00 == a
+0:21:00 == b
+0:30:00 off
@end example
A SBG script can mix absolute and relative timestamps. If the script uses
either only absolute timestamps (including the script start time) or only
relative ones, then its layout is fixed, and the conversion is
straightforward. On the other hand, if the script mixes both kind of
timestamps, then the @var{NOW} reference for relative timestamps will be
taken from the current time of day at the time the script is read, and the
script layout will be frozen according to that reference. That means that if
the script is directly played, the actual times will match the absolute
timestamps up to the sound controller's clock accuracy, but if the user
somehow pauses the playback or seeks, all times will be shifted accordingly.
@c man end INPUT DEVICES
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\input texinfo @c -*- texinfo -*-
@settitle Developer Documentation
@titlepage
@center @titlefont{Developer Documentation}
@end titlepage
@top
@contents
@chapter Developers Guide
@section API
@itemize @bullet
@item libavcodec is the library containing the codecs (both encoding and
decoding). Look at @file{libavcodec/apiexample.c} to see how to use it.
@item libavformat is the library containing the file format handling (mux and
demux code for several formats). Look at @file{ffplay.c} to use it in a
player. See @file{libavformat/output-example.c} to use it to generate
audio or video streams.
@end itemize
@section Integrating libavcodec or libavformat in your program
You can integrate all the source code of the libraries to link them
statically to avoid any version problem. All you need is to provide a
'config.mak' and a 'config.h' in the parent directory. See the defines
generated by ./configure to understand what is needed.
You can use libavcodec or libavformat in your commercial program, but
@emph{any patch you make must be published}. The best way to proceed is
to send your patches to the FFmpeg mailing list.
@section Contributing
There are 3 ways by which code gets into ffmpeg.
@itemize @bullet
@item Submitting Patches to the main developer mailing list
see @ref{Submitting patches} for details.
@item Directly committing changes to the main tree.
@item Committing changes to a git clone, for example on github.com or
gitorious.org. And asking us to merge these changes.
@end itemize
Whichever way, changes should be reviewed by the maintainer of the code
before they are committed. And they should follow the @ref{Coding Rules}.
The developer making the commit and the author are responsible for their changes
and should try to fix issues their commit causes.
@anchor{Coding Rules}
@section Coding Rules
@subsection Code formatting conventions
There are the following guidelines regarding the indentation in files:
@itemize @bullet
@item
Indent size is 4.
@item
The TAB character is forbidden outside of Makefiles as is any
form of trailing whitespace. Commits containing either will be
rejected by the git repository.
@item
You should try to limit your code lines to 80 characters; however, do so if
and only if this improves readability.
@end itemize
The presentation is one inspired by 'indent -i4 -kr -nut'.
The main priority in FFmpeg is simplicity and small code size in order to
minimize the bug count.
@subsection Comments
Use the JavaDoc/Doxygen format (see examples below) so that code documentation
can be generated automatically. All nontrivial functions should have a comment
above them explaining what the function does, even if it is just one sentence.
All structures and their member variables should be documented, too.
Avoid Qt-style and similar Doxygen syntax with @code{!} in it, i.e. replace
@code{//!} with @code{///} and similar. Also @@ syntax should be employed
for markup commands, i.e. use @code{@@param} and not @code{\param}.
@example
/**
* @@file
* MPEG codec.
* @@author ...
*/
/**
* Summary sentence.
* more text ...
* ...
*/
typedef struct Foobar@{
int var1; /**< var1 description */
int var2; ///< var2 description
/** var3 description */
int var3;
@} Foobar;
/**
* Summary sentence.
* more text ...
* ...
* @@param my_parameter description of my_parameter
* @@return return value description
*/
int myfunc(int my_parameter)
...
@end example
@subsection C language features
FFmpeg is programmed in the ISO C90 language with a few additional
features from ISO C99, namely:
@itemize @bullet
@item
the @samp{inline} keyword;
@item
@samp{//} comments;
@item
designated struct initializers (@samp{struct s x = @{ .i = 17 @};})
@item
compound literals (@samp{x = (struct s) @{ 17, 23 @};})
@end itemize
These features are supported by all compilers we care about, so we will not
accept patches to remove their use unless they absolutely do not impair
clarity and performance.
All code must compile with recent versions of GCC and a number of other
currently supported compilers. To ensure compatibility, please do not use
additional C99 features or GCC extensions. Especially watch out for:
@itemize @bullet
@item
mixing statements and declarations;
@item
@samp{long long} (use @samp{int64_t} instead);
@item
@samp{__attribute__} not protected by @samp{#ifdef __GNUC__} or similar;
@item
GCC statement expressions (@samp{(x = (@{ int y = 4; y; @})}).
@end itemize
@subsection Naming conventions
All names are using underscores (_), not CamelCase. For example, @samp{avfilter_get_video_buffer} is
a valid function name and @samp{AVFilterGetVideo} is not. The exception from this are type names, like
for example structs and enums; they should always be in the CamelCase
There are following conventions for naming variables and functions:
@itemize @bullet
@item
For local variables no prefix is required.
@item
For variables and functions declared as @code{static} no prefixes are required.
@item
For variables and functions used internally by the library, @code{ff_} prefix
should be used.
For example, @samp{ff_w64_demuxer}.
@item
For variables and functions used internally across multiple libraries, use
@code{avpriv_}. For example, @samp{avpriv_aac_parse_header}.
@item
For exported names, each library has its own prefixes. Just check the existing
code and name accordingly.
@end itemize
@subsection Miscellanous conventions
@itemize @bullet
@item
fprintf and printf are forbidden in libavformat and libavcodec,
please use av_log() instead.
@item
Casts should be used only when necessary. Unneeded parentheses
should also be avoided if they don't make the code easier to understand.
@end itemize
@subsection Editor configuration
In order to configure Vim to follow FFmpeg formatting conventions, paste
the following snippet into your @file{.vimrc}:
@example
" indentation rules for FFmpeg: 4 spaces, no tabs
set expandtab
set shiftwidth=4
set softtabstop=4
" allow tabs in Makefiles
autocmd FileType make set noexpandtab shiftwidth=8 softtabstop=8
" Trailing whitespace and tabs are forbidden, so highlight them.
highlight ForbiddenWhitespace ctermbg=red guibg=red
match ForbiddenWhitespace /\s\+$\|\t/
" Do not highlight spaces at the end of line while typing on that line.
autocmd InsertEnter * match ForbiddenWhitespace /\t\|\s\+\%#\@@<!$/
@end example
For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
@example
(setq c-default-style "k&r")
(setq-default c-basic-offset 4)
(setq-default indent-tabs-mode nil)
(setq-default show-trailing-whitespace t)
@end example
@section Development Policy
@enumerate
@item
Contributions should be licensed under the LGPL 2.1, including an
"or any later version" clause, or the MIT license. GPL 2 including
an "or any later version" clause is also acceptable, but LGPL is
preferred.
@item
You must not commit code which breaks FFmpeg! (Meaning unfinished but
enabled code which breaks compilation or compiles but does not work or
breaks the regression tests)
You can commit unfinished stuff (for testing etc), but it must be disabled
(#ifdef etc) by default so it does not interfere with other developers'
work.
@item
You do not have to over-test things. If it works for you, and you think it
should work for others, then commit. If your code has problems
(portability, triggers compiler bugs, unusual environment etc) they will be
reported and eventually fixed.
@item
Do not commit unrelated changes together, split them into self-contained
pieces. Also do not forget that if part B depends on part A, but A does not
depend on B, then A can and should be committed first and separate from B.
Keeping changes well split into self-contained parts makes reviewing and
understanding them on the commit log mailing list easier. This also helps
in case of debugging later on.
Also if you have doubts about splitting or not splitting, do not hesitate to
ask/discuss it on the developer mailing list.
@item
Do not change behavior of the programs (renaming options etc) or public
API or ABI without first discussing it on the ffmpeg-devel mailing list.
Do not remove functionality from the code. Just improve!
Note: Redundant code can be removed.
@item
Do not commit changes to the build system (Makefiles, configure script)
which change behavior, defaults etc, without asking first. The same
applies to compiler warning fixes, trivial looking fixes and to code
maintained by other developers. We usually have a reason for doing things
the way we do. Send your changes as patches to the ffmpeg-devel mailing
list, and if the code maintainers say OK, you may commit. This does not
apply to files you wrote and/or maintain.
@item
We refuse source indentation and other cosmetic changes if they are mixed
with functional changes, such commits will be rejected and removed. Every
developer has his own indentation style, you should not change it. Of course
if you (re)write something, you can use your own style, even though we would
prefer if the indentation throughout FFmpeg was consistent (Many projects
force a given indentation style - we do not.). If you really need to make
indentation changes (try to avoid this), separate them strictly from real
changes.
NOTE: If you had to put if()@{ .. @} over a large (> 5 lines) chunk of code,
then either do NOT change the indentation of the inner part within (do not
move it to the right)! or do so in a separate commit
@item
Always fill out the commit log message. Describe in a few lines what you
changed and why. You can refer to mailing list postings if you fix a
particular bug. Comments such as "fixed!" or "Changed it." are unacceptable.
Recommended format:
area changed: Short 1 line description
details describing what and why and giving references.
@item
Make sure the author of the commit is set correctly. (see git commit --author)
If you apply a patch, send an
answer to ffmpeg-devel (or wherever you got the patch from) saying that
you applied the patch.
@item
When applying patches that have been discussed (at length) on the mailing
list, reference the thread in the log message.
@item
Do NOT commit to code actively maintained by others without permission.
Send a patch to ffmpeg-devel instead. If no one answers within a reasonable
timeframe (12h for build failures and security fixes, 3 days small changes,
1 week for big patches) then commit your patch if you think it is OK.
Also note, the maintainer can simply ask for more time to review!
@item
Subscribe to the ffmpeg-cvslog mailing list. The diffs of all commits
are sent there and reviewed by all the other developers. Bugs and possible
improvements or general questions regarding commits are discussed there. We
expect you to react if problems with your code are uncovered.
@item
Update the documentation if you change behavior or add features. If you are
unsure how best to do this, send a patch to ffmpeg-devel, the documentation
maintainer(s) will review and commit your stuff.
@item
Try to keep important discussions and requests (also) on the public
developer mailing list, so that all developers can benefit from them.
@item
Never write to unallocated memory, never write over the end of arrays,
always check values read from some untrusted source before using them
as array index or other risky things.
@item
Remember to check if you need to bump versions for the specific libav*
parts (libavutil, libavcodec, libavformat) you are changing. You need
to change the version integer.
Incrementing the first component means no backward compatibility to
previous versions (e.g. removal of a function from the public API).
Incrementing the second component means backward compatible change
(e.g. addition of a function to the public API or extension of an
existing data structure).
Incrementing the third component means a noteworthy binary compatible
change (e.g. encoder bug fix that matters for the decoder).
@item
Compiler warnings indicate potential bugs or code with bad style. If a type of
warning always points to correct and clean code, that warning should
be disabled, not the code changed.
Thus the remaining warnings can either be bugs or correct code.
If it is a bug, the bug has to be fixed. If it is not, the code should
be changed to not generate a warning unless that causes a slowdown
or obfuscates the code.
@item
If you add a new file, give it a proper license header. Do not copy and
paste it from a random place, use an existing file as template.
@end enumerate
We think our rules are not too hard. If you have comments, contact us.
Note, these rules are mostly borrowed from the MPlayer project.
@anchor{Submitting patches}
@section Submitting patches
First, read the @ref{Coding Rules} above if you did not yet, in particular
the rules regarding patch submission.
When you submit your patch, please use @code{git format-patch} or
@code{git send-email}. We cannot read other diffs :-)
Also please do not submit a patch which contains several unrelated changes.
Split it into separate, self-contained pieces. This does not mean splitting
file by file. Instead, make the patch as small as possible while still
keeping it as a logical unit that contains an individual change, even
if it spans multiple files. This makes reviewing your patches much easier
for us and greatly increases your chances of getting your patch applied.
Use the patcheck tool of FFmpeg to check your patch.
The tool is located in the tools directory.
Run the @ref{Regression tests} before submitting a patch in order to verify
it does not cause unexpected problems.
Patches should be posted as base64 encoded attachments (or any other
encoding which ensures that the patch will not be trashed during
transmission) to the ffmpeg-devel mailing list, see
@url{http://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel}
It also helps quite a bit if you tell us what the patch does (for example
'replaces lrint by lrintf'), and why (for example '*BSD isn't C99 compliant
and has no lrint()')
Also please if you send several patches, send each patch as a separate mail,
do not attach several unrelated patches to the same mail.
Your patch will be reviewed on the mailing list. You will likely be asked
to make some changes and are expected to send in an improved version that
incorporates the requests from the review. This process may go through
several iterations. Once your patch is deemed good enough, some developer
will pick it up and commit it to the official FFmpeg tree.
Give us a few days to react. But if some time passes without reaction,
send a reminder by email. Your patch should eventually be dealt with.
@section New codecs or formats checklist
@enumerate
@item
Did you use av_cold for codec initialization and close functions?
@item
Did you add a long_name under NULL_IF_CONFIG_SMALL to the AVCodec or
AVInputFormat/AVOutputFormat struct?
@item
Did you bump the minor version number (and reset the micro version
number) in @file{libavcodec/version.h} or @file{libavformat/version.h}?
@item
Did you register it in @file{allcodecs.c} or @file{allformats.c}?
@item
Did you add the CodecID to @file{avcodec.h}?
@item
If it has a fourCC, did you add it to @file{libavformat/riff.c},
even if it is only a decoder?
@item
Did you add a rule to compile the appropriate files in the Makefile?
Remember to do this even if you're just adding a format to a file that is
already being compiled by some other rule, like a raw demuxer.
@item
Did you add an entry to the table of supported formats or codecs in
@file{doc/general.texi}?
@item
Did you add an entry in the Changelog?
@item
If it depends on a parser or a library, did you add that dependency in
configure?
@item
Did you @code{git add} the appropriate files before committing?
@item
Did you make sure it compiles standalone, i.e. with
@code{configure --disable-everything --enable-decoder=foo}
(or @code{--enable-demuxer} or whatever your component is)?
@end enumerate
@section patch submission checklist
@enumerate
@item
Does @code{make fate} pass with the patch applied?
@item
Was the patch generated with git format-patch or send-email?
@item
Did you sign off your patch? (git commit -s)
See @url{http://kerneltrap.org/files/Jeremy/DCO.txt} for the meaning
of sign off.
@item
Did you provide a clear git commit log message?
@item
Is the patch against latest FFmpeg git master branch?
@item
Are you subscribed to ffmpeg-devel?
(the list is subscribers only due to spam)
@item
Have you checked that the changes are minimal, so that the same cannot be
achieved with a smaller patch and/or simpler final code?
@item
If the change is to speed critical code, did you benchmark it?
@item
If you did any benchmarks, did you provide them in the mail?
@item
Have you checked that the patch does not introduce buffer overflows or
other security issues?
@item
Did you test your decoder or demuxer against damaged data? If no, see
tools/trasher and the noise bitstream filter. Your decoder or demuxer
should not crash or end in a (near) infinite loop when fed damaged data.
@item
Does the patch not mix functional and cosmetic changes?
@item
Did you add tabs or trailing whitespace to the code? Both are forbidden.
@item
Is the patch attached to the email you send?
@item
Is the mime type of the patch correct? It should be text/x-diff or
text/x-patch or at least text/plain and not application/octet-stream.
@item
If the patch fixes a bug, did you provide a verbose analysis of the bug?
@item
If the patch fixes a bug, did you provide enough information, including
a sample, so the bug can be reproduced and the fix can be verified?
Note please do not attach samples >100k to mails but rather provide a
URL, you can upload to ftp://upload.ffmpeg.org
@item
Did you provide a verbose summary about what the patch does change?
@item
Did you provide a verbose explanation why it changes things like it does?
@item
Did you provide a verbose summary of the user visible advantages and
disadvantages if the patch is applied?
@item
Did you provide an example so we can verify the new feature added by the
patch easily?
@item
If you added a new file, did you insert a license header? It should be
taken from FFmpeg, not randomly copied and pasted from somewhere else.
@item
You should maintain alphabetical order in alphabetically ordered lists as
long as doing so does not break API/ABI compatibility.
@item
Lines with similar content should be aligned vertically when doing so
improves readability.
@item
Consider to add a regression test for your code.
@item
If you added YASM code please check that things still work with --disable-yasm
@end enumerate
@section Patch review process
All patches posted to ffmpeg-devel will be reviewed, unless they contain a
clear note that the patch is not for the git master branch.
Reviews and comments will be posted as replies to the patch on the
mailing list. The patch submitter then has to take care of every comment,
that can be by resubmitting a changed patch or by discussion. Resubmitted
patches will themselves be reviewed like any other patch. If at some point
a patch passes review with no comments then it is approved, that can for
simple and small patches happen immediately while large patches will generally
have to be changed and reviewed many times before they are approved.
After a patch is approved it will be committed to the repository.
We will review all submitted patches, but sometimes we are quite busy so
especially for large patches this can take several weeks.
If you feel that the review process is too slow and you are willing to try to
take over maintainership of the area of code you change then just clone
git master and maintain the area of code there. We will merge each area from
where its best maintained.
When resubmitting patches, please do not make any significant changes
not related to the comments received during review. Such patches will
be rejected. Instead, submit significant changes or new features as
separate patches.
@anchor{Regression tests}
@section Regression tests
Before submitting a patch (or committing to the repository), you should at least
test that you did not break anything.
Running 'make fate' accomplishes this, please see @url{fate.html} for details.
[Of course, some patches may change the results of the regression tests. In
this case, the reference results of the regression tests shall be modified
accordingly].
@bye
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@chapter Encoders
@c man begin ENCODERS
Encoders are configured elements in FFmpeg which allow the encoding of
multimedia streams.
When you configure your FFmpeg build, all the supported native encoders
are enabled by default. Encoders requiring an external library must be enabled
manually via the corresponding @code{--enable-lib} option. You can list all
available encoders using the configure option @code{--list-encoders}.
You can disable all the encoders with the configure option
@code{--disable-encoders} and selectively enable / disable single encoders
with the options @code{--enable-encoder=@var{ENCODER}} /
@code{--disable-encoder=@var{ENCODER}}.
The option @code{-codecs} of the ff* tools will display the list of
enabled encoders.
@c man end ENCODERS
@chapter Audio Encoders
@c man begin AUDIO ENCODERS
A description of some of the currently available audio encoders
follows.
@section ac3 and ac3_fixed
AC-3 audio encoders.
These encoders implement part of ATSC A/52:2010 and ETSI TS 102 366, as well as
the undocumented RealAudio 3 (a.k.a. dnet).
The @var{ac3} encoder uses floating-point math, while the @var{ac3_fixed}
encoder only uses fixed-point integer math. This does not mean that one is
always faster, just that one or the other may be better suited to a
particular system. The floating-point encoder will generally produce better
quality audio for a given bitrate. The @var{ac3_fixed} encoder is not the
default codec for any of the output formats, so it must be specified explicitly
using the option @code{-acodec ac3_fixed} in order to use it.
@subsection AC-3 Metadata
The AC-3 metadata options are used to set parameters that describe the audio,
but in most cases do not affect the audio encoding itself. Some of the options
do directly affect or influence the decoding and playback of the resulting
bitstream, while others are just for informational purposes. A few of the
options will add bits to the output stream that could otherwise be used for
audio data, and will thus affect the quality of the output. Those will be
indicated accordingly with a note in the option list below.
These parameters are described in detail in several publicly-available
documents.
@itemize
@item @uref{http://www.atsc.org/cms/standards/a_52-2010.pdf,A/52:2010 - Digital Audio Compression (AC-3) (E-AC-3) Standard}
@item @uref{http://www.atsc.org/cms/standards/a_54a_with_corr_1.pdf,A/54 - Guide to the Use of the ATSC Digital Television Standard}
@item @uref{http://www.dolby.com/uploadedFiles/zz-_Shared_Assets/English_PDFs/Professional/18_Metadata.Guide.pdf,Dolby Metadata Guide}
@item @uref{http://www.dolby.com/uploadedFiles/zz-_Shared_Assets/English_PDFs/Professional/46_DDEncodingGuidelines.pdf,Dolby Digital Professional Encoding Guidelines}
@end itemize
@subsubsection Metadata Control Options
@table @option
@item -per_frame_metadata @var{boolean}
Allow Per-Frame Metadata. Specifies if the encoder should check for changing
metadata for each frame.
@table @option
@item 0
The metadata values set at initialization will be used for every frame in the
stream. (default)
@item 1
Metadata values can be changed before encoding each frame.
@end table
@end table
@subsubsection Downmix Levels
@table @option
@item -center_mixlev @var{level}
Center Mix Level. The amount of gain the decoder should apply to the center
channel when downmixing to stereo. This field will only be written to the
bitstream if a center channel is present. The value is specified as a scale
factor. There are 3 valid values:
@table @option
@item 0.707
Apply -3dB gain
@item 0.595
Apply -4.5dB gain (default)
@item 0.500
Apply -6dB gain
@end table
@item -surround_mixlev @var{level}
Surround Mix Level. The amount of gain the decoder should apply to the surround
channel(s) when downmixing to stereo. This field will only be written to the
bitstream if one or more surround channels are present. The value is specified
as a scale factor. There are 3 valid values:
@table @option
@item 0.707
Apply -3dB gain
@item 0.500
Apply -6dB gain (default)
@item 0.000
Silence Surround Channel(s)
@end table
@end table
@subsubsection Audio Production Information
Audio Production Information is optional information describing the mixing
environment. Either none or both of the fields are written to the bitstream.
@table @option
@item -mixing_level @var{number}
Mixing Level. Specifies peak sound pressure level (SPL) in the production
environment when the mix was mastered. Valid values are 80 to 111, or -1 for
unknown or not indicated. The default value is -1, but that value cannot be
used if the Audio Production Information is written to the bitstream. Therefore,
if the @code{room_type} option is not the default value, the @code{mixing_level}
option must not be -1.
@item -room_type @var{type}
Room Type. Describes the equalization used during the final mixing session at
the studio or on the dubbing stage. A large room is a dubbing stage with the
industry standard X-curve equalization; a small room has flat equalization.
This field will not be written to the bitstream if both the @code{mixing_level}
option and the @code{room_type} option have the default values.
@table @option
@item 0
@itemx notindicated
Not Indicated (default)
@item 1
@itemx large
Large Room
@item 2
@itemx small
Small Room
@end table
@end table
@subsubsection Other Metadata Options
@table @option
@item -copyright @var{boolean}
Copyright Indicator. Specifies whether a copyright exists for this audio.
@table @option
@item 0
@itemx off
No Copyright Exists (default)
@item 1
@itemx on
Copyright Exists
@end table
@item -dialnorm @var{value}
Dialogue Normalization. Indicates how far the average dialogue level of the
program is below digital 100% full scale (0 dBFS). This parameter determines a
level shift during audio reproduction that sets the average volume of the
dialogue to a preset level. The goal is to match volume level between program
sources. A value of -31dB will result in no volume level change, relative to
the source volume, during audio reproduction. Valid values are whole numbers in
the range -31 to -1, with -31 being the default.
@item -dsur_mode @var{mode}
Dolby Surround Mode. Specifies whether the stereo signal uses Dolby Surround
(Pro Logic). This field will only be written to the bitstream if the audio
stream is stereo. Using this option does @b{NOT} mean the encoder will actually
apply Dolby Surround processing.
@table @option
@item 0
@itemx notindicated
Not Indicated (default)
@item 1
@itemx off
Not Dolby Surround Encoded
@item 2
@itemx on
Dolby Surround Encoded
@end table
@item -original @var{boolean}
Original Bit Stream Indicator. Specifies whether this audio is from the
original source and not a copy.
@table @option
@item 0
@itemx off
Not Original Source
@item 1
@itemx on
Original Source (default)
@end table
@end table
@subsection Extended Bitstream Information
The extended bitstream options are part of the Alternate Bit Stream Syntax as
specified in Annex D of the A/52:2010 standard. It is grouped into 2 parts.
If any one parameter in a group is specified, all values in that group will be
written to the bitstream. Default values are used for those that are written
but have not been specified. If the mixing levels are written, the decoder
will use these values instead of the ones specified in the @code{center_mixlev}
and @code{surround_mixlev} options if it supports the Alternate Bit Stream
Syntax.
@subsubsection Extended Bitstream Information - Part 1
@table @option
@item -dmix_mode @var{mode}
Preferred Stereo Downmix Mode. Allows the user to select either Lt/Rt
(Dolby Surround) or Lo/Ro (normal stereo) as the preferred stereo downmix mode.
@table @option
@item 0
@itemx notindicated
Not Indicated (default)
@item 1
@itemx ltrt
Lt/Rt Downmix Preferred
@item 2
@itemx loro
Lo/Ro Downmix Preferred
@end table
@item -ltrt_cmixlev @var{level}
Lt/Rt Center Mix Level. The amount of gain the decoder should apply to the
center channel when downmixing to stereo in Lt/Rt mode.
@table @option
@item 1.414
Apply +3dB gain
@item 1.189
Apply +1.5dB gain
@item 1.000
Apply 0dB gain
@item 0.841
Apply -1.5dB gain
@item 0.707
Apply -3.0dB gain
@item 0.595
Apply -4.5dB gain (default)
@item 0.500
Apply -6.0dB gain
@item 0.000
Silence Center Channel
@end table
@item -ltrt_surmixlev @var{level}
Lt/Rt Surround Mix Level. The amount of gain the decoder should apply to the
surround channel(s) when downmixing to stereo in Lt/Rt mode.
@table @option
@item 0.841
Apply -1.5dB gain
@item 0.707
Apply -3.0dB gain
@item 0.595
Apply -4.5dB gain
@item 0.500
Apply -6.0dB gain (default)
@item 0.000
Silence Surround Channel(s)
@end table
@item -loro_cmixlev @var{level}
Lo/Ro Center Mix Level. The amount of gain the decoder should apply to the
center channel when downmixing to stereo in Lo/Ro mode.
@table @option
@item 1.414
Apply +3dB gain
@item 1.189
Apply +1.5dB gain
@item 1.000
Apply 0dB gain
@item 0.841
Apply -1.5dB gain
@item 0.707
Apply -3.0dB gain
@item 0.595
Apply -4.5dB gain (default)
@item 0.500
Apply -6.0dB gain
@item 0.000
Silence Center Channel
@end table
@item -loro_surmixlev @var{level}
Lo/Ro Surround Mix Level. The amount of gain the decoder should apply to the
surround channel(s) when downmixing to stereo in Lo/Ro mode.
@table @option
@item 0.841
Apply -1.5dB gain
@item 0.707
Apply -3.0dB gain
@item 0.595
Apply -4.5dB gain
@item 0.500
Apply -6.0dB gain (default)
@item 0.000
Silence Surround Channel(s)
@end table
@end table
@subsubsection Extended Bitstream Information - Part 2
@table @option
@item -dsurex_mode @var{mode}
Dolby Surround EX Mode. Indicates whether the stream uses Dolby Surround EX
(7.1 matrixed to 5.1). Using this option does @b{NOT} mean the encoder will actually
apply Dolby Surround EX processing.
@table @option
@item 0
@itemx notindicated
Not Indicated (default)
@item 1
@itemx on
Dolby Surround EX Off
@item 2
@itemx off
Dolby Surround EX On
@end table
@item -dheadphone_mode @var{mode}
Dolby Headphone Mode. Indicates whether the stream uses Dolby Headphone
encoding (multi-channel matrixed to 2.0 for use with headphones). Using this
option does @b{NOT} mean the encoder will actually apply Dolby Headphone
processing.
@table @option
@item 0
@itemx notindicated
Not Indicated (default)
@item 1
@itemx on
Dolby Headphone Off
@item 2
@itemx off
Dolby Headphone On
@end table
@item -ad_conv_type @var{type}
A/D Converter Type. Indicates whether the audio has passed through HDCD A/D
conversion.
@table @option
@item 0
@itemx standard
Standard A/D Converter (default)
@item 1
@itemx hdcd
HDCD A/D Converter
@end table
@end table
@subsection Other AC-3 Encoding Options
@table @option
@item -stereo_rematrixing @var{boolean}
Stereo Rematrixing. Enables/Disables use of rematrixing for stereo input. This
is an optional AC-3 feature that increases quality by selectively encoding
the left/right channels as mid/side. This option is enabled by default, and it
is highly recommended that it be left as enabled except for testing purposes.
@end table
@subsection Floating-Point-Only AC-3 Encoding Options
These options are only valid for the floating-point encoder and do not exist
for the fixed-point encoder due to the corresponding features not being
implemented in fixed-point.
@table @option
@item -channel_coupling @var{boolean}
Enables/Disables use of channel coupling, which is an optional AC-3 feature
that increases quality by combining high frequency information from multiple
channels into a single channel. The per-channel high frequency information is
sent with less accuracy in both the frequency and time domains. This allows
more bits to be used for lower frequencies while preserving enough information
to reconstruct the high frequencies. This option is enabled by default for the
floating-point encoder and should generally be left as enabled except for
testing purposes or to increase encoding speed.
@table @option
@item -1
@itemx auto
Selected by Encoder (default)
@item 0
@itemx off
Disable Channel Coupling
@item 1
@itemx on
Enable Channel Coupling
@end table
@item -cpl_start_band @var{number}
Coupling Start Band. Sets the channel coupling start band, from 1 to 15. If a
value higher than the bandwidth is used, it will be reduced to 1 less than the
coupling end band. If @var{auto} is used, the start band will be determined by
the encoder based on the bit rate, sample rate, and channel layout. This option
has no effect if channel coupling is disabled.
@table @option
@item -1
@itemx auto
Selected by Encoder (default)
@end table
@end table
@c man end AUDIO ENCODERS
@chapter Video Encoders
@c man begin VIDEO ENCODERS
A description of some of the currently available video encoders
follows.
@section libvpx
VP8 format supported through libvpx.
Requires the presence of the libvpx headers and library during configuration.
You need to explicitly configure the build with @code{--enable-libvpx}.
@subsection Options
Mapping from FFmpeg to libvpx options with conversion notes in parentheses.
@table @option
@item threads
g_threads
@item profile
g_profile
@item vb
rc_target_bitrate
@item g
kf_max_dist
@item keyint_min
kf_min_dist
@item qmin
rc_min_quantizer
@item qmax
rc_max_quantizer
@item bufsize, vb
rc_buf_sz
@code{(bufsize * 1000 / vb)}
rc_buf_optimal_sz
@code{(bufsize * 1000 / vb * 5 / 6)}
@item rc_init_occupancy, vb
rc_buf_initial_sz
@code{(rc_init_occupancy * 1000 / vb)}
@item rc_buffer_aggressivity
rc_undershoot_pct
@item skip_threshold
rc_dropframe_thresh
@item qcomp
rc_2pass_vbr_bias_pct
@item maxrate, vb
rc_2pass_vbr_maxsection_pct
@code{(maxrate * 100 / vb)}
@item minrate, vb
rc_2pass_vbr_minsection_pct
@code{(minrate * 100 / vb)}
@item minrate, maxrate, vb
@code{VPX_CBR}
@code{(minrate == maxrate == vb)}
@item crf
@code{VPX_CQ}, @code{VP8E_SET_CQ_LEVEL}
@item quality
@table @option
@item @var{best}
@code{VPX_DL_BEST_QUALITY}
@item @var{good}
@code{VPX_DL_GOOD_QUALITY}
@item @var{realtime}
@code{VPX_DL_REALTIME}
@end table
@item speed
@code{VP8E_SET_CPUUSED}
@item nr
@code{VP8E_SET_NOISE_SENSITIVITY}
@item mb_threshold
@code{VP8E_SET_STATIC_THRESHOLD}
@item slices
@code{VP8E_SET_TOKEN_PARTITIONS}
@item Alternate reference frame related
@table @option
@item vp8flags altref
@code{VP8E_SET_ENABLEAUTOALTREF}
@item @var{arnr_max_frames}
@code{VP8E_SET_ARNR_MAXFRAMES}
@item @var{arnr_type}
@code{VP8E_SET_ARNR_TYPE}
@item @var{arnr_strength}
@code{VP8E_SET_ARNR_STRENGTH}
@item @var{rc_lookahead}
g_lag_in_frames
@end table
@item vp8flags error_resilient
g_error_resilient
@end table
For more information about libvpx see:
@url{http://www.webmproject.org/}
@section libx264
H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 format supported through
libx264.
Requires the presence of the libx264 headers and library during
configuration. You need to explicitly configure the build with
@code{--enable-libx264}.
@subsection Options
@table @option
@item preset @var{preset_name}
Set the encoding preset.
@item tune @var{tune_name}
Tune the encoding params.
@item fastfirstpass @var{bool}
Use fast settings when encoding first pass, default value is 1.
@item profile @var{profile_name}
Set profile restrictions.
@item level @var{level}
Specify level (as defined by Annex A).
Deprecated in favor of @var{x264opts}.
@item passlogfile @var{filename}
Specify filename for 2 pass stats.
Deprecated in favor of @var{x264opts} (see @var{stats} libx264 option).
@item wpredp @var{wpred_type}
Specify Weighted prediction for P-frames.
Deprecated in favor of @var{x264opts} (see @var{weightp} libx264 option).
@item x264opts @var{options}
Allow to set any x264 option, see x264 --fullhelp for a list.
@var{options} is a list of @var{key}=@var{value} couples separated by
":".
@end table
For example to specify libx264 encoding options with @command{ffmpeg}:
@example
ffmpeg -i foo.mpg -vcodec libx264 -x264opts keyint=123:min-keyint=20 -an out.mkv
@end example
For more information about libx264 and the supported options see:
@url{http://www.videolan.org/developers/x264.html}
@c man end VIDEO ENCODERS
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The following table lists most error codes found in various operating
systems supported by FFmpeg.
OS
Code Std F LBMWwb Text (YMMV)
E2BIG POSIX ++++++ Argument list too long
EACCES POSIX ++++++ Permission denied
EADDRINUSE POSIX +++..+ Address in use
EADDRNOTAVAIL POSIX +++..+ Cannot assign requested address
EADV +..... Advertise error
EAFNOSUPPORT POSIX +++..+ Address family not supported
EAGAIN POSIX + ++++++ Resource temporarily unavailable
EALREADY POSIX +++..+ Operation already in progress
EAUTH .++... Authentication error
EBADARCH ..+... Bad CPU type in executable
EBADE +..... Invalid exchange
EBADEXEC ..+... Bad executable
EBADF POSIX ++++++ Bad file descriptor
EBADFD +..... File descriptor in bad state
EBADMACHO ..+... Malformed Macho file
EBADMSG POSIX ++4... Bad message
EBADR +..... Invalid request descriptor
EBADRPC .++... RPC struct is bad
EBADRQC +..... Invalid request code
EBADSLT +..... Invalid slot
EBFONT +..... Bad font file format
EBUSY POSIX - ++++++ Device or resource busy
ECANCELED POSIX +++... Operation canceled
ECHILD POSIX ++++++ No child processes
ECHRNG +..... Channel number out of range
ECOMM +..... Communication error on send
ECONNABORTED POSIX +++..+ Software caused connection abort
ECONNREFUSED POSIX - +++ss+ Connection refused
ECONNRESET POSIX +++..+ Connection reset
EDEADLK POSIX ++++++ Resource deadlock avoided
EDEADLOCK +..++. File locking deadlock error
EDESTADDRREQ POSIX +++... Destination address required
EDEVERR ..+... Device error
EDOM C89 - ++++++ Numerical argument out of domain
EDOOFUS .F.... Programming error
EDOTDOT +..... RFS specific error
EDQUOT POSIX +++... Disc quota exceeded
EEXIST POSIX ++++++ File exists
EFAULT POSIX - ++++++ Bad address
EFBIG POSIX - ++++++ File too large
EFTYPE .++... Inappropriate file type or format
EHOSTDOWN +++... Host is down
EHOSTUNREACH POSIX +++..+ No route to host
EHWPOISON +..... Memory page has hardware error
EIDRM POSIX +++... Identifier removed
EILSEQ C99 ++++++ Illegal byte sequence
EINPROGRESS POSIX - +++ss+ Operation in progress
EINTR POSIX - ++++++ Interrupted system call
EINVAL POSIX + ++++++ Invalid argument
EIO POSIX + ++++++ I/O error
EISCONN POSIX +++..+ Socket is already connected
EISDIR POSIX ++++++ Is a directory
EISNAM +..... Is a named type file
EKEYEXPIRED +..... Key has expired
EKEYREJECTED +..... Key was rejected by service
EKEYREVOKED +..... Key has been revoked
EL2HLT +..... Level 2 halted
EL2NSYNC +..... Level 2 not synchronized
EL3HLT +..... Level 3 halted
EL3RST +..... Level 3 reset
ELIBACC +..... Can not access a needed shared library
ELIBBAD +..... Accessing a corrupted shared library
ELIBEXEC +..... Cannot exec a shared library directly
ELIBMAX +..... Too many shared libraries
ELIBSCN +..... .lib section in a.out corrupted
ELNRNG +..... Link number out of range
ELOOP POSIX +++..+ Too many levels of symbolic links
EMEDIUMTYPE +..... Wrong medium type
EMFILE POSIX ++++++ Too many open files
EMLINK POSIX ++++++ Too many links
EMSGSIZE POSIX +++..+ Message too long
EMULTIHOP POSIX ++4... Multihop attempted
ENAMETOOLONG POSIX - ++++++ Filen ame too long
ENAVAIL +..... No XENIX semaphores available
ENEEDAUTH .++... Need authenticator
ENETDOWN POSIX +++..+ Network is down
ENETRESET SUSv3 +++..+ Network dropped connection on reset
ENETUNREACH POSIX +++..+ Network unreachable
ENFILE POSIX ++++++ Too many open files in system
ENOANO +..... No anode
ENOATTR .++... Attribute not found
ENOBUFS POSIX - +++..+ No buffer space available
ENOCSI +..... No CSI structure available
ENODATA XSR +N4... No message available
ENODEV POSIX - ++++++ No such device
ENOENT POSIX - ++++++ No such file or directory
ENOEXEC POSIX ++++++ Exec format error
ENOFILE ...++. No such file or directory
ENOKEY +..... Required key not available
ENOLCK POSIX ++++++ No locks available
ENOLINK POSIX ++4... Link has been severed
ENOMEDIUM +..... No medium found
ENOMEM POSIX ++++++ Not enough space
ENOMSG POSIX +++..+ No message of desired type
ENONET +..... Machine is not on the network
ENOPKG +..... Package not installed
ENOPROTOOPT POSIX +++..+ Protocol not available
ENOSPC POSIX ++++++ No space left on device
ENOSR XSR +N4... No STREAM resources
ENOSTR XSR +N4... Not a STREAM
ENOSYS POSIX + ++++++ Function not implemented
ENOTBLK +++... Block device required
ENOTCONN POSIX +++..+ Socket is not connected
ENOTDIR POSIX ++++++ Not a directory
ENOTEMPTY POSIX ++++++ Directory not empty
ENOTNAM +..... Not a XENIX named type file
ENOTRECOVERABLE SUSv4 - +..... State not recoverable
ENOTSOCK POSIX +++..+ Socket operation on non-socket
ENOTSUP POSIX +++... Operation not supported
ENOTTY POSIX ++++++ Inappropriate I/O control operation
ENOTUNIQ +..... Name not unique on network
ENXIO POSIX ++++++ No such device or address
EOPNOTSUPP POSIX +++..+ Operation not supported (on socket)
EOVERFLOW POSIX +++..+ Value too large to be stored in data type
EOWNERDEAD SUSv4 +..... Owner died
EPERM POSIX - ++++++ Operation not permitted
EPFNOSUPPORT +++..+ Protocol family not supported
EPIPE POSIX - ++++++ Broken pipe
EPROCLIM .++... Too many processes
EPROCUNAVAIL .++... Bad procedure for program
EPROGMISMATCH .++... Program version wrong
EPROGUNAVAIL .++... RPC prog. not avail
EPROTO POSIX ++4... Protocol error
EPROTONOSUPPORT POSIX - +++ss+ Protocol not supported
EPROTOTYPE POSIX +++..+ Protocol wrong type for socket
EPWROFF ..+... Device power is off
ERANGE C89 - ++++++ Result too large
EREMCHG +..... Remote address changed
EREMOTE +++... Object is remote
EREMOTEIO +..... Remote I/O error
ERESTART +..... Interrupted system call should be restarted
ERFKILL +..... Operation not possible due to RF-kill
EROFS POSIX ++++++ Read-only file system
ERPCMISMATCH .++... RPC version wrong
ESHLIBVERS ..+... Shared library version mismatch
ESHUTDOWN +++..+ Cannot send after socket shutdown
ESOCKTNOSUPPORT +++... Socket type not supported
ESPIPE POSIX ++++++ Illegal seek
ESRCH POSIX ++++++ No such process
ESRMNT +..... Srmount error
ESTALE POSIX +++..+ Stale NFS file handle
ESTRPIPE +..... Streams pipe error
ETIME XSR +N4... Stream ioctl timeout
ETIMEDOUT POSIX - +++ss+ Connection timed out
ETOOMANYREFS +++... Too many references: cannot splice
ETXTBSY POSIX +++... Text file busy
EUCLEAN +..... Structure needs cleaning
EUNATCH +..... Protocol driver not attached
EUSERS +++... Too many users
EWOULDBLOCK POSIX +++..+ Operation would block
EXDEV POSIX ++++++ Cross-device link
EXFULL +..... Exchange full
Notations:
F: used in FFmpeg (-: a few times, +: a lot)
SUSv3: Single Unix Specification, version 3
SUSv4: Single Unix Specification, version 4
XSR: XSI STREAMS (obsolete)
OS: availability on some supported operating systems
L: GNU/Linux
B: BSD (F: FreeBSD, N: NetBSD)
M: MacOS X
W: Microsoft Windows (s: emulated with winsock, see libavformat/network.h)
w: Mingw32 (3.17) and Mingw64 (2.0.1)
b: BeOS
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@chapter Expression Evaluation
@c man begin EXPRESSION EVALUATION
When evaluating an arithmetic expression, FFmpeg uses an internal
formula evaluator, implemented through the @file{libavutil/eval.h}
interface.
An expression may contain unary, binary operators, constants, and
functions.
Two expressions @var{expr1} and @var{expr2} can be combined to form
another expression "@var{expr1};@var{expr2}".
@var{expr1} and @var{expr2} are evaluated in turn, and the new
expression evaluates to the value of @var{expr2}.
The following binary operators are available: @code{+}, @code{-},
@code{*}, @code{/}, @code{^}.
The following unary operators are available: @code{+}, @code{-}.
The following functions are available:
@table @option
@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 isnan(x)
Return 1.0 if @var{x} is NAN, 0.0 otherwise.
@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 st(var, expr)
Allow to store the value of the expression @var{expr} in an internal
variable. @var{var} specifies the number of the variable where to
store the value, and it is a value ranging from 0 to 9. The function
returns the value stored in the internal variable.
Note, Variables are currently not shared between expressions.
@item ld(var)
Allow to load the value of the internal variable with number
@var{var}, which was previously stored with st(@var{var}, @var{expr}).
The function returns the loaded value.
@item while(cond, expr)
Evaluate expression @var{expr} while the expression @var{cond} is
non-zero, and returns the value of the last @var{expr} evaluation, or
NAN if @var{cond} was always false.
@item ceil(expr)
Round the value of expression @var{expr} upwards to the nearest
integer. For example, "ceil(1.5)" is "2.0".
@item floor(expr)
Round the value of expression @var{expr} downwards to the nearest
integer. For example, "floor(-1.5)" is "-2.0".
@item trunc(expr)
Round the value of expression @var{expr} towards zero to the nearest
integer. For example, "trunc(-1.5)" is "-1.0".
@item sqrt(expr)
Compute the square root of @var{expr}. This is equivalent to
"(@var{expr})^.5".
@item not(expr)
Return 1.0 if @var{expr} is zero, 0.0 otherwise.
@item pow(x, y)
Compute the power of @var{x} elevated @var{y}, it is equivalent to
"(@var{x})^(@var{y})".
@item random(x)
Return a pseudo random value between 0.0 and 1.0. @var{x} is the index of the
internal variable which will be used to save the seed/state.
@item hypot(x, y)
This function is similar to the C function with the same name; it returns
"sqrt(@var{x}*@var{x} + @var{y}*@var{y})", the length of the hypotenuse of a
right triangle with sides of length @var{x} and @var{y}, or the distance of the
point (@var{x}, @var{y}) from the origin.
@item gcd(x, y)
Return the greatest common divisor of @var{x} and @var{y}. If both @var{x} and
@var{y} are 0 or either or both are less than zero then behavior is undefined.
@item if(x, y)
Evaluate @var{x}, and if the result is non-zero return the result of
the evaluation of @var{y}, return 0 otherwise.
@item ifnot(x, y)
Evaluate @var{x}, and if the result is zero return the result of the
evaluation of @var{y}, return 0 otherwise.
@end table
The following constants are available:
@table @option
@item PI
area of the unit disc, approximately 3.14
@item E
exp(1) (Euler's number), approximately 2.718
@item PHI
golden ratio (1+sqrt(5))/2, approximately 1.618
@end table
Assuming that an expression is considered "true" if it has a non-zero
value, note that:
@code{*} works like AND
@code{+} works like OR
and the construct:
@example
if A then B else C
@end example
is equivalent to
@example
if(A,B) + ifnot(A,C)
@end example
In your C code, you can extend the list of unary and binary functions,
and define recognized constants, so that they are available for your
expressions.
The evaluator also recognizes the International System number
postfixes. If 'i' is appended after the postfix, powers of 2 are used
instead of powers of 10. The 'B' postfix multiplies the value for 8,
and can be appended after another postfix or used alone. This allows
using for example 'KB', 'MiB', 'G' and 'B' as postfix.
Follows the list of available International System postfixes, with
indication of the corresponding powers of 10 and of 2.
@table @option
@item y
-24 / -80
@item z
-21 / -70
@item a
-18 / -60
@item f
-15 / -50
@item p
-12 / -40
@item n
-9 / -30
@item u
-6 / -20
@item m
-3 / -10
@item c
-2
@item d
-1
@item h
2
@item k
3 / 10
@item K
3 / 10
@item M
6 / 20
@item G
9 / 30
@item T
12 / 40
@item P
15 / 40
@item E
18 / 50
@item Z
21 / 60
@item Y
24 / 70
@end table
@c man end
-21
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@@ -1,21 +0,0 @@
# use pkg-config for getting CFLAGS abd LDFLAGS
FFMPEG_LIBS=libavdevice libavformat libavfilter libavcodec libswscale libavutil
CFLAGS+=$(shell pkg-config --cflags $(FFMPEG_LIBS))
LDFLAGS+=$(shell pkg-config --libs $(FFMPEG_LIBS))
EXAMPLES=decoding_encoding filtering metadata muxing
OBJS=$(addsuffix .o,$(EXAMPLES))
%: %.o
$(CC) $< $(LDFLAGS) -o $@
%.o: %.c
$(CC) $< $(CFLAGS) -c -o $@
.phony: all clean
all: $(OBJS) $(EXAMPLES)
clean:
rm -rf $(EXAMPLES) $(OBJS)
-477
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@@ -1,477 +0,0 @@
/*
* Copyright (c) 2001 Fabrice Bellard
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* libavcodec API use example.
*
* Note that libavcodec only handles codecs (mpeg, mpeg4, etc...),
* not file formats (avi, vob, mp4, mov, mkv, mxf, flv, mpegts, mpegps, etc...). See library 'libavformat' for the
* format handling
*/
#include "libavutil/imgutils.h"
#include "libavutil/opt.h"
#include "libavcodec/avcodec.h"
#include "libavutil/mathematics.h"
#include "libavutil/samplefmt.h"
#define INBUF_SIZE 4096
#define AUDIO_INBUF_SIZE 20480
#define AUDIO_REFILL_THRESH 4096
/*
* Audio encoding example
*/
static void audio_encode_example(const char *filename)
{
AVCodec *codec;
AVCodecContext *c= NULL;
int frame_size, i, j, out_size, outbuf_size;
FILE *f;
short *samples;
float t, tincr;
uint8_t *outbuf;
printf("Audio encoding\n");
/* find the MP2 encoder */
codec = avcodec_find_encoder(CODEC_ID_MP2);
if (!codec) {
fprintf(stderr, "codec not found\n");
exit(1);
}
c = avcodec_alloc_context3(codec);
/* put sample parameters */
c->bit_rate = 64000;
c->sample_rate = 44100;
c->channels = 2;
c->sample_fmt = AV_SAMPLE_FMT_S16;
/* open it */
if (avcodec_open(c, codec) < 0) {
fprintf(stderr, "could not open codec\n");
exit(1);
}
/* the codec gives us the frame size, in samples */
frame_size = c->frame_size;
samples = malloc(frame_size * 2 * c->channels);
outbuf_size = 10000;
outbuf = malloc(outbuf_size);
f = fopen(filename, "wb");
if (!f) {
fprintf(stderr, "could not open %s\n", filename);
exit(1);
}
/* encode a single tone sound */
t = 0;
tincr = 2 * M_PI * 440.0 / c->sample_rate;
for(i=0;i<200;i++) {
for(j=0;j<frame_size;j++) {
samples[2*j] = (int)(sin(t) * 10000);
samples[2*j+1] = samples[2*j];
t += tincr;
}
/* encode the samples */
out_size = avcodec_encode_audio(c, outbuf, outbuf_size, samples);
fwrite(outbuf, 1, out_size, f);
}
fclose(f);
free(outbuf);
free(samples);
avcodec_close(c);
av_free(c);
}
/*
* Audio decoding.
*/
static void audio_decode_example(const char *outfilename, const char *filename)
{
AVCodec *codec;
AVCodecContext *c= NULL;
int len;
FILE *f, *outfile;
uint8_t inbuf[AUDIO_INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
AVPacket avpkt;
AVFrame *decoded_frame = NULL;
av_init_packet(&avpkt);
printf("Audio decoding\n");
/* find the mpeg audio decoder */
codec = avcodec_find_decoder(CODEC_ID_MP2);
if (!codec) {
fprintf(stderr, "codec not found\n");
exit(1);
}
c = avcodec_alloc_context3(codec);
/* open it */
if (avcodec_open(c, codec) < 0) {
fprintf(stderr, "could not open codec\n");
exit(1);
}
f = fopen(filename, "rb");
if (!f) {
fprintf(stderr, "could not open %s\n", filename);
exit(1);
}
outfile = fopen(outfilename, "wb");
if (!outfile) {
av_free(c);
exit(1);
}
/* decode until eof */
avpkt.data = inbuf;
avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
while (avpkt.size > 0) {
int got_frame = 0;
if (!decoded_frame) {
if (!(decoded_frame = avcodec_alloc_frame())) {
fprintf(stderr, "out of memory\n");
exit(1);
}
} else
avcodec_get_frame_defaults(decoded_frame);
len = avcodec_decode_audio4(c, decoded_frame, &got_frame, &avpkt);
if (len < 0) {
fprintf(stderr, "Error while decoding\n");
exit(1);
}
if (got_frame) {
/* if a frame has been decoded, output it */
int data_size = av_samples_get_buffer_size(NULL, c->channels,
decoded_frame->nb_samples,
c->sample_fmt, 1);
fwrite(decoded_frame->data[0], 1, data_size, outfile);
}
avpkt.size -= len;
avpkt.data += len;
avpkt.dts =
avpkt.pts = AV_NOPTS_VALUE;
if (avpkt.size < AUDIO_REFILL_THRESH) {
/* Refill the input buffer, to avoid trying to decode
* incomplete frames. Instead of this, one could also use
* a parser, or use a proper container format through
* libavformat. */
memmove(inbuf, avpkt.data, avpkt.size);
avpkt.data = inbuf;
len = fread(avpkt.data + avpkt.size, 1,
AUDIO_INBUF_SIZE - avpkt.size, f);
if (len > 0)
avpkt.size += len;
}
}
fclose(outfile);
fclose(f);
avcodec_close(c);
av_free(c);
av_free(decoded_frame);
}
/*
* Video encoding example
*/
static void video_encode_example(const char *filename, int codec_id)
{
AVCodec *codec;
AVCodecContext *c= NULL;
int i, out_size, size, x, y, outbuf_size;
FILE *f;
AVFrame *picture;
uint8_t *outbuf;
printf("Video encoding\n");
/* find the mpeg1 video encoder */
codec = avcodec_find_encoder(codec_id);
if (!codec) {
fprintf(stderr, "codec not found\n");
exit(1);
}
c = avcodec_alloc_context3(codec);
picture= avcodec_alloc_frame();
/* put sample parameters */
c->bit_rate = 400000;
/* resolution must be a multiple of two */
c->width = 352;
c->height = 288;
/* frames per second */
c->time_base= (AVRational){1,25};
c->gop_size = 10; /* emit one intra frame every ten frames */
c->max_b_frames=1;
c->pix_fmt = PIX_FMT_YUV420P;
if(codec_id == CODEC_ID_H264)
av_opt_set(c->priv_data, "preset", "slow", 0);
/* open it */
if (avcodec_open(c, codec) < 0) {
fprintf(stderr, "could not open codec\n");
exit(1);
}
f = fopen(filename, "wb");
if (!f) {
fprintf(stderr, "could not open %s\n", filename);
exit(1);
}
/* alloc image and output buffer */
outbuf_size = 100000;
outbuf = malloc(outbuf_size);
/* the image can be allocated by any means and av_image_alloc() is
* just the most convenient way if av_malloc() is to be used */
av_image_alloc(picture->data, picture->linesize,
c->width, c->height, c->pix_fmt, 1);
/* encode 1 second of video */
for(i=0;i<25;i++) {
fflush(stdout);
/* prepare a dummy image */
/* Y */
for(y=0;y<c->height;y++) {
for(x=0;x<c->width;x++) {
picture->data[0][y * picture->linesize[0] + x] = x + y + i * 3;
}
}
/* Cb and Cr */
for(y=0;y<c->height/2;y++) {
for(x=0;x<c->width/2;x++) {
picture->data[1][y * picture->linesize[1] + x] = 128 + y + i * 2;
picture->data[2][y * picture->linesize[2] + x] = 64 + x + i * 5;
}
}
/* encode the image */
out_size = avcodec_encode_video(c, outbuf, outbuf_size, picture);
printf("encoding frame %3d (size=%5d)\n", i, out_size);
fwrite(outbuf, 1, out_size, f);
}
/* get the delayed frames */
for(; out_size; i++) {
fflush(stdout);
out_size = avcodec_encode_video(c, outbuf, outbuf_size, NULL);
printf("write frame %3d (size=%5d)\n", i, out_size);
fwrite(outbuf, 1, out_size, f);
}
/* add sequence end code to have a real mpeg file */
outbuf[0] = 0x00;
outbuf[1] = 0x00;
outbuf[2] = 0x01;
outbuf[3] = 0xb7;
fwrite(outbuf, 1, 4, f);
fclose(f);
free(outbuf);
avcodec_close(c);
av_free(c);
av_free(picture->data[0]);
av_free(picture);
printf("\n");
}
/*
* Video decoding example
*/
static void pgm_save(unsigned char *buf, int wrap, int xsize, int ysize,
char *filename)
{
FILE *f;
int i;
f=fopen(filename,"w");
fprintf(f,"P5\n%d %d\n%d\n",xsize,ysize,255);
for(i=0;i<ysize;i++)
fwrite(buf + i * wrap,1,xsize,f);
fclose(f);
}
static void video_decode_example(const char *outfilename, const char *filename)
{
AVCodec *codec;
AVCodecContext *c= NULL;
int frame, got_picture, len;
FILE *f;
AVFrame *picture;
uint8_t inbuf[INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
char buf[1024];
AVPacket avpkt;
av_init_packet(&avpkt);
/* set end of buffer to 0 (this ensures that no overreading happens for damaged mpeg streams) */
memset(inbuf + INBUF_SIZE, 0, FF_INPUT_BUFFER_PADDING_SIZE);
printf("Video decoding\n");
/* find the mpeg1 video decoder */
codec = avcodec_find_decoder(CODEC_ID_MPEG1VIDEO);
if (!codec) {
fprintf(stderr, "codec not found\n");
exit(1);
}
c = avcodec_alloc_context3(codec);
picture= avcodec_alloc_frame();
if(codec->capabilities&CODEC_CAP_TRUNCATED)
c->flags|= CODEC_FLAG_TRUNCATED; /* we do not send complete frames */
/* For some codecs, such as msmpeg4 and mpeg4, width and height
MUST be initialized there because this information is not
available in the bitstream. */
/* open it */
if (avcodec_open(c, codec) < 0) {
fprintf(stderr, "could not open codec\n");
exit(1);
}
/* the codec gives us the frame size, in samples */
f = fopen(filename, "rb");
if (!f) {
fprintf(stderr, "could not open %s\n", filename);
exit(1);
}
frame = 0;
for(;;) {
avpkt.size = fread(inbuf, 1, INBUF_SIZE, f);
if (avpkt.size == 0)
break;
/* NOTE1: some codecs are stream based (mpegvideo, mpegaudio)
and this is the only method to use them because you cannot
know the compressed data size before analysing it.
BUT some other codecs (msmpeg4, mpeg4) are inherently frame
based, so you must call them with all the data for one
frame exactly. You must also initialize 'width' and
'height' before initializing them. */
/* NOTE2: some codecs allow the raw parameters (frame size,
sample rate) to be changed at any frame. We handle this, so
you should also take care of it */
/* here, we use a stream based decoder (mpeg1video), so we
feed decoder and see if it could decode a frame */
avpkt.data = inbuf;
while (avpkt.size > 0) {
len = avcodec_decode_video2(c, picture, &got_picture, &avpkt);
if (len < 0) {
fprintf(stderr, "Error while decoding frame %d\n", frame);
exit(1);
}
if (got_picture) {
printf("saving frame %3d\n", frame);
fflush(stdout);
/* the picture is allocated by the decoder. no need to
free it */
snprintf(buf, sizeof(buf), outfilename, frame);
pgm_save(picture->data[0], picture->linesize[0],
c->width, c->height, buf);
frame++;
}
avpkt.size -= len;
avpkt.data += len;
}
}
/* some codecs, such as MPEG, transmit the I and P frame with a
latency of one frame. You must do the following to have a
chance to get the last frame of the video */
avpkt.data = NULL;
avpkt.size = 0;
len = avcodec_decode_video2(c, picture, &got_picture, &avpkt);
if (got_picture) {
printf("saving last frame %3d\n", frame);
fflush(stdout);
/* the picture is allocated by the decoder. no need to
free it */
snprintf(buf, sizeof(buf), outfilename, frame);
pgm_save(picture->data[0], picture->linesize[0],
c->width, c->height, buf);
frame++;
}
fclose(f);
avcodec_close(c);
av_free(c);
av_free(picture);
printf("\n");
}
int main(int argc, char **argv)
{
const char *filename;
/* must be called before using avcodec lib */
avcodec_init();
/* register all the codecs */
avcodec_register_all();
if (argc <= 1) {
audio_encode_example("/tmp/test.mp2");
audio_decode_example("/tmp/test.sw", "/tmp/test.mp2");
video_encode_example("/tmp/test.h264", CODEC_ID_H264);
video_encode_example("/tmp/test.mpg", CODEC_ID_MPEG1VIDEO);
filename = "/tmp/test.mpg";
} else {
filename = argv[1];
}
// audio_decode_example("/tmp/test.sw", filename);
video_decode_example("/tmp/test%d.pgm", filename);
return 0;
}
-229
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@@ -1,229 +0,0 @@
/*
* Copyright (c) 2010 Nicolas George
* Copyright (c) 2011 Stefano Sabatini
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* API example for decoding and filtering
*/
#define _XOPEN_SOURCE 600 /* for usleep */
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavfilter/avfiltergraph.h>
#include <libavfilter/vsrc_buffer.h>
const char *filter_descr = "scale=78:24";
static AVFormatContext *fmt_ctx;
static AVCodecContext *dec_ctx;
AVFilterContext *buffersink_ctx;
AVFilterContext *buffersrc_ctx;
AVFilterGraph *filter_graph;
static int video_stream_index = -1;
static int64_t last_pts = AV_NOPTS_VALUE;
static int open_input_file(const char *filename)
{
int ret, i;
AVCodec *dec;
if ((ret = avformat_open_input(&fmt_ctx, filename, NULL, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
return ret;
}
if ((ret = av_find_stream_info(fmt_ctx)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
return ret;
}
/* select the video stream */
ret = av_find_best_stream(fmt_ctx, AVMEDIA_TYPE_VIDEO, -1, -1, &dec, 0);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find a video stream in the input file\n");
return ret;
}
video_stream_index = ret;
dec_ctx = fmt_ctx->streams[video_stream_index]->codec;
/* init the video decoder */
if ((ret = avcodec_open(dec_ctx, dec)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open video decoder\n");
return ret;
}
return 0;
}
static int init_filters(const char *filters_descr)
{
char args[512];
int ret;
AVFilter *buffersrc = avfilter_get_by_name("buffer");
AVFilter *buffersink = avfilter_get_by_name("buffersink");
AVFilterInOut *outputs = avfilter_inout_alloc();
AVFilterInOut *inputs = avfilter_inout_alloc();
enum PixelFormat pix_fmts[] = { PIX_FMT_GRAY8, PIX_FMT_NONE };
filter_graph = avfilter_graph_alloc();
/* buffer video source: the decoded frames from the decoder will be inserted here. */
snprintf(args, sizeof(args), "%d:%d:%d:%d:%d:%d:%d",
dec_ctx->width, dec_ctx->height, dec_ctx->pix_fmt,
dec_ctx->time_base.num, dec_ctx->time_base.den,
dec_ctx->sample_aspect_ratio.num, dec_ctx->sample_aspect_ratio.den);
ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in",
args, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create buffer source\n");
return ret;
}
/* buffer video sink: to terminate the filter chain. */
ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out",
NULL, pix_fmts, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create buffer sink\n");
return ret;
}
/* Endpoints for the filter graph. */
outputs->name = av_strdup("in");
outputs->filter_ctx = buffersrc_ctx;
outputs->pad_idx = 0;
outputs->next = NULL;
inputs->name = av_strdup("out");
inputs->filter_ctx = buffersink_ctx;
inputs->pad_idx = 0;
inputs->next = NULL;
if ((ret = avfilter_graph_parse(filter_graph, filter_descr,
&inputs, &outputs, NULL)) < 0)
return ret;
if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
return ret;
}
static void display_picref(AVFilterBufferRef *picref, AVRational time_base)
{
int x, y;
uint8_t *p0, *p;
int64_t delay;
if (picref->pts != AV_NOPTS_VALUE) {
if (last_pts != AV_NOPTS_VALUE) {
/* sleep roughly the right amount of time;
* usleep is in microseconds, just like AV_TIME_BASE. */
delay = av_rescale_q(picref->pts - last_pts,
time_base, AV_TIME_BASE_Q);
if (delay > 0 && delay < 1000000)
usleep(delay);
}
last_pts = picref->pts;
}
/* Trivial ASCII grayscale display. */
p0 = picref->data[0];
puts("\033c");
for (y = 0; y < picref->video->h; y++) {
p = p0;
for (x = 0; x < picref->video->w; x++)
putchar(" .-+#"[*(p++) / 52]);
putchar('\n');
p0 += picref->linesize[0];
}
fflush(stdout);
}
int main(int argc, char **argv)
{
int ret;
AVPacket packet;
AVFrame frame;
int got_frame;
if (argc != 2) {
fprintf(stderr, "Usage: %s file\n", argv[0]);
exit(1);
}
avcodec_register_all();
av_register_all();
avfilter_register_all();
if ((ret = open_input_file(argv[1])) < 0)
goto end;
if ((ret = init_filters(filter_descr)) < 0)
goto end;
/* read all packets */
while (1) {
AVFilterBufferRef *picref;
if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
break;
if (packet.stream_index == video_stream_index) {
avcodec_get_frame_defaults(&frame);
got_frame = 0;
ret = avcodec_decode_video2(dec_ctx, &frame, &got_frame, &packet);
av_free_packet(&packet);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error decoding video\n");
break;
}
if (got_frame) {
if (frame.pts == AV_NOPTS_VALUE)
frame.pts = frame.pkt_dts == AV_NOPTS_VALUE ?
frame.pkt_dts : frame.pkt_pts;
/* push the decoded frame into the filtergraph */
av_vsrc_buffer_add_frame(buffersrc_ctx, &frame);
/* pull filtered pictures from the filtergraph */
while (avfilter_poll_frame(buffersink_ctx->inputs[0])) {
av_vsink_buffer_get_video_buffer_ref(buffersink_ctx, &picref, 0);
if (picref) {
display_picref(picref, buffersink_ctx->inputs[0]->time_base);
avfilter_unref_buffer(picref);
}
}
}
}
}
end:
avfilter_graph_free(&filter_graph);
if (dec_ctx)
avcodec_close(dec_ctx);
av_close_input_file(fmt_ctx);
if (ret < 0 && ret != AVERROR_EOF) {
char buf[1024];
av_strerror(ret, buf, sizeof(buf));
fprintf(stderr, "Error occurred: %s\n", buf);
exit(1);
}
exit(0);
}
-55
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@@ -1,55 +0,0 @@
/*
* Copyright (c) 2011 Reinhard Tartler
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* Shows how the metadata API can be used in application programs.
*/
#include <stdio.h>
#include <libavformat/avformat.h>
#include <libavutil/dict.h>
int main (int argc, char **argv)
{
AVFormatContext *fmt_ctx = NULL;
AVDictionaryEntry *tag = NULL;
int ret;
if (argc != 2) {
printf("usage: %s <input_file>\n"
"example program to demonstrate the use of the libavformat metadata API.\n"
"\n", argv[0]);
return 1;
}
av_register_all();
if ((ret = avformat_open_input(&fmt_ctx, argv[1], NULL, NULL)))
return ret;
while ((tag = av_dict_get(fmt_ctx->metadata, "", tag, AV_DICT_IGNORE_SUFFIX)))
printf("%s=%s\n", tag->key, tag->value);
avformat_free_context(fmt_ctx);
return 0;
}
-556
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@@ -1,556 +0,0 @@
/*
* Copyright (c) 2003 Fabrice Bellard
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* libavformat API example.
*
* Output a media file in any supported libavformat format.
* The default codecs are used.
*/
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include <math.h>
#include "libavutil/mathematics.h"
#include "libavformat/avformat.h"
#include "libswscale/swscale.h"
#undef exit
/* 5 seconds stream duration */
#define STREAM_DURATION 200.0
#define STREAM_FRAME_RATE 25 /* 25 images/s */
#define STREAM_NB_FRAMES ((int)(STREAM_DURATION * STREAM_FRAME_RATE))
#define STREAM_PIX_FMT PIX_FMT_YUV420P /* default pix_fmt */
static int sws_flags = SWS_BICUBIC;
/**************************************************************/
/* audio output */
static float t, tincr, tincr2;
static int16_t *samples;
static uint8_t *audio_outbuf;
static int audio_outbuf_size;
static int audio_input_frame_size;
/*
* add an audio output stream
*/
static AVStream *add_audio_stream(AVFormatContext *oc, enum CodecID codec_id)
{
AVCodecContext *c;
AVStream *st;
st = avformat_new_stream(oc, NULL);
if (!st) {
fprintf(stderr, "Could not alloc stream\n");
exit(1);
}
st->id = 1;
c = st->codec;
c->codec_id = codec_id;
c->codec_type = AVMEDIA_TYPE_AUDIO;
/* put sample parameters */
c->sample_fmt = AV_SAMPLE_FMT_S16;
c->bit_rate = 64000;
c->sample_rate = 44100;
c->channels = 2;
// some formats want stream headers to be separate
if (oc->oformat->flags & AVFMT_GLOBALHEADER)
c->flags |= CODEC_FLAG_GLOBAL_HEADER;
return st;
}
static void open_audio(AVFormatContext *oc, AVStream *st)
{
AVCodecContext *c;
AVCodec *codec;
c = st->codec;
/* find the audio encoder */
codec = avcodec_find_encoder(c->codec_id);
if (!codec) {
fprintf(stderr, "codec not found\n");
exit(1);
}
/* open it */
if (avcodec_open(c, codec) < 0) {
fprintf(stderr, "could not open codec\n");
exit(1);
}
/* init signal generator */
t = 0;
tincr = 2 * M_PI * 110.0 / c->sample_rate;
/* increment frequency by 110 Hz per second */
tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate;
audio_outbuf_size = 10000;
audio_outbuf = av_malloc(audio_outbuf_size);
/* ugly hack for PCM codecs (will be removed ASAP with new PCM
support to compute the input frame size in samples */
if (c->frame_size <= 1) {
audio_input_frame_size = audio_outbuf_size / c->channels;
switch(st->codec->codec_id) {
case CODEC_ID_PCM_S16LE:
case CODEC_ID_PCM_S16BE:
case CODEC_ID_PCM_U16LE:
case CODEC_ID_PCM_U16BE:
audio_input_frame_size >>= 1;
break;
default:
break;
}
} else {
audio_input_frame_size = c->frame_size;
}
samples = av_malloc(audio_input_frame_size * 2 * c->channels);
}
/* prepare a 16 bit dummy audio frame of 'frame_size' samples and
'nb_channels' channels */
static void get_audio_frame(int16_t *samples, int frame_size, int nb_channels)
{
int j, i, v;
int16_t *q;
q = samples;
for (j = 0; j < frame_size; j++) {
v = (int)(sin(t) * 10000);
for(i = 0; i < nb_channels; i++)
*q++ = v;
t += tincr;
tincr += tincr2;
}
}
static void write_audio_frame(AVFormatContext *oc, AVStream *st)
{
AVCodecContext *c;
AVPacket pkt;
av_init_packet(&pkt);
c = st->codec;
get_audio_frame(samples, audio_input_frame_size, c->channels);
pkt.size = avcodec_encode_audio(c, audio_outbuf, audio_outbuf_size, samples);
if (c->coded_frame && c->coded_frame->pts != AV_NOPTS_VALUE)
pkt.pts= av_rescale_q(c->coded_frame->pts, c->time_base, st->time_base);
pkt.flags |= AV_PKT_FLAG_KEY;
pkt.stream_index = st->index;
pkt.data = audio_outbuf;
/* write the compressed frame in the media file */
if (av_interleaved_write_frame(oc, &pkt) != 0) {
fprintf(stderr, "Error while writing audio frame\n");
exit(1);
}
}
static void close_audio(AVFormatContext *oc, AVStream *st)
{
avcodec_close(st->codec);
av_free(samples);
av_free(audio_outbuf);
}
/**************************************************************/
/* video output */
static AVFrame *picture, *tmp_picture;
static uint8_t *video_outbuf;
static int frame_count, video_outbuf_size;
/* add a video output stream */
static AVStream *add_video_stream(AVFormatContext *oc, enum CodecID codec_id)
{
AVCodecContext *c;
AVStream *st;
AVCodec *codec;
st = avformat_new_stream(oc, NULL);
if (!st) {
fprintf(stderr, "Could not alloc stream\n");
exit(1);
}
c = st->codec;
/* find the video encoder */
codec = avcodec_find_encoder(codec_id);
if (!codec) {
fprintf(stderr, "codec not found\n");
exit(1);
}
avcodec_get_context_defaults3(c, codec);
c->codec_id = codec_id;
/* put sample parameters */
c->bit_rate = 400000;
/* resolution must be a multiple of two */
c->width = 352;
c->height = 288;
/* time base: this is the fundamental unit of time (in seconds) in terms
of which frame timestamps are represented. for fixed-fps content,
timebase should be 1/framerate and timestamp increments should be
identically 1. */
c->time_base.den = STREAM_FRAME_RATE;
c->time_base.num = 1;
c->gop_size = 12; /* emit one intra frame every twelve frames at most */
c->pix_fmt = STREAM_PIX_FMT;
if (c->codec_id == CODEC_ID_MPEG2VIDEO) {
/* just for testing, we also add B frames */
c->max_b_frames = 2;
}
if (c->codec_id == CODEC_ID_MPEG1VIDEO){
/* Needed to avoid using macroblocks in which some coeffs overflow.
This does not happen with normal video, it just happens here as
the motion of the chroma plane does not match the luma plane. */
c->mb_decision=2;
}
// some formats want stream headers to be separate
if (oc->oformat->flags & AVFMT_GLOBALHEADER)
c->flags |= CODEC_FLAG_GLOBAL_HEADER;
return st;
}
static AVFrame *alloc_picture(enum PixelFormat pix_fmt, int width, int height)
{
AVFrame *picture;
uint8_t *picture_buf;
int size;
picture = avcodec_alloc_frame();
if (!picture)
return NULL;
size = avpicture_get_size(pix_fmt, width, height);
picture_buf = av_malloc(size);
if (!picture_buf) {
av_free(picture);
return NULL;
}
avpicture_fill((AVPicture *)picture, picture_buf,
pix_fmt, width, height);
return picture;
}
static void open_video(AVFormatContext *oc, AVStream *st)
{
AVCodec *codec;
AVCodecContext *c;
c = st->codec;
/* find the video encoder */
codec = avcodec_find_encoder(c->codec_id);
if (!codec) {
fprintf(stderr, "codec not found\n");
exit(1);
}
/* open the codec */
if (avcodec_open(c, codec) < 0) {
fprintf(stderr, "could not open codec\n");
exit(1);
}
video_outbuf = NULL;
if (!(oc->oformat->flags & AVFMT_RAWPICTURE)) {
/* allocate output buffer */
/* XXX: API change will be done */
/* buffers passed into lav* can be allocated any way you prefer,
as long as they're aligned enough for the architecture, and
they're freed appropriately (such as using av_free for buffers
allocated with av_malloc) */
video_outbuf_size = 200000;
video_outbuf = av_malloc(video_outbuf_size);
}
/* allocate the encoded raw picture */
picture = alloc_picture(c->pix_fmt, c->width, c->height);
if (!picture) {
fprintf(stderr, "Could not allocate picture\n");
exit(1);
}
/* if the output format is not YUV420P, then a temporary YUV420P
picture is needed too. It is then converted to the required
output format */
tmp_picture = NULL;
if (c->pix_fmt != PIX_FMT_YUV420P) {
tmp_picture = alloc_picture(PIX_FMT_YUV420P, c->width, c->height);
if (!tmp_picture) {
fprintf(stderr, "Could not allocate temporary picture\n");
exit(1);
}
}
}
/* prepare a dummy image */
static void fill_yuv_image(AVFrame *pict, int frame_index, int width, int height)
{
int x, y, i;
i = frame_index;
/* Y */
for (y = 0; y < height; y++) {
for (x = 0; x < width; x++) {
pict->data[0][y * pict->linesize[0] + x] = x + y + i * 3;
}
}
/* Cb and Cr */
for (y = 0; y < height/2; y++) {
for (x = 0; x < width/2; x++) {
pict->data[1][y * pict->linesize[1] + x] = 128 + y + i * 2;
pict->data[2][y * pict->linesize[2] + x] = 64 + x + i * 5;
}
}
}
static void write_video_frame(AVFormatContext *oc, AVStream *st)
{
int out_size, ret;
AVCodecContext *c;
static struct SwsContext *img_convert_ctx;
c = st->codec;
if (frame_count >= STREAM_NB_FRAMES) {
/* no more frame to compress. The codec has a latency of a few
frames if using B frames, so we get the last frames by
passing the same picture again */
} else {
if (c->pix_fmt != PIX_FMT_YUV420P) {
/* as we only generate a YUV420P picture, we must convert it
to the codec pixel format if needed */
if (img_convert_ctx == NULL) {
img_convert_ctx = sws_getContext(c->width, c->height,
PIX_FMT_YUV420P,
c->width, c->height,
c->pix_fmt,
sws_flags, NULL, NULL, NULL);
if (img_convert_ctx == NULL) {
fprintf(stderr, "Cannot initialize the conversion context\n");
exit(1);
}
}
fill_yuv_image(tmp_picture, frame_count, c->width, c->height);
sws_scale(img_convert_ctx, tmp_picture->data, tmp_picture->linesize,
0, c->height, picture->data, picture->linesize);
} else {
fill_yuv_image(picture, frame_count, c->width, c->height);
}
}
if (oc->oformat->flags & AVFMT_RAWPICTURE) {
/* raw video case. The API will change slightly in the near
future for that. */
AVPacket pkt;
av_init_packet(&pkt);
pkt.flags |= AV_PKT_FLAG_KEY;
pkt.stream_index = st->index;
pkt.data = (uint8_t *)picture;
pkt.size = sizeof(AVPicture);
ret = av_interleaved_write_frame(oc, &pkt);
} else {
/* encode the image */
out_size = avcodec_encode_video(c, video_outbuf, video_outbuf_size, picture);
/* if zero size, it means the image was buffered */
if (out_size > 0) {
AVPacket pkt;
av_init_packet(&pkt);
if (c->coded_frame->pts != AV_NOPTS_VALUE)
pkt.pts= av_rescale_q(c->coded_frame->pts, c->time_base, st->time_base);
if(c->coded_frame->key_frame)
pkt.flags |= AV_PKT_FLAG_KEY;
pkt.stream_index = st->index;
pkt.data = video_outbuf;
pkt.size = out_size;
/* write the compressed frame in the media file */
ret = av_interleaved_write_frame(oc, &pkt);
} else {
ret = 0;
}
}
if (ret != 0) {
fprintf(stderr, "Error while writing video frame\n");
exit(1);
}
frame_count++;
}
static void close_video(AVFormatContext *oc, AVStream *st)
{
avcodec_close(st->codec);
av_free(picture->data[0]);
av_free(picture);
if (tmp_picture) {
av_free(tmp_picture->data[0]);
av_free(tmp_picture);
}
av_free(video_outbuf);
}
/**************************************************************/
/* media file output */
int main(int argc, char **argv)
{
const char *filename;
AVOutputFormat *fmt;
AVFormatContext *oc;
AVStream *audio_st, *video_st;
double audio_pts, video_pts;
int i;
/* initialize libavcodec, and register all codecs and formats */
av_register_all();
if (argc != 2) {
printf("usage: %s output_file\n"
"API example program to output a media file with libavformat.\n"
"The output format is automatically guessed according to the file extension.\n"
"Raw images can also be output by using '%%d' in the filename\n"
"\n", argv[0]);
return 1;
}
filename = argv[1];
/* allocate the output media context */
avformat_alloc_output_context2(&oc, NULL, NULL, filename);
if (!oc) {
printf("Could not deduce output format from file extension: using MPEG.\n");
avformat_alloc_output_context2(&oc, NULL, "mpeg", filename);
}
if (!oc) {
return 1;
}
fmt = oc->oformat;
/* add the audio and video streams using the default format codecs
and initialize the codecs */
video_st = NULL;
audio_st = NULL;
if (fmt->video_codec != CODEC_ID_NONE) {
video_st = add_video_stream(oc, fmt->video_codec);
}
if (fmt->audio_codec != CODEC_ID_NONE) {
audio_st = add_audio_stream(oc, fmt->audio_codec);
}
av_dump_format(oc, 0, filename, 1);
/* now that all the parameters are set, we can open the audio and
video codecs and allocate the necessary encode buffers */
if (video_st)
open_video(oc, video_st);
if (audio_st)
open_audio(oc, audio_st);
/* open the output file, if needed */
if (!(fmt->flags & AVFMT_NOFILE)) {
if (avio_open(&oc->pb, filename, AVIO_FLAG_WRITE) < 0) {
fprintf(stderr, "Could not open '%s'\n", filename);
return 1;
}
}
/* write the stream header, if any */
av_write_header(oc);
picture->pts = 0;
for(;;) {
/* compute current audio and video time */
if (audio_st)
audio_pts = (double)audio_st->pts.val * audio_st->time_base.num / audio_st->time_base.den;
else
audio_pts = 0.0;
if (video_st)
video_pts = (double)video_st->pts.val * video_st->time_base.num / video_st->time_base.den;
else
video_pts = 0.0;
if ((!audio_st || audio_pts >= STREAM_DURATION) &&
(!video_st || video_pts >= STREAM_DURATION))
break;
/* write interleaved audio and video frames */
if (!video_st || (video_st && audio_st && audio_pts < video_pts)) {
write_audio_frame(oc, audio_st);
} else {
write_video_frame(oc, video_st);
picture->pts++;
}
}
/* write the trailer, if any. the trailer must be written
* before you close the CodecContexts open when you wrote the
* header; otherwise write_trailer may try to use memory that
* was freed on av_codec_close() */
av_write_trailer(oc);
/* close each codec */
if (video_st)
close_video(oc, video_st);
if (audio_st)
close_audio(oc, audio_st);
/* free the streams */
for(i = 0; i < oc->nb_streams; i++) {
av_freep(&oc->streams[i]->codec);
av_freep(&oc->streams[i]);
}
if (!(fmt->flags & AVFMT_NOFILE)) {
/* close the output file */
avio_close(oc->pb);
}
/* free the stream */
av_free(oc);
return 0;
}
-430
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@@ -1,430 +0,0 @@
\input texinfo @c -*- texinfo -*-
@settitle FFmpeg FAQ
@titlepage
@center @titlefont{FFmpeg FAQ}
@end titlepage
@top
@contents
@chapter General Questions
@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 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 mpeg1video
.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_s16le
always
@item libmp3lame
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...
If you have large number of pictures to rename, you can use the
following command to ease the burden. The command, using the bourne
shell syntax, symbolically links all files in the current directory
that match @code{*jpg} to the @file{/tmp} directory in the sequence of
@file{img001.jpg}, @file{img002.jpg} and so on.
@example
x=1; for i in *jpg; do counter=$(printf %03d $x); ln -s "$i" /tmp/img"$counter".jpg; x=$(($x+1)); done
@end example
If you want to sequence them by oldest modified first, substitute
@code{$(ls -r -t *jpg)} in place of @code{*jpg}.
Then run:
@example
ffmpeg -f image2 -i /tmp/img%03d.jpg /tmp/a.mpg
@end example
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 -c:v ppm
@item -c:v png
@item -c:v mjpeg
@end table
to force the encoding.
Applying that to the previous example:
@example
ffmpeg -i movie.mpg -f image2 -c:v 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 -f jpeg doesn't work.
Try '-f image2 test%d.jpg'.
@section Why can I not change the frame rate?
Some codecs, like MPEG-1/2, only allow a small number of fixed frame rates.
Choose a different codec with the -c:v 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 '-c:v 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 Which are good parameters for encoding high quality MPEG-4?
'-mbd rd -flags +mv4+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.
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 the
@uref{http://www.avisynth.org/, Avisynth homepage}.
@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 -same_quant intermediate1.mpg
ffmpeg -i input2.avi -same_quant intermediate2.mpg
cat intermediate1.mpg intermediate2.mpg > intermediate_all.mpg
ffmpeg -i intermediate_all.mpg -same_quant output.avi
@end example
Notice that you should either use @code{-same_quant} 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 -same_quant -y intermediate1.mpg < /dev/null &
ffmpeg -i input2.avi -same_quant -y intermediate2.mpg < /dev/null &
cat intermediate1.mpg intermediate2.mpg |\
ffmpeg -f mpeg -i - -same_quant -c:v 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.
When using multiple yuv4mpegpipe(s), the first line needs to be discarded
from all but the first stream. This can be accomplished by piping through
@code{tail} as seen below. Note that when piping through @code{tail} you
must use command grouping, @code{@{ ;@}}, to background properly.
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 - < /dev/null | tail -n +2 > temp2.v ; @} &
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 \
-same_quant -y output.flv
rm temp[12].[av] all.[av]
@end example
@section -profile option fails when encoding H.264 video with AAC audio
@command{ffmpeg} prints an error like
@example
Undefined constant or missing '(' in 'baseline'
Unable to parse option value "baseline"
Error setting option profile to value baseline.
@end example
Short answer: write @option{-profile:v} instead of @option{-profile}.
Long answer: this happens because the @option{-profile} option can apply to both
video and audio. Specifically the AAC encoder also defines some profiles, none
of which are named @var{baseline}.
The solution is to apply the @option{-profile} option to the video stream only
by using @url{http://ffmpeg.org/ffmpeg.html#Stream-specifiers-1, Stream specifiers}.
Appending @code{:v} to it will do exactly that.
@section Using @option{-f lavfi}, audio becomes mono for no apparent reason.
Use @option{-dumpgraph -} to find out exactly where the channel layout is
lost.
Most likely, it is through @code{auto-inserted aconvert}. Try to understand
why the converting filter was needed at that place.
Just before the output is a likely place, as @option{-f lavfi} currently
only support packed S16.
Then insert the correct @code{aconvert} explicitly in the filter graph,
specifying the exact format.
@example
aconvert=s16:stereo:packed
@end example
@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 @uref{http://www.tux.org/lkml/#s15, "Programming Religion"}.
@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, use 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'm using FFmpeg 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'm using libavutil from within my C++ application but the compiler complains about 'UINT64_C' was not declared in this scope
FFmpeg is a pure C project using C99 math features, in order to enable C++
to use them you have to append -D__STDC_CONSTANT_MACROS to your CXXFLAGS
@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 @file{libavformat/file.c} in
FFmpeg and @file{libmpdemux/demux_lavf.c} in MPlayer sources.
@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://www.ffmpeg.org/~michael/}
@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 frame rate.
r_frame_rate is NOT the average frame rate, it is the smallest frame rate
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.
@section Why is @code{make fate} not running all tests?
Make sure you have the fate-suite samples and the @code{SAMPLES} Make variable
or @code{FATE_SAMPLES} environment variable or the @code{--samples}
@command{configure} option is set to the right path.
@section Why is @code{make fate} not finding the samples?
Do you happen to have a @code{~} character in the samples path to indicate a
home directory? The value is used in ways where the shell cannot expand it,
causing FATE to not find files. Just replace @code{~} by the full path.
@bye
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\input texinfo @c -*- texinfo -*-
@settitle FATE Automated Testing Environment
@titlepage
@center @titlefont{FATE Automated Testing Environment}
@end titlepage
@node Top
@top
@contents
@chapter Introduction
FATE is an extended regression suite on the client-side and a means
for results aggregation and presentation on the server-side.
The first part of this document explains how you can use FATE from
your FFmpeg source directory to test your ffmpeg binary. The second
part describes how you can run FATE to submit the results to FFmpeg's
FATE server.
In any way you can have a look at the publicly viewable FATE results
by visiting this website:
@url{http://fate.ffmpeg.org/}
This is especially recommended for all people contributing source
code to FFmpeg, as it can be seen if some test on some platform broke
with there recent contribution. This usually happens on the platforms
the developers could not test on.
The second part of this document describes how you can run FATE to
submit your results to FFmpeg's FATE server. If you want to submit your
results be sure to check that your combination of CPU, OS and compiler
is not already listed on the above mentioned website.
In the third part you can find a comprehensive listing of FATE makefile
targets and variables.
@chapter Using FATE from your FFmpeg source directory
If you want to run FATE on your machine you need to have the samples
in place. You can get the samples via the build target fate-rsync.
Use this command from the top-level source directory:
@example
make fate-rsync SAMPLES=fate-suite/
make fate SAMPLES=fate-suite/
@end example
The above commands set the samples location by passing a makefile
variable via command line. It is also possible to set the samples
location at source configuration time by invoking configure with
`--samples=<path to the samples directory>'. Afterwards you can
invoke the makefile targets without setting the SAMPLES makefile
variable. This is illustrated by the following commands:
@example
./configure --samples=fate-suite/
make fate-rsync
make fate
@end example
Yet another way to tell FATE about the location of the sample
directory is by making sure the environment variable FATE_SAMPLES
contains the path to your samples directory. This can be achieved
by e.g. putting that variable in your shell profile or by setting
it in your interactive session.
@example
FATE_SAMPLES=fate-suite/ make fate
@end example
@float NOTE
Do not put a '~' character in the samples path to indicate a home
directory. Because of shell nuances, this will cause FATE to fail.
@end float
@chapter Submitting the results to the FFmpeg result aggregation server
To submit your results to the server you should run fate through the
shell script tests/fate.sh from the FFmpeg sources. This script needs
to be invoked with a configuration file as its first argument.
@example
tests/fate.sh /path/to/fate_config
@end example
A configuration file template with comments describing the individual
configuration variables can be found at @file{tests/fate_config.sh.template}.
@ifhtml
The mentioned configuration template is also available here:
@verbatiminclude ../tests/fate_config.sh.template
@end ifhtml
Create a configuration that suits your needs, based on the configuration
template. The `slot' configuration variable can be any string that is not
yet used, but it is suggested that you name it adhering to the following
pattern <arch>-<os>-<compiler>-<compiler version>. The configuration file
itself will be sourced in a shell script, therefore all shell features may
be used. This enables you to setup the environment as you need it for your
build.
For your first test runs the `fate_recv' variable should be empty or
commented out. This will run everything as normal except that it will omit
the submission of the results to the server. The following files should be
present in $workdir as specified in the configuration file:
@itemize
@item configure.log
@item compile.log
@item test.log
@item report
@item version
@end itemize
When you have everything working properly you can create an SSH key and
send its public part to the FATE server administrator.
Configure your SSH client to use public key authentication with that key
when connecting to the FATE server. Also do not forget to check the identity
of the server and to accept its host key. This can usually be achieved by
running your SSH client manually and killing it after you accepted the key.
The FATE server's fingerprint is:
b1:31:c8:79:3f:04:1d:f8:f2:23:26:5a:fd:55:fa:92
The only thing left is to automate the execution of the fate.sh script and
the synchronisation of the samples directory.
@chapter FATE makefile targets and variables
@section Makefile targets
@table @option
@item fate-rsync
Download/synchronize sample files to the configured samples directory.
@item fate-list
Will list all fate/regression test targets.
@item fate
Run the FATE test suite (requires the fate-suite dataset).
@end table
@section Makefile variables
@table @option
@item V
Verbosity level, can be set to 0, 1 or 2.
@itemize
@item 0: show just the test arguments
@item 1: show just the command used in the test
@item 2: show everything
@end itemize
@item SAMPLES
Specify or override the path to the FATE samples at make time, it has a
meaning only while running the regression tests.
@item THREADS
Specify how many threads to use while running regression tests, it is
quite useful to detect thread-related regressions.
@end table
Example:
@example
make V=1 SAMPLES=/var/fate/samples THREADS=2 fate
@end example
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:
ffmpeg.c : libav*
======== : ======
:
:
--------------------------------:---> AVStream...
InputStream input_streams[] / :
/ :
InputFile input_files[] +==========================+ / ^ :
------> 0 | : st ---:-----------:--/ : :
^ +------+-----------+-----+ / +--------------------------+ : :
: | :ist_index--:-----:---------/ 1 | : st : | : :
: +------+-----------+-----+ +==========================+ : :
nb_input_files : | :ist_index--:-----:------------------> 2 | : st : | : :
: +------+-----------+-----+ +--------------------------+ : nb_input_streams :
: | :ist_index : | 3 | ... | : :
v +------+-----------+-----+ +--------------------------+ : :
--> 4 | | : :
| +--------------------------+ : :
| 5 | | : :
| +==========================+ v :
| :
| :
| :
| :
--------- --------------------------------:---> AVStream...
\ / :
OutputStream output_streams[] / :
\ / :
+======\======================/======+ ^ :
------> 0 | : source_index : st-:--- | : :
OuputFile output_files[] / +------------------------------------+ : :
/ 1 | : : : | : :
^ +------+------------+-----+ / +------------------------------------+ : :
: | : ost_index -:-----:------/ 2 | : : : | : :
nb_output_files : +------+------------+-----+ +====================================+ : :
: | : ost_index -:-----|-----------------> 3 | : : : | : :
: +------+------------+-----+ +------------------------------------+ : nb_output_streams :
: | : : | 4 | | : :
: +------+------------+-----+ +------------------------------------+ : :
: | : : | 5 | | : :
v +------+------------+-----+ +------------------------------------+ : :
6 | | : :
+------------------------------------+ : :
7 | | : :
+====================================+ v :
:
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\input texinfo @c -*- texinfo -*-
@settitle ffplay Documentation
@titlepage
@center @titlefont{ffplay Documentation}
@end titlepage
@top
@contents
@chapter Synopsis
@example
@c man begin SYNOPSIS
ffplay [options] [@file{input_file}]
@c man end
@end example
@chapter Description
@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 Options
@c man begin OPTIONS
@include avtools-common-opts.texi
@section Main options
@table @option
@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 do
not contain a header with the frame size like raw YUV. This option
has been deprecated in favor of private options, try -video_size.
@item -an
Disable audio.
@item -vn
Disable video.
@item -ss @var{pos}
Seek to a given position in seconds.
@item -t @var{duration}
play <duration> seconds of audio/video
@item -bytes
Seek by bytes.
@item -nodisp
Disable graphical display.
@item -f @var{fmt}
Force format.
@item -window_title @var{title}
Set window title (default is the input filename).
@item -loop @var{number}
Loops movie playback <number> times. 0 means forever.
@item -showmode @var{mode}
Set the show mode to use.
Available values for @var{mode} are:
@table @samp
@item 0, video
show video
@item 1, waves
show audio waves
@item 2, rdft
show audio frequency band using RDFT ((Inverse) Real Discrete Fourier Transform)
@end table
Default value is "video", if video is not present or cannot be played
"rdft" is automatically selected.
You can interactively cycle through the available show modes by
pressing the key @key{w}.
@item -vf @var{filter_graph}
@var{filter_graph} is a description of the filter graph to apply to
the input video.
Use the option "-filters" to show all the available filters (including
also sources and sinks).
@item -i @var{input_file}
Read @var{input_file}.
@end table
@section Advanced options
@table @option
@item -pix_fmt @var{format}
Set pixel format.
This option has been deprecated in favor of private options, try -pixel_format.
@item -stats
Show the stream duration, the codec parameters, the current position in
the stream and the audio/video synchronisation drift.
@item -bug
Work around bugs.
@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.
@item -autoexit
Exit when video is done playing.
@item -exitonkeydown
Exit if any key is pressed.
@item -exitonmousedown
Exit if any mouse button is pressed.
@item -codec:@var{stream_type}
Force a specific decoder implementation
@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 page down/page up
Seek backward/forward 10 minutes.
@item mouse click
Seek to percentage in file corresponding to fraction of width.
@end table
@c man end
@include eval.texi
@include decoders.texi
@include demuxers.texi
@include muxers.texi
@include indevs.texi
@include outdevs.texi
@include protocols.texi
@include filters.texi
@ignore
@setfilename ffplay
@settitle FFplay media player
@c man begin SEEALSO
ffmpeg(1), ffprobe(1), ffserver(1) and the FFmpeg HTML documentation
@c man end
@c man begin AUTHORS
The FFmpeg developers
@c man end
@end ignore
@bye
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\input texinfo @c -*- texinfo -*-
@settitle ffprobe Documentation
@titlepage
@center @titlefont{ffprobe Documentation}
@end titlepage
@top
@contents
@chapter Synopsis
The generic syntax is:
@example
@c man begin SYNOPSIS
ffprobe [options] [@file{input_file}]
@c man end
@end example
@chapter Description
@c man begin DESCRIPTION
ffprobe gathers information from multimedia streams and prints it in
human- and machine-readable fashion.
For example it can be used to check the format of the container used
by a multimedia stream and the format and type of each media stream
contained in it.
If a filename is specified in input, ffprobe will try to open and
probe the file content. If the file cannot be opened or recognized as
a multimedia file, a positive exit code is returned.
ffprobe may be employed both as a standalone application or in
combination with a textual filter, which may perform more
sophisticated processing, e.g. statistical processing or plotting.
Options are used to list some of the formats supported by ffprobe or
for specifying which information to display, and for setting how
ffprobe will show it.
ffprobe output is designed to be easily parsable by a textual filter,
and consists of one or more sections of a form defined by the selected
writer, which is specified by the @option{print_format} option.
Metadata tags stored in the container or in the streams are recognized
and printed in the corresponding "FORMAT" or "STREAM" section.
@c man end
@chapter Options
@c man begin OPTIONS
@include avtools-common-opts.texi
@section Main options
@table @option
@item -f @var{format}
Force format to use.
@item -unit
Show the unit of the displayed values.
@item -prefix
Use SI prefixes for the displayed values.
Unless the "-byte_binary_prefix" option is used all the prefixes
are decimal.
@item -byte_binary_prefix
Force the use of binary prefixes for byte values.
@item -sexagesimal
Use sexagesimal format HH:MM:SS.MICROSECONDS for time values.
@item -pretty
Prettify the format of the displayed values, it corresponds to the
options "-unit -prefix -byte_binary_prefix -sexagesimal".
@item -print_format @var{writer_name}[=@var{writer_options}]
Set the output printing format.
@var{writer_name} specifies the name of the writer, and
@var{writer_options} specifies the options to be passed to the writer.
For example for printing the output in JSON format, specify:
@example
-print_format json
@end example
For more details on the available output printing formats, see the
Writers section below.
@item -show_error
Show information about the error found when trying to probe the input.
The error information is printed within a section with name "ERROR".
@item -show_format
Show information about the container format of the input multimedia
stream.
All the container format information is printed within a section with
name "FORMAT".
@item -show_packets
Show information about each packet contained in the input multimedia
stream.
The information for each single packet is printed within a dedicated
section with name "PACKET".
@item -show_frames
Show information about each frame contained in the input multimedia
stream.
The information for each single frame is printed within a dedicated
section with name "FRAME".
@item -show_streams
Show information about each media stream contained in the input
multimedia stream.
Each media stream information is printed within a dedicated section
with name "STREAM".
@item -show_private_data, -private
Show private data, that is data depending on the format of the
particular shown element.
This option is enabled by default, but you may need to disable it
for specific uses, for example when creating XSD-compliant XML output.
@item -show_program_version
Show information related to program version.
Version information is printed within a section with name
"PROGRAM_VERSION".
@item -show_library_versions
Show information related to library versions.
Version information for each library is printed within a section with
name "LIBRARY_VERSION".
@item -show_versions
Show information related to program and library versions. This is the
equivalent of setting both @option{-show_program_version} and
@option{-show_library_versions} options.
@item -i @var{input_file}
Read @var{input_file}.
@end table
@c man end
@chapter Writers
@c man begin WRITERS
A writer defines the output format adopted by @command{ffprobe}, and will be
used for printing all the parts of the output.
A writer may accept one or more arguments, which specify the options to
adopt.
A description of the currently available writers follows.
@section default
Default format.
Print each section in the form:
@example
[SECTION]
key1=val1
...
keyN=valN
[/SECTION]
@end example
Metadata tags are printed as a line in the corresponding FORMAT or
STREAM section, and are prefixed by the string "TAG:".
@section compact
Compact format.
Each section is printed on a single line.
If no option is specifid, the output has the form:
@example
section|key1=val1| ... |keyN=valN
@end example
Metadata tags are printed in the corresponding "format" or "stream"
section. A metadata tag key, if printed, is prefixed by the string
"tag:".
This writer accepts options as a list of @var{key}=@var{value} pairs,
separated by ":".
The description of the accepted options follows.
@table @option
@item item_sep, s
Specify the character to use for separating fields in the output line.
It must be a single printable character, it is "|" by default.
@item nokey, nk
If set to 1 specify not to print the key of each field. Its default
value is 0.
@item escape, e
Set the escape mode to use, default to "c".
It can assume one of the following values:
@table @option
@item c
Perform C-like escaping. Strings containing a newline ('\n') or
carriage return ('\r'), the escaping character ('\') or the item
separator character @var{SEP} are escaped using C-like fashioned
escaping, so that a newline is converted to the sequence "\n", a
carriage return to "\r", '\' to "\\" and the separator @var{SEP} is
converted to "\@var{SEP}".
@item csv
Perform CSV-like escaping, as described in RFC4180. Strings
containing a newline ('\n'), a carriage return ('\r'), a double quote
('"'), or @var{SEP} are enclosed in double-quotes.
@item none
Perform no escaping.
@end table
@end table
@section csv
CSV format.
This writer is equivalent to
@code{compact=item_sep=,:nokey=1:escape=csv}.
@section json
JSON based format.
Each section is printed using JSON notation.
This writer accepts options as a list of @var{key}=@var{value} pairs,
separated by ":".
The description of the accepted options follows.
@table @option
@item compact, c
If set to 1 enable compact output, that is each section will be
printed on a single line. Default value is 0.
@end table
For more information about JSON, see @url{http://www.json.org/}.
@section xml
XML based format.
The XML output is described in the XML schema description file
@file{ffprobe.xsd} installed in the FFmpeg datadir.
Note that the output issued will be compliant to the
@file{ffprobe.xsd} schema only when no special global output options
(@option{unit}, @option{prefix}, @option{byte_binary_prefix},
@option{sexagesimal} etc.) are specified.
This writer accepts options as a list of @var{key}=@var{value} pairs,
separated by ":".
The description of the accepted options follows.
@table @option
@item fully_qualified, q
If set to 1 specify if the output should be fully qualified. Default
value is 0.
This is required for generating an XML file which can be validated
through an XSD file.
@item xsd_compliant, x
If set to 1 perform more checks for ensuring that the output is XSD
compliant. Default value is 0.
This option automatically sets @option{fully_qualified} to 1.
@end table
For more information about the XML format, see
@url{http://www.w3.org/XML/}.
@chapter Timecode
@command{ffprobe} supports Timecode extraction:
@itemize
@item MPEG1/2 timecode is extracted from the GOP, and is available in the video
stream details (@option{-show_streams}, see @var{timecode}).
@item MOV timecode is extracted from tmcd track, so is available in the tmcd
stream metadata (@option{-show_streams}, see @var{TAG:timecode}).
@item DV and GXF timecodes are available in format metadata
(@option{-show_format}, see @var{TAG:timecode}).
@end itemize
@c man end WRITERS
@include decoders.texi
@include demuxers.texi
@include protocols.texi
@include indevs.texi
@ignore
@setfilename ffprobe
@settitle ffprobe media prober
@c man begin SEEALSO
ffmpeg(1), ffplay(1), ffserver(1) and the FFmpeg HTML documentation
@c man end
@c man begin AUTHORS
The FFmpeg developers
@c man end
@end ignore
@bye
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@@ -1,164 +0,0 @@
<?xml version="1.0" encoding="UTF-8"?>
<xsd:schema xmlns:xsd="http://www.w3.org/2001/XMLSchema"
targetNamespace="http://www.ffmpeg.org/schema/ffprobe"
xmlns:ffprobe="http://www.ffmpeg.org/schema/ffprobe">
<xsd:element name="ffprobe" type="ffprobe:ffprobeType"/>
<xsd:complexType name="ffprobeType">
<xsd:sequence>
<xsd:element name="packets" type="ffprobe:packetsType" minOccurs="0" maxOccurs="1" />
<xsd:element name="frames" type="ffprobe:framesType" minOccurs="0" maxOccurs="1" />
<xsd:element name="streams" type="ffprobe:streamsType" minOccurs="0" maxOccurs="1" />
<xsd:element name="format" type="ffprobe:formatType" minOccurs="0" maxOccurs="1" />
<xsd:element name="error" type="ffprobe:errorType" minOccurs="0" maxOccurs="1" />
<xsd:element name="program_version" type="ffprobe:programVersionType" minOccurs="0" maxOccurs="1" />
<xsd:element name="library_versions" type="ffprobe:libraryVersionsType" minOccurs="0" maxOccurs="1" />
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="packetsType">
<xsd:sequence>
<xsd:element name="packet" type="ffprobe:packetType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="framesType">
<xsd:sequence>
<xsd:element name="frame" type="ffprobe:frameType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="packetType">
<xsd:attribute name="codec_type" type="xsd:string" use="required" />
<xsd:attribute name="stream_index" type="xsd:int" use="required" />
<xsd:attribute name="pts" type="xsd:long" />
<xsd:attribute name="pts_time" type="xsd:float" />
<xsd:attribute name="dts" type="xsd:long" />
<xsd:attribute name="dts_time" type="xsd:float" />
<xsd:attribute name="duration" type="xsd:long" />
<xsd:attribute name="duration_time" type="xsd:float" />
<xsd:attribute name="size" type="xsd:long" use="required" />
<xsd:attribute name="pos" type="xsd:long" />
<xsd:attribute name="flags" type="xsd:string" use="required" />
</xsd:complexType>
<xsd:complexType name="frameType">
<xsd:attribute name="media_type" type="xsd:string" use="required"/>
<xsd:attribute name="key_frame" type="xsd:int" use="required"/>
<xsd:attribute name="pts" type="xsd:long" />
<xsd:attribute name="pts_time" type="xsd:float"/>
<xsd:attribute name="pkt_pts" type="xsd:long" />
<xsd:attribute name="pkt_pts_time" type="xsd:float"/>
<xsd:attribute name="pkt_dts" type="xsd:long" />
<xsd:attribute name="pkt_dts_time" type="xsd:float"/>
<xsd:attribute name="pkt_pos" type="xsd:long" />
<!-- audio attributes -->
<xsd:attribute name="sample_fmt" type="xsd:string"/>
<xsd:attribute name="nb_samples" type="xsd:long" />
<!-- video attributes -->
<xsd:attribute name="width" type="xsd:long" />
<xsd:attribute name="height" type="xsd:long" />
<xsd:attribute name="pix_fmt" type="xsd:string"/>
<xsd:attribute name="sample_aspect_ratio" type="xsd:string"/>
<xsd:attribute name="pict_type" type="xsd:string"/>
<xsd:attribute name="coded_picture_number" type="xsd:long" />
<xsd:attribute name="display_picture_number" type="xsd:long" />
<xsd:attribute name="interlaced_frame" type="xsd:int" />
<xsd:attribute name="top_field_first" type="xsd:int" />
<xsd:attribute name="repeat_pict" type="xsd:int" />
<xsd:attribute name="reference" type="xsd:int" />
</xsd:complexType>
<xsd:complexType name="streamsType">
<xsd:sequence>
<xsd:element name="stream" type="ffprobe:streamType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="streamType">
<xsd:attribute name="index" type="xsd:int" use="required"/>
<xsd:attribute name="codec_name" type="xsd:string" />
<xsd:attribute name="codec_long_name" type="xsd:string" />
<xsd:attribute name="codec_type" type="xsd:string" />
<xsd:attribute name="codec_time_base" type="xsd:string" use="required"/>
<xsd:attribute name="codec_tag" type="xsd:string" use="required"/>
<xsd:attribute name="codec_tag_string" type="xsd:string" use="required"/>
<!-- video attributes -->
<xsd:attribute name="width" type="xsd:int"/>
<xsd:attribute name="height" type="xsd:int"/>
<xsd:attribute name="has_b_frames" type="xsd:int"/>
<xsd:attribute name="sample_aspect_ratio" type="xsd:string"/>
<xsd:attribute name="display_aspect_ratio" type="xsd:string"/>
<xsd:attribute name="pix_fmt" type="xsd:string"/>
<xsd:attribute name="level" type="xsd:int"/>
<xsd:attribute name="timecode" type="xsd:string"/>
<!-- audio attributes -->
<xsd:attribute name="sample_fmt" type="xsd:string"/>
<xsd:attribute name="sample_rate" type="xsd:int"/>
<xsd:attribute name="channels" type="xsd:int"/>
<xsd:attribute name="bits_per_sample" type="xsd:int"/>
<xsd:attribute name="id" type="xsd:string"/>
<xsd:attribute name="r_frame_rate" type="xsd:string" use="required"/>
<xsd:attribute name="avg_frame_rate" type="xsd:string" use="required"/>
<xsd:attribute name="time_base" type="xsd:string" use="required"/>
<xsd:attribute name="start_time" type="xsd:float"/>
<xsd:attribute name="duration" type="xsd:float"/>
<xsd:attribute name="nb_frames" type="xsd:int"/>
</xsd:complexType>
<xsd:complexType name="formatType">
<xsd:sequence>
<xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
<xsd:attribute name="filename" type="xsd:string" use="required"/>
<xsd:attribute name="nb_streams" type="xsd:int" use="required"/>
<xsd:attribute name="format_name" type="xsd:string" use="required"/>
<xsd:attribute name="format_long_name" type="xsd:string" use="required"/>
<xsd:attribute name="start_time" type="xsd:float"/>
<xsd:attribute name="duration" type="xsd:float"/>
<xsd:attribute name="size" type="xsd:long"/>
<xsd:attribute name="bit_rate" type="xsd:long"/>
</xsd:complexType>
<xsd:complexType name="tagType">
<xsd:attribute name="key" type="xsd:string" use="required"/>
<xsd:attribute name="value" type="xsd:string" use="required"/>
</xsd:complexType>
<xsd:complexType name="errorType">
<xsd:attribute name="code" type="xsd:int" use="required"/>
<xsd:attribute name="string" type="xsd:string" use="required"/>
</xsd:complexType>
<xsd:complexType name="programVersionType">
<xsd:attribute name="version" type="xsd:string" use="required"/>
<xsd:attribute name="copyright" type="xsd:string" use="required"/>
<xsd:attribute name="build_date" type="xsd:string" use="required"/>
<xsd:attribute name="build_time" type="xsd:string" use="required"/>
<xsd:attribute name="compiler_type" type="xsd:string" use="required"/>
<xsd:attribute name="compiler_version" type="xsd:string" use="required"/>
<xsd:attribute name="configuration" type="xsd:string" use="required"/>
</xsd:complexType>
<xsd:complexType name="libraryVersionType">
<xsd:attribute name="name" type="xsd:string" use="required"/>
<xsd:attribute name="major" type="xsd:int" use="required"/>
<xsd:attribute name="minor" type="xsd:int" use="required"/>
<xsd:attribute name="micro" type="xsd:int" use="required"/>
<xsd:attribute name="version" type="xsd:int" use="required"/>
</xsd:complexType>
<xsd:complexType name="libraryVersionsType">
<xsd:sequence>
<xsd:element name="library_version" type="ffprobe:libraryVersionType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
</xsd:schema>
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@@ -1,377 +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>
# Transcode an incoming live feed to another live feed,
# using libx264 and video presets
#<Stream live.h264>
#Format rtp
#Feed feed1.ffm
#VideoCodec libx264
#VideoFrameRate 24
#VideoBitRate 100
#VideoSize 480x272
#AVPresetVideo default
#AVPresetVideo baseline
#AVOptionVideo flags +global_header
#
#AudioCodec libfaac
#AudioBitRate 32
#AudioChannels 2
#AudioSampleRate 22050
#AVOptionAudio flags +global_header
#</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>
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\input texinfo @c -*- texinfo -*-
@settitle ffserver Documentation
@titlepage
@center @titlefont{ffserver Documentation}
@end titlepage
@top
@contents
@chapter Synopsys
The generic syntax is:
@example
@c man begin SYNOPSIS
ffserver [options]
@c man end
@end example
@chapter Description
@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.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 lose 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 Options
@c man begin OPTIONS
@include avtools-common-opts.texi
@section Main options
@table @option
@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. Since ffserver will not launch
any ffmpeg instances, 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), ffprobe(1), the @file{ffserver.conf}
example and the FFmpeg HTML documentation
@c man end
@c man begin AUTHORS
The FFmpeg developers
@c man end
@end ignore
@bye
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\input texinfo @c -*- texinfo -*-
@settitle General Documentation
@titlepage
@center @titlefont{General Documentation}
@end titlepage
@top
@contents
@chapter External libraries
FFmpeg can be hooked up with a number of external libraries to add support
for more formats. None of them are used by default, their use has to be
explicitly requested by passing the appropriate flags to @file{./configure}.
@section OpenJPEG
FFmpeg can use the OpenJPEG libraries for encoding/decoding J2K videos. Go to
@url{http://www.openjpeg.org/} to get the libraries and follow the installation
instructions. To enable using OpenJPEG in FFmpeg, pass @code{--enable-libopenjpeg} to
@file{./configure}.
@section OpenCORE and VisualOn libraries
Spun off Google Android sources, OpenCore and VisualOn libraries provide
encoders for a number of audio codecs.
@float NOTE
OpenCORE and VisualOn libraries are under the Apache License 2.0
(see @url{http://www.apache.org/licenses/LICENSE-2.0} for details), which is
incompatible with the LGPL version 2.1 and GPL version 2. You have to
upgrade FFmpeg's license to LGPL version 3 (or if you have enabled
GPL components, GPL version 3) to use it.
@end float
@subsection OpenCORE AMR
FFmpeg can make use of the OpenCORE libraries for AMR-NB
decoding/encoding and AMR-WB decoding.
Go to @url{http://sourceforge.net/projects/opencore-amr/} and follow the
instructions for installing the libraries.
Then pass @code{--enable-libopencore-amrnb} and/or
@code{--enable-libopencore-amrwb} to configure to enable them.
@subsection VisualOn AAC encoder library
FFmpeg can make use of the VisualOn AACenc library for AAC encoding.
Go to @url{http://sourceforge.net/projects/opencore-amr/} and follow the
instructions for installing the library.
Then pass @code{--enable-libvo-aacenc} to configure to enable it.
@subsection VisualOn AMR-WB encoder library
FFmpeg can make use of the VisualOn AMR-WBenc library for AMR-WB encoding.
Go to @url{http://sourceforge.net/projects/opencore-amr/} and follow the
instructions for installing the library.
Then pass @code{--enable-libvo-amrwbenc} to configure to enable it.
@section LAME
FFmpeg can make use of the LAME library for MP3 encoding.
Go to @url{http://lame.sourceforge.net/} and follow the
instructions for installing the library.
Then pass @code{--enable-libmp3lame} to configure to enable it.
@section libvpx
FFmpeg can make use of the libvpx library for VP8 encoding.
Go to @url{http://www.webmproject.org/} and follow the instructions for
installing the library. Then pass @code{--enable-libvpx} to configure to
enable it.
@section x264
FFmpeg can make use of the x264 library for H.264 encoding.
Go to @url{http://www.videolan.org/developers/x264.html} and follow the
instructions for installing the library. Then pass @code{--enable-libx264} to
configure to enable it.
@float NOTE
x264 is under the GNU Public License Version 2 or later
(see @url{http://www.gnu.org/licenses/old-licenses/gpl-2.0.html} for
details), you must upgrade FFmpeg's license to GPL in order to use it.
@end float
@chapter Supported File Formats, Codecs or Features
You can use the @code{-formats} and @code{-codecs} options to have an exhaustive list.
@section File Formats
FFmpeg supports the following file formats through the @code{libavformat}
library:
@multitable @columnfractions .4 .1 .1 .4
@item Name @tab Encoding @tab Decoding @tab Comments
@item 4xm @tab @tab X
@tab 4X Technologies format, used in some games.
@item 8088flex TMV @tab @tab X
@item ACT Voice @tab @tab X
@tab contains G.729 audio
@item Adobe Filmstrip @tab X @tab X
@item Audio IFF (AIFF) @tab X @tab X
@item American Laser Games MM @tab @tab X
@tab Multimedia format used in games like Mad Dog McCree.
@item 3GPP AMR @tab X @tab X
@item Apple HTTP Live Streaming @tab @tab X
@item Artworx Data Format @tab @tab X
@item ASF @tab X @tab X
@item AVI @tab X @tab X
@item AVISynth @tab @tab X
@item AVS @tab @tab X
@tab Multimedia format used by the Creature Shock game.
@item Beam Software SIFF @tab @tab X
@tab Audio and video format used in some games by Beam Software.
@item Bethesda Softworks VID @tab @tab X
@tab Used in some games from Bethesda Softworks.
@item Binary text @tab @tab X
@item Bink @tab @tab X
@tab Multimedia format used by many games.
@item Bitmap Brothers JV @tab @tab X
@tab Used in Z and Z95 games.
@item Brute Force & Ignorance @tab @tab X
@tab Used in the game Flash Traffic: City of Angels.
@item BWF @tab X @tab X
@item CRI ADX @tab X @tab X
@tab Audio-only format used in console video games.
@item Discworld II BMV @tab @tab X
@item Interplay C93 @tab @tab X
@tab Used in the game Cyberia from Interplay.
@item Delphine Software International CIN @tab @tab X
@tab Multimedia format used by Delphine Software games.
@item CD+G @tab @tab X
@tab Video format used by CD+G karaoke disks
@item Core Audio Format @tab X @tab X
@tab Apple Core Audio Format
@item CRC testing format @tab X @tab
@item Creative Voice @tab X @tab X
@tab Created for the Sound Blaster Pro.
@item CRYO APC @tab @tab X
@tab Audio format used in some games by CRYO Interactive Entertainment.
@item D-Cinema audio @tab X @tab X
@item Deluxe Paint Animation @tab @tab X
@item DFA @tab @tab X
@tab This format is used in Chronomaster game
@item DV video @tab X @tab X
@item DXA @tab @tab X
@tab This format is used in the non-Windows version of the Feeble Files
game and different game cutscenes repacked for use with ScummVM.
@item Electronic Arts cdata @tab @tab X
@item Electronic Arts Multimedia @tab @tab X
@tab Used in various EA games; files have extensions like WVE and UV2.
@item FFM (FFserver live feed) @tab X @tab X
@item Flash (SWF) @tab X @tab X
@item Flash 9 (AVM2) @tab X @tab X
@tab Only embedded audio is decoded.
@item FLI/FLC/FLX animation @tab @tab X
@tab .fli/.flc files
@item Flash Video (FLV) @tab X @tab X
@tab Macromedia Flash video files
@item framecrc testing format @tab X @tab
@item FunCom ISS @tab @tab X
@tab Audio format used in various games from FunCom like The Longest Journey.
@item G.723.1 @tab X @tab X
@item G.729 BIT @tab X @tab X
@item G.729 raw @tab @tab X
@item GIF Animation @tab X @tab
@item GXF @tab X @tab X
@tab General eXchange Format SMPTE 360M, used by Thomson Grass Valley
playout servers.
@item iCEDraw File @tab @tab X
@item ICO @tab @tab X
@tab Microsoft Windows ICO
@item id Quake II CIN video @tab @tab X
@item id RoQ @tab X @tab X
@tab Used in Quake III, Jedi Knight 2 and other computer games.
@item IEC61937 encapsulation @tab X @tab X
@item IFF @tab @tab X
@tab Interchange File Format
@item Interplay MVE @tab @tab X
@tab Format used in various Interplay computer games.
@item IV8 @tab @tab X
@tab A format generated by IndigoVision 8000 video server.
@item IVF (On2) @tab X @tab X
@tab A format used by libvpx
@item LATM @tab X @tab X
@item LMLM4 @tab @tab X
@tab Used by Linux Media Labs MPEG-4 PCI boards
@item LOAS @tab @tab X
@tab contains LATM multiplexed AAC audio
@item LXF @tab @tab X
@tab VR native stream format, used by Leitch/Harris' video servers.
@item Matroska @tab X @tab X
@item Matroska audio @tab X @tab
@item FFmpeg metadata @tab X @tab X
@tab Metadata in text format.
@item MAXIS XA @tab @tab X
@tab Used in Sim City 3000; file extension .xa.
@item MD Studio @tab @tab X
@item Mobotix .mxg @tab @tab X
@item Monkey's Audio @tab @tab X
@item Motion Pixels MVI @tab @tab X
@item MOV/QuickTime/MP4 @tab X @tab X
@tab 3GP, 3GP2, PSP, iPod variants supported
@item MP2 @tab X @tab X
@item MP3 @tab X @tab X
@item MPEG-1 System @tab X @tab X
@tab muxed audio and video, VCD format supported
@item MPEG-PS (program stream) @tab X @tab X
@tab also known as @code{VOB} file, SVCD and DVD format supported
@item MPEG-TS (transport stream) @tab X @tab X
@tab also known as DVB Transport Stream
@item MPEG-4 @tab X @tab X
@tab MPEG-4 is a variant of QuickTime.
@item MIME multipart JPEG @tab X @tab
@item MSN TCP webcam @tab @tab X
@tab Used by MSN Messenger webcam streams.
@item MTV @tab @tab X
@item Musepack @tab @tab X
@item Musepack SV8 @tab @tab X
@item Material eXchange Format (MXF) @tab X @tab X
@tab SMPTE 377M, used by D-Cinema, broadcast industry.
@item Material eXchange Format (MXF), D-10 Mapping @tab X @tab X
@tab SMPTE 386M, D-10/IMX Mapping.
@item NC camera feed @tab @tab X
@tab NC (AVIP NC4600) camera streams
@item NTT TwinVQ (VQF) @tab @tab X
@tab Nippon Telegraph and Telephone Corporation TwinVQ.
@item Nullsoft Streaming Video @tab @tab X
@item NuppelVideo @tab @tab X
@item NUT @tab X @tab X
@tab NUT Open Container Format
@item Ogg @tab X @tab X
@item Playstation Portable PMP @tab @tab X
@item TechnoTrend PVA @tab @tab X
@tab Used by TechnoTrend DVB PCI boards.
@item QCP @tab @tab X
@item raw ADTS (AAC) @tab X @tab X
@item raw AC-3 @tab X @tab X
@item raw Chinese AVS video @tab X @tab X
@item raw CRI ADX @tab X @tab X
@item raw Dirac @tab X @tab X
@item raw DNxHD @tab X @tab X
@item raw DTS @tab X @tab X
@item raw E-AC-3 @tab X @tab X
@item raw FLAC @tab X @tab X
@item raw GSM @tab @tab X
@item raw H.261 @tab X @tab X
@item raw H.263 @tab X @tab X
@item raw H.264 @tab X @tab X
@item raw Ingenient MJPEG @tab @tab X
@item raw MJPEG @tab X @tab X
@item raw MLP @tab @tab X
@item raw MPEG @tab @tab X
@item raw MPEG-1 @tab @tab X
@item raw MPEG-2 @tab @tab X
@item raw MPEG-4 @tab X @tab X
@item raw NULL @tab X @tab
@item raw video @tab X @tab X
@item raw id RoQ @tab X @tab
@item raw Shorten @tab @tab X
@item raw TrueHD @tab X @tab X
@item raw VC-1 @tab @tab X
@item raw PCM A-law @tab X @tab X
@item raw PCM mu-law @tab X @tab X
@item raw PCM signed 8 bit @tab X @tab X
@item raw PCM signed 16 bit big-endian @tab X @tab X
@item raw PCM signed 16 bit little-endian @tab X @tab X
@item raw PCM signed 24 bit big-endian @tab X @tab X
@item raw PCM signed 24 bit little-endian @tab X @tab X
@item raw PCM signed 32 bit big-endian @tab X @tab X
@item raw PCM signed 32 bit little-endian @tab X @tab X
@item raw PCM unsigned 8 bit @tab X @tab X
@item raw PCM unsigned 16 bit big-endian @tab X @tab X
@item raw PCM unsigned 16 bit little-endian @tab X @tab X
@item raw PCM unsigned 24 bit big-endian @tab X @tab X
@item raw PCM unsigned 24 bit little-endian @tab X @tab X
@item raw PCM unsigned 32 bit big-endian @tab X @tab X
@item raw PCM unsigned 32 bit little-endian @tab X @tab X
@item raw PCM floating-point 32 bit big-endian @tab X @tab X
@item raw PCM floating-point 32 bit little-endian @tab X @tab X
@item raw PCM floating-point 64 bit big-endian @tab X @tab X
@item raw PCM floating-point 64 bit little-endian @tab X @tab X
@item RDT @tab @tab X
@item REDCODE R3D @tab @tab X
@tab File format used by RED Digital cameras, contains JPEG 2000 frames and PCM audio.
@item RealMedia @tab X @tab X
@item Redirector @tab @tab X
@item Renderware TeXture Dictionary @tab @tab X
@item RL2 @tab @tab X
@tab Audio and video format used in some games by Entertainment Software Partners.
@item RPL/ARMovie @tab @tab X
@item Lego Mindstorms RSO @tab X @tab X
@item RTMP @tab X @tab X
@tab Output is performed by publishing stream to RTMP server
@item RTP @tab X @tab X
@item RTSP @tab X @tab X
@item SAP @tab X @tab X
@item SBG @tab @tab X
@item SDP @tab @tab X
@item Sega FILM/CPK @tab @tab X
@tab Used in many Sega Saturn console games.
@item Sierra SOL @tab @tab X
@tab .sol files used in Sierra Online games.
@item Sierra VMD @tab @tab X
@tab Used in Sierra CD-ROM games.
@item Smacker @tab @tab X
@tab Multimedia format used by many games.
@item SMJPEG @tab X @tab X
@tab Used in certain Loki game ports.
@item Sony OpenMG (OMA) @tab X @tab X
@tab Audio format used in Sony Sonic Stage and Sony Vegas.
@item Sony PlayStation STR @tab @tab X
@item Sony Wave64 (W64) @tab @tab X
@item SoX native format @tab X @tab X
@item SUN AU format @tab X @tab X
@item Text files @tab @tab X
@item THP @tab @tab X
@tab Used on the Nintendo GameCube.
@item Tiertex Limited SEQ @tab @tab X
@tab Tiertex .seq files used in the DOS CD-ROM version of the game Flashback.
@item True Audio @tab @tab X
@item VC-1 test bitstream @tab X @tab X
@item WAV @tab X @tab X
@item WavPack @tab @tab X
@item WebM @tab X @tab X
@item Windows Televison (WTV) @tab X @tab X
@item Wing Commander III movie @tab @tab X
@tab Multimedia format used in Origin's Wing Commander III computer game.
@item Westwood Studios audio @tab @tab X
@tab Multimedia format used in Westwood Studios games.
@item Westwood Studios VQA @tab @tab X
@tab Multimedia format used in Westwood Studios games.
@item XMV @tab @tab X
@tab Microsoft video container used in Xbox games.
@item xWMA @tab @tab X
@tab Microsoft audio container used by XAudio 2.
@item eXtended BINary text (XBIN) @tab @tab X
@item YUV4MPEG pipe @tab X @tab X
@item Psygnosis YOP @tab @tab X
@end multitable
@code{X} means that encoding (resp. decoding) is supported.
@section Image Formats
FFmpeg can read and write images for each frame of a video sequence. The
following image formats are supported:
@multitable @columnfractions .4 .1 .1 .4
@item Name @tab Encoding @tab Decoding @tab Comments
@item .Y.U.V @tab X @tab X
@tab one raw file per component
@item animated GIF @tab X @tab X
@tab Only uncompressed GIFs are generated.
@item BMP @tab X @tab X
@tab Microsoft BMP image
@item DPX @tab X @tab X
@tab Digital Picture Exchange
@item JPEG @tab X @tab X
@tab Progressive JPEG is not supported.
@item JPEG 2000 @tab X @tab X
@item JPEG-LS @tab X @tab X
@item LJPEG @tab X @tab
@tab Lossless JPEG
@item PAM @tab X @tab X
@tab PAM is a PNM extension with alpha support.
@item PBM @tab X @tab X
@tab Portable BitMap image
@item PCX @tab X @tab X
@tab PC Paintbrush
@item PGM @tab X @tab X
@tab Portable GrayMap image
@item PGMYUV @tab X @tab X
@tab PGM with U and V components in YUV 4:2:0
@item PIC @tab @tab X
@tab Pictor/PC Paint
@item PNG @tab X @tab X
@item PPM @tab X @tab X
@tab Portable PixelMap image
@item PTX @tab @tab X
@tab V.Flash PTX format
@item SGI @tab X @tab X
@tab SGI RGB image format
@item Sun Rasterfile @tab @tab X
@tab Sun RAS image format
@item TIFF @tab X @tab X
@tab YUV, JPEG and some extension is not supported yet.
@item Truevision Targa @tab X @tab X
@tab Targa (.TGA) image format
@item XWD @tab X @tab X
@tab X Window Dump image format
@end multitable
@code{X} means that encoding (resp. decoding) is supported.
@code{E} means that support is provided through an external library.
@section Video Codecs
@multitable @columnfractions .4 .1 .1 .4
@item Name @tab Encoding @tab Decoding @tab Comments
@item 4X Movie @tab @tab X
@tab Used in certain computer games.
@item 8088flex TMV @tab @tab X
@item 8SVX exponential @tab @tab X
@item 8SVX fibonacci @tab @tab X
@item A64 multicolor @tab X @tab
@tab Creates video suitable to be played on a commodore 64 (multicolor mode).
@item American Laser Games MM @tab @tab X
@tab Used in games like Mad Dog McCree.
@item AMV Video @tab X @tab X
@tab Used in Chinese MP3 players.
@item ANSI/ASCII art @tab @tab X
@item Apple MJPEG-B @tab @tab X
@item Apple ProRes @tab X @tab X
@item Apple QuickDraw @tab @tab X
@tab fourcc: qdrw
@item Asus v1 @tab X @tab X
@tab fourcc: ASV1
@item Asus v2 @tab X @tab X
@tab fourcc: ASV2
@item ATI VCR1 @tab @tab X
@tab fourcc: VCR1
@item ATI VCR2 @tab @tab X
@tab fourcc: VCR2
@item Auravision Aura @tab @tab X
@item Auravision Aura 2 @tab @tab X
@item Autodesk Animator Flic video @tab @tab X
@item Autodesk RLE @tab @tab X
@tab fourcc: AASC
@item Avid 1:1 10-bit RGB Packer @tab X @tab X
@tab fourcc: AVrp
@item AVS (Audio Video Standard) video @tab @tab X
@tab Video encoding used by the Creature Shock game.
@item Beam Software VB @tab @tab X
@item Bethesda VID video @tab @tab X
@tab Used in some games from Bethesda Softworks.
@item Bink Video @tab @tab X
@item Bitmap Brothers JV video @tab @tab X
@item y41p Brooktree uncompressed 4:1:1 12-bit @tab X @tab X
@item Brute Force & Ignorance @tab @tab X
@tab Used in the game Flash Traffic: City of Angels.
@item C93 video @tab @tab X
@tab Codec used in Cyberia game.
@item CamStudio @tab @tab X
@tab fourcc: CSCD
@item CD+G @tab @tab X
@tab Video codec for CD+G karaoke disks
@item Chinese AVS video @tab E @tab X
@tab AVS1-P2, JiZhun profile, encoding through external library libxavs
@item Delphine Software International CIN video @tab @tab X
@tab Codec used in Delphine Software International games.
@item Discworld II BMV Video @tab @tab X
@item Cinepak @tab @tab X
@item Cirrus Logic AccuPak @tab X @tab X
@tab fourcc: CLJR
@item Creative YUV (CYUV) @tab @tab X
@item DFA @tab @tab X
@tab Codec used in Chronomaster game.
@item Dirac @tab E @tab X
@tab supported through external libdirac/libschroedinger libraries
@item Deluxe Paint Animation @tab @tab X
@item DNxHD @tab X @tab X
@tab aka SMPTE VC3
@item Duck TrueMotion 1.0 @tab @tab X
@tab fourcc: DUCK
@item Duck TrueMotion 2.0 @tab @tab X
@tab fourcc: TM20
@item DV (Digital Video) @tab X @tab X
@item Dxtory capture format @tab @tab X
@item Feeble Files/ScummVM DXA @tab @tab X
@tab Codec originally used in Feeble Files game.
@item Electronic Arts CMV video @tab @tab X
@tab Used in NHL 95 game.
@item Electronic Arts Madcow video @tab @tab X
@item Electronic Arts TGV video @tab @tab X
@item Electronic Arts TGQ video @tab @tab X
@item Electronic Arts TQI video @tab @tab X
@item Escape 124 @tab @tab X
@item Escape 130 @tab @tab X
@item FFmpeg video codec #1 @tab X @tab X
@tab experimental lossless codec (fourcc: FFV1)
@item Flash Screen Video v1 @tab X @tab X
@tab fourcc: FSV1
@item Flash Screen Video v2 @tab X @tab X
@item Flash Video (FLV) @tab X @tab X
@tab Sorenson H.263 used in Flash
@item Fraps @tab @tab X
@item H.261 @tab X @tab X
@item H.263 / H.263-1996 @tab X @tab X
@item H.263+ / H.263-1998 / H.263 version 2 @tab X @tab X
@item H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 @tab E @tab X
@tab encoding supported through external library libx264
@item H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 (VDPAU acceleration) @tab E @tab X
@item HuffYUV @tab X @tab X
@item HuffYUV FFmpeg variant @tab X @tab X
@item IBM Ultimotion @tab @tab X
@tab fourcc: ULTI
@item id Cinematic video @tab @tab X
@tab Used in Quake II.
@item id RoQ video @tab X @tab X
@tab Used in Quake III, Jedi Knight 2, other computer games.
@item IFF ILBM @tab @tab X
@tab IFF interleaved bitmap
@item IFF ByteRun1 @tab @tab X
@tab IFF run length encoded bitmap
@item Intel H.263 @tab @tab X
@item Intel Indeo 2 @tab @tab X
@item Intel Indeo 3 @tab @tab X
@item Intel Indeo 4 @tab @tab X
@item Intel Indeo 5 @tab @tab X
@item Interplay C93 @tab @tab X
@tab Used in the game Cyberia from Interplay.
@item Interplay MVE video @tab @tab X
@tab Used in Interplay .MVE files.
@item J2K @tab X @tab X
@item Karl Morton's video codec @tab @tab X
@tab Codec used in Worms games.
@item Kega Game Video (KGV1) @tab @tab X
@tab Kega emulator screen capture codec.
@item Lagarith @tab @tab X
@item LCL (LossLess Codec Library) MSZH @tab @tab X
@item LCL (LossLess Codec Library) ZLIB @tab E @tab E
@item LOCO @tab @tab X
@item lossless MJPEG @tab X @tab X
@item Microsoft RLE @tab @tab X
@item Microsoft Video 1 @tab @tab X
@item Mimic @tab @tab X
@tab Used in MSN Messenger Webcam streams.
@item Miro VideoXL @tab @tab X
@tab fourcc: VIXL
@item MJPEG (Motion JPEG) @tab X @tab X
@item Mobotix MxPEG video @tab @tab X
@item Motion Pixels video @tab @tab X
@item MPEG-1 video @tab X @tab X
@item MPEG-1/2 video XvMC (X-Video Motion Compensation) @tab @tab X
@item MPEG-1/2 video (VDPAU acceleration) @tab @tab X
@item MPEG-2 video @tab X @tab X
@item MPEG-4 part 2 @tab X @tab X
@tab libxvidcore can be used alternatively for encoding.
@item MPEG-4 part 2 Microsoft variant version 1 @tab @tab X
@item MPEG-4 part 2 Microsoft variant version 2 @tab X @tab X
@item MPEG-4 part 2 Microsoft variant version 3 @tab X @tab X
@item Nintendo Gamecube THP video @tab @tab X
@item NuppelVideo/RTjpeg @tab @tab X
@tab Video encoding used in NuppelVideo files.
@item On2 VP3 @tab @tab X
@tab still experimental
@item On2 VP5 @tab @tab X
@tab fourcc: VP50
@item On2 VP6 @tab @tab X
@tab fourcc: VP60,VP61,VP62
@item VP8 @tab E @tab X
@tab fourcc: VP80, encoding supported through external library libvpx
@item planar RGB @tab @tab X
@tab fourcc: 8BPS
@item Prores @tab @tab X
@tab fourcc: apch,apcn,apcs,apco
@item Q-team QPEG @tab @tab X
@tab fourccs: QPEG, Q1.0, Q1.1
@item QuickTime 8BPS video @tab @tab X
@item QuickTime Animation (RLE) video @tab X @tab X
@tab fourcc: 'rle '
@item QuickTime Graphics (SMC) @tab @tab X
@tab fourcc: 'smc '
@item QuickTime video (RPZA) @tab @tab X
@tab fourcc: rpza
@item R10K AJA Kona 10-bit RGB Codec @tab X @tab X
@item R210 Quicktime Uncompressed RGB 10-bit @tab X @tab X
@item Raw Video @tab X @tab X
@item RealVideo 1.0 @tab X @tab X
@item RealVideo 2.0 @tab X @tab X
@item RealVideo 3.0 @tab @tab X
@tab still far from ideal
@item RealVideo 4.0 @tab @tab X
@item Renderware TXD (TeXture Dictionary) @tab @tab X
@tab Texture dictionaries used by the Renderware Engine.
@item RL2 video @tab @tab X
@tab used in some games by Entertainment Software Partners
@item Sierra VMD video @tab @tab X
@tab Used in Sierra VMD files.
@item Smacker video @tab @tab X
@tab Video encoding used in Smacker.
@item SMPTE VC-1 @tab @tab X
@item Snow @tab X @tab X
@tab experimental wavelet codec (fourcc: SNOW)
@item Sony PlayStation MDEC (Motion DECoder) @tab @tab X
@item Sorenson Vector Quantizer 1 @tab X @tab X
@tab fourcc: SVQ1
@item Sorenson Vector Quantizer 3 @tab @tab X
@tab fourcc: SVQ3
@item Sunplus JPEG (SP5X) @tab @tab X
@tab fourcc: SP5X
@item TechSmith Screen Capture Codec @tab @tab X
@tab fourcc: TSCC
@item Theora @tab E @tab X
@tab encoding supported through external library libtheora
@item Tiertex Limited SEQ video @tab @tab X
@tab Codec used in DOS CD-ROM FlashBack game.
@item Ut Video @tab @tab X
@item v210 QuickTime uncompressed 4:2:2 10-bit @tab X @tab X
@item v308 QuickTime uncompressed 4:4:4 @tab X @tab X
@item v410 QuickTime uncompressed 4:4:4 10-bit @tab X @tab X
@item VBLE Lossless Codec @tab @tab X
@item VMware Screen Codec / VMware Video @tab @tab X
@tab Codec used in videos captured by VMware.
@item Westwood Studios VQA (Vector Quantized Animation) video @tab @tab X
@item Windows Media Image @tab @tab X
@item Windows Media Video 7 @tab X @tab X
@item Windows Media Video 8 @tab X @tab X
@item Windows Media Video 9 @tab @tab X
@tab not completely working
@item Wing Commander III / Xan @tab @tab X
@tab Used in Wing Commander III .MVE files.
@item Wing Commander IV / Xan @tab @tab X
@tab Used in Wing Commander IV.
@item Winnov WNV1 @tab @tab X
@item WMV7 @tab X @tab X
@item YAMAHA SMAF @tab X @tab X
@item Psygnosis YOP Video @tab @tab X
@item yuv4 @tab X @tab X
@tab libquicktime uncompressed packed 4:2:0
@item ZLIB @tab X @tab X
@tab part of LCL, encoder experimental
@item Zip Motion Blocks Video @tab X @tab X
@tab Encoder works only in PAL8.
@end multitable
@code{X} means that encoding (resp. decoding) is supported.
@code{E} means that support is provided through an external library.
@section Audio Codecs
@multitable @columnfractions .4 .1 .1 .4
@item Name @tab Encoding @tab Decoding @tab Comments
@item 8SVX audio @tab @tab X
@item AAC+ @tab E @tab X
@tab encoding supported through external library libaacplus
@item AAC @tab E @tab X
@tab encoding supported through external library libfaac and libvo-aacenc
@item AC-3 @tab IX @tab X
@item ADPCM 4X Movie @tab @tab X
@item ADPCM CDROM XA @tab @tab X
@item ADPCM Creative Technology @tab @tab X
@tab 16 -> 4, 8 -> 4, 8 -> 3, 8 -> 2
@item ADPCM Electronic Arts @tab @tab X
@tab Used in various EA titles.
@item ADPCM Electronic Arts Maxis CDROM XS @tab @tab X
@tab Used in Sim City 3000.
@item ADPCM Electronic Arts R1 @tab @tab X
@item ADPCM Electronic Arts R2 @tab @tab X
@item ADPCM Electronic Arts R3 @tab @tab X
@item ADPCM Electronic Arts XAS @tab @tab X
@item ADPCM G.722 @tab X @tab X
@item ADPCM G.726 @tab X @tab X
@item ADPCM IMA AMV @tab @tab X
@tab Used in AMV files
@item ADPCM IMA Electronic Arts EACS @tab @tab X
@item ADPCM IMA Electronic Arts SEAD @tab @tab X
@item ADPCM IMA Funcom @tab @tab X
@item ADPCM IMA QuickTime @tab X @tab X
@item ADPCM IMA Loki SDL MJPEG @tab @tab X
@item ADPCM IMA WAV @tab X @tab X
@item ADPCM IMA Westwood @tab @tab X
@item ADPCM ISS IMA @tab @tab X
@tab Used in FunCom games.
@item ADPCM IMA Duck DK3 @tab @tab X
@tab Used in some Sega Saturn console games.
@item ADPCM IMA Duck DK4 @tab @tab X
@tab Used in some Sega Saturn console games.
@item ADPCM Microsoft @tab X @tab X
@item ADPCM MS IMA @tab X @tab X
@item ADPCM Nintendo Gamecube THP @tab @tab X
@item ADPCM QT IMA @tab X @tab X
@item ADPCM SEGA CRI ADX @tab X @tab X
@tab Used in Sega Dreamcast games.
@item ADPCM Shockwave Flash @tab X @tab X
@item ADPCM SMJPEG IMA @tab @tab X
@tab Used in certain Loki game ports.
@item ADPCM Sound Blaster Pro 2-bit @tab @tab X
@item ADPCM Sound Blaster Pro 2.6-bit @tab @tab X
@item ADPCM Sound Blaster Pro 4-bit @tab @tab X
@item ADPCM Westwood Studios IMA @tab @tab X
@tab Used in Westwood Studios games like Command and Conquer.
@item ADPCM Yamaha @tab X @tab X
@item AMR-NB @tab E @tab X
@tab encoding supported through external library libopencore-amrnb
@item AMR-WB @tab E @tab X
@tab encoding supported through external library libvo-amrwbenc
@item Apple lossless audio @tab X @tab X
@tab QuickTime fourcc 'alac'
@item Atrac 1 @tab @tab X
@item Atrac 3 @tab @tab X
@item Bink Audio @tab @tab X
@tab Used in Bink and Smacker files in many games.
@item CELT @tab @tab E
@tab decoding supported through external library libcelt
@item Delphine Software International CIN audio @tab @tab X
@tab Codec used in Delphine Software International games.
@item Discworld II BMV Audio @tab @tab X
@item COOK @tab @tab X
@tab All versions except 5.1 are supported.
@item DCA (DTS Coherent Acoustics) @tab X @tab X
@item DPCM id RoQ @tab X @tab X
@tab Used in Quake III, Jedi Knight 2 and other computer games.
@item DPCM Interplay @tab @tab X
@tab Used in various Interplay computer games.
@item DPCM Sierra Online @tab @tab X
@tab Used in Sierra Online game audio files.
@item DPCM Sol @tab @tab X
@item DPCM Xan @tab @tab X
@tab Used in Origin's Wing Commander IV AVI files.
@item DSP Group TrueSpeech @tab @tab X
@item DV audio @tab @tab X
@item Enhanced AC-3 @tab X @tab X
@item FLAC (Free Lossless Audio Codec) @tab X @tab IX
@item G.723.1 @tab X @tab X
@item G.729 @tab @tab X
@item GSM @tab E @tab X
@tab encoding supported through external library libgsm
@item GSM Microsoft variant @tab E @tab X
@tab encoding supported through external library libgsm
@item IMC (Intel Music Coder) @tab @tab X
@item MACE (Macintosh Audio Compression/Expansion) 3:1 @tab @tab X
@item MACE (Macintosh Audio Compression/Expansion) 6:1 @tab @tab X
@item MLP (Meridian Lossless Packing) @tab @tab X
@tab Used in DVD-Audio discs.
@item Monkey's Audio @tab @tab X
@tab Only versions 3.97-3.99 are supported.
@item MP1 (MPEG audio layer 1) @tab @tab IX
@item MP2 (MPEG audio layer 2) @tab IX @tab IX
@item MP3 (MPEG audio layer 3) @tab E @tab IX
@tab encoding supported through external library LAME, ADU MP3 and MP3onMP4 also supported
@item MPEG-4 Audio Lossless Coding (ALS) @tab @tab X
@item Musepack SV7 @tab @tab X
@item Musepack SV8 @tab @tab X
@item Nellymoser Asao @tab X @tab X
@item PCM A-law @tab X @tab X
@item PCM mu-law @tab X @tab X
@item PCM 16-bit little-endian planar @tab @tab X
@item PCM 32-bit floating point big-endian @tab X @tab X
@item PCM 32-bit floating point little-endian @tab X @tab X
@item PCM 64-bit floating point big-endian @tab X @tab X
@item PCM 64-bit floating point little-endian @tab X @tab X
@item PCM D-Cinema audio signed 24-bit @tab X @tab X
@item PCM signed 8-bit @tab X @tab X
@item PCM signed 16-bit big-endian @tab X @tab X
@item PCM signed 16-bit little-endian @tab X @tab X
@item PCM signed 24-bit big-endian @tab X @tab X
@item PCM signed 24-bit little-endian @tab X @tab X
@item PCM signed 32-bit big-endian @tab X @tab X
@item PCM signed 32-bit little-endian @tab X @tab X
@item PCM signed 16/20/24-bit big-endian in MPEG-TS @tab @tab X
@item PCM unsigned 8-bit @tab X @tab X
@item PCM unsigned 16-bit big-endian @tab X @tab X
@item PCM unsigned 16-bit little-endian @tab X @tab X
@item PCM unsigned 24-bit big-endian @tab X @tab X
@item PCM unsigned 24-bit little-endian @tab X @tab X
@item PCM unsigned 32-bit big-endian @tab X @tab X
@item PCM unsigned 32-bit little-endian @tab X @tab X
@item PCM Zork @tab @tab X
@item QCELP / PureVoice @tab @tab X
@item QDesign Music Codec 2 @tab @tab X
@tab There are still some distortions.
@item RealAudio 1.0 (14.4K) @tab X @tab X
@tab Real 14400 bit/s codec
@item RealAudio 2.0 (28.8K) @tab @tab X
@tab Real 28800 bit/s codec
@item RealAudio 3.0 (dnet) @tab IX @tab X
@tab Real low bitrate AC-3 codec
@item RealAudio SIPR / ACELP.NET @tab @tab X
@item Shorten @tab @tab X
@item Sierra VMD audio @tab @tab X
@tab Used in Sierra VMD files.
@item Smacker audio @tab @tab X
@item SMPTE 302M AES3 audio @tab @tab X
@item Sonic @tab X @tab X
@tab experimental codec
@item Sonic lossless @tab X @tab X
@tab experimental codec
@item Speex @tab E @tab E
@tab supported through external library libspeex
@item True Audio (TTA) @tab @tab X
@item TrueHD @tab @tab X
@tab Used in HD-DVD and Blu-Ray discs.
@item TwinVQ (VQF flavor) @tab @tab X
@item Vorbis @tab E @tab X
@tab A native but very primitive encoder exists.
@item WavPack @tab @tab X
@item Westwood Audio (SND1) @tab @tab X
@item Windows Media Audio 1 @tab X @tab X
@item Windows Media Audio 2 @tab X @tab X
@item Windows Media Audio Pro @tab @tab X
@item Windows Media Audio Voice @tab @tab X
@end multitable
@code{X} means that encoding (resp. decoding) is supported.
@code{E} means that support is provided through an external library.
@code{I} means that an integer-only version is available, too (ensures high
performance on systems without hardware floating point support).
@section Subtitle Formats
@multitable @columnfractions .4 .1 .1 .1 .1
@item Name @tab Muxing @tab Demuxing @tab Encoding @tab Decoding
@item SSA/ASS @tab X @tab X @tab X @tab X
@item DVB @tab X @tab X @tab X @tab X
@item DVD @tab X @tab X @tab X @tab X
@item MicroDVD @tab X @tab X @tab @tab
@item PGS @tab @tab @tab @tab X
@item SubRip (SRT) @tab X @tab X @tab X @tab X
@item XSUB @tab @tab @tab X @tab X
@end multitable
@code{X} means that the feature is supported.
@section Network Protocols
@multitable @columnfractions .4 .1
@item Name @tab Support
@item Apple HTTP Live Streaming @tab X
@item file @tab X
@item Gopher @tab X
@item HTTP @tab X
@item MMS @tab X
@item pipe @tab X
@item RTP @tab X
@item TCP @tab X
@item UDP @tab X
@end multitable
@code{X} means that the protocol is supported.
@section Input/Output Devices
@multitable @columnfractions .4 .1 .1
@item Name @tab Input @tab Output
@item ALSA @tab X @tab X
@item BKTR @tab X @tab
@item DV1394 @tab X @tab
@item JACK @tab X @tab
@item LIBDC1394 @tab X @tab
@item OSS @tab X @tab X
@item Pulseaudio @tab X @tab
@item Video4Linux @tab X @tab
@item Video4Linux2 @tab X @tab
@item VfW capture @tab X @tab
@item X11 grabbing @tab X @tab
@end multitable
@code{X} means that input/output is supported.
@section Timecode
@multitable @columnfractions .4 .1 .1
@item Codec/format @tab Read @tab Write
@item DV @tab X @tab X
@item GXF @tab X @tab X
@item MOV @tab X @tab
@item MPEG1/2 @tab X @tab X
@item MXF @tab @tab X
@end multitable
@bye
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\input texinfo @c -*- texinfo -*-
@settitle Using git to develop FFmpeg
@titlepage
@center @titlefont{Using git to develop FFmpeg}
@end titlepage
@top
@contents
@chapter Introduction
This document aims in giving some quick references on a set of useful git
commands. You should always use the extensive and detailed documentation
provided directly by git:
@example
git --help
man git
@end example
shows you the available subcommands,
@example
git <command> --help
man git-<command>
@end example
shows information about the subcommand <command>.
Additional information could be found on the
@url{http://gitref.org, Git Reference} website
For more information about the Git project, visit the
@url{http://git-scm.com/, Git website}
Consult these resources whenever you have problems, they are quite exhaustive.
What follows now is a basic introduction to Git and some FFmpeg-specific
guidelines to ease the contribution to the project
@chapter Basics Usage
@section Get GIT
You can get git from @url{http://git-scm.com/}
Most distribution and operating system provide a package for it.
@section Cloning the source tree
@example
git clone git://source.ffmpeg.org/ffmpeg <target>
@end example
This will put the FFmpeg sources into the directory @var{<target>}.
@example
git clone git@@source.ffmpeg.org:ffmpeg <target>
@end example
This will put the FFmpeg sources into the directory @var{<target>} and let
you push back your changes to the remote repository.
@section Updating the source tree to the latest revision
@example
git pull (--rebase)
@end example
pulls in the latest changes from the tracked branch. The tracked branch
can be remote. By default the master branch tracks the branch master in
the remote origin.
@float IMPORTANT
@command{--rebase} (see below) is recommended.
@end float
@section Rebasing your local branches
@example
git pull --rebase
@end example
fetches the changes from the main repository and replays your local commits
over it. This is required to keep all your local changes at the top of
FFmpeg's master tree. The master tree will reject pushes with merge commits.
@section Adding/removing files/directories
@example
git add [-A] <filename/dirname>
git rm [-r] <filename/dirname>
@end example
GIT needs to get notified of all changes you make to your working
directory that makes files appear or disappear.
Line moves across files are automatically tracked.
@section Showing modifications
@example
git diff <filename(s)>
@end example
will show all local modifications in your working directory as unified diff.
@section Inspecting the changelog
@example
git log <filename(s)>
@end example
You may also use the graphical tools like gitview or gitk or the web
interface available at http://source.ffmpeg.org/
@section Checking source tree status
@example
git status
@end example
detects all the changes you made and lists what actions will be taken in case
of a commit (additions, modifications, deletions, etc.).
@section Committing
@example
git diff --check
@end example
to double check your changes before committing them to avoid trouble later
on. All experienced developers do this on each and every commit, no matter
how small.
Every one of them has been saved from looking like a fool by this many times.
It's very easy for stray debug output or cosmetic modifications to slip in,
please avoid problems through this extra level of scrutiny.
For cosmetics-only commits you should get (almost) empty output from
@example
git diff -w -b <filename(s)>
@end example
Also check the output of
@example
git status
@end example
to make sure you don't have untracked files or deletions.
@example
git add [-i|-p|-A] <filenames/dirnames>
@end example
Make sure you have told git your name and email address
@example
git config --global user.name "My Name"
git config --global user.email my@@email.invalid
@end example
Use @var{--global} to set the global configuration for all your git checkouts.
Git will select the changes to the files for commit. Optionally you can use
the interactive or the patch mode to select hunk by hunk what should be
added to the commit.
@example
git commit
@end example
Git will commit the selected changes to your current local branch.
You will be prompted for a log message in an editor, which is either
set in your personal configuration file through
@example
git config --global core.editor
@end example
or set by one of the following environment variables:
@var{GIT_EDITOR}, @var{VISUAL} or @var{EDITOR}.
Log messages should be concise but descriptive. Explain why you made a change,
what you did will be obvious from the changes themselves most of the time.
Saying just "bug fix" or "10l" is bad. Remember that people of varying skill
levels look at and educate themselves while reading through your code. Don't
include filenames in log messages, Git provides that information.
Possibly make the commit message have a terse, descriptive first line, an
empty line and then a full description. The first line will be used to name
the patch by git format-patch.
@section Preparing a patchset
@example
git format-patch <commit> [-o directory]
@end example
will generate a set of patches for each commit between @var{<commit>} and
current @var{HEAD}. E.g.
@example
git format-patch origin/master
@end example
will generate patches for all commits on current branch which are not
present in upstream.
A useful shortcut is also
@example
git format-patch -n
@end example
which will generate patches from last @var{n} commits.
By default the patches are created in the current directory.
@section Sending patches for review
@example
git send-email <commit list|directory>
@end example
will send the patches created by @command{git format-patch} or directly
generates them. All the email fields can be configured in the global/local
configuration or overridden by command line.
Note that this tool must often be installed separately (e.g. @var{git-email}
package on Debian-based distros).
@section Renaming/moving/copying files or contents of files
Git automatically tracks such changes, making those normal commits.
@example
mv/cp path/file otherpath/otherfile
git add [-A] .
git commit
@end example
@chapter FFmpeg specific
@section Reverting broken commits
@example
git reset <commit>
@end example
@command{git reset} will uncommit the changes till @var{<commit>} rewriting
the current branch history.
@example
git commit --amend
@end example
allows to amend the last commit details quickly.
@example
git rebase -i origin/master
@end example
will replay local commits over the main repository allowing to edit, merge
or remove some of them in the process.
@float NOTE
@command{git reset}, @command{git commit --amend} and @command{git rebase}
rewrite history, so you should use them ONLY on your local or topic branches.
The main repository will reject those changes.
@end float
@example
git revert <commit>
@end example
@command{git revert} will generate a revert commit. This will not make the
faulty commit disappear from the history.
@section Pushing changes to remote trees
@example
git push
@end example
Will push the changes to the default remote (@var{origin}).
Git will prevent you from pushing changes if the local and remote trees are
out of sync. Refer to and to sync the local tree.
@example
git remote add <name> <url>
@end example
Will add additional remote with a name reference, it is useful if you want
to push your local branch for review on a remote host.
@example
git push <remote> <refspec>
@end example
Will push the changes to the @var{<remote>} repository.
Omitting @var{<refspec>} makes @command{git push} update all the remote
branches matching the local ones.
@section Finding a specific svn revision
Since version 1.7.1 git supports @var{:/foo} syntax for specifying commits
based on a regular expression. see man gitrevisions
@example
git show :/'as revision 23456'
@end example
will show the svn changeset @var{r23456}. With older git versions searching in
the @command{git log} output is the easiest option (especially if a pager with
search capabilities is used).
This commit can be checked out with
@example
git checkout -b svn_23456 :/'as revision 23456'
@end example
or for git < 1.7.1 with
@example
git checkout -b svn_23456 $SHA1
@end example
where @var{$SHA1} is the commit hash from the @command{git log} output.
@chapter Server Issues
Contact the project admins @email{root@@ffmpeg.org} if you have technical
problems with the GIT server.
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About Git write access:
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
Before everything else, you should know how to use GIT properly.
Luckily Git comes with excellent documentation.
git --help
man git
shows you the available subcommands,
git <command> --help
man git-<command>
shows information about the subcommand <command>.
The most comprehensive manual is the website Git Reference
http://gitref.org/
For more information about the Git project, visit
http://git-scm.com/
Consult these resources whenever you have problems, they are quite exhaustive.
You do not need a special username or password.
All you need is to provide a ssh public key to the Git server admin.
What follows now is a basic introduction to Git and some FFmpeg-specific
guidelines. Read it at least once, if you are granted commit privileges to the
FFmpeg project you are expected to be familiar with these rules.
I. BASICS:
==========
0. Get GIT:
Most distributions have a git package, if not
You can get git from http://git-scm.com/
1. Cloning the source tree:
git clone git://source.ffmpeg.org/ffmpeg <target>
This will put the FFmpeg sources into the directory <target>.
git clone git@source.ffmpeg.org:ffmpeg <target>
This will put the FFmpeg sources into the directory <target> and let
you push back your changes to the remote repository.
2. Updating the source tree to the latest revision:
git pull (--ff-only)
pulls in the latest changes from the tracked branch. The tracked branch
can be remote. By default the master branch tracks the branch master in
the remote origin.
Caveat: Since merge commits are forbidden at least for the initial
months of git --ff-only or --rebase (see below) are recommended.
--ff-only will fail and not create merge commits if your branch
has diverged (has a different history) from the tracked branch.
2.a Rebasing your local branches:
git pull --rebase
fetches the changes from the main repository and replays your local commits
over it. This is required to keep all your local changes at the top of
FFmpeg's master tree. The master tree will reject pushes with merge commits.
3. Adding/removing files/directories:
git add [-A] <filename/dirname>
git rm [-r] <filename/dirname>
GIT needs to get notified of all changes you make to your working
directory that makes files appear or disappear.
Line moves across files are automatically tracked.
4. Showing modifications:
git diff <filename(s)>
will show all local modifications in your working directory as unified diff.
5. Inspecting the changelog:
git log <filename(s)>
You may also use the graphical tools like gitview or gitk or the web
interface available at http://source.ffmpeg.org
6. Checking source tree status:
git status
detects all the changes you made and lists what actions will be taken in case
of a commit (additions, modifications, deletions, etc.).
7. Committing:
git diff --check
to double check your changes before committing them to avoid trouble later
on. All experienced developers do this on each and every commit, no matter
how small.
Every one of them has been saved from looking like a fool by this many times.
It's very easy for stray debug output or cosmetic modifications to slip in,
please avoid problems through this extra level of scrutiny.
For cosmetics-only commits you should get (almost) empty output from
git diff -w -b <filename(s)>
Also check the output of
git status
to make sure you don't have untracked files or deletions.
git add [-i|-p|-A] <filenames/dirnames>
Make sure you have told git your name and email address, e.g. by running
git config --global user.name "My Name"
git config --global user.email my@email.invalid
(--global to set the global configuration for all your git checkouts).
Git will select the changes to the files for commit. Optionally you can use
the interactive or the patch mode to select hunk by hunk what should be
added to the commit.
git commit
Git will commit the selected changes to your current local branch.
You will be prompted for a log message in an editor, which is either
set in your personal configuration file through
git config core.editor
or set by one of the following environment variables:
GIT_EDITOR, VISUAL or EDITOR.
Log messages should be concise but descriptive. Explain why you made a change,
what you did will be obvious from the changes themselves most of the time.
Saying just "bug fix" or "10l" is bad. Remember that people of varying skill
levels look at and educate themselves while reading through your code. Don't
include filenames in log messages, Git provides that information.
Possibly make the commit message have a terse, descriptive first line, an
empty line and then a full description. The first line will be used to name
the patch by git format-patch.
8. Renaming/moving/copying files or contents of files:
Git automatically tracks such changes, making those normal commits.
mv/cp path/file otherpath/otherfile
git add [-A] .
git commit
Do not move, rename or copy files of which you are not the maintainer without
discussing it on the mailing list first!
9. Reverting broken commits
git revert <commit>
git revert will generate a revert commit. This will not make the faulty
commit disappear from the history.
git reset <commit>
git reset will uncommit the changes till <commit> rewriting the current
branch history.
git commit --amend
allows to amend the last commit details quickly.
git rebase -i origin/master
will replay local commits over the main repository allowing to edit,
merge or remove some of them in the process.
Note that the reset, commit --amend and rebase rewrite history, so you
should use them ONLY on your local or topic branches.
The main repository will reject those changes.
10. Preparing a patchset.
git format-patch <commit> [-o directory]
will generate a set of patches for each commit between <commit> and
current HEAD. E.g.
git format-patch origin/master
will generate patches for all commits on current branch which are not
present in upstream.
A useful shortcut is also
git format-patch -n
which will generate patches from last n commits.
By default the patches are created in the current directory.
11. Sending patches for review
git send-email <commit list|directory>
will send the patches created by git format-patch or directly generates
them. All the email fields can be configured in the global/local
configuration or overridden by command line.
Note that this tool must often be installed separately (e.g. git-email
package on Debian-based distros).
12. Pushing changes to remote trees
git push
Will push the changes to the default remote (origin).
Git will prevent you from pushing changes if the local and remote trees are
out of sync. Refer to 2 and 2.a to sync the local tree.
git remote add <name> <url>
Will add additional remote with a name reference, it is useful if you want
to push your local branch for review on a remote host.
git push <remote> <refspec>
Will push the changes to the remote repository. Omitting refspec makes git
push update all the remote branches matching the local ones.
13. Finding a specific svn revision
Since version 1.7.1 git supports ':/foo' syntax for specifying commits
based on a regular expression. see man gitrevisions
git show :/'as revision 23456'
will show the svn changeset r23456. With older git versions searching in
the git log output is the easiest option (especially if a pager with
search capabilities is used).
This commit can be checked out with
git checkout -b svn_23456 :/'as revision 23456'
or for git < 1.7.1 with
git checkout -b svn_23456 $SHA1
where $SHA1 is the commit SHA1 from the 'git log' output.
Contact the project admins <root at ffmpeg dot org> if you have technical
problems with the GIT server.
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@chapter Input Devices
@c man begin INPUT DEVICES
Input devices are configured elements in FFmpeg which allow to access
the data coming from a multimedia device attached to your system.
When you configure your FFmpeg build, all the supported input devices
are enabled by default. You can list all available ones using the
configure option "--list-indevs".
You can disable all the input devices using the configure option
"--disable-indevs", and selectively enable an input device using the
option "--enable-indev=@var{INDEV}", or you can disable a particular
input device using the option "--disable-indev=@var{INDEV}".
The option "-formats" of the ff* tools will display the list of
supported input devices (amongst the demuxers).
A description of the currently available input devices follows.
@section alsa
ALSA (Advanced Linux Sound Architecture) input device.
To enable this input device during configuration you need libasound
installed on your system.
This device allows capturing from an ALSA device. The name of the
device to capture has to be an ALSA card identifier.
An ALSA identifier has the syntax:
@example
hw:@var{CARD}[,@var{DEV}[,@var{SUBDEV}]]
@end example
where the @var{DEV} and @var{SUBDEV} components are optional.
The three arguments (in order: @var{CARD},@var{DEV},@var{SUBDEV})
specify card number or identifier, device number and subdevice number
(-1 means any).
To see the list of cards currently recognized by your system check the
files @file{/proc/asound/cards} and @file{/proc/asound/devices}.
For example to capture with @command{ffmpeg} from an ALSA device with
card id 0, you may run the command:
@example
ffmpeg -f alsa -i hw:0 alsaout.wav
@end example
For more information see:
@url{http://www.alsa-project.org/alsa-doc/alsa-lib/pcm.html}
@section bktr
BSD video input device.
@section dshow
Windows DirectShow input device.
DirectShow support is enabled when FFmpeg is built with mingw-w64.
Currently only audio and video devices are supported.
Multiple devices may be opened as separate inputs, but they may also be
opened on the same input, which should improve synchronism between them.
The input name should be in the format:
@example
@var{TYPE}=@var{NAME}[:@var{TYPE}=@var{NAME}]
@end example
where @var{TYPE} can be either @var{audio} or @var{video},
and @var{NAME} is the device's name.
@subsection Options
If no options are specified, the device's defaults are used.
If the device does not support the requested options, it will
fail to open.
@table @option
@item video_size
Set the video size in the captured video.
@item framerate
Set the framerate in the captured video.
@item sample_rate
Set the sample rate (in Hz) of the captured audio.
@item sample_size
Set the sample size (in bits) of the captured audio.
@item channels
Set the number of channels in the captured audio.
@item list_devices
If set to @option{true}, print a list of devices and exit.
@item list_options
If set to @option{true}, print a list of selected device's options
and exit.
@item video_device_number
Set video device number for devices with same name (starts at 0,
defaults to 0).
@item audio_device_number
Set audio device number for devices with same name (starts at 0,
defaults to 0).
@end table
@subsection Examples
@itemize
@item
Print the list of DirectShow supported devices and exit:
@example
$ ffmpeg -list_devices true -f dshow -i dummy
@end example
@item
Open video device @var{Camera}:
@example
$ ffmpeg -f dshow -i video="Camera"
@end example
@item
Open second video device with name @var{Camera}:
@example
$ ffmpeg -f dshow -video_device_number 1 -i video="Camera"
@end example
@item
Open video device @var{Camera} and audio device @var{Microphone}:
@example
$ ffmpeg -f dshow -i video="Camera":audio="Microphone"
@end example
@item
Print the list of supported options in selected device and exit:
@example
$ ffmpeg -list_options true -f dshow -i video="Camera"
@end example
@end itemize
@section dv1394
Linux DV 1394 input device.
@section fbdev
Linux framebuffer input device.
The Linux framebuffer is a graphic hardware-independent abstraction
layer to show graphics on a computer monitor, typically on the
console. It is accessed through a file device node, usually
@file{/dev/fb0}.
For more detailed information read the file
Documentation/fb/framebuffer.txt included in the Linux source tree.
To record from the framebuffer device @file{/dev/fb0} with
@command{ffmpeg}:
@example
ffmpeg -f fbdev -r 10 -i /dev/fb0 out.avi
@end example
You can take a single screenshot image with the command:
@example
ffmpeg -f fbdev -frames:v 1 -r 1 -i /dev/fb0 screenshot.jpeg
@end example
See also @url{http://linux-fbdev.sourceforge.net/}, and fbset(1).
@section jack
JACK input device.
To enable this input device during configuration you need libjack
installed on your system.
A JACK input device creates one or more JACK writable clients, one for
each audio channel, with name @var{client_name}:input_@var{N}, where
@var{client_name} is the name provided by the application, and @var{N}
is a number which identifies the channel.
Each writable client will send the acquired data to the FFmpeg input
device.
Once you have created one or more JACK readable clients, you need to
connect them to one or more JACK writable clients.
To connect or disconnect JACK clients you can use the @command{jack_connect}
and @command{jack_disconnect} programs, or do it through a graphical interface,
for example with @command{qjackctl}.
To list the JACK clients and their properties you can invoke the command
@command{jack_lsp}.
Follows an example which shows how to capture a JACK readable client
with @command{ffmpeg}.
@example
# Create a JACK writable client with name "ffmpeg".
$ ffmpeg -f jack -i ffmpeg -y out.wav
# Start the sample jack_metro readable client.
$ jack_metro -b 120 -d 0.2 -f 4000
# List the current JACK clients.
$ jack_lsp -c
system:capture_1
system:capture_2
system:playback_1
system:playback_2
ffmpeg:input_1
metro:120_bpm
# Connect metro to the ffmpeg writable client.
$ jack_connect metro:120_bpm ffmpeg:input_1
@end example
For more information read:
@url{http://jackaudio.org/}
@section lavfi
Libavfilter input virtual device.
This input device reads data from the open output pads of a libavfilter
filtergraph.
For each filtergraph open output, the input device will create a
corresponding stream which is mapped to the generated output. Currently
only video data is supported. The filtergraph is specified through the
option @option{graph}.
@subsection Options
@table @option
@item graph
Specify the filtergraph to use as input. Each video open output must be
labelled by a unique string of the form "out@var{N}", where @var{N} is a
number starting from 0 corresponding to the mapped input stream
generated by the device.
The first unlabelled output is automatically assigned to the "out0"
label, but all the others need to be specified explicitly.
If not specified defaults to the filename specified for the input
device.
@end table
@subsection Examples
@itemize
@item
Create a color video stream and play it back with @command{ffplay}:
@example
ffplay -f lavfi -graph "color=pink [out0]" dummy
@end example
@item
As the previous example, but use filename for specifying the graph
description, and omit the "out0" label:
@example
ffplay -f lavfi color=pink
@end example
@item
Create three different video test filtered sources and play them:
@example
ffplay -f lavfi -graph "testsrc [out0]; testsrc,hflip [out1]; testsrc,negate [out2]" test3
@end example
@item
Read an audio stream from a file using the amovie source and play it
back with @command{ffplay}:
@example
ffplay -f lavfi "amovie=test.wav"
@end example
@item
Read an audio stream and a video stream and play it back with
@command{ffplay}:
@example
ffplay -f lavfi "movie=test.avi[out0];amovie=test.wav[out1]"
@end example
@end itemize
@section libdc1394
IIDC1394 input device, based on libdc1394 and libraw1394.
@section openal
The OpenAL input device provides audio capture on all systems with a
working OpenAL 1.1 implementation.
To enable this input device during configuration, you need OpenAL
headers and libraries installed on your system, and need to configure
FFmpeg with @code{--enable-openal}.
OpenAL headers and libraries should be provided as part of your OpenAL
implementation, or as an additional download (an SDK). Depending on your
installation you may need to specify additional flags via the
@code{--extra-cflags} and @code{--extra-ldflags} for allowing the build
system to locate the OpenAL headers and libraries.
An incomplete list of OpenAL implementations follows:
@table @strong
@item Creative
The official Windows implementation, providing hardware acceleration
with supported devices and software fallback.
See @url{http://openal.org/}.
@item OpenAL Soft
Portable, open source (LGPL) software implementation. Includes
backends for the most common sound APIs on the Windows, Linux,
Solaris, and BSD operating systems.
See @url{http://kcat.strangesoft.net/openal.html}.
@item Apple
OpenAL is part of Core Audio, the official Mac OS X Audio interface.
See @url{http://developer.apple.com/technologies/mac/audio-and-video.html}
@end table
This device allows to capture from an audio input device handled
through OpenAL.
You need to specify the name of the device to capture in the provided
filename. If the empty string is provided, the device will
automatically select the default device. You can get the list of the
supported devices by using the option @var{list_devices}.
@subsection Options
@table @option
@item channels
Set the number of channels in the captured audio. Only the values
@option{1} (monaural) and @option{2} (stereo) are currently supported.
Defaults to @option{2}.
@item sample_size
Set the sample size (in bits) of the captured audio. Only the values
@option{8} and @option{16} are currently supported. Defaults to
@option{16}.
@item sample_rate
Set the sample rate (in Hz) of the captured audio.
Defaults to @option{44.1k}.
@item list_devices
If set to @option{true}, print a list of devices and exit.
Defaults to @option{false}.
@end table
@subsection Examples
Print the list of OpenAL supported devices and exit:
@example
$ ffmpeg -list_devices true -f openal -i dummy out.ogg
@end example
Capture from the OpenAL device @file{DR-BT101 via PulseAudio}:
@example
$ ffmpeg -f openal -i 'DR-BT101 via PulseAudio' out.ogg
@end example
Capture from the default device (note the empty string '' as filename):
@example
$ ffmpeg -f openal -i '' out.ogg
@end example
Capture from two devices simultaneously, writing to two different files,
within the same @command{ffmpeg} command:
@example
$ ffmpeg -f openal -i 'DR-BT101 via PulseAudio' out1.ogg -f openal -i 'ALSA Default' out2.ogg
@end example
Note: not all OpenAL implementations support multiple simultaneous capture -
try the latest OpenAL Soft if the above does not work.
@section oss
Open Sound System input device.
The filename to provide to the input device is the device node
representing the OSS input device, and is usually set to
@file{/dev/dsp}.
For example to grab from @file{/dev/dsp} using @command{ffmpeg} use the
command:
@example
ffmpeg -f oss -i /dev/dsp /tmp/oss.wav
@end example
For more information about OSS see:
@url{http://manuals.opensound.com/usersguide/dsp.html}
@section pulse
pulseaudio input device.
To enable this input device during configuration you need libpulse-simple
installed in your system.
The filename to provide to the input device is a source device or the
string "default"
To list the pulse source devices and their properties you can invoke
the command @command{pactl list sources}.
@example
ffmpeg -f pulse -i default /tmp/pulse.wav
@end example
@subsection @var{server} AVOption
The syntax is:
@example
-server @var{server name}
@end example
Connects to a specific server.
@subsection @var{name} AVOption
The syntax is:
@example
-name @var{application name}
@end example
Specify the application name pulse will use when showing active clients,
by default it is the LIBAVFORMAT_IDENT string
@subsection @var{stream_name} AVOption
The syntax is:
@example
-stream_name @var{stream name}
@end example
Specify the stream name pulse will use when showing active streams,
by default it is "record"
@subsection @var{sample_rate} AVOption
The syntax is:
@example
-sample_rate @var{samplerate}
@end example
Specify the samplerate in Hz, by default 48kHz is used.
@subsection @var{channels} AVOption
The syntax is:
@example
-channels @var{N}
@end example
Specify the channels in use, by default 2 (stereo) is set.
@subsection @var{frame_size} AVOption
The syntax is:
@example
-frame_size @var{bytes}
@end example
Specify the number of byte per frame, by default it is set to 1024.
@subsection @var{fragment_size} AVOption
The syntax is:
@example
-fragment_size @var{bytes}
@end example
Specify the minimal buffering fragment in pulseaudio, it will affect the
audio latency. By default it is unset.
@section sndio
sndio input device.
To enable this input device during configuration you need libsndio
installed on your system.
The filename to provide to the input device is the device node
representing the sndio input device, and is usually set to
@file{/dev/audio0}.
For example to grab from @file{/dev/audio0} using @command{ffmpeg} use the
command:
@example
ffmpeg -f sndio -i /dev/audio0 /tmp/oss.wav
@end example
@section video4linux and video4linux2
Video4Linux and Video4Linux2 input video devices.
The name of the device to grab is a file device node, usually Linux
systems tend to automatically create such nodes when the device
(e.g. an USB webcam) is plugged into the system, and has a name of the
kind @file{/dev/video@var{N}}, where @var{N} is a number associated to
the device.
Video4Linux and Video4Linux2 devices only support a limited set of
@var{width}x@var{height} sizes and framerates. You can check which are
supported for example with the command @command{dov4l} for Video4Linux
devices and using @command{-list_formats all} for Video4Linux2 devices.
If the size for the device is set to 0x0, the input device will
try to auto-detect the size to use.
Only for the video4linux2 device, if the frame rate is set to 0/0 the
input device will use the frame rate value already set in the driver.
Video4Linux support is deprecated since Linux 2.6.30, and will be
dropped in later versions.
Note that if FFmpeg is build with v4l-utils support ("--enable-libv4l2"
option), it will always be used.
Follow some usage examples of the video4linux devices with the ff*
tools.
@example
# Grab and show the input of a video4linux device, frame rate is set
# to the default of 25/1.
ffplay -s 320x240 -f video4linux /dev/video0
# Grab and show the input of a video4linux2 device, auto-adjust size.
ffplay -f video4linux2 /dev/video0
# Grab and record the input of a video4linux2 device, auto-adjust size,
# frame rate value defaults to 0/0 so it is read from the video4linux2
# driver.
ffmpeg -f video4linux2 -i /dev/video0 out.mpeg
@end example
"v4l" and "v4l2" can be used as aliases for the respective "video4linux" and
"video4linux2".
@section vfwcap
VfW (Video for Windows) capture input device.
The filename passed as input is the capture driver number, ranging from
0 to 9. You may use "list" as filename to print a list of drivers. Any
other filename will be interpreted as device number 0.
@section x11grab
X11 video input device.
This device allows to capture a region of an X11 display.
The filename passed as input has the syntax:
@example
[@var{hostname}]:@var{display_number}.@var{screen_number}[+@var{x_offset},@var{y_offset}]
@end example
@var{hostname}:@var{display_number}.@var{screen_number} specifies the
X11 display name of the screen to grab from. @var{hostname} can be
omitted, and defaults to "localhost". The environment variable
@env{DISPLAY} contains the default display name.
@var{x_offset} and @var{y_offset} specify the offsets of the grabbed
area with respect to the top-left border of the X11 screen. They
default to 0.
Check the X11 documentation (e.g. man X) for more detailed information.
Use the @command{dpyinfo} program for getting basic information about the
properties of your X11 display (e.g. grep for "name" or "dimensions").
For example to grab from @file{:0.0} using @command{ffmpeg}:
@example
ffmpeg -f x11grab -r 25 -s cif -i :0.0 out.mpg
# Grab at position 10,20.
ffmpeg -f x11grab -r 25 -s cif -i :0.0+10,20 out.mpg
@end example
@subsection @var{follow_mouse} AVOption
The syntax is:
@example
-follow_mouse centered|@var{PIXELS}
@end example
When it is specified with "centered", the grabbing region follows the mouse
pointer and keeps the pointer at the center of region; otherwise, the region
follows only when the mouse pointer reaches within @var{PIXELS} (greater than
zero) to the edge of region.
For example:
@example
ffmpeg -f x11grab -follow_mouse centered -r 25 -s cif -i :0.0 out.mpg
# Follows only when the mouse pointer reaches within 100 pixels to edge
ffmpeg -f x11grab -follow_mouse 100 -r 25 -s cif -i :0.0 out.mpg
@end example
@subsection @var{show_region} AVOption
The syntax is:
@example
-show_region 1
@end example
If @var{show_region} AVOption is specified with @var{1}, then the grabbing
region will be indicated on screen. With this option, it's easy to know what is
being grabbed if only a portion of the screen is grabbed.
For example:
@example
ffmpeg -f x11grab -show_region 1 -r 25 -s cif -i :0.0+10,20 out.mpg
# With follow_mouse
ffmpeg -f x11grab -follow_mouse centered -show_region 1 -r 25 -s cif -i :0.0 out.mpg
@end example
@c man end INPUT DEVICES
-213
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FFmpeg's bug/patch/feature request tracker manual
=================================================
NOTE: This is a draft.
Overview:
---------
FFmpeg uses Trac for tracking issues, new issues and changes to
existing issues can be done through a web interface.
Issues can be different kinds of things we want to keep track of
but that do not belong into the source tree itself. This includes
bug reports, patches, feature requests and license violations. We
might add more items to this list in the future, so feel free to
propose a new `type of issue' on the ffmpeg-devel mailing list if
you feel it is worth tracking.
It is possible to subscribe to individual issues by adding yourself to the
Cc list or to subscribe to the ffmpeg-trac mailing list which receives
a mail for every change to every issue.
(the above does all work already after light testing)
The subscription URL for the ffmpeg-trac list is:
http(s)://ffmpeg.org/mailman/listinfo/ffmpeg-trac
The URL of the webinterface of the tracker is:
http(s)://ffmpeg.org/trac/ffmpeg
Type:
-----
bug / defect
An error, flaw, mistake, failure, or fault in FFmpeg or libav* that
prevents it from behaving as intended.
feature request / enhancement
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.
license violation
ticket to keep track of (L)GPL violations of ffmpeg by others
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.
Regressions also should be marked as important, regressions are bugs that
don't exist in a past revision or another branch.
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
Analyzed flag:
--------------
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.
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/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/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 substatus.
Component:
----------
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
ffprobe
issues in or related to ffprobe.c
ffserver
issues in or related to ffserver.c
build system
issues in or related to configure/Makefile
regression
bugs which were not present in a past revision
trac
issues related to our issue tracker
-92
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\input texinfo @c -*- texinfo -*-
@settitle Libavfilter Documentation
@titlepage
@center @titlefont{Libavfilter Documentation}
@end titlepage
@top
@contents
@chapter Introduction
Libavfilter is the filtering API of FFmpeg. It is the substitute of the
now deprecated 'vhooks' and started as a Google Summer of Code project.
Audio filtering integration into the main FFmpeg repository is a work in
progress, so audio API and ABI should not be considered stable yet.
@chapter Tutorial
In libavfilter, it is possible for filters to have multiple inputs and
multiple outputs.
To illustrate the sorts of things that are possible, we can
use a complex filter graph. For example, the following one:
@example
input --> split --> fifo -----------------------> overlay --> output
| ^
| |
+------> fifo --> crop --> vflip --------+
@end example
splits the stream in two streams, sends one stream through the crop filter
and the vflip filter before merging it back with the other stream by
overlaying it on top. You can use the following command to achieve this:
@example
ffmpeg -i input -vf "[in] split [T1], fifo, [T2] overlay=0:H/2 [out]; [T1] fifo, crop=iw:ih/2:0:ih/2, vflip [T2]" output
@end example
The result will be that in output the top half of the video is mirrored
onto the bottom half.
Video filters are loaded using the @var{-vf} option passed to
@command{ffmpeg} or to @command{ffplay}. Filters in the same linear
chain are separated by commas. In our example, @var{split, fifo,
overlay} are in one linear chain, and @var{fifo, crop, vflip} are in
another. The points where the linear chains join are labeled by names
enclosed in square brackets. In our example, that is @var{[T1]} and
@var{[T2]}. The magic labels @var{[in]} and @var{[out]} are the points
where video is input and output.
Some filters take in input a list of parameters: they are specified
after the filter name and an equal sign, and are separated each other
by a semicolon.
There exist so-called @var{source filters} that do not have a video
input, and we expect in the future some @var{sink filters} that will
not have video output.
@chapter graph2dot
The @file{graph2dot} program included in the FFmpeg @file{tools}
directory can be used to parse a filter graph description and issue a
corresponding textual representation in the dot language.
Invoke the command:
@example
graph2dot -h
@end example
to see how to use @file{graph2dot}.
You can then pass the dot description to the @file{dot} program (from
the graphviz suite of programs) and obtain a graphical representation
of the filter graph.
For example the sequence of commands:
@example
echo @var{GRAPH_DESCRIPTION} | \
tools/graph2dot -o graph.tmp && \
dot -Tpng graph.tmp -o graph.png && \
display graph.png
@end example
can be used to create and display an image representing the graph
described by the @var{GRAPH_DESCRIPTION} string.
@include filters.texi
@bye
-68
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@chapter Metadata
@c man begin METADATA
FFmpeg is able to dump metadata from media files into a simple UTF-8-encoded
INI-like text file and then load it back using the metadata muxer/demuxer.
The file format is as follows:
@enumerate
@item
A file consists of a header and a number of metadata tags divided into sections,
each on its own line.
@item
The header is a ';FFMETADATA' string, followed by a version number (now 1).
@item
Metadata tags are of the form 'key=value'
@item
Immediately after header follows global metadata
@item
After global metadata there may be sections with per-stream/per-chapter
metadata.
@item
A section starts with the section name in uppercase (i.e. STREAM or CHAPTER) in
brackets ('[', ']') and ends with next section or end of file.
@item
At the beginning of a chapter section there may be an optional timebase to be
used for start/end values. It must be in form 'TIMEBASE=num/den', where num and
den are integers. If the timebase is missing then start/end times are assumed to
be in milliseconds.
Next a chapter section must contain chapter start and end times in form
'START=num', 'END=num', where num is a positive integer.
@item
Empty lines and lines starting with ';' or '#' are ignored.
@item
Metadata keys or values containing special characters ('=', ';', '#', '\' and a
newline) must be escaped with a backslash '\'.
@item
Note that whitespace in metadata (e.g. foo = bar) is considered to be a part of
the tag (in the example above key is 'foo ', value is ' bar').
@end enumerate
A ffmetadata file might look like this:
@example
;FFMETADATA1
title=bike\\shed
;this is a comment
artist=FFmpeg troll team
[CHAPTER]
TIMEBASE=1/1000
START=0
#chapter ends at 0:01:00
END=60000
title=chapter \#1
[STREAM]
title=multi\
line
@end example
@c man end METADATA
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FFmpeg multithreading methods
==============================================
FFmpeg provides two methods for multithreading codecs.
Slice threading decodes multiple parts of a frame at the same time, using
AVCodecContext execute() and execute2().
Frame threading decodes multiple frames at the same time.
It accepts N future frames and delays decoded pictures by N-1 frames.
The later frames are decoded in separate threads while the user is
displaying the current one.
Restrictions on clients
==============================================
Slice threading -
* The client's draw_horiz_band() must be thread-safe according to the comment
in avcodec.h.
Frame threading -
* Restrictions with slice threading also apply.
* For best performance, the client should set thread_safe_callbacks if it
provides a thread-safe get_buffer() callback.
* There is one frame of delay added for every thread beyond the first one.
Clients must be able to handle this; the pkt_dts and pkt_pts fields in
AVFrame will work as usual.
Restrictions on codec implementations
==============================================
Slice threading -
None except that there must be something worth executing in parallel.
Frame threading -
* Codecs can only accept entire pictures per packet.
* Codecs similar to ffv1, whose streams don't reset across frames,
will not work because their bitstreams cannot be decoded in parallel.
* The contents of buffers must not be read before ff_thread_await_progress()
has been called on them. reget_buffer() and buffer age optimizations no longer work.
* The contents of buffers must not be written to after ff_thread_report_progress()
has been called on them. This includes draw_edges().
Porting codecs to frame threading
==============================================
Find all context variables that are needed by the next frame. Move all
code changing them, as well as code calling get_buffer(), up to before
the decode process starts. Call ff_thread_finish_setup() afterwards. If
some code can't be moved, have update_thread_context() run it in the next
thread.
If the codec allocates writable tables in its init(), add an init_thread_copy()
which re-allocates them for other threads.
Add CODEC_CAP_FRAME_THREADS to the codec capabilities. There will be very little
speed gain at this point but it should work.
Call ff_thread_report_progress() after some part of the current picture has decoded.
A good place to put this is where draw_horiz_band() is called - add this if it isn't
called anywhere, as it's useful too and the implementation is trivial when you're
doing this. Note that draw_edges() needs to be called before reporting progress.
Before accessing a reference frame or its MVs, call ff_thread_await_progress().
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@chapter Muxers
@c man begin MUXERS
Muxers are configured elements in FFmpeg which allow writing
multimedia streams to a particular type of file.
When you configure your FFmpeg build, all the supported muxers
are enabled by default. You can list all available muxers using the
configure option @code{--list-muxers}.
You can disable all the muxers with the configure option
@code{--disable-muxers} and selectively enable / disable single muxers
with the options @code{--enable-muxer=@var{MUXER}} /
@code{--disable-muxer=@var{MUXER}}.
The option @code{-formats} of the ff* tools will display the list of
enabled muxers.
A description of some of the currently available muxers follows.
@anchor{crc}
@section crc
CRC (Cyclic Redundancy Check) testing format.
This muxer computes and prints the Adler-32 CRC of all the input audio
and video frames. By default audio frames are converted to signed
16-bit raw audio and video frames to raw video before computing the
CRC.
The output of the muxer consists of a single line of the form:
CRC=0x@var{CRC}, where @var{CRC} is a hexadecimal number 0-padded to
8 digits containing the CRC for all the decoded input frames.
For example to compute the CRC of the input, and store it in the file
@file{out.crc}:
@example
ffmpeg -i INPUT -f crc out.crc
@end example
You can print the CRC to stdout with the command:
@example
ffmpeg -i INPUT -f crc -
@end example
You can select the output format of each frame with @command{ffmpeg} by
specifying the audio and video codec and format. For example to
compute the CRC of the input audio converted to PCM unsigned 8-bit
and the input video converted to MPEG-2 video, use the command:
@example
ffmpeg -i INPUT -c:a pcm_u8 -c:v mpeg2video -f crc -
@end example
See also the @ref{framecrc} muxer.
@anchor{framecrc}
@section framecrc
Per-frame CRC (Cyclic Redundancy Check) testing format.
This muxer computes and prints the Adler-32 CRC for each decoded audio
and video frame. By default audio frames are converted to signed
16-bit raw audio and video frames to raw video before computing the
CRC.
The output of the muxer consists of a line for each audio and video
frame of the form: @var{stream_index}, @var{frame_dts},
@var{frame_size}, 0x@var{CRC}, where @var{CRC} is a hexadecimal
number 0-padded to 8 digits containing the CRC of the decoded frame.
For example to compute the CRC of each decoded frame in the input, and
store it in the file @file{out.crc}:
@example
ffmpeg -i INPUT -f framecrc out.crc
@end example
You can print the CRC of each decoded frame to stdout with the command:
@example
ffmpeg -i INPUT -f framecrc -
@end example
You can select the output format of each frame with @command{ffmpeg} by
specifying the audio and video codec and format. For example, to
compute the CRC of each decoded input audio frame converted to PCM
unsigned 8-bit and of each decoded input video frame converted to
MPEG-2 video, use the command:
@example
ffmpeg -i INPUT -c:a pcm_u8 -c:v mpeg2video -f framecrc -
@end example
See also the @ref{crc} muxer.
@anchor{image2}
@section image2
Image file muxer.
The image file muxer writes video frames to image files.
The output filenames are specified by a pattern, which can be used to
produce sequentially numbered series of files.
The pattern may contain the string "%d" or "%0@var{N}d", this string
specifies the position of the characters representing a numbering in
the filenames. If the form "%0@var{N}d" is used, the string
representing the number in each filename is 0-padded to @var{N}
digits. The literal character '%' can be specified in the pattern with
the string "%%".
If the pattern contains "%d" or "%0@var{N}d", the first filename of
the file list specified will contain the number 1, all the following
numbers will be sequential.
The pattern may contain a suffix which is used to automatically
determine the format of the image files to write.
For example the pattern "img-%03d.bmp" will specify a sequence of
filenames of the form @file{img-001.bmp}, @file{img-002.bmp}, ...,
@file{img-010.bmp}, etc.
The pattern "img%%-%d.jpg" will specify a sequence of filenames of the
form @file{img%-1.jpg}, @file{img%-2.jpg}, ..., @file{img%-10.jpg},
etc.
The following example shows how to use @command{ffmpeg} for creating a
sequence of files @file{img-001.jpeg}, @file{img-002.jpeg}, ...,
taking one image every second from the input video:
@example
ffmpeg -i in.avi -vsync 1 -r 1 -f image2 'img-%03d.jpeg'
@end example
Note that with @command{ffmpeg}, if the format is not specified with the
@code{-f} option and the output filename specifies an image file
format, the image2 muxer is automatically selected, so the previous
command can be written as:
@example
ffmpeg -i in.avi -vsync 1 -r 1 'img-%03d.jpeg'
@end example
Note also that the pattern must not necessarily contain "%d" or
"%0@var{N}d", for example to create a single image file
@file{img.jpeg} from the input video you can employ the command:
@example
ffmpeg -i in.avi -f image2 -frames:v 1 img.jpeg
@end example
The image muxer supports the .Y.U.V image file format. This format is
special in that that each image frame consists of three files, for
each of the YUV420P components. To read or write this image file format,
specify the name of the '.Y' file. The muxer will automatically open the
'.U' and '.V' files as required.
@section mov
MOV / MP4 muxer
The muxer options are:
@table @option
@item -moov_size @var{bytes}
Reserves space for the moov atom at the beginning of the file instead of placing the
moov atom at the end. If the space reserved is insufficient, muxing will fail.
@end table
@section mpegts
MPEG transport stream muxer.
This muxer implements ISO 13818-1 and part of ETSI EN 300 468.
The muxer options are:
@table @option
@item -mpegts_original_network_id @var{number}
Set the original_network_id (default 0x0001). This is unique identifier
of a network in DVB. Its main use is in the unique identification of a
service through the path Original_Network_ID, Transport_Stream_ID.
@item -mpegts_transport_stream_id @var{number}
Set the transport_stream_id (default 0x0001). This identifies a
transponder in DVB.
@item -mpegts_service_id @var{number}
Set the service_id (default 0x0001) also known as program in DVB.
@item -mpegts_pmt_start_pid @var{number}
Set the first PID for PMT (default 0x1000, max 0x1f00).
@item -mpegts_start_pid @var{number}
Set the first PID for data packets (default 0x0100, max 0x0f00).
@end table
The recognized metadata settings in mpegts muxer are @code{service_provider}
and @code{service_name}. If they are not set the default for
@code{service_provider} is "FFmpeg" and the default for
@code{service_name} is "Service01".
@example
ffmpeg -i file.mpg -c copy \
-mpegts_original_network_id 0x1122 \
-mpegts_transport_stream_id 0x3344 \
-mpegts_service_id 0x5566 \
-mpegts_pmt_start_pid 0x1500 \
-mpegts_start_pid 0x150 \
-metadata service_provider="Some provider" \
-metadata service_name="Some Channel" \
-y out.ts
@end example
@section null
Null muxer.
This muxer does not generate any output file, it is mainly useful for
testing or benchmarking purposes.
For example to benchmark decoding with @command{ffmpeg} you can use the
command:
@example
ffmpeg -benchmark -i INPUT -f null out.null
@end example
Note that the above command does not read or write the @file{out.null}
file, but specifying the output file is required by the @command{ffmpeg}
syntax.
Alternatively you can write the command as:
@example
ffmpeg -benchmark -i INPUT -f null -
@end example
@section matroska
Matroska container muxer.
This muxer implements the matroska and webm container specs.
The recognized metadata settings in this muxer are:
@table @option
@item title=@var{title name}
Name provided to a single track
@end table
@table @option
@item language=@var{language name}
Specifies the language of the track in the Matroska languages form
@end table
@table @option
@item stereo_mode=@var{mode}
Stereo 3D video layout of two views in a single video track
@table @option
@item mono
video is not stereo
@item left_right
Both views are arranged side by side, Left-eye view is on the left
@item bottom_top
Both views are arranged in top-bottom orientation, Left-eye view is at bottom
@item top_bottom
Both views are arranged in top-bottom orientation, Left-eye view is on top
@item checkerboard_rl
Each view is arranged in a checkerboard interleaved pattern, Left-eye view being first
@item checkerboard_lr
Each view is arranged in a checkerboard interleaved pattern, Right-eye view being first
@item row_interleaved_rl
Each view is constituted by a row based interleaving, Right-eye view is first row
@item row_interleaved_lr
Each view is constituted by a row based interleaving, Left-eye view is first row
@item col_interleaved_rl
Both views are arranged in a column based interleaving manner, Right-eye view is first column
@item col_interleaved_lr
Both views are arranged in a column based interleaving manner, Left-eye view is first column
@item anaglyph_cyan_red
All frames are in anaglyph format viewable through red-cyan filters
@item right_left
Both views are arranged side by side, Right-eye view is on the left
@item anaglyph_green_magenta
All frames are in anaglyph format viewable through green-magenta filters
@item block_lr
Both eyes laced in one Block, Left-eye view is first
@item block_rl
Both eyes laced in one Block, Right-eye view is first
@end table
@end table
For example a 3D WebM clip can be created using the following command line:
@example
ffmpeg -i sample_left_right_clip.mpg -an -c:v libvpx -metadata stereo_mode=left_right -y stereo_clip.webm
@end example
@section segment
Basic stream segmenter.
The segmenter muxer outputs streams to a number of separate files of nearly
fixed duration. Output filename pattern can be set in a fashion similar to
@ref{image2}.
Every segment starts with a video keyframe, if a video stream is present.
The segment muxer works best with a single constant frame rate video.
Optionally it can generate a flat list of the created segments, one segment
per line.
@table @option
@item segment_format @var{format}
Override the inner container format, by default it is guessed by the filename
extension.
@item segment_time @var{t}
Set segment duration to @var{t} seconds.
@item segment_list @var{name}
Generate also a listfile named @var{name}.
@item segment_list_size @var{size}
Overwrite the listfile once it reaches @var{size} entries.
@end table
@example
ffmpeg -i in.mkv -c copy -map 0 -f segment -list out.list out%03d.nut
@end example
@c man end MUXERS
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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 (web frontends for the various FFmpeg
branches are listed at http://ffmpeg.org/download.html).
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.ffmpeg.org/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_instruction ....
Do not use C loops:
do{
__asm__(
...
}while()
For x86, mark registers that are clobbered in your asm. This means both
general x86 registers (e.g. eax) as well as XMM registers. This last one is
particularly important on Win64, where xmm6-15 are callee-save, and not
restoring their contents leads to undefined results. In external asm (e.g.
yasm), you do this by using:
cglobal functon_name, num_args, num_regs, num_xmm_regs
In inline asm, you specify clobbered registers at the end of your asm:
__asm__(".." ::: "%eax").
If gcc is not set to support sse (-msse) it will not accept xmm registers
in the clobber list. For that we use two macros to declare the clobbers.
XMM_CLOBBERS should be used when there are other clobbers, for example:
__asm__(".." ::: XMM_CLOBBERS("xmm0",) "eax");
and XMM_CLOBBERS_ONLY should be used when the only clobbers are xmm registers:
__asm__(".." :: XMM_CLOBBERS_ONLY("xmm0"));
Do not expect a compiler to maintain values in your registers between separate
(inline) asm code blocks. It is not required to. For example, this is bad:
__asm__("movdqa %0, %%xmm7" : src);
/* do something */
__asm__("movdqa %%xmm7, %1" : dst);
- first of all, you're assuming that the compiler will not use xmm7 in
between the two asm blocks. It probably won't when you test it, but it's
a poor assumption that will break at some point for some --cpu compiler flag
- secondly, you didn't mark xmm7 as clobbered. If you did, the compiler would
have restored the original value of xmm7 after the first asm block, thus
rendering the combination of the two blocks of code invalid
Code that depends on data in registries being untouched, should be written as
a single __asm__() statement. Ideally, a single function contains only one
__asm__() block.
Use external asm (nasm/yasm) or inline asm (__asm__()), do not use intrinsics.
The latter requires a good optimizing compiler which gcc is not.
Inline asm vs. external asm
---------------------------
Both inline asm (__asm__("..") in a .c file, handled by a compiler such as gcc)
and external asm (.s or .asm files, handled by an assembler such as yasm/nasm)
are accepted in FFmpeg. Which one to use differs per specific case.
- if your code is intended to be inlined in a C function, inline asm is always
better, because external asm cannot be inlined
- if your code calls external functions, yasm is always better
- if your code takes huge and complex structs as function arguments (e.g.
MpegEncContext; note that this is not ideal and is discouraged if there
are alternatives), then inline asm is always better, because predicting
member offsets in complex structs is almost impossible. It's safest to let
the compiler take care of that
- in many cases, both can be used and it just depends on the preference of the
person writing the asm. For new asm, the choice is up to you. For existing
asm, you'll likely want to maintain whatever form it is currently in unless
there is a good reason to change it.
- if, for some reason, you believe that a particular chunk of existing external
asm could be improved upon further if written in inline asm (or the other
way around), then please make the move from external asm <-> inline asm a
separate patch before your patches that actually improve the asm.
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
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@chapter Output Devices
@c man begin OUTPUT DEVICES
Output devices are configured elements in FFmpeg which allow to write
multimedia data to an output device attached to your system.
When you configure your FFmpeg build, all the supported output devices
are enabled by default. You can list all available ones using the
configure option "--list-outdevs".
You can disable all the output devices using the configure option
"--disable-outdevs", and selectively enable an output device using the
option "--enable-outdev=@var{OUTDEV}", or you can disable a particular
input device using the option "--disable-outdev=@var{OUTDEV}".
The option "-formats" of the ff* tools will display the list of
enabled output devices (amongst the muxers).
A description of the currently available output devices follows.
@section alsa
ALSA (Advanced Linux Sound Architecture) output device.
@section oss
OSS (Open Sound System) output device.
@section sdl
SDL (Simple DirectMedia Layer) output device.
This output devices allows to show a video stream in an SDL
window. Only one SDL window is allowed per application, so you can
have only one instance of this output device in an application.
To enable this output device you need libsdl installed on your system
when configuring your build.
For more information about SDL, check:
@url{http://www.libsdl.org/}
@subsection Options
@table @option
@item window_title
Set the SDL window title, if not specified default to the filename
specified for the output device.
@item icon_title
Set the name of the iconified SDL window, if not specified it is set
to the same value of @var{window_title}.
@item window_size
Set the SDL window size, can be a string of the form
@var{width}x@var{height} or a video size abbreviation.
If not specified it defaults to the size of the input video.
@end table
@subsection Examples
The following command shows the @command{ffmpeg} output is an
SDL window, forcing its size to the qcif format:
@example
ffmpeg -i INPUT -vcodec rawvideo -pix_fmt yuv420p -window_size qcif -f sdl "SDL output"
@end example
@section sndio
sndio audio output device.
@c man end OUTPUT DEVICES
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\input texinfo @c -*- texinfo -*-
@settitle Platform Specific information
@titlepage
@center @titlefont{Platform Specific information}
@end titlepage
@top
@contents
@chapter Unix-like
Some parts of FFmpeg cannot be built with version 2.15 of the GNU
assembler which is still provided by a few AMD64 distributions. To
make sure your compiler really uses the required version of gas
after a binutils upgrade, run:
@example
$(gcc -print-prog-name=as) --version
@end example
If not, then you should install a different compiler that has no
hard-coded path to gas. In the worst case pass @code{--disable-asm}
to configure.
@section BSD
BSD make will not build FFmpeg, you need to install and use GNU Make
(@file{gmake}).
@section (Open)Solaris
GNU Make is required to build FFmpeg, so you have to invoke (@file{gmake}),
standard Solaris Make will not work. When building with a non-c99 front-end
(gcc, generic suncc) add either @code{--extra-libs=/usr/lib/values-xpg6.o}
or @code{--extra-libs=/usr/lib/64/values-xpg6.o} to the configure options
since the libc is not c99-compliant by default. The probes performed by
configure may raise an exception leading to the death of configure itself
due to a bug in the system shell. Simply invoke a different shell such as
bash directly to work around this:
@example
bash ./configure
@end example
@anchor{Darwin}
@section Darwin (Mac OS X, iPhone)
The toolchain provided with Xcode is sufficient to build the basic
unacelerated code.
Mac OS X on PowerPC or ARM (iPhone) requires a preprocessor from
@url{http://github.com/yuvi/gas-preprocessor} to build the optimized
assembler functions. Just download the Perl script and put it somewhere
in your PATH, FFmpeg's configure will pick it up automatically.
Mac OS X on amd64 and x86 requires @command{yasm} to build most of the
optimized assembler functions. @uref{http://www.finkproject.org/, Fink},
@uref{http://www.gentoo.org/proj/en/gentoo-alt/prefix/bootstrap-macos.xml, Gentoo Prefix},
@uref{http://mxcl.github.com/homebrew/, Homebrew}
or @uref{http://www.macports.org, MacPorts} can easily provide it.
@chapter DOS
Using a cross-compiler is preferred for various reasons.
@url{http://www.delorie.com/howto/djgpp/linux-x-djgpp.html}
@chapter OS/2
For information about compiling FFmpeg on OS/2 see
@url{http://www.edm2.com/index.php/FFmpeg}.
@chapter Windows
To get help and instructions for building FFmpeg under Windows, check out
the FFmpeg Windows Help Forum at
@url{http://ffmpeg.arrozcru.org/}.
@section Native Windows compilation
FFmpeg can be built to run natively on Windows using the MinGW tools. Install
the latest versions of MSYS and MinGW from @url{http://www.mingw.org/}.
You can find detailed installation instructions in the download
section and the FAQ.
FFmpeg does not build out-of-the-box with the packages the automated MinGW
installer provides. It also requires coreutils to be installed and many other
packages updated to the latest version. The minimum version for some packages
are listed below:
@itemize
@item bash 3.1
@item msys-make 3.81-2 (note: not mingw32-make)
@item w32api 3.13
@item mingw-runtime 3.15
@end itemize
FFmpeg automatically passes @code{-fno-common} to the compiler to work around
a GCC bug (see @url{http://gcc.gnu.org/bugzilla/show_bug.cgi?id=37216}).
Notes:
@itemize
@item Building natively using MSYS can be sped up by disabling implicit rules
in the Makefile by calling @code{make -r} instead of plain @code{make}. This
speed up is close to non-existent for normal one-off builds and is only
noticeable when running make for a second time (for example in
@code{make install}).
@item In order to compile FFplay, you must have the MinGW development library
of @uref{http://www.libsdl.org/, SDL}.
Edit the @file{bin/sdl-config} script so that it points to the correct prefix
where SDL was installed. Verify that @file{sdl-config} can be launched from
the MSYS command line.
@item By using @code{./configure --enable-shared} when configuring FFmpeg,
you can build the FFmpeg libraries (e.g. libavutil, libavcodec,
libavformat) as DLLs.
@end itemize
@section Microsoft Visual C++ compatibility
As stated in the FAQ, FFmpeg will not compile under MSVC++. However, if you
want to use the libav* libraries in your own applications, you can still
compile those applications using MSVC++. But the libav* libraries you link
to @emph{must} be built with MinGW. However, you will not be able to debug
inside the libav* libraries, since MSVC++ does not recognize the debug
symbols generated by GCC.
We strongly recommend you to move over from MSVC++ to MinGW tools.
This description of how to use the FFmpeg libraries with MSVC++ is based on
Microsoft Visual C++ 2005 Express Edition. If you have a different version,
you might have to modify the procedures slightly.
@subsection Using static libraries
Assuming you have just built and installed FFmpeg in @file{/usr/local}.
@enumerate
@item Create a new console application ("File / New / Project") and then
select "Win32 Console Application". On the appropriate page of the
Application Wizard, uncheck the "Precompiled headers" option.
@item Write the source code for your application, or, for testing, just
copy the code from an existing sample application into the source file
that MSVC++ has already created for you. For example, you can copy
@file{libavformat/output-example.c} from the FFmpeg distribution.
@item Open the "Project / Properties" dialog box. In the "Configuration"
combo box, select "All Configurations" so that the changes you make will
affect both debug and release builds. In the tree view on the left hand
side, select "C/C++ / General", then edit the "Additional Include
Directories" setting to contain the path where the FFmpeg includes were
installed (i.e. @file{c:\msys\1.0\local\include}).
Do not add MinGW's include directory here, or the include files will
conflict with MSVC's.
@item Still in the "Project / Properties" dialog box, select
"Linker / General" from the tree view and edit the
"Additional Library Directories" setting to contain the @file{lib}
directory where FFmpeg was installed (i.e. @file{c:\msys\1.0\local\lib}),
the directory where MinGW libs are installed (i.e. @file{c:\mingw\lib}),
and the directory where MinGW's GCC libs are installed
(i.e. @file{C:\mingw\lib\gcc\mingw32\4.2.1-sjlj}). Then select
"Linker / Input" from the tree view, and add the files @file{libavformat.a},
@file{libavcodec.a}, @file{libavutil.a}, @file{libmingwex.a},
@file{libgcc.a}, and any other libraries you used (i.e. @file{libz.a})
to the end of "Additional Dependencies".
@item Now, select "C/C++ / Code Generation" from the tree view. Select
"Debug" in the "Configuration" combo box. Make sure that "Runtime
Library" is set to "Multi-threaded Debug DLL". Then, select "Release" in
the "Configuration" combo box and make sure that "Runtime Library" is
set to "Multi-threaded DLL".
@item Click "OK" to close the "Project / Properties" dialog box.
@item MSVC++ lacks some C99 header files that are fundamental for FFmpeg.
Get msinttypes from @url{http://code.google.com/p/msinttypes/downloads/list}
and install it in MSVC++'s include directory
(i.e. @file{C:\Program Files\Microsoft Visual Studio 8\VC\include}).
@item MSVC++ also does not understand the @code{inline} keyword used by
FFmpeg, so you must add this line before @code{#include}ing libav*:
@example
#define inline _inline
@end example
@item Build your application, everything should work.
@end enumerate
@subsection Using shared libraries
This is how to create DLL and LIB files that are compatible with MSVC++:
@enumerate
@item Add a call to @file{vcvars32.bat} (which sets up the environment
variables for the Visual C++ tools) as the first line of @file{msys.bat}.
The standard location for @file{vcvars32.bat} is
@file{C:\Program Files\Microsoft Visual Studio 8\VC\bin\vcvars32.bat},
and the standard location for @file{msys.bat} is @file{C:\msys\1.0\msys.bat}.
If this corresponds to your setup, add the following line as the first line
of @file{msys.bat}:
@example
call "C:\Program Files\Microsoft Visual Studio 8\VC\bin\vcvars32.bat"
@end example
Alternatively, you may start the @file{Visual Studio 2005 Command Prompt},
and run @file{c:\msys\1.0\msys.bat} from there.
@item Within the MSYS shell, run @code{lib.exe}. If you get a help message
from @file{Microsoft (R) Library Manager}, this means your environment
variables are set up correctly, the @file{Microsoft (R) Library Manager}
is on the path and will be used by FFmpeg to create
MSVC++-compatible import libraries.
@item Build FFmpeg with
@example
./configure --enable-shared
make
make install
@end example
Your install path (@file{/usr/local/} by default) should now have the
necessary DLL and LIB files under the @file{bin} directory.
@end enumerate
Alternatively, build the libraries with a cross compiler, according to
the instructions below in @ref{Cross compilation for Windows with Linux}.
To use those files with MSVC++, do the same as you would do with
the static libraries, as described above. But in Step 4,
you should only need to add the directory where the LIB files are installed
(i.e. @file{c:\msys\usr\local\bin}). This is not a typo, the LIB files are
installed in the @file{bin} directory. And instead of adding the static
libraries (@file{libxxx.a} files) you should add the MSVC import libraries
(@file{avcodec.lib}, @file{avformat.lib}, and
@file{avutil.lib}). Note that you should not use the GCC import
libraries (@file{libxxx.dll.a} files), as these will give you undefined
reference errors. There should be no need for @file{libmingwex.a},
@file{libgcc.a}, and @file{wsock32.lib}, nor any other external library
statically linked into the DLLs.
FFmpeg headers do not declare global data for Windows DLLs through the usual
dllexport/dllimport interface. Such data will be exported properly while
building, but to use them in your MSVC++ code you will have to edit the
appropriate headers and mark the data as dllimport. For example, in
libavutil/pixdesc.h you should have:
@example
extern __declspec(dllimport) const AVPixFmtDescriptor av_pix_fmt_descriptors[];
@end example
Note that using import libraries created by dlltool requires
the linker optimization option to be set to
"References: Keep Unreferenced Data (@code{/OPT:NOREF})", otherwise
the resulting binaries will fail during runtime. This isn't
required when using import libraries generated by lib.exe.
This issue is reported upstream at
@url{http://sourceware.org/bugzilla/show_bug.cgi?id=12633}.
To create import libraries that work with the @code{/OPT:REF} option
(which is enabled by default in Release mode), follow these steps:
@enumerate
@item Open @file{Visual Studio 2005 Command Prompt}.
Alternatively, in a normal command line prompt, call @file{vcvars32.bat}
which sets up the environment variables for the Visual C++ tools
(the standard location for this file is
@file{C:\Program Files\Microsoft Visual Studio 8\VC\bin\vcvars32.bat}).
@item Enter the @file{bin} directory where the created LIB and DLL files
are stored.
@item Generate new import libraries with @file{lib.exe}:
@example
lib /machine:i386 /def:..\lib\avcodec-53.def /out:avcodec.lib
lib /machine:i386 /def:..\lib\avdevice-53.def /out:avdevice.lib
lib /machine:i386 /def:..\lib\avfilter-2.def /out:avfilter.lib
lib /machine:i386 /def:..\lib\avformat-53.def /out:avformat.lib
lib /machine:i386 /def:..\lib\avutil-51.def /out:avutil.lib
lib /machine:i386 /def:..\lib\swscale-2.def /out:swscale.lib
@end example
@end enumerate
@anchor{Cross compilation for Windows with Linux}
@section Cross compilation for Windows with Linux
You must use the MinGW cross compilation tools available at
@url{http://www.mingw.org/}.
Then configure FFmpeg with the following options:
@example
./configure --target-os=mingw32 --cross-prefix=i386-mingw32msvc-
@end example
(you can change the cross-prefix according to the prefix chosen for the
MinGW tools).
Then you can easily test FFmpeg with @uref{http://www.winehq.com/, Wine}.
@section Compilation under Cygwin
Please use Cygwin 1.7.x as the obsolete 1.5.x Cygwin versions lack
llrint() in its C library.
Install your Cygwin with all the "Base" packages, plus the
following "Devel" ones:
@example
binutils, gcc4-core, make, git, mingw-runtime, texi2html
@end example
And the following "Utils" one:
@example
diffutils
@end example
Then run
@example
./configure
@end example
to make a static build.
The current @code{gcc4-core} package is buggy and needs this flag to build
shared libraries:
@example
./configure --enable-shared --disable-static --extra-cflags=-fno-reorder-functions
@end example
If you want to build FFmpeg with additional libraries, download Cygwin
"Devel" packages for Ogg and Vorbis from any Cygwin packages repository:
@example
libogg-devel, libvorbis-devel
@end example
These library packages are only available from
@uref{http://sourceware.org/cygwinports/, Cygwin Ports}:
@example
yasm, libSDL-devel, libdirac-devel, libfaac-devel, libaacplus-devel, libgsm-devel,
libmp3lame-devel, libschroedinger1.0-devel, speex-devel, libtheora-devel,
libxvidcore-devel
@end example
The recommendation for libnut and x264 is to build them from source by
yourself, as they evolve too quickly for Cygwin Ports to be up to date.
Cygwin 1.7.x has IPv6 support. You can add IPv6 to Cygwin 1.5.x by means
of the @code{libgetaddrinfo-devel} package, available at Cygwin Ports.
@section Crosscompilation for Windows under Cygwin
With Cygwin you can create Windows binaries that do not need the cygwin1.dll.
Just install your Cygwin as explained before, plus these additional
"Devel" packages:
@example
gcc-mingw-core, mingw-runtime, mingw-zlib
@end example
and add some special flags to your configure invocation.
For a static build run
@example
./configure --target-os=mingw32 --extra-cflags=-mno-cygwin --extra-libs=-mno-cygwin
@end example
and for a build with shared libraries
@example
./configure --target-os=mingw32 --enable-shared --disable-static --extra-cflags=-mno-cygwin --extra-libs=-mno-cygwin
@end example
@bye
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@chapter Protocols
@c man begin PROTOCOLS
Protocols are configured elements in FFmpeg which allow to access
resources which require the use of a particular protocol.
When you configure your FFmpeg build, all the supported protocols are
enabled by default. You can list all available ones using the
configure option "--list-protocols".
You can disable all the protocols using the configure option
"--disable-protocols", and selectively enable a protocol using the
option "--enable-protocol=@var{PROTOCOL}", or you can disable a
particular protocol using the option
"--disable-protocol=@var{PROTOCOL}".
The option "-protocols" of the ff* tools will display the list of
supported protocols.
A description of the currently available protocols follows.
@section applehttp
Read Apple HTTP Live Streaming compliant segmented stream as
a uniform one. The M3U8 playlists describing the segments can be
remote HTTP resources or local files, accessed using the standard
file protocol.
HTTP is default, specific protocol can be declared by specifying
"+@var{proto}" after the applehttp URI scheme name, where @var{proto}
is either "file" or "http".
@example
applehttp://host/path/to/remote/resource.m3u8
applehttp+http://host/path/to/remote/resource.m3u8
applehttp+file://path/to/local/resource.m3u8
@end example
@section concat
Physical concatenation protocol.
Allow to read and seek from many resource in sequence as if they were
a unique resource.
A URL accepted by this protocol has the syntax:
@example
concat:@var{URL1}|@var{URL2}|...|@var{URLN}
@end example
where @var{URL1}, @var{URL2}, ..., @var{URLN} are the urls of the
resource to be concatenated, each one possibly specifying a distinct
protocol.
For example to read a sequence of files @file{split1.mpeg},
@file{split2.mpeg}, @file{split3.mpeg} with @command{ffplay} use the
command:
@example
ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg
@end example
Note that you may need to escape the character "|" which is special for
many shells.
@section file
File access protocol.
Allow to read from or read to a file.
For example to read from a file @file{input.mpeg} with @command{ffmpeg}
use the command:
@example
ffmpeg -i file:input.mpeg output.mpeg
@end example
The ff* tools default to the file protocol, that is a resource
specified with the name "FILE.mpeg" is interpreted as the URL
"file:FILE.mpeg".
@section gopher
Gopher protocol.
@section http
HTTP (Hyper Text Transfer Protocol).
@section mmst
MMS (Microsoft Media Server) protocol over TCP.
@section mmsh
MMS (Microsoft Media Server) protocol over HTTP.
The required syntax is:
@example
mmsh://@var{server}[:@var{port}][/@var{app}][/@var{playpath}]
@end example
@section md5
MD5 output protocol.
Computes the MD5 hash of the data to be written, and on close writes
this to the designated output or stdout if none is specified. It can
be used to test muxers without writing an actual file.
Some examples follow.
@example
# Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
ffmpeg -i input.flv -f avi -y md5:output.avi.md5
# Write the MD5 hash of the encoded AVI file to stdout.
ffmpeg -i input.flv -f avi -y md5:
@end example
Note that some formats (typically MOV) require the output protocol to
be seekable, so they will fail with the MD5 output protocol.
@section pipe
UNIX pipe access protocol.
Allow to read and write from UNIX pipes.
The accepted syntax is:
@example
pipe:[@var{number}]
@end example
@var{number} is the number corresponding to the file descriptor of the
pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If @var{number}
is not specified, by default the stdout file descriptor will be used
for writing, stdin for reading.
For example to read from stdin with @command{ffmpeg}:
@example
cat test.wav | ffmpeg -i pipe:0
# ...this is the same as...
cat test.wav | ffmpeg -i pipe:
@end example
For writing to stdout with @command{ffmpeg}:
@example
ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
# ...this is the same as...
ffmpeg -i test.wav -f avi pipe: | cat > test.avi
@end example
Note that some formats (typically MOV), require the output protocol to
be seekable, so they will fail with the pipe output protocol.
@section rtmp
Real-Time Messaging Protocol.
The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia
content across a TCP/IP network.
The required syntax is:
@example
rtmp://@var{server}[:@var{port}][/@var{app}][/@var{playpath}]
@end example
The accepted parameters are:
@table @option
@item server
The address of the RTMP server.
@item port
The number of the TCP port to use (by default is 1935).
@item app
It is the name of the application to access. It usually corresponds to
the path where the application is installed on the RTMP server
(e.g. @file{/ondemand/}, @file{/flash/live/}, etc.).
@item playpath
It is the path or name of the resource to play with reference to the
application specified in @var{app}, may be prefixed by "mp4:".
@end table
For example to read with @command{ffplay} a multimedia resource named
"sample" from the application "vod" from an RTMP server "myserver":
@example
ffplay rtmp://myserver/vod/sample
@end example
@section rtmp, rtmpe, rtmps, rtmpt, rtmpte
Real-Time Messaging Protocol and its variants supported through
librtmp.
Requires the presence of the librtmp headers and library during
configuration. You need to explicitly configure the build with
"--enable-librtmp". If enabled this will replace the native RTMP
protocol.
This protocol provides most client functions and a few server
functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT),
encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled
variants of these encrypted types (RTMPTE, RTMPTS).
The required syntax is:
@example
@var{rtmp_proto}://@var{server}[:@var{port}][/@var{app}][/@var{playpath}] @var{options}
@end example
where @var{rtmp_proto} is one of the strings "rtmp", "rtmpt", "rtmpe",
"rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and
@var{server}, @var{port}, @var{app} and @var{playpath} have the same
meaning as specified for the RTMP native protocol.
@var{options} contains a list of space-separated options of the form
@var{key}=@var{val}.
See the librtmp manual page (man 3 librtmp) for more information.
For example, to stream a file in real-time to an RTMP server using
@command{ffmpeg}:
@example
ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
@end example
To play the same stream using @command{ffplay}:
@example
ffplay "rtmp://myserver/live/mystream live=1"
@end example
@section rtp
Real-Time Protocol.
@section rtsp
RTSP is not technically a protocol handler in libavformat, it is a demuxer
and muxer. The demuxer supports both normal RTSP (with data transferred
over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with
data transferred over RDT).
The muxer can be used to send a stream using RTSP ANNOUNCE to a server
supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's
@uref{http://github.com/revmischa/rtsp-server, RTSP server}).
The required syntax for a RTSP url is:
@example
rtsp://@var{hostname}[:@var{port}]/@var{path}
@end example
The following options (set on the @command{ffmpeg}/@command{ffplay} command
line, or set in code via @code{AVOption}s or in @code{avformat_open_input}),
are supported:
Flags for @code{rtsp_transport}:
@table @option
@item udp
Use UDP as lower transport protocol.
@item tcp
Use TCP (interleaving within the RTSP control channel) as lower
transport protocol.
@item udp_multicast
Use UDP multicast as lower transport protocol.
@item http
Use HTTP tunneling as lower transport protocol, which is useful for
passing proxies.
@end table
Multiple lower transport protocols may be specified, in that case they are
tried one at a time (if the setup of one fails, the next one is tried).
For the muxer, only the @code{tcp} and @code{udp} options are supported.
Flags for @code{rtsp_flags}:
@table @option
@item filter_src
Accept packets only from negotiated peer address and port.
@end table
When receiving data over UDP, the demuxer tries to reorder received packets
(since they may arrive out of order, or packets may get lost totally). In
order for this to be enabled, a maximum delay must be specified in the
@code{max_delay} field of AVFormatContext.
When watching multi-bitrate Real-RTSP streams with @command{ffplay}, the
streams to display can be chosen with @code{-vst} @var{n} and
@code{-ast} @var{n} for video and audio respectively, and can be switched
on the fly by pressing @code{v} and @code{a}.
Example command lines:
To watch a stream over UDP, with a max reordering delay of 0.5 seconds:
@example
ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
@end example
To watch a stream tunneled over HTTP:
@example
ffplay -rtsp_transport http rtsp://server/video.mp4
@end example
To send a stream in realtime to a RTSP server, for others to watch:
@example
ffmpeg -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
@end example
@section sap
Session Announcement Protocol (RFC 2974). This is not technically a
protocol handler in libavformat, it is a muxer and demuxer.
It is used for signalling of RTP streams, by announcing the SDP for the
streams regularly on a separate port.
@subsection Muxer
The syntax for a SAP url given to the muxer is:
@example
sap://@var{destination}[:@var{port}][?@var{options}]
@end example
The RTP packets are sent to @var{destination} on port @var{port},
or to port 5004 if no port is specified.
@var{options} is a @code{&}-separated list. The following options
are supported:
@table @option
@item announce_addr=@var{address}
Specify the destination IP address for sending the announcements to.
If omitted, the announcements are sent to the commonly used SAP
announcement multicast address 224.2.127.254 (sap.mcast.net), or
ff0e::2:7ffe if @var{destination} is an IPv6 address.
@item announce_port=@var{port}
Specify the port to send the announcements on, defaults to
9875 if not specified.
@item ttl=@var{ttl}
Specify the time to live value for the announcements and RTP packets,
defaults to 255.
@item same_port=@var{0|1}
If set to 1, send all RTP streams on the same port pair. If zero (the
default), all streams are sent on unique ports, with each stream on a
port 2 numbers higher than the previous.
VLC/Live555 requires this to be set to 1, to be able to receive the stream.
The RTP stack in libavformat for receiving requires all streams to be sent
on unique ports.
@end table
Example command lines follow.
To broadcast a stream on the local subnet, for watching in VLC:
@example
ffmpeg -re -i @var{input} -f sap sap://224.0.0.255?same_port=1
@end example
Similarly, for watching in @command{ffplay}:
@example
ffmpeg -re -i @var{input} -f sap sap://224.0.0.255
@end example
And for watching in @command{ffplay}, over IPv6:
@example
ffmpeg -re -i @var{input} -f sap sap://[ff0e::1:2:3:4]
@end example
@subsection Demuxer
The syntax for a SAP url given to the demuxer is:
@example
sap://[@var{address}][:@var{port}]
@end example
@var{address} is the multicast address to listen for announcements on,
if omitted, the default 224.2.127.254 (sap.mcast.net) is used. @var{port}
is the port that is listened on, 9875 if omitted.
The demuxers listens for announcements on the given address and port.
Once an announcement is received, it tries to receive that particular stream.
Example command lines follow.
To play back the first stream announced on the normal SAP multicast address:
@example
ffplay sap://
@end example
To play back the first stream announced on one the default IPv6 SAP multicast address:
@example
ffplay sap://[ff0e::2:7ffe]
@end example
@section tcp
Trasmission Control Protocol.
The required syntax for a TCP url is:
@example
tcp://@var{hostname}:@var{port}[?@var{options}]
@end example
@table @option
@item listen
Listen for an incoming connection
@example
ffmpeg -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
ffplay tcp://@var{hostname}:@var{port}
@end example
@end table
@section udp
User Datagram Protocol.
The required syntax for a UDP url is:
@example
udp://@var{hostname}:@var{port}[?@var{options}]
@end example
@var{options} contains a list of &-seperated options of the form @var{key}=@var{val}.
Follow the list of supported options.
@table @option
@item buffer_size=@var{size}
set the UDP buffer size in bytes
@item localport=@var{port}
override the local UDP port to bind with
@item localaddr=@var{addr}
Choose the local IP address. This is useful e.g. if sending multicast
and the host has multiple interfaces, where the user can choose
which interface to send on by specifying the IP address of that interface.
@item pkt_size=@var{size}
set the size in bytes of UDP packets
@item reuse=@var{1|0}
explicitly allow or disallow reusing UDP sockets
@item ttl=@var{ttl}
set the time to live value (for multicast only)
@item connect=@var{1|0}
Initialize the UDP socket with @code{connect()}. In this case, the
destination address can't be changed with ff_udp_set_remote_url later.
If the destination address isn't known at the start, this option can
be specified in ff_udp_set_remote_url, too.
This allows finding out the source address for the packets with getsockname,
and makes writes return with AVERROR(ECONNREFUSED) if "destination
unreachable" is received.
For receiving, this gives the benefit of only receiving packets from
the specified peer address/port.
@end table
Some usage examples of the udp protocol with @command{ffmpeg} follow.
To stream over UDP to a remote endpoint:
@example
ffmpeg -i @var{input} -f @var{format} udp://@var{hostname}:@var{port}
@end example
To stream in mpegts format over UDP using 188 sized UDP packets, using a large input buffer:
@example
ffmpeg -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535
@end example
To receive over UDP from a remote endpoint:
@example
ffmpeg -i udp://[@var{multicast-address}]:@var{port}
@end example
@c man end PROTOCOLS
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A Quick Description Of Rate Distortion Theory.
We want to encode a video, picture or piece of music optimally. What does
"optimally" really mean? It means that we want to get the best quality at a
given filesize OR we want to get the smallest filesize at a given quality
(in practice, these 2 goals are usually the same).
Solving this directly is not practical; trying all byte sequences 1
megabyte in length and selecting the "best looking" sequence will yield
256^1000000 cases to try.
But first, a word about quality, which is also called distortion.
Distortion can be quantified by almost any quality measurement one chooses.
Commonly, the sum of squared differences is used but more complex methods
that consider psychovisual effects can be used as well. It makes no
difference in this discussion.
First step: that rate distortion factor called lambda...
Let's consider the problem of minimizing:
distortion + lambda*rate
rate is the filesize
distortion is the quality
lambda is a fixed value choosen as a tradeoff between quality and filesize
Is this equivalent to finding the best quality for a given max
filesize? The answer is yes. For each filesize limit there is some lambda
factor for which minimizing above will get you the best quality (using your
chosen quality measurement) at the desired (or lower) filesize.
Second step: splitting the problem.
Directly splitting the problem of finding the best quality at a given
filesize is hard because we do not know how many bits from the total
filesize should be allocated to each of the subproblems. But the formula
from above:
distortion + lambda*rate
can be trivially split. Consider:
(distortion0 + distortion1) + lambda*(rate0 + rate1)
This creates a problem made of 2 independent subproblems. The subproblems
might be 2 16x16 macroblocks in a frame of 32x16 size. To minimize:
(distortion0 + distortion1) + lambda*(rate0 + rate1)
we just have to minimize:
distortion0 + lambda*rate0
and
distortion1 + lambda*rate1
I.e, the 2 problems can be solved independently.
Author: Michael Niedermayer
Copyright: LGPL
-630
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=============================================
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
-24
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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 development 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 git
* separation of cosmetic from non-cosmetic changes (this is almost entirely
ignored by mentors and students in soc 2006 which might lead to a surprise
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
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The official guide to swresample for confused developers.
=========================================================
Current (simplified) Architecture:
---------------------------------
Input
v
__________________/|\___________
/ | \
/ input sample format convert v
/ | ___________/
| |/
| v
| ___________/|\___________ _____________
| / | \ | |
| Rematrix | resample <---->| Buffers |
| \___________ | ___________/ |_____________|
v \|/
Special Converter v
v ___________/|\___________ _____________
| / | \ | |
| Rematrix | resample <---->| Buffers |
| \___________ | ___________/ |_____________|
| \|/
| v
| |\___________
\ | \
\ output sample format convert v
\_________________ | ___________/
\|/
v
Output
Planar/Packed convertion is done when needed during sample format convertion
Every step can be skiped without memcpy when its not needed.
Either Resampling and Rematrixing can be performed first depending on which
way its faster.
The Buffers are needed for resampling due to resamplng being a process that
requires future and past data, it thus also introduces inevitably a delay when
used.
Internally 32bit float and 16bit int is supported currently, other formats can
easily be added
Externally all sample formats in packed and planar configuration are supported
Its also trivial to add special converters for common cases
If only sample format and or packed/planar convertion is needed it
is performed from input to output directly in a single pass with no intermediates.
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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 -> RGB12/15/16/24/32. 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, ...
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# no horiz rules between sections
$end_section = \&FFmpeg_end_section;
sub FFmpeg_end_section($$)
{
}
$EXTRA_HEAD =
'<link rel="icon" href="favicon.png" type="image/png" />
<link rel="stylesheet" type="text/css" href="default.css" />
';
$CSS_LINES = <<EOT;
<style type="text/css">
<!--
a.summary-letter { text-decoration: none }
a { color: #2D6198; }
a:visited { color: #884488; }
h1 a, h2 a, h3 a { text-decoration: inherit; color: inherit; }
p { margin-left: 1em; margin-right: 1em; }
table { margin-left: 2em; }
pre { margin-left: 2em; }
#footer { text-align: center; }
#body { margin-left: 1em; margin-right: 1em; }
body { background-color: #313131; margin: 0; }
#container {
background-color: white;
color: #202020;
margin-left: 1em;
margin-right: 1em;
}
h1 {
background-color: #7BB37B;
border: 1px solid #6A996A;
color: #151515;
font-size: 1.2em;
padding-bottom: 0.2em;
padding-left: 0.4em;
padding-top: 0.2em;
}
h2 {
color: #313131;
font-size: 1.2em;
}
h3 {
color: #313131;
font-size: 0.8em;
margin-bottom: -8px;
}
.note {
margin: 1em;
border: 1px solid #bbc9d8;
background-color: #dde1e1;
}
.important {
margin: 1em;
border: 1px solid #d26767;
background-color: #f8e1e1;
}
-->
</style>
EOT
my $FFMPEG_NAVBAR = $ENV{"FFMPEG_NAVBAR"} || '';
$AFTER_BODY_OPEN =
'<div id="container">' .
"\n$FFMPEG_NAVBAR\n" .
'<div id="body">';
$PRE_BODY_CLOSE = '</div></div>';
$SMALL_RULE = '';
$BODYTEXT = '';
$print_page_foot = \&FFmpeg_print_page_foot;
sub FFmpeg_print_page_foot($$)
{
my $fh = shift;
print $fh '<div id="footer">' . "\n";
T2H_DEFAULT_print_page_foot($fh);
print $fh "</div>\n";
}
$float = \&FFmpeg_float;
sub FFmpeg_float($$$$)
{
my $text = shift;
my $float = shift;
my $caption = shift;
my $shortcaption = shift;
my $label = '';
if (exists($float->{'id'}))
{
$label = &$anchor($float->{'id'});
}
my $class = '';
my $subject = '';
if ($caption =~ /NOTE/)
{
$class = "note";
}
elsif ($caption =~ /IMPORTANT/)
{
$class = "important";
}
return '<div class="float ' . $class . '">' . "$label\n" . $text . '</div>';
}
$print_page_head = \&FFmpeg_print_page_head;
sub FFmpeg_print_page_head($$)
{
my $fh = shift;
my $longtitle = "$Texi2HTML::THISDOC{'title_no_texi'}";
$longtitle .= ": $Texi2HTML::NO_TEXI{'This'}" if exists $Texi2HTML::NO_TEXI{'This'};
my $description = $DOCUMENT_DESCRIPTION;
$description = $longtitle if (!defined($description));
$description = "<meta name=\"description\" content=\"$description\">" if
($description ne '');
$description = $Texi2HTML::THISDOC{'documentdescription'} if (defined($Texi2HTML::THISDOC{'documentdescription'}));
my $encoding = '';
$encoding = "<meta http-equiv=\"Content-Type\" content=\"text/html; charset=$ENCODING\">" if (defined($ENCODING) and ($ENCODING ne ''));
$longtitle =~ s/Documentation.*//g;
$longtitle = "FFmpeg documentation : " . $longtitle;
print $fh <<EOT;
$DOCTYPE
<html>
$Texi2HTML::THISDOC{'copying'}<!-- Created on $Texi2HTML::THISDOC{today} by $Texi2HTML::THISDOC{program} -->
<!--
$Texi2HTML::THISDOC{program_authors}
-->
<head>
<title>$longtitle</title>
$description
<meta name="keywords" content="$longtitle">
<meta name="resource-type" content="document">
<meta name="distribution" content="global">
<meta name="Generator" content="$Texi2HTML::THISDOC{program}">
$encoding
$CSS_LINES
$EXTRA_HEAD
</head>
<body $BODYTEXT>
$AFTER_BODY_OPEN
EOT
}
# no navigation elements
$SECTION_NAVIGATION = 0;
# the same for texi2html 5.0
$HEADERS = 0;
# TOC and Chapter headings link
$TOC_LINKS = 1;
# print the TOC where @contents is used
$INLINE_CONTENTS = 1;
-70
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Writing a table generator
This documentation is preliminary.
Parts of the API are not good and should be changed.
Basic concepts
A table generator consists of two files, *_tablegen.c and *_tablegen.h.
The .h file will provide the variable declarations and initialization
code for the tables, the .c calls the initialization code and then prints
the tables as a header file using the tableprint.h helpers.
Both of these files will be compiled for the host system, so to avoid
breakage with cross-compilation neither of them may include, directly
or indirectly, config.h or avconfig.h.
This means that e.g. libavutil/mathematics.h is ok but libavutil/libm.h is not.
Due to this, the .c file or Makefile may have to provide additional defines
or stubs, though if possible this should be avoided.
In particular, CONFIG_HARDCODED_TABLES should always be defined to 0.
The .c file
This file should include the *_tablegen.h and tableprint.h files and
anything else it needs as long as it does not depend on config.h or
avconfig.h.
In addition to that it must contain a main() function which initializes
all tables by calling the init functions from the .h file and then prints
them.
The printing code typically looks like this:
write_fileheader();
printf("static const uint8_t my_array[100] = {\n");
write_uint8_t_array(my_array, 100);
printf("};\n");
This is the more generic form, in case you need to do something special.
Usually you should instead use the short form:
write_fileheader();
WRITE_ARRAY("static const", uint8_t, my_array);
write_fileheader() adds some minor things like a "this is a generated file"
comment and some standard includes.
tablegen.h defines some write functions for one- and two-dimensional arrays
for standard types - they print only the "core" parts so they are easier
to reuse for multi-dimensional arrays so the outermost {} must be printed
separately.
If there's no standard function for printing the type you need, the
WRITE_1D_FUNC_ARGV macro is a very quick way to create one.
See libavcodec/dv_tablegen.c for an example.
The .h file
This file should contain:
- one or more initialization functions
- the table variable declarations
If CONFIG_HARDCODED_TABLES is set, the initialization functions should
not do anything, and instead of the variable declarations the
generated *_tables.h file should be included.
Since that will be generated in the build directory, the path must be
included, i.e.
#include "libavcodec/example_tables.h"
not
#include "example_tables.h"
Makefile changes
To make the automatic table creation work, you must manually declare the
new dependency.
For this add a line similar to this:
$(SUBDIR)example.o: $(SUBDIR)example_tables.h
under the "ifdef CONFIG_HARDCODED_TABLES" section in the Makefile.
-427
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@@ -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 = ();
@sects_sequence = ();
$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($2);
}
next;
};
/^\@include\s+(.+)$/ 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;
};
# 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-Za-z ]+)/ and $sect = $1, push (@sects_sequence, $sect), $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.
/^\@(?:section|unnumbered|unnumberedsec|center|heading)\s+(.+)$/
and $_ = "\n=head2 $1\n";
/^\@(?:subsection|subheading)\s+(.+)$/
and $_ = "\n=head3 $1\n";
/^\@(?:subsubsection|subsubheading)\s+(.+)$/
and $_ = "\n=head4 $1\n";
# Block command handlers:
/^\@itemize\s*(\@[a-z]+|\*|-)?/ and do {
push @endwstack, $endw;
push @icstack, $ic;
$ic = $1 ? $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};
unshift @sects_sequence, "NAME";
for $sect (@sects_sequence) {
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/\@anchor{(?:[^\}]*)\}//g;
s/\(?\@xref\{(?:[^\}]*)\}(?:[^.<]|(?:<[^<>]*>))*\.\)?//g;
s/\s+\(\@pxref\{(?:[^\}]*)\}\)//g;
s/;\s+\@pxref\{(?:[^\}]*)\}//g;
s/\@ref\{([^\}]*)\}/$1/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;
}
}
-110
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@@ -1,110 +0,0 @@
This is a quick description of the viterbi aka dynamic programing
algorthm.
Its reason for existence is that wikipedia has become very poor on
describing algorithms in a way that makes it useable for understanding
them or anything else actually. It tends now to describe the very same
algorithm under 50 different names and pages with few understandable
by even people who fully understand the algorithm and the theory behind.
Problem description: (that is what it can solve)
assume we have a 2d table, or you could call it a graph or matrix if you
prefer
O O O O O O O
O O O O O O O
O O O O O O O
O O O O O O O
That table has edges connecting points from each column to the next column
and each edge has a score like: (only some edge and scores shown to keep it
readable)
O--5--O-----O-----O-----O-----O
2 / 7 / \ / \ / \ /
\ / \ / \ / \ / \ /
O7-/--O--/--O--/--O--/--O--/--O
\/ \/ 1/ \/ \/ \/ \/ \/ \/ \/
/\ /\ 2\ /\ /\ /\ /\ /\ /\ /\
O3-/--O--/--O--/--O--/--O--/--O
/ \ / \ / \ / \ / \
1 \ 9 \ / \ / \ / \
O--2--O--1--O--5--O--3--O--8--O
Our goal is to find a path from left to right through it which
minimizes the sum of the score of all edges.
(and of course left/right is just a convention here it could be top down too)
Similarly the minimum could be the maximum by just fliping the sign,
Example of a path with scores:
O O O O O O O
>---O. O O .O-2-O O O
5. .7 .
O O-1-O O O 8 O O
.
O O O O O O-1-O---> (sum here is 24)
The viterbi algorthm now solves this simply column by column
For the previous column each point has a best path and a associated
score:
O-----5 O
\
\
O \ 1 O
\/
/\
O / 2 O
/
/
O-----2 O
To move one column forward we just need to find the best path and associated
scores for the next column
here are some edges we could choose from:
O-----5--3--O
\ \8
\ \
O \ 1--9--O
\/ \3
/\ \
O / 2--1--O
/ \2
/ \
O-----2--4--O
Finding the new best pathes and scores for each point of our new column is
trivial given we know the previous column best pathes and scores:
O-----0-----8
\
\
O \ 0----10
\/
/\
O / 0-----3
/ \
/ \
O 0 4
the viterbi algorthm continues exactly like this column for column until the
end and then just picks the path with the best score (above that would be the
one with score 3)
Author: Michael niedermayer
Copyright LGPL
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@@ -1,17 +0,0 @@
vcodec=libvpx
g=120
rc_lookahead=16
quality=good
speed=0
profile=1
qmax=51
qmin=11
slices=4
vb=2M
#ignored unless using -pass 2
maxrate=24M
minrate=100k
arnr_max_frames=7
arnr_strength=5
arnr_type=3
-17
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@@ -1,17 +0,0 @@
vcodec=libvpx
g=120
rc_lookahead=25
quality=good
speed=0
profile=1
qmax=51
qmin=11
slices=4
vb=2M
#ignored unless using -pass 2
maxrate=24M
minrate=100k
arnr_max_frames=7
arnr_strength=5
arnr_type=3
-16
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@@ -1,16 +0,0 @@
vcodec=libvpx
g=120
rc_lookahead=16
quality=good
speed=0
profile=0
qmax=63
qmin=0
vb=768k
#ignored unless using -pass 2
maxrate=1.5M
minrate=40k
arnr_max_frames=7
arnr_strength=5
arnr_type=3
-17
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@@ -1,17 +0,0 @@
vcodec=libvpx
g=120
rc_lookahead=16
quality=good
speed=0
profile=0
qmax=51
qmin=11
slices=4
vb=2M
#ignored unless using -pass 2
maxrate=24M
minrate=100k
arnr_max_frames=7
arnr_strength=5
arnr_type=3
-17
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@@ -1,17 +0,0 @@
vcodec=libvpx
g=120
rc_lookahead=25
quality=good
speed=0
profile=0
qmax=51
qmin=11
slices=4
vb=2M
#ignored unless using -pass 2
maxrate=24M
minrate=100k
arnr_max_frames=7
arnr_strength=5
arnr_type=3
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-938
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@@ -1,938 +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
* 4XM codec.
*/
#include "libavutil/intreadwrite.h"
#include "avcodec.h"
#include "dsputil.h"
#include "get_bits.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;
GetByteContext g;
GetByteContext g2;
int mv[256];
VLC pre_vlc;
int last_dc;
DECLARE_ALIGNED(16, DCTELEM, block)[6][64];
void *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){
static VLC_TYPE table[8][32][2];
int i;
for(i=0; i<8; i++){
block_type_vlc[0][i].table= table[i];
block_type_vlc[0][i].table_allocated= 32;
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, INIT_VLC_USE_NEW_STATIC);
}
}
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;
}
}
#if HAVE_BIGENDIAN
#define LE_CENTRIC_MUL(dst, src, scale, dc) \
{ \
unsigned tmpval = AV_RN32(src); \
tmpval = (tmpval << 16) | (tmpval >> 16); \
tmpval = tmpval * (scale) + (dc); \
tmpval = (tmpval << 16) | (tmpval >> 16); \
AV_WN32A(dst, tmpval); \
}
#else
#define LE_CENTRIC_MUL(dst, src, scale, dc) \
{ \
unsigned tmpval = AV_RN32(src) * (scale) + (dc); \
AV_WN32A(dst, tmpval); \
}
#endif
static inline void mcdc(uint16_t *dst, uint16_t *src, int log2w, int h, int stride, int scale, unsigned 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++){
LE_CENTRIC_MUL(dst, src, scale, dc);
if(scale) src += stride;
dst += stride;
}
break;
case 2:
for(i=0; i<h; i++){
LE_CENTRIC_MUL(dst, src, scale, dc);
LE_CENTRIC_MUL(dst + 2, src + 2, scale, dc);
if(scale) src += stride;
dst += stride;
}
break;
case 3:
for(i=0; i<h; i++){
LE_CENTRIC_MUL(dst, src, scale, dc);
LE_CENTRIC_MUL(dst + 2, src + 2, scale, dc);
LE_CENTRIC_MUL(dst + 4, src + 4, scale, dc);
LE_CENTRIC_MUL(dst + 6, src + 6, scale, 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->g.buffer_end - f->g.buffer < 1){
av_log(f->avctx, AV_LOG_ERROR, "bytestream overread\n");
return;
}
src += f->mv[ *f->g.buffer++ ];
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->g.buffer_end - f->g.buffer < 1){
av_log(f->avctx, AV_LOG_ERROR, "bytestream overread\n");
return;
}
src += f->mv[ *f->g.buffer++ ];
if(start > src || src > end){
av_log(f->avctx, AV_LOG_ERROR, "mv out of pic\n");
return;
}
if (f->g2.buffer_end - f->g2.buffer < 1){
av_log(f->avctx, AV_LOG_ERROR, "wordstream overread\n");
return;
}
mcdc(dst, src, log2w, h, stride, 1, bytestream2_get_le16(&f->g2));
}else if(code == 5){
if (f->g2.buffer_end - f->g2.buffer < 1){
av_log(f->avctx, AV_LOG_ERROR, "wordstream overread\n");
return;
}
mcdc(dst, src, log2w, h, stride, 0, bytestream2_get_le16(&f->g2));
}else if(code == 6){
if (f->g2.buffer_end - f->g2.buffer < 2){
av_log(f->avctx, AV_LOG_ERROR, "wordstream overread\n");
return;
}
if(log2w){
dst[0] = bytestream2_get_le16(&f->g2);
dst[1] = bytestream2_get_le16(&f->g2);
}else{
dst[0 ] = bytestream2_get_le16(&f->g2);
dst[stride] = bytestream2_get_le16(&f->g2);
}
}
}
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, bytestream_offset, wordstream_offset;
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;
}
av_fast_malloc(&f->bitstream_buffer, &f->bitstream_buffer_size, bitstream_size + FF_INPUT_BUFFER_PADDING_SIZE);
if (!f->bitstream_buffer)
return AVERROR(ENOMEM);
f->dsp.bswap_buf(f->bitstream_buffer, (const uint32_t*)(buf + extra), bitstream_size/4);
memset((uint8_t*)f->bitstream_buffer + bitstream_size, 0, FF_INPUT_BUFFER_PADDING_SIZE);
init_get_bits(&f->gb, f->bitstream_buffer, 8*bitstream_size);
wordstream_offset = extra + bitstream_size;
bytestream_offset = extra + bitstream_size + wordstream_size;
bytestream2_init(&f->g2, buf + wordstream_offset, length - wordstream_offset);
bytestream2_init(&f->g, buf + bytestream_offset, length - bytestream_offset);
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;
}
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;
if(get_bits_left(&f->gb) < 2){
av_log(f->avctx, AV_LOG_ERROR, "%d bits left before decode_i_block()\n", get_bits_left(&f->gb));
return -1;
}
/* 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;
}
if (init_vlc(&f->pre_vlc, ACDC_VLC_BITS, 257,
len_tab , 1, 1,
bits_tab, 4, 4, 0))
return NULL;
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;
const int mbs = (FFALIGN(width, 16) >> 4) * (FFALIGN(height, 16) >> 4);
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;
GetByteContext g3;
if(length < mbs * 8) {
av_log(f->avctx, AV_LOG_ERROR, "packet size too small\n");
return AVERROR_INVALIDDATA;
}
bytestream2_init(&g3, 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]= bytestream2_get_le16u(&g3);
color[1]= bytestream2_get_le16u(&g3);
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= bytestream2_get_le32u(&g3);
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 - x;
}
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;
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) {
av_log(f->avctx, AV_LOG_ERROR, "packet size too small\n");
return AVERROR_INVALIDDATA;
}
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;
av_fast_malloc(&f->bitstream_buffer, &f->bitstream_buffer_size, prestream_size + FF_INPUT_BUFFER_PADDING_SIZE);
if (!f->bitstream_buffer)
return AVERROR(ENOMEM);
f->dsp.bswap_buf(f->bitstream_buffer, (const uint32_t*)prestream, prestream_size/4);
memset((uint8_t*)f->bitstream_buffer + prestream_size, 0, FF_INPUT_BUFFER_PADDING_SIZE);
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);
}
}
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,
AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->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){
av_log(f->avctx, AV_LOG_ERROR, "sizes invalid\n");
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
p->reference= 3;
if (avctx->reget_buffer(avctx, p) < 0) {
av_log(avctx, AV_LOG_ERROR, "reget_buffer() failed\n");
return -1;
}
if(frame_4cc == AV_RL32("ifr2")){
p->pict_type= AV_PICTURE_TYPE_I;
if(decode_i2_frame(f, buf-4, frame_size + 4) < 0) {
av_log(f->avctx, AV_LOG_ERROR, "decode i2 frame failed\n");
return -1;
}
}else if(frame_4cc == AV_RL32("ifrm")){
p->pict_type= AV_PICTURE_TYPE_I;
if(decode_i_frame(f, buf, frame_size) < 0){
av_log(f->avctx, AV_LOG_ERROR, "decode i frame failed\n");
return -1;
}
}else if(frame_4cc == AV_RL32("pfrm") || frame_4cc == AV_RL32("pfr2")){
if(!f->last_picture.data[0]){
f->last_picture.reference= 3;
if(avctx->get_buffer(avctx, &f->last_picture) < 0){
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return -1;
}
}
p->pict_type= AV_PICTURE_TYPE_P;
if(decode_p_frame(f, buf, frame_size) < 0){
av_log(f->avctx, AV_LOG_ERROR, "decode p frame failed\n");
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 == AV_PICTURE_TYPE_I;
*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;
}
if((avctx->width % 16) || (avctx->height % 16)) {
av_log(avctx, AV_LOG_ERROR, "unsupported width/height\n");
return AVERROR_INVALIDDATA;
}
avcodec_get_frame_defaults(&f->current_picture);
avcodec_get_frame_defaults(&f->last_picture);
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_BGR555;
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);
if(f->current_picture.data[0])
avctx->release_buffer(avctx, &f->current_picture);
if(f->last_picture.data[0])
avctx->release_buffer(avctx, &f->last_picture);
return 0;
}
AVCodec ff_fourxm_decoder = {
.name = "4xm",
.type = AVMEDIA_TYPE_VIDEO,
.id = CODEC_ID_4XM,
.priv_data_size = sizeof(FourXContext),
.init = decode_init,
.close = decode_end,
.decode = decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("4X Movie"),
};
-227
View File
@@ -1,227 +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
* 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 alpha)
*
*/
#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];
uint32_t pal[256];
} EightBpsContext;
/*
*
* Decode a frame
*
*/
static int decode_frame(AVCodecContext *avctx, void *data, int *data_size, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->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 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);
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 = av_be2ne16(*(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 * planes > pixptr_end)
break;
if(dp + count > buf+buf_size) return -1;
while(count--) {
*pixptr = *dp++;
pixptr += planes;
}
} else {
count = 257 - count;
if (pixptr + count * planes > pixptr_end)
break;
while(count--) {
*pixptr = *dp;
pixptr += planes;
}
dp++;
dlen -= 2;
}
}
}
}
if (avctx->bits_per_coded_sample <= 8) {
const uint8_t *pal = av_packet_get_side_data(avpkt,
AV_PKT_DATA_PALETTE,
NULL);
if (pal) {
c->pic.palette_has_changed = 1;
memcpy(c->pal, pal, AVPALETTE_SIZE);
}
memcpy (c->pic.data[1], c->pal, AVPALETTE_SIZE);
}
*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;
avcodec_get_frame_defaults(&c->pic);
c->pic.data[0] = NULL;
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
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;
#if HAVE_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 ff_eightbps_decoder = {
.name = "8bps",
.type = AVMEDIA_TYPE_VIDEO,
.id = CODEC_ID_8BPS,
.priv_data_size = sizeof(EightBpsContext),
.init = decode_init,
.close = decode_end,
.decode = decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("QuickTime 8BPS video"),
};
-245
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@@ -1,245 +0,0 @@
/*
* Copyright (C) 2008 Jaikrishnan Menon
* Copyright (C) 2011 Stefano Sabatini
*
* 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
* 8svx audio decoder
* @author Jaikrishnan Menon
*
* supports: fibonacci delta encoding
* : exponential encoding
*
* For more information about the 8SVX format:
* http://netghost.narod.ru/gff/vendspec/iff/iff.txt
* http://sox.sourceforge.net/AudioFormats-11.html
* http://aminet.net/package/mus/misc/wavepak
* http://amigan.1emu.net/reg/8SVX.txt
*
* Samples can be found here:
* http://aminet.net/mods/smpl/
*/
#include "avcodec.h"
/** decoder context */
typedef struct EightSvxContext {
AVFrame frame;
const int8_t *table;
/* buffer used to store the whole audio decoded/interleaved chunk,
* which is sent with the first packet */
uint8_t *samples;
size_t samples_size;
int samples_idx;
} EightSvxContext;
static const int8_t fibonacci[16] = { -34, -21, -13, -8, -5, -3, -2, -1, 0, 1, 2, 3, 5, 8, 13, 21 };
static const int8_t exponential[16] = { -128, -64, -32, -16, -8, -4, -2, -1, 0, 1, 2, 4, 8, 16, 32, 64 };
#define MAX_FRAME_SIZE 2048
/**
* Interleave samples in buffer containing all left channel samples
* at the beginning, and right channel samples at the end.
* Each sample is assumed to be in signed 8-bit format.
*
* @param size the size in bytes of the dst and src buffer
*/
static void interleave_stereo(uint8_t *dst, const uint8_t *src, int size)
{
uint8_t *dst_end = dst + size;
size = size>>1;
while (dst < dst_end) {
*dst++ = *src;
*dst++ = *(src+size);
src++;
}
}
/**
* Delta decode the compressed values in src, and put the resulting
* decoded n samples in dst.
*
* @param val starting value assumed by the delta sequence
* @param table delta sequence table
* @return size in bytes of the decoded data, must be src_size*2
*/
static int delta_decode(int8_t *dst, const uint8_t *src, int src_size,
int8_t val, const int8_t *table)
{
int n = src_size;
int8_t *dst0 = dst;
while (n--) {
uint8_t d = *src++;
val = av_clip(val + table[d & 0x0f], -127, 128);
*dst++ = val;
val = av_clip(val + table[d >> 4] , -127, 128);
*dst++ = val;
}
return dst-dst0;
}
/** decode a frame */
static int eightsvx_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
EightSvxContext *esc = avctx->priv_data;
int n, out_data_size, ret;
uint8_t *src, *dst;
/* decode and interleave the first packet */
if (!esc->samples && avpkt) {
uint8_t *deinterleaved_samples, *p = NULL;
esc->samples_size = avctx->codec->id == CODEC_ID_8SVX_RAW || avctx->codec->id ==CODEC_ID_PCM_S8_PLANAR?
avpkt->size : avctx->channels + (avpkt->size-avctx->channels) * 2;
if (!(esc->samples = av_malloc(esc->samples_size)))
return AVERROR(ENOMEM);
/* decompress */
if (avctx->codec->id == CODEC_ID_8SVX_FIB || avctx->codec->id == CODEC_ID_8SVX_EXP) {
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
int n = esc->samples_size;
if (buf_size < 2) {
av_log(avctx, AV_LOG_ERROR, "packet size is too small\n");
return AVERROR(EINVAL);
}
if (!(deinterleaved_samples = av_mallocz(n)))
return AVERROR(ENOMEM);
p = deinterleaved_samples;
/* the uncompressed starting value is contained in the first byte */
if (avctx->channels == 2) {
delta_decode(deinterleaved_samples , buf+1, buf_size/2-1, buf[0], esc->table);
buf += buf_size/2;
delta_decode(deinterleaved_samples+n/2-1, buf+1, buf_size/2-1, buf[0], esc->table);
} else
delta_decode(deinterleaved_samples , buf+1, buf_size-1 , buf[0], esc->table);
} else {
deinterleaved_samples = avpkt->data;
}
if (avctx->channels == 2)
interleave_stereo(esc->samples, deinterleaved_samples, esc->samples_size);
else
memcpy(esc->samples, deinterleaved_samples, esc->samples_size);
av_freep(&p);
}
/* get output buffer */
esc->frame.nb_samples = (FFMIN(MAX_FRAME_SIZE, esc->samples_size - esc->samples_idx) +avctx->channels-1) / avctx->channels;
if ((ret = avctx->get_buffer(avctx, &esc->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
*got_frame_ptr = 1;
*(AVFrame *)data = esc->frame;
dst = esc->frame.data[0];
src = esc->samples + esc->samples_idx;
out_data_size = esc->frame.nb_samples * avctx->channels;
for (n = out_data_size; n > 0; n--)
*dst++ = *src++ + 128;
esc->samples_idx += out_data_size;
return avctx->codec->id == CODEC_ID_8SVX_FIB || avctx->codec->id == CODEC_ID_8SVX_EXP ?
(avctx->frame_number == 0)*2 + out_data_size / 2 :
out_data_size;
}
static av_cold int eightsvx_decode_init(AVCodecContext *avctx)
{
EightSvxContext *esc = avctx->priv_data;
if (avctx->channels < 1 || avctx->channels > 2) {
av_log(avctx, AV_LOG_ERROR, "8SVX does not support more than 2 channels\n");
return AVERROR_INVALIDDATA;
}
switch (avctx->codec->id) {
case CODEC_ID_8SVX_FIB: esc->table = fibonacci; break;
case CODEC_ID_8SVX_EXP: esc->table = exponential; break;
case CODEC_ID_PCM_S8_PLANAR:
case CODEC_ID_8SVX_RAW: esc->table = NULL; break;
default:
av_log(avctx, AV_LOG_ERROR, "Invalid codec id %d.\n", avctx->codec->id);
return AVERROR_INVALIDDATA;
}
avctx->sample_fmt = AV_SAMPLE_FMT_U8;
avcodec_get_frame_defaults(&esc->frame);
avctx->coded_frame = &esc->frame;
return 0;
}
static av_cold int eightsvx_decode_close(AVCodecContext *avctx)
{
EightSvxContext *esc = avctx->priv_data;
av_freep(&esc->samples);
esc->samples_size = 0;
esc->samples_idx = 0;
return 0;
}
AVCodec ff_eightsvx_fib_decoder = {
.name = "8svx_fib",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_8SVX_FIB,
.priv_data_size = sizeof (EightSvxContext),
.init = eightsvx_decode_init,
.decode = eightsvx_decode_frame,
.close = eightsvx_decode_close,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("8SVX fibonacci"),
};
AVCodec ff_eightsvx_exp_decoder = {
.name = "8svx_exp",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_8SVX_EXP,
.priv_data_size = sizeof (EightSvxContext),
.init = eightsvx_decode_init,
.decode = eightsvx_decode_frame,
.close = eightsvx_decode_close,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("8SVX exponential"),
};
AVCodec ff_pcm_s8_planar_decoder = {
.name = "pcm_s8_planar",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_PCM_S8_PLANAR,
.priv_data_size = sizeof(EightSvxContext),
.init = eightsvx_decode_init,
.close = eightsvx_decode_close,
.decode = eightsvx_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("PCM signed 8-bit planar"),
};
-791
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@@ -1,791 +0,0 @@
include $(SUBDIR)../config.mak
NAME = avcodec
FFLIBS = avutil
HEADERS = avcodec.h avfft.h dxva2.h opt.h vaapi.h vda.h vdpau.h version.h xvmc.h
OBJS = allcodecs.o \
audioconvert.o \
avpacket.o \
bitstream.o \
bitstream_filter.o \
dsputil.o \
faanidct.o \
fmtconvert.o \
imgconvert.o \
jrevdct.o \
options.o \
parser.o \
raw.o \
rawdec.o \
resample.o \
resample2.o \
simple_idct.o \
utils.o \
# parts needed for many different codecs
OBJS-$(CONFIG_AANDCT) += aandcttab.o
OBJS-$(CONFIG_AC3DSP) += ac3dsp.o
OBJS-$(CONFIG_CRYSTALHD) += crystalhd.o
OBJS-$(CONFIG_ENCODERS) += faandct.o jfdctfst.o jfdctint.o
OBJS-$(CONFIG_DCT) += dct.o dct32_fixed.o dct32_float.o
OBJS-$(CONFIG_DWT) += dwt.o
OBJS-$(CONFIG_DXVA2) += dxva2.o
FFT-OBJS-$(CONFIG_HARDCODED_TABLES) += cos_tables.o cos_fixed_tables.o
OBJS-$(CONFIG_FFT) += avfft.o fft_fixed.o fft_float.o \
$(FFT-OBJS-yes)
OBJS-$(CONFIG_GOLOMB) += golomb.o
OBJS-$(CONFIG_H264DSP) += h264dsp.o h264idct.o
OBJS-$(CONFIG_H264PRED) += h264pred.o
OBJS-$(CONFIG_HUFFMAN) += huffman.o
OBJS-$(CONFIG_LPC) += lpc.o
OBJS-$(CONFIG_LSP) += lsp.o
OBJS-$(CONFIG_MDCT) += mdct_fixed.o mdct_float.o
OBJS-$(CONFIG_MPEGAUDIODSP) += mpegaudiodsp.o \
mpegaudiodsp_fixed.o \
mpegaudiodsp_float.o
RDFT-OBJS-$(CONFIG_HARDCODED_TABLES) += sin_tables.o
OBJS-$(CONFIG_RDFT) += rdft.o $(RDFT-OBJS-yes)
OBJS-$(CONFIG_SINEWIN) += sinewin.o
OBJS-$(CONFIG_VAAPI) += vaapi.o
OBJS-$(CONFIG_VDA) += vda.o
OBJS-$(CONFIG_VDPAU) += vdpau.o
# decoders/encoders/hardware accelerators
OBJS-$(CONFIG_A64MULTI_ENCODER) += a64multienc.o elbg.o
OBJS-$(CONFIG_A64MULTI5_ENCODER) += a64multienc.o elbg.o
OBJS-$(CONFIG_AAC_DECODER) += aacdec.o aactab.o aacsbr.o aacps.o \
aacadtsdec.o mpeg4audio.o kbdwin.o
OBJS-$(CONFIG_AAC_ENCODER) += aacenc.o aaccoder.o \
aacpsy.o aactab.o \
psymodel.o iirfilter.o \
mpeg4audio.o kbdwin.o
OBJS-$(CONFIG_AASC_DECODER) += aasc.o msrledec.o
OBJS-$(CONFIG_AC3_DECODER) += ac3dec.o ac3dec_data.o ac3.o kbdwin.o
OBJS-$(CONFIG_AC3_ENCODER) += ac3enc_float.o ac3enc.o ac3tab.o \
ac3.o kbdwin.o
OBJS-$(CONFIG_AC3_FIXED_ENCODER) += ac3enc_fixed.o ac3enc.o ac3tab.o ac3.o
OBJS-$(CONFIG_ALAC_DECODER) += alac.o
OBJS-$(CONFIG_ALAC_ENCODER) += alacenc.o
OBJS-$(CONFIG_ALS_DECODER) += alsdec.o bgmc.o mpeg4audio.o
OBJS-$(CONFIG_AMRNB_DECODER) += amrnbdec.o celp_filters.o \
celp_math.o acelp_filters.o \
acelp_vectors.o \
acelp_pitch_delay.o
OBJS-$(CONFIG_AMRWB_DECODER) += amrwbdec.o celp_filters.o \
celp_math.o acelp_filters.o \
acelp_vectors.o \
acelp_pitch_delay.o lsp.o
OBJS-$(CONFIG_AMV_DECODER) += sp5xdec.o mjpegdec.o mjpeg.o
OBJS-$(CONFIG_ANM_DECODER) += anm.o
OBJS-$(CONFIG_ANSI_DECODER) += ansi.o cga_data.o
OBJS-$(CONFIG_APE_DECODER) += apedec.o
OBJS-$(CONFIG_ASS_DECODER) += assdec.o ass.o ass_split.o
OBJS-$(CONFIG_ASS_ENCODER) += assenc.o ass.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-$(CONFIG_ATRAC1_DECODER) += atrac1.o atrac.o
OBJS-$(CONFIG_ATRAC3_DECODER) += atrac3.o atrac.o
OBJS-$(CONFIG_AURA_DECODER) += cyuv.o
OBJS-$(CONFIG_AURA2_DECODER) += aura.o
OBJS-$(CONFIG_AVRP_DECODER) += r210dec.o
OBJS-$(CONFIG_AVRP_ENCODER) += r210enc.o
OBJS-$(CONFIG_AVS_DECODER) += avs.o
OBJS-$(CONFIG_BETHSOFTVID_DECODER) += bethsoftvideo.o
OBJS-$(CONFIG_BFI_DECODER) += bfi.o
OBJS-$(CONFIG_BINK_DECODER) += bink.o binkdsp.o
OBJS-$(CONFIG_BINKAUDIO_DCT_DECODER) += binkaudio.o wma.o
OBJS-$(CONFIG_BINKAUDIO_RDFT_DECODER) += binkaudio.o wma.o
OBJS-$(CONFIG_BINTEXT_DECODER) += bintext.o cga_data.o
OBJS-$(CONFIG_BMP_DECODER) += bmp.o msrledec.o
OBJS-$(CONFIG_BMP_ENCODER) += bmpenc.o
OBJS-$(CONFIG_BMV_VIDEO_DECODER) += bmv.o
OBJS-$(CONFIG_BMV_AUDIO_DECODER) += bmv.o
OBJS-$(CONFIG_C93_DECODER) += c93.o
OBJS-$(CONFIG_CAVS_DECODER) += cavs.o cavsdec.o cavsdsp.o \
mpeg12data.o mpegvideo.o
OBJS-$(CONFIG_CDGRAPHICS_DECODER) += cdgraphics.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-$(CONFIG_DCA_DECODER) += dca.o synth_filter.o dcadsp.o
OBJS-$(CONFIG_DCA_ENCODER) += dcaenc.o
OBJS-$(CONFIG_DIRAC_DECODER) += diracdec.o dirac.o diracdsp.o \
dirac_arith.o mpeg12data.o dwt.o
OBJS-$(CONFIG_DFA_DECODER) += dfa.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-$(CONFIG_DPX_DECODER) += dpx.o
OBJS-$(CONFIG_DPX_ENCODER) += dpxenc.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-$(CONFIG_DVVIDEO_DECODER) += dv.o dvdata.o
OBJS-$(CONFIG_DVVIDEO_ENCODER) += dv.o dvdata.o
OBJS-$(CONFIG_DXA_DECODER) += dxa.o
OBJS-$(CONFIG_DXTORY_DECODER) += dxtory.o
OBJS-$(CONFIG_EAC3_DECODER) += eac3dec.o eac3_data.o
OBJS-$(CONFIG_EAC3_ENCODER) += eac3enc.o ac3enc.o ac3enc_float.o \
ac3tab.o ac3.o kbdwin.o eac3_data.o
OBJS-$(CONFIG_EACMV_DECODER) += eacmv.o
OBJS-$(CONFIG_EAMAD_DECODER) += eamad.o eaidct.o mpeg12.o \
mpeg12data.o mpegvideo.o \
error_resilience.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-$(CONFIG_EIGHTSVX_RAW_DECODER) += 8svx.o
OBJS-$(CONFIG_ESCAPE124_DECODER) += escape124.o
OBJS-$(CONFIG_ESCAPE130_DECODER) += escape130.o
OBJS-$(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_FFWAVESYNTH_DECODER) += ffwavesynth.o
OBJS-$(CONFIG_FLAC_DECODER) += flacdec.o flacdata.o flac.o vorbis_data.o
OBJS-$(CONFIG_FLAC_ENCODER) += flacenc.o flacdata.o flac.o vorbis_data.o
OBJS-$(CONFIG_FLASHSV_DECODER) += flashsv.o
OBJS-$(CONFIG_FLASHSV_ENCODER) += flashsvenc.o
OBJS-$(CONFIG_FLASHSV2_ENCODER) += flashsv2enc.o
OBJS-$(CONFIG_FLASHSV2_DECODER) += flashsv.o
OBJS-$(CONFIG_FLIC_DECODER) += flicvideo.o
OBJS-$(CONFIG_FOURXM_DECODER) += 4xm.o
OBJS-$(CONFIG_FRAPS_DECODER) += fraps.o
OBJS-$(CONFIG_FRWU_DECODER) += frwu.o
OBJS-$(CONFIG_G723_1_DECODER) += g723_1.o acelp_vectors.o \
celp_filters.o celp_math.o
OBJS-$(CONFIG_G723_1_ENCODER) += g723_1.o
OBJS-$(CONFIG_G729_DECODER) += g729dec.o lsp.o celp_math.o acelp_filters.o acelp_pitch_delay.o acelp_vectors.o g729postfilter.o
OBJS-$(CONFIG_GIF_DECODER) += gifdec.o lzw.o
OBJS-$(CONFIG_GIF_ENCODER) += gif.o lzwenc.o
OBJS-$(CONFIG_GSM_DECODER) += gsmdec.o gsmdec_data.o msgsmdec.o
OBJS-$(CONFIG_GSM_MS_DECODER) += gsmdec.o gsmdec_data.o msgsmdec.o
OBJS-$(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 ituh263dec.o \
mpeg4video.o mpeg4videodec.o flvdec.o\
intelh263dec.o mpegvideo.o \
error_resilience.o
OBJS-$(CONFIG_H263_VAAPI_HWACCEL) += vaapi_mpeg4.o
OBJS-$(CONFIG_H263_ENCODER) += mpegvideo_enc.o mpeg4video.o \
mpeg4videoenc.o motion_est.o \
ratecontrol.o h263.o ituh263enc.o \
flvenc.o mpeg12data.o \
mpegvideo.o error_resilience.o
OBJS-$(CONFIG_H264_DECODER) += h264.o \
h264_loopfilter.o h264_direct.o \
cabac.o h264_sei.o h264_ps.o \
h264_refs.o h264_cavlc.o h264_cabac.o\
mpegvideo.o error_resilience.o
OBJS-$(CONFIG_H264_DXVA2_HWACCEL) += dxva2_h264.o
OBJS-$(CONFIG_H264_VAAPI_HWACCEL) += vaapi_h264.o
OBJS-$(CONFIG_H264_VDA_HWACCEL) += vda_h264.o
OBJS-$(CONFIG_HUFFYUV_DECODER) += huffyuv.o
OBJS-$(CONFIG_HUFFYUV_ENCODER) += huffyuv.o
OBJS-$(CONFIG_IDCIN_DECODER) += idcinvideo.o
OBJS-$(CONFIG_IDF_DECODER) += bintext.o cga_data.o
OBJS-$(CONFIG_IFF_BYTERUN1_DECODER) += iff.o
OBJS-$(CONFIG_IFF_ILBM_DECODER) += iff.o
OBJS-$(CONFIG_IMC_DECODER) += imc.o
OBJS-$(CONFIG_INDEO2_DECODER) += indeo2.o
OBJS-$(CONFIG_INDEO3_DECODER) += indeo3.o
OBJS-$(CONFIG_INDEO4_DECODER) += indeo4.o ivi_common.o ivi_dsp.o
OBJS-$(CONFIG_INDEO5_DECODER) += indeo5.o ivi_common.o ivi_dsp.o
OBJS-$(CONFIG_INTERPLAY_DPCM_DECODER) += dpcm.o
OBJS-$(CONFIG_INTERPLAY_VIDEO_DECODER) += interplayvideo.o
OBJS-$(CONFIG_JPEG2000_DECODER) += j2kdec.o mqcdec.o mqc.o j2k.o j2k_dwt.o
OBJS-$(CONFIG_JPEG2000_ENCODER) += j2kenc.o mqcenc.o mqc.o j2k.o j2k_dwt.o
OBJS-$(CONFIG_JPEGLS_DECODER) += jpeglsdec.o jpegls.o \
mjpegdec.o mjpeg.o
OBJS-$(CONFIG_JPEGLS_ENCODER) += jpeglsenc.o jpegls.o
OBJS-$(CONFIG_JV_DECODER) += jvdec.o
OBJS-$(CONFIG_KGV1_DECODER) += kgv1dec.o
OBJS-$(CONFIG_KMVC_DECODER) += kmvc.o
OBJS-$(CONFIG_LAGARITH_DECODER) += lagarith.o lagarithrac.o
OBJS-$(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-$(CONFIG_MLP_DECODER) += mlpdec.o mlpdsp.o
OBJS-$(CONFIG_MMVIDEO_DECODER) += mmvideo.o
OBJS-$(CONFIG_MOTIONPIXELS_DECODER) += motionpixels.o
OBJS-$(CONFIG_MP1_DECODER) += mpegaudiodec.o mpegaudiodecheader.o \
mpegaudio.o mpegaudiodata.o
OBJS-$(CONFIG_MP1FLOAT_DECODER) += mpegaudiodec_float.o mpegaudiodecheader.o \
mpegaudio.o mpegaudiodata.o
OBJS-$(CONFIG_MP2_DECODER) += mpegaudiodec.o mpegaudiodecheader.o \
mpegaudio.o mpegaudiodata.o
OBJS-$(CONFIG_MP2_ENCODER) += mpegaudioenc.o mpegaudio.o \
mpegaudiodata.o
OBJS-$(CONFIG_MP2FLOAT_DECODER) += mpegaudiodec_float.o mpegaudiodecheader.o \
mpegaudio.o mpegaudiodata.o
OBJS-$(CONFIG_MP3ADU_DECODER) += mpegaudiodec.o mpegaudiodecheader.o \
mpegaudio.o mpegaudiodata.o
OBJS-$(CONFIG_MP3ADUFLOAT_DECODER) += mpegaudiodec_float.o mpegaudiodecheader.o \
mpegaudio.o mpegaudiodata.o
OBJS-$(CONFIG_MP3ON4_DECODER) += mpegaudiodec.o mpegaudiodecheader.o \
mpegaudio.o mpegaudiodata.o \
mpeg4audio.o
OBJS-$(CONFIG_MP3ON4FLOAT_DECODER) += mpegaudiodec_float.o mpegaudiodecheader.o \
mpegaudio.o mpegaudiodata.o \
mpeg4audio.o
OBJS-$(CONFIG_MP3_DECODER) += mpegaudiodec.o mpegaudiodecheader.o \
mpegaudio.o mpegaudiodata.o
OBJS-$(CONFIG_MP3FLOAT_DECODER) += mpegaudiodec_float.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-$(CONFIG_MPEGVIDEO_DECODER) += mpeg12.o mpeg12data.o \
mpegvideo.o error_resilience.o
OBJS-$(CONFIG_MPEG_XVMC_DECODER) += mpegvideo_xvmc.o
OBJS-$(CONFIG_MPEG1VIDEO_DECODER) += mpeg12.o mpeg12data.o \
mpegvideo.o error_resilience.o
OBJS-$(CONFIG_MPEG1VIDEO_ENCODER) += mpeg12enc.o mpegvideo_enc.o \
timecode.o \
motion_est.o ratecontrol.o \
mpeg12.o mpeg12data.o \
mpegvideo.o error_resilience.o
OBJS-$(CONFIG_MPEG2_DXVA2_HWACCEL) += dxva2_mpeg2.o
OBJS-$(CONFIG_MPEG2_VAAPI_HWACCEL) += vaapi_mpeg2.o
OBJS-$(CONFIG_MPEG2VIDEO_DECODER) += mpeg12.o mpeg12data.o \
mpegvideo.o error_resilience.o
OBJS-$(CONFIG_MPEG2VIDEO_ENCODER) += mpeg12enc.o mpegvideo_enc.o \
timecode.o \
motion_est.o ratecontrol.o \
mpeg12.o mpeg12data.o \
mpegvideo.o error_resilience.o
OBJS-$(CONFIG_MPEG4_VAAPI_HWACCEL) += vaapi_mpeg4.o
OBJS-$(CONFIG_MSMPEG4V1_DECODER) += msmpeg4.o msmpeg4data.o
OBJS-$(CONFIG_MSMPEG4V2_DECODER) += msmpeg4.o msmpeg4data.o h263dec.o \
h263.o ituh263dec.o mpeg4videodec.o
OBJS-$(CONFIG_MSMPEG4V2_ENCODER) += msmpeg4.o msmpeg4data.o h263dec.o \
h263.o ituh263dec.o mpeg4videodec.o
OBJS-$(CONFIG_MSMPEG4V3_DECODER) += msmpeg4.o msmpeg4data.o h263dec.o \
h263.o ituh263dec.o mpeg4videodec.o
OBJS-$(CONFIG_MSMPEG4V3_ENCODER) += msmpeg4.o msmpeg4data.o h263dec.o \
h263.o ituh263dec.o mpeg4videodec.o
OBJS-$(CONFIG_MSRLE_DECODER) += msrle.o msrledec.o
OBJS-$(CONFIG_MSVIDEO1_DECODER) += msvideo1.o
OBJS-$(CONFIG_MSVIDEO1_ENCODER) += msvideo1enc.o elbg.o
OBJS-$(CONFIG_MSZH_DECODER) += lcldec.o
OBJS-$(CONFIG_MXPEG_DECODER) += mxpegdec.o mjpegdec.o mjpeg.o
OBJS-$(CONFIG_NELLYMOSER_DECODER) += nellymoserdec.o nellymoser.o
OBJS-$(CONFIG_NELLYMOSER_ENCODER) += nellymoserenc.o nellymoser.o
OBJS-$(CONFIG_NUV_DECODER) += nuv.o rtjpeg.o
OBJS-$(CONFIG_PAM_DECODER) += pnmdec.o pnm.o
OBJS-$(CONFIG_PAM_ENCODER) += pamenc.o pnm.o
OBJS-$(CONFIG_PBM_DECODER) += pnmdec.o pnm.o
OBJS-$(CONFIG_PBM_ENCODER) += pnmenc.o pnm.o
OBJS-$(CONFIG_PCX_DECODER) += pcx.o
OBJS-$(CONFIG_PCX_ENCODER) += pcxenc.o
OBJS-$(CONFIG_PGM_DECODER) += pnmdec.o pnm.o
OBJS-$(CONFIG_PGM_ENCODER) += pnmenc.o pnm.o
OBJS-$(CONFIG_PGMYUV_DECODER) += pnmdec.o pnm.o
OBJS-$(CONFIG_PGMYUV_ENCODER) += pnmenc.o pnm.o
OBJS-$(CONFIG_PGSSUB_DECODER) += pgssubdec.o
OBJS-$(CONFIG_PICTOR_DECODER) += pictordec.o cga_data.o
OBJS-$(CONFIG_PNG_DECODER) += png.o pngdec.o
OBJS-$(CONFIG_PNG_ENCODER) += png.o pngenc.o
OBJS-$(CONFIG_PPM_DECODER) += pnmdec.o pnm.o
OBJS-$(CONFIG_PPM_ENCODER) += pnmenc.o pnm.o
OBJS-$(CONFIG_PRORES_DECODER) += proresdec2.o
OBJS-$(CONFIG_PRORES_LGPL_DECODER) += proresdec_lgpl.o proresdsp.o
OBJS-$(CONFIG_PRORES_ENCODER) += proresenc.o
OBJS-$(CONFIG_PTX_DECODER) += ptx.o
OBJS-$(CONFIG_QCELP_DECODER) += qcelpdec.o celp_math.o \
celp_filters.o acelp_vectors.o \
acelp_filters.o
OBJS-$(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-$(CONFIG_R10K_DECODER) += r210dec.o
OBJS-$(CONFIG_R10K_ENCODER) += r210enc.o
OBJS-$(CONFIG_R210_DECODER) += r210dec.o
OBJS-$(CONFIG_R210_ENCODER) += r210enc.o
OBJS-$(CONFIG_RA_144_DECODER) += ra144dec.o ra144.o celp_filters.o
OBJS-$(CONFIG_RA_144_ENCODER) += ra144enc.o 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
OBJS-$(CONFIG_RV10_ENCODER) += rv10enc.o
OBJS-$(CONFIG_RV20_DECODER) += rv10.o
OBJS-$(CONFIG_RV20_ENCODER) += rv20enc.o
OBJS-$(CONFIG_RV30_DECODER) += rv30.o rv34.o rv30dsp.o rv34dsp.o \
mpegvideo.o error_resilience.o
OBJS-$(CONFIG_RV40_DECODER) += rv40.o rv34.o rv34dsp.o rv40dsp.o \
mpegvideo.o error_resilience.o
OBJS-$(CONFIG_S302M_DECODER) += s302m.o
OBJS-$(CONFIG_SGI_DECODER) += sgidec.o
OBJS-$(CONFIG_SGI_ENCODER) += sgienc.o rle.o
OBJS-$(CONFIG_SHORTEN_DECODER) += shorten.o
OBJS-$(CONFIG_SIPR_DECODER) += sipr.o acelp_pitch_delay.o \
celp_math.o acelp_vectors.o \
acelp_filters.o celp_filters.o \
sipr16k.o
OBJS-$(CONFIG_SMACKAUD_DECODER) += smacker.o
OBJS-$(CONFIG_SMACKER_DECODER) += smacker.o
OBJS-$(CONFIG_SMC_DECODER) += smc.o
OBJS-$(CONFIG_SNOW_DECODER) += snowdec.o snow.o rangecoder.o
OBJS-$(CONFIG_SNOW_ENCODER) += snowenc.o snow.o rangecoder.o \
motion_est.o ratecontrol.o \
h263.o mpegvideo.o \
error_resilience.o ituh263enc.o \
mpegvideo_enc.o mpeg12data.o
OBJS-$(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-$(CONFIG_SRT_DECODER) += srtdec.o ass.o
OBJS-$(CONFIG_SRT_ENCODER) += srtenc.o ass_split.o
OBJS-$(CONFIG_SUNRAST_DECODER) += sunrast.o
OBJS-$(CONFIG_SVQ1_DECODER) += svq1dec.o svq1.o h263.o \
mpegvideo.o error_resilience.o
OBJS-$(CONFIG_SVQ1_ENCODER) += svq1enc.o svq1.o \
motion_est.o h263.o \
mpegvideo.o error_resilience.o \
ituh263enc.o mpegvideo_enc.o \
ratecontrol.o mpeg12data.o
OBJS-$(CONFIG_SVQ3_DECODER) += h264.o svq3.o \
h264_loopfilter.o h264_direct.o \
h264_sei.o h264_ps.o h264_refs.o \
h264_cavlc.o h264_cabac.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-$(CONFIG_THEORA_DECODER) += xiph.o
OBJS-$(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_TMV_DECODER) += tmv.o cga_data.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_TWINVQ_DECODER) += twinvq.o celp_math.o
OBJS-$(CONFIG_TXD_DECODER) += txd.o s3tc.o
OBJS-$(CONFIG_ULTI_DECODER) += ulti.o
OBJS-$(CONFIG_UTVIDEO_DECODER) += utvideo.o
OBJS-$(CONFIG_V210_DECODER) += v210dec.o
OBJS-$(CONFIG_V210_ENCODER) += v210enc.o
OBJS-$(CONFIG_V308_DECODER) += v308dec.o
OBJS-$(CONFIG_V308_ENCODER) += v308enc.o
OBJS-$(CONFIG_V410_DECODER) += v410dec.o
OBJS-$(CONFIG_V410_ENCODER) += v410enc.o
OBJS-$(CONFIG_V210X_DECODER) += v210x.o
OBJS-$(CONFIG_VB_DECODER) += vb.o
OBJS-$(CONFIG_VBLE_DECODER) += vble.o
OBJS-$(CONFIG_VC1_DECODER) += vc1dec.o vc1.o vc1data.o vc1dsp.o \
msmpeg4.o msmpeg4data.o \
intrax8.o intrax8dsp.o
OBJS-$(CONFIG_VC1_DXVA2_HWACCEL) += dxva2_vc1.o
OBJS-$(CONFIG_VC1_VAAPI_HWACCEL) += vaapi_vc1.o
OBJS-$(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-$(CONFIG_VORBIS_DECODER) += vorbisdec.o vorbis.o \
vorbis_data.o xiph.o
OBJS-$(CONFIG_VORBIS_ENCODER) += vorbisenc.o vorbis.o \
vorbis_data.o
OBJS-$(CONFIG_VP3_DECODER) += vp3.o vp3dsp.o
OBJS-$(CONFIG_VP5_DECODER) += vp5.o vp56.o vp56data.o vp56dsp.o \
vp3dsp.o vp56rac.o
OBJS-$(CONFIG_VP6_DECODER) += vp6.o vp56.o vp56data.o vp56dsp.o \
vp3dsp.o vp6dsp.o vp56rac.o
OBJS-$(CONFIG_VP8_DECODER) += vp8.o vp8dsp.o vp56rac.o
OBJS-$(CONFIG_VQA_DECODER) += vqavideo.o
OBJS-$(CONFIG_WAVPACK_DECODER) += wavpack.o
OBJS-$(CONFIG_WMALOSSLESS_DECODER) += wmalosslessdec.o wma.o
OBJS-$(CONFIG_WMAPRO_DECODER) += wmaprodec.o wma.o
OBJS-$(CONFIG_WMAV1_DECODER) += wmadec.o wma.o aactab.o
OBJS-$(CONFIG_WMAV1_ENCODER) += wmaenc.o wma.o aactab.o
OBJS-$(CONFIG_WMAV2_DECODER) += wmadec.o wma.o aactab.o
OBJS-$(CONFIG_WMAV2_ENCODER) += wmaenc.o wma.o aactab.o
OBJS-$(CONFIG_WMAVOICE_DECODER) += wmavoice.o \
celp_math.o celp_filters.o \
acelp_vectors.o acelp_filters.o
OBJS-$(CONFIG_WMV1_DECODER) += msmpeg4.o msmpeg4data.o
OBJS-$(CONFIG_WMV2_DECODER) += wmv2dec.o wmv2.o \
msmpeg4.o msmpeg4data.o \
intrax8.o intrax8dsp.o
OBJS-$(CONFIG_WMV2_ENCODER) += wmv2enc.o wmv2.o \
msmpeg4.o msmpeg4data.o \
mpeg4videodec.o ituh263dec.o h263dec.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-$(CONFIG_XAN_WC4_DECODER) += xxan.o
OBJS-$(CONFIG_XBIN_DECODER) += bintext.o cga_data.o
OBJS-$(CONFIG_XL_DECODER) += xl.o
OBJS-$(CONFIG_XSUB_DECODER) += xsubdec.o
OBJS-$(CONFIG_XSUB_ENCODER) += xsubenc.o
OBJS-$(CONFIG_XWD_DECODER) += xwddec.o
OBJS-$(CONFIG_XWD_ENCODER) += xwdenc.o
OBJS-$(CONFIG_Y41P_DECODER) += y41pdec.o
OBJS-$(CONFIG_Y41P_ENCODER) += y41penc.o
OBJS-$(CONFIG_YOP_DECODER) += yop.o
OBJS-$(CONFIG_YUV4_DECODER) += yuv4dec.o
OBJS-$(CONFIG_YUV4_ENCODER) += yuv4enc.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-$(CONFIG_PCM_BLURAY_DECODER) += pcm-mpeg.o
OBJS-$(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_LXF_DECODER) += 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-$(CONFIG_PCM_S8_PLANAR_DECODER) += 8svx.o
OBJS-$(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_ADPCM_4XM_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_ADX_DECODER) += adxdec.o adx.o
OBJS-$(CONFIG_ADPCM_ADX_ENCODER) += adxenc.o adx.o
OBJS-$(CONFIG_ADPCM_CT_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_EA_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_EA_MAXIS_XA_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_EA_R1_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_EA_R2_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_EA_R3_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_EA_XAS_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_G722_DECODER) += g722.o g722dec.o
OBJS-$(CONFIG_ADPCM_G722_ENCODER) += g722.o g722enc.o
OBJS-$(CONFIG_ADPCM_G726_DECODER) += g726.o
OBJS-$(CONFIG_ADPCM_G726_ENCODER) += g726.o
OBJS-$(CONFIG_ADPCM_IMA_AMV_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_IMA_APC_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_IMA_DK3_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_IMA_DK4_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_IMA_EA_EACS_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_IMA_EA_SEAD_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_IMA_ISS_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_IMA_QT_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_IMA_QT_ENCODER) += adpcmenc.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_IMA_SMJPEG_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_IMA_WAV_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_IMA_WAV_ENCODER) += adpcmenc.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_IMA_WS_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_MS_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_MS_ENCODER) += adpcmenc.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_SBPRO_2_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_SBPRO_3_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_SBPRO_4_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_SWF_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_SWF_ENCODER) += adpcmenc.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_THP_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_XA_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_YAMAHA_DECODER) += adpcm.o adpcm_data.o
OBJS-$(CONFIG_ADPCM_YAMAHA_ENCODER) += adpcmenc.o adpcm_data.o
# libavformat dependencies
OBJS-$(CONFIG_ADTS_MUXER) += mpeg4audio.o
OBJS-$(CONFIG_ADX_DEMUXER) += adx.o
OBJS-$(CONFIG_CAF_DEMUXER) += mpeg4audio.o mpegaudiodata.o
OBJS-$(CONFIG_DV_DEMUXER) += dvdata.o
OBJS-$(CONFIG_DV_MUXER) += dvdata.o timecode.o
OBJS-$(CONFIG_FLAC_DEMUXER) += flacdec.o flacdata.o flac.o vorbis_data.o
OBJS-$(CONFIG_FLAC_MUXER) += flacdec.o flacdata.o flac.o vorbis_data.o
OBJS-$(CONFIG_FLV_DEMUXER) += mpeg4audio.o
OBJS-$(CONFIG_GXF_DEMUXER) += mpeg12data.o
OBJS-$(CONFIG_IFF_DEMUXER) += iff.o
OBJS-$(CONFIG_LATM_MUXER) += mpeg4audio.o
OBJS-$(CONFIG_MATROSKA_AUDIO_MUXER) += xiph.o mpeg4audio.o vorbis_data.o \
flacdec.o flacdata.o flac.o
OBJS-$(CONFIG_MATROSKA_DEMUXER) += mpeg4audio.o mpegaudiodata.o
OBJS-$(CONFIG_MATROSKA_MUXER) += xiph.o mpeg4audio.o \
flacdec.o flacdata.o flac.o \
mpegaudiodata.o vorbis_data.o
OBJS-$(CONFIG_MP3_MUXER) += mpegaudiodata.o mpegaudiodecheader.o
OBJS-$(CONFIG_MOV_DEMUXER) += mpeg4audio.o mpegaudiodata.o timecode.o
OBJS-$(CONFIG_MOV_MUXER) += mpeg4audio.o mpegaudiodata.o
OBJS-$(CONFIG_MPEGTS_MUXER) += mpegvideo.o mpeg4audio.o
OBJS-$(CONFIG_MPEGTS_DEMUXER) += mpeg4audio.o mpegaudiodata.o
OBJS-$(CONFIG_MXF_MUXER) += timecode.o
OBJS-$(CONFIG_NUT_MUXER) += mpegaudiodata.o
OBJS-$(CONFIG_OGG_DEMUXER) += flacdec.o flacdata.o flac.o \
dirac.o mpeg12data.o vorbis_data.o
OBJS-$(CONFIG_OGG_MUXER) += xiph.o flacdec.o flacdata.o flac.o \
vorbis_data.o
OBJS-$(CONFIG_RTP_MUXER) += mpeg4audio.o mpegvideo.o xiph.o
OBJS-$(CONFIG_SPDIF_DEMUXER) += aacadtsdec.o mpeg4audio.o
OBJS-$(CONFIG_WEBM_MUXER) += xiph.o mpeg4audio.o \
flacdec.o flacdata.o flac.o \
mpegaudiodata.o vorbis_data.o
OBJS-$(CONFIG_WTV_DEMUXER) += mpeg4audio.o mpegaudiodata.o
# external codec libraries
OBJS-$(CONFIG_LIBAACPLUS_ENCODER) += libaacplus.o
OBJS-$(CONFIG_LIBCELT_DECODER) += libcelt_dec.o
OBJS-$(CONFIG_LIBDIRAC_DECODER) += libdiracdec.o
OBJS-$(CONFIG_LIBDIRAC_ENCODER) += libdiracenc.o libdirac_libschro.o
OBJS-$(CONFIG_LIBFAAC_ENCODER) += libfaac.o
OBJS-$(CONFIG_LIBGSM_DECODER) += libgsm.o
OBJS-$(CONFIG_LIBGSM_ENCODER) += libgsm.o
OBJS-$(CONFIG_LIBGSM_MS_DECODER) += libgsm.o
OBJS-$(CONFIG_LIBGSM_MS_ENCODER) += libgsm.o
OBJS-$(CONFIG_LIBMP3LAME_ENCODER) += libmp3lame.o
OBJS-$(CONFIG_LIBOPENCORE_AMRNB_DECODER) += libopencore-amr.o
OBJS-$(CONFIG_LIBOPENCORE_AMRNB_ENCODER) += libopencore-amr.o
OBJS-$(CONFIG_LIBOPENCORE_AMRWB_DECODER) += libopencore-amr.o
OBJS-$(CONFIG_LIBOPENJPEG_DECODER) += libopenjpegdec.o
OBJS-$(CONFIG_LIBOPENJPEG_ENCODER) += libopenjpegenc.o
OBJS-$(CONFIG_LIBSCHROEDINGER_DECODER) += libschroedingerdec.o \
libschroedinger.o \
libdirac_libschro.o
OBJS-$(CONFIG_LIBSCHROEDINGER_ENCODER) += libschroedingerenc.o \
libschroedinger.o \
libdirac_libschro.o
OBJS-$(CONFIG_LIBSPEEX_DECODER) += libspeexdec.o
OBJS-$(CONFIG_LIBSPEEX_ENCODER) += libspeexenc.o
OBJS-$(CONFIG_LIBSTAGEFRIGHT_H264_DECODER)+= libstagefright.o
OBJS-$(CONFIG_LIBTHEORA_ENCODER) += libtheoraenc.o
OBJS-$(CONFIG_LIBUTVIDEO_DECODER) += libutvideo.o
OBJS-$(CONFIG_LIBVO_AACENC_ENCODER) += libvo-aacenc.o mpeg4audio.o
OBJS-$(CONFIG_LIBVO_AMRWBENC_ENCODER) += libvo-amrwbenc.o
OBJS-$(CONFIG_LIBVORBIS_ENCODER) += libvorbis.o vorbis_data.o
OBJS-$(CONFIG_LIBVPX_DECODER) += libvpxdec.o
OBJS-$(CONFIG_LIBVPX_ENCODER) += libvpxenc.o
OBJS-$(CONFIG_LIBX264_ENCODER) += libx264.o
OBJS-$(CONFIG_LIBXAVS_ENCODER) += libxavs.o
OBJS-$(CONFIG_LIBXVID) += libxvidff.o libxvid_rc.o
# parsers
OBJS-$(CONFIG_AAC_PARSER) += aac_parser.o aac_ac3_parser.o \
aacadtsdec.o mpeg4audio.o
OBJS-$(CONFIG_AC3_PARSER) += ac3_parser.o ac3tab.o \
aac_ac3_parser.o
OBJS-$(CONFIG_ADX_PARSER) += adx_parser.o adx.o
OBJS-$(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_FLAC_PARSER) += flac_parser.o flacdata.o flac.o \
vorbis_data.o
OBJS-$(CONFIG_GSM_PARSER) += gsm_parser.o
OBJS-$(CONFIG_H261_PARSER) += h261_parser.o
OBJS-$(CONFIG_H263_PARSER) += h263_parser.o
OBJS-$(CONFIG_H264_PARSER) += h264_parser.o h264.o \
cabac.o \
h264_refs.o h264_sei.o h264_direct.o \
h264_loopfilter.o h264_cabac.o \
h264_cavlc.o h264_ps.o \
mpegvideo.o error_resilience.o
OBJS-$(CONFIG_AAC_LATM_PARSER) += latm_parser.o
OBJS-$(CONFIG_MJPEG_PARSER) += mjpeg_parser.o
OBJS-$(CONFIG_MLP_PARSER) += mlp_parser.o mlp.o
OBJS-$(CONFIG_MPEG4VIDEO_PARSER) += mpeg4video_parser.o h263.o \
mpegvideo.o error_resilience.o \
mpeg4videodec.o mpeg4video.o \
ituh263dec.o h263dec.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_RV30_PARSER) += rv34_parser.o
OBJS-$(CONFIG_RV40_PARSER) += rv34_parser.o
OBJS-$(CONFIG_VC1_PARSER) += vc1_parser.o vc1.o vc1data.o \
msmpeg4.o msmpeg4data.o mpeg4video.o \
h263.o mpegvideo.o error_resilience.o
OBJS-$(CONFIG_VP3_PARSER) += vp3_parser.o
OBJS-$(CONFIG_VP8_PARSER) += vp8_parser.o
# bitstream filters
OBJS-$(CONFIG_AAC_ADTSTOASC_BSF) += aac_adtstoasc_bsf.o aacadtsdec.o \
mpeg4audio.o
OBJS-$(CONFIG_CHOMP_BSF) += chomp_bsf.o
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_MJPEG2JPEG_BSF) += mjpeg2jpeg_bsf.o mjpeg.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_PTHREADS) += pthread.o
OBJS-$(HAVE_W32THREADS) += pthread.o
OBJS-$(HAVE_OS2THREADS) += pthread.o
OBJS-$(CONFIG_MLIB) += mlib/dsputil_mlib.o \
# inverse.o contains the ff_inverse table definition, which is used by
# the FASTDIV macro (from libavutil); since referencing the external
# table has a negative effect on performance, copy it in libavcodec as
# well.
OBJS-$(!CONFIG_SMALL) += inverse.o
SKIPHEADERS += %_tablegen.h \
%_tables.h \
aac_tablegen_decl.h \
fft-internal.h \
tableprint.h \
$(ARCH)/vp56_arith.h
SKIPHEADERS-$(CONFIG_DXVA2) += dxva2.h dxva2_internal.h
SKIPHEADERS-$(CONFIG_LIBDIRAC) += libdirac.h
SKIPHEADERS-$(CONFIG_LIBSCHROEDINGER) += libschroedinger.h
SKIPHEADERS-$(CONFIG_VAAPI) += vaapi_internal.h
SKIPHEADERS-$(CONFIG_VDA) += vda_internal.h
SKIPHEADERS-$(CONFIG_VDPAU) += vdpau.h
SKIPHEADERS-$(CONFIG_XVMC) += xvmc.h
SKIPHEADERS-$(HAVE_W32THREADS) += w32pthreads.h
TESTPROGS = cabac dct fft fft-fixed h264 iirfilter rangecoder snowenc
TESTPROGS-$(HAVE_MMX) += motion
TESTOBJS = dctref.o
HOSTPROGS = aac_tablegen aacps_tablegen cbrt_tablegen cos_tablegen \
dv_tablegen motionpixels_tablegen mpegaudio_tablegen \
pcm_tablegen qdm2_tablegen sinewin_tablegen
DIRS = alpha arm bfin mlib ppc ps2 sh4 sparc x86
CLEANFILES = *_tables.c *_tables.h *_tablegen$(HOSTEXESUF)
$(SUBDIR)dct-test$(EXESUF): $(SUBDIR)dctref.o
TRIG_TABLES = cos cos_fixed sin
TRIG_TABLES := $(TRIG_TABLES:%=$(SUBDIR)%_tables.c)
$(TRIG_TABLES): $(SUBDIR)%_tables.c: $(SUBDIR)cos_tablegen$(HOSTEXESUF)
$(M)./$< $* > $@
ifdef CONFIG_SMALL
$(SUBDIR)%_tablegen$(HOSTEXESUF): HOSTCFLAGS += -DCONFIG_SMALL=1
else
$(SUBDIR)%_tablegen$(HOSTEXESUF): HOSTCFLAGS += -DCONFIG_SMALL=0
endif
GEN_HEADERS = cbrt_tables.h aacps_tables.h aac_tables.h dv_tables.h \
sinewin_tables.h mpegaudio_tables.h motionpixels_tables.h \
pcm_tables.h qdm2_tables.h
GEN_HEADERS := $(addprefix $(SUBDIR), $(GEN_HEADERS))
$(GEN_HEADERS): $(SUBDIR)%_tables.h: $(SUBDIR)%_tablegen$(HOSTEXESUF)
$(M)./$< > $@
ifdef CONFIG_HARDCODED_TABLES
$(SUBDIR)aacdec.o: $(SUBDIR)cbrt_tables.h
$(SUBDIR)aacps.o: $(SUBDIR)aacps_tables.h
$(SUBDIR)aactab.o: $(SUBDIR)aac_tables.h
$(SUBDIR)dv.o: $(SUBDIR)dv_tables.h
$(SUBDIR)sinewin.o: $(SUBDIR)sinewin_tables.h
$(SUBDIR)mpegaudiodec.o: $(SUBDIR)mpegaudio_tables.h
$(SUBDIR)mpegaudiodec_float.o: $(SUBDIR)mpegaudio_tables.h
$(SUBDIR)motionpixels.o: $(SUBDIR)motionpixels_tables.h
$(SUBDIR)pcm.o: $(SUBDIR)pcm_tables.h
$(SUBDIR)qdm2.o: $(SUBDIR)qdm2_tables.h
endif
CODEC_NAMES_SH := $(SRC_PATH)/$(SUBDIR)codec_names.sh
AVCODEC_H := $(SRC_PATH)/$(SUBDIR)avcodec.h
$(SUBDIR)codec_names.h: $(CODEC_NAMES_SH) config.h $(AVCODEC_H)
$(CC) $(CPPFLAGS) $(CFLAGS) -E $(AVCODEC_H) | \
$(CODEC_NAMES_SH) config.h $@
$(SUBDIR)utils.o: $(SUBDIR)codec_names.h
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/*
* a64 video encoder - c64 colors in rgb (Pepto)
* Copyright (c) 2009 Tobias Bindhammer
*
* 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
* a64 video encoder - c64 colors in rgb
*/
#ifndef AVCODEC_A64COLORS_H
#define AVCODEC_A64COLORS_H
#include <stdint.h>
/* c64 palette in RGB */
static const uint8_t a64_palette[16][3] = {
{0x00, 0x00, 0x00},
{0xff, 0xff, 0xff},
{0x68, 0x37, 0x2b},
{0x70, 0xa4, 0xb2},
{0x6f, 0x3d, 0x86},
{0x58, 0x8d, 0x43},
{0x35, 0x28, 0x79},
{0xb8, 0xc7, 0x6f},
{0x6f, 0x4f, 0x25},
{0x43, 0x39, 0x00},
{0x9a, 0x67, 0x59},
{0x44, 0x44, 0x44},
{0x6c, 0x6c, 0x6c},
{0x9a, 0xd2, 0x84},
{0x6c, 0x5e, 0xb5},
{0x95, 0x95, 0x95},
};
#endif /* AVCODEC_A64COLORS_H */
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/*
* a64 video encoder - basic headers
* Copyright (c) 2009 Tobias Bindhammer
*
* 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
* a64 video encoder - basic headers
*/
#ifndef AVCODEC_A64ENC_H
#define AVCODEC_A64ENC_H
#include "libavutil/lfg.h"
#include "avcodec.h"
#define C64XRES 320
#define C64YRES 200
typedef struct A64Context {
/* general variables */
AVFrame picture;
/* variables for multicolor modes */
AVLFG randctx;
int mc_lifetime;
int mc_use_5col;
int mc_frame_counter;
int *mc_meta_charset;
int *mc_charmap;
int *mc_best_cb;
int mc_luma_vals[5];
uint8_t *mc_charset;
uint8_t *mc_colram;
uint8_t *mc_palette;
int mc_pal_size;
} A64Context;
#endif /* AVCODEC_A64ENC_H */
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/*
* a64 video encoder - multicolor modes
* Copyright (c) 2009 Tobias Bindhammer
*
* 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
* a64 video encoder - multicolor modes
*/
#include "a64enc.h"
#include "a64colors.h"
#include "a64tables.h"
#include "elbg.h"
#include "libavutil/intreadwrite.h"
#define DITHERSTEPS 8
#define CHARSET_CHARS 256
#define INTERLACED 1
#define CROP_SCREENS 1
/* gray gradient */
static const int mc_colors[5]={0x0,0xb,0xc,0xf,0x1};
/* other possible gradients - to be tested */
//static const int mc_colors[5]={0x0,0x8,0xa,0xf,0x7};
//static const int mc_colors[5]={0x0,0x9,0x8,0xa,0x3};
static void to_meta_with_crop(AVCodecContext *avctx, AVFrame *p, int *dest)
{
int blockx, blocky, x, y;
int luma = 0;
int height = FFMIN(avctx->height, C64YRES);
int width = FFMIN(avctx->width , C64XRES);
uint8_t *src = p->data[0];
for (blocky = 0; blocky < C64YRES; blocky += 8) {
for (blockx = 0; blockx < C64XRES; blockx += 8) {
for (y = blocky; y < blocky + 8 && y < C64YRES; y++) {
for (x = blockx; x < blockx + 8 && x < C64XRES; x += 2) {
if(x < width && y < height) {
/* build average over 2 pixels */
luma = (src[(x + 0 + y * p->linesize[0])] +
src[(x + 1 + y * p->linesize[0])]) / 2;
/* write blocks as linear data now so they are suitable for elbg */
dest[0] = luma;
}
dest++;
}
}
}
}
}
static void render_charset(AVCodecContext *avctx, uint8_t *charset,
uint8_t *colrammap)
{
A64Context *c = avctx->priv_data;
uint8_t row1, row2;
int charpos, x, y;
int a, b;
uint8_t pix;
int lowdiff, highdiff;
int *best_cb = c->mc_best_cb;
static uint8_t index1[256];
static uint8_t index2[256];
static uint8_t dither[256];
int i;
int distance;
/* generate lookup-tables for dither and index before looping */
i = 0;
for (a=0; a < 256; a++) {
if(i < c->mc_pal_size -1 && a == c->mc_luma_vals[i + 1]) {
distance = c->mc_luma_vals[i + 1] - c->mc_luma_vals[i];
for(b = 0; b <= distance; b++) {
dither[c->mc_luma_vals[i] + b] = b * (DITHERSTEPS - 1) / distance;
}
i++;
}
if(i >= c->mc_pal_size - 1) dither[a] = 0;
index1[a] = i;
index2[a] = FFMIN(i + 1, c->mc_pal_size - 1);
}
/* and render charset */
for (charpos = 0; charpos < CHARSET_CHARS; charpos++) {
lowdiff = 0;
highdiff = 0;
for (y = 0; y < 8; y++) {
row1 = 0; row2 = 0;
for (x = 0; x < 4; x++) {
pix = best_cb[y * 4 + x];
/* accumulate error for brightest/darkest color */
if (index1[pix] >= 3)
highdiff += pix - c->mc_luma_vals[3];
if (index1[pix] < 1)
lowdiff += c->mc_luma_vals[1] - pix;
row1 <<= 2;
if (INTERLACED) {
row2 <<= 2;
if (interlaced_dither_patterns[dither[pix]][(y & 3) * 2 + 0][x & 3])
row1 |= 3-(index2[pix] & 3);
else
row1 |= 3-(index1[pix] & 3);
if (interlaced_dither_patterns[dither[pix]][(y & 3) * 2 + 1][x & 3])
row2 |= 3-(index2[pix] & 3);
else
row2 |= 3-(index1[pix] & 3);
}
else {
if (multi_dither_patterns[dither[pix]][(y & 3)][x & 3])
row1 |= 3-(index2[pix] & 3);
else
row1 |= 3-(index1[pix] & 3);
}
}
charset[y+0x000] = row1;
if (INTERLACED) charset[y+0x800] = row2;
}
/* do we need to adjust pixels? */
if (highdiff > 0 && lowdiff > 0 && c->mc_use_5col) {
if (lowdiff > highdiff) {
for (x = 0; x < 32; x++)
best_cb[x] = FFMIN(c->mc_luma_vals[3], best_cb[x]);
} else {
for (x = 0; x < 32; x++)
best_cb[x] = FFMAX(c->mc_luma_vals[1], best_cb[x]);
}
charpos--; /* redo now adjusted char */
/* no adjustment needed, all fine */
} else {
/* advance pointers */
best_cb += 32;
charset += 8;
/* remember colorram value */
colrammap[charpos] = (highdiff > 0);
}
}
}
static av_cold int a64multi_close_encoder(AVCodecContext *avctx)
{
A64Context *c = avctx->priv_data;
av_free(c->mc_meta_charset);
av_free(c->mc_best_cb);
av_free(c->mc_charset);
av_free(c->mc_charmap);
av_free(c->mc_colram);
return 0;
}
static av_cold int a64multi_init_encoder(AVCodecContext *avctx)
{
A64Context *c = avctx->priv_data;
int a;
av_lfg_init(&c->randctx, 1);
if (avctx->global_quality < 1) {
c->mc_lifetime = 4;
} else {
c->mc_lifetime = avctx->global_quality /= FF_QP2LAMBDA;
}
av_log(avctx, AV_LOG_INFO, "charset lifetime set to %d frame(s)\n", c->mc_lifetime);
c->mc_frame_counter = 0;
c->mc_use_5col = avctx->codec->id == CODEC_ID_A64_MULTI5;
c->mc_pal_size = 4 + c->mc_use_5col;
/* precalc luma values for later use */
for (a = 0; a < c->mc_pal_size; a++) {
c->mc_luma_vals[a]=a64_palette[mc_colors[a]][0] * 0.30 +
a64_palette[mc_colors[a]][1] * 0.59 +
a64_palette[mc_colors[a]][2] * 0.11;
}
if (!(c->mc_meta_charset = av_malloc(32000 * c->mc_lifetime * sizeof(int))) ||
!(c->mc_best_cb = av_malloc(CHARSET_CHARS * 32 * sizeof(int))) ||
!(c->mc_charmap = av_mallocz(1000 * c->mc_lifetime * sizeof(int))) ||
!(c->mc_colram = av_mallocz(CHARSET_CHARS * sizeof(uint8_t))) ||
!(c->mc_charset = av_malloc(0x800 * (INTERLACED+1) * sizeof(uint8_t)))) {
av_log(avctx, AV_LOG_ERROR, "Failed to allocate buffer memory.\n");
return AVERROR(ENOMEM);
}
/* set up extradata */
if (!(avctx->extradata = av_mallocz(8 * 4 + FF_INPUT_BUFFER_PADDING_SIZE))) {
av_log(avctx, AV_LOG_ERROR, "Failed to allocate memory for extradata.\n");
return AVERROR(ENOMEM);
}
avctx->extradata_size = 8 * 4;
AV_WB32(avctx->extradata, c->mc_lifetime);
AV_WB32(avctx->extradata + 16, INTERLACED);
avcodec_get_frame_defaults(&c->picture);
avctx->coded_frame = &c->picture;
avctx->coded_frame->pict_type = AV_PICTURE_TYPE_I;
avctx->coded_frame->key_frame = 1;
if (!avctx->codec_tag)
avctx->codec_tag = AV_RL32("a64m");
return 0;
}
static void a64_compress_colram(unsigned char *buf, int *charmap, uint8_t *colram)
{
int a;
uint8_t temp;
/* only needs to be done in 5col mode */
/* XXX could be squeezed to 0x80 bytes */
for (a = 0; a < 256; a++) {
temp = colram[charmap[a + 0x000]] << 0;
temp |= colram[charmap[a + 0x100]] << 1;
temp |= colram[charmap[a + 0x200]] << 2;
if (a < 0xe8) temp |= colram[charmap[a + 0x300]] << 3;
buf[a] = temp << 2;
}
}
static int a64multi_encode_frame(AVCodecContext *avctx, unsigned char *buf,
int buf_size, void *data)
{
A64Context *c = avctx->priv_data;
AVFrame *pict = data;
AVFrame *const p = (AVFrame *) & c->picture;
int frame;
int x, y;
int b_height;
int b_width;
int req_size;
int *charmap = c->mc_charmap;
uint8_t *colram = c->mc_colram;
uint8_t *charset = c->mc_charset;
int *meta = c->mc_meta_charset;
int *best_cb = c->mc_best_cb;
int charset_size = 0x800 * (INTERLACED + 1);
int colram_size = 0x100 * c->mc_use_5col;
int screen_size;
if(CROP_SCREENS) {
b_height = FFMIN(avctx->height,C64YRES) >> 3;
b_width = FFMIN(avctx->width ,C64XRES) >> 3;
screen_size = b_width * b_height;
} else {
b_height = C64YRES >> 3;
b_width = C64XRES >> 3;
screen_size = 0x400;
}
/* no data, means end encoding asap */
if (!data) {
/* all done, end encoding */
if (!c->mc_lifetime) return 0;
/* no more frames in queue, prepare to flush remaining frames */
if (!c->mc_frame_counter) {
c->mc_lifetime = 0;
}
/* still frames in queue so limit lifetime to remaining frames */
else c->mc_lifetime = c->mc_frame_counter;
/* still new data available */
} else {
/* fill up mc_meta_charset with data until lifetime exceeds */
if (c->mc_frame_counter < c->mc_lifetime) {
*p = *pict;
p->pict_type = AV_PICTURE_TYPE_I;
p->key_frame = 1;
to_meta_with_crop(avctx, p, meta + 32000 * c->mc_frame_counter);
c->mc_frame_counter++;
/* lifetime is not reached so wait for next frame first */
return 0;
}
}
/* lifetime reached so now convert X frames at once */
if (c->mc_frame_counter == c->mc_lifetime) {
req_size = 0;
/* any frames to encode? */
if (c->mc_lifetime) {
/* calc optimal new charset + charmaps */
ff_init_elbg(meta, 32, 1000 * c->mc_lifetime, best_cb, CHARSET_CHARS, 50, charmap, &c->randctx);
ff_do_elbg (meta, 32, 1000 * c->mc_lifetime, best_cb, CHARSET_CHARS, 50, charmap, &c->randctx);
/* create colorram map and a c64 readable charset */
render_charset(avctx, charset, colram);
/* copy charset to buf */
memcpy(buf,charset, charset_size);
/* advance pointers */
buf += charset_size;
charset += charset_size;
req_size += charset_size;
}
/* no charset so clean buf */
else memset(buf, 0, charset_size);
/* write x frames to buf */
for (frame = 0; frame < c->mc_lifetime; frame++) {
/* copy charmap to buf. buf is uchar*, charmap is int*, so no memcpy here, sorry */
for (y = 0; y < b_height; y++) {
for (x = 0; x < b_width; x++) {
buf[y * b_width + x] = charmap[y * b_width + x];
}
}
/* advance pointers */
buf += screen_size;
req_size += screen_size;
/* compress and copy colram to buf */
if (c->mc_use_5col) {
a64_compress_colram(buf, charmap, colram);
/* advance pointers */
buf += colram_size;
req_size += colram_size;
}
/* advance to next charmap */
charmap += 1000;
}
AV_WB32(avctx->extradata + 4, c->mc_frame_counter);
AV_WB32(avctx->extradata + 8, charset_size);
AV_WB32(avctx->extradata + 12, screen_size + colram_size);
/* reset counter */
c->mc_frame_counter = 0;
if (req_size > buf_size) {
av_log(avctx, AV_LOG_ERROR, "buf size too small (need %d, got %d)\n", req_size, buf_size);
return -1;
}
return req_size;
}
return 0;
}
AVCodec ff_a64multi_encoder = {
.name = "a64multi",
.type = AVMEDIA_TYPE_VIDEO,
.id = CODEC_ID_A64_MULTI,
.priv_data_size = sizeof(A64Context),
.init = a64multi_init_encoder,
.encode = a64multi_encode_frame,
.close = a64multi_close_encoder,
.pix_fmts = (const enum PixelFormat[]) {PIX_FMT_GRAY8, PIX_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("Multicolor charset for Commodore 64"),
.capabilities = CODEC_CAP_DELAY,
};
AVCodec ff_a64multi5_encoder = {
.name = "a64multi5",
.type = AVMEDIA_TYPE_VIDEO,
.id = CODEC_ID_A64_MULTI5,
.priv_data_size = sizeof(A64Context),
.init = a64multi_init_encoder,
.encode = a64multi_encode_frame,
.close = a64multi_close_encoder,
.pix_fmts = (const enum PixelFormat[]) {PIX_FMT_GRAY8, PIX_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("Multicolor charset for Commodore 64, extended with 5th color (colram)"),
.capabilities = CODEC_CAP_DELAY,
};
-150
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@@ -1,150 +0,0 @@
/*
* a64 video encoder - tables used by a64 encoders
* Copyright (c) 2009 Tobias Bindhammer
*
* 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
* a64 video encoder - tables used by a64 encoders
*/
#ifndef AVCODEC_A64TABLES_H
#define AVCODEC_A64TABLES_H
#include <stdint.h>
/**
* dither patterns used vor rendering the multicolor charset
*/
static const uint8_t multi_dither_patterns[9][4][4] = {
{
{0, 0, 0, 0},
{0, 0, 0, 0},
{0, 0, 0, 0},
{0, 0, 0, 0}
},
{
{1, 0, 0, 0},
{0, 0, 0, 0},
{0, 0, 1, 0},
{0, 0, 0, 0}
},
{
{1, 0, 0, 0},
{0, 0, 1, 0},
{0, 1, 0, 0},
{0, 0, 0, 1}
},
{
{1, 0, 0, 0},
{0, 1, 0, 1},
{0, 0, 1, 0},
{0, 1, 0, 1}
},
{
{1, 0, 1, 0},
{0, 1, 0, 1},
{1, 0, 1, 0},
{0, 1, 0, 1}
},
{
{1, 1, 1, 0},
{0, 1, 0, 1},
{1, 0, 1, 1},
{0, 1, 0, 1}
},
{
{0, 1, 1, 1},
{1, 1, 0, 1},
{1, 0, 1, 1},
{1, 1, 1, 0}
},
{
{0, 1, 1, 1},
{1, 1, 1, 1},
{1, 1, 0, 1},
{1, 1, 1, 1}
},
{
{1, 1, 1, 1},
{1, 1, 1, 1},
{1, 1, 1, 1},
{1, 1, 1, 1}
},
};
static const uint8_t interlaced_dither_patterns[9][8][4] = {
{
{0, 0, 0, 0}, {0, 0, 0, 0},
{0, 0, 0, 0}, {0, 0, 0, 0},
{0, 0, 0, 0}, {0, 0, 0, 0},
{0, 0, 0, 0}, {0, 0, 0, 0},
},
{
{1, 0, 1, 0}, {0, 0, 0, 0},
{0, 0, 0, 0}, {0, 0, 0, 0},
{1, 0, 1, 0}, {0, 0, 0, 0},
{0, 0, 0, 0}, {0, 0, 0, 0},
},
{
{1, 0, 1, 0}, {0, 0, 0, 0},
{0, 0, 0, 0}, {0, 1, 0, 1},
{1, 0, 1, 0}, {0, 0, 0, 0},
{0, 0, 0, 0}, {0, 1, 0, 1},
},
{
{1, 0, 1, 0}, {0, 1, 0, 1},
{0, 1, 0, 1}, {0, 0, 0, 0},
{1, 0, 1, 0}, {0, 1, 0, 1},
{0, 1, 0, 1}, {0, 0, 0, 0},
},
{
{1, 0, 1, 0}, {0, 1, 0, 1},
{0, 1, 0, 1}, {1, 0, 1, 0},
{1, 0, 1, 0}, {0, 1, 0, 1},
{0, 1, 0, 1}, {1, 0, 1, 0},
},
{
{1, 0, 1, 0}, {0, 1, 0, 1},
{1, 1, 1, 1}, {1, 0, 1, 0},
{1, 0, 1, 0}, {0, 1, 0, 1},
{1, 1, 1, 1}, {1, 0, 1, 0},
},
{
{1, 0, 1, 0}, {1, 1, 1, 1},
{1, 1, 1, 1}, {0, 1, 0, 1},
{1, 0, 1, 0}, {1, 1, 1, 1},
{1, 1, 1, 1}, {0, 1, 0, 1},
},
{
{1, 1, 1, 1}, {1, 1, 1, 1},
{1, 1, 1, 1}, {0, 1, 0, 1},
{1, 1, 1, 1}, {1, 1, 1, 1},
{1, 1, 1, 1}, {0, 1, 0, 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, 1}, {1, 1, 1, 1},
}
};
#endif /* AVCODEC_A64TABLES_H */
-307
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@@ -1,307 +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
* 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 "avcodec.h"
#include "dsputil.h"
#include "fft.h"
#include "mpeg4audio.h"
#include "sbr.h"
#include "fmtconvert.h"
#include <stdint.h>
#define MAX_CHANNELS 64
#define MAX_ELEM_ID 16
#define TNS_MAX_ORDER 20
#define MAX_LTP_LONG_SFB 40
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_OFF = 0,
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,
};
/**
* Output configuration status
*/
enum OCStatus {
OC_NONE, ///< Output unconfigured
OC_TRIAL_PCE, ///< Output configuration under trial specified by an inband PCE
OC_TRIAL_FRAME, ///< Output configuration under trial specified by a frame header
OC_GLOBAL_HDR, ///< Output configuration set in a global header but not yet locked
OC_LOCKED, ///< Output configuration locked in place
};
/**
* Predictor State
*/
typedef struct {
float cor0;
float cor1;
float var0;
float var1;
float r0;
float r1;
} PredictorState;
#define MAX_PREDICTORS 672
#define SCALE_DIV_512 36 ///< scalefactor difference that corresponds to scale difference in 512 times
#define SCALE_ONE_POS 140 ///< scalefactor index that corresponds to scale=1.0
#define SCALE_MAX_POS 255 ///< scalefactor index maximum value
#define SCALE_MAX_DIFF 60 ///< maximum scalefactor difference allowed by standard
#define SCALE_DIFF_ZERO 60 ///< codebook index corresponding to zero scalefactor indices difference
#define POW_SF2_ZERO 200 ///< ff_aac_pow2sf_tab index corresponding to pow(2, 0);
/**
* Long Term Prediction
*/
typedef struct {
int8_t present;
int16_t lag;
float coef;
int8_t used[MAX_LTP_LONG_SFB];
} LongTermPrediction;
/**
* 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];
LongTermPrediction ltp;
const uint16_t *swb_offset; ///< table of offsets to the lowest spectral coefficient of a scalefactor band, sfb, for a particular window
const uint8_t *swb_sizes; ///< table of scalefactor band sizes 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 start;
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;
Pulse pulse;
enum BandType band_type[128]; ///< band types
int band_type_run_end[120]; ///< band type run end points
float sf[120]; ///< scalefactors
int sf_idx[128]; ///< scalefactor indices (used by encoder)
uint8_t zeroes[128]; ///< band is not coded (used by encoder)
DECLARE_ALIGNED(32, float, coeffs)[1024]; ///< coefficients for IMDCT
DECLARE_ALIGNED(32, float, saved)[1024]; ///< overlap
DECLARE_ALIGNED(32, float, ret)[2048]; ///< PCM output
DECLARE_ALIGNED(16, float, ltp_state)[3072]; ///< time signal for LTP
PredictorState predictor_state[MAX_PREDICTORS];
} SingleChannelElement;
/**
* channel element - generic struct for SCE/CPE/CCE/LFE
*/
typedef struct {
// CPE specific
int common_window; ///< Set if channels share a common 'IndividualChannelStream' in bitstream.
int ms_mode; ///< Signals mid/side stereo flags coding mode (used by encoder)
uint8_t ms_mask[128]; ///< Set if mid/side stereo is used for each scalefactor window band
// shared
SingleChannelElement ch[2];
// CCE specific
ChannelCoupling coup;
SpectralBandReplication sbr;
} ChannelElement;
/**
* main AAC context
*/
typedef struct {
AVCodecContext *avctx;
AVFrame frame;
MPEG4AudioConfig m4ac;
int is_saved; ///< Set if elements have stored overlap from previous frame.
DynamicRangeControl che_drc;
/**
* @name 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;
/** @} */
/**
* @name temporary aligned temporary buffers
* (We do not want to have these on the stack.)
* @{
*/
DECLARE_ALIGNED(32, float, buf_mdct)[1024];
/** @} */
/**
* @name Computed / set up during initialization
* @{
*/
FFTContext mdct;
FFTContext mdct_small;
FFTContext mdct_ltp;
DSPContext dsp;
FmtConvertContext fmt_conv;
int random_state;
/** @} */
/**
* @name Members used for output interleaving
* @{
*/
float *output_data[MAX_CHANNELS]; ///< Points to each element's 'ret' buffer (PCM output).
/** @} */
DECLARE_ALIGNED(32, float, temp)[128];
enum OCStatus output_configured;
int warned_num_aac_frames;
} AACContext;
#endif /* AVCODEC_AAC_H */
-103
View File
@@ -1,103 +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 */
if(s->codec_id)
avctx->codec_id = s->codec_id;
/* Due to backwards compatible HE-AAC the sample rate, channel count,
and total number of samples found in an AAC ADTS header are not
reliable. Bit rate is still accurate because the total frame duration in
seconds is still correct (as is the number of bits in the frame). */
if (avctx->codec_id != CODEC_ID_AAC) {
avctx->sample_rate = s->sample_rate;
/* 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->channel_layout = s->channel_layout;
}
avctx->frame_size = s->samples;
avctx->audio_service_type = s->service_type;
}
avctx->bit_rate = s->bit_rate;
return i;
}
-66
View File
@@ -1,66 +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;
uint64_t channel_layout;
int service_type;
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 */

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