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Even if ffmpeg can read the file format, it may not support all its codecs. Please consult the supported codec list in the ffmpeg documentation.
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If the JPEGs are named img1.jpg, img2.jpg, img3.jpg,..., use:
ffmpeg -f image2 -i img%d.jpg /tmp/a.mpg |
`%d' is replaced by the image number.
`img%03d.jpg' generates `img001.jpg', `img002.jpg', etc...
The same system is used for the other image formats.
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Use:
ffmpeg -i movie.mpg movie%d.jpg |
The `movie.mpg' used as input will be converted to `movie1.jpg', `movie2.jpg', etc...
Instead of relying on file format self-recognition, you may also use
to force the encoding.
Applying that to the previous example:
ffmpeg -i movie.mpg -f image2 -vcodec mjpeg menu%d.jpg |
Beware that there is no "jpeg" codec. Use "mjpeg" instead.
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No. FFmpeg only supports open source codecs. Windows DLLs are not portable, bloated and often slow.
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This is the Qcelp codec, FFmpeg has no support for that codec currently. Try mencoder/mplayer it might work.
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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.
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Use `-' as filename.
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The audio is AC3 (a.k.a. A/52). AC3 decoding is an optional component in ffmpeg
as the component that handles AC3 decoding (liba52) is currently released under
the GPL. If you have liba52 installed on your system, enable AC3 decoding
with ./configure --enable-liba52 --enable-gpl
. Take care: by
enabling AC3, you automatically change the license of libavcodec from
LGPL to GPL.
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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:
.avi/.asf
.asf only
.asf only
.asf only
only if you have some MPEG-4 codec installed like ffdshow or XviD
.mpg only
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:
if some MP3 codec like LAME is installed
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This is a well-known bug in the bt8x8 driver. For 2.4.26 there is a patch at (http://svn.mplayerhq.hu/michael/trunk/patches/bttv-420-2.4.26.patch?view=co). This may also apply cleanly to other 2.4-series kernels.
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Pass 'combfilter=1 lumafilter=1' to the bttv driver. Note though that 'combfilter=1' will cause somewhat too strong filtering. A fix is to apply (http://svn.mplayerhq.hu/michael/trunk/patches/bttv-comb-2.4.26.patch?view=co) or (http://svn.mplayerhq.hu/michael/trunk/patches/bttv-comb-2.6.6.patch?view=co) and pass 'combfilter=2'.
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Nowhere. Upgrade to the latest release or if there is no recent release upgrade to Subversion HEAD. You could also try to report it. Maybe you will get lucky and become the first person in history to get an answer different from "upgrade to Subversion HEAD".
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Try '-f image2 test%d.jpg'.
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Some codecs, like MPEG-1/2, only allow a small number of fixed framerates. Choose a different codec with the -vcodec command line option.
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Try a 'make distclean' in the ffmpeg source directory before the build. If this does not help see (http://ffmpeg.org/bugreports.php).
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Both XviD and DivX (version 4+) are implementations of the ISO MPEG-4 standard (note that there are many other coding formats that use this same standard). Thus, use '-vcodec mpeg4' to encode 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.
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-acodec aac -vcodec mpeg4 width<=320 height<=240
4mv, title
B-frames
ffmpeg -i input -acodec aac -ab 128kb -vcodec mpeg4 -b 1200kb -mbd 2 -flags +4mv+trell -aic 2 -cmp 2 -subcmp 2 -s 320x180 -title X output.mp4
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-acodec aac -vcodec mpeg4 width*height<=76800 width%16=0 height%16=0 -ar 24000 -r 30000/1001 or 15000/1001 -f psp
4mv, title
B-frames
ffmpeg -i input -acodec aac -ab 128kb -vcodec mpeg4 -b 1200kb -ar 24000 -mbd 2 -flags +4mv+trell -aic 2 -cmp 2 -subcmp 2 -s 368x192 -r 30000/1001 -title X -f psp output.mp4
-acodec aac -vcodec h264 width*height<=76800 width%16=0? height%16=0? -ar 48000 -coder 1 -r 30000/1001 or 15000/1001 -f psp
title, loop filter
CAVLC
ffmpeg -i input -acodec aac -ab 128kb -vcodec h264 -b 1200kb -ar 48000 -mbd 2 -coder 1 -cmp 2 -subcmp 2 -s 368x192 -r 30000/1001 -title X -f psp -flags loop -trellis 2 -partitions parti4x4+parti8x8+partp4x4+partp8x8+partb8x8 output.mp4
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'-mbd rd -flags +4mv+trell+aic -cmp 2 -subcmp 2 -g 300 -pass 1/2', things to try: '-bf 2', '-flags qprd', '-flags mv0', '-flags skiprd'.
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'-mbd rd -flags +trell -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.
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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.
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If you have built FFmpeg with ./configure --enable-avisynth
(only possible on MinGW/Cygwin platforms),
then you may use any file that DirectShow can read as input.
(Be aware that this feature has been recently added,
so you will need to help yourself in case of problems.)
Just create an "input.avs" text file with this single line ...
DirectShowSource("C:\path to your file\yourfile.asf") |
... and then feed that text file to FFmpeg:
ffmpeg -i input.avs |
For ANY other help on Avisynth, please visit http://www.avisynth.org/.
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Likely reasons
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Like most open source projects FFmpeg suffers from a certain lack of manpower. For this reason the developers have to prioritize the work they do and putting out releases is not at the top of the list, fixing bugs and reviewing patches takes precedence. Please don't complain or request more timely and/or frequent releases unless you are willing to help out creating them.
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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.
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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 (projects.php).
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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 #ifdef
s related to the compiler.
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Visual C++ is not compliant to the C standard and does not support the inline assembly used in FFmpeg. If you wish - for whatever weird reason - to use Visual C++ for your project then you can link the Visual C++ code with libav* as long as you compile the latter with a working C compiler. For more information, see the Visual C++ compatibility section in the FFmpeg documentation.
There have been efforts to make FFmpeg compatible with Visual C++ in the past. However, they have all been rejected as too intrusive, especially since MinGW does the job perfectly adequately. None of the core developers work with Visual C++ and thus this item is low priority. Should you find the silver bullet that solves this problem, feel free to shoot it at us.
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Yes, but the MinGW tools must be used to compile FFmpeg. You can link the resulting DLLs with any other Windows program. Read the Native Windows Compilation and Visual C++ compatibility sections in the FFmpeg documentation to find more information.
To get help and instructions for using FFmpeg under Windows, check out the FFmpeg Windows Help Forum at http://arrozcru.no-ip.org/ffmpeg/.
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No. These tools are too bloated and they complicate the build.
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ffmpeg is already organized in a highly modular manner and does not need to be rewritten in a formal object language. Further, many of the developers favor straight C; it works for them. For more arguments on this matter, read "Programming Religion" at (http://www.tux.org/lkml/#s15).
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The build process creates ffmpeg_g, ffplay_g, etc. which contain full debug information. Those binaries are strip'd to create ffmpeg, ffplay, etc. If you need the debug information, used the *_g versions.
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Yes, as long as the code is optional and can easily and cleanly be placed under #ifdef CONFIG_GPL without breaking anything. So for example a new codec or filter would be OK under GPL while a bugfix to LGPL code would not.
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Common code is in its own files in libav* and is used by the individual codecs. They will not work without the common parts, you have to compile the whole libav*. If you wish, disable some parts with configure switches. You can also try to hack it and remove more, but if you had problems fixing the compilation failure then you are probably not qualified for this.
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You have to implement a URLProtocol, see libavformat/file.c in FFmpeg and libmpdemux/demux_lavf.c in MPlayer sources.
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The standard MSys bash (2.04) is broken. You need to install 2.05 or later.
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Of course not, you MUST strip ALL RTP headers and assemble valid packets first, an MP3 decoder decodes MP3 packets not bastardized MP3 packets encapsulated in RTP. The same applies to all decoders, this is not specific to ffmpeg or libavcodec.
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see http://www.iversenit.dk/dev/ffmpeg-headers/
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see http://svn.mplayerhq.hu/michael/trunk/docs/
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