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https://github.com/FFmpeg/FFmpeg.git
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0c9ddc0fd5
Reported-By: Clément Bœsch
431 lines
15 KiB
Plaintext
431 lines
15 KiB
Plaintext
\input texinfo @c -*- texinfo -*-
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@settitle FFmpeg FAQ
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@titlepage
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@center @titlefont{FFmpeg FAQ}
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@end titlepage
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@top
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@contents
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@chapter General Questions
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@section Why doesn't FFmpeg support feature [xyz]?
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Because no one has taken on that task yet. FFmpeg development is
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driven by the tasks that are important to the individual developers.
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If there is a feature that is important to you, the best way to get
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it implemented is to undertake the task yourself or sponsor a developer.
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@section FFmpeg does not support codec XXX. Can you include a Windows DLL loader to support it?
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No. Windows DLLs are not portable, bloated and often slow.
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Moreover FFmpeg strives to support all codecs natively.
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A DLL loader is not conducive to that goal.
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@section I cannot read this file although this format seems to be supported by ffmpeg.
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Even if ffmpeg can read the container format, it may not support all its
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codecs. Please consult the supported codec list in the ffmpeg
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documentation.
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@section Which codecs are supported by Windows?
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Windows does not support standard formats like MPEG very well, unless you
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install some additional codecs.
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The following list of video codecs should work on most Windows systems:
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@table @option
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@item msmpeg4v2
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.avi/.asf
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@item msmpeg4
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.asf only
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@item wmv1
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.asf only
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@item wmv2
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.asf only
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@item mpeg4
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Only if you have some MPEG-4 codec like ffdshow or Xvid installed.
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@item mpeg1video
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.mpg only
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@end table
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Note, ASF files often have .wmv or .wma extensions in Windows. It should also
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be mentioned that Microsoft claims a patent on the ASF format, and may sue
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or threaten users who create ASF files with non-Microsoft software. It is
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strongly advised to avoid ASF where possible.
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The following list of audio codecs should work on most Windows systems:
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@table @option
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@item adpcm_ima_wav
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@item adpcm_ms
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@item pcm_s16le
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always
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@item libmp3lame
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If some MP3 codec like LAME is installed.
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@end table
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@chapter Compilation
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@section @code{error: can't find a register in class 'GENERAL_REGS' while reloading 'asm'}
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This is a bug in gcc. Do not report it to us. Instead, please report it to
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the gcc developers. Note that we will not add workarounds for gcc bugs.
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Also note that (some of) the gcc developers believe this is not a bug or
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not a bug they should fix:
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@url{http://gcc.gnu.org/bugzilla/show_bug.cgi?id=11203}.
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Then again, some of them do not know the difference between an undecidable
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problem and an NP-hard problem...
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@chapter Usage
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@section ffmpeg does not work; what is wrong?
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Try a @code{make distclean} in the ffmpeg source directory before the build.
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If this does not help see
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(@url{http://ffmpeg.org/bugreports.html}).
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@section How do I encode single pictures into movies?
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First, rename your pictures to follow a numerical sequence.
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For example, img1.jpg, img2.jpg, img3.jpg,...
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Then you may run:
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@example
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ffmpeg -f image2 -i img%d.jpg /tmp/a.mpg
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@end example
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Notice that @samp{%d} is replaced by the image number.
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@file{img%03d.jpg} means the sequence @file{img001.jpg}, @file{img002.jpg}, etc...
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If you have large number of pictures to rename, you can use the
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following command to ease the burden. The command, using the bourne
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shell syntax, symbolically links all files in the current directory
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that match @code{*jpg} to the @file{/tmp} directory in the sequence of
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@file{img001.jpg}, @file{img002.jpg} and so on.
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@example
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x=1; for i in *jpg; do counter=$(printf %03d $x); ln -s "$i" /tmp/img"$counter".jpg; x=$(($x+1)); done
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@end example
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If you want to sequence them by oldest modified first, substitute
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@code{$(ls -r -t *jpg)} in place of @code{*jpg}.
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Then run:
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@example
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ffmpeg -f image2 -i /tmp/img%03d.jpg /tmp/a.mpg
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@end example
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The same logic is used for any image format that ffmpeg reads.
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@section How do I encode movie to single pictures?
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Use:
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@example
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ffmpeg -i movie.mpg movie%d.jpg
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@end example
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The @file{movie.mpg} used as input will be converted to
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@file{movie1.jpg}, @file{movie2.jpg}, etc...
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Instead of relying on file format self-recognition, you may also use
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@table @option
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@item -c:v ppm
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@item -c:v png
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@item -c:v mjpeg
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@end table
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to force the encoding.
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Applying that to the previous example:
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@example
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ffmpeg -i movie.mpg -f image2 -c:v mjpeg menu%d.jpg
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@end example
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Beware that there is no "jpeg" codec. Use "mjpeg" instead.
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@section Why do I see a slight quality degradation with multithreaded MPEG* encoding?
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For multithreaded MPEG* encoding, the encoded slices must be independent,
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otherwise thread n would practically have to wait for n-1 to finish, so it's
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quite logical that there is a small reduction of quality. This is not a bug.
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@section How can I read from the standard input or write to the standard output?
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Use @file{-} as file name.
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@section -f jpeg doesn't work.
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Try '-f image2 test%d.jpg'.
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@section Why can I not change the frame rate?
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Some codecs, like MPEG-1/2, only allow a small number of fixed frame rates.
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Choose a different codec with the -c:v command line option.
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@section How do I encode Xvid or DivX video with ffmpeg?
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Both Xvid and DivX (version 4+) are implementations of the ISO MPEG-4
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standard (note that there are many other coding formats that use this
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same standard). Thus, use '-c:v mpeg4' to encode in these formats. The
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default fourcc stored in an MPEG-4-coded file will be 'FMP4'. If you want
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a different fourcc, use the '-vtag' option. E.g., '-vtag xvid' will
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force the fourcc 'xvid' to be stored as the video fourcc rather than the
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default.
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@section Which are good parameters for encoding high quality MPEG-4?
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'-mbd rd -flags +mv4+aic -trellis 2 -cmp 2 -subcmp 2 -g 300 -pass 1/2',
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things to try: '-bf 2', '-flags qprd', '-flags mv0', '-flags skiprd'.
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@section Which are good parameters for encoding high quality MPEG-1/MPEG-2?
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'-mbd rd -trellis 2 -cmp 2 -subcmp 2 -g 100 -pass 1/2'
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but beware the '-g 100' might cause problems with some decoders.
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Things to try: '-bf 2', '-flags qprd', '-flags mv0', '-flags skiprd.
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@section Interlaced video looks very bad when encoded with ffmpeg, what is wrong?
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You should use '-flags +ilme+ildct' and maybe '-flags +alt' for interlaced
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material, and try '-top 0/1' if the result looks really messed-up.
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@section How can I read DirectShow files?
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If you have built FFmpeg with @code{./configure --enable-avisynth}
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(only possible on MinGW/Cygwin platforms),
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then you may use any file that DirectShow can read as input.
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Just create an "input.avs" text file with this single line ...
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@example
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DirectShowSource("C:\path to your file\yourfile.asf")
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@end example
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... and then feed that text file to ffmpeg:
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@example
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ffmpeg -i input.avs
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@end example
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For ANY other help on Avisynth, please visit the
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@uref{http://www.avisynth.org/, Avisynth homepage}.
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@section How can I join video files?
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A few multimedia containers (MPEG-1, MPEG-2 PS, DV) allow to join video files by
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merely concatenating them.
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Hence you may concatenate your multimedia files by first transcoding them to
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these privileged formats, then using the humble @code{cat} command (or the
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equally humble @code{copy} under Windows), and finally transcoding back to your
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format of choice.
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@example
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ffmpeg -i input1.avi -same_quant intermediate1.mpg
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ffmpeg -i input2.avi -same_quant intermediate2.mpg
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cat intermediate1.mpg intermediate2.mpg > intermediate_all.mpg
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ffmpeg -i intermediate_all.mpg -same_quant output.avi
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@end example
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Notice that you should either use @code{-same_quant} or set a reasonably high
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bitrate for your intermediate and output files, if you want to preserve
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video quality.
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Also notice that you may avoid the huge intermediate files by taking advantage
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of named pipes, should your platform support it:
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@example
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mkfifo intermediate1.mpg
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mkfifo intermediate2.mpg
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ffmpeg -i input1.avi -same_quant -y intermediate1.mpg < /dev/null &
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ffmpeg -i input2.avi -same_quant -y intermediate2.mpg < /dev/null &
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cat intermediate1.mpg intermediate2.mpg |\
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ffmpeg -f mpeg -i - -same_quant -c:v mpeg4 -acodec libmp3lame output.avi
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@end example
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Similarly, the yuv4mpegpipe format, and the raw video, raw audio codecs also
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allow concatenation, and the transcoding step is almost lossless.
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When using multiple yuv4mpegpipe(s), the first line needs to be discarded
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from all but the first stream. This can be accomplished by piping through
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@code{tail} as seen below. Note that when piping through @code{tail} you
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must use command grouping, @code{@{ ;@}}, to background properly.
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For example, let's say we want to join two FLV files into an output.flv file:
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@example
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mkfifo temp1.a
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mkfifo temp1.v
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mkfifo temp2.a
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mkfifo temp2.v
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mkfifo all.a
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mkfifo all.v
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ffmpeg -i input1.flv -vn -f u16le -acodec pcm_s16le -ac 2 -ar 44100 - > temp1.a < /dev/null &
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ffmpeg -i input2.flv -vn -f u16le -acodec pcm_s16le -ac 2 -ar 44100 - > temp2.a < /dev/null &
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ffmpeg -i input1.flv -an -f yuv4mpegpipe - > temp1.v < /dev/null &
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@{ ffmpeg -i input2.flv -an -f yuv4mpegpipe - < /dev/null | tail -n +2 > temp2.v ; @} &
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cat temp1.a temp2.a > all.a &
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cat temp1.v temp2.v > all.v &
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ffmpeg -f u16le -acodec pcm_s16le -ac 2 -ar 44100 -i all.a \
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-f yuv4mpegpipe -i all.v \
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-same_quant -y output.flv
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rm temp[12].[av] all.[av]
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@end example
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@section -profile option fails when encoding H.264 video with AAC audio
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@command{ffmpeg} prints an error like
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@example
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Undefined constant or missing '(' in 'baseline'
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Unable to parse option value "baseline"
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Error setting option profile to value baseline.
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@end example
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Short answer: write @option{-profile:v} instead of @option{-profile}.
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Long answer: this happens because the @option{-profile} option can apply to both
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video and audio. Specifically the AAC encoder also defines some profiles, none
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of which are named @var{baseline}.
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The solution is to apply the @option{-profile} option to the video stream only
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by using @url{http://ffmpeg.org/ffmpeg.html#Stream-specifiers-1, Stream specifiers}.
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Appending @code{:v} to it will do exactly that.
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@section Using @option{-f lavfi}, audio becomes mono for no apparent reason.
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Use @option{-dumpgraph -} to find out exactly where the channel layout is
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lost.
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Most likely, it is through @code{auto-inserted aconvert}. Try to understand
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why the converting filter was needed at that place.
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Just before the output is a likely place, as @option{-f lavfi} currently
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only support packed S16.
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Then insert the correct @code{aconvert} explicitly in the filter graph,
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specifying the exact format.
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@example
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aconvert=s16:stereo:packed
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@end example
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@chapter Development
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@section Are there examples illustrating how to use the FFmpeg libraries, particularly libavcodec and libavformat?
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Yes. Read the Developers Guide of the FFmpeg documentation. Alternatively,
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examine the source code for one of the many open source projects that
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already incorporate FFmpeg at (@url{projects.html}).
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@section Can you support my C compiler XXX?
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It depends. If your compiler is C99-compliant, then patches to support
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it are likely to be welcome if they do not pollute the source code
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with @code{#ifdef}s related to the compiler.
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@section Is Microsoft Visual C++ supported?
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No. Microsoft Visual C++ is not compliant to the C99 standard and does
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not - among other things - support the inline assembly used in FFmpeg.
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If you wish to use MSVC++ for your
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project then you can link the MSVC++ code with libav* as long as
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you compile the latter with a working C compiler. For more information, see
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the @emph{Microsoft Visual C++ compatibility} section in the FFmpeg
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documentation.
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There have been efforts to make FFmpeg compatible with MSVC++ in the
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past. However, they have all been rejected as too intrusive, especially
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since MinGW does the job adequately. None of the core developers
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work with MSVC++ and thus this item is low priority. Should you find
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the silver bullet that solves this problem, feel free to shoot it at us.
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We strongly recommend you to move over from MSVC++ to MinGW tools.
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@section Can I use FFmpeg or libavcodec under Windows?
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Yes, but the Cygwin or MinGW tools @emph{must} be used to compile FFmpeg.
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Read the @emph{Windows} section in the FFmpeg documentation to find more
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information.
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To get help and instructions for building FFmpeg under Windows, check out
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the FFmpeg Windows Help Forum at
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@url{http://ffmpeg.arrozcru.org/}.
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@section Can you add automake, libtool or autoconf support?
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No. These tools are too bloated and they complicate the build.
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@section Why not rewrite FFmpeg in object-oriented C++?
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FFmpeg is already organized in a highly modular manner and does not need to
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be rewritten in a formal object language. Further, many of the developers
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favor straight C; it works for them. For more arguments on this matter,
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read @uref{http://www.tux.org/lkml/#s15, "Programming Religion"}.
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@section Why are the ffmpeg programs devoid of debugging symbols?
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The build process creates ffmpeg_g, ffplay_g, etc. which contain full debug
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information. Those binaries are stripped to create ffmpeg, ffplay, etc. If
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you need the debug information, use the *_g versions.
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@section I do not like the LGPL, can I contribute code under the GPL instead?
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Yes, as long as the code is optional and can easily and cleanly be placed
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under #if CONFIG_GPL without breaking anything. So, for example, a new codec
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or filter would be OK under GPL while a bug fix to LGPL code would not.
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@section I'm using FFmpeg from within my C++ application but the linker complains about missing symbols which seem to be available.
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FFmpeg is a pure C project, so to use the libraries within your C++ application
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you need to explicitly state that you are using a C library. You can do this by
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encompassing your FFmpeg includes using @code{extern "C"}.
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See @url{http://www.parashift.com/c++-faq-lite/mixing-c-and-cpp.html#faq-32.3}
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@section I'm using libavutil from within my C++ application but the compiler complains about 'UINT64_C' was not declared in this scope
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FFmpeg is a pure C project using C99 math features, in order to enable C++
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to use them you have to append -D__STDC_CONSTANT_MACROS to your CXXFLAGS
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@section I have a file in memory / a API different from *open/*read/ libc how do I use it with libavformat?
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You have to implement a URLProtocol, see @file{libavformat/file.c} in
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FFmpeg and @file{libmpdemux/demux_lavf.c} in MPlayer sources.
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@section Where can I find libav* headers for Pascal/Delphi?
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see @url{http://www.iversenit.dk/dev/ffmpeg-headers/}
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@section Where is the documentation about ffv1, msmpeg4, asv1, 4xm?
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see @url{http://www.ffmpeg.org/~michael/}
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@section How do I feed H.263-RTP (and other codecs in RTP) to libavcodec?
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Even if peculiar since it is network oriented, RTP is a container like any
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other. You have to @emph{demux} RTP before feeding the payload to libavcodec.
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In this specific case please look at RFC 4629 to see how it should be done.
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@section AVStream.r_frame_rate is wrong, it is much larger than the frame rate.
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r_frame_rate is NOT the average frame rate, it is the smallest frame rate
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that can accurately represent all timestamps. So no, it is not
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wrong if it is larger than the average!
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For example, if you have mixed 25 and 30 fps content, then r_frame_rate
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will be 150.
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@section Why is @code{make fate} not running all tests?
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Make sure you have the fate-suite samples and the @code{SAMPLES} Make variable
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or @code{FATE_SAMPLES} environment variable or the @code{--samples}
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@command{configure} option is set to the right path.
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@section Why is @code{make fate} not finding the samples?
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Do you happen to have a @code{~} character in the samples path to indicate a
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home directory? The value is used in ways where the shell cannot expand it,
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causing FATE to not find files. Just replace @code{~} by the full path.
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@bye
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