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533 lines
20 KiB
Plaintext
533 lines
20 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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@section I have installed this library with my distro's package manager. Why does @command{configure} not see it?
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Distributions usually split libraries in several packages. The main package
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contains the files necessary to run programs using the library. The
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development package contains the files necessary to build programs using the
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library. Sometimes, docs and/or data are in a separate package too.
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To build FFmpeg, you need to install the development package. It is usually
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called @file{libfoo-dev} or @file{libfoo-devel}. You can remove it after the
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build is finished, but be sure to keep the main package.
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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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To "join" video files is quite ambiguous. The following list explains the
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different kinds of "joining" and points out how those are addressed in
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FFmpeg. To join video files may mean:
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@itemize
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@item
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To put them one after the other: this is called to @emph{concatenate} them
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(in short: concat) and is addressed
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@ref{How can I concatenate video files, in this very faq}.
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@item
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To put them together in the same file, to let the user choose between the
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different versions (example: different audio languages): this is called to
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@emph{multiplex} them together (in short: mux), and is done by simply
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invoking ffmpeg with several @option{-i} options.
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@item
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For audio, to put all channels together in a single stream (example: two
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mono streams into one stereo stream): this is sometimes called to
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@emph{merge} them, and can be done using the
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@url{http://ffmpeg.org/ffmpeg.html#amerge, @code{amerge}} filter.
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@item
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For audio, to play one on top of the other: this is called to @emph{mix}
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them, and can be done by first merging them into a single stream and then
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using the @url{http://ffmpeg.org/ffmpeg.html#pan, @code{pan}} filter to mix
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the channels at will.
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@item
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For video, to display both together, side by side or one on top of a part of
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the other; it can be done using the
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@url{http://ffmpeg.org/ffmpeg.html#overlay, @code{overlay}} video filter.
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@end itemize
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@anchor{How can I concatenate video files}
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@section How can I concatenate video files?
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There are several solutions, depending on the exact circumstances.
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@subsection Concatenating using filters
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FFmpeg has a @url{http://ffmpeg.org/ffmpeg.html#concat-1, @code{concat}}
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filter designed specifically for that, with examples in the documentation.
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@subsection Concatenating at the file level
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A few multimedia containers (MPEG-1, MPEG-2 PS, DV) allow to concatenate
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video by merely concatenating the files 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 -qscale:v 1 intermediate1.mpg
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ffmpeg -i input2.avi -qscale:v 1 intermediate2.mpg
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cat intermediate1.mpg intermediate2.mpg > intermediate_all.mpg
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ffmpeg -i intermediate_all.mpg -qscale:v 2 output.avi
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@end example
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Additionally, you can use the @code{concat} protocol instead of @code{cat} or
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@code{copy} which will avoid creation of a potentially huge intermediate file.
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@example
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ffmpeg -i input1.avi -qscale:v 1 intermediate1.mpg
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ffmpeg -i input2.avi -qscale:v 1 intermediate2.mpg
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ffmpeg -i concat:"intermediate1.mpg|intermediate2.mpg" -c copy intermediate_all.mpg
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ffmpeg -i intermediate_all.mpg -qscale:v 2 output.avi
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@end example
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Note that you may need to escape the character "|" which is special for many
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shells.
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Another option is usage 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 -qscale:v 1 -y intermediate1.mpg < /dev/null &
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ffmpeg -i input2.avi -qscale:v 1 -y intermediate2.mpg < /dev/null &
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cat intermediate1.mpg intermediate2.mpg |\
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ffmpeg -f mpeg -i - -c:v mpeg4 -acodec libmp3lame output.avi
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@end example
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@subsection Concatenating using raw audio and video
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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 concatenate two FLV files into an
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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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-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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@section Why does FFmpeg not see the subtitles in my VOB file?
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VOB and a few other formats do not have a global header that describes
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everything present in the file. Instead, applications are supposed to scan
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the file to see what it contains. Since VOB files are frequently large, only
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the beginning is scanned. If the subtitles happen only later in the file,
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they will not be initally detected.
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Some applications, including the @code{ffmpeg} command-line tool, can only
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work with streams that were detected during the initial scan; streams that
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are detected later are ignored.
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The size of the initial scan is controlled by two options: @code{probesize}
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(default ~5 Mo) and @code{analyzeduration} (default 5,000,000 µs = 5 s). For
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the subtitle stream to be detected, both values must be large enough.
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@section Why was the @command{ffmpeg} @option{-sameq} option removed? What to use instead?
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The @option{-sameq} option meant "same quantizer", and made sense only in a
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very limited set of cases. Unfortunately, a lot of people mistook it for
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"same quality" and used it in places where it did not make sense: it had
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roughly the expected visible effect, but achieved it in a very inefficient
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way.
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Each encoder has its own set of options to set the quality-vs-size balance,
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use the options for the encoder you are using to set the quality level to a
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point acceptable for your tastes. The most common options to do that are
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@option{-qscale} and @option{-qmax}, but you should peruse the documentation
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of the encoder you chose.
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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. Check the @file{doc/examples} directory in the source
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repository, also available online at:
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@url{https://github.com/FFmpeg/FFmpeg/tree/master/doc/examples}.
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Examples are also installed by default, usually in
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@code{$PREFIX/share/ffmpeg/examples}.
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Also you may 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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Yes. Please see the @uref{platform.html, Microsoft Visual C++}
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section in the FFmpeg documentation.
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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 from the libraries themselves.
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FFmpeg builds static libraries by default. In static libraries, dependencies
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are not handled. That has two consequences. First, you must specify the
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libraries in dependency order: @code{-lavdevice} must come before
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@code{-lavformat}, @code{-lavutil} must come after everything else, etc.
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Second, external libraries that are used in FFmpeg have to be specified too.
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An easy way to get the full list of required libraries in dependency order
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||
is to use @code{pkg-config}.
|
||
|
||
@example
|
||
c99 -o program program.c $(pkg-config --cflags --libs libavformat libavcodec)
|
||
@end example
|
||
|
||
See @file{doc/example/Makefile} and @file{doc/example/pc-uninstalled} for
|
||
more details.
|
||
|
||
@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 create a custom AVIOContext using @code{avio_alloc_context},
|
||
see @file{libavformat/aviobuf.c} in FFmpeg and @file{libmpdemux/demux_lavf.c} in MPlayer or MPlayer2 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
|