doc/faq.texi - third_party/ffmpeg - Git at Google

 \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
	\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