\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...

@section I have installed this library with my distro's package manager. Why does @command{configure} not see it?

Distributions usually split libraries in several packages. The main package
contains the files necessary to run programs using the library. The
development package contains the files necessary to build programs using the
library. Sometimes, docs and/or data are in a separate package too.

To build FFmpeg, you need to install the development package. It is usually
called @file{libfoo-dev} or @file{libfoo-devel}. You can remove it after the
build is finished, but be sure to keep the main package.

@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.

Use the @option{-start_number} option to declare a starting number for
the sequence. This is useful if your sequence does not start with
@file{img001.jpg} but is still in a numerical order. The following
example will start with @file{img100.jpg}:

@example
ffmpeg -f image2 -start_number 100 -i img%d.jpg /tmp/a.mpg
@end example

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.

You can also use @command{cat} to pipe images to ffmpeg:

@example
cat *.jpg | ffmpeg -f image2pipe -c:v mjpeg -i - output.mpg
@end example

@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?

To "join" video files is quite ambiguous. The following list explains the
different kinds of "joining" and points out how those are addressed in
FFmpeg. To join video files may mean:

@itemize

@item
To put them one after the other: this is called to @emph{concatenate} them
(in short: concat) and is addressed
@ref{How can I concatenate video files, in this very faq}.

@item
To put them together in the same file, to let the user choose between the
different versions (example: different audio languages): this is called to
@emph{multiplex} them together (in short: mux), and is done by simply
invoking ffmpeg with several @option{-i} options.

@item
For audio, to put all channels together in a single stream (example: two
mono streams into one stereo stream): this is sometimes called to
@emph{merge} them, and can be done using the
@url{http://ffmpeg.org/ffmpeg-filters.html#amerge, @code{amerge}} filter.

@item
For audio, to play one on top of the other: this is called to @emph{mix}
them, and can be done by first merging them into a single stream and then
using the @url{http://ffmpeg.org/ffmpeg-filters.html#pan, @code{pan}} filter to mix
the channels at will.

@item
For video, to display both together, side by side or one on top of a part of
the other; it can be done using the
@url{http://ffmpeg.org/ffmpeg-filters.html#overlay, @code{overlay}} video filter.

@end itemize

@anchor{How can I concatenate video files}
@section How can I concatenate video files?

There are several solutions, depending on the exact circumstances.

@subsection Concatenating using the concat @emph{filter}

FFmpeg has a @url{http://ffmpeg.org/ffmpeg-filters.html#concat,
@code{concat}} filter designed specifically for that, with examples in the
documentation. This operation is recommended if you need to re-encode.

@subsection Concatenating using the concat @emph{demuxer}

FFmpeg has a @url{http://www.ffmpeg.org/ffmpeg-formats.html#concat,
@code{concat}} demuxer which you can use when you want to avoid a re-encode and
your format doesn't support file level concatenation.

@subsection Concatenating using the concat @emph{protocol} (file level)

FFmpeg has a @url{http://ffmpeg.org/ffmpeg-protocols.html#concat,
@code{concat}} protocol designed specifically for that, with examples in the
documentation.

A few multimedia containers (MPEG-1, MPEG-2 PS, DV) allow to concatenate
video by merely concatenating the files containing 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 -qscale:v 1 intermediate1.mpg
ffmpeg -i input2.avi -qscale:v 1 intermediate2.mpg
cat intermediate1.mpg intermediate2.mpg > intermediate_all.mpg
ffmpeg -i intermediate_all.mpg -qscale:v 2 output.avi
@end example

Additionally, you can use the @code{concat} protocol instead of @code{cat} or
@code{copy} which will avoid creation of a potentially huge intermediate file.

@example
ffmpeg -i input1.avi -qscale:v 1 intermediate1.mpg
ffmpeg -i input2.avi -qscale:v 1 intermediate2.mpg
ffmpeg -i concat:"intermediate1.mpg|intermediate2.mpg" -c copy intermediate_all.mpg
ffmpeg -i intermediate_all.mpg -qscale:v 2 output.avi
@end example

Note that you may need to escape the character "|" which is special for many
shells.

Another option is usage of named pipes, should your platform support it:

@example
mkfifo intermediate1.mpg
mkfifo intermediate2.mpg
ffmpeg -i input1.avi -qscale:v 1 -y intermediate1.mpg < /dev/null &
ffmpeg -i input2.avi -qscale:v 1 -y intermediate2.mpg < /dev/null &
cat intermediate1.mpg intermediate2.mpg |\
ffmpeg -f mpeg -i - -c:v mpeg4 -acodec libmp3lame output.avi
@end example

@subsection Concatenating using raw audio and video

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 concatenate 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 \
       -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 aresample}. 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{aformat} explicitly in the filtergraph,
specifying the exact format.

@example
aformat=sample_fmts=s16:channel_layouts=stereo
@end example

@section Why does FFmpeg not see the subtitles in my VOB file?

VOB and a few other formats do not have a global header that describes
everything present in the file. Instead, applications are supposed to scan
the file to see what it contains. Since VOB files are frequently large, only
the beginning is scanned. If the subtitles happen only later in the file,
they will not be initally detected.

Some applications, including the @code{ffmpeg} command-line tool, can only
work with streams that were detected during the initial scan; streams that
are detected later are ignored.

The size of the initial scan is controlled by two options: @code{probesize}
(default ~5 Mo) and @code{analyzeduration} (default 5,000,000 µs = 5 s). For
the subtitle stream to be detected, both values must be large enough.

@section Why was the @command{ffmpeg} @option{-sameq} option removed? What to use instead?

The @option{-sameq} option meant "same quantizer", and made sense only in a
very limited set of cases. Unfortunately, a lot of people mistook it for
"same quality" and used it in places where it did not make sense: it had
roughly the expected visible effect, but achieved it in a very inefficient
way.

Each encoder has its own set of options to set the quality-vs-size balance,
use the options for the encoder you are using to set the quality level to a
point acceptable for your tastes. The most common options to do that are
@option{-qscale} and @option{-qmax}, but you should peruse the documentation
of the encoder you chose.

@chapter Development

@section Are there examples illustrating how to use the FFmpeg libraries, particularly libavcodec and libavformat?

Yes. Check the @file{doc/examples} directory in the source
repository, also available online at:
@url{https://github.com/FFmpeg/FFmpeg/tree/master/doc/examples}.

Examples are also installed by default, usually in
@code{$PREFIX/share/ffmpeg/examples}.

Also you may 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?

Yes. Please see the @uref{platform.html, Microsoft Visual C++}
section in the FFmpeg documentation.

@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 @command{ffmpeg_g}, @command{ffplay_g}, etc. which
contain full debug information. Those binaries are stripped to create
@command{ffmpeg}, @command{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 from the libraries themselves.

FFmpeg builds static libraries by default. In static libraries, dependencies
are not handled. That has two consequences. First, you must specify the
libraries in dependency order: @code{-lavdevice} must come before
@code{-lavformat}, @code{-lavutil} must come after everything else, etc.
Second, external libraries that are used in FFmpeg have to be specified too.

An easy way to get the full list of required libraries in dependency order
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 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.

@code{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 @code{r_frame_rate}
will be 150 (it is the least common multiple).
If you are looking for the average frame rate, see @code{AVStream.avg_frame_rate}.

@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
