make fate not running all tests?make fate not finding the samples?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.
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.
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.
Windows does not support standard formats like MPEG very well, unless you install some additional codecs.
The following list of video codecs should work on most Windows systems:
.avi/.asf
.asf only
.asf only
.asf only
Only if you have some MPEG-4 codec like ffdshow or Xvid installed.
.mpg only
Note, ASF files often have .wmv or .wma extensions in Windows. It should also be mentioned that Microsoft claims a patent on the ASF format, and may sue or threaten users who create ASF files with non-Microsoft software. It is strongly advised to avoid ASF where possible.
The following list of audio codecs should work on most Windows systems:
always
If some MP3 codec like LAME is installed.
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: 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...
Try a make distclean in the ffmpeg source directory before the build.
If this does not help see
(http://ffmpeg.org/bugreports.html).
First, rename your pictures to follow a numerical sequence. For example, img1.jpg, img2.jpg, img3.jpg,... Then you may run:
ffmpeg -f image2 -i img%d.jpg /tmp/a.mpg |
Notice that ‘%d’ is replaced by the image number.
‘img%03d.jpg’ means the sequence ‘img001.jpg’, ‘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 *jpg to the ‘/tmp’ directory in the sequence of
‘img001.jpg’, ‘img002.jpg’ and so on.
x=1; for i in *jpg; do counter=$(printf %03d $x); ln -s "$i" /tmp/img"$counter".jpg; x=$(($x+1)); done |
If you want to sequence them by oldest modified first, substitute
$(ls -r -t *jpg) in place of *jpg.
Then run:
ffmpeg -f image2 -i /tmp/img%03d.jpg /tmp/a.mpg |
The same logic is used for any image format that ffmpeg reads.
Use:
ffmpeg -i movie.mpg movie%d.jpg |
The ‘movie.mpg’ used as input will be converted to ‘movie1.jpg’, ‘movie2.jpg’, etc...
Instead of relying on file format self-recognition, you may also use
to force the encoding.
Applying that to the previous example:
ffmpeg -i movie.mpg -f image2 -c:v mjpeg menu%d.jpg |
Beware that there is no "jpeg" codec. Use "mjpeg" instead.
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.
Use ‘-’ as file name.
Try ’-f image2 test%d.jpg’.
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.
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.
’-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’.
’-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.
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.
If you have built FFmpeg with ./configure --enable-avisynth
(only possible on MinGW/Cygwin platforms),
then you may use any file that DirectShow can read as input.
Just create an "input.avs" text file with this single line ...
DirectShowSource("C:\path to your file\yourfile.asf")
|
... and then feed that text file to ffmpeg:
ffmpeg -i input.avs |
For ANY other help on Avisynth, please visit the Avisynth homepage.
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:
amerge filter.
pan filter to mix
the channels at will.
overlay video filter.
There are several solutions, depending on the exact circumstances.
FFmpeg has a concat
filter 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 them.
Hence you may concatenate your multimedia files by first transcoding them to
these privileged formats, then using the humble cat command (or the
equally humble copy under Windows), and finally transcoding back to your
format of choice.
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 |
Additionally, you can use the concat protocol instead of cat or
copy which will avoid creation of a potentially huge intermediate file.
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 |
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:
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 |
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
tail as seen below. Note that when piping through tail you
must use command grouping, { ;}, to background properly.
For example, let’s say we want to concatenate two FLV files into an output.flv file:
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]
|
ffmpeg prints an error like
Undefined constant or missing '(' in 'baseline'
Unable to parse option value "baseline"
Error setting option profile to value baseline.
|
Short answer: write ‘-profile:v’ instead of ‘-profile’.
Long answer: this happens because the ‘-profile’ option can apply to both video and audio. Specifically the AAC encoder also defines some profiles, none of which are named baseline.
The solution is to apply the ‘-profile’ option to the video stream only
by using Stream specifiers.
Appending :v to it will do exactly that.
Use ‘-dumpgraph -’ to find out exactly where the channel layout is lost.
Most likely, it is through auto-inserted aconvert. Try to understand
why the converting filter was needed at that place.
Just before the output is a likely place, as ‘-f lavfi’ currently only support packed S16.
Then insert the correct aconvert explicitly in the filter graph,
specifying the exact format.
aconvert=s16:stereo:packed |
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 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: probesize
(default ~5 Mo) and analyzeduration (default 5,000,000 µs = 5 s). For
the subtitle stream to be detected, both values must be large enough.
Yes. Check the ‘doc/examples’ directory in the source repository, also available online at: https://github.com/FFmpeg/FFmpeg/tree/master/doc/examples.
Examples are also installed by default, usually in
$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 (projects.html).
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 #ifdefs related to the compiler.
Yes. Please see the Microsoft Visual C++ section in the FFmpeg documentation.
No. These tools are too bloated and they complicate the build.
FFmpeg is already organized in a highly modular manner and does not need to be rewritten in a formal object language. Further, many of the developers favor straight C; it works for them. For more arguments on this matter, read "Programming Religion".
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.
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.
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: -lavdevice must come before
-lavformat, -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 pkg-config.
c99 -o program program.c $(pkg-config --cflags --libs libavformat libavcodec) |
See ‘doc/example/Makefile’ and ‘doc/example/pc-uninstalled’ for more details.
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 extern "C".
See http://www.parashift.com/c++-faq-lite/mixing-c-and-cpp.html#faq-32.3
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
You have to implement a URLProtocol, see ‘libavformat/file.c’ in FFmpeg and ‘libmpdemux/demux_lavf.c’ in MPlayer sources.
see http://www.iversenit.dk/dev/ffmpeg-headers/
see http://www.ffmpeg.org/~michael/
Even if peculiar since it is network oriented, RTP is a container like any other. You have to demux RTP before feeding the payload to libavcodec. In this specific case please look at RFC 4629 to see how it should be done.
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.
make fate not running all tests?Make sure you have the fate-suite samples and the SAMPLES Make variable
or FATE_SAMPLES environment variable or the --samples
configure option is set to the right path.
make fate not finding the samples?Do you happen to have a ~ 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 ~ by the full path.