ffmpeg -sameq option removed? What to use instead?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: +https://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... +
+ +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 libfoo-dev or libfoo-devel. You can remove it after the +build is finished, but be sure to keep the main package. +
+ +pkg-config find my libraries? Somewhere along with your libraries, there is a .pc file (or several)
+in a pkgconfig directory. You need to set environment variables to
+point pkg-config to these files.
+
If you need to add directories to pkg-config’s search list
+(typical use case: library installed separately), add it to
+$PKG_CONFIG_PATH:
+
export PKG_CONFIG_PATH=/opt/x264/lib/pkgconfig:/opt/opus/lib/pkgconfig +
If you need to replace pkg-config’s search list
+(typical use case: cross-compiling), set it in
+$PKG_CONFIG_LIBDIR:
+
export PKG_CONFIG_LIBDIR=/home/me/cross/usr/lib/pkgconfig:/home/me/cross/usr/local/lib/pkgconfig +
If you need to know the library’s internal dependencies (typical use: static
+linking), add the --static option to pkg-config:
+
./configure --pkg-config-flags=--static +
pkg-config when cross-compiling? The best way is to install pkg-config in your cross-compilation
+environment. It will automatically use the cross-compilation libraries.
+
You can also use pkg-config from the host environment by
+specifying explicitly --pkg-config=pkg-config to configure.
+In that case, you must point pkg-config to the correct directories
+using the PKG_CONFIG_LIBDIR, as explained in the previous entry.
+
As an intermediate solution, you can place in your cross-compilation
+environment a script that calls the host pkg-config with
+PKG_CONFIG_LIBDIR set. That script can look like that:
+
#!/bin/sh +PKG_CONFIG_LIBDIR=/path/to/cross/lib/pkgconfig +export PKG_CONFIG_LIBDIR +exec /usr/bin/pkg-config "$@" +
Try a make distclean in the ffmpeg source directory before the build.
+If this does not help see
+(https://ffmpeg.org/bugreports.html).
+
First, rename your pictures to follow a numerical sequence. +For example, img1.jpg, img2.jpg, img3.jpg,... +Then you may run: +
+ffmpeg -f image2 -i img%d.jpg /tmp/a.mpg +
Notice that ‘%d’ is replaced by the image number. +
+img%03d.jpg means the sequence img001.jpg, img002.jpg, etc. +
+Use the -start_number option to declare a starting number for +the sequence. This is useful if your sequence does not start with +img001.jpg but is still in a numerical order. The following +example will start with img100.jpg: +
+ffmpeg -f image2 -start_number 100 -i img%d.jpg /tmp/a.mpg +
If you have large number of pictures to rename, you can use the
+following command to ease the burden. The command, using the bourne
+shell syntax, symbolically links all files in the current directory
+that match *jpg to the /tmp directory in the sequence of
+img001.jpg, img002.jpg and so on.
+
x=1; for i in *jpg; do counter=$(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. +
+You can also use cat to pipe images to ffmpeg:
+
cat *.jpg | ffmpeg -f image2pipe -c:v mjpeg -i - output.mpg +
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’, ’-mpv_flags qp_rd’, ’-mpv_flags mv0’, ’-mpv_flags skip_rd’. +
+ +’-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’, ’-mpv_flags qp_rd’, ’-mpv_flags mv0’, ’-mpv_flags skip_rd’. +
+ +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. This operation is recommended if you need to re-encode.
+
FFmpeg has a concat demuxer which you can use when you want to avoid a re-encode and
+your format doesn’t support file level concatenation.
+
FFmpeg has a concat protocol designed specifically for that, with examples in the
+documentation.
+
A few multimedia containers (MPEG-1, MPEG-2 PS, DV) allow one 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 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 -c:a 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 -c:a pcm_s16le -ac 2 -ar 44100 - > temp1.a < /dev/null &
+ffmpeg -i input2.flv -vn -f u16le -c:a 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 -c:a pcm_s16le -ac 2 -ar 44100 -i all.a \
+ -f yuv4mpegpipe -i all.v \
+ -y output.flv
+rm temp[12].[av] all.[av]
+Use -dumpgraph - to find out exactly where the channel layout is +lost. +
+Most likely, it is through auto-inserted aresample. 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 aformat explicitly in the filtergraph,
+specifying the exact format.
+
aformat=sample_fmts=s16:channel_layouts=stereo +
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 initially 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.
+
ffmpeg -sameq option removed? What to use instead? The -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 +-qscale and -qmax, but you should peruse the documentation +of the encoder you chose. +
+ +A lot of video codecs and formats can store the aspect ratio of the +video: this is the ratio between the width and the height of either the full +image (DAR, display aspect ratio) or individual pixels (SAR, sample aspect +ratio). For example, EGA screens at resolution 640×350 had 4:3 DAR and 35:48 +SAR. +
+Most still image processing work with square pixels, i.e. 1:1 SAR, but a lot +of video standards, especially from the analogic-numeric transition era, use +non-square pixels. +
+Most processing filters in FFmpeg handle the aspect ratio to avoid +stretching the image: cropping adjusts the DAR to keep the SAR constant, +scaling adjusts the SAR to keep the DAR constant. +
+If you want to stretch, or “unstretch”, the image, you need to override the
+information with the
+setdar or setsar filters.
+
Do not forget to examine carefully the original video to check whether the +stretching comes from the image or from the aspect ratio information. +
+For example, to fix a badly encoded EGA capture, use the following commands, +either the first one to upscale to square pixels or the second one to set +the correct aspect ratio or the third one to avoid transcoding (may not work +depending on the format / codec / player / phase of the moon): +
+ffmpeg -i ega_screen.nut -vf scale=640:480,setsar=1 ega_screen_scaled.nut +ffmpeg -i ega_screen.nut -vf setdar=4/3 ega_screen_anamorphic.nut +ffmpeg -i ega_screen.nut -aspect 4/3 -c copy ega_screen_overridden.nut +
ffmpeg normally checks the console input, for entries like "q" to stop +and "?" to give help, while performing operations. ffmpeg does not have a way of +detecting when it is running as a background task. +When it checks the console input, that can cause the process running ffmpeg +in the background to suspend. +
+To prevent those input checks, allowing ffmpeg to run as a background task,
+use the -nostdin option
+in the ffmpeg invocation. This is effective whether you run ffmpeg in a shell
+or invoke ffmpeg in its own process via an operating system API.
+
As an alternative, when you are running ffmpeg in a shell, you can redirect
+standard input to /dev/null (on Linux and macOS)
+or NUL (on Windows). You can do this redirect either
+on the ffmpeg invocation, or from a shell script which calls ffmpeg.
+
For example: +
+ffmpeg -nostdin -i INPUT OUTPUT +
or (on Linux, macOS, and other UNIX-like shells): +
+ffmpeg -i INPUT OUTPUT </dev/null +
or (on Windows): +
+ffmpeg -i INPUT OUTPUT <NUL +
If you run ffmpeg in the background, you may find that its process suspends. +There may be a message like suspended (tty output). The question is how +to prevent the process from being suspended. +
+For example: +
+% ffmpeg -i INPUT OUTPUT &> ~/tmp/log.txt & +[1] 93352 +% +[1] + suspended (tty output) ffmpeg -i INPUT OUTPUT &> +
The message "tty output" notwithstanding, the problem here is that +ffmpeg normally checks the console input when it runs. The operating system +detects this, and suspends the process until you can bring it to the +foreground and attend to it. +
+The solution is to use the right techniques to tell ffmpeg not to consult
+console input. You can use the
+-nostdin option,
+or redirect standard input with < /dev/null.
+See FAQ
+How do I run ffmpeg as a background task?
+for details.
+
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 create a custom AVIOContext using avio_alloc_context,
+see libavformat/aviobuf.c in FFmpeg and libmpdemux/demux_lavf.c in MPlayer or MPlayer2 sources.
+
see https://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 (it is the least common multiple).
+If you are looking for the average frame rate, see AVStream.avg_frame_rate.
+
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.
+
+ This document was generated using makeinfo. +
+