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FFmpeg is a very fast video and audio converter. It can also grab from a live audio/video source.
The command line interface is designed to be intuitive, in the sense that ffmpeg tries to figure out all the parameters, when possible. You have usually to give only the target bitrate you want.
FFmpeg can also convert from any sample rate to any other, and resize video on the fly with a high quality polyphase filter.
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FFmpeg can use a video4linux compatible video source and any Open Sound System audio source:
ffmpeg /tmp/out.mpg |
Note that you must activate the right video source and channel before launching ffmpeg. You can use any TV viewer such as xawtv (http://bytesex.org/xawtv/) by Gerd Knorr which I find very good. You must also set correctly the audio recording levels with a standard mixer.
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* ffmpeg can use any supported file format and protocol as input:
Examples:
* You can input from YUV files:
ffmpeg -i /tmp/test%d.Y /tmp/out.mpg |
It will use the files:
/tmp/test0.Y, /tmp/test0.U, /tmp/test0.V, /tmp/test1.Y, /tmp/test1.U, /tmp/test1.V, etc... |
The Y files use twice the resolution of the U and V files. They are raw files, without header. They can be generated by all decent video decoders. You must specify the size of the image with the `-s' option if ffmpeg cannot guess it.
* You can input from a RAW YUV420P file:
ffmpeg -i /tmp/test.yuv /tmp/out.avi |
The RAW YUV420P is a file containing RAW YUV planar, for each frame first come the Y plane followed by U and V planes, which are half vertical and horizontal resolution.
* You can output to a RAW YUV420P file:
ffmpeg -i mydivx.avi -o hugefile.yuv |
* You can set several input files and output files:
ffmpeg -i /tmp/a.wav -s 640x480 -i /tmp/a.yuv /tmp/a.mpg |
Convert the audio file a.wav and the raw yuv video file a.yuv to mpeg file a.mpg
* You can also do audio and video conversions at the same time:
ffmpeg -i /tmp/a.wav -ar 22050 /tmp/a.mp2 |
Convert the sample rate of a.wav to 22050 Hz and encode it to MPEG audio.
* You can encode to several formats at the same time and define a mapping from input stream to output streams:
ffmpeg -i /tmp/a.wav -ab 64 /tmp/a.mp2 -ab 128 /tmp/b.mp2 -map 0:0 -map 0:0 |
Convert a.wav to a.mp2 at 64 kbits and b.mp2 at 128 kbits. '-map file:index' specify which input stream is used for each output stream, in the order of the definition of output streams.
* You can transcode decrypted VOBs
ffmpeg -i snatch_1.vob -f avi -vcodec mpeg4 -b 800 -g 300 -bf 2 -acodec mp3 -ab 128 snatch.avi |
This is a typical DVD ripper example, input from a VOB file, output
to an AVI file with MPEG-4 video and MP3 audio, note that in this
command we use B frames so the MPEG-4 stream is DivX5 compatible, GOP
size is 300 that means an INTRA frame every 10 seconds for 29.97 fps
input video. Also the audio stream is MP3 encoded so you need LAME
support which is enabled using --enable-mp3lame
when
configuring. The mapping is particularly useful for DVD transcoding
to get the desired audio language.
NOTE: to see the supported input formats, use ffmpeg -formats
.
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The generic syntax is:
ffmpeg [[options][`-i' input_file]]... {[options] output_file}... |
If no input file is given, audio/video grabbing is done.
As a general rule, options are applied to the next specified file. For example, if you give the `-b 64' option, it sets the video bitrate of the next file. Format option may be needed for raw input files.
By default, ffmpeg tries to convert as losslessly as possible: it uses the same audio and video parameter for the outputs as the one specified for the inputs.
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show license
show help
show available formats, codecs, protocols, ...
force format
input file name
overwrite output files
set the recording time in seconds. hh:mm:ss[.xxx]
syntax is also
supported.
seek to given time position. hh:mm:ss[.xxx]
syntax is also
supported.
set the title
set the author
set the copyright
set the comment
specify target file type ("vcd", "svcd" or "dvd"). All the format options (bitrate, codecs, buffer sizes) are automatically set by this option. You can just type:
ffmpeg -i myfile.avi -target vcd /tmp/vcd.mpg |
activate high quality settings
set the input time offset in seconds. [-]hh:mm:ss[.xxx]
syntax
is also supported. This option affects all the input files that
follow it. The offset is added to the input files' timestamps;
specifying a positive offset means that the corresponding streams are
delayed by 'offset' seconds.
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set the video bitrate in kbit/s (default = 200 kb/s)
set frame rate (default = 25)
set frame size. The format is `WxH' (default 160x128). The following abbreviations are recognized:
128x96
176x144
352x288
704x576
set aspect ratio (4:3, 16:9 or 1.3333, 1.7777)
set top crop band size (in pixels)
set bottom crop band size (in pixels)
set left crop band size (in pixels)
set right crop band size (in pixels)
set top pad band size (in pixels)
set bottom pad band size (in pixels)
set left pad band size (in pixels)
set right pad band size (in pixels)
set right pad band size (hex). The value for pad color is expressed as a six digit hexidecimal number where the first two digits represent red, middle two digits green and last two digits blue. Defaults to 000000 (black)
disable video recording
set video bitrate tolerance (in kbit/s)
set max video bitrate tolerance (in kbit/s)
set min video bitrate tolerance (in kbit/s)
set ratecontrol buffere size (in kbit)
force video codec to codec. Use the copy
special value to
tell that the raw codec data must be copied as is.
use same video quality as source (implies VBR)
select the pass number (1 or 2). It is useful to do two pass encoding. The statistics of the video are recorded in the first pass and the video at the exact requested bit rate is generated in the second pass.
select two pass log file name to file.
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set the group of picture size
use only intra frames
use fixed video quantiser scale (VBR)
min video quantiser scale (VBR)
max video quantiser scale (VBR)
max difference between the quantiser scale (VBR)
video quantiser scale blur (VBR)
video quantiser scale compression (VBR)
initial complexity for 1-pass encoding
qp factor between p and b frames
qp factor between p and i frames
qp offset between p and b frames
qp offset between p and i frames
set rate control equation (see section FFmpeg formula evaluator). Default is tex^qComp
.
rate control override for specific intervals
set motion estimation method to method. Available methods are (from lower to best quality):
Try just the (0, 0) vector.
(default method)
exhaustive search (slow and marginally better than epzs)
set dct algorithm to algo. Available values are:
FF_DCT_AUTO (default)
FF_DCT_FASTINT
FF_DCT_INT
FF_DCT_MMX
FF_DCT_MLIB
FF_DCT_ALTIVEC
set idct algorithm to algo. Available values are:
FF_IDCT_AUTO (default)
FF_IDCT_INT
FF_IDCT_SIMPLE
FF_IDCT_SIMPLEMMX
FF_IDCT_LIBMPEG2MMX
FF_IDCT_PS2
FF_IDCT_MLIB
FF_IDCT_ARM
FF_IDCT_ALTIVEC
FF_IDCT_SH4
FF_IDCT_SIMPLEARM
set error resilience to n.
FF_ER_CAREFULL (default)
FF_ER_COMPLIANT
FF_ER_AGGRESSIVE
FF_ER_VERY_AGGRESSIVE
set error concealment to bit_mask. bit_mask is a bit mask of the following values:
FF_EC_GUESS_MVS (default=enabled)
FF_EC_DEBLOCK (default=enabled)
use 'frames' B frames (supported for MPEG-1, MPEG-2 and MPEG-4)
macroblock decision
FF_MB_DECISION_SIMPLE: use mb_cmp (cannot change it yet in ffmpeg)
FF_MB_DECISION_BITS: chooses the one which needs the fewest bits
FF_MB_DECISION_RD: rate distoration
use four motion vector by macroblock (only MPEG-4)
use data partitioning (only MPEG-4)
workaround not auto detected encoder bugs
how strictly to follow the standarts
enable Advanced intra coding (h263+)
enable Unlimited Motion Vector (h263+)
deinterlace pictures
force interlacing support in encoder (only MPEG-2 and MPEG-4). Use this option if your input file is interlaced and if you want to keep the interlaced format for minimum losses. The alternative is to deinterlace the input stream with `-deinterlace', but deinterlacing introduces more losses.
calculate PSNR of compressed frames
dump video coding statistics to `vstats_HHMMSS.log'.
insert video processing module. module contains the module name and its parameters separated by spaces.
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set audio bitrate (in kbit/s)
set the audio sampling freq (default = 44100 Hz)
set the audio bitrate in kbit/s (default = 64)
set the number of audio channels (default = 1)
disable audio recording
force audio codec to codec. Use the copy
special value to
tell that the raw codec data must be copied as is.
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set video grab device (e.g. `/dev/video0')
set video grab channel (DV1394 only)
set television standard (NTSC, PAL (SECAM))
set DV1394 grab
set audio device (e.g. `/dev/dsp')
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set input stream mapping
print specific debug info
add timings for benchmarking
dump each input packet
only use bit exact algorithms (for codec testing)
set packet size in bits
read input at native frame rate. Mainly used to simulate a grab device.
loop over the input stream. Currently it works only for image streams. This option is used for ffserver automatic testing.
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When evaluating a rate control string, FFmpeg uses an internal formula evaluator.
The following binary operators are available: +
, -
,
*
, /
, ^
.
The following unary operators are available: +
, -
,
(...)
.
The following functions are available:
The following constants are available:
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The filename can be `-' to read from the standard input or to write to the standard output.
ffmpeg handles also many protocols specified with the URL syntax.
Use 'ffmpeg -formats' to have a list of the supported protocols.
The protocol http:
is currently used only to communicate with
ffserver (see the ffserver documentation). When ffmpeg will be a
video player it will also be used for streaming :-)
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ffmpeg -g 3 -r 3 -t 10 -b 50 -s qcif -f rv10 /tmp/b.rm |
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You can use the -formats
option to have an exhaustive list.
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FFmpeg supports the following file formats through the libavformat
library:
Supported File Format | Encoding | Decoding |
MPEG audio | X | X |
MPEG1 systems | X | X |
MPEG2 PS | X | X |
MPEG2 TS | X | |
ASF | X | X |
AVI | X | X |
WAV | X | X |
Macromedia Flash | X | X |
FLV | X | X |
Real Audio and Video | X | X |
Raw AC3 | X | X |
Raw MJPEG | X | X |
Raw MPEG video | X | X |
Raw PCM8/16 bits, mulaw/Alaw | X | X |
Raw CRI ADX audio | X | X |
SUN AU format | X | X |
NUT | X | X |
Quicktime | X | X |
MPEG4 | X | X |
Raw MPEG4 video | X | X |
DV | X | X |
4xm | X | |
Playstation STR | X | |
Id RoQ | X | |
Interplay MVE | X | |
WC3 Movie | X | |
Sega FILM/CPK | X | |
Westwood Studios VQA/AUD | X | |
Id Cinematic (.cin) | X | |
FLIC format | X | |
Sierra VMD | X | |
Matroska | X |
X
means that the encoding (resp. decoding) is supported.
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FFmpeg can read and write images for each frame of a video sequence. The following image formats are supported:
Supported Image Format | Encoding | Decoding |
PGM, PPM | X | X |
PAM | X | X |
PGMYUV | X | X |
JPEG | X | X |
.Y.U.V | X | X |
Animated GIF | X | X |
PNG | X | X |
SGI | X | X |
X
means that the encoding (resp. decoding) is supported.
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Supported Codec | Encoding | Decoding | Comments |
MPEG1 video | X | X | |
MPEG2 video | X | X | |
MPEG4 | X | X | Also known as DIVX4/5 |
MSMPEG4 V1 | X | X | |
MSMPEG4 V2 | X | X | |
MSMPEG4 V3 | X | X | Also known as DIVX3 |
WMV7 | X | X | |
WMV8 | X | X | Not completely working |
H263(+) | X | X | Also known as Real Video 1.0 |
MJPEG | X | X | |
Lossless MJPEG | X | X | |
Apple MJPEG-B | X | ||
Sunplus MJPEG | X | fourcc: SP5X | |
DV | X | X | |
Huff YUV | X | X | |
FFmpeg Video 1 | X | X | Lossless codec (fourcc: FFV1) |
Asus v1 | X | X | fourcc: ASV1 |
Asus v2 | X | X | fourcc: ASV2 |
Creative YUV | X | fourcc: CYUV | |
H.264 | X | ||
Sorenson Video 1 | X | X | fourcc: SVQ1 |
Sorenson Video 3 | X | fourcc: SVQ3 | |
On2 VP3 | X | still experimental | |
Theora | X | still experimental | |
Intel Indeo 3 | X | only works on i386 right now | |
FLV | X | X | Flash H263 variant |
ATI VCR1 | X | fourcc: VCR1 | |
ATI VCR2 | X | fourcc: VCR2 | |
Cirrus Logic AccuPak | X | fourcc: CLJR | |
4X Video | X | used in certain computer games | |
Sony Playstation MDEC | X | ||
Id RoQ | X | used in Quake III, Jedi Knight 2, other computer games | |
Xan/WC3 | X | used in Wing Commander III .MVE files | |
Interplay Video | X | used in Interplay .MVE files | |
Apple Animation | X | fourcc: 'rle ' | |
Apple Graphics | X | fourcc: 'smc ' | |
Apple Video | X | fourcc: rpza | |
Cinepak | X | ||
Microsoft RLE | X | ||
Microsoft Video-1 | X | ||
Westwood VQA | X | ||
Id Cinematic Video | X | used in Quake II | |
Planar RGB | X | fourcc: 8BPS | |
FLIC video | X | ||
Duck TrueMotion v1 | X | fourcc: DUCK | |
VMD Video | X | used in Sierra VMD files | |
MSZH | X | Part of LCL | |
ZLIB | X | X | Part of LCL, encoder experimental |
X
means that the encoding (resp. decoding) is supported.
Check at http://www.mplayerhq.hu/~michael/codec-features.html to get a precise comparison of FFmpeg MPEG4 codec compared to the other solutions.
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Supported Codec | Encoding | Decoding | Comments |
MPEG audio layer 2 | IX | IX | |
MPEG audio layer 1/3 | IX | IX | MP3 encoding is supported through the external library LAME |
AC3 | IX | X | liba52 is used internally for decoding |
Vorbis | X | X | supported through the external library libvorbis |
WMA V1/V2 | X | ||
AAC | X | X | supported through the external library libfaac/libfaad |
Microsoft ADPCM | X | X | |
MS IMA ADPCM | X | X | |
QT IMA ADPCM | X | ||
4X IMA ADPCM | X | ||
G.726 ADPCM | X | X | |
Duck DK3 IMA ADPCM | X | used in some Sega Saturn console games | |
Duck DK4 IMA ADPCM | X | used in some Sega Saturn console games | |
Westwood Studios IMA ADPCM | X | used in Westwood Studios games like Command and Conquer | |
SMJPEG IMA ADPCM | X | used in certain Loki game ports | |
CD-ROM XA ADPCM | X | ||
CRI ADX ADPCM | X | X | used in Sega Dreamcast games |
Electronic Arts ADPCM | X | used in various EA titles | |
RA144 | X | Real 14400 bit/s codec | |
RA288 | X | Real 28800 bit/s codec | |
AMR-NB | X | X | supported through an external library |
AMR-WB | X | X | supported through an external library |
DV audio | X | ||
Id RoQ DPCM | X | used in Quake III, Jedi Knight 2, other computer games | |
Interplay MVE DPCM | X | used in various Interplay computer games | |
Xan DPCM | X | used in Origin's Wing Commander IV AVI files | |
Apple MACE 3 | X | ||
Apple MACE 6 | X | ||
FLAC | X |
X
means that the encoding (resp. decoding) is supported.
I
means that an integer only version is available too (ensures highest
performances on systems without hardware floating point support).
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ffmpeg should be compiled with at least GCC 2.95.3. GCC 3.2 is the preferred compiler now for ffmpeg. All future optimizations will depend on features only found in GCC 3.2.
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Notes:
./configure --enable-shared
when configuring ffmpeg,
you can build `avcodec.dll' and `avformat.dll'. With
make install
you install the FFmpeg DLLs and the associated
headers in `Program Files/FFmpeg'.
./configure --enable-shared
when configuring FFmpeg, then FFmpeg tries to use the Microsoft Visual
C++ lib
tool to build avcodec.lib
and
avformat.lib
. With these libraries, you can link your Visual C++
code directly with the FFmpeg DLLs.
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You must use the MinGW cross compilation tools available at http://www.mingw.org/.
Then configure ffmpeg with the following options:
./configure --enable-mingw32 --cross-prefix=i386-mingw32msvc- |
(you can change the cross-prefix according to the prefix choosen for the MinGW tools).
Then you can easily test ffmpeg with wine (http://www.winehq.com/).
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The configure script should guess the configuration itself. Networking support is currently not finished. errno issues fixed by Andrew Bachmann.
Old stuff:
François Revol - revol at free dot fr - April 2002
The configure script should guess the configuration itself, however I still didn't tested building on net_server version of BeOS.
ffserver is broken (needs poll() implementation).
There is still issues with errno codes, which are negative in BeOs, and that ffmpeg negates when returning. This ends up turning errors into valid results, then crashes. (To be fixed)
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You can integrate all the source code of the libraries to link them statically to avoid any version problem. All you need is to provide a 'config.mak' and a 'config.h' in the parent directory. See the defines generated by ./configure to understand what is needed.
You can use libavcodec or libavformat in your commercial program, but any patch you make must be published. The best way to proceed is to send your patches to the ffmpeg mailing list.
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ffmpeg is programmed in the ISO C90 language with a few additional features from ISO C99, namely:
These features are supported by all compilers we care about, so we won't accept patches to remove their use unless they absolutely don't impair clarity and performance.
All code must compile with gcc 2.95 and gcc 3.3. Currently, ffmpeg also compiles with several other compilers, such as the Compaq ccc compiler or Sun Studio 9, and we would like to keep it that way unless it would be exceedingly involved. To ensure compatibility, please don't use any additional C99 features or gcc extensions. Watch out especially for:
Indent size is 4. The TAB character should not be used. The presentation is the one specified by 'indent -i4 -kr'.
Main priority in ffmpeg is simplicity and small code size (=less bugs).
Comments: for functions visible from other modules, use the JavaDoc format (see examples in `libav/utils.c') so that a documentation can be generated automatically.
fprintf and printf are forbidden in libavformat and libavcodec, please use av_log() instead.
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When you submit your patch, try to send a unified diff (diff '-up' option). I cannot read other diffs :-)
Run the regression tests before submitting a patch so that you can verify that there are no big problems.
Patches should be posted as base64 encoded attachments (or any other encoding which ensures that the patch wont be trashed during transmission) to the ffmpeg-devel mailinglist, see http://lists.sourceforge.net/lists/listinfo/ffmpeg-devel
It also helps quite a bit if you tell us what the patch does (for example 'replaces lrint by lrintf') , and why (for example '*bsd isnt c99 compliant and has no lrint()')
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Before submitting a patch (or committing with CVS), you should at least test that you did not break anything.
The regression test build a synthetic video stream and a synthetic audio stream. Then these are encoded then decoded with all codecs or formats. The CRC (or MD5) of each generated file is recorded in a result file. Then a 'diff' is launched with the reference results and the result file.
The regression test then goes on to test the ffserver code with a limited set of streams. It is important that this step runs correctly as well.
Run 'make test' to test all the codecs and formats.
Run 'make fulltest' to test all the codecs, formats and ffserver.
[Of course, some patches may change the regression tests results. In this case, the regression tests reference results shall be modified accordingly].
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