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FFmpeg Documentation





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1 Introduction

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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2 Quick Start


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2.1 Video and Audio grabbing

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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2.2 Video and Audio file format conversion

* 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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3 Invocation


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3.1 Syntax

The generic syntax is:

ffmpeg [[options][‘-iinput_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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3.2 Main options

-L

show license

-h

show help

-formats

show available formats, codecs, protocols, ...

-f fmt

force format

-i filename

input file name

-y

overwrite output files

-t duration

set the recording time in seconds. hh:mm:ss[.xxx] syntax is also supported.

-title string

set the title

-author string

set the author

-copyright string

set the copyright

-comment string

set the comment

-hq

activate high quality settings


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3.3 Video Options

-b bitrate

set the video bitrate in kbit/s (default = 200 kb/s)

-r fps

set frame rate (default = 25)

-s size

set frame size. The format is ‘WxH’ (default 160x128). The following abbreviations are recognized:

sqcif

128x96

qcif

176x144

cif

352x288

4cif

704x576

-aspect aspect

set aspect ratio (4:3, 16:9 or 1.3333, 1.7777)

-croptop size

set top crop band size (in pixels)

-cropbottom size

set bottom crop band size (in pixels)

-cropleft size

set left crop band size (in pixels)

-cropright size

set right crop band size (in pixels)

-vn

disable video recording

-bt tolerance

set video bitrate tolerance (in kbit/s)

-maxrate bitrate

set max video bitrate tolerance (in kbit/s)

-minrate bitrate

set min video bitrate tolerance (in kbit/s)

-bufsize size

set ratecontrol buffere size (in kbit)

-vcodec codec

force video codec to codec. Use the copy special value to tell that the raw codec data must be copied as is.

-sameq

use same video quality as source (implies VBR)

-pass n

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.

-passlogfile file

select two pass log file name to file.


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3.4 Advanced Video Options

-g gop_size

set the group of picture size

-intra

use only intra frames

-qscale q

use fixed video quantiser scale (VBR)

-qmin q

min video quantiser scale (VBR)

-qmax q

max video quantiser scale (VBR)

-qdiff q

max difference between the quantiser scale (VBR)

-qblur blur

video quantiser scale blur (VBR)

-qcomp compression

video quantiser scale compression (VBR)

-rc_init_cplx complexity

initial complexity for 1-pass encoding

-b_qfactor factor

qp factor between p and b frames

-i_qfactor factor

qp factor between p and i frames

-b_qoffset offset

qp offset between p and b frames

-i_qoffset offset

qp offset between p and i frames

-rc_eq equation

set rate control equation (see section FFmpeg formula evaluator). Default is tex^qComp.

-rc_override override

rate control override for specific intervals

-me method

set motion estimation method to method. Available methods are (from lower to best quality):

zero

Try just the (0, 0) vector.

phods
log
x1
epzs

(default method)

full

exhaustive search (slow and marginally better than epzs)

-dct_algo algo

set dct algorithm to algo. Available values are:

0

FF_DCT_AUTO (default)

1

FF_DCT_FASTINT

2

FF_DCT_INT

3

FF_DCT_MMX

4

FF_DCT_MLIB

5

FF_DCT_ALTIVEC

-idct_algo algo

set idct algorithm to algo. Available values are:

0

FF_IDCT_AUTO (default)

1

FF_IDCT_INT

2

FF_IDCT_SIMPLE

3

FF_IDCT_SIMPLEMMX

4

FF_IDCT_LIBMPEG2MMX

5

FF_IDCT_PS2

6

FF_IDCT_MLIB

7

FF_IDCT_ARM

8

FF_IDCT_ALTIVEC

9

FF_IDCT_SH4

10

FF_IDCT_SIMPLEARM

-er n

set error resilience to n.

1

FF_ER_CAREFULL (default)

2

FF_ER_COMPLIANT

3

FF_ER_AGGRESSIVE

4

FF_ER_VERY_AGGRESSIVE

-ec bit_mask

set error concealment to bit_mask. bit_mask is a bit mask of the following values:

1

FF_EC_GUESS_MVS (default=enabled)

2

FF_EC_DEBLOCK (default=enabled)

-bf frames

use ’frames’ B frames (supported for MPEG-1, MPEG-2 and MPEG-4)

-mbd mode

macroblock decision

0

FF_MB_DECISION_SIMPLE: use mb_cmp (cannot change it yet in ffmpeg)

1

FF_MB_DECISION_BITS: chooses the one which needs the fewest bits

2

FF_MB_DECISION_RD: rate distoration

-4mv

use four motion vector by macroblock (only MPEG-4)

-part

use data partitioning (only MPEG-4)

-bug param

workaround not auto detected encoder bugs

-strict strictness

how strictly to follow the standarts

-aic

enable Advanced intra coding (h263+)

-umv

enable Unlimited Motion Vector (h263+)

-deinterlace

deinterlace pictures

-psnr

calculate PSNR of compressed frames

-vstats

dump video coding statistics to ‘vstats_HHMMSS.log’.

-vhook module

insert video processing module. module contains the module name and its parameters separated by spaces.


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3.5 Audio Options

-ab bitrate

set audio bitrate (in kbit/s)

-ar freq

set the audio sampling freq (default = 44100 Hz)

-ab bitrate

set the audio bitrate in kbit/s (default = 64)

-ac channels

set the number of audio channels (default = 1)

-an

disable audio recording

-acodec codec

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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3.6 Audio/Video grab options

-vd device

set video grab device (e.g. ‘/dev/video0’)

-vc channel

set video grab channel (DV1394 only)

-tvstd standard

set television standard (NTSC, PAL (SECAM))

-dv1394

set DV1394 grab

-ad device

set audio device (e.g. ‘/dev/dsp’)


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3.7 Advanced options

-map file:stream

set input stream mapping

-debug

print specific debug info

-benchmark

add timings for benchmarking

-hex

dump each input packet

-bitexact

only use bit exact algorithms (for codec testing)

-ps size

set packet size in bits

-re

read input at native frame rate. Mainly used to simulate a grab device.

-loop

loop over the input stream. Currently it works only for image streams. This option is used for ffserver automatic testing.


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3.8 FFmpeg formula evaluator

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:

sinh(x)
cosh(x)
tanh(x)
sin(x)
cos(x)
tan(x)
exp(x)
log(x)
squish(x)
gauss(x)
abs(x)
max(x, y)
min(x, y)
gt(x, y)
lt(x, y)
eq(x, y)
bits2qp(bits)
qp2bits(qp)

The following constants are available:

PI
E
iTex
pTex
tex
mv
fCode
iCount
mcVar
var
isI
isP
isB
avgQP
qComp
avgIITex
avgPITex
avgPPTex
avgBPTex
avgTex

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3.9 Protocols

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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4 Tips


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5 Supported File Formats and Codecs

You can use the -formats option to have an exhaustive list.


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5.1 File Formats

FFmpeg supports the following file formats through the libavformat library:

Supported File FormatEncodingDecoding
MPEG audioXX
MPEG1 systemsXX
MPEG2 PSXX
MPEG2 TSX
ASFXX
AVIXX
WAVXX
Macromedia FlashXX
FLVXX
Real Audio and VideoXX
Raw AC3XX
Raw MJPEGXX
Raw MPEG videoXX
Raw PCM8/16 bits, mulaw/AlawXX
SUN AU formatXX
QuicktimeXX
MPEG4XX
Raw MPEG4 videoXX
DVX
4xmX
Playstation STRX
Id RoQX
Interplay MVEX
WC3 MovieX

X means that the encoding (resp. decoding) is supported.


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5.2 Image Formats

FFmpeg can read and write images for each frame of a video sequence. The following image formats are supported:

Supported Image FormatEncodingDecoding
PGM, PPMXX
PAMXX
PGMYUVXX
JPEGXX
.Y.U.VXX
Animated GIFXX
PNGXX

X means that the encoding (resp. decoding) is supported.


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5.3 Video Codecs

Supported CodecEncodingDecodingComments
MPEG1 videoXX
MPEG2 videoXX
MPEG4XXAlso known as DIVX4/5
MSMPEG4 V1XX
MSMPEG4 V2XX
MSMPEG4 V3XXAlso known as DIVX3
WMV7XX
WMV8XXNot completely working
H263(+)XXAlso known as Real Video 1.0
MJPEGXX
DVX
Huff YUVXX
Asus v1XXfourcc: ASV1
Asus v2XXfourcc: ASV2
Creative YUVXfourcc: CYUV
H.264X
Sorenson Video 1Xfourcc: SVQ1
Sorenson Video 3Xfourcc: SVQ3
On2 VP3Xstill experimental
Intel Indeo 3Xonly works on i386 right now
FLVXXFlash H263 variant
ATI VCR1Xfourcc: VCR1
Cirrus Logic AccuPakXfourcc: CLJR
4X VideoXused in certain computer games
Sony Playstation MDECX
Id RoQXused in Quake III, Jedi Knight 2, other computer games
Xan/WC3Xused in Wing Commander III .MVE files
Interplay VideoXused in Interplay .MVE files

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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5.4 Audio Codecs

Supported CodecEncodingDecodingComments
MPEG audio layer 2IXIX
MPEG audio layer 1/3IXIXMP3 encoding is supported through the external library LAME
AC3IXXliba52 is used internally for decoding
VorbisXXsupported through the external library libvorbis
WMA V1/V2X
Microsoft ADPCMXX
MS IMA ADPCMXX
QT IMA ADPCMX
4X IMA ADPCMX
Duck DK3 IMA ADPCMXused in some Sega Saturn console games
Duck DK4 IMA ADPCMXused in some Sega Saturn console games
RA144XReal 14400 bit/s codec
RA288XReal 28800 bit/s codec
AMR-NBXXsupported through an external library
DV audioX
Id RoQ DPCMXused in Quake III, Jedi Knight 2, other computer games
Interplay MVE DPCMXused in various Interplay computer games
Xan DPCMXused in Origin’s Wing Commander IV AVI files

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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6 Platform Specific information


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6.1 Linux

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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6.2 BSD


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6.3 Windows


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6.3.1 Native Windows compilation

Notes:


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6.3.2 Cross compilation for Windows with Linux

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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6.4 MacOS X


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6.5 BeOS

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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7 Developers Guide


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7.1 API


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7.2 Integrating libavcodec or libavformat in your program

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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7.3 Coding Rules

ffmpeg is programmed in ANSI C language. GCC extensions are tolerated. 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.


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7.4 Submitting patches

When you submit your patch, try to send a unified diff (diff ’-u’ 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


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7.5 Regression tests

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.

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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