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Originally committed as revision 2155 to svn://svn.ffmpeg.org/ffmpeg/trunk
850 lines
23 KiB
Plaintext
850 lines
23 KiB
Plaintext
\input texinfo @c -*- texinfo -*-
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@settitle FFmpeg Documentation
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@titlepage
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@sp 7
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@center @titlefont{FFmpeg Documentation}
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@sp 3
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@end titlepage
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@chapter Introduction
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FFmpeg is a very fast video and audio converter. It can also grab from
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a live audio/video source.
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The command line interface is designed to be intuitive, in the sense
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that ffmpeg tries to figure out all the parameters, when
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possible. You have usually to give only the target bitrate you want.
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FFmpeg can also convert from any sample rate to any other, and resize
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video on the fly with a high quality polyphase filter.
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@chapter Quick Start
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@c man begin EXAMPLES
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@section Video and Audio grabbing
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FFmpeg can use a video4linux compatible video source and any Open Sound
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System audio source:
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@example
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ffmpeg /tmp/out.mpg
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@end example
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Note that you must activate the right video source and channel before
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launching ffmpeg. You can use any TV viewer such as xawtv
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(@url{http://bytesex.org/xawtv/}) by Gerd Knorr which I find very
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good. You must also set correctly the audio recording levels with a
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standard mixer.
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@section Video and Audio file format conversion
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* ffmpeg can use any supported file format and protocol as input:
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Examples:
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* You can input from YUV files:
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@example
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ffmpeg -i /tmp/test%d.Y /tmp/out.mpg
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@end example
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It will use the files:
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@example
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/tmp/test0.Y, /tmp/test0.U, /tmp/test0.V,
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/tmp/test1.Y, /tmp/test1.U, /tmp/test1.V, etc...
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@end example
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The Y files use twice the resolution of the U and V files. They are
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raw files, without header. They can be generated by all decent video
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decoders. You must specify the size of the image with the @option{-s} option
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if ffmpeg cannot guess it.
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* You can input from a RAW YUV420P file:
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@example
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ffmpeg -i /tmp/test.yuv /tmp/out.avi
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@end example
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The RAW YUV420P is a file containing RAW YUV planar, for each frame first
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come the Y plane followed by U and V planes, which are half vertical and
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horizontal resolution.
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* You can output to a RAW YUV420P file:
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@example
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ffmpeg -i mydivx.avi -o hugefile.yuv
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@end example
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* You can set several input files and output files:
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@example
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ffmpeg -i /tmp/a.wav -s 640x480 -i /tmp/a.yuv /tmp/a.mpg
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@end example
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Convert the audio file a.wav and the raw yuv video file a.yuv
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to mpeg file a.mpg
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* You can also do audio and video conversions at the same time:
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@example
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ffmpeg -i /tmp/a.wav -ar 22050 /tmp/a.mp2
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@end example
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Convert the sample rate of a.wav to 22050 Hz and encode it to MPEG audio.
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* You can encode to several formats at the same time and define a
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mapping from input stream to output streams:
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@example
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ffmpeg -i /tmp/a.wav -ab 64 /tmp/a.mp2 -ab 128 /tmp/b.mp2 -map 0:0 -map 0:0
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@end example
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Convert a.wav to a.mp2 at 64 kbits and b.mp2 at 128 kbits. '-map
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file:index' specify which input stream is used for each output
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stream, in the order of the definition of output streams.
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* You can transcode decrypted VOBs
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@example
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ffmpeg -i snatch_1.vob -f avi -vcodec mpeg4 -b 800 -g 300 -bf 2 -acodec mp3 -ab 128 snatch.avi
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@end example
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This is a typical DVD ripper example, input from a VOB file, output
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to an AVI file with MPEG-4 video and MP3 audio, note that in this
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command we use B frames so the MPEG-4 stream is DivX5 compatible, GOP
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size is 300 that means an INTRA frame every 10 seconds for 29.97 fps
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input video. Also the audio stream is MP3 encoded so you need LAME
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support which is enabled using @code{--enable-mp3lame} when
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configuring. The mapping is particularly useful for DVD transcoding
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to get the desired audio language.
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NOTE: to see the supported input formats, use @code{ffmpeg -formats}.
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@c man end
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@chapter Invocation
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@section Syntax
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The generic syntax is:
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@example
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@c man begin SYNOPSIS
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ffmpeg [[options][@option{-i} @var{input_file}]]... @{[options] @var{output_file}@}...
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@c man end
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@end example
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@c man begin DESCRIPTION
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If no input file is given, audio/video grabbing is done.
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As a general rule, options are applied to the next specified
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file. For example, if you give the @option{-b 64} option, it sets the video
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bitrate of the next file. Format option may be needed for raw input
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files.
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By default, ffmpeg tries to convert as losslessly as possible: it
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uses the same audio and video parameter for the outputs as the one
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specified for the inputs.
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@c man end
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@c man begin OPTIONS
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@section Main options
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@table @option
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@item -L
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show license
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@item -h
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show help
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@item -formats
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show available formats, codecs, protocols, ...
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@item -f fmt
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force format
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@item -i filename
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input file name
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@item -y
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overwrite output files
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@item -t duration
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set the recording time in seconds. @code{hh:mm:ss[.xxx]} syntax is also
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supported.
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@item -title string
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set the title
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@item -author string
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set the author
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@item -copyright string
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set the copyright
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@item -comment string
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set the comment
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@item -hq
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activate high quality settings
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@end table
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@section Video Options
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@table @option
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@item -b bitrate
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set the video bitrate in kbit/s (default = 200 kb/s)
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@item -r fps
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set frame rate (default = 25)
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@item -s size
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set frame size. The format is @samp{WxH} (default 160x128). The
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following abbreviations are recognized:
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@table @samp
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@item sqcif
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128x96
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@item qcif
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176x144
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@item cif
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352x288
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@item 4cif
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704x576
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@end table
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@item -aspect aspect
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set aspect ratio (4:3, 16:9 or 1.3333, 1.7777)
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@item -croptop size
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set top crop band size (in pixels)
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@item -cropbottom size
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set bottom crop band size (in pixels)
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@item -cropleft size
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set left crop band size (in pixels)
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@item -cropright size
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set right crop band size (in pixels)
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@item -vn
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disable video recording
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@item -bt tolerance
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set video bitrate tolerance (in kbit/s)
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@item -maxrate bitrate
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set max video bitrate tolerance (in kbit/s)
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@item -minrate bitrate
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set min video bitrate tolerance (in kbit/s)
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@item -bufsize size
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set ratecontrol buffere size (in kbit)
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@item -vcodec codec
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force video codec to @var{codec}. Use the @code{copy} special value to
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tell that the raw codec data must be copied as is.
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@item -sameq
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use same video quality as source (implies VBR)
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@item -pass n
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select the pass number (1 or 2). It is useful to do two pass
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encoding. The statistics of the video are recorded in the first pass and
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the video at the exact requested bit rate is generated in the second
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pass.
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@item -passlogfile file
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select two pass log file name to @var{file}.
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@end table
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@section Advanced Video Options
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@table @option
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@item -g gop_size
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set the group of picture size
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@item -intra
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use only intra frames
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@item -qscale q
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use fixed video quantiser scale (VBR)
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@item -qmin q
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min video quantiser scale (VBR)
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@item -qmax q
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max video quantiser scale (VBR)
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@item -qdiff q
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max difference between the quantiser scale (VBR)
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@item -qblur blur
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video quantiser scale blur (VBR)
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@item -qcomp compression
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video quantiser scale compression (VBR)
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@item -rc_init_cplx complexity
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initial complexity for 1-pass encoding
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@item -b_qfactor factor
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qp factor between p and b frames
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@item -i_qfactor factor
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qp factor between p and i frames
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@item -b_qoffset offset
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qp offset between p and b frames
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@item -i_qoffset offset
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qp offset between p and i frames
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@item -rc_eq equation
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set rate control equation (@pxref{FFmpeg formula
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evaluator}). Default is @code{tex^qComp}.
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@item -rc_override override
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rate control override for specific intervals
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@item -me method
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set motion estimation method to @var{method}. Available methods are
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(from lower to best quality):
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@table @samp
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@item zero
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Try just the (0, 0) vector.
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@item phods
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@item log
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@item x1
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@item epzs
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(default method)
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@item full
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exhaustive search (slow and marginally better than epzs)
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@end table
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@item -dct_algo algo
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set dct algorithm to @var{algo}. Available values are:
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@table @samp
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@item 0
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FF_DCT_AUTO (default)
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@item 1
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FF_DCT_FASTINT
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@item 2
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FF_DCT_INT
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@item 3
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FF_DCT_MMX
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@item 4
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FF_DCT_MLIB
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@item 5
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FF_DCT_ALTIVEC
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@end table
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@item -idct_algo algo
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set idct algorithm to @var{algo}. Available values are:
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@table @samp
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@item 0
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FF_IDCT_AUTO (default)
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@item 1
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FF_IDCT_INT
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@item 2
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FF_IDCT_SIMPLE
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@item 3
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FF_IDCT_SIMPLEMMX
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@item 4
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FF_IDCT_LIBMPEG2MMX
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@item 5
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FF_IDCT_PS2
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@item 6
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FF_IDCT_MLIB
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@item 7
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FF_IDCT_ARM
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@item 8
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FF_IDCT_ALTIVEC
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@item 9
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FF_IDCT_SH4
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@item 10
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FF_IDCT_SIMPLEARM
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@end table
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@item -er n
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set error resilience to @var{n}.
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@table @samp
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@item 1
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FF_ER_CAREFULL (default)
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@item 2
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FF_ER_COMPLIANT
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@item 3
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FF_ER_AGGRESSIVE
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@item 4
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FF_ER_VERY_AGGRESSIVE
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@end table
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@item -ec bit_mask
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set error concealment to @var{bit_mask}. @var{bit_mask} is a bit mask of
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the following values:
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@table @samp
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@item 1
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FF_EC_GUESS_MVS (default=enabled)
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@item 2
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FF_EC_DEBLOCK (default=enabled)
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@end table
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@item -bf frames
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use 'frames' B frames (only MPEG-4)
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@item -mbd mode
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macroblock decision
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@table @samp
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@item 0
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FF_MB_DECISION_SIMPLE: use mb_cmp (cannot change it yet in ffmpeg)
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@item 1
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FF_MB_DECISION_BITS: chooses the one which needs the fewest bits
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@item 2
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FF_MB_DECISION_RD: rate distoration
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@end table
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@item -4mv
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use four motion vector by macroblock (only MPEG-4)
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@item -part
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use data partitioning (only MPEG-4)
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@item -bug param
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workaround not auto detected encoder bugs
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@item -strict strictness
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how strictly to follow the standarts
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@item -aic
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enable Advanced intra coding (h263+)
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@item -umv
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enable Unlimited Motion Vector (h263+)
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@item -deinterlace
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deinterlace pictures
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@item -psnr
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calculate PSNR of compressed frames
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@item -vstats
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dump video coding statistics to @file{vstats_HHMMSS.log}.
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@item -vhook module
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insert video processing @var{module}. @var{module} contains the module
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name and its parameters separated by spaces.
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@end table
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@section Audio Options
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@table @option
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@item -ab bitrate
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set audio bitrate (in kbit/s)
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@item -ar freq
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set the audio sampling freq (default = 44100 Hz)
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@item -ab bitrate
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set the audio bitrate in kbit/s (default = 64)
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@item -ac channels
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set the number of audio channels (default = 1)
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@item -an
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disable audio recording
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@item -acodec codec
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force audio codec to @var{codec}. Use the @code{copy} special value to
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tell that the raw codec data must be copied as is.
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@end table
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@section Audio/Video grab options
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@table @option
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@item -vd device
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set video grab device (e.g. @file{/dev/video0})
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@item -vc channel
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set video grab channel (DV1394 only)
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@item -tvstd standard
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set television standard (NTSC, PAL (SECAM))
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@item -dv1394
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set DV1394 grab
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@item -ad device
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set audio device (e.g. @file{/dev/dsp})
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@end table
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@section Advanced options
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@table @option
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@item -map file:stream
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set input stream mapping
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@item -debug
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print specific debug info
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@item -benchmark
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add timings for benchmarking
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@item -hex
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dump each input packet
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@item -bitexact
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only use bit exact algorithms (for codec testing)
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@item -ps size
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set packet size in bits
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@item -re
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read input at native frame rate. Mainly used to simulate a grab device.
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@item -loop
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loop over the input stream. Currently it works only for image
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streams. This option is used for ffserver automatic testing.
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@end table
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@node FFmpeg formula evaluator
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@section FFmpeg formula evaluator
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When evaluating a rate control string, FFmpeg uses an internal formula
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evaluator.
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The following binary operators are available: @code{+}, @code{-},
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@code{*}, @code{/}, @code{^}.
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The following unary operators are available: @code{+}, @code{-},
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@code{(...)}.
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The following functions are available:
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@table @var
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@item sinh(x)
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@item cosh(x)
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@item tanh(x)
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@item sin(x)
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@item cos(x)
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@item tan(x)
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@item exp(x)
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@item log(x)
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@item squish(x)
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@item gauss(x)
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@item abs(x)
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@item max(x, y)
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@item min(x, y)
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@item gt(x, y)
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@item lt(x, y)
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@item eq(x, y)
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@item bits2qp(bits)
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@item qp2bits(qp)
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@end table
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The following constants are available:
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@table @var
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@item PI
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@item E
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@item iTex
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@item pTex
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@item tex
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@item mv
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@item fCode
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@item iCount
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@item mcVar
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@item var
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@item isI
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@item isP
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@item isB
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@item avgQP
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@item qComp
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@item avgIITex
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@item avgPITex
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@item avgPPTex
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@item avgBPTex
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@item avgTex
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@end table
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@c man end
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@ignore
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@setfilename ffmpeg
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@settitle FFmpeg video converter
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@c man begin SEEALSO
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ffserver(1), ffplay(1) and the html documentation of @file{ffmpeg}.
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@c man end
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@c man begin AUTHOR
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Fabrice Bellard
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@c man end
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@end ignore
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@section Protocols
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The filename can be @file{-} to read from the standard input or to write
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to the standard output.
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ffmpeg handles also many protocols specified with the URL syntax.
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Use 'ffmpeg -formats' to have a list of the supported protocols.
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The protocol @code{http:} is currently used only to communicate with
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ffserver (see the ffserver documentation). When ffmpeg will be a
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video player it will also be used for streaming :-)
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@chapter Tips
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@itemize
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@item For streaming at very low bit rate application, use a low frame rate
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and a small gop size. This is especially true for real video where
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the Linux player does not seem to be very fast, so it can miss
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frames. An example is:
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@example
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ffmpeg -g 3 -r 3 -t 10 -b 50 -s qcif -f rv10 /tmp/b.rm
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@end example
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@item The parameter 'q' which is displayed while encoding is the current
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quantizer. The value of 1 indicates that a very good quality could
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be achieved. The value of 31 indicates the worst quality. If q=31
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too often, it means that the encoder cannot compress enough to meet
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your bit rate. You must either increase the bit rate, decrease the
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frame rate or decrease the frame size.
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|
||
@item If your computer is not fast enough, you can speed up the
|
||
compression at the expense of the compression ratio. You can use
|
||
'-me zero' to speed up motion estimation, and '-intra' to disable
|
||
completely motion estimation (you have only I frames, which means it
|
||
is about as good as JPEG compression).
|
||
|
||
@item To have very low bitrates in audio, reduce the sampling frequency
|
||
(down to 22050 kHz for mpeg audio, 22050 or 11025 for ac3).
|
||
|
||
@item To have a constant quality (but a variable bitrate), use the option
|
||
'-qscale n' when 'n' is between 1 (excellent quality) and 31 (worst
|
||
quality).
|
||
|
||
@item When converting video files, you can use the '-sameq' option which
|
||
uses in the encoder the same quality factor than in the decoder. It
|
||
allows to be almost lossless in encoding.
|
||
|
||
@end itemize
|
||
|
||
@chapter Supported File Formats and Codecs
|
||
|
||
You can use the @code{-formats} option to have an exhaustive list.
|
||
|
||
@section File Formats
|
||
|
||
FFmpeg supports the following file formats through the @code{libavformat}
|
||
library:
|
||
|
||
@multitable @columnfractions .4 .1 .1
|
||
@item Supported File Format @tab Encoding @tab Decoding @tab Comments
|
||
@item MPEG audio @tab X @tab X
|
||
@item MPEG1 systems @tab X @tab X
|
||
@tab muxed audio and video
|
||
@item MPEG2 PS @tab X @tab X
|
||
@tab also known as @code{VOB} file
|
||
@item MPEG2 TS @tab @tab X
|
||
@tab also known as DVB Transport Stream
|
||
@item ASF@tab X @tab X
|
||
@item AVI@tab X @tab X
|
||
@item WAV@tab X @tab X
|
||
@item Macromedia Flash@tab X @tab X
|
||
@tab Only embedded audio is decoded
|
||
@item FLV @tab X @tab X
|
||
@tab Macromedia Flash video files
|
||
@item Real Audio and Video @tab X @tab X
|
||
@item Raw AC3 @tab X @tab X
|
||
@item Raw MJPEG @tab X @tab X
|
||
@item Raw MPEG video @tab X @tab X
|
||
@item Raw PCM8/16 bits, mulaw/Alaw@tab X @tab X
|
||
@item SUN AU format @tab X @tab X
|
||
@item Quicktime @tab @tab X
|
||
@item MPEG4 @tab @tab X
|
||
@tab MPEG4 is a variant of Quicktime
|
||
@item Raw MPEG4 video @tab X @tab X
|
||
@item DV @tab @tab X
|
||
@item 4xm @tab @tab X
|
||
@tab 4X Technologies format, used in some games
|
||
@end multitable
|
||
|
||
@code{X} means that the encoding (resp. decoding) is supported.
|
||
|
||
@section Image Formats
|
||
|
||
FFmpeg can read and write images for each frame of a video sequence. The
|
||
following image formats are supported:
|
||
|
||
@multitable @columnfractions .4 .1 .1
|
||
@item Supported Image Format @tab Encoding @tab Decoding @tab Comments
|
||
@item PGM, PPM @tab X @tab X
|
||
@item PAM @tab X @tab X @tab PAM is a PNM extension with alpha support
|
||
@item PGMYUV @tab X @tab X @tab PGM with U and V components in YUV 4:2:0
|
||
@item JPEG @tab X @tab X @tab Progressive JPEG is not supported
|
||
@item .Y.U.V @tab X @tab X @tab One raw file per component
|
||
@item Animated GIF @tab X @tab X @tab Only uncompressed GIFs are generated
|
||
@item PNG @tab X @tab X @tab 2 bit and 4 bit/pixel not supported yet
|
||
@end multitable
|
||
|
||
@code{X} means that the encoding (resp. decoding) is supported.
|
||
|
||
@section Video Codecs
|
||
|
||
@multitable @columnfractions .4 .1 .1 .7
|
||
@item Supported Codec @tab Encoding @tab Decoding @tab Comments
|
||
@item MPEG1 video @tab X @tab X
|
||
@item MPEG2 video @tab @tab X
|
||
@item MPEG4 @tab X @tab X @tab Also known as DIVX4/5
|
||
@item MSMPEG4 V1 @tab X @tab X
|
||
@item MSMPEG4 V2 @tab X @tab X
|
||
@item MSMPEG4 V3 @tab X @tab X @tab Also known as DIVX3
|
||
@item WMV7 @tab X @tab X
|
||
@item WMV8 @tab X @tab X @tab Not completely working
|
||
@item H263(+) @tab X @tab X @tab Also known as Real Video 1.0
|
||
@item MJPEG @tab X @tab X
|
||
@item DV @tab @tab X
|
||
@item Huff YUV @tab X @tab X
|
||
@item Asus v1 @tab X @tab X @tab fourcc: ASV1
|
||
@item Creative YUV @tab @tab X @tab fourcc: CYUV
|
||
@item H.264 @tab @tab X
|
||
@item Sorenson Video 1 @tab @tab X @tab fourcc: SVQ1
|
||
@item Sorenson Video 3 @tab @tab X @tab fourcc: SVQ3
|
||
@item On2 VP3 @tab @tab X @tab still experimental
|
||
@item Intel Indeo 3 @tab @tab X @tab only works on i386 right now
|
||
@item FLV @tab X @tab X @tab Flash H263 variant
|
||
@end multitable
|
||
|
||
@code{X} means that the encoding (resp. decoding) is supported.
|
||
|
||
Check at @url{http://www.mplayerhq.hu/~michael/codec-features.html} to
|
||
get a precise comparison of FFmpeg MPEG4 codec compared to the other
|
||
solutions.
|
||
|
||
@section Audio Codecs
|
||
|
||
@multitable @columnfractions .4 .1 .1 .1 .7
|
||
@item Supported Codec @tab Encoding @tab Decoding @tab Comments
|
||
@item MPEG audio layer 2 @tab IX @tab IX
|
||
@item MPEG audio layer 1/3 @tab IX @tab IX
|
||
@tab MP3 encoding is supported through the external library LAME
|
||
@item AC3 @tab IX @tab X
|
||
@tab liba52 is used internally for decoding
|
||
@item Vorbis @tab X @tab X
|
||
@tab supported through the external library libvorbis
|
||
@item WMA V1/V2 @tab @tab X
|
||
@item Microsoft ADPCM @tab X @tab X
|
||
@item IMA ADPCM @tab X @tab X
|
||
@item RA144 @tab @tab X
|
||
@tab Real 14400 bit/s codec
|
||
@item RA288 @tab @tab X
|
||
@tab Real 28800 bit/s codec
|
||
@item AMR-NB @tab X @tab X
|
||
@tab supported through an external library
|
||
@item DV audio @tab @tab X
|
||
@end multitable
|
||
|
||
@code{X} means that the encoding (resp. decoding) is supported.
|
||
|
||
@code{I} means that an integer only version is available too (ensures highest
|
||
performances on systems without hardware floating point support).
|
||
|
||
@chapter Platform Specific information
|
||
|
||
@section 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.
|
||
|
||
@section BSD
|
||
|
||
@section Windows
|
||
|
||
@subsection Native Windows compilation
|
||
|
||
@itemize
|
||
@item Install the current versions of MSYS and MinGW from
|
||
@url{http://www.mingw.org/}. You can find detailed installation
|
||
instructions in the download section and the FAQ.
|
||
|
||
@item Extract the current version of ffmpeg (the latest release version or the current CVS snapshot whichever is recommended).
|
||
|
||
@item Start the MSYS shell (file @file{msys.bat}).
|
||
|
||
@item Change to the FFMPEG directory and create the libavcodec.a and libavformat.a libraries, following the instructions of how to compile ffmpeg (file @file{INSTALL}).
|
||
|
||
@end itemize
|
||
|
||
@subsection How to create a single Windows DLL from libavcodec and libavformat
|
||
|
||
Read @url{http://ffmpeg.org/win32.html}.
|
||
|
||
@subsection Cross compilation for Windows with Linux
|
||
|
||
You must use the MinGW cross compilation tools available at
|
||
@url{http://www.mingw.org/}.
|
||
|
||
Then configure ffmpeg with the following options:
|
||
@example
|
||
./configure --enable-mingw32 --cross-prefix=i386-mingw32msvc-
|
||
@end example
|
||
(you can change the cross-prefix according to the prefix choosen for the
|
||
MinGW tools).
|
||
|
||
Then you can easily test ffmpeg with wine
|
||
(@url{http://www.winehq.com/}).
|
||
|
||
@section MacOS X
|
||
|
||
@section BeOS
|
||
|
||
The configure script should guess the configuration itself.
|
||
Networking support is currently not finished.
|
||
errno issues fixed by Andrew Bachmann.
|
||
|
||
Old stuff:
|
||
|
||
Fran<EFBFBD>ois 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)
|
||
|
||
@chapter Developers Guide
|
||
|
||
@section API
|
||
@itemize
|
||
@item libavcodec is the library containing the codecs (both encoding and
|
||
decoding). See @file{libavcodec/apiexample.c} to see how to use it.
|
||
|
||
@item libavformat is the library containing the file formats handling (mux and
|
||
demux code for several formats). See @file{ffplay.c} to use it in a
|
||
player. See @file{output_example.c} to use it to generate audio or video
|
||
streams.
|
||
|
||
@end itemize
|
||
|
||
@section 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
|
||
@emph{any patch you make must be published}. The best way to proceed is
|
||
to send your patches to the ffmpeg mailing list.
|
||
|
||
@section 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 @file{libav/utils.c}) so that a documentation
|
||
can be generated automatically.
|
||
|
||
@section 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
|
||
@url{http://lists.sourceforge.net/lists/listinfo/ffmpeg-devel}
|
||
|
||
@section 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].
|
||
|
||
@bye
|