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added ffplay documentation - added automatic man page generation for ffmpeg, ffserver and ffplay
Originally committed as revision 2136 to svn://svn.ffmpeg.org/ffmpeg/trunk
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@@ -22,19 +22,21 @@ 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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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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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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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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@@ -45,109 +47,115 @@ 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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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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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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/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 '-s' option
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if ffmpeg cannot guess it.
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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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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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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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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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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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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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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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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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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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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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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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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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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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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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The generic syntax is:
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@example
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ffmpeg [[options][-i input_file]]... {[options] output_file}...
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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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If no input file is given, audio/video grabbing is done.
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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 '-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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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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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 @samp
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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 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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@@ -173,13 +181,11 @@ set the copyright
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@item -comment string
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set the comment
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@item -b bitrate
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set video bitrate (in kbit/s)
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@end table
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@section Video Options
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@table @samp
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@table @option
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@item -s size
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set frame size [160x128]
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@item -r fps
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@@ -203,7 +209,7 @@ select two pass log file name
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@section Audio Options
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@table @samp
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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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@@ -218,7 +224,7 @@ set audio bitrate (in kbit/s)
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@section Advanced options
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@table @samp
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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 -g gop_size
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@@ -264,6 +270,22 @@ calculate PSNR of compressed frames
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@item -vstats
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dump video coding statistics to file
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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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@@ -272,47 +294,47 @@ 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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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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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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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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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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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
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compression at the expense of the compression ratio. You can use
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'-me zero' to speed up motion estimation, and '-intra' to disable
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completely motion estimation (you have only I frames, which means it
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is about as good as JPEG compression).
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compression at the expense of the compression ratio. You can use
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'-me zero' to speed up motion estimation, and '-intra' to disable
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completely motion estimation (you have only I frames, which means it
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is about as good as JPEG compression).
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@item To have very low bitrates in audio, reduce the sampling frequency
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(down to 22050 kHz for mpeg audio, 22050 or 11025 for ac3).
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(down to 22050 kHz for mpeg audio, 22050 or 11025 for ac3).
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@item To have a constant quality (but a variable bitrate), use the option
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'-qscale n' when 'n' is between 1 (excellent quality) and 31 (worst
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quality).
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'-qscale n' when 'n' is between 1 (excellent quality) and 31 (worst
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quality).
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@item When converting video files, you can use the '-sameq' option which
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uses in the encoder the same quality factor than in the decoder. It
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allows to be almost lossless in encoding.
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uses in the encoder the same quality factor than in the decoder. It
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allows to be almost lossless in encoding.
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@end itemize
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