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\input texinfo @c -*- texinfo -*-
@settitle FFmpeg Documentation
@sp 7
@center @titlefont{FFmpeg Documentation}
@sp 3
@end titlepage
@chapter 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 parameters that can possibly be
derived automatically. You usually only have to specify 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.
@chapter Quick Start
@c man begin EXAMPLES
@section Video and Audio grabbing
FFmpeg can grab video and audio from devices given that you specify the input
format and device.
ffmpeg -f oss -i /dev/dsp -f video4linux2 -i /dev/video0 /tmp/out.mpg
@end example
Note that you must activate the right video source and channel before
launching FFmpeg with any TV viewer such as xawtv
(@url{}) by Gerd Knorr. You also
have to set the audio recording levels correctly with a
standard mixer.
@section X11 grabbing
FFmpeg can grab the X11 display.
ffmpeg -f x11grab -s cif -i :0.0 /tmp/out.mpg
@end example
0.0 is display.screen number of your X11 server, same as
the DISPLAY environment variable.
ffmpeg -f x11grab -s cif -i :0.0+10,20 /tmp/out.mpg
@end example
0.0 is display.screen number of your X11 server, same as the DISPLAY environment
variable. 10 is the x-offset and 20 the y-offset for the grabbing.
@section Video and Audio file format conversion
* FFmpeg can use any supported file format and protocol as input:
* You can use YUV files as input:
ffmpeg -i /tmp/test%d.Y /tmp/out.mpg
@end example
It will use the files:
/tmp/test0.Y, /tmp/test0.U, /tmp/test0.V,
/tmp/test1.Y, /tmp/test1.U, /tmp/test1.V, etc...
@end example
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 @option{-s} option
if FFmpeg cannot guess it.
* You can input from a raw YUV420P file:
ffmpeg -i /tmp/test.yuv /tmp/out.avi
@end example
test.yuv is a file containing raw YUV planar data. Each frame is composed
of the Y plane followed by the U and V planes at half vertical and
horizontal resolution.
* You can output to a raw YUV420P file:
ffmpeg -i mydivx.avi hugefile.yuv
@end example
* You can set several input files and output files:
ffmpeg -i /tmp/a.wav -s 640x480 -i /tmp/a.yuv /tmp/a.mpg
@end example
Converts 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
@end example
Converts a.wav to MPEG audio at 22050 Hz sample rate.
* 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 64k /tmp/a.mp2 -ab 128k /tmp/b.mp2 -map 0:0 -map 0:0
@end example
Converts a.wav to a.mp2 at 64 kbits and to b.mp2 at 128 kbits. '-map
file:index' specifies 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 800k -g 300 -bf 2 -acodec libmp3lame -ab 128k snatch.avi
@end example
This is a typical DVD ripping example; the input is a VOB file, the
output 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, and
GOP size is 300 which means one intra frame every 10 seconds for 29.97fps
input video. Furthermore, the audio stream is MP3-encoded so you need
to enable LAME support by passing @code{--enable-libmp3lame} to configure.
The mapping is particularly useful for DVD transcoding
to get the desired audio language.
NOTE: To see the supported input formats, use @code{ffmpeg -formats}.
* You can extract images from a video, or create a video from many images:
For extracting images from a video:
ffmpeg -i foo.avi -r 1 -s WxH -f image2 foo-%03d.jpeg
@end example
This will extract one video frame per second from the video and will
output them in files named @file{foo-001.jpeg}, @file{foo-002.jpeg},
etc. Images will be rescaled to fit the new WxH values.
If you want to extract just a limited number of frames, you can use the
above command in combination with the -vframes or -t option, or in
combination with -ss to start extracting from a certain point in time.
For creating a video from many images:
ffmpeg -f image2 -i foo-%03d.jpeg -r 12 -s WxH foo.avi
@end example
The syntax @code{foo-%03d.jpeg} specifies to use a decimal number
composed of three digits padded with zeroes to express the sequence
number. It is the same syntax supported by the C printf function, but
only formats accepting a normal integer are suitable.
* You can put many streams of the same type in the output:
ffmpeg -i test1.avi -i test2.avi -vcodec copy -acodec copy -vcodec copy -acodec copy test12.avi -newvideo -newaudio
@end example
In addition to the first video and audio streams, the resulting
output file @file{test12.avi} will contain the second video
and the second audio stream found in the input streams list.
The @code{-newvideo}, @code{-newaudio} and @code{-newsubtitle}
options have to be specified immediately after the name of the output
file to which you want to add them.
@c man end
@chapter Invocation
@section Syntax
The generic syntax is:
@c man begin SYNOPSIS
ffmpeg [[infile options][@option{-i} @var{infile}]]... @{[outfile options] @var{outfile}@}...
@c man end
@end example
@c man begin DESCRIPTION
As a general rule, options are applied to the next specified
file. Therefore, order is important, and you can have the same
option on the command line multiple times. Each occurrence is
then applied to the next input or output file.
* To set the video bitrate of the output file to 64kbit/s:
ffmpeg -i input.avi -b 64k output.avi
@end example
* To force the frame rate of the output file to 24 fps:
ffmpeg -i input.avi -r 24 output.avi
@end example
* To force the frame rate of the input file (valid for raw formats only)
to 1 fps and the frame rate of the output file to 24 fps:
ffmpeg -r 1 -i input.m2v -r 24 output.avi
@end example
The 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 parameters for the outputs as the one
specified for the inputs.
@c man end
@c man begin OPTIONS
@section Main options
@table @option
@item -L
Show license.
@item -h
Show help.
@item -version
Show version.
@item -formats
Show available formats, codecs, bitstream filters, protocols, and frame size and frame rate abbreviations.
The fields preceding the format and codec names have the following meanings:
@table @samp
@item D
Decoding available
@item E
Encoding available
@item V/A/S
Video/audio/subtitle codec
@item S
Codec supports slices
@item D
Codec supports direct rendering
@item T
Codec can handle input truncated at random locations instead of only at frame boundaries
@end table
@item -f @var{fmt}
Force format.
@item -i @var{filename}
input file name
@item -y
Overwrite output files.
@item -t @var{duration}
Restrict the transcoded/captured video sequence
to the duration specified in seconds.
@code{hh:mm:ss[.xxx]} syntax is also supported.
@item -fs @var{limit_size}
Set the file size limit.
@item -ss @var{position}
Seek to given time position in seconds.
@code{hh:mm:ss[.xxx]} syntax is also supported.
@item -itsoffset @var{offset}
Set the input time offset in seconds.
@code{[-]hh:mm:ss[.xxx]} syntax is also supported.
This option affects all the input files that follow it.
The offset is added to the timestamps of the input files.
Specifying a positive offset means that the corresponding
streams are delayed by 'offset' seconds.
@item -title @var{string}
Set the title.
@item -timestamp @var{time}
Set the timestamp.
@item -author @var{string}
Set the author.
@item -copyright @var{string}
Set the copyright.
@item -comment @var{string}
Set the comment.
@item -album @var{string}
Set the album.
@item -track @var{number}
Set the track.
@item -year @var{number}
Set the year.
@item -v @var{number}
Set the logging verbosity level.
@item -target @var{type}
Specify target file type ("vcd", "svcd", "dvd", "dv", "dv50", "pal-vcd",
"ntsc-svcd", ... ). All the format options (bitrate, codecs,
buffer sizes) are then set automatically. You can just type:
ffmpeg -i myfile.avi -target vcd /tmp/vcd.mpg
@end example
Nevertheless you can specify additional options as long as you know
they do not conflict with the standard, as in:
ffmpeg -i myfile.avi -target vcd -bf 2 /tmp/vcd.mpg
@end example
@item -dframes @var{number}
Set the number of data frames to record.
@item -scodec @var{codec}
Force subtitle codec ('copy' to copy stream).
@item -newsubtitle
Add a new subtitle stream to the current output stream.
@item -slang @var{code}
Set the ISO 639 language code (3 letters) of the current subtitle stream.
@end table
@section Video Options
@table @option
@item -b @var{bitrate}
Set the video bitrate in bit/s (default = 200 kb/s).
@item -vframes @var{number}
Set the number of video frames to record.
@item -r @var{fps}
Set frame rate (Hz value, fraction or abbreviation), (default = 25).
@item -s @var{size}
Set frame size. The format is @samp{wxh} (ffserver default = 160x128, ffmpeg default = same as source).
The following abbreviations are recognized:
@table @samp
@item sqcif
@item qcif
@item cif
@item 4cif
@item qqvga
@item qvga
@item vga
@item svga
@item xga
@item uxga
@item qxga
@item sxga
@item qsxga
@item hsxga
@item wvga
@item wxga
@item wsxga
@item wuxga
@item woxga
@item wqsxga
@item wquxga
@item whsxga
@item whuxga
@item cga
@item ega
@item hd480
@item hd720
@item hd1080
@end table
@item -aspect @var{aspect}
Set aspect ratio (4:3, 16:9 or 1.3333, 1.7777).
@item -croptop @var{size}
Set top crop band size (in pixels).
@item -cropbottom @var{size}
Set bottom crop band size (in pixels).
@item -cropleft @var{size}
Set left crop band size (in pixels).
@item -cropright @var{size}
Set right crop band size (in pixels).
@item -padtop @var{size}
Set top pad band size (in pixels).
@item -padbottom @var{size}
Set bottom pad band size (in pixels).
@item -padleft @var{size}
Set left pad band size (in pixels).
@item -padright @var{size}
Set right pad band size (in pixels).
@item -padcolor @var{hex_color}
Set color of padded bands. The value for padcolor is expressed
as a six digit hexadecimal number where the first two digits
represent red, the middle two digits green and last two digits
blue (default = 000000 (black)).
@item -vn
Disable video recording.
@item -bt @var{tolerance}
Set video bitrate tolerance (in bits, default 4000k).
Has a minimum value of: (target_bitrate/target_framerate).
In 1-pass mode, bitrate tolerance specifies how far ratecontrol is
willing to deviate from the target average bitrate value. This is
not related to min/max bitrate. Lowering tolerance too much has
an adverse effect on quality.
@item -maxrate @var{bitrate}
Set max video bitrate (in bit/s).
Requires -bufsize to be set.
@item -minrate @var{bitrate}
Set min video bitrate (in bit/s).
Most useful in setting up a CBR encode:
ffmpeg -i myfile.avi -b 4000k -minrate 4000k -maxrate 4000k -bufsize 1835k out.m2v
@end example
It is of little use elsewise.
@item -bufsize @var{size}
Set video buffer verifier buffer size (in bits).
@item -vcodec @var{codec}
Force video codec to @var{codec}. Use the @code{copy} special value to
tell that the raw codec data must be copied as is.
@item -sameq
Use same video quality as source (implies VBR).
@item -pass @var{n}
Select the pass number (1 or 2). It is used to do two-pass
video encoding. The statistics of the video are recorded in the first
pass into a log file (see also the option -passlogfile),
and in the second pass that log file is used to generate the video
at the exact requested bitrate.
On pass 1, you may just deactivate audio and set output to null,
examples for Windows and Unix:
ffmpeg -i -vcodec libxvid -pass 1 -an -f rawvideo -y NUL
ffmpeg -i -vcodec libxvid -pass 1 -an -f rawvideo -y /dev/null
@end example
@item -passlogfile @var{prefix}
Set two-pass log file name prefix to @var{prefix}, the default file name
prefix is ``ffmpeg2pass''. The complete file name will be
@file{PREFIX-N.log}, where N is a number specific to the output
@item -newvideo
Add a new video stream to the current output stream.
@end table
@section Advanced Video Options
@table @option
@item -pix_fmt @var{format}
Set pixel format. Use 'list' as parameter to show all the supported
pixel formats.
@item -sws_flags @var{flags}
Set SwScaler flags (only available when compiled with swscale support).
@item -g @var{gop_size}
Set the group of pictures size.
@item -intra
Use only intra frames.
@item -vdt @var{n}
Discard threshold.
@item -qscale @var{q}
Use fixed video quantizer scale (VBR).
@item -qmin @var{q}
minimum video quantizer scale (VBR)
@item -qmax @var{q}
maximum video quantizer scale (VBR)
@item -qdiff @var{q}
maximum difference between the quantizer scales (VBR)
@item -qblur @var{blur}
video quantizer scale blur (VBR) (range 0.0 - 1.0)
@item -qcomp @var{compression}
video quantizer scale compression (VBR) (default 0.5).
Constant of ratecontrol equation. Recommended range for default rc_eq: 0.0-1.0
@item -lmin @var{lambda}
minimum video lagrange factor (VBR)
@item -lmax @var{lambda}
max video lagrange factor (VBR)
@item -mblmin @var{lambda}
minimum macroblock quantizer scale (VBR)
@item -mblmax @var{lambda}
maximum macroblock quantizer scale (VBR)
These four options (lmin, lmax, mblmin, mblmax) use 'lambda' units,
but you may use the QP2LAMBDA constant to easily convert from 'q' units:
ffmpeg -i src.ext -lmax 21*QP2LAMBDA dst.ext
@end example
@item -rc_init_cplx @var{complexity}
initial complexity for single pass encoding
@item -b_qfactor @var{factor}
qp factor between P- and B-frames
@item -i_qfactor @var{factor}
qp factor between P- and I-frames
@item -b_qoffset @var{offset}
qp offset between P- and B-frames
@item -i_qoffset @var{offset}
qp offset between P- and I-frames
@item -rc_eq @var{equation}
Set rate control equation (@pxref{FFmpeg formula
evaluator}) (default = @code{tex^qComp}).
@item -rc_override @var{override}
rate control override for specific intervals
@item -me_method @var{method}
Set motion estimation method to @var{method}.
Available methods are (from lowest to best quality):
@table @samp
@item zero
Try just the (0, 0) vector.
@item phods
@item log
@item x1
@item hex
@item umh
@item epzs
(default method)
@item full
exhaustive search (slow and marginally better than epzs)
@end table
@item -dct_algo @var{algo}
Set DCT algorithm to @var{algo}. Available values are:
@table @samp
@item 0
FF_DCT_AUTO (default)
@item 1
@item 2
@item 3
@item 4
@item 5
@end table
@item -idct_algo @var{algo}
Set IDCT algorithm to @var{algo}. Available values are:
@table @samp
@item 0
FF_IDCT_AUTO (default)
@item 1
@item 2
@item 3
@item 4
@item 5
@item 6
@item 7
@item 8
@item 9
@item 10
@end table
@item -er @var{n}
Set error resilience to @var{n}.
@table @samp
@item 1
FF_ER_CAREFUL (default)
@item 2
@item 3
@item 4
@end table
@item -ec @var{bit_mask}
Set error concealment to @var{bit_mask}. @var{bit_mask} is a bit mask of
the following values:
@table @samp
@item 1
FF_EC_GUESS_MVS (default = enabled)
@item 2
FF_EC_DEBLOCK (default = enabled)
@end table
@item -bf @var{frames}
Use 'frames' B-frames (supported for MPEG-1, MPEG-2 and MPEG-4).
@item -mbd @var{mode}
macroblock decision
@table @samp
@item 0
FF_MB_DECISION_SIMPLE: Use mb_cmp (cannot change it yet in FFmpeg).
@item 1
FF_MB_DECISION_BITS: Choose the one which needs the fewest bits.
@item 2
FF_MB_DECISION_RD: rate distortion
@end table
@item -4mv
Use four motion vector by macroblock (MPEG-4 only).
@item -part
Use data partitioning (MPEG-4 only).
@item -bug @var{param}
Work around encoder bugs that are not auto-detected.
@item -strict @var{strictness}
How strictly to follow the standards.
@item -aic
Enable Advanced intra coding (h263+).
@item -umv
Enable Unlimited Motion Vector (h263+)
@item -deinterlace
Deinterlace pictures.
@item -ilme
Force interlacing support in encoder (MPEG-2 and MPEG-4 only).
Use this option if your input file is interlaced and you want
to keep the interlaced format for minimum losses.
The alternative is to deinterlace the input stream with
@option{-deinterlace}, but deinterlacing introduces losses.
@item -psnr
Calculate PSNR of compressed frames.
@item -vstats
Dump video coding statistics to @file{vstats_HHMMSS.log}.
@item -vstats_file @var{file}
Dump video coding statistics to @var{file}.
@item -top @var{n}
top=1/bottom=0/auto=-1 field first
@item -dc @var{precision}
@item -vtag @var{fourcc/tag}
Force video tag/fourcc.
@item -qphist
Show QP histogram.
@item -vbsf @var{bitstream_filter}
Bitstream filters available are "dump_extra", "remove_extra", "noise", "h264_mp4toannexb", "imxdump", "mjpegadump".
ffmpeg -i h264.mp4 -vcodec copy -vbsf h264_mp4toannexb -an out.h264
@end example
@end table
@section Audio Options
@table @option
@item -aframes @var{number}
Set the number of audio frames to record.
@item -ar @var{freq}
Set the audio sampling frequency (default = 44100 Hz).
@item -ab @var{bitrate}
Set the audio bitrate in bit/s (default = 64k).
@item -ac @var{channels}
Set the number of audio channels (default = 1).
@item -an
Disable audio recording.
@item -acodec @var{codec}
Force audio codec to @var{codec}. Use the @code{copy} special value to
specify that the raw codec data must be copied as is.
@item -newaudio
Add a new audio track to the output file. If you want to specify parameters,
do so before @code{-newaudio} (@code{-acodec}, @code{-ab}, etc..).
Mapping will be done automatically, if the number of output streams is equal to
the number of input streams, else it will pick the first one that matches. You
can override the mapping using @code{-map} as usual.
ffmpeg -i file.mpg -vcodec copy -acodec ac3 -ab 384k test.mpg -acodec mp2 -ab 192k -newaudio
@end example
@item -alang @var{code}
Set the ISO 639 language code (3 letters) of the current audio stream.
@end table
@section Advanced Audio options:
@table @option
@item -atag @var{fourcc/tag}
Force audio tag/fourcc.
@item -absf @var{bitstream_filter}
Bitstream filters available are "dump_extra", "remove_extra", "noise", "mp3comp", "mp3decomp".
@end table
@section Subtitle options:
@table @option
@item -scodec @var{codec}
Force subtitle codec ('copy' to copy stream).
@item -newsubtitle
Add a new subtitle stream to the current output stream.
@item -slang @var{code}
Set the ISO 639 language code (3 letters) of the current subtitle stream.
@item -sbsf @var{bitstream_filter}
Bitstream filters available are "mov2textsub", "text2movsub".
ffmpeg -i -an -vn -sbsf mov2textsub -scodec copy -f rawvideo sub.txt
@end example
@end table
@section Audio/Video grab options
@table @option
@item -vc @var{channel}
Set video grab channel (DV1394 only).
@item -tvstd @var{standard}
Set television standard (NTSC, PAL (SECAM)).
@item -isync
Synchronize read on input.
@end table
@section Advanced options
@table @option
@item -map @var{input_stream_id}[:@var{sync_stream_id}]
Set stream mapping from input streams to output streams.
Just enumerate the input streams in the order you want them in the output.
@var{sync_stream_id} if specified sets the input stream to sync
@item -map_meta_data @var{outfile}:@var{infile}
Set meta data information of @var{outfile} from @var{infile}.
@item -debug
Print specific debug info.
@item -benchmark
Add timings for benchmarking.
@item -dump
Dump each input packet.
@item -hex
When dumping packets, also dump the payload.
@item -bitexact
Only use bit exact algorithms (for codec testing).
@item -ps @var{size}
Set packet size in bits.
@item -re
Read input at native frame rate. Mainly used to simulate a grab device.
@item -loop_input
Loop over the input stream. Currently it works only for image
streams. This option is used for automatic FFserver testing.
@item -loop_output @var{number_of_times}
Repeatedly loop output for formats that support looping such as animated GIF
(0 will loop the output infinitely).
@item -threads @var{count}
Thread count.
@item -vsync @var{parameter}
Video sync method. Video will be stretched/squeezed to match the timestamps,
it is done by duplicating and dropping frames. With -map you can select from
which stream the timestamps should be taken. You can leave either video or
audio unchanged and sync the remaining stream(s) to the unchanged one.
@item -async @var{samples_per_second}
Audio sync method. "Stretches/squeezes" the audio stream to match the timestamps,
the parameter is the maximum samples per second by which the audio is changed.
-async 1 is a special case where only the start of the audio stream is corrected
without any later correction.
@item -copyts
Copy timestamps from input to output.
@item -shortest
Finish encoding when the shortest input stream ends.
@item -dts_delta_threshold
Timestamp discontinuity delta threshold.
@item -muxdelay @var{seconds}
Set the maximum demux-decode delay.
@item -muxpreload @var{seconds}
Set the initial demux-decode delay.
@end table
@section Preset files
A preset file contains a sequence of @var{option}=@var{value} pairs,
one for each line, specifying a sequence of options which would be
awkward to specify on the command line. Lines starting with the hash
('#') character are ignored and are used to provide comments. Check
the @file{ffpresets} directory in the FFmpeg source tree for examples.
Preset files are specified with the @code{vpre}, @code{apre} and
@code{spre} options. The options specified in a preset file are
applied to the currently selected codec of the same type as the preset
The argument passed to the preset options identifies the preset file
to use according to the following rules.
First ffmpeg searches for a file named @var{arg}.ffpreset in the
directories @file{$HOME/.ffmpeg}, and in the datadir defined at
configuration time (usually @file{PREFIX/share/ffmpeg}) in that
order. For example, if the argument is @code{libx264-max}, it will
search for the file @file{libx264-max.ffpreset}.
If no such file is found, then ffmpeg will search for a file named
@var{codec_name}-@var{arg}.ffpreset in the above-mentioned
directories, where @var{codec_name} is the name of the codec to which
the preset file options will be applied. For example, if you select
the video codec with @code{-vcodec libx264} and use @code{-vpre max},
then it will search for the file @file{libx264-max.ffpreset}.
Finally, if the above rules failed and the argument specifies an
absolute pathname, ffmpeg will search for that filename. This way you
can specify the absolute and complete filename of the preset file, for
example @file{./ffpresets/libx264-max.ffpreset}.
@node FFmpeg formula evaluator
@section FFmpeg formula evaluator
When evaluating a rate control string, FFmpeg uses an internal formula
The following binary operators are available: @code{+}, @code{-},
@code{*}, @code{/}, @code{^}.
The following unary operators are available: @code{+}, @code{-},
The following statements are available: @code{ld}, @code{st},
The following functions are available:
@table @var
@item sinh(x)
@item cosh(x)
@item tanh(x)
@item sin(x)
@item cos(x)
@item tan(x)
@item atan(x)
@item asin(x)
@item acos(x)
@item exp(x)
@item log(x)
@item abs(x)
@item squish(x)
@item gauss(x)
@item mod(x, y)
@item max(x, y)
@item min(x, y)
@item eq(x, y)
@item gte(x, y)
@item gt(x, y)
@item lte(x, y)
@item lt(x, y)
@item bits2qp(bits)
@item qp2bits(qp)
@end table
The following constants are available:
@table @var
@item PI
@item E
@item iTex
@item pTex
@item tex
@item mv
@item fCode
@item iCount
@item mcVar
@item var
@item isI
@item isP
@item isB
@item avgQP
@item qComp
@item avgIITex
@item avgPITex
@item avgPPTex
@item avgBPTex
@item avgTex
@end table
@c man end
@setfilename ffmpeg
@settitle FFmpeg video converter
@c man begin SEEALSO
ffserver(1), ffplay(1) and the HTML documentation of @file{ffmpeg}.
@c man end
@c man begin AUTHOR
Fabrice Bellard
@c man end
@end ignore
@section Protocols
The file name can be @file{-} to read from standard input or to write
to standard output.
FFmpeg also handles many protocols specified with an URL syntax.
Use 'ffmpeg -formats' to see a list of the supported protocols.
The protocol @code{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 :-)
@chapter Tips
@item For streaming at very low bitrate application, use a low frame rate
and a small GOP size. This is especially true for RealVideo where
the Linux player does not seem to be very fast, so it can miss
frames. An example is:
ffmpeg -g 3 -r 3 -t 10 -b 50k -s qcif -f rv10 /tmp/b.rm
@end example
@item The parameter 'q' which is displayed while encoding is the current
quantizer. The value 1 indicates that a very good quality could
be achieved. The value 31 indicates the worst quality. If q=31 appears
too often, it means that the encoder cannot compress enough to meet
your bitrate. You must either increase the bitrate, decrease the
frame rate or decrease the frame size.
@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
motion estimation completely (you have only I-frames, which means it
is about as good as JPEG compression).
@item To have very low audio bitrates, reduce the sampling frequency
(down to 22050 Hz for MPEG audio, 22050 or 11025 for AC-3).
@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
@item When converting video files, you can use the '-sameq' option which
uses the same quality factor in the encoder as in the decoder.
It allows almost lossless encoding.
@end itemize