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*************** FFMPEG soft VCR documentation *****************
0) Introduction
---------------
FFmpeg is a very fast video and audio encoder. It can grab from
files or from a live audio/video source.
The command line interface is designed to be intuitive, in the sense
that ffmpeg tries to figure out all the paramters, when
possible. You have usually to give only the target bitrate you want.
FFmpeg can also convert from any sample rate to any other, and
resize video on the fly with a high quality polyphase filter.
1) Video and Audio grabbing
---------------------------
* ffmpeg can use a video4linux compatible video source and any Open
Sound System audio source:
ffmpeg /tmp/out.mpg
Note that you must activate the right video source and channel
before launching ffmpeg. You can use any TV viewer such as xawtv by
Gerd Knorr which I find very good. You must also set correctly the
audio recording levels with a standard mixer.
2) Video and Audio file format convertion
-----------------------------------------
* ffmpeg can use any supported file format and protocol as input :
examples:
ffmpeg -i /tmp/test%d.Y /tmp/out.mpg
If will use the files:
/tmp/test0.Y, /tmp/test0.U, /tmp/test0.V,
/tmp/test1.Y, /tmp/test1.U, /tmp/test1.V, etc...
The Y files use twice the resolution of the U and V files. They are
raw files, without header. They can be generated by all decent video
decoders. You must specify the size of the image with the '-s' option
if ffmpeg cannot guess it.
* You can set several input files and output files:
ffmpeg -i /tmp/a.wav -s 640x480 -i /tmp/a.yuv /tmp/a.mpg
Convert the audio file a.wav and the raw yuv video file a.yuv to mpeg file a.mpg
* you can also do audio and video convertions at the same time:
ffmpeg -i /tmp/a.wav -ar 22050 /tmp/a.mp2
Convert the sample rate of a.wav to 22050 Hz and encode it to MPEG audio.
* you can encode to several formats at the same time and define a
mapping from input stream to output streams:
ffmpeg -i /tmp/a.wav -ab 64 /tmp/a.mp2 -ab 128 /tmp/b.mp2 -map 0:0 -map 0:0
convert a.wav to a.mp2 at 64 kbits and b.mp2 at 128 kbits. '-map
file:index' specify which input stream is used for each output
stream, in the order of the definition of output streams.
NOTE: to see the supported input formats, use 'ffmpeg -formats'.
2) Invocation
-------------
* The generic syntax is :
ffmpeg [[options][-i input_file]]... {[options] output_file}...
If no input file is given, audio/video grabbing is done.
As a general rule, options are applied to the next specified
file. For example, if you give the '-b 64' option, it sets the video
bitrate of the next file. Format option may be needed for raw input
files.
By default, ffmpeg tries to convert as losslessly as possible: it
uses the same audio and video parameter fors the outputs as the one
specified for the inputs.
* Main options are:
-h show help
-formats show available formats, codecs and protocols
-L print the LICENSE
-i filename input file name
-y overwrite output files
-t duration set recording time in seconds
-f format set encoding format [guessed]
-title string set the title
-author string set the author
-copyright string set the copyright
-comment string set the comment
* Video Options are:
-s size set frame size [160x128]
-r fps set frame rate [25]
-b bitrate set the video bitrate in kbit/s [200]
-vn disable video recording [no]
* Audio Options are:
-ar freq set the audio sampling freq [44100]
-ab bitrate set the audio bitrate in kbit/s [64]
-ac channels set the number of audio channels [1]
-an disable audio recording [no]
Advanced options are:
-map file:stream set input stream mapping
-g gop_size set the group of picture size [12]
-intra use only intra frames [no]
-qscale q use fixed video quantiser scale (VBR)
-c comment set the comment string
-vd device set video4linux device name [/dev/video]
-vcodec codec force audio codec
-me method set motion estimation method
-ad device set audio device name [/dev/dsp]
-acodec codec force audio codec
The output file can be "-" to output to a pipe. This is only possible
with mpeg1 and h263 formats.
3) Protocols
ffmpeg handles also many protocols specified with the URL syntax.
Use 'ffmpeg -formats' to have a list of the supported protocols.
The protocol 'http:' is currently used only to communicate with
ffserver (see the ffserver documentation). When ffmpeg will be a
video player it will also be used for streaming :-)
4) File formats and codecs
--------------------------
Use 'ffmpeg -formats' to have a list of the supported output
formats. Only some formats are handled as input, but it will improve
in the next versions.
5) Tips
-------
- For streaming at very low bit rate application, use a low frame rate
and a small gop size. This is especially true for real video 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 50 -s qcif -f rv10 /tmp/b.rm
- The parameter 'q' which is displayed while encoding is the current
quantizer. The value of 1 indicates that a very good quality could
be achieved. The value of 31 indicates the worst quality. If q=31
too often, it means that the encoder cannot compress enough to meet
your bit rate. You must either increase the bit rate, decrease the
frame rate or decrease the frame size.
- 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
completly motion estimation (you have only I frames, which means it
is about as good as JPEG compression).
- To have very low bitrates in audio, reduce the sampling frequency
(down to 22050 kHz for mpeg audio, 22050 or 11025 for ac3).
- 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).
- 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.
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