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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 parameters, when
possible. You have usually to give only the target bitrate you want.
FFmpeg can also convert from any sample rate to any other, and
resize video on the fly with a high quality polyphase filter.
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:
* You can input from YUV files:
ffmpeg -i /tmp/test%d.Y /tmp/out.mpg
It will use the files:
/tmp/test0.Y, /tmp/test0.U, /tmp/test0.V,
/tmp/test1.Y, /tmp/test1.U, /tmp/test1.V, etc...
The Y files use twice the resolution of the U and V files. They are
raw files, without header. They can be generated by all decent video
decoders. You must specify the size of the image with the '-s' option
if ffmpeg cannot guess it.
* You can input from a RAW YUV420P file:
ffmpeg -i /tmp/test.yuv /tmp/out.avi
The RAW YUV420P is a file containing RAW YUV planar, for each frame first
come the Y plane followed by U and V planes, which are half vertical and
horizontal resolution.
* You can output to a RAW YUV420P file:
ffmpeg -i mydivx.avi -o hugefile.yuv
* You can set several input files and output files:
ffmpeg -i /tmp/a.wav -s 640x480 -i /tmp/a.yuv /tmp/a.mpg
Convert the audio file a.wav and the raw yuv video file a.yuv
to mpeg file a.mpg
* You can also do audio and video 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.
* You can transcode decrypted VOBs
ffmpeg -i snatch_1.vob -f avi -vcodec mpeg4 -b 800 -g 300 -bf 2 -acodec
mp3 -ab 128 snatch.avi
This is a typicall DVD ripper example, input from a VOB file, output to
an AVI file with MPEG-4 video and MP3 audio, note that in this command we
use B frames so the MPEG-4 stream is DivX5 compatible, GOP size is 300
that means an INTRA frame every 10 seconds for 29.97 fps input video.
Also the audio stream is MP3 encoded so you need LAME support which is
enabled using '--enable-mp3lame' when configuring.
The mapping is particullary usefull for DVD transcoding to get the desired
audio language.
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:
-L show license
-h show help
-formats show available formats, codecs, protocols, ...
-f fmt force format
-i filename input file name
-y overwrite output files
-t duration set the recording time
-title string set the title
-author string set the author
-copyright string set the copyright
-comment string set the comment
-b bitrate set video bitrate (in kbit/s)
* 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]
-bt tolerance set video bitrate tolerance (in kbit/s)
-sameq use same video quality as source (implies VBR)
-ab bitrate set audio bitrate (in kbit/s)
* 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
-intra use only intra frames
-qscale q use fixed video quantiser scale (VBR)
-qmin q min video quantiser scale (VBR)
-qmax q max video quantiser scale (VBR)
-qdiff q max difference between the quantiser scale (VBR)
-qblur blur video quantiser scale blur (VBR)
-qcomp compression video quantiser scale compression (VBR)
-vd device set video device
-vcodec codec force video codec
-me method set motion estimation method
-bf frames use 'frames' B frames (only MPEG-4)
-hq activate high quality settings
-4mv use four motion vector by macroblock (only MPEG-4)
-ad device set audio device
-acodec codec force audio codec
-deinterlace deinterlace pictures
-benchmark add timings for benchmarking
-hex dump each input packet
-psnr calculate PSNR of compressed frames
-vstats dump video coding statistics to file
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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