FFmpeg/doc/ffmpeg.txt

*************** 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.