path: root/doc/ffmpeg-doc.texi

\input texinfo @c -*- texinfo -*-

@settitle FFmpeg Documentation
@titlepage
@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 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.

@chapter Quick Start

@section Video and Audio grabbing

 FFmpeg can use a video4linux compatible video source and any Open Sound
  System audio source:
@example
  ffmpeg /tmp/out.mpg 
@end example

  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.

@section Video and Audio file format conversion

* ffmpeg can use any supported file format and protocol as input: 

Examples:

* You can input from YUV files:

@example
  ffmpeg -i /tmp/test%d.Y /tmp/out.mpg 
@end example

  It will use the files: 
@example
       /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 '-s' option
  if ffmpeg cannot guess it.

* You can input from a RAW YUV420P file:

@example
  ffmpeg -i /tmp/test.yuv /tmp/out.avi
@end example

  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:

@example
  ffmpeg -i mydivx.avi -o hugefile.yuv
@end example

* You can set several input files and output files:

@example
  ffmpeg -i /tmp/a.wav -s 640x480 -i /tmp/a.yuv /tmp/a.mpg
@end example

  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 conversions at the same time:

@example
  ffmpeg -i /tmp/a.wav -ar 22050 /tmp/a.mp2
@end example

  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:

@example
  ffmpeg -i /tmp/a.wav -ab 64 /tmp/a.mp2 -ab 128 /tmp/b.mp2 -map 0:0 -map 0:0
@end example

  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

@example
  ffmpeg -i snatch_1.vob -f avi -vcodec mpeg4 -b 800 -g 300 -bf 2 -acodec mp3 -ab 128 snatch.avi
@end example

  This is a typical 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 @code{--enable-mp3lame} when
  configuring.  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}.

@chapter Invocation

@section Syntax

 The generic syntax is:

@example 
  ffmpeg [[options][-i input_file]]... {[options] output_file}...
@end example
  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 for the outputs as the one
  specified for the inputs.

@section Main options

@table @samp
@item -L
show license
@item -h
                  show help
@item -formats            
show available formats, codecs, protocols, ...
@item -f fmt              
force format
@item -i filename         
input file name

@item -y                  
overwrite output files

@item -t duration         
set the recording time in seconds. @code{hh:mm:ss[.xxx]} syntax is also
supported.

@item -title string       
set the title

@item -author string      
set the author

@item -copyright string   
set the copyright

@item -comment string     
set the comment

@item -b bitrate          
set video bitrate (in kbit/s)
@end table

@section Video Options

@table @samp
@item -s size             
set frame size                       [160x128]
@item -r fps           
set frame rate                       [25]
@item -b bitrate
set the video bitrate in kbit/s      [200]
@item -vn
disable video recording              [no]
@item -bt tolerance       
set video bitrate tolerance (in kbit/s)
@item -sameq              
use same video quality as source (implies VBR)

@item -pass n  
select the pass number (1 or 2). It is useful to do two pass encoding. The statistics of the video are recorded in the first pass and the video at the exact requested bit rate is generated in the second pass.

@item -passlogfile file   
select two pass log file name

@end table

@section Audio Options

@table @samp
@item -ab bitrate         
set audio bitrate (in kbit/s)
@item -ar freq    
 set the audio sampling freq          [44100]
@item -ab bitrate 
 set the audio bitrate in kbit/s      [64]
@item -ac channels
 set the number of audio channels     [1]
@item -an     
 disable audio recording              [no]
@end table

@section Advanced options

@table @samp
@item -map file:stream    
set input stream mapping
@item -g gop_size         
set the group of picture size
@item -intra              
use only intra frames
@item -qscale q           
use fixed video quantiser scale (VBR)
@item -qmin q             
min video quantiser scale (VBR)
@item -qmax q             
max video quantiser scale (VBR)
@item -qdiff q            
max difference between the quantiser scale (VBR)
@item -qblur blur         
video quantiser scale blur (VBR)
@item -qcomp compression  
video quantiser scale compression (VBR)
@item -vd device          
set video device
@item -vcodec codec       
force video codec
@item -me method          
set motion estimation method
@item -bf frames          
use 'frames' B frames (only MPEG-4)
@item -hq                 
activate high quality settings
@item -4mv                
use four motion vector by macroblock (only MPEG-4)
@item -ad device          
set audio device
@item -acodec codec       
force audio codec
@item -deinterlace        
deinterlace pictures
@item -benchmark          
add timings for benchmarking
@item -hex                
dump each input packet
@item -psnr               
calculate PSNR of compressed frames
@item -vstats             
dump video coding statistics to file
@end table

@section Protocols

The filename can be @file{-} to read from the standard input or to write
to the standard output.

ffmpeg handles also many protocols specified with the URL syntax.

 Use 'ffmpeg -formats' to have 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

@itemize
@item 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:

@example
  ffmpeg -g 3 -r 3 -t 10 -b 50 -s qcif -f rv10 /tmp/b.rm
@end example

@item  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.

@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
  completely motion estimation (you have only I frames, which means it
  is about as good as JPEG compression).

@item To have very low bitrates in audio, reduce the sampling frequency
  (down to 22050 kHz for mpeg audio, 22050 or 11025 for ac3).

@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
  quality).

@item 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.

@end itemize

@chapter Supported File Formats and Codecs

You can use the @code{-formats} option to have an exhaustive list.

@section File Formats

FFmpeg supports the following file formats through the @code{libavformat}
library:

@multitable @columnfractions .4 .1 .1
@item Supported File Format @tab Encoding @tab Decoding @tab Comments
@item MPEG audio @tab X @tab X
@item MPEG1 systems @tab X  @tab  X 
@tab muxed audio and video
@item MPEG2 PS @tab X  @tab  X 
@tab also known as @code{VOB} file
@item MPEG2 TS @tab    @tab  X 
@tab also known as DVB Transport Stream
@item ASF@tab X @tab X 
@item AVI@tab X @tab X 
@item WAV@tab X @tab X 
@item Macromedia Flash@tab X @tab X
@tab Only embedded audio is decoded
@item Real Audio and Video @tab X @tab X 
@item Raw AC3 @tab X  @tab  X 
@item Raw MJPEG @tab X  @tab  X 
@item Raw MPEG video @tab X  @tab  X 
@item Raw PCM8/16 bits, mulaw/Alaw@tab X  @tab  X 
@item SUN AU format @tab X  @tab  X 
@item Quicktime        @tab   @tab  X 
@item MPEG4            @tab  @tab  X 
@tab MPEG4 is a variant of Quicktime
@item Raw MPEG4 video  @tab  X @tab  X 
@item DV               @tab  @tab X
@tab Only the video track is decoded.
@end multitable

@code{X} means that the encoding (resp. decoding) is supported.

@section Image Formats

FFmpeg can read and write images for each frame of a video sequence. The
following image formats are supported:

@multitable @columnfractions .4 .1 .1
@item Supported Image Format @tab Encoding @tab Decoding @tab Comments
@item PGM, PPM     @tab X @tab X 
@item PGMYUV       @tab X @tab X @tab PGM with U and V components in 420
@item JPEG         @tab X @tab X @tab Progressive JPEG is not supported
@item .Y.U.V       @tab X @tab X @tab One raw file per component
@item Animated GIF @tab X @tab   @tab Only uncompressed GIFs are generated
@end multitable

@code{X} means that the encoding (resp. decoding) is supported.

@section Video Codecs

@multitable @columnfractions .4 .1 .1 .7
@item Supported Codec @tab Encoding @tab Decoding @tab Comments
@item MPEG1 video            @tab  X  @tab  X
@item MPEG2 video            @tab     @tab  X 
@item MPEG4                  @tab  X  @tab  X @tab Also known as DIVX4/5
@item MSMPEG4 V1             @tab  X  @tab  X
@item MSMPEG4 V2             @tab  X  @tab  X
@item MSMPEG4 V3             @tab  X  @tab  X @tab Also known as DIVX3
@item WMV7                   @tab  X  @tab  X
@item H263(+)                @tab  X  @tab  X @tab Also known as Real Video 1.0
@item MJPEG                  @tab  X  @tab  X 
@item DV                     @tab     @tab  X 
@item Huff YUV               @tab  X  @tab  X
@end multitable

@code{X} means that the encoding (resp. decoding) is supported.

Check at @url{http://www.mplayerhq.hu/~michael/codec-features.html} to
get a precise comparison of FFmpeg MPEG4 codec compared to the other
solutions.

@section Audio Codecs

@multitable @columnfractions .4 .1 .1 .1 .7
@item Supported Codec @tab Encoding @tab Decoding @tab Comments
@item MPEG audio layer 2     @tab  IX  @tab  IX 
@item MPEG audio layer 1/3   @tab IX   @tab  IX
@tab MP3 encoding is supported through the external library LAME
@item AC3                    @tab  IX  @tab  X
@tab liba52 is used internally for decoding.
@item Vorbis                 @tab  X   @tab  X
@tab supported through the external library libvorbis.
@item WMA V1/V2              @tab      @tab X

@end multitable

@code{X} means that the encoding (resp. decoding) is supported.

@code{I} means that an integer only version is available too (ensures highest
performances on systems without hardware floating point support).

@chapter Platform Specific information

@section Linux

ffmpeg should be compiled with at least GCC 2.95.3. GCC 3.2 is the
preferred compiler now for ffmpeg. All future optimizations will depend on
features only found in GCC 3.2.

@section BSD

@section Windows

@section MacOS X

@section BeOS

The configure script should guess the configuration itself.
Networking support is currently not finished.
errno issues fixed by Andrew Bachmann.

Old stuff:

Fran�ois Revol - revol at free dot fr - April 2002

The configure script should guess the configuration itself, 
however I still didn't tested building on net_server version of BeOS.

ffserver is broken (needs poll() implementation).

There is still issues with errno codes, which are negative in BeOs, and
that ffmpeg negates when returning. This ends up turning errors into 
valid results, then crashes.
(To be fixed)

@chapter Developers Guide

@section API
@itemize
@item libavcodec is the library containing the codecs (both encoding and
  decoding). See @file{libavcodec/apiexample.c} to see how to use it.

@item libavformat is the library containing the file formats handling (mux and
  demux code for several formats). (no example yet, the API is likely to
  evolve).
@end itemize

@section Integrating libavcodec or libavformat in your program

You can integrate all the source code of the libraries to link them
statically to avoid any version problem. All you need is to provide a
'config.mak' and a 'config.h' in the parent directory. See the defines
generated by ./configure to understand what is needed.

You can use libavcodec or libavformat in your commercial program, but
@emph{any patch you make must be published}. The best way to proceed is
to send your patches to the ffmpeg mailing list.

@section Coding Rules

ffmpeg is programmed in ANSI C language. GCC extensions are
tolerated. Indent size is 4. The TAB character should not be used.

The presentation is the one specified by 'indent -i4 -kr'.

Main priority in ffmpeg is simplicity and small code size (=less
bugs).

Comments: for functions visible from other modules, use the JavaDoc
format (see examples in @file{libav/utils.c}) so that a documentation
can be generated automatically.

@section Submitting patches

When you submit your patch, try to send a unified diff (diff '-u'
option). I cannot read other diffs :-)

Run the regression tests before submitting a patch so that you can
verify that there are no big problems.

Except if your patch is really big and adds an important feature, by
submitting it to me, you accept implicitly to put it under my
copyright. I prefer to do this to avoid potential problems if
licensing of ffmpeg changes.

Patches should be posted as base64 encoded attachments (or any other
encoding which ensures that the patch wont be trashed during 
transmission) to the ffmpeg-devel mailinglist, see 
@url{http://lists.sourceforge.net/lists/listinfo/ffmpeg-devel}

@section Regression tests

Before submitting a patch (or committing with CVS), you should at least
test that you did not break anything.

The regression test build a synthetic video stream and a synthetic
audio stream. Then these are encoded then decoded with all codecs or
formats. The CRC (or MD5) of each generated file is recorded in a
result file. Then a 'diff' is launched with the reference results and
the result file.

The regression test then goes on to test the ffserver code with a 
limited set of streams. It is important that this step runs correctly
as well.

Run 'make test' to test all the codecs.

Run 'make libavtest' to test all the codecs.

[Of course, some patches may change the regression tests results. In
this case, the regression tests reference results shall be modified
accordingly].

@bye