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.

Quick Start
***********

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 (http://bytesex.org/xawtv/) by Gerd Knorr which I find very good. You must also set correctly the audio recording levels with a standard mixer.

Video and Audio file format conversion
======================================

* 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 conversions 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 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 --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 ffmpeg -formats.

Invocation
**********

Syntax
======

The generic syntax is:

     ffmpeg [[infile options][-i INFILE]]... {[outfile options] OUTFILE}...

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.

Main options
============

-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 in seconds. hh:mm:ss[.xxx] syntax is also supported.

-ss position
     seek to given time position. hh:mm:ss[.xxx] syntax is also supported.

-title string
     set the title

-author string
     set the author

-copyright string
     set the copyright

-comment string
     set the comment

-target type
     specify target file type ("vcd", "svcd", "dvd", "pal-vcd", "ntsc-svcd", ... ). All the format options (bitrate, codecs, buffer sizes) are automatically set by this option. You can just type:

          ffmpeg -i myfile.avi -target vcd /tmp/vcd.mpg

     Nevertheless you can specify additional options as long as you know they do not compromise the standard, as in:

          ffmpeg -i myfile.avi -target vcd -bf 2 /tmp/vcd.mpg

-hq
     activate high quality settings

-itsoffset offset
     set the input time offset in seconds. [-]hh:mm:ss[.xxx] syntax is also supported.  This option affects all the input files that follow it.  The offset is added to the input files' timestamps; specifying a positive offset means that the corresponding streams are delayed by 'offset' seconds.

Video Options
=============

-b bitrate
     set the video bitrate in kbit/s (default = 200 kb/s)

-r fps
     set frame rate (default = 25)

-s size
     set frame size. The format is WxH (default 160x128).  The following abbreviations are recognized:

     sqcif
          128x96

     qcif
          176x144

     cif
          352x288

     4cif
          704x576

-aspect aspect
     set aspect ratio (4:3, 16:9 or 1.3333, 1.7777)

-croptop size
     set top crop band size (in pixels)

-cropbottom size
     set bottom crop band size (in pixels)

-cropleft size
     set left crop band size (in pixels)

-cropright size
     set right crop band size (in pixels)

-padtop size
     set top pad band size (in pixels)

-padbottom size
     set bottom pad band size (in pixels)

-padleft size
     set left pad band size (in pixels)

-padright size
     set right pad band size (in pixels)

-padcolor (hex color)
     set color of padded bands. The value for pad color is expressed as a six digit hexidecimal number where the first two digits represent red, middle two digits green and last two digits blue. Defaults to 000000 (black)

-vn
     disable video recording

-bt tolerance
     set video bitrate tolerance (in kbit/s)

-maxrate bitrate
     set max video bitrate tolerance (in kbit/s)

-minrate bitrate
     set min video bitrate tolerance (in kbit/s)

-bufsize size
     set ratecontrol buffere size (in kbit)

-vcodec codec
     force video codec to CODEC. Use the copy special value to tell that the raw codec data must be copied as is.

-sameq
     use same video quality as source (implies VBR)

-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.

-passlogfile file
     select two pass log file name to FILE.

Advanced Video Options
======================

-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)

-rc_init_cplx complexity
     initial complexity for 1-pass encoding

-b_qfactor factor
     qp factor between p and b frames

-i_qfactor factor
     qp factor between p and i frames

-b_qoffset offset
     qp offset between p and b frames

-i_qoffset offset
     qp offset between p and i frames

-rc_eq equation
     set rate control equation (see FFmpeg formula evaluator). Default is tex^qComp.

-rc_override override
     rate control override for specific intervals

-me method
     set motion estimation method to METHOD. Available methods are (from lower to best quality):

     zero
          Try just the (0, 0) vector.

     phods

     log

     x1

     epzs
          (default method)

     full
          exhaustive search (slow and marginally better than epzs)

-dct_algo algo
     set dct algorithm to ALGO. Available values are:

     0
          FF_DCT_AUTO (default)

     1
          FF_DCT_FASTINT

     2
          FF_DCT_INT

     3
          FF_DCT_MMX

     4
          FF_DCT_MLIB

     5
          FF_DCT_ALTIVEC

-idct_algo algo
     set idct algorithm to ALGO. Available values are:

     0
          FF_IDCT_AUTO (default)

     1
          FF_IDCT_INT

     2
          FF_IDCT_SIMPLE

     3
          FF_IDCT_SIMPLEMMX

     4
          FF_IDCT_LIBMPEG2MMX

     5
          FF_IDCT_PS2

     6
          FF_IDCT_MLIB

     7
          FF_IDCT_ARM

     8
          FF_IDCT_ALTIVEC

     9
          FF_IDCT_SH4

     10
          FF_IDCT_SIMPLEARM

-er n
     set error resilience to N.

     1
          FF_ER_CAREFULL (default)

     2
          FF_ER_COMPLIANT

     3
          FF_ER_AGGRESSIVE

     4
          FF_ER_VERY_AGGRESSIVE

-ec bit_mask
     set error concealment to BIT_MASK. BIT_MASK is a bit mask of the following values:

     1
          FF_EC_GUESS_MVS (default=enabled)

     2
          FF_EC_DEBLOCK (default=enabled)

-bf frames
     use 'frames' B frames (supported for MPEG-1, MPEG-2 and MPEG-4)

-mbd mode
     macroblock decision

     0
          FF_MB_DECISION_SIMPLE: use mb_cmp (cannot change it yet in ffmpeg)

     1
          FF_MB_DECISION_BITS: chooses the one which needs the fewest bits

     2
          FF_MB_DECISION_RD: rate distoration

-4mv
     use four motion vector by macroblock (only MPEG-4)

-part
     use data partitioning (only MPEG-4)

-bug param
     workaround not auto detected encoder bugs

-strict strictness
     how strictly to follow the standarts

-aic
     enable Advanced intra coding (h263+)

-umv
     enable Unlimited Motion Vector (h263+)

-deinterlace
     deinterlace pictures

-interlace
     force interlacing support in encoder (only MPEG-2 and MPEG-4). Use this option if your input file is interlaced and if you want to keep the interlaced format for minimum losses. The alternative is to deinterlace the input stream with -deinterlace, but deinterlacing introduces more losses.

-psnr
     calculate PSNR of compressed frames

-vstats
     dump video coding statistics to vstats_HHMMSS.log.

-vhook module
     insert video processing MODULE. MODULE contains the module name and its parameters separated by spaces.

Audio Options
=============

-ar freq
     set the audio sampling freq (default = 44100 Hz)

-ab bitrate
     set the audio bitrate in kbit/s (default = 64)

-ac channels
     set the number of audio channels (default = 1)

-an
     disable audio recording

-acodec codec
     force audio codec to CODEC. Use the copy special value to tell that the raw codec data must be copied as is.

Audio/Video grab options
========================

-vd device
     set video grab device (e.g. /dev/video0)

-vc channel
     set video grab channel (DV1394 only)

-tvstd standard
     set television standard (NTSC, PAL (SECAM))

-dv1394
     set DV1394 grab

-ad device
     set audio device (e.g. /dev/dsp)

Advanced options
================

-map file:stream
     set input stream mapping

-debug
     print specific debug info

-benchmark
     add timings for benchmarking

-hex
     dump each input packet

-bitexact
     only use bit exact algorithms (for codec testing)

-ps size
     set packet size in bits

-re
     read input at native frame rate. Mainly used to simulate a grab device.

-loop
     loop over the input stream. Currently it works only for image streams. This option is used for ffserver automatic testing.

FFmpeg formula evaluator
========================

When evaluating a rate control string, FFmpeg uses an internal formula evaluator.

The following binary operators are available: +, -, *, /, ^.

The following unary operators are available: +, -, (...).

The following functions are available:

sinh(x)

cosh(x)

tanh(x)

sin(x)

cos(x)

tan(x)

exp(x)

log(x)

squish(x)

gauss(x)

abs(x)

max(x, y)

min(x, y)

gt(x, y)

lt(x, y)

eq(x, y)

bits2qp(bits)

qp2bits(qp)

The following constants are available:

PI

E

iTex

pTex

tex

mv

fCode

iCount

mcVar

var

isI

isP

isB

avgQP

qComp

avgIITex

avgPITex

avgPPTex

avgBPTex

avgTex

Protocols
=========

The filename can be - 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 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 :-)

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 completely 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.

Supported File Formats and Codecs
*********************************

You can use the -formats option to have an exhaustive list.

File Formats
============

FFmpeg supports the following file formats through the libavformat library:

Supported File Format   Encoding   Decoding   Comments
MPEG audio   X   X
MPEG1 systems   X     X
muxed audio and video
MPEG2 PS   X     X
also known as VOB file
MPEG2 TS         X
also known as DVB Transport Stream
ASF  X   X
AVI  X   X
WAV  X   X
Macromedia Flash  X   X
Only embedded audio is decoded
FLV                 X   X
Macromedia Flash video files
Real Audio and Video   X   X
Raw AC3   X     X
Raw MJPEG   X     X
Raw MPEG video   X     X
Raw PCM8/16 bits, mulaw/Alaw  X     X
Raw CRI ADX audio   X     X
SUN AU format   X     X
NUT   X   X   NUT Open Container Format
Quicktime          X    X
MPEG4              X    X
MPEG4 is a variant of Quicktime
Raw MPEG4 video     X    X
DV                  X    X
4xm                     X
4X Technologies format, used in some games
Playstation STR         X
Id RoQ                  X
used in Quake III, Jedi Knight 2, other computer games
Interplay MVE           X
format used in various Interplay computer games
WC3 Movie               X
multimedia format used in Origin's Wing Commander III computer game
Sega FILM/CPK           X
used in many Sega Saturn console games
Westwood Studios VQA/AUD         X
Multimedia formats used in Westwood Studios games
Id Cinematic (.cin)        X
Used in Quake II
FLIC format             X
.fli/.flc files
Sierra VMD              X
used in Sierra CD-ROM games
Sierra Online           X
.sol files used in Sierra Online games
Matroska                X

Electronic Arts Multimedia           X
used in various EA games; files have extensions like WVE and UV2

X means that the encoding (resp. decoding) is supported.

Image Formats
=============

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

Supported Image Format   Encoding   Decoding   Comments
PGM, PPM       X   X
PAM            X   X   PAM is a PNM extension with alpha support
PGMYUV         X   X   PGM with U and V components in YUV 4:2:0
JPEG           X   X   Progressive JPEG is not supported
.Y.U.V         X   X   One raw file per component
Animated GIF   X   X   Only uncompressed GIFs are generated
PNG            X   X   2 bit and 4 bit/pixel not supported yet
SGI            X   X   SGI RGB image format

X means that the encoding (resp. decoding) is supported.

Video Codecs
============

Supported Codec   Encoding   Decoding   Comments
MPEG1 video               X     X
MPEG2 video               X     X
MPEG4                     X     X   Also known as DIVX4/5
MSMPEG4 V1                X     X
MSMPEG4 V2                X     X
MSMPEG4 V3                X     X   Also known as DIVX3
WMV7                      X     X
WMV8                      X     X   Not completely working
H.261                     X     X
H.263(+)                  X     X   Also known as Real Video 1.0
H.264                           X
MJPEG                     X     X
Lossless MJPEG            X     X
Apple MJPEG-B                   X
Sunplus MJPEG                   X   fourcc: SP5X
DV                        X     X
Huff YUV                  X     X
FFmpeg Video 1            X     X   Experimental lossless codec (fourcc: FFV1)
FFmpeg Snow               X     X   Experimental wavelet codec (fourcc: SNOW)
Asus v1                   X     X   fourcc: ASV1
Asus v2                   X     X   fourcc: ASV2
Creative YUV                    X   fourcc: CYUV
Sorenson Video 1          X     X   fourcc: SVQ1
Sorenson Video 3                X   fourcc: SVQ3
On2 VP3                         X   still experimental
Theora                          X   still experimental
Intel Indeo 3                   X   only works on i386 right now
FLV                       X     X   Sorenson H.263 used in Flash
ATI VCR1                        X   fourcc: VCR1
ATI VCR2                        X   fourcc: VCR2
Cirrus Logic AccuPak            X   fourcc: CLJR
4X Video                        X   used in certain computer games
Sony Playstation MDEC           X
Id RoQ                          X   used in Quake III, Jedi Knight 2, other computer games
Xan/WC3                         X   used in Wing Commander III .MVE files
Interplay Video                 X   used in Interplay .MVE files
Apple Animation                 X   fourcc: 'rle '
Apple Graphics                  X   fourcc: 'smc '
Apple Video                     X   fourcc: rpza
Apple QuickDraw                 X   fourcc: qdrw
Cinepak                         X
Microsoft RLE                   X
Microsoft Video-1               X
Westwood VQA                    X
Id Cinematic Video              X   used in Quake II
Planar RGB                      X   fourcc: 8BPS
FLIC video                      X
Duck TrueMotion v1              X   fourcc: DUCK
VMD Video                       X   used in Sierra VMD files
MSZH                            X   Part of LCL
ZLIB                      X     X   Part of LCL, encoder experimental
TechSmith Camtasia              X   fourcc: TSCC
IBM Ultimotion                  X   fourcc: ULTI
Miro VideoXL                    X   fourcc: VIXL

X means that the encoding (resp. decoding) is supported.

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

Audio Codecs
============

Supported Codec   Encoding   Decoding   Comments
MPEG audio layer 2        IX     IX
MPEG audio layer 1/3     IX      IX
MP3 encoding is supported through the external library LAME
AC3                       IX     IX
liba52 is used internally for decoding
Vorbis                    X      X
supported through the external library libvorbis
WMA V1/V2                       X
AAC                      X      X
supported through the external library libfaac/libfaad
Microsoft ADPCM          X      X
MS IMA ADPCM             X      X
QT IMA ADPCM                    X
4X IMA ADPCM                    X
G.726  ADPCM             X      X
Duck DK3 IMA ADPCM              X
used in some Sega Saturn console games
Duck DK4 IMA ADPCM              X
used in some Sega Saturn console games
Westwood Studios IMA ADPCM          X
used in Westwood Studios games like Command and Conquer
SMJPEG IMA ADPCM                X
used in certain Loki game ports
CD-ROM XA ADPCM                 X
CRI ADX ADPCM            X      X
used in Sega Dreamcast games
Electronic Arts ADPCM           X
used in various EA titles
Creative ADPCM                  X
RA144                           X
Real 14400 bit/s codec
RA288                           X
Real 28800 bit/s codec
RADnet                   X      IX
Real lowbitrate AC3 codec, liba52 is used for decoding
AMR-NB                   X      X
supported through an external library
AMR-WB                   X      X
supported through an external library
DV audio                        X
Id RoQ DPCM                     X
used in Quake III, Jedi Knight 2, other computer games
Interplay MVE DPCM              X
used in various Interplay computer games
Xan DPCM                        X
used in Origin's Wing Commander IV AVI files
Sierra Online DPCM              X
used in Sierra Online game audio files
Apple MACE 3                    X
Apple MACE 6                    X
FLAC                            X
FFmpeg Sonic             X      X
Experimental lossy/lossless codec

X means that the encoding (resp. decoding) is supported.

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

Platform Specific information
*****************************

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.

BSD
===

Windows
=======

Native Windows compilation
--------------------------

  * Install the current versions of MSYS and MinGW from http://www.mingw.org/. You can find detailed installation instructions in the download section and the FAQ.

  * If you want to test the FFmpeg Simple Media Player, also download the MinGW development library of SDL 1.2.x (SDL-devel-1.2.x-mingw32.tar.gz) from http://www.libsdl.org. Unpack it in a temporary place, and unpack the archive i386-mingw32msvc.tar.gz in the MinGW tool directory. Edit the sdl-config script so that it gives the correct SDL directory when invoked.

  * Extract the current version of FFmpeg (the latest release version or the current CVS snapshot whichever is recommended).

  * Start the MSYS shell (file msys.bat).

  * Change to the FFMPEG directory and follow the instructions of how to compile ffmpeg (file INSTALL). Usually, launching ./configure and make suffices. If you have problems using SDL, verify that sdl-config can be launched from the MSYS command line.

  * You can install FFmpeg in Program Files/FFmpeg by typing make install. Don't forget to copy SDL.dll at the place you launch ffplay.

Notes:

  * The target make wininstaller can be used to create a Nullsoft based Windows installer for FFmpeg and FFplay. SDL.dll must be copied in the ffmpeg directory in order to build the installer.

  * By using ./configure --enable-shared when configuring ffmpeg, you can build avcodec.dll and avformat.dll. With make install you install the FFmpeg DLLs and the associated headers in Program Files/FFmpeg.

  * Visual C++ compatibility: if you used ./configure --enable-shared when configuring FFmpeg, then FFmpeg tries to use the Microsoft Visual C++ lib tool to build avcodec.lib and avformat.lib. With these libraries, you can link your Visual C++ code directly with the FFmpeg DLLs.

Cross compilation for Windows with Linux
----------------------------------------

You must use the MinGW cross compilation tools available at http://www.mingw.org/.

Then configure ffmpeg with the following options:

     ./configure --enable-mingw32 --cross-prefix=i386-mingw32msvc-

(you can change the cross-prefix according to the prefix choosen for the MinGW tools).

Then you can easily test ffmpeg with wine (http://www.winehq.com/).

MacOS X
=======

BeOS
====

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

Old stuff:

François 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)

Developers Guide
****************

API
===

  * libavcodec is the library containing the codecs (both encoding and decoding). See libavcodec/apiexample.c to see how to use it.

  * libavformat is the library containing the file formats handling (mux and demux code for several formats). See ffplay.c to use it in a player. See output_example.c to use it to generate audio or video streams.

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 any patch you make must be published. The best way to proceed is to send your patches to the ffmpeg mailing list.

Coding Rules
============

ffmpeg is programmed in the ISO C90 language with a few additional features from ISO C99, namely:

  * the inline keyword;

  * // comments;

  * designated struct initializers (struct s x = { .i = 17 };)

  * compound literals (x = (struct s)  17, 23 };)

These features are supported by all compilers we care about, so we won't accept patches to remove their use unless they absolutely don't impair clarity and performance.

All code must compile with gcc 2.95 and gcc 3.3. Currently, ffmpeg also compiles with several other compilers, such as the Compaq ccc compiler or Sun Studio 9, and we would like to keep it that way unless it would be exceedingly involved. To ensure compatibility, please don't use any additional C99 features or gcc extensions. Watch out especially for:

  * mixing statements and declarations;

  * long long (use int64_t instead);

  * __attribute__ not protected by #ifdef __GNUC__ or similar;

  * gcc statement expressions ((x = ({ int y = 4; y; })).

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: use the JavaDoc/Doxygen format (see examples below) so that a documentation can be generated automatically. All non trivial functions should have a comment above it explaining what the function does, even if its just one sentance. All Structures and their member variables should be documented too.

     /**
      * mpeg.c
      * mpeg codec.
      *...
      */

     /**
      * Summary sentance.
      * more text ...
      * ...
      */
     typedef struct Foobar{
         int var1; /**< var1 description */
         int var2; ///< var2 description
         /** var3 description */
         int var3;
     } Foobar;

     /**
      * Summary sentance.
      * more text ...
      * ...
      * my_parameter description of my_parameter
      * return value description
      */
     int myfunc(int my_parameter)
     ...

fprintf and printf are forbidden in libavformat and libavcodec, please use av_log() instead.

Submitting patches
==================

First, (see Coding Rules) above if you didn't yet.

When you submit your patch, try to send a unified diff (diff '-up' 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.

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 http://lists.sourceforge.net/lists/listinfo/ffmpeg-devel

It also helps quite a bit if you tell us what the patch does (for example 'replaces lrint by lrintf') , and why (for example '*bsd isnt c99 compliant and has no lrint()')

We reply to all patches submitted and either apply or reject with some explanation why, but sometimes we are quite busy so it can take a week or 2

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 and formats.

Run 'make fulltest' to test all the codecs, formats and ffserver.

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