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 parameters that can possibly be derived automatically. You usually only have to specify 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 grab video and audio from devices given that you specify the input format and device.

     ffmpeg -f audio_device -i /dev/dsp -f video4linux2 -i /dev/video0 /tmp/out.mpg

Note that you must activate the right video source and channel before launching FFmpeg with any TV viewer such as xawtv (http://bytesex.org/xawtv/) by Gerd Knorr. You also have to set the audio recording levels correctly with a standard mixer.

X11 grabbing
============

FFmpeg can grab the X11 display.

     ffmpeg -f x11grab -i :0.0 /tmp/out.mpg

0.0 is display.screen number of your X11 server, same as the DISPLAY environment variable.

     ffmpeg -f x11grab -i :0.0+10,20 /tmp/out.mpg

0.0 is display.screen number of your X11 server, same as the DISPLAY environment variable. 10 is the x-offset and 20 the y-offset for the grabbing.

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

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

Examples:

* You can use YUV files as input:

     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

test.yuv is a file containing raw YUV planar data. Each frame is composed of the Y plane followed by the U and V planes at half vertical and horizontal resolution.

* You can output to a raw YUV420P file:

     ffmpeg -i mydivx.avi 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

Converts 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

Converts a.wav to MPEG audio at 22050Hz sample rate.

* 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 64k /tmp/a.mp2 -ab 128k /tmp/b.mp2 -map 0:0 -map 0:0

Converts a.wav to a.mp2 at 64 kbits and to b.mp2 at 128 kbits. '-map file:index' specifies 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 800k -g 300 -bf 2 -acodec mp3 -ab 128k snatch.avi

This is a typical DVD ripping example; the input is a VOB file, the output 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, and GOP size is 300 which means one intra frame every 10 seconds for 29.97fps input video. Furthermore, the audio stream is MP3-encoded so you need to enable LAME support by passing --enable-mp3lame to configure. 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}...

As a general rule, options are applied to the next specified file. Therefore, order is important, and you can have the same option on the command line multiple times. Each occurrence is then applied to the next input or output file.

* To set the video bitrate of the output file to 64kbit/s:

     ffmpeg -i input.avi -b 64k output.avi

* To force the frame rate of the input and output file to 24 fps:

     ffmpeg -r 24 -i input.avi output.avi

* To force the frame rate of the output file to 24 fps:

     ffmpeg -i input.avi -r 24 output.avi

* To force the frame rate of input file to 1 fps and the output file to 24 fps:

     ffmpeg -r 1 -i input.avi -r 24 output.avi

The 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 parameters for the outputs as the one specified for the inputs.

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

-L
     Show license.

-h
     Show help.

-version
     Show version.

-formats
     Show available formats, codecs, protocols, ...

-f fmt
     Force format.

-i filename
     input filename

-y
     Overwrite output files.

-t duration
     Set the recording time in seconds. hh:mm:ss[.xxx] syntax is also supported.

-fs limit_size
     Set the file size limit.

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

-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 timestamps of the input files. Specifying a positive offset means that the corresponding streams are delayed by 'offset' seconds.

-title string
     Set the title.

-timestamp time
     Set the timestamp.

-author string
     Set the author.

-copyright string
     Set the copyright.

-comment string
     Set the comment.

-album string
     Set the album.

-track number
     Set the track.

-year number
     Set the year.

-v verbose
     Control amount of logging.

-target type
     Specify target file type ("vcd", "svcd", "dvd", "dv", "dv50", "pal-vcd", "ntsc-svcd", ... ). All the format options (bitrate, codecs, buffer sizes) are then set automatically. 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 conflict with the standard, as in:

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

-dframes number
     Set the number of data frames to record.

-scodec codec
     Force subtitle codec ('copy' to copy stream).

-newsubtitle
     Add a new subtitle stream to the current output stream.

-slang code
     Set the ISO 639 language code (3 letters) of the current subtitle stream.

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

-b bitrate
     Set the video bitrate in bit/s (default = 200 kb/s).

-vframes number
     Set the number of video frames to record.

-r fps
     Set frame rate (Hz value, fraction or abbreviation), (default = 25).

-s size
     Set frame size. The format is wxh (ffserver default = 160x128, ffmpeg default = same as source). 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 padcolor is expressed as a six digit hexadecimal number where the first two digits represent red, the middle two digits green and last two digits blue (default = 000000 (black)).

-vn
     Disable video recording.

-bt tolerance
     Set video bitrate tolerance (in bit/s).

-maxrate bitrate
     Set max video bitrate tolerance (in bit/s).

-minrate bitrate
     Set min video bitrate tolerance (in bit/s).

-bufsize size
     Set rate control buffer size (in bits).

-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 is generated at the exact requested bitrate in the second pass.

-passlogfile file
     Set two pass logfile name to FILE.

-newvideo
     Add a new video stream to the current output stream.

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

-pix_fmt format
     Set pixel format.

-g gop_size
     Set the group of pictures size.

-intra
     Use only intra frames.

-vdt n
     Discard threshold.

-qscale q
     Use fixed video quantizer scale (VBR).

-qmin q
     minimum video quantizer scale (VBR)

-qmax q
     maximum video quantizer scale (VBR)

-qdiff q
     maximum difference between the quantizer scales (VBR)

-qblur blur
     video quantizer scale blur (VBR)

-qcomp compression
     video quantizer scale compression (VBR)

-lmin lambda
     minimum video lagrange factor (VBR)

-lmax lambda
     max video lagrange factor (VBR)

-mblmin lambda
     minimum macroblock quantizer scale (VBR)

-mblmax lambda
     maximum macroblock quantizer scale (VBR)

     These four options (lmin, lmax, mblmin, mblmax) use 'lambda' units, but you may use the QP2LAMBDA constant to easily convert from 'q' units:

          ffmpeg -i src.ext -lmax 21*QP2LAMBDA dst.ext

-rc_init_cplx complexity
     initial complexity for single 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 = tex^qComp).

-rc_override override
     rate control override for specific intervals

-me method
     Set motion estimation method to METHOD. Available methods are (from lowest 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_CAREFUL (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: Choose the one which needs the fewest bits.

     2
          FF_MB_DECISION_RD: rate distortion

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

-part
     Use data partitioning (MPEG-4 only).

-bug param
     Work around encoder bugs that are not auto-detected.

-strict strictness
     How strictly to follow the standards.

-aic
     Enable Advanced intra coding (h263+).

-umv
     Enable Unlimited Motion Vector (h263+)

-deinterlace
     Deinterlace pictures.

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

-top n
     top=1/bottom=0/auto=-1 field first

-dc precision
     Intra_dc_precision.

-vtag fourcc/tag
     Force video tag/fourcc.

-qphist
     Show QP histogram.

-vbsf bitstream filter
     Bitstream filters available are "dump_extra", "remove_extra", "noise".

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

-aframes number
     Set the number of audio frames to record.

-ar freq
     Set the audio sampling frequency (default = 44100 Hz).

-ab bitrate
     Set the audio bitrate in bit/s (default = 64k).

-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 specify that the raw codec data must be copied as is.

-newaudio
     Add a new audio track to the output file. If you want to specify parameters, do so before -newaudio (-acodec, -ab, etc..).

     Mapping will be done automatically, if the number of output streams is equal to the number of input streams, else it will pick the first one that matches. You can override the mapping using -map as usual.

     Example:

          ffmpeg -i file.mpg -vcodec copy -acodec ac3 -ab 384k test.mpg -acodec mp2 -ab 192k -newaudio

-alang code
     Set the ISO 639 language code (3 letters) of the current audio stream.

Advanced Audio options:
=======================

-atag fourcc/tag
     Force audio tag/fourcc.

-absf bitstream filter
     Bitstream filters available are "dump_extra", "remove_extra", "noise", "mp3comp", "mp3decomp".

Subtitle options:
=================

-scodec codec
     Force subtitle codec ('copy' to copy stream).

-newsubtitle
     Add a new subtitle stream to the current output stream.

-slang code
     Set the ISO 639 language code (3 letters) of the current subtitle stream.

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

-vc channel
     Set video grab channel (DV1394 only).

-tvstd standard
     Set television standard (NTSC, PAL (SECAM)).

-isync
     Synchronize read on input.

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

-map input stream id[:input stream id]
     Set stream mapping from input streams to output streams. Just enumerate the input streams in the order you want them in the output. [input stream id] sets the (input) stream to sync against.

-map_meta_data outfile:infile
     Set meta data information of outfile from infile.

-debug
     Print specific debug info.

-benchmark
     Add timings for benchmarking.

-dump
     Dump each input packet.

-hex
     When dumping packets, also dump the payload.

-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_input
     Loop over the input stream. Currently it works only for image streams. This option is used for automatic FFserver testing.

-loop_output number_of_times
     Repeatedly loop output for formats that support looping such as animated GIF (0 will loop the output infinitely).

-threads count
     Thread count.

-vsync parameter
     Video sync method. Video will be stretched/squeezed to match the timestamps, it is done by duplicating and dropping frames. With -map you can select from which stream the timestamps should be taken. You can leave either video or audio unchanged and sync the remaining stream(s) to the unchanged one.

-async samples_per_second
     Audio sync method. "Stretches/squeezes" the audio stream to match the timestamps, the parameter is the maximum samples per second by which the audio is changed. -async 1 is a special case where only the start of the audio stream is corrected without any later correction.

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 standard input or to write to standard output.

FFmpeg also handles many protocols specified with an URL syntax.

Use 'ffmpeg -formats' to see 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 bitrate application, use a low frame rate and a small GOP size. This is especially true for RealVideo 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 50k -s qcif -f rv10 /tmp/b.rm

  * The parameter 'q' which is displayed while encoding is the current quantizer. The value 1 indicates that a very good quality could be achieved. The value 31 indicates the worst quality. If q=31 appears too often, it means that the encoder cannot compress enough to meet your bitrate. You must either increase the bitrate, 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 motion estimation completely (you have only I-frames, which means it is about as good as JPEG compression).

  * To have very low audio bitrates, 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 the same quality factor in the encoder as in the decoder. It allows almost lossless encoding.

external libraries
******************

FFmpeg can be hooked up with a number of external libraries to add support for more formats. None of them are used by default, their use has to be explicitly requested by passing the appropriate flags to ./configure.

AMR
===

AMR comes in two different flavors, WB and NB. FFmpeg can make use of the AMR WB (floating-point mode) and the AMR NB (both floating-point and fixed-point mode) reference decoders and encoders.

  * For AMR WB floating-point and AMR NB floating point support, go to http://www.penguin.cz/~utx/amr and follow the instructions for building the libraries.

  * For AMR NB fixed-point download TS26.073 REL-6 V6.0.0 from http://www.3gpp.org/ftp/Specs/archive/26_series/26.073/26073-600.zip and extract the source to the directory libavcodec/amr. You must also add -DMMS_IO to CFLAGS in libavcodec/amr/makefile, i.e. ``CFLAGS = -Wall -pedantic-errors -I. $(CFLAGS_$(MODE)) -D$(VAD) -DMMS_IO''.

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
MPEG-1 systems   X     X
muxed audio and video
MPEG-2 PS   X     X
also known as VOB file
MPEG-2 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
Raw Shorten audio         X
SUN AU format   X     X
NUT   X   X   NUT Open Container Format
QuickTime          X    X
MPEG-4             X    X
MPEG-4 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.
Nullsoft Video (NSV) format        X
ADTS AAC audio   X   X
Creative VOC   X   X   Created for the Sound Blaster Pro.
American Laser Games MM         X
Multimedia format used in games like Mad Dog McCree
AVS        X
Multimedia format used by the Creature Shock game.
Smacker        X
Multimedia format used by many games.
GXF    X   X
General eXchange Format SMPTE 360M, used by Thomson Grass Valley playout servers.
CIN        X
Multimedia format used by Delphine Software games.
MXF        X
Material eXchange Format SMPTE 377M, used by D-Cinema, broadcast industry.
SEQ        X
Tiertex .seq files used in the DOS CDROM version of the game Flashback.
DXA        X
This format is used in non-Windows version of Feeble Files game and
different game cutscenes repacked for use with ScummVM.
THP        X
Used on the Nintendo GameCube.
C93        X
Used in the game Cyberia from Interplay.
Bethsoft VID        X
Used in some games from Bethesda Softworks.
CRYO APC        X
Audio format used in some games by CRYO Interactive Entertainment.

X means that 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.
Targa              X   Targa (.TGA) image format.
TIFF           X   X   YUV, JPEG and some extension is not supported yet.
SGI            X   X   SGI RGB image format

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

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

Supported Codec   Encoding   Decoding   Comments
MPEG-1 video              X     X
MPEG-2 video              X     X
MPEG-4                    X     X
MSMPEG4 V1                X     X
MSMPEG4 V2                X     X
MSMPEG4 V3                X     X
WMV7                      X     X
WMV8                      X     X   not completely working
WMV9                            X   not completely working
VC1                             X
H.261                     X     X
H.263(+)                  X     X   also known as RealVideo 1.0
H.264                           X
RealVideo 1.0             X     X
RealVideo 2.0             X     X
MJPEG                     X     X
lossless MJPEG            X     X
JPEG-LS                   X     X   fourcc: MJLS, lossless and near-lossless is supported
Apple MJPEG-B                   X
Sunplus MJPEG                   X   fourcc: SP5X
DV                        X     X
HuffYUV                   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
On2 VP5                         X   fourcc: VP50
On2 VP6                         X   fourcc: VP60,VP61,VP62
Theora                    X     X   still experimental
Intel Indeo 3                   X
FLV                       X     X   Sorenson H.263 used in Flash
Flash Screen Video        X     X   fourcc: FSV1
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
Duck TrueMotion v2              X   fourcc: TM20
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
QPEG                            X   fourccs: QPEG, Q1.0, Q1.1
LOCO                            X
Winnov WNV1                     X
Autodesk Animator Studio Codec           X   fourcc: AASC
Fraps FPS1                      X
CamStudio                       X   fourcc: CSCD
American Laser Games Video         X   Used in games like Mad Dog McCree
ZMBV                       X    X   Encoder works only on PAL8
AVS Video                       X   Video encoding used by the Creature Shock game.
Smacker Video                   X   Video encoding used in Smacker.
RTjpeg                          X   Video encoding used in NuppelVideo files.
KMVC                            X   Codec used in Worms games.
VMware Video                    X   Codec used in videos captured by VMware.
Cin Video                       X   Codec used in Delphine Software games.
Tiertex Seq Video               X   Codec used in DOS CDROM FlashBack game.
DXA Video                       X   Codec originally used in Feeble Files game.
AVID DNxHD                      X   aka SMPTE VC3
C93 Video                       X   Codec used in Cyberia game.
THP                             X   Used on the Nintendo GameCube.
Bethsoft VID                    X   Used in some games from Bethesda Softworks.

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

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
WMA V1/V2                X      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
16 -> 4, 8 -> 4, 8 -> 3, 8 -> 2
THP ADPCM                       X
Used on the Nintendo GameCube.
RA144                           X
Real 14400 bit/s codec
RA288                           X
Real 28800 bit/s codec
RADnet                   X      IX
Real low bitrate 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 lossless audio      X      X
Shorten lossless audio          X
Apple lossless audio            X
QuickTime fourcc 'alac'
FFmpeg Sonic             X      X
experimental lossy/lossless codec
Qdesign QDM2                    X
there are still some distortions
Real COOK                       X
All versions except 5.1 are supported
DSP Group TrueSpeech            X
True Audio (TTA)                X
Smacker Audio                   X
WavPack Audio                   X
Cin Audio                       X
Codec used in Delphine Software games.
Intel Music Coder               X
Musepack                        X
Only SV7 is supported
DT$ Coherent Audio              X

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

I means that an integer-only version is available, too (ensures high performance on systems without hardware floating point support).

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

BSD
===

BSD make will not build FFmpeg, you need to install and use GNU Make (gmake).

Windows
=======

To get help and instructions for using FFmpeg under Windows, check out the FFmpeg Windows Help Forum at http://arrozcru.no-ip.org/ffmpeg/.

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.

     NOTE: Use at least bash 3.1. Older versions are known to be failing on the configure script.

  * If you want to test the FFplay, 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 directory, 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.

  * 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. Do not forget to copy SDL.dll to the place you launch ffplay from.

Notes:

  * The target make wininstaller can be used to create a Nullsoft based Windows installer for FFmpeg and FFplay. SDL.dll must be copied to 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, 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 (see below).

Visual C++ compatibility
------------------------

FFmpeg will not compile under Visual C++ -- and it has too many dependencies on the GCC compiler to make a port viable. However, if you want to use the FFmpeg libraries in your own applications, you can still compile those applications using Visual C++. An important restriction to this is that you have to use the dynamically linked versions of the FFmpeg libraries (i.e. the DLLs), and you have to make sure that Visual-C++-compatible import libraries are created during the FFmpeg build process.

This description of how to use the FFmpeg libraries with Visual C++ is based on Visual C++ 2005 Express Edition Beta 2. If you have a different version, you might have to modify the procedures slightly.

Here are the step-by-step instructions for building the FFmpeg libraries so they can be used with Visual C++:

 1. Install Visual C++ (if you have not done so already).

 2. Install MinGW and MSYS as described above.

 3. Add a call to vcvars32.bat (which sets up the environment variables for the Visual C++ tools) as the first line of msys.bat. The standard location for vcvars32.bat is C:\Program Files\Microsoft Visual Studio 8\VC\bin\vcvars32.bat, and the standard location for msys.bat is C:\msys\1.0\msys.bat. If this corresponds to your setup, add the following line as the first line of msys.bat:

     call "C:\Program Files\Microsoft Visual Studio 8\VC\bin\vcvars32.bat"

 4. Start the MSYS shell (file msys.bat) and type link.exe. If you get a help message with the command line options of link.exe, this means your environment variables are set up correctly, the Microsoft linker is on the path and will be used by FFmpeg to create Visual-C++-compatible import libraries.

 5. Extract the current version of FFmpeg and change to the FFmpeg directory.

 6. Type the command ./configure --enable-shared --disable-static --enable-memalign-hack to configure and, if that did not produce any errors, type make to build FFmpeg.

 7. The subdirectories libavformat, libavcodec, and libavutil should now contain the files avformat.dll, avformat.lib, avcodec.dll, avcodec.lib, avutil.dll, and avutil.lib, respectively. Copy the three DLLs to your System32 directory (typically C:\Windows\System32).

And here is how to use these libraries with Visual C++:

 1. Create a new console application ("File / New / Project") and then select "Win32 Console Application". On the appropriate page of the Application Wizard, uncheck the "Precompiled headers" option.

 2. Write the source code for your application, or, for testing, just copy the code from an existing sample application into the source file that Visual C++ has already created for you. (Note that your source filehas to have a .cpp extension; otherwise, Visual C++ will not compile the FFmpeg headers correctly because in C mode, it does not recognize the inline keyword.)  For example, you can copy output_example.c from the FFmpeg distribution (but you will have to make minor modifications so the code will compile under C++, see below).

 3. Open the "Project / Properties" dialog box. In the "Configuration" combo box, select "All Configurations" so that the changes you make will affect both debug and release builds. In the tree view on the left hand side, select "C/C++ / General", then edit the "Additional Include Directories" setting to contain the complete paths to the libavformat, libavcodec, and libavutil subdirectories of your FFmpeg directory. Note that the directories have to be separated using semicolons. Now select "Linker / General" from the tree view and edit the "Additional Library Directories" setting to contain the same three directories.

 4. Still in the "Project / Properties" dialog box, select "Linker / Input" from the tree view, then add the files avformat.lib, avcodec.lib, and avutil.lib to the end of the "Additional Dependencies". Note that the names of the libraries have to be separated using spaces.

 5. Now, select "C/C++ / Code Generation" from the tree view. Select "Debug" in the "Configuration" combo box. Make sure that "Runtime Library" is set to "Multi-threaded Debug DLL". Then, select "Release" in the "Configuration" combo box and make sure that "Runtime Library" is set to "Multi-threaded DLL".

 6. Click "OK" to close the "Project / Properties" dialog box and build the application. Hopefully, it should compile and run cleanly. If you used output_example.c as your sample application, you will get a few compiler errors, but they are easy to fix. The first type of error occurs because Visual C++ does not allow an int to be converted to an enum without a cast. To solve the problem, insert the required casts (this error occurs once for a CodecID and once for a CodecType).  The second type of error occurs because C++ requires the return value of malloc to be cast to the exact type of the pointer it is being assigned to. Visual C++ will complain that, for example, (void *) is being assigned to (uint8_t *) without an explicit cast. So insert an explicit cast in these places to silence the compiler. The third type of error occurs because the snprintf library function is called _snprintf under Visual C++.  So just add an underscore to fix the problem. With these changes, output_example.c should compile under Visual C++, and the resulting executable should produce valid video files.

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 --target-os=mingw32 --cross-prefix=i386-mingw32msvc-

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

Then you can easily test FFmpeg with Wine (http://www.winehq.com/).

Compilation under Cygwin
------------------------

Cygwin works very much like Unix.

Just install your Cygwin with all the "Base" packages, plus the following "Devel" ones:

     binutils, gcc-core, make, subversion

Do not install binutils-20060709-1 (they are buggy on shared builds); use binutils-20050610-1 instead.

Then run

     ./configure --enable-static --disable-shared

to make a static build or

     ./configure --enable-shared --disable-static

to build shared libraries.

If you want to build FFmpeg with additional libraries, download Cygwin "Devel" packages for Ogg and Vorbis from any Cygwin packages repository and/or SDL, xvid, faac, faad2 packages from Cygwin Ports, (http://cygwinports.dotsrc.org/).

Crosscompilation for Windows under Cygwin
-----------------------------------------

With Cygwin you can create Windows binaries that do not need the cygwin1.dll.

Just install your Cygwin as explained before, plus these additional "Devel" packages:

     gcc-mingw-core, mingw-runtime, mingw-zlib

and add some special flags to your configure invocation.

For a static build run

     ./configure --target-os=mingw32 --enable-memalign-hack --enable-static --disable-shared --extra-cflags=-mno-cygwin --extra-libs=-mno-cygwin

and for a build with shared libraries

     ./configure --target-os=mingw32 --enable-memalign-hack --enable-shared --disable-static --extra-cflags=-mno-cygwin --extra-libs=-mno-cygwin

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 did not test building on the net_server version of BeOS.

FFserver is broken (needs poll() implementation).

There are 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). Look at libavcodec/apiexample.c to see how to use it.

  * libavformat is the library containing the file format handling (mux and demux code for several formats). Look at 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 will not accept patches to remove their use unless they absolutely do not 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 do not use any additional C99 features or GCC extensions. Especially watch out 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 presentation is the one specified by 'indent -i4 -kr -nut'. The TAB character is forbidden outside of Makefiles as is any form of trailing whitespace. Commits containing either will be rejected by the Subversion repository.

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

Comments: Use the JavaDoc/Doxygen format (see examples below) so that code documentation can be generated automatically. All nontrivial functions should have a comment above them explaining what the function does, even if it is just one sentence. All structures and their member variables should be documented, too.

     /**
      * @file mpeg.c
      * MPEG codec.
      * @author ...
      */

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

     /**
      * Summary sentence.
      * more text ...
      * ...
      * @param my_parameter description of my_parameter
      * @return return value description
      */
     int myfunc(int my_parameter)
     ...

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

Development Policy
==================

 1. You must not commit code which breaks FFmpeg! (Meaning unfinished but enabled code which breaks compilation or compiles but does not work or breaks the regression tests) You can commit unfinished stuff (for testing etc), but it must be disabled (#ifdef etc) by default so it does not interfere with other developers' work.

 2. You do not have to over-test things. If it works for you, and you think it should work for others, then commit. If your code has problems (portability, triggers compiler bugs, unusual environment etc) they will be reported and eventually fixed.

 3. Do not commit unrelated changes together, split them into self-contained pieces. Also do not forget that if part B depends on part A, but A does not depend on B, then A can and should be committed first and separate from B. Keeping changes well split into self-contained parts makes reviewing and understanding them on the commit log mailing list easier. This also helps in case of debugging later on. Also if you have doubts about splitting or not splitting, do not hesitate to ask/disscuss it on the developer mailing list.

 4. Do not change behavior of the program (renaming options etc) without first discussing it on the ffmpeg-devel mailing list. Do not remove functionality from the code. Just improve!

     Note: Redundant code can be removed.

 5. Do not commit changes to the build system (Makefiles, configure script) which change behavior, defaults etc, without asking first. The same applies to compiler warning fixes, trivial looking fixes and to code maintained by other developers. We usually have a reason for doing things the way we do. Send your changes as patches to the ffmpeg-devel mailing list, and if the code maintainers say OK, you may commit. This does not apply to files you wrote and/or maintain.

 6. We refuse source indentation and other cosmetic changes if they are mixed with functional changes, such commits will be rejected and removed. Every developer has his own indentation style, you should not change it. Of course if you (re)write something, you can use your own style, even though we would prefer if the indentation throughout FFmpeg was consistent (Many projects force a given indentation style - we do not.). If you really need to make indentation changes (try to avoid this), separate them strictly from real changes.

     NOTE: If you had to put if(){ .. } over a large (> 5 lines) chunk of code, then either do NOT change the indentation of the inner part within (do not move it to the right)! or do so in a separate commit

 7. Always fill out the commit log message. Describe in a few lines what you changed and why. You can refer to mailing list postings if you fix a particular bug. Comments such as "fixed!" or "Changed it." are unacceptable.

 8. If you apply a patch by someone else, include the name and email address in the log message. Since the ffmpeg-cvslog mailing list is publicly archived you should add some SPAM protection to the email address. Send an answer to ffmpeg-devel (or wherever you got the patch from) saying that you applied the patch.

 9. Do NOT commit to code actively maintained by others without permission. Send a patch to ffmpeg-devel instead. If noone answers within a reasonable timeframe (12h for build failures and security fixes, 3 days small changes, 1 week for big patches) then commit your patch if you think it is OK. Also note, the maintainer can simply ask for more time to review!

 10. Subscribe to the ffmpeg-cvslog mailing list. The diffs of all commits are sent there and reviewed by all the other developers. Bugs and possible improvements or general questions regarding commits are discussed there. We expect you to react if problems with your code are uncovered.

 11. Update the documentation if you change behavior or add features. If you are unsure how best to do this, send a patch to ffmpeg-devel, the documentation maintainer(s) will review and commit your stuff.

 12. Try to keep important discussions and requests (also) on the public developer mailing list, so that all developers can benefit from them.

 13. Never write to unallocated memory, never write over the end of arrays, always check values read from some untrusted source before using them as array index or other risky things.

 14. Remember to check if you need to bump versions for the specific libav parts (libavutil, libavcodec, libavformat) you are changing. You need to change the version integer and the version string. Incrementing the first component means no backward compatibility to previous versions (e.g. removal of a function from the public API). Incrementing the second component means backward compatible change (e.g. addition of a function to the public API). Incrementing the third component means a noteworthy binary compatible change (e.g. encoder bug fix that matters for the decoder).

 15. If you add a new codec, remember to update the changelog, add it to the supported codecs table in the documentation and bump the second component of the libavcodec version number appropriately. If it has a fourcc, add it to libavformat/avienc.c, even if it is only a decoder.

 16. Do not change code to hide warnings without ensuring that the underlying logic is correct and thus the warning was inappropriate.

We think our rules are not too hard. If you have comments, contact us.

Note, these rules are mostly borrowed from the MPlayer project.

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

First, (see Coding Rules) above if you did not yet.

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

Also please do not submit patches which contain several unrelated changes. Split them into individual self-contained patches; this makes reviewing them much easier.

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 will not be trashed during transmission) to the ffmpeg-devel mailing list, see http://lists.mplayerhq.hu/mailman/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 isn't C99 compliant and has no lrint()')

Also please if you send several patches, send each patch as seperate mail, dont attach several unrelated patches to the same mail.

Patch review process
====================

All patches posted to ffmpeg-devel will be reviewed, unless they contain a clear note that the patch is not for SVN. Reviews and comments will be posted as replies to the patch on the mailing list. The patch submitter then has to take care of every comment, that can be by resubmitting a changed patch or by disscussion. Resubmitted patches will themselves be reviewed like any other patch. If at some point a patch passes review with no comments then it is approved, that can for simple and small patches happen immediately while large patches will generally have to be changed and reviewed many times before they are approved. After a patch is approved it will be committed to the repository.

We will review all submitted patches, but sometimes we are quite busy so especially for large patches this can take several weeks.

When resubmitting patches, please do not make any significant changes not related to the comments received during review. Such patches will be rejected. Instead, submit  significant changes or new features as separate patches.

Regression tests
================

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

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

The regression tests then go 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 results of the regression tests. In this case, the reference results of the regression tests shall be modified accordingly].