FFmpeg
FFmpeg is a hyper fast real time audio/video encoder, a
streaming server and a generic audio and video file converter. It is a
command line tool that is composed of a collection of free software /
open source libraries. It includes libavcodec, an audio/video codec
library used by several other projects, and libavformat, an audio/video
container mux and demux library.
It can grab from a standard Video4Linux video source and
convert it
into several
file formats based on DCT/motion compensation encoding. Sound is
compressed in MPEG audio layer 2 or using an AC3 compatible stream.
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.
The FFmpeg project consists of two main parts: FFmpeg, which
encodes
and decodes the multimedia streams, and FFserver, which provides
streams via HTTP for various multimedia clients. FFMpeg is completely
portable since it does not rely on proprietary DLLs. Other components
of the project include ffplay (a simple media player), and various
libraries.
Features include:
- Grabs video and audio
- Can also grab the X11 display
- Convert from one audio and video format to another format
in real time
- FFserver:
- Streams several formats at several bit rates and
resolutions simultaneously
- Handles thousands of concurrent users
- Uses the HTTP protocol to be compatible with all major
players
including Icecast (aka mpg123, xmms, winamp), MTV (to play streaming
MPEG video and audio), "Real" compatible players and Windows Media
Player
- Allows time shifting of live streams
- One FFserver instance can stream from multiple live
encoding sources and/or multiple static files
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Last Updated Tuesday, March 10 2009 @ 01:56 PM EST
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