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.
- Grabs video and audio.
- Can also grab the X11 display.
- Convert from one audio and video format to another format in real time.
- 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.
- FFprobe tool.
- VorbisComment writing for FLAC, Ogg FLAC and Ogg Speex files.
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