FFmpeg is a computer program that can record, convert and stream digital audio and video in numerous formats. FFmpeg 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. The name of the project comes from the MPEG video standards group, together with “FF” for “quick forward”. The logo uses a zigzag pattern that shows how MPEG video codecs handle entropy encoding.

FFmpeg is free software and is licensed under the LGPL or GPL depending on your choice of configuration options. If you use FFmpeg or its constituent libraries, you must adhere to the terms of the license in question.

FFmpeg is developed under GNU/Linux, but it can be compiled under most operating systems, including Apple Inc. Mac OS X, Microsoft Windows and AmigaOS. Most computing platforms and microprocessor instruction set architecture are also supported, like x86 (IA-32 and x86-64), PPC (PowerPC), ARM, DEC Alpha, SPARC, and MIPS architecture.

Windows DLLs are not portable, bloated and often slow. Moreover FFmpeg strives to support all codecs natively. A DLL loader is not conducive to that goal. For multithreaded MPEG* encoding, the encoded slices must be independent, otherwise thread n would practically have to wait for n-1 to end, so it’s quite logical that there is a small saving of quality. This is not a bug.

There are two video codecs and one video container invented in the FFmpeg project during its development. The two video codecs are the lossless “FFV1″, and the lossless or lossy “Snow codec”, for which a version 1.0 is still in development, and the video container is “NUT” which is also currently being actively developed.

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categories: ffmpeg,video encode,video codec,streaming,open source,transcode,encoder,video convert,audio,video streaming

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