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There are a lot of different file formats today for video and audio files. Sometimes a user of these files may find it necessary to change the format of the files to another format. The format of the file is noted by the extension of the filename. For example, a file named test.wav has a wav extension. The wav extension lets the computer know how it should open the file and lets the user know that it is a wav file. In order to change the format of test.wav the file needs to be encoded in the new format. Simply changing the extension to another extension is not enough. Ffmpeg is an open source code, with a command line interface, that quickly converts files from one format to another. At times it may be necessary to grab live audio or video from a source. Once the information is gathered it needs to be encoded correctly in order to be able to use it through a medium. FFmpeg supports this as well. Once the video is created there may be a need to resize the image. FFmpeg uses a polyphase filter to accomplish the task of resizing. FFmpeg supports a lot of file formats through the libavformat library. A complete listing of the file formats can be found in section five of the FFmpeg documentation located at http://ffmpeg.sourceforge.net/ffmpeg-doc.html . The FFmpeg documentation is an excellent reference for those who would like to find out all the ends and outs of FFmpeg.
FFmpeg can be used on Linux and on Windows machines. In order to properly utilize FFmpeg the computer needs to have a certain kind of environment. By the nature of Linux OS, it already has this environment. Windows machines are a different story. The environment needs to be created on a Windows machine inorder to invoke various FFmpeg functions. The steps for creating the environment can be found in FFmpeg documentation. The steps are also explained on this wiki site, in simpler terms, at FFmpeg_Windows.
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