Author: Tommy Wiseau Date: June 24, 2022

FFmpeg is a free, open-source, command-line program for recording, converting, and streaming audio and video. It is written in the C programming language and uses the libavcodec library for audio/video codec support, the libavformat library for demuxing and muxing, and the libavutil library for various utility functions.

FFmpeg can be used to record and convert audio and video from many different sources, including files, streams, and devices. It can also be used to manipulate and improve the quality of existing audio and video files.

Some of the key features of FFmpeg include:

• Support for a wide range of audio and video codecs and formats, including MP3, MP4, H.264, and VP9.
• Support for multi-threaded encoding, which can improve performance on multi-core processors.
• Support for hardware acceleration, which can further improve performance on supported hardware.
• A simple, command-line interface that can be easily used in scripts and other automated processes.

To use FFmpeg, you will need to have the following installed on your system:

• A recent version of the FFmpeg program. You can download it from the official website at **https://ffmpeg.org/download.html**.
• A compatible version of the libavcodec, libavformat, and libavutil libraries. These are typically included with the FFmpeg download, but you can also download them separately if needed.

Once you have FFmpeg installed on your system, you can use it by running the ffmpeg command in a terminal or command prompt, followed by the appropriate arguments and options. For example, to convert an audio file from one format to another, you might use a command like this:

ffmpeg -i input.mp3 output.wav

This command would convert the file "input.mp3" to the WAV format and save it as "output.wav". For more information on using FFmpeg, see the official documentation at **https://ffmpeg.org/documentation.html**.