ffmpeg

Video format conversion with FFmpeg involves specifying input and output containers while selecting compatible codecs. FFmpeg automatically handles the encoding; you control quality through CRF values (0–51, lower is better) and presets (ultrafast to veryslow) to balance file size and speed. Common workflows convert MP4 to MKV or WebM by selecting the target container and codec pair.

Batch processing applies identical FFmpeg commands across multiple files using shell loops or scripting. Define your encoding parameters once—codec, CRF, bitrate mode, preset—then iterate over input files, outputting to a destination directory. This scales audio extraction, format conversion, and filtering across entire directories efficiently.

Audio extraction with FFmpeg isolates the audio stream from video containers using codec copy or re-encoding. You can extract to MP3, AAC, Opus, or FLAC by selecting the target codec and bitrate. FFmpeg handles container demuxing and audio codec conversion in a single pass.

ffprobe inspects video and audio files to extract metadata—duration, codec, bitrate, resolution, frame rate—without decoding. Query this metadata programmatically to guide conditional encoding: for example, skip re-encoding if source already matches target specifications, or adjust CRF based on detected bitrate.

FFmpeg supports H.264/AVC, H.265/HEVC, VP9, and AV1 video codecs with AAC, MP3, Opus, and FLAC audio across MP4, MKV, WebM, MOV, and AVI containers. Codec and container compatibility varies; H.264 + AAC in MP4 is widely compatible, while WebM requires VP9 or AV1 video with Opus audio.

FFmpeg's filter graph chains multiple operations—scaling, trimming, deinterlacing, color correction—in sequence before encoding. Apply filters using the -filter or -vf flag to stack operations; this reduces file I/O and re-encoding overhead compared to separate passes and produces final output in one command.