Open-source audio normalization powered by FFmpeg, LUFS and True Peak processing. Built for music production, podcasts, gaming audio and large batch workflows - without requiring command-line knowledge.
To use melcom's FFmpeg Audio Normalizer, the complete FFmpeg suite is required:
ffmpeg.exe - Audio analysis and normalizationffplay.exe - Audio preview playbackffprobe.exe - Reading technical audio informationffmpeg-master-latest-win64-gpl.zip After extracting FFmpeg, configure the path to the FFmpeg folder inside the application's options menu.
AudioNormalizer.exe.File -> Options ffmpeg.exe, ffplay.exe and ffprobe.exe Recommended for music production, mastering and archival work. The audio file is analyzed during the first pass and normalized with a fixed gain during the second pass. This preserves the original dynamics as accurately as possible.
Recommended for speech, podcasts, streams and broadcast material. Loudness differences are actively balanced during processing.
Pure loudness normalization without additional sound shaping.
Adds gentle compression and subtle soft clipping for a tighter overall mix.
Delivers stronger impact, more energy and more noticeable dynamic processing.
The strongest processing mode for dense, sharp or aggressive electronic material.
When exporting to WAV or FLAC, the original sample rate and bit depth are preserved automatically. No unnecessary downsampling is performed.
Artist, album, title, year and additional metadata are copied automatically to the normalized output file.
MP3 files automatically use the ID3v2.3 standard for maximum compatibility with Windows Explorer and hardware players.
ffprobe.exe does not affect normalization itself, but disables technical audio information and duration detection. This project was created by melcom and released under the MIT License.
Project website:
melcom-creations.github.io/melcom-music