Audio Converter
Convert audio between MP3, WAV, AAC/M4A, OGG, FLAC and Opus. Control bitrate, sample rate, channels, and loudness normalization, all on a real FFmpeg engine running fully inside your browser.
Click or drop any audio file
How This Audio Converter Works
This audio converter runs a real build of FFmpeg compiled to WebAssembly, right inside your browser tab. That is what lets it handle genuinely any format to any format conversion between MP3, WAV, AAC/M4A, OGG Vorbis, FLAC and Opus, rather than being limited to whatever a browser can natively decode.
Your file is loaded into an in memory virtual file system, FFmpeg reencodes it with your chosen codec, bitrate, sample rate and channel settings, and the output is handed back as a downloadable blob, all without a network request after the engine itself loads.
Lossy formats, MP3, AAC, OGG, Opus, expose a bitrate control since compression amount matters for those. Lossless formats, WAV and FLAC, skip that option since there is nothing to trade off. The optional loudness normalization pass runs FFmpeg’s loudnorm filter, which targets the EBU R128 broadcast standard used by podcast platforms and streaming services.
What This Tool Does
How to Convert an Audio File
Upload audio
Drop any audio file, the engine detects its format automatically.
Pick output format
Choose MP3, WAV, AAC, OGG, FLAC or Opus.
Set bitrate and sample rate
Tune quality against file size, or leave settings on the sensible defaults.
Convert and download
Click Convert Audio, preview, then download the finished file.
Command Line Equivalent
If you already run FFmpeg locally, here is what this tool generates and executes for an MP3 conversion at 192 kbps with normalization enabled.
ffmpeg -i input.wav -af loudnorm=I=-16:TP=-1.5:LRA=11 -vn -acodec libmp3lame -b:a 192k output.mp3
The “loudnorm” filter targets an integrated loudness of negative 16 LUFS, a true peak ceiling of negative 1.5 dB, and a loudness range of 11 LU, the standard values used across most streaming platforms.
FAQ: Audio Converter
Output can be MP3, WAV, AAC/M4A, OGG Vorbis, FLAC or Opus. Input can be virtually any audio format FFmpeg supports, which covers essentially every common audio file you will encounter, including less common ones like WMA or AIFF.
192 kbps is a strong general purpose default that is hard to distinguish from the source on most listening setups. Use 320 kbps for archival or audiophile use, or drop to 96 to 128 kbps for spoken word content and podcasts where file size matters more than fidelity.
It runs FFmpeg’s loudnorm filter to bring your file to a consistent integrated loudness of negative 16 LUFS, the EBU R128 target most podcast hosts and streaming platforms expect. This is useful when combining clips recorded at different volumes or preparing audio for a platform with strict loudness requirements.
No, converting an MP3 to OGG for example cannot recover data the original MP3 encoding already discarded. Reencoding a lossy file into another lossy format at the same or higher bitrate just avoids adding further loss, it never restores lost detail.
No. Conversion runs on ffmpeg.wasm, a real FFmpeg build compiled to WebAssembly that executes entirely inside your browser. Your file never leaves your device.