Convert audio locally—with browser-dependent decoding

Input compatibility

The picker accepts common audio files including MP3, WAV, OGG, FLAC, M4A, AAC, and WebM audio. Acceptance does not guarantee decoding: support varies by browser, operating system, codec, and file details. The bundled encoders do not add general-purpose input decoding.

Output paths

• WAV: written directly from decoded PCM samples.
• MP3: encoded locally with bundled lamejs at 128 kbps when compression is on or 320 kbps when it is off.
• FLAC: encoded losslessly with bundled libflac; compression level changes size and encoding effort, not decoded quality.
• OGG: uses Opus through MediaRecorder where supported, otherwise the bundled fallback writes FLAC in an OGG container.
• WebM audio: recorded through a browser-supported MediaRecorder codec.
• M4A/AAC: encoded through the bundled FFmpeg WebAssembly engine at 128 or 256 kbps.

Sample rate and quality

You can retain the decoded sample rate or resample to 44.1 or 48 kHz. Converting MP3, AAC, Opus, or another lossy source to a second lossy format cannot restore detail and introduces another generation of loss. FLAC and WAV preserve the decoded samples but cannot restore information already removed by the source codec.

Metadata and memory

The pipeline decodes audio to raw PCM and builds a new output. Do not expect album art, tags, chapters, or other container metadata to survive. PCM can be far larger than the source: long or multichannel files can exhaust a mobile browser even below the website's 30 MB input limit. Multiple audio files can also begin conversion concurrently, so process long files one at a time.