UtilityKit

500+ fast, free tools. Most run in your browser only; Image & PDF tools upload files to the backend when you run them.

Audio Format Converter

Convert audio between MP3, WAV, OGG, M4A and FLAC in your browser. Choose bitrate or sample rate, then download. Runs locally via ffmpeg.wasm — no uploads.

About Audio Format Converter

Audio Format Converter turns any common audio file into MP3, WAV, OGG Vorbis, M4A AAC or FLAC right in your browser. Drop in your source, pick the output container and codec, optionally pick a bitrate (for lossy formats) and a sample rate, then download the converted file. The work happens locally via ffmpeg.wasm — files never leave your device, there is no upload bar, no signup, no watermark. Use it to make a FLAC archive playable on a phone, fit lossless WAVs into the smaller MP3 format, or re-encode a podcast at a different bitrate. The tool keeps audio metadata behaviour transparent: lossless ↔ lossless conversions are bit-perfect on the audio data, lossy ↔ lossy is a re-encode (always lossy on top of lossy), and lossy ↔ lossless will not magically restore detail that was discarded earlier.

Why use Audio Format Converter

  • Five output formats covering every consumer device and DAW
  • Bitrate and sample rate controls for fine-grained results
  • 100% browser-side — no upload, no privacy compromise
  • Free and unlimited — no daily caps or watermark
  • Lossless options (WAV, FLAC) for archive-quality conversions
  • Cached engine — second run starts instantly

How to use Audio Format Converter

  1. Click or drag your audio file onto the upload area.
  2. Pick the output format — MP3 for universal support, FLAC for lossless, WAV for editing.
  3. If the format is lossy (MP3 / OGG / M4A), set the bitrate. 192 kbps is a good default for music.
  4. Optionally pick a sample rate, or leave 'Keep original' to preserve the source.
  5. Click 'Convert & Download' and wait for ffmpeg.wasm to finish.
  6. Audition the result in the inline player, then download.

When to use Audio Format Converter

  • Making a FLAC library playable on iPhone (convert to M4A)
  • Converting WAV stems into MP3 for sharing with collaborators
  • Turning OGG game audio into MP3 for a video editor that can't read OGG
  • Compressing voice notes from M4A to small MP3s
  • Re-saving a 48 kHz file to 44.1 kHz for CD-style projects
  • Extracting AAC audio from M4A and rewrapping as MP3

Examples

FLAC archive to phone

Input: FLAC 1411 kbps, 44.1 kHz

Output: M4A AAC 192 kbps — same listening quality at ~12% the size

Voice memo cleanup

Input: M4A 256 kbps stereo

Output: MP3 96 kbps mono — about 80% smaller, still clear for speech

DAW import prep

Input: MP3 128 kbps

Output: WAV 16-bit 44.1 kHz — ready for editing without re-decoding mid-project

Tips

  • FLAC → MP3 is fine; MP3 → FLAC will not recover any quality, it just makes a bigger file at the same audible level.
  • If your source is music at 44.1 kHz, leave sample rate alone — resampling adds nothing and can subtly soften the result.
  • 192 kbps MP3 is the historic sweet spot; 256 kbps is generally indistinguishable from CD on consumer playback.
  • Match output sample rate to your project — video edits typically use 48 kHz, music projects 44.1 kHz.
  • Use WAV when you plan to re-edit the file later — re-encoding lossy audio multiple times compounds artifacts.

Frequently Asked Questions

Is this free?
Yes — no signup, no watermark, no daily cap. The conversion runs in your browser.
Are my files uploaded?
No. ffmpeg.wasm runs locally; your audio never leaves your device.
Will FLAC sound better than MP3 made from the same source?
FLAC is lossless so it preserves every sample. Whether you can hear the difference depends on the bitrate of the MP3 and your playback gear; above 256 kbps most listeners cannot.
Does converting MP3 to FLAC improve quality?
No. FLAC is lossless, but it can only encode whatever the MP3 already contains — the discarded detail is gone for good.
What sample rate should I pick?
Match the source unless you have a specific reason. Video editors usually want 48 kHz; CDs and music streaming use 44.1 kHz.
Why is there no bitrate slider for WAV / FLAC?
Both are lossless and have no bitrate setting in the same sense — file size depends on input length and channel count.
Why does the first run take a while?
ffmpeg.wasm is ~30 MB and downloads once. It's cached after that, so later conversions start instantly.
What's the maximum file size?
There's no server-imposed cap. Browser memory is the practical limit; multi-hour files work but take longer to encode.

Explore the category

Glossary

Container format
The wrapper file (MP3, WAV, M4A, OGG, FLAC) that holds compressed audio plus metadata.
Codec
The compression algorithm inside the container — MP3, AAC, Vorbis, FLAC, or PCM (uncompressed).
Lossless
Encoding where the decoded audio is bit-identical to the source. WAV and FLAC are lossless here.
Lossy
Encoding that discards inaudible detail to shrink the file. MP3, OGG, AAC are lossy.
Sample rate
Audio samples per second. CDs use 44.1 kHz; video typically 48 kHz; voice can use 22.05 kHz.
Bitrate
Kilobits per second of compressed audio data. Higher bitrate = better fidelity in lossy formats.