Audio

MP3 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.

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About this tool

This MP3 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 format (the matching codec is chosen for you), 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. Keep the trade-offs in mind: FLAC output stores the decoded samples losslessly, WAV output is always 16-bit PCM (so 24-bit or 32-bit float sources are reduced to 16-bit), lossy ↔ lossy is a re-encode (lossy on top of lossy), and lossy → lossless will not restore detail that was discarded earlier.

Why use it

  • 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

  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 it helps

  • 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
  • Turning a FLAC master into a WAV for an editor that can't read FLAC

Examples

FLAC archive to phone

Input

FLAC album track, 44.1 kHz stereo

Expected result

M4A AAC at 192 kbps — about 14% of the 1,411 kbps uncompressed CD rate

Voice memo cleanup

Input

M4A voice memo at 256 kbps

Expected result

MP3 at 96 kbps — about 60% smaller, same channel count as the source

DAW import prep

Input

MP3 128 kbps

Expected result

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. The whole file is held in browser memory and in ffmpeg.wasm's in-memory file system, so very large inputs or outputs (roughly 1–2 GB, such as hours of audio converted to WAV) can fail.

Explore the category

Audio Compressor Shrink MP3, WAV, M4A, OGG and FLAC files by re-encoding at a lower bitrate. Choose 32–320 kbps, pick MP3, OGG or M4A output, then download. 100% browser-side via ffmpeg.wasm. Audio Reverser Reverse any audio file so it plays backwards. Drop MP3, WAV, M4A, OGG or FLAC, choose output format, and download. ffmpeg.wasm areverse — no upload. Audio Merger Concatenate multiple MP3, WAV, M4A, OGG or FLAC files into one continuous audio track. Reorder with ↑ / ↓ buttons, set output format and bitrate, then download. Normalize Audio (LUFS) Normalize loudness of MP3, WAV, M4A, OGG and FLAC files using ffmpeg loudnorm (EBU R128). Hit Spotify, podcast or broadcast targets — runs locally via ffmpeg.wasm. Audio Speed & Pitch Changer Speed audio up or slow it down (0.25×–4×) while keeping pitch, shift pitch ±12 semitones at the same speed, or link both vinyl-style. MP3 Cutter & Audio Trimmer Trim or cut uploaded audio with draggable waveform handles, precise preview, undo, and exports to WAV, MP3, M4A, OGG, or FLAC.
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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. FLAC output here keeps the decoded samples exactly; WAV output is 16-bit PCM, exact only for sources of 16 bits or less.
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.
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