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WAV to MP3 Converter

A minute of CD-quality WAV weighs about 10 MB; at the default 192 kbit/s the MP3 weighs about 1.4 MB. Encoded by ffmpeg inside your own browser tab.

The recording is decoded and encoded by WebAssembly in this tab; only the audio engine is ever downloaded.

Upload Audio Files

Drag and drop your audio files here or click to browse. Supports MP3, WAV, OGG, AAC, FLAC, and M4A formats.

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What the MP3 costs you

WAV is a thin container around raw PCM: nothing is modelled, nothing is predicted, every sample is written down as it was captured. At the usual 16-bit, 44.1 kHz, stereo settings that works out to 1,411 kbit/s, or roughly 10 MB per minute. MP3 keeps a psychoacoustic model of what an ear is likely to notice and discards the rest, so at this page's default of 192 kbit/s a minute costs about 1.4 MB — one seventh of the original.

What disappears is mostly sound that was masked by something louder nearby, plus the top of the spectrum. Encoding a white-noise file with exactly the arguments this page uses, everything above roughly 19 kHz was gone at 192 kbit/s, and everything above roughly 17 kHz at 128. Through normal speakers, on a finished mix, that is hard to hear. On a sparse recording, a solo cymbal or anything that will be encoded again afterwards, it is not.

Exactly how this page encodes

The audio is handled by ffmpeg compiled to WebAssembly, running libmp3lame at a constant bitrate: the rate you set is passed straight through as -b:a, and the finished file lands within about one percent of it once frame headers are counted. The slider under Settings runs from 128 to 320 kbit/s in steps of 32 and starts at 192, so a minute of audio costs about 1.0 MB at 128, 1.4 MB at 192 and 2.4 MB at 320 whatever the material is.

Pick 128 for speech and podcasts, 192 for general music, and 320 when the source is a lossless master. Above roughly 192 the audible gain gets small while the file keeps growing in a straight line — 320 is two thirds larger than 192. Two other defaults matter: sample rate sits at 44.1 kHz, so a 48 kHz session file is resampled unless you change it, and Channels sits at Stereo, which on a mono recording spends half the budget storing the same signal twice.

When to keep the WAV

Keep the WAV whenever the recording is still in production. Anything that will be edited, time-stretched, pitch-shifted, mastered, loaded into a sampler or exported again belongs in PCM, because every MP3 round trip re-quantises the signal and the damage accumulates silently. Archive masters, stem exports and sample libraries are all WAV cases, and storage is cheaper than a re-recording session.

MP3 also breaks seamless loops. The format works in frames of 1,152 samples and the encoder adds its own lead-in delay, so a precisely trimmed one-bar loop gains a sliver of silence at each end unless the player reads LAME's gapless header — and plenty of players do not. For loops, game audio and anything triggered from a pad, export WAV or OGG instead.

Where MP3 still wins

Delivery, and essentially nothing else. Podcast hosts, radio ingest, voicemail systems, car stereos, DJ controllers, cheap players and every messaging app accept MP3 without argument, which is more than can be said for WAV. A 60-minute stereo interview drops from around 600 MB to roughly 86 MB at 192 kbit/s, or 43 MB at 128 — often the difference between an upload that completes and one that times out.

The MP3 that comes out carries no cover art: the encoder is told to leave picture streams behind, so nothing is embedded even if something was attached to the source. Text tags travel as far as the WAV carried them, which is usually not at all — WAV has no widely honoured place to keep them. Add title, artist and artwork in your podcast host or a tagging app afterwards; it takes seconds and survives every later copy of the file.

WAV and MP3 side by side

The same recording before and after a psychoacoustic model gets to it.

 WAVMP3
CompressionNone — linear PCM samples stored as capturedLossy; constant bitrate at whichever rate you select
Bitrate1,411 kbit/s at 16-bit, 44.1 kHz, stereo128 to 320 kbit/s in steps of 32, starting at 192
Size per minuteAbout 10 MBAbout 1.0 MB at 128, 1.4 MB at 192, 2.4 MB at 320
Frequency rangeEverything up to half the sample rateCut above about 19 kHz at 192 kbit/s, about 17 kHz at 128
Tags and artworkNo widely honoured tagging; most files carry noneID3 text tags only — cover art is never written
Editing and loopingSample-accurate, safe to re-edit indefinitelyFrame-based with encoder delay; loops gain silence
SupportEvery DAW and editor; refused by many upload formsAccepted everywhere, from podcast hosts to car stereos

How to convert WAV to MP3

Three steps, and the audio never leaves your device

  1. Add your WAV files

    Up to 10 recordings of 100 MB each — roughly ten minutes of CD-quality stereo per file. Session exports and Audacity renders are fine.

  2. Pick a bitrate

    The slider starts at 192 kbit/s; drop it to 128 for speech or raise it to 320 for a master you care about. Sample rate and channels sit in the same panel.

  3. Convert and download

    The first run loads the audio engine once, then every file is encoded in the tab and saved straight to your downloads folder.

FAQ

WAV to MP3 questions

Bitrate, limits and what the encoder actually does

Exactly the rate shown on the slider, as a constant-bitrate file. The default is 192 kbit/s; measured on a test encode, the finished file sat within about one percent of the nominal rate once frame headers were counted. Nothing varies with the material the way a VBR encoder would.
Yes. Open Settings and move the slider: 128 to 320 kbit/s in steps of 32. Use 128 for speech and podcasts, 192 for general music, and 320 when the source is a lossless master that will be listened to closely. Broadcasters who insist on constant-bitrate 128 are catered for exactly.
The cap is 100 MB per file, which is about ten minutes of 16-bit, 44.1 kHz stereo or twenty minutes in mono. The real constraint is memory: ffmpeg holds the file inside the browser tab, so a laptop copes with the full limit comfortably while an older phone may not.
Slightly, and by design. The encoder discards sound it expects to be masked by louder material, and it also cuts the top of the spectrum: measured with these settings, nothing survives above about 19 kHz at 192 kbit/s or about 17 kHz at 128. For delivery — podcasts, previews, players — that is inaudible to most listeners on most systems. For anything that will be edited or encoded again, work from the WAV.
MP3 has no bit depth of its own, so the samples are converted to what the encoder expects, and the sample rate is resampled to whatever Settings says — 44.1 kHz by default, with 48 kHz available. A high-resolution master therefore comes out as an ordinary 44.1 kHz MP3 unless you change that first.
No. The only network request the conversion makes is the one-time download of the WebAssembly audio engine, which you can watch in the DevTools Network panel. Your recording is written into a virtual filesystem inside the tab and never crosses the network.

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