Convert Audio

Turn an audio file into an MP3 or a WAV, choosing the bitrate, channels and sample rate — without the file leaving your machine.

Runs on your deviceAudioWorkingNo account needed

What this tool does

Convert Audio reads the file you give it down to raw samples, applies whatever channel and sample-rate change you asked for, and writes the result back out as MP3 or WAV. All of that happens in this browser tab: the LAME encoder that writes the MP3 is a script served from this site rather than a service at the far end of a connection, so nothing is uploaded, and the only real ceiling is how much audio your device can hold in memory at once. WAV input is read by this page's own RIFF reader, which takes 8, 16, 24 and 32-bit integer PCM as well as 32-bit float; every other format is handed to the browser, so the list of things that open here is really your browser's list.

That hand-off has a consequence worth knowing before you judge the result. The browser's decoder returns audio at the rate your sound output happens to be running at, commonly 48 kHz, so a 44.1 kHz MP3 arrives here as 48 kHz and the rate reported back to you is the one after that conversion rather than the number written in the file's header. WAV normally escapes it, because the RIFF reader runs first — though a WAV holding compressed data that reader will not open drops back to the browser decoder like anything else. If you want a specific rate on the way out, set it in the sample rate list instead of assuming the original one survived the trip.

The output menu holds two entries, MP3 and WAV, and that is not an oversight. No browser exposes an AAC, FLAC or OGG encoder, so those formats are accepted as input and simply not offered as output rather than being faked by writing something else under the name. Four other things go missing on the way through. ID3 tags are not carried across, so a converted MP3 arrives with a filename and no title, artist or album. The MP3 path is mono or stereo only, and it narrows a wider file by keeping its first two channels and discarding the rest, so a 5.1 mix loses its centre and surround content outright instead of being mixed down into the pair. MP3 is also written at a fixed set of sample rates, so an unusual one such as 96 kHz is moved to the nearest that fits. And rate changes use linear interpolation, which is fine for moving 48 kHz to 44.1 kHz and is not a mastering-grade resampler.

How to use it

  1. Choose one or more audio files. MP3, WAV, M4A, AAC, FLAC, OGG, Opus and WebA can all be selected. Each file is converted in turn and saved as its own output, using the settings you set once.
  2. Pick MP3 or WAV. MP3 for something to send on or keep on a phone. WAV when the file is heading into an editor and you would rather not stack another lossy generation on top.
  3. Set the bitrate or the bit depth. MP3 runs from 64 kbps for speech to 320 kbps, with 192 as the default. WAV offers 16-bit, 24-bit or 32-bit float. Whichever field does not apply stays hidden.
  4. Adjust the channels and sample rate if you need to. Mono averages the channels together, stereo duplicates a mono track, and the rate list runs from 48 kHz down to 16 kHz for speech. Keep as is leaves both alone, except that an MP3 cannot hold an unusual rate and is written at the nearest one it supports.

Processing & privacy

Runs on your device. Your file is processed directly in this browser.

Common use cases

Getting a wav to mp3 conversion for sharing

A minute of 44.1 kHz stereo WAV is roughly ten megabytes; the same minute at 192 kbps is under one and a half. That is often the difference between an attachment that sends and one that bounces.

Opening a FLAC file in software that refuses it

Older editors, car stereos and a few DAWs will not touch FLAC. WAV gives you plain PCM that almost anything will open, and MP3 gives you something a stubborn device will actually play.

Preparing a recording for transcription

Plenty of services want mono and cap the sample rate. Folding to mono at 16 kHz cuts a long interview down sharply while leaving speech entirely intelligible.

Shrinking a phone voice memo before emailing it

An M4A from a handset recorder becomes a 64 kbps mono MP3 in a single pass, usually around a tenth of the size and still clear enough to listen back to.

Questions

Is my audio uploaded anywhere?
No. Reading the file, resampling it and encoding the result all happen inside this tab, with the MP3 encoder loaded as an ordinary script from this site. The network is used once, to fetch that encoder and the audio code the first time a page load needs them; the audio itself is never sent anywhere.
Can I convert to FLAC, AAC or OGG?
No, and the menu says as much rather than listing them. Browsers ship decoders for those formats but no encoders, so offering FLAC would mean writing a WAV and labelling it wrongly. They are input formats here; MP3 and WAV are the outputs.
Why is the sample rate not the one in my file?
For anything except WAV, the browser decodes to whatever rate your audio output runs at — usually 48 kHz — before this page sees a single sample. Choosing 44.1 kHz in the list converts it back, though that is a second resample rather than an undo.
Do my ID3 tags survive?
No. The output is written from the audio alone, so title, artist, album and cover art are all dropped. Audio Metadata can put title, artist, album, year, genre and track back onto a converted MP3 afterwards — but not cover art, and not onto a WAV, whose tags it does not write.
Does converting an MP3 to WAV improve the quality?
No, it only makes the file bigger. The WAV holds exactly what the MP3 decoded to, including the absence of everything the original encoder threw away, and no format change can restore that.
What happens to a file with more than two channels?
MP3 is written here as mono or stereo only, and a wider file keeps its first two channels while the rest are dropped, not mixed into them — anything carried in a centre or surround channel is lost. WAV keeps every channel the decoder handed over, so choose WAV when a surround mix matters.
Why did my file come back with a different name?
When the output extension would match the input's — a WAV converted to WAV, say — the word converted is added to the name, so a result never quietly shares a filename with the file it came from.

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