How to Convert Any Audio File Between MP3, WAV, FLAC, and AAC

Three kinds of audio file, not two

People tend to sort audio into good quality and small, which hides the more useful distinction. There are really three categories, and knowing which one you are holding answers most questions before you ask them.

A WAV is uncompressed. It holds raw PCM samples, one number per channel per tick of the sample clock, behind a short RIFF header. Nothing is analysed and nothing is discarded, which is why the size is pure arithmetic: sample rate times bit depth times channels.

FLAC is compressed but lossless. It predicts where the waveform is going, stores the difference between its prediction and the real samples, and packs that efficiently. Decode a FLAC and you get back the identical samples that went in, bit for bit. It is meaningfully smaller than a WAV and nowhere near as small as a lossy file.

MP3, AAC and Vorbis (the codec inside most .ogg files) are lossy. They run a psychoacoustic model over the signal to decide which parts your ear will not notice, then simply do not store those parts. The discarded audio is gone. You can decode a lossy file back to PCM, but you are decoding what the encoder chose to keep, not what was originally recorded.

Every conversion goes through the same middle step

There is no direct route from MP3 to AAC. The MP3 is decoded back to PCM samples, and the AAC encoder then looks at those samples fresh and makes its own decisions about what to throw away. That is transcoding, and it costs you something every single time.

The damage is worse than it sounds, because the second encoder cannot tell which parts of the signal are music and which are the first encoder's artifacts. It spends bits preserving both. Do it often enough and you can hear the result: cymbals turn to hiss, sharp transients pick up a faint smear in front of them, and the top of the spectrum quietly goes missing.

One rule follows from this, and it covers most cases. Always convert from the best copy you have, not from the copy you converted last week. If you own the FLAC, make the AAC from the FLAC. Converting a lossy file up to WAV or FLAC recovers nothing either; it just stores the damage in a larger container.

What the converter on this site actually does

The tool hands your file to the browser's own decoder, which is why it opens more or less anything your browser can play: MP3, AAC, M4A, OGG, WAV, and FLAC where the browser supports it. What happens next depends on the format you pick.

• WAV is written straight from the decoded samples as 16 bit PCM.
• MP3 is encoded by LAME at 192 kbps, in a background worker so the page stays usable on a long file.
• AAC is encoded at 192 kbps by the audio encoder built into your browser, and saved as a raw .aac stream. A few apps expect AAC inside an .m4a container and will not open that.
• OGG uses the same built-in encoder to write Opus at 128 kbps. Opus is newer than the Vorbis codec most older .ogg files contain. Current browsers and players such as VLC handle it, but some older hardware players only understand Vorbis.
• AAC and OGG depend on that built-in encoder, so on a browser without one those two options are switched off rather than quietly producing something else.

What changes on the way through

A few details catch people out. The converter decodes everything at 44.1 kHz, or at 48 kHz for OGG because Opus only runs at that rate, so a 96 kHz source is resampled on the way through. Surround files are mixed down to stereo for MP3, AAC and OGG. And WAV output is written at 16 bits per sample, so a 24 bit master is truncated. None of that matters for listening. All of it matters if someone asked you for a specific format.

Tags do not survive at all. The decoder returns samples and nothing else, so artist, album, track number and embedded artwork are all dropped. Expect a bare file with a bare name.

Size depends on the format. WAV follows the arithmetic: CD rate stereo PCM runs to roughly 10 MB per minute, so a four minute song lands near 40 MB. MP3 and AAC at 192 kbps take about 1.4 MB per minute, and OGG at 128 kbps a little under 1 MB. Long files also have to be held in memory as 32 bit floats while they convert, and that is where a browser tab starts to struggle.

What this is genuinely good for

Getting an awkward file into a form every editor will open, without installing anything. Audio software all speaks PCM WAV. If a colleague has sent you an OGG that your editing software refuses, or you need a plain waveform to drop onto a video timeline, thirty seconds in a browser tab beats twenty minutes of hunting for an installer and clicking through it.

It is also the right shape for feeding a transcription tool or a noise reduction pass, both of which want uncompressed input and would only have to decode an MP3 again on arrival.

It is just as useful in the other direction: turning a WAV voice memo or rehearsal take into an MP3 small enough to send, without installing an encoder first.

When a desktop encoder is the better choice

FLAC, which this tool does not write, and anything that needs control the page does not offer: a variable bitrate, a particular sample rate, 24 bit output, or a whole folder in one go. ffmpeg covers all of that from the command line. Desktop players such as foobar2000 wrap the same encoders in a converter that also writes tags, which is what you want when the files are going into a music library rather than into an editor.

Nothing is uploaded

Your audio never reaches a server. The decode and the encode both happen in the page you have open, so unreleased recordings, client work and private voice memos stay on your machine. Open the Network tab in your browser's developer tools and convert something. The first MP3 conversion downloads the encoder library, and no request ever carries your file.