Audio File Compressor

Make an audio file smaller by re-encoding it at a lower bitrate, in your browser. This changes the file size, not the dynamics, which is a different thing entirely.

The converter is ffmpeg compiled to WebAssembly. It downloads once (30 MB) when you pick your first file, then stays in your browser cache.

A higher bitrate keeps more detail and costs more bytes. 192 kbps is transparent for most music on ordinary gear, 128 is plenty for speech, and 320 is the practical ceiling for MP3.

Your file is processed by your own browser and never uploaded. The engine itself is a one-time 30 MB download the first time you use it, and after that it comes out of your browser cache.

How to use Audio Compressor

  1. 1Drop your audio files onto the box above, or click to pick them.
  2. 2Choose a target bitrate. 128 kbps is a good default, and 64 kbps is enough for speech.
  3. 3Check the before and after sizes in the list, then download. Adjust the bitrate and apply again if it is not small enough.

Two very different things called compression

If you work with music, "compression" means dynamic range compression: a processor that pulls loud peaks down and lifts quiet passages, so the whole thing sits at a more even level. That is a creative tool and it changes how the audio sounds.

This page does the other kind. An audio compressor in the file-size sense re-encodes the audio at a lower bitrate so it occupies fewer bytes. Loudness, dynamics and level are untouched. It is worth being explicit about, because searching for one and finding the other wastes everyone's afternoon.

Nearly everyone who wants to compress audio file by file is trying to clear a limit: an email attachment cap, a submission form, a podcast host's ceiling, a messaging app that refuses anything over a few megabytes. That is the job an audio compressor of this kind is for.

Where the size actually goes

Bitrate is the single lever, and it is close to linear: halve the bitrate and you roughly halve the file. Two other things matter almost as much, though, and neither is a quality setting.

Duration is the obvious one. Nothing reduces a file like removing the parts you do not need, and a trim costs no fidelity at all.

Channel count is the underrated one. A great deal of audio that is stored in stereo is functionally mono: a single voice, a phone recording, a screen capture, an interview into one microphone. Converting that to mono halves the data and loses nothing that was ever there.

What happens to lossless sources

A WAV or a FLAC has no bitrate to lower, because they store the samples rather than an approximation of them. To reduce audio file size in those cases means choosing a lossy codec, so the output becomes an MP3 at the bitrate you picked. That is the honest behaviour, and it is what you wanted if you came here to make a WAV manageable.

For lossy sources, the format is kept: an MP3 comes back as an MP3 and an M4A as an M4A, so nothing downstream has to change. The list above shows the before and after size for every file, so you can see whether the trade was worth it before you download.

Everything runs in your browser using ffmpeg compiled to WebAssembly, cached after the first file.

Frequently asked questions

Is this the same as a compressor in a DAW?
No, and the shared word causes real confusion. A compressor in music production reduces dynamic range, making quiet parts louder relative to loud ones. This page does data compression: it makes the file take fewer bytes. Nothing about the loudness or the dynamics is touched.
How small can it get?
That depends on the source and how much quality you will accept. Halving the bitrate roughly halves the size. Speech at 64 kbps mono is about 0.5 MB per minute, music at 128 kbps stereo about 1 MB per minute, and 320 kbps about 2.4 MB per minute.
Does it keep my format?
Where it can. An MP3 stays an MP3, an M4A stays an M4A, an OGG stays an OGG. Lossless sources are different: a WAV or FLAC has no bitrate to lower, so making one smaller means encoding it to MP3, and that is what happens.
Will re-compressing an MP3 make it noticeably worse?
It depends how far you go. Dropping a 320 kbps MP3 to 192 kbps is usually inaudible on normal equipment. Dropping it to 64 kbps is obvious immediately. The important part is that this is a second lossy pass, so the loss adds to whatever the first encode already did.
What if I need it smaller than the quality allows?
Then shorten it or convert it to mono. Both save more than a bitrate change and neither pretends to be free. A stereo file that is really one person talking loses nothing meaningful as mono, and halves in size.

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