Audio Converter

A free audio converter for every common format

This audio converter changes music, podcasts and recordings between the seven formats people actually use: MP3, WAV, FLAC, M4A, AAC, Ogg Vorbis and Opus. Drop in one file or a whole folder, pick the format you need, and each file is converted on your own device — nothing is uploaded, there is no sign-up and no watermark. Titles, artists, album names and cover art travel with the audio, translated into whatever tag system the new format uses.

Under the hood the page decodes your audio with the codecs built into your browser and encodes it with the open-source encoders desktop software relies on: LAME for MP3, libFLAC for FLAC, libvorbis for Ogg Vorbis, and the browser’s own AAC and Opus encoders. Every file shows its format, length, sample rate, channels and bitrate first, with a line saying exactly what will happen to it.

How to use the Audio Converter

1. Drop your audio files on the upload area, or click it to choose them. MP3, WAV, FLAC, M4A, AAC, OGG, Opus and WebM files can be mixed freely.
2. Under Convert to, pick the format you want. The line under each file switches between Encode (decoded and compressed again) and Copy (moved across unchanged).
3. Set the quality — Bitrate, Vorbis quality or Bit depth, depending on the format. Change Sample rate or Channels only if a device asks for it.
4. Press Convert. A progress bar tracks each file, and Cancel stops at any time.
5. Play the original and the converted file side by side, then download each one or press Download all (ZIP).

Audio Converter features

Which audio format should you choose?

Converting between two lossy formats always costs a little quality, because the second encoder works from an already-compressed signal — so convert from the best source you have, at a bitrate at least as high as the original’s.

Copy or encode: what happens to each file

An audio file is a container — MP3, MP4/M4A, Ogg, WebM, WAV or FLAC — around a stream compressed with a codec. When the container you choose can hold the stream you already have, the encoded packets are copied into a new file with fresh headers and tags. Nothing is decoded, so quality and size stay the same: an iPhone .m4a becomes a raw .aac in about a second.

Everything else is decoded, resampled or mixed down if you asked for that, and encoded again — every change of codec, bitrate, sample rate or channel count. Untick the copy option to re-encode a copyable file anyway.

Settings that matter

• Bitrate is the main size dial for MP3, AAC and Opus. Size per minute = bitrate × 60 ÷ 8, so 192 kbps is 1.44 MB a minute.
• Sample rate: 44.1 kHz is the CD standard, 48 kHz the video one; changes use a windowed-sinc filter. 22.05 or 16 kHz suit speech only.
• Channels: mono halves the data for speech and single-microphone recordings. Mixing stereo music down to mono can make wide-panned parts quieter.
• Bit depth applies to WAV and FLAC. 16-bit (CD quality) is plenty for listening; 24-bit or 32-bit float keep headroom for editing.
• Dedicated pages go deeper for single jobs: MP3 to WAV, FLAC to MP3, M4A to MP3 and MP3 to Opus.

Libraries, licences and limits

Files are read and written by Mediabunny (MPL-2.0); MP3 is encoded by LAME (LGPL), FLAC by libFLAC (BSD), Ogg Vorbis by libvorbis via wasm-media-encoders (MIT/BSD), and AAC and Opus by your browser, with libavcodec’s AAC encoder (LGPL) for bitrates the browser refuses. All load unmodified from a CDN when first needed. Audio your browser can’t decode — Dolby, DTS, DRM-protected purchases and, outside Safari, Apple Lossless — is reported, not converted badly. To shorten a song first, use the Audio Cutter; for a video’s soundtrack, Video to MP3.