Audio Format Converter
Convert any audio file into MP3, WAV, FLAC, M4A, OGG, Opus or AAC while keeping full control of bitrate, sample rate, channel count and bit depth.
Convert with this tool
or choose a file from your device — it is processed in this tab and never uploaded
Accepted: MP3, WAV, M4A/AAC, FLAC, OGG Vorbis, Opus, WebM/Opus, MP4 audio
Files stay on your device. There is no upload, no queue and no watermark, so a conversion works even with the network switched off after the page has loaded.
Supported formats & processing
| Accepted input | MP3, WAV, M4A/AAC, FLAC, OGG Vorbis, Opus, WebM/Opus, MP4 audio |
|---|---|
| Produced output | MP3, WAV (16/24/32-bit), FLAC, M4A, OGG, Opus, AAC |
| Processing | Web Audio API + canvas + native JavaScript encoders |
| Upload required | No — the file is read locally and never transmitted |
How to use this tool
- Drop your audio file onto the upload areaDrag it from your desktop, or use Choose File to browse. One file at a time keeps the settings unambiguous.
- Pick the output formatMP3 for compatibility, WAV for editing, FLAC when you need a lossless archive, M4A for Apple devices, OGG or Opus for web playback.
- Set the bitrate, sample rate and channelsLeave the sample rate on Keep original unless a target device demands otherwise. Downsampling to 44.1 kHz is safe for almost everything.
- Press ConvertThe progress bar shows the decode, the offline render and then the encode. You can cancel at any point.
- Listen to the preview and downloadThe result card shows the measured peak and average levels before you commit to the download.
About this tool
This is the general-purpose converter behind every other audio page on Ajee Convert. It decodes your file with the browser's own audio engine, re-renders it offline at the sample rate and channel layout you choose, and then encodes the result. Because the decode and the encode both happen inside this tab, the file never leaves your device and there is no queue, no waiting room and no file-size cap.
MP3 and WAV output are produced natively in JavaScript. FLAC, M4A/AAC, OGG Vorbis, Opus and raw AAC are produced by the FFmpeg engine compiled to WebAssembly, which the page loads on first use. Every technical value — sample rate, channel count, bit depth, peak level, average level — is reported in the result card so you can see exactly what changed.
Features
- MP3, WAV, FLAC, M4A, OGG, Opus and AAC output from a single page
- Sample rates from 8 kHz to 96 kHz, or keep the source rate untouched
- Mono or stereo conversion with automatic channel mapping
- 16-bit, 24-bit and 32-bit float WAV export for audio editing
- Measured peak and RMS levels reported after every conversion
- Runs offline after first load — no file ever reaches a server
Frequently asked questions
Does converting to a higher bitrate improve quality?
No. Encoding a 128 kbps file at 320 kbps only adds bytes; the detail that was discarded during the original lossy encode is gone for good. A higher bitrate is useful when the source is lossless (WAV or FLAC), because then nothing has been thrown away yet.
Which format should I choose?
Choose MP3 for the widest compatibility, WAV when another program will edit the audio, FLAC to archive without losing anything, and M4A/OGG/Opus for smaller files at the same perceived quality.
Why did my sample rate change on its own?
MP3 cannot store every sample rate. It supports 8, 11.025, 12, 16, 22.05, 24, 32, 44.1 and 48 kHz, so an unusual source rate is snapped to the nearest supported value and the exact figure is reported in the result card.
Is there a file-size limit?
The practical limit is your device memory rather than a server quota. Files up to roughly 100 MB convert comfortably on a modern phone or laptop.