ToolPilot

Audio format converter

Convert MP3, WAV, FLAC, M4A, AAC, OGG or Opus to the format you need. Everything runs in your browser — no file is uploaded.

Drop your audio files here

or click to browse — MP3, WAV, FLAC, M4A, AAC, OGG, Opus, WebM

Everything about converting audio formats

Why convert an audio file from one format to another?

Every audio format strikes a different balance between quality, file size and compatibility. MP3 remains the most universally playable format, WAV keeps the signal intact for editing, AAC is native to the Apple ecosystem and much of the streaming world, and Opus delivers the best efficiency at low bitrates. Converting is simply choosing the trade-off that matches the final use.

Conversion also clears technical roadblocks: a podcast host that only accepts MP3, a digital audio workstation that expects WAV, a car stereo that ignores FLAC, or an upload form with a hard size limit.

This tool runs entirely client-side through the browser's Web Audio API. Your files never leave your device: nothing is uploaded, stored or analysed on a server. You can convert several files in a row and download them as a single ZIP archive.

Common use cases

Podcasters
Export episodes as MP3 at 128 or 192 kbps, the format expected by virtually every podcast host and directory.
Musicians and audio engineers
Turn a compressed file into 16-bit PCM WAV for import into a digital audio workstation, or shrink a demo to MP3 to share it quickly.
Web and game developers
Produce Opus files in an OGG or WebM container — noticeably lighter than MP3 at comparable perceived quality — to speed up page loads and reduce build size.
Video editors
Match your tracks to the 48 kHz sample rate used in video, or convert to M4A/AAC for direct import into editing software.
Teachers and trainers
Shrink a lecture recording by switching to mono at a moderate bitrate, so it fits the size limits of e-learning platforms.

How to convert an audio file

Drop one or more files into the drop zone, or click to select them. The browser decodes each file and shows its duration, sample rate and channel count.

Pick the output format, then adjust bitrate, sample rate and channels if needed. Leave the last two on "Automatic" to keep the source characteristics.

Start the conversion, preview the result in the built-in player, then download each file individually or all of them as a ZIP archive.

Frequently asked questions

Which formats are supported?
On input, the tool accepts any format your browser can decode: MP3, WAV, FLAC, M4A/AAC, OGG, Opus and WebM are supported by recent versions of Chrome, Firefox, Edge and Safari. On output, MP3 and WAV are always available; M4A (AAC), OGG (Opus) and WebM (Opus) only appear if your browser can encode them. The output list therefore adapts automatically to your browser.
Why do some formats take longer to convert?
MP3 and WAV are encoded in JavaScript, much faster than the actual length of the track. M4A, OGG and WebM go through the browser's MediaRecorder API, which encodes a live stream in real time: converting a four-minute track takes about four minutes. The tool shows an estimate before you start.
Does converting MP3 to WAV improve quality?
No. A lossy encoder such as MP3 permanently discards part of the signal. Going back to WAV produces a larger file that carries no more information than the original MP3. The conversion is still useful for editing, since a WAV can be opened and cut without another round of encoding.
Is there a size or file-count limit?
The tool imposes no limit. The real constraint is browser memory: each file is fully decoded into uncompressed samples, which is roughly 10 MB per minute of 44.1 kHz stereo. Recent desktop machines handle hour-long files without trouble; on mobile, work in smaller batches.
Are my files sent to a server?
No. Decoding and encoding happen entirely inside your browser, on your device. No audio file, excerpt or metadata is transmitted, stored or collected. The tool even keeps working offline once the page has loaded.
Are metadata (title, artist, cover art) preserved?
No. The conversion works on the decoded audio signal, not on the source file's tags: ID3 tags, cover art and chapters are not copied into the output file. Re-add them with a metadata editor if you need them.

Understanding digital audio formats

What is the difference between lossy and lossless compression?

A lossless codec such as FLAC, or an uncompressed format such as WAV, reproduces the original samples exactly. A lossy codec such as MP3, AAC or Opus uses psychoacoustic models to discard the least perceptible information, cutting file size by a factor of five to ten. The difference is often inaudible at high bitrates, but it is irreversible: the discarded data cannot be recovered.

What do bitrate and sample rate mean?

Bitrate, measured in kilobits per second, is the amount of data allocated to each second of audio in a compressed format: the higher it is, the more faithful the playback and the larger the file. Sample rate, measured in kilohertz, is how many times per second the signal is measured. 44.1 kHz is the compact disc rate, 48 kHz the rate used in video and professional broadcast. By the Nyquist-Shannon sampling theorem, 44.1 kHz can represent components up to about 22 kHz, above the upper limit of human hearing.

Should you avoid converting one lossy format into another?

Where possible, yes. Going from MP3 to AAC, or from Opus to MP3, means a second lossy encoding pass: the new codec re-analyses an already degraded signal and discards information of its own. This build-up, known as generation loss, becomes audible after several successive conversions. Start from the source closest to the original whenever you can — a WAV, a FLAC or the raw export from your recorder.

Which format should you choose for each use?

For universal distribution, MP3 at 192 kbps remains the safest choice: no player refuses it. For editing or working archives, WAV avoids any re-encoding. For the web and applications, Opus in an OGG or WebM container offers the best quality per kilobyte, especially for speech. For the Apple ecosystem and video, M4A/AAC is the expected native format.