How to use the audio converter
- Drop an audio file onto the box or click to choose one.
- Pick Save as: WAV (16-bit, 24-bit or 32-bit float), M4A (AAC), WebM (Opus), OGG (Opus) or MP3.
- For compressed formats, choose a Bitrate from 64 to 320 kbps.
- Optionally set Channels (keep, mono or stereo) and Sample rate (keep, 44.1 kHz, 48 kHz, 22.05 kHz or 16 kHz for speech).
- For MP3 output or rare input formats, tick Allow the ffmpeg fallback. Then press Convert and download the file.
What it does and when to use it
Different devices and apps want different audio formats. A recorder saves huge WAV files, an Android phone records voice notes as Ogg Opus, a website upload form asks for MP3, and video editors often prefer WAV at 48 kHz. The converter moves a recording between these formats and can also turn stereo into mono or change the sample rate along the way.
The result shows the source format (for example “WAV pcm-s16 → M4A (AAC)”) and the source codec, sample rate and channels, so you know exactly what you started with.
To keep only part of a file, use the audio trimmer first. If the volume is uneven, the audio normalizer can even it out, and for the sound from a video file, the video to audio extractor can copy the soundtrack without re-encoding.
How it works
- Decode. WAV files are read by the site’s own code. Other formats are opened with the Mediabunny library (about 0.7 MB, loaded the first time) and decoded with your browser’s WebCodecs audio decoder. Formats the browser cannot decode, such as AIFF, WMA or AMR, go through ffmpeg.wasm if you allow it.
- Process. If you change the sample rate, the audio is resampled with a band-limited (windowed-sinc) filter. If you change channels, stereo is averaged to mono, or mono is copied to both sides.
- Encode. WAV is written by the site’s code; AAC and Opus by WebCodecs; MP3 by ffmpeg.wasm.
All of this runs in a Web Worker, so the page stays usable and you can cancel a long job.
Example
A 3-minute stereo WAV recorded at 44.1 kHz and 16-bit uses about 1,411 kbps (44,100 samples × 16 bits × 2 channels), which is roughly 31.8 MB. Saved as M4A at 192 kbps, the same 3 minutes is about 4.3 MB, roughly a seventh of the size. Choosing 96 kbps and Mono for a spoken recording would make it smaller still.
Limits and tips
- Lossy to lossy costs quality. Turning an MP3 into M4A or Opus re-compresses it, and the tool reminds you of this. Convert from the original whenever you can.
- Upsampling adds no detail. Changing 22.05 kHz audio to 48 kHz only makes the file bigger.
- Encoder support varies. If your browser lacks an AAC or Opus encoder, the tool says so; WAV output always works.
- Tags such as title and artist may not carry over to every output format.
Frequently asked questions
Why does MP3 need an extra download?
Which format should I choose?
Will converting MP3 to WAV improve the sound?
What bitrate should I use?
Are my files uploaded?
How do we know nothing is uploaded? Test it yourself on the privacy proof page.