How to use Audio Converter
- 1Drop in an audio file — it is decoded and drawn as a waveform
- 2Pick an output format from the nine available
- 3Set the bitrate, bit depth or compression level the format offers
- 4Change the sample rate or force mono or stereo if you need to
- 5Export and download the converted file
What Audio Converter does
An audio file is a container wrapping a stream that some codec produced. Converting means fully decoding that stream back to raw samples and re-encoding them with a different codec — there is no way to translate compressed data directly from one format to another. That single fact explains most of what people find surprising about conversion, starting with why quality never improves.
Because decoding is exact, converting from a lossless source is safe: FLAC to WAV, or WAV to ALAC, reproduces the samples bit for bit. Converting from a lossy source is not. An MP3 has already discarded data permanently, and re-encoding those reconstructed samples discards more, so MP3 to AAC is always a second generation of loss even at a higher bitrate.
When to use it
- A file a device refuses to play
- Car stereos, older hardware players and some editing software accept a narrow set of formats. Converting to MP3 or WAV is usually the shortest route to a file that simply opens.
- Archiving without loss
- FLAC stores the exact samples of a WAV at roughly half the size. For master recordings or a CD rip you intend to keep, that is a straight saving with no trade-off at all.
- Speech at a small size
- Opus delivers the best quality per byte of any format here, and the gap is widest on voice. A podcast episode at 64 kbps Opus is comfortably listenable where the same bitrate in MP3 is not.
- Video and surround workflows
- AC3 is what DVD and broadcast chains expect, and editors often want ALAC or WAV rather than a lossy file. Converting first avoids the editor doing it silently at a setting you did not choose.
Audio Converter settings explained
- Output format
- Nine formats. MP3 and WAV are written by the browser itself and are effectively instant. The other seven — FLAC, M4A, AAC, ALAC, OGG, Opus and AC3 — go through an ffmpeg engine that downloads once, about 32 MB, then stays cached.
- Bitrate
- For the lossy formats: 64, 96, 128, 192, 256 or 320 kbps. 192 is a sensible default for music and 128 is fine for speech. AC3 snaps to the nearest value on its own fixed ladder, and Opus is capped at 512.
- Bit depth
- For WAV, FLAC and ALAC. 16-bit is CD quality and right for anything you are simply listening to; 24-bit preserves the extra headroom of a studio recording and makes the file half again as large.
- Compression level
- FLAC only, 0 to 12. Higher takes longer and produces a smaller file, but the audio is bit-identical at every setting — this trades encoding time for size, never quality.
- Sample rate
- 48000, 44100, 32000, 22050, 16000 or 8000 Hz, or keep the source rate. 44100 is CD standard, 48000 is the video standard, and dropping to 16000 is a large saving for voice recordings.
- Channels
- Keep the source layout, or force mono or stereo. Converting a stereo voice recording to mono roughly halves the file with no audible loss, since both channels usually carry the same signal.
Which format to choose
| Format | Lossless | Encoded by | Best for |
|---|---|---|---|
| MP3 | No | Browser (instant) | Maximum compatibility |
| WAV | Yes | Browser (instant) | Editing and mastering |
| FLAC | Yes | ffmpeg | Archiving at half the size of WAV |
| M4A / AAC | No | ffmpeg | Better than MP3 at the same bitrate |
| ALAC | Yes | ffmpeg | Lossless in the Apple ecosystem |
| OGG | No | ffmpeg | Games and open-format projects |
| Opus | No | ffmpeg | Speech and low bitrates |
| AC3 | No | ffmpeg | Video and surround delivery |
Limits and what to expect
- 150 MB per file, 20 tracks per project and 400 MB across one project.
- Files above 30 MB or an hour long still work but decode noticeably more slowly.
- The first export to any non-MP3, non-WAV format downloads a ~32 MB engine; after that it is cached and instant.
- Converting a lossy file to another lossy format always adds a second generation of loss, whatever bitrate you choose.
- Multi-channel surround input is mixed down rather than preserved as discrete channels.
Troubleshooting
- The first FLAC or Opus export took a long time
- That was the one-time engine download, roughly 32 MB. MP3 and WAV skip it because the browser can encode them natively. Once it is cached, later exports in any format start immediately.
- The converted file sounds worse than the original
- The source was already lossy. Decoding an MP3 and re-encoding it discards data a second time. Convert from a lossless source where you have one, and if you must go lossy to lossy, raise the bitrate to limit the damage.
- The file got much bigger
- You converted to a lossless format. WAV stores every sample uncompressed, so a 4 MB MP3 becomes roughly 40 MB. FLAC is about half of WAV. This is expected, and the audio is not improved by it.
- A long recording will not load
- Decoded audio is held uncompressed in memory — an hour of stereo at 44.1 kHz is around 600 MB of samples regardless of how small the compressed file is. Split long recordings before converting.
Your files and your privacy
The file is decoded with the browser's own audio engine and re-encoded either by the browser or by an ffmpeg build compiled to WebAssembly and running in the page. Both happen on your device. The audio is never uploaded, and the only network request the tool ever makes is for the engine itself, which contains no part of your file.
What to use next
FAQ
Does converting audio reduce quality?
It depends on the source and the target. Lossless to lossless — FLAC to WAV, WAV to ALAC — is bit-exact and loses nothing. Anything to a lossy format discards data, and lossy to lossy discards it twice, because the encoder is working from an already-degraded signal rather than the original recording.
Which audio format has the best quality?
For exactness, FLAC, ALAC and WAV are equal — all three reproduce the original samples, differing only in file size. For lossy formats at a given bitrate, Opus is generally best, then AAC, then MP3. MP3 survives mainly because everything can play it.
What bitrate should I use for MP3?
192 kbps is a good default for music and is transparent to most listeners. 320 is the maximum and worth it for critical listening or if the file will be re-encoded later. For speech, 96 or even 64 kbps is usually indistinguishable and much smaller.
Why is the first conversion slow?
Seven of the nine formats are encoded by an ffmpeg build that has to download once — about 32 MB. MP3 and WAV avoid it entirely because browsers can write those natively. After the first download the engine is cached and every later export starts straight away.
Can I convert several files at once?
Up to 20 tracks within 400 MB can be loaded into one project. Each is decoded to raw samples first, so the practical limit is memory rather than file size — a few long recordings will fill it faster than many short ones.
Is my audio uploaded to a server?
No. Decoding and encoding both run inside the browser tab on your own device. Nothing is transmitted or stored. The one network request is for the encoding engine, which is generic code and carries none of your data.