OPUS to AIFF Converter

Convert OPUS to AIF Online

Turn a .opus file — usually a voice note from a messaging app, a browser recording, or a web-audio export — into a .aif that Mac software and audio editors open without arguing. The Opus stream is fully decoded to uncompressed PCM and written into the AIFF chunk structure, which is what samplers, DAWs and QuickTime expect. There are no bitrate or quality controls on this page and that is not an omission: uncompressed PCM has no bitrate to choose. What you control is sample rate, channels and which PCM flavour goes in the file — and those three, multiplied together, are the entire file size.

How to Convert OPUS to AIF

1. Upload Your OPUS File: Drag your .opus onto the page or click "Add Files" to browse; queue several recordings and they all run on the same settings. Files are uploaded over an encrypted connection, processed on our servers, and deleted automatically a few hours after conversion — no sign-up, no watermark, never shared or made public.
2. Leave Audio Codec on PCM 16-bit Big Endian: That is the AIFF default and the only one of the four choices that writes a textbook FORM…AIFF file. The other three — PCM 16-bit Little Endian, PCM A-law, PCM mu-law — all switch the file to the AIFF-C form even though the name still ends .aif.
3. Set Audio Sample Rate and Audio Channel: Leave both on Original to copy the source layout, or pick a rate from 8000, 12000, 16000, 24000, 44100 or 48000 Hz to match your session, and force Mono or Stereo if the project expects a specific channel count.
4. Trim and Convert: Trim stays out of the way unless you switch it on — and if you do, check the Duration box, which arrives pre-filled with 10 seconds. Then click Convert and download the .aif on its own or as a ZIP.

What Each Audio Codec Choice Actually Writes

sowt is "twos" spelled backwards — Apple's little-endian PCM variant, which macOS itself often writes. It carries identical audio to the big-endian default, but it is technically an AIFF-C file, so a strict reader that accepts only plain AIFF can refuse it. There is no 24-bit or 32-bit option on this page; everything here is 16-bit or 8-bit companded.

Working Out the File Size Before You Convert

Uncompressed PCM has no compression to guess at, so the size is arithmetic: bytes = sample rate × channels × bytes per sample × seconds, plus a header of a few dozen bytes.

Because there is no bitrate control, those two levers — Audio Sample Rate and Audio Channel — are the only ones that change the size. Halving the rate halves the file; collapsing stereo to Mono halves it again.

The jump from the source is steep and there is no way around it. A .opus voice note carries maybe 16–32 kbit/s, or a few kilobytes per second; the AIF that comes out of it carries 768 kbit/s at 48 kHz mono, or 1,536 kbit/s in stereo. Expect the file to grow by roughly one to two orders of magnitude. That is the cost of an editor-ready, uncompressed working copy — not a setting anyone can tune away.

Frequently Asked Questions

Why does the converted .aif come out 48 kHz when my recording was not?

Because every Opus stream decodes at 48 kHz. The codec is defined in RFC 6716 around internal bandwidths of 8, 12, 16, 24 and 48 kHz, and the decoder resamples its output — so a voice note captured at 16 kHz still presents itself as a 48 kHz stream to anything reading it. Leaving Audio Sample Rate on Original therefore gives you a 48 kHz AIF regardless of the capture rate. If your session runs at 44.1 kHz, select 44100 Hz explicitly so the file drops in without a resample prompt. Unlike the reverse direction, PCM has no restricted rate list, so every value in the dropdown is written verbatim.

Will the AIF sound better than the Opus it came from?

No, and it is worth being blunt about it. Opus is a lossy codec: whatever it discarded at record time is gone, and decoding to uncompressed PCM reproduces the surviving signal faithfully without rebuilding anything. What you gain is a working copy that will not degrade further — every subsequent edit and save is lossless, whereas re-encoding lossy audio repeatedly compounds artefacts. The trade is size: you are paying one to two orders of magnitude more bytes for zero fidelity gain, purely to buy compatibility and a clean editing baseline.

Why are there no bitrate or quality settings on this page?

Because the output is uncompressed. A bitrate control exists to tell a lossy encoder how much data it may spend; linear PCM stores every sample in full, so its data rate is fixed by sample rate × channels × bit depth and there is nothing left to negotiate. That is why the compression section is absent here entirely rather than present-but-greyed-out. If you want to choose a bitrate, you want a lossy target instead — Opus to MP3 gives you that.

Is .aif the same thing as .aiff?

Yes. .aif is the three-letter spelling forced by the old DOS/Windows 8.3 filename limit and .aiff is the four-letter one; the chunk structure, the byte order and the samples are identical. macOS tends to write the longer form, which is why plenty of Mac-origin files arrive as .aiff. If a tool insists on the four-letter name, use Opus to AIFF instead — it is the same conversion with the other extension. The genuinely different member of the family is .aifc; see Opus to AIFC.

Should I ever pick A-law or mu-law?

Only for telephony or legacy speech systems that ask for them by name. Both are 8-bit companded codings: they map a wider dynamic range onto a single byte per sample, so they halve the file at the cost of real quality — they are lossy, despite living in a container people associate with lossless audio. For music, sampling, or anything you plan to edit, stay on PCM 16-bit Big Endian. And note that choosing either one also flips the file to the AIFF-C form with an alaw or ulaw compression tag, so a strict AIFF-only reader may not open it.

Does my AIF file actually say AIFF inside, or AIFC?

It depends entirely on the codec you pick, not on the extension. We inspected the emitted headers directly: with the default PCM 16-bit Big Endian, the file opens FORM, then a size, then the literal AIFF form type with no FVER chunk — a textbook plain AIFF. Switch to PCM 16-bit Little Endian, A-law or mu-law and the same four bytes become AIFC, an FVER chunk appears, and the COMM chunk gains a four-character compression tag (sowt, alaw, ulaw). Most software does not care; strict validators do.

My Trim came back as a 10-second clip. What happened?

The Trim control's Duration field arrives pre-filled with 10, and switching Trim on without changing it exports ten seconds from your start point rather than the rest of the file. Either set Duration to cover the span you want, or leave Trim alone entirely to convert the whole recording. If you would rather cut the audio visually — or trim the resulting AIF afterwards — audio cutter accepts .opus, .aif, .aiff and .aifc alike.

Should I use AIF at all, or WAV?

Both are uncompressed PCM and sound identical; the difference is heritage and byte order. AIFF is big-endian and the traditional Apple interchange format, so it slots naturally into Logic Pro, GarageBand, Final Cut Pro and hardware samplers. WAV is little-endian and the de facto standard everywhere else, and it is the safer choice if your collaborators are on Windows or your tool chain is cross-platform — Opus to WAV produces that. Going the other way later, AIF to Opus compresses an AIF back down for delivery.