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Supports: OPUS
Everyone converting a .opus file to an uncompressed format is solving the same problem: the editor, sampler or DAW in front of them will not import Opus, and it will import the AIFF family. The real question is which uncompressed target to land in. Short answer: Apple-centric session → AIFF; cross-platform or Windows collaborators → WAV; you want lossless but not enormous → FLAC. All three carry identical audio out of the same Opus source, so the choice is about who has to open the file, not about quality.
| Property | AIFF | WAV | FLAC |
|---|---|---|---|
| Origin | Apple, 1988, from the EA IFF 85 chunk model | Microsoft and IBM, 1991, RIFF chunk model | Josh Coalson, first release 2001; now maintained by Xiph.Org |
| Compression | None — linear PCM | None — linear PCM | Lossless compression |
| Byte order | Big-endian | Little-endian | n/a — coded stream |
| Typical size, 1 min stereo 44.1 kHz | 10.6 MB | 10.6 MB | Roughly 40–70% of that, content-dependent |
| Chunk structure | FORM / COMM / SSND |
RIFF / fmt / data |
STREAMINFO + frames |
| Container size ceiling | ~4 GiB (32-bit chunk sizes) | ~4 GiB (32-bit chunk sizes) | Effectively none |
| Can carry sampler loop points | Yes, via INST and MARK chunks |
Via the smpl chunk |
Via tags, less standardised |
| Native on macOS | Yes — QuickTime, Music, Logic, Final Cut | Yes | Yes in modern versions |
| Native on Windows | Widely supported, not the default | Yes, the native format | Widely supported |
| Hardware samplers | Long-standing first-class support | Long-standing first-class support | Patchy on older units |
| What this page writes | 16-bit big-endian PCM | — | — |
INST and MARK chunks exist for exactly that, though a freshly converted file arrives without them and you set loops in the sampler..opus onto the page or click "Add Files"; queue several recordings to 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.FORM…AIFF file. PCM 16-bit Little Endian, PCM A-law and PCM mu-law all switch the output to the AIFF-C form instead.10 seconds. Click Convert and download the .aiff individually or as a ZIP.There is no bitrate or quality control on this page, because linear PCM has no compression to negotiate — the data rate is sample rate × channels × bytes per sample, full stop.
| Settings | Data rate | One minute | Ten minutes |
|---|---|---|---|
| 48 kHz, stereo, 16-bit | 1,536 kbit/s | 11.5 MB | 115 MB |
| 48 kHz, mono, 16-bit | 768 kbit/s | 5.76 MB | 57.6 MB |
| 44.1 kHz, stereo, 16-bit | 1,411 kbit/s | 10.6 MB | 106 MB |
| 44.1 kHz, mono, 16-bit | 706 kbit/s | 5.3 MB | 53 MB |
| 16 kHz, mono, 16-bit | 256 kbit/s | 1.9 MB | 19 MB |
For audio quality it is a coin flip — both store identical linear PCM samples and neither adds or removes anything. The differences are ecosystem and byte order: AIFF is big-endian and grew up on the Mac, WAV is little-endian and grew up on Windows. Pick AIFF when the destination is a Logic, GarageBand or Final Cut session or a sampler that names AIFF in its documentation; pick WAV when the chain is mixed or Windows-first. One genuine format difference worth knowing: AIFF's INST and MARK chunks are the classic way to store loop points and a base note for samplers, whereas WAV uses the smpl chunk for the same job.
No. Opus is lossy by design — RFC 6716 defines it as a hybrid of a linear-prediction layer for speech and an MDCT layer for music, spanning 6 to 510 kbit/s — and what it discarded at record time is not recoverable. Decoding to uncompressed PCM reproduces the surviving signal faithfully; it does not reconstruct anything. The real benefit is forward-looking: from here on, every edit and save is lossless, so you stop compounding artefacts the way repeated lossy re-encodes do. You pay for that with size, not with quality.
Because Opus is doing its job and AIFF is not trying to. A voice note might be coded at 16–32 kbit/s; the same audio as 16-bit PCM is 768 kbit/s in mono at 48 kHz and 1,536 kbit/s in stereo. That ratio is the whole point of a lossy codec, and no setting on this page reduces it — there is no bitrate control because uncompressed audio has no bitrate to set. The only levers are Audio Sample Rate and Audio Channel: halve the rate or collapse to Mono and you halve the file each time. For lossless audio at a fraction of the size, use FLAC instead.
Because every Opus stream decodes at 48 kHz. Opus codes internally at 8, 12, 16, 24 or 48 kHz bandwidths and its decoder resamples the output, so anything reading the stream sees 48000 Hz regardless of what the microphone captured. Leaving Audio Sample Rate on Original therefore always produces a 48 kHz AIFF here. Set 44100 Hz explicitly if your session runs at that rate and you want the file to import without a resample prompt — PCM has no restricted rate list, so whatever you pick is written verbatim.
A minimal, standards-clean file. Inspecting the emitted bytes shows a FORM header declaring the AIFF form type, an 18-byte COMM chunk carrying channel count, frame count, 16-bit sample size and the sample rate as an 80-bit extended float, and then the SSND chunk holding the raw big-endian samples — 54 bytes of overhead in total. There is no FVER chunk, because that belongs to AIFF-C, and no INST, MARK or metadata chunks, because the converter writes only what the audio needs. Loop points and markers are things you add in your editor afterwards.
On macOS, effectively everything: QuickTime, the Music app, Logic Pro, GarageBand and Final Cut Pro all treat AIFF as a native uncompressed format. On Windows and Linux, most serious audio editors and DAWs import AIFF without a plugin, and hardware samplers have supported it for decades. What is patchy is the consumer end — many phones, car stereos, web players and lightweight media apps do not handle the AIFF family at all. That gap is precisely why converting away from .opus is worth doing for editing but not for playback.
Rarely, and knowing what changes. PCM 16-bit Little Endian writes Apple's sowt variant — identical samples in reversed byte order — but doing so flips the file to the AIFF-C form with an FVER chunk, so a strict AIFF-only reader may refuse it even though the name still ends .aiff. A-law and mu-law halve the file by packing each sample into one byte, but they are lossy companded codings meant for telephony, not music, and they also produce AIFF-C. If you specifically want the AIFF-C form, Opus to AIFC covers that case properly; if you want the short spelling, Opus to AIF is the same conversion.
Yes. Switch Trim on, set Start time, and set Duration — both fields take plain seconds or HH:MM:SS.sss. The catch to watch is that Duration arrives pre-filled with 10, so leaving it untouched exports a ten-second excerpt rather than everything from your start point onward. Set it to the length you actually want. If you would rather cut visually, or trim the AIFF after conversion, audio cutter accepts .opus and the whole AIFF family. To go back the other way and shrink a finished AIFF for delivery, use AIFF to Opus.