OPUS to WAV Converter

Opus vs WAV — Which One Do You Actually Need?

Opus is a lossy codec built for streaming and voice: the IETF published it as RFC 6716 in September 2012, and it is what WhatsApp voice notes, Discord recordings, WebRTC calls and YouTube audio tracks are encoded with. WAV is not a codec at all — it is Microsoft's RIFF container holding raw PCM samples. The short answer: if the file only has to play, stay on Opus, because it is a tenth of the size for the same listening experience. Convert to WAV when a digital audio workstation, a transcription tool, or an old editor refuses to open Opus, and accept that WAV is bigger without being better.

Side-by-side Comparison

When to Stay on Opus

• The file only needs to be played, forwarded or stored — a voice note is already at its most efficient.
• You are keeping a long archive of calls or lectures and storage matters; a WAV of the same audio is roughly ten to fifty times larger.
• The playback target is a browser, a phone, Discord, or anything built this decade, all of which decode Opus natively.
• You are tempted to convert "to improve the quality." Decoding cannot rebuild what the Opus encoder discarded, so this buys nothing.

When to Convert to WAV

• Your DAW or video editor will not import .opus at all — Pro Tools, older Logic and Premiere versions, and plenty of Windows audio tools are in this camp.
• You are about to run noise reduction, EQ, normalisation or a stem export, and you want a working file that will not accumulate generation loss with each re-render.
• A transcription or speech-recognition pipeline specifies 16 kHz mono 16-bit PCM as its input contract.
• You need to hand the audio to a tool that reads only headered PCM — test rigs, embedded players, telephony gear.

How to Convert OPUS to WAV

1. Upload Your OPUS File: Drag and drop the .opus file onto the page or click "Add Files". Several files can be queued and converted with the same settings.
2. Pick the Audio Codec: The Audio Codec list holds the PCM formats WAV can carry — PCM 16-bit Little Endian is the default and the right answer for almost everyone; 24-bit and 32-bit are there for mastering chains, and A-law / mu-law for telephony targets.
3. Set Audio Sample Rate and Audio Channel: Leave Audio Sample Rate on "Original" to keep the 48 kHz an Opus stream always decodes at, or choose 44100 Hz for a CD-era workflow. Audio Channel offers Original, Mono or Stereo — "Original" keeps the layout the recording already has.
4. Convert and Download: Click Convert and save the WAV. No sign-up, no watermark.

Picking a PCM Format, and What It Costs

There is no bitrate or quality slider on this page, and that is correct rather than an omission: PCM is uncompressed, so the only things that change the file size are the sample rate, the bit depth and the channel count. Multiply them and you have the answer exactly — sample rate × channels × bytes per sample is the byte rate, with no encoder guesswork in between.

Two things worth knowing before you choose. First, an Opus stream always decodes at 48 kHz no matter what rate the original recording used — encode a 16 kHz source and the file still reports 48 kHz — so "Original" here means 48 kHz every time, and picking 44100 Hz is a genuine resample, not a relabelling. Second, moving from 16-bit to 24-bit does not recover anything the Opus encoder dropped; it only widens the headroom for processing you are about to do.

Frequently Asked Questions

Does converting Opus to WAV improve the audio quality?

No. Opus is lossy, so the detail its encoder discarded is gone permanently and decoding cannot rebuild it. What WAV gives you is a container that stops the bleeding: once the audio is uncompressed PCM, editing and re-exporting it does not stack another round of lossy encoding on top. That is the whole reason production workflows convert.

What sample rate and bit depth should I pick?

Leave Audio Sample Rate on "Original" and Audio Codec on PCM 16-bit Little Endian unless something downstream says otherwise. "Original" resolves to 48 kHz for every Opus file, which avoids a pointless resample. Choose 44100 Hz only if a CD-mastering chain or an older project session expects it, and 24-bit only if you are about to apply gain, EQ or dynamics and want the extra processing headroom.

Why is the WAV so much larger than the Opus file?

Because Opus stores compressed frames and WAV stores every sample. A voice note encoded around 24 kbit/s becomes a 768 kbit/s mono WAV at 48 kHz 16-bit — about thirty times the data rate — and the ratio does not depend on the source at all. WAV's size is fixed by sample rate, bit depth and channel count, so a heavily compressed original produces the largest apparent jump.

Will the WAV keep stereo, or does it collapse to mono?

It keeps whatever the Opus file holds when Audio Channel is set to "Original". Opus supports mono, stereo and multichannel, and most messaging apps record voice notes in mono, so a mono WAV is the expected result there rather than a fault. Choose Mono explicitly only when a downstream tool insists on one channel; choosing Stereo on a mono source just writes the same audio twice.

Should I choose A-law or mu-law instead of PCM?

Only for telephony. Both are 8-bit companded G.711 formats — half the size of 16-bit PCM but lossy, and band-limited in practice to a phone channel. They exist on this page for people feeding a PBX, an IVR or an embedded voice system that will not accept linear PCM. For editing, transcription or anything you plan to listen to, use one of the linear PCM options.

Is there a length limit on the WAV output?

There is a structural one. RIFF, the container WAV is built on, records chunk lengths in 32-bit fields, which puts the ceiling near 4 GB — roughly six hours of 48 kHz 16-bit stereo, or twelve hours in mono. Plenty of editors get uncomfortable well before that, so for a very long recording it is usually better to trim the section you need first with the Audio Cutter than to produce one enormous file.

Can I get back to Opus, or to something smaller, afterwards?

Yes, but expect a second round of loss. Re-encoding a WAV that came from Opus means the audio has been through two lossy passes, which is exactly the generation loss the WAV step was meant to avoid mid-workflow. Convert to a compressed format once at the end. If you want a small distributable file, Opus to MP3 straight from the original is the cleaner route; if you want a compact but lossless master of the finished WAV, WAV to FLAC typically halves it with no quality change at all.

Why does my editor still show 48 kHz when the original recording was 16 kHz?

Because that is what Opus does. The codec works internally at 8, 12, 16, 24 or 48 kHz depending on the content, but decoders in general use — including this one — hand back 48 kHz, so every Opus file presents as a 48 kHz stream regardless of what went into it. The extra samples carry no extra information; they are the decoder's output rate, not evidence of higher fidelity.

How are my files handled during conversion?

Your file is uploaded over an encrypted connection, decoded on our servers, and deleted automatically a few hours after conversion — no sign-up, no watermark, never shared or made public. Because WAV output is many times the size of the Opus input, the practical constraint is upload and download time rather than anything about the decode itself.