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Supports: WAV
WAV is the lossless PCM container Windows shipped with in 1991 — perfect for editing masters, but every minute of stereo 44.1 kHz / 16-bit audio is about 10 MB, and 24-bit / 96 kHz tracks balloon to 33 MB per minute. Opus, the IETF-standardised codec published as RFC 6716 in September 2012, was engineered to replace MP3, AAC, Vorbis, and Speex in one container. It hybridises Skype's SILK (speech) and Xiph's CELT (music) and runs end-to-end from 6 kbps voice to 510 kbps stereo. Re-encoding a WAV master to Opus typically cuts size 10x to 20x with no audible difference at 96 kbps and up.
| Property | WAV | Opus |
|---|---|---|
| Compression | None (PCM, lossless) | Lossy (SILK + CELT hybrid) |
| Standardised | Microsoft / IBM, 1991 (RIFF) | IETF RFC 6716, Sept 2012 |
| Typical bitrate (stereo music) | 1,411 kbps (CD) to 4,608 kbps (24/96) | 96-256 kbps (transparent ~128 kbps) |
| Bitrate range | Fixed by sample rate / depth | 6 to 510 kbps |
| Sample rates | 8 kHz to 384 kHz+ | 8 / 12 / 16 / 24 / 48 kHz |
| Max channels | 65,535 (RIFF) | 255 |
| Algorithmic latency | ~0 ms | 26.5 ms default, 2.5 ms minimum |
| Royalty | Public domain | Royalty-free (IETF) |
| Browser playback | All (PCM is universal) | Chrome 33+, Firefox 15+, Edge 14+, Safari 14.1+ (iOS 17+) |
| Best use | Editing masters, archive | Streaming, voice, downloads, podcasts |
| Bitrate | Mode | Best for | Quality |
|---|---|---|---|
| 6-12 kbps | SILK (voice) | IVR, transcription audio | Intelligible speech |
| 16-32 kbps | SILK (voice) | Audiobooks, telephony | Good speech, no music |
| 48-64 kbps | Hybrid | Podcasts (mono), voice notes | Excellent speech |
| 96 kbps | CELT | Stereo podcasts, near-transparent | Very good music |
| 128 kbps | CELT | Streaming music | Transparent for most |
| 192-256 kbps | CELT | Archival lossy, high-fidelity | Indistinguishable from source |
| 320-510 kbps | CELT | Mastering proofs, surround | Reference quality |
For most listeners, yes — at 128 kbps stereo, Opus is considered "transparent" in blind ABX tests against the original PCM source, and at 96 kbps it beats MP3 at 128 kbps. If you're a mastering engineer doing critical listening on studio monitors, keep the WAV; for anything you'd upload or stream, Opus 128-192 kbps is sonically indistinguishable in normal listening.
Mono voice content at 48 kbps Opus is excellent quality and roughly 22 MB per hour. Stereo with intro music or two-host bouncing audio benefits from 64-80 kbps. Don't go below 32 kbps on stereo — you'll hear pre-echo on sibilants. The Quality Preset "Very High (Recommended)" maps to about 128 kbps which is overkill for speech but bulletproof for music interludes.
Opus is the codec; .opus and .ogg are both Ogg containers that carry Opus data. xConvert outputs .opus by default per RFC 5334's recommendation that Opus-only streams use the .opus extension. If a service requires .ogg, try WAV to OGG instead — same Ogg container, Vorbis or Opus codec inside.
Yes since Safari 14.1 (April 2021) on macOS and iOS, including iOS 17+ and current macOS Sonoma/Sequoia. Older Safari versions silently fail playback in <audio> tags. If you need universal playback on iOS 13-14, fall back to WAV to MP3 or WAV to AAC. Chrome, Firefox, and Edge have full Opus support back to versions released in 2014.
You can decode the Opus back to WAV with Opus to WAV, but it will be the lossy Opus audio re-expanded to PCM — not the original samples. Opus is a lossy codec by design (the SILK and CELT layers discard inaudible information), so always archive the source WAV separately if you need bit-exact recovery later.
Free anonymous users get a generous per-file limit and unlimited file count per session. Multi-gigabyte 24-bit / 96 kHz session bounces convert successfully; processing time scales roughly linearly with duration. If you have a 4 GB+ RIFF/RF64 WAV, split it first with the Audio Cutter or upgrade for higher caps.
For streaming and playback, Variable Bitrate gives the best size-to-quality ratio — Opus VBR allocates bits to complex passages and skips silence. Constant Bitrate is required only for certain RTP / WebRTC scenarios where the stream needs a predictable rate. Quality Preset is a friendly wrapper that picks sensible VBR settings; pick "Very High (Recommended)" for music, "High" or "Medium" for voice.
If you leave Audio Channel and Audio Sample Rate on "ORIGINAL", xConvert preserves your WAV's source values — but Opus internally normalises to one of 8/12/16/24/48 kHz, so a 44.1 kHz WAV is resampled to 48 kHz inside the encoder. To downmix 5.1 to stereo or force mono for a voice note, set Audio Channel explicitly. Need to compress an existing Opus file further? Use Compress WAV on the source before converting, or re-encode with a lower bitrate.