The Opus file (.opus)

Opus is a lossy audio codec, standardised by the IETF in 2012 as RFC 6716. It was designed to work for everything from a phone call at a few kilobits per second to high-quality stereo music, with very low delay, and to be free to use. It's one of the two mandatory audio codecs of WebRTC, the technology behind video calls in the browser, and it's what many voice and chat apps send over the network.

An .opus file is Opus audio saved in an Ogg container, the same container as Ogg Vorbis files, following RFC 7845. In practice you meet them as voice messages, podcast and audiobook downloads, recordings from browser-based tools, and audio ripped from web videos.

A few things worth knowing up front:

• Opus is the codec, Ogg the container. The same Opus sound can sit in a .webm or .weba file, an .mp4, or an .ogg file. Only .opus promises Ogg with Opus inside and nothing else.
• It's lossy. Like MP3 or AAC, it throws away detail you're unlikely to hear. At about 128 kbps, stereo music is close to indistinguishable from the original for most listeners.
• It's always 48 kHz. Whatever the original sample rate, an Opus file plays back at 48,000 samples per second. More on that below.

Opus combines two codecs that were developed separately. SILK, from Skype, models the voice the way speech codecs do, with linear prediction. CELT, from Xiph.Org, transforms short blocks of sound into frequencies (an MDCT, as in AAC and Vorbis) and suits music and anything that isn't speech. The encoder picks one, or both at once, for every packet:

The first byte of every packet, the TOC byte, names one of 32 configurations (mode, bandwidth and frame size), says whether the packet is mono or stereo, and how many frames it holds. A packet can carry up to 120 ms of sound. Because the mode can change from one packet to the next, a single file can switch between speech and music coding as the content changes, which is why one codec can replace several.

Opus uses 20 ms frames by default, a compromise between delay and quality. Calls can use shorter frames for less delay; recordings sometimes use longer packets. In our test, Chrome's MediaRecorder packed three 20 ms frames into each 60 ms packet.

Opus covers 6 to 510 kbps. Unlike MP3, it gives usable results at very low rates, because the encoder narrows the audio bandwidth and switches to speech coding as the budget shrinks. Xiph.Org's recommended settings:

Xiph.Org describes 128 kbps VBR stereo as "pretty much transparent". Opus encodes with a variable bit rate by default, so the figure is an average target: in our tiny example the first packet is the largest, as the encoder spends bits on the start of the tone. The opusenc command-line tool defaults to 64 kbps per mono stream and 96 kbps per stereo pair.

Opus doesn't store a sample rate the way WAV or MP3 do. Internally it works at 8, 12, 16, 24 or 48 kHz depending on the bandwidth, and RFC 7845 tells players to decode at 48 kHz whenever the hardware allows. Granule positions, which mark time in the Ogg pages, always count 48 kHz samples. The input sample rate in OpusHead is only a note of where the audio came from.

• A 44.1 kHz song comes back at 48 kHz. 44.1 kHz isn't one of Opus's internal rates, so encoders resample it, usually to 48 kHz. That's not a loss: Opus's fullband mode stops at 20 kHz anyway. If you need 44.1 kHz again, resample after decoding (-ar 44100 in FFmpeg).
• Pre-skip and end trimming make lengths exact. The decoder drops the pre-skip samples at the start and uses the last page's granule position to cut the padding at the end, so a 10-second recording decodes to exactly 10 seconds. Tools that ignore this add a few milliseconds of silence, which matters for gapless albums and loops.
• Output gain is applied for you. The gain field in OpusHead, and optional R128_TRACK_GAIN and R128_ALBUM_GAIN tags, adjust loudness during decoding. RFC 7845 prefers these to older ReplayGain tags.

• WebRTC: RFC 7874 makes Opus mandatory for every WebRTC endpoint, alongside G.711. Browser-based calls and meeting tools use it.
• Discord: its voice documentation requires voice data to be Opus, stereo, at 48 kHz.
• WhatsApp voice messages: Meta's WhatsApp Business Platform documentation says voice messages must be .ogg files encoded with Opus: Ogg Opus with a different extension.
• Browser recordings: Chrome records audio with the MediaRecorder API as Opus in WebM, often saved as .weba, and Firefox can also record Ogg Opus.
• Files and streams: podcasts, audiobooks and internet radio, where Opus gives good quality at small sizes.

On a web page, ask the browser before relying on it: audio.canPlayType('audio/ogg; codecs=opus') returns "probably" where Ogg Opus is supported and an empty string where it isn't. For older Safari versions, offer an AAC or MP3 fallback in a second <source> element.

New libopus releases stay compatible with RFC 6716, so an Opus file made today plays in a decoder from 2012. Improvements come from the encoder and from optional extras in the reference library:

These features matter mostly for calls over bad networks. The ML features stay compatible with RFC 6716: a file or stream made with them still plays in an older decoder, which just doesn't use the extra data. DRED and the HD extension are still being standardised, so their formats may change.

• Won't play on an iPhone, iPad or Mac: Safari before 18.4 (macOS 15.4, iOS 18.4) doesn't play Ogg Opus, and many other apps on Apple devices still don't. Update, use VLC, or convert to M4A (AAC).
• The file says 48 kHz but the original was 44.1 kHz: that's how Opus works. Resample after decoding if your workflow needs 44.1 kHz.
• No duration in the browser, or a duration of Infinity: like any Ogg file, the length comes from the last page. If the server doesn't support range requests, the browser can't read the end without downloading everything. In our test, Chrome showed Infinity from a server without range requests.
• A .ogg file that won't play in a Vorbis-only player: it may contain Opus. Check with ffprobe and rename it to .opus.
• Slight silence at the start or end after converting: the tool ignored the pre-skip or end trimming. FFmpeg handles both: our 0.1-second example decodes to exactly 4,800 samples.
• A WhatsApp voice note won't open on a computer: it's Ogg Opus with a .ogg extension. VLC plays it; to share it widely, convert it to MP3 or M4A.

• Play it: drag it into Chrome, Edge, Firefox or Safari 18.4 and later, or use VLC, foobar2000 or mpv. Android plays it natively.
• Edit it: Audacity imports and exports Opus. Every export re-encodes, so keep a lossless master (WAV or FLAC) and export once at the end.
• Inspect it: ffprobe file.opus shows the codec, channels and tags; opusinfo from opus-tools gives more detail about the stream.
• Edit tags: Mp3tag, Kid3, foobar2000 and MusicBrainz Picard read and write OpusTags.

# WAV or FLAC to Opus, 96 kbps
ffmpeg -i in.wav -c:a libopus -b:a 96k out.opus
opusenc --bitrate 96 in.wav out.opus

# speech at a low bit rate
ffmpeg -i talk.wav -c:a libopus -b:a 24k -application voip talk.opus

# Opus to MP3 or M4A (AAC)
ffmpeg -i in.opus -c:a libmp3lame -q:a 2 out.mp3
ffmpeg -i in.opus -c:a aac -b:a 160k out.m4a

# Opus in WebM to .opus, without re-encoding
ffmpeg -i in.webm -vn -c:a copy out.opus

Converting Opus to MP3 or AAC is lossy to lossy, so use a generous bit rate. Converting it to WAV or FLAC makes a bigger file but can't bring back what the Opus encoder removed.