Opus is a lossy audio coding format developed by the Xiph.Org Foundation and standardized by the Internet Engineering Task Force, designed to efficiently code speech and general audio in a single format, while remaining low-latency enough for real-time interactive communication and low-complexity enough for low-end embedded processors.

[4][5] Opus replaces both Vorbis and Speex for new applications, and several blind listening tests have ranked it higher-quality than any other standard audio format at any given bitrate until transparency is reached, including MP3, AAC, and HE-AAC.

Opus has very short latency (26.5 ms using the default 20 ms frames and default application setting), which makes it suitable for real-time applications such as telephony, Voice over IP and videoconferencing; research by Xiph led to the CELT codec, which allows the highest quality while maintaining low delay.

Unlike Vorbis, Opus does not require large codebooks for each individual file, making it more efficient for short clips of audio and more resilient.

In Opus, both were modified to support more frame sizes, as well as further algorithmic improvements and integration, such as using CELT's range encoder for both types.

Opus can transparently switch between modes, frame sizes, bandwidths, and channel counts on a per-packet basis, although specific applications may choose to limit this.

Opus compression does not depend on the input sample rate; timestamps are measured in 48 kHz units even if the full bandwidth is not used.

[23] Opus was proposed for the standardization of a new audio format at the IETF, which was eventually accepted and granted by the codec working group.

The SILK part has been under development at Skype since January 2007 as the successor of their SVOPC, an internal project to make the company independent from third-party codecs like iSAC and iLBC and respective license payments.

[24] Representatives of several companies which were taking part in the standardization of patent-encumbered competing format, including Polycom and Ericsson—the creators and licensors of G.719—as well as France Télécom, Huawei and the Orange Labs (department of France Télécom), which were involved in the creation of G.718, stated objections against the start of the standardization process for a royalty-free format.

On July 11, 2013, libopus 1.0.3 brought bug fixes and a new Surround sound API that improves channel allocation and quality, especially for LFE.

July 15, 2016 saw the release of version 1.1.3 and includes bug fixes, optimizations, documentation updates and experimental Ambisonics work.

[36] libopus 1.2 includes optional support for the decoder specification changes made in drafts of RFC 8251, improving the quality of output from such low-rate streams.

The deep redundancy (DRED) algorithm was developed by among others Jean-Marc Valin, Ahmed Mustafa, Jan Büthe, Timothy Terriberry, Chris Montgomery, Michael Klingbeil, and Paris Smaragdis from Amazon Web Services[46] with sponsorship to open source the algorithm and subsequently extend the IETF standard from Sid Rao.

[47] This encoder is a backwards compatible change to the codec enabling customers to easily upgrade applications to take advantage of this machine learning capability.

[9] In listening tests around 64 kbit/s, Opus shows superior quality compared to HE-AAC codecs, which were previously dominant due to their use of the patented spectral band replication (SBR) technology.

[7][50] Opus has very low algorithmic delay,[4] a necessity for use as part of a low-audio-latency communication link, which can permit natural conversation, networked music performances, or lip sync at live events.

[51] Total one-way latency below 150 ms is the preferred target of most VoIP systems,[52] to enable natural conversation with turn-taking little affected by delay.

[55] Opus permits trading-off reduced quality or increased bitrate to achieve an even smaller algorithmic delay (5.0 ms minimum).

The Wikimedia Foundation sponsored a free and open source online JavaScript Opus encoder for browsers supporting the required HTML5 features.

[66] In 2021, the Danish journalism website Zetland switched from MP3 to Opus for its articles' audio recordings, which attained a 35 percent reduction in bandwidth and reduced climate footprint.

Native Opus codec support is implemented in most major multimedia frameworks for Unix-like operating systems, including GStreamer, FFmpeg, and Libav libraries.

[78] macOS Sonoma added support for mono and stereo Opus audio encapsulated in MPEG-4 and WebM containers in Safari.

[79][80] On Windows 10, version 1607, Microsoft provided native support for Opus audio encapsulated in Matroska and WebM containers.

[81] On version 1709, support for Opus audio encapsulated in Ogg containers was made available through a pre-installed add-on called Web Media Extensions.

[84] iOS 17 includes support for natively encoding and decoding the Opus codec through the operating system's AudioToolbox framework.

(substitute with a pseudo file extension such as .m4a)[88] On Windows 8.1 and older:- Requires installation of a third-party multimedia framework, LAV Filters.

Several SIP clients, including Acrobits Softphone, CSipSimple (via additional plug-in), Empathy (via GStreamer), Jitsi,[113] Tuenti,[114] Line2 (currently only on iOS),[115] Linphone,[116] Phoner and PhonerLite,[117] SFLphone,[118] Telephone, Mumble, Discord[11] and TeamSpeak 3 voice chat software also support Opus.

[126] Classified-ads distributed messaging app sends raw opus frames inside TLS socket in its VoIP implementation.

[135][136] Android 13 supports Opus as a bluetooth headphone (A2DP) codec, motivated by its low latency, which is important for head-tracked spatial audio.