Opus (audio format)

Opus is a free and open source lossy audio coding format developed by the Xiph.Org Foundation and standardized by the Internet Engineering Task Force, designed for efficient low-latency encoding of both speech and general audio. Due to its lower latency relative to other standard codecs, Opus finds specific use cases in real-time interactive communication for low-end embedded processors.^[4][5] Opus replaces both Vorbis and Speex for new applications.

Opus combines the speech-oriented LPC-based SILK algorithm and the lower-latency MDCT-based CELT algorithm, switching between or combining them as needed.^[4] Bitrate, audio bandwidth, complexity, and algorithm choice can be adjusted for each individual frame. Opus has low algorithmic delay (26.5 ms by default)^[6] ideal for use as part of a real-time communication link, networked music performances, and live lip sync; by trading off quality or bitrate, the delay can be further reduced down to 5 ms. Its delay thus is significantly lower compared to competing codecs, which require well over 100 ms. Opus remains competitive with these formats in terms of quality per bitrate.^[7]

As an open format standardized through RFC 6716, a reference implementation called libopus is available under the New BSD License. The reference has both fixed-point and floating-point optimizations for low- and high-end devices, with SIMD optimizations on platforms that support them. All known software patents that cover Opus are licensed under royalty-free terms.^[8] Opus is widely used as a voice over IP (VoIP) codec in applications such as Discord,^[9] WhatsApp,^[10][11][12] and the PlayStation 4.^[13] Listening tests have ranked it higher-quality than other standard audio formats at any given bitrate until transparency is reached, including MP3, AAC, and HE-AAC.^[14][15]