After a year and a half of development, the Xiph.Org organization, which creates free video and audio codecs, presented the release of the audio codec Opus 1.6. This new release offers high quality encoding and minimal latency for both high-bitrate streaming audio compression and voice compression in bandwidth-constrained VoIP telephony applications. The encoder and decoder reference implementations are distributed under the BSD license. The full specifications for the Opus format are publicly available, free, and approved as an Internet standard (RFC 6716).
The Opus codec is the result of combining the best technologies from the CELT codec developed by Xiph.org and the open source Skype codec SILK. Companies such as Mozilla, Octasic, Broadcom, and Google also participated in the development of Opus. The patents involved in Opus are provided by the companies involved for unlimited use without the need for license fees. All Opus-related intellectual rights and patent licenses are automatically delegated to applications and products using Opus, without requiring additional approval. However, these rights will be revoked if there is a patent litigation involving Opus technology against any Opus user.
Opus is known for its high encoding quality and minimal latency, making it ideal for high-bitrate streaming audio compression and voice compression in bandwidth-constrained VoIP telephony applications. It has been recognized as the best codec at a 64Kbit bitrate, surpassing competitors such as Apple HE-AAC, Nero HE-AAC, Vorbis, and AAC LC. Opus is supported by products like the Firefox browser, the GStreamer framework, and the FFmpeg package.
The main features of Opus include:
- Bitrate from 5 Kbps to 2 Mbps;
- Sampling frequency from 8 to 96KHz;
- Frame duration from 2.5 to 120 milliseconds;
- Support for constant (CBR) and variable (VBR) bitrates;
- Support for narrowband and wideband audio;