Google hitches Opus audio technology to WebRTC star
The developer version of Chrome now relies by default on Opus, a royalty-free audio compression technology designed for voice and music.
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Stephen Shankland has been a reporter at CNET since 1998 and writes about processors, digital photography, AI, quantum computing, computer science, materials science, supercomputers, drones, browsers, 3D printing, USB, and new computing technology in general. He has a soft spot in his heart for standards groups and I/O interfaces. His first big scoop was about radioactive cat poop.
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Opus is what's called a codec -- a technology to encode and decode streams of information, in this case audio. Technically, it's actually two codecs in one, an approach that lets it span a range of uses from Internet telephony on slow networks to streaming high-quality music on fast networks.
One of its chief virtues is low latency: there's not a long wait for audio to be encoded or decoded, something that's not a big problem with streaming music but can cripple a real-time conversation. Another advantage from a programmer's perspective is that unlike MP3 and AAC audio codecs, it's available royalty-free.
Google is among Opus' fans, and last week, it enabled the use of Opus by default in Chrome when establishing connections with the nascent WebRTC standard for browser-based voice and video chats. The change is in effect for Chrome 27, which is in the developer channel now, headed toward the beta and then the stable releases.
The move isn't a great surprise: the Internet Engineering Task Force, which standardizes both Opus and WebRTC, decided last year to make Opus a mandatory-to-implement (MTI) codec for WebRTC. But it is an important step in spreading the codec from the drawing board to the real world.
That's particularly true given that WebRTC has significant momentum. Its adoption will bring Skype-like abilities to the browser (although Microsoft, which owns Skype doesn't like WebRTC and prefers a lower-level alternative it proposed).