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Opus audio codecBig news for VoIP world! Opus audio codec has now been standardized by IETF as RFC 6716 . Just a little overview for those who is lazy to read about the codec on its website:

Opus is a totally open, royalty-free, highly versatile audio codec. Opus is unmatched for interactive speech and music transmission over the Internet, but also intended for storage and streaming applications. It is standardized by the Internet Engineering Task Force (IETF) as RFC 6716 which incorporated technology from Skype’s SILK codec and Xiph.Org’s CELT codec.

Opus can handle a wide range of audio applications, including Voice over IP, videoconferencing, in-game chat, and even remote live music performances. It can scale from low bit-rate narrowband speech to very high quality stereo music. Supported features are:

  • Bit-rates from 6 kb/s to 510 kb/s
  • Sampling rates from 8 kHz (narrowband) to 48 kHz (fullband)
  • Frame sizes from 2.5 ms to 60 ms
  • Support for both constant bit-rate (CBR) and variable bit-rate (VBR)
  • Audio bandwidth from narrowband to full-band
  • Support for speech and music
  • Support for mono and stereo
  • Support for up to 255 channels (multistream frames)
  • Dynamically adjustable bitrate, audio bandwidth, and frame size
  • Good loss robustness and packet loss concealment (PLC)
  • Floating point and fixed-point implementation

WebRTC working group is considering making Opus (together with G.711) a mandatory audio codec for browsers vendors that are going to support WebRTC standard. It makes Opus a default choice for all web-based audio and music. It’s a big day for open standards and all of us, hopefully we will be able to do achieve something similar with video codecs (MPEG-LA, do you hear us?) in the future. Looking forward to see Opus support in Chrome Canary and WebRTC in the nearest future.

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