RSS Feed for This PostCurrent Article

Opus–Open Source Audio Codec

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

Opus has been in development since early 2007. Programmers associated with Xiph.Org, Skype, and several other organizations have contributed to its development and to the standardization process as part of the IETF’s codec working group.

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. More information and 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 fullband
  • 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


Trackback URL


Sorry, comments for this entry are closed at this time.