Opus Interactive Audio Codec


Opus is a totally open, royalty-free, highly versatile audio codec. Opus is unmatched for interactive speech and music transmission over the Internet, but is 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 bitrate narrowband speech to very high quality stereo music. Supported features are:

  • Bitrates 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 bitrate (CBR) and variable bitrate (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

You can read the full specification, including the reference implementation, in RFC 6716. An up-to-date implementation of the Opus standard is also available from the downloads page.


  • Version 1.1.1-beta released

    15 October, 2014

    The is a beta release for the upcoming 1.1.1. This release focuses on code optimizations, especially on x86, MIPS and ARM. The quality should be unchanged. downloads page.

  • Version 1.1 released

    5 December, 2013

    After more than two years of development, we have released Opus 1.1. This includes:

    • new analysis code and tuning that significantly improves encoding quality, especially for variable-bitrate (VBR),
    • automatic detection of speech or music to decide which encoding mode to use,
    • surround with good quality at 128 kbps for 5.1 and usable down to 48 kbps, and
    • speed improvements on all architectures, especially ARM, where decoding uses around 40% less CPU and encoding uses around 30% less CPU.
    These improvements are explained in more details in Monty's demo (updated from the 1.1 beta demo).
  • Version 1.1-rc3 released

    3 December, 2013

    This is the third and likely last release candidate for 1.1. It fixes several bugs in the fixed-point build. Floating point is unaffected. The code is available from the downloads page.

  • Version 1.1-rc2 released

    27 November, 2013

    This second release candidate fixes an alignment issue on ARMv5E and improves the Neon optimizations. Other platforms are unaffected. The code is available from the downloads page.

  • Version 1.1-rc released

    26 November, 2013

    We're getting closer to the final 1.1 with this new release that further improves surround encoding quality. It also includes better tuning of surround and stereo for lower bitrates. The complexity has been reduced on all CPUs, but especially ARM, which now has Neon assembly for the encoder. At last, there are a few bugfixes. The code is available from the downloads page.

  • Versions 1.1-beta and 1.0.3 released

    11 July, 2013

    Following up on the alpha, here's now the beta release for version 1.1. Monty wrote a nice demo page showing off all the improvements in 1.1. As for version 1.0.3 it includes a backport of the new 1.1 surround API (see the demo) as well as some minor bugfixes. The code is available from the downloads page.

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