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Opus Audio Codec 1.1

Opus is a codec for interactive speech and audio transmission over the Internet.

Both, Opus decoder (opusdec.exe) and Opus encoder (opusenc.exe) are packed into Opus tools archive. You can also download the DLLs: libogg.dll & libopus.dll

- In order to play .opus files, foobar2000 is a good choice.
- If you want to add support for opus files to other media player, we recommend to install DC-Bass Source Mod.
- To convert other audio files (.mp3, .flac, .aac, etc) into opus files and vice versa, use one of these audio converters: TAudioConverter or LameXP.

The Opus Audio Codec was developed by a consortium of researchers from Mozilla, Xiph.org, Skype, Microsoft, and Broadcomm.

Opus is able to switch between different codecs based on how much bandwidth it detects, so it can reduce latency and drop-outs on any connection.

Opus can handle a wide range of interactive 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.

Opus, when coupled with an appropriate container format, is also suitable for non-realtime stored-file applications such as music distribution, game soundtracks, portable music players, jukeboxes, and other applications that have historically used high latency formats such as MP3, AAC, or Vorbis.

Features of Opus Audio Codec:

- 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