libopus-dev - 1.5.2-2 main

The Opus codec is designed for interactive speech and audio transmission over
the Internet. It is designed by the IETF Codec Working Group and incorporates
technology from Skype's SILK codec and Xiph.Org's CELT codec.
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It is intended to suit 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. The current features are:
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Bit-rates from 6 kb/s 510 kb/s
Sampling rates from 8 to 48 kHz
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
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This package provides the Opus library headers and development files.