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Title: Opus codec and library API documentation
Author: IETF codec working group
Abstract: A low-latency interactive speech and audio codec
.
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.
.
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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Sampling rates from 8 to 48 kHz
Bit-rates from 6 kb/s 510 kb/s
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
Frame sizes from 2.5 ms to 60 ms
Good loss robustness and packet loss concealment (PLC)
Section: Programming/C
Format: HTML
Index: /usr/share/doc/libopus-doc/html/index.html
Files: /usr/share/doc/libopus-doc/html/*
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