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Source: opus
Section: sound
Priority: optional
Maintainer: Debian Multimedia Maintainers <debian-multimedia@lists.debian.org>
Uploaders:
IOhannes m zmölnig (Debian/GNU) <umlaeute@debian.org>,
Ron Lee <ron@debian.org>,
Build-Depends:
debhelper-compat (= 13),
Build-Depends-Indep:
doxygen,
graphviz,
Standards-Version: 4.6.2
Rules-Requires-Root: no
Homepage: http://www.opus-codec.org
Vcs-Git: https://salsa.debian.org/multimedia-team/opus.git
Vcs-Browser: https://salsa.debian.org/multimedia-team/opus
Package: libopus0
Section: libs
Architecture: any
Multi-Arch: same
Depends:
${misc:Depends},
${shlibs:Depends},
Suggests:
opus-tools,
Description: Opus codec runtime library
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:
.
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
.
This package provides the Opus runtime library.
Package: libopus-dev
Section: libdevel
Architecture: any
Multi-Arch: same
Depends:
libopus0 (= ${binary:Version}),
${misc:Depends},
Description: Opus codec library development files
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:
.
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
.
This package provides the Opus library headers and development files.
Package: libopus-doc
Section: doc
Architecture: all
Depends:
${misc:Depends},
Multi-Arch: foreign
Description: libopus API documentation
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.
.
This package contains the developer documentation for libopus.
|
