Ubuntu
opus package

opus 1.5.2-1 source package in Ubuntu

Changelog

opus (1.5.2-1) unstable; urgency=medium

  * New upstream version 1.5.2
  * Re-generate d/copyright_hints

 -- IOhannes m zmölnig (Debian/GNU) <email address hidden>  Mon, 24 Jun 2024 22:42:32 +0200

Upload details

Uploaded by:
Debian Multimedia Team
Uploaded to:
Sid
Original maintainer:
Debian Multimedia Team
Architectures:
any all
Section:
sound
Urgency:
Medium Urgency

See full publishing history Publishing

Series Pocket Published Component Section

Downloads

File Size SHA-256 Checksum
opus_1.5.2-1.dsc 2.2 KiB 6d6ec8c9423412b1549dba00685f85b61ede58d8603b0598356924b312b4f030
opus_1.5.2.orig.tar.gz 7.5 MiB 65c1d2f78b9f2fb20082c38cbe47c951ad5839345876e46941612ee87f9a7ce1
opus_1.5.2-1.debian.tar.xz 107.8 KiB e9e490d1041407dd64fbf6754f2bdd0f781d470852f330fa33d281b836266c7c

No changes file available.

Binary packages built by this source

libopus-dev: 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.

libopus-doc: 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.

libopus0: 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.

libopus0-dbgsym: debug symbols for libopus0