opus 1.3.1-3 source package in Ubuntu

Changelog

opus (1.3.1-3) unstable; urgency=medium

  [ Debian Janitor ]
  * Apply multi-arch hints.
    + libopus-doc: Add Multi-Arch: foreign.

  [ IOhannes m zmölnig (Debian/GNU) ]
  * Modernize 'licensecheck' target
    + Re-generate d/copyright_hints
  * Bump standards version to 4.6.2

 -- IOhannes m zmölnig (Debian/GNU) <email address hidden>  Sat, 28 Jan 2023 00:10:34 +0100

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
Lunar release main sound

Downloads

File Size SHA-256 Checksum
opus_1.3.1-3.dsc 2.2 KiB c373ac82a1afb824819618c12e279ee47b6ac2fa18083b9ef2bb9e8e9e588419
opus_1.3.1.orig.tar.gz 1015.7 KiB 65b58e1e25b2a114157014736a3d9dfeaad8d41be1c8179866f144a2fb44ff9d
opus_1.3.1-3.debian.tar.xz 106.4 KiB f4a145f0d5b9140106396fc0b0c515b1c3abd59f2aa5fb62b36b69886aa411bc

Available diffs

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