Ubuntu
opus package

opus 1.4-1 source package in Ubuntu

Changelog

opus (1.4-1) unstable; urgency=medium

  * New upstream version 1.4
  * Update d/copyright
    + Bump copyright dates
    + Re-generate d/copyright_hints

 -- IOhannes m zmölnig (Debian/GNU) <email address hidden>  Tue, 13 Jun 2023 18:10:36 +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
Mantic release main sound

Downloads

File Size SHA-256 Checksum
opus_1.4-1.dsc 2.2 KiB 70dc728fa1fc7fd7b3fae962a4eaf04f34b0d4df586fec81987274c1f1b804e6
opus_1.4.orig.tar.gz 1.0 MiB c9b32b4253be5ae63d1ff16eea06b94b5f0f2951b7a02aceef58e3a3ce49c51f
opus_1.4-1.debian.tar.xz 106.7 KiB b1e8d9d827d46754cd94edfd60e15bf25e2200ef38249eb4fcee29200dc20543

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