opus 1.4-1build1 source package in Ubuntu

Changelog

opus (1.4-1build1) noble; urgency=high

  * No change rebuild for 64-bit time_t and frame pointers.

 -- Julian Andres Klode <email address hidden>  Mon, 08 Apr 2024 18:14:57 +0200

Upload details

Uploaded by:
Julian Andres Klode
Uploaded to:
Noble
Original maintainer:
Ubuntu Developers
Architectures:
any all
Section:
sound
Urgency:
Very Urgent

See full publishing history Publishing

Series Pocket Published Component Section
Oracular release main sound
Noble release main sound

Downloads

File Size SHA-256 Checksum
opus_1.4.orig.tar.gz 1.0 MiB c9b32b4253be5ae63d1ff16eea06b94b5f0f2951b7a02aceef58e3a3ce49c51f
opus_1.4-1build1.debian.tar.xz 106.7 KiB cea2249c7aa45747ef64e8a10ae5f3f40eb7e895e2acfff44dceaf453eed6da3
opus_1.4-1build1.dsc 2.2 KiB 873892e1f22d3297a79adbec277e53da490da249855ba1b69f19431d9ef806bf

Available diffs

View changes file

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