Ubuntu
opus package

opus 1.1.2-1ubuntu1 source package in Ubuntu

Changelog

opus (1.1.2-1ubuntu1) xenial; urgency=medium

  * Merge with Debian; remaining changes:
    - Convert to Debhelper 9 with a simpler rules file.
    - Build using dh-autoreconf.
    - Add a watch file.
    - debian/control: close some Lintian errors.
      - Pre-depend on multiarch-support.
      - Add ${misc:Depends} to package dependencies.
    - Convert to debian source format 3.0 (quilt).
    - Run the tests.

opus (1.1.2-1) unstable; urgency=medium

  * Fixes the transient detector on silence.
  * Fixes discontinuities in background noise after extended PLC.
  * Make the CELT background noise estimator adapt more quickly on DTX update.
  * Fixes max_decay for LFE in fixed-point.
  * Fixes patch_transient_decision() for hybrid mode.
  * Don't reset the RTCD arch on encoder/decoder reset.

 -- Matthias Klose <email address hidden>  Wed, 17 Feb 2016 16:33:28 +0100

Upload details

Uploaded by:
Matthias Klose
Uploaded to:
Xenial
Original maintainer:
Ron Lee
Architectures:
any all
Section:
sound
Urgency:
Medium Urgency

See full publishing history Publishing

Series Pocket Published Component Section
Bionic release main sound
Xenial release main sound

Downloads

File Size SHA-256 Checksum
opus_1.1.2.orig.tar.gz 988.4 KiB 7aaa84f06ec89cbf19d08c1dd1ceac965a11b28b3ff504cc52893f9be78eb5d1
opus_1.1.2-1ubuntu1.debian.tar.xz 5.8 KiB d31fc55216fbda1a05ea6d59f43b71988a5b748c3d292f0297dc97e5585c9535
opus_1.1.2-1ubuntu1.dsc 1.9 KiB 8f8ccf72939b5c7923df4a189ca1d7a0dcf0cd5151e70b5623029c095c54849e

Available diffs

View changes file

Binary packages built by this source

libopus-dbg: debugging symbols for libopus

 This package provides the detached debug symbols for libopus.

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-dev-dbgsym: debug symbols for package libopus-dev

 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

 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 package libopus0

 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.