opus 1.3.1-1 source package in Ubuntu

Changelog

opus (1.3.1-1) unstable; urgency=medium

  * Salvaging package on behalf of the Multimedia Team (Closes: #1011281)

 -- IOhannes m zmölnig (Debian/GNU) <email address hidden>  Fri, 10 Jun 2022 08:51:39 +0200

Upload details

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

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Series Pocket Published Component Section

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File Size SHA-256 Checksum
opus_1.3.1-1.dsc 2.2 KiB a156456f34e01b276cea0fb3dcd4b47f5bec7f49b549c4d960b42c39fdc7648a
opus_1.3.1.orig.tar.gz 1015.7 KiB 65b58e1e25b2a114157014736a3d9dfeaad8d41be1c8179866f144a2fb44ff9d
opus_1.3.1-1.diff.gz 8.9 KiB 9afaf234e08a79e85dd65151b30d682f66799f3cb72ef14bbe42c2bfcebe17f9

No changes file available.

Binary packages built by this source

libopus-dbg: No summary available for libopus-dbg in ubuntu kinetic.

No description available for libopus-dbg in ubuntu kinetic.

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.