openKylin opus package

opus 1.4-ok1 source package in openKylin

Changelog

opus (1.4-ok1) nile; urgency=high

  * Build for openKylin.

 -- Luoyaoming <email address hidden>  Wed, 24 Apr 2024 16:34:32 +0800

Upload details

Uploaded by:
luoyaoming
Sponsored by:
Cibot
Uploaded to:
Nile V2.0
Original maintainer:
Openkylin Developers
Architectures:
any all
Section:
sound
Urgency:
Very Urgent

Publishing See full publishing history

Series Pocket Published Component Section
Huanghe V3.0 proposed main sound
Huanghe V3.0 release main sound
Nile V2.0 release main sound
Nile V2.0 proposed main sound

Downloads

File Size SHA-256 Checksum
opus_1.4.orig.tar.gz 1.0 MiB c9b32b4253be5ae63d1ff16eea06b94b5f0f2951b7a02aceef58e3a3ce49c51f
opus_1.4-ok1.debian.tar.xz 102.4 KiB fd6a7efc77bc6ac78a00e8b902b4c68675d3742f3c93313601ab9659e675823d
opus_1.4-ok1.dsc 2.0 KiB 4fab204294c9893a309e5db90b6ad95d23807472bbcae27d418b38abf6b2cff4

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