opus 1.5.1-1 source package in Ubuntu
Changelog
opus (1.5.1-1) unstable; urgency=medium
* New upstream version 1.5.1
+ Refresh patches
* Enable Deep Packet Loss Concealment (deep PLC)
* Enable Low-Bitrate Speech Quality Enhancement (LACE/NoLACE)
* Update d/copyright information
+ Update d/copyright
+ Re-generate d/copyright_hints
* Update symbols file
* Clarify machine learning usage in d/README.source
* Bump standards version to 4.7.0
-- IOhannes m zmölnig (Debian/GNU) <email address hidden> Mon, 24 Jun 2024 10:29:41 +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 | Published | Component | Section |
|---|
Downloads
| File | Size | SHA-256 Checksum |
|---|---|---|
| opus_1.5.1-1.dsc | 2.2 KiB | 9f437a8c4d6c007eb3ee76ff4f661d96edd84cfe2a52131953bf424a68986ded |
| opus_1.5.1.orig.tar.gz | 7.5 MiB | b84610959b8d417b611aa12a22565e0a3732097c6389d19098d844543e340f85 |
| opus_1.5.1-1.debian.tar.xz | 107.7 KiB | 2365fa0f6334443be4840c185334843e479e130e036bcadb9ca8b00091c774f6 |
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
