The organization Xiph.Org, which develops free video and audio codecs, has released audio codec Opus 1.4.0, providing high coding quality and minimal delay for both high-bitrate streaming audio compression and voice compression in bandwidth-limited VoIP applications. The reference implementations of the encoder and decoder are distributed under the BSD license. Complete specifications for the Opus format are publicly available, free, and have been approved as an internet standard (RFC 6716).
The codec was created by combining the best technologies from the CELT codec developed by Xiph.org and the SILK codec from Skype. In addition to Skype and Xiph.Org, companies such as Mozilla, Octasic, Broadcom, and Google also participated in the development of Opus. The patents involved in Opus are provided by the participating companies for unlimited use without paying licensing fees. All intellectual property rights and patent licenses related to Opus are automatically delegated to applications and products using Opus, without the need for additional agreements. There are no restrictions on the scope of application and the creation of alternative third-party implementations. However, all granted rights are revoked in the event of a patent dispute related to Opus technologies against any Opus user.
Opus offers high coding quality and minimal delay for both high-bitrate streaming audio compression and voice compression in bandwidth-limited VoIP applications. Previously, Opus was recognized as the best codec when using a bitrate of 64Kbit (Opus outperformed competitors like Apple HE-AAC, Nero HE-AAC, Vorbis, and AAC LC). Notable products that support Opus out of the box include the Firefox browser, the GStreamer framework, and the FFmpeg package.
Key features of Opus:
- Bitrate from 5 to 510 Kbit/s;
- Sampling frequency from 8 to 48 KHz;
- Frame duration from 2.5 to 120 milliseconds;
- Support for constant (CBR) and variable (VBR) bitrates;
- Support for narrowband and wideband audio;
- Support for speech and music;
- Support for stereo and mono;
- Support for dynamic adjustment of bitrate, bandwidth, and frame size;
- Ability to recover the audio stream in case of frame loss (PLC);
- Support for up to 255 channels (multi-stream frames)
- Availability of implementations utilizing both floating-point and fixed-point arithmetic.
Key innovations in Opus 1.4:
- Optimizations of encoding parameters have been implemented, aimed at enhancing subjective audio quality when enabling FEC (Forward Error Correction) to recover damaged or lost packets at bitrates from 16 to 24kbs (LBRR, Low Bit-Rate Redundancy).
- An OPUS_SET_INBAND_FEC option has been added to enable FEC error correction mode without forcing the activation of SILK mode (FEC will not be used in CELT mode).
- Improvements have been made to the DTX (Discontinuous Transmission) mode, allowing traffic transmission to pause when no sound is detected.
- Support for the Meson build system has been added, along with enhanced support for building with CMake.
- An experimental 'Real-Time Packet Loss Concealment' mechanism has been introduced to restore speech fragments lost due to packet loss, utilizing machine learning technologies.
- An experimental implementation of a 'deep redundancy' mechanism has been added, which uses a machine learning system to enhance audio recovery after packet loss.
Source: opennet.ru
