Free video and audio codec developer Xiph.Org has released the Opus 1.4.0 audio codec, which provides high-quality encoding and minimal latency for both high-bitrate streaming audio and voice compression in bandwidth-constrained VoIP applications. telephony. The encoder and decoder reference implementations are distributed under the BSD license. The complete specifications for the Opus format are publicly available, free of charge, and approved as an Internet standard (RFC 6716).
The codec is created by combining the best technologies from Xiph.org's CELT codec and Skype's open source SILK codec. In addition to Skype and Xiph.Org, companies such as Mozilla, Octasic, Broadcom and Google also took part in the development of Opus. The patents involved in Opus are granted by the companies involved in the development for unlimited use without payment of royalties. All intellectual property rights and patent licenses related to Opus are automatically delegated to applications and products using Opus, without the need for additional approval. There are no restrictions on the scope and creation of alternative third-party implementations. However, all rights granted are revoked in the event of patent proceedings affecting Opus technologies against any user of Opus.
Opus features high coding quality and minimal latency for both high-bitrate streaming audio compression and voice compression for bandwidth-constrained VoIP telephony applications. Previously, Opus was voted the best codec at 64Kbit (Opus outperformed competitors like Apple HE-AAC, Nero HE-AAC, Vorbis and AAC LC). Products that support Opus out of the box include the Firefox browser, the GStreamer framework, and the FFmpeg package.
Main features of Opus:
- Bitrate from 5 to 510 Kbit/s;
- Sampling frequency from 8 to 48KHz;
- Frame duration from 2.5 to 120 milliseconds;
- Support for constant (CBR) and variable (VBR) bitrates;
- Support for narrowband and wideband audio;
- Voice and music support;
- Stereo and mono support;
- Support for dynamic setting of bitrate, bandwidth and frame size;
- Ability to restore the audio stream in case of frame loss (PLC);
- Support up to 255 channels (multi-stream frames)
- Availability of implementations using floating and fixed point arithmetic.
Key innovations in Opus 1.4:
- Optimization of encoding parameters has been carried out, aimed at improving the subjective indicators of sound quality when FEC (Forward Error Correction) is enabled to restore damaged or lost packets at bit rates from 16 to 24kbs (LBRR, Low Bit-Rate Redundancy).
- Added option OPUS_SET_INBAND_FEC to enable FEC error correction but without forcing SILK mode (FEC will not be used in CELT mode).
- Improved implementation of the DTX (Discontinuous Transmission) mode, which provides suspension of traffic transmission in the absence of sound.
- Added support for the Meson build system and improved support for building using CMake.
- An experimental mechanism "Real-Time Packet Loss Concealment" has been added to restore fragments of speech lost as a result of packet loss, working through the use of machine learning technologies.
- An experimental implementation of the "deep redundancy" mechanism has been added, which uses a machine learning system to improve the efficiency of audio recovery after packet loss.
Source: opennet.ru