The Open Media Alliance (AOMedia), which oversees the development of the AV1/AV2 video coding format, the IAMF surround sound format, and the AVIF image format, has begun developing the Open Audio Codec (OAC), which is billed as a continuation of the Opus codec developed by Xiph.Org. The liboac library, based on the libopus library, is proposed as the reference implementation of the OAC encoder and decoder under the BSD license.
The primary purpose of OAC is to transmit voice and audio over the internet. The codec is suitable for a variety of applications related to interactive audio transmission and can be used in a variety of fields, from VoIP telephony and video conferencing to gaming chats and remote concerts. OAC is scalable for both high-bitrate streaming audio compression, such as high-quality stereo music, and voice compression over bandwidth-constrained communication channels.
Development is still in the early alpha stage. The bitstream generated by the liboac library should not be used for distributing OAC files at this stage of development, as it includes additional debugging data and does not support stream position seeking.
With the exception of the lack of mention of dynamic bitrate, audio frequency range, and frame size adjustments, the OAC feature list mirrors the Opus 1.5 codec description:
- Bitrate from 6 Kbps to 510 Kbps;
- Sampling frequency from 8 to 48KHz;
- Frame duration from 2.5 to 60 milliseconds;
- Support for constant (CBR) and variable (VBR) bitrates;
- Support up to 255 channels;
- Support for narrowband and wideband audio;
- Voice and music support;
- Stereo and mono support;
- Ability to restore the audio stream in case of frame loss (PLC);
- Availability of implementations using floating and fixed point arithmetic.
Source: opennet.ru
