The Open Media Alliance (AOMedia), overseeing the development of the AV1/AV2 video encoding format, the IAMF audio format, and the AVIF image format, has begun developing the Open Audio Codec (OAC), touted as a continuation of the Opus codec developed by Xiph.Org. As a reference implementation of the OAC encoder and decoder under the BSD license, the liboac library has been proposed, built upon the code of the libopus library.
The primary purpose of OAC is to transmit speech and sound over the internet. The codec is suitable for various applications related to interactive audio transmission and can be used in contexts ranging from VoIP telephony and video conferencing to gaming chats and remote concerts. OAC is scalable for compressing high-bitrate streaming audio, such as high-quality stereo music, as well as for compressing voice over bandwidth-restricted communication channels.
Development is currently at an 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 debug data and does not support seeking within the stream.
Except for the lack of mention of dynamic bitrate adjustment, audio frequency range, and frame size, the capabilities list of OAC mirrors the description of the Opus 1.5 codec:
- Bitrate from 6 Kbps to 510 Kbps;
- Sampling frequency from 8 to 48 KHz;
- Frame durations from 2.5 to 60 milliseconds;
- Support for constant (CBR) and variable (VBR) bitrates;
- Support for up to 255 channels;
- Support for narrowband and wideband audio;
- Support for speech and music;
- Support for stereo and mono;
- Ability to recover the audio stream in case of frame loss (PLC);
- Availability of implementations utilizing both floating-point and fixed-point arithmetic.
Source: opennet.ru
