Open Media Alliance (AOMedia), which oversees the development of the AV1/AV2 video coding format, the IAMF spatial audio format, and the AVIF image format, has begun developing the Open Audio Codec (OAC), which is presented as a continuation of the Opus codec developed by Xiph.Org. The reference implementation of the OAC encoder and decoder under the BSD license is the liboac library, built on the code of the libopus library.
The primary purpose of OAC is to transmit speech and audio over the Internet. The codec is suitable for various applications related to interactive sound transmission and can be used in areas ranging from VoIP telephony and video conferencing to game chats and remote concerts. OAC is scalable for compressing high-bitrate streaming audio, for example, for high-quality stereo music transmission, as well as for compressing voice in bandwidth-constrained communication channels.
The development is currently at an early alpha stage. The bitstream generated by the liboac library at this stage of development should not be used for distributing OAC files, as it includes additional debugging data and does not support seeking in the stream.
With the exception of the lack of mention of dynamic bitrate adjustment, audio frequency range, and frame size, the list of OAC features replicates the description of the Opus codec 1.5:
- Bitrate from 6 kbps to 510 kbps;
- Sampling frequency from 8 to 48 kHz;
- Frame duration 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 both voice and music;
- Support for stereo and mono;
- Ability to recover the audio stream in case of packet loss (PLC);
- Availability of implementations using both floating-point and fixed-point arithmetic.
Allikas: opennet.ru
