Existing speech codecs are highly optimized for two-way speech communication and audio codecs for one-way music distribution and streaming. Furthermore, playback quality is limited by the linear and nonlinear distortions of small loudspeakers in mobile terminals.

The Audio project tackles these issues through the development of a low-latency hi-fidelity unified speech and audio codec (USAC), the delivery of realistic multimedia experiences using spatial audio, and the improvement of playback quality by compensating for loudspeaker distortions using nonlinear signal processing.

Transmission of speech over the network is also a part of this project. Multimedia services, such as audiovisual streaming, need to adapt to wired and wireless networks with complex Quality-of-Service (QoS) characteristics. Future packet-based mobile networks will place considerable constraints on the seamless distribution and transmission of media, particularly in real-time delivery scenarios. There is therefore a need for codec and network integration technology to enable advanced and seamless multimedia services delivering optimal QoS for each application. Research issues include new channel coding techniques targeted at broadcast and multicast media distribution, and joint source and channel coding. .

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Transmission of speech over the network is also a part of this project. Multimedia services, such as audiovisual streaming, need to adapt to wired and wireless networks with complex Quality-of-Service (QoS) characteristics. Future packet-based mobile networks will place considerable constraints on the seamless distribution and transmission of media, particularly in real-time delivery scenarios. There is therefore a need for codec and network integration technology to enable advanced and seamless multimedia services delivering optimal QoS for each application. Research issues include new channel coding techniques targeted at broadcast and multicast media distribution, and joint source and channel coding.