last modified 9/12/2005 11:58 by Mr Nicholson Warner
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Telecommunications The University of Adelaide Australia
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Prof Michael Rumsewicz
University of Adelaide
Australia, 5005

Telephone: +61 8 8303 5413
Facsimile: +61 8 8303 4395
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Smart Internet Co-operative Research Centre – Distributed Audio Servers

The Problem:

A growing area of technical importance is that of distributed virtual environments for work and play. For the audio component of such environments to be useful, great emphasis must be placed on the delivery of high quality audio scenes in which participants may change their relative positions. By maintaining synchronization, an end-user is ensured a more stable audio environment, which will assist in maximizing the usability of the application especially when the application characteristics are highly dynamic.

 

The Smart Internet CRC was interested in designing a multimedia application focussed on an immersive virtual environment. In this project, it was apparent that an integral part of constructing such an environment was the realism of the audio component. To further add complexity, there was a desire to enable a large number of users to communicate with and over-hear other users in the environment, such as would occur in a cafe environment in the real world. The Teletraffic Research Centre’s contribution to this project was the design of synchronization algorithm to assist in the delivery of high quality interactive audio to the end-users.

Solution:

The delivered solution was in the form of an efficient algorithm that can achieve and maintain relative synchronization between audio streams in a real time audio mixing environment. The algorithm does not attempt to maintain absolute synchronization between audio streams, as this would require the use of global timing. Rather, the algorithm attempts to maintain consistency within the mixing process such that the alignment of audio samples from each stream remains as constant as possible given the random elements of network delay. The algorithm is able to adapt quickly to gross changes in the underlying delays of each stream, as might result from network link failures. At the same time, the proposed algorithm is robust to short term variations in delay resulting from audio stream packets being queued at routers.

 


last modified 7/02/2006 09:57 by Mr Nicholson Warner