Increasingly there is a need for high quality audio streaming in devices such as modular home audio networking systems, PMPs and wireless speakers. As wireless transmission capacities continue to increase it is desirable to utilize this increasing data bandwidth to perform real-time wireless streaming of audio content coded in a lossless or near-lossless format. In this paper we discuss the development of an adaptive audio coding algorithm that balances the design goals of low latency, low complexity, error robustness and a dynamically-variable bit rate that scales to mathematically-lossless coding under suitable conditions. Particular emphasis is placed on the optimization of the algorithm structure for real-time audio processing applications and the mechanism by which "hybrid" lossless and near-lossless coding is achieved.
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