Excerpts from the Abstract:

With the popularization of high-quality audio, there is an increasing demand for an efective compression and transmission technology. Despite of the success of current perceptual audio coding techniques, Some problems still remain open and need improvement. This dissertation addresses two system issues that may arise in practice in high-fidelity audio coding technology: (i) an inter-channel redundancy removal approach designed for high-quality multichannel audio compression and (ii) a progressive syntax-rich multichannel audio coding algorithm and its error-resilient codec design.

The first contribution of this dissertation is to develop a Modified Advanced Audio Coding with Karhunen-Loµeve Transform (MAACKLT) algorithm. In MAACKLT, we exploit the inter-channel redundancy inherent in most multichannel audio sources, and prioritize the transformed channel transmission policy. Experimental results show that, compared with MPEG AAC (Advanced Audio Coding), the MAACKLT algorithm not only reconstructs better quality of multichannel audio at a regular low bit rate of 64 kbit/s/ch but also achieves coarse-grain quality scalability.

The second contribution of this dissertation is the design of a Progressive Syntax-Rich Multichannel Audio Coding (PSMAC) system. PSMAC inherits the efficient inter-channel de-correlation block in the MAACKLT algorithm while adding a scalable quantization coding block and a context-based QM noiseless coding block. The final bitstream generated by this multichannel audio coding system provides fine grain scalability and three user-defined functionalities which are not available in other existing multichannel audio codecs. The reconstructed audio files generated by our proposed algorithm achieve an excellent performance in a formal subjective listening test at various bit rates. Based on the PSMAC algorithm, we extend its error-free version to an error-resilient codec by re-organizing the bitstream and modifying the noiseless coding modules. The performance of the proposed algorithm has been tested under different error patterns in WCDMA channels using several single channel audio materials. Our experimental results show that the proposed approach has excellent error resiliency at a regular user bit rate of 64 kbit/s.