Abstract
Wavelet filtering is a promising tool for use in audio signal compression. What is still lacking, however, is a thorough understanding of wavelet filters performance relative to the more sophisticated examples of conventional filters. So, as we seek to apply wavelet filters to low bit rate audio coding, attention must be focused not only to the bit-rate/signal-quality trade-off, but also the complexity and processing delay should not be underemphasised. The results presented in this paper attempt to clarify these issues. To assess the coding gain of wavelet and conventional filters, various codec models have been designed and implemented based on a wavelet packet algorithm, an auditory perception model and entropy noiseless coding. The wavelet packet based coding approach is compared to the MPEG-audio international standard in terms of objective and subjective measurements and is shown to be superior to MPEG-audio layer I and competitive with layer II.
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