Abstract
The paper describes a technique to design a complexity-scalable audio codec based on the partial reconstruction of the spectrum of a signal. The technique can be applied to any subband coding system based on cosine-modulated filterbanks. In the decoder, only a part of the subband is reconstructed, achieving an output signal with a lower sampling rate. The new synthesis filterbank, with a reduced number of channels and low computational cost, must be suitably designed. Hence, computational saving is traded for lower quality in the reconstructed signal. The technique can be used to design low-complexity decoders compatible with the MPEG audio standard, i.e. able to decode only a part of an MPEG bit stream. In this way, a single MPEG-coded audio can be decoded by different receivers, i.e. standard and low-complexity ones. The problem of having fixed bit rate streams for all the decoders will be discussed.
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