This paper presents a digital signal processing algorithm for efficient and high-quality audio equalization. In this approach, the original full-band audio signal is first downsampled and separated into two or more sub-band signals using a multirate filterbank, after which the equalization is performed in the downsampled domains. After the equalization, the signal is upsampled and combined back to a full-band audio signal. Linear-phase FIR filters, designed based on user-controlled parameters, are used to implement the actual equalization. The method presented in this paper helps in designing an implementation that results in computational savings, while still preserving optimal sound quality with any equalization parameter setting.
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