Efficient Audio Equalization Using Multirate Processing

A digital signal processing algorithm for efficient audio equalization is presented. In this approach the original full-band audio signal is first downsampled and separated into two or more subband signals using a multirate filter bank, 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 provides a framework for an audio equalizer design with optimization possibilities between audio quality, computational complexity, and memory requirements of the algorithm, making it suitable for various applications and computing platforms.