Audio analyzers use filters for many reasons: to define the measurement bandwidth, to isolate tones for measurement, to remove fundamental signals, and so on. In modern instruments, the majority of this filtering is done digitally, following analog-to-digital conversion if the signal is not already digital. Digital filter design is a mature field that encompasses a broad range of techniques, from classical analog filter design to advanced iterative design methods. However, the filter design considerations and techniques unique to audio analyzers do not seem to occupy much space in the published literature. This paper aims to correct this with a discussion of filter design, implementation, and optimization for modern Intel x86 architectures.
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