Michael John

Jun 25, 2011
Sydney, Australia
Although not a new technology, manufacturers are increasingly including FIR filtering in loudspeaker processors and DSP based amplifiers due to the significant increase in performance-versus-cost of microprocessors and DSP hardware.
The advantages of FIR filtering include more arbitrary and fine control of a filter’s magnitude and phase characteristics, independent control of magnitude and phase, and the opportunity for maximum-phase characteristics (at the expense of some bulk time delay). The primary disadvantage is efficiency; FIR filters are generally more CPU intensive than IIR filters. For very long FIR filters, segmented frequency-domain and multi-rate methods help to reduce the computational load, but these methods come with increased algorithmic complexity.
In pro audio, the terms FIR Filter and FIR Filtering are often used when referring to specific implementations, such as:

linear-phase crossover and linear-phase brick-wall crossover filters
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You know, it's almost painful to see a good three semesters of my degree summarized on one page.

There's an important thing that I'd like to hammer home about the FIR process. The ability to arbitrarily design a FIR filter's response is the key takeaway here. While a FIR filter may never be perfect at LF, it excels at medium length filters where latency can be low and corrections aggressive. It's hard to express how amazing it is after fighting with IIR filters for years when you discover that we have enough processing power to approximate any sampled LTI impulse response, essentially for free.
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Wonderful article Michael (and Bennett). I always appreciate when knowledgeable folks are willing to share openly with the community. Kudos for making what can be a very complex topic very accessible. This is what makes communities such as these such a treasure.