![]() ![]() If you stick with linear phase filters, the latency quickly gets prohibitively large and you loose causality. Another possible option are FFT based filter (overlap add, etc.) but that presents a non-trivial trade off between latency and efficiency. In some cases you can use multi rate filters to implement this very efficiently, but I'm not sure if this is universally applicable. To do anything meaningful at 40 Hz when your sample rate is 48 kHz, an FIR filter needs to be thousands of taps long. The most straight forward reason is simply filter length. It's not that one cares much about matching or copying the "analog" but that digital IIR filters have some very nice and useful properties.įor example in audio, IIR filter are very common place and I use Butterworths on a daily basis. A second best answer if it's only an assumed statement or suspicion is to at least demonstrate a specific case of such a mapped filter to allow for testing against an "optimized" direct digital solution.Īnd simulation where "copying the analog" would result in the better solution. The best answer will list out applications for common low pass, high pass and band pass filters designs (not notch filters where an IIR would certainly rule) that the optimized algorithms specific to FIR filters (including optimized multi-rate structures) cannot possibly surpass in performance, for any of the class filter types (or prove why the optimized algorithms are always preferred if that is the case). ![]() That said I could be missing succinct and good practical applications beyond modelling and simulation where "copying the analog" would result in the better solution. My use of the mapping from s to z (as was common prior to the late 1960's given the wealth of knowledge in analog filter design) is mostly limited to simulation and modelling of existing analog filters, but not for the creation of new digital filters for common low pass, high pass and band pass structures. DIGITAL FILTER DESIGNER FULLI have been taught (fred harris and others) to avoid the trap of "copying the analog" given those techniques with the classic types are limited to what we can feasibly do with a relatively low number of inductors and capacitors, while in the digital world we have the full power of the underlying mathematics and scalability with simple delays and multiplies (and non-linear commutators for multi-rate design resulting in very efficient FIR structures). My question is specific to the approach of designing higher performance low pass, high pass or band pass structures specifically by copying the analog classics- there may be actual utility in doing this beyond my current narrow view). (NOTE: This question is not in regards to the common and useful application of simple IIR structures for loop filters, leaky accumulators, notch filters, or in regards to using optimized IIR structures as direct digital designs. There are also examples of undesirable filtering, such as the uneven reinforcement of certain frequencies in a room with “bad acoustics.” A well-known signal processing wizard is said to have remarked, “When you think about it, everything is a filter.Of the four classic analog filter types: Butterworth, Chebyshev, Elliptic and Bessel- are any of these relegated to obsolescence for purposes of digital filter design in comparison to optimized algorithms such as least squares ( firls), Parks-McClellan ( firpm or remez), maximally flat ( maxflat), etc? Graphic equalizers, reverberators, echo devices, phase shifters, and speaker crossover networks are further examples of useful filters in audio. The tone control circuit in an ordinary car radio is a filter, as are the bass, midrange, and treble boosts in a stereo preamplifier. The different vowel sounds in speech are produced primarily by changing the shape of the mouth cavity, which changes the resonances and hence the filtering characteristics of the vocal tract. ![]() For example, speaker wire is not considered a filter, but the speaker is (unfortunately). However, we do not usually think of something as a filter unless it can modify the sound in some way. Any medium through which the music signal passes, whatever its form, can be regarded as a filter. ![]()
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