ANALYSIS OF THE CONTROLLABILITY OF SOME DIGITAL FILTERS WITH A FINITE IMPULSE RESPONSE
Abstract
This overview article covers finite impulse response filters and filter banks. The use of these filters
for hearing aids is considered. Ways to compensate for hearing loss and ways to increase loudness
using broadband amplification are considered. A schematic diagram of a method for digital
signal processing using a bank of filters, as well as a technique for synthesizing interpolation filters
with low computational complexity, is presented. Also, the application of the MATLAB system for the
synthesis of narrow-band non-recursive FIR filters, their design procedure, methodology and examples
are considered. Finite Impulse Response (FIR) filters and filter banks have specific properties
that guarantee stability. Therefore, they are popular in many applications such as communication
systems, audio signal processing, biomedical instruments, and so on. Unfortunately, due to the longer
wavelength, the cost of implementing an FIR filter is usually not higher than an infinite impulse response
(IIR) filter that meets the same requirements. It is well known that the length of an FIR filter is
inversely proportional to its transition bandwidth. Therefore, the disadvantage becomes acute when a
given filter has a narrow transition band. The main goal is to consider computationally efficient
methods for designing FIR filters and filter banks. The masking method (FRM) results in significant
savings in the number of multipliers. Next, a 16-band, low group delay, non-equal-spacing digital
FIR filter bank is considered. Overall latency is significantly reduced as a result of a new filter structure
that reduces the interpolation factor for prototype filters. Masking filter may be an interpolated
finite impulse response (IFIR) filter that helps reduce complexity.
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