Novel Digital Alias-Free Signal Processing Approaches to FIR Filtering Estimation

Darawsheh, H. 2021. Novel Digital Alias-Free Signal Processing Approaches to FIR Filtering Estimation. PhD thesis University of Westminster Computer Science and Engineering https://doi.org/10.34737/v6w03

TitleNovel Digital Alias-Free Signal Processing Approaches to FIR Filtering Estimation
TypePhD thesis
AuthorsDarawsheh, H.
Abstract

This thesis aims at developing a new methodology of filtering continuous-time bandlimited signals and piecewise-continuous signals from their discrete-time samples. Unlike the existing state-of-the-art filters, my filters are not adversely affected by aliasing, allowing the designers to flexibly select the sampling rates of the processed signal to reach the required accuracy of signal filtering rather than meeting stiff and often demanding constraints imposed by the classical theory of digital signal processing (DSP). The impact of this thesis is cost reduction of alias-free sampling, filtering and other digital processing blocks, particularly when the processed signals have sparse and unknown spectral support.

Novel approaches are proposed which can mitigate the negative effects of aliasing, thanks to the use of nonuniform random/pseudorandom sampling and processing algorithms. As such, the proposed approaches belong to the family of digital alias-free signal processing (DASP). Namely, three main approaches are considered: total random (ToRa), stratified (StSa) and antithetical stratified (AnSt) random sampling techniques.

First, I introduce a finite impulse response (FIR) filter estimator for each of the three considered techniques. In addition, a generalised estimator that encompasses the three filter estimators is also proposed. Then, statistical properties of all estimators are investigated to assess their quality. Properties such as expected value, bias, variance, convergence rate, and consistency are all inspected and unveiled. Moreover, closed-form mathematical expression is devised for the variance of each single estimator.

Furthermore, quality assessment of the proposed estimators is examined in two main cases related to the smoothness status of the filter convolution’s integrand function, 𝑔(𝑑,𝜏)∢=π‘₯(𝜏)β„Ž(π‘‘βˆ’πœ), and its first two derivatives. The first main case is continuous and differentiable functions 𝑔(𝑑,𝜏), 𝑔′(𝑑,𝜏), and 𝑔′′(𝑑,𝜏). Whereas in the second main case, I cover all possible instances where some/all of such functions are piecewise-continuous and involving a finite number of bounded discontinuities.

Primarily obtained results prove that all considered filter estimators are unbiassed and consistent. Hence, variances of the estimators converge to zero after certain number of sample points. However, the convergence rate depends on the selected estimator and which case of smoothness is being considered.

In the first case (i.e. continuous 𝑔(𝑑,𝜏) and its derivatives), ToRa, StSa and AnSt filter estimators converge uniformly at rates of π‘βˆ’1, π‘βˆ’3, and π‘βˆ’5 respectively, where 2𝑁 is the total number of sample points. More interestingly, in the second main case, the convergence rates of StSa and AnSt estimators are maintained even if there are some discontinuities in the first-order derivative (FOD) with respect to 𝜏 of 𝑔(𝑑,𝜏) (for StSa estimator) or in the second-order derivative (SOD) with respect to 𝜏 of 𝑔(𝑑,𝜏) (for AnSt). Whereas these rates drop to π‘βˆ’2 and π‘βˆ’4 (for StSa and AnSt, respectively) if the zero-order derivative (ZOD) (for StSa) and FOD (for AnSt) are piecewise-continuous. Finally, if the ZOD of 𝑔(𝑑,𝜏) is piecewise-continuous, then the uniform convergence rate of the AnSt estimator further drops to π‘βˆ’2.

For practical reasons, I also introduce the utilisation of the three estimators in a special situation where the input signal is pseudorandomly sampled from otherwise uniform and dense grid. An FIR filter model with an oversampled finite-duration impulse response, timely aligned with the grid, is proposed and meant to be stored in a lookup table of the implemented filter’s memory to save processing time. Then, a synchronised convolution sum operation is conducted to estimate the filter output.

Finally, a new unequally spaced Lagrange interpolation-based rule is proposed. The so-called composite 3-nonuniform-sample (C3NS) rule is employed to estimate area under the curve (AUC) of an integrand function rather than the simple Rectangular rule. I then carry out comparisons for the convergence rates of different estimators based on the two interpolation rules. The proposed C3NS estimator outperforms other Rectangular rule estimators on the expense of higher computational complexity. Of course, this extra cost could only be justifiable for some specific applications where more accurate estimation is required.

Year2021
File
File Access Level
Open (open metadata and files)
PublisherUniversity of Westminster
Publication dates
PublishedJun 2021
Digital Object Identifier (DOI)https://doi.org/10.34737/v6w03

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