This research report is devoted to an investigation of finite impulse response (FIR) filters and their implementation via windowing techniques. Analysis is restricted to those FIR filters that possess both a nonrecursive structure and a linear phase response. Initially an informative yet concise overview of FIR filters is presented. Some knowledge of "discrete – time" systems is assumed. Next, the concepts associated with FIR filter implementation through the utilization of window functions is developed. Analysis is performed in both the time and frequency domains. Results obtained define optimal window function criteria. The last design technique is devoted to computer aided FIR filter design via windows. The purpose of this section is to demonstrate the advantages of using the computer to solve the FIR filter – window design problem. An example problem supplements this section. Conclusions, along with a brief summary are documented.

The relationship between internal precision and input/output precision and the ratio of sample frequency to signal frequency has been described in qualitative terms for the general case and in quantitative terms for many special cases. This paper attempts to generate quantitative approximated guidelines for the general case. Lowpass and bandpass digital filters are implemented using cascaded second order sections via a program that varies the amount of internal precision available for the emulation, and calculates the error produced compared to a 64 bit standard. The errors are plotted, and general equations relating input/output precision and internal precision and the ratio of sample frequency to signal frequency and the order of the filter are developed for both the bandpass and lowpass cases.

The Fujitsu MB8764 digital signal processing chip is designed to operate with a machine cycle of up to 10 MHz. The chip’s ability to perform a 16-by-16 bit multiply and add operation in one machine cycle makes it a good candidate for real time digital signal processing. Unlike the Intel 2920 the Fujitsu MB8764 does not have an onboard analog-to-digital, digital-to-analog converter. Therefore, this paper will be restricted to the operation of this device with digital data input and output. The use of the MB8764 as a digital filter is analyzed. Performance limitations due to finite word length, memory size and configuration, and clock rate are considered. The MB8764 capabilities in computing fast Fourier transforms are discussed. Development of a working digital filter with the MB8764 work station is presented.

When a digital process is realized on a general-purpose computer or a special-purpose hardware, errors due to finite register length are introduced. These errors are due primarily to arithmetic roundoff, coefficient quantization, and scaling rules. This paper addresses the effects of finite word length on a direct-form implementation of a high order H (z) transfer function. The development and analysis of a modified direct-form realization suggested by Dr. Fred O. Simons, are carried out via FORTRAN emulation of a fourth-order low-pass Butterworth filter. The results are presented as a parametric tradeoff of signal-to-noise ratio at the filter output versus word length. Conclusions are drawn by comparing the modified direct-form with the canonic direct-form. The analysis presented here is intended to illustrate how a high order transfer function can be realized directly without decomposing into a group of low-order subfilters.

A deterministic and stochastic signal analysis is presented on switched capacitor filters. This relatively new discrete-time circuit design technology provides a wide field of applications in the areas of filters, circuitry, and communications. It also raises numerous problems and challenges in both discrete-time design implementation and noise analysis. Basic principles of switched capacitor integration/summation networks and their interfacing is first presented. A review of fundamental discrete-time system design techniques which are, and can be, adopted in the design of switched capacitor filters is then provided along with a discussion of frequency design criteria to deal with frequency warping and aliasing effects. An analysis of computer and test results of a 7th-order bilinear and an 8th-order LDI leapfrog switched capacitor lowpass filter is then presented. Finally, with the intent of presenting an alternative approach to current analytical procedures in switched capacitor noise spectrum analysis, various spectrum estimation techniques are presented along with test results of noise spectrum estimation of a typical switched capacitor integration circuit. / M.S.

This paper presents a high speed digital signal processor (DSP) based system architecture for motor drive control. The system achieves fast speed performance by using the 50 MHz TMS320C25 DSP and specialized digital hardware to perform data acquisition and output control tasks usually performed in software. The peripheral hardware has been designed for easy interface to many types of motor drive systems, to make the system generally applicable in the motion control field. The specifications, systematic design, and realization of this general purpose controller are described. Software to support the features of the system is discussed. Experimental results using the proposed system to control a switched reluctance motor drive, both in torque mode and four quadrant speed operation, verify the speed performance of the DSP based system. / Master of Science

The last few decades have witnessed what is commonly known as an "information explosion". The advent of the Internet has enabled the sharing of large amounts of information between users in almost every part of the world. This had led to increased bandwidth and storage requirements. These requirements necessitate the development of new techniques to optimize the utilization of existing bandwidth and to minimize storage requirements. In emerging areas of real-time multimedia applications, researchers have successfully developed new approaches, employing techniques such as transform coding, prediction, vector quantization, etc, to efficiently represent and consequently compress data, especially image data. Recently, multiple transform domain representation techniques have been reported which successfully compress one and multidimensional signals. In this thesis, a novel and efficient codec using multiple transform domain representation in conjunction with split vector quantization is presented, which provides superior coding performance for images. An adaptive scheme that further enhances the representation accuracy of the above codec is also described. Results of simulations of the proposed technique are presented that confirm the improved performance of the codec.

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Within the last decade, a new type of signal acquisition has emerged called Compressive Sensing that has proven especially useful in providing a recoverable representation of sparse signals. This thesis presents similar results for Compressive Parametric Estimation. Here, signals known to lie on some unknown parameterized subspace may be recovered via randomized compressive measurements, provided the number of compressive measurements is a small factor above the product of the parametric dimension with the subspace dimension with an additional logarithmic term. In addition to potential applications that simplify the acquisition hardware, there is also the potential to reduce the computational burden in other applications, and we explore one such application in depth in this thesis. Source localization by matched-field processing (MFP) generally involves solving a number of computationally intensive partial differential equations. We introduce a technique that mitigates this computational workload by ``compressing'' these computations. Drawing on key concepts from the recently developed field of compressed sensing, we show how a low-dimensional proxy for the Green's function can be constructed by backpropagating a small set of random receiver vectors. Then, the source can be located by performing a number of ``short'' correlations between this proxy and the projection of the recorded acoustic data in the compressed space. Numerical experiments in a Pekeris ocean waveguide are presented which demonstrate that this compressed version of MFP is as effective as traditional MFP even when the compression is significant. The results are particularly promising in the broadband regime where using as few as two random backpropagations per frequency performs almost as well as the traditional broadband MFP, but with the added benefit of generic applicability. That is, the computationally intensive backpropagations may be computed offline independently from the received signals, and may be reused to locate any source within the search grid area. This thesis also introduces a round-robin approach for multi-source localization based on Matched-Field Processing. Each new source location is estimated from the ambiguity function after nulling from the data vector the current source location estimates using a robust projection matrix. This projection matrix effectively minimizes mean-square energy near current source location estimates subject to a rank constraint that prevents excessive interference with sources outside of these neighborhoods. Numerical simulations are presented for multiple sources transmitting through a generic Pekeris ocean waveguide that illustrate the performance of the proposed approach which compares favorably against other previously published approaches. Furthermore, the efficacy with which randomized back-propagations may also be incorporated for computational advantage (as in the case of compressive parametric estimation) is also presented.

Typescript (photocopy). / Digitized by Kansas State University Libraries