Thesis (MEng) -- Stellenbosch University, 2002. / ENGLISH ABSTRACT: The tuning of coupled-resonator filters is an essential part of its production. It is, however, a time consuming and expensive process. The use of model-based parameter extraction algorithms as a semi-automated tuning procedure is examined. This thesis describes the development of such a procedure. Different models for the filter are compared. The extraction algorithm is based on the least squares method. A more robust procedure is presented which makes the parameter extraction possible, without the availability of good starting values. The algorithm is evaluated using a fourth order single mode direct coupled cavity filter with a centre frequency of 10GHz. The algorithm produces good results and makes the tuning of this type of filter possible in an effective way. It also allows for the diagnosis of the filter, which is a great advantage over the more conventional direct tuning methods. / AFRIKAANSE OPSOMMING: Die instelling van gekoppelde-resoneerderfilters is ’n essensiele deel van die vervaardiging daarvan. Dit is egter ’n tydsame en duur proses. Die gebruik van modelgebaseerde parameteronttrekking as ’n semi-geoutomatiseerde instellingsproses is ondersoek. Hierdie tesis beskryf die ontwikkeling van so ’n prosedure. Verskillende modelle vir die filter is vergelyk. Die onttrekkingsalgoritme is gebaseer op die kleinste-kwadraatmetode. ’n Meer robuuste metode word voorgestel wat die parameteronttrekking moontlik maak, sonder dat daar goeie beginwaardes beskikbaar is. Die algoritme word geevalueer aan die hand van ’n vierde orde enkelmodus direkgekoppelderesoneerderfilter met ’n senterfrekwensie van 10GHz. Die algoritme lewer goeie resultate en maak die instelling van hierdie tipe filter moontlik op ’n effektiewe wyse. Die metode kan ook gebruik word vir die diagnose van die filter wat ’n groot voordeel is bo meer konvensionele direkte instellingsmetodes.

This thesis presents a study on the design of two-dimensional (2-D) digital filters by using the singular-value decomposition (SVD). A new method for the design of 2-D quadrantally symmetric FIR filters with linear phase response is proposed. It is shown that three realizations are possible, namely, a direct realization, a modified version of the direct realization, and a realization based on the combined application of the SV and LU decompositions. Each of the three realizations consists of a parallel arrangement of cascaded pairs of 1-D filters; hence extensive parallel processing and pipelining can be applied. The three realizations are compared and it is shown that the realization based on the SV and LU decomposition leads to the lowest approximation error and involves the smallest number of multiplications. It is shown that the SVD of the sampled amplitude response of a 2-D digital filter with real coefficients possesses a special structure: every singular vector is either mirror-image symmetric or antisymmetric with respect to its midpoint. Consequently, the SVD method can be applied along with 1-D FIR techniques for the design of linear-phase 2-D filters with arbitrary prescribed amplitude responses which are symmetrical with respect to the origin of the (ω1, ω2) plane. A method for the design of 2-D IIR digital filters based on the combined application of the SVD and the balanced approximation (BA) is proposed. It is shown that the approximation error in the phase angle is bounded by the sum of the neglected Hankel singular values of the filter. Consequently, the phase response of the resulting filter is approximately linear over the passband region provided that only small Hankel singular values are neglected. It is also shown that the resulting 2-D filter is nearly balanced, which implies that the filter has low roundoff noise as well as low parameter sensitivity. Furthermore, the 2-D filter obtained is more economical and computationally more efficient than the original 2-D FIR filter, and in the case where an IIR filter is obtained the stability of the filter is guaranteed. Efficient general algorithms for the evaluation of the 1-D and 2-D gramians for 1-D and 2-D, causal, stable, recursive digital filters are proposed, which facilitate the application of the BA method in the design of digital filters. The algorithms obtained are based on a two-stage extension of the Astrom-Jury-Agniel (AJA) algorithm. It is shown that the AJA algorithm can be modified to solve a 1-D Lyapunov equation in a recursive manner. The recursive algorithm is then extended to the case where the rational function vector involved depends on two complex variables. It is shown that the two algorithms obtained can be combined to evaluate the 2-D gramians. The proposed algorithms are also useful in obtaining optimal digital filter structures that minimize the output-noise power due to the roundoff of products. / Graduate

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The focus of this thesis is on developing theories and techniques to extract lossy microwave filter parameters from data. In the literature, the Cauchy methods have been used to extract filters’ characteristic polynomials from measured scattering parameters. These methods are described and some examples are constructed to test their performance. The results suggest that the Cauchy method does not work well when the Q factors representing the loss of filters are not even. Based on some prototype filters and the relationship between Q factors and the loss, we conduct preliminary studies on alternative representations of the characteristic polynomials. The parameters in these new models are extracted using the Levenberg–Marquardt algorithm to accurately estimate characteristic polynomials and the loss information.

No description available.

A general, non-iterative design technique for low shapefactor, transversal filters is presented. This design approach uses two cosine series to specify appropriate eigenfunctions. An infinite set of such eigenfunctions are defined and the method for choosing the coefficients is discussed. The total filter response is specified as the product of two individual frequency responses. The impulse response of each is then determined by applying the superposition of appropriate eigenfunctions. The criteria for choosing the appropriate eigenfunctions is discussed. A synthesis procedure for designing surface acoustic wave filters is presented. The effects of truncating the impulse response are also explored. A design example is shown for a filter with 10 percent fractional bandwidth and a shapefactor of 1.15.

Fast Recursive Least-Squares Transversal Filters (FTF), an important class of algorithms for adaptive filtering, have the well known problem of numerical instability. Several recent papers have suggested methods to modify the algorithm presented by Cioffi [3] to improve the algorithm’s stability. This paper explores the relationship between precision and stability of the adaptive filter. The effect of changing the adaptive time constant and the filter order are also investigated. These effects are studied for a floating point implementation of the FTF filter that allows for limiting the number of bits used in the mantissa of the results of all additions and multiplications. / Master of Science

This work is a study of a hybrid adaptive controller that blends fixed feedback control and adaptive feedback control techniques. This type of adaptive controller removes the requirement that information about the disturbance is known apriori. Additionally, the control structure is implemented in such a way that as long as the adaptive controller is stable during adaptation, the system consisting of the controller and plant remain stable. The objective is to design and implement an adaptive controller that damps the structural vibrations induced in a multi-modal structure. The adaptive controller utilizes an adaptive infinite impulse response lattice filter for improved damping over the fixed feedback controller alone. An adaptive finite impulse response LMS filter is also implemented for comparison of the ability for both algorithms to reject harmonic, narrow bandwidth and wide bandwidth disturbances. It is demonstrated that the lattice filter algorithm performs slightly better than the LMS filter algorithm in all three disturbance cases. The lattice filter also requires less than half the order of the LMS filter to get the same performance. / Master of Science

The objective of this thesis is to introduce novel procedures and guidelines to design bandstop microwave filters for modern terrestrial and satellite wireless communication systems. Among all available microwave filter technologies, planar structures of microstrip and substrate integrated waveguide (SIW) are chosen, due to ease of fabrication, low profile, weight and manufacturing cost. Particularly, SIW structures are more attractive because they have a better insertion loss, quality factor, and power handling capability in comparison to their microstrip counterparts, and can also be easily integrated into other planar circuitries. A comprehensive hybrid analytic-optimization method is developed to synthesize any single-band as well as multi-band bandstop coupling matrix. In this method, the location of reflection zeros (RZs) and the attenuations in stopbands can be determined in advance. Several novel single-band, dual-band, and triple-band bandstop filters are designed using regular and ridged SIW resonators, in-line coupled singlet resonators, cross-coupled resonators, and bandstop stubs. The designed filters have fractional bandwidths up to 23% . Moreover, a tunable ridged SIW bandstop resonator and a tunable CPW resonator, etched into the top plate of the SIW transmission line, are introduced. Combining these two resonators, a dual-band SIW filter is designed that permits one of its stopband to be tuned while another stopband is fixed. All introduced filters in this thesis are verified by commercial electromagnetic software, analytic investigations using Matlab codes, and measurements. / Graduate