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  • 1
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    Circuits, systems and signal processing 13 (1994), S. 273-293 
    ISSN: 1531-5878
    Source: Springer Online Journal Archives 1860-2000
    Topics: Electrical Engineering, Measurement and Control Technology
    Notes: Abstract A basic control engineer's adage-the poles of a feedback compensator become zeros of the closed-loop system-admits difficulties of interpretation even in the most simple of cases; that of single-input, single-output. An earlier investigation has provided an analysis of this adage in a module-theoretic context for systems in state space form while avoiding restrictive assumptions on system minimality or squareness. The main result is expressed concisely in terms of an exact sequence of modules which include Ω-zero modules corresponding to the feedback system and the plant. Extended zero modules of Ω-type incorporate both finite invariant zero structure, and generic zero information which occurs when a system fails to be right-invertible. In the case of compensation in the feedback path, this main exact sequence reduces to a mathematically clear expression of the aforementioned adage: the Ω-zero module of the feedback system is precisely the direct sum of the Ω-zero module of the plant and the system pole module of the feedback compensator. This paper extends the previous work in order to avoid assumptions on causality in the plant. Implicit dynamical systems are employed, in lieu of systems in state space form. Once again, it is not assumed that the system is one-to-one or onto; and so the concepts of generic zeros and their modules are brought into the arena of implicit systems. The implicit system itself is assumed in this work to be regular; however, decoupling zeros are permitted. Moreover, input-decoupling zeros and system pole feedback relationships are considered.
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  • 2
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    Circuits, systems and signal processing 13 (1994), S. 387-388 
    ISSN: 1531-5878
    Source: Springer Online Journal Archives 1860-2000
    Topics: Electrical Engineering, Measurement and Control Technology
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  • 3
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    Circuits, systems and signal processing 13 (1994), S. 373-384 
    ISSN: 1531-5878
    Source: Springer Online Journal Archives 1860-2000
    Topics: Electrical Engineering, Measurement and Control Technology
    Notes: Abstract When the problem is considered of obtaining a periodic description in state-space form of a linear process which can be modelled by linear difference equations with periodic coefficients, it is natural to ask whether it is possible to preliminarily derive a polynomial equivalent form of such equations, which in the periodic case plays a role similar to the Rosenbrock's polynomial matrix description of a linear time-invariant process. In this paper a polynomial time-invariant description of a linear periodic process is introduced. It is shown that such a polynomial description gives a simple characterization of the dimension of the space of the solutions corresponding to the null input function, i.e., of the order of the periodic model under consideration. In addition, it allows us to introduce a transfer matrix for the computation of the output responses corresponding to null initial conditions, and to deduce conditions for the periodic model to be causal. These results, as well as the possibility of defining strict system equivalence between two periodic models through their time-invariant polynomial descriptions, in a similar sense as in the time-invariant case, show the relevance of such a polynomial time-invariant description for the problem under consideration.
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  • 4
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    Circuits, systems and signal processing 13 (1994), S. 435-453 
    ISSN: 1531-5878
    Source: Springer Online Journal Archives 1860-2000
    Topics: Electrical Engineering, Measurement and Control Technology
    Notes: Abstract An actual sampling process can be modeled as a random process, which consists of the regular (uniform) deterministic sampling process plus an error in the sampling times which constitutes a zero-mean noise (the jitter). In this paper we discuss the problem of estimating the jitter process. By assuming that the jitter process is an i.i.d. one, with standard deviation that is small compared to the regular sampling time, we show that the variance of the jitter process can be estimated from thenth order spectrum of the sampled data,n=2, 3, i.e., the jitter variance can be extracted from the 2nd-order spectrum or the 3rd-order spectrum (the bispectrum) of the sampled data, provided the continuous signal spectrum is known. However when the signal skewness exceeds a certain level, the potential performance of the bispectrum-based estimation is better than that of the spectrum-based estimation. Moreover, the former can also provide jitter variance estimates when the continuous signal spectrum is unknown while the latter cannot. This suggests that the bispectrum of the sampled data is potentially better for estimating any parameter of the sampling jitter process, once the signal skewness is sufficiently large.
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  • 5
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    Circuits, systems and signal processing 14 (1995), S. 57-67 
    ISSN: 1531-5878
    Source: Springer Online Journal Archives 1860-2000
    Topics: Electrical Engineering, Measurement and Control Technology
    Notes: Abstract The paper presents a simple method for solving the optimal (LQG) control problem on an infinite time horizon for linear plants described by proper rational transfer function matrices. Since the class of proper plants discussed in the paper is more general than the one commonly used, additional properties of solutions are presented. Two numerical examples illustrate the theoretical considerations. The examples have been performed on an IBM PC using the proposed algorithm. This algorithm is a part of a public ASMCS package developed by the author for analysis, synthesis, and simulations of multi-input, multi-output (MIMO) control systems.
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  • 6
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    Circuits, systems and signal processing 14 (1995), S. 135-143 
    ISSN: 1531-5878
    Source: Springer Online Journal Archives 1860-2000
    Topics: Electrical Engineering, Measurement and Control Technology
    Notes: Abstract We study the behavior of a Hilbert network (i.e., a finite or countably infinite network whose variables are in a Hilbert space and in which the associated total energy is finite) whose elements are affected by perturbations. More specifically, we will give estimates for a change of the current distribution caused by (a) perturbations of the elements of the nominal network when the voltage sources are fixed, and (b) a change of voltage sources in a network whose elements are perturbed. The conditions given in our theorems imply insensitivity and robust stability of the nominal network. The applications of the results are illustrated by an example of an infinite network.
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  • 7
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    Circuits, systems and signal processing 14 (1995), S. 473-494 
    ISSN: 1531-5878
    Source: Springer Online Journal Archives 1860-2000
    Topics: Electrical Engineering, Measurement and Control Technology
    Notes: Abstract A new lattice filter structure to model two-dimensional (2-D) autoregressive (AR) fields is proposed. The proposed structure utilizes and extracts the information contained in the backward prediction error fields and their delayed versions. The main idea is to use two sets of reflection coefficients corresponding to two quadrant filters and to increase the number of reflection coefficients with the order of the lattice filter. Increasing the number of reflection coefficients at each stage produces a sufficient number of independent parameters to model AR fields up to order three, which is an improvement over the existing 2-D lattice filter structures. The improvement is confirmed by computer simulations. In addition, a relationship between the reflection coefficients and the AR coefficients is derived. It is also shown that the entropy contained in the backward prediction error field vector of the proposed structure is closer to the input entropy when compared to those contained in existing 2-D lattice filters.
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  • 8
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    Circuits, systems and signal processing 14 (1995), S. 555-561 
    ISSN: 1531-5878
    Source: Springer Online Journal Archives 1860-2000
    Topics: Electrical Engineering, Measurement and Control Technology
    Notes: Abstract This paper addresses the problem of global asymptotic stability in recursive first hyperquadrant causalm-D digital filters. A set of simple-to-check 1-D conditions necessary for stability of them-D system is given. Generalizations tov-dimensional (v〈m) subsystems are also provided. These derived conditions are useful in determining asymptotic convergence ofm-D digital filters implemented in fixed or floating point arithmetic. The set of 1-D conditions can be considered the analogous result to practical bounded input bounded output (BIBO) stability for linearm-D systems.
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  • 9
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    Circuits, systems and signal processing 14 (1995), S. 633-637 
    ISSN: 1531-5878
    Source: Springer Online Journal Archives 1860-2000
    Topics: Electrical Engineering, Measurement and Control Technology
    Notes: Abstract A new efficient algorithm for calculating the weighting coefficients of maximally linear, FIR digital differentiators is presented. Simple closed-form explicit and recursive formulas are derived in a very straightforward manner. Moreover, a simple recursive equation is established, relating coefficients of two digital differentiators of adjacent ranks.
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  • 10
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    Circuits, systems and signal processing 14 (1995), S. 661-667 
    ISSN: 1531-5878
    Source: Springer Online Journal Archives 1860-2000
    Topics: Electrical Engineering, Measurement and Control Technology
    Notes: Abstract This paper presents a sequence of pseudorandom arrays with triangular symmetry. The sequence of arrays is also pseudorandom in nature, so it can be called a pseudorandom sequence of arrays, or PRSA. A circuit for the PRSA generator is given and some interesting properties of this type of PRSA are discussed. Also, some of the concepts presented in this paper are clarified with an example.
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  • 11
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    Circuits, systems and signal processing 13 (1994), S. 19-30 
    ISSN: 1531-5878
    Source: Springer Online Journal Archives 1860-2000
    Topics: Electrical Engineering, Measurement and Control Technology
    Notes: Abstract A linear-quadratic (LQ) control problem subject to a standard continuous-time system is called regular if the input weighting matrix is invertible, and singular if this is not the case. Consequently, optimal inputs for regular LQ problems are ordinary functions (state feedbacks), whereas optical controls for singular problems are in general distributions, e.g., impulses. We will show that regularity and singularity in LQ problems subject to ageneral (implicit) system depends not so much on the input weighting matrix, as on the property that the integrand of the cost criterion is a function only if inputs and state trajectories are, as is the case for LQ problems, subject to astandard system. In particular, we will provide a simple criterion for distinguishing between regularity and singularity in LQ problems subject to a general system. Our criterion is expressed in the system coefficients only and reduces to the classical one if the underlying system is standard.
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  • 12
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    Circuits, systems and signal processing 13 (1994), S. 119-119 
    ISSN: 1531-5878
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    Topics: Electrical Engineering, Measurement and Control Technology
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  • 13
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    Circuits, systems and signal processing 13 (1994), S. 185-199 
    ISSN: 1531-5878
    Source: Springer Online Journal Archives 1860-2000
    Topics: Electrical Engineering, Measurement and Control Technology
    Notes: Abstract Completions of linear time varying singular systems of the formE(t)x′(t)+F(t)x′(t)=f(t) are explicitly computed using recent results on rational matrix functions. The algorithm and the theory behind it are carefully described. Computational issues are discussed.
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  • 14
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    Circuits, systems and signal processing 13 (1994), S. 225-239 
    ISSN: 1531-5878
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    Topics: Electrical Engineering, Measurement and Control Technology
    Notes: Abstract In this paper a spectral method using orthogonal periodic basis functions for the analysis of linear time invariant descriptor systems is discussed, and the case of the trigonometric Fourier functions is investigated in detail. The method is shown to be convergent, in the distributional sense. However, for any finite number of basis functions, the periodicity induced by the chosen basis can give rise to spurious impulsive components in the computed system response, even in the case of correct initial conditions.
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  • 15
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    Circuits, systems and signal processing 13 (1994), S. 295-308 
    ISSN: 1531-5878
    Source: Springer Online Journal Archives 1860-2000
    Topics: Electrical Engineering, Measurement and Control Technology
    Notes: Abstract In this paper we will study topological properties of the class of proper and improperp×m transfer functions of a fixed McMillan degreen. A natural generalization of this class is all autoregressive systems of degreen under external system equivalence. The subset of irreducible systems has in a natural way the structure of a manifold and we show how to extend this topology to the set of all autoregressive systems of degree at mostn. We will describe the subset of systems with fixed Kronecker indicesv=(v1,...,v p ) as an orbit space, which will enable us to calculate the topological dimension for each collection of indicesv. Finally, we will describe the topological closure of those sets in the space of all autoregressive systems.
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  • 16
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    Circuits, systems and signal processing 13 (1994), S. 349-359 
    ISSN: 1531-5878
    Source: Springer Online Journal Archives 1860-2000
    Topics: Electrical Engineering, Measurement and Control Technology
    Notes: Abstract It has been shown in [B.M.90] that non-square implicit differential equations allow for the description of variable structure systems (variable order, variable sign, variable parameters). We combine here the possible control strategy developed in [L.91] for rectangular systems (insuring a unique output behavior for the system compensated with a proportional or proportional and derivative state feedback) with the detector and the observer introduced in [B.M.90] in order to obtain a closed-loop system where the initial structure variation disappears on the output. We also give necessary and sufficient conditions for the free assignment of the associated output dynamics.
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  • 17
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    Circuits, systems and signal processing 13 (1994), S. 391-402 
    ISSN: 1531-5878
    Source: Springer Online Journal Archives 1860-2000
    Topics: Electrical Engineering, Measurement and Control Technology
    Notes: Abstract An exposition of joint cumulants and cumulant spectra is presented. A distinction is emphasized in this paper between the cumulant spectrum of a time series and its stationary version, here called apolyspectrum. The variance and covariance of the sample bispectrum is then derived using a relationship between cumulant spectra of the finite Fourier transform for the 2nd and 4th cumulant function, and the bispectrum and trispectrum of the time series.
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  • 18
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    Circuits, systems and signal processing 13 (1994), S. 467-479 
    ISSN: 1531-5878
    Source: Springer Online Journal Archives 1860-2000
    Topics: Electrical Engineering, Measurement and Control Technology
    Notes: Abstract The detection of a general class of transient (i.e., finite energy) signals in additive stationary interference using the spectral correlation function (second order cumulant spectrum) is presented. Observable features in the two-dimensional spectral correlation function due to properties of signals in the assumed class of transients are exploited to derive a detection statistic. The performance of the proposed detection statistic relative to a conventional power spectral detector is presented.
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  • 19
    ISSN: 1531-5878
    Keywords: Statistics: Nonparametric time series estimation for pattern analysis ; Industries: Health monitoring and durability of rotating machinery ; Reliability: Incipient failure inspection/quality control/system safety
    Source: Springer Online Journal Archives 1860-2000
    Topics: Electrical Engineering, Measurement and Control Technology
    Notes: Abstract Vibroacoustic signals of rotating machinery are composed of sums of modulated periodicities, broadband random components, and occasionally a set of transient responses. These signals are not ergodic as the modulated periodicities are partially coherent. Progressive wear of the rotating machine causes the nonlinear structure of the received signal to intensify, and nonlinearity results in transfer of energy between harmonics of the signal's periodic components. Statistics developed from bispectrum and second-order cumulant spectrum estimates of the measured signal are combined with power spectrum amplitudes as feature inputs for standard multivariate classifiers. The higher-order statistics measure, respectively, the extent of nonlinearity and intermodulation of the received signal. Classification results of simulated and actual incipient wear data collected from a controlled experiment drilling circuit boards illustrate the potential of this novel statistical signal processing approach.
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  • 20
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    Circuits, systems and signal processing 14 (1995), S. 1-16 
    ISSN: 1531-5878
    Source: Springer Online Journal Archives 1860-2000
    Topics: Electrical Engineering, Measurement and Control Technology
    Notes: Abstract In this paper, we present a new class of two-dimensional FIR-median hybrid (FMH) filters, which we call separable FMH filters, for image noise smoothing. In a separable FMH filter, a one-dimensional FMH filter is applied to the rows and the columns of an image successively. The deterministic properties of separable and cross window FMH filters are discussed. Under certain assumptions, it is proved that a root of a separable FMH filter is a root of the corresponding cross window FMH filter. The noise attenuating properties of a separable FMH filter are studied and compared with those of the separable median filter, the two-dimensional median filter with a square window, and the cross window FMH filter.
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  • 21
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    Circuits, systems and signal processing 14 (1995), S. 69-85 
    ISSN: 1531-5878
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    Topics: Electrical Engineering, Measurement and Control Technology
    Notes: Abstract The Chinese remainder theorem is a fundamental technique widely employed in digital signal processing for designing fast algorithms for computing convolutions. Classically, it has two versions. One is over a ring of integers and the second is over a ring of polynomials with coefficients defined over a field. In our previous papers, we developed an extension to this well-known theorem for the case of a ring of polynomials with coefficients defined over a finite ring of integers. The objective was to generalize number-theoretictransforms, which turn out to be a special case of this extension. This paper focuses on the extension of the Chinese remainder theorem for processing complex-valued integer sequences. Once again, the present work generalizes the complex-number-theoretic transforms. The impetus for this work is provided by the occurrence of complex integer sequences in digital signal processing and the desire to process them using exact arithmetic.
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    Circuits, systems and signal processing 14 (1995), S. 145-166 
    ISSN: 1531-5878
    Source: Springer Online Journal Archives 1860-2000
    Topics: Electrical Engineering, Measurement and Control Technology
    Notes: Abstract A method to improve the stability of the forced response of a recursive digital filter with nonlinearities due to finite wordlength is presented. An analysis of applying a stability criterion on the nonlinear filter is performed by transforming the problem into a mathematical problem of finding a zero set. As a result of the mathematical analysis, one can find a maximal sector size that bounds an equivalent nonlinearity, so that stability is ensured. This sector size is influenced by the choice of the desired nonlinearity characteristic appropriate to the problem, and the amount of the input scaling needed. Another improvement of the filter's stability is obtained by adding a feedback gain, the value of which is determined by using the zero set-finding method. Simulation results showing the improved stability properties of a filter designed by this method are presented.
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    Circuits, systems and signal processing 14 (1995), S. 285-298 
    ISSN: 1531-5878
    Source: Springer Online Journal Archives 1860-2000
    Topics: Electrical Engineering, Measurement and Control Technology
    Notes: Abstract In this paper a fully systolic, bit level, three-port unconstrained adaptor, which constitutes the main nontrivial building block of ladder wave digital filters, is generated. The one-dimensional binary convolution is used as the underlying algorithm for the implementation of a multiplication. The Isb-first input data organization approach is adopted and thecanonical mapping methodology is used to fully systolize the unconstrained parallel three-port adaptor at the bit level with piplining period a=1. The technique is based on a transformation of the adaptor's signal-flow graph, so that unidirectional data flow takes place. A ring-systolic scheme is proposed for implementing communications among adaptors.
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    Circuits, systems and signal processing 14 (1995), S. 317-349 
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    Topics: Electrical Engineering, Measurement and Control Technology
    Notes: Abstract We consider the problem of robust stochastic adaptive control of not necessarily minimum phase systems in the presence of unmodelled dynamics. Stochastic gradient algorithms with parameter projection and modified gain sequence are used for the estimation of the unknown controller parameters. Global stability of the adaptive system is achieved without requiring the strictly positive real condition and the persistency exciting condition to be satisfied.
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    Circuits, systems and signal processing 14 (1995), S. 427-443 
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    Topics: Electrical Engineering, Measurement and Control Technology
    Notes: Abstract A theoretical framework for the investigation of the qualitative behavior of differential-algebraic equations (DAEs) near an equilibrium point is established. The key notion of our approach is the notion of regularity. A DAE is called regular locally around an equilibrium point if there is a unique vector field such that the solutions of the DAE and the vector field are in one-to-one correspondence in a neighborhood of this equilibrium point. Sufficient conditions for the regularity of an equilibrium point are stated. This in turn allows us to translate several local results, as formulated for vector fields, to DAEs that are regular locally around a given equilibrium point (e.g. Local Stable and Unstable Manifold Theorem, Hopf theorem). It is important that these theorems are stated in terms of the given problem and not in terms of the corresponding vector field.
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    Circuits, systems and signal processing 14 (1995), S. 495-524 
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    Topics: Electrical Engineering, Measurement and Control Technology
    Notes: Abstract The classical notion of the λ-generalized nullspace, defined on a matrixA εR n×n,where λ is an eigenvalue, is extended to the case of ordered pairs of matrices(F, G), F, G ε R m×nwhere the associated pencilsF − G is right regular. It is shown that for every α εC ∪ {∞} generalized eigenvalue of (F, G), an ascending nested sequence of spaces {P α i ,i=1, 2,...} and a descending nested sequence of spaces {ie495-02 i=1, 2,...} are defined from the α-Toeplitz matrices of (F, G); the first sequence has a maximal elementM α * , the α-generalized nullspace of (F, G), which is the element of the sequence corresponding to the index τα, the α-index of annihilation of (F, G), whereas the second sequence has the first elementP α * as its maximal element, the α-prime space of (F, G). The geometric properties of the {M α i ,i=1, 2,...,τα and {P α i ,i=1, 2,...sets, as well as their interrelations are investigated and are shown to be intimately related to the existence of nested basis matrices of the nullspaces of the α-Toeplitz matrices of (F, G). These nested basis matrices characterize completely the geometry ofM α * and provide a systematic procedure for the selection of maximal length linearly independent vector chains characterizing theα-Segre characteristic of (F, G).
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    Circuits, systems and signal processing 14 (1995), S. 563-586 
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    Topics: Electrical Engineering, Measurement and Control Technology
    Notes: Abstract The robustness problem of stability for large-scale uncertain systems with a class of multiple time delays is addressed in this paper. By applying the complex Lyapunov stability theorem, the matrix measure techniques, and norm inequalities, a new approach for solving a general case of the above problem is proposed. Several robust stability conditions, delay-dependent or delay-independent, are derived to guarantee the asymptotic stability and exponential stability of the uncertain large-scale time-delay systems. Moreover, these obtained results can also be applied to the stabilization design.
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    Circuits, systems and signal processing 14 (1995), S. 615-632 
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    Topics: Electrical Engineering, Measurement and Control Technology
    Notes: Abstract Successful speech recognition is highly dependent on appropriate speech segmentation. The poor efficiency of the sequential detection of abrupt changes in the signals with relatively short stationary intervals, as is the case with speech signals, can be improved by the off-line maximum likelihood segmentation algorithm. In this paper the new segmentation algorithm is presented. For the a priori known number of segments, the algorithm determines such signal partitions for which the sum of segment distortion is minimal. The generalized maximum likelihood distortion measure has been introduced, and has proven to be particularly efficient on short signal segments. In the case of an unknown number of segments, its estimate is obtained comparing the reduction of the distortion. The asymptotic properties of the distortion sequence have been analyzed, which led to the definition of the presented segmentation algorithm. The introduced measure can be applied both to the AR and ARMA models. The segmentation algorithm is verified on test signals as well as on the natural speech signal, for which the pitch synchronous framing scheme is applied. The experimental results also include a comparison of the AR and ARMA model-based segmentations. The first results show that ARMA model-based segmentation gives somewhat better results than the AR model algorithm.
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    Circuits, systems and signal processing 14 (1995), S. 639-651 
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    Source: Springer Online Journal Archives 1860-2000
    Topics: Electrical Engineering, Measurement and Control Technology
    Notes: Abstract A new bit-serial architecture for implementation of high order FIR filters is introduced, as well as example FPGA and CMOS realizations. This structure exploits the simplicity of coefficients that consist of two power-of-two terms to yield efficient implementations. Quantization effects are discussed and a simple block scaling method for reducing rounding and truncation noise in high order filters is also presented.
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    Circuits, systems and signal processing 13 (1994), S. 255-272 
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    Topics: Electrical Engineering, Measurement and Control Technology
    Notes: Abstract A geometric interpretation of the Lewis Structure Algorithm (LSA) is given in terms of precise projection maps directly defined from the (E, A, B, C) maps of the system. An extended version of LSA is offered which, in addition to this geometric information, also provides in a direct way, and within the same (E, A, B, C) class of models, a left inverse (if any) of the system. An example is given.
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    Circuits, systems and signal processing 13 (1994), S. 329-345 
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    Topics: Electrical Engineering, Measurement and Control Technology
    Notes: Abstract We consider the problem of control of linear, time-invariant, multivariable descriptor (implicit) systems. In particular we examine the effectiveness of an algorithm (which is a generalization of previous work in state space systems) for the design of an output feedback control giving pole placement in such systems. Conditions are presented which ensure that the algorithm produces the required control. We also address the important issue of uniqueness of solutions.
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    Circuits, systems and signal processing 13 (1994), S. 389-390 
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    Circuits, systems and signal processing 13 (1994), S. 455-466 
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    Notes: Abstract Detecting active sonar returns in multipath media is a central underwater signal processing problem. This paper studies a new approach to active sonar detection using bispectral analysis of sonar data. Its sensitivity to non-stationarities is used to develop a threshold detector that can be applied to broad classes of signals and noise. Precise statistical descriptions of the underwater medium and noise are not required. Theoretical analyses predicting its performance as a function of signal-to-noise ratio and time-bandwidth product are presented. Computer simulation experiments verify the results and show that its performance compares favorably to that of conventional detectors.
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    Circuits, systems and signal processing 13 (1994), S. 481-496 
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    Notes: Abstract The noise suppression capability of higher-order moments and spectra has made them attractive when the goal is to extract or reconstruct a signal that is contaminated by multivariate Gaussian noise or certain types of non-Gaussian noise. Two new detectors, one centralized and one distributed, which are based on the third-order moment of the data are proposed. The asymptotic performance of the centralized detector and the asymptotic distribution of the components of the distributed detector are analyzed. Further, the performance of these detectors is simulated and compared to that of the matched filter for three different types of interference: Gaussian noise, Gaussian noise corrupted by a sinusoid with random phase, and Arctic under-ice noise.
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    Circuits, systems and signal processing 14 (1995), S. 39-55 
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    Notes: Abstract This paper describes an implementation of 2-D FIR and IIR linear digital filters via VLSI array processors. The underlying realization structures are based on the matrix decomposition approach. The 2-D concurrent processing is used in order to implement the row and column delays within the cycle time. A high degree of concurrency is achieved by exploiting the pipelining of the array processors with the inherent parallelism of the matrix decomposition structure. The resulting structures are modular, and regular, use only local communication and internal local feedback loops, and achieve high throughput and sampling rates.
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    Circuits, systems and signal processing 14 (1995), S. 111-134 
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    Notes: Abstract This article studies dynamical systems under perturbations. We prove an Equilibrium Equivalence Theorem that guarantees that the dynamics of the system remains unchanged under perturbations with certain fairly general assumptions. We also prove that a power system is robust with respect to parameter changes in generic situations. Concepts of equilibrium equivalence and equilibrium equivalence structural stability are developed and are applied to studies of bifurcations of vector fields on noncompact manifolds. A constructive approach to equilibrium equivalence structural stability verification is emphasized. General results on structural stability of vector fields on differential manifolds are established and important applications of this theory to stability analysis are considered.
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    Circuits, systems and signal processing 14 (1995), S. 167-185 
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    Notes: Abstract The problem of designing a stabilizing compensator for a control system to achieve prescribed initial value constraintsυ (i)(0+)=yi is considered. Indeed, modulo certain technical conditions, such a compensator exists if and only if yi=0;i= 0,1,...,r〈p〉 +r〈t〉 −2; wherer〈p〉 is the relative degree of the plant andr〈t〉 is the relative degree of the system input. This theorem is derived and a complete parameterization of the set of compensators that achieve the prescribed design constraints is formulated.
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    Circuits, systems and signal processing 14 (1995), S. 255-278 
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    Notes: Abstract Wideband source location in array signal processing has received much attention in the literature lately. Methods such as the Coherent Signal Subspace (CSS) method proposed by Wang and Kaveh [12], and the signal subspace method used by Cadzow [3], are typical of the approaches used to tackle the multiple wideband source location problem. Most of these methods are variations of the narrowband high-resolution methods. Grenier [5], on the other hand, has applied the idea of time-dependent Auto-Regressive (AR) modeling [7] for a nonstationary process to the frequency domain AR modeling of the sensor outputs in a linear array and has been able to produce good results for a wideband signal. The AR coefficients in the model are expanded in a set of frequency-dependent basis functions. The choice of the basis functions was deemed immaterial and the method works even when only one snapshot of the array output is available. In this paper, we re-examine this method and present an extension of the frequency-dependent AR modeling approach to a planar array. It is shown that the use of a set of sinc functions for representing the frequency-dependent AR coefficients accurately tracks their evolution in the frequency domain, and gives superior performance compared to that when power or Legendre functions are used. We also propose two methods for smoothing the spatial spectra, from which the source locations are determined. Comparison with the CSS method are also presented.
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    Circuits, systems and signal processing 13 (1994), S. 361-372 
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    Notes: Abstract We investigate the controller design problem for linear systems in which the state and the controls are subject to static linear constraints. We give necessary and sufficient conditions for the existence, and present a complete parametrization of all stabilizing controllers. This parametrization allows us to transform the constrained control problem into a standard problem which can be solved using usualH 2 orH ∞ optimization methods. The approach is illustrated by a simple numerical example showing the various steps of the proposed algorithm.
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    Circuits, systems and signal processing 13 (1994), S. 403-410 
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    Notes: Abstract A continuous stationary signal possessing non-Gaussian higher order statistics cannot be correctly modelled by any discrete process based on passing independently and identically distributed noise through a linear filter. In particular, it is shown that at third order there exists no discrete skewed linear model with a discrete bispectrum that is the same as that obtained from the Nyquist samples of any continuous stationary process. The nature of the problem is elucidated and an alternative method for modelling the third order statistics of continuous stationary processes is proposed.
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    Circuits, systems and signal processing 14 (1995), S. 17-38 
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    Notes: Abstract Centralized methods for source location using sensor arrays have computational and communication burdens that increase significantly with the number of sensors in the array. Therefore, these methods may not be usable in the applications involving very large arrays. In such applications, the data processing may need to be decentralized. This paper introduces two methods for decentralized array processing, based on the recently proposed MODE algorithm. For prescribed nonoverlapping subarrays, both methods are shown to be statistically optimal in the sense that asymptotically they provide the most accurate decentralized estimates of source location parameters. The problem of subarray selection to further optimize the estimation accuracy is only briefly addressed. The two methods are intended for different types of applications: the first should be preferred when there exist significant possibilities for local processing or for parallel computation in the central processor; otherwise the second method should be preferred. The accuracy of the two decentralized methods is compared to the centralized Cramér-Rao bound, both analytically and numerically, in order to provide indications about the loss of accuracy associated with decentralized processing.
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    Circuits, systems and signal processing 14 (1995), S. 87-110 
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    Notes: Abstract We present in this paper a recursive-in-order least-squares (LS) algorithm to compute efficiently the parameters of a 2-D Gaussian Markov random field (GMRF) model. The algorithm is based on the fact that the least-squares estimation of the parameters of a 2-D noncausal GMRF model is consistent and the coefficient matrix in the normal equation has near-to-block-Toeplitz structure. Hence, it has a Levinson-like form for the updating of model parameters by introducing auxiliary variables. Moreover, this paper proposes the concept ofrecursive path for 2-D recursive-in-order algorithms, and points out that there exists a tradeoff between fast computation of the parameters and accurate choice of model support; a compromise recursive path is then suggested where the orders change alternately in two directions. The computational complexity of the developed algorithm is analyzed, and the results show that the algorithm is more efficient when either the image size or the model support is larger. It is found that the total number of multiplications (mps) involved in the new algorithm is only about 14% of that in the conventional LS method when the image size is 512 × 512 and the neighbor set of the model is a 17 × 17 window. Computer simulation results using the recursive-in-order algorithm developed in this paper and the conventional LS method are given to verify the correctness of the new algorithm.
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    Circuits, systems and signal processing 14 (1995), S. 539-553 
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    Notes: Abstract Cross terms are an inherent consequence of the second order nature of Cohen's class TFDs (Time-Frequency Distributions) [5], [6]. They are manifest in a TFD of multicomponent signals as spurious artifacts arising from interactions between the various signal components, and they can often appear at times and/or frequencies inconsistent with the underlying physical nature of the signal, causing misinterpretation [2], [3], [4]. There are many time frequency distributions that avoid the cross term effect; the best are the Choi-Williams ED (Exponential Distribution) [1] and Levin's IPS (Instantaneous Power Spectrum) [9]. In this paper we combine the cross term reducing philosophy of the ED and IPS to obtain a new TFD that most effectively reduces the cross term effect. Surprisingly, the new TFD also satisfies most desired TFD properties.
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    Circuits, systems and signal processing 18 (1999), S. 27-42 
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    Notes: Abstract Of concern is the propagation of distortionless surface waves in a medium that may be nonuniform relative to depth. Distortionless wave propagation in inhomogeneous media was discussed by V. Burke, R. J. Duffin and D. Hazony, inQuart. Appl. Math., 183–194 (1976). Accordingly, the media could be modeled by a distributed electrical ladder network, nonuniform along the axis. We give a two-dimensional development based on Hooke's law and Newton's law which leads to the well-known case of Rayleigh waves in homogeneous media. It will be seen that the available pool of propagation modes greatly increases when high-pass propagation is included. The emphasis is on media where the elastic coefficients track one another as a function of depth. Special cases are studied in detail showing that as a disturbance travels along the surface, it may assume a broadband phase change, which translates into a shape distortion in the time domain, which is periodic with distance. Applications may be found in acousto-optics, in situ monitoring of elongated bodies, high-frequency SAW filters, microstrips, and any situations where surface waves are used in an environment of high precision or relatively large distances.
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    Circuits, systems and signal processing 18 (1999), S. 131-147 
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    Notes: Abstract This study presents a linear output-based controller for stabilizing a rigid-link flexible-joint electrically driven (RLFJED) robot manipulator. The proposed controller ensures local exponential stability under some uncertainty conditions. It is assumed that the velocity signals from the link side are not measurable. The controller is analyzed by using tools for pole placement by an output-feedback in the framework of the linear system theory. Some useful structural properties of the systems under consideration have been studied. Applications of the results to the set-point regulation control problem are considered.
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    Circuits, systems and signal processing 18 (1999), S. 205-223 
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    Notes: Abstract In this paper we consider an adaptive controller with vanishing gain and excitation of the reference signal. We use the burst recovery concept to show that all signals in the adaptive loop remain uniformly bounded. We also show that the mean-square performance converges so that the adaptive system is optimal in the sense that the parameter estimation error and the one-step ahead prediction error are uncorrelated in the mean despite the presence of the unmodeled dynamics.
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    Circuits, systems and signal processing 18 (1999), S. 191-204 
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    Notes: Abstract Finite homogeneous Markov chains ξ, which admit invariant probability distributions, can be defined by the cycloids { $$\bar C_k $$ } (closed polygonal lines whose consecutive edges have various orientations that do not necessarily determine a common direction for $$\bar C_k $$ ) occurring in their graphs. These Markov chains are called cycloid chains, and the corresponding finite-dimensional distributions are linear expressions on the cycloids { $$\bar C_k $$ } with the real coefficients αk. Then the collection {{ $$\bar C_k $$ }, {αk}}, called the cycloid decomposition of ξ, gives a minimal description of the finite-dimensional distributions that, except for a choice of the maximal tree, uniquely determines the chain ξ. Furthermore, the cycloid decompositions have an interpretation in terms of the transition probability functions expressing the same essence as the known Chapman-Kolmogorov equations.
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    Circuits, systems and signal processing 18 (1999), S. 241-267 
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    Notes: Abstract We study solutions of the “linear system in a saturated mode” $$\begin{array}{*{20}c} {(M)} & {x' \in Tx + c - \partial I_{D^n } x.} \\ \end{array} $$ We show that a trajectory is in a constant face of the cubeD n on some interval (0,d]. We answer a question about comparing the two systems: (M) and $$\begin{array}{*{20}c} {(H)} & {\begin{array}{*{20}c} {Cu' = T\upsilon + c - R^{ - 1} u,} & {\upsilon = G(\lambda } \\ \end{array} u)} \\ \end{array} $$ . As λ→∞, limits ofv corresponding to asymptotically stable equilibrium points of (H) are asymptotically stable equilibrium points of (M), and the converse is also true. We study the assumptions to see which are required and which may be weakened.
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    Circuits, systems and signal processing 18 (1999), S. 291-314 
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    Notes: Abstract In this paper we introduce a new computational method for solving the diffusion equation. In particular, we construct a “generalized” state-space system and compute the impulse response of an equivalent truncated state-space system. In this effort, we use a 3D finite element method (FEM) to obtain the state-space system. We then use the Arnoldi iteration to approximate the state impulse response by projecting on the dominant controllable subspace. The idea exploited here is the approximation of the impulse response of the linear system. We study the homogeneous and heterogeneous cases and discuss the approximation error. Finally, we compare our computational results to our experimental setup.
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    Circuits, systems and signal processing 18 (1999), S. 351-364 
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    Notes: Abstract A simple state-space approach for the four-block singular nonlinearH ∞ control problem is proposed in this paper. This approach combines a (J, J′)-lossless and a class of conjugate (J, J′)-expansive systems to yield a family of nonlinearH ∞ output feedback controllers. The singular nonlinearH ∞ control problem is thus transformed into a simple lossless network problem that is easy to deal with in a network-theory context.
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    Circuits, systems and signal processing 18 (1999), S. 395-406 
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    Notes: Abstract The stability of time-varying autoregressive (TVAR) models is an important issue in many applications such as time-varying spectral estimation, EEG simulation and analysis, and time-varying linear prediction coding (TVLPC). For stationary AR models there are methods that guarantee stability, but the for nonadaptive time-varying approaches there are no such methods. On the other hand, in some situations, such as in EEG analysis, the models that temporarily exhibit roots with almost unit moduli are difficult to use. Thus we may need a tighter stability condition such as stability with margin 1−ϱ. In this paper we propose a method for the estimation of TVAR models that guarantees stability with margin 1−ϱ, that is, the moduli of the roots of the time-varying characteristic polynomial are less than or equal to some arbitrary positive number ϱ for every time instant. The model class is the Subba Rao-Liporace class, in which the time-varying coefficients are constrained to a subspace of the coefficient time evolutions. The method is based on sequential linearization of the associated nonlinear constraints and the subsequent use of a Gauss-Newton-type algorithm. The method is also applied to a simulated autoregressive process.
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    Circuits, systems and signal processing 18 (1999), S. 443-443 
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    Circuits, systems and signal processing 18 (1999), S. 17-25 
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    Notes: Abstract A complete analytic characterization and solution construction (done either explicitly or by recursion) for the minimax control problem using optimal rate feedback is given for the case when the plant consists of a known fixed set of coupled oscillators of cardinality not exceeding three. When this is not the case, the problem appears to be analytically intractable, and suboptimal solutions based on numerical techniques are currently the only recourse.
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    Circuits, systems and signal processing 18 (1999), S. 1-16 
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    Notes: Abstract The task of constructing an energy function is essential for direct stability analysis of electric power systems. This paper presents a general procedure for constructing analytical energy functions for detailed lossless network-reduction power system stability models. This paper primarily (i) develops canonical representations for lossless networkreduction power system models and shows that such canonical representations cover existing stability models, (ii) derives theoretical results regarding the existence of analytical energy functions for the canonical representations, and (iii) presents a systematic procedure to construct corresponding energy functions.
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    Circuits, systems and signal processing 18 (1999), S. 89-110 
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    Notes: Abstract The problem of finding all the DC solutions of a certain class of piecewise-linear electronic circuits containing locally passive and locally active one-ports is considered in this paper. An effective method enabling us to locate the solutions is developed. The method constitutes the crucial point of an algorithm based on the idea of successive contraction, division, and elimination that is capable of determining all the solutions. Several numerical examples are given, and some comparison analyses are performed confirming the usefulness of the proposed approach.
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    Circuits, systems and signal processing 18 (1999), S. 269-290 
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    Notes: Abstract The controllability and observability properties of a singular system are extensively studied. The definitions of controllability,R-controllability, and impulse controllability are introduced via characteristics of the original state vector. Analogous definitions are presented for the case of observability. The criteria established for controllability and observability are simple rank criteria related to the Markov parameters from the inputs to the states and from the initial conditions to the outputs, respectively. The present results can be considered as the direct extension of Kalman's controllability and observability criteria to the case of singular systems. Finally, the controllability and observability subspaces are derived from the image and the kernel of the controllability and the observability matrices, respectively.
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    Notes: Abstract Given a stationary time seriesX and another stationary time seriesY (with a different power spectral density), we describe an algorithm for constructing a stationary time series Z that contains exactly the same values asX permuted in an order such that the power spectral density ofZ closely resembles that ofY. We call this methodspectral mimicry. We prove (under certain restrictions) that, if the univariate cumulative distribution function (CDF) ofX is identical to the CDF ofY, then the power spectral density ofZ equals the power spectral density ofY. We also show, for a class of examples, that when the CDFs ofX andY differ modestly, the power spectral density ofZ closely approximates the power spectral density ofY. The algorithm, developed to design an experiment in microbial population dynamics, has a variety of other applications.
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    Circuits, systems and signal processing 18 (1999), S. 479-487 
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    Keywords: Cramér-Rao bounds ; direction-of-arrival estimation ; unknown noise
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    Notes: Abstract The deterministic and stochastic direction estimation Cramér-Rao bounds (CRBs) are studied in the presence of one signal and spatially uncorrelated sensor noise with unknown nonequal variances in array sensors. The explicit CRB expressions are obtained, and their relationship is studied showing some typical properties inherent in the nonidentical noise case.
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    Keywords: Nonlinear circuit theory ; co-content ; functional minimization ; image processing
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    Notes: Abstract The solutions of many physical-mathematical problems can be obtained by minimizing proper functionals. In the literature, some methods for the synthesis of analog circuits (mainly cellular neural networks) are presented that find the solution of some of these problems by implementing the discretized Euler-Lagrange equations associated with the pertinent functionals. In this paper, we propose a method for defining analog circuits that directly minimize (in a parallel way) a class of discretized functionals in the frequently occurring case where the solution depends on two spatial variables. The method is a generalization of the one presented in Parodi et al.,Internat. J. Circuit Theory Appl., 26, 477–498, 1998. The analog circuits consist of both a (nonlinear) resistive part and a set of linear capacitors, whose steady-state voltages represent the discrete solution to the problem. The method is based on the potential (co-content) functions associated with voltage-controlled resistive elements. es an example, we describe an application in the field of image processing: the restoration of color images corrupted by additive noise.
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    Circuits, systems and signal processing 14 (1995), S. 669-674 
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    Notes: Abstract The stochastic likelihood function [(STO)LF] associated with the narrowband signal processing problem can be concentrated with respect to the signal covariance matrix elements and the noise power. Although this is a known fact, no clear-cut derivation of the concentrated (STO)LF appears to be available in the literature. In this short paper we provide a simple, complete proof of the concentrated (STO)LF formula.
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    Circuits, systems and signal processing 14 (1995), S. 725-734 
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    Notes: Abstract We give bounds on the self-sustained limit cycles in fixed-poin implementations of state-variable-form digital filters having general stable system matrices. These bounds, which are period independent and which concern limit cycles due to quantization errors in physical realizations of digital filters, are given in closed form for second-order sections. A numerical comparison shows that the bounds are smaller than corresponding ones reported in the literature. This is due to our use of real similarity transformations.
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    Circuits, systems and signal processing 14 (1995), S. 787-815 
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    Notes: Abstract We consider the concept reachability for Polynomial Matrix Descriptions (PMDs); i.e., systems of the form ∑: A(ρ)β(t)=B(ρ)u(t),y(t)=C(ρ)β(t), whereρ:=d/dt the differential operator,A(ρ)=A0+A1 ρ+...+ Av ρ v εR r×r[ρ], AiεR r×r,i=0, 1,..., ν ≥ 1 with rank R A v ≤r B(ρ) =B 0+B 1ρ+...+B σρσ εR r×m[ρ], Bi εR r×m,i=0,1,...,σ ≥ 0 C(ρ)=C0+C1 ρ+...+Cσ1 ρ σ1 εR m1×r[ρ],C i εR m1×r ,i=0, 1,..., σ1 ≥ 0, β(t): (0−, ∞) →R r is the pseudostate of (∑),u(t): [0, ∞) →R m is the control input to (∑), and y(t) is the output of the system (∑). Starting from the fact that generalized state space systems, i.e., systems of the form ∑1: Ex(t)=Ax(t)+ Bu(t), y(t)=Cx(t), whereE εR r×r, rank R E 〈r, A εR r×r,B εR r×m,C εR m1×r represent a particular case of PMDs, we generalize various known results regarding the smooth and impulsive solutions of the homogeneous and the nonhomogeneous system (∑1) to the more general case of PMDs (∑). Relying on the above generalizations we develop a theory regarding the reachability of PMDs using time-domain analysis, which takes into account finite and infinite zeros of the matrix A(s)=L.[A(ρ)]. The present analysis extends in a general way many results previously known only for regular and generalized state space systems.
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    Circuits, systems and signal processing 14 (1995), S. 445-463 
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    Notes: Abstract This paper proposes a scaling and squaring geometric series method along with the inverse-geometric series method for finding discrete-time (continuous-time) structured uncertain linear models from continuous-time (discrete-time) structured uncertain linear systems. The above methods allow the use of well-developed theorems and algorithms in the discrete-time (continuous-time) domain to indirectly solve the continuous-time (discretetime) domain problems. Moreover, these methods enhance the flexibility in modeling and control of a hybrid composite system. It has been shown that the commonly used bilinear approximation model is a specific class of the proposed geometric series model.
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    Circuits, systems and signal processing 14 (1995), S. 525-538 
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    Notes: Abstract Of concern is the undistorted propagation of pulse stress waves in long thin bodies. The primary signals are arbitrarily sharp pulses projected along thex-axis. It will be seen that these pulses travel close to the speed of sound in unbounded media. Secondary pulses follow representing interactive disturbances traveling along they-axis. But the process brings out certain anomalies in circuit realizability requirements. These issues, transmission parameters, and coupling between the various signals are studied in detail. The results are supported by experiments.
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    Circuits, systems and signal processing 18 (1999), S. i 
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    Circuits, systems and signal processing 18 (1999), S. 111-130 
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    Notes: Abstract The cumulants defined in terms of moments are basic to the study of higher-order statistics (HOS) of a stationary stochastic process. This paper presents a concurrent systolic array system for the computation of higher-order moments. The system allows for the simultaneous computation of the second-, third-, and fourth-order moments. The architecture achieves good speedup through its excellent exploitation of parallelism, pipelining, and reusability of some intermediate results. The computational complexity and system performance issues related to the architecture are discussed. The concurrent system is designed with the CMOS VLSI technology and is capable of operating at 3.9 MHz.
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    Circuits, systems and signal processing 18 (1999), S. 183-187 
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    Notes: Abstract Necessary and sufficient conditions for the invertibility of a (not necessarily linear) operatorN between normed linear spaces are given. It is shown thatN is invertible precisely if a certain operator associated withN is a contraction.
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    Circuits, systems and signal processing 14 (1995), S. 187-211 
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    Notes: Abstract This paper investigates circuit Markov processes under the standpoint of duality principle.
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    Circuits, systems and signal processing 14 (1995), S. 237-254 
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    Notes: Abstract In this paper, a high resolution technique for estimating DOAs of spatially close source signals is presented. It is observed that the array manifold over a sector of interest is rank deficient and the dimension of the array manifold space, which is the range space of the array manifold, is less than the number of sensors in the array. The true signal subspace is a subspace in the array manifold space. A novel technique is provided that searches for the signal subspace in this array manifold space. The resulting estimated signal subspace has minimum principal angles with the data signal subspace generated by eigen-decomposing the covariance matrix of the array data vector. It is proved that the proposed estimator is asymptotically consistent and the estimated signal subspace is closer to the true signal subspace than the data signal subspace formed by MUSIC. The proposed novel technique has better performance than the MUSIC algorithm. Its performance is comparable to MLE and MD-MUSIC yet it requires only one-dimensional searches and is computationally much less intense. Simulation results are presented to show the effectiveness of the proposed technique, and comparisons with MUSIC, MLE, and MD-MUSIC algorithms are also included.
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    Circuits, systems and signal processing 14 (1995), S. 351-400 
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    Notes: Abstract High-level synthesis is becoming increasingly important in the area of VLSI CAD. During the last decade, much research has been carried out in this area and some results have been achieved. This paper presents a survey on the state-of-the-art of current high-level synthesis research and applications. Besides basic knowledge, a comprehensive overview has been conducted among the various theories, methods, algorithms, and applications of the existing systems. Their advantages, limitations, and feasibilities are also described. Finally, some points on current status and future directions of high-level synthesis are addressed.
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    Circuits, systems and signal processing 14 (1995), S. 401-414 
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    Notes: Abstract This paper introduces the running kernels that yield recursive structures for time-frequency distributions (TFDs). The running kernels offer important properties not possessed by the commonly used block distribution kernels. The introduced kernels allow an invariance in computations with respect to the extent of the kernel in the time or the lag variable. However, contrary to the wide class of block kernels that satisfy the desired timefrequency (t-f) properties, most recursive (running) time-frequency distributions (RTFDs) violate the marginal and the support properties. This paper considers both the direct and the indirect types of recursion and presents examples for illustration.
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    Circuits, systems and signal processing 14 (1995), S. 465-472 
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    Notes: Abstract In this paper the dual topics of robust signal detection and robust estimation of a random variable are considered, where the data may be both dependent and nonstationary. We note that classical saddlepoint techniques for robustness do not readily apply in the dependent and/or nonstationary situation, and thus our results have application in a larger domain than what was feasible heretofore. In addition, our methods make possible the quantitative measurement of robustness and admit essentially arbitrary perturbations in an underlying joint statistical distribution away from the nominal. In particular, our methods show that the presence of dependency can result in a reduction of the robustness of the linear detector by approximately 50% and that appropriate censoring can improve this situation. We also show that, somewhat surprisingly, a weak amount of censoring can actually reduce robustness rather than increase it, even with dependent data that is “almost” independent. This calls into question the common practice, inspired by classical saddlepoint results for independent data, of employing censoring in cases where residual dependency is conceded. When applied to estimation, our work shows that for nominally Gaussian data, the conditional expectation estimator is optimal not only in terms of performance but also robustness (under appropriate performance measures), thus reinforcing the appeal of this estimator. On the other hand, for other performance measures, we also note that the conditional expectation estimator can be completely unrobust, regardless of whether the data is nominally Gaussian or not. Finally, our results establish a bound on estimator robustness.
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    Circuits, systems and signal processing 14 (1995), S. 707-724 
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    Notes: Abstract Selection of test nodes is an important phase of the fault dictionary approach. It is demonstrated in this paper that the techniques used for this purpose in other approaches of analog fault diagnosis like fault analysis and fault verification are not in general suitable for the fault dictionary approach. The ambiguity set is a simple and effective concept for choosing test nodes in the context of dictionaries. These sets are formed such that each faulty condition lies in only one ambiguity set. Deviating from this thinking, overlapping ambiguity sets are proposed in this paper, giving rise to a generalized fault dictionary. These sets use information more fully and hence reduce the number of test nodes. The concept of hashing is applied in this paper for selecting test nodes. This gives a linear time algorithm (linear in the number of fault voltage specificationsf′) and it isf′ times faster than the existing methods. It is not possible to select test nodes faster than this. This technique can also be used to select test nodes by the process of elimination of nodes. This is also linear inf′ per node elimination. Even a group of nodes can be eliminated or selected within the same computation. This freedom is not possible with the existing methods.
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    Circuits, systems and signal processing 14 (1995), S. 689-705 
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    Notes: Abstract A major reason for the success of linear autoregressive (AR) modeling is that Kolmogrorov proved that every linear system could be represented by a linear AR model of infinite order. The computation of a finite order AR approximation is, of course, the practical goal. In this paper, we prove that every nonlinear system with a Volterra series expansion can be represented as a nonlinear AR model of infinite order. Our method shows how an approximation to any desired order and degree can be achieved.
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    Circuits, systems and signal processing 14 (1995), S. 213-236 
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    Notes: Abstract This paper presents an improved direct control architecture for the on-line learning control of dynamical systems using backpropagation neural networks. The proposed architecture is compared with the other direct control schemes. In this scheme the neural network interconnection strengths are updated based on the output error of the dynamical system directly, rather than using a transformed version of the error employed in other schemes. The ill effects of the controlled dynamics on the on-line updating of the network weights are moderated by including a compensating gain layer. An error feedback is introduced to improve the dynamic response of the control system. Simulation studies are performed using the nonlinear dynamics of an underwater vehicle and the promising results support the effectiveness of the proposed scheme.
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    Circuits, systems and signal processing 14 (1995), S. 299-316 
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    Notes: Abstract The analysis of initial value problems associated with singular linear dynamic systems using spectral methods is discussed. These methods, though extensively studied in the literature, can be very sensitive to numerical errors and to the presence of noisy input data. Moreover, using piecewise constant functions, like Walsh or Block-Pulse, spectral methods have been shown to be unstable with respect to incorrect initial conditions. In this paper regularization techniques are used to reduce arbitrarily the sensitivity of the spectral methods, and at the same time to stabilize them. Although the regularized spectral methods maintain the structure of the original algorithm, they can also be rewritten in recursive form, leading to the definition of a class of discrete time filters. Finally, some applicative numerical examples are discussed in order to show the effectiveness of the proposed algorithms.
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    Circuits, systems and signal processing 14 (1995), S. 415-425 
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    Notes: Abstract A complete solution is given to the problem of determining when a certain iteration will converge to the minimum of an important type of quadratic functional. It is shown that convergence occurs whenever the minimum exists, and that the iterates produced by the iteration will be unbounded for every starting point if the minimum does not exist. Applications are given concerning an adaptive filtering algorithm and nonharmonic series expansions.
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    Circuits, systems and signal processing 13 (1994), S. 65-75 
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    Notes: Abstract A method will be presented for the approximation of a desired two-dimensional frequency response by the frequency response of a two-dimensional finite-impulse-response digital filter. It is possible to match the functional values and an arbitrary number of derivatives of both responses for zero frequency, thus making the error flat up to a desired degree. Remaining degrees of freedom are used for anL 2-approximation. Closed form design formulae will be given.
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    Circuits, systems and signal processing 13 (1994), S. 31-64 
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    Notes: Abstract This paper presents an efficient procedure for the design of interpolated FIR (IFIR) filters with linear phase. The algorithm uses the uniform B-spline function as an interpolator and solves the optimal Chebyshev approximation problem on the optimal subinterval. The technique can be used for the design of general lowpass, highpass and bandpass filters. While the number of multiplications of the IFIR filter is dependent on the bandwidth and the center frequency of the desired filter, it provides the minimum number of multiplications achievable and nearly always provides a substantial reduction when compared to Parks-McClellan designs.
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    Circuits, systems and signal processing 13 (1994), S. 117-118 
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    Circuits, systems and signal processing 13 (1994), S. 139-154 
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    Notes: Conclusion We have seen that the stationary solutions of the regularized equations are asymptotically stable if the stationary solution of the original quasilinear index-2 tractable DAE is so. In other words, the given regularizations supply a solution sharing the stability properties of the solution of the unperturbed equation.
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    Circuits, systems and signal processing 13 (1994), S. 123-138 
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    Notes: Abstract A method has been proposed for numerically solving lower dimensional, nonlinear, higher index differential algebraic equations for which more classical methods such as backward differentiation or implicit Runge-Kutta may not be appropriate. This method is based on solving nonlinear DAE derivative arrays. This paper discusses progress on the implementation of this method, resolves some of the issues involved, and lists some remaining problems. Computational experience on two prescribed path control problems is presented showing that the approach should prove practical for many applications.
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    Circuits, systems and signal processing 13 (1994), S. 213-222 
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    Notes: Abstract Some issues in the stability of differential delay systems in the linear and the nonlinear case are investigated. In particular, sufficient robustness conditions are derived under which a system remains stable, independent of the length of the delay(s). Applications in the design of delayed feedback systems are given. Two approaches are presented, one based on Lyapunov theory, the other on a transformation to Jordan form. In the former, sufficient conditions are obtained in the form of certain Riccati-type equations.
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    Circuits, systems and signal processing 14 (1995), S. 587-602 
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    Notes: Abstract The least mean square (LMS) algorithm is investigated for stability when implemented with two's complement quantization. The study is restricted to algorithms with periodically varying inputs. Such inputs are common in a variety of applications, and for system identification, they can always be generated as shown with an example. It is shown that the quantized LMS algorithm is just a special case of a quantized periodically shift-varying (PSV) filter. Two different sufficient conditions are obtained for the bounded input bounded output (BIBO) stability of the PSV filter. When the filter is BIBO stable, two different bounds on the filter output are also derived. These conditions and bounds are then applied to the quantized LMS algorithm. The results are illustrated with examples.
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    Circuits, systems and signal processing 14 (1995), S. 603-614 
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    Notes: Abstract An improved version of the Fast Polynomial Transform for the computation of two-dimensional cyclic convolutions is proposed in this paper. This new version of the Fast Polynomial Transform comes from the concept of splitting product-fields and gives more than a 20% to 50% reduction in the numbers of multiplications. An average of 20% improvement on overall computation time compared with the previous approach is also achievable.
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    Circuits, systems and signal processing 14 (1995), S. 653-660 
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    Notes: Abstract An algorithm is presented for the spectrum of experimentally obtained exponentially damped sinusoidal signals. The implementation of this algorithm is straightforward and requires only simple basic mathematical operations. Applications of the algorithm are illustrated by several examples.
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    Circuits, systems and signal processing 14 (1995), S. 675-688 
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    Notes: Abstract Exact analysis of second-order sigma-delta modulators with constant input is presented. Some properties of general limit cycles are established. Sufficient conditions for the existence of a general class of limit cycles of the state vectors of the modulator are determined. The conditions interrelate the input, an initial vector, and the period of the limit cycle. Also, sufficient conditions are determined for a class of initial condition vectors to converge to a limit cycle.
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    Circuits, systems and signal processing 14 (1995), S. 735-747 
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    Notes: Abstract A weighted median filter is a nonlinear digital filter consisting of a window of length 2N + 1 and a weight vector W=(W− N,..., W0,..., WN). A root signal of a median type filter is a signal that is invariant to the filter. However, not all weighted median filters possess the convergence property. In this paper, we shall study the root structures and the convergence behavior of a subclass of weighted median filters, calledclass- 1 filters, which is symmetric in its weight vector. We shall introduce an important parameter, calledfeature value, and show that any one-dimensional unappended signal of lengthL will converge to a root signal in at most $$3\left\lceil {\frac{{L - 2}}{{2(2N + 2 - p)}}} \right\rceil$$ passes of aclass −1 filter with window width 2N + 1 and thefeature value p.
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    Circuits, systems and signal processing 14 (1995), S. 771-786 
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    Notes: Abstract A multirate modeling theory of the ARMA stochastic signals is derived from a state-space viewpoint in this work. Its application to the signal reconstruction problem for the recovery of the complete ARMA signal from its noise-corrupted, missing-sample sequence is then developed in detail. The proposed estimation-interpolation problem can be resolved by using the multirate optimal state estimation scheme of this work. Theoretically, the multirate Kalman reconstruction filters derived in this paper produce the minimum variance estimation and interpolation of the original complete, clean ARMA signal. Practically, the numerical examples show that the multirate Kalman reconstruction filters illustrate good estimation/interpolation performances, not only for synthetic ARMA sequences but also for human speech signals.
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    Circuits, systems and signal processing 13 (1994), S. 3-18 
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    Notes: Abstract Image coding can be implemented through DPCM, transform, hybrid, or segmentation coding techniques. Some transform coding techniques, such as cosine and Hadamard, have been exhaustively analyzed and evaluated, while others, such as Legendre, have not. This paper introduces the use of Legendre transform in image coding. The transform matrix for different block sizes is calculated, the fast algorithm is derived, and the performance is evaluated through both mean square error and subjective quality. The results obtained have indicated that the system performance is comparable with that of optimum KLT and cosine transforms; moreover, it is simpler in implementation.
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    Circuits, systems and signal processing 13 (1994), S. 77-97 
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    Notes: Abstract A method will be presented for the approximation of a desired frequency response by the frequency response of a FIR filter. It is possible to match the functional values and an arbitrary number of derivatives of both responses for zero frequency, thus making the error flat up to a desired degree. Remaining degrees of freedom are used for a weightedL 2-approximation. Closed form design formulae will be given.
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    Circuits, systems and signal processing 13 (1994), S. 167-183 
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    Notes: Abstract The Stewart platform manipulator is a fully parallel kinematic linkage system that has great advantages over typical serial link robots. However, one major problem in controlling this kind of manipulator is that, due to its closed kinematic chain and parallel-link constrained structure, it is impossible to derive an explicit kinematic solution. This paper proposes a simplified algorithm to numerically solve the forward kinematics of a six-links Stewart platform. Taking advantage of some fundamental geometric operations, the proposed algorithm involves only 3 nonlinear simultaneous equations. The explicit expressions are derived for some special configurations which can directly give the geometric limitations to motion in terms of the geometric dimensions of the platforms and the legs.
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    Circuits, systems and signal processing 13 (1994), S. 155-165 
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    Notes: Abstract We consider the numerical solution of an unconfined detonation problem modelled by differential/algebraic equations in a boundary value format. The problem has index one throughout the interval of integration, except at a free boundary point, where it has index two. Moreover, a nonlinear constraint is active at the free boundary point. We consider a simple mathematical model, as well as a more realistic and functionally complex model. In both cases, the resulting boundary value problem is solved directly, using a collocation method. This approach compares favorably with the solution techniques proposed previously in the literature and enables us to examine some interesting properties of the solution to the detonation problem.
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    Circuits, systems and signal processing 13 (1994), S. 241-254 
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    Notes: Abstract In [2] a relation of covering for implicit systems has been introduced and studied. It has been shown that this notion is useful when studying the problems of minimal left inverse and minimal observer design. In the present paper we study a slightly more general notion: a relation of inclusion for implicit systems that enjoy the outputuniqueness property. We are chiefly interested in systems of a minimal size including a given system. A particular application we have in mind is a minimal order observer or a minimal left inverse construction. We show that study of including systems may allow us in some cases to reduce the size of an observer or an inverse beyond the one given by minimization under external equivalence. A similar approach has been applied in [2] to minimal covering systems. However, including systems have an important advantage over the covering ones: since they constitute a broader class, they offer us more freedom in picking a minimal one of a desired sort.
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    Circuits, systems and signal processing 13 (1994), S. 311-327 
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    Notes: Abstract The proportional canonical form and its complete orbital invariants, described by Loiseau and Lebret in [11] or [8], are used to obtain structural equivalent characterizations of the solutions to the Disturbance Decoupling Problems which have been defined and geometrically studied by Banaszuk, Kociecki and Przyluski in [3]. In addition, a new problem, based on deeper input-output consideration, is introduced. Necessary conditions for this last problem are given.
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    Circuits, systems and signal processing 18 (1999), S. 315-329 
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    Notes: Abstract In earlier studies a multiclass vector quantization (MVQ)-based neural network design was explored for pattern classification. We reconsider that design here in the context of function emulation. With proper adjustment, the MVQ design demonstrates excellent performance. Moreover, the design algorithms sense discontinuities in the data and replicate them in the network.
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    Circuits, systems and signal processing 18 (1999), S. 365-376 
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    Notes: Abstract In earlier studies the concept of a fast form was generalized to a format that unified such discrete transform examples as the FFT, the FCT, the FST, and the FHT. In this study we consider the approximation of arbitrary linear maps by fast forms. Using simulation we evaluate the approximation capabilities of the generalized fast form.
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    Circuits, systems and signal processing 18 (1999), S. 377-393 
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    Notes: Abstract On-line running spectral analysis is of considerable interest in many electrophysiological signals, such as the EEG (electroencephalograph). This paper presents a new method of implementing the fast Fourier transform (FFT) algorithm. Our “real-time FFT algorithm” efficiently utilizes computer time to perform the FFT computation while data acquisition proceeds so that local butterfly modules are built using the data points that are already available. The real-time FFT algorithm is developed using the decimation-in-time split-radix FFT (DIT sr-FFT) butterfly structure. In order to demonstate the synchronization ability of the proposed algorithm, the authors develop a method of evaluating the number of arithmetic operations that it requires. Both the derivation and the experimental result show that the real-time FFT algorithm is superior to the conventional whole-block FFT algorithm in synchronizing with the data acquisition process. Given that the FFT sizeN=2 r , real-time implementation of the FFT algorithm requires only 2/r the computational time required by the whole-block FFT algorithm.
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    Circuits, systems and signal processing 18 (1999), S. 523-523 
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    Circuits, systems and signal processing 18 (1999), S. 539-551 
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    Keywords: H 2 deconvolution filter ; envelope-constrained filter ; finite impulse response filter ; linear matrix inequality
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    Notes: Abstract In this paper, we consider an envelope-constrained (EC)H 2 optimal finite impulse response (FIR) filtering problem. Our aim is to design a filter such that theH 2 norm of the filtering error transfer function is minimized subject to the constraint that the filter output with a given input to the signal system is contained or bounded by a prescribed envelope. The filter design problem is formulated as a standard optimization problem with linear matrix inequality (LMI) constraints. Furthermore, by relaxing theH 2 norm constraint, we propose a robust ECFIR filter design algorithm based on the LMI approach.
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