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  • Articles  (167,894)
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  • 1
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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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  • 2
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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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  • 3
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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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  • 4
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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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  • 5
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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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  • 6
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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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  • 7
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    Circuits, systems and signal processing 15 (1996), S. 555-572 
    ISSN: 1531-5878
    Source: Springer Online Journal Archives 1860-2000
    Topics: Electrical Engineering, Measurement and Control Technology
    Notes: Abstract This work addresses the design of LoG filters in the frequency domain within a structure formed by the cascade of quasi-Gaussian and discrete Laplacian filters. The main feature of such a structure is that it requires half the number of convolutions of the classical structure in which the LoG transfer function is expressed as the sum of two separable transfer functions of 1-D Gaussian and LoG type. Such a perspective allows one to rephrase the design of IIR and FIR filters for edge detection as a frequency domain approximation problem solvable by standard digital filter design tools. The zero-phase IIR solutions have a good performance at low orders and approximation errors practically independent of the aperture parameter. The characteristics of the nearly linear-phase IIR filters solving the problem suggest the consideration of linear-phase FIR filters with zeros constrained on the unit circle. The use of such filters leads to remarkable computational savings with respect to the filters designed by impulse response sampling. The agreement between the edge values obtained by the filters designed according to the scheme proposed in this work and those obtained by standard techniques is very good.
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  • 8
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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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  • 9
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    Circuits, systems and signal processing 14 (1995), S. 69-85 
    ISSN: 1531-5878
    Source: Springer Online Journal Archives 1860-2000
    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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  • 11
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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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  • 12
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    Circuits, systems and signal processing 14 (1995), S. 317-349 
    ISSN: 1531-5878
    Source: Springer Online Journal Archives 1860-2000
    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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  • 13
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    Circuits, systems and signal processing 14 (1995), S. 427-443 
    ISSN: 1531-5878
    Source: Springer Online Journal Archives 1860-2000
    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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  • 14
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    Circuits, systems and signal processing 14 (1995), S. 495-524 
    ISSN: 1531-5878
    Source: Springer Online Journal Archives 1860-2000
    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 
    ISSN: 1531-5878
    Source: Springer Online Journal Archives 1860-2000
    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 
    ISSN: 1531-5878
    Source: Springer Online Journal Archives 1860-2000
    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 
    ISSN: 1531-5878
    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 14 (1995), S. 39-55 
    ISSN: 1531-5878
    Source: Springer Online Journal Archives 1860-2000
    Topics: Electrical Engineering, Measurement and Control Technology
    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 
    ISSN: 1531-5878
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    Topics: Electrical Engineering, Measurement and Control Technology
    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 
    ISSN: 1531-5878
    Source: Springer Online Journal Archives 1860-2000
    Topics: Electrical Engineering, Measurement and Control Technology
    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 
    ISSN: 1531-5878
    Source: Springer Online Journal Archives 1860-2000
    Topics: Electrical Engineering, Measurement and Control Technology
    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 15 (1996), S. 505-514 
    ISSN: 1531-5878
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    Topics: Electrical Engineering, Measurement and Control Technology
    Notes: Abstract A novel and complete method is presented for constructing the generalized Bezout identity in polynomial form of the transfer matrix from its state-space representation of a reachable and observable system. As a result, one can easily apply well-developed synthesis and analysis theories to systems described in the frequency domain.
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    Circuits, systems and signal processing 15 (1996), S. 543-554 
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    Topics: Electrical Engineering, Measurement and Control Technology
    Notes: Abstract The aim of this paper is to obtain the uncertain value set in the complex plane for systems with real and complex parameters that are known to lie inside a ball in a weightedl p-norm. It generalizes previously available results and may be used to test the robust stability of polynomials whose coefficients lie in a weighted lp-ball.
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    Circuits, systems and signal processing 15 (1996), S. 581-595 
    ISSN: 1531-5878
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    Topics: Electrical Engineering, Measurement and Control Technology
    Notes: Abstract This paper describes a new approach—a Fixed-Level Least Squares (FLLS) method for linear-phase FIR filter design. It is intended for rejection of the Gibbs phenomenon through the introduction of a set of equally spaced fixed levels in the transition band and subsequent redefinition of the approximated and weighted functions. Detailed mathematical solutions of the problem as well as many examples are given. The results in graphical form are shown as an output of the FLLS software model.
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    Circuits, systems and signal processing 15 (1996), S. 631-648 
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    Topics: Electrical Engineering, Measurement and Control Technology
    Notes: Abstract In this paper we examine order reduction of parabolic systems using modal truncation. The parabolic distributed system is first approximated using the Galerkin method. The system matrices have a special structure that allows us to find the approximate spectrum of the parabolic system. To do this we compute approximate inverses of tridiagonal, diagonally dominant symmetric matrices. The approximation leads to algorithms of orderO(n), as opposed to traditional algorithms of orderO(n), wheren is the order of the system. Finally, an example is presented to illustrate the proposed algorithm.
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    Circuits, systems and signal processing 15 (1996), S. 711-725 
    ISSN: 1531-5878
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    Topics: Electrical Engineering, Measurement and Control Technology
    Notes: Abstract We give results concerning the problem of approximating the input-output maps of nonlinear discrete-time approximately finite-memory systems. Here the focus is on the linear dynamical parts of the approximating structures, and we give examples showing that these linear parts can be derived from asingle prespecified function that meets certain conditions. This is done in the context of an approximation theorem in which attention is focused on what we call “basic sets.” We also consider the related but very different problem of approximating nonlinear functionals using lattice operations or the usual linear ring operations. For this problem we give criteria, not just sufficient conditions, for approximation on compact subsets of reflexive Banach spaces (any Hilbert space is a reflexive Banach space).
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    Circuits, systems and signal processing 15 (1996), S. 763-770 
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    Topics: Electrical Engineering, Measurement and Control Technology
    Notes: Abstract New calibration methods are presented for bearing estimation with sensor location uncertainties, which are based on eigenvalue decomposition of the sample covariance matrix and three or more nondisjoint sources in known bearings. The methods can be applied to arbitrary arrays, including linear arrays, require fewer computations, and are suitable for low signal-to-noise ratio (SNR) cases. Computer simulations are given to illustrate the performance of the proposed methods.
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    Circuits, systems and signal processing 15 (1996), S. 145-164 
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    Topics: Electrical Engineering, Measurement and Control Technology
    Notes: Abstract A doubly recursive algorithm for time domain convolution with a piecewise linear weighting function is presented that combines the speed of a recursive (IIR) digital filter with the flexibility and ease of design of a nonrecursive (FIR) digital filter. The approach approximates the desired FIR weighting function by a sum-of-triangles weighting function. ForL triangles (or triangle pairs for a linear phase filter) the algorithm is of orderLN. The approximation improves with the number of triangles. A significant advantage of the algorithm compared to FFT filtering or direct convolution is that there is no necessity of a tradeoff between frequency response accuracy and computation time per output point as the data spacing decreases in the filtered signal. The computational complexity is dependent on the number of triangles chosen, not the width of the weighting function, so the algorithm is especially effective for filters with an inherently wide FIR weighting function.
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    Circuits, systems and signal processing 15 (1996), S. 335-342 
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    Topics: Electrical Engineering, Measurement and Control Technology
    Notes: Abstract A solution of the problem of correlation analysis of stochastic amplitude-angle modulated signals is presented. The solution is obtained on the basis of the theory of linear stochastic processes for the general case of nonstationary correlated Gaussian noises in both modulation channels. These noises are regarded as the results of a linear filtration of Gaussian white noise processes. As an example of the analysis a case of band-pass modulation noises is considered.
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    Circuits, systems and signal processing 15 (1996), S. 395-413 
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    Topics: Electrical Engineering, Measurement and Control Technology
    Notes: Abstract Two-dimensional (2-D) periodically shift-variant (PSV) digital filters are considered. These filters have applications in processing video signals with cyclostationary noise, scrambling digital images, and 2-D multirate signal processing. The filters are formulated in the form of a Givone-Roesser (GR) state-space model with periodic coefficients. This PSV model is then presented in block form as a shift-invariant system that also has the same GR state-space form. This block form has reduced computations and ease of analysis. An algorithm is developed that transforms any given 2-D PSV GR system to its equivalent shift-invariant model. Invertibility of this model is an important consideration, especially in image scrambling applications. A condition is established for the invertibility of the shift-invariant model of the 2-D PSV system. Also, the inverse system can be easily computed from the original. The established results are illustrated with an example.
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    Circuits, systems and signal processing 15 (1996), S. 515-518 
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    Topics: Electrical Engineering, Measurement and Control Technology
    Notes: Abstract In a recent paper a method is described for constructing certain approximations to a general element in the closure of the convex hull of a subset of an inner product space. This is of interest in connection with neural networks. Here we give an algorithm that generates simpler approximants with somewhat less computational cost.
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    Circuits, systems and signal processing 15 (1996), S. 573-580 
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    Topics: Electrical Engineering, Measurement and Control Technology
    Notes: Abstract In this paper, we gave an exponential estimate for the solutions of singular systems of differential equations with a delay and establish a comparison inequality for the systems. Based on this we estabish the criteria for asymptotic stability for the composite singular systems.
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    Circuits, systems and signal processing 15 (1996), S. 685-693 
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    Notes: Abstract This paper presents the VLSI architectures for three-level correlator design based on 1-μm CMOS technology. The architecture performs very high speed, real-time, three-level cross-correlation of signals. Two architectures, one for serial incoming samples of signals (serial data) and the other for stored signal samples (parallel data), are described in the paper.
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    Circuits, systems and signal processing 15 (1996), S. 727-733 
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    Notes: Abstract This paper gives the existence of and bounds for the branch voltages and currents in a given infinite network, which on each branch are related by a maximal monotone resistance function. We give a simpler proof than that of Minty, with smaller bounds, in the finite network case.
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    Circuits, systems and signal processing 15 (1996), S. 807-818 
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    Notes: Abstract An efficient procedure is presented for computing the frequency-response templates of uncertain rational transfer functions depending on two independent interval polynomials. It consists of a modified Cohen-Sutherland algorithm for the intersection test of two rectangles and a pivoting algorithm. Because the proposed procedure traces out the boundary of a frequency-response template directly, it is, in general, more efficient than the previously proposed algorithms.
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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 15 (1996), S. 597-607 
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    Notes: Abstract In this paper, a novel VLSI algorithm for the computation of a two-dimensional discrete cosine transform is proposed. The 2D-DCT equation can be expressed by the sum of high order cosine functions, and the algorithm can be realized by combining a highly efficient first order recursive structure with some simplified matrix multiplications, which results in highly regular hardware architecture and simple routing. The algorithm has temporal and spatial locality of connection and can be segmentized for pipeline operations, so the computation time is greatly reduced. Owing to the simplicity in hardware structure, it is especially good for VLSI implementation.
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    Circuits, systems and signal processing 15 (1996), S. 609-630 
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    Notes: Abstract We examine a generalized matched-filter problem in which the interference is a nonstationary process generated by passing white noise through a general linear time-varying filter. First a matched filter is constructed by transforming the problem into an equivalent formulation involving stationary interference and a time-varying propagation channel. Whereas the response of a time-invariant matched filter is sampled at its peak, the response of this time-varying matched filter is normalized before sampling to account for variations in the signal power. Next a matched filter is constructed using a spectral characterization of the nonstationary interference. This construction is then used to formulate a simplified solution for the case where the rate of variation in the nonstationary interference is sufficiently small. The different solutions are illustrated by a numerical example.
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    Circuits, systems and signal processing 15 (1996), S. 671-683 
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    Notes: Abstract The purpose of this paper is to investigate the approximation capability of elliptic basis function (EBF) neural networks. The main results are: (1) A necessary and sufficient condition for a functionS′ (R 1) to be qualified as an activation function in the hidden layer of an EBF neural network is given. (2) Every nonzero functionG ε L 2(R n ) is qualified to be an activation function in the elliptic neural network to approximate any function in L2(Rn). (3) As applications, we give new proofs of the theorems concerning the approximation capability of affine basis function (ABF) neural networks and generalized radial basis function (GRBF) neural networks (including radial basis function neural networks) with arbitrary activation functions. In particular, we solve the problem of the approximation capability of sigma-pi neural networks.
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    Circuits, systems and signal processing 15 (1996), S. 735-748 
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    Notes: Abstract The frequency response of linear interval systems under PID controllers is analyzed. A two-stage elimination algorithm is presented to describe the boundary of the response in the complex plane. Illustrative examples are included.
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    Circuits, systems and signal processing 15 (1996), S. 771-805 
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    Notes: Abstract This article deals with a new intelligent analog circuit design system. Unlike previous design methods, this approach introduces a formal circuit representation for both the numerical and heuristic knowledge of the design system. The predicate logic circuit representation is proposed as a new formal analog circuit description language. The language's syntax and semantics provide a precise symbolic description of analog circuit functionality at higher levels of hierarchy and network component connectivities, together with CMOS transistor sizes as the lowest level of hierarchy. It is shown how sentence conversion rules of language grammer can be used to derive transistor level circuits from input performance specifications. Language sentences have the form of clauses. In addition to clause representation, a frame representation to reflect VLSI design hierarchy is also introduced. An original bidirectional inference mechanism with elements of hypothesis has been introduced to infer designs from the knowledge in clause and frame representations. The unique feature of this circuit knowledge representation is its ability to automate the analog CMOS circuit design process. The design methodology is described in detail. The proposed iterative closed loop design system adopts an expert system approach to provide topological synthesis, uses a SPICE circuit simulator to evaluate the circuit performance, and uses a new diagnostic expert system to provide advice on how to improve the design. The implementation of the methodology and associated experimental results for simple CMOS operational amplifier designs are also presented.
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    Circuits, systems and signal processing 15 (1996), S. 343-359 
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    Notes: Abstract In this paper, a closed-form identification of possibly nonminimum phase multichannel moving average (MA) processes is derived by exploiting the eigenstructures of the observation cumulant matrices using the ESPRIT algorithm. The proposed approach allows the combination of statistics of different orders for better performance and offers reduced computation complexity when compared with existing iterative approaches. Simulations are also presented to demonstrate the performance of the proposed algorithm.
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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 15 (1996), S. 137-144 
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    Notes: Abstract This paper presents an approach to estimate the directions-of-arrival (DOAs) of narrowband coherent (completely corrected) signals by an arbitrary geometry sensor array. The concept of focussing matrices is introduced to estimate the outputs of a virtual uniform linear array from the outputs of the real array in order to use spatial smoothing techniques. Unlike the work presented in [5], the proposed approach avoids multiple singular value decomposition (SVD) computations in the various sectors because it is not necessary to divide the field of view of the array into sectors. Simulation results are given to support the approach.
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    Circuits, systems and signal processing 15 (1996), S. 213-231 
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    Topics: Electrical Engineering, Measurement and Control Technology
    Notes: Abstract An optimal deconvolution filter design method is proposed in this paper for signal transmission systems with small perturbation of parameters. The perturbative parameters of the transmission channel and noise model are of probabilistic structures. A realizable filter is derived to minimize the mean square estimation error from the viewpoint of frequency domain. The calculus of variation technique and the spectral factorization method are used in the design procedure. The design method is suitable for the deconvolution of both minimum-phase and nonminimum-phase perturbative transmission systems. The minimum mean square error of the optimal deconvolution filter is also discussed. Finally, an example is given to illustrate the simulation results of the proposed optimal deconvolution filter.
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    Circuits, systems and signal processing 15 (1996), S. 291-311 
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    Notes: Abstract In this paper we present a simple and practical algorithm for the estimation of uncertain parameters of linear systems. The uncertainty is twofold, involving random observation noise, and possible jumps in the parameter values. The jumps may occur at unknown points in time, and are of unknown magnitudes and directions. The algorithm is based on the Kalman filter, with a single-sample hypothesis test, which is used to employ a three-state decision rule (yes, no, maybe). The “maybe” choice invokes a fading memory Kalman filter. The overall algorithm contains the constant parameter filter, fading memory filter, and the set of tests and rules that enable it to switch back and forth between the two filters. Application examples are presented.
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    Circuits, systems and signal processing 15 (1996), S. 361-376 
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    Notes: Abstract A novel approach that employs ordered binary decision diagrams (OBDDs) is contributed to factorize multi-level logic functions by requiring as few literals as possible. A logic function with PLA format is represented as an OBDD form first. A heuristic decision method of variable ordering, called theorder lookahead method, is derived for the construction of OBDDs. This method is based on the constant cofactor and the number of erasable logic terms for each input variable. The total execution time of the OBDD construction by the above ordering decisions is very fast for some MCNC benchmarks. With the above OBDDs, we introduce a simple yet effective graph manipulation, calledEXT, to obtain a minimal number of literals in the Boolean function. This greedyEXT algorithm consists mainly of two phases. The first phase, calledgraph analysis, identifies the similarities between nodes on the same level in the OBDD. The second phase, calledtree analysis, utilizes the above features to extract the common parts of the nodes. TheEXT procedure runs from the bottom level up to the top level of the OBDD. The computational complexity depends on the number of nodes in the OBDD. The results of simulations show thatEXT has a very fast CPU execution time and a competitive literal ratio with other methods for some MCNC benchmarks.EXT will produce the smallest literal number, especially for structured or symmetric circuits.
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    Circuits, systems and signal processing 15 (1996), S. 467-480 
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    Notes: Abstract In this study we explore the use of nonlinear embedding maps to expand the dimension of the input space. The efficacy of such maps to speed training and to enhance performance is illustrated through several examples. A natural connection to nonlinear synaptic interconnects is also developed.
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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 15 (1996), S. 313-333 
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    Notes: Abstract We construct a general model of a nonlinear input-output system consisting of finitely many blocks whose variables are in extended spaces. This model encompasses all configurations used in control which have several feedback and feedforward loops. We show that such a system is essentially the traditional MI-MO feedback system. Moreover, assuming that one block of the system is a plant affected by perturbations, we derive conditions guaranteeing linear insensitivity and/or robust stability of the whole system. In particular, we consider systems whose variables are continuous, vector-valued functions on [0, ∞), and whose blocks are described by nonlinear Volterra operators. The applications of the results are illustrated by several examples of feedback-feedforward systems.
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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 15 (1996), S. 453-466 
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    Notes: Abstract The design of a constant resistance multiplexer composed of three reactance ladder networks connected either in series or in parallel is a very practical problem. In this paper, an improvement on Norton's method and a set of necessary and sufficient conditions on the realizability of a Butterworth or elliptic constant resistance multiplexer are presented. Two illustrative examples are given to show the design procedure.
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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 15 (1996), S. 1-22 
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    Notes: Abstract This paper presents a new interval Pade approximation method to convert a continuous-time (discrete-time) uncertain linear system to an equivalent discrete-time (continuous-time) uncertain model via interval arithmetic operations. Based on the inclusion theorem related to the interval arithmetic, the interval Pade's approximants and their associated interval error matrices with interval arguments are obtained via the Pade's approximants and their associated error matrices with degenerate (real) arguments, respectively. Tighter error bounds of various approximate uncertain models with respect to their exact uncertain models are determined and used to modify the obtained Pade's approximants, so that the resulting approximate uncertain models are able to tightly enclose the original uncertain systems. Thus, the analysis and design of the original uncertain systems can be indirectly carried out using the converted uncertain models in either the continuous-time or the discrete-time domain.
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    Circuits, systems and signal processing 15 (1996), S. 165-211 
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    Notes: Abstract Three new concepts — breakdown points, breakdown probabilities, and midpoint sensitivity curves — for stack filter analysis are introduced and analyzed in this paper. Breakdown points and probabilities can be used as measures of the robustness of stack filters. Midpoint sensitivity curves in turn give information on how sensitive the output of a stack filter is to the changes of a single value in the input window. The second major contribution of this paper is the extension of the current optimality theory of stack filters. This theory combines noise attenuation and different constraints on the filter's behavior. New constraints are introduced in this paper. A new optimization approach based on breakdown probability as a noise attenuation measure is also derived. In certain special cases it is shown that the optimal stack filter that achieves the best noise attenuation subject to given constraints can be obtained in closed form. An algorithm for finding this form is given in this paper, and its modification for finding a stack filter having (approximately) a required rank selection vector is presented.
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    Circuits, systems and signal processing 15 (1996), S. 275-290 
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    Notes: Abstract This paper considers the estimation of the parameters of a linear discrete-time system from noise-perturbed input and output measurements. The conditions imposed on the system are fairly general. In particular, the input and output noises are allowed to be auto-correlated and they may be cross-correlated as well. The proposed method makes use of an instrumental variable (IV)-vector to produce a covariance matrix that is pre- and postmultiplied by some prechosen weights. The singular vectors of the so-obtained matrix possess complete information on the system parameters. A weighted subspace fitting (WSF) method is then applied to the aforementioned singular vectors to consistently estimate the parameters of the system. The IV-WSF technique suggested herein is noniterative and easy to implement, and has a small computational burden. The asymptotic distribution of its estimation errors is derived and the result is used to motivate the choice of the weighting matrix in the WSF step and also to predict the estimation accuracy. Numerical examples are included to illustrate the performance achievable by the method.
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    Circuits, systems and signal processing 15 (1996), S. 649-669 
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    Notes: Abstract This paper considers a (J, J′)-spectral factorization of a signed spectral matrix associated with a discrete-time improper transfer matrix by using the descriptor system representation and the generalized algebraic Riccati equation (GARE). Under the assumption that the improper transfer matrix is stable, a numerical algorithm for (J, J′)-spectral factorization is developed based on solutions of the generalized eigenvalue problem (GEP) and a related matrix equation. For an unstable transfer matrix, aJ-unitary conjugation method is applied to obtain a stable transfer matrix generating the same signed spectral matrix. Then a Hankel norm model reduction problem is briefly discussed. A simple example is also included to illustrate the numerical procedure.
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    Circuits, systems and signal processing 15 (1996), S. 695-709 
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    Notes: Abstract By means of norm,M-matrix, and matrix measure techniques, this paper estimates several restricted regions in the complex plane in which all eigenvalues of a class of discrete time-delay systems subjected to highly structured parametric perturbations are located. Both the stability and the instability conditions for these systems are also investigated via the proposed schemes. Two numerical examples are given to verify the correctness and demonstrate the applicability of the quantitative results.
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    Circuits, systems and signal processing 15 (1996), S. 749-762 
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    Notes: Abstract In this paper, we discuss a unified approach and its simple algorithm for estimating and tracking the location of the desired talker in the far field and the near field by microphone arrays. The approach is unified for the far field and the near field and can enhance the effective power of the talker's signal. The simple algorithm further avoids singular-value decomposition (SVD) computations and is suitable for real-time processing. Computer simulations and experiments using actual array data support the techniques.
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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 15 (1996), S. 93-110 
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    Notes: Abstract Systematic methods are developed for simplified implementations of cascaded stack and WOS filters. For recursive stack and WOS filters, corresponding nonrecursive implementations are given, with linear complexity with respect to the number of iterations. Dynamic domino logic is proposed for VLSI hardware implementation of positive Boolean functions, and a pipelined stack filter architecture is described.
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    Circuits, systems and signal processing 18 (1999), S. 315-329 
    ISSN: 1531-5878
    Source: Springer Online Journal Archives 1860-2000
    Topics: Electrical Engineering, Measurement and Control Technology
    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.
    Type of Medium: Electronic Resource
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    Electronic Resource
    Electronic Resource
    Springer
    Circuits, systems and signal processing 18 (1999), S. 365-376 
    ISSN: 1531-5878
    Source: Springer Online Journal Archives 1860-2000
    Topics: Electrical Engineering, Measurement and Control Technology
    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.
    Type of Medium: Electronic Resource
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