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International Conference on Acoustics, Speech, and signal processing (ICASSP)

作者： C. Nikias Hsing-Hsing Chiang Communications and Digital Signal Processing Laboratory Department of Electrical and Computer Engineering Northeastern University Boston MA USA

In this paper we propose a new method for non-minimum phase wavelet estimation and deconvolution based on non-causal autoregressive (AR) modeling of third moments of the output seismic trace. The magnitude and phase response of the wavelet are expressed in terms of the parameters of the noncausal AR model. The set of the AR parameters is obtained by solving an overdetermined linear system of equations which is formed using third moment estimates of the output data. A comparison of the non-causal AR bispectrum approach against more conventional approaches for wavelet magnitude and phase reconstruction is a major component of the paper. It is demonstrated by means of standard example that the proposed procedure achieves improved performance.

关键词： Deconvolution Phase estimation Reflectivity Digital signal processing Wavelet domain Laboratories Linear systems Equations Autocorrelation Nonlinear systems

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Nonminimum phase system identification via cepstrum modeling of higher-order cumulants

Nonminimum phase system identification via cepstrum modeling...

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International Conference on Acoustics, Speech, and signal processing (ICASSP)

作者： C. Nikias Renlong Pan Communications and Digital Signal Processing Laboratory Department of Electrical and Computer Engineering Northeastern University Boston MA USA

A new computationally efficient identification procedure is proposed for a non-Gaussian white noise driven linear, time-invariant, non-minimum phase system. The method is based on the idea of computing the complex cepstrum of higher-order cumulants of the system output. In particular, the differential cepstrum parameters of the system's impulse response are computed directly from higher-order cumulants via least-squares solution. The method is flexible enough to reconstruct the minimum and maximum phase components of the impulse response of MA, AR or ARMA systems without any prior knowledge of the type of the system. It does not require model order selection criteria and is shown to provide estimates with small bias and variance even with "short" length data records.

关键词： System identification Cepstrum White noise Transfer functions Digital signal processing Laboratories signal processing Convolution Frequency signal reconstruction

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VLSI Array processing strctures of quadratic digital filters with LMS algorithm

VLSI Array processing strctures of quadratic digital filters...

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International Conference on Acoustics, Speech, and signal processing (ICASSP)

作者： Yuang Lou C. Nikias A. Venetsanopoulos Communications and Digital Signal Processing Laboratory Department of Electrical and Computer Engineering Northeastern University Boston MA USA Department of Electrical Engineering University of Toronto Toronto Canada

There are many significant applications of nonlinear adaptive digital filters such as the cancellation of echoes and intersymbol interference, the equalization of transmission channels, adaptive noise cancellation and design of optimal predictors in communication systems. In this paper, we introduce a class of efficient architectures for adaptive quadratic digital filters based on the LMS algorithm and on rank compressed lower-upper (LU) triangular decomposition method. The architectures exhibit high parallelism as well as modularity and regularity. They are mapped into parallel pipeline and systolic array implementations and are evaluated on hardware cost (in bits), and data throughput delay.

关键词： Very large scale integration Array signal processing Digital filters Least squares approximation Interference cancellation Intersymbol interference Adaptive equalizers Noise cancellation Pipelines Systolic arrays

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THE RESIDENTIAL POWER CIRCUIT AS A COMMUNICATION MEDIUM

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IEEE TRANSACTIONS ON CONSUMER ELECTRONICS 1986年第3期32卷 567-577页

作者： ONEAL, JB Communications and Signal Processing Department of Electrical and Computer Engineering North Carolina State University Raleigh NC USA

This paper is a summary of work done to characterize the residential power circuit as a communication medium. Here the term residential power circuit refers to that part of the power circuit on the secondary side of the distribution transformer. This circuit is often used for communication signals for controlling appliances, security devices, and for other purposes. The frequencies of most interest in these systems range from about 100 to 200 kHz. The interface to this circuit is usually a 120 V or 240 V receptacle used to deliver power. These communication systems are called residential power line carrier systems. Other types of power line carrier are distribution line carrier and transmission line carrier both of which are used by the power utilities for communications purposes.

关键词： Circuits Impedance Frequency Wiring Home appliances Attenuation Capacitance signal processing Communication system control Communication system security

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DIGITAL signal RESTORATION USING FUZZY-SETS

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IEEE TRANSACTIONS ON ACOUSTICS SPEECH AND signal processing 1986年第4期34卷 919-936页

作者： CIVANLAR, MR TRUSSELL, HJ Center for Communication and Signal Processing North Carolina State University Raleigh NC USA Department of Electrical and Computer Engineering North Carolina State University Raleigh NC USA

A new signal restoration method with considerable flexibility in incorporating a priori information is developed. The method defines a fuzzy set for each piece of information to restrict the set of acceptable solutions. Using fuzzy sets makes it possible to model partially defined information as well as exact knowledge. The intersection of all the fuzzy sets is the feasibility set. The original signal is a member of this set with a high membership value, and any high membership valued element of this set is a nonrejectable solution. Such solutions can be computed by using optimization techniques. Ideally, the feasibility set contains only the original signal. The chance of recovering the original signal decreases as the feasibility set gets larger. Thus, the size of the feasibility set gives a quality measure for the solution. The method generated successful results in many restorations for which the conventional techniques have failed, and may be applied in image coding and tomography.

关键词： signal restoration Fuzzy sets Image restoration Degradation Wiener filter Convergence Pollution measurement Size measurement Image coding Tomography

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FACTOR INVERSE MATCHED FILTERING

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JOURNAL OF THE ACOUSTICAL SOCIETY OF AMERICA 1986年第1期79卷 91-99页

作者： BIRDSALL, TG METZGER, K Communications and Signal Processing Laboratory Electrical Engineering and Computer Science Department The University of Michigan Ann Arbor Michigan 48109

Factor inverse matched filtering (FIMF) and factor inverse filtering (FIF) are signal processing techniques used to obtain desired signal responses. Both are especially useful procedures for ‘‘pulse‐compression’’ processing and channel measurements. The theory is developed for a simple channel and known noise power spectral density so that comparison may be made with matched filtering. Expressions for the pulse‐compression energy gain, nonflatness loss NFL, and total performance are derived. The NFL is useful in selecting the best among practical pulse‐compression modulations, and with FIMF and FIF, has been used extensively since 1974 by the authors and their co‐workers in underwater acoustic propagation measurements and ocean acoustictomography.

关键词： Acoustic tomography Acoustical measurements Ocean acoustics signal processing Tomography

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Floating point error analysis of recursive least squares and least means squares adaptive filters

Floating point error analysis of recursive least squares and...

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International Conference on Acoustics, Speech, and signal processing (ICASSP)

作者： S. Ardalan Center for Communications and Signal Processing Department of Electrical and Computer Engineering North Carolina State University USA

A floating point error analysis of the Recursive Least Squares and Least Mean Squares (LMS) algorithms is presented. Both the prewindowed growing memory RLS algorithm (λ=1) for stationary systems and the exponential sliding window RLS algorithm (λ < 1) for time varying systems are studied. For both algorithms the expression for the mean square prediction error and the expected value of the weight error vector norm are derived in terms of the variance of the floating point noise sources. The results point to a trade off in the choice of the forgetting factor, λ. In order to reduce the effects of additive noise and the floating point noise due to the inner product calculation of the desired signal, λ must be chosen close to one. On the other hand, the floating point noise due to floating point addition in the weight vector update recursion increases as \lambda\rightarrow1 . Floating point errors in the calculation of the weight vector correction term, however, do not effect the steady state error and have a transient effect. Similar results are obtained for the LMS algorithm where a tradeoff exists in the choice of the loop gain.

关键词： Error analysis Least squares methods Adaptive filters Least squares approximation Resonance light scattering Additive noise Error correction Time varying systems Noise reduction Steady-state

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Derivation of the fast pole-zero (ARMA) recursive least squares algorithm using geometric projections

Derivation of the fast pole-zero (ARMA) recursive least squa...

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International Conference on Acoustics, Speech, and signal processing (ICASSP)

作者： S. Ardalan Center for Communications and Signal Processing Department of Electrical and Computer Engineering North Carolina State University USA

The Fast Pole-Zero (ARMA) Recursive Least Squares Algorithm is derived using geometric projections. The new algorithm has a reduced number of computations compared to the pole-zero fast Kalman algorithm. In particular, four vector inner products and scalar-times-vector operations are eliminated. This is particularly important in multiprocessor implementations of the algorithm. The geometric approach provides insight and useful interpretations of the various filters and residuals that form the algorithm. Simulations verifying the performance of the algorithm are presented where the poles and zeroes of an IIR filter are estimated.

关键词： Least squares methods signal processing algorithms Resonance light scattering Kalman filters Transversal filters Solid modeling IIR filters Echo cancellers Poles and zeros Vectors

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DIFFERENTIAL AND INTEGRAL METHODS FOR MULTIDIMENSIONAL INVERSE SCATTERING PROBLEMS

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JOURNAL OF MATHEMATICAL PHYSICS 1986年第10期27卷 2584-2591页

作者： YAGLE, AE Communications and Signal Processing Laboratory Department of Electrical Engineering and Computer Science The University of Michigan Ann Arbor Michigan 48109

A layer stripping procedure for solving three‐dimensional Schr?dinger equationinverse scattering problems is developed. This procedure operates by recursively reconstructing the Radon transform of the potential from the jump in the Radon transform of the scattered field at the wave front. This reconstructed potential is then used to propagate the wave front and scattered field differentially further into the support of the potential. The connections between this differential procedure and integral equation procedures are then illustrated by the derivations of two well known exact integral equation procedures using the Radon transform and a generalized Radon transform. These procedures, as well as the layer stripping procedure, are then reduced to the familiar Born approximation result for this problem by neglecting multiple scattering events. This illustrates the central role of the Radon transform in both exact and approximate inversion procedures.

关键词： Integral transforms Integral equations Acoustic wave front reconstruction Inverse scattering Differential equations

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A Derivation of a General Order, Multichannel Fast Transversal, Recursive Least Squares Filter Algorithm

A Derivation of a General Order, Multichannel Fast Transvers...

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MILCOM, Military communications Conference

作者： L.J. Faber S. H. Ardalan S.T. Alexander Center for Communications and Signal Processing Department of Electrical and Computer Engineering North Carolina State University Raleigh NC USA

This paper examines the derivation of a general order, multiple input version of the fast transversal recursive least squares filter algorithm. The new algorithm, in which the order (or number of taps) associated with each input channel is independent, is distinguished from the previously published multichannel form of the fast transversal RLS algorithm, wherein each input channel was con strained to have the same order. This additional flexibility allows assignment of filter resources to particular channels, or independent assignment to the pole/zero estimators in an ARMA system identification application. A summary of the new algorithm is given, in proper order of execution. An operations count is also provided for each equation

关键词： Least squares methods Transversal filters signal processing algorithms Partitioning algorithms Resonance light scattering Poles and zeros System identification Equations Delay Computational efficiency

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