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correlation function provides all of the information about the signal in many well-known signal processing algorithms (e.g.,<br />

maximum entropy spectral analysis, linear predictive modeling, <strong>and</strong> Wiener filters). When a non-Gaussian time series is part<br />

of the problem, many of the quantities used in signal processing algorithms provide inadequate information about critical<br />

aspects of the observed data. For example, if Laplacian noise is present, the absolute error (L 1 norm) is more appropriate than<br />

mean-square error: large deviations from the mean of the distribution a_ more likely in ‘,.he Laplacian case <strong>and</strong> should not<br />

be weighted as much by the choice of norm. Furthermore, the correlation function captures the dependency structure of a time<br />

series only when the time series model for it is Gaussian; only the multivariate Gaussian density depends solely on the<br />

covariance matrix. Thus, algorithms which rely on mean-square error <strong>and</strong> the correlation function are implicitly tailored for<br />

Gaussian signals <strong>and</strong> may not yield the best possible solutions in many other circumstances. The question becomes what<br />

aspects of a non-Gaussian signal are required to specify its properties <strong>and</strong> how they can be used in signal processing<br />

algorithms. A subtle, but key issue, in developing analysis tools from both theoretical <strong>and</strong> practical viewpoints is how<br />

non-Gaussian signals are generated: given a specification of the non-Gaussian signal, how can it be produced by operations<br />

on an elementary r<strong>and</strong>om sequence? Gaussian signals can be generated by passing white, Gaussian noise through the<br />

appropriate linear system. For non-Gaussian signals, nonlinear systems are usually required, but not necessary. In this initial<br />

study, the approach taken is to determine which measurements are necessary to specify the system which models the<br />

generation of the signal.<br />

Derived from text<br />

Signal Processing; Algorithms; Maximum Entropy Method; Spectrum Analysis; Time Series Analysis; R<strong>and</strong>om Noise;<br />

Prediction Analysis Techniques<br />

20060001676 Tampere Univ. of Technology, Finl<strong>and</strong><br />

Suppression <strong>and</strong> Detection of Impulse Type Interference Using Adaptive Median Hybrid Filters<br />

Nieminen, Ari; Heinomen, Pekkra; Neuvo, Yrjo; IEEE International Conference on Acoustics, Speech, <strong>and</strong> Signal Processing<br />

(ICASSP ‘87); Volume 1; 1987, pp. 4.4.1-4.4.4; In English; See also 20060001583; Copyright; Avail.: Other Sources<br />

In this paper, we introduce a new type of nonlinear filters, the Adaptive Median Hybrid (AMH) filters, for the suppression<br />

<strong>and</strong> detection of short duration interferences. In the AMH filters, adaptive filter substructures are used to estimate the current<br />

signal value from the future <strong>and</strong> past signal values. The output of the overall filter is the median of the adaptive filter outputs<br />

<strong>and</strong> the current signal value. This kind of nonlinear filter structure is shown to adapt <strong>and</strong> preserve rapid changes in signal<br />

characteristics well. However, it filters out short duration interferences. By examining the difference between the original <strong>and</strong><br />

filtered data, interferences can be detected. We introduce two types of AMH filters, the AMH filter with separate adaptive<br />

substructures (SAMH) <strong>and</strong> the AMH filter with coupled substructures (CAMH), which have different convergence properties<br />

<strong>and</strong> implementation. We use both synthetic <strong>and</strong> real data (speech <strong>and</strong> electroencephalogram (EEG)) to show the applicability<br />

of the proposed filters.<br />

Author<br />

Adaptive Filters; Electroencephalography; Nonlinear Filters; Impulses<br />

20060001687 British Columbia Univ., Vancouver, British Columbia, Canada<br />

A Criticism of the Parametric EEG Spike Detector<br />

Beddoes, M. P.; Panych, L.; Qian, Juan; Wada, J. A.; IEEE International Conference on Acoustics, Speech, <strong>and</strong> Signal<br />

Processing (ICASSP ‘87); Volume 1; 1987, pp. 4.5.1-4.5.4; In English; See also 20060001583<br />

Contract(s)/Grant(s): 67-3290; Copyright; Avail.: Other Sources<br />

The role of the parametric stage is studied under various conditions <strong>and</strong> the following points have been demonstrated: 1)<br />

With high sampling rate (200 Hz) but otherwise favourable conditions, as the filter order, p, is increased from zero to nineteen<br />

the signal-to-noise ratio at the output of the parametric stage remains the same as at the input. Under less favourable<br />

conditions, it can fall as p is increased. 2) We find that comparable performance (in terms of spikes detected) is obtained when<br />

the parametric stage is omitted entirely, <strong>and</strong> detection is based only on the very simple non-parametric stage.<br />

Author<br />

Electroencephalography; Signal to Noise Ratios; Signal Processing<br />

20060001688 Massachusetts Inst. of Tech., Lexington, MA, USA<br />

A Methodology for Evaluating the Performance of Dynamic Range Control Algorithms for Speech Enhancement<br />

Lynch, John T.; IEEE International Conference on Acoustics, Speech, <strong>and</strong> Signal Processing (ICASSP ‘87); Volume 1; 1987,<br />

pp. 5.4.1-5.4.4; In English; See also 20060001583; Copyright; Avail.: Other Sources<br />

57

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