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A fast Gauss-Newton parallel-cascade adaptive truncated Volterra filter

Authors:

Thomas Panicker, University of Utah (U.S.A.)
V. Mathews, University of Utah (U.S.A.)

Volume 3, Page 1873

Abstract:

This paper introduces a computationally efficient Gauss-Newton type adaptation algorithm for parallel-cascade realizations of truncated Volterra systems with arbitrary, but finite order nonlinearity. Parallel-cascade realizations implement higher-order Volterra systems using parallel and multiplicative combinations of lower-order Volterra systems. The complexity of our system is comparable to the complexity of the system model itself, and is considerably less than that of the fast RLS Volterra filters. Results of experiments comparing the Gauss-Newton method with a competing structure with similar computational complexity as well as demonstrating the capability of parallel-cascade systems to approximate truncated Volterra systems are also included in the paper.

ic971873.pdf

ic971873.pdf

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Sufficient Stability Bounds for Slowly Varying Discrete-Time Recursive Linear Filters

Authors:

Alberto Carini, DEEI, University Trieste (Italy)
V. Mathews, University of Utah (U.S.A.)
Giovanni L. Sicuranza, DEEI, University Trieste (Italy)

Volume 3, Page 1877

Abstract:

This paper derives sufficient time-varying bounds on the maximum variation of the coefficients of an exponentially stable, linear, time-varying and recursive filter. The stability bound is less conservative than all previously derived bounds for time-varying IIR systems. The bound is then applied to control the step size of output error adaptive IIR filters to achieve exponentially stable operation. Experimental results that demonstrate the good stability characteristics of the resulting algorithms are included in this paper.

ic971877.pdf

ic971877.pdf

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Spread Spectrum Interference Suppression Using Adaptive Time-Frequency Tilings

Authors:

Brian S. Krongold, University of Illinois (U.S.A.)
Kannan Ramchandran, University of Illinois (U.S.A.)
Douglas L. Jones, University of Illinois (U.S.A.)
Michael L. Kramer, University of Illinois (U.S.A.)

Volume 3, Page 1881

Abstract:

Interference suppression in spread spectrum communication systems is often essential for achieving maximum system performance. Existing interference suppression methods do not perform well for most types of nonstationary interference. We first consider interference suppression schemes based on adaptive orthogonal time-frequency decompositions, such as wavelet packet and arbitrary dyadic time-frequency tilings. These methods often reduce interference substantially, but their performance can vary dramatically with minor changes in interference characteristics such as the center frequency. To circumvent these drawbacks, we propose a multiple overdetermined tiling (MODT) with an accompanying blind interference excision scheme which appears very promising for mitigating time-frequency-concentrated interference. Simulations with narrowband, impulsive, and simultaneous impulsive and narrowband interference compare the performance of the various methods and illustrate the promise of approaches based on multiple overdetermined tilings.

ic971881.pdf

ic971881.pdf

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A DSP Based Long Distance Echo Canceller Using Short Length Centered Adaptive Filters

Authors:

Paulo Alexandre Marques, ISEL (Portugal)
Fernando Manuel Sousa, ISEL (Portugal)
José Manuel Leitão, IST (Portugal)

Volume 3, Page 1885

Abstract:

This paper describes an implementation of a long distance echo canceller which copes with double talking situations and exceeds the CCITT G.165 recommendation. The proposed solution is based on short length adaptive filters centered on the positions of the most significant echoes, which are tracked by time-delay estimators. To deal with double talking situations a speech detector is employed. The resulting algorithm enables long-distance echo cancellation with low computational requirements. It reaches greater echo return loss enhancement and shows faster convergence speed as compared with results reported in recent literature.

ic971885.pdf

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Optimal and Robust Shockwave Detection and Estimation

Authors:

Brian M. Sadler, ARL (U.S.A.)
Laurel C. Sadler, ARL (U.S.A.)
Tien Pham, ARL (U.S.A.)

Volume 3, Page 1889

Abstract:

We consider detection and estimation of aeroacoustic shockwaves generated by supersonic projectiles. The shockwave is an N-shaped acoustic wave. The optimal detection/estimation scheme is considered based on an additive white Gaussian noise model. The introduction of an invertible linear transformation, such as the Fourier transform or the wavelet transform, does not improve detection performance under this model. However, if unknown interference and/or model mismatch is present, linear transforms may be of use. In addition, they may significantly reduce complexity at the cost of sub-optimality. We consider the use of the wavelet transform as a means of detecting the very fast rise and fall times of the shockwave, resulting in a 1-D edge detection problem. This method is effective at moderate to high SNR and is robust with respect to unknown environmental interference that will generally not exhibit singularities as sharp as the N-wave edges.

ic971889.pdf

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Automatic Fault Monitoring using Acoustic Emissions

Authors:

Gopal Venkatesan, Dept. of EE, University of Minnesota (U.S.A.)
Dennis West, Dept. of EE, University of Minnesota (U.S.A.)
Kevin Buckley, Dept. of EE, University of Minnesota (U.S.A.)
Ahmed H. Tewfik, Dept. of EE, University of Minnesota (U.S.A.)
Mostafa Kaveh, Dept. of EE, University of Minnesota (U.S.A.)

Volume 3, Page 1893

Abstract:

Techniques for automatic monitoring of faults in machinery are being considered as a means to safely simplify or dispense with expensive periodic fault inspection procedures. This paper presents results from an ongoing investigation into the feasibility of using Acoustic Emissions (AEs) for automatic detection of microcrack formation/growth in machine components.

ic971893.pdf

ic971893.pdf

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A new Algorithm for double talk detection and separation in the context of digital mobile radio telephone

Authors:

Hassan Ezzadi, DSA Univ. du Quebec (Canada)
Jean Rouat, DSA Univ. du Quebec (Canada)
Ivan Bourmeyster, Alcatel (France)

Volume 3, Page 1897

Abstract:

This paper describes a new technique that enhances the Voice Activity Detection (V.A.D) performance between the remote speaker (receive signal) and the local speaker (located in the vehicle) in the context of mobile radio telephone environment. We use an Auditory Pitch and voiced/unvoiced Detection (A.P.D) in conjunction with an Auto Regressive (A.R) analysis in order to remove the remote speaker's voice signal from the car hands-free microphone signal. Results are compared with the reference system that doesn't include the APD.

ic971897.pdf

ic971897.pdf

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Transmission of chosen transform coefficients of normalized cardiac beats for compression

Authors:

Supratim Saha, University of Erlangen (Germany)
Ramakrishnan Angarai G., Department of Electrical Engineering, Indian Institute of Science (India)

Volume 3, Page 1901

Abstract:

A new technique for ECG compression is presented. Each delineated ECG beat is period normalized by multirate processing and then amplitude normalized. Discrete Wavelet Transform (DWT), based on Daubechies-4 basis functions is applied on these normalized beats, after shifting each of them to the origin. The concatenation of ordered DWT coefficients of these beats is a near-cyclostationary signal. An algorithm is proposed to select a set of common positions of the significant coefficients to be retained from each beat. Linear Prediction is then applied to predict only these DWT coefficients of the current beat from the corresponding coefficients of a certain number of previous beats. Transmitting only the residuals of selected coefficients improves compression. A significant advantage of this technique is that the maximum reconstruction error in any cycle does not occur in the diagnostically crucial QRS region, while achieving a compression of about 15:1 and a normalized root mean square error of about 10

ic971901.pdf

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FIR Filters in Envelope Constrained Filter Design

Authors:

Ba-Ngu Vo, ATRI, Curtin University of Technology (Australia)
Thi-Ngoc Ho, EE, UWA. (Australia)
Antonio Cantoni, ATRI, Curtin University of Technology (Australia)
Victor Sreeram, EE, UWA. (Australia)

Volume 3, Page 1905

Abstract:

Consider a continuous-time filter which in structure is comprised of an A/D converter, an FIR filter, a D/A converter and an analog post-filter. The envelope constrained (EC) filtering problem for this filter structure is to design it's digital component so as to minimize the effect of input noise whilst satisfying the constraint that the noiseless response of the filter to a specified excitation fits into a prescribed envelope. This problem is formulated as a quadratic programming (QP) problem with functional inequality constraints. Approximating this continuum of constraints by a finite set, the problem is solved by QP via active set strategy.

ic971905.pdf

ic971905.pdf

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Modified Cepstral Analysis For Accurate Estimation Of Echo Parameters In Telecommunication Networks

Authors:

Matteo Bertocco, Padova University (Italy)
Dionisio Lorenzin, Necsy (Italy)
Pietro Paglierani, Padova University (Italy)

Volume 3, Page 1909

Abstract:

A modified cepstral analysis for accurate estimation of the echo delay and the echo loss in a telecommunication system is presented. It is based on the optimization of a parametric transformation of the observed signal energy spectrum. Simulation results that show the effectiveness and the accuracy of the proposed method are reported and discussed.

ic971909.pdf

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Bispectral Reconstruction using Incomplete Phase Knowledge: a Neuroelectric Signal Estimation Application

Authors:

Olivier Meste, University of Nice-Sophia Antipolis (France)

Volume 3, Page 1913

Abstract:

The bispectral averaging technique is often used in order to analyze signal with variable signal delay, in presence of noise. Unfortunately, as the bispectrum is time-shift invariant, the initial phase of the signal can't be recovered. When studying somatosensory evoked potentials (neuroelectric signals) this phase is generally the major information, especially when it characterizes pathologies. We show that some informations about this phase can be extracted from the averaged signal. An attempt to include this knowledge in the magnitude and phase recovery algorithms is made. We illustrate the benefits of this approach on a simulation and a real application leading to a details enhancement of the analyzed signal.

ic971913.pdf

ic971913.pdf

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Optimal Phase-Locked Loop Design with Kalman Predictors for Synchronous Networks

Authors:

Gustavo A. Hirchoren, UNICAMP (Brazil)
Dalton S. Arantes, UNICAMP (Brazil)

Volume 3, Page 1917

Abstract:

A systematic technique for the optimal design of phase-locked loops for synchronous networks is presented. The method is based on Kalman estimation theory under self-similar random noise processes. This approach is optimal for certain noise models and for linear phase-detectors. The results are then extended in order to maintain the minimum mean-square phase error when the reference signal of a master-slave network is lost.

ic971917.pdf

ic971917.pdf

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