Thomas Panicker, University of Utah (U.S.A.)
V. Mathews, University of Utah (U.S.A.)
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.
Alberto Carini, DEEI, University Trieste (Italy)
V. Mathews, University of Utah (U.S.A.)
Giovanni L. Sicuranza, DEEI, University Trieste (Italy)
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.
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.)
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.
Paulo Alexandre Marques, ISEL (Portugal)
Fernando Manuel Sousa, ISEL (Portugal)
José Manuel Leitão, IST (Portugal)
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.
Brian M. Sadler, ARL (U.S.A.)
Laurel C. Sadler, ARL (U.S.A.)
Tien Pham, ARL (U.S.A.)
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.
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.)
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.
Hassan Ezzadi, DSA Univ. du Quebec (Canada)
Jean Rouat, DSA Univ. du Quebec (Canada)
Ivan Bourmeyster, Alcatel (France)
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.
Supratim Saha, University of Erlangen (Germany)
Ramakrishnan Angarai G., Department of Electrical Engineering, Indian Institute of Science (India)
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
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)
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.
Matteo Bertocco, Padova University (Italy)
Dionisio Lorenzin, Necsy (Italy)
Pietro Paglierani, Padova University (Italy)
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.
Olivier Meste, University of Nice-Sophia Antipolis (France)
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.
Gustavo A. Hirchoren, UNICAMP (Brazil)
Dalton S. Arantes, UNICAMP (Brazil)
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.