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Blind source separation using higher-order time-frequency representations

Blind source separation using higher-order time-frequency re...

引用

1st ieee sensor array and multichannel signal processing workshop (SAM 2000)

作者： Kamran, ZM Leyman, AR Abed-Meraim, K Nanyang Technol Univ Sch Elect & Elect Engn Singapore 639798 Singapore

ISBN: (纸本)0780363396

In this paper, a novel blind separation approach using higher-order time-frequency distributions is presented. The concept of higher-order time-frequency distribution matrix is also introduced. It is devised to primarily separate sources with temporal nonstationary signal characteristics. So far, this problem has been solved using statistical information available on the source signals. In contrast to well known blind source separation approaches using second-order statistics (SOS) and/or higher-order statistics (HOS) which rely on stationarity properties of the signals, the proposed approach allows separation of the sources with nonstationarity properties. In addition, the effect of spreading the noise power while localizing of the source energy in the time-frequency domain amounts to increasing the signal to noise ratio. A computationally feasible implementation is presented based on joint diagonalization of the matrices of the principal slices of the time-multifrequency domain of support of the cumulant-based Wigner trispectrums. Numerical examples demonstrate the effectiveness of the proposed approach.

关键词： Additive noise array signal processing Biomedical signal processing Blind source separation Narrowband sensor arrays signal processing algorithms signal to noise ratio Source separation Time frequency analysis

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Passive range estimation and range rate detection

Passive range estimation and range rate detection

引用

1st ieee sensor array and multichannel signal processing workshop (SAM 2000)

作者： Jeffers, R Breed, B Gallemore, B Digital Syst Resources Inc Fairfax VA 22033 USA

ISBN: (纸本)0780363396

The problem of interest is rapidly closing object (e.g a torpedo) emitting broadband and narrowband acoustic signatures. A high negative range rate is indicative of a torpedo threat, and good range-bearing estimates provide good target localization. A linear, towed hydrophone array is used as a receiver, and in many cases the received signals will display high signal-to-noise ratio. It is desirable to detect at ranges greater than the nearfield range of the array. High-resolution range-bearing estimators have been developed using narrowband signatures and broadband signatures. The range difference to the target across an array causes target signals to display a wave-front curvature. Range estimation is based on wave-front curvature determination using a matched-filtering post beamformer approach. First a target is selected for range estimation in which a subset of beams is chosen for sector inverse beamforming. Curvature pattern replicas over range and bearing are computed and used in a Capon Form beamformer. Simulations and experiments with in-water data compare algorithm performance to a derived benchmark. In addition, a cepstrum-based range rate indicator was demonstrated using experimental data displaying multipath propagation (Lloyd mirror).

关键词： Acoustic beams Acoustic emission Acoustic signal detection array signal processing Computational modeling Displays Mirrors Narrowband signal to noise ratio Sonar equipment

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Statistical performance of the MVDR beamformer in the presence of diagonal loading

Statistical performance of the MVDR beamformer in the presen...

引用

1st ieee sensor array and multichannel signal processing workshop (SAM 2000)

作者： Fertig, LB MIT Lincoln Lab Lexington MA 02173 USA

ISBN: (纸本)0780363396

New results are derived that describe the statistical performance of the MVDR (minimum variance distortionless response) adaptive beamformer when the weights are computed utilizing the sample covariance matrix and diagonal loading. It is well known that undesirably high sidelobes are generated when the unaugmented sample covariance matrix is used as an estimate of the true covariance matrix in adaptive beamforming algorithms. In practice, diagonal loading is frequently employed to reduce this deleterious effect. However, surprisingly little information is available in the technical literature that enables the user to predict the statistical performance of the MVDR algorithm when the sample covariance matrix and diagonal loading are employed. We help fill this gap by generating an approximate probability density function for the adaptive beam response for the diagonally loaded MVDR algorithm. The approximations are valid for an important class of signal scenarios. The utility of the new analysis is demonstrated via simulation. In particular, the new results are used to estimate the probability density function of the range sidelobe level for a wideband radar system in which adaptive beamforming and pulse compression are implemented in subbands.

关键词： Adaptive arrays Adaptive systems Analytical models array signal processing Covariance matrix Interference Laboratories Phased arrays Probability density function Radar

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Time-frequency matched field processing for time varying sources

Time-frequency matched field processing for time varying sou...

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1st ieee sensor array and multichannel signal processing workshop (SAM 2000)

作者： Dizaji, RM Chapman, NB Kirlin, RL Univ Victoria Sch Earth & Ocean Sci Dept Elect & Comp Engn Victoria BC V8W 3P6 Canada

ISBN: (纸本)0780363396

Deterministic broadband time-varying pulses such as chirp signals are widely used in seismic applications to determine the layering structure of the earth. Various broadband matched field processors (MFP) which have been already developed for use in inversion [1,2] rely on only their frequency distribution so their performance fail in the presence of similar powerful broadband correlated noises. Because chirp and other deterministic time-varying signals have distinct signatures in time-frequency space, we have introduced a cross relation based MFP in time-frequency space that incorporates these signatures in the processing to counter the effect of broadband correlated noises in the received data. Additionally, white noise can be removed by using a Cross-cross relation (CCR) time-frequency (TF)-MFP that contains only cross components of the received signals in its structure. We demonstrate the advantages of this process over the conventional MFPs to date.

关键词： Acoustic measurements Acoustic sensors Equations Oceans Sea measurements sensor arrays signal processing Time frequency analysis Transfer functions White noise

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The time-reversal technique re-interpreted: Subspace-based signal processing for multi-static target location

The time-reversal technique re-interpreted: Subspace-based s...

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1st ieee sensor array and multichannel signal processing workshop (SAM 2000)

作者： Lev-Ari, H Devaney, AJ Northeastern Univ Dept Elect & Comp Engn Boston MA 02115 USA

ISBN: (纸本)0780363396

The time-reversal field processing technique for active remote sensing is re-interpreted in the conceptual framework of subspace-based signal processing. Although both of these subjects have a long and extensive history, we believe that our work is the first to relate these two rather different signal processing approaches. The concept at the heart of both approaches is the multi-static Frequency Response Matrix (FRM) of the active antenna array. It is a function of array/scatterer geometry (via a suitable Green function) and of the scatterers' reflection coefficients. The time-reversal technique locates the most reflective target by determining the largest singular value of the FRM and the corresponding (right) singular vector. Consequently, it performs well only when the scatterers are "well-resolved," i.e., when the Green function vectors used in forming the FRM are approximately orthogonal to each other. In contrast, we show that subspace based signal processing can be successfully used even when the targets are not "well-resolved.'' Moreover, the performance of our subspace-based schemes can be further enhanced by using wideband signals and combining FRM information from multiple frequencies. The coupling of time-reversal field. processing with subspace methods leads to a powerful approach for locating targets under arbitrary wave-propagation conditions (both near-field and far-field), including arbitrary non-homogeneous media and arbitrary geometries.

关键词： Antenna arrays array signal processing Frequency response Geometry Green function Heart History Remote sensing Scattering signal processing

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Airborne phased array radar: Clutter and jamming suppression and moving target detection and feature extraction

Airborne phased array radar: Clutter and jamming suppression...

引用

1st ieee sensor array and multichannel signal processing workshop (SAM 2000)

作者： Li, J Liu, GQ Jiang, NZ Stoica, P Univ Florida Dept Elect & Comp Engn Gainesville FL 32611 USA

ISBN: (纸本)0780363396

We study the clutter and jamming suppression and moving target detection and feature extraction for airborne phased array radars. For the Low Range Resolution (LRR) case, we use a Vector Autoregressive (VAR) filtering technique for clutter suppression and compare it to the Displaced Phase Center Antenna (DPCA) processing technique in the presence of various system errors. For the High Range Resolution (HRR) case, we first show how to use the VAR filter to suppress the ground clutter. We divide the HRR range profiles into LRR segments to avoid the range migration problems that occur in the HRR radar data. Since each LRR segment contains a sequence of HRR range bins, no loss of resolution occurs due to the division. We then propose new algorithms for moving target detection and super resolution feature extraction. Numerical results are provided to demonstrate the effectiveness of the proposed algorithms.

关键词： Airborne radar Feature extraction Filtering Filters Jamming Object detection Phased arrays Radar antennas Radar clutter Reactive power

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Stochastic partial update LMS algorithm for adaptive arrays

Stochastic partial update LMS algorithm for adaptive arrays

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1st ieee sensor array and multichannel signal processing workshop (SAM 2000)

作者： Godavarti, M Hero, AO Univ Michigan Dept Elect Engn & Comp Sci Ann Arbor MI 48109 USA

ISBN: (纸本)0780363396

Partial updating of LMS filter coefficients is an effective method for reducing the computational load and the power consumption in adaptive filter implementations. Several algorithms have been proposed in the literature based on partial updating. Unfortunately, it has been observed that these algorithms don't have good convergence properties in practice. In particular, there generally exist signals for which these algorithms stagnate or diverge. In this paper, we propose a new algorithm, called the Stochastic Partial Update LMS (SPU-LMS) algorithm which attempts to remedy some of the drawbacks of existing algorithms. The SPU-LMS algorithm differs from the existing algorithms in that the subsets to be updated are chosen in a random manner at each iteration. We derive conditions for filter stability, convergence rate, and steady state error for the proposed algorithm under an independent snapshots assumption in the context of adaptive beamforming for antenna arrays. We verify the analysis via computer simulations and illustrate the advantages of the new algorithm through examples.

关键词： Adaptive arrays Adaptive filters array signal processing Computer errors Convergence Energy consumption Least squares approximation Stability Steady-state Stochastic processes

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Beamforming, Doppler compensation, and differential space-time coding

Beamforming, Doppler compensation, and differential space-ti...

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1st ieee sensor array and multichannel signal processing workshop (SAM 2000)

作者： Ingram, MA Li, KH Nguyen, AV Pratt, T Georgia Inst Technol Sch Elect & Comp Engn Atlanta GA 30332 USA

ISBN: (纸本)0780363396

We consider the design of two beams, formed at the base station, that will serve as the two channels for differential space-time block coding in a high-mobility outdoor scenario. The mobile is assumed to have one omnidirectional antenna. In one approach, we consider selecting, assuming no interference, the two most powerful beams produced by an RF beamformer. In the other approach, we make a compromise between power and Doppler spectrum partitioning. One channel is a beam that points directly at the mobile. The other beam is the difference of left and right beams, symmetric about the mobile. These left and right beams are Doppler compensated prior to forming their difference. Using computer simulation, we find that selecting beams based on power is more effective than selecting beams based on Doppler partitioning. We also find that using the two most powerful beams formed from eight antenna elements yields 7 dB and 8 dB SNR improvements, at 1.9 GHz and 5.8 GHz, respectively, relative to spatial diversity with two omnidirectional antennas.

关键词： array signal processing Azimuth Base stations Block codes Chebyshev approximation Design engineering Diversity reception Downlink Mobile antennas Space technology

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Early experimental results for a large-aperture microphone-array system

Early experimental results for a large-aperture microphone-a...

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1st ieee sensor array and multichannel signal processing workshop (SAM 2000)

作者： Silverman, HF Patterson, WR Sachar, JM Brown Univ Div Engn LEMS Providence RI 02912 USA

ISBN: (纸本)0780363396

Microphone arrays offer a potential solution to the important problem of obtaining high-quality sound pick-up from remote acoustic sources. Most reported results for microphone array technology have been theoretical, proposed algorithms with predicted performance, or based on ideal or near-ideal simulations of microphonearray systems and room environments. Previous work that is based on real data has usually been for a small number of microphones. In this paper, for the first time, measured results for a large number of microphones, 256, in a large-aperture, real-time, all DSP system are reported. The array system is mounted in a moderate-sized (8Mx8M) room with a significant reverberation time. While listeners to the locationing/tracking/beamforming system claim to hear near-close-talking microphone performance when the system acquires a talker, our measured results of the real-time system indicate that a true 9dB separation of a source from other sources in the room has been achieved and that reverberant effects have been significantly reduced. This level of isolation is maintained at all locations throughout the room. In addition to reporting these early performance measures, this paper discusses the nuances of the real-time algorithms, the latency in such systems, and introduces some of the issues of making real-time, real-room measurements.

关键词： Acoustic arrays array signal processing Delay Digital signal processing Microphone arrays Prediction algorithms Predictive models Real time systems Reverberation Time measurement

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Approximate Maximum Likelihood DOA estimation in multiplicative noise environments

Approximate Maximum Likelihood DOA estimation in multiplicat...

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1st ieee sensor array and multichannel signal processing workshop (SAM 2000)

作者： Besson, O Vincent, F Stoica, P Gershman, AB ENSICA Dept Avion & Syst Toulouse France

ISBN: (纸本)0780363396

We consider the problem of localizing a source by means of a uniform linear array of sensors when the received signal is corrupted by multiplicative noise. Since the exact Maximum Likelihood (ML) estimator is computationally intensive, two approximate solutions are proposed, originating from the analysis of the high and low signal to Noise Ratio (SNR) cases, respectively. First, starting with the no additive noise case, a very simple approximate ML (AML(1)) estimator is derived. A theoretical expression for its asymptotic variance in the presence of additive noise is derived. It shows that the AML1 estimator has a performance close to the Cramer-Rao bound (CRB) for moderate to high SNR. Next, the low SNR case is considered and the corresponding AML(2) solution is derived. It is shown that the approximate ML criterion can be concentrated with respect to (w.r.t.) both the multiplicative and additive noise powers, leaving out a 2-D minimization problem instead of a 4-D problem required by the exact ML. Numerical results illustrate the performance of the estimators and confirm the validity of the theoretical analysis.

关键词： Additive noise Direction of arrival estimation Integrated circuit modeling Integrated circuit noise Maximum likelihood estimation Parameter estimation Scattering sensor arrays signal to noise ratio Working environment noise

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