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Seismic-array signal processing for moving source localization

Seismic-array signal processing for moving source localizati...

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conference on advanced signal processing algorithms, architectures, and implementations XVII

作者： Stafsudd, Jing Z. Hudson, Ralph E. Taciroglu, Ertugrul Yao, Kung UCLA Dept Elect Engn Los Angeles CA 90095 USA UCLA Dept Civil & Environm Engn Los Angeles CA 90095 USA

ISBN: (纸本)9780819468451

In this paper, we consider the use of a seismic sensor array for the localization and tracking of a wideband moving source. The proposed solution consists of two steps: source Direction-Of-Arrival (DOA) estimation and localization via DOA estimates. Three DOA estimation methods are considered. The Covariance Matrix Analysis and the Surface Wave Analysis are previously published DOA estimation algorithms shown to be effective in the localization of a stationary wideband source. This paper investigates their performance on moving wideband sources. A novel DOA estimation algorithm, the Modified Kirlin's Method was also developed for the localization of a moving Source. The DOAs estimated by these algorithms are combined rising a least-squares optimization for source localization. The application of these algorithms to real-life data show the effectiveness of both the Surface Wave Analysis and the Modified Kirlin's Method in locating and tracking a wideband moving source.

关键词： direction-of-arrival (DOA) source localization mover localization seismic signal processing surface waves

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Integrated numeric and symbolic signal processing using a heterogeneous design environment 6

Integrated numeric and symbolic signal processing using a he...

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conference on advanced signal processing algorithms, architectures, and implementations VI

作者： Mani, R Nawab, SH Winograd, JM Evans, BL BOSTON UNIV BOSTONMA 02215

ISBN: (纸本)0819422347

We present a solution to a complex multi-tone transient detection problem to illustrate the integrated use of symbolic and numeric processing techniques which are supported by well-established underlying models. Examples of such models include synchronous dataflow for numeric processing and the blackboard paradigm for symbolic heuristic search. Our transient detection solution serves to emphasize the importance of developing system design methods and tools which can support the integrated use of well- established symbolic and numerical models of computation. Recently, we incorporated a blackboard-based model of computation underlying the Integrated processing and Understanding of signals (IPUS) paradigm into a system-level design environment for numeric processing called Ptolemy. Using the IPUS/Ptolemy environment, we are implementing our solution to the multi-tone transient detection problem.

关键词： design environments symbolic signal processing heuristic search blackboard model synchronous dataflow model Ptolemy Integrated processing and Understanding of signals (IPUS) multi-tone transient detection Adaptive Time-Frequency Representation (ATFR)

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A novel signal detection technique based on generalized scale transforms - art. no. 63130Y

A novel signal detection technique based on generalized scal...

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conference on advanced signal processing algorithms, architectures, and implementations XVI

作者： Nickel, Robert M. Penn State Univ Dept Elect Engn University Pk PA 16802 USA

ISBN: (纸本)0819463922

We are presenting an extension to the classic multiple signal classification method (MUSIC) developed by Schmidt and Bienvenu in 1979.(1) While the classic MUSIC algorithm is limited to the detection of constant frequency sinusoids in white noise, the proposed new method is capable of detecting signals with a continuously varying instantaneous frequency. The method is based on the development of a discrete-time version of the generalized scale transform (GST) which was introduced by Nickel and Williams in 1999.(2) As a byproduct we obtain techniques for discrete-time warp-shift invariant filtering which can be used in addition to the signal detection to separate signals with different instantaneous frequency contours.

关键词： warping functions sampling theory generalized scale transform signal detection multiple signal classification (MUSIC)

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A unified Bayesian framework for algorithms to recover blocky signals

A unified Bayesian framework for algorithms to recover block...

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conference on advanced signal processing algorithms, architectures, and implementations XVII

作者： Calvetti, Daniela Somersalo, Erkki Case Western Reserve Univ Dept Math Cleveland OH 44106 USA Case Western Reserve Univ Ctr Modeling Integrated Metab Syst Cleveland OH 44106 USA Aalto Univ Dept Math Helsinki Finland

ISBN: (纸本)9780819468451

We consider the problem of recovering signals from noisy indirect observations under the additional a priori information that the signal is believed to be slowly varying except at all unknown number of points where it may have discontinuities of unknown size. The model problem is a linear deconvolution problem. To take advantage of the qualitative prior information available, we use a non-stationary Markov model with the variance of the innovation process also unknown, and apply Bayesian techniques to estimate both the signal and the prior variance. We propose a fast iterative method for computing a MAP estimates and we show that, with a rather standard choices of the hyperpriors, the algorithm produces the fixed point iterative solutions of the total variation and of the Perona-Malik regularization methods. We also demonstrate that, unlike the non-statistical estimation methods, the Bayesian approach leads to a very natural reliability assessment of edge detection by a Markov Chain Monte Carlo (MCMC) based analysis of the posterior.

关键词： signal processing Data modeling Reliability Stochastic processes Atrial fibrillation Autoregressive models Deconvolution Monte Carlo methods Statistical analysis Mathematics

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Missile signal processing common computer architecture for rapid technology upgrade

Missile signal processing common computer architecture for r...

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conference on advanced signal processing algorithms, architectures, and implementations XIV

作者： Rabinkin, D Rutledge, E Monticciolo, P MIT Lincoln Lab Lexington MA 02420 USA

ISBN: (纸本)0819454974

Interceptor missiles process IR images captured by a focal plane array for locating and guiding the interceptor toward an intended target. A typical interceptor signal processing chain comprises two parts. Front-end video processing operates on all pixels of the image and performs such operations as non-uniformity correction (NUC), image stabilization, frame integration and detection. Back-end target processing, which tracks and classifies targets detected in the image, performs such algorithms as Kalman tracking, spectral feature extraction and target discrimination. signal processing requirements increased as sensor bandwidth increases and interceptors operate against more sophisticated targets. In the past, video processing was implemented using ASICs or FPGAs because computation requirements exceeded the throughput of general-purpose processors. Target processing was performed using hybrid architectures that included ASICs. DSPs and general-purpose processors. The resulting systems tended to be very function-specific, and required custom software development. These systems were developed using non-integrated toolsets, and test equipment had to be developed along with the processor platform. The lifespan of the interceptor platform on which the signal processors operate often spans decades, while the specialized nature of processsor hardware and software makes it difficult and costly to upgrade. As a result., the signal processing systems often run on outdated components, signal processing algorithms are difficult to update. and system effectiveness is impaired by the inability to rapidly, respond to new threats. A new design approach to interceptor signal processing is made possible by three developments: Moore's Law - driven improvement in computational throughput;a newly introduced vector computing capability in general-purpose processors and a modern set of open interface software standards. Today's multiprocessor commercial-off-the-shelf (COTS) platforms hav

关键词： signal processing Digital signal processing Target detection Detection and tracking algorithms Image processing Standards development MATLAB Commercial off the shelf technology signal to noise ratio Software development

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Application of time-frequency analysis methods to speaker verification

Application of time-frequency analysis methods to speaker ve...

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conference on advanced signal processing algorithms, architectures, and implementations XVI

作者： Williams, W. J. Bossemeyer, R. W. Quantum Signal LLC Ann Arbor MI 48176 USA Univ Michigan EECS Ann Arbor MI 48109 USA Speech Technol Appl Res Inc St Charles IL USA

ISBN: (纸本)0819463922

Time-Frequency Analysis has previously been successfully applied to characterize and quantify a variety of acoustic signals, including marine mammal sounds. In this research, Time-Frequency analysis is applied to human speech signals in an effort to reveal signal structure salient to the biometric speaker verification challenge. Prior approaches to speaker verification have relied upon signal processing analysis such as linear prediction or weighted Cepstrum spectral representations of segments of speech and classification techniques based on stochastic pattern matching. The authors believe that the classification of identity of a speaker based on time-frequency representation of short time events occurring in speech could have substantial advantages. Using these ideas, a speaker verification algorithm was developed(1) and has been refined over the past several years. In this presentation, the authors describe the testing of the algorithm using a large speech database, the results obtained, and recommendations for further improvements.

关键词： time-frequency speaker verification multimodal biometrics homeland security pattern recognition

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Modern microphone array for hearing aid and speech processing 6

Modern microphone array for hearing aid and speech processin...

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conference on advanced signal processing algorithms, architectures, and implementations VI

作者： Wang, A Yao, K Hudson, RE Korompis, D Lorenzelli, F Soli, SD Gao, S UNIV CALIF LOS ANGELES DEPT ELECT ENGNLOS ANGELESCA 90095

ISBN: (纸本)0819422347

For various audio, teleconference, hearing aid, and voice recognition applications, a microphone array is known to be an effective method to enhance the SNR in noisy environments resulting in significant improvement of speech intelligibility or recognition. We propose an novel electronically steerable microphone array based on the Maximum Energy concentration criterion to form a focused beam toward the desired speech source, attenuating background noises and rejecting discrete spatial interferers. The design and implementation of a prototype DSP-based microphone array system are described. Some details on microphone measurement, calibration, and optimization needed to achieve a high performance microphone array are discussed. Computer simulated and measured array performances are presented to show the effectiveness of the array system.

关键词： Sensors Calibration signal to noise ratio Digital signal processing Computer simulations Near field Acoustics Computing systems signal processing Speaker recognition

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A multivariate speech activity detector based on the syllable rate

A multivariate speech activity detector based on the syllabl...

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SPIE conference on advanced signal processing algorithms, architectures, and implementations VIII

作者： Smith, D Townsend, J Nelson, DJ US Dept Def Ft George G Meade MD 20755 USA

ISBN: (纸本)0819429163

Computationally efficient algorithms which perform speech activity detection have significant potential economic and labor saving benefit, by automating an extremely tedious manual process. In many applications, it is desirable to extract intervals of speech which are bounded by segments of other signal types (fax/modem, music, static, dial tones, etc.). In the past, algorithms which successfully discriminate between speech and one specific other signal type have been developed. Frequently, these algorithms fail when the specific non-speech signal is replaced by a different non-speech signal. Little work has been done on combining such discriminators in order to solve the general speech vs. non-speech discrimination problem. Typically, several signal specific discriminators are blindly combined with predictable negative results. Moreover, when a large number of discriminators are involved, dimension reduction is achieved using Principal Components, which optimally compresses signal variance into the fewest number of dimensions. Unfortunately, these new coordinates are not necessarily optimal for discrimination. In this paper we apply graphical tools to determine a set of discriminators which produce excellent speech vs. non-speech clustering, thereby eliminating the guesswork in selecting good feature vectors. This cluster structure provides a basis for a general multivariate speech vs. non-speech discriminator, which compares very favorably with the TALKATIVE speech extraction algorithm.

关键词： signal processing

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Performance of coherent adaptation algorithms for the WCDMA smart antenna system in Rayleigh flat-fading channel

Performance of coherent adaptation algorithms for the WCDMA ...

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conference on advanced signal processing algorithms, architectures, and implementations XI

作者： Kong, DG Lee, K Chun, J Choi, P Yun, JC Lee, SJ Lee, JH Samsung Adv Inst Technol Human Comp Interact Lab Suwon South Korea

ISBN: (纸本)0819441880

Coherent adaptation algorithms are proposed for a beamformer in the WCDMA reverse link. The coherent adaptation algorithm improves the convergence speed by constraining the desired signal component of the filtered output to be always coherent in phase with the reference signal at each iteration. With the coherent constraint, the performance of coherent methods is significantly improved. We present the simulation results that show the various desirable characteristics of coherent methods such as a fast convergence speed. insensitiveness to fading parameters over conventional methods.

关键词： coherent adaptation algorithm smart antenna WCDMA array signal processing

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Performance of space-frequency codes for MIMO OFDM systems

Performance of space-frequency codes for MIMO OFDM systems

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conference on advanced signal processing algorithms architectures and implementations xiiI

作者： Wang, JB Yao, K Univ Calif Los Angeles Dept Elect Engn Los Angeles CA 90095 USA

ISBN: (纸本)0819450782

Concatenation of space-time (ST) coding with orthogonal frequency-division multiplexing (OFDM) has gained much interest recently. In this work, we derive the exact pairwise error probability (PEP) of space-frequency (SF) codes for MIMO OFDM Systems. Based on the exact PEP, we derive the tighter upper and lower bounds for the PEP. For asymptotically high SNRs, the design criteria for SF codes differ significantly from those for ST codes over flat fading channels. In this paper, by drawing an analogy between SF and ST codes, we show that when the number of receive antennas is large, the minimum Euclidean distance among code words dominates the performance of SF codes. Therefore, SF codes can be optimized by using the Euclidean-distance criterion valid for AWGN channels. Simulation results are given to show that the results valid for a number of receive antennas tending to infinity still provide correct indications when the number of antennas is small.

关键词： Orthogonal frequency division multiplexing Antennas Modulation Lutetium signal to noise ratio Computer simulations Demodulation signal processing

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