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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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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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signal processing algorithms by permutation test in radar application 6

Signal processing algorithms by permutation test in radar ap...

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Conference on advanced signal processing algorithms, architectures, and implementations vI

作者： Alvarezvaquero, F SanzGonzalez, JL UNIV POLITECN MADRID ETSI TELECOMUNDEPT SIGNAL SYST & RADIOCOMMUNE-28040 MADRIDSPAIN

ISBN: (纸本)0819422347

A hypothesis H is parametric if every distribution from the process defined by H belongs to a family of distributions characterized by a finite number of parameters; on the other hand, if the distribution can not be defined by a finite number of parameters, the hypothesis is nonparametric. In this paper, we analyze a detector based on the optimum permutation test, applied to nonparametric radar detection which provide good performances without a large computational work, and we compare it with the parametric test and rank test in the Neyman-Pearson sense. The computational complexity of the detector is high and its implementation in real time is difficult, due to the number of operations increase with the factorial of the number of samples. Also, we present an algorithm that reduces the computational work required. We also present the detectability characteristic of the optimum permutation test against rank test and parametric test under Gaussian noise environments and different types of target models (nonfluctuating, Swerling I and Swerling II). The detection probability versus signal-to-noise ratio is estimated by Monte-Carlo simulations for different parameter values (N pulse, M reference samples and false alarm probability P_fa).

关键词： nonparametric permutation test rank test radar detection detectability target models

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Design and evaluation of automated checks for signal processing applications 6

Design and evaluation of automated checks for signal process...

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Conference on advanced signal processing algorithms, architectures, and implementations vI

作者： Nair, vSS Kim, HC Krishnamurthy, N Abraham, JA SO METHODIST UNIV DEPT COMP SCI & ENGNDALLASTX 75275

ISBN: (纸本)0819422347

Most of the signal processing application programs involve computationally intensive iterative steps. In such programs, various failures in the underlying hardware manifest as control-flow errors that affect the reliability of the computed results. various techniques have been proposed in the past to detect and recover from such control-flow errors. Unfortunately, all these techniques need either additional hardware or modification of the hardware and are not portable across various platforms. To circumvent these limitations, recently we have developed a high-level control-flow checking approach using assertions (CCA). In CCA, branch-free intervals in a given high-level language program are identified and the entry and exit points of the intervals are fortified through pre-inserted assertions. In this paper we describe an implementation of CCA through a pre-processor that will automatically insert the necessary assertions into a high-level language program. Based on the implementation we study the fault detection capabilities of CCA with the help of fault injection experiments using FERRARI.

关键词： Simulation of CCA and DLA aggregates Switches signal processing Reliability Computing systems Computer engineering Data modeling Error analysis Error control coding Computer science

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Binary openings and closings in one pass using finite-state machines 6

Binary openings and closings in one pass using finite-state ...

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Conference on advanced signal processing algorithms, architectures, and implementations vI

作者： Waltz, FM Consultant (United States)

ISBN: (纸本)0819422347

In a series of ten papers published since 1994, a radically new technique for implementing a wide range of standard image processing operations has been presented, under the acronym SKIPSM (Separated-Kernel Image processing using finite State Machines). Key steps are: (1) the operation is separated into a row operation followed by a column operation, (2) these row and column operations are put in recursive form. That is, in a form compatible with either one-step software implementation or pipelined raster-scan hardware implementation, (3) the resulting operations are realized as FSMs (finite-state machines), and (4) these FSMs are implemented in software or in inexpensive off-the-shelf integrated circuits. Note that this technique does not require separability, in the usual sense. In this paper, the SKIPSM technique is applied to computing binary openings and closings in one pass using arbitrary binary structuring elements. Whether the resulting finite state machines are implemented in software or hardware, the result is generally much faster and/or much cheaper than conventional implementations. Furthermore, this same SKIPSM architecture is highly versatile and programmable, allowing it to be software-reconfigured to perform hundreds of other software- based or pipelined image processing operations, such a binary and grey-level morphology, the Grassfire Transform, binary and grey-level template matching, binary skeletonization, etc.

关键词： fast implementations image processing finite-state machines separability pipelining real-time inspection

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An adaptive algorithm for interference cancelling in array processing 6

An adaptive algorithm for interference cancelling in array p...

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Conference on advanced signal processing algorithms, architectures, and implementations vI

作者： Luk, FT Qiao, SZ RENSSELAER POLYTECH INST DEPT COMP SCITROYNY 12180

ISBN: (纸本)0819422347

In array processing, one technique for cancelling interference in the presence of colored noise is the ULLv decomposition of a pair of matrices. The factorization is stable and accurate, and is easy to update when a row is added to either one of the two matrices. In earlier work, we made the assumptions that both matrices must have at least as many rows as columns and that the matrix representing interference must have full column rank. In this paper, we relax the latter restriction and allow the interference matrix to be rank deficient. We present an algorithm for computing the decomposition and two techniques for updating it. The details on updating are quite different from those in the full rank case. © 1996 SPIE. All rights reserved.

关键词： ULLv decomposition matrix updating noise subspace tracking

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Time sequenced adaptive filtering using a modified P-vector algorithm 6

Time sequenced adaptive filtering using a modified P-vector ...

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Conference on advanced signal processing algorithms, architectures, and implementations vI

作者： Williams, R USAF WRIGHT LABAUTOMAT TARGET RECOGNIT DEV BRANCHWRIGHT PATTERSON AFBOH 45433

ISBN: (纸本)0819422347

An adaptive algorithm and two stage filter structure were developed for adaptive filtering of certain classes of signals that exhibit cyclostationary characteristics. The new modified P-vector algorithm (mPa) eliminates the need for a separate desired signal which is typically required by conventional adaptive algorithms. It is then implemented in a time-sequenced manner to counteract the nonstationary characteristics typically found in certain radar and bioelectromagnetic signals. Initial algorithm testing is performed on evoked responses generated by the visual cortex of the human brain with the objective, ultimately, to transition the results to radar signals. Each sample of the evoked response is modeled as the sum of three uncorrelated signal components, a time-varying mean (M), a noise component (N), and a random jitter component (Q). A two stage single channel time-sequenced adaptive filter structure was developed which improves convergence characteristics by de coupling the time-varying mean component from the `Q' and noise components in the first stage. The EEG statistics must be known a priori and are adaptively estimated from the pre stimulus data. The performance of the two stage mPa time-sequenced adaptive filter approaches the performance for the ideal case of an adaptive filter having a noiseless desired response.

关键词： time sequenced adaptive filter EEG biomedical signal processing radar signal processing

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Convergence results on adaptive approximate filtering 6

Convergence results on adaptive approximate filtering

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Conference on advanced signal processing algorithms, architectures, and implementations vI

作者： Ludwig, JT Nawab, SH Chandrakasan, AP MIT ELECT RES LABCAMBRIDGEMA 02139

ISBN: (纸本)0819422347

We present an IIR filtering technique based upon a recently proposed approach for reducing power consumption in implementation of frequency-selective filters. The basic idea in such techniques is to utilize the most recent input and output signal samples to estimate the current SNR (defined as the ratio of the in-band signal power to the out-of-band signal power) at the filter's input. This estimated input SNR is then used to update the filter order to the minimum value which would guarantee a minimum tolerable SNR at the filter's output. A key issue addressed in this paper is how well the estimated filter order converges to the theoretical minimum for situations satisfying the assumptions behind the derivation of the technique. Experimental results are used to verify that convergence to the correct filter order depends (1) upon the number of input and output samples used for estimating the input SNR, (2) upon the filter order applied in generating the output samples that are used in estimating the input SNR and (3) upon the proximity of the actual input SNR to boundaries in the input-SNR space corresponding to changes in the optimal choice for filter order.

关键词： low power approximate processing incremental refinement structure frequency-selective filtering Butterworth filters adaptive filtering

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A DSP implementation of source location using microphone arrays 6

A DSP implementation of source location using microphone arr...

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Conference on advanced signal processing algorithms, architectures, and implementations vI

作者： Rabinkin, Dv Renomeron, RJ Dahl, A French, JC Flanagan, JL Bianchi, MH RUTGERS STATE UNIV CTR COMP AIDS IND PRODPISCATAWAYNJ 08855

ISBN: (纸本)0819422347

The design, implementation, and performance of a low-cost, real-time DSP system for source location is discussed. The system consists of an 8-element electret microphone array connected to a signalogic DSP daughterboard hosted by a PC. The system determines the location of a speaker in the audience in an irregularly shaped auditorium. The auditorium presents a non-ideal acoustical environment;some of the walls are acoustically treated, but there still exists significant reverberation and a large amount of low frequency noise from fans in the ceiling. The source location algorithm is implemented in a two step process: The first step determines time delay of arrival (TDOA) for select microphone pairs. A modified version of the Cross-Power Spectrum Phase Method is used to compute TDOAs and is implemented on the DSP daughterboard. The second step uses the computed TDOAs in a least mean squares gradient descent search algorithm implemented on the PC to compute a location estimate. © 1996 SPIE. All rights reserved.

关键词： microphone arrays speech source location real-time processing

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CISC, RISC AND DSP PROCESSORS IN REAL-TIME signal-processing AND CONTROL

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MICROPROCESSORS AND MICROSYSTEMS 1995年第5期19卷 291-300页

作者： TOKHI, MO HOSSAIN, MA Department of Automatic Control and Systems Engineering The University of Sheffield PO Box 600 Mappin Street Sheffield S1 4DU UK

This paper presents an investigation into the performance evaluation of advanced complex instruction set computer (CISC) processors, reduced instruction set computer (RISC) processors and digital signal processing (DSP) devices in the real-time implementation of signal processing and control algorithms. The algorithms considered are of varying degree of regularity relative to one another. These include a fast Fourier transform algorithm, a second-order correlation algorithm, two adaptive filtering algorithms, a simulation algorithm of a flexible manipulator system and simulation, identification and active vibration control algorithms of a flexible beam system. The algorithms are implemented on an 80486DX2 CISC processor, an 80386DX CISC processor, a SPARC TMS390S10 RISC processor, a T805 RISC processor, an 80i860 RISC processor and a TMS320C40 DSP device. The hardware and software resources and capabilities of the processors and the characteristics of the algorithms are discussed to provide a matching between the algorithms and the architectures. Finally, a comparison of the results of the implementations, on the basis of real-time computation performance, is made and discussed.

关键词： DIGITAL signal processing COMPLEX INSTRUCTION SET COMPUTER REDUCED INSTRUCTION SET COMPUTER

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