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作者： B.M. Howe S.G. Anderson A. Baggeroer J.A. Colosi K.R. Hardy D. Horwitt F.W. Karig S. Leach J.A. Mercer K. Metzger L.O. Olson D.A. Peckham D.A. Reddaway R.R. Ryan R.P. Stein K. von der Heydt J.D. Watson S.L. Weslander P.F. Worcester Applied Physics Laboratory University of Washington Seattle WA USA Massachusetts Institute of Technology Cambridge MA USA Scripps Institution of Oceanography University of California San Diego Seattle WA USA Scripps Institution of Oceanography University of California San Diego La Jolla CA USA Communications and Signal Processing Laboratory Electrical Engineering and Computer Science Department University of Michigan Ann Arbor MI USA Woods Hole Oceanographic Institution Seattle WA USA Applied Physics Laboratory University of Washington Woods Hole MA USA

The prototype Pacific Ocean network for the Acoustic Thermometry of Ocean Climate (ATOC) program consists of two shore-cabled acoustic sources, two autonomous vertical hydrophone receiving arrays, and 16 U.S. Navy hydrophone arrays. The sources, receivers, and signaling strategy are designed to be capable of up to 10-megameter transmissions on a routine basis. A new broadband, low-frequency (75 Hz), acoustic source was designed and built by Alliant Techsystems. The bender-bar/barrel-stave transducer is pressure compensated for operation to 1000-m depth. signal waveform shaping is employed to broaden the effective bandwidth of the source to 37.5 Hz (Q=2; time resolution of 27 ms). Four-element vertical hydrophone arrays are attached to the sources to permit reciprocal transmissions and ambient noise monitoring. Autonomous vertical hydrophone arrays consisting of 40 elements at 35-m spacing (1400-m aperture) were developed to resolve low-order acoustic modes. The autonomous arrays are designed for deployments of up to 1 year, with intermediate readout of the data via ship-recoverable pop-up data modules. The U.S. Navy hydrophone arrays were instrumented with 80486-based data acquisition and processing systems, with communications links for transmitting the data in near real time to a central site for further processing and archiving.

关键词： Instruments Oceans Sonar equipment Acoustic arrays Prototypes signal design Acoustic transducers Bandwidth signal resolution Acoustic noise

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WIDE-AREA ADAPTIVE ACTIVE NOISE CANCELLATION

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IEEE TRANSACTIONS ON CIRCUITS AND SYSTEMS II-ANALOG AND DIGITAL signal processing 1994年第6期41卷 405-409页

作者： EFRON, AJ HAN, LC Signal and Image Research Lab Department of Electrical Engineering and Computer Science University of Illinois Chicago IL USA

This paper describes an adaptive active noise cancellation (ANC) system that is based on an output-whitening approach and is an extension of the work of Graupe and Efron on single point adaptive ANC. The controller works by continually estimating the time-series parameters of the noise to be canceled and by forcing the cancellation network to follow the identified parameters, so that the output is whitened. To address the problem of wide-area cancellation, a system that consists of a weighted array of single point output-whitening cancellation systems is proposed. Each controller is chosen to cancel the local temporal effects of the noise, and the array weights are chosen to match the wavefront shape. A comparison of cancellations based on single-point and wide-area systems demonstrates the superiority of the wide-area system.

关键词： Noise cancellation Acoustic noise Loudspeakers signal processing algorithms Frequency Parameter estimation Shape control Noise shaping Active noise reduction Noise level

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Adaptive multiuser detector for synchronous code division multiple access applications 13th

Adaptive multiuser detector for synchronous code division mu...

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13th International Zurich Seminar on Digital communications, 1994

作者： Siveski, Zoran Bar-Ness, Yeheskel Chen, David W. Center for Communications and Signal Processing Electrical and Computer Engineering Department New Jersey Institute of Technology University Heights NewarkNJ07102 United States Delft University of Technology Faculty of Electrical Engineering Telecommunications and Traffic-Control Systems Group Delft Netherlands Center for Communications and Signal Processing Research NJIT NewarkNJ07102 United States

ISBN: (纸本)9783540578567

A two-stage multiuser detector for the code division multiple access environment employing a combination of a decorrelator and a nonlinear multiuser interference canceler is considered. The weights of the canceler are adaptively adjusted, thus rendering the knowledge of the received signal energies unnecessary. The probability of error is evaluated and compared to those of the decorrelator and a similar two-stage, fixed-weights scheme which requires an estimation of signal energies. © Springer-Verlag Berlin Heidelberg 1994.

关键词： Code division multiple access

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2-D IIR digital filters for deinterlacing of video signals

2-D IIR digital filters for deinterlacing of video signals

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1993 Canadian Conference on electrical and computer engineering, CCECE 1993

作者： Gu, Q.S. Lee, L.C.K. Swamy, M.N.S. Centre for Signal Processing and Communications Department of Electrical and Computer Engineering Concordia University MontrealQCH3G 1M8 Canada

ISBN: (纸本)0780314433

Investigates the application of 2D IIR digital filters for deinterlacing of video signals. A special class of diamond filters is presented, which employs only two coefficients and has approximately zero-phase response. The implementation of this filter is very efficient and the stability is guaranteed even under finite arithmetic conditions. This filter is tested with a group of sequences. Very promising results are obtained. © 1993 IEEE.

关键词： IIR filters

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Performance of bus allocation policies for an ATM switch under bursty arrivals and correlated destinations 1

Performance of bus allocation policies for an ATM switch und...

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1st IEEE Symposium on Global Data Networking, GDN 1993

作者： Zaghloul, A.O. Perros, H.G. Viniotis, I. IBM Corporation Department C15 RTP NC27709 United States Department of Computer Science Center for Communications and Signal Processing North Carolina State University RaleighNC27695-8206 United States Department of Electrical and Computer Engineering Center for Communications and Signal Processing North Carolina State University RaleighNC27695-8208 United States

ISBN: (纸本)081864270X

Studies, via simulation, the performance of different bus allocation policies for a generic shared-medium ATM switch with input and output queueing. The policies differ in the amount of information used by the bus arbitrator to decide which input queue to serve next. Classical polling systems (e.g. Round-Robin) consider only the state of the input queues. This may waste bus bandwidth on cells that are destined to output queues which are full. The present authors consider a range of bus service policies that are based on the state of the input queues, or the state of both the input and the output queues. The performance of the switch is analyzed under realistic system characteristics such as bursty traffic, limited buffer size, asymmetric load conditions, and nonuniform destinations. Input traffic is modelled by an interrupted Bernoulli process (IBP). Furthermore, it is assumed that the destinations of the arriving cells within the same burst are correlated. Two types of destination correlations are introduced and their effect on the performance of the switch is analyzed. The performance measures considered are: cell loss probability, and mean cell delay. © 1993 IEEE.

关键词： Cells

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Performance of bus allocation policies for an ATM switch under bursty arrivals and correlated destinations

Performance of bus allocation policies for an ATM switch und...

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Proceedings of the Global Data Networking

作者： A.O. Zaghloul H.G. Perros I. Viniotis Department C15 IBM Corporation Research Triangle Park NC USA Department of Computer Science and Center for Communications and Signal Processing North Carolina State University Raleigh NC USA Department of Electrical and Computer Engineering and Center for Communications and Signal Processing North Carolina State University Raleigh NC USA

关键词： Switches Asynchronous transfer mode Packet switching Communication switching signal processing computer science Computational modeling computer simulation Bandwidth Performance loss

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A theoretical framework for a class of frequency estimation algorithms

A theoretical framework for a class of frequency estimation ...

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IEEE Workshop on Statistical signal and Array processing

作者： R.M. Todd J.R. Cruz Communications and Signal Processing Laboratory (CMP Lab) School of Electrical Engineering and Computer Science University of Oklahama Norman OK USA

This paper discusses a general class of algorithms for estimating the frequencies of a set of complex exponentials, and presents a corrected proof of the validity of the algorithms when applied to either real or complex data. The linear-prediction least-squares algorithms, involve the formulation of the estimation problem in terms of finding the roots of a polynomial in C(x) (the vector space of polynomials over the complex numbers C) that has minimum norm with respect to some inner product defined over C(x).< >

关键词： Frequency estimation Polynomials signal processing algorithms Kernel laboratories computer science Vectors Symmetric matrices

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Complex system identification methods for fast echo canceler initialization

Complex system identification methods for fast echo canceler...

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International Conference on Acoustics, Speech, and signal processing (ICASSP)

作者： S.A. Ahmed J.R. Cruz Communications and Signal Processing Laboratory (CMP Lab) School of Electrical Engineering and Computer Science University of Oklahama Norman OK USA

The authors propose a new recursive version of a technique proposed by G. Long and F. Ling (1990) for the initialization of a data-driven echo canceler (DDEC). They prove that the Long and Ling algorithm yields a least-squares solution, and then a new technique is presented which is comparable to the recursive-least-squares (RLS) algorithm. However, the use of a unique training sequence reduces the complexity of the RLS algorithm to that of the least-mean-square (LMS) algorithm. An analysis of the covariance of the estimated weight vector is presented, and simulation results show a remarkable improvement in both convergence speed and steady-state error.< >

关键词： System identification signal processing algorithms Least squares approximation Resonance light scattering Least squares methods Steady-state Computational efficiency signal processing laboratories computer science

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IMAGE SEQUENCE CODING BY SPLIT AND MERGE

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IEEE TRANSACTIONS ON communications 1991年第12期39卷 1845-1855页

作者： WILLEMIN, P REED, TR KUNT, M Signal and Image Processing Laboratory Swiss Federal Institute of Technology Lausanne Switzerland Department of Electrical Engineering and Computer Science University of California Davis CA USA

This paper describes a method for reducing the information contained in an image sequence, while retaining the information necessary for the interpretation of the sequence by a human observer. The method consists of first locating the redundant information, reducing the degree of redundancy, and coding the result. The sequence is treated as a single 3-D data volume, the voxels of which are grouped into several regions, obtained by a 3-D split and merge algorithm. To find these regions, we first obtain an initial region space by splitting the image sequence until the gray-level variation over each region can be approximated by a 3-D polynomial, to a specified accuracy. This results in a set of parallelepipedic regions of various sizes. To represent the gray-level variation over these regions, the coefficients of the approximating polynomial are used as features. The most similar regions are then merged, using a region adjacency graph. The information is coded by representing the borders of the regions using a pyramidal structure in the x, y, t space. The coefficients of the approximating polynomials are coded in a straightforward manner. For 256 x 256 pixel, 25 frames/s image sequences, compressions allowing transmission rates near 64 kbit/s are obtained.

关键词： Image sequences Image coding Pixel Bandwidth Layout Polynomials Image segmentation Humans Helium ISDN

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A communications architecture concept for ATIS

A communications architecture concept for ATIS

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VNIS 1991

作者： Ristenbatt, Marlin P. Communications and Signal Processing Lab. Department of Electrical Engrg. and Computer Science University of Michigan Ann Arbor MI United States

Three recent developments suggest cost-beneficial approaches to communications for Advanced Traveler Information Systems (ATIS): 1) the use of FM subcarrier for efficient paging-type (digital) one-way broadcast, along with the upcoming introduction of Radio Data System (RDS) vehicle receivers, 2) the rapidly growing use of RF "tags" for automatic tolling, which have recently been upgraded for discrete zone two-way communication, and 3) the "signal processing gain" and "demand access" capability of spread-spectrum-type coding, which together make event-driven communications more feasible and cost-beneficial. An architecture concept exploiting these thrusts is outlined in terms of a "baseline" and a higher level ATIS. The baseline emphasizes basic traffic exception and safety messages and supports higher level (optional) services such as navigators and invehicle route guidance computers. The baseline portion emphasizes minimum cost;overall the level-of-service choices of both the consumer and the traffic manager are maximized.

关键词： Costs

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