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IEEE International Symposium on Circuits and Systems (ISCAS)

作者： L.J. Karam J.H. McClellan Telecommunications Research Center Department of Electrical Engineering Arizona State University Tempe AZ USA Digital Signal Processing Laboratory School of Electrical and Computer Engineering Georgia Institute of Technology Atlanta GA USA

This paper presents a new multiple-exchange ascent algorithm for designing optimal Chebyshev digital FIR filters with arbitrary magnitude and phase specifications. Compared to existing Chebyshev design techniques, the new design algorithm exhibits faster convergence while maintaining high accuracy, and is guaranteed to converge to the optimal solution. In addition, the proposed algorithm reduces to the classic second Remez (Parks-McClellan) algorithm when real-only or imaginary-only filters are designed and is, therefore, a true generalization of the classic Remez algorithm to the complex case.

关键词： Finite impulse response filter Digital filters Chebyshev approximation Algorithm design and analysis signal processing algorithms Frequency response Electronic mail Design optimization Filtering Robustness

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Interference mitigation in orthogonal frequency division multiplexing (OFDM)

Interference mitigation in orthogonal frequency division mul...

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International Conference on Universal Personal Communications

作者： R.W. Lowdermilk F.J. Harris Communications and Signal Processing Laboratory Department of Electrical and Computer Engineering San Diego State University San Diego CA USA Advanced Development Operations Lockheed Martin Corporation San Diego CA USA

Orthogonal frequency division multiplexing (OFDM) is a modulation scheme purported to be insensitive to the multipath effects of a mobile communication channel as well as exhibiting immunity to narrow-band interference via hole punching in the OFDM spectrum. We examine and verify the OFDM bit error rate sensitivity to low level interference and introduce and demonstrate a spectrum spreading technique to reduce the effect of the narrow-band interferer.

关键词： Interference OFDM modulation Pulse modulation Pulse shaping methods Frequency domain analysis signal processing Narrowband Bit error rate Demodulation Fast Fourier transforms

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Partially adaptive radar signal processing: The cross-spectral approach 29

Partially adaptive radar signal processing: The cross-spectr...

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29th Asilomar Conference on signals, Systems and computers, ACSSC 1995

作者： Goldstein, J. Scott Kogon, Stephen M. Reed, Irving S. Williams, Douglas B. Holder, E. Jeff USAF Rome Laboratory RL/C3BA 525 Brooks Rd. Grifilss AFBNY13441-4505 United States Statistical Signal and Image Processing Laboratory Department of Electrical Engineering University of Southern California Los AngelesCA90089-2565 United States Digital Signal Processing Laboratory School of Electrical and Computer Engineering Georgia Institute of Technology AtlantaGA30332-0250 United States

ISBN: (纸本)0818673702

This paper is concerned with reducing the rank of the adaptive weight vector in radar array signal processing. The motivation for reducing the rank is that modern space-time processing requires many more weights than can be supported on airborne and spacesegment platforms. The loss incurred in partially adaptive radar processing is that the steady-state Wiener solution of the lower rank weight vector may not perform as well as the full rank solution. Hence, this paper will be concerned with the Wiener solution of partially adaptive radar arrays and compare the performance of principal component techniques with the recently developed crossspectral technique. © 1996 IEEE.

关键词： Array processing

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VLSI implementation of very-high-order FIR filters

VLSI implementation of very-high-order FIR filters

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IEEE International Symposium on Circuits and Systems (ISCAS)

作者： M.A. Soderstrand K. Al-Marayati Digital Signal Processing and Communications Laboratory Department of Electrical and Computer Engineering University of California Davis USA

Very-high-order FIR filters required for the new modulation schemes associated with wireless computer networks and cellular telephones can be implemented in VLSI circuitry using low-power CMOS technology and a novel application of Residue Number System (RNS) arithmetic. Through this approach 20-bit equivalent integer arithmetic can be obtained for filters with 8 to 256 taps with only a modest increase in hardware for filters above 8 taps. Simulations indicate that this new technique can increase dramatically the number of taps implemented on a single VLSI chip when compared with an FIR filter generated using FIRGEN.

关键词： Very large scale integration Finite impulse response filter CMOS technology computer networks Cellular networks Telephony Circuits Application software Digital arithmetic Hardware

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An efficient algorithm for DTMF decoding using the subband NDFT

An efficient algorithm for DTMF decoding using the subband N...

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IEEE International Symposium on Circuits and Systems (ISCAS)

作者： S. Bagchi S.K. Mitra Department of Electrical and Computer Engineering Signal and Image Processing Laboratory University of California Santa Barbara CA USA

An algorithm based on the subband nonuniform discrete Fourier transform (SB-NDFT) is proposed for decoding dual-tone multi-frequency (DTMF) signals. To decode a DTMF signal, its energy at the eight DTMF frequencies must be determined by evaluating samples of the NDFT at these frequencies. In the proposed SB-NDFT algorithm, these NDFT samples are computed by decomposing the input signal into two subbands. Since DTMF signals occupy the low-frequency part of the telephone bandwidth, the higher subband can be discarded for a fast, approximate computation. A performance comparison between algorithms based on the NDFT, SB-NDFT, DFT, and SB-DFT shows that the SB-NDFT requires the lowest number of computations to attain a specified level of performance.

关键词： Decoding Frequency Discrete Fourier transforms signal processing Image processing Laboratories Telephony Bandwidth Communication system signaling Telephone sets

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Delay models for the sea-of-wires array synthesis system 95

Delay models for the sea-of-wires array synthesis system

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European Design and Test (ED&TC) Conference

作者： Ing-Yi Chen Geng-Lin Chen Sy-Yen Kuo Department of Electronic Engineering Chung Yuan Christian University Chungli Taiwan Digital Signal Processing Department Computer & Communication Laboratory ITRI Hsinchu Taiwan Department of Electrical Engineering National Taiwan University Taipei Taiwan

ISBN: (纸本)9780818670398

This paper presents two simple, accurate and efficient delay models, the static delay model and the dynamic delay model, to support performance optimization of VLSI Sea-of-Wires Arrays (SWA). The SWA delay model treats each distributed gate as an attribute-based primitive gate with different internal and external connection wires. Instead of solving differential equations, the SWA model determines delays by lookup from a multi-dimensional table. Only a few microseconds of execution time are needed per gate. The propagation delay along a circuit path is the sum of the delay segments of distributed gates in the path. The critical path of an SWA design can be identified with an O(n) timing analysis algorithm. For most AHPL Benchmarks, the table-lookup method achieves 5 orders of magnitude speedup over SPICE for the same circuits with error margin less than 7%.< >

关键词： SPICE Circuit simulation Wires signal synthesis Very large scale integration Algorithm design and analysis Timing Equations Delay effects Table lookup

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Improvement of convergence speed for subband adaptive digital filter using the multirate repeating method

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ELECTRONICS AND COMMUNICATIONS IN JAPAN PART III-FUNDAMENTAL ELECTRONIC SCIENCE 1995年第10期78卷 37-45页

作者： Kiya, H Nishikawa, K Ashihara, K Faculty of Technology Tokyo Metropolitan University Hachioji Japan 192-03 Members Kiyoshi Nshikawa graduated in 1990 from the Department of Electrical Engineering Tokyo Metropolitan University where he completed the Master's Program in 1992 and affiliated with Nippon Steel Co. Computer Systems Laboratory. He was a Research Associate in 1993 in the Department of Elect. Inf. Eng. Tokyo Metropolitan University engaged in research onadaptive signal processing and information source coding. He is a member of the IEEE. Associate Member

The subband adaptive system, where the idea of the filter bank is applied to the adaptive filter, has a feature that a higher-order adaptation can be reduced to lower-order adaptations. The reduction of the order of the adaptive filter is related closely to the number of channels and the decimation ratio. The order of adaptive digital filter (ADF) is decreased greatly when the number of subbands is increased, and the decimation ratio is increased up to the maximum value. Then, however, the number of coefficient-updates per unit time is decreased, which results in the deterioration of the convergence Speed. From such a viewpoint, this paper discusses a method to improve the convergence speed, which is deteriorated in the subband adaptive system, due to the decimation. The idea of the proposed method is to utilize effectively the data which have been discarded in the decimation process and to improve the convergence speed. It is called the multirate repeating method. As the first step, the multirate repeating method is applied to the conventional subband adaptive system and the convergence speed is improved. Then a subband adaptive system is introduced in which the multirate repeating method can be utilized more effectively. As a result, a faster convergence is realized while retaining the ADF order-reduction effect, which is an advantage of the subband adaptive system.

关键词： adaptive signal processing subband adaptive system multirate repeating method rational decimation

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Instrumentation for the Acoustic Thermometry of Ocean Climate (ATOC) prototype Pacific Ocean network

Instrumentation for the Acoustic Thermometry of Ocean Climat...

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OCEANS

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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A class of order statistics learning vector quantizers

A class of order statistics learning vector quantizers

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IEEE International Symposium on Circuits and Systems (ISCAS)

作者： I. Pitas C. Kotropoulos N. Nikolaidis R. Yang M. Gabbouj Department of Electrical & Computer Engineering University of Thessaloniki Thessaloniki Greece Signal Processing Laboratory Tampere University of Technology Tampere Finland

A novel class of Learning Vector Quantizers (LVQs) based on multivariate order statistics is proposed in order to overcome the drawback that the estimators for obtaining the reference vectors in LVQ do not have robustness either against erroneous choices for the winner vector or against the outliers that may exist in vector-valued observations. The performance of the proposed variants of LVQ is demonstrated by experiments. In the case of marginal median LVQ, its asymptotic properties are derived as well.< >

关键词： Statistics Vectors Robustness Artificial neural networks signal processing algorithms signal processing Laboratories Neural networks Network topology Error correction codes

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Faster, higher-quality training of feedforward neural networks by select updating

Faster, higher-quality training of feedforward neural networ...

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IEEE International Symposium on Circuits and Systems (ISCAS)

作者： J.R. Deller S.D. Hunt Department of Electrical Engineering/260 EB Speech Processing Laboratory Michigan State University East Lansing MI USA Department of Electrical Engineering and Computer Science University of Puerto Rico RUM Puerto Rico

A new training method for feedforward neural networks is presented which exploits results from matrix perturbation theory for significant training time improvement. This theory is used to assess the effect of a particular training pattern on the weight estimates prior to its inclusion in any iteration. Data which do not significantly change the weights are not used in that iteration obviating the computation expense of updating.< >

关键词： Neural networks Feedforward neural networks speech processing Laboratories Least squares methods Joining processes Nonlinear equations Linear systems Vectors Least squares approximation

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