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IEEE Southeastcon

作者： T.D. Robinson D.B. Koch Communications Information and Signal Processing (CISP) Group Department of Electrical & Computer Engineering University of Tennessee Knoxville TN USA

Expressions are developed which can be used to determine the false-detection performance of complementary codes. The false-detection performance of complementary phase codes for various bit-error rates is presented and verified through simulation. It is shown that all complementary phase code pairs of the same length perform equally when preceded by random data. This shows that the performance of a two-sequence codeword depends only on the combined autocorrelation properties of the two sequences. When preceded by recovery data, all complementary phase code pairs are not found to perform equally; however, one pair (01001000, 10111000) is found to be nearly optimum for all five bit-error probabilities.< >

关键词： Digital communication Time division multiple access Phase change materials Satellite communication Radar detection Phase detection Pulse modulation Transmitters Binary sequences Quadrature phase shift keying

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A digital signal processing chip for iterative deconvolution restoration algorithms

A digital signal processing chip for iterative deconvolution...

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IEEE Instrumentation and Measurement Technology Conference

作者： R.B. Whitted P.B. Crilly Communications Information and Signal Processing (CISP) Group Department of Electrical and Computer Engineering University of Tennessee Knoxville TN USA

A VLSI DSP chip is presented that will significantly improve the processing throughput for a general class of iterative deconvolution algorithms. The design will be based on a systolic array concept. This will enable these algorithms to be used for real time DSP applications which formerly due to speed limitations were not possible. The increased class of applications will enable further understanding of these applications. The higher throughput will also enable the researcher to further take advantage of the features unique to iterative deconvolution.< >

关键词： Digital signal processing chips Deconvolution signal restoration signal processing algorithms Iterative algorithms Wiener filter Throughput Spectroscopy Additive noise Distortion

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A communications architecture concept for ATIS

A communications architecture concept for ATIS

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Vehicle Navigation and Information Systems Conference

作者： M.P. Ristenbatt Communications and Signal Processing Lab Department of Electrical Engineering & Computer Science University of Michigan Ann Arbor MI USA

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 in-vehicle 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： Information systems Radio broadcasting Data systems Vehicles Receivers Radio frequency Digital signal processing computer architecture Safety Navigation

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Multichannel Givens lattice adaptive algorithm

Multichannel Givens lattice adaptive algorithm

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

作者： K. Zhao F. Ling J.G. Proakis Communications and Digital Signal Processing Center Department of Electrical and Computer Engineering Northeastern University Boston MA USA Codex Corporation Mansfield MA USA

A multichannel Givens lattice algorithm is presented, which has a modular structure and is suitable for parallel implementation. The basic processing module requires a 2*2 plane rotation (Givens rotation), which has superior numerical properties due to its circular nature. The application of the multichannel Givens lattice algorithm to channel equalization is also discussed.< >

关键词： Lattices Adaptive algorithm signal processing algorithms Resonance light scattering Computational complexity Decision feedback equalizers State estimation Throughput Recursive estimation Partitioning algorithms

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Subband coding of digital images using absolute moment block truncation

Subband coding of digital images using absolute moment block...

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

作者： K.-K. Ma S.A. Rajala Department of Electrical and Computer Engineering Center of Communications and Signal Processing North Carolina State University Raleigh NC USA IBM Corporation Raleigh France

A combination of subband coding and absolute moment block truncation coding (AMBTC) is presented to effectively eliminate the blocking effect (or grid noise) that the AMBTC severely suffers. To dynamically allocate bits for each subband and efficiently achieve quality images for a given compression ratio, there are two fundamental questions; (1) how to choose subbands for applying AMBTC, and (2) how to choose the appropriate window sizes for these subbands. The concept of intra-subband and inter-subband bit allocation has been proposed to address the former issue, and a variable window size approach is suggested for the latter one. Each solution shows promising results.< >

关键词： Image coding Digital images Filter bank Quantization Channel bank filters Bit rate signal synthesis Pixel Decoding Image reconstruction

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Self-synchronized signal controlled constrained notch filter for rejection of nonstationary interference

Self-synchronized signal controlled constrained notch filter...

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

作者： W. Tong E.I. Plotkin D. Wulich M.N.S. Swamy Center of Signal Processing and Communications Department of Electrical and Computer Engineering Concordia University Montreal QUE Canada Department of ECE Ben-Gurion University of the Negev Beersheba Israel

A novel signal controlled constrained notch filter (SC-CNF) structure with global feedback is presented. The SC-CNF is combined with an adaptive linear enhancer to retrieve a multitone signal corrupted by a strong nonstationary FM interference. Using a time-warping technique, the SC-CNF, by its nature time-varying, is transformed into a CNF which is time variant. Such transformation is implemented by nonequally spaced sampling (NESS) of the input mixture. A closed-loop system is proposed to improve the strategy of NESS. By exploiting the self-synchronization process, nearly complete rejection of FM interference is achieved. The condition for the occurrence of self-synchronization is obtained. The self-synchronized SC-CNF provides significant improvement (up to 30 dB) of the signal-to-interference ratio of the entire system output.< >

关键词： Filters Interference constraints Communication system control signal sampling Closed loop systems Frequency synchronization Phase locked loops Phase estimation signal processing Adaptive signal processing

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Closed-loop self-synchronized signal controlled constrained notch filter with application for rejection of arbitrary frequency variation speed sinusoidal interference

Closed-loop self-synchronized signal controlled constrained ...

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IEEE International Symposium on Circuits and Systems (ISCAS)

作者： W. Tong E.I. Plotkin M.N.S. Swamy D. Wulich Center of Signal Processing and Communications Department of Electrical and Computer Engineering Concordia University Montreal QUE Canada Department of ECE Ben-Gurion University of the Negev Beersheba Israel

The closed-loop self-synchronized constrained notch filter (CLSS-CNF) is proposed, and its ability for retrieval of narrowband signals corrupted by a strong FM interference is studied. Self-synchronization is achieved through the estimation of the instantaneous phase of the FM interference to control time-varying coefficients of SC-CNF by using a closed-loop structure. The CLSS-CNF can overcome the tracking capability threshold of a conventional constrained adaptive notch filter (CANF), and reject a sinusoidal interference with arbitrary speed of frequency variation. Based on this approach, two such filters are discussed, and their system performances compared. It is shown that the proposed filters are superior to both the conventional CANF as well as the recently proposed SC-CNF with open loop structure. Use of closed loop and the specially developed projection scheme for time-varying coefficients allow one to achieve a signal to the interference ratio (SIR/sub O/) of up to 20 dB compared to the input SIR/sub i/ of less than -20 dB.< >

关键词： Interference constraints Phase estimation Frequency synchronization Phase locked loops Adaptive filters Communication system control Application software signal processing algorithms Open loop systems Time varying systems

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Externally controlled time-varying constrained notch filter

Externally controlled time-varying constrained notch filter

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IEEE International Symposium on Circuits and Systems (ISCAS)

作者： D. Wulich E.I. Plotkin M.N.S. Swamy W. Tong Center of Signal Processing and Communications Department of Electrical and Computer Engineering Concordia University Montreal QUE Canada Department of ECE Ben-Gurion University of the Negev Beersheba Israel

An application of a constrained notch filter (CNF) for filtering a sine signal with arbitrary varying frequency is considered. It is shown that the concept of CNF can be easily extended to time-varying CNF (TVCNF) whose coefficients vary with time according to the rule of frequency variation of the considered FM signal. Such a TVCNF is able to reject the FM signal and at the same time remain transparent to any other signal. The main problem in TVCNF implementation is to provide a proper control (synchronization) of its coefficients. An application of the TVCNF for the enhancement of an arbitrary signal corrupted by a strong FM interference is investigated. It is shown that if the synchronization is based on the input mixture, then for any negative signal-to-interference ratio (SIR<<0 dB) at the input of the TVCNF, the SIR at its output reaches 0 dB.< >

关键词： Frequency modulation Interference Frequency synchronization Adaptive filters Background noise Communication system control signal processing Application software Filtering signal design

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SCALED AND SCALING-FREE McCLELLAN TRANSFORMATIONS FOR THE DESIGN OF 2-D DIGITAL FILTERS

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Journal of Circuits, Systems and computers 1991年第1期1卷 105-124页

作者： N. NAGAMUTHU M. N. S. SWAMY Centre for Signal Processing and Communications (CENSIPCOM) Department of Electrical and Computer Engineering Concordia University Montreal Quebec H3G 1M8 Canada

In this paper, the scaling problem of McClellan transformation used for the design 2-D digital filters with elliptical and circular cutoff contours is critically analyzed. Several new and extremely simple formulas for calculating the maximum and minimum values of the transformation function are presented. Using these extreme values, simple formulas for scaling factors and scaled McClellan transformations are derived for the various cases of each type of contour. Using the necessary and sufficient conditions for a scaling-free transformation, the scaling-free McClellan transformations are also derived. It is shown that the number of independent coefficients in them to be optimized varies from 0 to 2 depending on the case, compared to 3 in the original transformation. They are very useful in deriving analytical expressions for the transformation coefficients, and the 2-D filters designed by employing them have a smaller number of multiplications per output sample and hence, attractive for real-time applications. A new scaling formula is presented and some of its properties and effects on the derived formulas are discussed. The major advantage of this new formula is that from the unscaled transformation coefficients for a given type of 2-D filter, we can generate the scaled transformation coefficients for a different type of 2-D filter. Some examples are presented to demonstrate the usefulness of the derived formulas.

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Nonlinear adaptive filtering of systems with hysteresis by quantized mean field annealing

Nonlinear adaptive filtering of systems with hysteresis by q...

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IEEE Workshop on Neural Networks for signal processing

作者： R.A. Nobakht S.H. Ardalan D.E. Van den Bout International Business Machines Corporation NC USA Center of Communications and Signal Processing Department of Electrical and Computer Engineering North Carolina State University Raleigh NC USA

A technique for nonlinear adaptive filtering of systems with hysteresis has been developed which combines quantized mean field annealing (QMFA) and conventional RLS/FTF adaptive filtering. Hysteresis is modeled as a nonlinear system with memory. Unlike other methods which rely on Volterra and Wiener models, this technique can efficiently handle large order nonlinearities with or without hysteresis effects. The nonlinear channel is divided into a memory nonlinearity followed by a dispersive linear system. Assuming that the dispersive linear system is stationary during initialization, and the nonlinearity does not change while the dispersive linear system varies with time, QMFA is applied to obtain the coefficients and the order of the memory of the nonlinearity and RLS/FTF is applied to determine the weights of the dispersive linear system. Application of this method to a full duplex digital subscriber loop is made. Simulations show the superior performance of our technique compared to that of ordinary RLS/FTF and steepest-descent algorithms.< >

关键词： Adaptive filters Hysteresis Dispersion Linear systems Resonance light scattering Nonlinear systems Neural networks Simulated annealing Least squares approximation Adaptive signal processing

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