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检索条件"任意字段=Advanced Algorithms and Architectures for Signal Processing IV 1989"

共 49 条记录，以下是21-30 订阅

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MODIFIED GivENS ROTATIONS FOR INVERSE UPDATING IN QR DECOMPOSITION 4

MODIFIED GIVENS ROTATIONS FOR INVERSE UPDATING IN QR DECOMPO...

引用

4TH CONF ON advanced signal processing algorithms, architectures, AND IMPLEMENTATIONS

作者： SHEPHERD, TJ MCWHIRTER, JG Defence Research Agency (United Kingdom)

The sequence of Givens rotations used to update the upper triangular matrix R in a recursive QR decomposition may also be used to update the inverse transpose matrix R-H. Alternative forms of square-root-free Givens r... 详细信息

ISBN: (纸本)0819412767

The sequence of Givens rotations used to update the upper triangular matrix R in a recursive QR decomposition may also be used to update the inverse transpose matrix R^-H. Alternative forms of square-root-free Givens rotation for updating the inverse transpose matrix are derived by representing and storing it in a different factorized form from that used before. The modified Givens rotations do not involve explicit division by the exponential forget factor and lead to an update algorithm equivalent to the one derived by Sakai using Recursive Modified Gram Schmidt orthogonalization.

关键词： Algorithm development Evolutionary algorithms Array processing Aluminum Chlorine Numerical simulations signal processing Defense and security Digital filtering Roads

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Homogeneous VLSI structures for high speed digital signal processing using number theoretic techniques 4

Homogeneous VLSI structures for high speed digital signal pr...

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advanced algorithms and architectures for signal processing iv 1989

作者： Jullien, G.A. Miller, W.C. VLSI Research Group University of Windsor WindsorONN9B 3P4 Canada

Exact computations, performed with residues, occur in Number Theoretic Transforms and Residue Number System implementations. Once thought awkward to implement with standard logic circuits, the application of efficient small lookup tables, constructed with pipelined dynamic ROM's, allows very efficient construction of hardware ideally suited to residue operations. Linear DSP operations that are compute bound (require many arithmetic computations per input/output sample), are best suited for implementation with systolic arrays. For very high throughput operations, bit-level systolic arrays appear to be ideally suited to their implementation. The dynamic ROM's used for the residue computations, are a perfect vehicle for implementing such operations at the bit-level in a systolic architecture. This paper discusses VLSI architectures based on finite ring computations, using linear systolic arrays of small look-up tables. The advantage of this approach is that fixed coefficient multiplication requires no increase in hardware over that required for general addition, and the resulting structure is homogeneous in that only one cell type is required to implement all of the processing functions. Several advantages accrue in the VLSI setting, particularly clock rate increase, ease of testability and natural fault tolerance Three different approaches to implementing the finite ring/field residue calculations are discussed. The first uses a bit-level steering mechanism around small dynamic ROM's;two applications of this technique to digital signal processing are outlined. The other two techniques are based on a redundant binary representation implemented with a pipelined adder configuration, and an iterative solution technique based on neural-like networks. © 1989 SPIE. All rights reserved.

关键词： Iterative methods

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BOOTSTRAP - THEORY AND APPLICATIONS 4

BOOTSTRAP - THEORY AND APPLICATIONS

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4TH CONF ON advanced signal processing algorithms, architectures, AND IMPLEMENTATIONS

作者： ZOUBIR, AM Queensland Univ. of Technology (Australia)

ISBN: (纸本)0819412767

This is a review of bootstrap methods, concentrating on basic ideas and applications. It begins with an exposition of the bootstrap principle and gives several examples of its use. Bootstrap methods for testing statistical hypotheses are then reviewed and an analysis of accuracy of bootstrap tests is provided. We discuss how the bootstrap can be used to estimate variance stabilizing transformations that are crucial for the level of accuracy of bootstrap tests. Finally, we describe an application of multiple bootstrap tests to the problem of finding optimum locations of vibration sensors for knock detection in spark ignition engines.

关键词： Statistical analysis Sensors signal processing Combustion Fourier transforms signal to noise ratio Autoregressive models Error analysis Monte Carlo methods Data modeling

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Novel approach to cross-terms elimination in Wigner distribution using adaptive filtering in frequency domain

Novel approach to cross-terms elimination in Wigner distribu...

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advanced algorithms and architectures for signal processing iv 1989

作者： Amin, Moeness G. Allen, Fred Villanova University Department of Electrical Engineering VillanovaPA19085 United States General Electric Company 751 Fifth Avenue King of PrussiaPA19406 United States

The proposed adaptive filter approach for cross-terms elimination relies on the fact that the Wigner distribution (WD), when windowed in time, due to computations and availability problems, does not change its spectral characteristics at the signal frequencies. The only time-varying components, as the window slides on stationary data, are those corresponding to the signal cross-terms, where they exhibit a phase change. The constant and the time-varying behavior in Pseudo Wigner Distribution (PWD) can be distinguished by adaptive filtering. In this paper, frequency-domain least mean squares (LMS) algorithm is used to both track and suppress the cross-terms, and therefore, allows a much better reading of the time-varying distribution of power over frequency. In the adaptive filtering approach, the desired input (primary) takes the PWD's for different data blocks, while the filter input (reference) is assigned unit values. Due to the nature of the primary and reference data, the frequency domain filter adapts to the constant sinusoidal peak values. However, because of the uncorrelatedness, it fails to track the time-varying components in the desired signal, leaving out the cross-terms to propagate to the output filter error. The cross-correlation between the constant values in the reference and the time-varying components in the desired signal can be controlled by the mechanism by which the PWD's are generated. Random sliding over the data may show improved performance compared to regular sliding with disjoint or overlapping blocks.

关键词： Digital filtering signal processing Filtering (signal processing) Fourier transforms Electronic filtering Linear filtering Evolutionary algorithms Optical filters Electrical engineering signal analyzers

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PIPELINED ADAPTivE DFE architectures 4

PIPELINED ADAPTIVE DFE ARCHITECTURES

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4TH CONF ON advanced signal processing algorithms, architectures, AND IMPLEMENTATIONS

作者： SHANBHAG, NR PARHI, KK Univ. of Minnesota/Twin Cities (United States)

ISBN: (纸本)0819412767

Fine-grain pipelined adaptive decision-feedback equalizer (ADFE) architectures are developed using the relaxed look-ahead technique. This technique, which is an approximation to the conventional look-ahead computation, maintains functionality of the algorithm rather than the input-output behavior. Thus, it results in substantial hardware savings as compared to either parallel processing or look-ahead techniques. The delay relaxation, delay transfer relaxation, and sum relaxation are introduced for purposes of pipelining. Both the conventional and the predictor form of ADFE have been pipelined. The performance of the pipelined algorithms for the equalization of a magnetic recording channel is studied. It is demonstrated via simulations that, for a byte error rate of 10^-7 or less, speed-ups of up to 8 can be easily achieved with the conventional ADFE. The predictor form of ADFE allows much higher speed-ups (up to 32) for less than 1 dB of SNR degradation.

关键词： signal to noise ratio Magnetism Algorithm development Error analysis Optical filters Digital filtering Finite impulse response filters Nonlinear filtering Parallel processing Transform theory

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DIRECTION-OF-ARRivAL TRACKING BY PARALLEL ARRAY-processing 4

DIRECTION-OF-ARRIVAL TRACKING BY PARALLEL ARRAY-PROCESSING

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4TH CONF ON advanced signal processing algorithms, architectures, AND IMPLEMENTATIONS

作者： PEREZNEIRA, A LAGUNAS, MA ETSI Telecommunication (Spain)

ISBN: (纸本)0819412767

Antenna arrays collect multidimensional data that contains signals arriving from different sources. Neural Network architectures can separate the different signals, thus enabling parallel processing structures. These structures can solve a multi-signal estimation problem more efficiently than the corresponding single signal estimator. Various of these parallel architectures are evaluated in the context of array signal processing. Specifically, the scheme developed in this paper (section 2) uses the spatial diversity supplied by the aperture associated with the sensors to separate the signals and to apply them to a bank of parallel adaptive filters. These filters are then designed in accordance with a mean square error minimization criterion (i.e., a criterion based on Second Order Statistics). As Second Order Statistics assume linearity or Gaussianity they are sometimes overly restrictive. It is shown how High Order Statistics can be very useful when more general criteria such as statistical independence between the signals to be separated are imposed.

关键词： Array processing Sensors Phased arrays signal processing Antennas Digital filtering Filtering (signal processing) Error analysis Electronic filtering Mobile communications

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COHERENT signal-SUBSPACE processing FOR NEAR-FIELD BROAD-BAND SOURCE LOCALIZATION 4

COHERENT SIGNAL-SUBSPACE PROCESSING FOR NEAR-FIELD BROAD-BAN...

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4TH CONF ON advanced signal processing algorithms, architectures, AND IMPLEMENTATIONS

作者： LIN, S HUNG, HS Iowa State Univ. (United States)

ISBN: (纸本)0819412767

In this paper, the coherent signal-subspace method (CSM) is extended for estimation of bearing, elevation and range of multiple, near-field, broad-band sources. Analytical results are also derived to justify the effectiveness of the near-field CSM. It is shown that the coherently averaged sample covariance matrix in the near-field CSM has complex Wishart distribution with number of degrees-of-freedom equaling the time-bandwidth product, provided that (1) the frequency components of the array output are statistically independent and Gaussian distributed, and (2) the errors between all the actual source locations and the preliminary estimate of the source cluster center are upperly bounded by a small constant. Some simulation results are provided to show the effectiveness of the near-field CSM.

关键词： Near field signal processing Error analysis Sensors Statistical analysis Data modeling Interference (communication) signal to noise ratio Source localization Spherical lenses

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NUMERICAL-INTEGRATION OF PDES BY DISCRETE PASSivE MODELING OF PHYSICAL SYSTEMS 4

NUMERICAL-INTEGRATION OF PDES BY DISCRETE PASSIVE MODELING O...

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4TH CONF ON advanced signal processing algorithms, architectures, AND IMPLEMENTATIONS

作者： FETTWEIS, A Ruhr-Univ. Bochum (Germany)

ISBN: (纸本)0819412767

A new method for numerically integrating partial differential equations (PDEs) has been under study for the last few years. This method is based on principles of multidimensional (MD) Kirchhoff circuits and multidimensional wave digital filters (MD WDFs), which explains why it has probably not been discovered earlier. It makes wide use of methods and results that have been developed by extensive research in the areas of circuit theory and digital signal processing, but that are rather unknown outside of the small circle of experts on MD WDFs. Instead of talking about MD WDFs we prefer using, within the context of numerical integration, the designation `discrete multidimensionally passive (MD-passive) dynamical systems.'

关键词： Palladium Wigner distribution functions Systems modeling Neodymium Numerical integration Digital electronics Digital signal processing signal processing Digital filtering Partial differential equations

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STABLE CHASING algorithms FOR MODIFYING COMPLETE AND PARTIAL SINGULAR-VALUE DECOMPOSITIONS 4

STABLE CHASING ALGORITHMS FOR MODIFYING COMPLETE AND PARTIAL...

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4TH CONF ON advanced signal processing algorithms, architectures, AND IMPLEMENTATIONS

作者： BARLOW, JL ZHA, HY YOON, PA The Pennsylvania State Univ. (United States)

ISBN: (纸本)0819412767

Methods for updating and downdating singular value decompositions (SVDs) and partially reduced bidiagonal forms (partial SVDs) are introduced. The methods are based upon chasing procedures for updating the SVD and downdating procedures for orthogonal decompositions. The main feature of these methods is the ability to separate the singular values into `large' and `small' sets and then obtain an updated bidiagonal form with corresponding `large' and `small' columns. This makes for a more accurate update or dosndate.

关键词： Matrices Computer science Feature extraction Tolerancing Applied research Error analysis Lithium signal processing

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signal-ADAPTivE DECOMPOSITION OF MULTICOMPONENT signalS 4

SIGNAL-ADAPTIVE DECOMPOSITION OF MULTICOMPONENT SIGNALS

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4TH CONF ON advanced signal processing algorithms, architectures, AND IMPLEMENTATIONS

作者： ASSALEH, KT MAMMONE, RJ Rutgers Univ. (United States)

ISBN: (纸本)0819412767

In this paper we present a new method for adaptively decomposing a multicomponent signal into its components. This method is based on fitting an autoregressive (AR) model to the short-time spectra of the signal. The AR parameters represent the coefficients of the linear predictive (LP) polynomial. The roots of this polynomial constitute a set of center frequencies and bandwidths that characterize the modes of the signal. The decomposition process is achieved by applying a time-varying filter bank to the original multicomponent signal. The characteristics of this filter bank are derived from a subset of the roots of the LP polynomial. We have developed a constraining algorithm to determine that subset based on the boundedness of the bandwidths, and the temporal continuity of the center frequencies of the components. We have applied the proposed decomposition method for the separation of the formants of speech signals.

关键词： Autoregressive models signal to noise ratio Electronic filtering Composites Digital filtering Algorithm development signal processing Image segmentation Resonators Time-frequency analysis

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