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

作者： S. Ardalan Center for Communications and Signal Processing Department of Electrical and Computer Engineering North Carolina State University USA

A floating point error analysis of the Recursive Least Squares and Least Mean Squares (LMS) algorithms is presented. Both the prewindowed growing memory RLS algorithm (λ=1) for stationary systems and the exponential sliding window RLS algorithm (λ < 1) for time varying systems are studied. For both algorithms the expression for the mean square prediction error and the expected value of the weight error vector norm are derived in terms of the variance of the floating point noise sources. The results point to a trade off in the choice of the forgetting factor, λ. In order to reduce the effects of additive noise and the floating point noise due to the inner product calculation of the desired signal, λ must be chosen close to one. On the other hand, the floating point noise due to floating point addition in the weight vector update recursion increases as \lambda\rightarrow1 . Floating point errors in the calculation of the weight vector correction term, however, do not effect the steady state error and have a transient effect. Similar results are obtained for the LMS algorithm where a tradeoff exists in the choice of the loop gain.

关键词： Error analysis Least squares methods Adaptive filters Least squares approximation Resonance light scattering Additive noise Error correction Time varying systems Noise reduction Steady-state

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Derivation of the fast pole-zero (ARMA) recursive least squares algorithm using geometric projections

Derivation of the fast pole-zero (ARMA) recursive least squa...

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

作者： S. Ardalan Center for Communications and Signal Processing Department of Electrical and Computer Engineering North Carolina State University USA

The Fast Pole-Zero (ARMA) Recursive Least Squares Algorithm is derived using geometric projections. The new algorithm has a reduced number of computations compared to the pole-zero fast Kalman algorithm. In particular, four vector inner products and scalar-times-vector operations are eliminated. This is particularly important in multiprocessor implementations of the algorithm. The geometric approach provides insight and useful interpretations of the various filters and residuals that form the algorithm. Simulations verifying the performance of the algorithm are presented where the poles and zeroes of an IIR filter are estimated.

关键词： Least squares methods signal processing algorithms Resonance light scattering Kalman filters Transversal filters Solid modeling IIR filters Echo cancellers Poles and zeros Vectors

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A Derivation of a General Order, Multichannel Fast Transversal, Recursive Least Squares Filter Algorithm

A Derivation of a General Order, Multichannel Fast Transvers...

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MILCOM, Military communications Conference

作者： L.J. Faber S. H. Ardalan S.T. Alexander Center for Communications and Signal Processing Department of Electrical and Computer Engineering North Carolina State University Raleigh NC USA

This paper examines the derivation of a general order, multiple input version of the fast transversal recursive least squares filter algorithm. The new algorithm, in which the order (or number of taps) associated with each input channel is independent, is distinguished from the previously published multichannel form of the fast transversal RLS algorithm, wherein each input channel was con strained to have the same order. This additional flexibility allows assignment of filter resources to particular channels, or independent assignment to the pole/zero estimators in an ARMA system identification application. A summary of the new algorithm is given, in proper order of execution. An operations count is also provided for each equation

关键词： Least squares methods Transversal filters signal processing algorithms Partitioning algorithms Resonance light scattering Poles and zeros System identification Equations Delay Computational efficiency

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Patterns of usage of radiographic studies: Implications for picture archival and communication systems development

Patterns of usage of radiographic studies: Implications for ...

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Application of Optical Instrumentation in Medicine XIV and Picture Archiving and Communication Systems (PACS IV) for Medical Applications 1986

作者： Choplin, Robert H. Nilsson, Arne Boehme, John M. D'Silva, Vijay Maynard, C. Douglas Williams, Rodney Green, Robert Crowe, James E. Howsam, Irish Dice, Lydia Department of Radiology Wake Forest University Medical Center 300 S. Hawthorne Road Winston-SalemNC27103 United States Department of Electrical and Computer Engineering Center for Communications and Signal Processing North Carolina State University RaleighNC United States Forsyth Memorial Hospital 3333 Silas Creek Parkway Winston-SalemNC United States Northern Hospital of Surry County Mt. AiryNC United States

Electronics storage systems, image display stations, protocols, and networks capable of managing a large amount of data must be developed in order for picture archival and communications systems to become a reality, These systems must be designed based on information about procedure volumes and information about the magnitude of data transfer rates needed at any given time. Since significant fluctuations can be expected with respect to the demands during a 24 hour period it is important to generate a traffic model that can predict these fluctuations. In addition to recognizing the demands at one institution, it should be noted that different institutions may have widely differing needs. Data has been collected from three differing radiological environments and an investigation with regards to traffic profile and fluctuations has been performed. © 1986 SPIE. All rights reserved.

关键词： Data transfer

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CANCELLATION OF HARMONIC NOISE IN DISTRIBUTION LINE communications

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IEEE TRANSACTIONS ON POWER APPARATUS AND SYSTEMS 1985年第12期104卷 3338-3344页

作者： TRUSSELL, HJ WANG, JD Center for Communications and Signal Processing Department of Electrical and Computer Engineering North Carolina State University Raleigh NC USA

Adaptive least mean square (LMS) filters are shown to be effective in reducing harmonic noise on distribution power lines. The efficiency of these filters can be improved by imposing constraints.

关键词： Noise cancellation Power harmonic filters Power system harmonics Adaptive filters Circuit noise Switches Frequency Least squares approximation Noise reduction Voltage

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A SIMPLE NONITERATIVE SPEECH EXCITATION ALGORITHM USING THE LPC RESIDUAL

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IEEE TRANSACTIONS ON ACOUSTICS SPEECH AND signal processing 1985年第2期33卷 432-434页

作者： ALEXANDER, ST Department of Electrical and Computer Engineering Center for Communications and Signal Processing North Carolina State University Raleigh NC USA

This paper provides an analytical derivation of a simple noniterative technique for extracting a multiple impulse excitation model for synthesized speech directly from the LPC residual sequence. While suboptimal with respect to "multipulse" techniques, this method is very applicable for speech enhancement where processor capability is limited. The results suggest an additional "orthogonality" requirement between the excitation sequence and the resulting prediction error, which aids in the intuitive understanding of the method.

关键词： Linear predictive coding Speech synthesis Speech enhancement Bit rate Detectors Degradation Speech analysis Filters signal processing algorithms Switches

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Adaptive reduction of interfering speaker noise using the least mean squares algorithm

Adaptive reduction of interfering speaker noise using the le...

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

作者： S. Alexander Center for Communications & Signal Processing Department of Electrical and Computer Engineering North Carolina State University NC USA

In this paper, the adaptive Least Mean Squares (LMS) algorithm is used to separate speaker-produced "information" from interferer-produced "noise" on the basis of the difference in power levels associated with the two phenomena. This method exploits the property of LMS that it rapidly adapts for the dominant excitation modes while simultaneously adapting very slowly for the weaker modes of excitation. This selective convergence property of LMS is next analyzed using an eigenvalue-eigenvector approach which easily displays the signal separation property. Lastly, computer simulations are presented which verify the analysis above for representative synthetic speech waveforms.

关键词： Noise reduction Least mean square algorithms Least squares approximation Speech analysis Noise level Convergence signal analysis computer displays Source separation computer simulation

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Concurrent process structured software models for time domain harmonic scaling of speech

Concurrent process structured software models for time domai...

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

作者： G. Poston R. Fornaro Department of Electrical and Computer Engineering Center for Communications and Signal Processing North Carolina State University Raleigh NC USA Department of Computer Science Center for Communications and Signal Processing North Carolina State University Raleigh NC USA

Concurrent process structured software can be used as a tool for studying the behavior of parallel digital signal processing systems. By modeling alternative architectures for a given application and characterizing the interactions of concurrent processes, an evaluation of relative efficiencies can be made. Presented here are models of three architectures for time domain harmonic scaling (TDHS) of speech. Their implementations and the software tools required to study their behavior are described.

关键词： Speech processing computer architecture Application software Digital signal processing chips Pipelines Very large scale integration Monitoring computer science Digital signal processing Software tools

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Motion-compensated interframe coding

Motion-compensated interframe coding

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

作者： R. Moorhead S. Rajala Center for Communications and Signal Processing Department of Electrical and Computer Engineering North Carolina State University Raleigh NC USA

This paper addresses various aspects of motion-compensated interframe coding. The image sequence data is investigated in an attempt to develop a more thorough analysis of the convergence requirements and the convergence rate. A new motion prediction technique is used which increases the validity of the assumptions made in proving convergence and which decreases the total prediction error. Simulation results are presented to indicate the improvement of the proposed motion prediction scheme and to indicate the results of the convergence analysis.

关键词： Convergence Motion analysis Image sequences Image motion analysis Image analysis Image sequence analysis Algorithm design and analysis signal processing Analytical models Predictive models

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Transform/Time domain coding using the method of projection onto convex sets

Transform/Time domain coding using the method of projection ...

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

作者： A. Faryar S. Rajala Center for Communications and Signal Processing Department of Electrical and Computer Engineering North Carolina State University Raleigh NC USA

In image coding, image information can often be transmitted with relatively little bandwidth requirement. This is the case in the pixel domain if there are large blocks of homogeneous intensity within a single frame, or when comparing adjacent frames there are large blocks of unchanged pixels. In the frequency domain, this is the case when only a few of the frequency coefficients are deemed significant. This paper presents an algorithm for improving the bandwidth compression of the error pattern generated by typical motion-compensated coding techniques. The algorithm involves two well established techniques; conventional transform coding and the method of successive Projection onto Convex Sets.

关键词： Image coding Bandwidth Transform coding signal processing algorithms Motion estimation Bit rate signal processing Frequency domain analysis Robustness Image sequences

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