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Conference on advanced signal processing algorithms, architectures, and implementations VI

作者： Williams, R USAF WRIGHT LABAUTOMAT TARGET RECOGNIT DEV BRANCHWRIGHT PATTERSON AFBOH 45433

ISBN: (纸本)0819422347

An adaptive algorithm and two stage filter structure were developed for adaptive filtering of certain classes of signals that exhibit cyclostationary characteristics. The new modified P-vector algorithm (mPa) eliminates the need for a separate desired signal which is typically required by conventional adaptive algorithms. It is then implemented in a time-sequenced manner to counteract the nonstationary characteristics typically found in certain radar and bioelectromagnetic signals. Initial algorithm testing is performed on evoked responses generated by the visual cortex of the human brain with the objective, ultimately, to transition the results to radar signals. Each sample of the evoked response is modeled as the sum of three uncorrelated signal components, a time-varying mean (M), a noise component (N), and a random jitter component (Q). A two stage single channel time-sequenced adaptive filter structure was developed which improves convergence characteristics by de coupling the time-varying mean component from the `Q' and noise components in the first stage. The EEG statistics must be known a priori and are adaptively estimated from the pre stimulus data. The performance of the two stage mPa time-sequenced adaptive filter approaches the performance for the ideal case of an adaptive filter having a noiseless desired response.

关键词： time sequenced adaptive filter EEG biomedical signal processing radar signal processing

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Convergence results on adaptive approximate filtering 6

Convergence results on adaptive approximate filtering

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Conference on advanced signal processing algorithms, architectures, and implementations VI

作者： Ludwig, JT Nawab, SH Chandrakasan, AP MIT ELECT RES LABCAMBRIDGEMA 02139

ISBN: (纸本)0819422347

We present an IIR filtering technique based upon a recently proposed approach for reducing power consumption in implementation of frequency-selective filters. The basic idea in such techniques is to utilize the most recent input and output signal samples to estimate the current SNR (defined as the ratio of the in-band signal power to the out-of-band signal power) at the filter's input. This estimated input SNR is then used to update the filter order to the minimum value which would guarantee a minimum tolerable SNR at the filter's output. A key issue addressed in this paper is how well the estimated filter order converges to the theoretical minimum for situations satisfying the assumptions behind the derivation of the technique. Experimental results are used to verify that convergence to the correct filter order depends (1) upon the number of input and output samples used for estimating the input SNR, (2) upon the filter order applied in generating the output samples that are used in estimating the input SNR and (3) upon the proximity of the actual input SNR to boundaries in the input-SNR space corresponding to changes in the optimal choice for filter order.

关键词： low power approximate processing incremental refinement structure frequency-selective filtering Butterworth filters adaptive filtering

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Application of higher-order cepstral techniques in problems of fetal heart signal extraction 6

Application of higher-order cepstral techniques in problems ...

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Conference on advanced signal processing algorithms, architectures, and implementations VI

作者： SabryRizk, M Zgallai, W Hardiman, P ORiordan, J CITY UNIV LONDON DEPT ELECT ENGNSIGNAL PROC GRPLONDON EC1V 0HBENGLAND

ISBN: (纸本)0819422347

Recently, cepstral analysis based on second order statistics and homomorphic filtering techniques have been used in the adaptive decomposition of overlapping, or otherwise, and noise contaminated ECG complexes of mothers and fetals obtained by a transabdominal surface electrodes connected to a monitoring instrument, an interface card, and a PC. Differential time delays of fetal heart beats measured from a reference point located on the mother complex after transformation to cepstra domains are first obtained and this is followed by fetal heart rate variability computations. Homomorphic filtering in the complex cepstral domain and the subuent transformation to the time domain results in fetal complex recovery. However, three problems have been identified with second-order based cepstral techniques that needed rectification in this paper. These are (1) errors resulting from the phase unwrapping algorithms and leading to fetal complex perturbation, (2) the unavoidable conversion of noise statistics from Gaussianess to non-Gaussianess due to the highly non-linear nature of homomorphic transform does warrant stringent noise cancellation routines, (3) due to the aforementioned problems in (1) and (2), it is difficult to adaptively optimize windows to include all individual fetal complexes in the time domain based on amplitude thresholding routines in the complex cepstral domain (i.e. the task of `zooming' in on weak fetal complexes requires more processing time). The use of third-order based high resolution differential cepstrum technique results in recovery of the delay of the order of 120 milliseconds.

关键词： advanced signal processing techniques non-linear adaptive filtering differential bicepstral applications non-invasive fetal ECG monitoring

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A DSP implementation of source location using microphone arrays 6

A DSP implementation of source location using microphone arr...

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Conference on advanced signal processing algorithms, architectures, and implementations VI

作者： Rabinkin, DV Renomeron, RJ Dahl, A French, JC Flanagan, JL Bianchi, MH RUTGERS STATE UNIV CTR COMP AIDS IND PRODPISCATAWAYNJ 08855

ISBN: (纸本)0819422347

The design, implementation, and performance of a low-cost, real-time DSP system for source location is discussed. The system consists of an 8-element electret microphone array connected to a signalogic DSP daughterboard hosted by a PC. The system determines the location of a speaker in the audience in an irregularly shaped auditorium. The auditorium presents a non-ideal acoustical environment;some of the walls are acoustically treated, but there still exists significant reverberation and a large amount of low frequency noise from fans in the ceiling. The source location algorithm is implemented in a two step process: The first step determines time delay of arrival (TDOA) for select microphone pairs. A modified version of the Cross-Power Spectrum Phase Method is used to compute TDOAs and is implemented on the DSP daughterboard. The second step uses the computed TDOAs in a least mean squares gradient descent search algorithm implemented on the PC to compute a location estimate. © 1996 SPIE. All rights reserved.

关键词： microphone arrays speech source location real-time processing

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Separating desired image and signal invariant components from extraneous variations 6

Separating desired image and signal invariant components fro...

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Conference on advanced signal processing algorithms, architectures, and implementations VI

作者： Williams, WJ Zalubas, EJ Hero, A UNIV MICHIGAN DEPT ELECT ENGN & COMP SCIANN ARBORMI 48109

ISBN: (纸本)0819422347

Images and signals can be characterized by representations invariant to time shifts, spatial shifts, frequency shifts, and scale changes as the situation dictates. Advances in time-frequency analysis and scale transform techniques have made this possible. The next step is to distinguish between invariant forms representing different classes of image or signal. Unfortunately, additional factors such as noise contamination and `style' differences complicate this. A ready example is found in text, where letters and words may vary in size and position within the image segment being examined. Examples of complicating variations include font used, corruption during fax transmission, and printer characteristics. The solution advanced in this paper is to cast the desired invariants into separate subspaces for each extraneous factor or group of factors. The first goal is to have minimal overlap between these subspaces and the second goal is to be able to identify each subspace accurately. Concepts borrowed from high-resolution spectral analysis, but adapted uniquely to this problem have been found to be useful in this context. Once the pertinent subspace is identified, the recognition of a particular invariant form within this subspace is relatively simple using well-known singular value decomposition techniques.

关键词： scale transform Mellin transform wordspotting singular value decomposition character recognition time-frequency

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CISC, RISC AND DSP PROCESSORS IN REAL-TIME signal-processing AND CONTROL

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MICROPROCESSORS AND MICROSYSTEMS 1995年第5期19卷 291-300页

作者： TOKHI, MO HOSSAIN, MA Department of Automatic Control and Systems Engineering The University of Sheffield PO Box 600 Mappin Street Sheffield S1 4DU UK

This paper presents an investigation into the performance evaluation of advanced complex instruction set computer (CISC) processors, reduced instruction set computer (RISC) processors and digital signal processing (DSP) devices in the real-time implementation of signal processing and control algorithms. The algorithms considered are of varying degree of regularity relative to one another. These include a fast Fourier transform algorithm, a second-order correlation algorithm, two adaptive filtering algorithms, a simulation algorithm of a flexible manipulator system and simulation, identification and active vibration control algorithms of a flexible beam system. The algorithms are implemented on an 80486DX2 CISC processor, an 80386DX CISC processor, a SPARC TMS390S10 RISC processor, a T805 RISC processor, an 80i860 RISC processor and a TMS320C40 DSP device. The hardware and software resources and capabilities of the processors and the characteristics of the algorithms are discussed to provide a matching between the algorithms and the architectures. Finally, a comparison of the results of the implementations, on the basis of real-time computation performance, is made and discussed.

关键词： DIGITAL signal processing COMPLEX INSTRUCTION SET COMPUTER REDUCED INSTRUCTION SET COMPUTER

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advanced signal processing: algorithms, architectures, and implementations V

Advanced Signal Processing: Algorithms, Architectures, and I...

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advanced signal processing: algorithms, architectures, and implementations V

The topics discussed are cumulants;time frequency signal analysis;mobile communications;nonlinear dynamics and signal processing;subspace tracking;array processing;fast solvers for structured systems;regularization me... 详细信息

ISBN: (纸本)0819416207

关键词： signal processing

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CONTRIBUTION OF THE POLARIZATION DIVERSITY IN HF DIRECTION FINDING SYSTEMS

CONTRIBUTION OF THE POLARIZATION DIVERSITY IN HF DIRECTION F...

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5th SPIE Conference on advanced signal processing - algorithms, architectures, and implementations

作者： ERHEL, Y EDJEOU, A BERTEL, L CTR RECH ECOLES COETQUIDAN F-56381 GUERFRANCE

ISBN: (纸本)0819416207

The aim of the paper is to estimate the contribution of the polarization diversity in high frequency (3 - 30 MHz) direction finding systems. We first describe the peculiarities of H.F. propagation and the resulting signal model involved in computer simulations. Next, we analyze the behavior of some particular direction finding systems using linear and circular geometries and polarization diversity. Some algorithms (non linear frequential analysis, M.U.S.I.C.) are tested in several conditions (narrowband or broadband signals, polarization filtering reiterated or no, sub-sampling). Theoretical and experimental results show that polarization diversity based upon the knowledge of the antenna complex responses improves greatly the efficiency of direction finding.

关键词： DIRECTION FINDING POLARIZATION ARRAY processing HIGH RESOLUTION algorithms HF PROPAGATION

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Architecture of a complex arithmetic processor for communication signal processing

Architecture of a complex arithmetic processor for communica...

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advanced signal processing: algorithms, architectures, and implementations V

作者： Gilfeather, Susan L. Gehman Jr., John B. Harrison, Calvin Motorola Government and Systems Technology Group Scottsdale AZ USA

ISBN: (纸本)0819416207

The Complex Arithmetic Processor (CAP) is a high performance, single chip Digital signal Processor optimized for communication signal processing operations. The CAP VLSI provides the communication system building block necessary to meet the increased signal processing requirements of complex modulation types, voice and image compression while maintaining the requirement for small, low power implementations. The chip is intended for high speed, low power digital communication system applications such as hand held spread spectrum communications systems. The CAP architecture has been developed specifically for the complex arithmetic functions required in communication signal processing. The CAP is a software programmable, highly integrated parallel array of processors containing the arithmetic resources, memories, address generation, bit manipulation and logic functions necessary to support the sophisticated processing required in advanced communication equipment. The CAP executes a 1024 point complex Fast Fourier Transform in 131 microseconds.

关键词： Digital integrated circuits

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Parallel digital signal processing architectures for image processing

Parallel digital signal processing architectures for image p...

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advanced signal processing: algorithms, architectures, and implementations V

作者： Kshirsagar, Shirish P. Hartley, David A. Harvey, David M. Hobson, Clifford A. Liverpool John Moores Univ. Maharashtra India

ISBN: (纸本)0819416207

This paper describes research into a high speed image processing system using parallel digital signal processors for the processing of electro-optic images. The objective of the system is to reduce the processing time of non-contact type inspection problems including industrial and medical applications. A single processor can not deliver sufficient processing power required for the use of applications hence, a MIMD system is designed and constructed to enable fast processing of electro-optic images. The Texas Instruments TMS320C40 digital signal processor is used due to its high speed floating point CPU and the support for the parallel processing environment. A custom designed VISION bus is provided to transfer images between processors. The system is being applied for solder joint inspection of high technology printed circuit boards.

关键词： Image processing

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