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

作者： V. Algazi Signal and Image Processing Laboratory Department Electrical and Computer Engineering University of California Davis USA

This paper describes a technique for the enhancement of images by FIR filters which compensates for the decreased response of human vision at high spatial frequencies. Because many images contain mainly horizontal and vertical features, and because vision is less acute along diagonals, it is possible to design anisotropic enhancement filters which do not increase greatly the background noise. The anisotropic design methodology also incorporates other results on filter design based on human vision which have recently been reported by the author and coworkers.

关键词： Finite impulse response filter Anisotropic filters Image enhancement Humans Nonlinear filters Frequency response Additive noise Image processing Anisotropic magnetoresistance Visual perception

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Improved pitch detection algorithm for noisy speech

Improved pitch detection algorithm for noisy speech

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

作者： Sanguoon Chung V. Algazi Signal and Image Processing Laboratory Department of Electrical and Computer Engineering University of California Davis CA USA

We propose a new, robust pitch detection algorithm for speech degraded by additive noise. Our algorithm exploits the high correlation between adjacent pitch periods that does not exist for the segment as a whole and performs well in the vicinity of voiced/unvoiced regions where the local SNR is low. The algorithm works as follows. We first determine an estimate of the pitch period near the short time peak energy where the local SNR is highest. We then adaptively estimate the local pitch period from the peak towards the transition region by using pitch synchronous cross-correlation with an updated waveform. The performance of this new algorithm is compared to the SIFT and CEPSTRUM algorithms.

关键词： Detection algorithms speech enhancement Cepstrum Acoustic noise Working environment noise Detectors Cepstral analysis Algorithm design and analysis Time measurement speech synthesis

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LEARNING CHARACTERISTICS OF STOCHASTIC-GRADIENT-DESCENT ALGORITHMS - A GENERAL STUDY, ANALYSIS, AND CRITIQUE

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signal processing 1984年第2期6卷 113-133页

作者： GARDNER, WA Signal and Image Processing Laboratory Department of Electrical and Computer Engineering University of California Davis 95616 USA

A comprehensive analysis of the mean-square learning characteristics of stochastic-descent algorithms is presented. The approach is based on the commonly exploited simplifying assumption of stationary independent training vectors. Characteristics analyzed include stability, steady-state misadjustment, initial rate of convergence, optimum step size, and steady-state autocovariance and spectral characteristics of the weight-vector. Effects on these characteristics due to degree of randomness of stochastic gradient, particular data distribution, and data corruption are isolated and analyzed. An objective of the work is to keep the number of simplifying assumptions and approximations to a minimum. Comparison of results with previous more approximate analyses are made. Lernkurven von stochastischen Gradienten Algorithmen werden untersucht. Die vereinfachende Annahme von stationären, unabhängigen Trainingsvektoren wird benutzt. Charakteristiken die untersucht werden beinhalten Stabilität, ‘steady state’ Fehlanpassung, Start Konvergenz, optimaler Schritt sowie ‘steady state’ Autokovarianz und spektrale Charakteristik des Gewichtsvektors. Die Effekten auf diese Charakteristiken von dem Zufälligkeitsgrad des stichastischen Gradienten, besonderer Daten Verteilung und Daten Verderbung werden isoliert und analysiert. Ein Zielpunkt dieser Arbeit ist es die Anzahl vereinfachenden Annahmen und Approximationen auf ein minimum zu beschränken. Vergleiche mit Resultaten die von approximativeren Analysen stammen werden gezogen. Une analyse d'ensemble des caractéristiques de convergence de l'erreur quadratique moyenne dans les algorithmes utilisant la décroissance du gradient stochastique est presentée. Cette approche est basée sur l'hypothése simplificatrice classique de stationnarité et indépendance des vecteurs de test. Les caractéristiques analysées comprennent la stabilité, l'écart d'état stable, la vitesse initiale de convergence, le pas optimum d'incrémentation, et l'autocovarian

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A PIPELINED RECURSIVE RESIDUE NUMBER SYSTEM DIGITAL-FILTER

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IEEE TRANSACTIONS ON CIRCUITS AND SYSTEMS 1984年第4期31卷 415-417页

作者： SODERSTRAND, MA SINHA, B Signal and Image Processing Laboratory Department of Electrical and Computer Engineering University of California Davis CA USA IBM Corporation Boca Raton FL USA University of California Davis CA USA

The well-known advantages of pipelining as applied to Finite Impulse Response (FIR) Residue Number System (RNS) arithmetic digital filters is extended to the important area of Infinite Impulse Response (IIR) digital filters through a new technique based on augmentation of the IIR transfer function. Through this technique, pipelined IIR filters based on RNS Read-Only-Memory (ROM) table look-up techniques can be designed which offer throughput rates equal to the table look-up time of the ROM's. This high-speed realization can be achieved even though the recursive filter algorithm requires multiple delays in realizing the output of the filter. For the example of a typical second-order IIR filter, the pipelined structure represents a five-fold increase in speed over standard techniques. Higher order realizations will yield proportionately higher speed improvements. Although the new technique does increase somewhat the hardware complexity of the filter, the increase in speed will often justify the additional hardware. The paper discusses the basic technique, stability considerations, and hardware realizations.

关键词： Digital filters IIR filters Finite impulse response filter Hardware Read only memory Pipeline processing Digital arithmetic Transfer functions Throughput Delay

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Radon transform theory for random fields and optimum image reconstruction from noisy projections

Radon transform theory for random fields and optimum image r...

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

作者： A. Jain S. Ansari Signal and Image Processing Laboratory Department of Electrical and Computer Engineering University of California Davis CA USA

In this paper we present some new results on Radon transform theory for stationary random fields. In particular we present a new projection theorem which gives the relation between the power spectrum density of one dimensional projections of a stationary random field and its two dimensional power spectrum density. This result yields the optimum mean square reconstruction filter from noisy projections and is useful in other problems such as multidimensional spectral estimation from one dimensional projections, noise analysis in computed tomography, etc. Example are given to demonstrate the usefulness of these results.

关键词： Image reconstruction Filters Frequency response Computed tomography Fourier transforms Convolution Stochastic processes signal processing Image processing Laboratories

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Application of a simple hearing model to the design of audio filters

Application of a simple hearing model to the design of audio...

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

作者： M. Ready V. Algazi Signal and Image Processing Laboratory Department of Electrical and Computer Engineering University of California Davis CA USA

Filtering audio signals with filters designed exclusively from frequency domain specifications may result in an audible distortion in the vicinity of sharp amplitude transitions. This paper considers the application of known psychoacoustical properties to the design of digital audio filters which minimizes this distortion while approximating some ideal frequency domain characteristics. Psychoacoustic properties and a simple model for hearing are reviewed. A weighted least squares design criteria based on the model and frequency domain specifications is given. Examples of FIR and IIR filters are given and compared to classical frequency domain filters.

关键词： Auditory system Finite impulse response filter Frequency domain analysis Psychoacoustic models IIR filters Psychology Filtering signal design Distortion Digital filters

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Two step color shrinking algorithm for encoding of graphics

Two step color shrinking algorithm for encoding of graphics

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

作者： P. Kelly V. Algazi Signal and Image Processing Laboratory Department of Electrical and Computer Engineering University of California Davis CA USA

The emphasis of many algorithms that have been proposed for the compression of binary images has been the efficient coding of local redundancy in data. We propose that increased compression may be achieved by a decomposition of the compression problem into two steps. The goal of the first step is to extract the global redundancy in an image. This is achieved by a color shrinking algorithm, The goal of the second step is to code the resulting localized data.

关键词： Encoding Image coding Data compression Statistics Source coding computer graphics signal processing Image processing Laboratories Data mining

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AN ADAPTIVE FILTER WHICH ADAPTS STRUCTURE AS WELL AS WEIGHTS

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INTERNATIONAL JOURNAL OF ELECTRONICS 1983年第4期55卷 533-549页

作者： SODERSTRAND, MA VIGIL, MC Department of Electrical and Computer Engineering Signal and Image Processing Laboratory University of California Davis CA 95616 United States Hewlett Packard Davis CA United States

A computer or microprocessor-based adaptive digital filter can easily be constructed to adapt structure as well as weights. These totally adaptive filters will always find a structure and set of weights which offer equal or better error than the standard FIR adaptive filter yet avoid most of the difficulties encountered with IIR adaptive filters. computer simulations of the totally adaptive filter in a host of filtering applications confirm its superior performance when compared with any fixed-structure adaptive filter with an equal number of weights.

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Convergence analysis of LMS and NLMS adaptive algorithms

Convergence analysis of LMS and NLMS adaptive algorithms

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

作者： T. Hsia Signal and Image Processing Laboratory Department of Electrical and Computer Engineering University of California Davis CA USA

The main contribution of this paper is the unified treatment of convergence analysis for both LMS and NLMS adaptive algorithms. The following new results are obtained: (i) necessary and sufficient conditions of convergence, (ii) optimal adjustment gains and optimal convergence rates, (iii) interrelationship between LMS and NLMS gains, and (iv) non-stationary algorithm design.

关键词： Convergence Algorithm design and analysis Least squares approximation Adaptive algorithm Recursive estimation Adaptive filters Random processes Equations Covariance matrix signal processing

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THEORETICAL PERFORMANCE OF ENTROPY-ENCODED DPCM

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IEEE TRANSACTIONS ON COMMUNICATIONS 1982年第5期30卷 1088-1094页

作者： ALGAZI, VR DEWITTE, JT Signal and Image Processing Laboratory Department of Electrical and Computer Engineering University of California Davis CA USA Department of Electrical and Computer Engineering University of California Davis CA USA

In this paper the authors derive simple approximate formulas for the performance of entropy-encoded DPCM for a Gaussian random process and a frequency-weighted mean-square distortion measure. Using these results they compare the performance of DPCM to the information theoretic rate-distortion bound. They study the effect on the performance of DPCM of the spectrum of the input process, the frequency weight in the distortion measure, and the number of prediction coefficients. They also examine briefly the case of achromatic still images using line-by-line and two-dimensional DPCM encoding with intrafield and intraframe information.

关键词： Distortion measurement Image coding Rate-distortion Frequency measurement Fast Fourier transforms Random processes Performance analysis Encoding Fourier transforms

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