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Connections between three-dimensional inverse scattering and linear least-squares estimation of random fields

Acta Applicandae Mathematicae 1988年第3期13卷 267-289页

作者： Yagle, Andrew E. Communications and Signal Processing Laboratory Department of Electrical Engineering and Computer Science The University of Michigan Ann Arbor United States

The three-dimensional Schrödinger equation inverse scattering problem with a nonspherically-symmetric potential is related to the filtering problem of computing the linear leastsquares estimate of the three-dimensional random field on the surface of a sphere from noisy observations inside the sphere. The relation consists of associating an estimation problem with the inverse scattering problem, and vice-versa. This association allows equations and quantities for one problem to be given interpretations in terms of the other problem. A new fast algorithm is obtained for the estimation of random fields using this association. The present work is an extension of the connections between estimation and inverse scattering already known to exist for stationary random processes and one-dimensional scattering potentials, and isotropic random fields and radial scattering protentials. © 1988 Kluwer Academic Publishers.

关键词： AMS subject classifications (1980): 35R30, 93E11, 35L20 fast algorithms Inverse scattering linear least-squares estimation random fields

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A computer-BASED SYNTHESIS METHOD OF STRUCTURALLY LBR DIGITAL ALLPASS NETWORKS

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IEEE TRANSACTIONS ON CIRCUITS AND SYSTEMS 1988年第6期35卷 755-760页

作者： SZCZUPAK, J MITRA, SK FADAVIARDEKANI, J Department of Electrical Engineering Pontificia Universidade Catolica do Rio de Janeiro Rio de Janeiro Brazil Department of Electrical and Computer Engineering Signal Processing Laboratory University of California Santa Barbara CA USA AT and T Bell Laboratories Inc. Allentown PA USA

A digital lossless bounded-real (LBR) transfer function, more commonly known as a stable allpass function, maintains the LBR property under multiplier coefficient quantization when implemented by networks that are structurally LBR. A systematic method is outlined for generating all possible such realizations for second-order transfer functions containing a minimum number of multipliers and delays, allowing the designer to take advantage of the flexibility offered by a catalog of such realizations. Illustrative examples are included.

关键词： TRANSFER FUNCTIONS catalogs Network Designers LBR gene Multiplying circuits

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speech training aids for hearing-impaired individuals: II. Configuration of the Johns Hopkins aids

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Journal of Rehabilitation Research and Development 1988年第4期25卷 63-68页

作者： Ferguson, J.B. Bernstein, L.E. Goldstein Jr., M.H. Speech Processing Laboratory Department of Electrical and Computer Engineering Johns Hopkins University Baltimore MD 21218 United States

Two interrelated personal computer (PC) based speech training aids have been developed: one for use in a school or clinic, the speech Training Station (STS);and the other for the deaf child's home, the speech Practice Station (SPS). The STS monitors speech production by microphone, electroglottograph, and pneumotachograph. The SPS system uses only the microphone input. Both systems utilize commercially available board-level hardware and a custom analog preprocessor board for the analysis of the acoustic and/or physiologic inputs. The school system has been used by speech therapists for diagnosis, training by game playing, and specification of exercises for the SPS. The home system provides directed speech practice between therapy sessions.

关键词： HUMAN REHABILITATION engineering

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Low bit-rate LPC speech communications via a new multipulse speech model

Low bit-rate LPC speech communications via a new multipulse ...

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IEEE International Conference on Communications (ICC)

作者： L.D. Paarmann C.A. Magner B.M. McCarthy Signal Processing Laboratory Department of Electrical and Computer Engineering Drexel University Philadelphia PA USA

A speech model with an excitation that is a multipulse sequence with additive white noise is presented. This model, when used in a 4.8-kb/s communication system, has the potential for improved synthetic speech quality. The algorithm is described and convergence for synthetic data of the assumed structure is proved. A method for significantly improved efficiency is presented, based on an algorithm given by M. Korenberg (1985). Initial experimental data that support both the authors' and Korenberg algorithms are given.< >

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A geometric approach to real time pitch detection

A geometric approach to real time pitch detection

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

作者： T.D. Nguyen J.B. Ferguson T.W. McNamara Speech Processing Laboratory Department of Electrical and Computer Engineering Johns Hopkins University Baltimore MD USA

An approach to time-domain pitch detection based on the concepts of center of mass is presented and evaluated. Excursions, or bumps, in speech signals are treated as geometric areas and replaced by their total mass lumped at the centers of mass. Intervals between masses are grouped into candidate classes. Coincidence and coherence indices of these classes are computed to determine the most likely pitch estimate. Postprocessing consists of a simple error-correction and silence-detection scheme. This algorithm compares favourably in performance with the autocorrelation method, using pitch contours from electroglottograph signals as a reference. The algorithm is tested in noisy environments simulated by uniformly distributed white noise and multitalker babble noise. Results show that the algorithm is robust and accurate. The implementation of this algorithm for a vibrotactile device to aid lipreading is described.< >

关键词： Time domain analysis speech processing Testing speech synthesis speech analysis White noise Detection algorithms Energy states Laboratories Error correction

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signal processing For Ocean Tomography With Moving Ships

Signal Processing For Ocean Tomography With Moving Ships

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Asilomar Conference on signals, Systems & computers

作者： T.G. Birdsall K. Metzger R.C. Spindel Communication and Signal Processing Laboratory Department of Electrical Engineering and Computer Science University of Michigan Ann Arbor MI USA Applied Physics Laboratory University of Washington Seattle WA USA

In ocean acoustic tomography, maximal length binary shift- register sequences, m-sequences, are used to modulate acoustic carriers to achieve high average power and good time and Doppler resolution. To date, the underwater transmitters and receivers have been in fixed positions, and signal processing has consisted of demodulation followed by factor inverse filtering. Ocean tomography now is being extended to include the use of moving, ship-towed, transmitters and receivers, where signal processing must account for Doppler time and frequency resealing. This paper describes the signal demodulation and processing methods developed for moving ship tomography and presents illustrative results.

关键词： signal processing Oceans Tomography Marine vehicles Transmitters Spatial resolution signal resolution Filtering Acoustic signal processing Demodulation

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AN ALTERNATIVE ADAPTIVE SEQUENTIAL REGRESSION ALGORITHM AND ITS APPLICATION TO THE RECOGNITION OF CEREBRAL-PALSY speech

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IEEE TRANSACTIONS ON CIRCUITS AND SYSTEMS 1987年第7期34卷 782-787页

作者： DELLER, JR DONG, H Digital Signal Processing Laboratory Speech Processing Sector Department of Electrical and Computer Engineering Northeastern University Boston MA USA

An algorithm is presented which can be used to compute a temporally adaptive, recursive solution to the problem of estimating the linear prediction parameters of speech. The method is theoretically equivalent to conventional adaptive sequential regression, but is slightly more efficient and computationally much more straightforward. Application to the problem of recognition of speech of the profoundly disabled is discussed.

关键词： speech recognition Birth disorders Equations signal processing algorithms Recursive estimation speech analysis Least squares methods speech processing Parameter estimation

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A NEW ALGORITHM FOR ADAPTIVE ARRAYS

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IEEE TRANSACTIONS ON ACOUSTICS speech AND signal processing 1987年第9期35卷 1314-1319页

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

A new deflected stochastic-gradient algorithm for adaptive sensor arrays is introduced. It is shown to have the capability of converging substantially faster than the nondeflected stochastic gradient (LMS) algorithm but slower than the recursive least squares (RLS) algorithm. Because of its amenability to reduction of computational complexity by data quantization, it has the potential for processing digital signals with bandwidths one to two orders of magnitude larger than the bandwidths manageable with the RLS algorithm for sensor arrays containing a number of adjustable weights in the approximate range of 10-50.

关键词： Adaptive arrays Sensor arrays Least squares approximation Resonance light scattering Bandwidth Stochastic processes Least squares methods Computational complexity Quantization signal processing

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SPECTRAL CORRELATION OF MODULATED signalS - .1. ANALOG MODULATION

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IEEE TRANSACTIONS ON COMMUNICATIONS 1987年第6期35卷 584-594页

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

The importance of the concept of cyclostationarity in design and analysis of signal detectors, synchronizers, and extractors in communication systems is briefly discussed, and the central role of spectral correlation, in the characterization of random processes that are cyclostationary in the wide sense, is explained. A spectral correlation function that is a generalization of the power spectral density function is described, and a corresponding generalization of the Wiener-Khinchine relation and several other fundamental spectral correlation relations also are described. Explicit formulas for the spectral correlation function for various types of analog-modulated signals are derived. This includes pulse and carrier amplitude modulation, quadrature amplitude carrier modulation, and phase and frequency carrier modulation. To illustrate the differing spectral correlation characteristics of different modulation types, the magnitudes of the spectral correlation functions are graphed or described in graphical terms as the heights of surfaces above a bifrequency plane.

关键词： Pulse modulation Amplitude modulation Phase modulation Quadrature amplitude modulation signal design signal analysis signal processing signal detection Detectors Frequency synchronization

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SPECTRAL CORRELATION OF MODULATED signalS .2. DIGITAL MODULATION

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IEEE TRANSACTIONS ON COMMUNICATIONS 1987年第6期35卷 595-601页

作者： GARDNER, WA BROWN, WA CHEN, CK Signal and Image Processing Laboratory Department of Electrical and Computer Engineering University of California Davis CA USA

As a continuation of Part I, the spectral correlation function is presented for a variety of types of digitally modulated signals. These include digital pulse-amplitude, pulse-width, and pulse-position modulation, and various types of phase-shift keying and frequency-shift keying. The magnitudes of the spectral correlation functions are graphed as the heights of surfaces above a bifrequency plane, and these graphs are used as visual aids for comparison and contrast of the spectral correlation properties of different modulation types.

关键词： Digital modulation Space vector pulse width modulation signal processing Pulse modulation Phase shift keying Frequency shift keying Pulse width modulation Filters Matrices Parameter estimation

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